From e58de7baf7de11f01a675cbbf6ecc8a2758b9ca5 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@alsa3.local>
Date: Sun, 28 Dec 2008 16:44:30 +0100
Subject: ALSA: Convert to snd_card_create() in sound/pci/*

Convert from snd_card_new() to the new snd_card_create() function
in sound/pci/*.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ad1889.c                  | 6 +++---
 sound/pci/ali5451/ali5451.c         | 6 +++---
 sound/pci/als300.c                  | 6 +++---
 sound/pci/als4000.c                 | 9 +++++----
 sound/pci/atiixp.c                  | 6 +++---
 sound/pci/atiixp_modem.c            | 6 +++---
 sound/pci/au88x0/au88x0.c           | 6 +++---
 sound/pci/aw2/aw2-alsa.c            | 6 +++---
 sound/pci/azt3328.c                 | 6 +++---
 sound/pci/bt87x.c                   | 6 +++---
 sound/pci/ca0106/ca0106_main.c      | 6 +++---
 sound/pci/cmipci.c                  | 6 +++---
 sound/pci/cs4281.c                  | 6 +++---
 sound/pci/cs46xx/cs46xx.c           | 6 +++---
 sound/pci/cs5530.c                  | 6 +++---
 sound/pci/cs5535audio/cs5535audio.c | 6 +++---
 sound/pci/echoaudio/echoaudio.c     | 6 +++---
 sound/pci/emu10k1/emu10k1.c         | 6 +++---
 sound/pci/emu10k1/emu10k1x.c        | 6 +++---
 sound/pci/ens1370.c                 | 6 +++---
 sound/pci/es1938.c                  | 6 +++---
 sound/pci/es1968.c                  | 6 +++---
 sound/pci/fm801.c                   | 6 +++---
 sound/pci/hda/hda_intel.c           | 6 +++---
 sound/pci/ice1712/ice1712.c         | 6 +++---
 sound/pci/ice1712/ice1724.c         | 6 +++---
 sound/pci/intel8x0.c                | 6 +++---
 sound/pci/intel8x0m.c               | 6 +++---
 sound/pci/korg1212/korg1212.c       | 6 +++---
 sound/pci/maestro3.c                | 6 +++---
 sound/pci/mixart/mixart.c           | 6 +++---
 sound/pci/nm256/nm256.c             | 6 +++---
 sound/pci/oxygen/oxygen_lib.c       | 8 ++++----
 sound/pci/pcxhr/pcxhr.c             | 6 +++---
 sound/pci/riptide/riptide.c         | 6 +++---
 sound/pci/rme32.c                   | 7 ++++---
 sound/pci/rme96.c                   | 7 ++++---
 sound/pci/rme9652/hdsp.c            | 6 ++++--
 sound/pci/rme9652/hdspm.c           | 8 ++++----
 sound/pci/rme9652/rme9652.c         | 8 ++++----
 sound/pci/sis7019.c                 | 5 ++---
 sound/pci/sonicvibes.c              | 6 +++---
 sound/pci/trident/trident.c         | 6 +++---
 sound/pci/via82xx.c                 | 6 +++---
 sound/pci/via82xx_modem.c           | 6 +++---
 sound/pci/vx222/vx222.c             | 6 +++---
 sound/pci/ymfpci/ymfpci.c           | 6 +++---
 47 files changed, 148 insertions(+), 144 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c
index a7f38e63303..d1f242bd0ac 100644
--- a/sound/pci/ad1889.c
+++ b/sound/pci/ad1889.c
@@ -995,10 +995,10 @@ snd_ad1889_probe(struct pci_dev *pci,
 	}
 
 	/* (2) */
-	card = snd_card_new(index[devno], id[devno], THIS_MODULE, 0);
+	err = snd_card_create(index[devno], id[devno], THIS_MODULE, 0, &card);
 	/* XXX REVISIT: we can probably allocate chip in this call */
-	if (card == NULL)
-		return -ENOMEM;
+	if (err < 0)
+		return err;
 
 	strcpy(card->driver, "AD1889");
 	strcpy(card->shortname, "Analog Devices AD1889");
diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index 1a0fd65ec28..b36c551da56 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -2307,9 +2307,9 @@ static int __devinit snd_ali_probe(struct pci_dev *pci,
 
 	snd_ali_printk("probe ...\n");
 
-	card = snd_card_new(index, id, THIS_MODULE, 0);
-	if (!card)
-		return -ENOMEM;
+	err = snd_card_create(index, id, THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	err = snd_ali_create(card, pci, pcm_channels, spdif, &codec);
 	if (err < 0)
diff --git a/sound/pci/als300.c b/sound/pci/als300.c
index 8df6824b51c..f557c155db4 100644
--- a/sound/pci/als300.c
+++ b/sound/pci/als300.c
@@ -812,10 +812,10 @@ static int __devinit snd_als300_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
 
-	if (card == NULL)
-		return -ENOMEM;
+	if (err < 0)
+		return err;
 
 	chip_type = pci_id->driver_data;
 
diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c
index ba570053d4d..542a0c65a92 100644
--- a/sound/pci/als4000.c
+++ b/sound/pci/als4000.c
@@ -889,12 +889,13 @@ static int __devinit snd_card_als4000_probe(struct pci_dev *pci,
 	pci_write_config_word(pci, PCI_COMMAND, word | PCI_COMMAND_IO);
 	pci_set_master(pci);
 	
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 
-			    sizeof(*acard) /* private_data: acard */);
-	if (card == NULL) {
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 
+			      sizeof(*acard) /* private_data: acard */,
+			      &card);
+	if (err < 0) {
 		pci_release_regions(pci);
 		pci_disable_device(pci);
-		return -ENOMEM;
+		return err;
 	}
 
 	acard = card->private_data;
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index 226fe8237d3..9ce8548c03e 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -1645,9 +1645,9 @@ static int __devinit snd_atiixp_probe(struct pci_dev *pci,
 	struct atiixp *chip;
 	int err;
 
-	card = snd_card_new(index, id, THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index, id, THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	strcpy(card->driver, spdif_aclink ? "ATIIXP" : "ATIIXP-SPDMA");
 	strcpy(card->shortname, "ATI IXP");
diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c
index 0e6e5cc1c50..c3136cccc55 100644
--- a/sound/pci/atiixp_modem.c
+++ b/sound/pci/atiixp_modem.c
@@ -1288,9 +1288,9 @@ static int __devinit snd_atiixp_probe(struct pci_dev *pci,
 	struct atiixp_modem *chip;
 	int err;
 
-	card = snd_card_new(index, id, THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index, id, THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	strcpy(card->driver, "ATIIXP-MODEM");
 	strcpy(card->shortname, "ATI IXP Modem");
diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c
index a36d4d1fd41..9ec122383ee 100644
--- a/sound/pci/au88x0/au88x0.c
+++ b/sound/pci/au88x0/au88x0.c
@@ -250,9 +250,9 @@ snd_vortex_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
 		return -ENOENT;
 	}
 	// (2)
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	// (3)
 	if ((err = snd_vortex_create(card, pci, &chip)) < 0) {
diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c
index 3f00ddf450f..eefcbf648ee 100644
--- a/sound/pci/aw2/aw2-alsa.c
+++ b/sound/pci/aw2/aw2-alsa.c
@@ -368,9 +368,9 @@ static int __devinit snd_aw2_probe(struct pci_dev *pci,
 	}
 
 	/* (2) Create card instance */
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	/* (3) Create main component */
 	err = snd_aw2_create(card, pci, &chip);
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 333007c523a..1df96e76c48 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -2216,9 +2216,9 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	strcpy(card->driver, "AZF3328");
 	strcpy(card->shortname, "Aztech AZF3328 (PCI168)");
diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
index 1aa1c040254..a299340519d 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -888,9 +888,9 @@ static int __devinit snd_bt87x_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (!card)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	err = snd_bt87x_create(card, pci, &chip);
 	if (err < 0)
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index 0e62205d408..b116456e770 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -1707,9 +1707,9 @@ static int __devinit snd_ca0106_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	err = snd_ca0106_create(dev, card, pci, &chip);
 	if (err < 0)
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index 1a74ca62c31..c7899c32aba 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -3272,9 +3272,9 @@ static int __devinit snd_cmipci_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 	
 	switch (pci->device) {
 	case PCI_DEVICE_ID_CMEDIA_CM8738:
diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c
index 192e7842e18..b9b07f46463 100644
--- a/sound/pci/cs4281.c
+++ b/sound/pci/cs4281.c
@@ -1925,9 +1925,9 @@ static int __devinit snd_cs4281_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	if ((err = snd_cs4281_create(card, pci, &chip, dual_codec[dev])) < 0) {
 		snd_card_free(card);
diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c
index e876b3263e4..c9b3e3d48cb 100644
--- a/sound/pci/cs46xx/cs46xx.c
+++ b/sound/pci/cs46xx/cs46xx.c
@@ -88,9 +88,9 @@ static int __devinit snd_card_cs46xx_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 	if ((err = snd_cs46xx_create(card, pci,
 				     external_amp[dev], thinkpad[dev],
 				     &chip)) < 0) {
diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c
index 6dea5b5cc77..dc464321d0f 100644
--- a/sound/pci/cs5530.c
+++ b/sound/pci/cs5530.c
@@ -258,10 +258,10 @@ static int __devinit snd_cs5530_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
 
-	if (card == NULL)
-		return -ENOMEM;
+	if (err < 0)
+		return err;
 
 	err = snd_cs5530_create(card, pci, &chip);
 	if (err < 0) {
diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c
index 826e6dec2e9..ac1d72e0a1e 100644
--- a/sound/pci/cs5535audio/cs5535audio.c
+++ b/sound/pci/cs5535audio/cs5535audio.c
@@ -353,9 +353,9 @@ static int __devinit snd_cs5535audio_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	if ((err = snd_cs5535audio_create(card, pci, &cs5535au)) < 0)
 		goto probefail_out;
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 8dbc5c4ba42..9d015a76eb6 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -1997,9 +1997,9 @@ static int __devinit snd_echo_probe(struct pci_dev *pci,
 
 	DE_INIT(("Echoaudio driver starting...\n"));
 	i = 0;
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	snd_card_set_dev(card, &pci->dev);
 
diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c
index 8354c1a8331..c7f3b994101 100644
--- a/sound/pci/emu10k1/emu10k1.c
+++ b/sound/pci/emu10k1/emu10k1.c
@@ -114,9 +114,9 @@ static int __devinit snd_card_emu10k1_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 	if (max_buffer_size[dev] < 32)
 		max_buffer_size[dev] = 32;
 	else if (max_buffer_size[dev] > 1024)
diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c
index 5ff4dbb62da..31542adc6b7 100644
--- a/sound/pci/emu10k1/emu10k1x.c
+++ b/sound/pci/emu10k1/emu10k1x.c
@@ -1544,9 +1544,9 @@ static int __devinit snd_emu10k1x_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	if ((err = snd_emu10k1x_create(card, pci, &chip)) < 0) {
 		snd_card_free(card);
diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c
index 9bf95367c88..e00614cbcef 100644
--- a/sound/pci/ens1370.c
+++ b/sound/pci/ens1370.c
@@ -2409,9 +2409,9 @@ static int __devinit snd_audiopci_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	if ((err = snd_ensoniq_create(card, pci, &ensoniq)) < 0) {
 		snd_card_free(card);
diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c
index 4cd9a1faaec..34a78afc26d 100644
--- a/sound/pci/es1938.c
+++ b/sound/pci/es1938.c
@@ -1799,9 +1799,9 @@ static int __devinit snd_es1938_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 	for (idx = 0; idx < 5; idx++) {
 		if (pci_resource_start(pci, idx) == 0 ||
 		    !(pci_resource_flags(pci, idx) & IORESOURCE_IO)) {
diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
index e9c3794bbcb..dc97e811614 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -2645,9 +2645,9 @@ static int __devinit snd_es1968_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (!card)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
                 
 	if (total_bufsize[dev] < 128)
 		total_bufsize[dev] = 128;
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index c129f9e2072..60cdb9e0b68 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -1468,9 +1468,9 @@ static int __devinit snd_card_fm801_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 	if ((err = snd_fm801_create(card, pci, tea575x_tuner[dev], &chip)) < 0) {
 		snd_card_free(card);
 		return err;
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index f04de115ee1..ad5df2ae6f7 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2335,10 +2335,10 @@ static int __devinit azx_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (!card) {
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0) {
 		snd_printk(KERN_ERR SFX "Error creating card!\n");
-		return -ENOMEM;
+		return err;
 	}
 
 	err = azx_create(card, pci, dev, pci_id->driver_data, &chip);
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index 58d7cda03de..bab1c700f49 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -2648,9 +2648,9 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	strcpy(card->driver, "ICE1712");
 	strcpy(card->shortname, "ICEnsemble ICE1712");
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index bb8d8c766b9..7ff36d3f0f4 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -2456,9 +2456,9 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	strcpy(card->driver, "ICE1724");
 	strcpy(card->shortname, "ICEnsemble ICE1724");
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 19d3391e229..671ff65db02 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -3058,9 +3058,9 @@ static int __devinit snd_intel8x0_probe(struct pci_dev *pci,
 	int err;
 	struct shortname_table *name;
 
-	card = snd_card_new(index, id, THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index, id, THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	if (spdif_aclink < 0)
 		spdif_aclink = check_default_spdif_aclink(pci);
diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c
index 93449e46456..33a843c1931 100644
--- a/sound/pci/intel8x0m.c
+++ b/sound/pci/intel8x0m.c
@@ -1269,9 +1269,9 @@ static int __devinit snd_intel8x0m_probe(struct pci_dev *pci,
 	int err;
 	struct shortname_table *name;
 
-	card = snd_card_new(index, id, THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index, id, THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	strcpy(card->driver, "ICH-MODEM");
 	strcpy(card->shortname, "Intel ICH");
diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c
index 5f8006b4275..8b79969034b 100644
--- a/sound/pci/korg1212/korg1212.c
+++ b/sound/pci/korg1212/korg1212.c
@@ -2443,9 +2443,9 @@ snd_korg1212_probe(struct pci_dev *pci,
 		dev++;
 		return -ENOENT;
 	}
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-        if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
         if ((err = snd_korg1212_create(card, pci, &korg1212)) < 0) {
 		snd_card_free(card);
diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c
index 59bbaf8f3e5..70141548f25 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -2691,9 +2691,9 @@ snd_m3_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	switch (pci->device) {
 	case PCI_DEVICE_ID_ESS_ALLEGRO:
diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c
index f23a73577c2..bfc19e36c4b 100644
--- a/sound/pci/mixart/mixart.c
+++ b/sound/pci/mixart/mixart.c
@@ -1365,12 +1365,12 @@ static int __devinit snd_mixart_probe(struct pci_dev *pci,
 		else
 			idx = index[dev] + i;
 		snprintf(tmpid, sizeof(tmpid), "%s-%d", id[dev] ? id[dev] : "MIXART", i);
-		card = snd_card_new(idx, tmpid, THIS_MODULE, 0);
+		err = snd_card_create(idx, tmpid, THIS_MODULE, 0, &card);
 
-		if (! card) {
+		if (err < 0) {
 			snd_printk(KERN_ERR "cannot allocate the card %d\n", i);
 			snd_mixart_free(mgr);
-			return -ENOMEM;
+			return err;
 		}
 
 		strcpy(card->driver, CARD_NAME);
diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c
index 50c9f8a0508..522a040855d 100644
--- a/sound/pci/nm256/nm256.c
+++ b/sound/pci/nm256/nm256.c
@@ -1668,9 +1668,9 @@ static int __devinit snd_nm256_probe(struct pci_dev *pci,
 		}
 	}
 
-	card = snd_card_new(index, id, THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index, id, THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	switch (pci->device) {
 	case PCI_DEVICE_ID_NEOMAGIC_NM256AV_AUDIO:
diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c
index 84f481d41ef..9c81e0b0511 100644
--- a/sound/pci/oxygen/oxygen_lib.c
+++ b/sound/pci/oxygen/oxygen_lib.c
@@ -459,10 +459,10 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
 	struct oxygen *chip;
 	int err;
 
-	card = snd_card_new(index, id, model->owner,
-			    sizeof *chip + model->model_data_size);
-	if (!card)
-		return -ENOMEM;
+	err = snd_card_create(index, id, model->owner,
+			      sizeof(*chip) + model->model_data_size, &card);
+	if (err < 0)
+		return err;
 
 	chip = card->private_data;
 	chip->card = card;
diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c
index 27cf2c28d11..7f95459c8b1 100644
--- a/sound/pci/pcxhr/pcxhr.c
+++ b/sound/pci/pcxhr/pcxhr.c
@@ -1510,12 +1510,12 @@ static int __devinit pcxhr_probe(struct pci_dev *pci,
 
 		snprintf(tmpid, sizeof(tmpid), "%s-%d",
 			 id[dev] ? id[dev] : card_name, i);
-		card = snd_card_new(idx, tmpid, THIS_MODULE, 0);
+		err = snd_card_create(idx, tmpid, THIS_MODULE, 0, &card);
 
-		if (! card) {
+		if (err < 0) {
 			snd_printk(KERN_ERR "cannot allocate the card %d\n", i);
 			pcxhr_free(mgr);
-			return -ENOMEM;
+			return err;
 		}
 
 		strcpy(card->driver, DRIVER_NAME);
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index 3caacfb9d8e..6f1034417a0 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -2102,9 +2102,9 @@ snd_card_riptide_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 	if ((err = snd_riptide_create(card, pci, &chip)) < 0) {
 		snd_card_free(card);
 		return err;
diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c
index e7ef3a1a25a..d7b966e7c4c 100644
--- a/sound/pci/rme32.c
+++ b/sound/pci/rme32.c
@@ -1941,9 +1941,10 @@ snd_rme32_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
 		return -ENOENT;
 	}
 
-	if ((card = snd_card_new(index[dev], id[dev], THIS_MODULE,
-				 sizeof(struct rme32))) == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE,
+			      sizeof(struct rme32), &card);
+	if (err < 0)
+		return err;
 	card->private_free = snd_rme32_card_free;
 	rme32 = (struct rme32 *) card->private_data;
 	rme32->card = card;
diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c
index 3fdd488d097..55fb1c131f5 100644
--- a/sound/pci/rme96.c
+++ b/sound/pci/rme96.c
@@ -2348,9 +2348,10 @@ snd_rme96_probe(struct pci_dev *pci,
 		dev++;
 		return -ENOENT;
 	}
-	if ((card = snd_card_new(index[dev], id[dev], THIS_MODULE,
-				 sizeof(struct rme96))) == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE,
+			      sizeof(struct rme96), &card);
+	if (err < 0)
+		return err;
 	card->private_free = snd_rme96_card_free;
 	rme96 = (struct rme96 *)card->private_data;	
 	rme96->card = card;
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index 44d0c15e2b7..05b3f795a16 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -5158,8 +5158,10 @@ static int __devinit snd_hdsp_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	if (!(card = snd_card_new(index[dev], id[dev], THIS_MODULE, sizeof(struct hdsp))))
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE,
+			      sizeof(struct hdsp), &card);
+	if (err < 0)
+		return err;
 
 	hdsp = (struct hdsp *) card->private_data;
 	card->private_free = snd_hdsp_card_free;
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 71231cf1b2b..d4b4e0d0fee 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -4503,10 +4503,10 @@ static int __devinit snd_hdspm_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev],
-			    THIS_MODULE, sizeof(struct hdspm));
-	if (!card)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev],
+			      THIS_MODULE, sizeof(struct hdspm), &card);
+	if (err < 0)
+		return err;
 
 	hdspm = card->private_data;
 	card->private_free = snd_hdspm_card_free;
diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c
index 2570907134d..bc539abb210 100644
--- a/sound/pci/rme9652/rme9652.c
+++ b/sound/pci/rme9652/rme9652.c
@@ -2594,11 +2594,11 @@ static int __devinit snd_rme9652_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE,
-			    sizeof(struct snd_rme9652));
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE,
+			      sizeof(struct snd_rme9652), &card);
 
-	if (!card)
-		return -ENOMEM;
+	if (err < 0)
+		return err;
 
 	rme9652 = (struct snd_rme9652 *) card->private_data;
 	card->private_free = snd_rme9652_card_free;
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index df2007e3be7..baf6d8e3dab 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -1387,9 +1387,8 @@ static int __devinit snd_sis7019_probe(struct pci_dev *pci,
 	if (!enable)
 		goto error_out;
 
-	rc = -ENOMEM;
-	card = snd_card_new(index, id, THIS_MODULE, sizeof(*sis));
-	if (!card)
+	rc = snd_card_create(index, id, THIS_MODULE, sizeof(*sis), &card);
+	if (rc < 0)
 		goto error_out;
 
 	strcpy(card->driver, "SiS7019");
diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c
index cd408b86c83..c5601b0ad7c 100644
--- a/sound/pci/sonicvibes.c
+++ b/sound/pci/sonicvibes.c
@@ -1423,9 +1423,9 @@ static int __devinit snd_sonic_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
  
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 	for (idx = 0; idx < 5; idx++) {
 		if (pci_resource_start(pci, idx) == 0 ||
 		    !(pci_resource_flags(pci, idx) & IORESOURCE_IO)) {
diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c
index d94b16ffb38..21cef97d478 100644
--- a/sound/pci/trident/trident.c
+++ b/sound/pci/trident/trident.c
@@ -89,9 +89,9 @@ static int __devinit snd_trident_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	if ((err = snd_trident_create(card, pci,
 				      pcm_channels[dev],
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 1aafe956ee2..d8705547dae 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -2433,9 +2433,9 @@ static int __devinit snd_via82xx_probe(struct pci_dev *pci,
 	unsigned int i;
 	int err;
 
-	card = snd_card_new(index, id, THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index, id, THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	card_type = pci_id->driver_data;
 	switch (card_type) {
diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c
index 5bd79d2a5a1..c086b762c15 100644
--- a/sound/pci/via82xx_modem.c
+++ b/sound/pci/via82xx_modem.c
@@ -1167,9 +1167,9 @@ static int __devinit snd_via82xx_probe(struct pci_dev *pci,
 	unsigned int i;
 	int err;
 
-	card = snd_card_new(index, id, THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index, id, THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	card_type = pci_id->driver_data;
 	switch (card_type) {
diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c
index acc352f4a44..fc9136c3e0d 100644
--- a/sound/pci/vx222/vx222.c
+++ b/sound/pci/vx222/vx222.c
@@ -204,9 +204,9 @@ static int __devinit snd_vx222_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	switch ((int)pci_id->driver_data) {
 	case VX_PCI_VX222_OLD:
diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c
index 2631a554845..4af66661f9b 100644
--- a/sound/pci/ymfpci/ymfpci.c
+++ b/sound/pci/ymfpci/ymfpci.c
@@ -187,9 +187,9 @@ static int __devinit snd_card_ymfpci_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	switch (pci_id->device) {
 	case 0x0004: str = "YMF724";  model = "DS-1"; break;
-- 
cgit v1.2.3


From f3a374e55a60f7ca57335c24ef875731b6683147 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 19 Jan 2009 14:30:48 +0100
Subject: ALSA: ca0106 - Add quirk for GA-G1975X mobo

Giga-byte GA-G1975X mobo has a CA0106 on-board chip.
Reference: bnc#395807
	https://bugzilla.novell.com/show_bug.cgi?id=395807

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ca0106/ca0106_main.c | 8 ++++++++
 1 file changed, 8 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index 0e62205d408..3aac7e6489c 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -255,6 +255,14 @@ static struct snd_ca0106_details ca0106_chip_details[] = {
 	   .gpio_type = 2,
 	   .i2c_adc = 1,
 	   .spi_dac = 1 } ,
+	/* Giga-byte GA-G1975X mobo
+	 * Novell bnc#395807
+	 */
+	/* FIXME: the GPIO and I2C setting aren't tested well */
+	{ .serial = 0x1458a006,
+	  .name = "Giga-byte GA-G1975X",
+	  .gpio_type = 1,
+	  .i2c_adc = 1 },
 	 /* Shuttle XPC SD31P which has an onboard Creative Labs
 	  * Sound Blaster Live! 24-bit EAX
 	  * high-definition 7.1 audio processor".
-- 
cgit v1.2.3


From 29fdbec2dcb1ce364812778271056aa9516ff3ed Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 20 Jan 2009 13:07:55 +0100
Subject: ALSA: hda - Add extra volume offset to standard volume amp macros

Added the volume offset to base for the standard volume controls
to handle elements with too big volume scales like -96dB..0dB.
For such elements, you can set the base volume to reduce the range.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 45 +++++++++++++++++++++++++++++++++++++--------
 sound/pci/hda/hda_local.h |  5 ++++-
 2 files changed, 41 insertions(+), 9 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index b7bba7dc7cf..0cf2424ada6 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1119,6 +1119,7 @@ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol,
 	u16 nid = get_amp_nid(kcontrol);
 	u8 chs = get_amp_channels(kcontrol);
 	int dir = get_amp_direction(kcontrol);
+	unsigned int ofs = get_amp_offset(kcontrol);
 	u32 caps;
 
 	caps = query_amp_caps(codec, nid, dir);
@@ -1130,6 +1131,8 @@ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol,
 		       kcontrol->id.name);
 		return -EINVAL;
 	}
+	if (ofs < caps)
+		caps -= ofs;
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
 	uinfo->count = chs == 3 ? 2 : 1;
 	uinfo->value.integer.min = 0;
@@ -1138,6 +1141,32 @@ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol,
 }
 EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_info);
 
+
+static inline unsigned int
+read_amp_value(struct hda_codec *codec, hda_nid_t nid,
+	       int ch, int dir, int idx, unsigned int ofs)
+{
+	unsigned int val;
+	val = snd_hda_codec_amp_read(codec, nid, ch, dir, idx);
+	val &= HDA_AMP_VOLMASK;
+	if (val >= ofs)
+		val -= ofs;
+	else
+		val = 0;
+	return val;
+}
+
+static inline int
+update_amp_value(struct hda_codec *codec, hda_nid_t nid,
+		 int ch, int dir, int idx, unsigned int ofs,
+		 unsigned int val)
+{
+	if (val > 0)
+		val += ofs;
+	return snd_hda_codec_amp_update(codec, nid, ch, dir, idx,
+					HDA_AMP_VOLMASK, val);
+}
+
 int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol,
 				 struct snd_ctl_elem_value *ucontrol)
 {
@@ -1146,14 +1175,13 @@ int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol,
 	int chs = get_amp_channels(kcontrol);
 	int dir = get_amp_direction(kcontrol);
 	int idx = get_amp_index(kcontrol);
+	unsigned int ofs = get_amp_offset(kcontrol);
 	long *valp = ucontrol->value.integer.value;
 
 	if (chs & 1)
-		*valp++ = snd_hda_codec_amp_read(codec, nid, 0, dir, idx)
-			& HDA_AMP_VOLMASK;
+		*valp++ = read_amp_value(codec, nid, 0, dir, idx, ofs);
 	if (chs & 2)
-		*valp = snd_hda_codec_amp_read(codec, nid, 1, dir, idx)
-			& HDA_AMP_VOLMASK;
+		*valp = read_amp_value(codec, nid, 1, dir, idx, ofs);
 	return 0;
 }
 EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_get);
@@ -1166,18 +1194,17 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol,
 	int chs = get_amp_channels(kcontrol);
 	int dir = get_amp_direction(kcontrol);
 	int idx = get_amp_index(kcontrol);
+	unsigned int ofs = get_amp_offset(kcontrol);
 	long *valp = ucontrol->value.integer.value;
 	int change = 0;
 
 	snd_hda_power_up(codec);
 	if (chs & 1) {
-		change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx,
-						  0x7f, *valp);
+		change = update_amp_value(codec, nid, 0, dir, idx, ofs, *valp);
 		valp++;
 	}
 	if (chs & 2)
-		change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx,
-						   0x7f, *valp);
+		change |= update_amp_value(codec, nid, 1, dir, idx, ofs, *valp);
 	snd_hda_power_down(codec);
 	return change;
 }
@@ -1189,6 +1216,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag,
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	hda_nid_t nid = get_amp_nid(kcontrol);
 	int dir = get_amp_direction(kcontrol);
+	unsigned int ofs = get_amp_offset(kcontrol);
 	u32 caps, val1, val2;
 
 	if (size < 4 * sizeof(unsigned int))
@@ -1197,6 +1225,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag,
 	val2 = (caps & AC_AMPCAP_STEP_SIZE) >> AC_AMPCAP_STEP_SIZE_SHIFT;
 	val2 = (val2 + 1) * 25;
 	val1 = -((caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT);
+	val1 += ofs;
 	val1 = ((int)val1) * ((int)val2);
 	if (put_user(SNDRV_CTL_TLVT_DB_SCALE, _tlv))
 		return -EFAULT;
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 1dd8716c387..d53ce1f8541 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -26,8 +26,10 @@
 /*
  * for mixer controls
  */
+#define HDA_COMPOSE_AMP_VAL_OFS(nid,chs,idx,dir,ofs)		\
+	((nid) | ((chs)<<16) | ((dir)<<18) | ((idx)<<19) | ((ofs)<<23))
 #define HDA_COMPOSE_AMP_VAL(nid,chs,idx,dir) \
-	((nid) | ((chs)<<16) | ((dir)<<18) | ((idx)<<19))
+	HDA_COMPOSE_AMP_VAL_OFS(nid, chs, idx, dir, 0)
 /* mono volume with index (index=0,1,...) (channel=1,2) */
 #define HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \
 	{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx,  \
@@ -456,6 +458,7 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec,
 #define get_amp_channels(kc)	(((kc)->private_value >> 16) & 0x3)
 #define get_amp_direction(kc)	(((kc)->private_value >> 18) & 0x1)
 #define get_amp_index(kc)	(((kc)->private_value >> 19) & 0xf)
+#define get_amp_offset(kc)	(((kc)->private_value >> 23) & 0x3f)
 
 /*
  * CEA Short Audio Descriptor data
-- 
cgit v1.2.3


From 7c7767ebe2fa847c91a0dd5551ca422aba359473 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 20 Jan 2009 15:28:38 +0100
Subject: ALSA: hda - Halve too large volume scales for STAC/IDT codecs

STAC/IDT codecs have often too large volume scales such as -96dB,
and exposing this as is results in too large scale in percentage
representation.

This patch adds the check of the volume scale and halves the
volume range if it's too large automatically.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 41 +++++++++++++++++++++++++++++++++--------
 1 file changed, 33 insertions(+), 8 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index a4d4afe6b4f..c2d4abee3b0 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -166,6 +166,7 @@ struct sigmatel_spec {
 	unsigned int alt_switch: 1;
 	unsigned int hp_detect: 1;
 	unsigned int spdif_mute: 1;
+	unsigned int check_volume_offset:1;
 
 	/* gpio lines */
 	unsigned int eapd_mask;
@@ -202,6 +203,8 @@ struct sigmatel_spec {
 	hda_nid_t hp_dacs[5];
 	hda_nid_t speaker_dacs[5];
 
+	int volume_offset;
+
 	/* capture */
 	hda_nid_t *adc_nids;
 	unsigned int num_adcs;
@@ -1297,6 +1300,8 @@ static int stac92xx_build_controls(struct hda_codec *codec)
 		unsigned int vmaster_tlv[4];
 		snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0],
 					HDA_OUTPUT, vmaster_tlv);
+		/* correct volume offset */
+		vmaster_tlv[2] += vmaster_tlv[3] * spec->volume_offset;
 		err = snd_hda_add_vmaster(codec, "Master Playback Volume",
 					  vmaster_tlv, slave_vols);
 		if (err < 0)
@@ -2980,14 +2985,34 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec)
 }
 
 /* create volume control/switch for the given prefx type */
-static int create_controls(struct sigmatel_spec *spec, const char *pfx, hda_nid_t nid, int chs)
+static int create_controls(struct hda_codec *codec, const char *pfx,
+			   hda_nid_t nid, int chs)
 {
+	struct sigmatel_spec *spec = codec->spec;
 	char name[32];
 	int err;
 
+	if (!spec->check_volume_offset) {
+		unsigned int caps, step, nums, db_scale;
+		caps = query_amp_caps(codec, nid, HDA_OUTPUT);
+		step = (caps & AC_AMPCAP_STEP_SIZE) >>
+			AC_AMPCAP_STEP_SIZE_SHIFT;
+		step = (step + 1) * 25; /* in .01dB unit */
+		nums = (caps & AC_AMPCAP_NUM_STEPS) >>
+			AC_AMPCAP_NUM_STEPS_SHIFT;
+		db_scale = nums * step;
+		/* if dB scale is over -64dB, and finer enough,
+		 * let's reduce it to half
+		 */
+		if (db_scale > 6400 && nums >= 0x1f)
+			spec->volume_offset = nums / 2;
+		spec->check_volume_offset = 1;
+	}
+
 	sprintf(name, "%s Playback Volume", pfx);
 	err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, name,
-				   HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT));
+		HDA_COMPOSE_AMP_VAL_OFS(nid, chs, 0, HDA_OUTPUT,
+					spec->volume_offset));
 	if (err < 0)
 		return err;
 	sprintf(name, "%s Playback Switch", pfx);
@@ -3053,10 +3078,10 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec,
 		nid = spec->multiout.dac_nids[i];
 		if (i == 2) {
 			/* Center/LFE */
-			err = create_controls(spec, "Center", nid, 1);
+			err = create_controls(codec, "Center", nid, 1);
 			if (err < 0)
 				return err;
-			err = create_controls(spec, "LFE", nid, 2);
+			err = create_controls(codec, "LFE", nid, 2);
 			if (err < 0)
 				return err;
 
@@ -3084,7 +3109,7 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec,
 					break;
 				}
 			}
-			err = create_controls(spec, name, nid, 3);
+			err = create_controls(codec, name, nid, 3);
 			if (err < 0)
 				return err;
 		}
@@ -3139,7 +3164,7 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec,
 		nid = spec->hp_dacs[i];
 		if (!nid)
 			continue;
-		err = create_controls(spec, pfxs[nums++], nid, 3);
+		err = create_controls(codec, pfxs[nums++], nid, 3);
 		if (err < 0)
 			return err;
 	}
@@ -3153,7 +3178,7 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec,
 		nid = spec->speaker_dacs[i];
 		if (!nid)
 			continue;
-		err = create_controls(spec, pfxs[nums++], nid, 3);
+		err = create_controls(codec, pfxs[nums++], nid, 3);
 		if (err < 0)
 			return err;
 	}
@@ -3729,7 +3754,7 @@ static int stac9200_auto_create_lfe_ctls(struct hda_codec *codec,
 	}
 
 	if (lfe_pin) {
-		err = create_controls(spec, "LFE", lfe_pin, 1);
+		err = create_controls(codec, "LFE", lfe_pin, 1);
 		if (err < 0)
 			return err;
 	}
-- 
cgit v1.2.3


From 89ce9e87083216389d2ff5740cc60f835537d8d0 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 20 Jan 2009 17:15:57 +0100
Subject: ALSA: hda - Add debug prints for digital I/O pin detections

Add the debug prints for digital I/O pin detections in
snd_hda_parse_pin_def_config() function.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 4 ++++
 1 file changed, 4 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index b7bba7dc7cf..c03de0bc399 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3499,6 +3499,8 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
 		   cfg->hp_pins[1], cfg->hp_pins[2],
 		   cfg->hp_pins[3], cfg->hp_pins[4]);
 	snd_printd("   mono: mono_out=0x%x\n", cfg->mono_out_pin);
+	if (cfg->dig_out_pin)
+		snd_printd("   dig-out=0x%x\n", cfg->dig_out_pin);
 	snd_printd("   inputs: mic=0x%x, fmic=0x%x, line=0x%x, fline=0x%x,"
 		   " cd=0x%x, aux=0x%x\n",
 		   cfg->input_pins[AUTO_PIN_MIC],
@@ -3507,6 +3509,8 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
 		   cfg->input_pins[AUTO_PIN_FRONT_LINE],
 		   cfg->input_pins[AUTO_PIN_CD],
 		   cfg->input_pins[AUTO_PIN_AUX]);
+	if (cfg->dig_out_pin)
+		snd_printd("   dig-in=0x%x\n", cfg->dig_in_pin);
 
 	return 0;
 }
-- 
cgit v1.2.3


From 1b52ae701fedf97f9984e73b6a1fe2444230871b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 20 Jan 2009 17:17:29 +0100
Subject: ALSA: hda - Detect non-SPDIF digital I/O

Accept non-SPDIF digital I/O pins as the digital pins.
These are usually corresponding to HDMI I/O.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index c03de0bc399..2d6f72ca014 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3390,9 +3390,11 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
 			cfg->input_pins[AUTO_PIN_AUX] = nid;
 			break;
 		case AC_JACK_SPDIF_OUT:
+		case AC_JACK_DIG_OTHER_OUT:
 			cfg->dig_out_pin = nid;
 			break;
 		case AC_JACK_SPDIF_IN:
+		case AC_JACK_DIG_OTHER_IN:
 			cfg->dig_in_pin = nid;
 			break;
 		}
-- 
cgit v1.2.3


From caa10b6e808a4d65eb0306f0006308244f2b8d79 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 20 Jan 2009 17:19:01 +0100
Subject: ALSA: hda - Improve auto-probing of STAC9872 codec

Use the standard STAC/IDT auto-probing routine for non-static
STAC9872 codec probing.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 58 ++++++++++++++++++++++++++++++++++--------
 1 file changed, 48 insertions(+), 10 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index a4d4afe6b4f..b6e797d1c21 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -5511,24 +5511,62 @@ static struct snd_pci_quirk stac9872_cfg_tbl[] = {
 	{}
 };
 
+static struct snd_kcontrol_new stac9872_mixer[] = {
+	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT),
+	STAC_INPUT_SOURCE(1),
+	{ } /* end */
+};
+
+static hda_nid_t stac9872_pin_nids[] = {
+	0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f,
+	0x11, 0x13, 0x14,
+};
+
+static hda_nid_t stac9872_adc_nids[] = {
+	0x8 /*,0x6*/
+};
+
+static hda_nid_t stac9872_mux_nids[] = {
+	0x15
+};
+
 static int patch_stac9872(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec;
-	int board_config;
 
-	board_config = snd_hda_check_board_config(codec, STAC_9872_MODELS,
-						  stac9872_models,
-						  stac9872_cfg_tbl);
-	if (board_config < 0)
-		/* unknown config, let generic-parser do its job... */
-		return snd_hda_parse_generic_codec(codec);
-	
 	spec  = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
 		return -ENOMEM;
-
 	codec->spec = spec;
-	switch (board_config) {
+
+	spec->board_config = snd_hda_check_board_config(codec, STAC_9872_MODELS,
+							stac9872_models,
+							stac9872_cfg_tbl);
+	if (spec->board_config < 0) {
+		int err;
+
+		spec->num_pins = ARRAY_SIZE(stac9872_pin_nids);
+		spec->pin_nids = stac9872_pin_nids;
+		spec->multiout.dac_nids = spec->dac_nids;
+		spec->num_adcs = ARRAY_SIZE(stac9872_adc_nids);
+		spec->adc_nids = stac9872_adc_nids;
+		spec->num_muxes = ARRAY_SIZE(stac9872_mux_nids);
+		spec->mux_nids = stac9872_mux_nids;
+		spec->mixer = stac9872_mixer;
+		spec->init = vaio_init;
+
+		err = stac92xx_parse_auto_config(codec, 0x10, 0x12);
+		if (err < 0) {
+			stac92xx_free(codec);
+			return -EINVAL;
+		}
+		spec->input_mux = &spec->private_imux;
+		codec->patch_ops = stac92xx_patch_ops;
+		return 0;
+	}
+	
+	switch (spec->board_config) {
 	case CXD9872RD_VAIO:
 	case STAC9872AK_VAIO:
 	case STAC9872K_VAIO:
-- 
cgit v1.2.3


From 41b5b01afb71226653282951965d5efa9d7b843d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 20 Jan 2009 18:21:23 +0100
Subject: ALSA: hda - Don't break the PCM creation loop

Don't break the loop in snd_hda_codec_build_pcms() even if the item
has no substreams.

It's possible that it's an empty item and the next item containing
the valid substreams (e.g. realtek codecs may create the analog
and alt-analog but no digitl streams).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 2d6f72ca014..0129e95672a 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2613,7 +2613,7 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec)
 		int dev;
 
 		if (!cpcm->stream[0].substreams && !cpcm->stream[1].substreams)
-			return 0; /* no substreams assigned */
+			continue; /* no substreams assigned */
 
 		if (!cpcm->pcm) {
 			dev = get_empty_pcm_device(codec->bus, cpcm->pcm_type);
-- 
cgit v1.2.3


From 2297bd6e526ce1469279284ffda9140f8d60ea84 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 20 Jan 2009 18:24:13 +0100
Subject: ALSA: hda - Check HDMI jack types in the auto configuration

Add dig_out_type and dig_in_type fields to autocfg struct.
A proper HDA_PCM_TYPE_* value is assigned to these fields according
to the pin-jack location type value.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 8 ++++++++
 sound/pci/hda/hda_local.h | 2 ++
 2 files changed, 10 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 0129e95672a..dd419ce43d9 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3392,10 +3392,18 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
 		case AC_JACK_SPDIF_OUT:
 		case AC_JACK_DIG_OTHER_OUT:
 			cfg->dig_out_pin = nid;
+			if (loc == AC_JACK_LOC_HDMI)
+				cfg->dig_out_type = HDA_PCM_TYPE_HDMI;
+			else
+				cfg->dig_out_type = HDA_PCM_TYPE_SPDIF;
 			break;
 		case AC_JACK_SPDIF_IN:
 		case AC_JACK_DIG_OTHER_IN:
 			cfg->dig_in_pin = nid;
+			if (loc == AC_JACK_LOC_HDMI)
+				cfg->dig_in_type = HDA_PCM_TYPE_HDMI;
+			else
+				cfg->dig_in_type = HDA_PCM_TYPE_SPDIF;
 			break;
 		}
 	}
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 1dd8716c387..a4ecd77a451 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -355,6 +355,8 @@ struct auto_pin_cfg {
 	hda_nid_t dig_out_pin;
 	hda_nid_t dig_in_pin;
 	hda_nid_t mono_out_pin;
+	int dig_out_type; /* HDA_PCM_TYPE_XXX */
+	int dig_in_type; /* HDA_PCM_TYPE_XXX */
 };
 
 #define get_defcfg_connect(cfg) \
-- 
cgit v1.2.3


From 8c441982fdc00f77b7aa609061c6411f47bcceda Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 20 Jan 2009 18:30:20 +0100
Subject: ALSA: hda - Assign proper digital I/O type for STAC/IDT

Assign the proper PCM digital I/O type (HDA_PCM_TYPE_*) for the digital
I/O on STAC/IDT codecs.  HDA_PCM_TYPE_HDMI is assigned for the HDMI I/O.

A similar framework is implemented to patch_realtek.c, but it's not
set up and still using only HDA_PCM_TYPE_SPDIF yet.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c  | 6 +++++-
 sound/pci/hda/patch_sigmatel.c | 2 +-
 2 files changed, 6 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 5d249a547fb..4fdae06162e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -269,6 +269,7 @@ struct alc_spec {
 					 * dig_out_nid and hp_nid are optional
 					 */
 	hda_nid_t alt_dac_nid;
+	int dig_out_type;
 
 	/* capture */
 	unsigned int num_adc_nids;
@@ -3087,7 +3088,10 @@ static int alc_build_pcms(struct hda_codec *codec)
 		codec->num_pcms = 2;
 		info = spec->pcm_rec + 1;
 		info->name = spec->stream_name_digital;
-		info->pcm_type = HDA_PCM_TYPE_SPDIF;
+		if (spec->dig_out_type)
+			info->pcm_type = spec->dig_out_type;
+		else
+			info->pcm_type = HDA_PCM_TYPE_SPDIF;
 		if (spec->multiout.dig_out_nid &&
 		    spec->stream_digital_playback) {
 			info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_digital_playback);
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index b6e797d1c21..1dd448e85bc 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2553,7 +2553,7 @@ static int stac92xx_build_pcms(struct hda_codec *codec)
 		codec->num_pcms++;
 		info++;
 		info->name = "STAC92xx Digital";
-		info->pcm_type = HDA_PCM_TYPE_SPDIF;
+		info->pcm_type = spec->autocfg.dig_out_type;
 		if (spec->multiout.dig_out_nid) {
 			info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_digital_playback;
 			info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid;
-- 
cgit v1.2.3


From e64f14f4e570d6ec5bc88abac92a3a27150756d7 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 20 Jan 2009 18:32:55 +0100
Subject: ALSA: hda - Allow digital-only I/O on ALC262 codec

Some laptops like VAIO have multiple codecs and uses ALC262 only for
the SPIDF output without analog I/O.  So far, the codec-parser assumes
the presence of analog I/O and returned an error for such a case.

This patch adds some hacks to allow the digital-only configuration for
ALC262.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 43 +++++++++++++++++++++++++++++++++----------
 1 file changed, 33 insertions(+), 10 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 4fdae06162e..4cfa78c5439 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -306,6 +306,9 @@ struct alc_spec {
 	unsigned int jack_present: 1;
 	unsigned int master_sw: 1;
 
+	/* other flags */
+	unsigned int no_analog :1; /* digital I/O only */
+
 	/* for virtual master */
 	hda_nid_t vmaster_nid;
 #ifdef CONFIG_SND_HDA_POWER_SAVE
@@ -2019,11 +2022,13 @@ static int alc_build_controls(struct hda_codec *codec)
 						    spec->multiout.dig_out_nid);
 		if (err < 0)
 			return err;
-		err = snd_hda_create_spdif_share_sw(codec,
-						    &spec->multiout);
-		if (err < 0)
-			return err;
-		spec->multiout.share_spdif = 1;
+		if (!spec->no_analog) {
+			err = snd_hda_create_spdif_share_sw(codec,
+							    &spec->multiout);
+			if (err < 0)
+				return err;
+			spec->multiout.share_spdif = 1;
+		}
 	}
 	if (spec->dig_in_nid) {
 		err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
@@ -2032,7 +2037,8 @@ static int alc_build_controls(struct hda_codec *codec)
 	}
 
 	/* if we have no master control, let's create it */
-	if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) {
+	if (!spec->no_analog &&
+	    !snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) {
 		unsigned int vmaster_tlv[4];
 		snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid,
 					HDA_OUTPUT, vmaster_tlv);
@@ -2041,7 +2047,8 @@ static int alc_build_controls(struct hda_codec *codec)
 		if (err < 0)
 			return err;
 	}
-	if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
+	if (!spec->no_analog &&
+	    !snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
 		err = snd_hda_add_vmaster(codec, "Master Playback Switch",
 					  NULL, alc_slave_sws);
 		if (err < 0)
@@ -3060,6 +3067,9 @@ static int alc_build_pcms(struct hda_codec *codec)
 	codec->num_pcms = 1;
 	codec->pcm_info = info;
 
+	if (spec->no_analog)
+		goto skip_analog;
+
 	info->name = spec->stream_name_analog;
 	if (spec->stream_analog_playback) {
 		if (snd_BUG_ON(!spec->multiout.dac_nids))
@@ -3083,6 +3093,7 @@ static int alc_build_pcms(struct hda_codec *codec)
 		}
 	}
 
+ skip_analog:
 	/* SPDIF for stream index #1 */
 	if (spec->multiout.dig_out_nid || spec->dig_in_nid) {
 		codec->num_pcms = 2;
@@ -3106,6 +3117,9 @@ static int alc_build_pcms(struct hda_codec *codec)
 		codec->spdif_status_reset = 1;
 	}
 
+	if (spec->no_analog)
+		return 0;
+
 	/* If the use of more than one ADC is requested for the current
 	 * model, configure a second analog capture-only PCM.
 	 */
@@ -10468,8 +10482,14 @@ static int alc262_parse_auto_config(struct hda_codec *codec)
 					   alc262_ignore);
 	if (err < 0)
 		return err;
-	if (!spec->autocfg.line_outs)
+	if (!spec->autocfg.line_outs) {
+		if (spec->autocfg.dig_out_pin || spec->autocfg.dig_in_pin) {
+			spec->multiout.max_channels = 2;
+			spec->no_analog = 1;
+			goto dig_only;
+		}
 		return 0; /* can't find valid BIOS pin config */
+	}
 	err = alc262_auto_create_multi_out_ctls(spec, &spec->autocfg);
 	if (err < 0)
 		return err;
@@ -10479,8 +10499,11 @@ static int alc262_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_out_pin)
+ dig_only:
+	if (spec->autocfg.dig_out_pin) {
 		spec->multiout.dig_out_nid = ALC262_DIGOUT_NID;
+		spec->dig_out_type = spec->autocfg.dig_out_type;
+	}
 	if (spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = ALC262_DIGIN_NID;
 
@@ -10875,7 +10898,7 @@ static int patch_alc262(struct hda_codec *codec)
 			spec->capsrc_nids = alc262_capsrc_nids;
 		}
 	}
-	if (!spec->cap_mixer)
+	if (!spec->cap_mixer && !spec->no_analog)
 		set_capture_mixer(spec);
 
 	spec->vmaster_nid = 0x0c;
-- 
cgit v1.2.3


From 1e137f929bb490ff615ea475ac3904d58b0cdd5e Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 21 Jan 2009 07:41:22 +0100
Subject: ALSA: hda - Clean up old VAIO hack codes for STAC9872

Get rid of old VAIO static hack codes for STAC9872 and use the BIOS
auto-parser for all models.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 238 ++++-------------------------------------
 1 file changed, 21 insertions(+), 217 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 775f8581906..dbe8b1201ef 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -5351,172 +5351,12 @@ static int patch_stac9205(struct hda_codec *codec)
  * STAC9872 hack
  */
 
-/* static config for Sony VAIO FE550G and Sony VAIO AR */
-static hda_nid_t vaio_dacs[] = { 0x2 };
-#define VAIO_HP_DAC	0x5
-static hda_nid_t vaio_adcs[] = { 0x8 /*,0x6*/ };
-static hda_nid_t vaio_mux_nids[] = { 0x15 };
-
-static struct hda_input_mux vaio_mux = {
-	.num_items = 3,
-	.items = {
-		/* { "HP", 0x0 }, */
-		{ "Mic Jack", 0x1 },
-		{ "Internal Mic", 0x2 },
-		{ "PCM", 0x3 },
-	}
-};
-
-static struct hda_verb vaio_init[] = {
-	{0x0a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP <- 0x2 */
-	{0x0a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | STAC_HP_EVENT},
-	{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Speaker <- 0x5 */
-	{0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? (<- 0x2) */
-	{0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */
-	{0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */
-	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* capture sw/vol -> 0x8 */
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* CD-in -> 0x6 */
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Mic-in -> 0x9 */
-	{}
-};
-
-static struct hda_verb vaio_ar_init[] = {
-	{0x0a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP <- 0x2 */
-	{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Speaker <- 0x5 */
-	{0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? (<- 0x2) */
-	{0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */
-/*	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },*/ /* Optical Out */
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */
+static struct hda_verb stac9872_core_init[] = {
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */
-	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */
-/*	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},*/ /* Optical Out */
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* capture sw/vol -> 0x8 */
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* CD-in -> 0x6 */
 	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Mic-in -> 0x9 */
 	{}
 };
 
-static struct snd_kcontrol_new vaio_mixer[] = {
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x02, 0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x02, 0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x05, 0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Speaker Playback Switch", 0x05, 0, HDA_OUTPUT),
-	/* HDA_CODEC_VOLUME("CD Capture Volume", 0x07, 0, HDA_INPUT), */
-	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.count = 1,
-		.info = stac92xx_mux_enum_info,
-		.get = stac92xx_mux_enum_get,
-		.put = stac92xx_mux_enum_put,
-	},
-	{}
-};
-
-static struct snd_kcontrol_new vaio_ar_mixer[] = {
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x02, 0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x02, 0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x05, 0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Speaker Playback Switch", 0x05, 0, HDA_OUTPUT),
-	/* HDA_CODEC_VOLUME("CD Capture Volume", 0x07, 0, HDA_INPUT), */
-	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT),
-	/*HDA_CODEC_MUTE("Optical Out Switch", 0x10, 0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Optical Out Volume", 0x10, 0, HDA_OUTPUT),*/
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.count = 1,
-		.info = stac92xx_mux_enum_info,
-		.get = stac92xx_mux_enum_get,
-		.put = stac92xx_mux_enum_put,
-	},
-	{}
-};
-
-static struct hda_codec_ops stac9872_patch_ops = {
-	.build_controls = stac92xx_build_controls,
-	.build_pcms = stac92xx_build_pcms,
-	.init = stac92xx_init,
-	.free = stac92xx_free,
-#ifdef SND_HDA_NEEDS_RESUME
-	.resume = stac92xx_resume,
-#endif
-};
-
-static int stac9872_vaio_init(struct hda_codec *codec)
-{
-	int err;
-
-	err = stac92xx_init(codec);
-	if (err < 0)
-		return err;
-	if (codec->patch_ops.unsol_event)
-		codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26);
-	return 0;
-}
-
-static void stac9872_vaio_hp_detect(struct hda_codec *codec, unsigned int res)
-{
-	if (get_pin_presence(codec, 0x0a)) {
-		stac92xx_reset_pinctl(codec, 0x0f, AC_PINCTL_OUT_EN);
-		stac92xx_set_pinctl(codec, 0x0a, AC_PINCTL_OUT_EN);
-	} else {
-		stac92xx_reset_pinctl(codec, 0x0a, AC_PINCTL_OUT_EN);
-		stac92xx_set_pinctl(codec, 0x0f, AC_PINCTL_OUT_EN);
-	}
-} 
-
-static void stac9872_vaio_unsol_event(struct hda_codec *codec, unsigned int res)
-{
-	switch (res >> 26) {
-	case STAC_HP_EVENT:
-		stac9872_vaio_hp_detect(codec, res);
-		break;
-	}
-}
-
-static struct hda_codec_ops stac9872_vaio_patch_ops = {
-	.build_controls = stac92xx_build_controls,
-	.build_pcms = stac92xx_build_pcms,
-	.init = stac9872_vaio_init,
-	.free = stac92xx_free,
-	.unsol_event = stac9872_vaio_unsol_event,
-#ifdef CONFIG_PM
-	.resume = stac92xx_resume,
-#endif
-};
-
-enum { /* FE and SZ series. id=0x83847661 and subsys=0x104D0700 or 104D1000. */
-       CXD9872RD_VAIO,
-       /* Unknown. id=0x83847662 and subsys=0x104D1200 or 104D1000. */
-       STAC9872AK_VAIO, 
-       /* Unknown. id=0x83847661 and subsys=0x104D1200. */
-       STAC9872K_VAIO,
-       /* AR Series. id=0x83847664 and subsys=104D1300 */
-       CXD9872AKD_VAIO,
-       STAC_9872_MODELS,
-};
-
-static const char *stac9872_models[STAC_9872_MODELS] = {
-	[CXD9872RD_VAIO]	= "vaio",
-	[CXD9872AKD_VAIO]	= "vaio-ar",
-};
-
-static struct snd_pci_quirk stac9872_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x104d, 0x81e6, "Sony VAIO F/S", CXD9872RD_VAIO),
-	SND_PCI_QUIRK(0x104d, 0x81ef, "Sony VAIO F/S", CXD9872RD_VAIO),
-	SND_PCI_QUIRK(0x104d, 0x81fd, "Sony VAIO AR", CXD9872AKD_VAIO),
-	SND_PCI_QUIRK(0x104d, 0x8205, "Sony VAIO AR", CXD9872AKD_VAIO),
-	{}
-};
-
 static struct snd_kcontrol_new stac9872_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT),
@@ -5540,72 +5380,36 @@ static hda_nid_t stac9872_mux_nids[] = {
 static int patch_stac9872(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec;
+	int err;
 
 	spec  = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
 		return -ENOMEM;
 	codec->spec = spec;
 
+#if 0 /* no model right now */
 	spec->board_config = snd_hda_check_board_config(codec, STAC_9872_MODELS,
 							stac9872_models,
 							stac9872_cfg_tbl);
-	if (spec->board_config < 0) {
-		int err;
-
-		spec->num_pins = ARRAY_SIZE(stac9872_pin_nids);
-		spec->pin_nids = stac9872_pin_nids;
-		spec->multiout.dac_nids = spec->dac_nids;
-		spec->num_adcs = ARRAY_SIZE(stac9872_adc_nids);
-		spec->adc_nids = stac9872_adc_nids;
-		spec->num_muxes = ARRAY_SIZE(stac9872_mux_nids);
-		spec->mux_nids = stac9872_mux_nids;
-		spec->mixer = stac9872_mixer;
-		spec->init = vaio_init;
-
-		err = stac92xx_parse_auto_config(codec, 0x10, 0x12);
-		if (err < 0) {
-			stac92xx_free(codec);
-			return -EINVAL;
-		}
-		spec->input_mux = &spec->private_imux;
-		codec->patch_ops = stac92xx_patch_ops;
-		return 0;
-	}
-	
-	switch (spec->board_config) {
-	case CXD9872RD_VAIO:
-	case STAC9872AK_VAIO:
-	case STAC9872K_VAIO:
-		spec->mixer = vaio_mixer;
-		spec->init = vaio_init;
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = ARRAY_SIZE(vaio_dacs);
-		spec->multiout.dac_nids = vaio_dacs;
-		spec->multiout.hp_nid = VAIO_HP_DAC;
-		spec->num_adcs = ARRAY_SIZE(vaio_adcs);
-		spec->adc_nids = vaio_adcs;
-		spec->num_pwrs = 0;
-		spec->input_mux = &vaio_mux;
-		spec->mux_nids = vaio_mux_nids;
-		codec->patch_ops = stac9872_vaio_patch_ops;
-		break;
-	
-	case CXD9872AKD_VAIO:
-		spec->mixer = vaio_ar_mixer;
-		spec->init = vaio_ar_init;
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = ARRAY_SIZE(vaio_dacs);
-		spec->multiout.dac_nids = vaio_dacs;
-		spec->multiout.hp_nid = VAIO_HP_DAC;
-		spec->num_adcs = ARRAY_SIZE(vaio_adcs);
-		spec->num_pwrs = 0;
-		spec->adc_nids = vaio_adcs;
-		spec->input_mux = &vaio_mux;
-		spec->mux_nids = vaio_mux_nids;
-		codec->patch_ops = stac9872_patch_ops;
-		break;
-	}
+#endif
 
+	spec->num_pins = ARRAY_SIZE(stac9872_pin_nids);
+	spec->pin_nids = stac9872_pin_nids;
+	spec->multiout.dac_nids = spec->dac_nids;
+	spec->num_adcs = ARRAY_SIZE(stac9872_adc_nids);
+	spec->adc_nids = stac9872_adc_nids;
+	spec->num_muxes = ARRAY_SIZE(stac9872_mux_nids);
+	spec->mux_nids = stac9872_mux_nids;
+	spec->mixer = stac9872_mixer;
+	spec->init = stac9872_core_init;
+
+	err = stac92xx_parse_auto_config(codec, 0x10, 0x12);
+	if (err < 0) {
+		stac92xx_free(codec);
+		return -EINVAL;
+	}
+	spec->input_mux = &spec->private_imux;
+	codec->patch_ops = stac92xx_patch_ops;
 	return 0;
 }
 
-- 
cgit v1.2.3


From 8ce8419829998c91b33200894a0db5e1441d6952 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 22 Jan 2009 16:59:20 +0100
Subject: ALSA: hda - Avoid to set the pin control again if already set

Check the present pin control bit and avoid the write if it's already
set in patch_sigmatel.c.  This will reduce the number of verb execs at
jack plugging.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 20 +++++++++++++-------
 1 file changed, 13 insertions(+), 7 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 0fa6c593d1d..11634a4478e 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4126,7 +4126,9 @@ static void stac92xx_free(struct hda_codec *codec)
 static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid,
 				unsigned int flag)
 {
-	unsigned int pin_ctl = snd_hda_codec_read(codec, nid,
+	unsigned int old_ctl, pin_ctl;
+
+	pin_ctl = snd_hda_codec_read(codec, nid,
 			0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00);
 
 	if (pin_ctl & AC_PINCTL_IN_EN) {
@@ -4140,14 +4142,17 @@ static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid,
 			return;
 	}
 
+	old_ctl = pin_ctl;
 	/* if setting pin direction bits, clear the current
 	   direction bits first */
 	if (flag & (AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN))
 		pin_ctl &= ~(AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN);
 	
-	snd_hda_codec_write_cache(codec, nid, 0,
-			AC_VERB_SET_PIN_WIDGET_CONTROL,
-			pin_ctl | flag);
+	pin_ctl |= flag;
+	if (old_ctl != pin_ctl)
+		snd_hda_codec_write_cache(codec, nid, 0,
+					  AC_VERB_SET_PIN_WIDGET_CONTROL,
+					  pin_ctl);
 }
 
 static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid,
@@ -4155,9 +4160,10 @@ static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid,
 {
 	unsigned int pin_ctl = snd_hda_codec_read(codec, nid,
 			0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00);
-	snd_hda_codec_write_cache(codec, nid, 0,
-			AC_VERB_SET_PIN_WIDGET_CONTROL,
-			pin_ctl & ~flag);
+	if (pin_ctl & flag)
+		snd_hda_codec_write_cache(codec, nid, 0,
+					  AC_VERB_SET_PIN_WIDGET_CONTROL,
+					  pin_ctl & ~flag);
 }
 
 static int get_pin_presence(struct hda_codec *codec, hda_nid_t nid)
-- 
cgit v1.2.3


From d9a4268ee92ba1a2355c892a3add1fa66856b510 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 22 Jan 2009 17:40:18 +0100
Subject: ALSA: hda - Add quirk for Gateway %1616 laptop

Gateway T1616 laptop needs EAPD always on while the current STAC9205
code turns off per HP plug.  Added a new model "eapd" to keep it on.

Reference: Novell bnc#467597
	https://bugzilla.novell.com/show_bug.cgi?id=467597

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 10 +++++++++-
 1 file changed, 9 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 3f85731055c..ed2fa431b03 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -66,6 +66,7 @@ enum {
 	STAC_9205_DELL_M42,
 	STAC_9205_DELL_M43,
 	STAC_9205_DELL_M44,
+	STAC_9205_EAPD,
 	STAC_9205_MODELS
 };
 
@@ -2240,6 +2241,7 @@ static unsigned int *stac9205_brd_tbl[STAC_9205_MODELS] = {
 	[STAC_9205_DELL_M42] = dell_9205_m42_pin_configs,
 	[STAC_9205_DELL_M43] = dell_9205_m43_pin_configs,
 	[STAC_9205_DELL_M44] = dell_9205_m44_pin_configs,
+	[STAC_9205_EAPD] = NULL,
 };
 
 static const char *stac9205_models[STAC_9205_MODELS] = {
@@ -2247,12 +2249,14 @@ static const char *stac9205_models[STAC_9205_MODELS] = {
 	[STAC_9205_DELL_M42] = "dell-m42",
 	[STAC_9205_DELL_M43] = "dell-m43",
 	[STAC_9205_DELL_M44] = "dell-m44",
+	[STAC_9205_EAPD] = "eapd",
 };
 
 static struct snd_pci_quirk stac9205_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
 		      "DFI LanParty", STAC_9205_REF),
+	/* Dell */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f1,
 		      "unknown Dell", STAC_9205_DELL_M42),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f2,
@@ -2283,6 +2287,8 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = {
 		      "Dell Inspiron", STAC_9205_DELL_M44),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0228,
 		      "Dell Vostro 1500", STAC_9205_DELL_M42),
+	/* Gateway */
+	SND_PCI_QUIRK(0x107b, 0x0565, "Gateway T1616", STAC_9205_EAPD),
 	{} /* terminator */
 };
 
@@ -5320,7 +5326,9 @@ static int patch_stac9205(struct hda_codec *codec)
 
 	spec->aloopback_mask = 0x40;
 	spec->aloopback_shift = 0;
-	spec->eapd_switch = 1;
+	/* Turn on/off EAPD per HP plugging */
+	if (spec->board_config != STAC_9205_EAPD)
+		spec->eapd_switch = 1;
 	spec->multiout.dac_nids = spec->dac_nids;
 	
 	switch (spec->board_config){
-- 
cgit v1.2.3


From 577aa2c195045599275b54356969ae19f34e7a66 Mon Sep 17 00:00:00 2001
From: Matthew Ranostay <mranostay@embeddedalley.com>
Date: Thu, 22 Jan 2009 22:55:44 -0500
Subject: ALSA: hda: add reference board SND_PCI_QUIRK

Add another LanParty reference board SND_PCI_QUIRK to quirk lists of
all codec families.

Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 15 +++++++++++++++
 1 file changed, 15 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 212d8c09a67..3fbe22053b3 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1517,6 +1517,8 @@ static struct snd_pci_quirk stac9200_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
 		      "DFI LanParty", STAC_REF),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101,
+		      "DFI LanParty", STAC_REF),
 	/* Dell laptops have BIOS problem */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01a8,
 		      "unknown Dell", STAC_9200_DELL_D21),
@@ -1666,6 +1668,7 @@ static struct snd_pci_quirk stac925x_codec_id_cfg_tbl[] = {
 static struct snd_pci_quirk stac925x_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_REF),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_REF),
 	SND_PCI_QUIRK(0x8384, 0x7632, "Stac9202 Reference Board", STAC_REF),
 
 	/* Default table for unknown ID */
@@ -1709,6 +1712,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
 				"DFI LanParty", STAC_92HD73XX_REF),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101,
+				"DFI LanParty", STAC_92HD73XX_REF),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0254,
 				"Dell Studio 1535", STAC_DELL_M6_DMIC),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0255,
@@ -1753,6 +1758,8 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
 		      "DFI LanParty", STAC_92HD83XXX_REF),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101,
+		      "DFI LanParty", STAC_92HD83XXX_REF),
 	{} /* terminator */
 };
 
@@ -1802,6 +1809,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
 		      "DFI LanParty", STAC_92HD71BXX_REF),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101,
+		      "DFI LanParty", STAC_92HD71BXX_REF),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f2,
 		      "HP dv5", STAC_HP_M4),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f4,
@@ -1992,6 +2001,8 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
 		      "DFI LanParty", STAC_D945_REF),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101,
+		      "DFI LanParty", STAC_D945_REF),
 	/* Intel 945G based systems */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0101,
 		      "Intel D945G", STAC_D945GTP3),
@@ -2148,6 +2159,8 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
 		      "DFI LanParty", STAC_D965_REF),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101,
+		      "DFI LanParty", STAC_D965_REF),
 	 /* Intel 946 based systems */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x3d01, "Intel D946", STAC_D965_3ST),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0xa301, "Intel D946", STAC_D965_3ST),
@@ -2259,6 +2272,8 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
 		      "DFI LanParty", STAC_9205_REF),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101,
+		      "DFI LanParty", STAC_9205_REF),
 	/* Dell */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f1,
 		      "unknown Dell", STAC_9205_DELL_M42),
-- 
cgit v1.2.3


From 8056d47e77a0f7e3c99c114deab4859d31496075 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 23 Jan 2009 09:09:43 +0100
Subject: ALSA: hda - Add model=ref for Intel board with STAC9221

An intel board (8086:0204) works only with model=ref.

Reference: Novell bug #406529
	https://bugzilla.novell.com/show_bug.cgi?id=406529

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 3fbe22053b3..4ee9f7fc772 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2056,6 +2056,9 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = {
 		      "Intel D945P", STAC_D945GTP3),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0707,
 		      "Intel D945P", STAC_D945GTP5),
+	/* other intel */
+	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0204,
+		      "Intel D945", STAC_D945_REF),
 	/* other systems  */
 	/* Apple Intel Mac (Mac Mini, MacBook, MacBook Pro...) */
 	SND_PCI_QUIRK(0x8384, 0x7680,
-- 
cgit v1.2.3


From e3c75964666a27cec46d2cccf2d9806336becd48 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 23 Jan 2009 11:57:22 +0100
Subject: ALSA: hda - Create "Input Source" control dynamically for STAC/IDT

Instead of fixed kcontrol_new element, build "Input Source" controls
dynamically.  If the number of input-source items is 0 or 1, we don't
need to create such a control.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 78 ++++++++++++++++++++++++++++--------------
 1 file changed, 53 insertions(+), 25 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index b3c3a02a422..80a4c288b31 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -958,16 +958,6 @@ static struct hda_verb stac9205_core_init[] = {
 		.private_value = HDA_COMPOSE_AMP_VAL(nid, chs, idx, dir) \
 	}
 
-#define STAC_INPUT_SOURCE(cnt) \
-	{ \
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
-		.name = "Input Source", \
-		.count = cnt, \
-		.info = stac92xx_mux_enum_info, \
-		.get = stac92xx_mux_enum_get, \
-		.put = stac92xx_mux_enum_put, \
-	}
-
 #define STAC_ANALOG_LOOPBACK(verb_read, verb_write, cnt) \
 	{ \
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
@@ -982,7 +972,6 @@ static struct hda_verb stac9205_core_init[] = {
 static struct snd_kcontrol_new stac9200_mixer[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0xb, 0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT),
-	STAC_INPUT_SOURCE(1),
 	HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x0a, 0, HDA_OUTPUT),
 	{ } /* end */
@@ -1098,7 +1087,6 @@ static struct snd_kcontrol_new stac92hd83xxx_mixer[] = {
 };
 
 static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = {
-	STAC_INPUT_SOURCE(2),
 	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2),
 
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT),
@@ -1127,7 +1115,6 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = {
 };
 
 static struct snd_kcontrol_new stac92hd71bxx_mixer[] = {
-	STAC_INPUT_SOURCE(2),
 	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2),
 
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT),
@@ -1141,14 +1128,12 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = {
 static struct snd_kcontrol_new stac925x_mixer[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x0e, 0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Master Playback Switch", 0x0e, 0, HDA_OUTPUT),
-	STAC_INPUT_SOURCE(1),
 	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x14, 0, HDA_OUTPUT),
 	{ } /* end */
 };
 
 static struct snd_kcontrol_new stac9205_mixer[] = {
-	STAC_INPUT_SOURCE(2),
 	STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0, 1),
 
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1b, 0x0, HDA_INPUT),
@@ -1161,7 +1146,6 @@ static struct snd_kcontrol_new stac9205_mixer[] = {
 
 /* This needs to be generated dynamically based on sequence */
 static struct snd_kcontrol_new stac922x_mixer[] = {
-	STAC_INPUT_SOURCE(2),
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x17, 0x0, HDA_INPUT),
 
@@ -1172,7 +1156,6 @@ static struct snd_kcontrol_new stac922x_mixer[] = {
 
 
 static struct snd_kcontrol_new stac927x_mixer[] = {
-	STAC_INPUT_SOURCE(3),
 	STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1),
 
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x18, 0x0, HDA_INPUT),
@@ -2777,22 +2760,37 @@ static struct snd_kcontrol_new stac92xx_control_templates[] = {
 };
 
 /* add dynamic controls */
-static int stac92xx_add_control_temp(struct sigmatel_spec *spec,
-				     struct snd_kcontrol_new *ktemp,
-				     int idx, const char *name,
-				     unsigned long val)
+static struct snd_kcontrol_new *
+stac_control_new(struct sigmatel_spec *spec,
+		 struct snd_kcontrol_new *ktemp,
+		 const char *name)
 {
 	struct snd_kcontrol_new *knew;
 
 	snd_array_init(&spec->kctls, sizeof(*knew), 32);
 	knew = snd_array_new(&spec->kctls);
 	if (!knew)
-		return -ENOMEM;
+		return NULL;
 	*knew = *ktemp;
-	knew->index = idx;
 	knew->name = kstrdup(name, GFP_KERNEL);
-	if (!knew->name)
+	if (!knew->name) {
+		/* roolback */
+		memset(knew, 0, sizeof(*knew));
+		spec->kctls.alloced--;
+		return NULL;
+	}
+	return knew;
+}
+
+static int stac92xx_add_control_temp(struct sigmatel_spec *spec,
+				     struct snd_kcontrol_new *ktemp,
+				     int idx, const char *name,
+				     unsigned long val)
+{
+	struct snd_kcontrol_new *knew = stac_control_new(spec, ktemp, name);
+	if (!knew)
 		return -ENOMEM;
+	knew->index = idx;
 	knew->private_value = val;
 	return 0;
 }
@@ -2814,6 +2812,29 @@ static inline int stac92xx_add_control(struct sigmatel_spec *spec, int type,
 	return stac92xx_add_control_idx(spec, type, 0, name, val);
 }
 
+static struct snd_kcontrol_new stac_input_src_temp = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Input Source",
+	.info = stac92xx_mux_enum_info,
+	.get = stac92xx_mux_enum_get,
+	.put = stac92xx_mux_enum_put,
+};
+
+static int stac92xx_add_input_source(struct sigmatel_spec *spec)
+{
+	struct snd_kcontrol_new *knew;
+	struct hda_input_mux *imux = &spec->private_imux;
+
+	if (!spec->num_adcs || imux->num_items <= 1)
+		return 0; /* no need for input source control */
+	knew = stac_control_new(spec, &stac_input_src_temp,
+				stac_input_src_temp.name);
+	if (!knew)
+		return -ENOMEM;
+	knew->count = spec->num_adcs;
+	return 0;
+}
+
 /* check whether the line-input can be used as line-out */
 static hda_nid_t check_line_out_switch(struct hda_codec *codec)
 {
@@ -3699,6 +3720,10 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 			return err;
 	}
 
+	err = stac92xx_add_input_source(spec);
+	if (err < 0)
+		return err;
+
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 	if (spec->multiout.max_channels > 2)
 		spec->surr_switch = 1;
@@ -3812,6 +3837,10 @@ static int stac9200_parse_auto_config(struct hda_codec *codec)
 			return err;
 	}
 
+	err = stac92xx_add_input_source(spec);
+	if (err < 0)
+		return err;
+
 	if (spec->autocfg.dig_out_pin)
 		spec->multiout.dig_out_nid = 0x05;
 	if (spec->autocfg.dig_in_pin)
@@ -5426,7 +5455,6 @@ static struct hda_verb stac9872_core_init[] = {
 static struct snd_kcontrol_new stac9872_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT),
-	STAC_INPUT_SOURCE(1),
 	{ } /* end */
 };
 
-- 
cgit v1.2.3


From 6d6e17de4f64131e9c976fd524d73aaec268178f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 23 Jan 2009 12:33:54 +0100
Subject: ALSA: hda - Fix initial verbs for mic-boosts on AD1981HD

The mic boosts (NID 0x08 and 0x18) are input-amps, not output-amps.
Fix the initial verbs for them.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 2e7371ec2e2..9a902c2f05a 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -1407,8 +1407,8 @@ static struct hda_verb ad1981_init_verbs[] = {
 	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
 	{0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
 	/* Mic boost: 0dB */
-	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	/* Record selector: Front mic */
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x0},
 	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-- 
cgit v1.2.3


From 19a2d3e9b99ffa264adf1138bd8d8aef8909dca9 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 23 Jan 2009 12:35:25 +0100
Subject: ALSA: hda - Remove invalid amp initializations for AD1988* codecs

The ADC widgets on AD1988* codecs have no amp controls.
Remove invalid initialization verbs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c | 12 ------------
 1 file changed, 12 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 9a902c2f05a..52bc85dd6f5 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -2288,10 +2288,6 @@ static struct hda_verb ad1988_capture_init_verbs[] = {
 	{0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
 	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x1},
 	{0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
-	/* ADCs; muted */
-	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 
 	{ }
 };
@@ -2399,10 +2395,6 @@ static struct hda_verb ad1988_3stack_init_verbs[] = {
 	{0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
 	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x1},
 	{0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
-	/* ADCs; muted */
-	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 	/* Analog Mix output amp */
 	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
 	{ }
@@ -2474,10 +2466,6 @@ static struct hda_verb ad1988_laptop_init_verbs[] = {
 	{0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
 	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x1},
 	{0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
-	/* ADCs; muted */
-	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 	/* Analog Mix output amp */
 	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
 	{ }
-- 
cgit v1.2.3


From 60e388e89c9e258a51a0995ddd9e18fdebcdbe12 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 23 Jan 2009 12:37:09 +0100
Subject: ALSA: hda - Fix invalid verbs for mic-boosts on AD1884*

The mic-boosts (0x14 and 0x15) on AD1884* codecs are input-amps,
not output-amps.  Fix the invalid initialization verbs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c | 8 ++++----
 1 file changed, 4 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 52bc85dd6f5..a7298d28a0d 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3183,10 +3183,10 @@ static struct hda_verb ad1884_init_verbs[] = {
 	{0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
 	/* Port-B (front mic) pin */
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	/* Port-C (rear mic) pin */
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	/* Analog mixer; mute as default */
 	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
@@ -3601,10 +3601,10 @@ static struct hda_verb ad1884a_init_verbs[] = {
 	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 	/* Port-B (front mic) pin */
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	/* Port-C (rear line-in) pin */
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	/* Port-E (rear mic) pin */
 	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
 	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-- 
cgit v1.2.3


From 4cfb91c6d764b18e81bfb6e6779e07bcecbb197c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 23 Jan 2009 12:53:09 +0100
Subject: ALSA: hda - Fix invalid amp init for ALC268 codec

Fix some invalid AMP initializations for ALC268 codecs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 18 ++----------------
 1 file changed, 2 insertions(+), 16 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 4cfa78c5439..863ab957204 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -11279,19 +11279,13 @@ static void alc267_quanta_il1_unsol_event(struct hda_codec *codec,
 static struct hda_verb alc268_base_init_verbs[] = {
 	/* Unmute DAC0-1 and set vol = 0 */
 	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
 	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
 
 	/*
 	 * Set up output mixers (0x0c - 0x0e)
 	 */
 	/* set vol=0 to output mixers */
 	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
         {0x0e, AC_VERB_SET_CONNECT_SEL, 0x00},
 
 	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
@@ -11310,9 +11304,7 @@ static struct hda_verb alc268_base_init_verbs[] = {
 	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 
 	/* set PCBEEP vol = 0, mute connections */
 	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
@@ -11334,10 +11326,8 @@ static struct hda_verb alc268_base_init_verbs[] = {
  */
 static struct hda_verb alc268_volume_init_verbs[] = {
 	/* set output DAC */
-	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
 
 	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
 	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
@@ -11345,16 +11335,12 @@ static struct hda_verb alc268_volume_init_verbs[] = {
 	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
 	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
 
-	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
 	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
 	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 
 	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 
 	/* set PCBEEP vol = 0, mute connections */
 	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-- 
cgit v1.2.3


From 70040c07402ef5a3fad2133daffb7ee61b0d4641 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 23 Jan 2009 14:18:11 +0100
Subject: ALSA: hda - Fix wrong initial verb for AD1984 thinkpad model

The docking mic-boost (0x25) has no mute bit.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index a7298d28a0d..e934e2c187d 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3337,7 +3337,7 @@ static struct hda_verb ad1984_thinkpad_init_verbs[] = {
 	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
 	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 	/* docking mic boost */
-	{0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
 	/* Analog mixer - docking mic; mute as default */
 	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
 	/* enable EAPD bit */
-- 
cgit v1.2.3


From 55aef4508598d59c2baea7e2a3e6dfed415bbfc0 Mon Sep 17 00:00:00 2001
From: Markus Bollinger <bollinger@digigram.com>
Date: Fri, 23 Jan 2009 14:45:41 +0100
Subject: ALSA: pcxhr - add support for gpio ports and minor bug fix

- add support for gpio ports (2 GPI, 2 GPO) of pcxhr stereo cards
- minor bugfixes : allow setting clock to internal by the mixer
                   even if there is no stream (but monitoring)

Signed-off-by: Markus Bollinger <bollinger@digigram.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/pcxhr/pcxhr.c       | 41 +++++++++++++++++++++++++++++++++++++++++
 sound/pci/pcxhr/pcxhr.h       |  5 +++--
 sound/pci/pcxhr/pcxhr_mix22.c | 40 ++++++++++++++++++++++++++++++++++++----
 sound/pci/pcxhr/pcxhr_mix22.h |  3 +++
 sound/pci/pcxhr/pcxhr_mixer.c |  8 ++++++--
 5 files changed, 89 insertions(+), 8 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c
index 27cf2c28d11..ca89106f8c5 100644
--- a/sound/pci/pcxhr/pcxhr.c
+++ b/sound/pci/pcxhr/pcxhr.c
@@ -1334,6 +1334,40 @@ static void pcxhr_proc_sync(struct snd_info_entry *entry,
 	snd_iprintf(buffer, "\n");
 }
 
+static void pcxhr_proc_gpio_read(struct snd_info_entry *entry,
+				 struct snd_info_buffer *buffer)
+{
+	struct snd_pcxhr *chip = entry->private_data;
+	struct pcxhr_mgr *mgr = chip->mgr;
+	/* commands available when embedded DSP is running */
+	if (mgr->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_MAIN_INDEX)) {
+		/* gpio ports on stereo boards only available */
+		int value = 0;
+		hr222_read_gpio(mgr, 1, &value);	/* GPI */
+		snd_iprintf(buffer, "GPI: 0x%x\n", value);
+		hr222_read_gpio(mgr, 0, &value);	/* GP0 */
+		snd_iprintf(buffer, "GPO: 0x%x\n", value);
+	} else
+		snd_iprintf(buffer, "no firmware loaded\n");
+	snd_iprintf(buffer, "\n");
+}
+static void pcxhr_proc_gpo_write(struct snd_info_entry *entry,
+				 struct snd_info_buffer *buffer)
+{
+	struct snd_pcxhr *chip = entry->private_data;
+	struct pcxhr_mgr *mgr = chip->mgr;
+	char line[64];
+	int value;
+	/* commands available when embedded DSP is running */
+	if (!(mgr->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_MAIN_INDEX)))
+		return;
+	while (!snd_info_get_line(buffer, line, sizeof(line))) {
+		if (sscanf(line, "GPO: 0x%x", &value) != 1)
+			continue;
+		hr222_write_gpo(mgr, value);	/* GP0 */
+	}
+}
+
 static void __devinit pcxhr_proc_init(struct snd_pcxhr *chip)
 {
 	struct snd_info_entry *entry;
@@ -1342,6 +1376,13 @@ static void __devinit pcxhr_proc_init(struct snd_pcxhr *chip)
 		snd_info_set_text_ops(entry, chip, pcxhr_proc_info);
 	if (! snd_card_proc_new(chip->card, "sync", &entry))
 		snd_info_set_text_ops(entry, chip, pcxhr_proc_sync);
+	/* gpio available on stereo sound cards only */
+	if (chip->mgr->is_hr_stereo &&
+	    !snd_card_proc_new(chip->card, "gpio", &entry)) {
+		snd_info_set_text_ops(entry, chip, pcxhr_proc_gpio_read);
+		entry->c.text.write = pcxhr_proc_gpo_write;
+		entry->mode |= S_IWUSR;
+	}
 }
 /* end of proc interface */
 
diff --git a/sound/pci/pcxhr/pcxhr.h b/sound/pci/pcxhr/pcxhr.h
index 84131a916c9..ac9c3b3bb4e 100644
--- a/sound/pci/pcxhr/pcxhr.h
+++ b/sound/pci/pcxhr/pcxhr.h
@@ -27,8 +27,8 @@
 #include <linux/mutex.h>
 #include <sound/pcm.h>
 
-#define PCXHR_DRIVER_VERSION		0x000905	/* 0.9.5 */
-#define PCXHR_DRIVER_VERSION_STRING	"0.9.5"		/* 0.9.5 */
+#define PCXHR_DRIVER_VERSION		0x000906	/* 0.9.6 */
+#define PCXHR_DRIVER_VERSION_STRING	"0.9.6"		/* 0.9.6 */
 
 
 #define PCXHR_MAX_CARDS		6
@@ -124,6 +124,7 @@ struct pcxhr_mgr {
 
 	unsigned char xlx_cfg;		/* copy of PCXHR_XLX_CFG register */
 	unsigned char xlx_selmic;	/* copy of PCXHR_XLX_SELMIC register */
+	unsigned char dsp_reset;	/* copy of PCXHR_DSP_RESET register */
 };
 
 
diff --git a/sound/pci/pcxhr/pcxhr_mix22.c b/sound/pci/pcxhr/pcxhr_mix22.c
index ff019126b67..1cb82c0a9cb 100644
--- a/sound/pci/pcxhr/pcxhr_mix22.c
+++ b/sound/pci/pcxhr/pcxhr_mix22.c
@@ -53,6 +53,8 @@
 #define PCXHR_DSP_RESET_DSP	0x01
 #define PCXHR_DSP_RESET_MUTE	0x02
 #define PCXHR_DSP_RESET_CODEC	0x08
+#define PCXHR_DSP_RESET_GPO_OFFSET	5
+#define PCXHR_DSP_RESET_GPO_MASK	0x60
 
 /* values for PCHR_XLX_CFG register */
 #define PCXHR_CFG_SYNCDSP_MASK		0x80
@@ -81,6 +83,8 @@
 /* values for PCHR_XLX_STATUS register - READ */
 #define PCXHR_STAT_SRC_LOCK		0x01
 #define PCXHR_STAT_LEVEL_IN		0x02
+#define PCXHR_STAT_GPI_OFFSET		2
+#define PCXHR_STAT_GPI_MASK		0x0C
 #define PCXHR_STAT_MIC_CAPS		0x10
 /* values for PCHR_XLX_STATUS register - WRITE */
 #define PCXHR_STAT_FREQ_SYNC_MASK	0x01
@@ -291,10 +295,11 @@ int hr222_sub_init(struct pcxhr_mgr *mgr)
 	PCXHR_OUTPB(mgr, PCXHR_DSP_RESET,
 		    PCXHR_DSP_RESET_DSP);
 	msleep(5);
-	PCXHR_OUTPB(mgr, PCXHR_DSP_RESET,
-		    PCXHR_DSP_RESET_DSP  |
-		    PCXHR_DSP_RESET_MUTE |
-		    PCXHR_DSP_RESET_CODEC);
+	mgr->dsp_reset = PCXHR_DSP_RESET_DSP  |
+			 PCXHR_DSP_RESET_MUTE |
+			 PCXHR_DSP_RESET_CODEC;
+	PCXHR_OUTPB(mgr, PCXHR_DSP_RESET, mgr->dsp_reset);
+	/* hr222_write_gpo(mgr, 0); does the same */
 	msleep(5);
 
 	/* config AKM */
@@ -496,6 +501,33 @@ int hr222_get_external_clock(struct pcxhr_mgr *mgr,
 }
 
 
+int hr222_read_gpio(struct pcxhr_mgr *mgr, int is_gpi, int *value)
+{
+	if (is_gpi) {
+		unsigned char reg = PCXHR_INPB(mgr, PCXHR_XLX_STATUS);
+		*value = (int)(reg & PCXHR_STAT_GPI_MASK) >>
+			      PCXHR_STAT_GPI_OFFSET;
+	} else {
+		*value = (int)(mgr->dsp_reset & PCXHR_DSP_RESET_GPO_MASK) >>
+			 PCXHR_DSP_RESET_GPO_OFFSET;
+	}
+	return 0;
+}
+
+
+int hr222_write_gpo(struct pcxhr_mgr *mgr, int value)
+{
+	unsigned char reg = mgr->dsp_reset & ~PCXHR_DSP_RESET_GPO_MASK;
+
+	reg |= (unsigned char)(value << PCXHR_DSP_RESET_GPO_OFFSET) &
+	       PCXHR_DSP_RESET_GPO_MASK;
+
+	PCXHR_OUTPB(mgr, PCXHR_DSP_RESET, reg);
+	mgr->dsp_reset = reg;
+	return 0;
+}
+
+
 int hr222_update_analog_audio_level(struct snd_pcxhr *chip,
 				    int is_capture, int channel)
 {
diff --git a/sound/pci/pcxhr/pcxhr_mix22.h b/sound/pci/pcxhr/pcxhr_mix22.h
index 6b318b2f010..5a37a0007e8 100644
--- a/sound/pci/pcxhr/pcxhr_mix22.h
+++ b/sound/pci/pcxhr/pcxhr_mix22.h
@@ -32,6 +32,9 @@ int hr222_get_external_clock(struct pcxhr_mgr *mgr,
 			     enum pcxhr_clock_type clock_type,
 			     int *sample_rate);
 
+int hr222_read_gpio(struct pcxhr_mgr *mgr, int is_gpi, int *value);
+int hr222_write_gpo(struct pcxhr_mgr *mgr, int value);
+
 #define HR222_LINE_PLAYBACK_LEVEL_MIN		0	/* -25.5 dB */
 #define HR222_LINE_PLAYBACK_ZERO_LEVEL		51	/* 0.0 dB */
 #define HR222_LINE_PLAYBACK_LEVEL_MAX		99	/* +24.0 dB */
diff --git a/sound/pci/pcxhr/pcxhr_mixer.c b/sound/pci/pcxhr/pcxhr_mixer.c
index 2436e374586..fec04934462 100644
--- a/sound/pci/pcxhr/pcxhr_mixer.c
+++ b/sound/pci/pcxhr/pcxhr_mixer.c
@@ -789,11 +789,15 @@ static int pcxhr_clock_type_put(struct snd_kcontrol *kcontrol,
 	if (mgr->use_clock_type != ucontrol->value.enumerated.item[0]) {
 		mutex_lock(&mgr->setup_mutex);
 		mgr->use_clock_type = ucontrol->value.enumerated.item[0];
-		if (mgr->use_clock_type)
+		rate = 0;
+		if (mgr->use_clock_type != PCXHR_CLOCK_TYPE_INTERNAL) {
 			pcxhr_get_external_clock(mgr, mgr->use_clock_type,
 						 &rate);
-		else
+		} else {
 			rate = mgr->sample_rate;
+			if (!rate)
+				rate = 48000;
+		}
 		if (rate) {
 			pcxhr_set_clock(mgr, rate);
 			if (mgr->sample_rate)
-- 
cgit v1.2.3


From ca8d33fc9fafe373362d35107f01fba1e73fb966 Mon Sep 17 00:00:00 2001
From: Matthew Ranostay <mranostay@embeddedalley.com>
Date: Mon, 26 Jan 2009 09:33:52 -0500
Subject: ALSA: hda: 92hd71xxx disable unmute support for codecs that don't
 have input amps

Some revisions of the 92hd71xxx codec families don't have input amps
on ports 0xa, 0xd and 0xf, so probe the widget caps on port 0xa and
check for support, if found run snd_hda_sequence_write_cache() on the
stac92hd71xxx_unmute_core_init verb list.

Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 20 ++++++++++++--------
 1 file changed, 12 insertions(+), 8 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 80a4c288b31..03b26426611 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -858,26 +858,25 @@ static struct hda_verb stac92hd83xxx_core_init[] = {
 static struct hda_verb stac92hd71bxx_core_init[] = {
 	/* set master volume and direct control */
 	{ 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
-	/* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */
-	{ 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{ 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{}
 };
 
-#define HD_DISABLE_PORTF 2
+#define HD_DISABLE_PORTF 1
 static struct hda_verb stac92hd71bxx_analog_core_init[] = {
 	/* start of config #1 */
 
 	/* connect port 0f to audio mixer */
 	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2},
-	/* unmute right and left channels for node 0x0f */
-	{ 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	/* start of config #2 */
 
 	/* set master volume and direct control */
 	{ 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
-	/* unmute right and left channels for nodes 0x0a, 0xd */
+	{}
+};
+
+static struct hda_verb stac92hd71bxx_unmute_core_init[] = {
+	/* unmute right and left channels for nodes 0x0f, 0xa, 0x0d */
+	{ 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{ 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{}
@@ -4942,6 +4941,7 @@ static struct hda_input_mux stac92hd71bxx_dmux = {
 static int patch_stac92hd71bxx(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec;
+	struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init;
 	int err = 0;
 
 	spec  = kzalloc(sizeof(*spec), GFP_KERNEL);
@@ -5015,6 +5015,7 @@ again:
 
 		/* disable VSW */
 		spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF];
+		unmute_init++;
 		stac_change_pin_config(codec, 0xf, 0x40f000f0);
 		break;
 	case 0x111d7603: /* 6 Port with Analog Mixer */
@@ -5031,6 +5032,9 @@ again:
 		codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
 	}
 
+	if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP)
+		snd_hda_sequence_write_cache(codec, unmute_init);
+
 	spec->aloopback_mask = 0x50;
 	spec->aloopback_shift = 0;
 
-- 
cgit v1.2.3


From e3e9c5e7096f6379ca8fa78413b2055fa29f4530 Mon Sep 17 00:00:00 2001
From: Thadeu Lima de Souza Cascardo <cascardo@holoscopio.com>
Date: Wed, 28 Jan 2009 12:40:42 -0200
Subject: ALSA: Don't cold reset AC97 codecs in some ICH chipsets

Check in a quirk list if it should do cold reset when AC97 power saving
is enabled. Some devices do not resume properly when cold reset,
although power saving works OK.

Signed-off-by: Thadeu Lima de Souza Cascardo <cascardo@holoscopio.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/intel8x0.c | 68 +++++++++++++++++++++++++++++++++++++++-------------
 1 file changed, 52 insertions(+), 16 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 19d3391e229..b37bd268301 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -2287,23 +2287,23 @@ static void do_ali_reset(struct intel8x0 *chip)
 	iputdword(chip, ICHREG(ALI_INTERRUPTSR), 0x00000000);
 }
 
-static int snd_intel8x0_ich_chip_init(struct intel8x0 *chip, int probing)
-{
-	unsigned long end_time;
-	unsigned int cnt, status, nstatus;
-	
-	/* put logic to right state */
-	/* first clear status bits */
-	status = ICH_RCS | ICH_MCINT | ICH_POINT | ICH_PIINT;
-	if (chip->device_type == DEVICE_NFORCE)
-		status |= ICH_NVSPINT;
-	cnt = igetdword(chip, ICHREG(GLOB_STA));
-	iputdword(chip, ICHREG(GLOB_STA), cnt & status);
+#ifdef CONFIG_SND_AC97_POWER_SAVE
+static struct snd_pci_quirk ich_chip_reset_mode[] = {
+	SND_PCI_QUIRK(0x1014, 0x051f, "Thinkpad R32", 1),
+	{ } /* end */
+};
 
+static int snd_intel8x0_ich_chip_cold_reset(struct intel8x0 *chip)
+{
+	unsigned int cnt;
 	/* ACLink on, 2 channels */
+
+	if (snd_pci_quirk_lookup(chip->pci, ich_chip_reset_mode))
+		return -EIO;
+
 	cnt = igetdword(chip, ICHREG(GLOB_CNT));
 	cnt &= ~(ICH_ACLINK | ICH_PCM_246_MASK);
-#ifdef CONFIG_SND_AC97_POWER_SAVE
+
 	/* do cold reset - the full ac97 powerdown may leave the controller
 	 * in a warm state but actually it cannot communicate with the codec.
 	 */
@@ -2312,22 +2312,58 @@ static int snd_intel8x0_ich_chip_init(struct intel8x0 *chip, int probing)
 	udelay(10);
 	iputdword(chip, ICHREG(GLOB_CNT), cnt | ICH_AC97COLD);
 	msleep(1);
+	return 0;
+}
+#define snd_intel8x0_ich_chip_can_cold_reset(chip) \
+	(!snd_pci_quirk_lookup(chip->pci, ich_chip_reset_mode))
 #else
+#define snd_intel8x0_ich_chip_cold_reset(x) do { } while (0)
+#define snd_intel8x0_ich_chip_can_cold_reset(chip) (0)
+#endif
+
+static int snd_intel8x0_ich_chip_reset(struct intel8x0 *chip)
+{
+	unsigned long end_time;
+	unsigned int cnt;
+	/* ACLink on, 2 channels */
+	cnt = igetdword(chip, ICHREG(GLOB_CNT));
+	cnt &= ~(ICH_ACLINK | ICH_PCM_246_MASK);
 	/* finish cold or do warm reset */
 	cnt |= (cnt & ICH_AC97COLD) == 0 ? ICH_AC97COLD : ICH_AC97WARM;
 	iputdword(chip, ICHREG(GLOB_CNT), cnt);
 	end_time = (jiffies + (HZ / 4)) + 1;
 	do {
 		if ((igetdword(chip, ICHREG(GLOB_CNT)) & ICH_AC97WARM) == 0)
-			goto __ok;
+			return 0;
 		schedule_timeout_uninterruptible(1);
 	} while (time_after_eq(end_time, jiffies));
 	snd_printk(KERN_ERR "AC'97 warm reset still in progress? [0x%x]\n",
 		   igetdword(chip, ICHREG(GLOB_CNT)));
 	return -EIO;
+}
+
+static int snd_intel8x0_ich_chip_init(struct intel8x0 *chip, int probing)
+{
+	unsigned long end_time;
+	unsigned int status, nstatus;
+	unsigned int cnt;
+	int err;
+
+	/* put logic to right state */
+	/* first clear status bits */
+	status = ICH_RCS | ICH_MCINT | ICH_POINT | ICH_PIINT;
+	if (chip->device_type == DEVICE_NFORCE)
+		status |= ICH_NVSPINT;
+	cnt = igetdword(chip, ICHREG(GLOB_STA));
+	iputdword(chip, ICHREG(GLOB_STA), cnt & status);
+
+	if (snd_intel8x0_ich_chip_can_cold_reset(chip))
+		err = snd_intel8x0_ich_chip_cold_reset(chip);
+	else
+		err = snd_intel8x0_ich_chip_reset(chip);
+	if (err < 0)
+		return err;
 
-      __ok:
-#endif
 	if (probing) {
 		/* wait for any codec ready status.
 		 * Once it becomes ready it should remain ready
-- 
cgit v1.2.3


From e167280070cccd2e0cde94f73ded0a4b08bf6412 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 28 Jan 2009 16:05:16 +0100
Subject: ALSA: intel8x0 - Fix build with CONFIG_SND_AC97_POWERSAVE=n

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/intel8x0.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index b37bd268301..b13ef1e2a4a 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -2317,7 +2317,7 @@ static int snd_intel8x0_ich_chip_cold_reset(struct intel8x0 *chip)
 #define snd_intel8x0_ich_chip_can_cold_reset(chip) \
 	(!snd_pci_quirk_lookup(chip->pci, ich_chip_reset_mode))
 #else
-#define snd_intel8x0_ich_chip_cold_reset(x) do { } while (0)
+#define snd_intel8x0_ich_chip_cold_reset(chip)	0
 #define snd_intel8x0_ich_chip_can_cold_reset(chip) (0)
 #endif
 
-- 
cgit v1.2.3


From 61b9b9b109217b2bfb128c3ca24d8f8c839a425f Mon Sep 17 00:00:00 2001
From: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Date: Wed, 28 Jan 2009 09:16:33 -0200
Subject: ALSA: hda - Consider additional capture source/selector in ALC889

Currently code for capture source support in ALC889 only considers
capture mixers. This change adds additional support for ADC+selector
present in ALC889, taking into account also the presence of an
additional DMIC connection item in the selector.

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 105 +++++++++++++++++++++++++++++++-----------
 1 file changed, 77 insertions(+), 28 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 863ab957204..d81cb5eb8c5 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -238,6 +238,13 @@ enum {
 	ALC883_MODEL_LAST,
 };
 
+/* styles of capture selection */
+enum {
+	CAPT_MUX = 0,	/* only mux based */
+	CAPT_MIX,	/* only mixer based */
+	CAPT_1MUX_MIX,	/* first mux and other mixers */
+};
+
 /* for GPIO Poll */
 #define GPIO_MASK	0x03
 
@@ -276,7 +283,7 @@ struct alc_spec {
 	hda_nid_t *adc_nids;
 	hda_nid_t *capsrc_nids;
 	hda_nid_t dig_in_nid;		/* digital-in NID; optional */
-	unsigned char is_mix_capture;	/* matrix-style capture (non-mux) */
+	int capture_style;		/* capture style (CAPT_*) */
 
 	/* capture source */
 	unsigned int num_mux_defs;
@@ -294,7 +301,7 @@ struct alc_spec {
 	/* dynamic controls, init_verbs and input_mux */
 	struct auto_pin_cfg autocfg;
 	struct snd_array kctls;
-	struct hda_input_mux private_imux;
+	struct hda_input_mux private_imux[3];
 	hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS];
 
 	/* hooks */
@@ -396,7 +403,8 @@ static int alc_mux_enum_put(struct snd_kcontrol *kcontrol,
 	mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx;
 	imux = &spec->input_mux[mux_idx];
 
-	if (spec->is_mix_capture) {
+	if (spec->capture_style &&
+	    !(spec->capture_style == CAPT_1MUX_MIX && !adc_idx)) {
 		/* Matrix-mixer style (e.g. ALC882) */
 		unsigned int *cur_val = &spec->cur_mux[adc_idx];
 		unsigned int i, idx;
@@ -4130,7 +4138,7 @@ static int new_analog_input(struct alc_spec *spec, hda_nid_t pin,
 static int alc880_auto_create_analog_input_ctls(struct alc_spec *spec,
 						const struct auto_pin_cfg *cfg)
 {
-	struct hda_input_mux *imux = &spec->private_imux;
+	struct hda_input_mux *imux = &spec->private_imux[0];
 	int i, err, idx;
 
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
@@ -4279,7 +4287,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
 	add_verb(spec, alc880_volume_init_verbs);
 
 	spec->num_mux_defs = 1;
-	spec->input_mux = &spec->private_imux;
+	spec->input_mux = &spec->private_imux[0];
 
 	store_pin_configs(codec);
 	return 1;
@@ -5487,7 +5495,7 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec,
 static int alc260_auto_create_analog_input_ctls(struct alc_spec *spec,
 						const struct auto_pin_cfg *cfg)
 {
-	struct hda_input_mux *imux = &spec->private_imux;
+	struct hda_input_mux *imux = &spec->private_imux[0];
 	int i, err, idx;
 
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
@@ -5647,7 +5655,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec)
 	add_verb(spec, alc260_volume_init_verbs);
 
 	spec->num_mux_defs = 1;
-	spec->input_mux = &spec->private_imux;
+	spec->input_mux = &spec->private_imux[0];
 
 	store_pin_configs(codec);
 	return 1;
@@ -7087,7 +7095,7 @@ static int patch_alc882(struct hda_codec *codec)
 	spec->stream_digital_playback = &alc882_pcm_digital_playback;
 	spec->stream_digital_capture = &alc882_pcm_digital_capture;
 
-	spec->is_mix_capture = 1; /* matrix-style capture */
+	spec->capture_style = CAPT_MIX; /* matrix-style capture */
 	if (!spec->adc_nids && spec->input_mux) {
 		/* check whether NID 0x07 is valid */
 		unsigned int wcap = get_wcaps(codec, 0x07);
@@ -7155,10 +7163,14 @@ static hda_nid_t alc883_adc_nids_rev[2] = {
 	0x09, 0x08
 };
 
+#define alc889_adc_nids		alc880_adc_nids
+
 static hda_nid_t alc883_capsrc_nids[2] = { 0x23, 0x22 };
 
 static hda_nid_t alc883_capsrc_nids_rev[2] = { 0x22, 0x23 };
 
+#define alc889_capsrc_nids	alc882_capsrc_nids
+
 /* input MUX */
 /* FIXME: should be a matrix-type input source selection */
 
@@ -8977,6 +8989,8 @@ static int alc883_parse_auto_config(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	int err = alc880_parse_auto_config(codec);
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	int i;
 
 	if (err < 0)
 		return err;
@@ -8990,6 +9004,26 @@ static int alc883_parse_auto_config(struct hda_codec *codec)
 	/* hack - override the init verbs */
 	spec->init_verbs[0] = alc883_auto_init_verbs;
 
+	/* setup input_mux for ALC889 */
+	if (codec->vendor_id == 0x10ec0889) {
+		/* digital-mic input pin is excluded in alc880_auto_create..()
+		 * because it's under 0x18
+		 */
+		if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 ||
+		    cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) {
+			struct hda_input_mux *imux = &spec->private_imux[0];
+			for (i = 1; i < 3; i++)
+				memcpy(&spec->private_imux[i],
+				       &spec->private_imux[0],
+				       sizeof(spec->private_imux[0]));
+			imux->items[imux->num_items].label = "Int DMic";
+			imux->items[imux->num_items].index = 0x0b;
+			imux->num_items++;
+			spec->num_mux_defs = 3;
+			spec->input_mux = spec->private_imux;
+		}
+	}
+
 	return 1; /* config found */
 }
 
@@ -9053,14 +9087,36 @@ static int patch_alc883(struct hda_codec *codec)
 			spec->stream_name_analog = "ALC888 Analog";
 			spec->stream_name_digital = "ALC888 Digital";
 		}
+		if (!spec->num_adc_nids) {
+			spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids);
+			spec->adc_nids = alc883_adc_nids;
+		}
+		if (!spec->capsrc_nids)
+			spec->capsrc_nids = alc883_capsrc_nids;
+		spec->capture_style = CAPT_MIX; /* matrix-style capture */
 		break;
 	case 0x10ec0889:
 		spec->stream_name_analog = "ALC889 Analog";
 		spec->stream_name_digital = "ALC889 Digital";
+		if (!spec->num_adc_nids) {
+			spec->num_adc_nids = ARRAY_SIZE(alc889_adc_nids);
+			spec->adc_nids = alc889_adc_nids;
+		}
+		if (!spec->capsrc_nids)
+			spec->capsrc_nids = alc889_capsrc_nids;
+		spec->capture_style = CAPT_1MUX_MIX; /* 1mux/Nmix-style
+							capture */
 		break;
 	default:
 		spec->stream_name_analog = "ALC883 Analog";
 		spec->stream_name_digital = "ALC883 Digital";
+		if (!spec->num_adc_nids) {
+			spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids);
+			spec->adc_nids = alc883_adc_nids;
+		}
+		if (!spec->capsrc_nids)
+			spec->capsrc_nids = alc883_capsrc_nids;
+		spec->capture_style = CAPT_MIX; /* matrix-style capture */
 		break;
 	}
 
@@ -9071,13 +9127,6 @@ static int patch_alc883(struct hda_codec *codec)
 	spec->stream_digital_playback = &alc883_pcm_digital_playback;
 	spec->stream_digital_capture = &alc883_pcm_digital_capture;
 
-	if (!spec->num_adc_nids) {
-		spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids);
-		spec->adc_nids = alc883_adc_nids;
-	}
-	if (!spec->capsrc_nids)
-		spec->capsrc_nids = alc883_capsrc_nids;
-	spec->is_mix_capture = 1; /* matrix-style capture */
 	if (!spec->cap_mixer)
 		set_capture_mixer(spec);
 
@@ -10512,7 +10561,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec)
 
 	add_verb(spec, alc262_volume_init_verbs);
 	spec->num_mux_defs = 1;
-	spec->input_mux = &spec->private_imux;
+	spec->input_mux = &spec->private_imux[0];
 
 	err = alc_auto_add_mic_boost(codec);
 	if (err < 0)
@@ -10881,7 +10930,7 @@ static int patch_alc262(struct hda_codec *codec)
 	spec->stream_digital_playback = &alc262_pcm_digital_playback;
 	spec->stream_digital_capture = &alc262_pcm_digital_capture;
 
-	spec->is_mix_capture = 1;
+	spec->capture_style = CAPT_MIX;
 	if (!spec->adc_nids && spec->input_mux) {
 		/* check whether NID 0x07 is valid */
 		unsigned int wcap = get_wcaps(codec, 0x07);
@@ -11539,7 +11588,7 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec,
 static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec,
 						const struct auto_pin_cfg *cfg)
 {
-	struct hda_input_mux *imux = &spec->private_imux;
+	struct hda_input_mux *imux = &spec->private_imux[0];
 	int i, idx1;
 
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
@@ -11657,7 +11706,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
 
 	add_verb(spec, alc268_volume_init_verbs);
 	spec->num_mux_defs = 1;
-	spec->input_mux = &spec->private_imux;
+	spec->input_mux = &spec->private_imux[0];
 
 	err = alc_auto_add_mic_boost(codec);
 	if (err < 0)
@@ -12511,7 +12560,7 @@ static int alc269_auto_create_analog_input_ctls(struct alc_spec *spec,
 	 */
 	if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 ||
 	    cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) {
-		struct hda_input_mux *imux = &spec->private_imux;
+		struct hda_input_mux *imux = &spec->private_imux[0];
 		imux->items[imux->num_items].label = "Int Mic";
 		imux->items[imux->num_items].index = 0x05;
 		imux->num_items++;
@@ -12567,7 +12616,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 
 	add_verb(spec, alc269_init_verbs);
 	spec->num_mux_defs = 1;
-	spec->input_mux = &spec->private_imux;
+	spec->input_mux = &spec->private_imux[0];
 	/* set default input source */
 	snd_hda_codec_write_cache(codec, alc269_capsrc_nids[0],
 				  0, AC_VERB_SET_CONNECT_SEL,
@@ -13483,7 +13532,7 @@ static int alc861_auto_create_hp_ctls(struct alc_spec *spec, hda_nid_t pin)
 static int alc861_auto_create_analog_input_ctls(struct alc_spec *spec,
 						const struct auto_pin_cfg *cfg)
 {
-	struct hda_input_mux *imux = &spec->private_imux;
+	struct hda_input_mux *imux = &spec->private_imux[0];
 	int i, err, idx, idx1;
 
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
@@ -13620,7 +13669,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec)
 	add_verb(spec, alc861_auto_init_verbs);
 
 	spec->num_mux_defs = 1;
-	spec->input_mux = &spec->private_imux;
+	spec->input_mux = &spec->private_imux[0];
 
 	spec->adc_nids = alc861_adc_nids;
 	spec->num_adc_nids = ARRAY_SIZE(alc861_adc_nids);
@@ -14724,7 +14773,7 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec)
 	add_verb(spec, alc861vd_volume_init_verbs);
 
 	spec->num_mux_defs = 1;
-	spec->input_mux = &spec->private_imux;
+	spec->input_mux = &spec->private_imux[0];
 
 	err = alc_auto_add_mic_boost(codec);
 	if (err < 0)
@@ -14803,7 +14852,7 @@ static int patch_alc861vd(struct hda_codec *codec)
 	spec->adc_nids = alc861vd_adc_nids;
 	spec->num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids);
 	spec->capsrc_nids = alc861vd_capsrc_nids;
-	spec->is_mix_capture = 1;
+	spec->capture_style = CAPT_MIX;
 
 	set_capture_mixer(spec);
 
@@ -16397,7 +16446,7 @@ static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin,
 static int alc662_auto_create_analog_input_ctls(struct alc_spec *spec,
 						const struct auto_pin_cfg *cfg)
 {
-	struct hda_input_mux *imux = &spec->private_imux;
+	struct hda_input_mux *imux = &spec->private_imux[0];
 	int i, err, idx;
 
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
@@ -16528,7 +16577,7 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
 		add_mixer(spec, spec->kctls.list);
 
 	spec->num_mux_defs = 1;
-	spec->input_mux = &spec->private_imux;
+	spec->input_mux = &spec->private_imux[0];
 
 	add_verb(spec, alc662_auto_init_verbs);
 	if (codec->vendor_id == 0x10ec0663)
@@ -16613,7 +16662,7 @@ static int patch_alc662(struct hda_codec *codec)
 	spec->adc_nids = alc662_adc_nids;
 	spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids);
 	spec->capsrc_nids = alc662_capsrc_nids;
-	spec->is_mix_capture = 1;
+	spec->capture_style = CAPT_MIX;
 
 	if (!spec->cap_mixer)
 		set_capture_mixer(spec);
-- 
cgit v1.2.3


From 328cc6dfaadad614449eca1c75559e64c5054fea Mon Sep 17 00:00:00 2001
From: Thadeu Lima de Souza Cascardo <cascardo@holoscopio.com>
Date: Wed, 28 Jan 2009 15:39:22 -0200
Subject: ALSA: AC97: Print AC97 flags in proc file to make debug it easier

While debugging some code paths in AC97 codec patches and its suspend
and resume functions, getting to know the flags has proved useful to
follow those code paths.

Signed-off-by: Thadeu Lima de Souza Cascardo <cascardo@holoscopio.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ac97/ac97_proc.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/ac97/ac97_proc.c b/sound/pci/ac97/ac97_proc.c
index 060ea59d5f0..73b17d526c8 100644
--- a/sound/pci/ac97/ac97_proc.c
+++ b/sound/pci/ac97/ac97_proc.c
@@ -125,6 +125,8 @@ static void snd_ac97_proc_read_main(struct snd_ac97 *ac97, struct snd_info_buffe
         snd_iprintf(buffer, "PCI Subsys Device: 0x%04x\n\n",
                     ac97->subsystem_device);
 
+	snd_iprintf(buffer, "Flags: %x\n", ac97->flags);
+
 	if ((ac97->ext_id & AC97_EI_REV_MASK) >= AC97_EI_REV_23) {
 		val = snd_ac97_read(ac97, AC97_INT_PAGING);
 		snd_ac97_update_bits(ac97, AC97_INT_PAGING,
-- 
cgit v1.2.3


From b833b5ec0411adc2255053a0e0ec536d97e5784e Mon Sep 17 00:00:00 2001
From: Thadeu Lima de Souza Cascardo <cascardo@holoscopio.com>
Date: Wed, 28 Jan 2009 18:20:06 -0200
Subject: ALSA: AC97: Fix function name type in comment s/updat/update/

Signed-off-by: Thadeu Lima de Souza Cascardo <cascardo@holoscopio.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ac97/ac97_codec.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index e2b843b4f9d..27551e963e5 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -383,7 +383,7 @@ int snd_ac97_update_bits(struct snd_ac97 *ac97, unsigned short reg, unsigned sho
 
 EXPORT_SYMBOL(snd_ac97_update_bits);
 
-/* no lock version - see snd_ac97_updat_bits() */
+/* no lock version - see snd_ac97_update_bits() */
 int snd_ac97_update_bits_nolock(struct snd_ac97 *ac97, unsigned short reg,
 				unsigned short mask, unsigned short value)
 {
-- 
cgit v1.2.3


From b98b7b347eed333d6fa2f74770beb8106e576cc6 Mon Sep 17 00:00:00 2001
From: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Date: Thu, 29 Jan 2009 13:18:31 -0200
Subject: ALSA: hda - make alc882_auto_init_input_src aware of selectors

In the case of having a selector instead of mixer while initing input
sources, the case that happens with ALC889, we must select instead
of muting input. Problem was found while testing with hda-emu.

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 21 ++++++++++++++++++---
 1 file changed, 18 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index d81cb5eb8c5..3666cc5dc3b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6924,18 +6924,21 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec)
 static void alc882_auto_init_input_src(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	const struct hda_input_mux *imux = spec->input_mux;
 	int c;
 
 	for (c = 0; c < spec->num_adc_nids; c++) {
 		hda_nid_t conn_list[HDA_MAX_NUM_INPUTS];
 		hda_nid_t nid = spec->capsrc_nids[c];
+		unsigned int mux_idx;
+		const struct hda_input_mux *imux;
 		int conns, mute, idx, item;
 
 		conns = snd_hda_get_connections(codec, nid, conn_list,
 						ARRAY_SIZE(conn_list));
 		if (conns < 0)
 			continue;
+		mux_idx = c >= spec->num_mux_defs ? 0 : c;
+		imux = &spec->input_mux[mux_idx];
 		for (idx = 0; idx < conns; idx++) {
 			/* if the current connection is the selected one,
 			 * unmute it as default - otherwise mute it
@@ -6948,8 +6951,20 @@ static void alc882_auto_init_input_src(struct hda_codec *codec)
 					break;
 				}
 			}
-			snd_hda_codec_write(codec, nid, 0,
-					    AC_VERB_SET_AMP_GAIN_MUTE, mute);
+			/* check if we have a selector or mixer
+			 * we could check for the widget type instead, but
+			 * just check for Amp-In presence (in case of mixer
+			 * without amp-in there is something wrong, this
+			 * function shouldn't be used or capsrc nid is wrong)
+			 */
+			if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)
+				snd_hda_codec_write(codec, nid, 0,
+						    AC_VERB_SET_AMP_GAIN_MUTE,
+						    mute);
+			else if (mute != AMP_IN_MUTE(idx))
+				snd_hda_codec_write(codec, nid, 0,
+						    AC_VERB_SET_CONNECT_SEL,
+						    idx);
 		}
 	}
 }
-- 
cgit v1.2.3


From d563ffa6b319a4e401d096db9014a947590ca081 Mon Sep 17 00:00:00 2001
From: Tim Blechmann <tim@klingt.org>
Date: Sat, 31 Jan 2009 18:01:13 +0100
Subject: ALSA: pcxhr: fix trivial typo

Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/pcxhr/pcxhr_core.h | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/pcxhr/pcxhr_core.h b/sound/pci/pcxhr/pcxhr_core.h
index bbbd66d13a6..be0173796cd 100644
--- a/sound/pci/pcxhr/pcxhr_core.h
+++ b/sound/pci/pcxhr/pcxhr_core.h
@@ -1,7 +1,7 @@
 /*
  * Driver for Digigram pcxhr compatible soundcards
  *
- * low level interface with interrupt ans message handling
+ * low level interface with interrupt and message handling
  *
  * Copyright (c) 2004 by Digigram <alsa@digigram.com>
  *
-- 
cgit v1.2.3


From e683ec4697c74c7d04ff8e90ec625ac34e25a7d8 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 12 Nov 2008 16:42:44 +0100
Subject: ALSA: ice1724 - Dynamic MIDI TX irq control

MIDI_TX IRQ seems always pending when any bytes on FIFO is available.
Thus, it's better to enable MPU_TX only when any bytres are really
stored in the substream, and disables immediately when the queue
becomes empty.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ice1712/ice1724.c | 43 +++++++++++++++++++++++++++----------------
 1 file changed, 27 insertions(+), 16 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index bb8d8c766b9..eb7872dec5a 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -241,6 +241,8 @@ get_rawmidi_substream(struct snd_ice1712 *ice, unsigned int stream)
 				struct snd_rawmidi_substream, list);
 }
 
+static void enable_midi_irq(struct snd_ice1712 *ice, u8 flag, int enable);
+
 static void vt1724_midi_write(struct snd_ice1712 *ice)
 {
 	struct snd_rawmidi_substream *s;
@@ -254,6 +256,11 @@ static void vt1724_midi_write(struct snd_ice1712 *ice)
 		for (i = 0; i < count; ++i)
 			outb(buffer[i], ICEREG1724(ice, MPU_DATA));
 	}
+	/* mask irq when all bytes have been transmitted.
+	 * enabled again in output_trigger when the new data comes in.
+	 */
+	enable_midi_irq(ice, VT1724_IRQ_MPU_TX,
+			!snd_rawmidi_transmit_empty(s));
 }
 
 static void vt1724_midi_read(struct snd_ice1712 *ice)
@@ -272,31 +279,34 @@ static void vt1724_midi_read(struct snd_ice1712 *ice)
 	}
 }
 
-static void vt1724_enable_midi_irq(struct snd_rawmidi_substream *substream,
-				   u8 flag, int enable)
+/* call with ice->reg_lock */
+static void enable_midi_irq(struct snd_ice1712 *ice, u8 flag, int enable)
 {
-	struct snd_ice1712 *ice = substream->rmidi->private_data;
-	u8 mask;
-
-	spin_lock_irq(&ice->reg_lock);
-	mask = inb(ICEREG1724(ice, IRQMASK));
+	u8 mask = inb(ICEREG1724(ice, IRQMASK));
 	if (enable)
 		mask &= ~flag;
 	else
 		mask |= flag;
 	outb(mask, ICEREG1724(ice, IRQMASK));
+}
+
+static void vt1724_enable_midi_irq(struct snd_rawmidi_substream *substream,
+				   u8 flag, int enable)
+{
+	struct snd_ice1712 *ice = substream->rmidi->private_data;
+
+	spin_lock_irq(&ice->reg_lock);
+	enable_midi_irq(ice, flag, enable);
 	spin_unlock_irq(&ice->reg_lock);
 }
 
 static int vt1724_midi_output_open(struct snd_rawmidi_substream *s)
 {
-	vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_TX, 1);
 	return 0;
 }
 
 static int vt1724_midi_output_close(struct snd_rawmidi_substream *s)
 {
-	vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_TX, 0);
 	return 0;
 }
 
@@ -311,6 +321,7 @@ static void vt1724_midi_output_trigger(struct snd_rawmidi_substream *s, int up)
 		vt1724_midi_write(ice);
 	} else {
 		ice->midi_output = 0;
+		enable_midi_irq(ice, VT1724_IRQ_MPU_TX, 0);
 	}
 	spin_unlock_irqrestore(&ice->reg_lock, flags);
 }
@@ -320,6 +331,7 @@ static void vt1724_midi_output_drain(struct snd_rawmidi_substream *s)
 	struct snd_ice1712 *ice = s->rmidi->private_data;
 	unsigned long timeout;
 
+	vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_TX, 0);
 	/* 32 bytes should be transmitted in less than about 12 ms */
 	timeout = jiffies + msecs_to_jiffies(15);
 	do {
@@ -389,24 +401,24 @@ static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id)
 		status &= status_mask;
 		if (status == 0)
 			break;
+		spin_lock(&ice->reg_lock);
 		if (++timeout > 10) {
 			status = inb(ICEREG1724(ice, IRQSTAT));
 			printk(KERN_ERR "ice1724: Too long irq loop, "
 			       "status = 0x%x\n", status);
 			if (status & VT1724_IRQ_MPU_TX) {
 				printk(KERN_ERR "ice1724: Disabling MPU_TX\n");
-				outb(inb(ICEREG1724(ice, IRQMASK)) |
-				     VT1724_IRQ_MPU_TX,
-				     ICEREG1724(ice, IRQMASK));
+				enable_midi_irq(ice, VT1724_IRQ_MPU_TX, 0);
 			}
+			spin_unlock(&ice->reg_lock);
 			break;
 		}
 		handled = 1;
 		if (status & VT1724_IRQ_MPU_TX) {
-			spin_lock(&ice->reg_lock);
 			if (ice->midi_output)
 				vt1724_midi_write(ice);
-			spin_unlock(&ice->reg_lock);
+			else
+				enable_midi_irq(ice, VT1724_IRQ_MPU_TX, 0);
 			/* Due to mysterical reasons, MPU_TX is always
 			 * generated (and can't be cleared) when a PCM
 			 * playback is going.  So let's ignore at the
@@ -415,15 +427,14 @@ static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id)
 			status_mask &= ~VT1724_IRQ_MPU_TX;
 		}
 		if (status & VT1724_IRQ_MPU_RX) {
-			spin_lock(&ice->reg_lock);
 			if (ice->midi_input)
 				vt1724_midi_read(ice);
 			else
 				vt1724_midi_clear_rx(ice);
-			spin_unlock(&ice->reg_lock);
 		}
 		/* ack MPU irq */
 		outb(status, ICEREG1724(ice, IRQSTAT));
+		spin_unlock(&ice->reg_lock);
 		if (status & VT1724_IRQ_MTPCM) {
 			/*
 			 * Multi-track PCM
-- 
cgit v1.2.3


From ba340e825f4b892782779abd0f93bcff7e763567 Mon Sep 17 00:00:00 2001
From: Tony Vroon <chainsaw@gentoo.org>
Date: Mon, 2 Feb 2009 19:01:30 +0000
Subject: ALSA: hda - Add tyan model for Realtek ALC262

The Realtek ALC262 on the Tyan Thunder n6650W (S2915-E) mainboard has a
rather odd configuration template. As a result, the white AUX connector
can not be used. This rewrites the default config to more accurately
reflect the connector layout, colour and function.
Unfortunately the black CD_IN connector, which is suspected to be widget
0x1c refuses to switch into input (0x20), instead opting to remain on 0x0.
As such, no mixer controls are exposed for it. Autoswitching is implemented
between the front headphone output and back line output.

Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 77 +++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 77 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 0c81d92c3d7..bd9ef336389 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -103,6 +103,7 @@ enum {
 	ALC262_NEC,
 	ALC262_TOSHIBA_S06,
 	ALC262_TOSHIBA_RX1,
+	ALC262_TYAN,
 	ALC262_AUTO,
 	ALC262_MODEL_LAST /* last tag */
 };
@@ -9509,6 +9510,67 @@ static struct snd_kcontrol_new alc262_benq_t31_mixer[] = {
 	{ } /* end */
 };
 
+static struct snd_kcontrol_new alc262_tyan_mixer[] = {
+	HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Aux Playback Volume", 0x0b, 0x06, HDA_INPUT),
+	HDA_CODEC_MUTE("Aux Playback Switch", 0x0b, 0x06, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
+	{ } /* end */
+};
+
+static struct hda_verb alc262_tyan_verbs[] = {
+	/* Headphone automute */
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	/* P11 AUX_IN, white 4-pin connector */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, 0xe1},
+	{0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, 0x93},
+	{0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, 0x19},
+
+	{}
+};
+
+/* unsolicited event for HP jack sensing */
+static void alc262_tyan_automute(struct hda_codec *codec)
+{
+	unsigned int mute;
+	unsigned int present;
+
+	snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0);
+	present = snd_hda_codec_read(codec, 0x1b, 0,
+				     AC_VERB_GET_PIN_SENSE, 0);
+	present = (present & 0x80000000) != 0;
+	if (present) {
+		/* mute line output on ATX panel */
+		snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+					 HDA_AMP_MUTE, HDA_AMP_MUTE);
+	} else {
+		/* unmute line output if necessary */
+		mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0);
+		snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+					 HDA_AMP_MUTE, mute);
+	}
+}
+
+static void alc262_tyan_unsol_event(struct hda_codec *codec,
+				       unsigned int res)
+{
+	if ((res >> 26) != ALC880_HP_EVENT)
+		return;
+	alc262_tyan_automute(codec);
+}
+
 #define alc262_capture_mixer		alc882_capture_mixer
 #define alc262_capture_alt_mixer	alc882_capture_alt_mixer
 
@@ -10626,6 +10688,7 @@ static const char *alc262_models[ALC262_MODEL_LAST] = {
 	[ALC262_ULTRA]		= "ultra",
 	[ALC262_LENOVO_3000]	= "lenovo-3000",
 	[ALC262_NEC]		= "nec",
+	[ALC262_TYAN]		= "tyan",
 	[ALC262_AUTO]		= "auto",
 };
 
@@ -10666,6 +10729,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06),
 	SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU),
 	SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU),
+	SND_PCI_QUIRK(0x10f1, 0x2915, "Tyan Thunder n6650W", ALC262_TYAN),
 	SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA),
 	SND_PCI_QUIRK(0x144d, 0xc039, "Samsung Q1U EL", ALC262_ULTRA),
 	SND_PCI_QUIRK(0x144d, 0xc510, "Samsung Q45", ALC262_HIPPO),
@@ -10884,6 +10948,19 @@ static struct alc_config_preset alc262_presets[] = {
 		.unsol_event = alc262_hippo_unsol_event,
 		.init_hook = alc262_hippo_automute,
 	},
+	[ALC262_TYAN] = {
+		.mixers = { alc262_tyan_mixer },
+		.init_verbs = { alc262_init_verbs, alc262_tyan_verbs},
+		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
+		.dac_nids = alc262_dac_nids,
+		.hp_nid = 0x02,
+		.dig_out_nid = ALC262_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc262_modes),
+		.channel_mode = alc262_modes,
+		.input_mux = &alc262_capture_source,
+		.unsol_event = alc262_tyan_unsol_event,
+		.init_hook = alc262_tyan_automute,
+	},
 };
 
 static int patch_alc262(struct hda_codec *codec)
-- 
cgit v1.2.3


From 680cd53652d8bfb2b97d8c0248d1afb82de6b61d Mon Sep 17 00:00:00 2001
From: Kusanagi Kouichi <slash@ma.neweb.ne.jp>
Date: Thu, 5 Feb 2009 00:00:58 +0900
Subject: ALSA: hda: Add digital beep generator support for Realtek codecs.

A digital beep generator can be used via input layer.

Signed-off-by: Kusanagi Kouichi <slash@ma.neweb.ne.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_beep.h      |  2 +-
 sound/pci/hda/patch_realtek.c | 62 +++++++++++++++++++++++++++++++++++++++++++
 2 files changed, 63 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h
index b9679f081ca..51bf6a5daf3 100644
--- a/sound/pci/hda/hda_beep.h
+++ b/sound/pci/hda/hda_beep.h
@@ -39,7 +39,7 @@ struct hda_beep {
 int snd_hda_attach_beep_device(struct hda_codec *codec, int nid);
 void snd_hda_detach_beep_device(struct hda_codec *codec);
 #else
-#define snd_hda_attach_beep_device(...)
+#define snd_hda_attach_beep_device(...)		0
 #define snd_hda_detach_beep_device(...)
 #endif
 #endif
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index bd9ef336389..0faa41bfc8b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -30,6 +30,7 @@
 #include <sound/core.h>
 #include "hda_codec.h"
 #include "hda_local.h"
+#include "hda_beep.h"
 
 #define ALC880_FRONT_EVENT		0x01
 #define ALC880_DCVOL_EVENT		0x02
@@ -3187,6 +3188,7 @@ static void alc_free(struct hda_codec *codec)
 
 	alc_free_kctls(codec);
 	kfree(spec);
+	snd_hda_detach_beep_device(codec);
 	codec->spec = NULL; /* to be sure */
 }
 
@@ -4355,6 +4357,12 @@ static int patch_alc880(struct hda_codec *codec)
 		}
 	}
 
+	err = snd_hda_attach_beep_device(codec, 0x1);
+	if (err < 0) {
+		alc_free(codec);
+		return err;
+	}
+
 	if (board_config != ALC880_AUTO)
 		setup_preset(spec, &alc880_presets[board_config]);
 
@@ -5882,6 +5890,12 @@ static int patch_alc260(struct hda_codec *codec)
 		}
 	}
 
+	err = snd_hda_attach_beep_device(codec, 0x1);
+	if (err < 0) {
+		alc_free(codec);
+		return err;
+	}
+
 	if (board_config != ALC260_AUTO)
 		setup_preset(spec, &alc260_presets[board_config]);
 
@@ -7093,6 +7107,12 @@ static int patch_alc882(struct hda_codec *codec)
 		}
 	}
 
+	err = snd_hda_attach_beep_device(codec, 0x1);
+	if (err < 0) {
+		alc_free(codec);
+		return err;
+	}
+
 	if (board_config != ALC882_AUTO)
 		setup_preset(spec, &alc882_presets[board_config]);
 
@@ -9093,6 +9113,12 @@ static int patch_alc883(struct hda_codec *codec)
 		}
 	}
 
+	err = snd_hda_attach_beep_device(codec, 0x1);
+	if (err < 0) {
+		alc_free(codec);
+		return err;
+	}
+
 	if (board_config != ALC883_AUTO)
 		setup_preset(spec, &alc883_presets[board_config]);
 
@@ -11013,6 +11039,12 @@ static int patch_alc262(struct hda_codec *codec)
 		}
 	}
 
+	err = snd_hda_attach_beep_device(codec, 0x1);
+	if (err < 0) {
+		alc_free(codec);
+		return err;
+	}
+
 	if (board_config != ALC262_AUTO)
 		setup_preset(spec, &alc262_presets[board_config]);
 
@@ -12051,6 +12083,12 @@ static int patch_alc268(struct hda_codec *codec)
 		}
 	}
 
+	err = snd_hda_attach_beep_device(codec, 0x1);
+	if (err < 0) {
+		alc_free(codec);
+		return err;
+	}
+
 	if (board_config != ALC268_AUTO)
 		setup_preset(spec, &alc268_presets[board_config]);
 
@@ -12885,6 +12923,12 @@ static int patch_alc269(struct hda_codec *codec)
 		}
 	}
 
+	err = snd_hda_attach_beep_device(codec, 0x1);
+	if (err < 0) {
+		alc_free(codec);
+		return err;
+	}
+
 	if (board_config != ALC269_AUTO)
 		setup_preset(spec, &alc269_presets[board_config]);
 
@@ -13978,6 +14022,12 @@ static int patch_alc861(struct hda_codec *codec)
 		}
 	}
 
+	err = snd_hda_attach_beep_device(codec, 0x23);
+	if (err < 0) {
+		alc_free(codec);
+		return err;
+	}
+
 	if (board_config != ALC861_AUTO)
 		setup_preset(spec, &alc861_presets[board_config]);
 
@@ -14924,6 +14974,12 @@ static int patch_alc861vd(struct hda_codec *codec)
 		}
 	}
 
+	err = snd_hda_attach_beep_device(codec, 0x23);
+	if (err < 0) {
+		alc_free(codec);
+		return err;
+	}
+
 	if (board_config != ALC861VD_AUTO)
 		setup_preset(spec, &alc861vd_presets[board_config]);
 
@@ -16733,6 +16789,12 @@ static int patch_alc662(struct hda_codec *codec)
 		}
 	}
 
+	err = snd_hda_attach_beep_device(codec, 0x1);
+	if (err < 0) {
+		alc_free(codec);
+		return err;
+	}
+
 	if (board_config != ALC662_AUTO)
 		setup_preset(spec, &alc662_presets[board_config]);
 
-- 
cgit v1.2.3


From 616f89e74cd954e04ae4f8bad6a3dc8730a4a47a Mon Sep 17 00:00:00 2001
From: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Date: Wed, 4 Feb 2009 11:23:19 -0500
Subject: ALSA: hda - Additional pin nids for STAC92HD71Bx and STAC92HD75Bx
 codecs

Current code for STAC92HD71Bx and STAC92HD75Bx doesn't consider pin
complexes 0x20 and 0x27. Also for 4 port models, nids 0x0e and 0x0f
are vendor reserved. This commit changes code so it'll consider the
additional pin complexes for models that have it, and avoid reserved
nids to be touched on 4 port models.

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 59 ++++++++++++++++++++++++++++++------------
 1 file changed, 43 insertions(+), 16 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index a7df81efed2..58c9ff9d27f 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -481,10 +481,17 @@ static hda_nid_t stac92hd83xxx_pin_nids[14] = {
 	0x0f, 0x10, 0x11, 0x12, 0x13,
 	0x1d, 0x1e, 0x1f, 0x20
 };
-static hda_nid_t stac92hd71bxx_pin_nids[11] = {
+
+#define STAC92HD71BXX_NUM_PINS 13
+static hda_nid_t stac92hd71bxx_pin_nids_4port[STAC92HD71BXX_NUM_PINS] = {
+	0x0a, 0x0b, 0x0c, 0x0d, 0x00,
+	0x00, 0x14, 0x18, 0x19, 0x1e,
+	0x1f, 0x20, 0x27
+};
+static hda_nid_t stac92hd71bxx_pin_nids_6port[STAC92HD71BXX_NUM_PINS] = {
 	0x0a, 0x0b, 0x0c, 0x0d, 0x0e,
 	0x0f, 0x14, 0x18, 0x19, 0x1e,
-	0x1f,
+	0x1f, 0x20, 0x27
 };
 
 static hda_nid_t stac927x_pin_nids[14] = {
@@ -1745,28 +1752,32 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
 	{} /* terminator */
 };
 
-static unsigned int ref92hd71bxx_pin_configs[11] = {
+static unsigned int ref92hd71bxx_pin_configs[STAC92HD71BXX_NUM_PINS] = {
 	0x02214030, 0x02a19040, 0x01a19020, 0x01014010,
 	0x0181302e, 0x01014010, 0x01019020, 0x90a000f0,
-	0x90a000f0, 0x01452050, 0x01452050,
+	0x90a000f0, 0x01452050, 0x01452050, 0x00000000,
+	0x00000000
 };
 
-static unsigned int dell_m4_1_pin_configs[11] = {
+static unsigned int dell_m4_1_pin_configs[STAC92HD71BXX_NUM_PINS] = {
 	0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110,
 	0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0,
-	0x40f000f0, 0x4f0000f0, 0x4f0000f0,
+	0x40f000f0, 0x4f0000f0, 0x4f0000f0, 0x00000000,
+	0x00000000
 };
 
-static unsigned int dell_m4_2_pin_configs[11] = {
+static unsigned int dell_m4_2_pin_configs[STAC92HD71BXX_NUM_PINS] = {
 	0x0421101f, 0x04a11221, 0x90a70330, 0x90170110,
 	0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0,
-	0x40f000f0, 0x044413b0, 0x044413b0,
+	0x40f000f0, 0x044413b0, 0x044413b0, 0x00000000,
+	0x00000000
 };
 
-static unsigned int dell_m4_3_pin_configs[11] = {
+static unsigned int dell_m4_3_pin_configs[STAC92HD71BXX_NUM_PINS] = {
 	0x0421101f, 0x04a11221, 0x90a70330, 0x90170110,
 	0x40f000f0, 0x40f000f0, 0x40f000f0, 0x90a000f0,
-	0x40f000f0, 0x044413b0, 0x044413b0,
+	0x40f000f0, 0x044413b0, 0x044413b0, 0x00000000,
+	0x00000000
 };
 
 static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = {
@@ -2311,7 +2322,9 @@ static int stac92xx_save_bios_config_regs(struct hda_codec *codec)
 	for (i = 0; i < spec->num_pins; i++) {
 		hda_nid_t nid = spec->pin_nids[i];
 		unsigned int pin_cfg;
-		
+
+		if (!nid)
+			continue;
 		pin_cfg = snd_hda_codec_read(codec, nid, 0, 
 			AC_VERB_GET_CONFIG_DEFAULT, 0x00);	
 		snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x bios pin config %8.8x\n",
@@ -2354,8 +2367,9 @@ static void stac92xx_set_config_regs(struct hda_codec *codec)
  		return;
 
 	for (i = 0; i < spec->num_pins; i++)
-		stac92xx_set_config_reg(codec, spec->pin_nids[i],
-					spec->pin_configs[i]);
+		if (spec->pin_nids[i] && spec->pin_configs[i])
+			stac92xx_set_config_reg(codec, spec->pin_nids[i],
+						spec->pin_configs[i]);
 }
 
 static int stac_save_pin_cfgs(struct hda_codec *codec, unsigned int *pins)
@@ -4952,9 +4966,21 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
 
 	codec->spec = spec;
 	codec->patch_ops = stac92xx_patch_ops;
-	spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids);
+	spec->num_pins = STAC92HD71BXX_NUM_PINS;
+	switch (codec->vendor_id) {
+	case 0x111d76b6:
+	case 0x111d76b7:
+		spec->pin_nids = stac92hd71bxx_pin_nids_4port;
+		break;
+	case 0x111d7603:
+	case 0x111d7608:
+		/* On 92HD75Bx 0x27 isn't a pin nid */
+		spec->num_pins--;
+		/* fallthrough */
+	default:
+		spec->pin_nids = stac92hd71bxx_pin_nids_6port;
+	}
 	spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids);
-	spec->pin_nids = stac92hd71bxx_pin_nids;
 	memcpy(&spec->private_dimux, &stac92hd71bxx_dmux,
 			sizeof(stac92hd71bxx_dmux));
 	spec->board_config = snd_hda_check_board_config(codec,
@@ -5018,7 +5044,8 @@ again:
 		/* disable VSW */
 		spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF];
 		unmute_init++;
-		stac_change_pin_config(codec, 0xf, 0x40f000f0);
+		stac_change_pin_config(codec, 0x0f, 0x40f000f0);
+		stac_change_pin_config(codec, 0x19, 0x40f000f3);
 		break;
 	case 0x111d7603: /* 6 Port with Analog Mixer */
 		if ((codec->revision_id & 0xf) == 1)
-- 
cgit v1.2.3


From 6df703aefc81252447c69d24d2863007de2338e9 Mon Sep 17 00:00:00 2001
From: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Date: Wed, 4 Feb 2009 11:34:22 -0500
Subject: ALSA: hda - Dynamic detection of dmics/dmuxes/smuxes in stac92hd71bxx

Detect the number of connected ports and number of smuxes dynamically,
looking at pin configs, using new introduced functions
stac92hd71bxx_connected_ports and stac92hd71bxx_connected_smuxes. Also
use proper input mux configuration for 4port and 5port models.

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 99 +++++++++++++++++++++++++++++++++++++-----
 1 file changed, 87 insertions(+), 12 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 58c9ff9d27f..c36c1c0f957 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4944,7 +4944,16 @@ again:
 	return 0;
 }
 
-static struct hda_input_mux stac92hd71bxx_dmux = {
+static struct hda_input_mux stac92hd71bxx_dmux_nomixer = {
+	.num_items = 3,
+	.items = {
+		{ "Analog Inputs", 0x00 },
+		{ "Digital Mic 1", 0x02 },
+		{ "Digital Mic 2", 0x03 },
+	}
+};
+
+static struct hda_input_mux stac92hd71bxx_dmux_amixer = {
 	.num_items = 4,
 	.items = {
 		{ "Analog Inputs", 0x00 },
@@ -4954,11 +4963,57 @@ static struct hda_input_mux stac92hd71bxx_dmux = {
 	}
 };
 
+static int stac92hd71bxx_connected_ports(struct hda_codec *codec,
+					 hda_nid_t *nids, int num_nids)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	int idx, num;
+	unsigned int def_conf;
+
+	for (num = 0; num < num_nids; num++) {
+		for (idx = 0; idx < spec->num_pins; idx++)
+			if (spec->pin_nids[idx] == nids[num])
+				break;
+		if (idx >= spec->num_pins)
+			break;
+		def_conf = get_defcfg_connect(spec->pin_configs[idx]);
+		if (def_conf == AC_JACK_PORT_NONE)
+			break;
+	}
+	return num;
+}
+
+static int stac92hd71bxx_connected_smuxes(struct hda_codec *codec,
+					  hda_nid_t dig0pin)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	int idx;
+
+	for (idx = 0; idx < spec->num_pins; idx++)
+		if (spec->pin_nids[idx] == dig0pin)
+			break;
+	if ((idx + 2) >= spec->num_pins)
+		return 0;
+
+	/* dig1pin case */
+	if (get_defcfg_connect(spec->pin_configs[idx+1]) != AC_JACK_PORT_NONE)
+		return 2;
+
+	/* dig0pin + dig2pin case */
+	if (get_defcfg_connect(spec->pin_configs[idx+2]) != AC_JACK_PORT_NONE)
+		return 2;
+	if (get_defcfg_connect(spec->pin_configs[idx]) != AC_JACK_PORT_NONE)
+		return 1;
+	else
+		return 0;
+}
+
 static int patch_stac92hd71bxx(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec;
 	struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init;
 	int err = 0;
+	unsigned int ndmic_nids = 0;
 
 	spec  = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -4981,8 +5036,6 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
 		spec->pin_nids = stac92hd71bxx_pin_nids_6port;
 	}
 	spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids);
-	memcpy(&spec->private_dimux, &stac92hd71bxx_dmux,
-			sizeof(stac92hd71bxx_dmux));
 	spec->board_config = snd_hda_check_board_config(codec,
 							STAC_92HD71BXX_MODELS,
 							stac92hd71bxx_models,
@@ -5007,16 +5060,32 @@ again:
 		spec->gpio_data = 0x01;
 	}
 
+	spec->dmic_nids = stac92hd71bxx_dmic_nids;
+	spec->dmux_nids = stac92hd71bxx_dmux_nids;
+
 	switch (codec->vendor_id) {
 	case 0x111d76b6: /* 4 Port without Analog Mixer */
 	case 0x111d76b7:
 	case 0x111d76b4: /* 6 Port without Analog Mixer */
 	case 0x111d76b5:
+		memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_nomixer,
+		       sizeof(stac92hd71bxx_dmux_nomixer));
 		spec->mixer = stac92hd71bxx_mixer;
 		spec->init = stac92hd71bxx_core_init;
 		codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
+		spec->num_dmics = stac92hd71bxx_connected_ports(codec,
+					stac92hd71bxx_dmic_nids,
+					STAC92HD71BXX_NUM_DMICS);
+		if (spec->num_dmics) {
+			spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
+			spec->dinput_mux = &spec->private_dimux;
+			ndmic_nids = ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1;
+		}
 		break;
 	case 0x111d7608: /* 5 Port with Analog Mixer */
+		memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_amixer,
+		       sizeof(stac92hd71bxx_dmux_amixer));
+		spec->private_dimux.num_items--;
 		switch (spec->board_config) {
 		case STAC_HP_M4:
 			/* Enable VREF power saving on GPIO1 detect */
@@ -5046,6 +5115,12 @@ again:
 		unmute_init++;
 		stac_change_pin_config(codec, 0x0f, 0x40f000f0);
 		stac_change_pin_config(codec, 0x19, 0x40f000f3);
+		stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS - 1] = 0;
+		spec->num_dmics = stac92hd71bxx_connected_ports(codec,
+					stac92hd71bxx_dmic_nids,
+					STAC92HD71BXX_NUM_DMICS - 1);
+		spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
+		ndmic_nids = ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 2;
 		break;
 	case 0x111d7603: /* 6 Port with Analog Mixer */
 		if ((codec->revision_id & 0xf) == 1)
@@ -5055,10 +5130,17 @@ again:
 		spec->num_pwrs = 0;
 		/* fallthru */
 	default:
+		memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_amixer,
+		       sizeof(stac92hd71bxx_dmux_amixer));
 		spec->dinput_mux = &spec->private_dimux;
 		spec->mixer = stac92hd71bxx_analog_mixer;
 		spec->init = stac92hd71bxx_analog_core_init;
 		codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
+		spec->num_dmics = stac92hd71bxx_connected_ports(codec,
+					stac92hd71bxx_dmic_nids,
+					STAC92HD71BXX_NUM_DMICS);
+		spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
+		ndmic_nids = ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1;
 	}
 
 	if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP)
@@ -5071,13 +5153,12 @@ again:
 	spec->digbeep_nid = 0x26;
 	spec->mux_nids = stac92hd71bxx_mux_nids;
 	spec->adc_nids = stac92hd71bxx_adc_nids;
-	spec->dmic_nids = stac92hd71bxx_dmic_nids;
-	spec->dmux_nids = stac92hd71bxx_dmux_nids;
 	spec->smux_nids = stac92hd71bxx_smux_nids;
 	spec->pwr_nids = stac92hd71bxx_pwr_nids;
 
 	spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids);
 	spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids);
+	spec->num_smuxes = stac92hd71bxx_connected_smuxes(codec, 0x1e);
 
 	switch (spec->board_config) {
 	case STAC_HP_M4:
@@ -5097,17 +5178,11 @@ again:
 		spec->num_smuxes = 0;
 		spec->num_dmuxes = 0;
 		break;
-	default:
-		spec->num_dmics = STAC92HD71BXX_NUM_DMICS;
-		spec->num_smuxes = ARRAY_SIZE(stac92hd71bxx_smux_nids);
-		spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
 	};
 
 	spec->multiout.dac_nids = spec->dac_nids;
 	if (spec->dinput_mux)
-		spec->private_dimux.num_items +=
-			spec->num_dmics -
-				(ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1);
+		spec->private_dimux.num_items += spec->num_dmics - ndmic_nids;
 
 	err = stac92xx_parse_auto_config(codec, 0x21, 0x23);
 	if (!err) {
-- 
cgit v1.2.3


From 29d4ab4d6e996ef4c71910c915611151c34f1c75 Mon Sep 17 00:00:00 2001
From: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Date: Wed, 4 Feb 2009 11:37:27 -0500
Subject: ALSA: hda - Don't call stac92xx_parse_auto_config with wrong dig_in

Don't use uneeded/wrong third parameter for stac92xx_parse_auto_config
in patch_stac92hd71bxx (no SPDIF in).

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index c36c1c0f957..0b00110a5a0 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -5184,7 +5184,7 @@ again:
 	if (spec->dinput_mux)
 		spec->private_dimux.num_items += spec->num_dmics - ndmic_nids;
 
-	err = stac92xx_parse_auto_config(codec, 0x21, 0x23);
+	err = stac92xx_parse_auto_config(codec, 0x21, 0);
 	if (!err) {
 		if (spec->board_config < 0) {
 			printk(KERN_WARNING "hda_codec: No auto-config is "
-- 
cgit v1.2.3


From 45c1d85bcc6438454d104966c30fd2497ae1cdd7 Mon Sep 17 00:00:00 2001
From: Matthew Ranostay <mranostay@embeddedalley.com>
Date: Wed, 4 Feb 2009 17:49:41 -0500
Subject: ALSA: hda: Added stac378x digital slave out struct

Added the ADATOut nid to a slave digital outs struct to allow output
via the DigOut pin.

Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 0b00110a5a0..85dc642d113 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -404,6 +404,10 @@ static hda_nid_t stac922x_mux_nids[2] = {
         0x12, 0x13,
 };
 
+static hda_nid_t stac927x_slave_dig_outs[2] = {
+	0x1f, 0,
+};
+
 static hda_nid_t stac927x_adc_nids[3] = {
         0x07, 0x08, 0x09
 };
@@ -5320,6 +5324,7 @@ static int patch_stac927x(struct hda_codec *codec)
 		return -ENOMEM;
 
 	codec->spec = spec;
+	codec->slave_dig_outs = stac927x_slave_dig_outs;
 	spec->num_pins = ARRAY_SIZE(stac927x_pin_nids);
 	spec->pin_nids = stac927x_pin_nids;
 	spec->board_config = snd_hda_check_board_config(codec, STAC_927X_MODELS,
-- 
cgit v1.2.3


From 28b7e343ee63454d563a71d2d5f769fc297fd5ad Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 5 Feb 2009 09:28:08 +0100
Subject: ALSA: Remove superfluous hwdep ops

Remove NOP hwdep ops in sound drivers.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/mixart/mixart_hwdep.c | 12 ------------
 sound/pci/pcxhr/pcxhr_hwdep.c   | 12 ------------
 sound/pci/rme9652/hdsp.c        |  9 ---------
 sound/pci/rme9652/hdspm.c       |  9 ---------
 4 files changed, 42 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/mixart/mixart_hwdep.c b/sound/pci/mixart/mixart_hwdep.c
index 3782b52bc0e..fa4de985fc4 100644
--- a/sound/pci/mixart/mixart_hwdep.c
+++ b/sound/pci/mixart/mixart_hwdep.c
@@ -581,16 +581,6 @@ MODULE_FIRMWARE("mixart/miXart8AES.xlx");
 /* miXart hwdep interface id string */
 #define SND_MIXART_HWDEP_ID       "miXart Loader"
 
-static int mixart_hwdep_open(struct snd_hwdep *hw, struct file *file)
-{
-	return 0;
-}
-
-static int mixart_hwdep_release(struct snd_hwdep *hw, struct file *file)
-{
-	return 0;
-}
-
 static int mixart_hwdep_dsp_status(struct snd_hwdep *hw,
 				   struct snd_hwdep_dsp_status *info)
 {
@@ -643,8 +633,6 @@ int snd_mixart_setup_firmware(struct mixart_mgr *mgr)
 
 	hw->iface = SNDRV_HWDEP_IFACE_MIXART;
 	hw->private_data = mgr;
-	hw->ops.open = mixart_hwdep_open;
-	hw->ops.release = mixart_hwdep_release;
 	hw->ops.dsp_status = mixart_hwdep_dsp_status;
 	hw->ops.dsp_load = mixart_hwdep_dsp_load;
 	hw->exclusive = 1;
diff --git a/sound/pci/pcxhr/pcxhr_hwdep.c b/sound/pci/pcxhr/pcxhr_hwdep.c
index 592743a298b..17cb1233a90 100644
--- a/sound/pci/pcxhr/pcxhr_hwdep.c
+++ b/sound/pci/pcxhr/pcxhr_hwdep.c
@@ -471,16 +471,6 @@ static int pcxhr_hwdep_dsp_load(struct snd_hwdep *hw,
 	return 0;
 }
 
-static int pcxhr_hwdep_open(struct snd_hwdep *hw, struct file *file)
-{
-	return 0;
-}
-
-static int pcxhr_hwdep_release(struct snd_hwdep *hw, struct file *file)
-{
-	return 0;
-}
-
 int pcxhr_setup_firmware(struct pcxhr_mgr *mgr)
 {
 	int err;
@@ -495,8 +485,6 @@ int pcxhr_setup_firmware(struct pcxhr_mgr *mgr)
 
 	hw->iface = SNDRV_HWDEP_IFACE_PCXHR;
 	hw->private_data = mgr;
-	hw->ops.open = pcxhr_hwdep_open;
-	hw->ops.release = pcxhr_hwdep_release;
 	hw->ops.dsp_status = pcxhr_hwdep_dsp_status;
 	hw->ops.dsp_load = pcxhr_hwdep_dsp_load;
 	hw->exclusive = 1;
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index 44d0c15e2b7..2434609b2d3 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -4413,13 +4413,6 @@ static int snd_hdsp_capture_release(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static int snd_hdsp_hwdep_dummy_op(struct snd_hwdep *hw, struct file *file)
-{
-	/* we have nothing to initialize but the call is required */
-	return 0;
-}
-
-
 /* helper functions for copying meter values */
 static inline int copy_u32_le(void __user *dest, void __iomem *src)
 {
@@ -4738,9 +4731,7 @@ static int snd_hdsp_create_hwdep(struct snd_card *card, struct hdsp *hdsp)
 	hw->private_data = hdsp;
 	strcpy(hw->name, "HDSP hwdep interface");
 
-	hw->ops.open = snd_hdsp_hwdep_dummy_op;
 	hw->ops.ioctl = snd_hdsp_hwdep_ioctl;
-	hw->ops.release = snd_hdsp_hwdep_dummy_op;
 		
 	return 0;
 }
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 71231cf1b2b..df2034eb235 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -4100,13 +4100,6 @@ static int snd_hdspm_capture_release(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static int snd_hdspm_hwdep_dummy_op(struct snd_hwdep * hw, struct file *file)
-{
-	/* we have nothing to initialize but the call is required */
-	return 0;
-}
-
-
 static int snd_hdspm_hwdep_ioctl(struct snd_hwdep * hw, struct file *file,
 				 unsigned int cmd, unsigned long arg)
 {
@@ -4213,9 +4206,7 @@ static int __devinit snd_hdspm_create_hwdep(struct snd_card *card,
 	hw->private_data = hdspm;
 	strcpy(hw->name, "HDSPM hwdep interface");
 
-	hw->ops.open = snd_hdspm_hwdep_dummy_op;
 	hw->ops.ioctl = snd_hdspm_hwdep_ioctl;
-	hw->ops.release = snd_hdspm_hwdep_dummy_op;
 
 	return 0;
 }
-- 
cgit v1.2.3


From 67f7857ab12e9f8005ef988f0b667396e07622c2 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 5 Feb 2009 12:14:52 +0100
Subject: ALSA: hda - Add quirk for HP zenith laptop

Added model=laptop for another HP laptop (103c:3072) with AD1984A codec.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index e934e2c187d..6e348d03b71 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3890,6 +3890,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE),
 	SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP),
 	SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE),
+	SND_PCI_QUIRK(0x103c, 0x3072, "HP", AD1884A_LAPTOP),
 	SND_PCI_QUIRK(0x103c, 0x30e6, "HP 6730b", AD1884A_LAPTOP),
 	SND_PCI_QUIRK(0x103c, 0x30e7, "HP EliteBook 8530p", AD1884A_LAPTOP),
 	SND_PCI_QUIRK(0x103c, 0x3614, "HP 6730s", AD1884A_LAPTOP),
-- 
cgit v1.2.3


From 632da7321b7e9fa5375956280f8a0f380836c22d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 5 Feb 2009 15:02:06 +0100
Subject: ALSA: hda - Add quirk for another HP laptop

Add model=laptop entry for another HP laptop (103c:3077) with AD1984A.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 6e348d03b71..30399cbf819 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3891,6 +3891,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP),
 	SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE),
 	SND_PCI_QUIRK(0x103c, 0x3072, "HP", AD1884A_LAPTOP),
+	SND_PCI_QUIRK(0x103c, 0x3077, "HP", AD1884A_LAPTOP),
 	SND_PCI_QUIRK(0x103c, 0x30e6, "HP 6730b", AD1884A_LAPTOP),
 	SND_PCI_QUIRK(0x103c, 0x30e7, "HP EliteBook 8530p", AD1884A_LAPTOP),
 	SND_PCI_QUIRK(0x103c, 0x3614, "HP 6730s", AD1884A_LAPTOP),
-- 
cgit v1.2.3


From 939778aedd9386e13051a9e1d57c14cba2b6ae13 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 5 Feb 2009 15:57:55 +0100
Subject: ALSA: hda - Add missing KERN_* prefix to printk

... and disable the annoying debug message.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 5218118f01b..d2812ab729c 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8265,7 +8265,7 @@ static void alc888_6st_dell_unsol_event(struct hda_codec *codec,
 {
 	switch (res >> 26) {
 	case ALC880_HP_EVENT:
-		printk("hp_event\n");
+		/* printk(KERN_DEBUG "hp_event\n"); */
 		alc888_6st_dell_front_automute(codec);
 		break;
 	}
@@ -16564,7 +16564,7 @@ static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin,
 
 	if (alc880_is_fixed_pin(pin)) {
 		nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin));
-                /* printk("DAC nid=%x\n",nid); */
+		/* printk(KERN_DEBUG "DAC nid=%x\n",nid); */
 		/* specify the DAC as the extra output */
 		if (!spec->multiout.hp_nid)
 			spec->multiout.hp_nid = nid;
-- 
cgit v1.2.3


From e2ea7cfc703cba3299d22db728516a0fc1a9717c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 5 Feb 2009 16:07:02 +0100
Subject: ALSA: Add missing KERN_* prefix to printk in sound/pci/ice1712

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ice1712/ice1712.c    |  2 +-
 sound/pci/ice1712/ice1724.c    | 17 ++++++++++++++---
 sound/pci/ice1712/juli.c       |  5 +++--
 sound/pci/ice1712/prodigy192.c | 13 +++++++++----
 4 files changed, 27 insertions(+), 10 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index 58d7cda03de..dcd3f4f89b4 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -458,7 +458,7 @@ static irqreturn_t snd_ice1712_interrupt(int irq, void *dev_id)
 			u16 pbkstatus;
 			struct snd_pcm_substream *substream;
 			pbkstatus = inw(ICEDS(ice, INTSTAT));
-			/* printk("pbkstatus = 0x%x\n", pbkstatus); */
+			/* printk(KERN_DEBUG "pbkstatus = 0x%x\n", pbkstatus); */
 			for (idx = 0; idx < 6; idx++) {
 				if ((pbkstatus & (3 << (idx * 2))) == 0)
 					continue;
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index eb7872dec5a..da8c111e9e3 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -756,7 +756,14 @@ static int snd_vt1724_playback_pro_prepare(struct snd_pcm_substream *substream)
 
 	spin_unlock_irq(&ice->reg_lock);
 
-	/* printk("pro prepare: ch = %d, addr = 0x%x, buffer = 0x%x, period = 0x%x\n", substream->runtime->channels, (unsigned int)substream->runtime->dma_addr, snd_pcm_lib_buffer_bytes(substream), snd_pcm_lib_period_bytes(substream)); */
+	/*
+	printk(KERN_DEBUG "pro prepare: ch = %d, addr = 0x%x, "
+	       "buffer = 0x%x, period = 0x%x\n",
+	       substream->runtime->channels,
+	       (unsigned int)substream->runtime->dma_addr,
+	       snd_pcm_lib_buffer_bytes(substream),
+	       snd_pcm_lib_period_bytes(substream));
+	*/
 	return 0;
 }
 
@@ -2133,7 +2140,9 @@ unsigned char snd_vt1724_read_i2c(struct snd_ice1712 *ice,
 	wait_i2c_busy(ice);
 	val = inb(ICEREG1724(ice, I2C_DATA));
 	mutex_unlock(&ice->i2c_mutex);
-	/* printk("i2c_read: [0x%x,0x%x] = 0x%x\n", dev, addr, val); */
+	/*
+	printk(KERN_DEBUG "i2c_read: [0x%x,0x%x] = 0x%x\n", dev, addr, val);
+	*/
 	return val;
 }
 
@@ -2142,7 +2151,9 @@ void snd_vt1724_write_i2c(struct snd_ice1712 *ice,
 {
 	mutex_lock(&ice->i2c_mutex);
 	wait_i2c_busy(ice);
-	/* printk("i2c_write: [0x%x,0x%x] = 0x%x\n", dev, addr, data); */
+	/*
+	printk(KERN_DEBUG "i2c_write: [0x%x,0x%x] = 0x%x\n", dev, addr, data);
+	*/
 	outb(addr, ICEREG1724(ice, I2C_BYTE_ADDR));
 	outb(data, ICEREG1724(ice, I2C_DATA));
 	outb(dev | VT1724_I2C_WRITE, ICEREG1724(ice, I2C_DEV_ADDR));
diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c
index c51659b9caf..fd948bfd9ae 100644
--- a/sound/pci/ice1712/juli.c
+++ b/sound/pci/ice1712/juli.c
@@ -345,8 +345,9 @@ static int juli_mute_put(struct snd_kcontrol *kcontrol,
 			new_gpio =  old_gpio &
 				~((unsigned int) kcontrol->private_value);
 	}
-	/* printk("JULI - mute/unmute: control_value: 0x%x, old_gpio: 0x%x, \
-		new_gpio 0x%x\n",
+	/* printk(KERN_DEBUG
+		"JULI - mute/unmute: control_value: 0x%x, old_gpio: 0x%x, "
+		"new_gpio 0x%x\n",
 		(unsigned int)ucontrol->value.integer.value[0], old_gpio,
 		new_gpio); */
 	if (old_gpio != new_gpio) {
diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c
index 48d3679292a..2a8e5cd8f2d 100644
--- a/sound/pci/ice1712/prodigy192.c
+++ b/sound/pci/ice1712/prodigy192.c
@@ -133,8 +133,10 @@ static int stac9460_dac_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e
 		idx  = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + STAC946X_LF_VOLUME;
 	/* due to possible conflicts with stac9460_set_rate_val, mutexing */
 	mutex_lock(&spec->mute_mutex);
-	/*printk("Mute put: reg 0x%02x, ctrl value: 0x%02x\n", idx,
-		ucontrol->value.integer.value[0]);*/
+	/*
+	printk(KERN_DEBUG "Mute put: reg 0x%02x, ctrl value: 0x%02x\n", idx,
+	       ucontrol->value.integer.value[0]);
+	*/
 	change = stac9460_dac_mute(ice, idx, ucontrol->value.integer.value[0]);
 	mutex_unlock(&spec->mute_mutex);
 	return change;
@@ -185,7 +187,10 @@ static int stac9460_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_el
 	change = (ovol != nvol);
 	if (change) {
 		ovol =  (0x7f - nvol) | (tmp & 0x80);
-		/*printk("DAC Volume: reg 0x%02x: 0x%02x\n", idx, ovol);*/
+		/*
+		printk(KERN_DEBUG "DAC Volume: reg 0x%02x: 0x%02x\n",
+		       idx, ovol);
+		*/
 		stac9460_put(ice, idx, (0x7f - nvol) | (tmp & 0x80));
 	}
 	return change;
@@ -344,7 +349,7 @@ static void stac9460_set_rate_val(struct snd_ice1712 *ice, unsigned int rate)
 	for (idx = 0; idx < 7 ; ++idx)
 		changed[idx] = stac9460_dac_mute(ice,
 				STAC946X_MASTER_VOLUME + idx, 0);
-	/*printk("Rate change: %d, new MC: 0x%02x\n", rate, new);*/
+	/*printk(KERN_DEBUG "Rate change: %d, new MC: 0x%02x\n", rate, new);*/
 	stac9460_put(ice, STAC946X_MASTER_CLOCKING, new);
 	udelay(10);
 	/* unmuting - only originally unmuted dacs -
-- 
cgit v1.2.3


From 28a97c194cec477073ae341f15b836437d8ef8e5 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 5 Feb 2009 16:08:14 +0100
Subject: ALSA: emu10k1 - Add missing KERN_* prefix to printk

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/emu10k1/emu10k1_callback.c |   7 ++-
 sound/pci/emu10k1/emu10k1_main.c     |   5 +-
 sound/pci/emu10k1/emufx.c            |  11 ++--
 sound/pci/emu10k1/emupcm.c           |  37 ++++++++++---
 sound/pci/emu10k1/io.c               |   4 +-
 sound/pci/emu10k1/p16v.c             | 100 +++++++++++++++++++++++++----------
 sound/pci/emu10k1/voice.c            |  12 +++--
 7 files changed, 130 insertions(+), 46 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/emu10k1/emu10k1_callback.c b/sound/pci/emu10k1/emu10k1_callback.c
index 0e649dcdbf6..7ef949d99a5 100644
--- a/sound/pci/emu10k1/emu10k1_callback.c
+++ b/sound/pci/emu10k1/emu10k1_callback.c
@@ -103,7 +103,10 @@ snd_emu10k1_synth_get_voice(struct snd_emu10k1 *hw)
 			int ch;
 			vp = &emu->voices[best[i].voice];
 			if ((ch = vp->ch) < 0) {
-				//printk("synth_get_voice: ch < 0 (%d) ??", i);
+				/*
+				printk(KERN_WARNING
+				       "synth_get_voice: ch < 0 (%d) ??", i);
+				*/
 				continue;
 			}
 			vp->emu->num_voices--;
@@ -335,7 +338,7 @@ start_voice(struct snd_emux_voice *vp)
 		return -EINVAL;
 	emem->map_locked++;
 	if (snd_emu10k1_memblk_map(hw, emem) < 0) {
-		// printk("emu: cannot map!\n");
+		/* printk(KERN_ERR "emu: cannot map!\n"); */
 		return -ENOMEM;
 	}
 	mapped_offset = snd_emu10k1_memblk_offset(emem) >> 1;
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 7958006a1d6..8343aecbd25 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -758,7 +758,8 @@ static int emu1010_firmware_thread(void *data)
 			snd_printk(KERN_INFO "emu1010: Audio Dock Firmware loaded\n");
 			snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp);
 			snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2);
-			snd_printk("Audio Dock ver:%d.%d\n", tmp, tmp2);
+			snd_printk(KERN_INFO "Audio Dock ver:%d.%d\n",
+				   tmp, tmp2);
 			/* Sync clocking between 1010 and Dock */
 			/* Allow DLL to settle */
 			msleep(10);
@@ -887,7 +888,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu)
 	snd_printk(KERN_INFO "emu1010: Hana Firmware loaded\n");
 	snd_emu1010_fpga_read(emu, EMU_HANA_MAJOR_REV, &tmp);
 	snd_emu1010_fpga_read(emu, EMU_HANA_MINOR_REV, &tmp2);
-	snd_printk("emu1010: Hana version: %d.%d\n", tmp, tmp2);
+	snd_printk(KERN_INFO "emu1010: Hana version: %d.%d\n", tmp, tmp2);
 	/* Enable 48Volt power to Audio Dock */
 	snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, EMU_HANA_DOCK_PWR_ON);
 
diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c
index 7dba08f0ab8..191e1cd9997 100644
--- a/sound/pci/emu10k1/emufx.c
+++ b/sound/pci/emu10k1/emufx.c
@@ -1519,7 +1519,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input))
 	/* A_PUT_STEREO_OUTPUT(A_EXTOUT_FRONT_L, A_EXTOUT_FRONT_R, playback + SND_EMU10K1_PLAYBACK_CHANNELS); */
 	if (emu->card_capabilities->emu_model) {
 		/* EMU1010 Outputs from PCM Front, Rear, Center, LFE, Side */
-		snd_printk("EMU outputs on\n");
+		snd_printk(KERN_INFO "EMU outputs on\n");
 		for (z = 0; z < 8; z++) {
 			if (emu->card_capabilities->ca0108_chip) {
 				A_OP(icode, &ptr, iACC3, A3_EMU32OUT(z), A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + z), A_C_00000000, A_C_00000000);
@@ -1567,7 +1567,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input))
 
 	if (emu->card_capabilities->emu_model) {
 		if (emu->card_capabilities->ca0108_chip) {
-			snd_printk("EMU2 inputs on\n");
+			snd_printk(KERN_INFO "EMU2 inputs on\n");
 			for (z = 0; z < 0x10; z++) {
 				snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, 
 									bit_shifter16,
@@ -1575,10 +1575,13 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input))
 									A_FXBUS2(z*2) );
 			}
 		} else {
-			snd_printk("EMU inputs on\n");
+			snd_printk(KERN_INFO "EMU inputs on\n");
 			/* Capture 16 (originally 8) channels of S32_LE sound */
 
-			/* printk("emufx.c: gpr=0x%x, tmp=0x%x\n",gpr, tmp); */
+			/*
+			printk(KERN_DEBUG "emufx.c: gpr=0x%x, tmp=0x%x\n",
+			       gpr, tmp);
+			*/
 			/* For the EMU1010: How to get 32bit values from the DSP. High 16bits into L, low 16bits into R. */
 			/* A_P16VIN(0) is delayed by one sample,
 			 * so all other A_P16VIN channels will need to also be delayed
diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c
index cf9276ddad4..78f62fd404c 100644
--- a/sound/pci/emu10k1/emupcm.c
+++ b/sound/pci/emu10k1/emupcm.c
@@ -44,7 +44,7 @@ static void snd_emu10k1_pcm_interrupt(struct snd_emu10k1 *emu,
 	if (epcm->substream == NULL)
 		return;
 #if 0
-	printk("IRQ: position = 0x%x, period = 0x%x, size = 0x%x\n",
+	printk(KERN_DEBUG "IRQ: position = 0x%x, period = 0x%x, size = 0x%x\n",
 			epcm->substream->runtime->hw->pointer(emu, epcm->substream),
 			snd_pcm_lib_period_bytes(epcm->substream),
 			snd_pcm_lib_buffer_bytes(epcm->substream));
@@ -146,7 +146,11 @@ static int snd_emu10k1_pcm_channel_alloc(struct snd_emu10k1_pcm * epcm, int voic
 					      1,
 					      &epcm->extra);
 		if (err < 0) {
-			/* printk("pcm_channel_alloc: failed extra: voices=%d, frame=%d\n", voices, frame); */
+			/*
+			printk(KERN_DEBUG "pcm_channel_alloc: "
+			       "failed extra: voices=%d, frame=%d\n",
+			       voices, frame);
+			*/
 			for (i = 0; i < voices; i++) {
 				snd_emu10k1_voice_free(epcm->emu, epcm->voices[i]);
 				epcm->voices[i] = NULL;
@@ -737,7 +741,10 @@ static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream,
 	struct snd_emu10k1_pcm_mixer *mix;
 	int result = 0;
 
-	/* printk("trigger - emu10k1 = 0x%x, cmd = %i, pointer = %i\n", (int)emu, cmd, substream->ops->pointer(substream)); */
+	/*
+	printk(KERN_DEBUG "trigger - emu10k1 = 0x%x, cmd = %i, pointer = %i\n",
+	       (int)emu, cmd, substream->ops->pointer(substream))
+	*/
 	spin_lock(&emu->reg_lock);
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
@@ -786,7 +793,10 @@ static int snd_emu10k1_capture_trigger(struct snd_pcm_substream *substream,
 		/* hmm this should cause full and half full interrupt to be raised? */
 		outl(epcm->capture_ipr, emu->port + IPR);
 		snd_emu10k1_intr_enable(emu, epcm->capture_inte);
-		/* printk("adccr = 0x%x, adcbs = 0x%x\n", epcm->adccr, epcm->adcbs); */
+		/*
+		printk(KERN_DEBUG "adccr = 0x%x, adcbs = 0x%x\n",
+		       epcm->adccr, epcm->adcbs);
+		*/
 		switch (epcm->type) {
 		case CAPTURE_AC97ADC:
 			snd_emu10k1_ptr_write(emu, ADCCR, 0, epcm->capture_cr_val);
@@ -857,7 +867,11 @@ static snd_pcm_uframes_t snd_emu10k1_playback_pointer(struct snd_pcm_substream *
 			ptr -= runtime->buffer_size;
 	}
 #endif
-	/* printk("ptr = 0x%x, buffer_size = 0x%x, period_size = 0x%x\n", ptr, runtime->buffer_size, runtime->period_size); */
+	/*
+	printk(KERN_DEBUG
+	       "ptr = 0x%x, buffer_size = 0x%x, period_size = 0x%x\n",
+	       ptr, runtime->buffer_size, runtime->period_size);
+	*/
 	return ptr;
 }
 
@@ -1546,7 +1560,11 @@ static void snd_emu10k1_fx8010_playback_tram_poke1(unsigned short *dst_left,
 						   unsigned int count,
 						   unsigned int tram_shift)
 {
-	/* printk("tram_poke1: dst_left = 0x%p, dst_right = 0x%p, src = 0x%p, count = 0x%x\n", dst_left, dst_right, src, count); */
+	/*
+	printk(KERN_DEBUG "tram_poke1: dst_left = 0x%p, dst_right = 0x%p, "
+	       "src = 0x%p, count = 0x%x\n",
+	       dst_left, dst_right, src, count);
+	*/
 	if ((tram_shift & 1) == 0) {
 		while (count--) {
 			*dst_left-- = *src++;
@@ -1623,7 +1641,12 @@ static int snd_emu10k1_fx8010_playback_prepare(struct snd_pcm_substream *substre
 	struct snd_emu10k1_fx8010_pcm *pcm = &emu->fx8010.pcm[substream->number];
 	unsigned int i;
 	
-	/* printk("prepare: etram_pages = 0x%p, dma_area = 0x%x, buffer_size = 0x%x (0x%x)\n", emu->fx8010.etram_pages, runtime->dma_area, runtime->buffer_size, runtime->buffer_size << 2); */
+	/*
+	printk(KERN_DEBUG "prepare: etram_pages = 0x%p, dma_area = 0x%x, "
+	       "buffer_size = 0x%x (0x%x)\n",
+	       emu->fx8010.etram_pages, runtime->dma_area,
+	       runtime->buffer_size, runtime->buffer_size << 2);
+	*/
 	memset(&pcm->pcm_rec, 0, sizeof(pcm->pcm_rec));
 	pcm->pcm_rec.hw_buffer_size = pcm->buffer_size * 2; /* byte size */
 	pcm->pcm_rec.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream);
diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c
index b5a802bdeb7..4bfc31d1b28 100644
--- a/sound/pci/emu10k1/io.c
+++ b/sound/pci/emu10k1/io.c
@@ -226,7 +226,9 @@ int snd_emu10k1_i2c_write(struct snd_emu10k1 *emu,
 				break;
 
 			if (timeout > 1000) {
-                		snd_printk("emu10k1:I2C:timeout status=0x%x\n", status);
+                		snd_printk(KERN_WARNING
+					   "emu10k1:I2C:timeout status=0x%x\n",
+					   status);
 				break;
 			}
 		}
diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c
index 749a21b6bd0..e617acaf10e 100644
--- a/sound/pci/emu10k1/p16v.c
+++ b/sound/pci/emu10k1/p16v.c
@@ -168,7 +168,7 @@ static void snd_p16v_pcm_free_substream(struct snd_pcm_runtime *runtime)
 	struct snd_emu10k1_pcm *epcm = runtime->private_data;
   
 	if (epcm) {
-        	//snd_printk("epcm free: %p\n", epcm);
+        	/* snd_printk(KERN_DEBUG "epcm free: %p\n", epcm); */
 		kfree(epcm);
 	}
 }
@@ -183,14 +183,16 @@ static int snd_p16v_pcm_open_playback_channel(struct snd_pcm_substream *substrea
 	int err;
 
 	epcm = kzalloc(sizeof(*epcm), GFP_KERNEL);
-        //snd_printk("epcm kcalloc: %p\n", epcm);
+        /* snd_printk(KERN_DEBUG "epcm kcalloc: %p\n", epcm); */
 
 	if (epcm == NULL)
 		return -ENOMEM;
 	epcm->emu = emu;
 	epcm->substream = substream;
-        //snd_printk("epcm device=%d, channel_id=%d\n", substream->pcm->device, channel_id);
-  
+	/*
+	snd_printk(KERN_DEBUG "epcm device=%d, channel_id=%d\n",
+		   substream->pcm->device, channel_id);
+	*/
 	runtime->private_data = epcm;
 	runtime->private_free = snd_p16v_pcm_free_substream;
   
@@ -200,10 +202,15 @@ static int snd_p16v_pcm_open_playback_channel(struct snd_pcm_substream *substrea
         channel->number = channel_id;
 
         channel->use=1;
-	//snd_printk("p16v: open channel_id=%d, channel=%p, use=0x%x\n", channel_id, channel, channel->use);
-        //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel);
-        //channel->interrupt = snd_p16v_pcm_channel_interrupt;
-        channel->epcm=epcm;
+#if 0 /* debug */
+	snd_printk(KERN_DEBUG
+		   "p16v: open channel_id=%d, channel=%p, use=0x%x\n",
+		   channel_id, channel, channel->use);
+	printk(KERN_DEBUG "open:channel_id=%d, chip=%p, channel=%p\n",
+	       channel_id, chip, channel);
+#endif /* debug */
+	/* channel->interrupt = snd_p16v_pcm_channel_interrupt; */
+	channel->epcm = epcm;
 	if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
                 return err;
 
@@ -224,14 +231,16 @@ static int snd_p16v_pcm_open_capture_channel(struct snd_pcm_substream *substream
 	int err;
 
 	epcm = kzalloc(sizeof(*epcm), GFP_KERNEL);
-	//snd_printk("epcm kcalloc: %p\n", epcm);
+	/* snd_printk(KERN_DEBUG "epcm kcalloc: %p\n", epcm); */
 
 	if (epcm == NULL)
 		return -ENOMEM;
 	epcm->emu = emu;
 	epcm->substream = substream;
-	//snd_printk("epcm device=%d, channel_id=%d\n", substream->pcm->device, channel_id);
-
+	/*
+	snd_printk(KERN_DEBUG "epcm device=%d, channel_id=%d\n",
+		   substream->pcm->device, channel_id);
+	*/
 	runtime->private_data = epcm;
 	runtime->private_free = snd_p16v_pcm_free_substream;
   
@@ -241,10 +250,15 @@ static int snd_p16v_pcm_open_capture_channel(struct snd_pcm_substream *substream
 	channel->number = channel_id;
 
 	channel->use=1;
-	//snd_printk("p16v: open channel_id=%d, channel=%p, use=0x%x\n", channel_id, channel, channel->use);
-	//printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel);
-	//channel->interrupt = snd_p16v_pcm_channel_interrupt;
-	channel->epcm=epcm;
+#if 0 /* debug */
+	snd_printk(KERN_DEBUG
+		   "p16v: open channel_id=%d, channel=%p, use=0x%x\n",
+		   channel_id, channel, channel->use);
+	printk(KERN_DEBUG "open:channel_id=%d, chip=%p, channel=%p\n",
+	       channel_id, chip, channel);
+#endif /* debug */
+	/* channel->interrupt = snd_p16v_pcm_channel_interrupt; */
+	channel->epcm = epcm;
 	if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
 		return err;
 
@@ -334,9 +348,19 @@ static int snd_p16v_pcm_prepare_playback(struct snd_pcm_substream *substream)
 	int i;
 	u32 tmp;
 	
-        //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1));
-        //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base);
-	//snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->p16v_buffer.addr, emu->p16v_buffer.area, emu->p16v_buffer.bytes);
+#if 0 /* debug */
+	snd_printk(KERN_DEBUG "prepare:channel_number=%d, rate=%d, "
+		   "format=0x%x, channels=%d, buffer_size=%ld, "
+		   "period_size=%ld, periods=%u, frames_to_bytes=%d\n",
+		   channel, runtime->rate, runtime->format, runtime->channels,
+		   runtime->buffer_size, runtime->period_size,
+		   runtime->periods, frames_to_bytes(runtime, 1));
+	snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, table_base=%p\n",
+		   runtime->dma_addr, runtime->dma_area, table_base);
+	snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",
+		   emu->p16v_buffer.addr, emu->p16v_buffer.area,
+		   emu->p16v_buffer.bytes);
+#endif /* debug */
 	tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, channel);
         switch (runtime->rate) {
 	case 44100:
@@ -379,7 +403,15 @@ static int snd_p16v_pcm_prepare_capture(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	int channel = substream->pcm->device - emu->p16v_device_offset;
 	u32 tmp;
-	//printk("prepare capture:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size,  frames_to_bytes(runtime, 1));
+
+	/*
+	printk(KERN_DEBUG "prepare capture:channel_number=%d, rate=%d, "
+	       "format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, "
+	       "frames_to_bytes=%d\n",
+	       channel, runtime->rate, runtime->format, runtime->channels,
+	       runtime->buffer_size, runtime->period_size,
+	       frames_to_bytes(runtime, 1));
+	*/
 	tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, channel);
         switch (runtime->rate) {
 	case 44100:
@@ -459,13 +491,13 @@ static int snd_p16v_pcm_trigger_playback(struct snd_pcm_substream *substream,
 		runtime = s->runtime;
 		epcm = runtime->private_data;
 		channel = substream->pcm->device-emu->p16v_device_offset;
-		//snd_printk("p16v channel=%d\n",channel);
+		/* snd_printk(KERN_DEBUG "p16v channel=%d\n", channel); */
 		epcm->running = running;
 		basic |= (0x1<<channel);
 		inte |= (INTE2_PLAYBACK_CH_0_LOOP<<channel);
                 snd_pcm_trigger_done(s, substream);
         }
-	//snd_printk("basic=0x%x, inte=0x%x\n",basic, inte);
+	/* snd_printk(KERN_DEBUG "basic=0x%x, inte=0x%x\n", basic, inte); */
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
@@ -558,8 +590,13 @@ snd_p16v_pcm_pointer_capture(struct snd_pcm_substream *substream)
 		ptr -= runtime->buffer_size;
 		printk(KERN_WARNING "buffer capture limited!\n");
 	}
-	//printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate);
-
+	/*
+	printk(KERN_DEBUG "ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, "
+	       "buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n",
+	       ptr1, ptr2, ptr, (int)runtime->buffer_size,
+	       (int)runtime->period_size, (int)runtime->frame_bits,
+	       (int)runtime->rate);
+	*/
 	return ptr;
 }
 
@@ -592,7 +629,10 @@ int snd_p16v_free(struct snd_emu10k1 *chip)
 	// release the data
 	if (chip->p16v_buffer.area) {
 		snd_dma_free_pages(&chip->p16v_buffer);
-		//snd_printk("period lables free: %p\n", &chip->p16v_buffer);
+		/*
+		snd_printk(KERN_DEBUG "period lables free: %p\n",
+			   &chip->p16v_buffer);
+		*/
 	}
 	return 0;
 }
@@ -604,7 +644,7 @@ int __devinit snd_p16v_pcm(struct snd_emu10k1 *emu, int device, struct snd_pcm *
 	int err;
         int capture=1;
   
-	//snd_printk("snd_p16v_pcm called. device=%d\n", device);
+	/* snd_printk("KERN_DEBUG snd_p16v_pcm called. device=%d\n", device); */
 	emu->p16v_device_offset = device;
 	if (rpcm)
 		*rpcm = NULL;
@@ -631,7 +671,10 @@ int __devinit snd_p16v_pcm(struct snd_emu10k1 *emu, int device, struct snd_pcm *
 							 snd_dma_pci_data(emu->pci), 
 							 ((65536 - 64) * 8), ((65536 - 64) * 8))) < 0) 
 			return err;
-		//snd_printk("preallocate playback substream: err=%d\n", err);
+		/*
+		snd_printk(KERN_DEBUG
+			   "preallocate playback substream: err=%d\n", err);
+		*/
 	}
 
 	for (substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; 
@@ -642,7 +685,10 @@ int __devinit snd_p16v_pcm(struct snd_emu10k1 *emu, int device, struct snd_pcm *
 	                                           snd_dma_pci_data(emu->pci), 
 	                                           65536 - 64, 65536 - 64)) < 0)
 			return err;
-		//snd_printk("preallocate capture substream: err=%d\n", err);
+		/*
+		snd_printk(KERN_DEBUG
+			   "preallocate capture substream: err=%d\n", err);
+		*/
 	}
   
 	if (rpcm)
diff --git a/sound/pci/emu10k1/voice.c b/sound/pci/emu10k1/voice.c
index d7300a1aa26..20b8da250bd 100644
--- a/sound/pci/emu10k1/voice.c
+++ b/sound/pci/emu10k1/voice.c
@@ -53,7 +53,10 @@ static int voice_alloc(struct snd_emu10k1 *emu, int type, int number,
 	*rvoice = NULL;
 	first_voice = last_voice = 0;
 	for (i = emu->next_free_voice, j = 0; j < NUM_G ; i += number, j += number) {
-		// printk("i %d j %d next free %d!\n", i, j, emu->next_free_voice);
+		/*
+		printk(KERN_DEBUG "i %d j %d next free %d!\n",
+		       i, j, emu->next_free_voice);
+		*/
 		i %= NUM_G;
 
 		/* stereo voices must be even/odd */
@@ -71,7 +74,7 @@ static int voice_alloc(struct snd_emu10k1 *emu, int type, int number,
 			}
 		}
 		if (!skip) {
-			// printk("allocated voice %d\n", i);
+			/* printk(KERN_DEBUG "allocated voice %d\n", i); */
 			first_voice = i;
 			last_voice = (i + number) % NUM_G;
 			emu->next_free_voice = last_voice;
@@ -84,7 +87,10 @@ static int voice_alloc(struct snd_emu10k1 *emu, int type, int number,
 	
 	for (i = 0; i < number; i++) {
 		voice = &emu->voices[(first_voice + i) % NUM_G];
-		// printk("voice alloc - %i, %i of %i\n", voice->number, idx-first_voice+1, number);
+		/*
+		printk(kERN_DEBUG "voice alloc - %i, %i of %i\n",
+		       voice->number, idx-first_voice+1, number);
+		*/
 		voice->use = 1;
 		switch (type) {
 		case EMU10K1_PCM:
-- 
cgit v1.2.3


From 14ab08610971eb1db572ad8ca63acd13bc4d4caf Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 5 Feb 2009 16:09:57 +0100
Subject: ALSA: intel8x0 - Add missing KERN_* prefix to printk

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/intel8x0.c  | 11 +++++++----
 sound/pci/intel8x0m.c | 14 ++++++++++----
 2 files changed, 17 insertions(+), 8 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index b13ef1e2a4a..0f7d1291190 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -689,7 +689,7 @@ static void snd_intel8x0_setup_periods(struct intel8x0 *chip, struct ichdev *ich
 			bdbar[idx + 1] = cpu_to_le32(0x80000000 | /* interrupt on completion */
 						     ichdev->fragsize >> ichdev->pos_shift);
 #if 0
-			printk("bdbar[%i] = 0x%x [0x%x]\n",
+			printk(KERN_DEBUG "bdbar[%i] = 0x%x [0x%x]\n",
 			       idx + 0, bdbar[idx + 0], bdbar[idx + 1]);
 #endif
 		}
@@ -701,8 +701,10 @@ static void snd_intel8x0_setup_periods(struct intel8x0 *chip, struct ichdev *ich
 	ichdev->lvi_frag = ICH_REG_LVI_MASK % ichdev->frags;
 	ichdev->position = 0;
 #if 0
-	printk("lvi_frag = %i, frags = %i, period_size = 0x%x, period_size1 = 0x%x\n",
-			ichdev->lvi_frag, ichdev->frags, ichdev->fragsize, ichdev->fragsize1);
+	printk(KERN_DEBUG "lvi_frag = %i, frags = %i, period_size = 0x%x, "
+	       "period_size1 = 0x%x\n",
+	       ichdev->lvi_frag, ichdev->frags, ichdev->fragsize,
+	       ichdev->fragsize1);
 #endif
 	/* clear interrupts */
 	iputbyte(chip, port + ichdev->roff_sr, ICH_FIFOE | ICH_BCIS | ICH_LVBCI);
@@ -768,7 +770,8 @@ static inline void snd_intel8x0_update(struct intel8x0 *chip, struct ichdev *ich
 		ichdev->lvi_frag %= ichdev->frags;
 		ichdev->bdbar[ichdev->lvi * 2] = cpu_to_le32(ichdev->physbuf + ichdev->lvi_frag * ichdev->fragsize1);
 #if 0
-	printk("new: bdbar[%i] = 0x%x [0x%x], prefetch = %i, all = 0x%x, 0x%x\n",
+	printk(KERN_DEBUG "new: bdbar[%i] = 0x%x [0x%x], prefetch = %i, "
+	       "all = 0x%x, 0x%x\n",
 	       ichdev->lvi * 2, ichdev->bdbar[ichdev->lvi * 2],
 	       ichdev->bdbar[ichdev->lvi * 2 + 1], inb(ICH_REG_OFF_PIV + port),
 	       inl(port + 4), inb(port + ICH_REG_OFF_CR));
diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c
index 93449e46456..7c819fd824a 100644
--- a/sound/pci/intel8x0m.c
+++ b/sound/pci/intel8x0m.c
@@ -411,7 +411,10 @@ static void snd_intel8x0_setup_periods(struct intel8x0m *chip, struct ichdev *ic
 			bdbar[idx + 0] = cpu_to_le32(ichdev->physbuf + (((idx >> 1) * ichdev->fragsize) % ichdev->size));
 			bdbar[idx + 1] = cpu_to_le32(0x80000000 | /* interrupt on completion */
 						     ichdev->fragsize >> chip->pcm_pos_shift);
-			// printk("bdbar[%i] = 0x%x [0x%x]\n", idx + 0, bdbar[idx + 0], bdbar[idx + 1]);
+			/*
+			printk(KERN_DEBUG "bdbar[%i] = 0x%x [0x%x]\n",
+			       idx + 0, bdbar[idx + 0], bdbar[idx + 1]);
+			*/
 		}
 		ichdev->frags = ichdev->size / ichdev->fragsize;
 	}
@@ -421,8 +424,10 @@ static void snd_intel8x0_setup_periods(struct intel8x0m *chip, struct ichdev *ic
 	ichdev->lvi_frag = ICH_REG_LVI_MASK % ichdev->frags;
 	ichdev->position = 0;
 #if 0
-	printk("lvi_frag = %i, frags = %i, period_size = 0x%x, period_size1 = 0x%x\n",
-			ichdev->lvi_frag, ichdev->frags, ichdev->fragsize, ichdev->fragsize1);
+	printk(KERN_DEBUG "lvi_frag = %i, frags = %i, period_size = 0x%x, "
+	       "period_size1 = 0x%x\n",
+	       ichdev->lvi_frag, ichdev->frags, ichdev->fragsize,
+	       ichdev->fragsize1);
 #endif
 	/* clear interrupts */
 	iputbyte(chip, port + ichdev->roff_sr, ICH_FIFOE | ICH_BCIS | ICH_LVBCI);
@@ -465,7 +470,8 @@ static inline void snd_intel8x0_update(struct intel8x0m *chip, struct ichdev *ic
 							     ichdev->lvi_frag *
 							     ichdev->fragsize1);
 #if 0
-		printk("new: bdbar[%i] = 0x%x [0x%x], prefetch = %i, all = 0x%x, 0x%x\n",
+		printk(KERN_DEBUG "new: bdbar[%i] = 0x%x [0x%x], "
+		       "prefetch = %i, all = 0x%x, 0x%x\n",
 		       ichdev->lvi * 2, ichdev->bdbar[ichdev->lvi * 2],
 		       ichdev->bdbar[ichdev->lvi * 2 + 1], inb(ICH_REG_OFF_PIV + port),
 		       inl(port + 4), inb(port + ICH_REG_OFF_CR));
-- 
cgit v1.2.3


From ee419653a38de93b75a577851d9e4003cf0bbe07 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 5 Feb 2009 16:11:31 +0100
Subject: ALSA: Fix missing KERN_* prefix to printk in sound/pci

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ac97/ac97_codec.c         |   5 +-
 sound/pci/ak4531_codec.c            |   3 +-
 sound/pci/als300.c                  |   2 +-
 sound/pci/au88x0/au88x0_a3d.c       |   7 ++-
 sound/pci/au88x0/au88x0_core.c      |  19 +++++--
 sound/pci/au88x0/au88x0_synth.c     |  39 ++++++++++---
 sound/pci/azt3328.c                 |   8 +--
 sound/pci/ca0106/ca0106_main.c      |  91 +++++++++++++++++++++++-------
 sound/pci/cs4281.c                  |   6 +-
 sound/pci/cs46xx/cs46xx_lib.c       |   6 +-
 sound/pci/cs46xx/cs46xx_lib.h       |   6 +-
 sound/pci/cs5535audio/cs5535audio.c |   2 +-
 sound/pci/ens1370.c                 |   3 +-
 sound/pci/es1938.c                  |  23 +++++---
 sound/pci/mixart/mixart_hwdep.c     |  46 ++++++++-------
 sound/pci/sonicvibes.c              | 109 ++++++++++++++++++++++++------------
 sound/pci/trident/trident_main.c    |  57 ++++++++++---------
 sound/pci/via82xx.c                 |   5 +-
 sound/pci/via82xx_modem.c           |   5 +-
 sound/pci/vx222/vx222_ops.c         |   8 ++-
 sound/pci/ymfpci/ymfpci_main.c      |  14 ++++-
 21 files changed, 318 insertions(+), 146 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index e2b843b4f9d..bc707b60385 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -1643,7 +1643,10 @@ static int snd_ac97_modem_build(struct snd_card *card, struct snd_ac97 * ac97)
 {
 	int err, idx;
 
-	//printk("AC97_GPIO_CFG = %x\n",snd_ac97_read(ac97,AC97_GPIO_CFG));
+	/*
+	printk(KERN_DEBUG "AC97_GPIO_CFG = %x\n",
+	       snd_ac97_read(ac97,AC97_GPIO_CFG));
+	*/
 	snd_ac97_write(ac97, AC97_GPIO_CFG, 0xffff & ~(AC97_GPIO_LINE1_OH));
 	snd_ac97_write(ac97, AC97_GPIO_POLARITY, 0xffff & ~(AC97_GPIO_LINE1_OH));
 	snd_ac97_write(ac97, AC97_GPIO_STICKY, 0xffff);
diff --git a/sound/pci/ak4531_codec.c b/sound/pci/ak4531_codec.c
index 0f819ddb3eb..fd135e3d8a8 100644
--- a/sound/pci/ak4531_codec.c
+++ b/sound/pci/ak4531_codec.c
@@ -51,7 +51,8 @@ static void snd_ak4531_dump(struct snd_ak4531 *ak4531)
 	int idx;
 	
 	for (idx = 0; idx < 0x19; idx++)
-		printk("ak4531 0x%x: 0x%x\n", idx, ak4531->regs[idx]);
+		printk(KERN_DEBUG "ak4531 0x%x: 0x%x\n",
+		       idx, ak4531->regs[idx]);
 }
 
 #endif
diff --git a/sound/pci/als300.c b/sound/pci/als300.c
index 8df6824b51c..a2c35c1081c 100644
--- a/sound/pci/als300.c
+++ b/sound/pci/als300.c
@@ -91,7 +91,7 @@
 #define DEBUG_PLAY_REC	0
 
 #if DEBUG_CALLS
-#define snd_als300_dbgcalls(format, args...) printk(format, ##args)
+#define snd_als300_dbgcalls(format, args...) printk(KERN_DEBUG format, ##args)
 #define snd_als300_dbgcallenter() printk(KERN_ERR "--> %s\n", __func__)
 #define snd_als300_dbgcallleave() printk(KERN_ERR "<-- %s\n", __func__)
 #else
diff --git a/sound/pci/au88x0/au88x0_a3d.c b/sound/pci/au88x0/au88x0_a3d.c
index 649849e540d..f4aa8ff6f5f 100644
--- a/sound/pci/au88x0/au88x0_a3d.c
+++ b/sound/pci/au88x0/au88x0_a3d.c
@@ -462,9 +462,10 @@ static void a3dsrc_ZeroSliceIO(a3dsrc_t * a)
 /* Reset Single A3D source. */
 static void a3dsrc_ZeroState(a3dsrc_t * a)
 {
-
-	//printk("vortex: ZeroState slice: %d, source %d\n", a->slice, a->source);
-
+	/*
+	printk(KERN_DEBUG "vortex: ZeroState slice: %d, source %d\n",
+	       a->slice, a->source);
+	*/
 	a3dsrc_SetAtmosState(a, 0, 0, 0, 0);
 	a3dsrc_SetHrtfState(a, A3dHrirZeros, A3dHrirZeros);
 	a3dsrc_SetItdDline(a, A3dItdDlineZeros);
diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c
index b070e571451..e6a04d037c1 100644
--- a/sound/pci/au88x0/au88x0_core.c
+++ b/sound/pci/au88x0/au88x0_core.c
@@ -1135,7 +1135,10 @@ vortex_adbdma_setbuffers(vortex_t * vortex, int adbdma,
 			snd_pcm_sgbuf_get_addr(dma->substream, 0));
 		break;
 	}
-	//printk("vortex: cfg0 = 0x%x\nvortex: cfg1=0x%x\n", dma->cfg0, dma->cfg1);
+	/*
+	printk(KERN_DEBUG "vortex: cfg0 = 0x%x\nvortex: cfg1=0x%x\n",
+	       dma->cfg0, dma->cfg1);
+	*/
 	hwwrite(vortex->mmio, VORTEX_ADBDMA_BUFCFG0 + (adbdma << 3), dma->cfg0);
 	hwwrite(vortex->mmio, VORTEX_ADBDMA_BUFCFG1 + (adbdma << 3), dma->cfg1);
 
@@ -1959,7 +1962,7 @@ vortex_connect_codecplay(vortex_t * vortex, int en, unsigned char mixers[])
 					  ADB_CODECOUT(0 + 4));
 		vortex_connection_mix_adb(vortex, en, 0x11, mixers[3],
 					  ADB_CODECOUT(1 + 4));
-		//printk("SDAC detected ");
+		/* printk(KERN_DEBUG "SDAC detected "); */
 	}
 #else
 	// Use plain direct output to codec.
@@ -2013,7 +2016,11 @@ vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, int restype)
 					resmap[restype] |= (1 << i);
 				else
 					vortex->dma_adb[i].resources[restype] |= (1 << i);
-				//printk("vortex: ResManager: type %d out %d\n", restype, i);
+				/*
+				printk(KERN_DEBUG
+				       "vortex: ResManager: type %d out %d\n",
+				       restype, i);
+				*/
 				return i;
 			}
 		}
@@ -2024,7 +2031,11 @@ vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, int restype)
 		for (i = 0; i < qty; i++) {
 			if (resmap[restype] & (1 << i)) {
 				resmap[restype] &= ~(1 << i);
-				//printk("vortex: ResManager: type %d in %d\n",restype, i);
+				/*
+				printk(KERN_DEBUG
+				       "vortex: ResManager: type %d in %d\n",
+				       restype, i);
+				*/
 				return i;
 			}
 		}
diff --git a/sound/pci/au88x0/au88x0_synth.c b/sound/pci/au88x0/au88x0_synth.c
index 978b856f562..2805e34bd41 100644
--- a/sound/pci/au88x0/au88x0_synth.c
+++ b/sound/pci/au88x0/au88x0_synth.c
@@ -213,38 +213,59 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt,
 	switch (reg) {
 		/* Voice specific parameters */
 	case 0:		/* running */
-		//printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_RUN(wt), (int)val);
+		/*
+		printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n",
+		       WT_RUN(wt), (int)val);
+		*/
 		hwwrite(vortex->mmio, WT_RUN(wt), val);
 		return 0xc;
 		break;
 	case 1:		/* param 0 */
-		//printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,0), (int)val);
+		/*
+		printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n",
+		       WT_PARM(wt,0), (int)val);
+		*/
 		hwwrite(vortex->mmio, WT_PARM(wt, 0), val);
 		return 0xc;
 		break;
 	case 2:		/* param 1 */
-		//printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,1), (int)val);
+		/*
+		printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n",
+		       WT_PARM(wt,1), (int)val);
+		*/
 		hwwrite(vortex->mmio, WT_PARM(wt, 1), val);
 		return 0xc;
 		break;
 	case 3:		/* param 2 */
-		//printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,2), (int)val);
+		/*
+		printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n",
+		       WT_PARM(wt,2), (int)val);
+		*/
 		hwwrite(vortex->mmio, WT_PARM(wt, 2), val);
 		return 0xc;
 		break;
 	case 4:		/* param 3 */
-		//printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,3), (int)val);
+		/*
+		printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n",
+		       WT_PARM(wt,3), (int)val);
+		*/
 		hwwrite(vortex->mmio, WT_PARM(wt, 3), val);
 		return 0xc;
 		break;
 	case 6:		/* mute */
-		//printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_MUTE(wt), (int)val);
+		/*
+		printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n",
+		       WT_MUTE(wt), (int)val);
+		*/
 		hwwrite(vortex->mmio, WT_MUTE(wt), val);
 		return 0xc;
 		break;
 	case 0xb:
 		{		/* delay */
-			//printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_DELAY(wt,0), (int)val);
+			/*
+			printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n",
+			       WT_DELAY(wt,0), (int)val);
+			*/
 			hwwrite(vortex->mmio, WT_DELAY(wt, 3), val);
 			hwwrite(vortex->mmio, WT_DELAY(wt, 2), val);
 			hwwrite(vortex->mmio, WT_DELAY(wt, 1), val);
@@ -272,7 +293,9 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt,
 		return 0;
 		break;
 	}
-	//printk("vortex: WT SetReg(0x%x) = 0x%08x\n", ecx, (int)val);
+	/*
+	printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", ecx, (int)val);
+	*/
 	hwwrite(vortex->mmio, ecx, val);
 	return 1;
 }
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 333007c523a..8121763b0c1 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -211,25 +211,25 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}");
 #endif
 
 #if DEBUG_MIXER
-#define snd_azf3328_dbgmixer(format, args...) printk(format, ##args)
+#define snd_azf3328_dbgmixer(format, args...) printk(KERN_DEBUG format, ##args)
 #else
 #define snd_azf3328_dbgmixer(format, args...)
 #endif
 
 #if DEBUG_PLAY_REC
-#define snd_azf3328_dbgplay(format, args...) printk(KERN_ERR format, ##args)
+#define snd_azf3328_dbgplay(format, args...) printk(KERN_DEBUG format, ##args)
 #else
 #define snd_azf3328_dbgplay(format, args...)
 #endif
 
 #if DEBUG_MISC
-#define snd_azf3328_dbgtimer(format, args...) printk(KERN_ERR format, ##args)
+#define snd_azf3328_dbgtimer(format, args...) printk(KERN_DEBUG format, ##args)
 #else
 #define snd_azf3328_dbgtimer(format, args...)
 #endif
 
 #if DEBUG_GAME
-#define snd_azf3328_dbggame(format, args...) printk(KERN_ERR format, ##args)
+#define snd_azf3328_dbggame(format, args...) printk(KERN_DEBUG format, ##args)
 #else
 #define snd_azf3328_dbggame(format, args...)
 #endif
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index 0e62205d408..f2f8fd17ea4 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -404,7 +404,9 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu,
 	}
 
 	tmp = reg << 25 | value << 16;
-	// snd_printk("I2C-write:reg=0x%x, value=0x%x\n", reg, value);
+	/*
+	snd_printk(KERN_DEBUG "I2C-write:reg=0x%x, value=0x%x\n", reg, value);
+	*/
 	/* Not sure what this I2C channel controls. */
 	/* snd_ca0106_ptr_write(emu, I2C_D0, 0, tmp); */
 
@@ -422,7 +424,7 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu,
 		/* Wait till the transaction ends */
 		while (1) {
 			status = snd_ca0106_ptr_read(emu, I2C_A, 0);
-                	//snd_printk("I2C:status=0x%x\n", status);
+			/*snd_printk(KERN_DEBUG "I2C:status=0x%x\n", status);*/
 			timeout++;
 			if ((status & I2C_A_ADC_START) == 0)
 				break;
@@ -521,7 +523,10 @@ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substr
         channel->number = channel_id;
 
 	channel->use = 1;
-        //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel);
+	/*
+	printk(KERN_DEBUG "open:channel_id=%d, chip=%p, channel=%p\n",
+	       channel_id, chip, channel);
+	*/
         //channel->interrupt = snd_ca0106_pcm_channel_interrupt;
 	channel->epcm = epcm;
 	if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
@@ -614,7 +619,10 @@ static int snd_ca0106_pcm_open_capture_channel(struct snd_pcm_substream *substre
         channel->number = channel_id;
 
 	channel->use = 1;
-        //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel);
+	/*
+        printk(KERN_DEBUG "open:channel_id=%d, chip=%p, channel=%p\n",
+	       channel_id, chip, channel);
+	*/
         //channel->interrupt = snd_ca0106_pcm_channel_interrupt;
         channel->epcm = epcm;
 	if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
@@ -705,9 +713,20 @@ static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream)
 	u32 reg71;
 	int i;
 	
-        //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1));
-        //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base);
-	//snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes);
+#if 0 /* debug */
+	snd_printk(KERN_DEBUG
+		   "prepare:channel_number=%d, rate=%d, format=0x%x, "
+		   "channels=%d, buffer_size=%ld, period_size=%ld, "
+		   "periods=%u, frames_to_bytes=%d\n",
+		   channel, runtime->rate, runtime->format,
+		   runtime->channels, runtime->buffer_size,
+		   runtime->period_size, runtime->periods,
+		   frames_to_bytes(runtime, 1));
+	snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, table_base=%p\n",
+		   runtime->dma_addr, runtime->dma_area, table_base);
+	snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",
+		   emu->buffer.addr, emu->buffer.area, emu->buffer.bytes);
+#endif /* debug */
 	/* Rate can be set per channel. */
 	/* reg40 control host to fifo */
 	/* reg71 controls DAC rate. */
@@ -799,9 +818,20 @@ static int snd_ca0106_pcm_prepare_capture(struct snd_pcm_substream *substream)
 	u32 reg71_set = 0;
 	u32 reg71;
 	
-        //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1));
-        //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base);
-	//snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes);
+#if 0 /* debug */
+	snd_printk(KERN_DEBUG
+		   "prepare:channel_number=%d, rate=%d, format=0x%x, "
+		   "channels=%d, buffer_size=%ld, period_size=%ld, "
+		   "periods=%u, frames_to_bytes=%d\n",
+		   channel, runtime->rate, runtime->format,
+		   runtime->channels, runtime->buffer_size,
+		   runtime->period_size, runtime->periods,
+		   frames_to_bytes(runtime, 1));
+        snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, table_base=%p\n",
+		   runtime->dma_addr, runtime->dma_area, table_base);
+	snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",
+		   emu->buffer.addr, emu->buffer.area, emu->buffer.bytes);
+#endif /* debug */
 	/* reg71 controls ADC rate. */
 	switch (runtime->rate) {
 	case 44100:
@@ -846,7 +876,14 @@ static int snd_ca0106_pcm_prepare_capture(struct snd_pcm_substream *substream)
 	}
 
 
-        //printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size,  frames_to_bytes(runtime, 1));
+	/*
+	printk(KERN_DEBUG
+	       "prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, "
+	       "buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",
+	       channel, runtime->rate, runtime->format, runtime->channels,
+	       runtime->buffer_size, runtime->period_size,
+	       frames_to_bytes(runtime, 1));
+	*/
 	snd_ca0106_ptr_write(emu, 0x13, channel, 0);
 	snd_ca0106_ptr_write(emu, CAPTURE_DMA_ADDR, channel, runtime->dma_addr);
 	snd_ca0106_ptr_write(emu, CAPTURE_BUFFER_SIZE, channel, frames_to_bytes(runtime, runtime->buffer_size)<<16); // buffer size in bytes
@@ -888,13 +925,13 @@ static int snd_ca0106_pcm_trigger_playback(struct snd_pcm_substream *substream,
 		runtime = s->runtime;
 		epcm = runtime->private_data;
 		channel = epcm->channel_id;
-		/* snd_printk("channel=%d\n",channel); */
+		/* snd_printk(KERN_DEBUG "channel=%d\n", channel); */
 		epcm->running = running;
 		basic |= (0x1 << channel);
 		extended |= (0x10 << channel);
                 snd_pcm_trigger_done(s, substream);
         }
-	/* snd_printk("basic=0x%x, extended=0x%x\n",basic, extended); */
+	/* snd_printk(KERN_DEBUG "basic=0x%x, extended=0x%x\n",basic, extended); */
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
@@ -972,8 +1009,13 @@ snd_ca0106_pcm_pointer_playback(struct snd_pcm_substream *substream)
 	ptr=ptr2;
         if (ptr >= runtime->buffer_size)
 		ptr -= runtime->buffer_size;
-	//printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate);
-
+	/*
+	printk(KERN_DEBUG "ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, "
+	       "buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n",
+	       ptr1, ptr2, ptr, (int)runtime->buffer_size,
+	       (int)runtime->period_size, (int)runtime->frame_bits,
+	       (int)runtime->rate);
+	*/
 	return ptr;
 }
 
@@ -995,8 +1037,13 @@ snd_ca0106_pcm_pointer_capture(struct snd_pcm_substream *substream)
 	ptr=ptr2;
         if (ptr >= runtime->buffer_size)
 		ptr -= runtime->buffer_size;
-	//printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate);
-
+	/*
+	printk(KERN_DEBUG "ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, "
+	       "buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n",
+	       ptr1, ptr2, ptr, (int)runtime->buffer_size,
+	       (int)runtime->period_size, (int)runtime->frame_bits,
+	       (int)runtime->rate);
+	*/
 	return ptr;
 }
 
@@ -1181,8 +1228,12 @@ static irqreturn_t snd_ca0106_interrupt(int irq, void *dev_id)
 		return IRQ_NONE;
 
         stat76 = snd_ca0106_ptr_read(chip, EXTENDED_INT, 0);
-	//snd_printk("interrupt status = 0x%08x, stat76=0x%08x\n", status, stat76);
-	//snd_printk("ptr=0x%08x\n",snd_ca0106_ptr_read(chip, PLAYBACK_POINTER, 0));
+	/*
+	snd_printk(KERN_DEBUG "interrupt status = 0x%08x, stat76=0x%08x\n",
+		   status, stat76);
+	snd_printk(KERN_DEBUG "ptr=0x%08x\n",
+		   snd_ca0106_ptr_read(chip, PLAYBACK_POINTER, 0));
+	*/
         mask = 0x11; /* 0x1 for one half, 0x10 for the other half period. */
 	for(i = 0; i < 4; i++) {
 		pchannel = &(chip->playback_channels[i]);
@@ -1470,7 +1521,7 @@ static void ca0106_init_chip(struct snd_ca0106 *chip, int resume)
 		int size, n;
 
 		size = ARRAY_SIZE(i2c_adc_init);
-		/* snd_printk("I2C:array size=0x%x\n", size); */
+		/* snd_printk(KERN_DEBUG "I2C:array size=0x%x\n", size); */
 		for (n = 0; n < size; n++)
 			snd_ca0106_i2c_write(chip, i2c_adc_init[n][0],
 					     i2c_adc_init[n][1]);
diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c
index 192e7842e18..415e88f2c62 100644
--- a/sound/pci/cs4281.c
+++ b/sound/pci/cs4281.c
@@ -834,7 +834,11 @@ static snd_pcm_uframes_t snd_cs4281_pointer(struct snd_pcm_substream *substream)
 	struct cs4281_dma *dma = runtime->private_data;
 	struct cs4281 *chip = snd_pcm_substream_chip(substream);
 
-	// printk("DCC = 0x%x, buffer_size = 0x%x, jiffies = %li\n", snd_cs4281_peekBA0(chip, dma->regDCC), runtime->buffer_size, jiffies);
+	/*
+	printk(KERN_DEBUG "DCC = 0x%x, buffer_size = 0x%x, jiffies = %li\n",
+	       snd_cs4281_peekBA0(chip, dma->regDCC), runtime->buffer_size,
+	       jiffies);
+	*/
 	return runtime->buffer_size -
 	       snd_cs4281_peekBA0(chip, dma->regDCC) - 1;
 }
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index 8ab07aa6365..1be96ead424 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -194,7 +194,7 @@ static unsigned short snd_cs46xx_codec_read(struct snd_cs46xx *chip,
 	 *  ACSDA = Status Data Register = 474h
 	 */
 #if 0
-	printk("e) reg = 0x%x, val = 0x%x, BA0_ACCAD = 0x%x\n", reg,
+	printk(KERN_DEBUG "e) reg = 0x%x, val = 0x%x, BA0_ACCAD = 0x%x\n", reg,
 			snd_cs46xx_peekBA0(chip, BA0_ACSDA),
 			snd_cs46xx_peekBA0(chip, BA0_ACCAD));
 #endif
@@ -428,8 +428,8 @@ static int cs46xx_wait_for_fifo(struct snd_cs46xx * chip,int retry_timeout)
 	}
   
 	if(status & SERBST_WBSY) {
-		snd_printk( KERN_ERR "cs46xx: failure waiting for FIFO command to complete\n");
-
+		snd_printk(KERN_ERR "cs46xx: failure waiting for "
+			   "FIFO command to complete\n");
 		return -EINVAL;
 	}
 
diff --git a/sound/pci/cs46xx/cs46xx_lib.h b/sound/pci/cs46xx/cs46xx_lib.h
index 018a7de5601..4eb55aa3361 100644
--- a/sound/pci/cs46xx/cs46xx_lib.h
+++ b/sound/pci/cs46xx/cs46xx_lib.h
@@ -62,7 +62,11 @@ static inline void snd_cs46xx_poke(struct snd_cs46xx *chip, unsigned long reg, u
 	unsigned int bank = reg >> 16;
 	unsigned int offset = reg & 0xffff;
 
-	/*if (bank == 0) printk("snd_cs46xx_poke: %04X - %08X\n",reg >> 2,val); */
+	/*
+	if (bank == 0)
+		printk(KERN_DEBUG "snd_cs46xx_poke: %04X - %08X\n",
+		       reg >> 2,val);
+	*/
 	writel(val, chip->region.idx[bank+1].remap_addr + offset);
 }
 
diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c
index 826e6dec2e9..6506201d56f 100644
--- a/sound/pci/cs5535audio/cs5535audio.c
+++ b/sound/pci/cs5535audio/cs5535audio.c
@@ -312,7 +312,7 @@ static int __devinit snd_cs5535audio_create(struct snd_card *card,
 
 	if (request_irq(pci->irq, snd_cs5535audio_interrupt,
 			IRQF_SHARED, "CS5535 Audio", cs5535au)) {
-		snd_printk("unable to grab IRQ %d\n", pci->irq);
+		snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
 		err = -EBUSY;
 		goto sndfail;
 	}
diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c
index 9bf95367c88..17674b3406b 100644
--- a/sound/pci/ens1370.c
+++ b/sound/pci/ens1370.c
@@ -584,7 +584,8 @@ static void snd_es1370_codec_write(struct snd_ak4531 *ak4531,
 	unsigned long end_time = jiffies + HZ / 10;
 
 #if 0
-	printk("CODEC WRITE: reg = 0x%x, val = 0x%x (0x%x), creg = 0x%x\n",
+	printk(KERN_DEBUG
+	       "CODEC WRITE: reg = 0x%x, val = 0x%x (0x%x), creg = 0x%x\n",
 	       reg, val, ES_1370_CODEC_WRITE(reg, val), ES_REG(ensoniq, 1370_CODEC));
 #endif
 	do {
diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c
index 4cd9a1faaec..e4ba84bed4a 100644
--- a/sound/pci/es1938.c
+++ b/sound/pci/es1938.c
@@ -1673,18 +1673,22 @@ static irqreturn_t snd_es1938_interrupt(int irq, void *dev_id)
 
 	status = inb(SLIO_REG(chip, IRQCONTROL));
 #if 0
-	printk("Es1938debug - interrupt status: =0x%x\n", status);
+	printk(KERN_DEBUG "Es1938debug - interrupt status: =0x%x\n", status);
 #endif
 	
 	/* AUDIO 1 */
 	if (status & 0x10) {
 #if 0
-                printk("Es1938debug - AUDIO channel 1 interrupt\n");
-		printk("Es1938debug - AUDIO channel 1 DMAC DMA count: %u\n",
+                printk(KERN_DEBUG
+		       "Es1938debug - AUDIO channel 1 interrupt\n");
+		printk(KERN_DEBUG
+		       "Es1938debug - AUDIO channel 1 DMAC DMA count: %u\n",
 		       inw(SLDM_REG(chip, DMACOUNT)));
-		printk("Es1938debug - AUDIO channel 1 DMAC DMA base: %u\n",
+		printk(KERN_DEBUG
+		       "Es1938debug - AUDIO channel 1 DMAC DMA base: %u\n",
 		       inl(SLDM_REG(chip, DMAADDR)));
-		printk("Es1938debug - AUDIO channel 1 DMAC DMA status: 0x%x\n",
+		printk(KERN_DEBUG
+		       "Es1938debug - AUDIO channel 1 DMAC DMA status: 0x%x\n",
 		       inl(SLDM_REG(chip, DMASTATUS)));
 #endif
 		/* clear irq */
@@ -1699,10 +1703,13 @@ static irqreturn_t snd_es1938_interrupt(int irq, void *dev_id)
 	/* AUDIO 2 */
 	if (status & 0x20) {
 #if 0
-                printk("Es1938debug - AUDIO channel 2 interrupt\n");
-		printk("Es1938debug - AUDIO channel 2 DMAC DMA count: %u\n",
+                printk(KERN_DEBUG
+		       "Es1938debug - AUDIO channel 2 interrupt\n");
+		printk(KERN_DEBUG
+		       "Es1938debug - AUDIO channel 2 DMAC DMA count: %u\n",
 		       inw(SLIO_REG(chip, AUDIO2DMACOUNT)));
-		printk("Es1938debug - AUDIO channel 2 DMAC DMA base: %u\n",
+		printk(KERN_DEBUG
+		       "Es1938debug - AUDIO channel 2 DMAC DMA base: %u\n",
 		       inl(SLIO_REG(chip, AUDIO2DMAADDR)));
 
 #endif
diff --git a/sound/pci/mixart/mixart_hwdep.c b/sound/pci/mixart/mixart_hwdep.c
index 3782b52bc0e..dda562081d7 100644
--- a/sound/pci/mixart/mixart_hwdep.c
+++ b/sound/pci/mixart/mixart_hwdep.c
@@ -345,8 +345,8 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw
 	status_daught = readl_be( MIXART_MEM( mgr,MIXART_PSEUDOREG_DXLX_STATUS_OFFSET ));
 
 	/* motherboard xilinx status 5 will say that the board is performing a reset */
-	if( status_xilinx == 5 ) {
-		snd_printk( KERN_ERR "miXart is resetting !\n");
+	if (status_xilinx == 5) {
+		snd_printk(KERN_ERR "miXart is resetting !\n");
 		return -EAGAIN; /* try again later */
 	}
 
@@ -354,13 +354,14 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw
 	case MIXART_MOTHERBOARD_XLX_INDEX:
 
 		/* xilinx already loaded ? */ 
-		if( status_xilinx == 4 ) {
-			snd_printk( KERN_DEBUG "xilinx is already loaded !\n");
+		if (status_xilinx == 4) {
+			snd_printk(KERN_DEBUG "xilinx is already loaded !\n");
 			return 0;
 		}
 		/* the status should be 0 == "idle" */
-		if( status_xilinx != 0 ) {
-			snd_printk( KERN_ERR "xilinx load error ! status = %d\n", status_xilinx);
+		if (status_xilinx != 0) {
+			snd_printk(KERN_ERR "xilinx load error ! status = %d\n",
+				   status_xilinx);
 			return -EIO; /* modprob -r may help ? */
 		}
 
@@ -389,21 +390,23 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw
 
 	case MIXART_MOTHERBOARD_ELF_INDEX:
 
-		if( status_elf == 4 ) {
-			snd_printk( KERN_DEBUG "elf file already loaded !\n");
+		if (status_elf == 4) {
+			snd_printk(KERN_DEBUG "elf file already loaded !\n");
 			return 0;
 		}
 
 		/* the status should be 0 == "idle" */
-		if( status_elf != 0 ) {
-			snd_printk( KERN_ERR "elf load error ! status = %d\n", status_elf);
+		if (status_elf != 0) {
+			snd_printk(KERN_ERR "elf load error ! status = %d\n",
+				   status_elf);
 			return -EIO; /* modprob -r may help ? */
 		}
 
 		/* wait for xilinx status == 4 */
 		err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_MXLX_STATUS_OFFSET, 1, 4, 500); /* 5sec */
 		if (err < 0) {
-			snd_printk( KERN_ERR "xilinx was not loaded or could not be started\n");
+			snd_printk(KERN_ERR "xilinx was not loaded or "
+				   "could not be started\n");
 			return err;
 		}
 
@@ -424,7 +427,7 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw
 		/* wait for elf status == 4 */
 		err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_ELF_STATUS_OFFSET, 1, 4, 300); /* 3sec */
 		if (err < 0) {
-			snd_printk( KERN_ERR "elf could not be started\n");
+			snd_printk(KERN_ERR "elf could not be started\n");
 			return err;
 		}
 
@@ -437,15 +440,16 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw
 	default:
 
 		/* elf and xilinx should be loaded */
-		if( (status_elf != 4) || (status_xilinx != 4) ) {
-			printk( KERN_ERR "xilinx or elf not successfully loaded\n");
+		if (status_elf != 4 || status_xilinx != 4) {
+			printk(KERN_ERR "xilinx or elf not "
+			       "successfully loaded\n");
 			return -EIO; /* modprob -r may help ? */
 		}
 
 		/* wait for daughter detection != 0 */
 		err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_DBRD_PRESENCE_OFFSET, 0, 0, 30); /* 300msec */
 		if (err < 0) {
-			snd_printk( KERN_ERR "error starting elf file\n");
+			snd_printk(KERN_ERR "error starting elf file\n");
 			return err;
 		}
 
@@ -460,8 +464,9 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw
 			return -EINVAL;
 
 		/* daughter should be idle */
-		if( status_daught != 0 ) {
-			printk( KERN_ERR "daughter load error ! status = %d\n", status_daught);
+		if (status_daught != 0) {
+			printk(KERN_ERR "daughter load error ! status = %d\n",
+			       status_daught);
 			return -EIO; /* modprob -r may help ? */
 		}
  
@@ -480,7 +485,7 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw
 		/* wait for status == 2 */
 		err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_DXLX_STATUS_OFFSET, 1, 2, 30); /* 300msec */
 		if (err < 0) {
-			snd_printk( KERN_ERR "daughter board load error\n");
+			snd_printk(KERN_ERR "daughter board load error\n");
 			return err;
 		}
 
@@ -502,7 +507,8 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw
         /* wait for daughter status == 3 */
         err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_DXLX_STATUS_OFFSET, 1, 3, 300); /* 3sec */
         if (err < 0) {
-		snd_printk( KERN_ERR "daughter board could not be initialised\n");
+		snd_printk(KERN_ERR
+			   "daughter board could not be initialised\n");
 		return err;
 	}
 
@@ -512,7 +518,7 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw
 	/* first communication with embedded */
 	err = mixart_first_init(mgr);
         if (err < 0) {
-		snd_printk( KERN_ERR "miXart could not be set up\n");
+		snd_printk(KERN_ERR "miXart could not be set up\n");
 		return err;
 	}
 
diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c
index cd408b86c83..e922b1887b1 100644
--- a/sound/pci/sonicvibes.c
+++ b/sound/pci/sonicvibes.c
@@ -273,7 +273,8 @@ static inline void snd_sonicvibes_setdmaa(struct sonicvibes * sonic,
 	outl(count, sonic->dmaa_port + SV_DMA_COUNT0);
 	outb(0x18, sonic->dmaa_port + SV_DMA_MODE);
 #if 0
-	printk("program dmaa: addr = 0x%x, paddr = 0x%x\n", addr, inl(sonic->dmaa_port + SV_DMA_ADDR0));
+	printk(KERN_DEBUG "program dmaa: addr = 0x%x, paddr = 0x%x\n",
+	       addr, inl(sonic->dmaa_port + SV_DMA_ADDR0));
 #endif
 }
 
@@ -288,7 +289,8 @@ static inline void snd_sonicvibes_setdmac(struct sonicvibes * sonic,
 	outl(count, sonic->dmac_port + SV_DMA_COUNT0);
 	outb(0x14, sonic->dmac_port + SV_DMA_MODE);
 #if 0
-	printk("program dmac: addr = 0x%x, paddr = 0x%x\n", addr, inl(sonic->dmac_port + SV_DMA_ADDR0));
+	printk(KERN_DEBUG "program dmac: addr = 0x%x, paddr = 0x%x\n",
+	       addr, inl(sonic->dmac_port + SV_DMA_ADDR0));
 #endif
 }
 
@@ -355,71 +357,104 @@ static unsigned char snd_sonicvibes_in(struct sonicvibes * sonic, unsigned char
 #if 0
 static void snd_sonicvibes_debug(struct sonicvibes * sonic)
 {
-	printk("SV REGS:          INDEX = 0x%02x  ", inb(SV_REG(sonic, INDEX)));
+	printk(KERN_DEBUG
+	       "SV REGS:          INDEX = 0x%02x  ", inb(SV_REG(sonic, INDEX)));
 	printk("                 STATUS = 0x%02x\n", inb(SV_REG(sonic, STATUS)));
-	printk("  0x00: left input      = 0x%02x  ", snd_sonicvibes_in(sonic, 0x00));
+	printk(KERN_DEBUG
+	       "  0x00: left input      = 0x%02x  ", snd_sonicvibes_in(sonic, 0x00));
 	printk("  0x20: synth rate low  = 0x%02x\n", snd_sonicvibes_in(sonic, 0x20));
-	printk("  0x01: right input     = 0x%02x  ", snd_sonicvibes_in(sonic, 0x01));
+	printk(KERN_DEBUG
+	       "  0x01: right input     = 0x%02x  ", snd_sonicvibes_in(sonic, 0x01));
 	printk("  0x21: synth rate high = 0x%02x\n", snd_sonicvibes_in(sonic, 0x21));
-	printk("  0x02: left AUX1       = 0x%02x  ", snd_sonicvibes_in(sonic, 0x02));
+	printk(KERN_DEBUG
+	       "  0x02: left AUX1       = 0x%02x  ", snd_sonicvibes_in(sonic, 0x02));
 	printk("  0x22: ADC clock       = 0x%02x\n", snd_sonicvibes_in(sonic, 0x22));
-	printk("  0x03: right AUX1      = 0x%02x  ", snd_sonicvibes_in(sonic, 0x03));
+	printk(KERN_DEBUG
+	       "  0x03: right AUX1      = 0x%02x  ", snd_sonicvibes_in(sonic, 0x03));
 	printk("  0x23: ADC alt rate    = 0x%02x\n", snd_sonicvibes_in(sonic, 0x23));
-	printk("  0x04: left CD         = 0x%02x  ", snd_sonicvibes_in(sonic, 0x04));
+	printk(KERN_DEBUG
+	       "  0x04: left CD         = 0x%02x  ", snd_sonicvibes_in(sonic, 0x04));
 	printk("  0x24: ADC pll M       = 0x%02x\n", snd_sonicvibes_in(sonic, 0x24));
-	printk("  0x05: right CD        = 0x%02x  ", snd_sonicvibes_in(sonic, 0x05));
+	printk(KERN_DEBUG
+	       "  0x05: right CD        = 0x%02x  ", snd_sonicvibes_in(sonic, 0x05));
 	printk("  0x25: ADC pll N       = 0x%02x\n", snd_sonicvibes_in(sonic, 0x25));
-	printk("  0x06: left line       = 0x%02x  ", snd_sonicvibes_in(sonic, 0x06));
+	printk(KERN_DEBUG
+	       "  0x06: left line       = 0x%02x  ", snd_sonicvibes_in(sonic, 0x06));
 	printk("  0x26: Synth pll M     = 0x%02x\n", snd_sonicvibes_in(sonic, 0x26));
-	printk("  0x07: right line      = 0x%02x  ", snd_sonicvibes_in(sonic, 0x07));
+	printk(KERN_DEBUG
+	       "  0x07: right line      = 0x%02x  ", snd_sonicvibes_in(sonic, 0x07));
 	printk("  0x27: Synth pll N     = 0x%02x\n", snd_sonicvibes_in(sonic, 0x27));
-	printk("  0x08: MIC             = 0x%02x  ", snd_sonicvibes_in(sonic, 0x08));
+	printk(KERN_DEBUG
+	       "  0x08: MIC             = 0x%02x  ", snd_sonicvibes_in(sonic, 0x08));
 	printk("  0x28: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x28));
-	printk("  0x09: Game port       = 0x%02x  ", snd_sonicvibes_in(sonic, 0x09));
+	printk(KERN_DEBUG
+	       "  0x09: Game port       = 0x%02x  ", snd_sonicvibes_in(sonic, 0x09));
 	printk("  0x29: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x29));
-	printk("  0x0a: left synth      = 0x%02x  ", snd_sonicvibes_in(sonic, 0x0a));
+	printk(KERN_DEBUG
+	       "  0x0a: left synth      = 0x%02x  ", snd_sonicvibes_in(sonic, 0x0a));
 	printk("  0x2a: MPU401          = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2a));
-	printk("  0x0b: right synth     = 0x%02x  ", snd_sonicvibes_in(sonic, 0x0b));
+	printk(KERN_DEBUG
+	       "  0x0b: right synth     = 0x%02x  ", snd_sonicvibes_in(sonic, 0x0b));
 	printk("  0x2b: drive ctrl      = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2b));
-	printk("  0x0c: left AUX2       = 0x%02x  ", snd_sonicvibes_in(sonic, 0x0c));
+	printk(KERN_DEBUG
+	       "  0x0c: left AUX2       = 0x%02x  ", snd_sonicvibes_in(sonic, 0x0c));
 	printk("  0x2c: SRS space       = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2c));
-	printk("  0x0d: right AUX2      = 0x%02x  ", snd_sonicvibes_in(sonic, 0x0d));
+	printk(KERN_DEBUG
+	       "  0x0d: right AUX2      = 0x%02x  ", snd_sonicvibes_in(sonic, 0x0d));
 	printk("  0x2d: SRS center      = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2d));
-	printk("  0x0e: left analog     = 0x%02x  ", snd_sonicvibes_in(sonic, 0x0e));
+	printk(KERN_DEBUG
+	       "  0x0e: left analog     = 0x%02x  ", snd_sonicvibes_in(sonic, 0x0e));
 	printk("  0x2e: wave source     = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2e));
-	printk("  0x0f: right analog    = 0x%02x  ", snd_sonicvibes_in(sonic, 0x0f));
+	printk(KERN_DEBUG
+	       "  0x0f: right analog    = 0x%02x  ", snd_sonicvibes_in(sonic, 0x0f));
 	printk("  0x2f: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2f));
-	printk("  0x10: left PCM        = 0x%02x  ", snd_sonicvibes_in(sonic, 0x10));
+	printk(KERN_DEBUG
+	       "  0x10: left PCM        = 0x%02x  ", snd_sonicvibes_in(sonic, 0x10));
 	printk("  0x30: analog power    = 0x%02x\n", snd_sonicvibes_in(sonic, 0x30));
-	printk("  0x11: right PCM       = 0x%02x  ", snd_sonicvibes_in(sonic, 0x11));
+	printk(KERN_DEBUG
+	       "  0x11: right PCM       = 0x%02x  ", snd_sonicvibes_in(sonic, 0x11));
 	printk("  0x31: analog power    = 0x%02x\n", snd_sonicvibes_in(sonic, 0x31));
-	printk("  0x12: DMA data format = 0x%02x  ", snd_sonicvibes_in(sonic, 0x12));
+	printk(KERN_DEBUG
+	       "  0x12: DMA data format = 0x%02x  ", snd_sonicvibes_in(sonic, 0x12));
 	printk("  0x32: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x32));
-	printk("  0x13: P/C enable      = 0x%02x  ", snd_sonicvibes_in(sonic, 0x13));
+	printk(KERN_DEBUG
+	       "  0x13: P/C enable      = 0x%02x  ", snd_sonicvibes_in(sonic, 0x13));
 	printk("  0x33: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x33));
-	printk("  0x14: U/D button      = 0x%02x  ", snd_sonicvibes_in(sonic, 0x14));
+	printk(KERN_DEBUG
+	       "  0x14: U/D button      = 0x%02x  ", snd_sonicvibes_in(sonic, 0x14));
 	printk("  0x34: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x34));
-	printk("  0x15: revision        = 0x%02x  ", snd_sonicvibes_in(sonic, 0x15));
+	printk(KERN_DEBUG
+	       "  0x15: revision        = 0x%02x  ", snd_sonicvibes_in(sonic, 0x15));
 	printk("  0x35: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x35));
-	printk("  0x16: ADC output ctrl = 0x%02x  ", snd_sonicvibes_in(sonic, 0x16));
+	printk(KERN_DEBUG
+	       "  0x16: ADC output ctrl = 0x%02x  ", snd_sonicvibes_in(sonic, 0x16));
 	printk("  0x36: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x36));
-	printk("  0x17: ---             = 0x%02x  ", snd_sonicvibes_in(sonic, 0x17));
+	printk(KERN_DEBUG
+	       "  0x17: ---             = 0x%02x  ", snd_sonicvibes_in(sonic, 0x17));
 	printk("  0x37: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x37));
-	printk("  0x18: DMA A upper cnt = 0x%02x  ", snd_sonicvibes_in(sonic, 0x18));
+	printk(KERN_DEBUG
+	       "  0x18: DMA A upper cnt = 0x%02x  ", snd_sonicvibes_in(sonic, 0x18));
 	printk("  0x38: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x38));
-	printk("  0x19: DMA A lower cnt = 0x%02x  ", snd_sonicvibes_in(sonic, 0x19));
+	printk(KERN_DEBUG
+	       "  0x19: DMA A lower cnt = 0x%02x  ", snd_sonicvibes_in(sonic, 0x19));
 	printk("  0x39: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x39));
-	printk("  0x1a: ---             = 0x%02x  ", snd_sonicvibes_in(sonic, 0x1a));
+	printk(KERN_DEBUG
+	       "  0x1a: ---             = 0x%02x  ", snd_sonicvibes_in(sonic, 0x1a));
 	printk("  0x3a: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3a));
-	printk("  0x1b: ---             = 0x%02x  ", snd_sonicvibes_in(sonic, 0x1b));
+	printk(KERN_DEBUG
+	       "  0x1b: ---             = 0x%02x  ", snd_sonicvibes_in(sonic, 0x1b));
 	printk("  0x3b: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3b));
-	printk("  0x1c: DMA C upper cnt = 0x%02x  ", snd_sonicvibes_in(sonic, 0x1c));
+	printk(KERN_DEBUG
+	       "  0x1c: DMA C upper cnt = 0x%02x  ", snd_sonicvibes_in(sonic, 0x1c));
 	printk("  0x3c: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3c));
-	printk("  0x1d: DMA C upper cnt = 0x%02x  ", snd_sonicvibes_in(sonic, 0x1d));
+	printk(KERN_DEBUG
+	       "  0x1d: DMA C upper cnt = 0x%02x  ", snd_sonicvibes_in(sonic, 0x1d));
 	printk("  0x3d: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3d));
-	printk("  0x1e: PCM rate low    = 0x%02x  ", snd_sonicvibes_in(sonic, 0x1e));
+	printk(KERN_DEBUG
+	       "  0x1e: PCM rate low    = 0x%02x  ", snd_sonicvibes_in(sonic, 0x1e));
 	printk("  0x3e: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3e));
-	printk("  0x1f: PCM rate high   = 0x%02x  ", snd_sonicvibes_in(sonic, 0x1f));
+	printk(KERN_DEBUG
+	       "  0x1f: PCM rate high   = 0x%02x  ", snd_sonicvibes_in(sonic, 0x1f));
 	printk("  0x3f: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3f));
 }
 
@@ -476,8 +511,8 @@ static void snd_sonicvibes_pll(unsigned int rate,
 	*res_m = m;
 	*res_n = n;
 #if 0
-	printk("metric = %i, xm = %i, xn = %i\n", metric, xm, xn);
-	printk("pll: m = 0x%x, r = 0x%x, n = 0x%x\n", reg, m, r, n);
+	printk(KERN_DEBUG "metric = %i, xm = %i, xn = %i\n", metric, xm, xn);
+	printk(KERN_DEBUG "pll: m = 0x%x, r = 0x%x, n = 0x%x\n", reg, m, r, n);
 #endif
 }
 
diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c
index c612b435ca2..a9da9c18466 100644
--- a/sound/pci/trident/trident_main.c
+++ b/sound/pci/trident/trident_main.c
@@ -68,40 +68,40 @@ static void snd_trident_print_voice_regs(struct snd_trident *trident, int voice)
 {
 	unsigned int val, tmp;
 
-	printk("Trident voice %i:\n", voice);
+	printk(KERN_DEBUG "Trident voice %i:\n", voice);
 	outb(voice, TRID_REG(trident, T4D_LFO_GC_CIR));
 	val = inl(TRID_REG(trident, CH_LBA));
-	printk("LBA: 0x%x\n", val);
+	printk(KERN_DEBUG "LBA: 0x%x\n", val);
 	val = inl(TRID_REG(trident, CH_GVSEL_PAN_VOL_CTRL_EC));
-	printk("GVSel: %i\n", val >> 31);
-	printk("Pan: 0x%x\n", (val >> 24) & 0x7f);
-	printk("Vol: 0x%x\n", (val >> 16) & 0xff);
-	printk("CTRL: 0x%x\n", (val >> 12) & 0x0f);
-	printk("EC: 0x%x\n", val & 0x0fff);
+	printk(KERN_DEBUG "GVSel: %i\n", val >> 31);
+	printk(KERN_DEBUG "Pan: 0x%x\n", (val >> 24) & 0x7f);
+	printk(KERN_DEBUG "Vol: 0x%x\n", (val >> 16) & 0xff);
+	printk(KERN_DEBUG "CTRL: 0x%x\n", (val >> 12) & 0x0f);
+	printk(KERN_DEBUG "EC: 0x%x\n", val & 0x0fff);
 	if (trident->device != TRIDENT_DEVICE_ID_NX) {
 		val = inl(TRID_REG(trident, CH_DX_CSO_ALPHA_FMS));
-		printk("CSO: 0x%x\n", val >> 16);
+		printk(KERN_DEBUG "CSO: 0x%x\n", val >> 16);
 		printk("Alpha: 0x%x\n", (val >> 4) & 0x0fff);
-		printk("FMS: 0x%x\n", val & 0x0f);
+		printk(KERN_DEBUG "FMS: 0x%x\n", val & 0x0f);
 		val = inl(TRID_REG(trident, CH_DX_ESO_DELTA));
-		printk("ESO: 0x%x\n", val >> 16);
-		printk("Delta: 0x%x\n", val & 0xffff);
+		printk(KERN_DEBUG "ESO: 0x%x\n", val >> 16);
+		printk(KERN_DEBUG "Delta: 0x%x\n", val & 0xffff);
 		val = inl(TRID_REG(trident, CH_DX_FMC_RVOL_CVOL));
 	} else {		// TRIDENT_DEVICE_ID_NX
 		val = inl(TRID_REG(trident, CH_NX_DELTA_CSO));
 		tmp = (val >> 24) & 0xff;
-		printk("CSO: 0x%x\n", val & 0x00ffffff);
+		printk(KERN_DEBUG "CSO: 0x%x\n", val & 0x00ffffff);
 		val = inl(TRID_REG(trident, CH_NX_DELTA_ESO));
 		tmp |= (val >> 16) & 0xff00;
-		printk("Delta: 0x%x\n", tmp);
-		printk("ESO: 0x%x\n", val & 0x00ffffff);
+		printk(KERN_DEBUG "Delta: 0x%x\n", tmp);
+		printk(KERN_DEBUG "ESO: 0x%x\n", val & 0x00ffffff);
 		val = inl(TRID_REG(trident, CH_NX_ALPHA_FMS_FMC_RVOL_CVOL));
-		printk("Alpha: 0x%x\n", val >> 20);
-		printk("FMS: 0x%x\n", (val >> 16) & 0x0f);
+		printk(KERN_DEBUG "Alpha: 0x%x\n", val >> 20);
+		printk(KERN_DEBUG "FMS: 0x%x\n", (val >> 16) & 0x0f);
 	}
-	printk("FMC: 0x%x\n", (val >> 14) & 3);
-	printk("RVol: 0x%x\n", (val >> 7) & 0x7f);
-	printk("CVol: 0x%x\n", val & 0x7f);
+	printk(KERN_DEBUG "FMC: 0x%x\n", (val >> 14) & 3);
+	printk(KERN_DEBUG "RVol: 0x%x\n", (val >> 7) & 0x7f);
+	printk(KERN_DEBUG "CVol: 0x%x\n", val & 0x7f);
 }
 #endif
 
@@ -496,12 +496,17 @@ void snd_trident_write_voice_regs(struct snd_trident * trident,
 	outl(regs[4], TRID_REG(trident, CH_START + 16));
 
 #if 0
-	printk("written %i channel:\n", voice->number);
-	printk("  regs[0] = 0x%x/0x%x\n", regs[0], inl(TRID_REG(trident, CH_START + 0)));
-	printk("  regs[1] = 0x%x/0x%x\n", regs[1], inl(TRID_REG(trident, CH_START + 4)));
-	printk("  regs[2] = 0x%x/0x%x\n", regs[2], inl(TRID_REG(trident, CH_START + 8)));
-	printk("  regs[3] = 0x%x/0x%x\n", regs[3], inl(TRID_REG(trident, CH_START + 12)));
-	printk("  regs[4] = 0x%x/0x%x\n", regs[4], inl(TRID_REG(trident, CH_START + 16)));
+	printk(KERN_DEBUG "written %i channel:\n", voice->number);
+	printk(KERN_DEBUG "  regs[0] = 0x%x/0x%x\n",
+	       regs[0], inl(TRID_REG(trident, CH_START + 0)));
+	printk(KERN_DEBUG "  regs[1] = 0x%x/0x%x\n",
+	       regs[1], inl(TRID_REG(trident, CH_START + 4)));
+	printk(KERN_DEBUG "  regs[2] = 0x%x/0x%x\n",
+	       regs[2], inl(TRID_REG(trident, CH_START + 8)));
+	printk(KERN_DEBUG "  regs[3] = 0x%x/0x%x\n",
+	       regs[3], inl(TRID_REG(trident, CH_START + 12)));
+	printk(KERN_DEBUG "  regs[4] = 0x%x/0x%x\n",
+	       regs[4], inl(TRID_REG(trident, CH_START + 16)));
 #endif
 }
 
@@ -583,7 +588,7 @@ static void snd_trident_write_vol_reg(struct snd_trident * trident,
 		outb(voice->Vol >> 2, TRID_REG(trident, CH_GVSEL_PAN_VOL_CTRL_EC + 2));
 		break;
 	case TRIDENT_DEVICE_ID_SI7018:
-		// printk("voice->Vol = 0x%x\n", voice->Vol);
+		/* printk(KERN_DEBUG "voice->Vol = 0x%x\n", voice->Vol); */
 		outw((voice->CTRL << 12) | voice->Vol,
 		     TRID_REG(trident, CH_GVSEL_PAN_VOL_CTRL_EC));
 		break;
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 1aafe956ee2..fc62d6380f8 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -466,7 +466,10 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre
 					flag = VIA_TBL_BIT_FLAG; /* period boundary */
 			} else
 				flag = 0; /* period continues to the next */
-			// printk("via: tbl %d: at %d  size %d (rest %d)\n", idx, ofs, r, rest);
+			/*
+			printk(KERN_DEBUG "via: tbl %d: at %d  size %d "
+			       "(rest %d)\n", idx, ofs, r, rest);
+			*/
 			((u32 *)dev->table.area)[(idx<<1) + 1] = cpu_to_le32(r | flag);
 			dev->idx_table[idx].offset = ofs;
 			dev->idx_table[idx].size = r;
diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c
index 5bd79d2a5a1..c0d9cc9dad4 100644
--- a/sound/pci/via82xx_modem.c
+++ b/sound/pci/via82xx_modem.c
@@ -328,7 +328,10 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre
 					flag = VIA_TBL_BIT_FLAG; /* period boundary */
 			} else
 				flag = 0; /* period continues to the next */
-			// printk("via: tbl %d: at %d  size %d (rest %d)\n", idx, ofs, r, rest);
+			/*
+			printk(KERN_DEBUG "via: tbl %d: at %d  size %d "
+			       "(rest %d)\n", idx, ofs, r, rest);
+			*/
 			((u32 *)dev->table.area)[(idx<<1) + 1] = cpu_to_le32(r | flag);
 			dev->idx_table[idx].offset = ofs;
 			dev->idx_table[idx].size = r;
diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c
index 7e87f398ff0..c0efe449111 100644
--- a/sound/pci/vx222/vx222_ops.c
+++ b/sound/pci/vx222/vx222_ops.c
@@ -107,7 +107,9 @@ static unsigned char vx2_inb(struct vx_core *chip, int offset)
 static void vx2_outb(struct vx_core *chip, int offset, unsigned char val)
 {
 	outb(val, vx2_reg_addr(chip, offset));
-	//printk("outb: %x -> %x\n", val, vx2_reg_addr(chip, offset));
+	/*
+	printk(KERN_DEBUG "outb: %x -> %x\n", val, vx2_reg_addr(chip, offset));
+	*/
 }
 
 /**
@@ -126,7 +128,9 @@ static unsigned int vx2_inl(struct vx_core *chip, int offset)
  */
 static void vx2_outl(struct vx_core *chip, int offset, unsigned int val)
 {
-	// printk("outl: %x -> %x\n", val, vx2_reg_addr(chip, offset));
+	/*
+	printk(KERN_DEBUG "outl: %x -> %x\n", val, vx2_reg_addr(chip, offset));
+	*/
 	outl(val, vx2_reg_addr(chip, offset));
 }
 
diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c
index 90d0d62bd0b..2f0925236a1 100644
--- a/sound/pci/ymfpci/ymfpci_main.c
+++ b/sound/pci/ymfpci/ymfpci_main.c
@@ -318,7 +318,12 @@ static void snd_ymfpci_pcm_interrupt(struct snd_ymfpci *chip, struct snd_ymfpci_
 		ypcm->period_pos += delta;
 		ypcm->last_pos = pos;
 		if (ypcm->period_pos >= ypcm->period_size) {
-			// printk("done - active_bank = 0x%x, start = 0x%x\n", chip->active_bank, voice->bank[chip->active_bank].start);
+			/*
+			printk(KERN_DEBUG
+			       "done - active_bank = 0x%x, start = 0x%x\n",
+			       chip->active_bank,
+			       voice->bank[chip->active_bank].start);
+			*/
 			ypcm->period_pos %= ypcm->period_size;
 			spin_unlock(&chip->reg_lock);
 			snd_pcm_period_elapsed(ypcm->substream);
@@ -366,7 +371,12 @@ static void snd_ymfpci_pcm_capture_interrupt(struct snd_pcm_substream *substream
 		ypcm->last_pos = pos;
 		if (ypcm->period_pos >= ypcm->period_size) {
 			ypcm->period_pos %= ypcm->period_size;
-			// printk("done - active_bank = 0x%x, start = 0x%x\n", chip->active_bank, voice->bank[chip->active_bank].start);
+			/*
+			printk(KERN_DEBUG
+			       "done - active_bank = 0x%x, start = 0x%x\n",
+			       chip->active_bank,
+			       voice->bank[chip->active_bank].start);
+			*/
 			spin_unlock(&chip->reg_lock);
 			snd_pcm_period_elapsed(substream);
 			spin_lock(&chip->reg_lock);
-- 
cgit v1.2.3


From dd542f169aaa35f4ac0d063e04b41c648a93887c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 5 Feb 2009 16:15:39 +0100
Subject: ALSA: ca0106 - Add missing KERN_* prefix to printk

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ca0106/ca0106_main.c | 91 ++++++++++++++++++++++++++++++++----------
 1 file changed, 71 insertions(+), 20 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index 3aac7e6489c..dac8a5f040e 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -412,7 +412,9 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu,
 	}
 
 	tmp = reg << 25 | value << 16;
-	// snd_printk("I2C-write:reg=0x%x, value=0x%x\n", reg, value);
+	/*
+	snd_printk(KERN_DEBUG "I2C-write:reg=0x%x, value=0x%x\n", reg, value);
+	*/
 	/* Not sure what this I2C channel controls. */
 	/* snd_ca0106_ptr_write(emu, I2C_D0, 0, tmp); */
 
@@ -430,7 +432,7 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu,
 		/* Wait till the transaction ends */
 		while (1) {
 			status = snd_ca0106_ptr_read(emu, I2C_A, 0);
-                	//snd_printk("I2C:status=0x%x\n", status);
+			/*snd_printk(KERN_DEBUG "I2C:status=0x%x\n", status);*/
 			timeout++;
 			if ((status & I2C_A_ADC_START) == 0)
 				break;
@@ -529,7 +531,10 @@ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substr
         channel->number = channel_id;
 
 	channel->use = 1;
-        //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel);
+	/*
+	printk(KERN_DEBUG "open:channel_id=%d, chip=%p, channel=%p\n",
+	       channel_id, chip, channel);
+	*/
         //channel->interrupt = snd_ca0106_pcm_channel_interrupt;
 	channel->epcm = epcm;
 	if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
@@ -622,7 +627,10 @@ static int snd_ca0106_pcm_open_capture_channel(struct snd_pcm_substream *substre
         channel->number = channel_id;
 
 	channel->use = 1;
-        //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel);
+	/*
+        printk(KERN_DEBUG "open:channel_id=%d, chip=%p, channel=%p\n",
+	       channel_id, chip, channel);
+	*/
         //channel->interrupt = snd_ca0106_pcm_channel_interrupt;
         channel->epcm = epcm;
 	if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
@@ -713,9 +721,20 @@ static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream)
 	u32 reg71;
 	int i;
 	
-        //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1));
-        //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base);
-	//snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes);
+#if 0 /* debug */
+	snd_printk(KERN_DEBUG
+		   "prepare:channel_number=%d, rate=%d, format=0x%x, "
+		   "channels=%d, buffer_size=%ld, period_size=%ld, "
+		   "periods=%u, frames_to_bytes=%d\n",
+		   channel, runtime->rate, runtime->format,
+		   runtime->channels, runtime->buffer_size,
+		   runtime->period_size, runtime->periods,
+		   frames_to_bytes(runtime, 1));
+	snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, table_base=%p\n",
+		   runtime->dma_addr, runtime->dma_area, table_base);
+	snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",
+		   emu->buffer.addr, emu->buffer.area, emu->buffer.bytes);
+#endif /* debug */
 	/* Rate can be set per channel. */
 	/* reg40 control host to fifo */
 	/* reg71 controls DAC rate. */
@@ -807,9 +826,20 @@ static int snd_ca0106_pcm_prepare_capture(struct snd_pcm_substream *substream)
 	u32 reg71_set = 0;
 	u32 reg71;
 	
-        //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1));
-        //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base);
-	//snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes);
+#if 0 /* debug */
+	snd_printk(KERN_DEBUG
+		   "prepare:channel_number=%d, rate=%d, format=0x%x, "
+		   "channels=%d, buffer_size=%ld, period_size=%ld, "
+		   "periods=%u, frames_to_bytes=%d\n",
+		   channel, runtime->rate, runtime->format,
+		   runtime->channels, runtime->buffer_size,
+		   runtime->period_size, runtime->periods,
+		   frames_to_bytes(runtime, 1));
+        snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, table_base=%p\n",
+		   runtime->dma_addr, runtime->dma_area, table_base);
+	snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",
+		   emu->buffer.addr, emu->buffer.area, emu->buffer.bytes);
+#endif /* debug */
 	/* reg71 controls ADC rate. */
 	switch (runtime->rate) {
 	case 44100:
@@ -854,7 +884,14 @@ static int snd_ca0106_pcm_prepare_capture(struct snd_pcm_substream *substream)
 	}
 
 
-        //printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size,  frames_to_bytes(runtime, 1));
+	/*
+	printk(KERN_DEBUG
+	       "prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, "
+	       "buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",
+	       channel, runtime->rate, runtime->format, runtime->channels,
+	       runtime->buffer_size, runtime->period_size,
+	       frames_to_bytes(runtime, 1));
+	*/
 	snd_ca0106_ptr_write(emu, 0x13, channel, 0);
 	snd_ca0106_ptr_write(emu, CAPTURE_DMA_ADDR, channel, runtime->dma_addr);
 	snd_ca0106_ptr_write(emu, CAPTURE_BUFFER_SIZE, channel, frames_to_bytes(runtime, runtime->buffer_size)<<16); // buffer size in bytes
@@ -896,13 +933,13 @@ static int snd_ca0106_pcm_trigger_playback(struct snd_pcm_substream *substream,
 		runtime = s->runtime;
 		epcm = runtime->private_data;
 		channel = epcm->channel_id;
-		/* snd_printk("channel=%d\n",channel); */
+		/* snd_printk(KERN_DEBUG "channel=%d\n", channel); */
 		epcm->running = running;
 		basic |= (0x1 << channel);
 		extended |= (0x10 << channel);
                 snd_pcm_trigger_done(s, substream);
         }
-	/* snd_printk("basic=0x%x, extended=0x%x\n",basic, extended); */
+	/* snd_printk(KERN_DEBUG "basic=0x%x, extended=0x%x\n",basic, extended); */
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
@@ -980,8 +1017,13 @@ snd_ca0106_pcm_pointer_playback(struct snd_pcm_substream *substream)
 	ptr=ptr2;
         if (ptr >= runtime->buffer_size)
 		ptr -= runtime->buffer_size;
-	//printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate);
-
+	/*
+	printk(KERN_DEBUG "ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, "
+	       "buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n",
+	       ptr1, ptr2, ptr, (int)runtime->buffer_size,
+	       (int)runtime->period_size, (int)runtime->frame_bits,
+	       (int)runtime->rate);
+	*/
 	return ptr;
 }
 
@@ -1003,8 +1045,13 @@ snd_ca0106_pcm_pointer_capture(struct snd_pcm_substream *substream)
 	ptr=ptr2;
         if (ptr >= runtime->buffer_size)
 		ptr -= runtime->buffer_size;
-	//printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate);
-
+	/*
+	printk(KERN_DEBUG "ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, "
+	       "buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n",
+	       ptr1, ptr2, ptr, (int)runtime->buffer_size,
+	       (int)runtime->period_size, (int)runtime->frame_bits,
+	       (int)runtime->rate);
+	*/
 	return ptr;
 }
 
@@ -1189,8 +1236,12 @@ static irqreturn_t snd_ca0106_interrupt(int irq, void *dev_id)
 		return IRQ_NONE;
 
         stat76 = snd_ca0106_ptr_read(chip, EXTENDED_INT, 0);
-	//snd_printk("interrupt status = 0x%08x, stat76=0x%08x\n", status, stat76);
-	//snd_printk("ptr=0x%08x\n",snd_ca0106_ptr_read(chip, PLAYBACK_POINTER, 0));
+	/*
+	snd_printk(KERN_DEBUG "interrupt status = 0x%08x, stat76=0x%08x\n",
+		   status, stat76);
+	snd_printk(KERN_DEBUG "ptr=0x%08x\n",
+		   snd_ca0106_ptr_read(chip, PLAYBACK_POINTER, 0));
+	*/
         mask = 0x11; /* 0x1 for one half, 0x10 for the other half period. */
 	for(i = 0; i < 4; i++) {
 		pchannel = &(chip->playback_channels[i]);
@@ -1478,7 +1529,7 @@ static void ca0106_init_chip(struct snd_ca0106 *chip, int resume)
 		int size, n;
 
 		size = ARRAY_SIZE(i2c_adc_init);
-		/* snd_printk("I2C:array size=0x%x\n", size); */
+		/* snd_printk(KERN_DEBUG "I2C:array size=0x%x\n", size); */
 		for (n = 0; n < size; n++)
 			snd_ca0106_i2c_write(chip, i2c_adc_init[n][0],
 					     i2c_adc_init[n][1]);
-- 
cgit v1.2.3


From b25c9da19889e33bb4ee2dff369fc46caa4543b0 Mon Sep 17 00:00:00 2001
From: Wu Fengguang <fengguang.wu@intel.com>
Date: Fri, 6 Feb 2009 15:02:27 +0800
Subject: ALSA: enable concurrent digital outputs for ALC1200

Add the SPDIF pin as slave digital out to enable concurrent
HDMI/SPDIF outputs for ASUS M3A-H/HDMI with ALC1200 codec.

Tested-by: Thomas Schneider <nailstudio@gmx.net>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_local.h     | 1 +
 sound/pci/hda/patch_realtek.c | 8 ++++++++
 2 files changed, 9 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index ec687b206c0..4086491ed33 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -229,6 +229,7 @@ struct hda_multi_out {
 	hda_nid_t hp_nid;	/* optional DAC for HP, 0 when not exists */
 	hda_nid_t extra_out_nid[3];	/* optional DACs, 0 when not exists */
 	hda_nid_t dig_out_nid;	/* digital out audio widget */
+	hda_nid_t *slave_dig_outs;
 	int max_channels;	/* currently supported analog channels */
 	int dig_out_used;	/* current usage of digital out (HDA_DIG_XXX) */
 	int no_share_stream;	/* don't share a stream with multiple pins */
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index d2812ab729c..5194a58fafa 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -349,6 +349,7 @@ struct alc_config_preset {
 	hda_nid_t *dac_nids;
 	hda_nid_t dig_out_nid;		/* optional */
 	hda_nid_t hp_nid;		/* optional */
+	hda_nid_t *slave_dig_outs;
 	unsigned int num_adc_nids;
 	hda_nid_t *adc_nids;
 	hda_nid_t *capsrc_nids;
@@ -824,6 +825,7 @@ static void setup_preset(struct alc_spec *spec,
 	spec->multiout.num_dacs = preset->num_dacs;
 	spec->multiout.dac_nids = preset->dac_nids;
 	spec->multiout.dig_out_nid = preset->dig_out_nid;
+	spec->multiout.slave_dig_outs = preset->slave_dig_outs;
 	spec->multiout.hp_nid = preset->hp_nid;
 
 	spec->num_mux_defs = preset->num_mux_defs;
@@ -3107,6 +3109,7 @@ static int alc_build_pcms(struct hda_codec *codec)
 	/* SPDIF for stream index #1 */
 	if (spec->multiout.dig_out_nid || spec->dig_in_nid) {
 		codec->num_pcms = 2;
+	        codec->slave_dig_outs = spec->multiout.slave_dig_outs;
 		info = spec->pcm_rec + 1;
 		info->name = spec->stream_name_digital;
 		if (spec->dig_out_type)
@@ -8603,6 +8606,10 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	{}
 };
 
+static hda_nid_t alc1200_slave_dig_outs[] = {
+	ALC883_DIGOUT_NID, 0,
+};
+
 static struct alc_config_preset alc883_presets[] = {
 	[ALC883_3ST_2ch_DIG] = {
 		.mixers = { alc883_3ST_2ch_mixer },
@@ -8943,6 +8950,7 @@ static struct alc_config_preset alc883_presets[] = {
 		.dac_nids = alc883_dac_nids,
 		.dig_out_nid = ALC1200_DIGOUT_NID,
 		.dig_in_nid = ALC883_DIGIN_NID,
+		.slave_dig_outs = alc1200_slave_dig_outs,
 		.num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
 		.channel_mode = alc883_sixstack_modes,
 		.input_mux = &alc883_capture_source,
-- 
cgit v1.2.3


From 45bdd1c1bbac56876cb9c71649300013281e4b22 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 6 Feb 2009 16:11:25 +0100
Subject: ALSA: hda - Create beep mixer controls dynamically for Realtek codecs

Create beep mixer controls dynamically for Realtek codecs instead of
static arrays.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 147 ++++++++++++++----------------------------
 1 file changed, 47 insertions(+), 100 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 5194a58fafa..3b3b483e2a9 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -255,6 +255,7 @@ struct alc_spec {
 	struct snd_kcontrol_new *mixers[5];	/* mixer arrays */
 	unsigned int num_mixers;
 	struct snd_kcontrol_new *cap_mixer;	/* capture mixer */
+	unsigned int beep_amp;	/* beep amp value, set via set_beep_amp() */
 
 	const struct hda_verb *init_verbs[5];	/* initialization verbs
 						 * don't forget NULL
@@ -937,7 +938,7 @@ static void alc_mic_automute(struct hda_codec *codec)
 			 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
 }
 #else
-#define alc_mic_automute(codec) /* NOP */
+#define alc_mic_automute(codec) do {} while(0) /* NOP */
 #endif /* disabled */
 
 /* unsolicited event for HP jack sensing */
@@ -1389,8 +1390,6 @@ static struct snd_kcontrol_new alc888_base_mixer[] = {
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -1497,8 +1496,6 @@ static struct snd_kcontrol_new alc880_three_stack_mixer[] = {
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x3, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x3, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x19, 0x0, HDA_OUTPUT),
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1720,8 +1717,6 @@ static struct snd_kcontrol_new alc880_six_stack_mixer[] = {
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Channel Mode",
@@ -1898,13 +1893,6 @@ static struct snd_kcontrol_new alc880_asus_w1v_mixer[] = {
 	{ } /* end */
 };
 
-/* additional mixers to alc880_asus_mixer */
-static struct snd_kcontrol_new alc880_pcbeep_mixer[] = {
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
-	{ } /* end */
-};
-
 /* TCL S700 */
 static struct snd_kcontrol_new alc880_tcl_s700_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
@@ -1937,8 +1925,6 @@ static struct snd_kcontrol_new alc880_uniwill_mixer[] = {
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Channel Mode",
@@ -2013,6 +1999,13 @@ static const char *alc_slave_sws[] = {
 
 static void alc_free_kctls(struct hda_codec *codec);
 
+/* additional beep mixers; the actual parameters are overwritten at build */
+static struct snd_kcontrol_new alc_beep_mixer[] = {
+	HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Beep Playback Switch", 0, 0, HDA_INPUT),
+	{ } /* end */
+};
+
 static int alc_build_controls(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -2048,6 +2041,21 @@ static int alc_build_controls(struct hda_codec *codec)
 			return err;
 	}
 
+	/* create beep controls if needed */
+	if (spec->beep_amp) {
+		struct snd_kcontrol_new *knew;
+		for (knew = alc_beep_mixer; knew->name; knew++) {
+			struct snd_kcontrol *kctl;
+			kctl = snd_ctl_new1(knew, codec);
+			if (!kctl)
+				return -ENOMEM;
+			kctl->private_value = spec->beep_amp;
+			err = snd_hda_ctl_add(codec, kctl);
+			if (err < 0)
+				return err;
+		}
+	}
+
 	/* if we have no master control, let's create it */
 	if (!spec->no_analog &&
 	    !snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) {
@@ -3812,7 +3820,7 @@ static struct alc_config_preset alc880_presets[] = {
 		.input_mux = &alc880_capture_source,
 	},
 	[ALC880_UNIWILL_DIG] = {
-		.mixers = { alc880_asus_mixer, alc880_pcbeep_mixer },
+		.mixers = { alc880_asus_mixer },
 		.init_verbs = { alc880_volume_init_verbs,
 				alc880_pin_asus_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
@@ -3850,8 +3858,7 @@ static struct alc_config_preset alc880_presets[] = {
 		.init_hook = alc880_uniwill_p53_hp_automute,
 	},
 	[ALC880_FUJITSU] = {
-		.mixers = { alc880_fujitsu_mixer,
-			    alc880_pcbeep_mixer, },
+		.mixers = { alc880_fujitsu_mixer },
 		.init_verbs = { alc880_volume_init_verbs,
 				alc880_uniwill_p53_init_verbs,
 	       			alc880_beep_init_verbs },
@@ -4310,10 +4317,6 @@ static void alc880_auto_init(struct hda_codec *codec)
 		alc_inithook(codec);
 }
 
-/*
- * OK, here we have finally the patch for ALC880
- */
-
 static void set_capture_mixer(struct alc_spec *spec)
 {
 	static struct snd_kcontrol_new *caps[3] = {
@@ -4325,6 +4328,13 @@ static void set_capture_mixer(struct alc_spec *spec)
 		spec->cap_mixer = caps[spec->num_adc_nids - 1];
 }
 
+#define set_beep_amp(spec, nid, idx, dir) \
+	((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir))
+
+/*
+ * OK, here we have finally the patch for ALC880
+ */
+
 static int patch_alc880(struct hda_codec *codec)
 {
 	struct alc_spec *spec;
@@ -4392,6 +4402,7 @@ static int patch_alc880(struct hda_codec *codec)
 		}
 	}
 	set_capture_mixer(spec);
+	set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 
 	spec->vmaster_nid = 0x0c;
 
@@ -4541,12 +4552,6 @@ static struct snd_kcontrol_new alc260_input_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc260_pc_beep_mixer[] = {
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x07, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x07, 0x05, HDA_INPUT),
-	{ } /* end */
-};
-
 /* update HP, line and mono out pins according to the master switch */
 static void alc260_hp_master_update(struct hda_codec *codec,
 				    hda_nid_t hp, hda_nid_t line,
@@ -4738,8 +4743,6 @@ static struct snd_kcontrol_new alc260_fujitsu_mixer[] = {
 	HDA_CODEC_VOLUME("Mic/Line Playback Volume", 0x07, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Mic/Line Playback Switch", 0x07, 0x0, HDA_INPUT),
 	ALC_PIN_MODE("Mic/Line Jack Mode", 0x12, ALC_PIN_DIR_IN),
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT),
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Speaker Playback Switch", 0x09, 2, HDA_INPUT),
 	{ } /* end */
@@ -4784,8 +4787,6 @@ static struct snd_kcontrol_new alc260_acer_mixer[] = {
 	HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
 	HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
 	ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -4803,8 +4804,6 @@ static struct snd_kcontrol_new alc260_will_mixer[] = {
 	ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -5308,8 +5307,6 @@ static struct snd_kcontrol_new alc260_test_mixer[] = {
 	HDA_CODEC_MUTE("LINE2 Playback Switch", 0x07, 0x03, HDA_INPUT),
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT),
 	HDA_CODEC_VOLUME("LINE-OUT loopback Playback Volume", 0x07, 0x06, HDA_INPUT),
 	HDA_CODEC_MUTE("LINE-OUT loopback Playback Switch", 0x07, 0x06, HDA_INPUT),
 	HDA_CODEC_VOLUME("HP-OUT loopback Playback Volume", 0x07, 0x7, HDA_INPUT),
@@ -5737,8 +5734,7 @@ static struct snd_pci_quirk alc260_cfg_tbl[] = {
 static struct alc_config_preset alc260_presets[] = {
 	[ALC260_BASIC] = {
 		.mixers = { alc260_base_output_mixer,
-			    alc260_input_mixer,
-			    alc260_pc_beep_mixer },
+			    alc260_input_mixer },
 		.init_verbs = { alc260_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc260_dac_nids),
 		.dac_nids = alc260_dac_nids,
@@ -5924,6 +5920,7 @@ static int patch_alc260(struct hda_codec *codec)
 		}
 	}
 	set_capture_mixer(spec);
+	set_beep_amp(spec, 0x07, 0x05, HDA_INPUT);
 
 	spec->vmaster_nid = 0x08;
 
@@ -6095,8 +6092,6 @@ static struct snd_kcontrol_new alc882_base_mixer[] = {
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -6123,8 +6118,6 @@ static struct snd_kcontrol_new alc882_w2jc_mixer[] = {
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -6176,8 +6169,6 @@ static struct snd_kcontrol_new alc882_asus_a7m_mixer[] = {
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -6286,8 +6277,10 @@ static struct snd_kcontrol_new alc882_macpro_mixer[] = {
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x18, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT),
 	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT),
+	/* FIXME: this looks suspicious...
 	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x02, HDA_INPUT),
 	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	*/
 	{ } /* end */
 };
 
@@ -7153,6 +7146,7 @@ static int patch_alc882(struct hda_codec *codec)
 		}
 	}
 	set_capture_mixer(spec);
+	set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 
 	spec->vmaster_nid = 0x0c;
 
@@ -7429,8 +7423,6 @@ static struct snd_kcontrol_new alc883_base_mixer[] = {
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -7493,8 +7485,6 @@ static struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = {
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -7518,8 +7508,6 @@ static struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = {
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -7544,8 +7532,6 @@ static struct snd_kcontrol_new alc883_3ST_6ch_intel_mixer[] = {
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x18, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -7569,8 +7555,6 @@ static struct snd_kcontrol_new alc883_fivestack_mixer[] = {
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -9183,6 +9167,7 @@ static int patch_alc883(struct hda_codec *codec)
 
 	if (!spec->cap_mixer)
 		set_capture_mixer(spec);
+	set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 
 	spec->vmaster_nid = 0x0c;
 
@@ -9235,8 +9220,6 @@ static struct snd_kcontrol_new alc262_base_mixer[] = {
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
-	/* HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	   HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), */
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
@@ -9257,8 +9240,6 @@ static struct snd_kcontrol_new alc262_hippo1_mixer[] = {
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
-	/* HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	   HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), */
 	/*HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT),*/
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
 	{ } /* end */
@@ -9367,8 +9348,6 @@ static struct snd_kcontrol_new alc262_HP_BPC_mixer[] = {
 	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	HDA_CODEC_VOLUME("AUX IN Playback Volume", 0x0b, 0x06, HDA_INPUT),
 	HDA_CODEC_MUTE("AUX IN Playback Switch", 0x0b, 0x06, HDA_INPUT),
 	{ } /* end */
@@ -9397,8 +9376,6 @@ static struct snd_kcontrol_new alc262_HP_BPC_WildWest_mixer[] = {
 	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT),
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -10073,8 +10050,6 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = {
 	},
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Switch", 0x0b, 0x05, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
@@ -11085,6 +11060,7 @@ static int patch_alc262(struct hda_codec *codec)
 	}
 	if (!spec->cap_mixer && !spec->no_analog)
 		set_capture_mixer(spec);
+	set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 
 	spec->vmaster_nid = 0x0c;
 
@@ -12205,8 +12181,6 @@ static struct snd_kcontrol_new alc269_base_mixer[] = {
 	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x0b, 0x4, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x0b, 0x4, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
@@ -12233,8 +12207,6 @@ static struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = {
 	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
 	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
 	HDA_CODEC_VOLUME("Internal Mic Boost", 0x19, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x04, HDA_INPUT),
 	{ }
 };
 
@@ -12258,8 +12230,6 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = {
 	HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x0b, 0x03, HDA_INPUT),
 	HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x0b, 0x03, HDA_INPUT),
 	HDA_CODEC_VOLUME("Dock Mic Boost", 0x1b, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x04, HDA_INPUT),
 	{ }
 };
 
@@ -12296,13 +12266,6 @@ static struct snd_kcontrol_new alc269_fujitsu_mixer[] = {
 	{ } /* end */
 };
 
-/* beep control */
-static struct snd_kcontrol_new alc269_beep_mixer[] = {
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x0b, 0x4, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x0b, 0x4, HDA_INPUT),
-	{ } /* end */
-};
-
 static struct hda_verb alc269_quanta_fl1_verbs[] = {
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
 	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
@@ -12749,13 +12712,6 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 	if (spec->kctls.list)
 		add_mixer(spec, spec->kctls.list);
 
-	/* create a beep mixer control if the pin 0x1d isn't assigned */
-	for (i = 0; i < ARRAY_SIZE(spec->autocfg.input_pins); i++)
-		if (spec->autocfg.input_pins[i] == 0x1d)
-			break;
-	if (i >= ARRAY_SIZE(spec->autocfg.input_pins))
-		add_mixer(spec, alc269_beep_mixer);
-
 	add_verb(spec, alc269_init_verbs);
 	spec->num_mux_defs = 1;
 	spec->input_mux = &spec->private_imux[0];
@@ -12868,7 +12824,7 @@ static struct alc_config_preset alc269_presets[] = {
 		.init_hook = alc269_eeepc_dmic_inithook,
 	},
 	[ALC269_FUJITSU] = {
-		.mixers = { alc269_fujitsu_mixer, alc269_beep_mixer },
+		.mixers = { alc269_fujitsu_mixer },
 		.cap_mixer = alc269_epc_capture_mixer,
 		.init_verbs = { alc269_init_verbs,
 				alc269_eeepc_dmic_init_verbs },
@@ -12955,6 +12911,7 @@ static int patch_alc269(struct hda_codec *codec)
 	spec->capsrc_nids = alc269_capsrc_nids;
 	if (!spec->cap_mixer)
 		set_capture_mixer(spec);
+	set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
 
 	codec->patch_ops = alc_patch_ops;
 	if (board_config == ALC269_AUTO)
@@ -13205,8 +13162,6 @@ static struct snd_kcontrol_new alc861_asus_mixer[] = {
 static struct snd_kcontrol_new alc861_asus_laptop_mixer[] = {
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x23, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PC Beep Playback Switch", 0x23, 0x0, HDA_OUTPUT),
 	{ }
 };
 
@@ -14049,6 +14004,8 @@ static int patch_alc861(struct hda_codec *codec)
 	spec->stream_digital_playback = &alc861_pcm_digital_playback;
 	spec->stream_digital_capture = &alc861_pcm_digital_capture;
 
+	set_beep_amp(spec, 0x23, 0, HDA_OUTPUT);
+
 	spec->vmaster_nid = 0x03;
 
 	codec->patch_ops = alc_patch_ops;
@@ -14205,9 +14162,6 @@ static struct snd_kcontrol_new alc861vd_6st_mixer[] = {
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
 
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
-
 	{ } /* end */
 };
 
@@ -14231,9 +14185,6 @@ static struct snd_kcontrol_new alc861vd_3st_mixer[] = {
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
 
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
-
 	{ } /* end */
 };
 
@@ -14272,8 +14223,6 @@ static struct snd_kcontrol_new alc861vd_dallas_mixer[] = {
 	HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Beep Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Beep Switch", 0x0b, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -15015,6 +14964,7 @@ static int patch_alc861vd(struct hda_codec *codec)
 	spec->capture_style = CAPT_MIX;
 
 	set_capture_mixer(spec);
+	set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 
 	spec->vmaster_nid = 0x02;
 
@@ -15203,8 +15153,6 @@ static struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = {
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -15226,8 +15174,6 @@ static struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = {
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -16832,6 +16778,7 @@ static int patch_alc662(struct hda_codec *codec)
 
 	if (!spec->cap_mixer)
 		set_capture_mixer(spec);
+	set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 
 	spec->vmaster_nid = 0x02;
 
-- 
cgit v1.2.3


From c8dcdf829ca1827a802eae841dd04de8c9d6653f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 6 Feb 2009 16:21:20 +0100
Subject: ALSA: hda - Add missing NULL check in snd_hda_create_spdif_in_ctls()

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index c9158799ccb..93412f335dc 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1984,6 +1984,8 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid)
 	}
 	for (dig_mix = dig_in_ctls; dig_mix->name; dig_mix++) {
 		kctl = snd_ctl_new1(dig_mix, codec);
+		if (!kctl)
+			return -ENOMEM;
 		kctl->private_value = nid;
 		err = snd_hda_ctl_add(codec, kctl);
 		if (err < 0)
-- 
cgit v1.2.3


From c44765b8c8bfc883c9868ab7aef37d27b5b14be8 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 6 Feb 2009 16:48:10 +0100
Subject: ALSA: hda - Clear codec->beep at release

Clear codec->beep field in snd_hda_detach_beep_device() to be sure.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_beep.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 960fd797038..4de5bacd392 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -138,6 +138,7 @@ void snd_hda_detach_beep_device(struct hda_codec *codec)
 
 		input_unregister_device(beep->dev);
 		kfree(beep);
+		codec->beep = NULL;
 	}
 }
 EXPORT_SYMBOL_HDA(snd_hda_detach_beep_device);
-- 
cgit v1.2.3


From a4ddeba9c8896cba8c6ce7a98c0b5c755c15a746 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 6 Feb 2009 17:21:09 +0100
Subject: ALSA: hda - Remove superfluous code in patch_realtek.c

codec->spec is reset in the caller side.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 76934bc8b48..3d933e307b1 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3202,7 +3202,6 @@ static void alc_free(struct hda_codec *codec)
 	alc_free_kctls(codec);
 	kfree(spec);
 	snd_hda_detach_beep_device(codec);
-	codec->spec = NULL; /* to be sure */
 }
 
 #ifdef SND_HDA_NEEDS_RESUME
-- 
cgit v1.2.3


From c5a4bcd0cac546c5d776af881c5e913ba4a9922d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 6 Feb 2009 17:22:05 +0100
Subject: ALSA: hda - Use digital beep for AD codecs

Use digital beep instead of analog pc-beep for AD codecs.
Create the beep mixer controls dynamically on demand.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c | 140 +++++++++++++++++++++++++++----------------
 1 file changed, 88 insertions(+), 52 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 30399cbf819..cc02f2df251 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -27,11 +27,12 @@
 #include <sound/core.h>
 #include "hda_codec.h"
 #include "hda_local.h"
+#include "hda_beep.h"
 
 struct ad198x_spec {
 	struct snd_kcontrol_new *mixers[5];
 	int num_mixers;
-
+	unsigned int beep_amp;	/* beep amp value, set via set_beep_amp() */
 	const struct hda_verb *init_verbs[5];	/* initialization verbs
 						 * don't forget NULL termination!
 						 */
@@ -154,6 +155,16 @@ static const char *ad_slave_sws[] = {
 
 static void ad198x_free_kctls(struct hda_codec *codec);
 
+/* additional beep mixers; the actual parameters are overwritten at build */
+static struct snd_kcontrol_new ad_beep_mixer[] = {
+	HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Beep Playback Switch", 0, 0, HDA_OUTPUT),
+	{ } /* end */
+};
+
+#define set_beep_amp(spec, nid, idx, dir) \
+	((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir)) /* mono */
+
 static int ad198x_build_controls(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec = codec->spec;
@@ -181,6 +192,21 @@ static int ad198x_build_controls(struct hda_codec *codec)
 			return err;
 	}
 
+	/* create beep controls if needed */
+	if (spec->beep_amp) {
+		struct snd_kcontrol_new *knew;
+		for (knew = ad_beep_mixer; knew->name; knew++) {
+			struct snd_kcontrol *kctl;
+			kctl = snd_ctl_new1(knew, codec);
+			if (!kctl)
+				return -ENOMEM;
+			kctl->private_value = spec->beep_amp;
+			err = snd_hda_ctl_add(codec, kctl);
+			if (err < 0)
+				return err;
+		}
+	}
+
 	/* if we have no master control, let's create it */
 	if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) {
 		unsigned int vmaster_tlv[4];
@@ -397,7 +423,8 @@ static void ad198x_free(struct hda_codec *codec)
 		return;
 
 	ad198x_free_kctls(codec);
-	kfree(codec->spec);
+	kfree(spec);
+	snd_hda_detach_beep_device(codec);
 }
 
 static struct hda_codec_ops ad198x_patch_ops = {
@@ -536,8 +563,6 @@ static struct snd_kcontrol_new ad1986a_mixers[] = {
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x18, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x18, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
@@ -601,8 +626,7 @@ static struct snd_kcontrol_new ad1986a_laptop_mixers[] = {
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT),
-	/* HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x18, 0x0, HDA_OUTPUT),
-	   HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x18, 0x0, HDA_OUTPUT),
+	/* 
 	   HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT),
 	   HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT), */
 	HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
@@ -800,8 +824,6 @@ static struct snd_kcontrol_new ad1986a_laptop_automute_mixers[] = {
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x18, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x18, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT),
 	{
@@ -1026,7 +1048,7 @@ static int is_jack_available(struct hda_codec *codec, hda_nid_t nid)
 static int patch_ad1986a(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
-	int board_config;
+	int err, board_config;
 
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -1034,6 +1056,13 @@ static int patch_ad1986a(struct hda_codec *codec)
 
 	codec->spec = spec;
 
+	err = snd_hda_attach_beep_device(codec, 0x19);
+	if (err < 0) {
+		ad198x_free(codec);
+		return err;
+	}
+	set_beep_amp(spec, 0x18, 0, HDA_OUTPUT);
+
 	spec->multiout.max_channels = 6;
 	spec->multiout.num_dacs = ARRAY_SIZE(ad1986a_dac_nids);
 	spec->multiout.dac_nids = ad1986a_dac_nids;
@@ -1213,8 +1242,6 @@ static struct snd_kcontrol_new ad1983_mixers[] = {
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Line Playback Volume", 0x13, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Line Playback Switch", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("PC Speaker Playback Volume", 0x10, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("PC Speaker Playback Switch", 0x10, 1, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Mic Boost", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
@@ -1285,6 +1312,7 @@ static struct hda_amp_list ad1983_loopbacks[] = {
 static int patch_ad1983(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
+	int err;
 
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -1292,6 +1320,13 @@ static int patch_ad1983(struct hda_codec *codec)
 
 	codec->spec = spec;
 
+	err = snd_hda_attach_beep_device(codec, 0x10);
+	if (err < 0) {
+		ad198x_free(codec);
+		return err;
+	}
+	set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
+
 	spec->multiout.max_channels = 2;
 	spec->multiout.num_dacs = ARRAY_SIZE(ad1983_dac_nids);
 	spec->multiout.dac_nids = ad1983_dac_nids;
@@ -1361,8 +1396,6 @@ static struct snd_kcontrol_new ad1981_mixers[] = {
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x1c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("PC Speaker Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("PC Speaker Playback Switch", 0x0d, 1, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x08, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
@@ -1685,7 +1718,7 @@ static struct snd_pci_quirk ad1981_cfg_tbl[] = {
 static int patch_ad1981(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
-	int board_config;
+	int err, board_config;
 
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -1693,6 +1726,13 @@ static int patch_ad1981(struct hda_codec *codec)
 
 	codec->spec = spec;
 
+	err = snd_hda_attach_beep_device(codec, 0x10);
+	if (err < 0) {
+		ad198x_free(codec);
+		return err;
+	}
+	set_beep_amp(spec, 0x0d, 0, HDA_OUTPUT);
+
 	spec->multiout.max_channels = 2;
 	spec->multiout.num_dacs = ARRAY_SIZE(ad1981_dac_nids);
 	spec->multiout.dac_nids = ad1981_dac_nids;
@@ -1979,9 +2019,6 @@ static struct snd_kcontrol_new ad1988_6stack_mixers2[] = {
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT),
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT),
 
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT),
-
 	HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT),
 
@@ -2025,9 +2062,6 @@ static struct snd_kcontrol_new ad1988_3stack_mixers2[] = {
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT),
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT),
 
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT),
-
 	HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT),
 
@@ -2057,9 +2091,6 @@ static struct snd_kcontrol_new ad1988_laptop_mixers[] = {
 	HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT),
 	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT),
 
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT),
-
 	HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT),
 
@@ -2919,7 +2950,7 @@ static struct snd_pci_quirk ad1988_cfg_tbl[] = {
 static int patch_ad1988(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
-	int board_config;
+	int err, board_config;
 
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -2939,7 +2970,7 @@ static int patch_ad1988(struct hda_codec *codec)
 
 	if (board_config == AD1988_AUTO) {
 		/* automatic parse from the BIOS config */
-		int err = ad1988_parse_auto_config(codec);
+		err = ad1988_parse_auto_config(codec);
 		if (err < 0) {
 			ad198x_free(codec);
 			return err;
@@ -2949,6 +2980,13 @@ static int patch_ad1988(struct hda_codec *codec)
 		}
 	}
 
+	err = snd_hda_attach_beep_device(codec, 0x10);
+	if (err < 0) {
+		ad198x_free(codec);
+		return err;
+	}
+	set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
+
 	switch (board_config) {
 	case AD1988_6STACK:
 	case AD1988_6STACK_DIG:
@@ -3105,12 +3143,6 @@ static struct snd_kcontrol_new ad1884_base_mixers[] = {
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT),
-	/*
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x20, 0x03, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x20, 0x03, HDA_INPUT),
-	HDA_CODEC_VOLUME("Digital Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Digital Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT),
-	*/
 	HDA_CODEC_VOLUME("Mic Boost", 0x15, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
@@ -3219,7 +3251,7 @@ static const char *ad1884_slave_vols[] = {
 	"CD Playback Volume",
 	"Internal Mic Playback Volume",
 	"Docking Mic Playback Volume"
-	"Beep Playback Volume",
+	/* "Beep Playback Volume", */
 	"IEC958 Playback Volume",
 	NULL
 };
@@ -3227,6 +3259,7 @@ static const char *ad1884_slave_vols[] = {
 static int patch_ad1884(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
+	int err;
 
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -3234,6 +3267,13 @@ static int patch_ad1884(struct hda_codec *codec)
 
 	codec->spec = spec;
 
+	err = snd_hda_attach_beep_device(codec, 0x10);
+	if (err < 0) {
+		ad198x_free(codec);
+		return err;
+	}
+	set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
+
 	spec->multiout.max_channels = 2;
 	spec->multiout.num_dacs = ARRAY_SIZE(ad1884_dac_nids);
 	spec->multiout.dac_nids = ad1884_dac_nids;
@@ -3300,8 +3340,6 @@ static struct snd_kcontrol_new ad1984_thinkpad_mixers[] = {
 	HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Internal Mic Boost", 0x15, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Docking Mic Boost", 0x25, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT),
 	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
@@ -3358,10 +3396,6 @@ static struct snd_kcontrol_new ad1984_dell_desktop_mixers[] = {
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
 	HDA_CODEC_VOLUME("Line-In Playback Volume", 0x20, 0x01, HDA_INPUT),
 	HDA_CODEC_MUTE("Line-In Playback Switch", 0x20, 0x01, HDA_INPUT),
-	/*
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x20, 0x03, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x20, 0x03, HDA_INPUT),
-	*/
 	HDA_CODEC_VOLUME("Line-In Boost", 0x15, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
@@ -3540,8 +3574,6 @@ static struct snd_kcontrol_new ad1884a_base_mixers[] = {
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Line Boost", 0x15, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Boost", 0x25, 0x0, HDA_OUTPUT),
@@ -3674,8 +3706,6 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = {
 	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
 	HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
 	HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Internal Mic Boost", 0x15, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Dock Mic Boost", 0x25, 0x0, HDA_OUTPUT),
@@ -3703,8 +3733,6 @@ static struct snd_kcontrol_new ad1884a_mobile_mixers[] = {
 	HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
 	HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Capture Volume", 0x14, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x15, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
@@ -3815,8 +3843,6 @@ static struct snd_kcontrol_new ad1984a_thinkpad_mixers[] = {
 	HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Internal Mic Boost", 0x17, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
@@ -3902,7 +3928,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
 static int patch_ad1884a(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
-	int board_config;
+	int err, board_config;
 
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -3910,6 +3936,13 @@ static int patch_ad1884a(struct hda_codec *codec)
 
 	codec->spec = spec;
 
+	err = snd_hda_attach_beep_device(codec, 0x10);
+	if (err < 0) {
+		ad198x_free(codec);
+		return err;
+	}
+	set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
+
 	spec->multiout.max_channels = 2;
 	spec->multiout.num_dacs = ARRAY_SIZE(ad1884a_dac_nids);
 	spec->multiout.dac_nids = ad1884a_dac_nids;
@@ -4064,8 +4097,6 @@ static struct snd_kcontrol_new ad1882_loopback_mixers[] = {
 	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x04, HDA_INPUT),
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT),
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x07, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x07, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -4078,8 +4109,6 @@ static struct snd_kcontrol_new ad1882a_loopback_mixers[] = {
 	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x01, HDA_INPUT),
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT),
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x07, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x07, HDA_INPUT),
 	HDA_CODEC_VOLUME("Digital Mic Boost", 0x1f, 0x0, HDA_INPUT),
 	{ } /* end */
 };
@@ -4238,7 +4267,7 @@ static const char *ad1882_models[AD1986A_MODELS] = {
 static int patch_ad1882(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
-	int board_config;
+	int err, board_config;
 
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -4246,6 +4275,13 @@ static int patch_ad1882(struct hda_codec *codec)
 
 	codec->spec = spec;
 
+	err = snd_hda_attach_beep_device(codec, 0x10);
+	if (err < 0) {
+		ad198x_free(codec);
+		return err;
+	}
+	set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
+
 	spec->multiout.max_channels = 6;
 	spec->multiout.num_dacs = 3;
 	spec->multiout.dac_nids = ad1882_dac_nids;
-- 
cgit v1.2.3


From cfb9fb5517faa9e61c7e874fc89ef9c9253a0902 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 6 Feb 2009 17:34:03 +0100
Subject: ALSA: hda - Fix unused variable compile warning
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

  sound/pci/hda/patch_realtek.c:12693: warning: unused variable ‘i’

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 3d933e307b1..f594a096029 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -12690,7 +12690,7 @@ static int alc269_auto_create_analog_input_ctls(struct alc_spec *spec,
 static int alc269_parse_auto_config(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	int i, err;
+	int err;
 	static hda_nid_t alc269_ignore[] = { 0x1d, 0 };
 
 	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
-- 
cgit v1.2.3


From 8663ae55f39e99c25242adb6242a191258a4eca1 Mon Sep 17 00:00:00 2001
From: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Date: Sun, 8 Feb 2009 19:50:34 -0200
Subject: ALSA: hda - Bind new ecs mobo id (1019:2950) to model=ecs202

This adds a new sound quirk entry (model=ecs202) for an ecs motherboard
with IDT STAC9221 codec (1019:2950).

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 85dc642d113..d16d5c60eec 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2108,6 +2108,8 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = {
 		      "ECS/PC chips", STAC_ECS_202),
 	SND_PCI_QUIRK(0x1019, 0x2820,
 		      "ECS/PC chips", STAC_ECS_202),
+	SND_PCI_QUIRK(0x1019, 0x2950,
+		      "ECS/PC chips", STAC_ECS_202),
 	{} /* terminator */
 };
 
-- 
cgit v1.2.3


From 23c7b521c250b261dd97a7a06d5a2e74b56233d5 Mon Sep 17 00:00:00 2001
From: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Date: Sun, 8 Feb 2009 19:51:28 -0200
Subject: ALSA: hda - Don't touch non-existent port f on 4-port 92hd71bxx
 codecs

When checking for input amps on pins 0x0a, 0x0d and 0x0f, and
initializing them for 92hd71xxx codec models, we must skip nid 0x0f
for 4-port models too like with 5-port models, as it is unused
(nid 0x0f is vendor reserved in 4-port models).

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index d16d5c60eec..2f4e090b055 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -5072,6 +5072,8 @@ again:
 	switch (codec->vendor_id) {
 	case 0x111d76b6: /* 4 Port without Analog Mixer */
 	case 0x111d76b7:
+		unmute_init++;
+		/* fallthru */
 	case 0x111d76b4: /* 6 Port without Analog Mixer */
 	case 0x111d76b5:
 		memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_nomixer,
-- 
cgit v1.2.3


From dea0a5095b5e21306a81c496567043798fac7815 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 9 Feb 2009 17:14:52 +0100
Subject: ALSA: hda - Clean up quirk lists

Clean up quirk lists with bit masks.
Also, sorted in numerical order for alc662_cfg_tbl[].

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c   | 10 ++---
 sound/pci/hda/patch_conexant.c | 20 +++------
 sound/pci/hda/patch_realtek.c  | 97 ++++++++++++++++++++----------------------
 sound/pci/hda/patch_sigmatel.c | 61 ++++----------------------
 4 files changed, 65 insertions(+), 123 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index cc02f2df251..6106dfe8ec0 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -1015,10 +1015,8 @@ static struct snd_pci_quirk ad1986a_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba", AD1986A_LAPTOP_EAPD),
 	SND_PCI_QUIRK(0x144d, 0xb03c, "Samsung R55", AD1986A_3STACK),
 	SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_LAPTOP),
-	SND_PCI_QUIRK(0x144d, 0xc023, "Samsung X60", AD1986A_SAMSUNG),
-	SND_PCI_QUIRK(0x144d, 0xc024, "Samsung R65", AD1986A_SAMSUNG),
-	SND_PCI_QUIRK(0x144d, 0xc026, "Samsung X11", AD1986A_SAMSUNG),
 	SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_ULTRA),
+	SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_SAMSUNG),
 	SND_PCI_QUIRK(0x144d, 0xc504, "Samsung Q35", AD1986A_3STACK),
 	SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_LAPTOP),
 	SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_3STACK),
@@ -1706,10 +1704,10 @@ static struct snd_pci_quirk ad1981_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1014, 0x0597, "Lenovo Z60", AD1981_THINKPAD),
 	SND_PCI_QUIRK(0x1014, 0x05b7, "Lenovo Z60m", AD1981_THINKPAD),
 	/* All HP models */
-	SND_PCI_QUIRK(0x103c, 0, "HP nx", AD1981_HP),
+	SND_PCI_QUIRK_VENDOR(0x103c, "HP nx", AD1981_HP),
 	SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba U205", AD1981_TOSHIBA),
 	/* Lenovo Thinkpad T60/X60/Z6xx */
-	SND_PCI_QUIRK(0x17aa, 0, "Lenovo Thinkpad", AD1981_THINKPAD),
+	SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1981_THINKPAD),
 	/* HP nx6320 (reversed SSID, H/W bug) */
 	SND_PCI_QUIRK(0x30b0, 0x103c, "HP nx6320", AD1981_HP),
 	{}
@@ -3481,7 +3479,7 @@ static const char *ad1984_models[AD1984_MODELS] = {
 
 static struct snd_pci_quirk ad1984_cfg_tbl[] = {
 	/* Lenovo Thinkpad T61/X61 */
-	SND_PCI_QUIRK(0x17aa, 0, "Lenovo Thinkpad", AD1984_THINKPAD),
+	SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1984_THINKPAD),
 	SND_PCI_QUIRK(0x1028, 0x0214, "Dell T3400", AD1984_DELL_DESKTOP),
 	{}
 };
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 0177ef8f4c9..fdf876be712 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1002,15 +1002,9 @@ static const char *cxt5045_models[CXT5045_MODELS] = {
 };
 
 static struct snd_pci_quirk cxt5045_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x103c, 0x30a5, "HP", CXT5045_LAPTOP_HPSENSE),
-	SND_PCI_QUIRK(0x103c, 0x30b2, "HP DV Series", CXT5045_LAPTOP_HPSENSE),
-	SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2120", CXT5045_LAPTOP_HPSENSE),
-	SND_PCI_QUIRK(0x103c, 0x30b7, "HP DV6000Z", CXT5045_LAPTOP_HPSENSE),
-	SND_PCI_QUIRK(0x103c, 0x30bb, "HP DV8000", CXT5045_LAPTOP_HPSENSE),
-	SND_PCI_QUIRK(0x103c, 0x30cd, "HP DV Series", CXT5045_LAPTOP_HPSENSE),
-	SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV9533EG", CXT5045_LAPTOP_HPSENSE),
 	SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP_HP530),
-	SND_PCI_QUIRK(0x103c, 0x30d9, "HP Spartan", CXT5045_LAPTOP_HPSENSE),
+	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP DV Series",
+			   CXT5045_LAPTOP_HPSENSE),
 	SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P105", CXT5045_LAPTOP_MICSENSE),
 	SND_PCI_QUIRK(0x152d, 0x0753, "Benq R55E", CXT5045_BENQ),
 	SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_LAPTOP_MICSENSE),
@@ -1020,8 +1014,8 @@ static struct snd_pci_quirk cxt5045_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1509, 0x1e40, "FIC", CXT5045_LAPTOP_HPMICSENSE),
 	SND_PCI_QUIRK(0x1509, 0x2f05, "FIC", CXT5045_LAPTOP_HPMICSENSE),
 	SND_PCI_QUIRK(0x1509, 0x2f06, "FIC", CXT5045_LAPTOP_HPMICSENSE),
-	SND_PCI_QUIRK(0x1631, 0xc106, "Packard Bell", CXT5045_LAPTOP_HPMICSENSE),
-	SND_PCI_QUIRK(0x1631, 0xc107, "Packard Bell", CXT5045_LAPTOP_HPMICSENSE),
+	SND_PCI_QUIRK_MASK(0x1631, 0xff00, 0xc100, "Packard Bell",
+			   CXT5045_LAPTOP_HPMICSENSE),
 	SND_PCI_QUIRK(0x8086, 0x2111, "Conexant Reference board", CXT5045_LAPTOP_HPSENSE),
 	{}
 };
@@ -1571,11 +1565,9 @@ static const char *cxt5047_models[CXT5047_MODELS] = {
 };
 
 static struct snd_pci_quirk cxt5047_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x103c, 0x30a0, "HP DV1000", CXT5047_LAPTOP),
 	SND_PCI_QUIRK(0x103c, 0x30a5, "HP DV5200T/DV8000T", CXT5047_LAPTOP_HP),
-	SND_PCI_QUIRK(0x103c, 0x30b2, "HP DV2000T/DV3000T", CXT5047_LAPTOP),
-	SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2000Z", CXT5047_LAPTOP),
-	SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6700", CXT5047_LAPTOP),
+	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP DV Series",
+			   CXT5047_LAPTOP),
 	SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P100", CXT5047_LAPTOP_EAPD),
 	{}
 };
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index f594a096029..7ae8fad0189 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3598,7 +3598,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x8181, "ASUS P4GPL", ALC880_ASUS_DIG),
 	SND_PCI_QUIRK(0x1043, 0x8196, "ASUS P5GD1", ALC880_6ST),
 	SND_PCI_QUIRK(0x1043, 0x81b4, "ASUS", ALC880_6ST),
-	SND_PCI_QUIRK(0x1043, 0, "ASUS", ALC880_ASUS), /* default ASUS */
+	SND_PCI_QUIRK_VENDOR(0x1043, "ASUS", ALC880_ASUS), /* default ASUS */
 	SND_PCI_QUIRK(0x104d, 0x81a0, "Sony", ALC880_3ST),
 	SND_PCI_QUIRK(0x104d, 0x81d6, "Sony", ALC880_3ST),
 	SND_PCI_QUIRK(0x107b, 0x3032, "Gateway", ALC880_5ST),
@@ -3641,7 +3641,8 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x8086, 0xe400, "Intel mobo", ALC880_5ST_DIG),
 	SND_PCI_QUIRK(0x8086, 0xe401, "Intel mobo", ALC880_5ST_DIG),
 	SND_PCI_QUIRK(0x8086, 0xe402, "Intel mobo", ALC880_5ST_DIG),
-	SND_PCI_QUIRK(0x8086, 0, "Intel mobo", ALC880_3ST), /* default Intel */
+	/* default Intel */
+	SND_PCI_QUIRK_VENDOR(0x8086, "Intel mobo", ALC880_3ST),
 	SND_PCI_QUIRK(0xa0a0, 0x0560, "AOpen i915GMm-HFS", ALC880_5ST_DIG),
 	SND_PCI_QUIRK(0xe803, 0x1019, NULL, ALC880_6ST_DIG),
 	{}
@@ -8521,7 +8522,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 		ALC888_ACER_ASPIRE_4930G),
 	SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G",
 		ALC888_ACER_ASPIRE_4930G),
-	SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), /* default Acer */
+	/* default Acer */
+	SND_PCI_QUIRK_VENDOR(0x1025, "Acer laptop", ALC883_ACER),
 	SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL),
 	SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP),
@@ -8566,7 +8568,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720),
 	SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720),
-	SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD),
+	SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC883_LAPTOP_EAPD),
 	SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch),
 	SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION),
 	SND_PCI_QUIRK(0x1734, 0x1107, "FSC AMILO Xi2550",
@@ -10707,14 +10709,10 @@ static const char *alc262_models[ALC262_MODEL_LAST] = {
 static struct snd_pci_quirk alc262_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1002, 0x437b, "Hippo", ALC262_HIPPO),
 	SND_PCI_QUIRK(0x1033, 0x8895, "NEC Versa S9100", ALC262_NEC),
-	SND_PCI_QUIRK(0x103c, 0x12fe, "HP xw9400", ALC262_HP_BPC),
-	SND_PCI_QUIRK(0x103c, 0x12ff, "HP xw4550", ALC262_HP_BPC),
-	SND_PCI_QUIRK(0x103c, 0x1306, "HP xw8600", ALC262_HP_BPC),
-	SND_PCI_QUIRK(0x103c, 0x1307, "HP xw6600", ALC262_HP_BPC),
-	SND_PCI_QUIRK(0x103c, 0x1308, "HP xw4600", ALC262_HP_BPC),
-	SND_PCI_QUIRK(0x103c, 0x1309, "HP xw4*00", ALC262_HP_BPC),
-	SND_PCI_QUIRK(0x103c, 0x130a, "HP xw6*00", ALC262_HP_BPC),
-	SND_PCI_QUIRK(0x103c, 0x130b, "HP xw8*00", ALC262_HP_BPC),
+	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1200, "HP xw series",
+			   ALC262_HP_BPC),
+	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1300, "HP xw series",
+			   ALC262_HP_BPC),
 	SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL),
 	SND_PCI_QUIRK(0x103c, 0x2801, "HP D7000", ALC262_HP_BPC_D7000_WF),
 	SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL),
@@ -10742,8 +10740,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU),
 	SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU),
 	SND_PCI_QUIRK(0x10f1, 0x2915, "Tyan Thunder n6650W", ALC262_TYAN),
-	SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA),
-	SND_PCI_QUIRK(0x144d, 0xc039, "Samsung Q1U EL", ALC262_ULTRA),
+	SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc032, "Samsung Q1",
+			   ALC262_ULTRA),
 	SND_PCI_QUIRK(0x144d, 0xc510, "Samsung Q45", ALC262_HIPPO),
 	SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 y410", ALC262_LENOVO_3000),
 	SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8),
@@ -14534,9 +14532,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1179, 0xff03, "Toshiba P205", ALC861VD_LENOVO),
 	SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba L30-149", ALC861VD_DALLAS),
 	SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG),
-	SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo", ALC861VD_LENOVO),
-	SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo 3000 C200", ALC861VD_LENOVO),
-	SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 N200", ALC861VD_LENOVO),
+	SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", ALC861VD_LENOVO),
 	SND_PCI_QUIRK(0x1849, 0x0862, "ASRock K8NF6G-VSTA", ALC861VD_6ST_DIG),
 	{}
 };
@@ -16150,56 +16146,55 @@ static const char *alc662_models[ALC662_MODEL_LAST] = {
 };
 
 static struct snd_pci_quirk alc662_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M51VA", ALC663_ASUS_M51VA),
-	SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS G50V", ALC663_ASUS_G50V),
-	SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG),
-	SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701),
-	SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20),
-	SND_PCI_QUIRK(0x1043, 0x1903, "ASUS F5GL", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M50Vr", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_ECS),
 	SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS F7Z", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x1953, "ASUS NB", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS NB", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC663_ASUS_MODE3),
+	SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS M70V", ALC663_ASUS_MODE3),
 	SND_PCI_QUIRK(0x1043, 0x11d3, "ASUS NB", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_ASUS_MODE2),
 	SND_PCI_QUIRK(0x1043, 0x1203, "ASUS NB", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x19e3, "ASUS NB", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x1993, "ASUS N20", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x19c3, "ASUS F5Z/F6x", ALC662_ASUS_MODE2),
 	SND_PCI_QUIRK(0x1043, 0x1339, "ASUS NB", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x1913, "ASUS NB", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x1843, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC663_ASUS_MODE6),
+	SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC663_ASUS_MODE6),
+	SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_ASUS_MODE2),
 	SND_PCI_QUIRK(0x1043, 0x1813, "ASUS NB", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x1876, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1823, "ASUS NB", ALC663_ASUS_MODE5),
+	SND_PCI_QUIRK(0x1043, 0x1833, "ASUS NB", ALC663_ASUS_MODE6),
+	SND_PCI_QUIRK(0x1043, 0x1843, "ASUS NB", ALC662_ASUS_MODE2),
 	SND_PCI_QUIRK(0x1043, 0x1864, "ASUS NB", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x1933, "ASUS F80Q", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1876, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M51VA", ALC663_ASUS_M51VA),
+	/*SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M50Vr", ALC663_ASUS_MODE1),*/
 	SND_PCI_QUIRK(0x1043, 0x1893, "ASUS M50Vm", ALC663_ASUS_MODE3),
-	SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS M70V", ALC663_ASUS_MODE3),
-	SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71C", ALC663_ASUS_MODE3),
 	SND_PCI_QUIRK(0x1043, 0x1894, "ASUS X55", ALC663_ASUS_MODE3),
-	SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC663_ASUS_MODE3),
+	SND_PCI_QUIRK(0x1043, 0x1903, "ASUS F5GL", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1913, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1933, "ASUS F80Q", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1953, "ASUS NB", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71C", ALC663_ASUS_MODE3),
+	SND_PCI_QUIRK(0x1043, 0x1993, "ASUS N20", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS G50V", ALC663_ASUS_G50V),
+	/*SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS NB", ALC663_ASUS_MODE1),*/
+	SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS F7Z", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x19c3, "ASUS F5Z/F6x", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x19e3, "ASUS NB", ALC663_ASUS_MODE1),
 	SND_PCI_QUIRK(0x1043, 0x19f3, "ASUS NB", ALC663_ASUS_MODE4),
-	SND_PCI_QUIRK(0x1043, 0x1823, "ASUS NB", ALC663_ASUS_MODE5),
-	SND_PCI_QUIRK(0x1043, 0x1833, "ASUS NB", ALC663_ASUS_MODE6),
-	SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC663_ASUS_MODE6),
-	SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC663_ASUS_MODE6),
+	SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG),
+	SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701),
+	SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20),
+	SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS),
 	SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K",
 		      ALC662_3ST_6ch_DIG),
-	SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E),
-	SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_ECS),
-	SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS),
 	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L",
 		      ALC662_3ST_6ch_DIG),
 	SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG),
+	SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E),
 	SND_PCI_QUIRK(0x1849, 0x3662, "ASROCK K10N78FullHD-hSLI R3.0",
 					ALC662_3ST_6ch_DIG),
-	SND_PCI_QUIRK(0x1854, 0x2000, "ASUS H13-2000", ALC663_ASUS_H13),
-	SND_PCI_QUIRK(0x1854, 0x2001, "ASUS H13-2001", ALC663_ASUS_H13),
-	SND_PCI_QUIRK(0x1854, 0x2002, "ASUS H13-2002", ALC663_ASUS_H13),
+	SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x",
+			   ALC663_ASUS_H13),
 	{}
 };
 
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 2f4e090b055..12b30884843 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2082,33 +2082,7 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = {
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d7,
 		      "Dell XPS M1210", STAC_922X_DELL_M82),
 	/* ECS/PC Chips boards */
-	SND_PCI_QUIRK(0x1019, 0x2144,
-		      "ECS/PC chips", STAC_ECS_202),
-	SND_PCI_QUIRK(0x1019, 0x2608,
-		      "ECS/PC chips", STAC_ECS_202),
-	SND_PCI_QUIRK(0x1019, 0x2633,
-		      "ECS/PC chips P17G/1333", STAC_ECS_202),
-	SND_PCI_QUIRK(0x1019, 0x2811,
-		      "ECS/PC chips", STAC_ECS_202),
-	SND_PCI_QUIRK(0x1019, 0x2812,
-		      "ECS/PC chips", STAC_ECS_202),
-	SND_PCI_QUIRK(0x1019, 0x2813,
-		      "ECS/PC chips", STAC_ECS_202),
-	SND_PCI_QUIRK(0x1019, 0x2814,
-		      "ECS/PC chips", STAC_ECS_202),
-	SND_PCI_QUIRK(0x1019, 0x2815,
-		      "ECS/PC chips", STAC_ECS_202),
-	SND_PCI_QUIRK(0x1019, 0x2816,
-		      "ECS/PC chips", STAC_ECS_202),
-	SND_PCI_QUIRK(0x1019, 0x2817,
-		      "ECS/PC chips", STAC_ECS_202),
-	SND_PCI_QUIRK(0x1019, 0x2818,
-		      "ECS/PC chips", STAC_ECS_202),
-	SND_PCI_QUIRK(0x1019, 0x2819,
-		      "ECS/PC chips", STAC_ECS_202),
-	SND_PCI_QUIRK(0x1019, 0x2820,
-		      "ECS/PC chips", STAC_ECS_202),
-	SND_PCI_QUIRK(0x1019, 0x2950,
+	SND_PCI_QUIRK_MASK(0x1019, 0xf000, 0x2000,
 		      "ECS/PC chips", STAC_ECS_202),
 	{} /* terminator */
 };
@@ -2169,22 +2143,10 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = {
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x3d01, "Intel D946", STAC_D965_3ST),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0xa301, "Intel D946", STAC_D965_3ST),
 	/* 965 based 3 stack systems */
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2116, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2115, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2114, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2113, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2112, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2111, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2110, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2009, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2008, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2007, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2006, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2005, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2004, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2003, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2002, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2001, "Intel D965", STAC_D965_3ST),
+	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2100,
+			   "Intel D965", STAC_D965_3ST),
+	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2000,
+			   "Intel D965", STAC_D965_3ST),
 	/* Dell 3 stack systems */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x01f7, "Dell XPS M1730", STAC_DELL_3ST),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x01dd, "Dell Dimension E520", STAC_DELL_3ST),
@@ -2200,15 +2162,10 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = {
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x02ff, "Dell     ", STAC_DELL_BIOS),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x0209, "Dell XPS 1330", STAC_DELL_BIOS),
 	/* 965 based 5 stack systems */
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2301, "Intel D965", STAC_D965_5ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2302, "Intel D965", STAC_D965_5ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2303, "Intel D965", STAC_D965_5ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2304, "Intel D965", STAC_D965_5ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2305, "Intel D965", STAC_D965_5ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2501, "Intel D965", STAC_D965_5ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2502, "Intel D965", STAC_D965_5ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2503, "Intel D965", STAC_D965_5ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2504, "Intel D965", STAC_D965_5ST),
+	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2300,
+			   "Intel D965", STAC_D965_5ST),
+	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2500,
+			   "Intel D965", STAC_D965_5ST),
 	{} /* terminator */
 };
 
-- 
cgit v1.2.3


From a85165c66c5640c37b67a94aa4e00fe45273bca1 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 9 Feb 2009 17:15:50 +0100
Subject: ALSA: via82xx - Clean up quirk list

Use SND_PCI_QUIRK_VENDOR() macro.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/via82xx.c | 18 +++++++++---------
 1 file changed, 9 insertions(+), 9 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index fc62d6380f8..a027896a220 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -2363,14 +2363,14 @@ static struct snd_pci_quirk dxs_whitelist[] __devinitdata = {
 	SND_PCI_QUIRK(0x1019, 0x0996, "ESC Mobo", VIA_DXS_48K),
 	SND_PCI_QUIRK(0x1019, 0x0a81, "ECS K7VTA3 v8.0", VIA_DXS_NO_VRA),
 	SND_PCI_QUIRK(0x1019, 0x0a85, "ECS L7VMM2", VIA_DXS_NO_VRA),
-	SND_PCI_QUIRK(0x1019, 0, "ESC K8", VIA_DXS_SRC),
+	SND_PCI_QUIRK_VENDOR(0x1019, "ESC K8", VIA_DXS_SRC),
 	SND_PCI_QUIRK(0x1019, 0xaa01, "ESC K8T890-A", VIA_DXS_SRC),
 	SND_PCI_QUIRK(0x1025, 0x0033, "Acer Inspire 1353LM", VIA_DXS_NO_VRA),
 	SND_PCI_QUIRK(0x1025, 0x0046, "Acer Aspire 1524 WLMi", VIA_DXS_SRC),
-	SND_PCI_QUIRK(0x1043, 0, "ASUS A7/A8", VIA_DXS_NO_VRA),
-	SND_PCI_QUIRK(0x1071, 0, "Diverse Notebook", VIA_DXS_NO_VRA),
+	SND_PCI_QUIRK_VENDOR(0x1043, "ASUS A7/A8", VIA_DXS_NO_VRA),
+	SND_PCI_QUIRK_VENDOR(0x1071, "Diverse Notebook", VIA_DXS_NO_VRA),
 	SND_PCI_QUIRK(0x10cf, 0x118e, "FSC Laptop", VIA_DXS_ENABLE),
-	SND_PCI_QUIRK(0x1106, 0, "ASRock", VIA_DXS_SRC),
+	SND_PCI_QUIRK_VENDOR(0x1106, "ASRock", VIA_DXS_SRC),
 	SND_PCI_QUIRK(0x1297, 0xa231, "Shuttle AK31v2", VIA_DXS_SRC),
 	SND_PCI_QUIRK(0x1297, 0xa232, "Shuttle", VIA_DXS_SRC),
 	SND_PCI_QUIRK(0x1297, 0xc160, "Shuttle Sk41G", VIA_DXS_SRC),
@@ -2378,7 +2378,7 @@ static struct snd_pci_quirk dxs_whitelist[] __devinitdata = {
 	SND_PCI_QUIRK(0x1462, 0x3800, "MSI KT266", VIA_DXS_ENABLE),
 	SND_PCI_QUIRK(0x1462, 0x7120, "MSI KT4V", VIA_DXS_ENABLE),
 	SND_PCI_QUIRK(0x1462, 0x7142, "MSI K8MM-V", VIA_DXS_ENABLE),
-	SND_PCI_QUIRK(0x1462, 0, "MSI Mobo", VIA_DXS_SRC),
+	SND_PCI_QUIRK_VENDOR(0x1462, "MSI Mobo", VIA_DXS_SRC),
 	SND_PCI_QUIRK(0x147b, 0x1401, "ABIT KD7(-RAID)", VIA_DXS_ENABLE),
 	SND_PCI_QUIRK(0x147b, 0x1411, "ABIT VA-20", VIA_DXS_ENABLE),
 	SND_PCI_QUIRK(0x147b, 0x1413, "ABIT KV8 Pro", VIA_DXS_ENABLE),
@@ -2392,11 +2392,11 @@ static struct snd_pci_quirk dxs_whitelist[] __devinitdata = {
 	SND_PCI_QUIRK(0x161f, 0x2032, "m680x machines", VIA_DXS_48K),
 	SND_PCI_QUIRK(0x1631, 0xe004, "PB EasyNote 3174", VIA_DXS_ENABLE),
 	SND_PCI_QUIRK(0x1695, 0x3005, "EPoX EP-8K9A", VIA_DXS_ENABLE),
-	SND_PCI_QUIRK(0x1695, 0, "EPoX mobo", VIA_DXS_SRC),
-	SND_PCI_QUIRK(0x16f3, 0, "Jetway K8", VIA_DXS_SRC),
-	SND_PCI_QUIRK(0x1734, 0, "FSC Laptop", VIA_DXS_SRC),
+	SND_PCI_QUIRK_VENDOR(0x1695, "EPoX mobo", VIA_DXS_SRC),
+	SND_PCI_QUIRK_VENDOR(0x16f3, "Jetway K8", VIA_DXS_SRC),
+	SND_PCI_QUIRK_VENDOR(0x1734, "FSC Laptop", VIA_DXS_SRC),
 	SND_PCI_QUIRK(0x1849, 0x3059, "ASRock K7VM2", VIA_DXS_NO_VRA),
-	SND_PCI_QUIRK(0x1849, 0, "ASRock mobo", VIA_DXS_SRC),
+	SND_PCI_QUIRK_VENDOR(0x1849, "ASRock mobo", VIA_DXS_SRC),
 	SND_PCI_QUIRK(0x1919, 0x200a, "Soltek SL-K8",  VIA_DXS_NO_VRA),
 	SND_PCI_QUIRK(0x4005, 0x4710, "MSI K7T266", VIA_DXS_SRC),
 	{ } /* terminator */
-- 
cgit v1.2.3


From 22971e3a77f193579be525a12f3ab91dbf241517 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 10 Feb 2009 11:56:44 +0100
Subject: ALSA: hda - add digital beep support for ALC268

Added the digital beep support for ALC268.  It was missing in the
last patches.

However, ALC268 has a strange pin use for widget 0x1d, which could be
used as another purpose.  So, the patch adds a check of the beep control
before creating the hook for input layer.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 28 ++++++++++++++++++++++------
 1 file changed, 22 insertions(+), 6 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7ae8fad0189..97eaf3b1d97 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -11885,7 +11885,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = {
 
 static struct alc_config_preset alc268_presets[] = {
 	[ALC267_QUANTA_IL1] = {
-		.mixers = { alc267_quanta_il1_mixer },
+		.mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer },
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc267_quanta_il1_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -11967,7 +11967,8 @@ static struct alc_config_preset alc268_presets[] = {
 	},
 	[ALC268_ACER_ASPIRE_ONE] = {
 		.mixers = { alc268_acer_aspire_one_mixer,
-				alc268_capture_alt_mixer },
+			    alc268_beep_mixer,
+			    alc268_capture_alt_mixer },
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_acer_aspire_one_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -12036,7 +12037,7 @@ static int patch_alc268(struct hda_codec *codec)
 {
 	struct alc_spec *spec;
 	int board_config;
-	int err;
+	int i, has_beep, err;
 
 	spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -12091,13 +12092,28 @@ static int patch_alc268(struct hda_codec *codec)
 
 	spec->stream_digital_playback = &alc268_pcm_digital_playback;
 
-	if (!query_amp_caps(codec, 0x1d, HDA_INPUT))
-		/* override the amp caps for beep generator */
-		snd_hda_override_amp_caps(codec, 0x1d, HDA_INPUT,
+	has_beep = 0;
+	for (i = 0; i < spec->num_mixers; i++) {
+		if (spec->mixers[i] == alc268_beep_mixer) {
+			has_beep = 1;
+			break;
+		}
+	}
+
+	if (has_beep) {
+		err = snd_hda_attach_beep_device(codec, 0x1);
+		if (err < 0) {
+			alc_free(codec);
+			return err;
+		}
+		if (!query_amp_caps(codec, 0x1d, HDA_INPUT))
+			/* override the amp caps for beep generator */
+			snd_hda_override_amp_caps(codec, 0x1d, HDA_INPUT,
 					  (0x0c << AC_AMPCAP_OFFSET_SHIFT) |
 					  (0x0c << AC_AMPCAP_NUM_STEPS_SHIFT) |
 					  (0x07 << AC_AMPCAP_STEP_SIZE_SHIFT) |
 					  (0 << AC_AMPCAP_MUTE_SHIFT));
+	}
 
 	if (!spec->adc_nids && spec->input_mux) {
 		/* check whether NID 0x07 is valid */
-- 
cgit v1.2.3


From 32d2c7fa1344ddf51886eddf31e228d139501dc6 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 11 Feb 2009 11:33:13 +0100
Subject: ALSA: hda - Fix a wrong pin check in snd_hda_parse_pin_def_config()

Fixed a wrong pin check (a typo) for debug print of digital input pin.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 93412f335dc..95f10aec7a0 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3551,7 +3551,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
 		   cfg->input_pins[AUTO_PIN_FRONT_LINE],
 		   cfg->input_pins[AUTO_PIN_CD],
 		   cfg->input_pins[AUTO_PIN_AUX]);
-	if (cfg->dig_out_pin)
+	if (cfg->dig_in_pin)
 		snd_printd("   dig-in=0x%x\n", cfg->dig_in_pin);
 
 	return 0;
-- 
cgit v1.2.3


From 0852d7a654f75d22a3c09fd7da4a3551bbb37740 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 11 Feb 2009 11:35:15 +0100
Subject: ALSA: hda - Detect multiple digital-out pins

Detect multiple digital-out pins in snd_hda_parse_pin_defconfig().
The dig_out_pin and dig_out_type fields become arrays.

The codec parser still doesn't use this multiple pins detection, though.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c      | 17 ++++++++++-------
 sound/pci/hda/hda_local.h      |  5 +++--
 sound/pci/hda/patch_analog.c   |  2 +-
 sound/pci/hda/patch_realtek.c  | 20 ++++++++++----------
 sound/pci/hda/patch_sigmatel.c | 10 +++++-----
 sound/pci/hda/patch_via.c      | 10 +++++-----
 6 files changed, 34 insertions(+), 30 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 95f10aec7a0..29eeb748561 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3423,11 +3423,13 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
 			break;
 		case AC_JACK_SPDIF_OUT:
 		case AC_JACK_DIG_OTHER_OUT:
-			cfg->dig_out_pin = nid;
-			if (loc == AC_JACK_LOC_HDMI)
-				cfg->dig_out_type = HDA_PCM_TYPE_HDMI;
-			else
-				cfg->dig_out_type = HDA_PCM_TYPE_SPDIF;
+			if (cfg->dig_outs >= ARRAY_SIZE(cfg->dig_out_pins))
+				continue;
+			cfg->dig_out_pins[cfg->dig_outs] = nid;
+			cfg->dig_out_type[cfg->dig_outs] =
+				(loc == AC_JACK_LOC_HDMI) ?
+				HDA_PCM_TYPE_HDMI : HDA_PCM_TYPE_SPDIF;
+			cfg->dig_outs++;
 			break;
 		case AC_JACK_SPDIF_IN:
 		case AC_JACK_DIG_OTHER_IN:
@@ -3541,8 +3543,9 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
 		   cfg->hp_pins[1], cfg->hp_pins[2],
 		   cfg->hp_pins[3], cfg->hp_pins[4]);
 	snd_printd("   mono: mono_out=0x%x\n", cfg->mono_out_pin);
-	if (cfg->dig_out_pin)
-		snd_printd("   dig-out=0x%x\n", cfg->dig_out_pin);
+	if (cfg->dig_outs)
+		snd_printd("   dig-out=0x%x/0x%x\n",
+			   cfg->dig_out_pins[0], cfg->dig_out_pins[1]);
 	snd_printd("   inputs: mic=0x%x, fmic=0x%x, line=0x%x, fline=0x%x,"
 		   " cd=0x%x, aux=0x%x\n",
 		   cfg->input_pins[AUTO_PIN_MIC],
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 4086491ed33..2ae6b53a462 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -355,10 +355,11 @@ struct auto_pin_cfg {
 	int line_out_type;	/* AUTO_PIN_XXX_OUT */
 	hda_nid_t hp_pins[AUTO_CFG_MAX_OUTS];
 	hda_nid_t input_pins[AUTO_PIN_LAST];
-	hda_nid_t dig_out_pin;
+	int dig_outs;
+	hda_nid_t dig_out_pins[2];
 	hda_nid_t dig_in_pin;
 	hda_nid_t mono_out_pin;
-	int dig_out_type; /* HDA_PCM_TYPE_XXX */
+	int dig_out_type[2]; /* HDA_PCM_TYPE_XXX */
 	int dig_in_type; /* HDA_PCM_TYPE_XXX */
 };
 
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 6106dfe8ec0..d58c32b5b43 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -2898,7 +2898,7 @@ static int ad1988_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_out_pin)
+	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = AD1988_SPDIF_OUT;
 	if (spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = AD1988_SPDIF_IN;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 1db99df7950..e46251bceb9 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4291,7 +4291,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_out_pin)
+	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = ALC880_DIGOUT_NID;
 	if (spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = ALC880_DIGIN_NID;
@@ -5658,7 +5658,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = 2;
 
-	if (spec->autocfg.dig_out_pin)
+	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = ALC260_DIGOUT_NID;
 	if (spec->kctls.list)
 		add_mixer(spec, spec->kctls.list);
@@ -10626,7 +10626,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec)
 	if (err < 0)
 		return err;
 	if (!spec->autocfg.line_outs) {
-		if (spec->autocfg.dig_out_pin || spec->autocfg.dig_in_pin) {
+		if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) {
 			spec->multiout.max_channels = 2;
 			spec->no_analog = 1;
 			goto dig_only;
@@ -10643,9 +10643,9 @@ static int alc262_parse_auto_config(struct hda_codec *codec)
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
  dig_only:
-	if (spec->autocfg.dig_out_pin) {
+	if (spec->autocfg.dig_outs) {
 		spec->multiout.dig_out_nid = ALC262_DIGOUT_NID;
-		spec->dig_out_type = spec->autocfg.dig_out_type;
+		spec->dig_out_type = spec->autocfg.dig_out_type[0];
 	}
 	if (spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = ALC262_DIGIN_NID;
@@ -11807,7 +11807,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
 	spec->multiout.max_channels = 2;
 
 	/* digital only support output */
-	if (spec->autocfg.dig_out_pin)
+	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = ALC268_DIGOUT_NID;
 
 	if (spec->kctls.list)
@@ -12722,7 +12722,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_out_pin)
+	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = ALC269_DIGOUT_NID;
 
 	if (spec->kctls.list)
@@ -13779,7 +13779,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_out_pin)
+	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = ALC861_DIGOUT_NID;
 
 	if (spec->kctls.list)
@@ -14881,7 +14881,7 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_out_pin)
+	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = ALC861VD_DIGOUT_NID;
 
 	if (spec->kctls.list)
@@ -16689,7 +16689,7 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_out_pin)
+	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = ALC880_DIGOUT_NID;
 
 	if (spec->kctls.list)
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 12b30884843..1882c573587 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2546,7 +2546,7 @@ static int stac92xx_build_pcms(struct hda_codec *codec)
 		codec->num_pcms++;
 		info++;
 		info->name = "STAC92xx Digital";
-		info->pcm_type = spec->autocfg.dig_out_type;
+		info->pcm_type = spec->autocfg.dig_out_type[0];
 		if (spec->multiout.dig_out_nid) {
 			info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_digital_playback;
 			info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid;
@@ -3706,7 +3706,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 	if (spec->multiout.max_channels > 2)
 		spec->surr_switch = 1;
 
-	if (spec->autocfg.dig_out_pin)
+	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = dig_out;
 	if (dig_in && spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = dig_in;
@@ -3819,7 +3819,7 @@ static int stac9200_parse_auto_config(struct hda_codec *codec)
 	if (err < 0)
 		return err;
 
-	if (spec->autocfg.dig_out_pin)
+	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = 0x05;
 	if (spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = 0x04;
@@ -4069,8 +4069,8 @@ static int stac92xx_init(struct hda_codec *codec)
 	for (i = 0; i < spec->num_dmics; i++)
 		stac92xx_auto_set_pinctl(codec, spec->dmic_nids[i],
 					AC_PINCTL_IN_EN);
-	if (cfg->dig_out_pin)
-		stac92xx_auto_set_pinctl(codec, cfg->dig_out_pin,
+	if (cfg->dig_out_pins[0])
+		stac92xx_auto_set_pinctl(codec, cfg->dig_out_pins[0],
 					 AC_PINCTL_OUT_EN);
 	if (cfg->dig_in_pin)
 		stac92xx_auto_set_pinctl(codec, cfg->dig_in_pin,
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index c761394cbe8..639b2ff510a 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1354,7 +1354,7 @@ static int vt1708_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_out_pin)
+	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = VT1708_DIGOUT_NID;
 	if (spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = VT1708_DIGIN_NID;
@@ -1827,7 +1827,7 @@ static int vt1709_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_out_pin)
+	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = VT1709_DIGOUT_NID;
 	if (spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = VT1709_DIGIN_NID;
@@ -2371,7 +2371,7 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_out_pin)
+	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = VT1708B_DIGOUT_NID;
 	if (spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = VT1708B_DIGIN_NID;
@@ -2836,7 +2836,7 @@ static int vt1708S_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_out_pin)
+	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = VT1708S_DIGOUT_NID;
 
 	spec->extra_dig_out_nid = 0x15;
@@ -3155,7 +3155,7 @@ static int vt1702_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_out_pin)
+	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = VT1702_DIGOUT_NID;
 
 	spec->extra_dig_out_nid = 0x1B;
-- 
cgit v1.2.3


From c98041f7d71890ac6aa2257d78ef175db44d2cd3 Mon Sep 17 00:00:00 2001
From: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Date: Wed, 11 Feb 2009 20:33:15 -0200
Subject: ALSA: hda - Cleanup setting of pin_configs in patch_stac927x

After commit "ALSA: hda - Fix restore of pin configs at resume for
STAC/IDT codecs", the introduced stac_save_pin_cfgs function checks
already for pins == NULL case, saving then default pin configs from
machine with stac92xx_save_bios_config_regs. So we can remove the
extra checks when stac927x_brd_tbl[spec->board_config] == NULL.

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 7 +++----
 1 file changed, 3 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 1882c573587..3c84817ccd2 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -5292,10 +5292,9 @@ static int patch_stac927x(struct hda_codec *codec)
 							stac927x_models,
 							stac927x_cfg_tbl);
  again:
-	if (spec->board_config < 0 || !stac927x_brd_tbl[spec->board_config]) {
-		if (spec->board_config < 0)
-			snd_printdd(KERN_INFO "hda_codec: Unknown model for"
-				    "STAC927x, using BIOS defaults\n");
+	if (spec->board_config < 0) {
+		snd_printdd(KERN_INFO "hda_codec: Unknown model for"
+			    "STAC927x, using BIOS defaults\n");
 		err = stac92xx_save_bios_config_regs(codec);
 	} else
 		err = stac_save_pin_cfgs(codec,
-- 
cgit v1.2.3


From e930e99500e5bd055270c668cca8bd2f33056895 Mon Sep 17 00:00:00 2001
From: Harvey Harrison <harvey.harrison@gmail.com>
Date: Wed, 11 Feb 2009 14:49:30 -0800
Subject: ALSA: echoaudio - replace uses of __constant_{endian}

The base versions handle constant folding now.

Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/echoaudio/echo3g_dsp.c    | 2 +-
 sound/pci/echoaudio/echoaudio_3g.c  | 3 +--
 sound/pci/echoaudio/echoaudio_dsp.c | 6 +++---
 sound/pci/echoaudio/gina20_dsp.c    | 4 ++--
 sound/pci/echoaudio/layla20_dsp.c   | 4 ++--
 sound/pci/echoaudio/mia_dsp.c       | 4 ++--
 sound/pci/echoaudio/midi.c          | 4 ++--
 7 files changed, 13 insertions(+), 14 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/echoaudio/echo3g_dsp.c b/sound/pci/echoaudio/echo3g_dsp.c
index 417e25add82..57967e58057 100644
--- a/sound/pci/echoaudio/echo3g_dsp.c
+++ b/sound/pci/echoaudio/echo3g_dsp.c
@@ -56,7 +56,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 	}
 
 	chip->comm_page->e3g_frq_register =
-		__constant_cpu_to_le32((E3G_MAGIC_NUMBER / 48000) - 2);
+		cpu_to_le32((E3G_MAGIC_NUMBER / 48000) - 2);
 	chip->device_id = device_id;
 	chip->subdevice_id = subdevice_id;
 	chip->bad_board = TRUE;
diff --git a/sound/pci/echoaudio/echoaudio_3g.c b/sound/pci/echoaudio/echoaudio_3g.c
index c3736bbd819..e32a7489792 100644
--- a/sound/pci/echoaudio/echoaudio_3g.c
+++ b/sound/pci/echoaudio/echoaudio_3g.c
@@ -40,8 +40,7 @@ static int check_asic_status(struct echoaudio *chip)
 	if (wait_handshake(chip))
 		return -EIO;
 
-	chip->comm_page->ext_box_status =
-		__constant_cpu_to_le32(E3G_ASIC_NOT_LOADED);
+	chip->comm_page->ext_box_status = cpu_to_le32(E3G_ASIC_NOT_LOADED);
 	chip->asic_loaded = FALSE;
 	clear_handshake(chip);
 	send_vector(chip, DSP_VC_TEST_ASIC);
diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c
index be0e18192de..4df51ef5e09 100644
--- a/sound/pci/echoaudio/echoaudio_dsp.c
+++ b/sound/pci/echoaudio/echoaudio_dsp.c
@@ -926,11 +926,11 @@ static int init_dsp_comm_page(struct echoaudio *chip)
 
 	/* Init the comm page */
 	chip->comm_page->comm_size =
-		__constant_cpu_to_le32(sizeof(struct comm_page));
+		cpu_to_le32(sizeof(struct comm_page));
 	chip->comm_page->handshake = 0xffffffff;
 	chip->comm_page->midi_out_free_count =
-		__constant_cpu_to_le32(DSP_MIDI_OUT_FIFO_SIZE);
-	chip->comm_page->sample_rate = __constant_cpu_to_le32(44100);
+		cpu_to_le32(DSP_MIDI_OUT_FIFO_SIZE);
+	chip->comm_page->sample_rate = cpu_to_le32(44100);
 	chip->sample_rate = 44100;
 
 	/* Set line levels so we don't blast any inputs on startup */
diff --git a/sound/pci/echoaudio/gina20_dsp.c b/sound/pci/echoaudio/gina20_dsp.c
index db6c952e9d7..3f1e7475fae 100644
--- a/sound/pci/echoaudio/gina20_dsp.c
+++ b/sound/pci/echoaudio/gina20_dsp.c
@@ -208,10 +208,10 @@ static int set_professional_spdif(struct echoaudio *chip, char prof)
 	DE_ACT(("set_professional_spdif %d\n", prof));
 	if (prof)
 		chip->comm_page->flags |=
-			__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
+			cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
 	else
 		chip->comm_page->flags &=
-			~__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
+			~cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
 	chip->professional_spdif = prof;
 	return update_flags(chip);
 }
diff --git a/sound/pci/echoaudio/layla20_dsp.c b/sound/pci/echoaudio/layla20_dsp.c
index ede75c6ca0f..83750e9fd7b 100644
--- a/sound/pci/echoaudio/layla20_dsp.c
+++ b/sound/pci/echoaudio/layla20_dsp.c
@@ -284,10 +284,10 @@ static int set_professional_spdif(struct echoaudio *chip, char prof)
 	DE_ACT(("set_professional_spdif %d\n", prof));
 	if (prof)
 		chip->comm_page->flags |=
-			__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
+			cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
 	else
 		chip->comm_page->flags &=
-			~__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
+			~cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
 	chip->professional_spdif = prof;
 	return update_flags(chip);
 }
diff --git a/sound/pci/echoaudio/mia_dsp.c b/sound/pci/echoaudio/mia_dsp.c
index 227386602f9..3eca16cb7f7 100644
--- a/sound/pci/echoaudio/mia_dsp.c
+++ b/sound/pci/echoaudio/mia_dsp.c
@@ -222,10 +222,10 @@ static int set_professional_spdif(struct echoaudio *chip, char prof)
 	DE_ACT(("set_professional_spdif %d\n", prof));
 	if (prof)
 		chip->comm_page->flags |=
-			__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
+			cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
 	else
 		chip->comm_page->flags &=
-			~__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
+			~cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
 	chip->professional_spdif = prof;
 	return update_flags(chip);
 }
diff --git a/sound/pci/echoaudio/midi.c b/sound/pci/echoaudio/midi.c
index 77bf2a83d99..a953d142cb4 100644
--- a/sound/pci/echoaudio/midi.c
+++ b/sound/pci/echoaudio/midi.c
@@ -44,10 +44,10 @@ static int enable_midi_input(struct echoaudio *chip, char enable)
 	if (enable) {
 		chip->mtc_state = MIDI_IN_STATE_NORMAL;
 		chip->comm_page->flags |=
-			__constant_cpu_to_le32(DSP_FLAG_MIDI_INPUT);
+			cpu_to_le32(DSP_FLAG_MIDI_INPUT);
 	} else
 		chip->comm_page->flags &=
-			~__constant_cpu_to_le32(DSP_FLAG_MIDI_INPUT);
+			~cpu_to_le32(DSP_FLAG_MIDI_INPUT);
 
 	clear_handshake(chip);
 	return send_vector(chip, DSP_VC_UPDATE_FLAGS);
-- 
cgit v1.2.3


From f1eaaeec11982c6b529d4255987fdf507a5fa69e Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 13 Feb 2009 08:16:55 +0100
Subject: ALSA: hda - Allow fixed codec-probe mask

Some devices have broken BIOS and they don't set the codec probe-bit
properly after cleared by the driver.  This makes the driver skipping
the necessary codec slots.

Since BIOS update isn't always easy, now the semantics of probe_mask
option is changed a bit.  When it contains the bit 8 (0x100), the
lower bits are used to probe that slots regardless of codec-probe bits
returned by the hardware.

For example, probe_mask=0x103 will force to probe the codec slot #0
and #1.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 24 +++++++++++++++++-------
 1 file changed, 17 insertions(+), 7 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 11e791b965f..19886e4bc82 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -381,6 +381,7 @@ struct azx {
 
 	/* HD codec */
 	unsigned short codec_mask;
+	int  codec_probe_mask; /* copied from probe_mask option */
 	struct hda_bus *bus;
 
 	/* CORB/RIRB */
@@ -1228,7 +1229,6 @@ static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] __devinitdata = {
 };
 
 static int __devinit azx_codec_create(struct azx *chip, const char *model,
-				      unsigned int codec_probe_mask,
 				      int no_init)
 {
 	struct hda_bus_template bus_temp;
@@ -1261,7 +1261,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model,
 
 	/* First try to probe all given codec slots */
 	for (c = 0; c < max_slots; c++) {
-		if ((chip->codec_mask & (1 << c)) & codec_probe_mask) {
+		if ((chip->codec_mask & (1 << c)) & chip->codec_probe_mask) {
 			if (probe_codec(chip, c) < 0) {
 				/* Some BIOSen give you wrong codec addresses
 				 * that don't exist
@@ -1285,7 +1285,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model,
 
 	/* Then create codec instances */
 	for (c = 0; c < max_slots; c++) {
-		if ((chip->codec_mask & (1 << c)) & codec_probe_mask) {
+		if ((chip->codec_mask & (1 << c)) & chip->codec_probe_mask) {
 			struct hda_codec *codec;
 			err = snd_hda_codec_new(chip->bus, c, !no_init, &codec);
 			if (err < 0)
@@ -2101,20 +2101,31 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = {
 	{}
 };
 
+#define AZX_FORCE_CODEC_MASK	0x100
+
 static void __devinit check_probe_mask(struct azx *chip, int dev)
 {
 	const struct snd_pci_quirk *q;
 
-	if (probe_mask[dev] == -1) {
+	chip->codec_probe_mask = probe_mask[dev];
+	if (chip->codec_probe_mask == -1) {
 		q = snd_pci_quirk_lookup(chip->pci, probe_mask_list);
 		if (q) {
 			printk(KERN_INFO
 			       "hda_intel: probe_mask set to 0x%x "
 			       "for device %04x:%04x\n",
 			       q->value, q->subvendor, q->subdevice);
-			probe_mask[dev] = q->value;
+			chip->codec_probe_mask = q->value;
 		}
 	}
+
+	/* check forced option */
+	if (chip->codec_probe_mask != -1 &&
+	    (chip->codec_probe_mask & AZX_FORCE_CODEC_MASK)) {
+		chip->codec_mask = chip->codec_probe_mask & 0xff;
+		printk(KERN_INFO "hda_intel: codec_mask forced to 0x%x\n",
+		       chip->codec_mask);
+	}
 }
 
 
@@ -2347,8 +2358,7 @@ static int __devinit azx_probe(struct pci_dev *pci,
 	card->private_data = chip;
 
 	/* create codec instances */
-	err = azx_codec_create(chip, model[dev], probe_mask[dev],
-			       probe_only[dev]);
+	err = azx_codec_create(chip, model[dev], probe_only[dev]);
 	if (err < 0)
 		goto out_free;
 
-- 
cgit v1.2.3


From 20db7cb0acd0ba5a3b12f686148d670294a69366 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 13 Feb 2009 08:18:48 +0100
Subject: ALSA: hda - Add forced codec-slots for ASUS W5F

ASUS W5F needs the fixed codec-slots to probe to override the BIOS
problem.

Tested-by: Giovanni Moser Frainer <giovanni@redix.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 19886e4bc82..e853e4a8bde 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2098,6 +2098,8 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x1028, 0x20ac, "Dell Studio Desktop", 0x01),
 	/* including bogus ALC268 in slot#2 that conflicts with ALC888 */
 	SND_PCI_QUIRK(0x17c0, 0x4085, "Medion MD96630", 0x01),
+	/* forced codec slots */
+	SND_PCI_QUIRK(0x1046, 0x1262, "ASUS W5F", 0x103),
 	{}
 };
 
-- 
cgit v1.2.3


From 8bb0ac5573ff0879fef511e1a80a4a4db0316daa Mon Sep 17 00:00:00 2001
From: Matthew Ranostay <mranostay@embeddedalley.com>
Date: Thu, 12 Feb 2009 16:50:01 -0500
Subject: ALSA: hda: Add STAC_DELL_S14 quirk

Add STAC_DELL_S14 quirk for new laptop series. Removed un-needed pins
in pin_nids for stac92hd83xxx. Also reorganized connection selection
code for the respective ports per quirk define.

Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 52 ++++++++++++++++++++++++++++--------------
 1 file changed, 35 insertions(+), 17 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 3c84817ccd2..1ebb36ca2e0 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -83,6 +83,7 @@ enum {
 enum {
 	STAC_92HD83XXX_REF,
 	STAC_92HD83XXX_PWR_REF,
+	STAC_DELL_S14,
 	STAC_92HD83XXX_MODELS
 };
 
@@ -480,10 +481,9 @@ static hda_nid_t stac92hd73xx_pin_nids[13] = {
 	0x14, 0x22, 0x23
 };
 
-static hda_nid_t stac92hd83xxx_pin_nids[14] = {
+static hda_nid_t stac92hd83xxx_pin_nids[10] = {
 	0x0a, 0x0b, 0x0c, 0x0d, 0x0e,
-	0x0f, 0x10, 0x11, 0x12, 0x13,
-	0x1d, 0x1e, 0x1f, 0x20
+	0x0f, 0x10, 0x11, 0x1f, 0x20,
 };
 
 #define STAC92HD71BXX_NUM_PINS 13
@@ -857,9 +857,9 @@ static struct hda_verb stac92hd73xx_10ch_core_init[] = {
 };
 
 static struct hda_verb stac92hd83xxx_core_init[] = {
-	{ 0xa, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{ 0xb, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{ 0xd, AC_VERB_SET_CONNECT_SEL, 0x1},
+	{ 0xa, AC_VERB_SET_CONNECT_SEL, 0x1},
+	{ 0xb, AC_VERB_SET_CONNECT_SEL, 0x1},
+	{ 0xd, AC_VERB_SET_CONNECT_SEL, 0x0},
 
 	/* power state controls amps */
 	{ 0x01, AC_VERB_SET_EAPD, 1 << 2},
@@ -1730,21 +1730,28 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
 	{} /* terminator */
 };
 
-static unsigned int ref92hd83xxx_pin_configs[14] = {
+static unsigned int ref92hd83xxx_pin_configs[10] = {
 	0x02214030, 0x02211010, 0x02a19020, 0x02170130,
 	0x01014050, 0x01819040, 0x01014020, 0x90a3014e,
-	0x40f000f0, 0x40f000f0, 0x40f000f0, 0x40f000f0,
 	0x01451160, 0x98560170,
 };
 
+static unsigned int dell_s14_pin_configs[10] = {
+	0x02214030, 0x02211010, 0x02a19020, 0x01014050,
+	0x40f000f0, 0x01819040, 0x40f000f0, 0x90a60160,
+	0x40f000f0, 0x40f000f0,
+};
+
 static unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = {
 	[STAC_92HD83XXX_REF] = ref92hd83xxx_pin_configs,
 	[STAC_92HD83XXX_PWR_REF] = ref92hd83xxx_pin_configs,
+	[STAC_DELL_S14] = dell_s14_pin_configs,
 };
 
 static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = {
 	[STAC_92HD83XXX_REF] = "ref",
 	[STAC_92HD83XXX_PWR_REF] = "mic-ref",
+	[STAC_DELL_S14] = "dell-s14",
 };
 
 static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
@@ -1753,6 +1760,8 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
 		      "DFI LanParty", STAC_92HD83XXX_REF),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101,
 		      "DFI LanParty", STAC_92HD83XXX_REF),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ba,
+		      "unknown Dell", STAC_DELL_S14),
 	{} /* terminator */
 };
 
@@ -4822,6 +4831,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
 	hda_nid_t conn[STAC92HD83_DAC_COUNT + 1];
 	int err;
 	int num_dacs;
+	hda_nid_t nid;
 
 	spec  = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -4840,15 +4850,6 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
 	spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids);
 	spec->multiout.dac_nids = spec->dac_nids;
 
-
-	/* set port 0xe to select the last DAC
-	 */
-	num_dacs = snd_hda_get_connections(codec, 0x0e,
-		conn, STAC92HD83_DAC_COUNT + 1) - 1;
-
-	snd_hda_codec_write_cache(codec, 0xe, 0,
-		AC_VERB_SET_CONNECT_SEL, num_dacs);
-
 	spec->init = stac92hd83xxx_core_init;
 	spec->mixer = stac92hd83xxx_mixer;
 	spec->num_pins = ARRAY_SIZE(stac92hd83xxx_pin_nids);
@@ -4900,6 +4901,23 @@ again:
 		return err;
 	}
 
+	switch (spec->board_config) {
+	case STAC_DELL_S14:
+		nid = 0xf;
+		break;
+	default:
+		nid = 0xe;
+		break;
+	}
+
+	num_dacs = snd_hda_get_connections(codec, nid,
+				conn, STAC92HD83_DAC_COUNT + 1) - 1;
+
+	/* set port X to select the last DAC
+	 */
+	snd_hda_codec_write_cache(codec, nid, 0,
+			AC_VERB_SET_CONNECT_SEL, num_dacs);
+
 	codec->patch_ops = stac92xx_patch_ops;
 
 	codec->proc_widget_hook = stac92hd_proc_hook;
-- 
cgit v1.2.3


From 27e089888fb1a3d1d13892262f9d522b03985044 Mon Sep 17 00:00:00 2001
From: Aristeu Sergio Rozanski Filho <aris@ruivo.org>
Date: Thu, 12 Feb 2009 17:50:37 -0500
Subject: ALSA: hda: add quirk for Lenovo X200 laptop dock

Currently the HP connector on X200 dock doesn't detect when a HP is connected
nor allows sound to be played using it. This patch fixes the problem by adding
a quirk for this specific model. It's possible that others have the same NID
(0x19) to report when dock HP is connected, but I don't have access to any.
Please Cc me in the reply since I'm not subscribed to alsa-devel@.

Signed-off-by: Aristeu Rozanski <aris@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 40 ++++++++++++++++++++++++++++++++++++++++
 1 file changed, 40 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index fdf876be712..b8de73ecfde 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1798,6 +1798,40 @@ static struct hda_verb cxt5051_init_verbs[] = {
 	{ } /* end */
 };
 
+static struct hda_verb cxt5051_lenovo_x200_init_verbs[] = {
+	/* Line in, Mic */
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
+	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
+	/* SPK  */
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* HP, Amp  */
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x16, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* Docking HP */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x19, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* DAC1 */
+	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Record selector: Int mic */
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44},
+	/* SPDIF route: PCM */
+	{0x1c, AC_VERB_SET_CONNECT_SEL, 0x0},
+	/* EAPD */
+	{0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
+	{0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT},
+	{0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTC_EVENT},
+	{0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT},
+	{ } /* end */
+};
+
 /* initialize jack-sensing, too */
 static int cxt5051_init(struct hda_codec *codec)
 {
@@ -1815,18 +1849,21 @@ static int cxt5051_init(struct hda_codec *codec)
 enum {
 	CXT5051_LAPTOP,	 /* Laptops w/ EAPD support */
 	CXT5051_HP,	/* no docking */
+	CXT5051_LENOVO_X200,	/* Lenovo X200 laptop */
 	CXT5051_MODELS
 };
 
 static const char *cxt5051_models[CXT5051_MODELS] = {
 	[CXT5051_LAPTOP]	= "laptop",
 	[CXT5051_HP]		= "hp",
+	[CXT5051_LENOVO_X200]	= "lenovo-x200",
 };
 
 static struct snd_pci_quirk cxt5051_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board",
 		      CXT5051_LAPTOP),
 	SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP),
+	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200),
 	{}
 };
 
@@ -1867,6 +1904,9 @@ static int patch_cxt5051(struct hda_codec *codec)
 		codec->patch_ops.unsol_event = cxt5051_hp_unsol_event;
 		spec->mixers[0] = cxt5051_hp_mixers;
 		break;
+	case CXT5051_LENOVO_X200:
+		spec->init_verbs[0] = cxt5051_lenovo_x200_init_verbs;
+		/* fallthru */
 	default:
 	case CXT5051_LAPTOP:
 		codec->patch_ops.unsol_event = cxt5051_hp_unsol_event;
-- 
cgit v1.2.3


From 946835074e026f4bbe9f3c2b091dca6346bd1474 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 13 Feb 2009 09:31:20 +0100
Subject: ALSA: hda - Add quirk for Acer AX1700-U3700A

Force model=auto for Acer AX1700-U3700A with ALC888 codec.
Since Acer devices are handlded as model=acer as default, the auto
parsing has to be specified explicitly.
(Maybe it's better rather to remove this default model=acer handling,
 though.)

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index e46251bceb9..2306cca1b69 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8520,6 +8520,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 		ALC888_ACER_ASPIRE_4930G),
 	SND_PCI_QUIRK(0x1025, 0x013f, "Acer Aspire 5930G",
 		ALC888_ACER_ASPIRE_4930G),
+	SND_PCI_QUIRK(0x1025, 0x0158, "Acer AX1700-U3700A", ALC883_AUTO),
 	SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G",
 		ALC888_ACER_ASPIRE_4930G),
 	/* default Acer */
-- 
cgit v1.2.3


From 9b5f12e5a4029c1cd03784754687faef6d9e54fa Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 13 Feb 2009 11:47:37 +0100
Subject: ALSA: hda - Add proper cleanup for multiout-dig for ALC codecs

The recent patch_realtek.c contains the slave digital-out support
as well.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 11 ++++++++++-
 1 file changed, 10 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 2306cca1b69..ef9b7ee3410 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2979,6 +2979,14 @@ static int alc880_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 					     stream_tag, format, substream);
 }
 
+static int alc880_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+					   struct hda_codec *codec,
+					   struct snd_pcm_substream *substream)
+{
+	struct alc_spec *spec = codec->spec;
+	return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout);
+}
+
 static int alc880_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
 					 struct hda_codec *codec,
 					 struct snd_pcm_substream *substream)
@@ -3062,7 +3070,8 @@ static struct hda_pcm_stream alc880_pcm_digital_playback = {
 	.ops = {
 		.open = alc880_dig_playback_pcm_open,
 		.close = alc880_dig_playback_pcm_close,
-		.prepare = alc880_dig_playback_pcm_prepare
+		.prepare = alc880_dig_playback_pcm_prepare,
+		.cleanup = alc880_dig_playback_pcm_cleanup
 	},
 };
 
-- 
cgit v1.2.3


From 6a05ac4afa90ac9c38fedd3f6940fe8da5d1fcf6 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 13 Feb 2009 11:19:09 +0100
Subject: ALSA: hda - Support multiple digital outs with auto-probing

Added the support of multiple digital outputs via auto-probing for
Realtek ALC88x codecs.  The multiple outputs are handled as slave
streams, so only one PCM stream (and the corresponding IEC958*
elements) is provided to control both digital outputs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 22 +++++++++++++++++++---
 1 file changed, 19 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ef9b7ee3410..244de597c5b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -279,6 +279,7 @@ struct alc_spec {
 					 * dig_out_nid and hp_nid are optional
 					 */
 	hda_nid_t alt_dac_nid;
+	hda_nid_t slave_dig_outs[3];	/* optional - for auto-parsing */
 	int dig_out_type;
 
 	/* capture */
@@ -4269,7 +4270,7 @@ static void alc880_auto_init_analog_input(struct hda_codec *codec)
 static int alc880_parse_auto_config(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	int err;
+	int i, err;
 	static hda_nid_t alc880_ignore[] = { 0x1d, 0 };
 
 	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
@@ -4300,8 +4301,23 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_outs)
-		spec->multiout.dig_out_nid = ALC880_DIGOUT_NID;
+	/* check multiple SPDIF-out (for recent codecs) */
+	for (i = 0; i < spec->autocfg.dig_outs; i++) {
+		hda_nid_t dig_nid;
+		err = snd_hda_get_connections(codec,
+					      spec->autocfg.dig_out_pins[i],
+					      &dig_nid, 1);
+		if (err < 0)
+			continue;
+		if (!i)
+			spec->multiout.dig_out_nid = dig_nid;
+		else {
+			spec->multiout.slave_dig_outs = spec->slave_dig_outs;
+			spec->slave_dig_outs[i - 1] = dig_nid;
+			if (i == ARRAY_SIZE(spec->slave_dig_outs) - 1)
+				break;
+		}
+	}
 	if (spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = ALC880_DIGIN_NID;
 
-- 
cgit v1.2.3


From e2ea57a8df6da45f5f63ab7b56528a552f36fb72 Mon Sep 17 00:00:00 2001
From: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Date: Mon, 16 Feb 2009 10:23:00 +0100
Subject: ALSA: hda - Fix speaker output on HP DV4 1155-SE

Force speaker pin config with model=hp-dv5 model for cases when bios
doesn't set it up properly. All reported hp laptops using model=hp-dv5
model have speaker at pin 0x0d with same config, so it's safe to add
this within hp-dv5 model.

Reference: alsa-devel mailing list thread on
    http://mailman.alsa-project.org/pipermail/alsa-devel/2009-February/014390.html

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 6 ++++++
 1 file changed, 6 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index aeb5d2126da..7320059b713 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1823,6 +1823,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
 		      "HP dv7", STAC_HP_DV5),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f7,
 		      "HP dv4", STAC_HP_DV5),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fb,
+		      "HP dv7", STAC_HP_DV5),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fc,
 		      "HP dv7", STAC_HP_M4),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3600,
@@ -5170,6 +5172,10 @@ again:
 		spec->num_smuxes = 0;
 		spec->num_dmuxes = 0;
 		break;
+	case STAC_HP_DV5:
+		stac_change_pin_config(codec, 0x0d, 0x90170010);
+		stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN);
+		break;
 	};
 
 	spec->multiout.dac_nids = spec->dac_nids;
-- 
cgit v1.2.3


From 2ae466f8cc522843fa9a456e46007dd98b052b13 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 16 Feb 2009 14:16:36 +0100
Subject: ALSA: hda - Cleanup IDT92HD7x HP quirks

Clean up IDT92HD7x quirks for HP laptops with SND_PCI_QUIRK_MASK().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 20 +++++---------------
 1 file changed, 5 insertions(+), 15 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 7320059b713..d00a211a813 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1817,22 +1817,12 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
 		      "DFI LanParty", STAC_92HD71BXX_REF),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101,
 		      "DFI LanParty", STAC_92HD71BXX_REF),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f2,
-		      "HP dv5", STAC_HP_M4),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f4,
-		      "HP dv7", STAC_HP_DV5),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f7,
-		      "HP dv4", STAC_HP_DV5),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fb,
-		      "HP dv7", STAC_HP_DV5),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fc,
-		      "HP dv7", STAC_HP_M4),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3600,
-		      "HP dv5", STAC_HP_DV5),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3603,
-		      "HP dv5", STAC_HP_DV5),
+	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x30f0,
+		      "HP dv4-7", STAC_HP_DV5),
+	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3600,
+		      "HP dv4-7", STAC_HP_DV5),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361a,
-				"unknown HP", STAC_HP_M4),
+		      "HP mini 1000", STAC_HP_M4),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233,
 				"unknown Dell", STAC_DELL_M4_1),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0234,
-- 
cgit v1.2.3


From c23127566c7a54c8413bf1b99becea76072f467e Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 16 Feb 2009 15:20:41 +0100
Subject: ALSA: hda - Clean up quirks for HP laptops with AD1984A

Clean up quirks for HP laptops with AD1984A using SND_PCI_QUIRK_MASK()

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index af6b0035e2e..2c58d7b05ab 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3923,8 +3923,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE),
 	SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP),
 	SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE),
-	SND_PCI_QUIRK(0x103c, 0x3072, "HP", AD1884A_LAPTOP),
-	SND_PCI_QUIRK(0x103c, 0x3077, "HP", AD1884A_LAPTOP),
+	SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x3070, "HP", AD1884A_MOBILE),
 	SND_PCI_QUIRK(0x103c, 0x30e6, "HP 6730b", AD1884A_LAPTOP),
 	SND_PCI_QUIRK(0x103c, 0x30e7, "HP EliteBook 8530p", AD1884A_LAPTOP),
 	SND_PCI_QUIRK(0x103c, 0x3614, "HP 6730s", AD1884A_LAPTOP),
-- 
cgit v1.2.3


From 83807400794a1d680a4fb70a610c5f486e734f45 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 17 Feb 2009 07:59:40 +0100
Subject: ALSA: au88x0 - Fix &&|| typo
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Fixed a typo of || and &&.
As it's in a disabled code section, there is no behavior change, though.

Reported-by: Jörg-Volker Peetz <jvpeetz@web.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/au88x0/au88x0_core.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c
index e6a04d037c1..3906f5afe27 100644
--- a/sound/pci/au88x0/au88x0_core.c
+++ b/sound/pci/au88x0/au88x0_core.c
@@ -2800,7 +2800,7 @@ vortex_translateformat(vortex_t * vortex, char bits, char nch, int encod)
 {
 	int a, this_194;
 
-	if ((bits != 8) || (bits != 16))
+	if ((bits != 8) && (bits != 16))
 		return -1;
 
 	switch (encod) {
-- 
cgit v1.2.3


From b3bdb30b6d1989129e297641fec791e9e555e4d8 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 18 Feb 2009 13:16:26 +0100
Subject: ALSA: hda - Add quirk for Acer X3200

Acer X3200 needs model=auto, otherwise model=acer is pre-selected.

Reference: Novell bnc#476268
	https://bugzilla.novell.com/show_bug.cgi?id=476268

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 244de597c5b..192c92a5af3 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8545,6 +8545,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 		ALC888_ACER_ASPIRE_4930G),
 	SND_PCI_QUIRK(0x1025, 0x013f, "Acer Aspire 5930G",
 		ALC888_ACER_ASPIRE_4930G),
+	SND_PCI_QUIRK(0x1025, 0x0157, "Acer X3200", ALC883_AUTO),
 	SND_PCI_QUIRK(0x1025, 0x0158, "Acer AX1700-U3700A", ALC883_AUTO),
 	SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G",
 		ALC888_ACER_ASPIRE_4930G),
-- 
cgit v1.2.3


From 07eba61dd68678e30b24b4776f59798f625e089d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 19 Feb 2009 08:06:35 +0100
Subject: ALSA: hda - Don't enable beep for digital-only ALC262

When ALC262 codec is configured as digital-only, it's meaningless to
add the digital beep input.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 13 ++++++++-----
 1 file changed, 8 insertions(+), 5 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 192c92a5af3..91da92259c8 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -11051,10 +11051,12 @@ static int patch_alc262(struct hda_codec *codec)
 		}
 	}
 
-	err = snd_hda_attach_beep_device(codec, 0x1);
-	if (err < 0) {
-		alc_free(codec);
-		return err;
+	if (!spec->no_analog) {
+		err = snd_hda_attach_beep_device(codec, 0x1);
+		if (err < 0) {
+			alc_free(codec);
+			return err;
+		}
 	}
 
 	if (board_config != ALC262_AUTO)
@@ -11087,7 +11089,8 @@ static int patch_alc262(struct hda_codec *codec)
 	}
 	if (!spec->cap_mixer && !spec->no_analog)
 		set_capture_mixer(spec);
-	set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
+	if (!spec->no_analog)
+		set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 
 	spec->vmaster_nid = 0x0c;
 
-- 
cgit v1.2.3


From ab9fec099b796b002b6996c4c5845167d8fe6dbd Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 19 Feb 2009 08:13:26 +0100
Subject: ALSA: hda - Avoid doubly beep attachment in patch_alc268()

Remove the doubly attachment in patch_alc268().
The input beep is attached conditionally only when needed.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 6 ------
 1 file changed, 6 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 91da92259c8..df32f9353e7 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -12100,12 +12100,6 @@ static int patch_alc268(struct hda_codec *codec)
 		}
 	}
 
-	err = snd_hda_attach_beep_device(codec, 0x1);
-	if (err < 0) {
-		alc_free(codec);
-		return err;
-	}
-
 	if (board_config != ALC268_AUTO)
 		setup_preset(spec, &alc268_presets[board_config]);
 
-- 
cgit v1.2.3


From 7e0e44d430281d398769f1d7864e161203252760 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 19 Feb 2009 08:15:49 +0100
Subject: ALSA: hda - Add digital-only mode for ALC268

ALC268 can be configured as digital-only, e.g. for HDMI, on some
machines.  Allow the parser to set up the digital-only mode.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 19 +++++++++++++------
 1 file changed, 13 insertions(+), 6 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index df32f9353e7..169b3837af5 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -11824,9 +11824,14 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
 					   alc268_ignore);
 	if (err < 0)
 		return err;
-	if (!spec->autocfg.line_outs)
+	if (!spec->autocfg.line_outs) {
+		if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) {
+			spec->multiout.max_channels = 2;
+			spec->no_analog = 1;
+			goto dig_only;
+		}
 		return 0; /* can't find valid BIOS pin config */
-
+	}
 	err = alc268_auto_create_multi_out_ctls(spec, &spec->autocfg);
 	if (err < 0)
 		return err;
@@ -11836,10 +11841,12 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = 2;
 
+ dig_only:
 	/* digital only support output */
-	if (spec->autocfg.dig_outs)
+	if (spec->autocfg.dig_outs) {
 		spec->multiout.dig_out_nid = ALC268_DIGOUT_NID;
-
+		spec->dig_out_type = spec->autocfg.dig_out_type[0];
+	}
 	if (spec->kctls.list)
 		add_mixer(spec, spec->kctls.list);
 
@@ -12140,7 +12147,7 @@ static int patch_alc268(struct hda_codec *codec)
 					  (0 << AC_AMPCAP_MUTE_SHIFT));
 	}
 
-	if (!spec->adc_nids && spec->input_mux) {
+	if (!spec->no_analog && !spec->adc_nids && spec->input_mux) {
 		/* check whether NID 0x07 is valid */
 		unsigned int wcap = get_wcaps(codec, 0x07);
 		int i;
@@ -12764,7 +12771,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 	if (err < 0)
 		return err;
 
-	if (!spec->cap_mixer)
+	if (!spec->cap_mixer && !spec->no_analog)
 		set_capture_mixer(spec);
 
 	store_pin_configs(codec);
-- 
cgit v1.2.3


From bb71858853a5c9616eea98512f4075d4f081154d Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Thu, 19 Feb 2009 08:37:13 +0100
Subject: sound: oxygen: make the owner module a parameter of the probe
 function

Move the owner field out of the oxygen_model structure and make it
a parameter of oxygen_pci_probe(), because the actual owner module does
not depend on the card model.  Furthermore, moving it out of the model
structure allows us to create the card structure before the actual model
is known.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/oxygen/hifier.c     | 3 +--
 sound/pci/oxygen/oxygen.c     | 3 +--
 sound/pci/oxygen/oxygen.h     | 2 +-
 sound/pci/oxygen/oxygen_lib.c | 3 ++-
 sound/pci/oxygen/virtuoso.c   | 5 +----
 5 files changed, 6 insertions(+), 10 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c
index 1ab833f843e..cc98bad9916 100644
--- a/sound/pci/oxygen/hifier.c
+++ b/sound/pci/oxygen/hifier.c
@@ -151,7 +151,6 @@ static const struct oxygen_model model_hifier = {
 	.shortname = "C-Media CMI8787",
 	.longname = "C-Media Oxygen HD Audio",
 	.chip = "CMI8788",
-	.owner = THIS_MODULE,
 	.init = hifier_init,
 	.control_filter = hifier_control_filter,
 	.cleanup = hifier_cleanup,
@@ -185,7 +184,7 @@ static int __devinit hifier_probe(struct pci_dev *pci,
 		++dev;
 		return -ENOENT;
 	}
-	err = oxygen_pci_probe(pci, index[dev], id[dev], &model_hifier, 0);
+	err = oxygen_pci_probe(pci, index[dev], id[dev], THIS_MODULE, &model_hifier, 0);
 	if (err >= 0)
 		++dev;
 	return err;
diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c
index de999c6d6dd..12b6c2137d5 100644
--- a/sound/pci/oxygen/oxygen.c
+++ b/sound/pci/oxygen/oxygen.c
@@ -315,7 +315,6 @@ static const struct oxygen_model model_generic = {
 	.shortname = "C-Media CMI8788",
 	.longname = "C-Media Oxygen HD Audio",
 	.chip = "CMI8788",
-	.owner = THIS_MODULE,
 	.probe = generic_probe,
 	.init = generic_init,
 	.cleanup = generic_cleanup,
@@ -353,7 +352,7 @@ static int __devinit generic_oxygen_probe(struct pci_dev *pci,
 		++dev;
 		return -ENOENT;
 	}
-	err = oxygen_pci_probe(pci, index[dev], id[dev],
+	err = oxygen_pci_probe(pci, index[dev], id[dev], THIS_MODULE,
 			       &model_generic, pci_id->driver_data);
 	if (err >= 0)
 		++dev;
diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h
index 19107c6307e..268bff4f29d 100644
--- a/sound/pci/oxygen/oxygen.h
+++ b/sound/pci/oxygen/oxygen.h
@@ -62,7 +62,6 @@ struct oxygen_model {
 	const char *shortname;
 	const char *longname;
 	const char *chip;
-	struct module *owner;
 	int (*probe)(struct oxygen *chip, unsigned long driver_data);
 	void (*init)(struct oxygen *chip);
 	int (*control_filter)(struct snd_kcontrol_new *template);
@@ -134,6 +133,7 @@ struct oxygen {
 /* oxygen_lib.c */
 
 int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
+		     struct module *owner,
 		     const struct oxygen_model *model,
 		     unsigned long driver_data);
 void oxygen_pci_remove(struct pci_dev *pci);
diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c
index 9c81e0b0511..b5560fa5a5e 100644
--- a/sound/pci/oxygen/oxygen_lib.c
+++ b/sound/pci/oxygen/oxygen_lib.c
@@ -452,6 +452,7 @@ static void oxygen_card_free(struct snd_card *card)
 }
 
 int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
+		     struct module *owner,
 		     const struct oxygen_model *model,
 		     unsigned long driver_data)
 {
@@ -459,7 +460,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
 	struct oxygen *chip;
 	int err;
 
-	err = snd_card_create(index, id, model->owner,
+	err = snd_card_create(index, id, owner,
 			      sizeof(*chip) + model->model_data_size, &card);
 	if (err < 0)
 		return err;
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index 6c870c12a17..c05f7e7bdb3 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -816,7 +816,6 @@ static int xonar_model_probe(struct oxygen *chip, unsigned long driver_data)
 static const struct oxygen_model model_xonar_d2 = {
 	.longname = "Asus Virtuoso 200",
 	.chip = "AV200",
-	.owner = THIS_MODULE,
 	.probe = xonar_model_probe,
 	.init = xonar_d2_init,
 	.control_filter = xonar_d2_control_filter,
@@ -849,7 +848,6 @@ static const struct oxygen_model model_xonar_d2 = {
 static const struct oxygen_model model_xonar_d1 = {
 	.longname = "Asus Virtuoso 100",
 	.chip = "AV200",
-	.owner = THIS_MODULE,
 	.probe = xonar_model_probe,
 	.init = xonar_d1_init,
 	.control_filter = xonar_d1_control_filter,
@@ -878,7 +876,6 @@ static const struct oxygen_model model_xonar_d1 = {
 static const struct oxygen_model model_xonar_hdav = {
 	.longname = "Asus Virtuoso 200",
 	.chip = "AV200",
-	.owner = THIS_MODULE,
 	.probe = xonar_model_probe,
 	.init = xonar_hdav_init,
 	.cleanup = xonar_hdav_cleanup,
@@ -925,7 +922,7 @@ static int __devinit xonar_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 	BUG_ON(pci_id->driver_data >= ARRAY_SIZE(models));
-	err = oxygen_pci_probe(pci, index[dev], id[dev],
+	err = oxygen_pci_probe(pci, index[dev], id[dev], THIS_MODULE,
 			       models[pci_id->driver_data],
 			       pci_id->driver_data);
 	if (err >= 0)
-- 
cgit v1.2.3


From 6ed91157093c60e26bf0215b752f07af52935afc Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Thu, 19 Feb 2009 08:38:25 +0100
Subject: sound: oxygen: allocate model_data dynamically

Allocate the model-specific data dynamically instead of including it in
the memory block of the card structure.  This will allow us to determine
the actual model after the card creation.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/oxygen/oxygen_lib.c | 14 +++++++++++---
 1 file changed, 11 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c
index b5560fa5a5e..228f30800fd 100644
--- a/sound/pci/oxygen/oxygen_lib.c
+++ b/sound/pci/oxygen/oxygen_lib.c
@@ -446,6 +446,7 @@ static void oxygen_card_free(struct snd_card *card)
 		free_irq(chip->irq, chip);
 	flush_scheduled_work();
 	chip->model.cleanup(chip);
+	kfree(chip->model_data);
 	mutex_destroy(&chip->mutex);
 	pci_release_regions(chip->pci);
 	pci_disable_device(chip->pci);
@@ -460,8 +461,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
 	struct oxygen *chip;
 	int err;
 
-	err = snd_card_create(index, id, owner,
-			      sizeof(*chip) + model->model_data_size, &card);
+	err = snd_card_create(index, id, owner, sizeof(*chip), &card);
 	if (err < 0)
 		return err;
 
@@ -470,7 +470,6 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
 	chip->pci = pci;
 	chip->irq = -1;
 	chip->model = *model;
-	chip->model_data = chip + 1;
 	spin_lock_init(&chip->reg_lock);
 	mutex_init(&chip->mutex);
 	INIT_WORK(&chip->spdif_input_bits_work,
@@ -496,6 +495,15 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
 	}
 	chip->addr = pci_resource_start(pci, 0);
 
+	if (chip->model.model_data_size) {
+		chip->model_data = kmalloc(chip->model.model_data_size,
+					   GFP_KERNEL);
+		if (!chip->model_data) {
+			err = -ENOMEM;
+			goto err_pci_regions;
+		}
+	}
+
 	pci_set_master(pci);
 	snd_card_set_dev(card, &pci->dev);
 	card->private_free = oxygen_card_free;
-- 
cgit v1.2.3


From a69bb3c3fe0881d986ec78e253cb8a6bb9c28230 Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Thu, 19 Feb 2009 08:38:55 +0100
Subject: sound: oxygen: use static driver name

When allocating resources, use a fixed name instead of reading it from
the model structure.  This allows us to allocate the resources before
the actual model is known.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/oxygen/oxygen_lib.c | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c
index 228f30800fd..516d94ad2bb 100644
--- a/sound/pci/oxygen/oxygen_lib.c
+++ b/sound/pci/oxygen/oxygen_lib.c
@@ -34,6 +34,7 @@ MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
 MODULE_DESCRIPTION("C-Media CMI8788 helper library");
 MODULE_LICENSE("GPL v2");
 
+#define DRIVER "oxygen"
 
 static inline int oxygen_uart_input_ready(struct oxygen *chip)
 {
@@ -481,7 +482,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
 	if (err < 0)
 		goto err_card;
 
-	err = pci_request_regions(pci, model->chip);
+	err = pci_request_regions(pci, DRIVER);
 	if (err < 0) {
 		snd_printk(KERN_ERR "cannot reserve PCI resources\n");
 		goto err_pci_enable;
@@ -517,7 +518,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
 	chip->model.init(chip);
 
 	err = request_irq(pci->irq, oxygen_interrupt, IRQF_SHARED,
-			  chip->model.chip, chip);
+			  DRIVER, chip);
 	if (err < 0) {
 		snd_printk(KERN_ERR "cannot grab interrupt %d\n", pci->irq);
 		goto err_card;
-- 
cgit v1.2.3


From 30459d7b1843cbdea56ca120c8cac10dc5613e90 Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Thu, 19 Feb 2009 08:42:44 +0100
Subject: sound: oxygen: handle cards with broken EEPROM

Under as yet unknown circumstances, the first word of the sound card's
EEPROM gets overwritten.  When this has happened, we cannot rely on the
subsystem IDs that the kernel reads from the PCI configuration
registers.  Instead, we read the IDs directly from the EEPROM and do the
ID matching manually.

Because the model-specific driver cannot determine the model before
calling oxygen_pci_probe(), that function now gets a get_model()
callback as parameter.  The customizing of the model structure, which
was formerly done by the probe() callback, also has moved into
get_model().

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/oxygen/hifier.c     |  11 +++-
 sound/pci/oxygen/oxygen.c     |  44 +++++++-------
 sound/pci/oxygen/oxygen.h     |  17 +++++-
 sound/pci/oxygen/oxygen_io.c  |  15 +++++
 sound/pci/oxygen/oxygen_lib.c |  51 +++++++++++++---
 sound/pci/oxygen/virtuoso.c   | 134 +++++++++++++++++++++++-------------------
 6 files changed, 179 insertions(+), 93 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c
index cc98bad9916..84ef1318341 100644
--- a/sound/pci/oxygen/hifier.c
+++ b/sound/pci/oxygen/hifier.c
@@ -45,6 +45,7 @@ MODULE_PARM_DESC(enable, "enable card");
 static struct pci_device_id hifier_ids[] __devinitdata = {
 	{ OXYGEN_PCI_SUBID(0x14c3, 0x1710) },
 	{ OXYGEN_PCI_SUBID(0x14c3, 0x1711) },
+	{ OXYGEN_PCI_SUBID_BROKEN_EEPROM },
 	{ }
 };
 MODULE_DEVICE_TABLE(pci, hifier_ids);
@@ -172,6 +173,13 @@ static const struct oxygen_model model_hifier = {
 	.adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
 };
 
+static int __devinit get_hifier_model(struct oxygen *chip,
+				      const struct pci_device_id *id)
+{
+	chip->model = model_hifier;
+	return 0;
+}
+
 static int __devinit hifier_probe(struct pci_dev *pci,
 				  const struct pci_device_id *pci_id)
 {
@@ -184,7 +192,8 @@ static int __devinit hifier_probe(struct pci_dev *pci,
 		++dev;
 		return -ENOENT;
 	}
-	err = oxygen_pci_probe(pci, index[dev], id[dev], THIS_MODULE, &model_hifier, 0);
+	err = oxygen_pci_probe(pci, index[dev], id[dev], THIS_MODULE,
+			       hifier_ids, get_hifier_model);
 	if (err >= 0)
 		++dev;
 	return err;
diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c
index 12b6c2137d5..f2c37f379d3 100644
--- a/sound/pci/oxygen/oxygen.c
+++ b/sound/pci/oxygen/oxygen.c
@@ -293,29 +293,10 @@ static void set_ak5385_params(struct oxygen *chip,
 
 static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0);
 
-static int generic_probe(struct oxygen *chip, unsigned long driver_data)
-{
-	if (driver_data == MODEL_MERIDIAN) {
-		chip->model.init = meridian_init;
-		chip->model.resume = meridian_resume;
-		chip->model.set_adc_params = set_ak5385_params;
-		chip->model.device_config = PLAYBACK_0_TO_I2S |
-					    PLAYBACK_1_TO_SPDIF |
-					    CAPTURE_0_FROM_I2S_2 |
-					    CAPTURE_1_FROM_SPDIF;
-	}
-	if (driver_data == MODEL_MERIDIAN || driver_data == MODEL_HALO) {
-		chip->model.misc_flags = OXYGEN_MISC_MIDI;
-		chip->model.device_config |= MIDI_OUTPUT | MIDI_INPUT;
-	}
-	return 0;
-}
-
 static const struct oxygen_model model_generic = {
 	.shortname = "C-Media CMI8788",
 	.longname = "C-Media Oxygen HD Audio",
 	.chip = "CMI8788",
-	.probe = generic_probe,
 	.init = generic_init,
 	.cleanup = generic_cleanup,
 	.resume = generic_resume,
@@ -340,6 +321,29 @@ static const struct oxygen_model model_generic = {
 	.adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
 };
 
+static int __devinit get_oxygen_model(struct oxygen *chip,
+				      const struct pci_device_id *id)
+{
+	chip->model = model_generic;
+	switch (id->driver_data) {
+	case MODEL_MERIDIAN:
+		chip->model.init = meridian_init;
+		chip->model.resume = meridian_resume;
+		chip->model.set_adc_params = set_ak5385_params;
+		chip->model.device_config = PLAYBACK_0_TO_I2S |
+					    PLAYBACK_1_TO_SPDIF |
+					    CAPTURE_0_FROM_I2S_2 |
+					    CAPTURE_1_FROM_SPDIF;
+		break;
+	}
+	if (id->driver_data == MODEL_MERIDIAN ||
+	    id->driver_data == MODEL_HALO) {
+		chip->model.misc_flags = OXYGEN_MISC_MIDI;
+		chip->model.device_config |= MIDI_OUTPUT | MIDI_INPUT;
+	}
+	return 0;
+}
+
 static int __devinit generic_oxygen_probe(struct pci_dev *pci,
 					  const struct pci_device_id *pci_id)
 {
@@ -353,7 +357,7 @@ static int __devinit generic_oxygen_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 	err = oxygen_pci_probe(pci, index[dev], id[dev], THIS_MODULE,
-			       &model_generic, pci_id->driver_data);
+			       oxygen_ids, get_oxygen_model);
 	if (err >= 0)
 		++dev;
 	return err;
diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h
index 268bff4f29d..c500d48ea34 100644
--- a/sound/pci/oxygen/oxygen.h
+++ b/sound/pci/oxygen/oxygen.h
@@ -49,7 +49,13 @@ enum {
 	.subvendor = sv, \
 	.subdevice = sd
 
+#define BROKEN_EEPROM_DRIVER_DATA ((unsigned long)-1)
+#define OXYGEN_PCI_SUBID_BROKEN_EEPROM \
+	OXYGEN_PCI_SUBID(PCI_VENDOR_ID_CMEDIA, 0x8788), \
+	.driver_data = BROKEN_EEPROM_DRIVER_DATA
+
 struct pci_dev;
+struct pci_device_id;
 struct snd_card;
 struct snd_pcm_substream;
 struct snd_pcm_hardware;
@@ -62,7 +68,6 @@ struct oxygen_model {
 	const char *shortname;
 	const char *longname;
 	const char *chip;
-	int (*probe)(struct oxygen *chip, unsigned long driver_data);
 	void (*init)(struct oxygen *chip);
 	int (*control_filter)(struct snd_kcontrol_new *template);
 	int (*mixer_init)(struct oxygen *chip);
@@ -82,6 +87,7 @@ struct oxygen_model {
 	void (*ac97_switch)(struct oxygen *chip,
 			    unsigned int reg, unsigned int mute);
 	const unsigned int *dac_tlv;
+	unsigned long private_data;
 	size_t model_data_size;
 	unsigned int device_config;
 	u8 dac_channels;
@@ -134,8 +140,11 @@ struct oxygen {
 
 int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
 		     struct module *owner,
-		     const struct oxygen_model *model,
-		     unsigned long driver_data);
+		     const struct pci_device_id *ids,
+		     int (*get_model)(struct oxygen *chip,
+				      const struct pci_device_id *id
+				     )
+		    );
 void oxygen_pci_remove(struct pci_dev *pci);
 #ifdef CONFIG_PM
 int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state);
@@ -180,6 +189,8 @@ void oxygen_write_i2c(struct oxygen *chip, u8 device, u8 map, u8 data);
 void oxygen_reset_uart(struct oxygen *chip);
 void oxygen_write_uart(struct oxygen *chip, u8 data);
 
+u16 oxygen_read_eeprom(struct oxygen *chip, unsigned int index);
+
 static inline void oxygen_set_bits8(struct oxygen *chip,
 				    unsigned int reg, u8 value)
 {
diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c
index 3126c4b403d..05f48ef1a44 100644
--- a/sound/pci/oxygen/oxygen_io.c
+++ b/sound/pci/oxygen/oxygen_io.c
@@ -254,3 +254,18 @@ void oxygen_write_uart(struct oxygen *chip, u8 data)
 	_write_uart(chip, 0, data);
 }
 EXPORT_SYMBOL(oxygen_write_uart);
+
+u16 oxygen_read_eeprom(struct oxygen *chip, unsigned int index)
+{
+	unsigned int timeout;
+
+	oxygen_write8(chip, OXYGEN_EEPROM_CONTROL,
+		      index | OXYGEN_EEPROM_DIR_READ);
+	for (timeout = 0; timeout < 100; ++timeout) {
+		udelay(1);
+		if (!(oxygen_read8(chip, OXYGEN_EEPROM_STATUS)
+		      & OXYGEN_EEPROM_BUSY))
+			break;
+	}
+	return oxygen_read16(chip, OXYGEN_EEPROM_DATA);
+}
diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c
index 516d94ad2bb..d83c3a95732 100644
--- a/sound/pci/oxygen/oxygen_lib.c
+++ b/sound/pci/oxygen/oxygen_lib.c
@@ -244,6 +244,34 @@ static void oxygen_proc_init(struct oxygen *chip)
 #define oxygen_proc_init(chip)
 #endif
 
+static const struct pci_device_id *
+oxygen_search_pci_id(struct oxygen *chip, const struct pci_device_id ids[])
+{
+	u16 subdevice;
+
+	/*
+	 * Make sure the EEPROM pins are available, i.e., not used for SPI.
+	 * (This function is called before we initialize or use SPI.)
+	 */
+	oxygen_clear_bits8(chip, OXYGEN_FUNCTION,
+			   OXYGEN_FUNCTION_ENABLE_SPI_4_5);
+	/*
+	 * Read the subsystem device ID directly from the EEPROM, because the
+	 * chip didn't if the first EEPROM word was overwritten.
+	 */
+	subdevice = oxygen_read_eeprom(chip, 2);
+	/*
+	 * We use only the subsystem device ID for searching because it is
+	 * unique even without the subsystem vendor ID, which may have been
+	 * overwritten in the EEPROM.
+	 */
+	for (; ids->vendor; ++ids)
+		if (ids->subdevice == subdevice &&
+		    ids->driver_data != BROKEN_EEPROM_DRIVER_DATA)
+			return ids;
+	return NULL;
+}
+
 static void oxygen_init(struct oxygen *chip)
 {
 	unsigned int i;
@@ -455,11 +483,15 @@ static void oxygen_card_free(struct snd_card *card)
 
 int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
 		     struct module *owner,
-		     const struct oxygen_model *model,
-		     unsigned long driver_data)
+		     const struct pci_device_id *ids,
+		     int (*get_model)(struct oxygen *chip,
+				      const struct pci_device_id *id
+				     )
+		    )
 {
 	struct snd_card *card;
 	struct oxygen *chip;
+	const struct pci_device_id *pci_id;
 	int err;
 
 	err = snd_card_create(index, id, owner, sizeof(*chip), &card);
@@ -470,7 +502,6 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
 	chip->card = card;
 	chip->pci = pci;
 	chip->irq = -1;
-	chip->model = *model;
 	spin_lock_init(&chip->reg_lock);
 	mutex_init(&chip->mutex);
 	INIT_WORK(&chip->spdif_input_bits_work,
@@ -496,6 +527,15 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
 	}
 	chip->addr = pci_resource_start(pci, 0);
 
+	pci_id = oxygen_search_pci_id(chip, ids);
+	if (!pci_id) {
+		err = -ENODEV;
+		goto err_pci_regions;
+	}
+	err = get_model(chip, pci_id);
+	if (err < 0)
+		goto err_pci_regions;
+
 	if (chip->model.model_data_size) {
 		chip->model_data = kmalloc(chip->model.model_data_size,
 					   GFP_KERNEL);
@@ -509,11 +549,6 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
 	snd_card_set_dev(card, &pci->dev);
 	card->private_free = oxygen_card_free;
 
-	if (chip->model.probe) {
-		err = chip->model.probe(chip, driver_data);
-		if (err < 0)
-			goto err_card;
-	}
 	oxygen_init(chip);
 	chip->model.init(chip);
 
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index c05f7e7bdb3..4ac49772da8 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -160,6 +160,7 @@ static struct pci_device_id xonar_ids[] __devinitdata = {
 	{ OXYGEN_PCI_SUBID(0x1043, 0x82b7), .driver_data = MODEL_D2X },
 	{ OXYGEN_PCI_SUBID(0x1043, 0x8314), .driver_data = MODEL_HDAV },
 	{ OXYGEN_PCI_SUBID(0x1043, 0x834f), .driver_data = MODEL_D1 },
+	{ OXYGEN_PCI_SUBID_BROKEN_EEPROM },
 	{ }
 };
 MODULE_DEVICE_TABLE(pci, xonar_ids);
@@ -188,7 +189,6 @@ MODULE_DEVICE_TABLE(pci, xonar_ids);
 #define I2C_DEVICE_CS4362A	0x30	/* 001100, AD0=0, /W=0 */
 
 struct xonar_data {
-	unsigned int model;
 	unsigned int anti_pop_delay;
 	unsigned int dacs;
 	u16 output_enable_bit;
@@ -334,15 +334,9 @@ static void xonar_d2_init(struct oxygen *chip)
 	struct xonar_data *data = chip->model_data;
 
 	data->anti_pop_delay = 300;
+	data->dacs = 4;
 	data->output_enable_bit = GPIO_D2_OUTPUT_ENABLE;
 	data->pcm1796_oversampling = PCM1796_OS_64;
-	if (data->model == MODEL_D2X) {
-		data->ext_power_reg = OXYGEN_GPIO_DATA;
-		data->ext_power_int_reg = OXYGEN_GPIO_INTERRUPT_MASK;
-		data->ext_power_bit = GPIO_D2X_EXT_POWER;
-		oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL,
-				    GPIO_D2X_EXT_POWER);
-	}
 
 	pcm1796_init(chip);
 
@@ -355,6 +349,18 @@ static void xonar_d2_init(struct oxygen *chip)
 	snd_component_add(chip->card, "CS5381");
 }
 
+static void xonar_d2x_init(struct oxygen *chip)
+{
+	struct xonar_data *data = chip->model_data;
+
+	data->ext_power_reg = OXYGEN_GPIO_DATA;
+	data->ext_power_int_reg = OXYGEN_GPIO_INTERRUPT_MASK;
+	data->ext_power_bit = GPIO_D2X_EXT_POWER;
+	oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_D2X_EXT_POWER);
+
+	xonar_d2_init(chip);
+}
+
 static void update_cs4362a_volumes(struct oxygen *chip)
 {
 	u8 mute;
@@ -422,11 +428,6 @@ static void xonar_d1_init(struct oxygen *chip)
 	data->cs4398_fm = CS4398_FM_SINGLE | CS4398_DEM_NONE | CS4398_DIF_LJUST;
 	data->cs4362a_fm = CS4362A_FM_SINGLE |
 		CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L;
-	if (data->model == MODEL_DX) {
-		data->ext_power_reg = OXYGEN_GPI_DATA;
-		data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK;
-		data->ext_power_bit = GPI_DX_EXT_POWER;
-	}
 
 	oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS,
 		       OXYGEN_2WIRE_LENGTH_8 |
@@ -447,6 +448,17 @@ static void xonar_d1_init(struct oxygen *chip)
 	snd_component_add(chip->card, "CS5361");
 }
 
+static void xonar_dx_init(struct oxygen *chip)
+{
+	struct xonar_data *data = chip->model_data;
+
+	data->ext_power_reg = OXYGEN_GPI_DATA;
+	data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK;
+	data->ext_power_bit = GPI_DX_EXT_POWER;
+
+	xonar_d1_init(chip);
+}
+
 static void xonar_hdav_init(struct oxygen *chip)
 {
 	struct xonar_data *data = chip->model_data;
@@ -458,6 +470,7 @@ static void xonar_hdav_init(struct oxygen *chip)
 		       OXYGEN_2WIRE_SPEED_FAST);
 
 	data->anti_pop_delay = 100;
+	data->dacs = chip->model.private_data == MODEL_HDAV_H6 ? 4 : 1;
 	data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE;
 	data->ext_power_reg = OXYGEN_GPI_DATA;
 	data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK;
@@ -773,50 +786,9 @@ static int xonar_d1_mixer_init(struct oxygen *chip)
 	return snd_ctl_add(chip->card, snd_ctl_new1(&front_panel_switch, chip));
 }
 
-static int xonar_model_probe(struct oxygen *chip, unsigned long driver_data)
-{
-	static const char *const names[] = {
-		[MODEL_D1]	= "Xonar D1",
-		[MODEL_DX]	= "Xonar DX",
-		[MODEL_D2]	= "Xonar D2",
-		[MODEL_D2X]	= "Xonar D2X",
-		[MODEL_HDAV]	= "Xonar HDAV1.3",
-		[MODEL_HDAV_H6]	= "Xonar HDAV1.3+H6",
-	};
-	static const u8 dacs[] = {
-		[MODEL_D1]	= 2,
-		[MODEL_DX]	= 2,
-		[MODEL_D2]	= 4,
-		[MODEL_D2X]	= 4,
-		[MODEL_HDAV]	= 1,
-		[MODEL_HDAV_H6]	= 4,
-	};
-	struct xonar_data *data = chip->model_data;
-
-	data->model = driver_data;
-	if (data->model == MODEL_HDAV) {
-		oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL,
-				    GPIO_HDAV_DB_MASK);
-		switch (oxygen_read16(chip, OXYGEN_GPIO_DATA) &
-			GPIO_HDAV_DB_MASK) {
-		case GPIO_HDAV_DB_H6:
-			data->model = MODEL_HDAV_H6;
-			break;
-		case GPIO_HDAV_DB_XX:
-			snd_printk(KERN_ERR "unknown daughterboard\n");
-			return -ENODEV;
-		}
-	}
-
-	data->dacs = dacs[data->model];
-	chip->model.shortname = names[data->model];
-	return 0;
-}
-
 static const struct oxygen_model model_xonar_d2 = {
 	.longname = "Asus Virtuoso 200",
 	.chip = "AV200",
-	.probe = xonar_model_probe,
 	.init = xonar_d2_init,
 	.control_filter = xonar_d2_control_filter,
 	.mixer_init = xonar_d2_mixer_init,
@@ -848,7 +820,6 @@ static const struct oxygen_model model_xonar_d2 = {
 static const struct oxygen_model model_xonar_d1 = {
 	.longname = "Asus Virtuoso 100",
 	.chip = "AV200",
-	.probe = xonar_model_probe,
 	.init = xonar_d1_init,
 	.control_filter = xonar_d1_control_filter,
 	.mixer_init = xonar_d1_mixer_init,
@@ -876,7 +847,6 @@ static const struct oxygen_model model_xonar_d1 = {
 static const struct oxygen_model model_xonar_hdav = {
 	.longname = "Asus Virtuoso 200",
 	.chip = "AV200",
-	.probe = xonar_model_probe,
 	.init = xonar_hdav_init,
 	.cleanup = xonar_hdav_cleanup,
 	.suspend = xonar_hdav_suspend,
@@ -902,8 +872,8 @@ static const struct oxygen_model model_xonar_hdav = {
 	.adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
 };
 
-static int __devinit xonar_probe(struct pci_dev *pci,
-				 const struct pci_device_id *pci_id)
+static int __devinit get_xonar_model(struct oxygen *chip,
+				     const struct pci_device_id *id)
 {
 	static const struct oxygen_model *const models[] = {
 		[MODEL_D1]	= &model_xonar_d1,
@@ -912,6 +882,50 @@ static int __devinit xonar_probe(struct pci_dev *pci,
 		[MODEL_D2X]	= &model_xonar_d2,
 		[MODEL_HDAV]	= &model_xonar_hdav,
 	};
+	static const char *const names[] = {
+		[MODEL_D1]	= "Xonar D1",
+		[MODEL_DX]	= "Xonar DX",
+		[MODEL_D2]	= "Xonar D2",
+		[MODEL_D2X]	= "Xonar D2X",
+		[MODEL_HDAV]	= "Xonar HDAV1.3",
+		[MODEL_HDAV_H6]	= "Xonar HDAV1.3+H6",
+	};
+	unsigned int model = id->driver_data;
+
+	if (model >= ARRAY_SIZE(models) || !models[model])
+		return -EINVAL;
+	chip->model = *models[model];
+
+	switch (model) {
+	case MODEL_D2X:
+		chip->model.init = xonar_d2x_init;
+		break;
+	case MODEL_DX:
+		chip->model.init = xonar_dx_init;
+		break;
+	case MODEL_HDAV:
+		oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL,
+				    GPIO_HDAV_DB_MASK);
+		switch (oxygen_read16(chip, OXYGEN_GPIO_DATA) &
+			GPIO_HDAV_DB_MASK) {
+		case GPIO_HDAV_DB_H6:
+			model = MODEL_HDAV_H6;
+			break;
+		case GPIO_HDAV_DB_XX:
+			snd_printk(KERN_ERR "unknown daughterboard\n");
+			return -ENODEV;
+		}
+		break;
+	}
+
+	chip->model.shortname = names[model];
+	chip->model.private_data = model;
+	return 0;
+}
+
+static int __devinit xonar_probe(struct pci_dev *pci,
+				 const struct pci_device_id *pci_id)
+{
 	static int dev;
 	int err;
 
@@ -921,10 +935,8 @@ static int __devinit xonar_probe(struct pci_dev *pci,
 		++dev;
 		return -ENOENT;
 	}
-	BUG_ON(pci_id->driver_data >= ARRAY_SIZE(models));
 	err = oxygen_pci_probe(pci, index[dev], id[dev], THIS_MODULE,
-			       models[pci_id->driver_data],
-			       pci_id->driver_data);
+			       xonar_ids, get_xonar_model);
 	if (err >= 0)
 		++dev;
 	return err;
-- 
cgit v1.2.3


From 1275d6f608abda23d101ada17dc39940192d4bc4 Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Thu, 19 Feb 2009 08:44:12 +0100
Subject: sound: oxygen: automatically restore overwritten EEPROM

If the EEPROM was partially overwritten (which seems to happen before the OS is
booted), restore its entire contents by deducing it from the remaining
information.

This does not have any effect on the Linux driver, which works even with
incomplete information in the EEPROM, but it makes other drivers work again.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/oxygen/oxygen.h     |  3 +++
 sound/pci/oxygen/oxygen_io.c  | 16 ++++++++++++++++
 sound/pci/oxygen/oxygen_lib.c | 29 +++++++++++++++++++++++++++++
 3 files changed, 48 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h
index c500d48ea34..bd615dbffad 100644
--- a/sound/pci/oxygen/oxygen.h
+++ b/sound/pci/oxygen/oxygen.h
@@ -18,6 +18,8 @@
 
 #define OXYGEN_IO_SIZE	0x100
 
+#define OXYGEN_EEPROM_ID	0x434d	/* "CM" */
+
 /* model-specific configuration of outputs/inputs */
 #define PLAYBACK_0_TO_I2S	0x0001
      /* PLAYBACK_0_TO_AC97_0		not implemented */
@@ -190,6 +192,7 @@ void oxygen_reset_uart(struct oxygen *chip);
 void oxygen_write_uart(struct oxygen *chip, u8 data);
 
 u16 oxygen_read_eeprom(struct oxygen *chip, unsigned int index);
+void oxygen_write_eeprom(struct oxygen *chip, unsigned int index, u16 value);
 
 static inline void oxygen_set_bits8(struct oxygen *chip,
 				    unsigned int reg, u8 value)
diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c
index 05f48ef1a44..c1eb923f2ac 100644
--- a/sound/pci/oxygen/oxygen_io.c
+++ b/sound/pci/oxygen/oxygen_io.c
@@ -269,3 +269,19 @@ u16 oxygen_read_eeprom(struct oxygen *chip, unsigned int index)
 	}
 	return oxygen_read16(chip, OXYGEN_EEPROM_DATA);
 }
+
+void oxygen_write_eeprom(struct oxygen *chip, unsigned int index, u16 value)
+{
+	unsigned int timeout;
+
+	oxygen_write16(chip, OXYGEN_EEPROM_DATA, value);
+	oxygen_write8(chip, OXYGEN_EEPROM_CONTROL,
+		      index | OXYGEN_EEPROM_DIR_WRITE);
+	for (timeout = 0; timeout < 10; ++timeout) {
+		msleep(1);
+		if (!(oxygen_read8(chip, OXYGEN_EEPROM_STATUS)
+		      & OXYGEN_EEPROM_BUSY))
+			return;
+	}
+	snd_printk(KERN_ERR "EEPROM write timeout\n");
+}
diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c
index d83c3a95732..6e1cdd2fd76 100644
--- a/sound/pci/oxygen/oxygen_lib.c
+++ b/sound/pci/oxygen/oxygen_lib.c
@@ -272,6 +272,34 @@ oxygen_search_pci_id(struct oxygen *chip, const struct pci_device_id ids[])
 	return NULL;
 }
 
+static void oxygen_restore_eeprom(struct oxygen *chip,
+				  const struct pci_device_id *id)
+{
+	if (oxygen_read_eeprom(chip, 0) != OXYGEN_EEPROM_ID) {
+		/*
+		 * This function gets called only when a known card model has
+		 * been detected, i.e., we know there is a valid subsystem
+		 * product ID at index 2 in the EEPROM.  Therefore, we have
+		 * been able to deduce the correct subsystem vendor ID, and
+		 * this is enough information to restore the original EEPROM
+		 * contents.
+		 */
+		oxygen_write_eeprom(chip, 1, id->subvendor);
+		oxygen_write_eeprom(chip, 0, OXYGEN_EEPROM_ID);
+
+		oxygen_set_bits8(chip, OXYGEN_MISC,
+				 OXYGEN_MISC_WRITE_PCI_SUBID);
+		pci_write_config_word(chip->pci, PCI_SUBSYSTEM_VENDOR_ID,
+				      id->subvendor);
+		pci_write_config_word(chip->pci, PCI_SUBSYSTEM_ID,
+				      id->subdevice);
+		oxygen_clear_bits8(chip, OXYGEN_MISC,
+				   OXYGEN_MISC_WRITE_PCI_SUBID);
+
+		snd_printk(KERN_INFO "EEPROM ID restored\n");
+	}
+}
+
 static void oxygen_init(struct oxygen *chip)
 {
 	unsigned int i;
@@ -532,6 +560,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
 		err = -ENODEV;
 		goto err_pci_regions;
 	}
+	oxygen_restore_eeprom(chip, pci_id);
 	err = get_model(chip, pci_id);
 	if (err < 0)
 		goto err_pci_regions;
-- 
cgit v1.2.3


From d91b424d6d7bda0773b6b6b606d48d089c4f5115 Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Fri, 20 Feb 2009 09:31:14 +0100
Subject: sound: oxygen: handle AK5385 ADC on Claro halo cards

The HT-Omega Claro halo's ADC is an AK5385 instead of a WM8785, so we
should handle the ADC parameters as we do with the X-Meridian.

Using the code for the wrong ADC does not seem to have any audible
effects, and the Windows driver does it, but it is nonetheless a good
idea to run the AK5385 with an oversampling ratio that is not outside
the documented limits.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/oxygen/oxygen.c | 16 ++++++++++++++++
 1 file changed, 16 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c
index f2c37f379d3..1d8e2b29745 100644
--- a/sound/pci/oxygen/oxygen.c
+++ b/sound/pci/oxygen/oxygen.c
@@ -196,6 +196,12 @@ static void meridian_init(struct oxygen *chip)
 	ak5385_init(chip);
 }
 
+static void halo_init(struct oxygen *chip)
+{
+	ak4396_init(chip);
+	ak5385_init(chip);
+}
+
 static void generic_cleanup(struct oxygen *chip)
 {
 }
@@ -211,6 +217,11 @@ static void meridian_resume(struct oxygen *chip)
 	ak4396_registers_init(chip);
 }
 
+static void halo_resume(struct oxygen *chip)
+{
+	ak4396_registers_init(chip);
+}
+
 static void set_ak4396_params(struct oxygen *chip,
 			      struct snd_pcm_hw_params *params)
 {
@@ -335,6 +346,11 @@ static int __devinit get_oxygen_model(struct oxygen *chip,
 					    CAPTURE_0_FROM_I2S_2 |
 					    CAPTURE_1_FROM_SPDIF;
 		break;
+	case MODEL_HALO:
+		chip->model.init = halo_init;
+		chip->model.resume = halo_resume;
+		chip->model.set_adc_params = set_ak5385_params;
+		break;
 	}
 	if (id->driver_data == MODEL_MERIDIAN ||
 	    id->driver_data == MODEL_HALO) {
-- 
cgit v1.2.3


From eacbb9dba6b4c982a0217ea2c7d15db88d4fda37 Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Fri, 20 Feb 2009 09:33:40 +0100
Subject: sound: virtuoso: increase minimum volume to -60 dB

Use -60 dB as the minimum value of the master volume mixer control.
While the DACs would support ranges down to about -120 dB, such
attenuations are not useful in practice.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/oxygen/virtuoso.c | 14 +++++++-------
 1 file changed, 7 insertions(+), 7 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index 4ac49772da8..00dc97806f1 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -758,8 +758,8 @@ static void xonar_line_mic_ac97_switch(struct oxygen *chip,
 	}
 }
 
-static const DECLARE_TLV_DB_SCALE(pcm1796_db_scale, -12000, 50, 0);
-static const DECLARE_TLV_DB_SCALE(cs4362a_db_scale, -12700, 100, 0);
+static const DECLARE_TLV_DB_SCALE(pcm1796_db_scale, -6000, 50, 0);
+static const DECLARE_TLV_DB_SCALE(cs4362a_db_scale, -6000, 100, 0);
 
 static int xonar_d2_control_filter(struct snd_kcontrol_new *template)
 {
@@ -808,8 +808,8 @@ static const struct oxygen_model model_xonar_d2 = {
 			 MIDI_OUTPUT |
 			 MIDI_INPUT,
 	.dac_channels = 8,
-	.dac_volume_min = 0x0f,
-	.dac_volume_max = 0xff,
+	.dac_volume_min = 255 - 2*60,
+	.dac_volume_max = 255,
 	.misc_flags = OXYGEN_MISC_MIDI,
 	.function_flags = OXYGEN_FUNCTION_SPI |
 			  OXYGEN_FUNCTION_ENABLE_SPI_4_5,
@@ -837,7 +837,7 @@ static const struct oxygen_model model_xonar_d1 = {
 			 PLAYBACK_1_TO_SPDIF |
 			 CAPTURE_0_FROM_I2S_2,
 	.dac_channels = 8,
-	.dac_volume_min = 0,
+	.dac_volume_min = 127 - 60,
 	.dac_volume_max = 127,
 	.function_flags = OXYGEN_FUNCTION_2WIRE,
 	.dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
@@ -864,8 +864,8 @@ static const struct oxygen_model model_xonar_hdav = {
 			 PLAYBACK_1_TO_SPDIF |
 			 CAPTURE_0_FROM_I2S_2,
 	.dac_channels = 8,
-	.dac_volume_min = 0x0f,
-	.dac_volume_max = 0xff,
+	.dac_volume_min = 255 - 2*60,
+	.dac_volume_max = 255,
 	.misc_flags = OXYGEN_MISC_MIDI,
 	.function_flags = OXYGEN_FUNCTION_2WIRE,
 	.dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
-- 
cgit v1.2.3


From 3be141494a080a9189b51fa78154c975ad8d9806 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 20 Feb 2009 14:11:16 +0100
Subject: ALSA: hda - Add generic pincfg initialization

Added the generic pincfg cache and save/restore functions.
Also introduced the pin-overriding via hwdep sysfs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 151 +++++++++++++++++++++++++++++++++++++++++++---
 sound/pci/hda/hda_codec.h |  15 +++++
 sound/pci/hda/hda_hwdep.c |  66 ++++++++++++++++++++
 3 files changed, 223 insertions(+), 9 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 98884bc8f35..6fa871f66a7 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -682,11 +682,132 @@ static int read_widget_caps(struct hda_codec *codec, hda_nid_t fg_node)
 	return 0;
 }
 
+/* read all pin default configurations and save codec->init_pins */
+static int read_pin_defaults(struct hda_codec *codec)
+{
+	int i;
+	hda_nid_t nid = codec->start_nid;
+
+	for (i = 0; i < codec->num_nodes; i++, nid++) {
+		struct hda_pincfg *pin;
+		unsigned int wcaps = get_wcaps(codec, nid);
+		unsigned int wid_type = (wcaps & AC_WCAP_TYPE) >>
+				AC_WCAP_TYPE_SHIFT;
+		if (wid_type != AC_WID_PIN)
+			continue;
+		pin = snd_array_new(&codec->init_pins);
+		if (!pin)
+			return -ENOMEM;
+		pin->nid = nid;
+		pin->cfg = snd_hda_codec_read(codec, nid, 0,
+					      AC_VERB_GET_CONFIG_DEFAULT, 0);
+	}
+	return 0;
+}
+
+/* look up the given pin config list and return the item matching with NID */
+static struct hda_pincfg *look_up_pincfg(struct hda_codec *codec,
+					 struct snd_array *array,
+					 hda_nid_t nid)
+{
+	int i;
+	for (i = 0; i < array->used; i++) {
+		struct hda_pincfg *pin = snd_array_elem(array, i);
+		if (pin->nid == nid)
+			return pin;
+	}
+	return NULL;
+}
+
+/* write a config value for the given NID */
+static void set_pincfg(struct hda_codec *codec, hda_nid_t nid,
+		       unsigned int cfg)
+{
+	int i;
+	for (i = 0; i < 4; i++) {
+		snd_hda_codec_write(codec, nid, 0,
+				    AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 + i,
+				    cfg & 0xff);
+		cfg >>= 8;
+	}
+}
+
+/* set the current pin config value for the given NID.
+ * the value is cached, and read via snd_hda_codec_get_pincfg()
+ */
+int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list,
+		       hda_nid_t nid, unsigned int cfg)
+{
+	struct hda_pincfg *pin;
+
+	pin = look_up_pincfg(codec, list, nid);
+	if (!pin) {
+		pin = snd_array_new(list);
+		if (!pin)
+			return -ENOMEM;
+		pin->nid = nid;
+	}
+	pin->cfg = cfg;
+	set_pincfg(codec, nid, cfg);
+	return 0;
+}
+
+int snd_hda_codec_set_pincfg(struct hda_codec *codec,
+			     hda_nid_t nid, unsigned int cfg)
+{
+	return snd_hda_add_pincfg(codec, &codec->cur_pins, nid, cfg);
+}
+EXPORT_SYMBOL_HDA(snd_hda_codec_set_pincfg);
+
+/* get the current pin config value of the given pin NID */
+unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid)
+{
+	struct hda_pincfg *pin;
+
+	pin = look_up_pincfg(codec, &codec->cur_pins, nid);
+	if (pin)
+		return pin->cfg;
+#ifdef CONFIG_SND_HDA_HWDEP
+	pin = look_up_pincfg(codec, &codec->override_pins, nid);
+	if (pin)
+		return pin->cfg;
+#endif
+	pin = look_up_pincfg(codec, &codec->init_pins, nid);
+	if (pin)
+		return pin->cfg;
+	return 0;
+}
+EXPORT_SYMBOL_HDA(snd_hda_codec_get_pincfg);
+
+/* restore all current pin configs */
+static void restore_pincfgs(struct hda_codec *codec)
+{
+	int i;
+	for (i = 0; i < codec->init_pins.used; i++) {
+		struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i);
+		set_pincfg(codec, pin->nid,
+			   snd_hda_codec_get_pincfg(codec, pin->nid));
+	}
+}
 
 static void init_hda_cache(struct hda_cache_rec *cache,
 			   unsigned int record_size);
 static void free_hda_cache(struct hda_cache_rec *cache);
 
+/* restore the initial pin cfgs and release all pincfg lists */
+static void restore_init_pincfgs(struct hda_codec *codec)
+{
+	/* first free cur_pins and override_pins, then call restore_pincfg
+	 * so that only the values in init_pins are restored
+	 */
+	snd_array_free(&codec->cur_pins);
+#ifdef CONFIG_SND_HDA_HWDEP
+	snd_array_free(&codec->override_pins);
+#endif
+	restore_pincfgs(codec);
+	snd_array_free(&codec->init_pins);
+}
+
 /*
  * codec destructor
  */
@@ -694,6 +815,7 @@ static void snd_hda_codec_free(struct hda_codec *codec)
 {
 	if (!codec)
 		return;
+	restore_init_pincfgs(codec);
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	cancel_delayed_work(&codec->power_work);
 	flush_workqueue(codec->bus->workq);
@@ -751,6 +873,8 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr
 	init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info));
 	init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head));
 	snd_array_init(&codec->mixers, sizeof(struct snd_kcontrol *), 32);
+	snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16);
+	snd_array_init(&codec->cur_pins, sizeof(struct hda_pincfg), 16);
 	if (codec->bus->modelname) {
 		codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL);
 		if (!codec->modelname) {
@@ -787,15 +911,18 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr
 	setup_fg_nodes(codec);
 	if (!codec->afg && !codec->mfg) {
 		snd_printdd("hda_codec: no AFG or MFG node found\n");
-		snd_hda_codec_free(codec);
-		return -ENODEV;
+		err = -ENODEV;
+		goto error;
 	}
 
-	if (read_widget_caps(codec, codec->afg ? codec->afg : codec->mfg) < 0) {
+	err = read_widget_caps(codec, codec->afg ? codec->afg : codec->mfg);
+	if (err < 0) {
 		snd_printk(KERN_ERR "hda_codec: cannot malloc\n");
-		snd_hda_codec_free(codec);
-		return -ENOMEM;
+		goto error;
 	}
+	err = read_pin_defaults(codec);
+	if (err < 0)
+		goto error;
 
 	if (!codec->subsystem_id) {
 		hda_nid_t nid = codec->afg ? codec->afg : codec->mfg;
@@ -808,10 +935,8 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr
 
 	if (do_init) {
 		err = snd_hda_codec_configure(codec);
-		if (err < 0) {
-			snd_hda_codec_free(codec);
-			return err;
-		}
+		if (err < 0)
+			goto error;
 	}
 	snd_hda_codec_proc_new(codec);
 
@@ -824,6 +949,10 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr
 	if (codecp)
 		*codecp = codec;
 	return 0;
+
+ error:
+	snd_hda_codec_free(codec);
+	return err;
 }
 EXPORT_SYMBOL_HDA(snd_hda_codec_new);
 
@@ -1334,6 +1463,9 @@ void snd_hda_codec_reset(struct hda_codec *codec)
 	free_hda_cache(&codec->cmd_cache);
 	init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info));
 	init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head));
+	/* free only cur_pins so that init_pins + override_pins are restored */
+	snd_array_free(&codec->cur_pins);
+	restore_pincfgs(codec);
 	codec->num_pcms = 0;
 	codec->pcm_info = NULL;
 	codec->preset = NULL;
@@ -2175,6 +2307,7 @@ static void hda_call_codec_resume(struct hda_codec *codec)
 	hda_set_power_state(codec,
 			    codec->afg ? codec->afg : codec->mfg,
 			    AC_PWRST_D0);
+	restore_pincfgs(codec); /* restore all current pin configs */
 	hda_exec_init_verbs(codec);
 	if (codec->patch_ops.resume)
 		codec->patch_ops.resume(codec);
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 09a332ada0c..6d01a8058f0 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -778,11 +778,14 @@ struct hda_codec {
 	unsigned short spdif_ctls;	/* SPDIF control bits */
 	unsigned int spdif_in_enable;	/* SPDIF input enable? */
 	hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */
+	struct snd_array init_pins;	/* initial (BIOS) pin configurations */
+	struct snd_array cur_pins;	/* current pin configurations */
 
 #ifdef CONFIG_SND_HDA_HWDEP
 	struct snd_hwdep *hwdep;	/* assigned hwdep device */
 	struct snd_array init_verbs;	/* additional init verbs */
 	struct snd_array hints;		/* additional hints */
+	struct snd_array override_pins;	/* default pin configs to override */
 #endif
 
 	/* misc flags */
@@ -855,6 +858,18 @@ void snd_hda_codec_resume_cache(struct hda_codec *codec);
 #define snd_hda_sequence_write_cache	snd_hda_sequence_write
 #endif
 
+/* the struct for codec->pin_configs */
+struct hda_pincfg {
+	hda_nid_t nid;
+	unsigned int cfg;
+};
+
+unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid);
+int snd_hda_codec_set_pincfg(struct hda_codec *codec, hda_nid_t nid,
+			     unsigned int cfg);
+int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list,
+		       hda_nid_t nid, unsigned int cfg); /* for hwdep */
+
 /*
  * Mixer
  */
diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index 4ae51dcb81a..71039a6dec2 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -109,6 +109,7 @@ static void clear_hwdep_elements(struct hda_codec *codec)
 	for (i = 0; i < codec->hints.used; i++, head++)
 		kfree(*head);
 	snd_array_free(&codec->hints);
+	snd_array_free(&codec->override_pins);
 }
 
 static void hwdep_free(struct snd_hwdep *hwdep)
@@ -141,6 +142,7 @@ int /*__devinit*/ snd_hda_create_hwdep(struct hda_codec *codec)
 
 	snd_array_init(&codec->init_verbs, sizeof(struct hda_verb), 32);
 	snd_array_init(&codec->hints, sizeof(char *), 32);
+	snd_array_init(&codec->override_pins, sizeof(struct hda_pincfg), 16);
 
 	return 0;
 }
@@ -316,6 +318,67 @@ static ssize_t hints_store(struct device *dev,
 	return count;
 }
 
+static ssize_t pin_configs_show(struct hda_codec *codec,
+				struct snd_array *list,
+				char *buf)
+{
+	int i, len = 0;
+	for (i = 0; i < list->used; i++) {
+		struct hda_pincfg *pin = snd_array_elem(list, i);
+		len += sprintf(buf + len, "0x%02x 0x%08x\n",
+			       pin->nid, pin->cfg);
+	}
+	return len;
+}
+
+static ssize_t init_pin_configs_show(struct device *dev,
+				     struct device_attribute *attr,
+				     char *buf)
+{
+	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
+	struct hda_codec *codec = hwdep->private_data;
+	return pin_configs_show(codec, &codec->init_pins, buf);
+}
+
+static ssize_t override_pin_configs_show(struct device *dev,
+					 struct device_attribute *attr,
+					 char *buf)
+{
+	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
+	struct hda_codec *codec = hwdep->private_data;
+	return pin_configs_show(codec, &codec->override_pins, buf);
+}
+
+static ssize_t cur_pin_configs_show(struct device *dev,
+				    struct device_attribute *attr,
+				    char *buf)
+{
+	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
+	struct hda_codec *codec = hwdep->private_data;
+	return pin_configs_show(codec, &codec->cur_pins, buf);
+}
+
+#define MAX_PIN_CONFIGS		32
+
+static ssize_t override_pin_configs_store(struct device *dev,
+					  struct device_attribute *attr,
+					  const char *buf, size_t count)
+{
+	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
+	struct hda_codec *codec = hwdep->private_data;
+	int nid, cfg;
+	int err;
+
+	if (sscanf(buf, "%i %i", &nid, &cfg) != 2)
+		return -EINVAL;
+	if (!nid)
+		return -EINVAL;
+	err = snd_hda_add_pincfg(codec, &codec->override_pins, nid, cfg);
+	if (err < 0)
+		return err;
+	return count;
+}
+
 #define CODEC_ATTR_RW(type) \
 	__ATTR(type, 0644, type##_show, type##_store)
 #define CODEC_ATTR_RO(type) \
@@ -333,6 +396,9 @@ static struct device_attribute codec_attrs[] = {
 	CODEC_ATTR_RW(modelname),
 	CODEC_ATTR_WO(init_verbs),
 	CODEC_ATTR_WO(hints),
+	CODEC_ATTR_RO(init_pin_configs),
+	CODEC_ATTR_RW(override_pin_configs),
+	CODEC_ATTR_RO(cur_pin_configs),
 	CODEC_ATTR_WO(reconfig),
 	CODEC_ATTR_WO(clear),
 };
-- 
cgit v1.2.3


From 0e8a21b59d48a63f45b3e6d2aca7fb91c5aec882 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 20 Feb 2009 14:13:06 +0100
Subject: ALSA: hda - Remove realtek codec-specific pin save/restore functions

Now it's done in the common code.
Also use the common access functions for pin defaults.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 78 ++-----------------------------------------
 1 file changed, 3 insertions(+), 75 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 169b3837af5..d7f255e3b91 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -329,13 +329,6 @@ struct alc_spec {
 	/* for PLL fix */
 	hda_nid_t pll_nid;
 	unsigned int pll_coef_idx, pll_coef_bit;
-
-#ifdef SND_HDA_NEEDS_RESUME
-#define ALC_MAX_PINS	16
-	unsigned int num_pins;
-	hda_nid_t pin_nids[ALC_MAX_PINS];
-	unsigned int pin_cfgs[ALC_MAX_PINS];
-#endif
 };
 
 /*
@@ -1009,8 +1002,7 @@ static void alc_subsystem_id(struct hda_codec *codec,
 	nid = 0x1d;
 	if (codec->vendor_id == 0x10ec0260)
 		nid = 0x17;
-	ass = snd_hda_codec_read(codec, nid, 0,
-				 AC_VERB_GET_CONFIG_DEFAULT, 0);
+	ass = snd_hda_codec_get_pincfg(codec, nid);
 	if (!(ass & 1) && !(ass & 0x100000))
 		return;
 	if ((ass >> 30) != 1)	/* no physical connection */
@@ -1184,16 +1176,8 @@ static void alc_fix_pincfg(struct hda_codec *codec,
 		return;
 
 	cfg = pinfix[quirk->value];
-	for (; cfg->nid; cfg++) {
-		int i;
-		u32 val = cfg->val;
-		for (i = 0; i < 4; i++) {
-			snd_hda_codec_write(codec, cfg->nid, 0,
-				    AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 + i,
-				    val & 0xff);
-			val >>= 8;
-		}
-	}
+	for (; cfg->nid; cfg++)
+		snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
 }
 
 /*
@@ -3215,61 +3199,13 @@ static void alc_free(struct hda_codec *codec)
 }
 
 #ifdef SND_HDA_NEEDS_RESUME
-static void store_pin_configs(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	hda_nid_t nid, end_nid;
-
-	end_nid = codec->start_nid + codec->num_nodes;
-	for (nid = codec->start_nid; nid < end_nid; nid++) {
-		unsigned int wid_caps = get_wcaps(codec, nid);
-		unsigned int wid_type =
-			(wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
-		if (wid_type != AC_WID_PIN)
-			continue;
-		if (spec->num_pins >= ARRAY_SIZE(spec->pin_nids))
-			break;
-		spec->pin_nids[spec->num_pins] = nid;
-		spec->pin_cfgs[spec->num_pins] =
-			snd_hda_codec_read(codec, nid, 0,
-					   AC_VERB_GET_CONFIG_DEFAULT, 0);
-		spec->num_pins++;
-	}
-}
-
-static void resume_pin_configs(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	int i;
-
-	for (i = 0; i < spec->num_pins; i++) {
-		hda_nid_t pin_nid = spec->pin_nids[i];
-		unsigned int pin_config = spec->pin_cfgs[i];
-		snd_hda_codec_write(codec, pin_nid, 0,
-				    AC_VERB_SET_CONFIG_DEFAULT_BYTES_0,
-				    pin_config & 0x000000ff);
-		snd_hda_codec_write(codec, pin_nid, 0,
-				    AC_VERB_SET_CONFIG_DEFAULT_BYTES_1,
-				    (pin_config & 0x0000ff00) >> 8);
-		snd_hda_codec_write(codec, pin_nid, 0,
-				    AC_VERB_SET_CONFIG_DEFAULT_BYTES_2,
-				    (pin_config & 0x00ff0000) >> 16);
-		snd_hda_codec_write(codec, pin_nid, 0,
-				    AC_VERB_SET_CONFIG_DEFAULT_BYTES_3,
-				    pin_config >> 24);
-	}
-}
-
 static int alc_resume(struct hda_codec *codec)
 {
-	resume_pin_configs(codec);
 	codec->patch_ops.init(codec);
 	snd_hda_codec_resume_amp(codec);
 	snd_hda_codec_resume_cache(codec);
 	return 0;
 }
-#else
-#define store_pin_configs(codec)
 #endif
 
 /*
@@ -4329,7 +4265,6 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
 	spec->num_mux_defs = 1;
 	spec->input_mux = &spec->private_imux[0];
 
-	store_pin_configs(codec);
 	return 1;
 }
 
@@ -5693,7 +5628,6 @@ static int alc260_parse_auto_config(struct hda_codec *codec)
 	spec->num_mux_defs = 1;
 	spec->input_mux = &spec->private_imux[0];
 
-	store_pin_configs(codec);
 	return 1;
 }
 
@@ -10688,7 +10622,6 @@ static int alc262_parse_auto_config(struct hda_codec *codec)
 	if (err < 0)
 		return err;
 
-	store_pin_configs(codec);
 	return 1;
 }
 
@@ -11861,7 +11794,6 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
 	if (err < 0)
 		return err;
 
-	store_pin_configs(codec);
 	return 1;
 }
 
@@ -12774,7 +12706,6 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 	if (!spec->cap_mixer && !spec->no_analog)
 		set_capture_mixer(spec);
 
-	store_pin_configs(codec);
 	return 1;
 }
 
@@ -13825,7 +13756,6 @@ static int alc861_parse_auto_config(struct hda_codec *codec)
 	spec->num_adc_nids = ARRAY_SIZE(alc861_adc_nids);
 	set_capture_mixer(spec);
 
-	store_pin_configs(codec);
 	return 1;
 }
 
@@ -14927,7 +14857,6 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec)
 	if (err < 0)
 		return err;
 
-	store_pin_configs(codec);
 	return 1;
 }
 
@@ -16737,7 +16666,6 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
 	if (err < 0)
 		return err;
 
-	store_pin_configs(codec);
 	return 1;
 }
 
-- 
cgit v1.2.3


From 330ee9957910826a072c2ad5d4045182335f9963 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 20 Feb 2009 14:33:36 +0100
Subject: ALSA: hda - Remove IDT codec-specific pin save/restore functions

Removed its own save/restore functions and replaced with the common code.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 255 +++++++++++------------------------------
 1 file changed, 65 insertions(+), 190 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index d00a211a813..da48d8c0b29 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -229,7 +229,6 @@ struct sigmatel_spec {
 	/* pin widgets */
 	hda_nid_t *pin_nids;
 	unsigned int num_pins;
-	unsigned int *pin_configs;
 
 	/* codec specific stuff */
 	struct hda_verb *init;
@@ -2272,101 +2271,19 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = {
 	{} /* terminator */
 };
 
-static int stac92xx_save_bios_config_regs(struct hda_codec *codec)
+static void stac92xx_set_config_regs(struct hda_codec *codec,
+				     unsigned int *pincfgs)
 {
 	int i;
 	struct sigmatel_spec *spec = codec->spec;
-	
-	kfree(spec->pin_configs);
-	spec->pin_configs = kcalloc(spec->num_pins, sizeof(*spec->pin_configs),
-				    GFP_KERNEL);
-	if (!spec->pin_configs)
-		return -ENOMEM;
-	
-	for (i = 0; i < spec->num_pins; i++) {
-		hda_nid_t nid = spec->pin_nids[i];
-		unsigned int pin_cfg;
-
-		if (!nid)
-			continue;
-		pin_cfg = snd_hda_codec_read(codec, nid, 0, 
-			AC_VERB_GET_CONFIG_DEFAULT, 0x00);	
-		snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x bios pin config %8.8x\n",
-					nid, pin_cfg);
-		spec->pin_configs[i] = pin_cfg;
-	}
-	
-	return 0;
-}
 
-static void stac92xx_set_config_reg(struct hda_codec *codec,
-				    hda_nid_t pin_nid, unsigned int pin_config)
-{
-	int i;
-	snd_hda_codec_write(codec, pin_nid, 0,
-			    AC_VERB_SET_CONFIG_DEFAULT_BYTES_0,
-			    pin_config & 0x000000ff);
-	snd_hda_codec_write(codec, pin_nid, 0,
-			    AC_VERB_SET_CONFIG_DEFAULT_BYTES_1,
-			    (pin_config & 0x0000ff00) >> 8);
-	snd_hda_codec_write(codec, pin_nid, 0,
-			    AC_VERB_SET_CONFIG_DEFAULT_BYTES_2,
-			    (pin_config & 0x00ff0000) >> 16);
-	snd_hda_codec_write(codec, pin_nid, 0,
-			    AC_VERB_SET_CONFIG_DEFAULT_BYTES_3,
-			    pin_config >> 24);
-	i = snd_hda_codec_read(codec, pin_nid, 0,
-			       AC_VERB_GET_CONFIG_DEFAULT,
-			       0x00);	
-	snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x pin config %8.8x\n",
-		    pin_nid, i);
-}
-
-static void stac92xx_set_config_regs(struct hda_codec *codec)
-{
-	int i;
-	struct sigmatel_spec *spec = codec->spec;
-
- 	if (!spec->pin_configs)
- 		return;
+	if (!pincfgs)
+		return;
 
 	for (i = 0; i < spec->num_pins; i++)
-		if (spec->pin_nids[i] && spec->pin_configs[i])
-			stac92xx_set_config_reg(codec, spec->pin_nids[i],
-						spec->pin_configs[i]);
-}
-
-static int stac_save_pin_cfgs(struct hda_codec *codec, unsigned int *pins)
-{
-	struct sigmatel_spec *spec = codec->spec;
-
-	if (!pins)
-		return stac92xx_save_bios_config_regs(codec);
-
-	kfree(spec->pin_configs);
-	spec->pin_configs = kmemdup(pins,
-				    spec->num_pins * sizeof(*pins),
-				    GFP_KERNEL);
-	if (!spec->pin_configs)
-		return -ENOMEM;
-
-	stac92xx_set_config_regs(codec);
-	return 0;
-}
-
-static void stac_change_pin_config(struct hda_codec *codec, hda_nid_t nid,
-				   unsigned int cfg)
-{
-	struct sigmatel_spec *spec = codec->spec;
-	int i;
-
-	for (i = 0; i < spec->num_pins; i++) {
-		if (spec->pin_nids[i] == nid) {
-			spec->pin_configs[i] = cfg;
-			stac92xx_set_config_reg(codec, nid, cfg);
-			break;
-		}
-	}
+		if (spec->pin_nids[i] && pincfgs[i])
+			snd_hda_codec_set_pincfg(codec, spec->pin_nids[i],
+						 pincfgs[i]);
 }
 
 /*
@@ -2853,8 +2770,7 @@ static hda_nid_t check_mic_out_switch(struct hda_codec *codec)
 	mic_pin = AUTO_PIN_MIC;
 	for (;;) {
 		hda_nid_t nid = cfg->input_pins[mic_pin];
-		def_conf = snd_hda_codec_read(codec, nid, 0,
-					      AC_VERB_GET_CONFIG_DEFAULT, 0);
+		def_conf = snd_hda_codec_get_pincfg(codec, nid);
 		/* some laptops have an internal analog microphone
 		 * which can't be used as a output */
 		if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) {
@@ -3426,11 +3342,7 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec,
 		unsigned int wcaps;
 		unsigned int def_conf;
 
-		def_conf = snd_hda_codec_read(codec,
-					      spec->dmic_nids[i],
-					      0,
-					      AC_VERB_GET_CONFIG_DEFAULT,
-					      0);
+		def_conf = snd_hda_codec_get_pincfg(codec, spec->dmic_nids[i]);
 		if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE)
 			continue;
 
@@ -3779,9 +3691,7 @@ static int stac9200_auto_create_lfe_ctls(struct hda_codec *codec,
 		for (i = 0; i < spec->autocfg.line_outs && lfe_pin == 0x0; i++) {
 			hda_nid_t pin = spec->autocfg.line_out_pins[i];
 			unsigned int defcfg;
-			defcfg = snd_hda_codec_read(codec, pin, 0,
-						 AC_VERB_GET_CONFIG_DEFAULT,
-						 0x00);
+			defcfg = snd_hda_codec_get_pincfg(codec, pin);
 			if (get_defcfg_device(defcfg) == AC_JACK_SPEAKER) {
 				unsigned int wcaps = get_wcaps(codec, pin);
 				wcaps &= (AC_WCAP_STEREO | AC_WCAP_OUT_AMP);
@@ -3885,8 +3795,7 @@ static int stac92xx_add_jack(struct hda_codec *codec,
 #ifdef CONFIG_SND_JACK
 	struct sigmatel_spec *spec = codec->spec;
 	struct sigmatel_jack *jack;
-	int def_conf = snd_hda_codec_read(codec, nid,
-			0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+	int def_conf = snd_hda_codec_get_pincfg(codec, nid);
 	int connectivity = get_defcfg_connect(def_conf);
 	char name[32];
 
@@ -4066,8 +3975,7 @@ static int stac92xx_init(struct hda_codec *codec)
 								 pinctl);
 				}
 			}
-			conf = snd_hda_codec_read(codec, nid, 0,
-					      AC_VERB_GET_CONFIG_DEFAULT, 0);
+			conf = snd_hda_codec_get_pincfg(codec, nid);
 			if (get_defcfg_connect(conf) != AC_JACK_PORT_FIXED) {
 				enable_pin_detect(codec, nid,
 						  STAC_INSERT_EVENT);
@@ -4108,8 +4016,7 @@ static int stac92xx_init(struct hda_codec *codec)
 			stac_toggle_power_map(codec, nid, 1);
 			continue;
 		}
-		def_conf = snd_hda_codec_read(codec, nid, 0,
-					      AC_VERB_GET_CONFIG_DEFAULT, 0);
+		def_conf = snd_hda_codec_get_pincfg(codec, nid);
 		def_conf = get_defcfg_connect(def_conf);
 		/* skip any ports that don't have jacks since presence
  		 * detection is useless */
@@ -4163,7 +4070,6 @@ static void stac92xx_free(struct hda_codec *codec)
 	if (! spec)
 		return;
 
-	kfree(spec->pin_configs);
 	stac92xx_free_jacks(codec);
 	snd_array_free(&spec->events);
 
@@ -4474,7 +4380,6 @@ static int stac92xx_resume(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec = codec->spec;
 
-	stac92xx_set_config_regs(codec);
 	stac92xx_init(codec);
 	snd_hda_codec_resume_amp(codec);
 	snd_hda_codec_resume_cache(codec);
@@ -4523,16 +4428,11 @@ static int patch_stac9200(struct hda_codec *codec)
 	spec->board_config = snd_hda_check_board_config(codec, STAC_9200_MODELS,
 							stac9200_models,
 							stac9200_cfg_tbl);
-	if (spec->board_config < 0) {
+	if (spec->board_config < 0)
 		snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9200, using BIOS defaults\n");
-		err = stac92xx_save_bios_config_regs(codec);
-	} else
-		err = stac_save_pin_cfgs(codec,
+	else
+		stac92xx_set_config_regs(codec,
 					 stac9200_brd_tbl[spec->board_config]);
-	if (err < 0) {
-		stac92xx_free(codec);
-		return err;
-	}
 
 	spec->multiout.max_channels = 2;
 	spec->multiout.num_dacs = 1;
@@ -4600,17 +4500,12 @@ static int patch_stac925x(struct hda_codec *codec)
 							stac925x_models,
 							stac925x_cfg_tbl);
  again:
-	if (spec->board_config < 0) {
+	if (spec->board_config < 0)
 		snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC925x,"
 				      "using BIOS defaults\n");
-		err = stac92xx_save_bios_config_regs(codec);
-	} else
-		err = stac_save_pin_cfgs(codec,
+	else
+		stac92xx_set_config_regs(codec,
 					 stac925x_brd_tbl[spec->board_config]);
-	if (err < 0) {
-		stac92xx_free(codec);
-		return err;
-	}
 
 	spec->multiout.max_channels = 2;
 	spec->multiout.num_dacs = 1;
@@ -4688,17 +4583,12 @@ static int patch_stac92hd73xx(struct hda_codec *codec)
 							stac92hd73xx_models,
 							stac92hd73xx_cfg_tbl);
 again:
-	if (spec->board_config < 0) {
+	if (spec->board_config < 0)
 		snd_printdd(KERN_INFO "hda_codec: Unknown model for"
 			" STAC92HD73XX, using BIOS defaults\n");
-		err = stac92xx_save_bios_config_regs(codec);
-	} else
-		err = stac_save_pin_cfgs(codec,
+	else
+		stac92xx_set_config_regs(codec,
 				stac92hd73xx_brd_tbl[spec->board_config]);
-	if (err < 0) {
-		stac92xx_free(codec);
-		return err;
-	}
 
 	num_dacs = snd_hda_get_connections(codec, 0x0a,
 			conn, STAC92HD73_DAC_COUNT + 2) - 1;
@@ -4758,18 +4648,18 @@ again:
 			spec->init = dell_m6_core_init;
 		switch (spec->board_config) {
 		case STAC_DELL_M6_AMIC: /* Analog Mics */
-			stac92xx_set_config_reg(codec, 0x0b, 0x90A70170);
+			snd_hda_codec_set_pincfg(codec, 0x0b, 0x90A70170);
 			spec->num_dmics = 0;
 			spec->private_dimux.num_items = 1;
 			break;
 		case STAC_DELL_M6_DMIC: /* Digital Mics */
-			stac92xx_set_config_reg(codec, 0x13, 0x90A60160);
+			snd_hda_codec_set_pincfg(codec, 0x13, 0x90A60160);
 			spec->num_dmics = 1;
 			spec->private_dimux.num_items = 2;
 			break;
 		case STAC_DELL_M6_BOTH: /* Both */
-			stac92xx_set_config_reg(codec, 0x0b, 0x90A70170);
-			stac92xx_set_config_reg(codec, 0x13, 0x90A60160);
+			snd_hda_codec_set_pincfg(codec, 0x0b, 0x90A70170);
+			snd_hda_codec_set_pincfg(codec, 0x13, 0x90A60160);
 			spec->num_dmics = 1;
 			spec->private_dimux.num_items = 2;
 			break;
@@ -4865,17 +4755,12 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
 							stac92hd83xxx_models,
 							stac92hd83xxx_cfg_tbl);
 again:
-	if (spec->board_config < 0) {
+	if (spec->board_config < 0)
 		snd_printdd(KERN_INFO "hda_codec: Unknown model for"
 			" STAC92HD83XXX, using BIOS defaults\n");
-		err = stac92xx_save_bios_config_regs(codec);
-	} else
-		err = stac_save_pin_cfgs(codec,
+	else
+		stac92xx_set_config_regs(codec,
 				stac92hd83xxx_brd_tbl[spec->board_config]);
-	if (err < 0) {
-		stac92xx_free(codec);
-		return err;
-	}
 
 	switch (codec->vendor_id) {
 	case 0x111d7604:
@@ -4945,6 +4830,16 @@ static struct hda_input_mux stac92hd71bxx_dmux_amixer = {
 	}
 };
 
+/* get the pin connection (fixed, none, etc) */
+static unsigned int stac_get_defcfg_connect(struct hda_codec *codec, int idx)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	unsigned int cfg;
+
+	cfg = snd_hda_codec_get_pincfg(codec, spec->pin_nids[idx]);
+	return get_defcfg_connect(cfg);
+}
+
 static int stac92hd71bxx_connected_ports(struct hda_codec *codec,
 					 hda_nid_t *nids, int num_nids)
 {
@@ -4958,7 +4853,7 @@ static int stac92hd71bxx_connected_ports(struct hda_codec *codec,
 				break;
 		if (idx >= spec->num_pins)
 			break;
-		def_conf = get_defcfg_connect(spec->pin_configs[idx]);
+		def_conf = stac_get_defcfg_connect(codec, idx);
 		if (def_conf == AC_JACK_PORT_NONE)
 			break;
 	}
@@ -4978,13 +4873,13 @@ static int stac92hd71bxx_connected_smuxes(struct hda_codec *codec,
 		return 0;
 
 	/* dig1pin case */
-	if (get_defcfg_connect(spec->pin_configs[idx+1]) != AC_JACK_PORT_NONE)
+	if (stac_get_defcfg_connect(codec, idx + 1) != AC_JACK_PORT_NONE)
 		return 2;
 
 	/* dig0pin + dig2pin case */
-	if (get_defcfg_connect(spec->pin_configs[idx+2]) != AC_JACK_PORT_NONE)
+	if (stac_get_defcfg_connect(codec, idx + 2) != AC_JACK_PORT_NONE)
 		return 2;
-	if (get_defcfg_connect(spec->pin_configs[idx]) != AC_JACK_PORT_NONE)
+	if (stac_get_defcfg_connect(codec, idx) != AC_JACK_PORT_NONE)
 		return 1;
 	else
 		return 0;
@@ -5023,17 +4918,12 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
 							stac92hd71bxx_models,
 							stac92hd71bxx_cfg_tbl);
 again:
-	if (spec->board_config < 0) {
+	if (spec->board_config < 0)
 		snd_printdd(KERN_INFO "hda_codec: Unknown model for"
 			" STAC92HD71BXX, using BIOS defaults\n");
-		err = stac92xx_save_bios_config_regs(codec);
-	} else
-		err = stac_save_pin_cfgs(codec,
+	else
+		stac92xx_set_config_regs(codec,
 				stac92hd71bxx_brd_tbl[spec->board_config]);
-	if (err < 0) {
-		stac92xx_free(codec);
-		return err;
-	}
 
 	if (spec->board_config > STAC_92HD71BXX_REF) {
 		/* GPIO0 = EAPD */
@@ -5097,8 +4987,8 @@ again:
 		/* disable VSW */
 		spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF];
 		unmute_init++;
-		stac_change_pin_config(codec, 0x0f, 0x40f000f0);
-		stac_change_pin_config(codec, 0x19, 0x40f000f3);
+		snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0);
+		snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3);
 		stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS - 1] = 0;
 		spec->num_dmics = stac92hd71bxx_connected_ports(codec,
 					stac92hd71bxx_dmic_nids,
@@ -5147,7 +5037,7 @@ again:
 	switch (spec->board_config) {
 	case STAC_HP_M4:
 		/* enable internal microphone */
-		stac_change_pin_config(codec, 0x0e, 0x01813040);
+		snd_hda_codec_set_pincfg(codec, 0x0e, 0x01813040);
 		stac92xx_auto_set_pinctl(codec, 0x0e,
 			AC_PINCTL_IN_EN | AC_PINCTL_VREF_80);
 		/* fallthru */
@@ -5163,7 +5053,7 @@ again:
 		spec->num_dmuxes = 0;
 		break;
 	case STAC_HP_DV5:
-		stac_change_pin_config(codec, 0x0d, 0x90170010);
+		snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010);
 		stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN);
 		break;
 	};
@@ -5247,17 +5137,12 @@ static int patch_stac922x(struct hda_codec *codec)
 	}
 
  again:
-	if (spec->board_config < 0) {
+	if (spec->board_config < 0)
 		snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC922x, "
 			"using BIOS defaults\n");
-		err = stac92xx_save_bios_config_regs(codec);
-	} else
-		err = stac_save_pin_cfgs(codec,
+	else
+		stac92xx_set_config_regs(codec,
 				stac922x_brd_tbl[spec->board_config]);
-	if (err < 0) {
-		stac92xx_free(codec);
-		return err;
-	}
 
 	spec->adc_nids = stac922x_adc_nids;
 	spec->mux_nids = stac922x_mux_nids;
@@ -5315,17 +5200,12 @@ static int patch_stac927x(struct hda_codec *codec)
 							stac927x_models,
 							stac927x_cfg_tbl);
  again:
-	if (spec->board_config < 0) {
+	if (spec->board_config < 0)
 		snd_printdd(KERN_INFO "hda_codec: Unknown model for"
 			    "STAC927x, using BIOS defaults\n");
-		err = stac92xx_save_bios_config_regs(codec);
-	} else
-		err = stac_save_pin_cfgs(codec,
+	else
+		stac92xx_set_config_regs(codec,
 				stac927x_brd_tbl[spec->board_config]);
-	if (err < 0) {
-		stac92xx_free(codec);
-		return err;
-	}
 
 	spec->digbeep_nid = 0x23;
 	spec->adc_nids = stac927x_adc_nids;
@@ -5354,15 +5234,15 @@ static int patch_stac927x(struct hda_codec *codec)
 		case 0x10280209:
 		case 0x1028022e:
 			/* correct the device field to SPDIF out */
-			stac_change_pin_config(codec, 0x21, 0x01442070);
+			snd_hda_codec_set_pincfg(codec, 0x21, 0x01442070);
 			break;
 		};
 		/* configure the analog microphone on some laptops */
-		stac_change_pin_config(codec, 0x0c, 0x90a79130);
+		snd_hda_codec_set_pincfg(codec, 0x0c, 0x90a79130);
 		/* correct the front output jack as a hp out */
-		stac_change_pin_config(codec, 0x0f, 0x0227011f);
+		snd_hda_codec_set_pincfg(codec, 0x0f, 0x0227011f);
 		/* correct the front input jack as a mic */
-		stac_change_pin_config(codec, 0x0e, 0x02a79130);
+		snd_hda_codec_set_pincfg(codec, 0x0e, 0x02a79130);
 		/* fallthru */
 	case STAC_DELL_3ST:
 		/* GPIO2 High = Enable EAPD */
@@ -5447,16 +5327,11 @@ static int patch_stac9205(struct hda_codec *codec)
 							stac9205_models,
 							stac9205_cfg_tbl);
  again:
-	if (spec->board_config < 0) {
+	if (spec->board_config < 0)
 		snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9205, using BIOS defaults\n");
-		err = stac92xx_save_bios_config_regs(codec);
-	} else
-		err = stac_save_pin_cfgs(codec,
+	else
+		stac92xx_set_config_regs(codec,
 					 stac9205_brd_tbl[spec->board_config]);
-	if (err < 0) {
-		stac92xx_free(codec);
-		return err;
-	}
 
 	spec->digbeep_nid = 0x23;
 	spec->adc_nids = stac9205_adc_nids;
@@ -5484,8 +5359,8 @@ static int patch_stac9205(struct hda_codec *codec)
 	switch (spec->board_config){
 	case STAC_9205_DELL_M43:
 		/* Enable SPDIF in/out */
-		stac_change_pin_config(codec, 0x1f, 0x01441030);
-		stac_change_pin_config(codec, 0x20, 0x1c410030);
+		snd_hda_codec_set_pincfg(codec, 0x1f, 0x01441030);
+		snd_hda_codec_set_pincfg(codec, 0x20, 0x1c410030);
 
 		/* Enable unsol response for GPIO4/Dock HP connection */
 		err = stac_add_event(spec, codec->afg, STAC_VREF_EVENT, 0x01);
-- 
cgit v1.2.3


From 2f334f92cfb44d17b9f24a43f8998cca03f9a3dd Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 20 Feb 2009 14:37:42 +0100
Subject: ALSA: hda - Remove codec-specific pin save/restore functions

Replace the accessor to pin defaults with the common code for caching.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c |  3 +--
 sound/pci/hda/patch_cmedia.c | 12 ++++++------
 sound/pci/hda/patch_via.c    |  7 ++-----
 3 files changed, 9 insertions(+), 13 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 2c58d7b05ab..53d0edaf04c 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -1047,8 +1047,7 @@ static struct hda_amp_list ad1986a_loopbacks[] = {
 
 static int is_jack_available(struct hda_codec *codec, hda_nid_t nid)
 {
-	unsigned int conf = snd_hda_codec_read(codec, nid, 0,
-					       AC_VERB_GET_CONFIG_DEFAULT, 0);
+	unsigned int conf = snd_hda_codec_get_pincfg(codec, nid);
 	return get_defcfg_connect(conf) != AC_JACK_PORT_NONE;
 }
 
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index f3ebe837f2d..c921264bbd7 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -680,13 +680,13 @@ static int patch_cmi9880(struct hda_codec *codec)
 		struct auto_pin_cfg cfg;
 
 		/* collect pin default configuration */
-		port_e = snd_hda_codec_read(codec, 0x0f, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
-		port_f = snd_hda_codec_read(codec, 0x10, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+		port_e = snd_hda_codec_get_pincfg(codec, 0x0f);
+		port_f = snd_hda_codec_get_pincfg(codec, 0x10);
 		spec->front_panel = 1;
 		if (get_defcfg_connect(port_e) == AC_JACK_PORT_NONE ||
 		    get_defcfg_connect(port_f) == AC_JACK_PORT_NONE) {
-			port_g = snd_hda_codec_read(codec, 0x1f, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
-			port_h = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+			port_g = snd_hda_codec_get_pincfg(codec, 0x1f);
+			port_h = snd_hda_codec_get_pincfg(codec, 0x20);
 			spec->channel_modes = cmi9880_channel_modes;
 			/* no front panel */
 			if (get_defcfg_connect(port_g) == AC_JACK_PORT_NONE ||
@@ -703,8 +703,8 @@ static int patch_cmi9880(struct hda_codec *codec)
 			spec->multiout.max_channels = cmi9880_channel_modes[0].channels;
 		} else {
 			spec->input_mux = &cmi9880_basic_mux;
-			port_spdifi = snd_hda_codec_read(codec, 0x13, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
-			port_spdifo = snd_hda_codec_read(codec, 0x12, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+			port_spdifi = snd_hda_codec_get_pincfg(codec, 0x13);
+			port_spdifo = snd_hda_codec_get_pincfg(codec, 0x12);
 			if (get_defcfg_connect(port_spdifo) != AC_JACK_PORT_NONE)
 				spec->multiout.dig_out_nid = CMI_DIG_OUT_NID;
 			if (get_defcfg_connect(port_spdifi) != AC_JACK_PORT_NONE)
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 639b2ff510a..b25a5cc637d 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1308,16 +1308,13 @@ static void vt1708_set_pinconfig_connect(struct hda_codec *codec, hda_nid_t nid)
 	unsigned int def_conf;
 	unsigned char seqassoc;
 
-	def_conf = snd_hda_codec_read(codec, nid, 0,
-				      AC_VERB_GET_CONFIG_DEFAULT, 0);
+	def_conf = snd_hda_codec_get_pincfg(codec, nid);
 	seqassoc = (unsigned char) get_defcfg_association(def_conf);
 	seqassoc = (seqassoc << 4) | get_defcfg_sequence(def_conf);
 	if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) {
 		if (seqassoc == 0xff) {
 			def_conf = def_conf & (~(AC_JACK_PORT_BOTH << 30));
-			snd_hda_codec_write(codec, nid, 0,
-					    AC_VERB_SET_CONFIG_DEFAULT_BYTES_3,
-					    def_conf >> 24);
+			snd_hda_codec_set_pincfg(codec, nid, def_conf);
 		}
 	}
 
-- 
cgit v1.2.3


From cc95948972576c3efa43c9ed05b4a265805a4c54 Mon Sep 17 00:00:00 2001
From: Michael Schwingen <michael@schwingen.org>
Date: Sun, 22 Feb 2009 18:58:45 +0100
Subject: ALSA: hda - add support for "Maxdata Favorit 100XS" (Intel
 HDA/ALC260)

Signed-off-by: Michael Schwingen <michael@schwingen.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 130 ++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 130 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 169b3837af5..abddabc1efa 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -78,6 +78,7 @@ enum {
 	ALC260_ACER,
 	ALC260_WILL,
 	ALC260_REPLACER_672V,
+	ALC260_FAVORIT100,
 #ifdef CONFIG_SND_DEBUG
 	ALC260_TEST,
 #endif
@@ -4537,6 +4538,26 @@ static struct hda_input_mux alc260_acer_capture_sources[2] = {
 		},
 	},
 };
+
+/* Maxdata Favorit 100XS */
+static struct hda_input_mux alc260_favorit100_capture_sources[2] = {
+	{
+		.num_items = 2,
+		.items = {
+			{ "Line/Mic", 0x0 },
+			{ "CD", 0x4 },
+		},
+	},
+	{
+		.num_items = 3,
+		.items = {
+			{ "Line/Mic", 0x0 },
+			{ "CD", 0x4 },
+			{ "Mixer", 0x5 },
+		},
+	},
+};
+
 /*
  * This is just place-holder, so there's something for alc_build_pcms to look
  * at when it calculates the maximum number of channels. ALC260 has no mixer
@@ -4817,6 +4838,18 @@ static struct snd_kcontrol_new alc260_acer_mixer[] = {
 	{ } /* end */
 };
 
+/* Maxdata Favorit 100XS: one output and one input (0x12) jack
+ */
+static struct snd_kcontrol_new alc260_favorit100_mixer[] = {
+	HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT),
+	ALC_PIN_MODE("Output Jack Mode", 0x0f, ALC_PIN_DIR_INOUT),
+	HDA_CODEC_VOLUME("Line/Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Line/Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
+	ALC_PIN_MODE("Line/Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
+	{ } /* end */
+};
+
 /* Packard bell V7900  ALC260 pin usage: HP = 0x0f, Mic jack = 0x12,
  * Line In jack = 0x14, CD audio =  0x16, pc beep = 0x17.
  */
@@ -5188,6 +5221,89 @@ static struct hda_verb alc260_acer_init_verbs[] = {
 	{ }
 };
 
+/* Initialisation sequence for Maxdata Favorit 100XS
+ * (adapted from Acer init verbs).
+ */
+static struct hda_verb alc260_favorit100_init_verbs[] = {
+	/* GPIO 0 enables the output jack.
+	 * Turn this on and rely on the standard mute
+	 * methods whenever the user wants to turn these outputs off.
+	 */
+	{0x01, AC_VERB_SET_GPIO_MASK, 0x01},
+	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
+	{0x01, AC_VERB_SET_GPIO_DATA, 0x01},
+	/* Line/Mic input jack is connected to Mic1 pin */
+	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
+	/* Ensure all other unused pins are disabled and muted. */
+	{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	/* Disable digital (SPDIF) pins */
+	{0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
+	{0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
+
+	/* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum
+	 * bus when acting as outputs.
+	 */
+	{0x0b, AC_VERB_SET_CONNECT_SEL, 0},
+	{0x0d, AC_VERB_SET_CONNECT_SEL, 0},
+
+	/* Start with output sum widgets muted and their output gains at min */
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+	/* Unmute Line-out pin widget amp left and right
+	 * (no equiv mixer ctrl)
+	 */
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Unmute Mic1 and Line1 pin widget input buffers since they start as
+	 * inputs. If the pin mode is changed by the user the pin mode control
+	 * will take care of enabling the pin's input/output buffers as needed.
+	 * Therefore there's no need to enable the input buffer at this
+	 * stage.
+	 */
+	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+	/* Mute capture amp left and right */
+	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	/* Set ADC connection select to match default mixer setting - mic
+	 * (on mic1 pin)
+	 */
+	{0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	/* Do similar with the second ADC: mute capture input amp and
+	 * set ADC connection to mic to match ALSA's default state.
+	 */
+	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	/* Mute all inputs to mixer widget (even unconnected ones) */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
+
+	{ }
+};
+
 static struct hda_verb alc260_will_verbs[] = {
 	{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
 	{0x0b, AC_VERB_SET_CONNECT_SEL, 0x00},
@@ -5730,6 +5846,7 @@ static const char *alc260_models[ALC260_MODEL_LAST] = {
 	[ALC260_ACER]		= "acer",
 	[ALC260_WILL]		= "will",
 	[ALC260_REPLACER_672V]	= "replacer",
+	[ALC260_FAVORIT100]	= "favorit100",
 #ifdef CONFIG_SND_DEBUG
 	[ALC260_TEST]		= "test",
 #endif
@@ -5739,6 +5856,7 @@ static const char *alc260_models[ALC260_MODEL_LAST] = {
 static struct snd_pci_quirk alc260_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x007b, "Acer C20x", ALC260_ACER),
 	SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER),
+	SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100),
 	SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013),
 	SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_HP_3013),
 	SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013),
@@ -5840,6 +5958,18 @@ static struct alc_config_preset alc260_presets[] = {
 		.num_mux_defs = ARRAY_SIZE(alc260_acer_capture_sources),
 		.input_mux = alc260_acer_capture_sources,
 	},
+	[ALC260_FAVORIT100] = {
+		.mixers = { alc260_favorit100_mixer },
+		.init_verbs = { alc260_favorit100_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc260_dac_nids),
+		.dac_nids = alc260_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
+		.adc_nids = alc260_dual_adc_nids,
+		.num_channel_mode = ARRAY_SIZE(alc260_modes),
+		.channel_mode = alc260_modes,
+		.num_mux_defs = ARRAY_SIZE(alc260_favorit100_capture_sources),
+		.input_mux = alc260_favorit100_capture_sources,
+	},
 	[ALC260_WILL] = {
 		.mixers = { alc260_will_mixer },
 		.init_verbs = { alc260_init_verbs, alc260_will_verbs },
-- 
cgit v1.2.3


From e588ed8304f76cbb396ee85e657a58990298a675 Mon Sep 17 00:00:00 2001
From: Tim Blechmann <tim@klingt.org>
Date: Fri, 20 Feb 2009 19:30:35 +0100
Subject: ALSA: hdsp - poll for iobox

sleeping for 2 seconds before checking for the iobox is not enough
on some systems.
this patch increases the timeout, but polls the card during that
time. it thus speeds up the module loading when the card has already
been initialized, while being more robust on systems, which require
a higher timeout than the predefined 2 seconds.

Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/rme9652/hdsp.c | 29 +++++++++++++++++++++++++----
 1 file changed, 25 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index bacfdd12619..12c6b4305ec 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -653,7 +653,6 @@ static unsigned int hdsp_read(struct hdsp *hdsp, int reg)
 
 static int hdsp_check_for_iobox (struct hdsp *hdsp)
 {
-
 	if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0;
 	if (hdsp_read (hdsp, HDSP_statusRegister) & HDSP_ConfigError) {
 		snd_printk ("Hammerfall-DSP: no Digiface or Multiface connected!\n");
@@ -661,7 +660,29 @@ static int hdsp_check_for_iobox (struct hdsp *hdsp)
 		return -EIO;
 	}
 	return 0;
+}
 
+static int hdsp_wait_for_iobox(struct hdsp *hdsp, unsigned int loops,
+			       unsigned int delay)
+{
+	unsigned int i;
+
+	if (hdsp->io_type == H9652 || hdsp->io_type == H9632)
+		return 0;
+
+	for (i = 0; i != loops; ++i) {
+		if (hdsp_read(hdsp, HDSP_statusRegister) & HDSP_ConfigError)
+			msleep(delay);
+		else {
+			snd_printd("Hammerfall-DSP: iobox found after %ums!\n",
+				   i * delay);
+			return 0;
+		}
+	}
+
+	snd_printk("Hammerfall-DSP: no Digiface or Multiface connected!\n");
+	hdsp->state &= ~HDSP_FirmwareLoaded;
+	return -EIO;
 }
 
 static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) {
@@ -5046,10 +5067,10 @@ static int __devinit snd_hdsp_create(struct snd_card *card,
 		return err;
 	
 	if (!is_9652 && !is_9632) {
-		/* we wait 2 seconds to let freshly inserted cardbus cards do their hardware init */
-		ssleep(2);
+		/* we wait a maximum of 10 seconds to let freshly
+		 * inserted cardbus cards do their hardware init */
+		err = hdsp_wait_for_iobox(hdsp, 1000, 10);
 
-		err = hdsp_check_for_iobox(hdsp);
 		if (err < 0)
 			return err;
 
-- 
cgit v1.2.3


From f9ffc5d6f0161b66202f2df9ecc42d1be241020d Mon Sep 17 00:00:00 2001
From: Tim Blechmann <tim@klingt.org>
Date: Fri, 20 Feb 2009 19:38:16 +0100
Subject: ALSA: hdsp - whitespace cleanup

Impact: remove trailing spaces

Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/rme9652/hdsp.c | 474 +++++++++++++++++++++++------------------------
 1 file changed, 237 insertions(+), 237 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index 12c6b4305ec..dc65fe1c9c6 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -113,7 +113,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin");
 
 /* the meters are regular i/o-mapped registers, but offset
    considerably from the rest. the peak registers are reset
-   when read; the least-significant 4 bits are full-scale counters; 
+   when read; the least-significant 4 bits are full-scale counters;
    the actual peak value is in the most-significant 24 bits.
 */
 
@@ -131,7 +131,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin");
    26*3 values are read in ss mode
    14*3 in ds mode, with no gap between values
 */
-#define HDSP_9652_peakBase	7164	
+#define HDSP_9652_peakBase	7164
 #define HDSP_9652_rmsBase	4096
 
 /* c.f. the hdsp_9632_meters_t struct */
@@ -173,12 +173,12 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin");
 #define HDSP_SPDIFEmphasis        (1<<10) /* 0=none, 1=on */
 #define HDSP_SPDIFNonAudio        (1<<11) /* 0=off, 1=on */
 #define HDSP_SPDIFOpticalOut      (1<<12) /* 1=use 1st ADAT connector for SPDIF, 0=do not */
-#define HDSP_SyncRef2             (1<<13) 
-#define HDSP_SPDIFInputSelect0    (1<<14) 
-#define HDSP_SPDIFInputSelect1    (1<<15) 
-#define HDSP_SyncRef0             (1<<16) 
+#define HDSP_SyncRef2             (1<<13)
+#define HDSP_SPDIFInputSelect0    (1<<14)
+#define HDSP_SPDIFInputSelect1    (1<<15)
+#define HDSP_SyncRef0             (1<<16)
 #define HDSP_SyncRef1             (1<<17)
-#define HDSP_AnalogExtensionBoard (1<<18) /* For H9632 cards */ 
+#define HDSP_AnalogExtensionBoard (1<<18) /* For H9632 cards */
 #define HDSP_XLRBreakoutCable     (1<<20) /* For H9632 cards */
 #define HDSP_Midi0InterruptEnable (1<<22)
 #define HDSP_Midi1InterruptEnable (1<<23)
@@ -314,7 +314,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin");
 #define HDSP_TimecodeSync       (1<<27)
 #define HDSP_AEBO          	(1<<28) /* H9632 specific Analog Extension Boards */
 #define HDSP_AEBI		(1<<29) /* 0 = present, 1 = absent */
-#define HDSP_midi0IRQPending    (1<<30) 
+#define HDSP_midi0IRQPending    (1<<30)
 #define HDSP_midi1IRQPending    (1<<31)
 
 #define HDSP_spdifFrequencyMask    (HDSP_spdifFrequency0|HDSP_spdifFrequency1|HDSP_spdifFrequency2)
@@ -391,7 +391,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin");
 #define HDSP_CHANNEL_BUFFER_BYTES    (4*HDSP_CHANNEL_BUFFER_SAMPLES)
 
 /* the size of the area we need to allocate for DMA transfers. the
-   size is the same regardless of the number of channels - the 
+   size is the same regardless of the number of channels - the
    Multiface still uses the same memory area.
 
    Note that we allocate 1 more channel than is apparently needed
@@ -460,7 +460,7 @@ struct hdsp {
 	unsigned char	      qs_in_channels;	     /* quad speed mode for H9632 */
 	unsigned char         ds_in_channels;
 	unsigned char         ss_in_channels;	    /* different for multiface/digiface */
-	unsigned char	      qs_out_channels;	    
+	unsigned char	      qs_out_channels;
 	unsigned char         ds_out_channels;
 	unsigned char         ss_out_channels;
 
@@ -502,9 +502,9 @@ static char channel_map_df_ss[HDSP_MAX_CHANNELS] = {
 
 static char channel_map_mf_ss[HDSP_MAX_CHANNELS] = { /* Multiface */
 	/* Analog */
-	0, 1, 2, 3, 4, 5, 6, 7, 
+	0, 1, 2, 3, 4, 5, 6, 7,
 	/* ADAT 2 */
-	16, 17, 18, 19, 20, 21, 22, 23, 
+	16, 17, 18, 19, 20, 21, 22, 23,
 	/* SPDIF */
 	24, 25,
 	-1, -1, -1, -1, -1, -1, -1, -1
@@ -525,11 +525,11 @@ static char channel_map_H9632_ss[HDSP_MAX_CHANNELS] = {
 	/* SPDIF */
 	8, 9,
 	/* Analog */
-	10, 11, 
+	10, 11,
 	/* AO4S-192 and AI4S-192 extension boards */
 	12, 13, 14, 15,
 	/* others don't exist */
-	-1, -1, -1, -1, -1, -1, -1, -1, 
+	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1
 };
 
@@ -539,7 +539,7 @@ static char channel_map_H9632_ds[HDSP_MAX_CHANNELS] = {
 	/* SPDIF */
 	8, 9,
 	/* Analog */
-	10, 11, 
+	10, 11,
 	/* AO4S-192 and AI4S-192 extension boards */
 	12, 13, 14, 15,
 	/* others don't exist */
@@ -587,7 +587,7 @@ static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_d
 static struct pci_device_id snd_hdsp_ids[] = {
 	{
 		.vendor = PCI_VENDOR_ID_XILINX,
-		.device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP, 
+		.device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP,
 		.subvendor = PCI_ANY_ID,
 		.subdevice = PCI_ANY_ID,
 	}, /* RME Hammerfall-DSP */
@@ -691,19 +691,19 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) {
 	unsigned long flags;
 
 	if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) {
-		
+
 		snd_printk ("Hammerfall-DSP: loading firmware\n");
 
 		hdsp_write (hdsp, HDSP_control2Reg, HDSP_S_PROGRAM);
 		hdsp_write (hdsp, HDSP_fifoData, 0);
-		
+
 		if (hdsp_fifo_wait (hdsp, 0, HDSP_LONG_WAIT)) {
 			snd_printk ("Hammerfall-DSP: timeout waiting for download preparation\n");
 			return -EIO;
 		}
-		
+
 		hdsp_write (hdsp, HDSP_control2Reg, HDSP_S_LOAD);
-		
+
 		for (i = 0; i < 24413; ++i) {
 			hdsp_write(hdsp, HDSP_fifoData, hdsp->firmware_cache[i]);
 			if (hdsp_fifo_wait (hdsp, 127, HDSP_LONG_WAIT)) {
@@ -713,7 +713,7 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) {
 		}
 
 		ssleep(3);
-		
+
 		if (hdsp_fifo_wait (hdsp, 0, HDSP_LONG_WAIT)) {
 			snd_printk ("Hammerfall-DSP: timeout at end of firmware loading\n");
 		    	return -EIO;
@@ -726,15 +726,15 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) {
 #endif
 		hdsp_write (hdsp, HDSP_control2Reg, hdsp->control2_register);
 		snd_printk ("Hammerfall-DSP: finished firmware loading\n");
-		
+
 	}
 	if (hdsp->state & HDSP_InitializationComplete) {
 		snd_printk(KERN_INFO "Hammerfall-DSP: firmware loaded from cache, restoring defaults\n");
 		spin_lock_irqsave(&hdsp->lock, flags);
 		snd_hdsp_set_defaults(hdsp);
-		spin_unlock_irqrestore(&hdsp->lock, flags); 
+		spin_unlock_irqrestore(&hdsp->lock, flags);
 	}
-	
+
 	hdsp->state |= HDSP_FirmwareLoaded;
 
 	return 0;
@@ -743,7 +743,7 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) {
 static int hdsp_get_iobox_version (struct hdsp *hdsp)
 {
 	if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) {
-	
+
 		hdsp_write (hdsp, HDSP_control2Reg, HDSP_PROGRAM);
 		hdsp_write (hdsp, HDSP_fifoData, 0);
 		if (hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT) < 0)
@@ -759,7 +759,7 @@ static int hdsp_get_iobox_version (struct hdsp *hdsp)
 			hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT);
 		} else {
 			hdsp->io_type = Digiface;
-		} 
+		}
 	} else {
 		/* firmware was already loaded, get iobox type */
 		if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version1)
@@ -807,13 +807,13 @@ static int hdsp_check_for_firmware (struct hdsp *hdsp, int load_on_demand)
 
 
 static int hdsp_fifo_wait(struct hdsp *hdsp, int count, int timeout)
-{    
+{
 	int i;
 
 	/* the fifoStatus registers reports on how many words
 	   are available in the command FIFO.
 	*/
-	
+
 	for (i = 0; i < timeout; i++) {
 
 		if ((int)(hdsp_read (hdsp, HDSP_fifoStatus) & 0xff) <= count)
@@ -845,11 +845,11 @@ static int hdsp_write_gain(struct hdsp *hdsp, unsigned int addr, unsigned short
 
 	if (addr >= HDSP_MATRIX_MIXER_SIZE)
 		return -1;
-	
+
 	if (hdsp->io_type == H9652 || hdsp->io_type == H9632) {
 
 		/* from martin bjornsen:
-		   
+
 		   "You can only write dwords to the
 		   mixer memory which contain two
 		   mixer values in the low and high
@@ -868,7 +868,7 @@ static int hdsp_write_gain(struct hdsp *hdsp, unsigned int addr, unsigned short
 
 		hdsp->mixer_matrix[addr] = data;
 
-		
+
 		/* `addr' addresses a 16-bit wide address, but
 		   the address space accessed via hdsp_write
 		   uses byte offsets. put another way, addr
@@ -877,17 +877,17 @@ static int hdsp_write_gain(struct hdsp *hdsp, unsigned int addr, unsigned short
 		   to access 0 to 2703 ...
 		*/
 		ad = addr/2;
-	
-		hdsp_write (hdsp, 4096 + (ad*4), 
-			    (hdsp->mixer_matrix[(addr&0x7fe)+1] << 16) + 
+
+		hdsp_write (hdsp, 4096 + (ad*4),
+			    (hdsp->mixer_matrix[(addr&0x7fe)+1] << 16) +
 			    hdsp->mixer_matrix[addr&0x7fe]);
-		
+
 		return 0;
 
 	} else {
 
 		ad = (addr << 16) + data;
-		
+
 		if (hdsp_fifo_wait(hdsp, 127, HDSP_LONG_WAIT))
 			return -1;
 
@@ -923,7 +923,7 @@ static int hdsp_spdif_sample_rate(struct hdsp *hdsp)
 
 	if (status & HDSP_SPDIFErrorFlag)
 		return 0;
-	
+
 	switch (rate_bits) {
 	case HDSP_spdifFrequency32KHz: return 32000;
 	case HDSP_spdifFrequency44_1KHz: return 44100;
@@ -931,13 +931,13 @@ static int hdsp_spdif_sample_rate(struct hdsp *hdsp)
 	case HDSP_spdifFrequency64KHz: return 64000;
 	case HDSP_spdifFrequency88_2KHz: return 88200;
 	case HDSP_spdifFrequency96KHz: return 96000;
-	case HDSP_spdifFrequency128KHz: 
+	case HDSP_spdifFrequency128KHz:
 		if (hdsp->io_type == H9632) return 128000;
 		break;
-	case HDSP_spdifFrequency176_4KHz: 
+	case HDSP_spdifFrequency176_4KHz:
 		if (hdsp->io_type == H9632) return 176400;
 		break;
-	case HDSP_spdifFrequency192KHz: 
+	case HDSP_spdifFrequency192KHz:
 		if (hdsp->io_type == H9632) return 192000;
 		break;
 	default:
@@ -1048,7 +1048,7 @@ static void hdsp_set_dds_value(struct hdsp *hdsp, int rate)
 {
 	u64 n;
 	u32 r;
-	
+
 	if (rate >= 112000)
 		rate /= 4;
 	else if (rate >= 56000)
@@ -1074,35 +1074,35 @@ static int hdsp_set_rate(struct hdsp *hdsp, int rate, int called_internally)
 	   there is no need for it (e.g. during module
 	   initialization).
 	*/
-	
-	if (!(hdsp->control_register & HDSP_ClockModeMaster)) {	
+
+	if (!(hdsp->control_register & HDSP_ClockModeMaster)) {
 		if (called_internally) {
 			/* request from ctl or card initialization */
 			snd_printk(KERN_ERR "Hammerfall-DSP: device is not running as a clock master: cannot set sample rate.\n");
 			return -1;
-		} else {		
+		} else {
 			/* hw_param request while in AutoSync mode */
 			int external_freq = hdsp_external_sample_rate(hdsp);
 			int spdif_freq = hdsp_spdif_sample_rate(hdsp);
-		
+
 			if ((spdif_freq == external_freq*2) && (hdsp_autosync_ref(hdsp) >= HDSP_AUTOSYNC_FROM_ADAT1))
 				snd_printk(KERN_INFO "Hammerfall-DSP: Detected ADAT in double speed mode\n");
 			else if (hdsp->io_type == H9632 && (spdif_freq == external_freq*4) && (hdsp_autosync_ref(hdsp) >= HDSP_AUTOSYNC_FROM_ADAT1))
-				snd_printk(KERN_INFO "Hammerfall-DSP: Detected ADAT in quad speed mode\n");			
+				snd_printk(KERN_INFO "Hammerfall-DSP: Detected ADAT in quad speed mode\n");
 			else if (rate != external_freq) {
 				snd_printk(KERN_INFO "Hammerfall-DSP: No AutoSync source for requested rate\n");
 				return -1;
-			}		
-		}	
+			}
+		}
 	}
 
 	current_rate = hdsp->system_sample_rate;
 
 	/* Changing from a "single speed" to a "double speed" rate is
 	   not allowed if any substreams are open. This is because
-	   such a change causes a shift in the location of 
+	   such a change causes a shift in the location of
 	   the DMA buffers and a reduction in the number of available
-	   buffers. 
+	   buffers.
 
 	   Note that a similar but essentially insoluble problem
 	   exists for externally-driven rate changes. All we can do
@@ -1110,7 +1110,7 @@ static int hdsp_set_rate(struct hdsp *hdsp, int rate, int called_internally)
 
 	if (rate > 96000 && hdsp->io_type != H9632)
 		return -EINVAL;
-	
+
 	switch (rate) {
 	case 32000:
 		if (current_rate > 48000)
@@ -1200,7 +1200,7 @@ static int hdsp_set_rate(struct hdsp *hdsp, int rate, int called_internally)
 			break;
 		}
 	}
-	
+
 	hdsp->system_sample_rate = rate;
 
 	return 0;
@@ -1266,16 +1266,16 @@ static int snd_hdsp_midi_output_write (struct hdsp_midi *hmidi)
 	unsigned char buf[128];
 
 	/* Output is not interrupt driven */
-		
+
 	spin_lock_irqsave (&hmidi->lock, flags);
 	if (hmidi->output) {
 		if (!snd_rawmidi_transmit_empty (hmidi->output)) {
 			if ((n_pending = snd_hdsp_midi_output_possible (hmidi->hdsp, hmidi->id)) > 0) {
 				if (n_pending > (int)sizeof (buf))
 					n_pending = sizeof (buf);
-				
+
 				if ((to_write = snd_rawmidi_transmit (hmidi->output, buf, n_pending)) > 0) {
-					for (i = 0; i < to_write; ++i) 
+					for (i = 0; i < to_write; ++i)
 						snd_hdsp_midi_write_byte (hmidi->hdsp, hmidi->id, buf[i]);
 				}
 			}
@@ -1346,14 +1346,14 @@ static void snd_hdsp_midi_output_timer(unsigned long data)
 {
 	struct hdsp_midi *hmidi = (struct hdsp_midi *) data;
 	unsigned long flags;
-	
+
 	snd_hdsp_midi_output_write(hmidi);
 	spin_lock_irqsave (&hmidi->lock, flags);
 
 	/* this does not bump hmidi->istimer, because the
 	   kernel automatically removed the timer when it
 	   expired, and we are now adding it back, thus
-	   leaving istimer wherever it was set before.  
+	   leaving istimer wherever it was set before.
 	*/
 
 	if (hmidi->istimer) {
@@ -1522,7 +1522,7 @@ static int snd_hdsp_control_spdif_info(struct snd_kcontrol *kcontrol, struct snd
 static int snd_hdsp_control_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	snd_hdsp_convert_to_aes(&ucontrol->value.iec958, hdsp->creg_spdif);
 	return 0;
 }
@@ -1532,7 +1532,7 @@ static int snd_hdsp_control_spdif_put(struct snd_kcontrol *kcontrol, struct snd_
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	u32 val;
-	
+
 	val = snd_hdsp_convert_from_aes(&ucontrol->value.iec958);
 	spin_lock_irq(&hdsp->lock);
 	change = val != hdsp->creg_spdif;
@@ -1551,7 +1551,7 @@ static int snd_hdsp_control_spdif_stream_info(struct snd_kcontrol *kcontrol, str
 static int snd_hdsp_control_spdif_stream_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	snd_hdsp_convert_to_aes(&ucontrol->value.iec958, hdsp->creg_spdif_stream);
 	return 0;
 }
@@ -1561,7 +1561,7 @@ static int snd_hdsp_control_spdif_stream_put(struct snd_kcontrol *kcontrol, stru
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	u32 val;
-	
+
 	val = snd_hdsp_convert_from_aes(&ucontrol->value.iec958);
 	spin_lock_irq(&hdsp->lock);
 	change = val != hdsp->creg_spdif_stream;
@@ -1623,7 +1623,7 @@ static int snd_hdsp_info_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_
 static int snd_hdsp_get_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.enumerated.item[0] = hdsp_spdif_in(hdsp);
 	return 0;
 }
@@ -1633,7 +1633,7 @@ static int snd_hdsp_put_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_e
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	unsigned int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.enumerated.item[0] % ((hdsp->io_type == H9632) ? 4 : 3);
@@ -1670,7 +1670,7 @@ static int hdsp_set_spdif_output(struct hdsp *hdsp, int out)
 static int snd_hdsp_get_spdif_out(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.integer.value[0] = hdsp_spdif_out(hdsp);
 	return 0;
 }
@@ -1680,7 +1680,7 @@ static int snd_hdsp_put_spdif_out(struct snd_kcontrol *kcontrol, struct snd_ctl_
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	unsigned int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.integer.value[0] & 1;
@@ -1714,7 +1714,7 @@ static int hdsp_set_spdif_professional(struct hdsp *hdsp, int val)
 static int snd_hdsp_get_spdif_professional(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.integer.value[0] = hdsp_spdif_professional(hdsp);
 	return 0;
 }
@@ -1724,7 +1724,7 @@ static int snd_hdsp_put_spdif_professional(struct snd_kcontrol *kcontrol, struct
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	unsigned int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.integer.value[0] & 1;
@@ -1758,7 +1758,7 @@ static int hdsp_set_spdif_emphasis(struct hdsp *hdsp, int val)
 static int snd_hdsp_get_spdif_emphasis(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.integer.value[0] = hdsp_spdif_emphasis(hdsp);
 	return 0;
 }
@@ -1768,7 +1768,7 @@ static int snd_hdsp_put_spdif_emphasis(struct snd_kcontrol *kcontrol, struct snd
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	unsigned int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.integer.value[0] & 1;
@@ -1802,7 +1802,7 @@ static int hdsp_set_spdif_nonaudio(struct hdsp *hdsp, int val)
 static int snd_hdsp_get_spdif_nonaudio(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.integer.value[0] = hdsp_spdif_nonaudio(hdsp);
 	return 0;
 }
@@ -1812,7 +1812,7 @@ static int snd_hdsp_put_spdif_nonaudio(struct snd_kcontrol *kcontrol, struct snd
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	unsigned int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.integer.value[0] & 1;
@@ -1849,7 +1849,7 @@ static int snd_hdsp_info_spdif_sample_rate(struct snd_kcontrol *kcontrol, struct
 static int snd_hdsp_get_spdif_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	switch (hdsp_spdif_sample_rate(hdsp)) {
 	case 32000:
 		ucontrol->value.enumerated.item[0] = 0;
@@ -1879,7 +1879,7 @@ static int snd_hdsp_get_spdif_sample_rate(struct snd_kcontrol *kcontrol, struct
 		ucontrol->value.enumerated.item[0] = 9;
 		break;
 	default:
-		ucontrol->value.enumerated.item[0] = 6;		
+		ucontrol->value.enumerated.item[0] = 6;
 	}
 	return 0;
 }
@@ -1903,7 +1903,7 @@ static int snd_hdsp_info_system_sample_rate(struct snd_kcontrol *kcontrol, struc
 static int snd_hdsp_get_system_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.enumerated.item[0] = hdsp->system_sample_rate;
 	return 0;
 }
@@ -1920,7 +1920,7 @@ static int snd_hdsp_get_system_sample_rate(struct snd_kcontrol *kcontrol, struct
 static int snd_hdsp_info_autosync_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	static char *texts[] = {"32000", "44100", "48000", "64000", "88200", "96000", "None", "128000", "176400", "192000"};	
+	static char *texts[] = {"32000", "44100", "48000", "64000", "88200", "96000", "None", "128000", "176400", "192000"};
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
 	uinfo->count = 1;
 	uinfo->value.enumerated.items = (hdsp->io_type == H9632) ? 10 : 7 ;
@@ -1933,7 +1933,7 @@ static int snd_hdsp_info_autosync_sample_rate(struct snd_kcontrol *kcontrol, str
 static int snd_hdsp_get_autosync_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	switch (hdsp_external_sample_rate(hdsp)) {
 	case 32000:
 		ucontrol->value.enumerated.item[0] = 0;
@@ -1961,9 +1961,9 @@ static int snd_hdsp_get_autosync_sample_rate(struct snd_kcontrol *kcontrol, stru
 		break;
 	case 192000:
 		ucontrol->value.enumerated.item[0] = 9;
-		break;	
+		break;
 	default:
-		ucontrol->value.enumerated.item[0] = 6;		
+		ucontrol->value.enumerated.item[0] = 6;
 	}
 	return 0;
 }
@@ -1989,7 +1989,7 @@ static int hdsp_system_clock_mode(struct hdsp *hdsp)
 static int snd_hdsp_info_system_clock_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
 	static char *texts[] = {"Master", "Slave" };
-	
+
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
 	uinfo->count = 1;
 	uinfo->value.enumerated.items = 2;
@@ -2002,7 +2002,7 @@ static int snd_hdsp_info_system_clock_mode(struct snd_kcontrol *kcontrol, struct
 static int snd_hdsp_get_system_clock_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.enumerated.item[0] = hdsp_system_clock_mode(hdsp);
 	return 0;
 }
@@ -2039,7 +2039,7 @@ static int hdsp_clock_source(struct hdsp *hdsp)
 		case 192000:
 			return 9;
 		default:
-			return 3;	
+			return 3;
 		}
 	} else {
 		return 0;
@@ -2053,7 +2053,7 @@ static int hdsp_set_clock_source(struct hdsp *hdsp, int mode)
 	case HDSP_CLOCK_SOURCE_AUTOSYNC:
 		if (hdsp_external_sample_rate(hdsp) != 0) {
 		    if (!hdsp_set_rate(hdsp, hdsp_external_sample_rate(hdsp), 1)) {
-			hdsp->control_register &= ~HDSP_ClockModeMaster;		
+			hdsp->control_register &= ~HDSP_ClockModeMaster;
 			hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
 			return 0;
 		    }
@@ -2064,7 +2064,7 @@ static int hdsp_set_clock_source(struct hdsp *hdsp, int mode)
 		break;
 	case HDSP_CLOCK_SOURCE_INTERNAL_44_1KHZ:
 		rate = 44100;
-		break;	    
+		break;
 	case HDSP_CLOCK_SOURCE_INTERNAL_48KHZ:
 		rate = 48000;
 		break;
@@ -2099,13 +2099,13 @@ static int snd_hdsp_info_clock_source(struct snd_kcontrol *kcontrol, struct snd_
 {
 	static char *texts[] = {"AutoSync", "Internal 32.0 kHz", "Internal 44.1 kHz", "Internal 48.0 kHz", "Internal 64.0 kHz", "Internal 88.2 kHz", "Internal 96.0 kHz", "Internal 128 kHz", "Internal 176.4 kHz", "Internal 192.0 KHz" };
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
 	uinfo->count = 1;
 	if (hdsp->io_type == H9632)
 	    uinfo->value.enumerated.items = 10;
 	else
-	    uinfo->value.enumerated.items = 7;	
+	    uinfo->value.enumerated.items = 7;
 	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
 		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
 	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
@@ -2115,7 +2115,7 @@ static int snd_hdsp_info_clock_source(struct snd_kcontrol *kcontrol, struct snd_
 static int snd_hdsp_get_clock_source(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.enumerated.item[0] = hdsp_clock_source(hdsp);
 	return 0;
 }
@@ -2125,7 +2125,7 @@ static int snd_hdsp_put_clock_source(struct snd_kcontrol *kcontrol, struct snd_c
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.enumerated.item[0];
@@ -2151,7 +2151,7 @@ static int snd_hdsp_put_clock_source(struct snd_kcontrol *kcontrol, struct snd_c
 static int snd_hdsp_get_clock_source_lock(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.integer.value[0] = hdsp->clock_source_locked;
 	return 0;
 }
@@ -2186,7 +2186,7 @@ static int hdsp_da_gain(struct hdsp *hdsp)
 	case HDSP_DAGainMinus10dBV:
 		return 2;
 	default:
-		return 1;	
+		return 1;
 	}
 }
 
@@ -2201,8 +2201,8 @@ static int hdsp_set_da_gain(struct hdsp *hdsp, int mode)
 		hdsp->control_register |= HDSP_DAGainPlus4dBu;
 		break;
 	case 2:
-		hdsp->control_register |= HDSP_DAGainMinus10dBV;		
-		break;	    
+		hdsp->control_register |= HDSP_DAGainMinus10dBV;
+		break;
 	default:
 		return -1;
 
@@ -2214,7 +2214,7 @@ static int hdsp_set_da_gain(struct hdsp *hdsp, int mode)
 static int snd_hdsp_info_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
 	static char *texts[] = {"Hi Gain", "+4 dBu", "-10 dbV"};
-	
+
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
 	uinfo->count = 1;
 	uinfo->value.enumerated.items = 3;
@@ -2227,7 +2227,7 @@ static int snd_hdsp_info_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_e
 static int snd_hdsp_get_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.enumerated.item[0] = hdsp_da_gain(hdsp);
 	return 0;
 }
@@ -2237,7 +2237,7 @@ static int snd_hdsp_put_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_el
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.enumerated.item[0];
@@ -2271,7 +2271,7 @@ static int hdsp_ad_gain(struct hdsp *hdsp)
 	case HDSP_ADGainLowGain:
 		return 2;
 	default:
-		return 1;	
+		return 1;
 	}
 }
 
@@ -2283,11 +2283,11 @@ static int hdsp_set_ad_gain(struct hdsp *hdsp, int mode)
 		hdsp->control_register |= HDSP_ADGainMinus10dBV;
 		break;
 	case 1:
-		hdsp->control_register |= HDSP_ADGainPlus4dBu;		
+		hdsp->control_register |= HDSP_ADGainPlus4dBu;
 		break;
 	case 2:
-		hdsp->control_register |= HDSP_ADGainLowGain;		
-		break;	    
+		hdsp->control_register |= HDSP_ADGainLowGain;
+		break;
 	default:
 		return -1;
 
@@ -2299,7 +2299,7 @@ static int hdsp_set_ad_gain(struct hdsp *hdsp, int mode)
 static int snd_hdsp_info_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
 	static char *texts[] = {"-10 dBV", "+4 dBu", "Lo Gain"};
-	
+
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
 	uinfo->count = 1;
 	uinfo->value.enumerated.items = 3;
@@ -2312,7 +2312,7 @@ static int snd_hdsp_info_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_e
 static int snd_hdsp_get_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.enumerated.item[0] = hdsp_ad_gain(hdsp);
 	return 0;
 }
@@ -2322,7 +2322,7 @@ static int snd_hdsp_put_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_el
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.enumerated.item[0];
@@ -2356,7 +2356,7 @@ static int hdsp_phone_gain(struct hdsp *hdsp)
 	case HDSP_PhoneGainMinus12dB:
 		return 2;
 	default:
-		return 0;	
+		return 0;
 	}
 }
 
@@ -2368,11 +2368,11 @@ static int hdsp_set_phone_gain(struct hdsp *hdsp, int mode)
 		hdsp->control_register |= HDSP_PhoneGain0dB;
 		break;
 	case 1:
-		hdsp->control_register |= HDSP_PhoneGainMinus6dB;		
+		hdsp->control_register |= HDSP_PhoneGainMinus6dB;
 		break;
 	case 2:
-		hdsp->control_register |= HDSP_PhoneGainMinus12dB;		
-		break;	    
+		hdsp->control_register |= HDSP_PhoneGainMinus12dB;
+		break;
 	default:
 		return -1;
 
@@ -2384,7 +2384,7 @@ static int hdsp_set_phone_gain(struct hdsp *hdsp, int mode)
 static int snd_hdsp_info_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
 	static char *texts[] = {"0 dB", "-6 dB", "-12 dB"};
-	
+
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
 	uinfo->count = 1;
 	uinfo->value.enumerated.items = 3;
@@ -2397,7 +2397,7 @@ static int snd_hdsp_info_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ct
 static int snd_hdsp_get_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.enumerated.item[0] = hdsp_phone_gain(hdsp);
 	return 0;
 }
@@ -2407,7 +2407,7 @@ static int snd_hdsp_put_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.enumerated.item[0];
@@ -2453,7 +2453,7 @@ static int hdsp_set_xlr_breakout_cable(struct hdsp *hdsp, int mode)
 static int snd_hdsp_get_xlr_breakout_cable(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.enumerated.item[0] = hdsp_xlr_breakout_cable(hdsp);
 	return 0;
 }
@@ -2463,7 +2463,7 @@ static int snd_hdsp_put_xlr_breakout_cable(struct snd_kcontrol *kcontrol, struct
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.integer.value[0] & 1;
@@ -2509,7 +2509,7 @@ static int hdsp_set_aeb(struct hdsp *hdsp, int mode)
 static int snd_hdsp_get_aeb(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.enumerated.item[0] = hdsp_aeb(hdsp);
 	return 0;
 }
@@ -2519,7 +2519,7 @@ static int snd_hdsp_put_aeb(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.integer.value[0] & 1;
@@ -2597,7 +2597,7 @@ static int snd_hdsp_info_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd
 {
 	static char *texts[] = {"Word", "IEC958", "ADAT1", "ADAT Sync", "ADAT2", "ADAT3" };
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
 	uinfo->count = 1;
 
@@ -2616,7 +2616,7 @@ static int snd_hdsp_info_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd
 		uinfo->value.enumerated.items = 0;
 		break;
 	}
-		
+
 	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
 		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
 	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
@@ -2626,7 +2626,7 @@ static int snd_hdsp_info_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd
 static int snd_hdsp_get_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.enumerated.item[0] = hdsp_pref_sync_ref(hdsp);
 	return 0;
 }
@@ -2636,7 +2636,7 @@ static int snd_hdsp_put_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd_
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change, max;
 	unsigned int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 
@@ -2685,7 +2685,7 @@ static int hdsp_autosync_ref(struct hdsp *hdsp)
 	case HDSP_SelSyncRef_SPDIF:
 		return HDSP_AUTOSYNC_FROM_SPDIF;
 	case HDSP_SelSyncRefMask:
-		return HDSP_AUTOSYNC_FROM_NONE;	
+		return HDSP_AUTOSYNC_FROM_NONE;
 	case HDSP_SelSyncRef_ADAT1:
 		return HDSP_AUTOSYNC_FROM_ADAT1;
 	case HDSP_SelSyncRef_ADAT2:
@@ -2701,7 +2701,7 @@ static int hdsp_autosync_ref(struct hdsp *hdsp)
 static int snd_hdsp_info_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
 	static char *texts[] = {"Word", "ADAT Sync", "IEC958", "None", "ADAT1", "ADAT2", "ADAT3" };
-	
+
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
 	uinfo->count = 1;
 	uinfo->value.enumerated.items = 7;
@@ -2714,7 +2714,7 @@ static int snd_hdsp_info_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_
 static int snd_hdsp_get_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.enumerated.item[0] = hdsp_autosync_ref(hdsp);
 	return 0;
 }
@@ -2748,7 +2748,7 @@ static int hdsp_set_line_output(struct hdsp *hdsp, int out)
 static int snd_hdsp_get_line_out(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	spin_lock_irq(&hdsp->lock);
 	ucontrol->value.integer.value[0] = hdsp_line_out(hdsp);
 	spin_unlock_irq(&hdsp->lock);
@@ -2760,7 +2760,7 @@ static int snd_hdsp_put_line_out(struct snd_kcontrol *kcontrol, struct snd_ctl_e
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	unsigned int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.integer.value[0] & 1;
@@ -2794,7 +2794,7 @@ static int hdsp_set_precise_pointer(struct hdsp *hdsp, int precise)
 static int snd_hdsp_get_precise_pointer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	spin_lock_irq(&hdsp->lock);
 	ucontrol->value.integer.value[0] = hdsp->precise_ptr;
 	spin_unlock_irq(&hdsp->lock);
@@ -2806,7 +2806,7 @@ static int snd_hdsp_put_precise_pointer(struct snd_kcontrol *kcontrol, struct sn
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	unsigned int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.integer.value[0] & 1;
@@ -2840,7 +2840,7 @@ static int hdsp_set_use_midi_tasklet(struct hdsp *hdsp, int use_tasklet)
 static int snd_hdsp_get_use_midi_tasklet(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	spin_lock_irq(&hdsp->lock);
 	ucontrol->value.integer.value[0] = hdsp->use_midi_tasklet;
 	spin_unlock_irq(&hdsp->lock);
@@ -2852,7 +2852,7 @@ static int snd_hdsp_put_use_midi_tasklet(struct snd_kcontrol *kcontrol, struct s
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	unsigned int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.integer.value[0] & 1;
@@ -2894,12 +2894,12 @@ static int snd_hdsp_get_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem
 
 	source = ucontrol->value.integer.value[0];
 	destination = ucontrol->value.integer.value[1];
-	
+
 	if (source >= hdsp->max_channels)
 		addr = hdsp_playback_to_output_key(hdsp,source-hdsp->max_channels,destination);
 	else
 		addr = hdsp_input_to_output_key(hdsp,source, destination);
-	
+
 	spin_lock_irq(&hdsp->lock);
 	ucontrol->value.integer.value[2] = hdsp_read_gain (hdsp, addr);
 	spin_unlock_irq(&hdsp->lock);
@@ -2947,7 +2947,7 @@ static int snd_hdsp_put_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem
 
 static int snd_hdsp_info_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = {"No Lock", "Lock", "Sync" };	
+	static char *texts[] = {"No Lock", "Lock", "Sync" };
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
 	uinfo->count = 1;
 	uinfo->value.enumerated.items = 3;
@@ -2992,7 +2992,7 @@ static int hdsp_spdif_sync_check(struct hdsp *hdsp)
 	int status = hdsp_read(hdsp, HDSP_statusRegister);
 	if (status & HDSP_SPDIFErrorFlag)
 		return 0;
-	else {	
+	else {
 		if (status & HDSP_SPDIFSync)
 			return 2;
 		else
@@ -3028,7 +3028,7 @@ static int hdsp_adatsync_sync_check(struct hdsp *hdsp)
 			return 1;
 	} else
 		return 0;
-}	
+}
 
 static int snd_hdsp_get_adatsync_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
@@ -3046,17 +3046,17 @@ static int snd_hdsp_get_adatsync_sync_check(struct snd_kcontrol *kcontrol, struc
 }
 
 static int hdsp_adat_sync_check(struct hdsp *hdsp, int idx)
-{	
+{
 	int status = hdsp_read(hdsp, HDSP_statusRegister);
-	
+
 	if (status & (HDSP_Lock0>>idx)) {
 		if (status & (HDSP_Sync0>>idx))
 			return 2;
 		else
-			return 1;		
+			return 1;
 	} else
 		return 0;
-} 
+}
 
 static int snd_hdsp_get_adat_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
@@ -3074,7 +3074,7 @@ static int snd_hdsp_get_adat_sync_check(struct snd_kcontrol *kcontrol, struct sn
 		break;
 	case Multiface:
 	case H9632:
-		if (offset >= 1) 
+		if (offset >= 1)
 			return -EINVAL;
 		break;
 	default:
@@ -3136,7 +3136,7 @@ static int snd_hdsp_info_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ct
 static int snd_hdsp_get_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.enumerated.item[0] = hdsp_dds_offset(hdsp);
 	return 0;
 }
@@ -3146,7 +3146,7 @@ static int snd_hdsp_put_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ctl
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.enumerated.item[0];
@@ -3191,7 +3191,7 @@ static struct snd_kcontrol_new snd_hdsp_controls[] = {
 	.get =		snd_hdsp_control_spdif_mask_get,
 	.private_value = IEC958_AES0_NONAUDIO |
   			 IEC958_AES0_PROFESSIONAL |
-			 IEC958_AES0_CON_EMPHASIS,	                                                                                      
+			 IEC958_AES0_CON_EMPHASIS,
 },
 {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READ,
@@ -3209,7 +3209,7 @@ HDSP_SPDIF_OUT("IEC958 Output also on ADAT1", 0),
 HDSP_SPDIF_PROFESSIONAL("IEC958 Professional Bit", 0),
 HDSP_SPDIF_EMPHASIS("IEC958 Emphasis Bit", 0),
 HDSP_SPDIF_NON_AUDIO("IEC958 Non-audio Bit", 0),
-/* 'Sample Clock Source' complies with the alsa control naming scheme */ 
+/* 'Sample Clock Source' complies with the alsa control naming scheme */
 HDSP_CLOCK_SOURCE("Sample Clock Source", 0),
 {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -3261,7 +3261,7 @@ static int snd_hdsp_create_controls(struct snd_card *card, struct hdsp *hdsp)
 				return err;
 		}
 	}
-	
+
 	/* DA, AD and Phone gain and XLR breakout cable controls for H9632 cards */
 	if (hdsp->io_type == H9632) {
 		for (idx = 0; idx < ARRAY_SIZE(snd_hdsp_9632_controls); idx++) {
@@ -3280,7 +3280,7 @@ static int snd_hdsp_create_controls(struct snd_card *card, struct hdsp *hdsp)
 }
 
 /*------------------------------------------------------------
-   /proc interface 
+   /proc interface
  ------------------------------------------------------------*/
 
 static void
@@ -3319,7 +3319,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
 			}
 		}
 	}
-	
+
 	status = hdsp_read(hdsp, HDSP_statusRegister);
 	status2 = hdsp_read(hdsp, HDSP_status2Register);
 
@@ -3383,17 +3383,17 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
 		break;
 		case HDSP_CLOCK_SOURCE_INTERNAL_192KHZ:
 		clock_source = "Internal 192 kHz";
-		break;	
+		break;
 	default:
-		clock_source = "Error";		
+		clock_source = "Error";
 	}
 	snd_iprintf (buffer, "Sample Clock Source: %s\n", clock_source);
-			
+
 	if (hdsp_system_clock_mode(hdsp))
 		system_clock_mode = "Slave";
 	else
 		system_clock_mode = "Master";
-	
+
 	switch (hdsp_pref_sync_ref (hdsp)) {
 	case HDSP_SYNC_FROM_WORD:
 		pref_sync_ref = "Word Clock";
@@ -3418,7 +3418,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
 		break;
 	}
 	snd_iprintf (buffer, "Preferred Sync Reference: %s\n", pref_sync_ref);
-	
+
 	switch (hdsp_autosync_ref (hdsp)) {
 	case HDSP_AUTOSYNC_FROM_WORD:
 		autosync_ref = "Word Clock";
@@ -3431,7 +3431,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
 		break;
 	case HDSP_AUTOSYNC_FROM_NONE:
 		autosync_ref = "None";
-		break;	
+		break;
 	case HDSP_AUTOSYNC_FROM_ADAT1:
 		autosync_ref = "ADAT1";
 		break;
@@ -3446,14 +3446,14 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
 		break;
 	}
 	snd_iprintf (buffer, "AutoSync Reference: %s\n", autosync_ref);
-	
+
 	snd_iprintf (buffer, "AutoSync Frequency: %d\n", hdsp_external_sample_rate(hdsp));
-	
+
 	snd_iprintf (buffer, "System Clock Mode: %s\n", system_clock_mode);
 
 	snd_iprintf (buffer, "System Clock Frequency: %d\n", hdsp->system_sample_rate);
 	snd_iprintf (buffer, "System Clock Locked: %s\n", hdsp->clock_source_locked ? "Yes" : "No");
-		
+
 	snd_iprintf(buffer, "\n");
 
 	switch (hdsp_spdif_in(hdsp)) {
@@ -3473,7 +3473,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
 		snd_iprintf(buffer, "IEC958 input: ???\n");
 		break;
 	}
-	
+
 	if (hdsp->control_register & HDSP_SPDIFOpticalOut)
 		snd_iprintf(buffer, "IEC958 output: Coaxial & ADAT1\n");
 	else
@@ -3531,13 +3531,13 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
 		snd_iprintf (buffer, "SPDIF: No Lock\n");
 	else
 		snd_iprintf (buffer, "SPDIF: %s\n", x ? "Sync" : "Lock");
-	
+
 	x = status2 & HDSP_wc_sync;
 	if (status2 & HDSP_wc_lock)
 		snd_iprintf (buffer, "Word Clock: %s\n", x ? "Sync" : "Lock");
 	else
 		snd_iprintf (buffer, "Word Clock: No Lock\n");
-	
+
 	x = status & HDSP_TimecodeSync;
 	if (status & HDSP_TimecodeLock)
 		snd_iprintf(buffer, "ADAT Sync: %s\n", x ? "Sync" : "Lock");
@@ -3545,11 +3545,11 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
 		snd_iprintf(buffer, "ADAT Sync: No Lock\n");
 
 	snd_iprintf(buffer, "\n");
-	
+
 	/* Informations about H9632 specific controls */
 	if (hdsp->io_type == H9632) {
 		char *tmp;
-	
+
 		switch (hdsp_ad_gain(hdsp)) {
 		case 0:
 			tmp = "-10 dBV";
@@ -3575,7 +3575,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
 			break;
 		}
 		snd_iprintf(buffer, "DA Gain : %s\n", tmp);
-		
+
 		switch (hdsp_phone_gain(hdsp)) {
 		case 0:
 			tmp = "0 dB";
@@ -3589,8 +3589,8 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
 		}
 		snd_iprintf(buffer, "Phones Gain : %s\n", tmp);
 
-		snd_iprintf(buffer, "XLR Breakout Cable : %s\n", hdsp_xlr_breakout_cable(hdsp) ? "yes" : "no");	
-		
+		snd_iprintf(buffer, "XLR Breakout Cable : %s\n", hdsp_xlr_breakout_cable(hdsp) ? "yes" : "no");
+
 		if (hdsp->control_register & HDSP_AnalogExtensionBoard)
 			snd_iprintf(buffer, "AEB : on (ADAT1 internal)\n");
 		else
@@ -3653,18 +3653,18 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp)
 
 	/* set defaults:
 
-	   SPDIF Input via Coax 
+	   SPDIF Input via Coax
 	   Master clock mode
 	   maximum latency (7 => 2^7 = 8192 samples, 64Kbyte buffer,
 	                    which implies 2 4096 sample, 32Kbyte periods).
-           Enable line out.			    
+           Enable line out.
 	 */
 
-	hdsp->control_register = HDSP_ClockModeMaster | 
-		                 HDSP_SPDIFInputCoaxial | 
-		                 hdsp_encode_latency(7) | 
+	hdsp->control_register = HDSP_ClockModeMaster |
+		                 HDSP_SPDIFInputCoaxial |
+		                 hdsp_encode_latency(7) |
 		                 HDSP_LineOut;
-	
+
 
 	hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
 
@@ -3682,7 +3682,7 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp)
 	hdsp_compute_period_size(hdsp);
 
 	/* silence everything */
-	
+
 	for (i = 0; i < HDSP_MATRIX_MIXER_SIZE; ++i)
 		hdsp->mixer_matrix[i] = MINUS_INFINITY_GAIN;
 
@@ -3690,7 +3690,7 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp)
 		if (hdsp_write_gain (hdsp, i, MINUS_INFINITY_GAIN))
 			return -EIO;
 	}
-	
+
 	/* H9632 specific defaults */
 	if (hdsp->io_type == H9632) {
 		hdsp->control_register |= (HDSP_DAGainPlus4dBu | HDSP_ADGainPlus4dBu | HDSP_PhoneGain0dB);
@@ -3708,12 +3708,12 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp)
 static void hdsp_midi_tasklet(unsigned long arg)
 {
 	struct hdsp *hdsp = (struct hdsp *)arg;
-	
+
 	if (hdsp->midi[0].pending)
 		snd_hdsp_midi_input_read (&hdsp->midi[0]);
 	if (hdsp->midi[1].pending)
 		snd_hdsp_midi_input_read (&hdsp->midi[1]);
-} 
+}
 
 static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id)
 {
@@ -3725,7 +3725,7 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id)
 	unsigned int midi0status;
 	unsigned int midi1status;
 	int schedule = 0;
-	
+
 	status = hdsp_read(hdsp, HDSP_statusRegister);
 
 	audio = status & HDSP_audioIRQPending;
@@ -3739,15 +3739,15 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id)
 
 	midi0status = hdsp_read (hdsp, HDSP_midiStatusIn0) & 0xff;
 	midi1status = hdsp_read (hdsp, HDSP_midiStatusIn1) & 0xff;
-	
+
 	if (audio) {
 		if (hdsp->capture_substream)
 			snd_pcm_period_elapsed(hdsp->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream);
-		
+
 		if (hdsp->playback_substream)
 			snd_pcm_period_elapsed(hdsp->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream);
 	}
-	
+
 	if (midi0 && midi0status) {
 		if (hdsp->use_midi_tasklet) {
 			/* we disable interrupts for this input until processing is done */
@@ -3790,10 +3790,10 @@ static char *hdsp_channel_buffer_location(struct hdsp *hdsp,
 
         if (snd_BUG_ON(channel < 0 || channel >= hdsp->max_channels))
 		return NULL;
-        
+
 	if ((mapped_channel = hdsp->channel_map[channel]) < 0)
 		return NULL;
-	
+
 	if (stream == SNDRV_PCM_STREAM_CAPTURE)
 		return hdsp->capture_buffer + (mapped_channel * HDSP_CHANNEL_BUFFER_BYTES);
 	else
@@ -3986,7 +3986,7 @@ static int snd_hdsp_trigger(struct snd_pcm_substream *substream, int cmd)
 	struct hdsp *hdsp = snd_pcm_substream_chip(substream);
 	struct snd_pcm_substream *other;
 	int running;
-	
+
 	if (hdsp_check_for_iobox (hdsp))
 		return -EIO;
 
@@ -4080,10 +4080,10 @@ static struct snd_pcm_hardware snd_hdsp_playback_subinfo =
 	.formats =		SNDRV_PCM_FMTBIT_S32_LE,
 #endif
 	.rates =		(SNDRV_PCM_RATE_32000 |
-				 SNDRV_PCM_RATE_44100 | 
-				 SNDRV_PCM_RATE_48000 | 
-				 SNDRV_PCM_RATE_64000 | 
-				 SNDRV_PCM_RATE_88200 | 
+				 SNDRV_PCM_RATE_44100 |
+				 SNDRV_PCM_RATE_48000 |
+				 SNDRV_PCM_RATE_64000 |
+				 SNDRV_PCM_RATE_88200 |
 				 SNDRV_PCM_RATE_96000),
 	.rate_min =		32000,
 	.rate_max =		96000,
@@ -4109,10 +4109,10 @@ static struct snd_pcm_hardware snd_hdsp_capture_subinfo =
 	.formats =		SNDRV_PCM_FMTBIT_S32_LE,
 #endif
 	.rates =		(SNDRV_PCM_RATE_32000 |
-				 SNDRV_PCM_RATE_44100 | 
-				 SNDRV_PCM_RATE_48000 | 
-				 SNDRV_PCM_RATE_64000 | 
-				 SNDRV_PCM_RATE_88200 | 
+				 SNDRV_PCM_RATE_44100 |
+				 SNDRV_PCM_RATE_48000 |
+				 SNDRV_PCM_RATE_64000 |
+				 SNDRV_PCM_RATE_88200 |
 				 SNDRV_PCM_RATE_96000),
 	.rate_min =		32000,
 	.rate_max =		96000,
@@ -4191,7 +4191,7 @@ static int snd_hdsp_hw_rule_in_channels_rate(struct snd_pcm_hw_params *params,
 			.max = hdsp->qs_in_channels,
 			.integer = 1,
 		};
-		return snd_interval_refine(c, &t);	
+		return snd_interval_refine(c, &t);
 	} else if (r->min > 48000 && r->max <= 96000) {
 		struct snd_interval t = {
 			.min = hdsp->ds_in_channels,
@@ -4222,7 +4222,7 @@ static int snd_hdsp_hw_rule_out_channels_rate(struct snd_pcm_hw_params *params,
 			.max = hdsp->qs_out_channels,
 			.integer = 1,
 		};
-		return snd_interval_refine(c, &t);	
+		return snd_interval_refine(c, &t);
 	} else if (r->min > 48000 && r->max <= 96000) {
 		struct snd_interval t = {
 			.min = hdsp->ds_out_channels,
@@ -4339,8 +4339,8 @@ static int snd_hdsp_playback_open(struct snd_pcm_substream *substream)
 	if (hdsp->io_type == H9632) {
 		runtime->hw.channels_min = hdsp->qs_out_channels;
 		runtime->hw.channels_max = hdsp->ss_out_channels;
-	}	
-	
+	}
+
 	snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
 			     snd_hdsp_hw_rule_out_channels, hdsp,
 			     SNDRV_PCM_HW_PARAM_CHANNELS, -1);
@@ -4550,7 +4550,7 @@ static int hdsp_get_peak(struct hdsp *hdsp, struct hdsp_peak_rms __user *peak_rm
 				hdsp->iobase + HDSP_playbackRmsLevel + i * 8 + 4,
 				hdsp->iobase + HDSP_playbackRmsLevel + i * 8))
 			return -EFAULT;
-		if (copy_u64_le(&peak_rms->input_rms[i], 
+		if (copy_u64_le(&peak_rms->input_rms[i],
 				hdsp->iobase + HDSP_inputRmsLevel + i * 8 + 4,
 				hdsp->iobase + HDSP_inputRmsLevel + i * 8))
 			return -EFAULT;
@@ -4560,7 +4560,7 @@ static int hdsp_get_peak(struct hdsp *hdsp, struct hdsp_peak_rms __user *peak_rm
 
 static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigned int cmd, unsigned long arg)
 {
-	struct hdsp *hdsp = (struct hdsp *)hw->private_data;	
+	struct hdsp *hdsp = (struct hdsp *)hw->private_data;
 	void __user *argp = (void __user *)arg;
 	int err;
 
@@ -4594,7 +4594,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne
 		struct hdsp_config_info info;
 		unsigned long flags;
 		int i;
-		
+
 		err = hdsp_check_for_iobox(hdsp);
 		if (err < 0)
 			return err;
@@ -4628,7 +4628,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne
 			info.ad_gain = (unsigned char)hdsp_ad_gain(hdsp);
 			info.phone_gain = (unsigned char)hdsp_phone_gain(hdsp);
 			info.xlr_breakout_cable = (unsigned char)hdsp_xlr_breakout_cable(hdsp);
-		
+
 		}
 		if (hdsp->io_type == H9632 || hdsp->io_type == H9652)
 			info.analog_extension_board = (unsigned char)hdsp_aeb(hdsp);
@@ -4639,7 +4639,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne
 	}
 	case SNDRV_HDSP_IOCTL_GET_9632_AEB: {
 		struct hdsp_9632_aeb h9632_aeb;
-		
+
 		if (hdsp->io_type != H9632) return -EINVAL;
 		h9632_aeb.aebi = hdsp->ss_in_channels - H9632_SS_CHANNELS;
 		h9632_aeb.aebo = hdsp->ss_out_channels - H9632_SS_CHANNELS;
@@ -4650,7 +4650,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne
 	case SNDRV_HDSP_IOCTL_GET_VERSION: {
 		struct hdsp_version hdsp_version;
 		int err;
-		
+
 		if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return -EINVAL;
 		if (hdsp->io_type == Undefined) {
 			if ((err = hdsp_get_iobox_version(hdsp)) < 0)
@@ -4666,7 +4666,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne
 		struct hdsp_firmware __user *firmware;
 		u32 __user *firmware_data;
 		int err;
-		
+
 		if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return -EINVAL;
 		/* SNDRV_HDSP_IOCTL_GET_VERSION must have been called */
 		if (hdsp->io_type == Undefined) return -EINVAL;
@@ -4679,25 +4679,25 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne
 
 		if (get_user(firmware_data, &firmware->firmware_data))
 			return -EFAULT;
-		
+
 		if (hdsp_check_for_iobox (hdsp))
 			return -EIO;
 
 		if (copy_from_user(hdsp->firmware_cache, firmware_data, sizeof(hdsp->firmware_cache)) != 0)
 			return -EFAULT;
-		
+
 		hdsp->state |= HDSP_FirmwareCached;
 
 		if ((err = snd_hdsp_load_firmware_from_cache(hdsp)) < 0)
 			return err;
-		
+
 		if (!(hdsp->state & HDSP_InitializationComplete)) {
 			if ((err = snd_hdsp_enable_io(hdsp)) < 0)
 				return err;
-			
-			snd_hdsp_initialize_channels(hdsp);		
+
+			snd_hdsp_initialize_channels(hdsp);
 			snd_hdsp_initialize_midi_flush(hdsp);
-	    
+
 			if ((err = snd_hdsp_create_alsa_devices(hdsp->card, hdsp)) < 0) {
 				snd_printk(KERN_ERR "Hammerfall-DSP: error creating alsa devices\n");
 				return err;
@@ -4744,16 +4744,16 @@ static int snd_hdsp_create_hwdep(struct snd_card *card, struct hdsp *hdsp)
 {
 	struct snd_hwdep *hw;
 	int err;
-	
+
 	if ((err = snd_hwdep_new(card, "HDSP hwdep", 0, &hw)) < 0)
 		return err;
-		
+
 	hdsp->hwdep = hw;
 	hw->private_data = hdsp;
 	strcpy(hw->name, "HDSP hwdep interface");
 
 	hw->ops.ioctl = snd_hdsp_hwdep_ioctl;
-		
+
 	return 0;
 }
 
@@ -4786,24 +4786,24 @@ static void snd_hdsp_9652_enable_mixer (struct hdsp *hdsp)
 static int snd_hdsp_enable_io (struct hdsp *hdsp)
 {
 	int i;
-	
+
 	if (hdsp_fifo_wait (hdsp, 0, 100)) {
 		snd_printk(KERN_ERR "Hammerfall-DSP: enable_io fifo_wait failed\n");
 		return -EIO;
 	}
-	
+
 	for (i = 0; i < hdsp->max_channels; ++i) {
 		hdsp_write (hdsp, HDSP_inputEnable + (4 * i), 1);
 		hdsp_write (hdsp, HDSP_outputEnable + (4 * i), 1);
 	}
-	
+
 	return 0;
 }
 
 static void snd_hdsp_initialize_channels(struct hdsp *hdsp)
 {
 	int status, aebi_channels, aebo_channels;
-	
+
 	switch (hdsp->io_type) {
 	case Digiface:
 		hdsp->card_name = "RME Hammerfall DSP + Digiface";
@@ -4816,7 +4816,7 @@ static void snd_hdsp_initialize_channels(struct hdsp *hdsp)
 		hdsp->ss_in_channels = hdsp->ss_out_channels = H9652_SS_CHANNELS;
 		hdsp->ds_in_channels = hdsp->ds_out_channels = H9652_DS_CHANNELS;
 		break;
-	
+
 	case H9632:
 		status = hdsp_read(hdsp, HDSP_statusRegister);
 		/* HDSP_AEBx bits are low when AEB are connected */
@@ -4836,7 +4836,7 @@ static void snd_hdsp_initialize_channels(struct hdsp *hdsp)
 		hdsp->ss_in_channels = hdsp->ss_out_channels = MULTIFACE_SS_CHANNELS;
 		hdsp->ds_in_channels = hdsp->ds_out_channels = MULTIFACE_DS_CHANNELS;
 		break;
-		
+
 	default:
  		/* should never get here */
 		break;
@@ -4852,12 +4852,12 @@ static void snd_hdsp_initialize_midi_flush (struct hdsp *hdsp)
 static int snd_hdsp_create_alsa_devices(struct snd_card *card, struct hdsp *hdsp)
 {
 	int err;
-	
+
 	if ((err = snd_hdsp_create_pcm(card, hdsp)) < 0) {
 		snd_printk(KERN_ERR "Hammerfall-DSP: Error creating pcm interface\n");
 		return err;
 	}
-	
+
 
 	if ((err = snd_hdsp_create_midi(card, hdsp, 0)) < 0) {
 		snd_printk(KERN_ERR "Hammerfall-DSP: Error creating first midi interface\n");
@@ -4888,19 +4888,19 @@ static int snd_hdsp_create_alsa_devices(struct snd_card *card, struct hdsp *hdsp
 		snd_printk(KERN_ERR "Hammerfall-DSP: Error setting default values\n");
 		return err;
 	}
-	
+
 	if (!(hdsp->state & HDSP_InitializationComplete)) {
 		strcpy(card->shortname, "Hammerfall DSP");
-		sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, 
+		sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name,
 			hdsp->port, hdsp->irq);
-	    
+
 		if ((err = snd_card_register(card)) < 0) {
 			snd_printk(KERN_ERR "Hammerfall-DSP: error registering card\n");
 			return err;
 		}
 		hdsp->state |= HDSP_InitializationComplete;
 	}
-	
+
 	return 0;
 }
 
@@ -4911,7 +4911,7 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp)
 	const char *fwfile;
 	const struct firmware *fw;
 	int err;
-		
+
 	if (hdsp->io_type == H9652 || hdsp->io_type == H9632)
 		return 0;
 	if (hdsp->io_type == Undefined) {
@@ -4920,7 +4920,7 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp)
 		if (hdsp->io_type == H9652 || hdsp->io_type == H9632)
 			return 0;
 	}
-	
+
 	/* caution: max length of firmware filename is 30! */
 	switch (hdsp->io_type) {
 	case Multiface:
@@ -4954,12 +4954,12 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp)
 	memcpy(hdsp->firmware_cache, fw->data, sizeof(hdsp->firmware_cache));
 
 	release_firmware(fw);
-		
+
 	hdsp->state |= HDSP_FirmwareCached;
 
 	if ((err = snd_hdsp_load_firmware_from_cache(hdsp)) < 0)
 		return err;
-		
+
 	if (!(hdsp->state & HDSP_InitializationComplete)) {
 		if ((err = snd_hdsp_enable_io(hdsp)) < 0)
 			return err;
@@ -5006,14 +5006,14 @@ static int __devinit snd_hdsp_create(struct snd_card *card,
 	hdsp->max_channels = 26;
 
 	hdsp->card = card;
-	
+
 	spin_lock_init(&hdsp->lock);
 
 	tasklet_init(&hdsp->midi_tasklet, hdsp_midi_tasklet, (unsigned long)hdsp);
-	
+
 	pci_read_config_word(hdsp->pci, PCI_CLASS_REVISION, &hdsp->firmware_rev);
 	hdsp->firmware_rev &= 0xff;
-	
+
 	/* From Martin Bjoernsen :
 	    "It is important that the card's latency timer register in
 	    the PCI configuration space is set to a value much larger
@@ -5022,7 +5022,7 @@ static int __devinit snd_hdsp_create(struct snd_card *card,
 	    to its maximum 255 to avoid problems with some computers."
 	*/
 	pci_write_config_byte(hdsp->pci, PCI_LATENCY_TIMER, 0xFF);
-	
+
 	strcpy(card->driver, "H-DSP");
 	strcpy(card->mixername, "Xilinx FPGA");
 
@@ -5036,7 +5036,7 @@ static int __devinit snd_hdsp_create(struct snd_card *card,
 	} else {
 		hdsp->card_name = "RME HDSP 9632";
 		hdsp->max_channels = 16;
-		is_9632 = 1;	
+		is_9632 = 1;
 	}
 
 	if ((err = pci_enable_device(pci)) < 0)
@@ -5065,7 +5065,7 @@ static int __devinit snd_hdsp_create(struct snd_card *card,
 
 	if ((err = snd_hdsp_initialize_memory(hdsp)) < 0)
 		return err;
-	
+
 	if (!is_9652 && !is_9632) {
 		/* we wait a maximum of 10 seconds to let freshly
 		 * inserted cardbus cards do their hardware init */
@@ -5092,35 +5092,35 @@ static int __devinit snd_hdsp_create(struct snd_card *card,
 				return err;
 			return 0;
 		} else {
-			snd_printk(KERN_INFO "Hammerfall-DSP: Firmware already present, initializing card.\n");	    
+			snd_printk(KERN_INFO "Hammerfall-DSP: Firmware already present, initializing card.\n");
 			if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version1)
 				hdsp->io_type = Multiface;
-			else 
+			else
 				hdsp->io_type = Digiface;
 		}
 	}
-	
+
 	if ((err = snd_hdsp_enable_io(hdsp)) != 0)
 		return err;
-	
+
 	if (is_9652)
 	        hdsp->io_type = H9652;
-	
+
 	if (is_9632)
 		hdsp->io_type = H9632;
 
 	if ((err = snd_hdsp_create_hwdep(card, hdsp)) < 0)
 		return err;
-	
+
 	snd_hdsp_initialize_channels(hdsp);
 	snd_hdsp_initialize_midi_flush(hdsp);
 
-	hdsp->state |= HDSP_FirmwareLoaded;	
+	hdsp->state |= HDSP_FirmwareLoaded;
 
 	if ((err = snd_hdsp_create_alsa_devices(card, hdsp)) < 0)
 		return err;
 
-	return 0;	
+	return 0;
 }
 
 static int snd_hdsp_free(struct hdsp *hdsp)
@@ -5136,13 +5136,13 @@ static int snd_hdsp_free(struct hdsp *hdsp)
 		free_irq(hdsp->irq, (void *)hdsp);
 
 	snd_hdsp_free_buffers(hdsp);
-	
+
 	if (hdsp->iobase)
 		iounmap(hdsp->iobase);
 
 	if (hdsp->port)
 		pci_release_regions(hdsp->pci);
-		
+
 	pci_disable_device(hdsp->pci);
 	return 0;
 }
@@ -5187,7 +5187,7 @@ static int __devinit snd_hdsp_probe(struct pci_dev *pci,
 	}
 
 	strcpy(card->shortname, "Hammerfall DSP");
-	sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, 
+	sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name,
 		hdsp->port, hdsp->irq);
 
 	if ((err = snd_card_register(card)) < 0) {
-- 
cgit v1.2.3


From c17a1abae2f29047a0f57324240b01609489261b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 23 Feb 2009 09:28:12 +0100
Subject: ALSA: hda - Use snd_hda_codec_get_pincfg() in the rest places

Replace with snd_hda_codec_get_pincfg() in the places where available.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c   | 3 +--
 sound/pci/hda/hda_generic.c | 2 +-
 2 files changed, 2 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 6fa871f66a7..8ec2dfca9a6 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3488,8 +3488,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
 		if (ignore_nids && is_in_nid_list(nid, ignore_nids))
 			continue;
 
-		def_conf = snd_hda_codec_read(codec, nid, 0,
-					      AC_VERB_GET_CONFIG_DEFAULT, 0);
+		def_conf = snd_hda_codec_get_pincfg(codec, nid);
 		if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE)
 			continue;
 		loc = get_defcfg_location(def_conf);
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 65745e96dc7..2c81a683e8f 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -146,7 +146,7 @@ static int add_new_node(struct hda_codec *codec, struct hda_gspec *spec, hda_nid
 	if (node->type == AC_WID_PIN) {
 		node->pin_caps = snd_hda_param_read(codec, node->nid, AC_PAR_PIN_CAP);
 		node->pin_ctl = snd_hda_codec_read(codec, node->nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		node->def_cfg = snd_hda_codec_read(codec, node->nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+		node->def_cfg = snd_hda_codec_get_pincfg(codec, node->nid);
 	}
 
 	if (node->wid_caps & AC_WCAP_OUT_AMP) {
-- 
cgit v1.2.3


From 346ff70fdbe9093947b9494fe714c89cafcceade Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 23 Feb 2009 09:42:57 +0100
Subject: ALSA: hda - Rename {override,cur}_pin with {user,driver}_pin

Rename from override_pin and cur_pin with user_pin and driver_pin,
respectively, to be a bit more intuitive.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 18 +++++++++---------
 sound/pci/hda/hda_codec.h |  4 ++--
 sound/pci/hda/hda_hwdep.c | 32 ++++++++++++++++----------------
 3 files changed, 27 insertions(+), 27 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 8ec2dfca9a6..df9453d0122 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -755,7 +755,7 @@ int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list,
 int snd_hda_codec_set_pincfg(struct hda_codec *codec,
 			     hda_nid_t nid, unsigned int cfg)
 {
-	return snd_hda_add_pincfg(codec, &codec->cur_pins, nid, cfg);
+	return snd_hda_add_pincfg(codec, &codec->driver_pins, nid, cfg);
 }
 EXPORT_SYMBOL_HDA(snd_hda_codec_set_pincfg);
 
@@ -764,11 +764,11 @@ unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid)
 {
 	struct hda_pincfg *pin;
 
-	pin = look_up_pincfg(codec, &codec->cur_pins, nid);
+	pin = look_up_pincfg(codec, &codec->driver_pins, nid);
 	if (pin)
 		return pin->cfg;
 #ifdef CONFIG_SND_HDA_HWDEP
-	pin = look_up_pincfg(codec, &codec->override_pins, nid);
+	pin = look_up_pincfg(codec, &codec->user_pins, nid);
 	if (pin)
 		return pin->cfg;
 #endif
@@ -797,12 +797,12 @@ static void free_hda_cache(struct hda_cache_rec *cache);
 /* restore the initial pin cfgs and release all pincfg lists */
 static void restore_init_pincfgs(struct hda_codec *codec)
 {
-	/* first free cur_pins and override_pins, then call restore_pincfg
+	/* first free driver_pins and user_pins, then call restore_pincfg
 	 * so that only the values in init_pins are restored
 	 */
-	snd_array_free(&codec->cur_pins);
+	snd_array_free(&codec->driver_pins);
 #ifdef CONFIG_SND_HDA_HWDEP
-	snd_array_free(&codec->override_pins);
+	snd_array_free(&codec->user_pins);
 #endif
 	restore_pincfgs(codec);
 	snd_array_free(&codec->init_pins);
@@ -874,7 +874,7 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr
 	init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head));
 	snd_array_init(&codec->mixers, sizeof(struct snd_kcontrol *), 32);
 	snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16);
-	snd_array_init(&codec->cur_pins, sizeof(struct hda_pincfg), 16);
+	snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16);
 	if (codec->bus->modelname) {
 		codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL);
 		if (!codec->modelname) {
@@ -1463,8 +1463,8 @@ void snd_hda_codec_reset(struct hda_codec *codec)
 	free_hda_cache(&codec->cmd_cache);
 	init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info));
 	init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head));
-	/* free only cur_pins so that init_pins + override_pins are restored */
-	snd_array_free(&codec->cur_pins);
+	/* free only driver_pins so that init_pins + user_pins are restored */
+	snd_array_free(&codec->driver_pins);
 	restore_pincfgs(codec);
 	codec->num_pcms = 0;
 	codec->pcm_info = NULL;
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 6d01a8058f0..2ea628478a9 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -779,13 +779,13 @@ struct hda_codec {
 	unsigned int spdif_in_enable;	/* SPDIF input enable? */
 	hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */
 	struct snd_array init_pins;	/* initial (BIOS) pin configurations */
-	struct snd_array cur_pins;	/* current pin configurations */
+	struct snd_array driver_pins;	/* pin configs set by codec parser */
 
 #ifdef CONFIG_SND_HDA_HWDEP
 	struct snd_hwdep *hwdep;	/* assigned hwdep device */
 	struct snd_array init_verbs;	/* additional init verbs */
 	struct snd_array hints;		/* additional hints */
-	struct snd_array override_pins;	/* default pin configs to override */
+	struct snd_array user_pins;	/* default pin configs to override */
 #endif
 
 	/* misc flags */
diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index 71039a6dec2..c660383ef38 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -109,7 +109,7 @@ static void clear_hwdep_elements(struct hda_codec *codec)
 	for (i = 0; i < codec->hints.used; i++, head++)
 		kfree(*head);
 	snd_array_free(&codec->hints);
-	snd_array_free(&codec->override_pins);
+	snd_array_free(&codec->user_pins);
 }
 
 static void hwdep_free(struct snd_hwdep *hwdep)
@@ -142,7 +142,7 @@ int /*__devinit*/ snd_hda_create_hwdep(struct hda_codec *codec)
 
 	snd_array_init(&codec->init_verbs, sizeof(struct hda_verb), 32);
 	snd_array_init(&codec->hints, sizeof(char *), 32);
-	snd_array_init(&codec->override_pins, sizeof(struct hda_pincfg), 16);
+	snd_array_init(&codec->user_pins, sizeof(struct hda_pincfg), 16);
 
 	return 0;
 }
@@ -340,29 +340,29 @@ static ssize_t init_pin_configs_show(struct device *dev,
 	return pin_configs_show(codec, &codec->init_pins, buf);
 }
 
-static ssize_t override_pin_configs_show(struct device *dev,
-					 struct device_attribute *attr,
-					 char *buf)
+static ssize_t user_pin_configs_show(struct device *dev,
+				     struct device_attribute *attr,
+				     char *buf)
 {
 	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
 	struct hda_codec *codec = hwdep->private_data;
-	return pin_configs_show(codec, &codec->override_pins, buf);
+	return pin_configs_show(codec, &codec->user_pins, buf);
 }
 
-static ssize_t cur_pin_configs_show(struct device *dev,
-				    struct device_attribute *attr,
-				    char *buf)
+static ssize_t driver_pin_configs_show(struct device *dev,
+				       struct device_attribute *attr,
+				       char *buf)
 {
 	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
 	struct hda_codec *codec = hwdep->private_data;
-	return pin_configs_show(codec, &codec->cur_pins, buf);
+	return pin_configs_show(codec, &codec->driver_pins, buf);
 }
 
 #define MAX_PIN_CONFIGS		32
 
-static ssize_t override_pin_configs_store(struct device *dev,
-					  struct device_attribute *attr,
-					  const char *buf, size_t count)
+static ssize_t user_pin_configs_store(struct device *dev,
+				      struct device_attribute *attr,
+				      const char *buf, size_t count)
 {
 	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
 	struct hda_codec *codec = hwdep->private_data;
@@ -373,7 +373,7 @@ static ssize_t override_pin_configs_store(struct device *dev,
 		return -EINVAL;
 	if (!nid)
 		return -EINVAL;
-	err = snd_hda_add_pincfg(codec, &codec->override_pins, nid, cfg);
+	err = snd_hda_add_pincfg(codec, &codec->user_pins, nid, cfg);
 	if (err < 0)
 		return err;
 	return count;
@@ -397,8 +397,8 @@ static struct device_attribute codec_attrs[] = {
 	CODEC_ATTR_WO(init_verbs),
 	CODEC_ATTR_WO(hints),
 	CODEC_ATTR_RO(init_pin_configs),
-	CODEC_ATTR_RW(override_pin_configs),
-	CODEC_ATTR_RO(cur_pin_configs),
+	CODEC_ATTR_RW(user_pin_configs),
+	CODEC_ATTR_RO(driver_pin_configs),
 	CODEC_ATTR_WO(reconfig),
 	CODEC_ATTR_WO(clear),
 };
-- 
cgit v1.2.3


From 5e7b8e0d87091ae21b291588817b5359a5e00795 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 23 Feb 2009 09:45:59 +0100
Subject: ALSA: hda - Make user_pin overriding the driver setup

Make user_pin overriding even the driver pincfg, e.g. the static / fixed
pin config table in patch_sigmatel.c.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 16 ++++++++++++----
 1 file changed, 12 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index df9453d0122..a13480fa8e7 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -739,7 +739,9 @@ int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list,
 		       hda_nid_t nid, unsigned int cfg)
 {
 	struct hda_pincfg *pin;
+	unsigned int oldcfg;
 
+	oldcfg = snd_hda_codec_get_pincfg(codec, nid);
 	pin = look_up_pincfg(codec, list, nid);
 	if (!pin) {
 		pin = snd_array_new(list);
@@ -748,7 +750,13 @@ int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list,
 		pin->nid = nid;
 	}
 	pin->cfg = cfg;
-	set_pincfg(codec, nid, cfg);
+
+	/* change only when needed; e.g. if the pincfg is already present
+	 * in user_pins[], don't write it
+	 */
+	cfg = snd_hda_codec_get_pincfg(codec, nid);
+	if (oldcfg != cfg)
+		set_pincfg(codec, nid, cfg);
 	return 0;
 }
 
@@ -764,14 +772,14 @@ unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid)
 {
 	struct hda_pincfg *pin;
 
-	pin = look_up_pincfg(codec, &codec->driver_pins, nid);
-	if (pin)
-		return pin->cfg;
 #ifdef CONFIG_SND_HDA_HWDEP
 	pin = look_up_pincfg(codec, &codec->user_pins, nid);
 	if (pin)
 		return pin->cfg;
 #endif
+	pin = look_up_pincfg(codec, &codec->driver_pins, nid);
+	if (pin)
+		return pin->cfg;
 	pin = look_up_pincfg(codec, &codec->init_pins, nid);
 	if (pin)
 		return pin->cfg;
-- 
cgit v1.2.3


From 13c989beba166b470b1e6b0fa117148bcbfa3dd1 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 23 Feb 2009 11:33:34 +0100
Subject: ALSA: hda - Don't give over 0dB volume for AD1984A HP laptops

Set the upper limit 0dB to the volume of mixer amp 0x20 for
AD1984A HP laptops.  The overloaded volume may damage the internal
speaker.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c | 8 ++++++++
 1 file changed, 8 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 2c58d7b05ab..b1680284146 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3986,6 +3986,14 @@ static int patch_ad1884a(struct hda_codec *codec)
 		spec->multiout.dig_out_nid = 0;
 		codec->patch_ops.unsol_event = ad1884a_hp_unsol_event;
 		codec->patch_ops.init = ad1884a_hp_init;
+		/* set the upper-limit for mixer amp to 0dB for avoiding the
+		 * possible damage by overloading
+		 */
+		snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT,
+					  (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
+					  (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
+					  (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
+					  (1 << AC_AMPCAP_MUTE_SHIFT));
 		break;
 	case AD1884A_THINKPAD:
 		spec->mixers[0] = ad1984a_thinkpad_mixers;
-- 
cgit v1.2.3


From a65d629ceb4cff5e7d5edadfd6bf1f64c370a517 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 23 Feb 2009 16:57:04 +0100
Subject: ALSA: hda - Add pseudo device-locking for clear/reconfig

Added the pseudo device-locking using card->shutdown flag to avoid
the crash via clear/reconfig during operations.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 54 +++++++++++++++++++++++++++++++++++++++++++----
 sound/pci/hda/hda_hwdep.c | 15 +++++++++++--
 sound/pci/hda/hda_local.h |  2 +-
 3 files changed, 64 insertions(+), 7 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index a13480fa8e7..5dceee8a113 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1445,9 +1445,52 @@ void snd_hda_ctls_clear(struct hda_codec *codec)
 	snd_array_free(&codec->mixers);
 }
 
-void snd_hda_codec_reset(struct hda_codec *codec)
+/* pseudo device locking
+ * toggle card->shutdown to allow/disallow the device access (as a hack)
+ */
+static int hda_lock_devices(struct snd_card *card)
 {
-	int i;
+	spin_lock(&card->files_lock);
+	if (card->shutdown) {
+		spin_unlock(&card->files_lock);
+		return -EINVAL;
+	}
+	card->shutdown = 1;
+	spin_unlock(&card->files_lock);
+	return 0;
+}
+
+static void hda_unlock_devices(struct snd_card *card)
+{
+	spin_lock(&card->files_lock);
+	card->shutdown = 0;
+	spin_unlock(&card->files_lock);
+}
+
+int snd_hda_codec_reset(struct hda_codec *codec)
+{
+	struct snd_card *card = codec->bus->card;
+	int i, pcm;
+
+	if (hda_lock_devices(card) < 0)
+		return -EBUSY;
+	/* check whether the codec isn't used by any mixer or PCM streams */
+	if (!list_empty(&card->ctl_files)) {
+		hda_unlock_devices(card);
+		return -EBUSY;
+	}
+	for (pcm = 0; pcm < codec->num_pcms; pcm++) {
+		struct hda_pcm *cpcm = &codec->pcm_info[pcm];
+		if (!cpcm->pcm)
+			continue;
+		if (cpcm->pcm->streams[0].substream_opened ||
+		    cpcm->pcm->streams[1].substream_opened) {
+			hda_unlock_devices(card);
+			return -EBUSY;
+		}
+	}
+
+	/* OK, let it free */
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	cancel_delayed_work(&codec->power_work);
@@ -1457,8 +1500,7 @@ void snd_hda_codec_reset(struct hda_codec *codec)
 	/* relase PCMs */
 	for (i = 0; i < codec->num_pcms; i++) {
 		if (codec->pcm_info[i].pcm) {
-			snd_device_free(codec->bus->card,
-					codec->pcm_info[i].pcm);
+			snd_device_free(card, codec->pcm_info[i].pcm);
 			clear_bit(codec->pcm_info[i].device,
 				  codec->bus->pcm_dev_bits);
 		}
@@ -1479,6 +1521,10 @@ void snd_hda_codec_reset(struct hda_codec *codec)
 	codec->preset = NULL;
 	module_put(codec->owner);
 	codec->owner = NULL;
+
+	/* allow device access again */
+	hda_unlock_devices(card);
+	return 0;
 }
 #endif /* CONFIG_SND_HDA_RECONFIG */
 
diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index c660383ef38..4af484b8240 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -155,7 +155,13 @@ int /*__devinit*/ snd_hda_create_hwdep(struct hda_codec *codec)
 
 static int clear_codec(struct hda_codec *codec)
 {
-	snd_hda_codec_reset(codec);
+	int err;
+
+	err = snd_hda_codec_reset(codec);
+	if (err < 0) {
+		snd_printk(KERN_ERR "The codec is being used, can't free.\n");
+		return err;
+	}
 	clear_hwdep_elements(codec);
 	return 0;
 }
@@ -165,7 +171,12 @@ static int reconfig_codec(struct hda_codec *codec)
 	int err;
 
 	snd_printk(KERN_INFO "hda-codec: reconfiguring\n");
-	snd_hda_codec_reset(codec);
+	err = snd_hda_codec_reset(codec);
+	if (err < 0) {
+		snd_printk(KERN_ERR
+			   "The codec is being used, can't reconfigure.\n");
+		return err;
+	}
 	err = snd_hda_codec_configure(codec);
 	if (err < 0)
 		return err;
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 84e2cf644fd..4bd82a37a4c 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -98,7 +98,7 @@ struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec,
 					    const char *name);
 int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
 			unsigned int *tlv, const char **slaves);
-void snd_hda_codec_reset(struct hda_codec *codec);
+int snd_hda_codec_reset(struct hda_codec *codec);
 int snd_hda_codec_configure(struct hda_codec *codec);
 
 /* amp value bits */
-- 
cgit v1.2.3


From 873dc78a8676b7ba6260b1d74c50d8ea5025ecbe Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 25 Feb 2009 18:12:13 +0100
Subject: ALSA: hda - Clean up / fix quirks for HP laptops with AD1984A

Use SND_PCI_QUIRK_MASK() to clean up / support better HP laptops with
AD1984A codec.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c | 5 ++---
 1 file changed, 2 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 0253cb93aa7..5bb48ee8b6c 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3923,9 +3923,8 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP),
 	SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE),
 	SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x3070, "HP", AD1884A_MOBILE),
-	SND_PCI_QUIRK(0x103c, 0x30e6, "HP 6730b", AD1884A_LAPTOP),
-	SND_PCI_QUIRK(0x103c, 0x30e7, "HP EliteBook 8530p", AD1884A_LAPTOP),
-	SND_PCI_QUIRK(0x103c, 0x3614, "HP 6730s", AD1884A_LAPTOP),
+	SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30e0, "HP laptop", AD1884A_LAPTOP),
+	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP),
 	SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD),
 	{}
 };
-- 
cgit v1.2.3


From f872a9194cb006994d47a58efc875218594e6072 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 26 Feb 2009 00:57:01 +0100
Subject: ALSA: hda - Clean up / fix quirk for Sony laptops with ALC262

Clean up / fix quirk entries for Sony laptops with ALC262 codec
using NSD_PCI_QUIRK_MASK().

This also fixes the kernel bug #12780
	http://bugme.linux-foundation.org/show_bug.cgi?id=12780

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 6 ++----
 1 file changed, 2 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 50ae8f33af5..d670d33cfa1 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -10824,10 +10824,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD),
 	SND_PCI_QUIRK(0x104d, 0x8203, "Sony UX-90", ALC262_HIPPO),
 	SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD),
-	SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD),
-	SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD),
-	SND_PCI_QUIRK(0x104d, 0x9033, "Sony VAIO VGN-SR19XN",
-		      ALC262_SONY_ASSAMD),
+	SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO",
+			   ALC262_SONY_ASSAMD),
 	SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1",
 		      ALC262_TOSHIBA_RX1),
 	SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06),
-- 
cgit v1.2.3


From 930738de602d2ceb0d1c1b368fe2a8d2a974ab72 Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Thu, 26 Feb 2009 09:27:20 +0100
Subject: sound: virtuoso: add Xonar Essence STX support

Add support for the Asus Xonar Essence STX sound card.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/Kconfig           |   3 +-
 sound/pci/oxygen/virtuoso.c | 192 ++++++++++++++++++++++++++++++++++++++++++++
 2 files changed, 194 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 82b9bddcdcd..21d117ada84 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -744,7 +744,8 @@ config SND_VIRTUOSO
 	select SND_OXYGEN_LIB
 	help
 	  Say Y here to include support for sound cards based on the
-	  Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2 and D2X.
+	  Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X, and
+	  Essence STX.
 	  Support for the HDAV1.3 (Deluxe) is very experimental.
 
 	  To compile this driver as a module, choose M here: the module
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index 00dc97806f1..bc5ce11c8b1 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -112,6 +112,34 @@
  * CS4362A: AD0 <- 0
  */
 
+/*
+ * Xonar Essence STX
+ * -----------------
+ *
+ * CMI8788:
+ *
+ * I²C <-> PCM1792A
+ *
+ * GPI 0 <- external power present
+ *
+ * GPIO 0 -> enable output to speakers
+ * GPIO 1 -> route HP to front panel (0) or rear jack (1)
+ * GPIO 2 -> M0 of CS5381
+ * GPIO 3 -> M1 of CS5381
+ * GPIO 7 -> route output to speaker jacks (0) or HP (1)
+ * GPIO 8 -> route input jack to line-in (0) or mic-in (1)
+ *
+ * PCM1792A:
+ *
+ * AD0 <- 0
+ *
+ * H6 daughterboard
+ * ----------------
+ *
+ * GPIO 4 <- 0
+ * GPIO 5 <- 0
+ */
+
 #include <linux/pci.h>
 #include <linux/delay.h>
 #include <linux/mutex.h>
@@ -152,6 +180,7 @@ enum {
 	MODEL_DX,
 	MODEL_HDAV,	/* without daughterboard */
 	MODEL_HDAV_H6,	/* with H6 daughterboard */
+	MODEL_STX,
 };
 
 static struct pci_device_id xonar_ids[] __devinitdata = {
@@ -160,6 +189,7 @@ static struct pci_device_id xonar_ids[] __devinitdata = {
 	{ OXYGEN_PCI_SUBID(0x1043, 0x82b7), .driver_data = MODEL_D2X },
 	{ OXYGEN_PCI_SUBID(0x1043, 0x8314), .driver_data = MODEL_HDAV },
 	{ OXYGEN_PCI_SUBID(0x1043, 0x834f), .driver_data = MODEL_D1 },
+	{ OXYGEN_PCI_SUBID(0x1043, 0x835c), .driver_data = MODEL_STX },
 	{ OXYGEN_PCI_SUBID_BROKEN_EEPROM },
 	{ }
 };
@@ -184,6 +214,9 @@ MODULE_DEVICE_TABLE(pci, xonar_ids);
 #define GPIO_HDAV_DB_H6		0x0000
 #define GPIO_HDAV_DB_XX		0x0020
 
+#define GPIO_ST_HP_REAR		0x0002
+#define GPIO_ST_HP		0x0080
+
 #define I2C_DEVICE_PCM1796(i)	(0x98 + ((i) << 1))	/* 10011, ADx=i, /W=0 */
 #define I2C_DEVICE_CS4398	0x9e	/* 10011, AD1=1, AD0=1, /W=0 */
 #define I2C_DEVICE_CS4362A	0x30	/* 001100, AD0=0, /W=0 */
@@ -497,6 +530,36 @@ static void xonar_hdav_init(struct oxygen *chip)
 	snd_component_add(chip->card, "CS5381");
 }
 
+static void xonar_stx_init(struct oxygen *chip)
+{
+	struct xonar_data *data = chip->model_data;
+
+	oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS,
+		       OXYGEN_2WIRE_LENGTH_8 |
+		       OXYGEN_2WIRE_INTERRUPT_MASK |
+		       OXYGEN_2WIRE_SPEED_FAST);
+
+	data->anti_pop_delay = 100;
+	data->dacs = 1;
+	data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE;
+	data->ext_power_reg = OXYGEN_GPI_DATA;
+	data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK;
+	data->ext_power_bit = GPI_DX_EXT_POWER;
+	data->pcm1796_oversampling = PCM1796_OS_64;
+
+	pcm1796_init(chip);
+
+	oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL,
+			  GPIO_DX_INPUT_ROUTE | GPIO_ST_HP_REAR | GPIO_ST_HP);
+	oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA,
+			    GPIO_DX_INPUT_ROUTE | GPIO_ST_HP_REAR | GPIO_ST_HP);
+
+	xonar_common_init(chip);
+
+	snd_component_add(chip->card, "PCM1792A");
+	snd_component_add(chip->card, "CS5381");
+}
+
 static void xonar_disable_output(struct oxygen *chip)
 {
 	struct xonar_data *data = chip->model_data;
@@ -524,6 +587,11 @@ static void xonar_hdav_cleanup(struct oxygen *chip)
 	xonar_disable_output(chip);
 }
 
+static void xonar_st_cleanup(struct oxygen *chip)
+{
+	xonar_disable_output(chip);
+}
+
 static void xonar_d2_suspend(struct oxygen *chip)
 {
 	xonar_d2_cleanup(chip);
@@ -540,6 +608,11 @@ static void xonar_hdav_suspend(struct oxygen *chip)
 	msleep(2);
 }
 
+static void xonar_st_suspend(struct oxygen *chip)
+{
+	xonar_st_cleanup(chip);
+}
+
 static void xonar_d2_resume(struct oxygen *chip)
 {
 	pcm1796_init(chip);
@@ -567,6 +640,12 @@ static void xonar_hdav_resume(struct oxygen *chip)
 	xonar_enable_output(chip);
 }
 
+static void xonar_st_resume(struct oxygen *chip)
+{
+	pcm1796_init(chip);
+	xonar_enable_output(chip);
+}
+
 static void xonar_hdav_pcm_hardware_filter(unsigned int channel,
 					   struct snd_pcm_hardware *hardware)
 {
@@ -746,6 +825,72 @@ static const struct snd_kcontrol_new front_panel_switch = {
 	.private_value = GPIO_DX_FRONT_PANEL,
 };
 
+static int st_output_switch_info(struct snd_kcontrol *ctl,
+				 struct snd_ctl_elem_info *info)
+{
+	static const char *const names[3] = {
+		"Speakers", "Headphones", "FP Headphones"
+	};
+
+	info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	info->count = 1;
+	info->value.enumerated.items = 3;
+	if (info->value.enumerated.item >= 3)
+		info->value.enumerated.item = 2;
+	strcpy(info->value.enumerated.name, names[info->value.enumerated.item]);
+	return 0;
+}
+
+static int st_output_switch_get(struct snd_kcontrol *ctl,
+				struct snd_ctl_elem_value *value)
+{
+	struct oxygen *chip = ctl->private_data;
+	u16 gpio;
+
+	gpio = oxygen_read16(chip, OXYGEN_GPIO_DATA);
+	if (!(gpio & GPIO_ST_HP))
+		value->value.enumerated.item[0] = 0;
+	else if (gpio & GPIO_ST_HP_REAR)
+		value->value.enumerated.item[0] = 1;
+	else
+		value->value.enumerated.item[0] = 2;
+	return 0;
+}
+
+
+static int st_output_switch_put(struct snd_kcontrol *ctl,
+				struct snd_ctl_elem_value *value)
+{
+	struct oxygen *chip = ctl->private_data;
+	u16 gpio_old, gpio;
+
+	mutex_lock(&chip->mutex);
+	gpio_old = oxygen_read16(chip, OXYGEN_GPIO_DATA);
+	gpio = gpio_old;
+	switch (value->value.enumerated.item[0]) {
+	case 0:
+		gpio &= ~(GPIO_ST_HP | GPIO_ST_HP_REAR);
+		break;
+	case 1:
+		gpio |= GPIO_ST_HP | GPIO_ST_HP_REAR;
+		break;
+	case 2:
+		gpio = (gpio | GPIO_ST_HP) & ~GPIO_ST_HP_REAR;
+		break;
+	}
+	oxygen_write16(chip, OXYGEN_GPIO_DATA, gpio);
+	mutex_unlock(&chip->mutex);
+	return gpio != gpio_old;
+}
+
+static const struct snd_kcontrol_new st_output_switch = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Analog Output",
+	.info = st_output_switch_info,
+	.get = st_output_switch_get,
+	.put = st_output_switch_put,
+};
+
 static void xonar_line_mic_ac97_switch(struct oxygen *chip,
 				       unsigned int reg, unsigned int mute)
 {
@@ -776,6 +921,15 @@ static int xonar_d1_control_filter(struct snd_kcontrol_new *template)
 	return 0;
 }
 
+static int xonar_st_control_filter(struct snd_kcontrol_new *template)
+{
+	if (!strncmp(template->name, "CD Capture ", 11))
+		return 1; /* no CD input */
+	if (!strcmp(template->name, "Stereo Upmixing"))
+		return 1; /* stereo only - we don't need upmixing */
+	return 0;
+}
+
 static int xonar_d2_mixer_init(struct oxygen *chip)
 {
 	return snd_ctl_add(chip->card, snd_ctl_new1(&alt_switch, chip));
@@ -786,6 +940,11 @@ static int xonar_d1_mixer_init(struct oxygen *chip)
 	return snd_ctl_add(chip->card, snd_ctl_new1(&front_panel_switch, chip));
 }
 
+static int xonar_st_mixer_init(struct oxygen *chip)
+{
+	return snd_ctl_add(chip->card, snd_ctl_new1(&st_output_switch, chip));
+}
+
 static const struct oxygen_model model_xonar_d2 = {
 	.longname = "Asus Virtuoso 200",
 	.chip = "AV200",
@@ -872,6 +1031,33 @@ static const struct oxygen_model model_xonar_hdav = {
 	.adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
 };
 
+static const struct oxygen_model model_xonar_st = {
+	.longname = "Asus Virtuoso 100",
+	.chip = "AV200",
+	.init = xonar_stx_init,
+	.control_filter = xonar_st_control_filter,
+	.mixer_init = xonar_st_mixer_init,
+	.cleanup = xonar_st_cleanup,
+	.suspend = xonar_st_suspend,
+	.resume = xonar_st_resume,
+	.set_dac_params = set_pcm1796_params,
+	.set_adc_params = set_cs53x1_params,
+	.update_dac_volume = update_pcm1796_volume,
+	.update_dac_mute = update_pcm1796_mute,
+	.ac97_switch = xonar_line_mic_ac97_switch,
+	.dac_tlv = pcm1796_db_scale,
+	.model_data_size = sizeof(struct xonar_data),
+	.device_config = PLAYBACK_0_TO_I2S |
+			 PLAYBACK_1_TO_SPDIF |
+			 CAPTURE_0_FROM_I2S_2,
+	.dac_channels = 2,
+	.dac_volume_min = 255 - 2*60,
+	.dac_volume_max = 255,
+	.function_flags = OXYGEN_FUNCTION_2WIRE,
+	.dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
+	.adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
+};
+
 static int __devinit get_xonar_model(struct oxygen *chip,
 				     const struct pci_device_id *id)
 {
@@ -881,6 +1067,7 @@ static int __devinit get_xonar_model(struct oxygen *chip,
 		[MODEL_D2]	= &model_xonar_d2,
 		[MODEL_D2X]	= &model_xonar_d2,
 		[MODEL_HDAV]	= &model_xonar_hdav,
+		[MODEL_STX]	= &model_xonar_st,
 	};
 	static const char *const names[] = {
 		[MODEL_D1]	= "Xonar D1",
@@ -889,6 +1076,7 @@ static int __devinit get_xonar_model(struct oxygen *chip,
 		[MODEL_D2X]	= "Xonar D2X",
 		[MODEL_HDAV]	= "Xonar HDAV1.3",
 		[MODEL_HDAV_H6]	= "Xonar HDAV1.3+H6",
+		[MODEL_STX]	= "Xonar Essence STX",
 	};
 	unsigned int model = id->driver_data;
 
@@ -916,6 +1104,10 @@ static int __devinit get_xonar_model(struct oxygen *chip,
 			return -ENODEV;
 		}
 		break;
+	case MODEL_STX:
+		oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL,
+				    GPIO_HDAV_DB_MASK);
+		break;
 	}
 
 	chip->model.shortname = names[model];
-- 
cgit v1.2.3


From 730d45f9130f81fd49009301e9dfbd19fe2b3e1f Mon Sep 17 00:00:00 2001
From: Hannes Eder <hannes@hanneseder.net>
Date: Wed, 25 Feb 2009 22:28:59 +0100
Subject: ALSA: sound/pci/emu10k1: fix sparse warning: different signedness

Fix this sparse warnings:
  sound/pci/emu10k1/emu10k1_main.c:723:66: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:724:68: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:748:74: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:751:66: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:759:73: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:760:73: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:837:50: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:845:50: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:881:50: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:889:57: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:890:57: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:895:60: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:897:60: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:899:60: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:910:56: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:914:57: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:918:56: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:922:57: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:924:58: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:936:60: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:1073:60: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:1088:60: warning: incorrect type in argument 3 (different signedness)
  sound/pci/emu10k1/emu10k1_main.c:1093:58: warning: incorrect type in argument 3 (different signedness)

Signed-off-by: Hannes Eder <hannes@hanneseder.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/emu10k1/emu10k1_main.c | 10 ++++------
 1 file changed, 4 insertions(+), 6 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 8343aecbd25..e6836fc3388 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -711,8 +711,7 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 *emu, const char *filena
 static int emu1010_firmware_thread(void *data)
 {
 	struct snd_emu10k1 *emu = data;
-	int tmp, tmp2;
-	int reg;
+	u32 tmp, tmp2, reg;
 	int err;
 
 	for (;;) {
@@ -758,7 +757,7 @@ static int emu1010_firmware_thread(void *data)
 			snd_printk(KERN_INFO "emu1010: Audio Dock Firmware loaded\n");
 			snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp);
 			snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2);
-			snd_printk(KERN_INFO "Audio Dock ver:%d.%d\n",
+			snd_printk(KERN_INFO "Audio Dock ver: %u.%u\n",
 				   tmp, tmp2);
 			/* Sync clocking between 1010 and Dock */
 			/* Allow DLL to settle */
@@ -805,8 +804,7 @@ static int emu1010_firmware_thread(void *data)
 static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu)
 {
 	unsigned int i;
-	int tmp, tmp2;
-	int reg;
+	u32 tmp, tmp2, reg;
 	int err;
 	const char *filename = NULL;
 
@@ -888,7 +886,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu)
 	snd_printk(KERN_INFO "emu1010: Hana Firmware loaded\n");
 	snd_emu1010_fpga_read(emu, EMU_HANA_MAJOR_REV, &tmp);
 	snd_emu1010_fpga_read(emu, EMU_HANA_MINOR_REV, &tmp2);
-	snd_printk(KERN_INFO "emu1010: Hana version: %d.%d\n", tmp, tmp2);
+	snd_printk(KERN_INFO "emu1010: Hana version: %u.%u\n", tmp, tmp2);
 	/* Enable 48Volt power to Audio Dock */
 	snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, EMU_HANA_DOCK_PWR_ON);
 
-- 
cgit v1.2.3


From 5d9b6c07831456b7a7d90eac31c853d60eaf8ab6 Mon Sep 17 00:00:00 2001
From: Hannes Eder <hannes@hanneseder.net>
Date: Wed, 25 Feb 2009 22:28:45 +0100
Subject: ALSA: sound/pci/hda: fix sparse warning: different signedness

Fix this sparse warning:
  sound/pci/hda/hda_codec.c:1544:19: warning: incorrect type in assignment (different signedness)
  sound/pci/hda/hda_codec.c:1544:19:    expected unsigned long *vals
  sound/pci/hda/hda_codec.c:1544:19:    got long *<noident>

Signed-off-by: Hannes Eder <hannes@hanneseder.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_local.h | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 4bd82a37a4c..03ee9dd0491 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -136,7 +136,7 @@ extern struct hda_ctl_ops snd_hda_bind_sw;	/* for bind-switch */
 
 struct hda_bind_ctls {
 	struct hda_ctl_ops *ops;
-	long values[];
+	unsigned long values[];
 };
 
 int snd_hda_mixer_bind_ctls_info(struct snd_kcontrol *kcontrol,
-- 
cgit v1.2.3


From 23f0c048ba59ad5c2f3fd85ed98360b631dbf6f8 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 26 Feb 2009 13:03:58 +0100
Subject: ALSA: hda - Clean up the input pin setup in automatic mode

Clean up the input-pin setup in automatic mode in patch_realtek.c.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 62 ++++++++++++++++++-------------------------
 1 file changed, 26 insertions(+), 36 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index d670d33cfa1..b3406302d06 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -760,6 +760,24 @@ static int alc_eapd_ctrl_put(struct snd_kcontrol *kcontrol,
 	  .private_value = nid | (mask<<16) }
 #endif   /* CONFIG_SND_DEBUG */
 
+/*
+ * set up the input pin config (depending on the given auto-pin type)
+ */
+static void alc_set_input_pin(struct hda_codec *codec, hda_nid_t nid,
+			      int auto_pin_type)
+{
+	unsigned int val = PIN_IN;
+
+	if (auto_pin_type <= AUTO_PIN_FRONT_MIC) {
+		unsigned int pincap;
+		pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
+		pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT;
+		if (pincap & AC_PINCAP_VREF_80)
+			val = PIN_VREF80;
+	}
+	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, val);
+}
+
 /*
  */
 static void add_mixer(struct alc_spec *spec, struct snd_kcontrol_new *mix)
@@ -4188,10 +4206,7 @@ static void alc880_auto_init_analog_input(struct hda_codec *codec)
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
 		hda_nid_t nid = spec->autocfg.input_pins[i];
 		if (alc880_is_input_pin(nid)) {
-			snd_hda_codec_write(codec, nid, 0,
-					    AC_VERB_SET_PIN_WIDGET_CONTROL,
-					    i <= AUTO_PIN_FRONT_MIC ?
-					    PIN_VREF80 : PIN_IN);
+			alc_set_input_pin(codec, nid, i);
 			if (nid != ALC880_PIN_CD_NID)
 				snd_hda_codec_write(codec, nid, 0,
 						    AC_VERB_SET_AMP_GAIN_MUTE,
@@ -5657,10 +5672,7 @@ static void alc260_auto_init_analog_input(struct hda_codec *codec)
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
 		hda_nid_t nid = spec->autocfg.input_pins[i];
 		if (nid >= 0x12) {
-			snd_hda_codec_write(codec, nid, 0,
-					    AC_VERB_SET_PIN_WIDGET_CONTROL,
-					    i <= AUTO_PIN_FRONT_MIC ?
-					    PIN_VREF80 : PIN_IN);
+			alc_set_input_pin(codec, nid, i);
 			if (nid != ALC260_PIN_CD_NID)
 				snd_hda_codec_write(codec, nid, 0,
 						    AC_VERB_SET_AMP_GAIN_MUTE,
@@ -7006,16 +7018,7 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec)
 		unsigned int vref;
 		if (!nid)
 			continue;
-		vref = PIN_IN;
-		if (1 /*i <= AUTO_PIN_FRONT_MIC*/) {
-			unsigned int pincap;
-			pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
-			if ((pincap >> AC_PINCAP_VREF_SHIFT) &
-			    AC_PINCAP_VREF_80)
-				vref = PIN_VREF80;
-		}
-		snd_hda_codec_write(codec, nid, 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL, vref);
+		alc_set_input_pin(codec, nid, AUTO_PIN_FRONT_MIC /*i*/);
 		if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)
 			snd_hda_codec_write(codec, nid, 0,
 					    AC_VERB_SET_AMP_GAIN_MUTE,
@@ -9100,10 +9103,7 @@ static void alc883_auto_init_analog_input(struct hda_codec *codec)
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
 		hda_nid_t nid = spec->autocfg.input_pins[i];
 		if (alc883_is_input_pin(nid)) {
-			snd_hda_codec_write(codec, nid, 0,
-					    AC_VERB_SET_PIN_WIDGET_CONTROL,
-					    (i <= AUTO_PIN_FRONT_MIC ?
-					     PIN_VREF80 : PIN_IN));
+			alc_set_input_pin(codec, nid, i);
 			if (nid != ALC883_PIN_CD_NID)
 				snd_hda_codec_write(codec, nid, 0,
 						    AC_VERB_SET_AMP_GAIN_MUTE,
@@ -13831,12 +13831,8 @@ static void alc861_auto_init_analog_input(struct hda_codec *codec)
 
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
 		hda_nid_t nid = spec->autocfg.input_pins[i];
-		if (nid >= 0x0c && nid <= 0x11) {
-			snd_hda_codec_write(codec, nid, 0,
-					    AC_VERB_SET_PIN_WIDGET_CONTROL,
-					    i <= AUTO_PIN_FRONT_MIC ?
-					    PIN_VREF80 : PIN_IN);
-		}
+		if (nid >= 0x0c && nid <= 0x11)
+			alc_set_input_pin(codec, nid, i);
 	}
 }
 
@@ -14803,10 +14799,7 @@ static void alc861vd_auto_init_analog_input(struct hda_codec *codec)
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
 		hda_nid_t nid = spec->autocfg.input_pins[i];
 		if (alc861vd_is_input_pin(nid)) {
-			snd_hda_codec_write(codec, nid, 0,
-					AC_VERB_SET_PIN_WIDGET_CONTROL,
-					i <= AUTO_PIN_FRONT_MIC ?
-							PIN_VREF80 : PIN_IN);
+			alc_set_input_pin(codec, nid, i);
 			if (nid != ALC861VD_PIN_CD_NID)
 				snd_hda_codec_write(codec, nid, 0,
 						AC_VERB_SET_AMP_GAIN_MUTE,
@@ -16732,10 +16725,7 @@ static void alc662_auto_init_analog_input(struct hda_codec *codec)
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
 		hda_nid_t nid = spec->autocfg.input_pins[i];
 		if (alc662_is_input_pin(nid)) {
-			snd_hda_codec_write(codec, nid, 0,
-					    AC_VERB_SET_PIN_WIDGET_CONTROL,
-					    (i <= AUTO_PIN_FRONT_MIC ?
-					     PIN_VREF80 : PIN_IN));
+			alc_set_input_pin(codec, nid, i);
 			if (nid != ALC662_PIN_CD_NID)
 				snd_hda_codec_write(codec, nid, 0,
 						    AC_VERB_SET_AMP_GAIN_MUTE,
-- 
cgit v1.2.3


From 1607b8ea0a4cc20752978fadb027daafc8a2d93c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 26 Feb 2009 16:50:43 +0100
Subject: ALSA: hda - Add model=auto for STAC/IDT codecs

Added the model=auto to STAC/IDT codecs to use the BIOS default setup
explicitly.  It can be used to disable the device-specific model quirk
in the driver.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 16 ++++++++++++++++
 1 file changed, 16 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index da48d8c0b29..37ffd96a9ff 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -43,6 +43,7 @@ enum {
 };
 
 enum {
+	STAC_AUTO,
 	STAC_REF,
 	STAC_9200_OQO,
 	STAC_9200_DELL_D21,
@@ -62,6 +63,7 @@ enum {
 };
 
 enum {
+	STAC_9205_AUTO,
 	STAC_9205_REF,
 	STAC_9205_DELL_M42,
 	STAC_9205_DELL_M43,
@@ -71,6 +73,7 @@ enum {
 };
 
 enum {
+	STAC_92HD73XX_AUTO,
 	STAC_92HD73XX_NO_JD, /* no jack-detection */
 	STAC_92HD73XX_REF,
 	STAC_DELL_M6_AMIC,
@@ -81,6 +84,7 @@ enum {
 };
 
 enum {
+	STAC_92HD83XXX_AUTO,
 	STAC_92HD83XXX_REF,
 	STAC_92HD83XXX_PWR_REF,
 	STAC_DELL_S14,
@@ -88,6 +92,7 @@ enum {
 };
 
 enum {
+	STAC_92HD71BXX_AUTO,
 	STAC_92HD71BXX_REF,
 	STAC_DELL_M4_1,
 	STAC_DELL_M4_2,
@@ -98,6 +103,7 @@ enum {
 };
 
 enum {
+	STAC_925x_AUTO,
 	STAC_925x_REF,
 	STAC_M1,
 	STAC_M1_2,
@@ -110,6 +116,7 @@ enum {
 };
 
 enum {
+	STAC_922X_AUTO,
 	STAC_D945_REF,
 	STAC_D945GTP3,
 	STAC_D945GTP5,
@@ -137,6 +144,7 @@ enum {
 };
 
 enum {
+	STAC_927X_AUTO,
 	STAC_D965_REF_NO_JD, /* no jack-detection */
 	STAC_D965_REF,
 	STAC_D965_3ST,
@@ -1488,6 +1496,7 @@ static unsigned int *stac9200_brd_tbl[STAC_9200_MODELS] = {
 };
 
 static const char *stac9200_models[STAC_9200_MODELS] = {
+	[STAC_AUTO] = "auto",
 	[STAC_REF] = "ref",
 	[STAC_9200_OQO] = "oqo",
 	[STAC_9200_DELL_D21] = "dell-d21",
@@ -1633,6 +1642,7 @@ static unsigned int *stac925x_brd_tbl[STAC_925x_MODELS] = {
 };
 
 static const char *stac925x_models[STAC_925x_MODELS] = {
+	[STAC_925x_AUTO] = "auto",
 	[STAC_REF] = "ref",
 	[STAC_M1] = "m1",
 	[STAC_M1_2] = "m1-2",
@@ -1692,6 +1702,7 @@ static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = {
 };
 
 static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = {
+	[STAC_92HD73XX_AUTO] = "auto",
 	[STAC_92HD73XX_NO_JD] = "no-jd",
 	[STAC_92HD73XX_REF] = "ref",
 	[STAC_DELL_M6_AMIC] = "dell-m6-amic",
@@ -1748,6 +1759,7 @@ static unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = {
 };
 
 static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = {
+	[STAC_92HD83XXX_AUTO] = "auto",
 	[STAC_92HD83XXX_REF] = "ref",
 	[STAC_92HD83XXX_PWR_REF] = "mic-ref",
 	[STAC_DELL_S14] = "dell-s14",
@@ -1802,6 +1814,7 @@ static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = {
 };
 
 static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = {
+	[STAC_92HD71BXX_AUTO] = "auto",
 	[STAC_92HD71BXX_REF] = "ref",
 	[STAC_DELL_M4_1] = "dell-m4-1",
 	[STAC_DELL_M4_2] = "dell-m4-2",
@@ -1973,6 +1986,7 @@ static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = {
 };
 
 static const char *stac922x_models[STAC_922X_MODELS] = {
+	[STAC_922X_AUTO] = "auto",
 	[STAC_D945_REF]	= "ref",
 	[STAC_D945GTP5]	= "5stack",
 	[STAC_D945GTP3]	= "3stack",
@@ -2125,6 +2139,7 @@ static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = {
 };
 
 static const char *stac927x_models[STAC_927X_MODELS] = {
+	[STAC_927X_AUTO]	= "auto",
 	[STAC_D965_REF_NO_JD]	= "ref-no-jd",
 	[STAC_D965_REF]		= "ref",
 	[STAC_D965_3ST]		= "3stack",
@@ -2222,6 +2237,7 @@ static unsigned int *stac9205_brd_tbl[STAC_9205_MODELS] = {
 };
 
 static const char *stac9205_models[STAC_9205_MODELS] = {
+	[STAC_9205_AUTO] = "auto",
 	[STAC_9205_REF] = "ref",
 	[STAC_9205_DELL_M42] = "dell-m42",
 	[STAC_9205_DELL_M43] = "dell-m43",
-- 
cgit v1.2.3


From bedfcebb4fb33fc9ebd395462e72afa103db0bec Mon Sep 17 00:00:00 2001
From: peerchen <peerchen@gmail.com>
Date: Fri, 27 Feb 2009 17:03:19 +0800
Subject: ALSA: hda - Add the Device IDs for MCP89 and remove IDs of MCP7B

Added the Device IDs for MCP89 HD audio controller.
Removed the IDs of MCP7B cause this chipset had been cancelled.

Signed-off-by: Peer Chen <peerchen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 8 ++++----
 1 file changed, 4 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index c5a5dc5698a..47a5833feb7 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2454,10 +2454,10 @@ static struct pci_device_id azx_ids[] = {
 	{ PCI_DEVICE(0x10de, 0x0ac1), .driver_data = AZX_DRIVER_NVIDIA },
 	{ PCI_DEVICE(0x10de, 0x0ac2), .driver_data = AZX_DRIVER_NVIDIA },
 	{ PCI_DEVICE(0x10de, 0x0ac3), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x0bd4), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x0bd5), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x0bd6), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x0bd7), .driver_data = AZX_DRIVER_NVIDIA },
+	{ PCI_DEVICE(0x10de, 0x0d94), .driver_data = AZX_DRIVER_NVIDIA },
+	{ PCI_DEVICE(0x10de, 0x0d95), .driver_data = AZX_DRIVER_NVIDIA },
+	{ PCI_DEVICE(0x10de, 0x0d96), .driver_data = AZX_DRIVER_NVIDIA },
+	{ PCI_DEVICE(0x10de, 0x0d97), .driver_data = AZX_DRIVER_NVIDIA },
 	/* Teradici */
 	{ PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA },
 	/* AMD Generic, PCI class code and Vendor ID for HD Audio */
-- 
cgit v1.2.3


From 82af308f658cf2193e5058bbbfd37c3437cfb4e7 Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Fri, 27 Feb 2009 09:27:44 +0100
Subject: sound: oxygen: zero-initialize model data

Model drivers assume that model_data is zeroed, so we better use
kzalloc() (like we did before when it was allocated together with the
card structure).

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/oxygen/oxygen_lib.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c
index 6e1cdd2fd76..312251d3969 100644
--- a/sound/pci/oxygen/oxygen_lib.c
+++ b/sound/pci/oxygen/oxygen_lib.c
@@ -566,7 +566,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
 		goto err_pci_regions;
 
 	if (chip->model.model_data_size) {
-		chip->model_data = kmalloc(chip->model.model_data_size,
+		chip->model_data = kzalloc(chip->model.model_data_size,
 					   GFP_KERNEL);
 		if (!chip->model_data) {
 			err = -ENOMEM;
-- 
cgit v1.2.3


From 53eff7e1e0de1cde8e8cbe619f401d2578dde946 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 27 Feb 2009 17:49:44 +0100
Subject: ALSA: hda - Match all 103c:17xx devices for HP BPC model

Use SND_PCI_QUIRK_MASK() to match all devices with 103c:17xx for
HP BPC model.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index e72b74efc69..0b4afa0a351 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -10807,7 +10807,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
 			   ALC262_HP_BPC),
 	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1300, "HP xw series",
 			   ALC262_HP_BPC),
-	SND_PCI_QUIRK(0x103c, 0x170b, "HP xw*", ALC262_HP_BPC),
+	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1700, "HP xw series",
+			   ALC262_HP_BPC),
 	SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL),
 	SND_PCI_QUIRK(0x103c, 0x2801, "HP D7000", ALC262_HP_BPC_D7000_WF),
 	SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL),
-- 
cgit v1.2.3


From c82c8abdeef53eb0bb0504becb4e91bbccceaee8 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 27 Feb 2009 17:52:22 +0100
Subject: ALSA: hda - Fix an "unused variable" compile warning
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Forgot to remove an unused variable.
  sound/pci/hda/patch_realtek.c: In function ‘alc882_auto_init_analog_input’:
  sound/pci/hda/patch_realtek.c:7018: warning: unused variable ‘vref’

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 0b4afa0a351..1cc31ac0352 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -7015,7 +7015,6 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec)
 
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
 		hda_nid_t nid = spec->autocfg.input_pins[i];
-		unsigned int vref;
 		if (!nid)
 			continue;
 		alc_set_input_pin(codec, nid, AUTO_PIN_FRONT_MIC /*i*/);
-- 
cgit v1.2.3


From 892981ffbe9a5c4cbc9d75f423b145f32c765f9c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 2 Mar 2009 08:04:35 +0100
Subject: ALSA: hda - Don't create a beep control for digital-only ALC268

When an ALC268 codec is set up as the digital-only (as found in Toshiba
laptops), it shouldn't contain any beep control that conflict with the
primary codec.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 1cc31ac0352..c60c86acd9b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -11915,7 +11915,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
 	if (spec->kctls.list)
 		add_mixer(spec, spec->kctls.list);
 
-	if (spec->autocfg.speaker_pins[0] != 0x1d)
+	if (!spec->no_analog && spec->autocfg.speaker_pins[0] != 0x1d)
 		add_mixer(spec, alc268_beep_mixer);
 
 	add_verb(spec, alc268_volume_init_verbs);
-- 
cgit v1.2.3


From 4c4531d64dd0442813c7307b860bf40a2aec51bc Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 2 Mar 2009 08:06:11 +0100
Subject: ALSA: hda - Remove Toshiba probe_mask quirk

Revert the Toshiba probe_mask quirk for 2.6.29 kernel
(commit 38f1df27e3191d76e983cb9c6b4392582fd32fda).
In the current tree, the digital-only codec is handled properly so
no codec conflict should occur.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 2 --
 1 file changed, 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 68a128fb487..47a5833feb7 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2095,8 +2095,6 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x1028, 0x20ac, "Dell Studio Desktop", 0x01),
 	/* including bogus ALC268 in slot#2 that conflicts with ALC888 */
 	SND_PCI_QUIRK(0x17c0, 0x4085, "Medion MD96630", 0x01),
-	/* conflict of ALC268 in slot#3 (digital I/O); a temporary fix */
-	SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba laptop", 0x03),
 	/* forced codec slots */
 	SND_PCI_QUIRK(0x1046, 0x1262, "ASUS W5F", 0x103),
 	{}
-- 
cgit v1.2.3


From d1f1af2dbf8207db590853a59bec465c4f68cfdc Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 2 Mar 2009 10:35:29 +0100
Subject: ALSA: hda - Intialize more codec fields in snd_hda_codec_reset()

Initiailize forgotten fields in snd_hda_codec_reset().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 5dceee8a113..3b44c789f23 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1519,6 +1519,9 @@ int snd_hda_codec_reset(struct hda_codec *codec)
 	codec->num_pcms = 0;
 	codec->pcm_info = NULL;
 	codec->preset = NULL;
+	memset(&codec->patch_ops, 0, sizeof(codec->patch_ops));
+	codec->slave_dig_outs = NULL;
+	codec->spdif_status_reset = 0;
 	module_put(codec->owner);
 	codec->owner = NULL;
 
-- 
cgit v1.2.3


From f93d461bcde6ac3db542361c00a7e4167f88176d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 2 Mar 2009 10:44:15 +0100
Subject: ALSA: hda - Revert the codec probe at control-creation errors

Revert the codec probe instead of returning the error to the driver
when any error occurs at creating the control elements.
The control element conflict can be non-fatal in many cases,
especially if it comes from the digital-only codec.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 14 ++++++++++----
 1 file changed, 10 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 3b44c789f23..1be34ed9c0e 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1434,7 +1434,6 @@ int snd_hda_ctl_add(struct hda_codec *codec, struct snd_kcontrol *kctl)
 }
 EXPORT_SYMBOL_HDA(snd_hda_ctl_add);
 
-#ifdef CONFIG_SND_HDA_RECONFIG
 /* Clear all controls assigned to the given codec */
 void snd_hda_ctls_clear(struct hda_codec *codec)
 {
@@ -1529,7 +1528,6 @@ int snd_hda_codec_reset(struct hda_codec *codec)
 	hda_unlock_devices(card);
 	return 0;
 }
-#endif /* CONFIG_SND_HDA_RECONFIG */
 
 /* create a virtual master control and add slaves */
 int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
@@ -2392,8 +2390,16 @@ int /*__devinit*/ snd_hda_build_controls(struct hda_bus *bus)
 
 	list_for_each_entry(codec, &bus->codec_list, list) {
 		int err = snd_hda_codec_build_controls(codec);
-		if (err < 0)
-			return err;
+		if (err < 0) {
+			printk(KERN_ERR "hda_codec: cannot build controls"
+			       "for #%d (error %d)\n", codec->addr, err); 
+			err = snd_hda_codec_reset(codec);
+			if (err < 0) {
+				printk(KERN_ERR
+				       "hda_codec: cannot revert codec\n");
+				return err;
+			}
+		}
 	}
 	return 0;
 }
-- 
cgit v1.2.3


From 6e655bf21697d2594243098a14e0699e8d4a4059 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 2 Mar 2009 10:46:03 +0100
Subject: ALSA: hda - Don't return a fatal error at PCM-creation errors

Don't return a fatal error to the driver but continue to probe when
any error occurs at creating PCM streams for each codec.
It's often non-fatal and keeping it would help debugging.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 22 +++++++++++++++++-----
 1 file changed, 17 insertions(+), 5 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 1be34ed9c0e..7c9ef5c18e7 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2833,8 +2833,16 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec)
 		if (!codec->patch_ops.build_pcms)
 			return 0;
 		err = codec->patch_ops.build_pcms(codec);
-		if (err < 0)
-			return err;
+		if (err < 0) {
+			printk(KERN_ERR "hda_codec: cannot build PCMs"
+			       "for #%d (error %d)\n", codec->addr, err); 
+			err = snd_hda_codec_reset(codec);
+			if (err < 0) {
+				printk(KERN_ERR
+				       "hda_codec: cannot revert codec\n");
+				return err;
+			}
+		}
 	}
 	for (pcm = 0; pcm < codec->num_pcms; pcm++) {
 		struct hda_pcm *cpcm = &codec->pcm_info[pcm];
@@ -2846,11 +2854,15 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec)
 		if (!cpcm->pcm) {
 			dev = get_empty_pcm_device(codec->bus, cpcm->pcm_type);
 			if (dev < 0)
-				return 0;
+				continue; /* no fatal error */
 			cpcm->device = dev;
 			err = snd_hda_attach_pcm(codec, cpcm);
-			if (err < 0)
-				return err;
+			if (err < 0) {
+				printk(KERN_ERR "hda_codec: cannot attach "
+				       "PCM stream %d for codec #%d\n",
+				       dev, codec->addr);
+				continue; /* no fatal error */
+			}
 		}
 	}
 	return 0;
-- 
cgit v1.2.3


From 43b62713f67d9f0655f3a61f5bd14d6297ddd3ce Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 2 Mar 2009 14:25:17 +0100
Subject: ALSA: hda - Add hint string helper functions

Added snd_hda_get_hint() and snd_hda_get_bool_hint() helper functions
to retrieve a hint value.

Internally, the hint is stored in a pair of two strings, key and val.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_hwdep.c | 112 ++++++++++++++++++++++++++++++++++++++++------
 sound/pci/hda/hda_local.h |  17 +++++++
 2 files changed, 116 insertions(+), 13 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index 4af484b8240..5e554de9cd9 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -30,6 +30,12 @@
 #include <sound/hda_hwdep.h>
 #include <sound/minors.h>
 
+/* hint string pair */
+struct hda_hint {
+	const char *key;
+	const char *val;	/* contained in the same alloc as key */
+};
+
 /*
  * write/read an out-of-bound verb
  */
@@ -99,15 +105,15 @@ static int hda_hwdep_open(struct snd_hwdep *hw, struct file *file)
 
 static void clear_hwdep_elements(struct hda_codec *codec)
 {
-	char **head;
 	int i;
 
 	/* clear init verbs */
 	snd_array_free(&codec->init_verbs);
 	/* clear hints */
-	head = codec->hints.list;
-	for (i = 0; i < codec->hints.used; i++, head++)
-		kfree(*head);
+	for (i = 0; i < codec->hints.used; i++) {
+		struct hda_hint *hint = snd_array_elem(&codec->hints, i);
+		kfree(hint->key); /* we don't need to free hint->val */
+	}
 	snd_array_free(&codec->hints);
 	snd_array_free(&codec->user_pins);
 }
@@ -141,7 +147,7 @@ int /*__devinit*/ snd_hda_create_hwdep(struct hda_codec *codec)
 #endif
 
 	snd_array_init(&codec->init_verbs, sizeof(struct hda_verb), 32);
-	snd_array_init(&codec->hints, sizeof(char *), 32);
+	snd_array_init(&codec->hints, sizeof(struct hda_hint), 32);
 	snd_array_init(&codec->user_pins, sizeof(struct hda_pincfg), 16);
 
 	return 0;
@@ -306,26 +312,81 @@ static ssize_t init_verbs_store(struct device *dev,
 	return count;
 }
 
+static struct hda_hint *get_hint(struct hda_codec *codec, const char *key)
+{
+	int i;
+
+	for (i = 0; i < codec->hints.used; i++) {
+		struct hda_hint *hint = snd_array_elem(&codec->hints, i);
+		if (!strcmp(hint->key, key))
+			return hint;
+	}
+	return NULL;
+}
+
+static void remove_trail_spaces(char *str)
+{
+	char *p;
+	if (!*str)
+		return;
+	p = str + strlen(str) - 1;
+	for (; isspace(*p); p--) {
+		*p = 0;
+		if (p == str)
+			return;
+	}
+}
+
+#define MAX_HINTS	1024
+
 static ssize_t hints_store(struct device *dev,
 			   struct device_attribute *attr,
 			   const char *buf, size_t count)
 {
 	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
 	struct hda_codec *codec = hwdep->private_data;
-	char *p;
-	char **hint;
+	char *key, *val;
+	struct hda_hint *hint;
 
-	if (!*buf || isspace(*buf) || *buf == '#' || *buf == '\n')
+	while (isspace(*buf))
+		buf++;
+	if (!*buf || *buf == '#' || *buf == '\n')
 		return count;
-	p = kstrndup_noeol(buf, 1024);
-	if (!p)
+	if (*buf == '=')
+		return -EINVAL;
+	key = kstrndup_noeol(buf, 1024);
+	if (!key)
 		return -ENOMEM;
-	hint = snd_array_new(&codec->hints);
+	/* extract key and val */
+	val = strchr(key, '=');
+	if (!val) {
+		kfree(key);
+		return -EINVAL;
+	}
+	*val++ = 0;
+	while (isspace(*val))
+		val++;
+	remove_trail_spaces(key);
+	remove_trail_spaces(val);
+	hint = get_hint(codec, key);
+	if (hint) {
+		/* replace */
+		kfree(hint->key);
+		hint->key = key;
+		hint->val = val;
+		return count;
+	}
+	/* allocate a new hint entry */
+	if (codec->hints.used >= MAX_HINTS)
+		hint = NULL;
+	else
+		hint = snd_array_new(&codec->hints);
 	if (!hint) {
-		kfree(p);
+		kfree(key);
 		return -ENOMEM;
 	}
-	*hint = p;
+	hint->key = key;
+	hint->val = val;
 	return count;
 }
 
@@ -428,4 +489,29 @@ int snd_hda_hwdep_add_sysfs(struct hda_codec *codec)
 	return 0;
 }
 
+/*
+ * Look for hint string
+ */
+const char *snd_hda_get_hint(struct hda_codec *codec, const char *key)
+{
+	struct hda_hint *hint = get_hint(codec, key);
+	return hint ? hint->val : NULL;
+}
+EXPORT_SYMBOL_HDA(snd_hda_get_hint);
+
+int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key)
+{
+	const char *p = snd_hda_get_hint(codec, key);
+	if (!p || !*p)
+		return -ENOENT;
+	switch (toupper(*p)) {
+	case 'T': /* true */
+	case 'Y': /* yes */
+	case '1':
+		return 1;
+	}
+	return 0;
+}
+EXPORT_SYMBOL_HDA(snd_hda_get_bool_hint);
+
 #endif /* CONFIG_SND_HDA_RECONFIG */
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 03ee9dd0491..27428c718fd 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -433,6 +433,23 @@ static inline int snd_hda_hwdep_add_sysfs(struct hda_codec *codec)
 }
 #endif
 
+#ifdef CONFIG_SND_HDA_RECONFIG
+const char *snd_hda_get_hint(struct hda_codec *codec, const char *key);
+int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key);
+#else
+static inline
+const char *snd_hda_get_hint(struct hda_codec *codec, const char *key)
+{
+	return NULL;
+}
+
+static inline
+int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key)
+{
+	return -ENOENT;
+}
+#endif
+
 /*
  * power-management
  */
-- 
cgit v1.2.3


From ab1726f920275b52991b2eff7538ac6d313bf9a2 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 2 Mar 2009 17:09:25 +0100
Subject: ALSA: hda - Add show for init_verbs and hints sysfs entries

Added the show method for init_verbs and hints hwdep sysfs entries.
They show the current values.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_hwdep.c | 35 +++++++++++++++++++++++++++++++++--
 1 file changed, 33 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index 5e554de9cd9..1e3ccc740af 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -290,6 +290,22 @@ static ssize_t type##_store(struct device *dev,			\
 CODEC_ACTION_STORE(reconfig);
 CODEC_ACTION_STORE(clear);
 
+static ssize_t init_verbs_show(struct device *dev,
+			       struct device_attribute *attr,
+			       char *buf)
+{
+	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
+	struct hda_codec *codec = hwdep->private_data;
+	int i, len = 0;
+	for (i = 0; i < codec->init_verbs.used; i++) {
+		struct hda_verb *v = snd_array_elem(&codec->init_verbs, i);
+		len += snprintf(buf + len, PAGE_SIZE - len,
+				"0x%02x 0x%03x 0x%04x\n",
+				v->nid, v->verb, v->param);
+	}
+	return len;
+}
+
 static ssize_t init_verbs_store(struct device *dev,
 				struct device_attribute *attr,
 				const char *buf, size_t count)
@@ -312,6 +328,21 @@ static ssize_t init_verbs_store(struct device *dev,
 	return count;
 }
 
+static ssize_t hints_show(struct device *dev,
+			  struct device_attribute *attr,
+			  char *buf)
+{
+	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
+	struct hda_codec *codec = hwdep->private_data;
+	int i, len = 0;
+	for (i = 0; i < codec->hints.used; i++) {
+		struct hda_hint *hint = snd_array_elem(&codec->hints, i);
+		len += snprintf(buf + len, PAGE_SIZE - len,
+				"%s = %s\n", hint->key, hint->val);
+	}
+	return len;
+}
+
 static struct hda_hint *get_hint(struct hda_codec *codec, const char *key)
 {
 	int i;
@@ -466,8 +497,8 @@ static struct device_attribute codec_attrs[] = {
 	CODEC_ATTR_RO(mfg),
 	CODEC_ATTR_RW(name),
 	CODEC_ATTR_RW(modelname),
-	CODEC_ATTR_WO(init_verbs),
-	CODEC_ATTR_WO(hints),
+	CODEC_ATTR_RW(init_verbs),
+	CODEC_ATTR_RW(hints),
 	CODEC_ATTR_RO(init_pin_configs),
 	CODEC_ATTR_RW(user_pin_configs),
 	CODEC_ATTR_RO(driver_pin_configs),
-- 
cgit v1.2.3


From d78d7a90adf793943cc29a414b6f4364a700aad5 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 2 Mar 2009 14:26:25 +0100
Subject: ALSA: hda - Create "Analog Loopback" controls optionally

Don't create "Analog Loopback" controls as default since these controls
are usually more harmful than useful for normal users.
Only created when "loopback = yes" hint is given.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 56 ++++++++++++++++++++++++++++++++----------
 1 file changed, 43 insertions(+), 13 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 13056429aa6..7381325b98f 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -190,6 +190,7 @@ struct sigmatel_spec {
 	unsigned int stream_delay;
 
 	/* analog loopback */
+	struct snd_kcontrol_new *aloopback_ctl;
 	unsigned char aloopback_mask;
 	unsigned char aloopback_shift;
 
@@ -1013,8 +1014,6 @@ static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = {
 	HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT),
 	HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT),
 
-	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3),
-
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT),
 
@@ -1024,9 +1023,22 @@ static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = {
+static struct snd_kcontrol_new stac92hd73xx_6ch_loopback[] = {
+	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3),
+	{}
+};
+
+static struct snd_kcontrol_new stac92hd73xx_8ch_loopback[] = {
 	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 4),
+	{}
+};
 
+static struct snd_kcontrol_new stac92hd73xx_10ch_loopback[] = {
+	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5),
+	{}
+};
+
+static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = {
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT),
 
@@ -1051,8 +1063,6 @@ static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = {
 };
 
 static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = {
-	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5),
-
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT),
 
@@ -1104,8 +1114,6 @@ static struct snd_kcontrol_new stac92hd83xxx_mixer[] = {
 };
 
 static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = {
-	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2),
-
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT),
 
@@ -1131,9 +1139,11 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new stac92hd71bxx_mixer[] = {
-	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2),
+static struct snd_kcontrol_new stac92hd71bxx_loopback[] = {
+	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2)
+};
 
+static struct snd_kcontrol_new stac92hd71bxx_mixer[] = {
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT),
 
@@ -1151,8 +1161,6 @@ static struct snd_kcontrol_new stac925x_mixer[] = {
 };
 
 static struct snd_kcontrol_new stac9205_mixer[] = {
-	STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0, 1),
-
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1b, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1d, 0x0, HDA_OUTPUT),
 
@@ -1161,6 +1169,11 @@ static struct snd_kcontrol_new stac9205_mixer[] = {
 	{ } /* end */
 };
 
+static struct snd_kcontrol_new stac9205_loopback[] = {
+	STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0, 1),
+	{}
+};
+
 /* This needs to be generated dynamically based on sequence */
 static struct snd_kcontrol_new stac922x_mixer[] = {
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_INPUT),
@@ -1173,8 +1186,6 @@ static struct snd_kcontrol_new stac922x_mixer[] = {
 
 
 static struct snd_kcontrol_new stac927x_mixer[] = {
-	STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1),
-
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x18, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1b, 0x0, HDA_OUTPUT),
 
@@ -1186,6 +1197,11 @@ static struct snd_kcontrol_new stac927x_mixer[] = {
 	{ } /* end */
 };
 
+static struct snd_kcontrol_new stac927x_loopback[] = {
+	STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1),
+	{}
+};
+
 static struct snd_kcontrol_new stac_dmux_mixer = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Digital Input Source",
@@ -1312,6 +1328,13 @@ static int stac92xx_build_controls(struct hda_codec *codec)
 			return err;
 	}
 
+	if (spec->aloopback_ctl &&
+	    snd_hda_get_bool_hint(codec, "loopback") == 1) {
+		err = snd_hda_add_new_ctls(codec, spec->aloopback_ctl);
+		if (err < 0)
+			return err;
+	}
+
 	stac92xx_free_kctls(codec); /* no longer needed */
 
 	/* create jack input elements */
@@ -4618,14 +4641,18 @@ again:
 	case 0x3: /* 6 Channel */
 		spec->mixer = stac92hd73xx_6ch_mixer;
 		spec->init = stac92hd73xx_6ch_core_init;
+		spec->aloopback_ctl = stac92hd73xx_6ch_loopback;
 		break;
 	case 0x4: /* 8 Channel */
 		spec->mixer = stac92hd73xx_8ch_mixer;
 		spec->init = stac92hd73xx_8ch_core_init;
+		spec->aloopback_ctl = stac92hd73xx_8ch_loopback;
 		break;
 	case 0x5: /* 10 Channel */
 		spec->mixer = stac92hd73xx_10ch_mixer;
 		spec->init = stac92hd73xx_10ch_core_init;
+		spec->aloopback_ctl = stac92hd73xx_10ch_loopback;
+		break;
 	}
 	spec->multiout.dac_nids = spec->dac_nids;
 
@@ -5036,6 +5063,7 @@ again:
 	if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP)
 		snd_hda_sequence_write_cache(codec, unmute_init);
 
+	spec->aloopback_ctl = stac92hd71bxx_loopback;
 	spec->aloopback_mask = 0x50;
 	spec->aloopback_shift = 0;
 
@@ -5285,6 +5313,7 @@ static int patch_stac927x(struct hda_codec *codec)
 	}
 
 	spec->num_pwrs = 0;
+	spec->aloopback_ctl = stac927x_loopback;
 	spec->aloopback_mask = 0x40;
 	spec->aloopback_shift = 0;
 	spec->eapd_switch = 1;
@@ -5364,6 +5393,7 @@ static int patch_stac9205(struct hda_codec *codec)
 
 	spec->init = stac9205_core_init;
 	spec->mixer = stac9205_mixer;
+	spec->aloopback_ctl = stac9205_loopback;
 
 	spec->aloopback_mask = 0x40;
 	spec->aloopback_shift = 0;
-- 
cgit v1.2.3


From 6565e4faca257fc51a4c55199d72e2701ba7e819 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 2 Mar 2009 14:38:35 +0100
Subject: ALSA: hda - Add more hint options for IDT/Sigmatel codecs

Allow more options to be set/reset via hwdep hint entry.
hp_detect, gpio_mask, gpio_dir, gpio_data, eapd_mask and eapd_switch
can be checked.

For example, to disable hp_detect on the fly,
	# echo "hp_detect=0" > /sys/class/sound/hwC0D0/hints

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 33 +++++++++++++++++++++++++++++++++
 1 file changed, 33 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 7381325b98f..e9331561a48 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -3949,6 +3949,36 @@ static void stac92xx_power_down(struct hda_codec *codec)
 static void stac_toggle_power_map(struct hda_codec *codec, hda_nid_t nid,
 				  int enable);
 
+/* override some hints from the hwdep entry */
+static void stac_store_hints(struct hda_codec *codec)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	const char *p;
+	int val;
+
+	val = snd_hda_get_bool_hint(codec, "hp_detect");
+	if (val >= 0)
+		spec->hp_detect = val;
+	p = snd_hda_get_hint(codec, "gpio_mask");
+	if (p) {
+		spec->gpio_mask = simple_strtoul(p, NULL, 0);
+		spec->eapd_mask = spec->gpio_dir = spec->gpio_data =
+			spec->gpio_mask;
+	}
+	p = snd_hda_get_hint(codec, "gpio_dir");
+	if (p)
+		spec->gpio_dir = simple_strtoul(p, NULL, 0) & spec->gpio_mask;
+	p = snd_hda_get_hint(codec, "gpio_data");
+	if (p)
+		spec->gpio_data = simple_strtoul(p, NULL, 0) & spec->gpio_mask;
+	p = snd_hda_get_hint(codec, "eapd_mask");
+	if (p)
+		spec->eapd_mask = simple_strtoul(p, NULL, 0) & spec->gpio_mask;
+	val = snd_hda_get_bool_hint(codec, "eapd_switch");
+	if (val >= 0)
+		spec->eapd_switch = val;
+}
+
 static int stac92xx_init(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec = codec->spec;
@@ -3965,6 +3995,9 @@ static int stac92xx_init(struct hda_codec *codec)
 				spec->adc_nids[i], 0,
 				AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
 
+	/* override some hints */
+	stac_store_hints(codec);
+
 	/* set up GPIO */
 	gpio = spec->gpio_data;
 	/* turn on EAPD statically when spec->eapd_switch isn't set.
-- 
cgit v1.2.3


From 82ad39f9391fca1d3177bd9f6a5264eff5b5346a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 3 Mar 2009 15:00:35 +0100
Subject: ALSA: hda - Fix gcc compile warning
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

It's false positive, but annoying.
  sound/pci/hda/hda_codec.c: In function ‘get_empty_pcm_device’:
  sound/pci/hda/hda_codec.c:2772: warning: ‘dev’ may be used uninitialized in this function

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 10 ++++------
 1 file changed, 4 insertions(+), 6 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 7c9ef5c18e7..04cb1251e3e 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2776,13 +2776,10 @@ static int get_empty_pcm_device(struct hda_bus *bus, int type)
 		for (i = 0; i < ARRAY_SIZE(audio_idx); i++) {
 			dev = audio_idx[i];
 			if (!test_bit(dev, bus->pcm_dev_bits))
-				break;
-		}
-		if (i >= ARRAY_SIZE(audio_idx)) {
-			snd_printk(KERN_WARNING "Too many audio devices\n");
-			return -EAGAIN;
+				goto ok;
 		}
-		break;
+		snd_printk(KERN_WARNING "Too many audio devices\n");
+		return -EAGAIN;
 	case HDA_PCM_TYPE_SPDIF:
 	case HDA_PCM_TYPE_HDMI:
 	case HDA_PCM_TYPE_MODEM:
@@ -2797,6 +2794,7 @@ static int get_empty_pcm_device(struct hda_bus *bus, int type)
 		snd_printk(KERN_WARNING "Invalid PCM type %d\n", type);
 		return -EINVAL;
 	}
+ ok:
 	set_bit(dev, bus->pcm_dev_bits);
 	return dev;
 }
-- 
cgit v1.2.3


From 79d7d5333b598e9a559bf27833f0ad2b8bf6ad2c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 4 Mar 2009 09:03:50 +0100
Subject: ALSA: hda - Fix HP dv6736 mic input

Fix the mic input of HP dv6736 with Conexant 5051 codec chip.
This laptop seems have no mic-switching per jack connection.
A new model hp-dv6736 is introduced to match with the h/w implementation.

Reference: Novell bnc#480753
	https://bugzilla.novell.com/show_bug.cgi?id=480753

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 63 ++++++++++++++++++++++++++++++++++++++----
 1 file changed, 58 insertions(+), 5 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index b8de73ecfde..1938e92e1f0 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -72,6 +72,7 @@ struct conexant_spec {
 					 */
 	unsigned int cur_eapd;
 	unsigned int hp_present;
+	unsigned int no_auto_mic;
 	unsigned int need_dac_fix;
 
 	/* capture */
@@ -1665,8 +1666,11 @@ static int cxt5051_hp_master_sw_put(struct snd_kcontrol *kcontrol,
 /* toggle input of built-in and mic jack appropriately */
 static void cxt5051_portb_automic(struct hda_codec *codec)
 {
+	struct conexant_spec *spec = codec->spec;
 	unsigned int present;
 
+	if (spec->no_auto_mic)
+		return;
 	present = snd_hda_codec_read(codec, 0x17, 0,
 				     AC_VERB_GET_PIN_SENSE, 0) &
 		AC_PINSENSE_PRESENCE;
@@ -1682,6 +1686,8 @@ static void cxt5051_portc_automic(struct hda_codec *codec)
 	unsigned int present;
 	hda_nid_t new_adc;
 
+	if (spec->no_auto_mic)
+		return;
 	present = snd_hda_codec_read(codec, 0x18, 0,
 				     AC_VERB_GET_PIN_SENSE, 0) &
 		AC_PINSENSE_PRESENCE;
@@ -1768,6 +1774,22 @@ static struct snd_kcontrol_new cxt5051_hp_mixers[] = {
 	{}
 };
 
+static struct snd_kcontrol_new cxt5051_hp_dv6736_mixers[] = {
+	HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x00, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Switch", 0x14, 0x00, HDA_INPUT),
+	HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Master Playback Switch",
+		.info = cxt_eapd_info,
+		.get = cxt_eapd_get,
+		.put = cxt5051_hp_master_sw_put,
+		.private_value = 0x1a,
+	},
+
+	{}
+};
+
 static struct hda_verb cxt5051_init_verbs[] = {
 	/* Line in, Mic */
 	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
@@ -1798,6 +1820,32 @@ static struct hda_verb cxt5051_init_verbs[] = {
 	{ } /* end */
 };
 
+static struct hda_verb cxt5051_hp_dv6736_init_verbs[] = {
+	/* Line in, Mic */
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
+	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0},
+	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0},
+	/* SPK  */
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* HP, Amp  */
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x16, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* DAC1 */
+	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Record selector: Int mic */
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44},
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x1},
+	/* SPDIF route: PCM */
+	{0x1c, AC_VERB_SET_CONNECT_SEL, 0x0},
+	/* EAPD */
+	{0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
+	{0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT},
+	{0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT},
+	{ } /* end */
+};
+
 static struct hda_verb cxt5051_lenovo_x200_init_verbs[] = {
 	/* Line in, Mic */
 	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
@@ -1849,6 +1897,7 @@ static int cxt5051_init(struct hda_codec *codec)
 enum {
 	CXT5051_LAPTOP,	 /* Laptops w/ EAPD support */
 	CXT5051_HP,	/* no docking */
+	CXT5051_HP_DV6736,	/* HP without mic switch */
 	CXT5051_LENOVO_X200,	/* Lenovo X200 laptop */
 	CXT5051_MODELS
 };
@@ -1856,10 +1905,12 @@ enum {
 static const char *cxt5051_models[CXT5051_MODELS] = {
 	[CXT5051_LAPTOP]	= "laptop",
 	[CXT5051_HP]		= "hp",
+	[CXT5051_HP_DV6736]	= "hp-dv6736",
 	[CXT5051_LENOVO_X200]	= "lenovo-x200",
 };
 
 static struct snd_pci_quirk cxt5051_cfg_tbl[] = {
+	SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6736", CXT5051_HP_DV6736),
 	SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board",
 		      CXT5051_LAPTOP),
 	SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP),
@@ -1896,20 +1947,22 @@ static int patch_cxt5051(struct hda_codec *codec)
 	spec->cur_adc = 0;
 	spec->cur_adc_idx = 0;
 
+	codec->patch_ops.unsol_event = cxt5051_hp_unsol_event;
+
 	board_config = snd_hda_check_board_config(codec, CXT5051_MODELS,
 						  cxt5051_models,
 						  cxt5051_cfg_tbl);
 	switch (board_config) {
 	case CXT5051_HP:
-		codec->patch_ops.unsol_event = cxt5051_hp_unsol_event;
 		spec->mixers[0] = cxt5051_hp_mixers;
 		break;
+	case CXT5051_HP_DV6736:
+		spec->init_verbs[0] = cxt5051_hp_dv6736_init_verbs;
+		spec->mixers[0] = cxt5051_hp_dv6736_mixers;
+		spec->no_auto_mic = 1;
+		break;
 	case CXT5051_LENOVO_X200:
 		spec->init_verbs[0] = cxt5051_lenovo_x200_init_verbs;
-		/* fallthru */
-	default:
-	case CXT5051_LAPTOP:
-		codec->patch_ops.unsol_event = cxt5051_hp_unsol_event;
 		break;
 	}
 
-- 
cgit v1.2.3


From bd6afe3f34d41ed81e0c62a5a2181bb7bd51aebf Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 4 Mar 2009 11:30:25 +0100
Subject: ALSA: hda - Fix conflict of mixer controls on Sony VAIO VGN-AR71S

The recent update enabled the model=sony-assamd for all ALC262 with
PCI SSID 104d:90xx.  But this includes the VAIO VGN-AR* that has the
primary codec of STAC92xx and the secondary ALC262 as a slave
digital-only codec.  For this device, the model=auto must be chosen
to work properly.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index c60c86acd9b..8c02f789e4f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -10825,6 +10825,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD),
 	SND_PCI_QUIRK(0x104d, 0x8203, "Sony UX-90", ALC262_HIPPO),
 	SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD),
+	SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */
 	SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO",
 			   ALC262_SONY_ASSAMD),
 	SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1",
-- 
cgit v1.2.3


From 3ea0d7cf472c6118bb8c0842d606f5436251e179 Mon Sep 17 00:00:00 2001
From: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Date: Wed, 4 Mar 2009 14:22:50 -0300
Subject: ALSA: hda - Add 4 channel mode for 3stack-hp model (ALC888)

Add additional 4 channel mode for 3stack-hp models.

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 27 ++++++++++++++++++++++++---
 1 file changed, 24 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 8c02f789e4f..3696ff31838 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8068,24 +8068,45 @@ static struct hda_verb alc888_6st_dell_verbs[] = {
 	{ }
 };
 
+/*
+ * 2ch mode
+ */
 static struct hda_verb alc888_3st_hp_2ch_init[] = {
 	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
 	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
 	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
 	{ 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-	{ }
+	{ } /* end */
 };
 
+/*
+ * 4ch mode
+ */
+static struct hda_verb alc888_3st_hp_4ch_init[] = {
+	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x16, AC_VERB_SET_CONNECT_SEL, 0x01 },
+	{ } /* end */
+};
+
+/*
+ * 6ch mode
+ */
 static struct hda_verb alc888_3st_hp_6ch_init[] = {
 	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
 	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ }
+	{ 0x16, AC_VERB_SET_CONNECT_SEL, 0x01 },
+	{ } /* end */
 };
 
-static struct hda_channel_mode alc888_3st_hp_modes[2] = {
+static struct hda_channel_mode alc888_3st_hp_modes[3] = {
 	{ 2, alc888_3st_hp_2ch_init },
+	{ 4, alc888_3st_hp_4ch_init },
 	{ 6, alc888_3st_hp_6ch_init },
 };
 
-- 
cgit v1.2.3


From 8718b700ccbcc3c6016d38a75e005293c3660f1c Mon Sep 17 00:00:00 2001
From: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Date: Wed, 4 Mar 2009 14:22:51 -0300
Subject: ALSA: hda - Add headphone automute support for 3stack-hp model
 (ALC888)

Mute speaker outputs on headphone insertion for machines that use
3stack-hp model.

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 38 +++++++++++++++++++++++++++++++++-----
 1 file changed, 33 insertions(+), 5 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 3696ff31838..251647d8b5b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8056,16 +8056,42 @@ static struct hda_verb alc888_lenovo_sky_verbs[] = {
 	{ } /* end */
 };
 
+static struct hda_verb alc888_6st_dell_verbs[] = {
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+	{ }
+};
+
+static void alc888_3st_hp_front_automute(struct hda_codec *codec)
+{
+	unsigned int present, bits;
+
+	present = snd_hda_codec_read(codec, 0x1b, 0,
+			AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+	bits = present ? HDA_AMP_MUTE : 0;
+	snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+				 HDA_AMP_MUTE, bits);
+	snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
+				 HDA_AMP_MUTE, bits);
+	snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0,
+				 HDA_AMP_MUTE, bits);
+}
+
+static void alc888_3st_hp_unsol_event(struct hda_codec *codec,
+				      unsigned int res)
+{
+	switch (res >> 26) {
+	case ALC880_HP_EVENT:
+		alc888_3st_hp_front_automute(codec);
+		break;
+	}
+}
+
 static struct hda_verb alc888_3st_hp_verbs[] = {
 	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},	/* Front: output 0 (0x0c) */
 	{0x16, AC_VERB_SET_CONNECT_SEL, 0x01},	/* Rear : output 1 (0x0d) */
 	{0x18, AC_VERB_SET_CONNECT_SEL, 0x02},	/* CLFE : output 2 (0x0e) */
-	{ }
-};
-
-static struct hda_verb alc888_6st_dell_verbs[] = {
 	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
-	{ }
+	{ } /* end */
 };
 
 /*
@@ -8950,6 +8976,8 @@ static struct alc_config_preset alc883_presets[] = {
 		.channel_mode = alc888_3st_hp_modes,
 		.need_dac_fix = 1,
 		.input_mux = &alc883_capture_source,
+		.unsol_event = alc888_3st_hp_unsol_event,
+		.init_hook = alc888_3st_hp_front_automute,
 	},
 	[ALC888_6ST_DELL] = {
 		.mixers = { alc883_base_mixer, alc883_chmode_mixer },
-- 
cgit v1.2.3


From 7ec30f0e7768985ab2ef6334840e3fc8fa253421 Mon Sep 17 00:00:00 2001
From: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Date: Wed, 4 Mar 2009 14:22:52 -0300
Subject: ALSA: hda - Map 3stack-hp model (ALC888) for HP Educ.ar

Added model=3stack-hp for HP Educ.ar desktop machine (103c:2a72).

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 251647d8b5b..91ef9f27b12 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8673,6 +8673,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP),
 	SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x103c, 0x2a66, "HP Acacia", ALC888_3ST_HP),
+	SND_PCI_QUIRK(0x103c, 0x2a72, "HP Educ.ar", ALC888_3ST_HP),
 	SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V),
 	SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG),
 	SND_PCI_QUIRK(0x1043, 0x8284, "Asus Z37E", ALC883_6ST_DIG),
-- 
cgit v1.2.3


From c2503cd3be9eacb1dd06ec5b6fba8bb06aac12a8 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 5 Mar 2009 09:37:40 +0100
Subject: ALSA: hdsp - Ignore MIDI and PCM events in interrupts until
 initialized

Ignore MIDI and PCM events in the interrupt handler until the device
gets initialized properly.  Otherwise you may get kernel panic by the
access to uninitialized devices via hotplugging.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/rme9652/hdsp.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index dc65fe1c9c6..314e73531bd 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -3740,6 +3740,9 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id)
 	midi0status = hdsp_read (hdsp, HDSP_midiStatusIn0) & 0xff;
 	midi1status = hdsp_read (hdsp, HDSP_midiStatusIn1) & 0xff;
 
+	if (!(hdsp->state & HDSP_InitializationComplete))
+		return IRQ_HANDLED;
+
 	if (audio) {
 		if (hdsp->capture_substream)
 			snd_pcm_period_elapsed(hdsp->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream);
-- 
cgit v1.2.3


From 37db623ae2a7bde234a8ed683d0d13d6f939199c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 5 Mar 2009 09:40:16 +0100
Subject: ALSA: hda - Fix check of ALC888S-VC in alc888_coef_init()

Fixed the wrong bits check to identify ALC888S-VC model in
alc888_coef_init().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 91ef9f27b12..6325ea43cf0 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -982,7 +982,7 @@ static void alc888_coef_init(struct hda_codec *codec)
 	snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0);
 	tmp = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_PROC_COEF, 0);
 	snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 7);
-	if ((tmp & 0xf0) == 2)
+	if ((tmp & 0xf0) == 0x20)
 		/* alc888S-VC */
 		snd_hda_codec_read(codec, 0x20, 0,
 				   AC_VERB_SET_PROC_COEF, 0x830);
-- 
cgit v1.2.3


From f03d3115a6bcb814019d945c50c2ef91e5f14477 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 5 Mar 2009 14:18:16 +0100
Subject: ALSA: Fix sample rate of Lenovo Ideapad to 44.1kHz

Noises can be heard on analog outputs of (some model of) Lenovo
Ideapad due to the hardware problem, and the only workaround right now
is to fix the sample rate to 44.1kHz.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 34 +++++++++++++++++++++++++++++++---
 1 file changed, 31 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 6325ea43cf0..b794cba494c 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -12845,6 +12845,27 @@ static int alc269_auto_create_analog_input_ctls(struct alc_spec *spec,
 #define alc269_pcm_digital_playback	alc880_pcm_digital_playback
 #define alc269_pcm_digital_capture	alc880_pcm_digital_capture
 
+static struct hda_pcm_stream alc269_44k_pcm_analog_playback = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 8,
+	.rates = SNDRV_PCM_RATE_44100, /* fixed rate */
+	/* NID is set in alc_build_pcms */
+	.ops = {
+		.open = alc880_playback_pcm_open,
+		.prepare = alc880_playback_pcm_prepare,
+		.cleanup = alc880_playback_pcm_cleanup
+	},
+};
+
+static struct hda_pcm_stream alc269_44k_pcm_analog_capture = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+	.rates = SNDRV_PCM_RATE_44100, /* fixed rate */
+	/* NID is set in alc_build_pcms */
+};
+
 /*
  * BIOS auto configuration
  */
@@ -13060,9 +13081,16 @@ static int patch_alc269(struct hda_codec *codec)
 		setup_preset(spec, &alc269_presets[board_config]);
 
 	spec->stream_name_analog = "ALC269 Analog";
-	spec->stream_analog_playback = &alc269_pcm_analog_playback;
-	spec->stream_analog_capture = &alc269_pcm_analog_capture;
-
+	if (codec->subsystem_id == 0x17aa3bf8) {
+		/* Due to a hardware problem on Lenovo Ideadpad, we need to
+		 * fix the sample rate of analog I/O to 44.1kHz
+		 */
+		spec->stream_analog_playback = &alc269_44k_pcm_analog_playback;
+		spec->stream_analog_capture = &alc269_44k_pcm_analog_capture;
+	} else {
+		spec->stream_analog_playback = &alc269_pcm_analog_playback;
+		spec->stream_analog_capture = &alc269_pcm_analog_capture;
+	}
 	spec->stream_name_digital = "ALC269 Digital";
 	spec->stream_digital_playback = &alc269_pcm_digital_playback;
 	spec->stream_digital_capture = &alc269_pcm_digital_capture;
-- 
cgit v1.2.3


From dc04d1b4d2043e2fca2d94d6d5542b930f2bc5b3 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 6 Mar 2009 10:00:05 +0100
Subject: ALSA: hda - Create output controls according to pin types for
 IDT/STAC

Improve the parser to pick up more intuitive control names for the
outputs judging from the pin type, instead of fixed names assigned
to channels.

Also, revive the multi-HP workaround since this change fixes the
problem with the multi-HP detection.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 141 +++++++++++++++++++++--------------------
 1 file changed, 72 insertions(+), 69 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 2e0a599f8c1..edd2ed7ebb4 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -3039,35 +3039,33 @@ static int add_spec_extra_dacs(struct sigmatel_spec *spec, hda_nid_t nid)
 	return 1;
 }
 
-static int is_unique_dac(struct sigmatel_spec *spec, hda_nid_t nid)
-{
-	int i;
-
-	if (spec->autocfg.line_outs != 1)
-		return 0;
-	if (spec->multiout.hp_nid == nid)
-		return 0;
-	for (i = 0; i < ARRAY_SIZE(spec->multiout.extra_out_nid); i++)
-		if (spec->multiout.extra_out_nid[i] == nid)
-			return 0;
-	return 1;
-}
-
-/* add playback controls from the parsed DAC table */
-static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec,
-					       const struct auto_pin_cfg *cfg)
+/* Create output controls
+ * The mixer elements are named depending on the given type (AUTO_PIN_XXX_OUT)
+ */
+static int create_multi_out_ctls(struct hda_codec *codec, int num_outs,
+				 const hda_nid_t *pins,
+				 const hda_nid_t *dac_nids,
+				 int type)
 {
 	struct sigmatel_spec *spec = codec->spec;
 	static const char *chname[4] = {
 		"Front", "Surround", NULL /*CLFE*/, "Side"
 	};
-	hda_nid_t nid = 0;
+	static const char *hp_pfxs[] = {
+		"Headphone", "Headphone2", "Headphone3", "Headphone4"
+	};
+	static const char *speaker_pfxs[] = {
+		"Speaker", "External Speaker", "Speaker2", "Speaker3"
+	};
+	hda_nid_t nid;
 	int i, err;
 	unsigned int wid_caps;
 
-	for (i = 0; i < cfg->line_outs && spec->multiout.dac_nids[i]; i++) {
-		nid = spec->multiout.dac_nids[i];
-		if (i == 2) {
+	for (i = 0; i < num_outs && i < ARRAY_SIZE(chname); i++) {
+		nid = dac_nids[i];
+		if (!nid)
+			continue;
+		if (type != AUTO_PIN_HP_OUT && i == 2) {
 			/* Center/LFE */
 			err = create_controls(codec, "Center", nid, 1);
 			if (err < 0)
@@ -3088,23 +3086,43 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec,
 			}
 
 		} else {
-			const char *name = chname[i];
-			/* if it's a single DAC, assign a better name */
-			if (!i && is_unique_dac(spec, nid)) {
-				switch (cfg->line_out_type) {
-				case AUTO_PIN_HP_OUT:
-					name = "Headphone";
-					break;
-				case AUTO_PIN_SPEAKER_OUT:
-					name = "Speaker";
-					break;
-				}
+			const char *name;
+			switch (type) {
+			case AUTO_PIN_HP_OUT:
+				name = hp_pfxs[i];
+				break;
+			case AUTO_PIN_SPEAKER_OUT:
+				name = speaker_pfxs[i];
+				break;
+			default:
+				name = chname[i];
+				break;
 			}
 			err = create_controls(codec, name, nid, 3);
 			if (err < 0)
 				return err;
+			if (type == AUTO_PIN_HP_OUT && !spec->hp_detect) {
+				wid_caps = get_wcaps(codec, pins[i]);
+				if (wid_caps & AC_WCAP_UNSOL_CAP)
+					spec->hp_detect = 1;
+			}
 		}
 	}
+	return 0;
+}
+
+/* add playback controls from the parsed DAC table */
+static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec,
+					       const struct auto_pin_cfg *cfg)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	int err;
+
+	err = create_multi_out_ctls(codec, cfg->line_outs, cfg->line_out_pins,
+				    spec->multiout.dac_nids,
+				    cfg->line_out_type);
+	if (err < 0)
+		return err;
 
 	if (cfg->hp_outs > 1 && cfg->line_out_type == AUTO_PIN_LINE_OUT) {
 		err = stac92xx_add_control(spec,
@@ -3139,40 +3157,18 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec,
 					struct auto_pin_cfg *cfg)
 {
 	struct sigmatel_spec *spec = codec->spec;
-	hda_nid_t nid;
-	int i, err, nums;
+	int err;
+
+	err = create_multi_out_ctls(codec, cfg->hp_outs, cfg->hp_pins,
+				    spec->hp_dacs, AUTO_PIN_HP_OUT);
+	if (err < 0)
+		return err;
+
+	err = create_multi_out_ctls(codec, cfg->speaker_outs, cfg->speaker_pins,
+				    spec->speaker_dacs, AUTO_PIN_SPEAKER_OUT);
+	if (err < 0)
+		return err;
 
-	nums = 0;
-	for (i = 0; i < cfg->hp_outs; i++) {
-		static const char *pfxs[] = {
-			"Headphone", "Headphone2", "Headphone3",
-		};
-		unsigned int wid_caps = get_wcaps(codec, cfg->hp_pins[i]);
-		if (wid_caps & AC_WCAP_UNSOL_CAP)
-			spec->hp_detect = 1;
-		if (nums >= ARRAY_SIZE(pfxs))
-			continue;
-		nid = spec->hp_dacs[i];
-		if (!nid)
-			continue;
-		err = create_controls(codec, pfxs[nums++], nid, 3);
-		if (err < 0)
-			return err;
-	}
-	nums = 0;
-	for (i = 0; i < cfg->speaker_outs; i++) {
-		static const char *pfxs[] = {
-			"Speaker", "External Speaker", "Speaker2",
-		};
-		if (nums >= ARRAY_SIZE(pfxs))
-			continue;
-		nid = spec->speaker_dacs[i];
-		if (!nid)
-			continue;
-		err = create_controls(codec, pfxs[nums++], nid, 3);
-		if (err < 0)
-			return err;
-	}
 	return 0;
 }
 
@@ -3505,6 +3501,7 @@ static void stac92xx_auto_init_hp_out(struct hda_codec *codec)
 static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out, hda_nid_t dig_in)
 {
 	struct sigmatel_spec *spec = codec->spec;
+	int hp_swap = 0;
 	int err;
 
 	if ((err = snd_hda_parse_pin_def_config(codec,
@@ -3514,7 +3511,6 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 	if (! spec->autocfg.line_outs)
 		return 0; /* can't find valid pin config */
 
-#if 0 /* FIXME: temporarily disabled */
 	/* If we have no real line-out pin and multiple hp-outs, HPs should
 	 * be set up as multi-channel outputs.
 	 */
@@ -3533,8 +3529,8 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 		spec->autocfg.line_outs = spec->autocfg.hp_outs;
 		spec->autocfg.line_out_type = AUTO_PIN_HP_OUT;
 		spec->autocfg.hp_outs = 0;
+		hp_swap = 1;
 	}
-#endif /* FIXME: temporarily disabled */
 	if (spec->autocfg.mono_out_pin) {
 		int dir = get_wcaps(codec, spec->autocfg.mono_out_pin) &
 			(AC_WCAP_OUT_AMP | AC_WCAP_IN_AMP);
@@ -3627,12 +3623,19 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 #endif
 
 	err = stac92xx_auto_create_hp_ctls(codec, &spec->autocfg);
-
 	if (err < 0)
 		return err;
 
-	err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg);
+	/* All output parsing done, now restore the swapped hp pins */
+	if (hp_swap) {
+		memcpy(spec->autocfg.hp_pins, spec->autocfg.line_out_pins,
+		       sizeof(spec->autocfg.hp_pins));
+		spec->autocfg.hp_outs = spec->autocfg.line_outs;
+		spec->autocfg.line_out_type = AUTO_PIN_HP_OUT;
+		spec->autocfg.line_outs = 0;
+	}
 
+	err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg);
 	if (err < 0)
 		return err;
 
-- 
cgit v1.2.3


From 7a411ee01bf3114ba2a2ae013eaae4e3c41f8eb5 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 6 Mar 2009 10:08:14 +0100
Subject: ALSA: hda - Allow slave controls with non-zero indices

Fix snd_hda_add_vmaster() to check the non-zero indices of slave controls.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 21 +++++++++++++--------
 1 file changed, 13 insertions(+), 8 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 04cb1251e3e..1885e764910 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1552,15 +1552,20 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
 	
 	for (s = slaves; *s; s++) {
 		struct snd_kcontrol *sctl;
-
-		sctl = snd_hda_find_mixer_ctl(codec, *s);
-		if (!sctl) {
-			snd_printdd("Cannot find slave %s, skipped\n", *s);
-			continue;
+		int i = 0;
+		for (;;) {
+			sctl = _snd_hda_find_mixer_ctl(codec, *s, i);
+			if (!sctl) {
+				if (!i)
+					snd_printdd("Cannot find slave %s, "
+						    "skipped\n", *s);
+				break;
+			}
+			err = snd_ctl_add_slave(kctl, sctl);
+			if (err < 0)
+				return err;
+			i++;
 		}
-		err = snd_ctl_add_slave(kctl, sctl);
-		if (err < 0)
-			return err;
 	}
 	return 0;
 }
-- 
cgit v1.2.3


From 668b9652be33510a2a42b290dd335d34d38e2068 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 6 Mar 2009 10:13:24 +0100
Subject: ALSA: hda - Create multiple HP / speaker controls with index

Create multiple "Headphone" and "Speaker" controls with non-zero index
numbers instead of "Headphone2", etc.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 33 ++++++++++++++-------------------
 1 file changed, 14 insertions(+), 19 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index edd2ed7ebb4..d19090fd2d1 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1227,10 +1227,7 @@ static const char *slave_vols[] = {
 	"LFE Playback Volume",
 	"Side Playback Volume",
 	"Headphone Playback Volume",
-	"Headphone2 Playback Volume",
 	"Speaker Playback Volume",
-	"External Speaker Playback Volume",
-	"Speaker2 Playback Volume",
 	NULL
 };
 
@@ -1241,10 +1238,7 @@ static const char *slave_sws[] = {
 	"LFE Playback Switch",
 	"Side Playback Switch",
 	"Headphone Playback Switch",
-	"Headphone2 Playback Switch",
 	"Speaker Playback Switch",
-	"External Speaker Playback Switch",
-	"Speaker2 Playback Switch",
 	"IEC958 Playback Switch",
 	NULL
 };
@@ -2976,8 +2970,8 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec)
 }
 
 /* create volume control/switch for the given prefx type */
-static int create_controls(struct hda_codec *codec, const char *pfx,
-			   hda_nid_t nid, int chs)
+static int create_controls_idx(struct hda_codec *codec, const char *pfx,
+			       int idx, hda_nid_t nid, int chs)
 {
 	struct sigmatel_spec *spec = codec->spec;
 	char name[32];
@@ -3001,19 +2995,22 @@ static int create_controls(struct hda_codec *codec, const char *pfx,
 	}
 
 	sprintf(name, "%s Playback Volume", pfx);
-	err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, name,
+	err = stac92xx_add_control_idx(spec, STAC_CTL_WIDGET_VOL, idx, name,
 		HDA_COMPOSE_AMP_VAL_OFS(nid, chs, 0, HDA_OUTPUT,
 					spec->volume_offset));
 	if (err < 0)
 		return err;
 	sprintf(name, "%s Playback Switch", pfx);
-	err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, name,
+	err = stac92xx_add_control_idx(spec, STAC_CTL_WIDGET_MUTE, idx, name,
 				   HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT));
 	if (err < 0)
 		return err;
 	return 0;
 }
 
+#define create_controls(codec, pfx, nid, chs) \
+	create_controls_idx(codec, pfx, 0, nid, chs)
+
 static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid)
 {
 	if (spec->multiout.num_dacs > 4) {
@@ -3051,12 +3048,6 @@ static int create_multi_out_ctls(struct hda_codec *codec, int num_outs,
 	static const char *chname[4] = {
 		"Front", "Surround", NULL /*CLFE*/, "Side"
 	};
-	static const char *hp_pfxs[] = {
-		"Headphone", "Headphone2", "Headphone3", "Headphone4"
-	};
-	static const char *speaker_pfxs[] = {
-		"Speaker", "External Speaker", "Speaker2", "Speaker3"
-	};
 	hda_nid_t nid;
 	int i, err;
 	unsigned int wid_caps;
@@ -3087,18 +3078,22 @@ static int create_multi_out_ctls(struct hda_codec *codec, int num_outs,
 
 		} else {
 			const char *name;
+			int idx;
 			switch (type) {
 			case AUTO_PIN_HP_OUT:
-				name = hp_pfxs[i];
+				name = "Headphone";
+				idx = i;
 				break;
 			case AUTO_PIN_SPEAKER_OUT:
-				name = speaker_pfxs[i];
+				name = "Speaker";
+				idx = i;
 				break;
 			default:
 				name = chname[i];
+				idx = 0;
 				break;
 			}
-			err = create_controls(codec, name, nid, 3);
+			err = create_controls_idx(codec, name, idx, nid, 3);
 			if (err < 0)
 				return err;
 			if (type == AUTO_PIN_HP_OUT && !spec->hp_detect) {
-- 
cgit v1.2.3


From ee58a7ca21b2acf0d7ad0e1eb2f8d916ecf9fadc Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 6 Mar 2009 12:00:24 +0100
Subject: ALSA: hda - Connect to primary DAC if no individual DAC is available

In stac92xx_auto_fill_dac_nids[], connect to the primary DAC if no
individual DAC is available for each pin.  This ensures that the pin
works somehow at least.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 10 ++++++++++
 1 file changed, 10 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index d19090fd2d1..ee119259183 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2871,6 +2871,16 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid)
 			return conn[j];
 		}
 	}
+	/* if all DACs are already assigned, connect to the primary DAC */
+	if (conn_len > 1) {
+		for (j = 0; j < conn_len; j++) {
+			if (conn[j] == spec->multiout.dac_nids[0]) {
+				snd_hda_codec_write_cache(codec, nid, 0,
+						  AC_VERB_SET_CONNECT_SEL, j);
+				break;
+			}
+		}
+	}
 	return 0;
 }
 
-- 
cgit v1.2.3


From 139e071b0ff37800ed0a68b10c4bb325f51786eb Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 6 Mar 2009 12:10:41 +0100
Subject: ALSA: hda - Assign HP and speaker DACs before mic/line-in

Assign DACs to HP and speaker before mic-in/line-in shared outputs.
This improves the usability as it results in more intuitive mixer
names.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 40 ++++++++++++++++++++--------------------
 1 file changed, 20 insertions(+), 20 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index ee119259183..123bcf7c3b2 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2921,6 +2921,26 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec)
 		add_spec_dacs(spec, dac);
 	}
 
+	for (i = 0; i < cfg->hp_outs; i++) {
+		nid = cfg->hp_pins[i];
+		dac = get_unassigned_dac(codec, nid);
+		if (dac) {
+			if (!spec->multiout.hp_nid)
+				spec->multiout.hp_nid = dac;
+			else
+				add_spec_extra_dacs(spec, dac);
+		}
+		spec->hp_dacs[i] = dac;
+	}
+
+	for (i = 0; i < cfg->speaker_outs; i++) {
+		nid = cfg->speaker_pins[i];
+		dac = get_unassigned_dac(codec, nid);
+		if (dac)
+			add_spec_extra_dacs(spec, dac);
+		spec->speaker_dacs[i] = dac;
+	}
+
 	/* add line-in as output */
 	nid = check_line_out_switch(codec);
 	if (nid) {
@@ -2948,26 +2968,6 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec)
 		}
 	}
 
-	for (i = 0; i < cfg->hp_outs; i++) {
-		nid = cfg->hp_pins[i];
-		dac = get_unassigned_dac(codec, nid);
-		if (dac) {
-			if (!spec->multiout.hp_nid)
-				spec->multiout.hp_nid = dac;
-			else
-				add_spec_extra_dacs(spec, dac);
-		}
-		spec->hp_dacs[i] = dac;
-	}
-
-	for (i = 0; i < cfg->speaker_outs; i++) {
-		nid = cfg->speaker_pins[i];
-		dac = get_unassigned_dac(codec, nid);
-		if (dac)
-			add_spec_extra_dacs(spec, dac);
-		spec->speaker_dacs[i] = dac;
-	}
-
 	snd_printd("stac92xx: dac_nids=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n",
 		   spec->multiout.num_dacs,
 		   spec->multiout.dac_nids[0],
-- 
cgit v1.2.3


From 90f349d96e1dc05b1f7916958282c30760eeacd6 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 6 Mar 2009 14:30:08 +0100
Subject: ALSA: ac97 - Add patch entry for Conexant CX20468-31 chip

Added the patch entry for Conexant CX20468-31 chip (4358:5430).

Reference: Novell bnc#471265
	https://bugzilla.novell.com/show_bug.cgi?id=471265

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ac97/ac97_codec.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index bc707b60385..44f2381b0ae 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -143,6 +143,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = {
 { 0x43525970, 0xfffffff8, "CS4202",		NULL,		NULL },
 { 0x43585421, 0xffffffff, "HSD11246",		NULL,		NULL },	// SmartMC II
 { 0x43585428, 0xfffffff8, "Cx20468",		patch_conexant,	NULL }, // SmartAMC fixme: the mask might be different
+{ 0x43585430, 0xffffffff, "Cx20468-31",		patch_conexant, NULL },
 { 0x43585431, 0xffffffff, "Cx20551",           patch_cx20551,  NULL },
 { 0x44543031, 0xfffffff0, "DT0398",		NULL,		NULL },
 { 0x454d4328, 0xffffffff, "EM28028",		NULL,		NULL },  // same as TR28028?
-- 
cgit v1.2.3


From 873591db59e66434fd0a484c92f69fc21100b33d Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Mon, 9 Mar 2009 09:12:55 +0100
Subject: sound: oxygen: enable headphone output on Claro cards

On the HT-Omega Claro (halo) sound cards, the headphone amplifier must
be enabled explicitly by setting a GPIO bit.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/oxygen/oxygen.c | 63 +++++++++++++++++++++++++++++++++++++++--------
 1 file changed, 53 insertions(+), 10 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c
index 1d8e2b29745..72db4c39007 100644
--- a/sound/pci/oxygen/oxygen.c
+++ b/sound/pci/oxygen/oxygen.c
@@ -1,5 +1,5 @@
 /*
- * C-Media CMI8788 driver for C-Media's reference design and for the X-Meridian
+ * C-Media CMI8788 driver for C-Media's reference design and similar models
  *
  * Copyright (c) Clemens Ladisch <clemens@ladisch.de>
  *
@@ -26,6 +26,7 @@
  *
  * GPIO 0 -> DFS0 of AK5385
  * GPIO 1 -> DFS1 of AK5385
+ * GPIO 8 -> enable headphone amplifier on HT-Omega models
  */
 
 #include <linux/delay.h>
@@ -61,7 +62,8 @@ MODULE_PARM_DESC(enable, "enable card");
 enum {
 	MODEL_CMEDIA_REF,	/* C-Media's reference design */
 	MODEL_MERIDIAN,		/* AuzenTech X-Meridian */
-	MODEL_HALO,		/* HT-Omega Claro halo */
+	MODEL_CLARO,		/* HT-Omega Claro */
+	MODEL_CLARO_HALO,	/* HT-Omega Claro halo */
 };
 
 static struct pci_device_id oxygen_ids[] __devinitdata = {
@@ -74,8 +76,8 @@ static struct pci_device_id oxygen_ids[] __devinitdata = {
 	{ OXYGEN_PCI_SUBID(0x147a, 0xa017), .driver_data = MODEL_CMEDIA_REF },
 	{ OXYGEN_PCI_SUBID(0x1a58, 0x0910), .driver_data = MODEL_CMEDIA_REF },
 	{ OXYGEN_PCI_SUBID(0x415a, 0x5431), .driver_data = MODEL_MERIDIAN },
-	{ OXYGEN_PCI_SUBID(0x7284, 0x9761), .driver_data = MODEL_CMEDIA_REF },
-	{ OXYGEN_PCI_SUBID(0x7284, 0x9781), .driver_data = MODEL_HALO },
+	{ OXYGEN_PCI_SUBID(0x7284, 0x9761), .driver_data = MODEL_CLARO },
+	{ OXYGEN_PCI_SUBID(0x7284, 0x9781), .driver_data = MODEL_CLARO_HALO },
 	{ }
 };
 MODULE_DEVICE_TABLE(pci, oxygen_ids);
@@ -86,6 +88,8 @@ MODULE_DEVICE_TABLE(pci, oxygen_ids);
 #define GPIO_AK5385_DFS_DOUBLE	0x0001
 #define GPIO_AK5385_DFS_QUAD	0x0002
 
+#define GPIO_CLARO_HP		0x0100
+
 struct generic_data {
 	u8 ak4396_ctl2;
 	u16 saved_wm8785_registers[2];
@@ -196,16 +200,46 @@ static void meridian_init(struct oxygen *chip)
 	ak5385_init(chip);
 }
 
-static void halo_init(struct oxygen *chip)
+static void claro_enable_hp(struct oxygen *chip)
+{
+	msleep(300);
+	oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_CLARO_HP);
+	oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, GPIO_CLARO_HP);
+}
+
+static void claro_init(struct oxygen *chip)
+{
+	ak4396_init(chip);
+	wm8785_init(chip);
+	claro_enable_hp(chip);
+}
+
+static void claro_halo_init(struct oxygen *chip)
 {
 	ak4396_init(chip);
 	ak5385_init(chip);
+	claro_enable_hp(chip);
 }
 
 static void generic_cleanup(struct oxygen *chip)
 {
 }
 
+static void claro_disable_hp(struct oxygen *chip)
+{
+	oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_CLARO_HP);
+}
+
+static void claro_cleanup(struct oxygen *chip)
+{
+	claro_disable_hp(chip);
+}
+
+static void claro_suspend(struct oxygen *chip)
+{
+	claro_disable_hp(chip);
+}
+
 static void generic_resume(struct oxygen *chip)
 {
 	ak4396_registers_init(chip);
@@ -217,9 +251,10 @@ static void meridian_resume(struct oxygen *chip)
 	ak4396_registers_init(chip);
 }
 
-static void halo_resume(struct oxygen *chip)
+static void claro_resume(struct oxygen *chip)
 {
 	ak4396_registers_init(chip);
+	claro_enable_hp(chip);
 }
 
 static void set_ak4396_params(struct oxygen *chip,
@@ -346,14 +381,22 @@ static int __devinit get_oxygen_model(struct oxygen *chip,
 					    CAPTURE_0_FROM_I2S_2 |
 					    CAPTURE_1_FROM_SPDIF;
 		break;
-	case MODEL_HALO:
-		chip->model.init = halo_init;
-		chip->model.resume = halo_resume;
+	case MODEL_CLARO:
+		chip->model.init = claro_init;
+		chip->model.cleanup = claro_cleanup;
+		chip->model.suspend = claro_suspend;
+		chip->model.resume = claro_resume;
+		break;
+	case MODEL_CLARO_HALO:
+		chip->model.init = claro_halo_init;
+		chip->model.cleanup = claro_cleanup;
+		chip->model.suspend = claro_suspend;
+		chip->model.resume = claro_resume;
 		chip->model.set_adc_params = set_ak5385_params;
 		break;
 	}
 	if (id->driver_data == MODEL_MERIDIAN ||
-	    id->driver_data == MODEL_HALO) {
+	    id->driver_data == MODEL_CLARO_HALO) {
 		chip->model.misc_flags = OXYGEN_MISC_MIDI;
 		chip->model.device_config |= MIDI_OUTPUT | MIDI_INPUT;
 	}
-- 
cgit v1.2.3


From ae6241fbf5c8863631532e8069037bae460607be Mon Sep 17 00:00:00 2001
From: Christoph Plattner <christoph.plattner@gmx.at>
Date: Sun, 8 Mar 2009 23:19:05 +0100
Subject: ALSA: hda - Added HP HDX16/HDX18 notebook support for HDA codecs
 (82HD71)

Added codec recognition of HP HDX platforms and added support of the
MUTE LED (orange/white). For this feature the CONFIG_SND_HDA_POWER_SAVE
is needed to use event handling for mute control.

Signed-off-by: Christoph Plattner <christoph.plattner@gmx.at>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 57 ++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 57 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 123bcf7c3b2..fb9f4ccba88 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -99,6 +99,7 @@ enum {
 	STAC_DELL_M4_3,
 	STAC_HP_M4,
 	STAC_HP_DV5,
+	STAC_HP_HDX,
 	STAC_92HD71BXX_MODELS
 };
 
@@ -1828,6 +1829,7 @@ static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = {
 	[STAC_DELL_M4_3]	= dell_m4_3_pin_configs,
 	[STAC_HP_M4]		= NULL,
 	[STAC_HP_DV5]		= NULL,
+	[STAC_HP_HDX]           = NULL,
 };
 
 static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = {
@@ -1838,6 +1840,7 @@ static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = {
 	[STAC_DELL_M4_3] = "dell-m4-3",
 	[STAC_HP_M4] = "hp-m4",
 	[STAC_HP_DV5] = "hp-dv5",
+	[STAC_HP_HDX] = "hp-hdx",
 };
 
 static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
@@ -1852,6 +1855,10 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
 		      "HP dv4-7", STAC_HP_DV5),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361a,
 		      "HP mini 1000", STAC_HP_M4),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361b,
+		                "HP HDX", STAC_HP_HDX),  /* HDX16 */
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3610,
+		                "HP HDX", STAC_HP_HDX),  /* HDX18 */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233,
 				"unknown Dell", STAC_DELL_M4_1),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0234,
@@ -4472,6 +4479,41 @@ static int stac92xx_resume(struct hda_codec *codec)
 	return 0;
 }
 
+
+/*
+ * using power check for controlling mute led of HP HDX notebooks
+ * check for mute state only on Speakers (nid = 0x10)
+ *
+ * For this feature CONFIG_SND_HDA_POWER_SAVE is needed, otherwise
+ * the LED is NOT working properly !
+ */
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static int stac92xx_check_power_status (struct hda_codec * codec, hda_nid_t nid)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	
+	/* only handle on HP HDX */
+	if (spec->board_config != STAC_HP_HDX)
+		return 0;
+	
+	if (nid == 0x10)
+	{
+		if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0)  & 
+		    HDA_AMP_MUTE)
+			spec->gpio_data &= ~0x08;  /* orange */
+		else
+			spec->gpio_data |= 0x08;   /* white */
+		
+		stac_gpio_set(codec, spec->gpio_mask, 
+			      spec->gpio_dir, 
+			      spec->gpio_data);
+	}
+
+	return 0;
+}
+#endif
+
 static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state)
 {
 	struct sigmatel_spec *spec = codec->spec;
@@ -4493,6 +4535,9 @@ static struct hda_codec_ops stac92xx_patch_ops = {
 	.suspend = stac92xx_suspend,
 	.resume = stac92xx_resume,
 #endif
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+	.check_power_status = stac92xx_check_power_status,
+#endif
 };
 
 static int patch_stac9200(struct hda_codec *codec)
@@ -5089,6 +5134,13 @@ again:
 		/* no output amps */
 		spec->num_pwrs = 0;
 		/* fallthru */
+	case 0x111d76b2: /* Codec of HP HDX16/HDX18 */
+
+		/* orange/white mute led on GPIO3, orange=0, white=1 */
+		spec->gpio_mask |= 0x08;
+		spec->gpio_dir  |= 0x08;
+		spec->gpio_data |= 0x08;  /* set to white */
+		/* fallthru */
 	default:
 		memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_amixer,
 		       sizeof(stac92hd71bxx_dmux_amixer));
@@ -5143,6 +5195,11 @@ again:
 		snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010);
 		stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN);
 		break;
+	case STAC_HP_HDX:
+		spec->num_dmics = 1;
+		spec->num_dmuxes = 1;
+		spec->num_smuxes = 1;
+		break;
 	};
 
 	spec->multiout.dac_nids = spec->dac_nids;
-- 
cgit v1.2.3


From 443e26d014c242623dd70cda054cc6e5ebf7993d Mon Sep 17 00:00:00 2001
From: Christoph Plattner <christoph.plattner@gmx.at>
Date: Tue, 10 Mar 2009 00:05:56 +0100
Subject: ALSA: hda - Rework on patch_sigmatel.c for HP HDX16/HDX18

Code rework, comments of mail tiwai@suse.de (2009-03-09) incorporated.
Code tested on HP HDX16 (not tested on HDX18 yet).

Signed-off-by: Christoph Plattner <christoph.plattner@gmx.at>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 30 +++++++++++++++---------------
 1 file changed, 15 insertions(+), 15 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index fb9f4ccba88..d119feed42c 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4489,14 +4489,10 @@ static int stac92xx_resume(struct hda_codec *codec)
  */
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-static int stac92xx_check_power_status (struct hda_codec * codec, hda_nid_t nid)
+static int stac92xx_hp_hdx_check_power_status (struct hda_codec * codec, hda_nid_t nid)
 {
 	struct sigmatel_spec *spec = codec->spec;
 	
-	/* only handle on HP HDX */
-	if (spec->board_config != STAC_HP_HDX)
-		return 0;
-	
 	if (nid == 0x10)
 	{
 		if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0)  & 
@@ -4535,9 +4531,6 @@ static struct hda_codec_ops stac92xx_patch_ops = {
 	.suspend = stac92xx_suspend,
 	.resume = stac92xx_resume,
 #endif
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-	.check_power_status = stac92xx_check_power_status,
-#endif
 };
 
 static int patch_stac9200(struct hda_codec *codec)
@@ -5134,13 +5127,6 @@ again:
 		/* no output amps */
 		spec->num_pwrs = 0;
 		/* fallthru */
-	case 0x111d76b2: /* Codec of HP HDX16/HDX18 */
-
-		/* orange/white mute led on GPIO3, orange=0, white=1 */
-		spec->gpio_mask |= 0x08;
-		spec->gpio_dir  |= 0x08;
-		spec->gpio_data |= 0x08;  /* set to white */
-		/* fallthru */
 	default:
 		memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_amixer,
 		       sizeof(stac92hd71bxx_dmux_amixer));
@@ -5199,6 +5185,20 @@ again:
 		spec->num_dmics = 1;
 		spec->num_dmuxes = 1;
 		spec->num_smuxes = 1;
+		/* 
+		 * For controlling MUTE LED on HP HDX16/HDX18 notebooks,
+		 * the CONFIG_SND_HDA_POWER_SAVE is needed to be set.
+		 */
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+		/* orange/white mute led on GPIO3, orange=0, white=1 */
+		spec->gpio_mask |= 0x08;
+		spec->gpio_dir  |= 0x08;
+		spec->gpio_data |= 0x08;  /* set to white */
+
+		/* register check_power_status callback. */
+		codec->patch_ops.check_power_status = 
+		    stac92xx_hp_hdx_check_power_status;
+#endif	
 		break;
 	};
 
-- 
cgit v1.2.3


From 6fce61aeaf0dc1dfa306092539397ab903a9afc4 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 10 Mar 2009 07:48:57 +0100
Subject: ALSA: hda - Fix coding style issues in last two patches

Also re-ordered the quirk entries per SSID.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 26 +++++++++++++-------------
 1 file changed, 13 insertions(+), 13 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index d119feed42c..72c87aa20bd 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1853,12 +1853,12 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
 		      "HP dv4-7", STAC_HP_DV5),
 	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3600,
 		      "HP dv4-7", STAC_HP_DV5),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3610,
+		      "HP HDX", STAC_HP_HDX),  /* HDX18 */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361a,
 		      "HP mini 1000", STAC_HP_M4),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361b,
-		                "HP HDX", STAC_HP_HDX),  /* HDX16 */
-	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3610,
-		                "HP HDX", STAC_HP_HDX),  /* HDX18 */
+		      "HP HDX", STAC_HP_HDX),  /* HDX16 */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233,
 				"unknown Dell", STAC_DELL_M4_1),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0234,
@@ -4489,20 +4489,20 @@ static int stac92xx_resume(struct hda_codec *codec)
  */
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-static int stac92xx_hp_hdx_check_power_status (struct hda_codec * codec, hda_nid_t nid)
+static int stac92xx_hp_hdx_check_power_status(struct hda_codec *codec,
+					      hda_nid_t nid)
 {
 	struct sigmatel_spec *spec = codec->spec;
-	
-	if (nid == 0x10)
-	{
-		if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0)  & 
+
+	if (nid == 0x10) {
+		if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) &
 		    HDA_AMP_MUTE)
 			spec->gpio_data &= ~0x08;  /* orange */
 		else
 			spec->gpio_data |= 0x08;   /* white */
-		
-		stac_gpio_set(codec, spec->gpio_mask, 
-			      spec->gpio_dir, 
+
+		stac_gpio_set(codec, spec->gpio_mask,
+			      spec->gpio_dir,
 			      spec->gpio_data);
 	}
 
@@ -5185,7 +5185,7 @@ again:
 		spec->num_dmics = 1;
 		spec->num_dmuxes = 1;
 		spec->num_smuxes = 1;
-		/* 
+		/*
 		 * For controlling MUTE LED on HP HDX16/HDX18 notebooks,
 		 * the CONFIG_SND_HDA_POWER_SAVE is needed to be set.
 		 */
@@ -5196,7 +5196,7 @@ again:
 		spec->gpio_data |= 0x08;  /* set to white */
 
 		/* register check_power_status callback. */
-		codec->patch_ops.check_power_status = 
+		codec->patch_ops.check_power_status =
 		    stac92xx_hp_hdx_check_power_status;
 #endif	
 		break;
-- 
cgit v1.2.3


From dd5746a85cb21ea5b3afca0b569586a05aa56846 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 10 Mar 2009 14:30:40 +0100
Subject: ALSA: hda - Create vmaster for conexant codecs

Instead of binding volumes, create a virtual master volume for Conexant
codecs.  This allows separate HP and speaker volume controls.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 47 ++++++++++++++++++++++++++++++++----------
 1 file changed, 36 insertions(+), 11 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 1938e92e1f0..e1476d6d8b3 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -58,6 +58,7 @@ struct conexant_spec {
 
 	struct snd_kcontrol_new *mixers[5];
 	int num_mixers;
+	hda_nid_t vmaster_nid;
 
 	const struct hda_verb *init_verbs[5];	/* initialization verbs
 						 * don't forget NULL
@@ -462,6 +463,18 @@ static void conexant_free(struct hda_codec *codec)
 	kfree(codec->spec);
 }
 
+static const char *slave_vols[] = {
+	"Headphone Playback Volume",
+	"Speaker Playback Volume",
+	NULL
+};
+
+static const char *slave_sws[] = {
+	"Headphone Playback Switch",
+	"Speaker Playback Switch",
+	NULL
+};
+
 static int conexant_build_controls(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
@@ -489,6 +502,26 @@ static int conexant_build_controls(struct hda_codec *codec)
 		if (err < 0)
 			return err;
 	}
+
+	/* if we have no master control, let's create it */
+	if (spec->vmaster_nid &&
+	    !snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) {
+		unsigned int vmaster_tlv[4];
+		snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid,
+					HDA_OUTPUT, vmaster_tlv);
+		err = snd_hda_add_vmaster(codec, "Master Playback Volume",
+					  vmaster_tlv, slave_vols);
+		if (err < 0)
+			return err;
+	}
+	if (spec->vmaster_nid &&
+	    !snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
+		err = snd_hda_add_vmaster(codec, "Master Playback Switch",
+					  NULL, slave_sws);
+		if (err < 0)
+			return err;
+	}
+
 	return 0;
 }
 
@@ -1182,16 +1215,6 @@ static int cxt5047_hp_master_sw_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
-/* bind volumes of both NID 0x13 (Headphones) and 0x1d (Speakers) */
-static struct hda_bind_ctls cxt5047_bind_master_vol = {
-	.ops = &snd_hda_bind_vol,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(0x13, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT),
-		0
-	},
-};
-
 /* mute internal speaker if HP is plugged */
 static void cxt5047_hp_automute(struct hda_codec *codec)
 {
@@ -1311,7 +1334,8 @@ static struct snd_kcontrol_new cxt5047_toshiba_mixers[] = {
 	HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT),
 	HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT),
 	HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT),
-	HDA_BIND_VOL("Master Playback Volume", &cxt5047_bind_master_vol),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x00, HDA_OUTPUT),
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Master Playback Switch",
@@ -1631,6 +1655,7 @@ static int patch_cxt5047(struct hda_codec *codec)
 		codec->patch_ops.unsol_event = cxt5047_hp_unsol_event;
 #endif	
 	}
+	spec->vmaster_nid = 0x13;
 	return 0;
 }
 
-- 
cgit v1.2.3


From b880c74adf7e79b97de710a152ea82f292f9abc7 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 10 Mar 2009 14:41:05 +0100
Subject: ALSA: hda - Create "Capture Source" control dynamically in
 patch_conexant.c

Create "Capture Source" control dynamically for Conexant codecs.
If only one capture item is available, don't create such a control
since it's just useless.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 61 ++++++++++++------------------------------
 1 file changed, 17 insertions(+), 44 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index e1476d6d8b3..d5d736ff7c6 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -463,6 +463,17 @@ static void conexant_free(struct hda_codec *codec)
 	kfree(codec->spec);
 }
 
+static struct snd_kcontrol_new cxt_capture_mixers[] = {
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Capture Source",
+		.info = conexant_mux_enum_info,
+		.get = conexant_mux_enum_get,
+		.put = conexant_mux_enum_put
+	},
+	{}
+};
+
 static const char *slave_vols[] = {
 	"Headphone Playback Volume",
 	"Speaker Playback Volume",
@@ -522,6 +533,12 @@ static int conexant_build_controls(struct hda_codec *codec)
 			return err;
 	}
 
+	if (spec->input_mux) {
+		err = snd_hda_add_new_ctls(codec, cxt_capture_mixers);
+		if (err < 0)
+			return err;
+	}
+
 	return 0;
 }
 
@@ -753,13 +770,6 @@ static void cxt5045_hp_unsol_event(struct hda_codec *codec,
 }
 
 static struct snd_kcontrol_new cxt5045_mixers[] = {
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = conexant_mux_enum_info,
-		.get = conexant_mux_enum_get,
-		.put = conexant_mux_enum_put
-	},
 	HDA_CODEC_VOLUME("Int Mic Capture Volume", 0x1a, 0x01, HDA_INPUT),
 	HDA_CODEC_MUTE("Int Mic Capture Switch", 0x1a, 0x01, HDA_INPUT),
 	HDA_CODEC_VOLUME("Ext Mic Capture Volume", 0x1a, 0x02, HDA_INPUT),
@@ -793,13 +803,6 @@ static struct snd_kcontrol_new cxt5045_benq_mixers[] = {
 };
 
 static struct snd_kcontrol_new cxt5045_mixers_hp530[] = {
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = conexant_mux_enum_info,
-		.get = conexant_mux_enum_get,
-		.put = conexant_mux_enum_put
-	},
 	HDA_CODEC_VOLUME("Int Mic Capture Volume", 0x1a, 0x02, HDA_INPUT),
 	HDA_CODEC_MUTE("Int Mic Capture Switch", 0x1a, 0x02, HDA_INPUT),
 	HDA_CODEC_VOLUME("Ext Mic Capture Volume", 0x1a, 0x01, HDA_INPUT),
@@ -1170,20 +1173,6 @@ static struct hda_channel_mode cxt5047_modes[1] = {
 	{ 2, NULL },
 };
 
-static struct hda_input_mux cxt5047_capture_source = {
-	.num_items = 1,
-	.items = {
-		{ "Mic", 0x2 },
-	}
-};
-
-static struct hda_input_mux cxt5047_hp_capture_source = {
-	.num_items = 1,
-	.items = {
-		{ "ExtMic", 0x2 },
-	}
-};
-
 static struct hda_input_mux cxt5047_toshiba_capture_source = {
 	.num_items = 2,
 	.items = {
@@ -1321,13 +1310,6 @@ static struct snd_kcontrol_new cxt5047_mixers[] = {
 };
 
 static struct snd_kcontrol_new cxt5047_toshiba_mixers[] = {
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = conexant_mux_enum_info,
-		.get = conexant_mux_enum_get,
-		.put = conexant_mux_enum_put
-	},
 	HDA_CODEC_VOLUME("Mic Bypass Capture Volume", 0x19, 0x02, HDA_INPUT),
 	HDA_CODEC_MUTE("Mic Bypass Capture Switch", 0x19, 0x02, HDA_INPUT),
 	HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT),
@@ -1349,13 +1331,6 @@ static struct snd_kcontrol_new cxt5047_toshiba_mixers[] = {
 };
 
 static struct snd_kcontrol_new cxt5047_hp_mixers[] = {
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = conexant_mux_enum_info,
-		.get = conexant_mux_enum_get,
-		.put = conexant_mux_enum_put
-	},
 	HDA_CODEC_VOLUME("Mic Bypass Capture Volume", 0x19, 0x02, HDA_INPUT),
 	HDA_CODEC_MUTE("Mic Bypass Capture Switch", 0x19,0x02,HDA_INPUT),
 	HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT),
@@ -1614,7 +1589,6 @@ static int patch_cxt5047(struct hda_codec *codec)
 	spec->num_adc_nids = 1;
 	spec->adc_nids = cxt5047_adc_nids;
 	spec->capsrc_nids = cxt5047_capsrc_nids;
-	spec->input_mux = &cxt5047_capture_source;
 	spec->num_mixers = 1;
 	spec->mixers[0] = cxt5047_mixers;
 	spec->num_init_verbs = 1;
@@ -1633,7 +1607,6 @@ static int patch_cxt5047(struct hda_codec *codec)
 		codec->patch_ops.unsol_event = cxt5047_hp2_unsol_event;
 		break;
 	case CXT5047_LAPTOP_HP:
-		spec->input_mux = &cxt5047_hp_capture_source;
 		spec->num_init_verbs = 2;
 		spec->init_verbs[1] = cxt5047_hp_init_verbs;
 		spec->mixers[0] = cxt5047_hp_mixers;
-- 
cgit v1.2.3


From 3b628867f328cfe1ad4811d63961579874f87041 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 10 Mar 2009 14:53:54 +0100
Subject: ALSA: hda - Remove superfluous verbs for Cxt5047 laptop-eapd model

Remove superfluous verbs from cxt5047_toshiba_init_verbs[].
Also fix comments and minor coding style issues.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 7 ++-----
 1 file changed, 2 insertions(+), 5 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index d5d736ff7c6..e9e47574c61 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1377,12 +1377,9 @@ static struct hda_verb cxt5047_init_verbs[] = {
 
 /* configuration for Toshiba Laptops */
 static struct hda_verb cxt5047_toshiba_init_verbs[] = {
-	{0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x0 }, /* default on */
-	/* pin sensing on HP and Mic jacks */
-	{0x13, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT},
-	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT},
+	{0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x0}, /* default off */
 	/* Speaker routing */
-	{0x1d, AC_VERB_SET_CONNECT_SEL,0x1},
+	{0x1d, AC_VERB_SET_CONNECT_SEL, 0x1},
 	{}
 };
 
-- 
cgit v1.2.3


From 5b3a7440cbabdda07cfb3dcf4a07e0115a3dff9a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 10 Mar 2009 15:10:55 +0100
Subject: ALSA: hda - Fix / clean up init verbs for Cxt5047 codec

Fix the initial connections of output pins 0x13 and 0x1d for Conexant
5047 codec to point to the mixer amp properly.

Removed unneeded (doubly) verbs from arrays, also removed the unneeded
changing of widget 0x1c, which is now completely unused.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 36 +++---------------------------------
 1 file changed, 3 insertions(+), 33 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index e9e47574c61..71822140294 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1165,7 +1165,7 @@ static int patch_cxt5045(struct hda_codec *codec)
 /* Conexant 5047 specific */
 #define CXT5047_SPDIF_OUT	0x11
 
-static hda_nid_t cxt5047_dac_nids[2] = { 0x10, 0x1c };
+static hda_nid_t cxt5047_dac_nids[1] = { 0x10 }; /* 0x1c */
 static hda_nid_t cxt5047_adc_nids[1] = { 0x12 };
 static hda_nid_t cxt5047_capsrc_nids[1] = { 0x1a };
 
@@ -1216,9 +1216,6 @@ static void cxt5047_hp_automute(struct hda_codec *codec)
 	bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0;
 	snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0,
 				 HDA_AMP_MUTE, bits);
-	/* Mute/Unmute PCM 2 for good measure - some systems need this */
-	snd_hda_codec_amp_stereo(codec, 0x1c, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, bits);
 }
 
 /* mute internal speaker if HP is plugged */
@@ -1233,9 +1230,6 @@ static void cxt5047_hp2_automute(struct hda_codec *codec)
 	bits = spec->hp_present ? HDA_AMP_MUTE : 0;
 	snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0,
 				 HDA_AMP_MUTE, bits);
-	/* Mute/Unmute PCM 2 for good measure - some systems need this */
-	snd_hda_codec_amp_stereo(codec, 0x1c, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, bits);
 }
 
 /* toggle input of built-in and mic jack appropriately */
@@ -1299,8 +1293,6 @@ static struct snd_kcontrol_new cxt5047_mixers[] = {
 	HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT),
 	HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT),
 	HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PCM-2 Volume", 0x1c, 0x00, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PCM-2 Switch", 0x1c, 0x00, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x00, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Speaker Playback Switch", 0x1d, 0x00, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT),
@@ -1356,8 +1348,8 @@ static struct hda_verb cxt5047_init_verbs[] = {
 	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_50 },
 	/* HP, Speaker  */
 	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
-	{0x13, AC_VERB_SET_CONNECT_SEL,0x1},
-	{0x1d, AC_VERB_SET_CONNECT_SEL,0x0},
+	{0x13, AC_VERB_SET_CONNECT_SEL, 0x0}, /* mixer(0x19) */
+	{0x1d, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mixer(0x19) */
 	/* Record selector: Mic */
 	{0x12, AC_VERB_SET_CONNECT_SEL,0x03},
 	{0x19, AC_VERB_SET_AMP_GAIN_MUTE,
@@ -1378,26 +1370,6 @@ static struct hda_verb cxt5047_init_verbs[] = {
 /* configuration for Toshiba Laptops */
 static struct hda_verb cxt5047_toshiba_init_verbs[] = {
 	{0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x0}, /* default off */
-	/* Speaker routing */
-	{0x1d, AC_VERB_SET_CONNECT_SEL, 0x1},
-	{}
-};
-
-/* configuration for HP Laptops */
-static struct hda_verb cxt5047_hp_init_verbs[] = {
-	/* pin sensing on HP jack */
-	{0x13, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT},
-	/* 0x13 is actually shared by both HP and speaker;
-	 * setting the connection to 0 (=0x19) makes the master volume control
-	 * working mysteriouslly...
-	 */
-	{0x13, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Record selector: Ext Mic */
-	{0x12, AC_VERB_SET_CONNECT_SEL,0x03},
-	{0x19, AC_VERB_SET_AMP_GAIN_MUTE,
-	 AC_AMP_SET_INPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x17},
-	/* Speaker routing */
-	{0x1d, AC_VERB_SET_CONNECT_SEL,0x1},
 	{}
 };
 
@@ -1604,8 +1576,6 @@ static int patch_cxt5047(struct hda_codec *codec)
 		codec->patch_ops.unsol_event = cxt5047_hp2_unsol_event;
 		break;
 	case CXT5047_LAPTOP_HP:
-		spec->num_init_verbs = 2;
-		spec->init_verbs[1] = cxt5047_hp_init_verbs;
 		spec->mixers[0] = cxt5047_hp_mixers;
 		codec->patch_ops.unsol_event = cxt5047_hp_unsol_event;
 		codec->patch_ops.init = cxt5047_hp_init;
-- 
cgit v1.2.3


From df481e41b963b7fc3d7e3543a0c7bb140a682146 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 10 Mar 2009 15:35:35 +0100
Subject: ALSA: hda - Clean up Cxt5047 parser

Clean up Conexant 5047 pareser code:
 - Split mixer elements to separate arrays to reduce the duplicated
   entires
 - Fix mixer element names to the standard ones
 - Remove unneeded cxt5047_hp2_unsol_event; the normal unsol_event
   handler works fine.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 89 +++++++++---------------------------------
 1 file changed, 19 insertions(+), 70 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 71822140294..d60ccb5bb12 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1218,20 +1218,6 @@ static void cxt5047_hp_automute(struct hda_codec *codec)
 				 HDA_AMP_MUTE, bits);
 }
 
-/* mute internal speaker if HP is plugged */
-static void cxt5047_hp2_automute(struct hda_codec *codec)
-{
-	struct conexant_spec *spec = codec->spec;
-	unsigned int bits;
-
-	spec->hp_present = snd_hda_codec_read(codec, 0x13, 0,
-				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
-
-	bits = spec->hp_present ? HDA_AMP_MUTE : 0;
-	snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, bits);
-}
-
 /* toggle input of built-in and mic jack appropriately */
 static void cxt5047_hp_automic(struct hda_codec *codec)
 {
@@ -1269,47 +1255,14 @@ static void cxt5047_hp_unsol_event(struct hda_codec *codec,
 	}
 }
 
-/* unsolicited event for HP jack sensing - non-EAPD systems */
-static void cxt5047_hp2_unsol_event(struct hda_codec *codec,
-				  unsigned int res)
-{
-	res >>= 26;
-	switch (res) {
-	case CONEXANT_HP_EVENT:
-		cxt5047_hp2_automute(codec);
-		break;
-	case CONEXANT_MIC_EVENT:
-		cxt5047_hp_automic(codec);
-		break;
-	}
-}
-
-static struct snd_kcontrol_new cxt5047_mixers[] = {
-	HDA_CODEC_VOLUME("Mic Bypass Capture Volume", 0x19, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Bypass Capture Switch", 0x19, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Gain Volume", 0x1a, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Mic Gain Switch", 0x1a, 0x0, HDA_OUTPUT),
+static struct snd_kcontrol_new cxt5047_base_mixers[] = {
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x19, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x19, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost", 0x1a, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT),
 	HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT),
 	HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x00, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Speaker Playback Switch", 0x1d, 0x00, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x13, 0x00, HDA_OUTPUT),
-
-	{}
-};
-
-static struct snd_kcontrol_new cxt5047_toshiba_mixers[] = {
-	HDA_CODEC_VOLUME("Mic Bypass Capture Volume", 0x19, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Bypass Capture Switch", 0x19, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT),
-	HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x00, HDA_OUTPUT),
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Master Playback Switch",
@@ -1322,22 +1275,14 @@ static struct snd_kcontrol_new cxt5047_toshiba_mixers[] = {
 	{}
 };
 
-static struct snd_kcontrol_new cxt5047_hp_mixers[] = {
-	HDA_CODEC_VOLUME("Mic Bypass Capture Volume", 0x19, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Bypass Capture Switch", 0x19,0x02,HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT),
-	HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT),
+static struct snd_kcontrol_new cxt5047_hp_spk_mixers[] = {
+	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x00, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT),
+	{}
+};
+
+static struct snd_kcontrol_new cxt5047_hp_only_mixers[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x13, 0x00, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Master Playback Switch",
-		.info = cxt_eapd_info,
-		.get = cxt_eapd_get,
-		.put = cxt5047_hp_master_sw_put,
-		.private_value = 0x13,
-	},
 	{ } /* end */
 };
 
@@ -1559,7 +1504,7 @@ static int patch_cxt5047(struct hda_codec *codec)
 	spec->adc_nids = cxt5047_adc_nids;
 	spec->capsrc_nids = cxt5047_capsrc_nids;
 	spec->num_mixers = 1;
-	spec->mixers[0] = cxt5047_mixers;
+	spec->mixers[0] = cxt5047_base_mixers;
 	spec->num_init_verbs = 1;
 	spec->init_verbs[0] = cxt5047_init_verbs;
 	spec->spdif_route = 0;
@@ -1573,18 +1518,22 @@ static int patch_cxt5047(struct hda_codec *codec)
 						  cxt5047_cfg_tbl);
 	switch (board_config) {
 	case CXT5047_LAPTOP:
-		codec->patch_ops.unsol_event = cxt5047_hp2_unsol_event;
+		spec->num_mixers = 2;
+		spec->mixers[1] = cxt5047_hp_spk_mixers;
+		codec->patch_ops.unsol_event = cxt5047_hp_unsol_event;
 		break;
 	case CXT5047_LAPTOP_HP:
-		spec->mixers[0] = cxt5047_hp_mixers;
+		spec->num_mixers = 2;
+		spec->mixers[1] = cxt5047_hp_only_mixers;
 		codec->patch_ops.unsol_event = cxt5047_hp_unsol_event;
 		codec->patch_ops.init = cxt5047_hp_init;
 		break;
 	case CXT5047_LAPTOP_EAPD:
 		spec->input_mux = &cxt5047_toshiba_capture_source;
+		spec->num_mixers = 2;
+		spec->mixers[1] = cxt5047_hp_spk_mixers;
 		spec->num_init_verbs = 2;
 		spec->init_verbs[1] = cxt5047_toshiba_init_verbs;
-		spec->mixers[0] = cxt5047_toshiba_mixers;
 		codec->patch_ops.unsol_event = cxt5047_hp_unsol_event;
 		break;
 #ifdef CONFIG_SND_DEBUG
-- 
cgit v1.2.3


From 5d75bc557859805f00eeddb09d7cc8ffc7e5334e Mon Sep 17 00:00:00 2001
From: Gregorio Guidi <gregorio.guidi@gmail.com>
Date: Thu, 12 Mar 2009 16:41:51 +0100
Subject: ALSA: hda - fix headphone settings and master volume (Conexant
 CX20551)

Update the places where the 0x1d widget is used for Conexant 5047, fixing
mismatch introduced after changing the connection.

Signed-off-by: Gregorio Guidi <gregorio.guidi@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index d60ccb5bb12..6cb184e9c2f 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1196,7 +1196,7 @@ static int cxt5047_hp_master_sw_put(struct snd_kcontrol *kcontrol,
 	 * the headphone jack
 	 */
 	bits = (!spec->hp_present && spec->cur_eapd) ? 0 : HDA_AMP_MUTE;
-	snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0,
+	snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0x01,
 				 HDA_AMP_MUTE, bits);
 	bits = spec->cur_eapd ? 0 : HDA_AMP_MUTE;
 	snd_hda_codec_amp_stereo(codec, 0x13, HDA_OUTPUT, 0,
@@ -1214,7 +1214,7 @@ static void cxt5047_hp_automute(struct hda_codec *codec)
 				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
 
 	bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0;
-	snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0,
+	snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0x01,
 				 HDA_AMP_MUTE, bits);
 }
 
@@ -1276,7 +1276,7 @@ static struct snd_kcontrol_new cxt5047_base_mixers[] = {
 };
 
 static struct snd_kcontrol_new cxt5047_hp_spk_mixers[] = {
-	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x00, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x01, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT),
 	{}
 };
-- 
cgit v1.2.3


From 3b7523fc828e41b2988feb400704e01b67859d78 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 12 Mar 2009 16:45:01 +0100
Subject: ALSA: hda - Add comments for the previous fix for conexant codecs

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 6 ++++++
 1 file changed, 6 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 6cb184e9c2f..bc016fade19 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1196,6 +1196,10 @@ static int cxt5047_hp_master_sw_put(struct snd_kcontrol *kcontrol,
 	 * the headphone jack
 	 */
 	bits = (!spec->hp_present && spec->cur_eapd) ? 0 : HDA_AMP_MUTE;
+	/* NOTE: Conexat codec needs the index for *OUTPUT* amp of
+	 * pin widgets unlike other codecs.  In this case, we need to
+	 * set index 0x01 for the volume from the mixer amp 0x19.
+	 */
 	snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0x01,
 				 HDA_AMP_MUTE, bits);
 	bits = spec->cur_eapd ? 0 : HDA_AMP_MUTE;
@@ -1214,6 +1218,7 @@ static void cxt5047_hp_automute(struct hda_codec *codec)
 				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
 
 	bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0;
+	/* See the note in cxt5047_hp_master_sw_put */
 	snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0x01,
 				 HDA_AMP_MUTE, bits);
 }
@@ -1276,6 +1281,7 @@ static struct snd_kcontrol_new cxt5047_base_mixers[] = {
 };
 
 static struct snd_kcontrol_new cxt5047_hp_spk_mixers[] = {
+	/* See the note in cxt5047_hp_master_sw_put */
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x01, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT),
 	{}
-- 
cgit v1.2.3


From 9421f9543b3a0a870499f64498406003de8214b4 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 12 Mar 2009 17:06:07 +0100
Subject: ALSA: hda - Print multiple out-amp values of pin widgets on Conext
 codecs

Add a flag to work around the non-standard amp-value handling on
Conexant codecs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.h      |  3 +++
 sound/pci/hda/hda_proc.c       | 10 ++++++++--
 sound/pci/hda/patch_conexant.c |  3 +++
 3 files changed, 14 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 2ea628478a9..079e1ab718d 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -793,6 +793,9 @@ struct hda_codec {
 					     * status change
 					     * (e.g. Realtek codecs)
 					     */
+	unsigned int pin_amp_workaround:1; /* pin out-amp takes index
+					    * (e.g. Conexant codecs)
+					    */
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	unsigned int power_on :1;	/* current (global) power-state */
 	unsigned int power_transition :1; /* power-state in transition */
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 144b85276d5..93b25ba4d00 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -554,8 +554,14 @@ static void print_codec_info(struct snd_info_entry *entry,
 			snd_iprintf(buffer, "  Amp-Out caps: ");
 			print_amp_caps(buffer, codec, nid, HDA_OUTPUT);
 			snd_iprintf(buffer, "  Amp-Out vals: ");
-			print_amp_vals(buffer, codec, nid, HDA_OUTPUT,
-				       wid_caps & AC_WCAP_STEREO, 1);
+			if (wid_type == AC_WID_PIN &&
+			    codec->pin_amp_workaround)
+				print_amp_vals(buffer, codec, nid, HDA_OUTPUT,
+					       wid_caps & AC_WCAP_STEREO,
+					       conn_len);
+			else
+				print_amp_vals(buffer, codec, nid, HDA_OUTPUT,
+					       wid_caps & AC_WCAP_STEREO, 1);
 		}
 
 		switch (wid_type) {
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index bc016fade19..1f2ad76ca94 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1066,6 +1066,7 @@ static int patch_cxt5045(struct hda_codec *codec)
 	if (!spec)
 		return -ENOMEM;
 	codec->spec = spec;
+	codec->pin_amp_workaround = 1;
 
 	spec->multiout.max_channels = 2;
 	spec->multiout.num_dacs = ARRAY_SIZE(cxt5045_dac_nids);
@@ -1501,6 +1502,7 @@ static int patch_cxt5047(struct hda_codec *codec)
 	if (!spec)
 		return -ENOMEM;
 	codec->spec = spec;
+	codec->pin_amp_workaround = 1;
 
 	spec->multiout.max_channels = 2;
 	spec->multiout.num_dacs = ARRAY_SIZE(cxt5047_dac_nids);
@@ -1847,6 +1849,7 @@ static int patch_cxt5051(struct hda_codec *codec)
 	if (!spec)
 		return -ENOMEM;
 	codec->spec = spec;
+	codec->pin_amp_workaround = 1;
 
 	codec->patch_ops = conexant_patch_ops;
 	codec->patch_ops.init = cxt5051_init;
-- 
cgit v1.2.3


From 307282c8990c5658604b9fda8a64a9a07079b850 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 12 Mar 2009 18:17:58 +0100
Subject: ALSA: hda - Add model=vaio for STAC9872

Add the default pin config for model=vaio (in case of broken BIOS).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 33 +++++++++++++++++++++++++++++++--
 1 file changed, 31 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 72c87aa20bd..e06fc7decd3 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -155,6 +155,12 @@ enum {
 	STAC_927X_MODELS
 };
 
+enum {
+	STAC_9872_AUTO,
+	STAC_9872_VAIO,
+	STAC_9872_MODELS
+};
+
 struct sigmatel_event {
 	hda_nid_t nid;
 	unsigned char type;
@@ -5588,6 +5594,25 @@ static hda_nid_t stac9872_mux_nids[] = {
 	0x15
 };
 
+static unsigned int stac9872_vaio_pin_configs[9] = {
+	0x03211020, 0x411111f0, 0x411111f0, 0x03a15030,
+	0x411111f0, 0x90170110, 0x411111f0, 0x411111f0,
+	0x90a7013e
+};
+
+static const char *stac9872_models[STAC_9872_MODELS] = {
+	[STAC_9872_AUTO] = "auto",
+	[STAC_9872_VAIO] = "vaio",
+};
+
+static unsigned int *stac9872_brd_tbl[STAC_9872_MODELS] = {
+	[STAC_9872_VAIO] = stac9872_vaio_pin_configs,
+};
+
+static struct snd_pci_quirk stac9872_cfg_tbl[] = {
+	{} /* terminator */
+};
+
 static int patch_stac9872(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec;
@@ -5598,11 +5623,15 @@ static int patch_stac9872(struct hda_codec *codec)
 		return -ENOMEM;
 	codec->spec = spec;
 
-#if 0 /* no model right now */
 	spec->board_config = snd_hda_check_board_config(codec, STAC_9872_MODELS,
 							stac9872_models,
 							stac9872_cfg_tbl);
-#endif
+	if (spec->board_config < 0)
+		snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9872, "
+			    "using BIOS defaults\n");
+	else
+		stac92xx_set_config_regs(codec,
+					 stac9872_brd_tbl[spec->board_config]);
 
 	spec->num_pins = ARRAY_SIZE(stac9872_pin_nids);
 	spec->pin_nids = stac9872_pin_nids;
-- 
cgit v1.2.3


From bb6ac72fb19c6676eb8bafa8e3b8bf970a2294a2 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 13 Mar 2009 09:02:42 +0100
Subject: ALSA: hda - power up before codec initialization

Change the power state of each widget before starting the initialization
work so that all verbs are executed properly.

Also, keep power-up during hwdep reconfiguration.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 15 ++++++++-------
 sound/pci/hda/hda_hwdep.c | 14 +++++++++-----
 2 files changed, 17 insertions(+), 12 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 1885e764910..cf6339436de 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -842,6 +842,9 @@ static void snd_hda_codec_free(struct hda_codec *codec)
 	kfree(codec);
 }
 
+static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
+				unsigned int power_state);
+
 /**
  * snd_hda_codec_new - create a HDA codec
  * @bus: the bus to assign
@@ -941,6 +944,11 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr
 	if (bus->modelname)
 		codec->modelname = kstrdup(bus->modelname, GFP_KERNEL);
 
+	/* power-up all before initialization */
+	hda_set_power_state(codec,
+			    codec->afg ? codec->afg : codec->mfg,
+			    AC_PWRST_D0);
+
 	if (do_init) {
 		err = snd_hda_codec_configure(codec);
 		if (err < 0)
@@ -2413,19 +2421,12 @@ EXPORT_SYMBOL_HDA(snd_hda_build_controls);
 int snd_hda_codec_build_controls(struct hda_codec *codec)
 {
 	int err = 0;
-	/* fake as if already powered-on */
-	hda_keep_power_on(codec);
-	/* then fire up */
-	hda_set_power_state(codec,
-			    codec->afg ? codec->afg : codec->mfg,
-			    AC_PWRST_D0);
 	hda_exec_init_verbs(codec);
 	/* continue to initialize... */
 	if (codec->patch_ops.init)
 		err = codec->patch_ops.init(codec);
 	if (!err && codec->patch_ops.build_controls)
 		err = codec->patch_ops.build_controls(codec);
-	snd_hda_power_down(codec);
 	if (err < 0)
 		return err;
 	return 0;
diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index 1e3ccc740af..1c57505c287 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -176,25 +176,29 @@ static int reconfig_codec(struct hda_codec *codec)
 {
 	int err;
 
+	snd_hda_power_up(codec);
 	snd_printk(KERN_INFO "hda-codec: reconfiguring\n");
 	err = snd_hda_codec_reset(codec);
 	if (err < 0) {
 		snd_printk(KERN_ERR
 			   "The codec is being used, can't reconfigure.\n");
-		return err;
+		goto error;
 	}
 	err = snd_hda_codec_configure(codec);
 	if (err < 0)
-		return err;
+		goto error;
 	/* rebuild PCMs */
 	err = snd_hda_codec_build_pcms(codec);
 	if (err < 0)
-		return err;
+		goto error;
 	/* rebuild mixers */
 	err = snd_hda_codec_build_controls(codec);
 	if (err < 0)
-		return err;
-	return snd_card_register(codec->bus->card);
+		goto error;
+	err = snd_card_register(codec->bus->card);
+ error:
+	snd_hda_power_down(codec);
+	return err;
 }
 
 /*
-- 
cgit v1.2.3


From 58d8395b74f78a2f4225c5faea8b5bffb8af1cf9 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 13 Mar 2009 17:04:34 +0100
Subject: ALSA: hda - Add another HP model with IDT92HD71bx codec

HP laptops require GPIO0 on as EAPD.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index e06fc7decd3..4da72403fc8 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1855,6 +1855,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
 		      "DFI LanParty", STAC_92HD71BXX_REF),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101,
 		      "DFI LanParty", STAC_92HD71BXX_REF),
+	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3080,
+		      "HP", STAC_HP_DV5),
 	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x30f0,
 		      "HP dv4-7", STAC_HP_DV5),
 	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3600,
-- 
cgit v1.2.3


From 9f5d790d1b0af8e3705df12fd5d49a1df2a45c47 Mon Sep 17 00:00:00 2001
From: Giuliano Pochini <pochini@shiny.it>
Date: Sun, 15 Mar 2009 21:33:34 +0100
Subject: ALSA: echoaudio: remove line-out volume from vmixer cards

There is a long standing bug in the drivers for cards with a vmixer because
I overlooked a detail in the c++ generic driver by echoaudio. Those cards
do not have a line-out volume control. It is a virtual control provided by
the generic driver. The bug is harmless because the DSP just ignores the
command to change the volume.
*NB:* It breaks alsa-tools/echomixer. A patch for it will follow.

This patch removes the line-out volume control from vmixer-equipped cards.

Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/echoaudio/echoaudio.c | 17 +++--------------
 1 file changed, 3 insertions(+), 14 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 8dbc5c4ba42..4b70ea1e4c9 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -950,6 +950,8 @@ static int __devinit snd_echo_new_pcm(struct echoaudio *chip)
 	Control interface
 ******************************************************************************/
 
+#ifndef ECHOCARD_HAS_VMIXER
+
 /******************* PCM output volume *******************/
 static int snd_echo_output_gain_info(struct snd_kcontrol *kcontrol,
 				     struct snd_ctl_elem_info *uinfo)
@@ -1001,18 +1003,6 @@ static int snd_echo_output_gain_put(struct snd_kcontrol *kcontrol,
 	return changed;
 }
 
-#ifdef ECHOCARD_HAS_VMIXER
-/* On Vmixer cards this one controls the line-out volume */
-static struct snd_kcontrol_new snd_echo_line_output_gain __devinitdata = {
-	.name = "Line Playback Volume",
-	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ,
-	.info = snd_echo_output_gain_info,
-	.get = snd_echo_output_gain_get,
-	.put = snd_echo_output_gain_put,
-	.tlv = {.p = db_scale_output_gain},
-};
-#else
 static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = {
 	.name = "PCM Playback Volume",
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1022,6 +1012,7 @@ static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = {
 	.put = snd_echo_output_gain_put,
 	.tlv = {.p = db_scale_output_gain},
 };
+
 #endif
 
 
@@ -2037,8 +2028,6 @@ static int __devinit snd_echo_probe(struct pci_dev *pci,
 
 #ifdef ECHOCARD_HAS_VMIXER
 	snd_echo_vmixer.count = num_pipes_out(chip) * num_busses_out(chip);
-	if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_line_output_gain, chip))) < 0)
-		goto ctl_error;
 	if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_vmixer, chip))) < 0)
 		goto ctl_error;
 #else
-- 
cgit v1.2.3


From 4c55bb0149b604901e4989d1ee0fddc53df8eb0c Mon Sep 17 00:00:00 2001
From: Giuliano Pochini <pochini@shiny.it>
Date: Sun, 15 Mar 2009 21:33:55 +0100
Subject: ALSA: echoaudio: remove line-out volume from vmixer cards

With this patch the drivers do not set the vmixer volume anymore at startup
because it is actually the output volume of the voices and ALSA mandates
that the volume must be 0 by default.

Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/echoaudio/indigo_dsp.c   | 12 ------------
 sound/pci/echoaudio/indigodj_dsp.c | 12 ------------
 sound/pci/echoaudio/indigoio_dsp.c | 12 ------------
 sound/pci/echoaudio/mia_dsp.c      | 12 ------------
 4 files changed, 48 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/echoaudio/indigo_dsp.c b/sound/pci/echoaudio/indigo_dsp.c
index f05e39f7aad..0b2cd9c8627 100644
--- a/sound/pci/echoaudio/indigo_dsp.c
+++ b/sound/pci/echoaudio/indigo_dsp.c
@@ -63,18 +63,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 	if ((err = init_line_levels(chip)) < 0)
 		return err;
 
-	/* Default routing of the virtual channels: all vchannels are routed
-	to the stereo output */
-	set_vmixer_gain(chip, 0, 0, 0);
-	set_vmixer_gain(chip, 1, 1, 0);
-	set_vmixer_gain(chip, 0, 2, 0);
-	set_vmixer_gain(chip, 1, 3, 0);
-	set_vmixer_gain(chip, 0, 4, 0);
-	set_vmixer_gain(chip, 1, 5, 0);
-	set_vmixer_gain(chip, 0, 6, 0);
-	set_vmixer_gain(chip, 1, 7, 0);
-	err = update_vmixer_level(chip);
-
 	DE_INIT(("init_hw done\n"));
 	return err;
 }
diff --git a/sound/pci/echoaudio/indigodj_dsp.c b/sound/pci/echoaudio/indigodj_dsp.c
index 90730a5ecb4..08392916691 100644
--- a/sound/pci/echoaudio/indigodj_dsp.c
+++ b/sound/pci/echoaudio/indigodj_dsp.c
@@ -63,18 +63,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 	if ((err = init_line_levels(chip)) < 0)
 		return err;
 
-	/* Default routing of the virtual channels: vchannels 0-3 and
-	vchannels 4-7 are routed to real channels 0-4 */
-	set_vmixer_gain(chip, 0, 0, 0);
-	set_vmixer_gain(chip, 1, 1, 0);
-	set_vmixer_gain(chip, 2, 2, 0);
-	set_vmixer_gain(chip, 3, 3, 0);
-	set_vmixer_gain(chip, 0, 4, 0);
-	set_vmixer_gain(chip, 1, 5, 0);
-	set_vmixer_gain(chip, 2, 6, 0);
-	set_vmixer_gain(chip, 3, 7, 0);
-	err = update_vmixer_level(chip);
-
 	DE_INIT(("init_hw done\n"));
 	return err;
 }
diff --git a/sound/pci/echoaudio/indigoio_dsp.c b/sound/pci/echoaudio/indigoio_dsp.c
index a7e09ec2107..0604c8a8522 100644
--- a/sound/pci/echoaudio/indigoio_dsp.c
+++ b/sound/pci/echoaudio/indigoio_dsp.c
@@ -63,18 +63,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 	if ((err = init_line_levels(chip)) < 0)
 		return err;
 
-	/* Default routing of the virtual channels: all vchannels are routed
-	to the stereo output */
-	set_vmixer_gain(chip, 0, 0, 0);
-	set_vmixer_gain(chip, 1, 1, 0);
-	set_vmixer_gain(chip, 0, 2, 0);
-	set_vmixer_gain(chip, 1, 3, 0);
-	set_vmixer_gain(chip, 0, 4, 0);
-	set_vmixer_gain(chip, 1, 5, 0);
-	set_vmixer_gain(chip, 0, 6, 0);
-	set_vmixer_gain(chip, 1, 7, 0);
-	err = update_vmixer_level(chip);
-
 	DE_INIT(("init_hw done\n"));
 	return err;
 }
diff --git a/sound/pci/echoaudio/mia_dsp.c b/sound/pci/echoaudio/mia_dsp.c
index 227386602f9..f7abe1b60a1 100644
--- a/sound/pci/echoaudio/mia_dsp.c
+++ b/sound/pci/echoaudio/mia_dsp.c
@@ -69,18 +69,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 	if ((err = init_line_levels(chip)))
 		return err;
 
-	/* Default routing of the virtual channels: vchannels 0-3 go to analog
-	outputs and vchannels 4-7 go to S/PDIF outputs */
-	set_vmixer_gain(chip, 0, 0, 0);
-	set_vmixer_gain(chip, 1, 1, 0);
-	set_vmixer_gain(chip, 0, 2, 0);
-	set_vmixer_gain(chip, 1, 3, 0);
-	set_vmixer_gain(chip, 2, 4, 0);
-	set_vmixer_gain(chip, 3, 5, 0);
-	set_vmixer_gain(chip, 2, 6, 0);
-	set_vmixer_gain(chip, 3, 7, 0);
-	err = update_vmixer_level(chip);
-
 	DE_INIT(("init_hw done\n"));
 	return err;
 }
-- 
cgit v1.2.3


From b8dbed0f095263b9ced5bd2e6d54743a7fa13f1b Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@perex.cz>
Date: Mon, 16 Mar 2009 14:56:58 +0100
Subject: ALSA: snd-hda-intel: Fix ALC662/ALC663 Beep Amplifier Index

ALC662/663 codecs have Beep Amplifier Index 0x04 not 0x05 in 0x0b NID.
Confirmed by testing on real hardware.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b794cba494c..672103d84ff 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -16951,7 +16951,7 @@ static int patch_alc662(struct hda_codec *codec)
 
 	if (!spec->cap_mixer)
 		set_capture_mixer(spec);
-	set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
+	set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
 
 	spec->vmaster_nid = 0x02;
 
-- 
cgit v1.2.3


From b9591448e5160ccd353d8547ade018cfdf2b3e09 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 16 Mar 2009 15:25:00 +0100
Subject: ALSA: hda - Fix ALC662 beep again

The previous commit breaks the (digital-) beep on ALC662.
ALC662 has the connection index 0x05 while ALC662 and ALC272 have the
index 0x04 for the beep widget.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 5 ++++-
 1 file changed, 4 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 672103d84ff..5ad0f8d72dd 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -16951,7 +16951,10 @@ static int patch_alc662(struct hda_codec *codec)
 
 	if (!spec->cap_mixer)
 		set_capture_mixer(spec);
-	set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
+	if (codec->vendor_id == 0x10ec0662)
+		set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
+	else
+		set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
 
 	spec->vmaster_nid = 0x02;
 
-- 
cgit v1.2.3


From ee5047102cf632351c418060bfbe3b6eb5c42e7b Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@perex.cz>
Date: Tue, 17 Mar 2009 14:30:31 +0100
Subject: ALSA: snd-hda-intel - add checks for invalid values to
 *query_supported_pcm()

If ratesp or formatsp values are zero, wrong values are passed to ALSA's
the PCM midlevel code. The bug is showed more later than expected.

Also, clean a bit the code.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 33 +++++++++++++++++++++++++--------
 1 file changed, 25 insertions(+), 8 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index cf6339436de..b90a2400f53 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2539,12 +2539,11 @@ EXPORT_SYMBOL_HDA(snd_hda_calc_stream_format);
 static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
 				u32 *ratesp, u64 *formatsp, unsigned int *bpsp)
 {
-	int i;
-	unsigned int val, streams;
+	unsigned int i, val, wcaps;
 
 	val = 0;
-	if (nid != codec->afg &&
-	    (get_wcaps(codec, nid) & AC_WCAP_FORMAT_OVRD)) {
+	wcaps = get_wcaps(codec, nid);
+	if (nid != codec->afg && (wcaps & AC_WCAP_FORMAT_OVRD)) {
 		val = snd_hda_param_read(codec, nid, AC_PAR_PCM);
 		if (val == -1)
 			return -EIO;
@@ -2558,15 +2557,20 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
 			if (val & (1 << i))
 				rates |= rate_bits[i].alsa_bits;
 		}
+		if (rates == 0) {
+			snd_printk(KERN_ERR "hda_codec: rates == 0 "
+				   "(nid=0x%x, val=0x%x, ovrd=%i)\n",
+					nid, val,
+					(wcaps & AC_WCAP_FORMAT_OVRD) ? 1 : 0);
+			return -EIO;
+		}
 		*ratesp = rates;
 	}
 
 	if (formatsp || bpsp) {
 		u64 formats = 0;
-		unsigned int bps;
-		unsigned int wcaps;
+		unsigned int streams, bps;
 
-		wcaps = get_wcaps(codec, nid);
 		streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM);
 		if (streams == -1)
 			return -EIO;
@@ -2619,6 +2623,15 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
 			formats |= SNDRV_PCM_FMTBIT_U8;
 			bps = 8;
 		}
+		if (formats == 0) {
+			snd_printk(KERN_ERR "hda_codec: formats == 0 "
+				   "(nid=0x%x, val=0x%x, ovrd=%i, "
+				   "streams=0x%x)\n",
+					nid, val,
+					(wcaps & AC_WCAP_FORMAT_OVRD) ? 1 : 0,
+					streams);
+			return -EIO;
+		}
 		if (formatsp)
 			*formatsp = formats;
 		if (bpsp)
@@ -2734,12 +2747,16 @@ static int hda_pcm_default_cleanup(struct hda_pcm_stream *hinfo,
 static int set_pcm_default_values(struct hda_codec *codec,
 				  struct hda_pcm_stream *info)
 {
+	int err;
+
 	/* query support PCM information from the given NID */
 	if (info->nid && (!info->rates || !info->formats)) {
-		snd_hda_query_supported_pcm(codec, info->nid,
+		err = snd_hda_query_supported_pcm(codec, info->nid,
 				info->rates ? NULL : &info->rates,
 				info->formats ? NULL : &info->formats,
 				info->maxbps ? NULL : &info->maxbps);
+		if (err < 0)
+			return err;
 	}
 	if (info->ops.open == NULL)
 		info->ops.open = hda_pcm_default_open_close;
-- 
cgit v1.2.3


From a2328d0249fce44381289525bd580b37d2105963 Mon Sep 17 00:00:00 2001
From: Giuliano Pochini <pochini@shiny.it>
Date: Thu, 19 Mar 2009 00:09:03 +0100
Subject: ALSA: Echoaudio: add support for Indigo express cards

This patch adds support for IndigoIOx and IndigoDJx.

Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/Kconfig                        |  20 ++++++
 sound/pci/echoaudio/Makefile             |   4 ++
 sound/pci/echoaudio/echoaudio.h          |   3 +
 sound/pci/echoaudio/echoaudio_dsp.h      |   9 ++-
 sound/pci/echoaudio/indigo_express_dsp.c | 119 +++++++++++++++++++++++++++++++
 sound/pci/echoaudio/indigodjx.c          | 107 +++++++++++++++++++++++++++
 sound/pci/echoaudio/indigodjx_dsp.c      |  68 ++++++++++++++++++
 sound/pci/echoaudio/indigoiox.c          | 109 ++++++++++++++++++++++++++++
 sound/pci/echoaudio/indigoiox_dsp.c      |  68 ++++++++++++++++++
 9 files changed, 505 insertions(+), 2 deletions(-)
 create mode 100644 sound/pci/echoaudio/indigo_express_dsp.c
 create mode 100644 sound/pci/echoaudio/indigodjx.c
 create mode 100644 sound/pci/echoaudio/indigodjx_dsp.c
 create mode 100644 sound/pci/echoaudio/indigoiox.c
 create mode 100644 sound/pci/echoaudio/indigoiox_dsp.c

(limited to 'sound/pci')

diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 82b9bddcdcd..9387ab00a41 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -400,6 +400,26 @@ config SND_INDIGODJ
 	  To compile this driver as a module, choose M here: the module
 	  will be called snd-indigodj
 
+config SND_INDIGOIOX
+	tristate "(Echoaudio) Indigo IOx"
+	select FW_LOADER
+	select SND_PCM
+	help
+	  Say 'Y' or 'M' to include support for Echoaudio Indigo IOx.
+
+	  To compile this driver as a module, choose M here: the module
+	  will be called snd-indigoiox
+
+config SND_INDIGODJX
+	tristate "(Echoaudio) Indigo DJx"
+	select FW_LOADER
+	select SND_PCM
+	help
+	  Say 'Y' or 'M' to include support for Echoaudio Indigo DJx.
+
+	  To compile this driver as a module, choose M here: the module
+	  will be called snd-indigodjx
+
 config SND_EMU10K1
 	tristate "Emu10k1 (SB Live!, Audigy, E-mu APS)"
 	select FW_LOADER
diff --git a/sound/pci/echoaudio/Makefile b/sound/pci/echoaudio/Makefile
index 7b576aeb3f8..1361de77e0c 100644
--- a/sound/pci/echoaudio/Makefile
+++ b/sound/pci/echoaudio/Makefile
@@ -15,6 +15,8 @@ snd-echo3g-objs := echo3g.o
 snd-indigo-objs := indigo.o
 snd-indigoio-objs := indigoio.o
 snd-indigodj-objs := indigodj.o
+snd-indigoiox-objs := indigoiox.o
+snd-indigodjx-objs := indigodjx.o
 
 obj-$(CONFIG_SND_DARLA20) += snd-darla20.o
 obj-$(CONFIG_SND_GINA20) += snd-gina20.o
@@ -28,3 +30,5 @@ obj-$(CONFIG_SND_ECHO3G) += snd-echo3g.o
 obj-$(CONFIG_SND_INDIGO) += snd-indigo.o
 obj-$(CONFIG_SND_INDIGOIO) += snd-indigoio.o
 obj-$(CONFIG_SND_INDIGODJ) += snd-indigodj.o
+obj-$(CONFIG_SND_INDIGOIOX) += snd-indigoiox.o
+obj-$(CONFIG_SND_INDIGODJX) += snd-indigodjx.o
diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h
index 1c88e051abf..f9490ae36c2 100644
--- a/sound/pci/echoaudio/echoaudio.h
+++ b/sound/pci/echoaudio/echoaudio.h
@@ -189,6 +189,9 @@
 #define INDIGO			0x0090
 #define INDIGO_IO		0x00a0
 #define INDIGO_DJ		0x00b0
+#define DC8			0x00c0
+#define INDIGO_IOX		0x00d0
+#define INDIGO_DJX		0x00e0
 #define ECHO3G			0x0100
 
 
diff --git a/sound/pci/echoaudio/echoaudio_dsp.h b/sound/pci/echoaudio/echoaudio_dsp.h
index e352f3ae292..cb7d75a0a50 100644
--- a/sound/pci/echoaudio/echoaudio_dsp.h
+++ b/sound/pci/echoaudio/echoaudio_dsp.h
@@ -576,8 +576,13 @@ SET_LAYLA24_FREQUENCY_REG command.
 #define E3G_ASIC_NOT_LOADED		0xffff
 #define E3G_BOX_TYPE_MASK		0xf0
 
-#define EXT_3GBOX_NC			0x01
-#define EXT_3GBOX_NOT_SET		0x02
+/* Indigo express control register values */
+#define INDIGO_EXPRESS_32000		0x02
+#define INDIGO_EXPRESS_44100		0x01
+#define INDIGO_EXPRESS_48000		0x00
+#define INDIGO_EXPRESS_DOUBLE_SPEED	0x10
+#define INDIGO_EXPRESS_QUAD_SPEED	0x04
+#define INDIGO_EXPRESS_CLOCK_MASK	0x17
 
 
 /*
diff --git a/sound/pci/echoaudio/indigo_express_dsp.c b/sound/pci/echoaudio/indigo_express_dsp.c
new file mode 100644
index 00000000000..9ab625e1565
--- /dev/null
+++ b/sound/pci/echoaudio/indigo_express_dsp.c
@@ -0,0 +1,119 @@
+/************************************************************************
+
+This file is part of Echo Digital Audio's generic driver library.
+Copyright Echo Digital Audio Corporation (c) 1998 - 2005
+All rights reserved
+www.echoaudio.com
+
+This library is free software; you can redistribute it and/or
+modify it under the terms of the GNU Lesser General Public
+License as published by the Free Software Foundation; either
+version 2.1 of the License, or (at your option) any later version.
+
+This library is distributed in the hope that it will be useful,
+but WITHOUT ANY WARRANTY; without even the implied warranty of
+MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+Lesser General Public License for more details.
+
+You should have received a copy of the GNU Lesser General Public
+License along with this library; if not, write to the Free Software
+Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+
+*************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+*************************************************************************/
+
+static int set_sample_rate(struct echoaudio *chip, u32 rate)
+{
+	u32 clock, control_reg, old_control_reg;
+
+	if (wait_handshake(chip))
+		return -EIO;
+
+	old_control_reg = le32_to_cpu(chip->comm_page->control_register);
+	control_reg = old_control_reg & ~INDIGO_EXPRESS_CLOCK_MASK;
+
+	switch (rate) {
+	case 32000:
+		clock = INDIGO_EXPRESS_32000;
+		break;
+	case 44100:
+		clock = INDIGO_EXPRESS_44100;
+		break;
+	case 48000:
+		clock = INDIGO_EXPRESS_48000;
+		break;
+	case 64000:
+		clock = INDIGO_EXPRESS_32000|INDIGO_EXPRESS_DOUBLE_SPEED;
+		break;
+	case 88200:
+		clock = INDIGO_EXPRESS_44100|INDIGO_EXPRESS_DOUBLE_SPEED;
+		break;
+	case 96000:
+		clock = INDIGO_EXPRESS_48000|INDIGO_EXPRESS_DOUBLE_SPEED;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	control_reg |= clock;
+	if (control_reg != old_control_reg) {
+		chip->comm_page->control_register = cpu_to_le32(control_reg);
+		chip->sample_rate = rate;
+		clear_handshake(chip);
+		return send_vector(chip, DSP_VC_UPDATE_CLOCKS);
+	}
+	return 0;
+}
+
+
+
+/* This function routes the sound from a virtual channel to a real output */
+static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
+			   int gain)
+{
+	int index;
+
+	if (snd_BUG_ON(pipe >= num_pipes_out(chip) ||
+		       output >= num_busses_out(chip)))
+		return -EINVAL;
+
+	if (wait_handshake(chip))
+		return -EIO;
+
+	chip->vmixer_gain[output][pipe] = gain;
+	index = output * num_pipes_out(chip) + pipe;
+	chip->comm_page->vmixer[index] = gain;
+
+	DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain));
+	return 0;
+}
+
+
+
+/* Tell the DSP to read and update virtual mixer levels in comm page. */
+static int update_vmixer_level(struct echoaudio *chip)
+{
+	if (wait_handshake(chip))
+		return -EIO;
+	clear_handshake(chip);
+	return send_vector(chip, DSP_VC_SET_VMIXER_GAIN);
+}
+
+
+
+static u32 detect_input_clocks(const struct echoaudio *chip)
+{
+	return ECHO_CLOCK_BIT_INTERNAL;
+}
+
+
+
+/* The IndigoIO has no ASIC. Just do nothing */
+static int load_asic(struct echoaudio *chip)
+{
+	return 0;
+}
diff --git a/sound/pci/echoaudio/indigodjx.c b/sound/pci/echoaudio/indigodjx.c
new file mode 100644
index 00000000000..3482ef69f49
--- /dev/null
+++ b/sound/pci/echoaudio/indigodjx.c
@@ -0,0 +1,107 @@
+/*
+ *  ALSA driver for Echoaudio soundcards.
+ *  Copyright (C) 2009 Giuliano Pochini <pochini@shiny.it>
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *  GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License
+ *  along with this program; if not, write to the Free Software
+ *  Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#define INDIGO_FAMILY
+#define ECHOCARD_INDIGO_DJX
+#define ECHOCARD_NAME "Indigo DJx"
+#define ECHOCARD_HAS_SUPER_INTERLEAVE
+#define ECHOCARD_HAS_VMIXER
+#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+
+/* Pipe indexes */
+#define PX_ANALOG_OUT	0	/* 8 */
+#define PX_DIGITAL_OUT	8	/* 0 */
+#define PX_ANALOG_IN	8	/* 0 */
+#define PX_DIGITAL_IN	8	/* 0 */
+#define PX_NUM		8
+
+/* Bus indexes */
+#define BX_ANALOG_OUT	0	/* 4 */
+#define BX_DIGITAL_OUT	4	/* 0 */
+#define BX_ANALOG_IN	4	/* 0 */
+#define BX_DIGITAL_IN	4	/* 0 */
+#define BX_NUM		4
+
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <linux/io.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/tlv.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include <asm/atomic.h>
+#include "echoaudio.h"
+
+MODULE_FIRMWARE("ea/loader_dsp.fw");
+MODULE_FIRMWARE("ea/indigo_djx_dsp.fw");
+
+#define FW_361_LOADER		0
+#define FW_INDIGO_DJX_DSP	1
+
+static const struct firmware card_fw[] = {
+	{0, "loader_dsp.fw"},
+	{0, "indigo_djx_dsp.fw"}
+};
+
+static struct pci_device_id snd_echo_ids[] = {
+	{0x1057, 0x3410, 0xECC0, 0x00E0, 0, 0, 0},	/* Indigo DJx*/
+	{0,}
+};
+
+static struct snd_pcm_hardware pcm_hardware_skel = {
+	.info = SNDRV_PCM_INFO_MMAP |
+		SNDRV_PCM_INFO_INTERLEAVED |
+		SNDRV_PCM_INFO_BLOCK_TRANSFER |
+		SNDRV_PCM_INFO_MMAP_VALID |
+		SNDRV_PCM_INFO_PAUSE |
+		SNDRV_PCM_INFO_SYNC_START,
+	.formats =	SNDRV_PCM_FMTBIT_U8 |
+			SNDRV_PCM_FMTBIT_S16_LE |
+			SNDRV_PCM_FMTBIT_S24_3LE |
+			SNDRV_PCM_FMTBIT_S32_LE |
+			SNDRV_PCM_FMTBIT_S32_BE,
+	.rates = 	SNDRV_PCM_RATE_32000 |
+			SNDRV_PCM_RATE_44100 |
+			SNDRV_PCM_RATE_48000 |
+			SNDRV_PCM_RATE_88200 |
+			SNDRV_PCM_RATE_96000,
+	.rate_min = 32000,
+	.rate_max = 96000,
+	.channels_min = 1,
+	.channels_max = 4,
+	.buffer_bytes_max = 262144,
+	.period_bytes_min = 32,
+	.period_bytes_max = 131072,
+	.periods_min = 2,
+	.periods_max = 220,
+};
+
+#include "indigodjx_dsp.c"
+#include "indigo_express_dsp.c"
+#include "echoaudio_dsp.c"
+#include "echoaudio.c"
diff --git a/sound/pci/echoaudio/indigodjx_dsp.c b/sound/pci/echoaudio/indigodjx_dsp.c
new file mode 100644
index 00000000000..f591fc2ed96
--- /dev/null
+++ b/sound/pci/echoaudio/indigodjx_dsp.c
@@ -0,0 +1,68 @@
+/************************************************************************
+
+This file is part of Echo Digital Audio's generic driver library.
+Copyright Echo Digital Audio Corporation (c) 1998 - 2005
+All rights reserved
+www.echoaudio.com
+
+This library is free software; you can redistribute it and/or
+modify it under the terms of the GNU Lesser General Public
+License as published by the Free Software Foundation; either
+version 2.1 of the License, or (at your option) any later version.
+
+This library is distributed in the hope that it will be useful,
+but WITHOUT ANY WARRANTY; without even the implied warranty of
+MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+Lesser General Public License for more details.
+
+You should have received a copy of the GNU Lesser General Public
+License along with this library; if not, write to the Free Software
+Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+
+*************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+*************************************************************************/
+
+static int update_vmixer_level(struct echoaudio *chip);
+static int set_vmixer_gain(struct echoaudio *chip, u16 output,
+			   u16 pipe, int gain);
+
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+	int err;
+
+	DE_INIT(("init_hw() - Indigo DJx\n"));
+	if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJX))
+		return -ENODEV;
+
+	err = init_dsp_comm_page(chip);
+	if (err < 0) {
+		DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+		return err;
+	}
+
+	chip->device_id = device_id;
+	chip->subdevice_id = subdevice_id;
+	chip->bad_board = TRUE;
+	chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJX_DSP];
+	/* Since this card has no ASIC, mark it as loaded so everything
+	   works OK */
+	chip->asic_loaded = TRUE;
+	chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL;
+
+	err = load_firmware(chip);
+	if (err < 0)
+		return err;
+	chip->bad_board = FALSE;
+
+	err = init_line_levels(chip);
+	if (err < 0)
+		return err;
+
+	DE_INIT(("init_hw done\n"));
+	return err;
+}
diff --git a/sound/pci/echoaudio/indigoiox.c b/sound/pci/echoaudio/indigoiox.c
new file mode 100644
index 00000000000..aebee27a40f
--- /dev/null
+++ b/sound/pci/echoaudio/indigoiox.c
@@ -0,0 +1,109 @@
+/*
+ *  ALSA driver for Echoaudio soundcards.
+ *  Copyright (C) 2009 Giuliano Pochini <pochini@shiny.it>
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *  GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License
+ *  along with this program; if not, write to the Free Software
+ *  Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#define INDIGO_FAMILY
+#define ECHOCARD_INDIGO_IOX
+#define ECHOCARD_NAME "Indigo IOx"
+#define ECHOCARD_HAS_MONITOR
+#define ECHOCARD_HAS_SUPER_INTERLEAVE
+#define ECHOCARD_HAS_VMIXER
+#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+
+/* Pipe indexes */
+#define PX_ANALOG_OUT	0	/* 8 */
+#define PX_DIGITAL_OUT	8	/* 0 */
+#define PX_ANALOG_IN	8	/* 2 */
+#define PX_DIGITAL_IN	10	/* 0 */
+#define PX_NUM		10
+
+/* Bus indexes */
+#define BX_ANALOG_OUT	0	/* 2 */
+#define BX_DIGITAL_OUT	2	/* 0 */
+#define BX_ANALOG_IN	2	/* 2 */
+#define BX_DIGITAL_IN	4	/* 0 */
+#define BX_NUM		4
+
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <linux/io.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/tlv.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include <asm/atomic.h>
+#include "echoaudio.h"
+
+MODULE_FIRMWARE("ea/loader_dsp.fw");
+MODULE_FIRMWARE("ea/indigo_iox_dsp.fw");
+
+#define FW_361_LOADER		0
+#define FW_INDIGO_IOX_DSP	1
+
+static const struct firmware card_fw[] = {
+	{0, "loader_dsp.fw"},
+	{0, "indigo_iox_dsp.fw"}
+};
+
+static struct pci_device_id snd_echo_ids[] = {
+	{0x1057, 0x3410, 0xECC0, 0x00D0, 0, 0, 0},	/* Indigo IOx */
+	{0,}
+};
+
+static struct snd_pcm_hardware pcm_hardware_skel = {
+	.info = SNDRV_PCM_INFO_MMAP |
+		SNDRV_PCM_INFO_INTERLEAVED |
+		SNDRV_PCM_INFO_BLOCK_TRANSFER |
+		SNDRV_PCM_INFO_MMAP_VALID |
+		SNDRV_PCM_INFO_PAUSE |
+		SNDRV_PCM_INFO_SYNC_START,
+	.formats =	SNDRV_PCM_FMTBIT_U8 |
+			SNDRV_PCM_FMTBIT_S16_LE |
+			SNDRV_PCM_FMTBIT_S24_3LE |
+			SNDRV_PCM_FMTBIT_S32_LE |
+			SNDRV_PCM_FMTBIT_S32_BE,
+	.rates = 	SNDRV_PCM_RATE_32000 |
+			SNDRV_PCM_RATE_44100 |
+			SNDRV_PCM_RATE_48000 |
+			SNDRV_PCM_RATE_88200 |
+			SNDRV_PCM_RATE_96000,
+	.rate_min = 32000,
+	.rate_max = 96000,
+	.channels_min = 1,
+	.channels_max = 8,
+	.buffer_bytes_max = 262144,
+	.period_bytes_min = 32,
+	.period_bytes_max = 131072,
+	.periods_min = 2,
+	.periods_max = 220,
+};
+
+#include "indigoiox_dsp.c"
+#include "indigo_express_dsp.c"
+#include "echoaudio_dsp.c"
+#include "echoaudio.c"
+
diff --git a/sound/pci/echoaudio/indigoiox_dsp.c b/sound/pci/echoaudio/indigoiox_dsp.c
new file mode 100644
index 00000000000..f357521c79e
--- /dev/null
+++ b/sound/pci/echoaudio/indigoiox_dsp.c
@@ -0,0 +1,68 @@
+/************************************************************************
+
+This file is part of Echo Digital Audio's generic driver library.
+Copyright Echo Digital Audio Corporation (c) 1998 - 2005
+All rights reserved
+www.echoaudio.com
+
+This library is free software; you can redistribute it and/or
+modify it under the terms of the GNU Lesser General Public
+License as published by the Free Software Foundation; either
+version 2.1 of the License, or (at your option) any later version.
+
+This library is distributed in the hope that it will be useful,
+but WITHOUT ANY WARRANTY; without even the implied warranty of
+MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+Lesser General Public License for more details.
+
+You should have received a copy of the GNU Lesser General Public
+License along with this library; if not, write to the Free Software
+Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+
+*************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+*************************************************************************/
+
+static int update_vmixer_level(struct echoaudio *chip);
+static int set_vmixer_gain(struct echoaudio *chip, u16 output,
+			   u16 pipe, int gain);
+
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+	int err;
+
+	DE_INIT(("init_hw() - Indigo IOx\n"));
+	if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IOX))
+		return -ENODEV;
+
+	err = init_dsp_comm_page(chip);
+	if (err < 0) {
+		DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+		return err;
+	}
+
+	chip->device_id = device_id;
+	chip->subdevice_id = subdevice_id;
+	chip->bad_board = TRUE;
+	chip->dsp_code_to_load = &card_fw[FW_INDIGO_IOX_DSP];
+	/* Since this card has no ASIC, mark it as loaded so everything
+	   works OK */
+	chip->asic_loaded = TRUE;
+	chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL;
+
+	err = load_firmware(chip);
+	if (err < 0)
+		return err;
+	chip->bad_board = FALSE;
+
+	err = init_line_levels(chip);
+	if (err < 0)
+		return err;
+
+	DE_INIT(("init_hw done\n"));
+	return err;
+}
-- 
cgit v1.2.3


From 97b71c94d691728b82052e9c4d6286fbc9965d7f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 18 Mar 2009 15:09:13 +0100
Subject: ALSA: hda - Don't reset BDL unnecessarily

So far, the prepare callback is called multiple times, BDL entries
are reset and re-programmed at each time.

This patch adds the check to avoid the reset of BDL entries when the
same parameters are used.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 46 ++++++++++++++++++++++++++++++++--------------
 1 file changed, 32 insertions(+), 14 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 6bcf5af6edc..ba97795d89c 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1076,8 +1076,7 @@ static int azx_setup_periods(struct azx *chip,
 	azx_sd_writel(azx_dev, SD_BDLPL, 0);
 	azx_sd_writel(azx_dev, SD_BDLPU, 0);
 
-	period_bytes = snd_pcm_lib_period_bytes(substream);
-	azx_dev->period_bytes = period_bytes;
+	period_bytes = azx_dev->period_bytes;
 	periods = azx_dev->bufsize / period_bytes;
 
 	/* program the initial BDL entries */
@@ -1124,9 +1123,6 @@ static int azx_setup_periods(struct azx *chip,
  error:
 	snd_printk(KERN_ERR "Too many BDL entries: buffer=%d, period=%d\n",
 		   azx_dev->bufsize, period_bytes);
-	/* reset */
-	azx_sd_writel(azx_dev, SD_BDLPL, 0);
-	azx_sd_writel(azx_dev, SD_BDLPU, 0);
 	return -EINVAL;
 }
 
@@ -1429,6 +1425,11 @@ static int azx_pcm_close(struct snd_pcm_substream *substream)
 static int azx_pcm_hw_params(struct snd_pcm_substream *substream,
 			     struct snd_pcm_hw_params *hw_params)
 {
+	struct azx_dev *azx_dev = get_azx_dev(substream);
+
+	azx_dev->bufsize = 0;
+	azx_dev->period_bytes = 0;
+	azx_dev->format_val = 0;
 	return snd_pcm_lib_malloc_pages(substream,
 					params_buffer_bytes(hw_params));
 }
@@ -1443,6 +1444,9 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream)
 	azx_sd_writel(azx_dev, SD_BDLPL, 0);
 	azx_sd_writel(azx_dev, SD_BDLPU, 0);
 	azx_sd_writel(azx_dev, SD_CTL, 0);
+	azx_dev->bufsize = 0;
+	azx_dev->period_bytes = 0;
+	azx_dev->format_val = 0;
 
 	hinfo->ops.cleanup(hinfo, apcm->codec, substream);
 
@@ -1456,23 +1460,37 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream)
 	struct azx_dev *azx_dev = get_azx_dev(substream);
 	struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream];
 	struct snd_pcm_runtime *runtime = substream->runtime;
+	unsigned int bufsize, period_bytes, format_val;
+	int err;
 
-	azx_dev->bufsize = snd_pcm_lib_buffer_bytes(substream);
-	azx_dev->format_val = snd_hda_calc_stream_format(runtime->rate,
-							 runtime->channels,
-							 runtime->format,
-							 hinfo->maxbps);
-	if (!azx_dev->format_val) {
+	format_val = snd_hda_calc_stream_format(runtime->rate,
+						runtime->channels,
+						runtime->format,
+						hinfo->maxbps);
+	if (!format_val) {
 		snd_printk(KERN_ERR SFX
 			   "invalid format_val, rate=%d, ch=%d, format=%d\n",
 			   runtime->rate, runtime->channels, runtime->format);
 		return -EINVAL;
 	}
 
+	bufsize = snd_pcm_lib_buffer_bytes(substream);
+	period_bytes = snd_pcm_lib_period_bytes(substream);
+
 	snd_printdd("azx_pcm_prepare: bufsize=0x%x, format=0x%x\n",
-		    azx_dev->bufsize, azx_dev->format_val);
-	if (azx_setup_periods(chip, substream, azx_dev) < 0)
-		return -EINVAL;
+		    bufsize, format_val);
+
+	if (bufsize != azx_dev->bufsize ||
+	    period_bytes != azx_dev->period_bytes ||
+	    format_val != azx_dev->format_val) {
+		azx_dev->bufsize = bufsize;
+		azx_dev->period_bytes = period_bytes;
+		azx_dev->format_val = format_val;
+		err = azx_setup_periods(chip, substream, azx_dev);
+		if (err < 0)
+			return err;
+	}
+
 	azx_setup_controller(chip, azx_dev);
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		azx_dev->fifo_size = azx_sd_readw(azx_dev, SD_FIFOSIZE) + 1;
-- 
cgit v1.2.3


From 1dddab400b7ad028b21d7d5b060e4a068d6d3cd9 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 18 Mar 2009 15:15:37 +0100
Subject: ALSA: hda - Don't reset stream at each prepare callback

Don't reset the stream at each prepare callback but do it only once
after the open.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 33 ++++++++++++++++++++++-----------
 1 file changed, 22 insertions(+), 11 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index ba97795d89c..8b2e4160de8 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -859,13 +859,18 @@ static void azx_stream_start(struct azx *chip, struct azx_dev *azx_dev)
 		      SD_CTL_DMA_START | SD_INT_MASK);
 }
 
-/* stop a stream */
-static void azx_stream_stop(struct azx *chip, struct azx_dev *azx_dev)
+/* stop DMA */
+static void azx_stream_clear(struct azx *chip, struct azx_dev *azx_dev)
 {
-	/* stop DMA */
 	azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) &
 		      ~(SD_CTL_DMA_START | SD_INT_MASK));
 	azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK); /* to be sure */
+}
+
+/* stop a stream */
+static void azx_stream_stop(struct azx *chip, struct azx_dev *azx_dev)
+{
+	azx_stream_clear(chip, azx_dev);
 	/* disable SIE */
 	azx_writeb(chip, INTCTL,
 		   azx_readb(chip, INTCTL) & ~(1 << azx_dev->index));
@@ -1126,18 +1131,14 @@ static int azx_setup_periods(struct azx *chip,
 	return -EINVAL;
 }
 
-/*
- * set up the SD for streaming
- */
-static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev)
+/* reset stream */
+static void azx_stream_reset(struct azx *chip, struct azx_dev *azx_dev)
 {
 	unsigned char val;
 	int timeout;
 
-	/* make sure the run bit is zero for SD */
-	azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) &
-		      ~SD_CTL_DMA_START);
-	/* reset stream */
+	azx_stream_clear(chip, azx_dev);
+
 	azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) |
 		      SD_CTL_STREAM_RESET);
 	udelay(3);
@@ -1154,7 +1155,15 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev)
 	while (((val = azx_sd_readb(azx_dev, SD_CTL)) & SD_CTL_STREAM_RESET) &&
 	       --timeout)
 		;
+}
 
+/*
+ * set up the SD for streaming
+ */
+static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev)
+{
+	/* make sure the run bit is zero for SD */
+	azx_stream_clear(chip, azx_dev);
 	/* program the stream_tag */
 	azx_sd_writel(azx_dev, SD_CTL,
 		      (azx_sd_readl(azx_dev, SD_CTL) & ~SD_CTL_STREAM_TAG_MASK)|
@@ -1399,6 +1408,8 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
 	runtime->private_data = azx_dev;
 	snd_pcm_set_sync(substream);
 	mutex_unlock(&chip->open_mutex);
+
+	azx_stream_reset(chip, azx_dev);
 	return 0;
 }
 
-- 
cgit v1.2.3


From 07a1e81355245ca65ab16c7b4ae2332e52ed7acd Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 19 Mar 2009 17:08:19 +0100
Subject: ALSA: hda - Don't show the current connection for power widgets

The power-widgets have no connection selection, so skip the check
in proc output, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_proc.c | 6 ++++--
 1 file changed, 4 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 93b25ba4d00..639cf0edaa9 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -399,8 +399,10 @@ static void print_conn_list(struct snd_info_buffer *buffer,
 {
 	int c, curr = -1;
 
-	if (conn_len > 1 && wid_type != AC_WID_AUD_MIX &&
-	    wid_type != AC_WID_VOL_KNB)
+	if (conn_len > 1 &&
+	    wid_type != AC_WID_AUD_MIX &&
+	    wid_type != AC_WID_VOL_KNB &&
+	    wid_type != AC_WID_POWER)
 		curr = snd_hda_codec_read(codec, nid, 0,
 					  AC_VERB_GET_CONNECT_SEL, 0);
 	snd_iprintf(buffer, "  Connection: %d\n", conn_len);
-- 
cgit v1.2.3


From c468ac29e63b9927275a94379d00b367f0f97c43 Mon Sep 17 00:00:00 2001
From: Wolfram Sang <w.sang@pengutronix.de>
Date: Fri, 20 Mar 2009 10:08:11 +0100
Subject: ALSA: sound/ali5451: typo: s/resouces/resources/

Signed-off-by: Wolfram Sang <w.sang@pengutronix.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ali5451/ali5451.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index 1a0fd65ec28..9069c78c2dc 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -2142,7 +2142,7 @@ static int __devinit snd_ali_resources(struct snd_ali *codec)
 {
 	int err;
 
-	snd_ali_printk("resouces allocation ...\n");
+	snd_ali_printk("resources allocation ...\n");
 	err = pci_request_regions(codec->pci, "ALI 5451");
 	if (err < 0)
 		return err;
@@ -2154,7 +2154,7 @@ static int __devinit snd_ali_resources(struct snd_ali *codec)
 		return -EBUSY;
 	}
 	codec->irq = codec->pci->irq;
-	snd_ali_printk("resouces allocated.\n");
+	snd_ali_printk("resources allocated.\n");
 	return 0;
 }
 static int snd_ali_dev_free(struct snd_device *device)
-- 
cgit v1.2.3


From 2d864c499a77129dc6aa4f7552ddf2885e4a9c47 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 20 Mar 2009 12:52:47 +0100
Subject: ALSA: hda - Detect digital-mic inputs on ALC663 / ALC272

Fix the detection of digital-mic inputs on ALC663 / ALC272 codecs
in the auto-detection mode.  The automatic mic switch via plugging
isn't implemented yet, though.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 63 ++++++++++++++++++++++++++++++++-----------
 1 file changed, 47 insertions(+), 16 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 5ad0f8d72dd..b69d9864f6f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -16725,26 +16725,58 @@ static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin,
 	return 0;
 }
 
+/* return the index of the src widget from the connection list of the nid.
+ * return -1 if not found
+ */
+static int alc662_input_pin_idx(struct hda_codec *codec, hda_nid_t nid,
+				hda_nid_t src)
+{
+	hda_nid_t conn_list[HDA_MAX_CONNECTIONS];
+	int i, conns;
+
+	conns = snd_hda_get_connections(codec, nid, conn_list,
+					ARRAY_SIZE(conn_list));
+	if (conns < 0)
+		return -1;
+	for (i = 0; i < conns; i++)
+		if (conn_list[i] == src)
+			return i;
+	return -1;
+}
+
+static int alc662_is_input_pin(struct hda_codec *codec, hda_nid_t nid)
+{
+	unsigned int pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
+	return (pincap & AC_PINCAP_IN) != 0;
+}
+
 /* create playback/capture controls for input pins */
-static int alc662_auto_create_analog_input_ctls(struct alc_spec *spec,
+static int alc662_auto_create_analog_input_ctls(struct hda_codec *codec,
 						const struct auto_pin_cfg *cfg)
 {
+	struct alc_spec *spec = codec->spec;
 	struct hda_input_mux *imux = &spec->private_imux[0];
 	int i, err, idx;
 
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
-		if (alc880_is_input_pin(cfg->input_pins[i])) {
-			idx = alc880_input_pin_idx(cfg->input_pins[i]);
-			err = new_analog_input(spec, cfg->input_pins[i],
-					       auto_pin_cfg_labels[i],
-					       idx, 0x0b);
-			if (err < 0)
-				return err;
-			imux->items[imux->num_items].label =
-				auto_pin_cfg_labels[i];
-			imux->items[imux->num_items].index =
-				alc880_input_pin_idx(cfg->input_pins[i]);
-			imux->num_items++;
+		if (alc662_is_input_pin(codec, cfg->input_pins[i])) {
+			idx = alc662_input_pin_idx(codec, 0x0b,
+						   cfg->input_pins[i]);
+			if (idx >= 0) {
+				err = new_analog_input(spec, cfg->input_pins[i],
+						       auto_pin_cfg_labels[i],
+						       idx, 0x0b);
+				if (err < 0)
+					return err;
+			}
+			idx = alc662_input_pin_idx(codec, 0x22,
+						   cfg->input_pins[i]);
+			if (idx >= 0) {
+				imux->items[imux->num_items].label =
+					auto_pin_cfg_labels[i];
+				imux->items[imux->num_items].index = idx;
+				imux->num_items++;
+			}
 		}
 	}
 	return 0;
@@ -16794,7 +16826,6 @@ static void alc662_auto_init_hp_out(struct hda_codec *codec)
 		alc662_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
 }
 
-#define alc662_is_input_pin(nid)	alc880_is_input_pin(nid)
 #define ALC662_PIN_CD_NID		ALC880_PIN_CD_NID
 
 static void alc662_auto_init_analog_input(struct hda_codec *codec)
@@ -16804,7 +16835,7 @@ static void alc662_auto_init_analog_input(struct hda_codec *codec)
 
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
 		hda_nid_t nid = spec->autocfg.input_pins[i];
-		if (alc662_is_input_pin(nid)) {
+		if (alc662_is_input_pin(codec, nid)) {
 			alc_set_input_pin(codec, nid, i);
 			if (nid != ALC662_PIN_CD_NID)
 				snd_hda_codec_write(codec, nid, 0,
@@ -16844,7 +16875,7 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
 					   "Headphone");
 	if (err < 0)
 		return err;
-	err = alc662_auto_create_analog_input_ctls(spec, &spec->autocfg);
+	err = alc662_auto_create_analog_input_ctls(codec, &spec->autocfg);
 	if (err < 0)
 		return err;
 
-- 
cgit v1.2.3


From 234b4346a064f8a2a488da10b3c1e640fb778a17 Mon Sep 17 00:00:00 2001
From: Pascal de Bruijn <pascal@unilogicnetworks.net>
Date: Mon, 23 Mar 2009 11:15:59 +0100
Subject: ALSA: hda - Add function id to proc output

This patch does two things:
 Output Intel HDA Function Id in /proc/asound/cardX/codec#X
 Align Vendor/Subsystem/Revision Ids to 8 characters, front-padded with zeros

Before:
 Vendor Id: 0x11d41884
 Subsystem Id: 0x103c281a
 Revision Id: 0x100100

After:
 Function Id: 0x1
 Vendor Id: 0x11d41884
 Subsystem Id: 0x103c281a
 Revision Id: 0x0100100

As report on the Kernel Bugzilla #12888

Signed-off-by: Pascal de Bruijn <pascal@unilogicnetworks.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 6 +++---
 sound/pci/hda/hda_codec.h | 1 +
 sound/pci/hda/hda_proc.c  | 5 +++--
 3 files changed, 7 insertions(+), 5 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index b90a2400f53..1b5575ecb0a 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -647,9 +647,9 @@ static void /*__devinit*/ setup_fg_nodes(struct hda_codec *codec)
 
 	total_nodes = snd_hda_get_sub_nodes(codec, AC_NODE_ROOT, &nid);
 	for (i = 0; i < total_nodes; i++, nid++) {
-		unsigned int func;
-		func = snd_hda_param_read(codec, nid, AC_PAR_FUNCTION_TYPE);
-		switch (func & 0xff) {
+		codec->function_id = snd_hda_param_read(codec, nid,
+						AC_PAR_FUNCTION_TYPE) & 0xff;
+		switch (codec->function_id) {
 		case AC_GRP_AUDIO_FUNCTION:
 			codec->afg = nid;
 			break;
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 079e1ab718d..2fdecf4b0eb 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -739,6 +739,7 @@ struct hda_codec {
 	hda_nid_t mfg;	/* MFG node id */
 
 	/* ids */
+	u32 function_id;
 	u32 vendor_id;
 	u32 subsystem_id;
 	u32 revision_id;
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 639cf0edaa9..93d7499350c 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -469,8 +469,9 @@ static void print_codec_info(struct snd_info_entry *entry,
 	snd_iprintf(buffer, "Codec: %s\n",
 		    codec->name ? codec->name : "Not Set");
 	snd_iprintf(buffer, "Address: %d\n", codec->addr);
-	snd_iprintf(buffer, "Vendor Id: 0x%x\n", codec->vendor_id);
-	snd_iprintf(buffer, "Subsystem Id: 0x%x\n", codec->subsystem_id);
+	snd_iprintf(buffer, "Function Id: 0x%x\n", codec->function_id);
+	snd_iprintf(buffer, "Vendor Id: 0x%08x\n", codec->vendor_id);
+	snd_iprintf(buffer, "Subsystem Id: 0x%08x\n", codec->subsystem_id);
 	snd_iprintf(buffer, "Revision Id: 0x%x\n", codec->revision_id);
 
 	if (codec->mfg)
-- 
cgit v1.2.3


From 52ca15b7c0c711eb37f5e4b769e8488e5c516d43 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 23 Mar 2009 12:51:55 +0100
Subject: ALSA: hda - Avoid output amp manipulation to digital mic pins

Don't set amp-out values to pins without PINCAP_OUT capability,
which are usually assigned for digital mics on ALC663/ALC272.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 9 ++++++++-
 1 file changed, 8 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b69d9864f6f..965a531d2fb 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -16750,6 +16750,12 @@ static int alc662_is_input_pin(struct hda_codec *codec, hda_nid_t nid)
 	return (pincap & AC_PINCAP_IN) != 0;
 }
 
+static int alc662_is_output_pin(struct hda_codec *codec, hda_nid_t nid)
+{
+	unsigned int pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
+	return (pincap & AC_PINCAP_OUT) != 0;
+}
+
 /* create playback/capture controls for input pins */
 static int alc662_auto_create_analog_input_ctls(struct hda_codec *codec,
 						const struct auto_pin_cfg *cfg)
@@ -16837,7 +16843,8 @@ static void alc662_auto_init_analog_input(struct hda_codec *codec)
 		hda_nid_t nid = spec->autocfg.input_pins[i];
 		if (alc662_is_input_pin(codec, nid)) {
 			alc_set_input_pin(codec, nid, i);
-			if (nid != ALC662_PIN_CD_NID)
+			if (nid != ALC662_PIN_CD_NID &&
+			    alc662_is_output_pin(codec, nid))
 				snd_hda_codec_write(codec, nid, 0,
 						    AC_VERB_SET_AMP_GAIN_MUTE,
 						    AMP_OUT_MUTE);
-- 
cgit v1.2.3


From 1327a32b878b5ed2113c63557b6f4f949f821857 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 23 Mar 2009 13:07:47 +0100
Subject: ALSA: hda - Cache pin-cap values

Added snd_hda_query_pin_caps() to read and cache pin-cap values
to avoid too frequently issuing the same verbs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c      | 16 ++++++++++++++++
 sound/pci/hda/hda_generic.c    |  2 +-
 sound/pci/hda/hda_local.h      |  1 +
 sound/pci/hda/patch_realtek.c  |  6 +++---
 sound/pci/hda/patch_sigmatel.c |  7 +++----
 5 files changed, 24 insertions(+), 8 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 1b5575ecb0a..0f70d2d102e 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1052,6 +1052,7 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_cleanup_stream);
 
 /* FIXME: more better hash key? */
 #define HDA_HASH_KEY(nid,dir,idx) (u32)((nid) + ((idx) << 16) + ((dir) << 24))
+#define HDA_HASH_PINCAP_KEY(nid) (u32)((nid) + (0x02 << 24))
 #define INFO_AMP_CAPS	(1<<0)
 #define INFO_AMP_VOL(ch)	(1 << (1 + (ch)))
 
@@ -1142,6 +1143,21 @@ int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir,
 }
 EXPORT_SYMBOL_HDA(snd_hda_override_amp_caps);
 
+u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid)
+{
+	struct hda_amp_info *info;
+
+	info = get_alloc_amp_hash(codec, HDA_HASH_PINCAP_KEY(nid));
+	if (!info)
+		return 0;
+	if (!info->head.val) {
+		info->amp_caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
+		info->head.val |= INFO_AMP_CAPS;
+	}
+	return info->amp_caps;
+}
+EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps);
+
 /*
  * read the current volume to info
  * if the cache exists, read the cache value.
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 2c81a683e8f..1d5797a9668 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -144,7 +144,7 @@ static int add_new_node(struct hda_codec *codec, struct hda_gspec *spec, hda_nid
 	node->type = (node->wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
 
 	if (node->type == AC_WID_PIN) {
-		node->pin_caps = snd_hda_param_read(codec, node->nid, AC_PAR_PIN_CAP);
+		node->pin_caps = snd_hda_query_pin_caps(codec, node->nid);
 		node->pin_ctl = snd_hda_codec_read(codec, node->nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
 		node->def_cfg = snd_hda_codec_get_pincfg(codec, node->nid);
 	}
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 27428c718fd..83349013b4d 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -411,6 +411,7 @@ static inline u32 get_wcaps(struct hda_codec *codec, hda_nid_t nid)
 u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction);
 int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir,
 			      unsigned int caps);
+u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid);
 
 int snd_hda_ctl_add(struct hda_codec *codec, struct snd_kcontrol *kctl);
 void snd_hda_ctls_clear(struct hda_codec *codec);
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 965a531d2fb..bf7e64e2c46 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -770,7 +770,7 @@ static void alc_set_input_pin(struct hda_codec *codec, hda_nid_t nid,
 
 	if (auto_pin_type <= AUTO_PIN_FRONT_MIC) {
 		unsigned int pincap;
-		pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
+		pincap = snd_hda_query_pin_caps(codec, nid);
 		pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT;
 		if (pincap & AC_PINCAP_VREF_80)
 			val = PIN_VREF80;
@@ -16746,13 +16746,13 @@ static int alc662_input_pin_idx(struct hda_codec *codec, hda_nid_t nid,
 
 static int alc662_is_input_pin(struct hda_codec *codec, hda_nid_t nid)
 {
-	unsigned int pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
+	unsigned int pincap = snd_hda_query_pin_caps(codec, nid);
 	return (pincap & AC_PINCAP_IN) != 0;
 }
 
 static int alc662_is_output_pin(struct hda_codec *codec, hda_nid_t nid)
 {
-	unsigned int pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
+	unsigned int pincap = snd_hda_query_pin_caps(codec, nid);
 	return (pincap & AC_PINCAP_OUT) != 0;
 }
 
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 4da72403fc8..b1c180a9e9b 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2537,8 +2537,7 @@ static int stac92xx_build_pcms(struct hda_codec *codec)
 
 static unsigned int stac92xx_get_vref(struct hda_codec *codec, hda_nid_t nid)
 {
-	unsigned int pincap = snd_hda_param_read(codec, nid,
-						 AC_PAR_PIN_CAP);
+	unsigned int pincap = snd_hda_query_pin_caps(codec, nid);
 	pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT;
 	if (pincap & AC_PINCAP_VREF_100)
 		return AC_PINCTL_VREF_100;
@@ -2799,7 +2798,7 @@ static hda_nid_t check_line_out_switch(struct hda_codec *codec)
 	if (cfg->line_out_type != AUTO_PIN_LINE_OUT)
 		return 0;
 	nid = cfg->input_pins[AUTO_PIN_LINE];
-	pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
+	pincap = snd_hda_query_pin_caps(codec, nid);
 	if (pincap & AC_PINCAP_OUT)
 		return nid;
 	return 0;
@@ -2822,7 +2821,7 @@ static hda_nid_t check_mic_out_switch(struct hda_codec *codec)
 		/* some laptops have an internal analog microphone
 		 * which can't be used as a output */
 		if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) {
-			pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
+			pincap = snd_hda_query_pin_caps(codec, nid);
 			if (pincap & AC_PINCAP_OUT)
 				return nid;
 		}
-- 
cgit v1.2.3


From e82c025b501a1ca62dec40989817dbb17c0b9167 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 23 Mar 2009 15:17:38 +0100
Subject: ALSA: hda - Fix the wrong pin-cap check in patch_realtek.c

The check for the amp-output must be done for widget-caps rather than
pin-caps as implemented in the recent change...  Simply a thinko.

Also, add the similar checks to all places that put output-amp mutes
in the initialization.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 20 +++++++++-----------
 1 file changed, 9 insertions(+), 11 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index bf7e64e2c46..8dcbb04e57b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4207,7 +4207,8 @@ static void alc880_auto_init_analog_input(struct hda_codec *codec)
 		hda_nid_t nid = spec->autocfg.input_pins[i];
 		if (alc880_is_input_pin(nid)) {
 			alc_set_input_pin(codec, nid, i);
-			if (nid != ALC880_PIN_CD_NID)
+			if (nid != ALC880_PIN_CD_NID &&
+			    (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP))
 				snd_hda_codec_write(codec, nid, 0,
 						    AC_VERB_SET_AMP_GAIN_MUTE,
 						    AMP_OUT_MUTE);
@@ -5673,7 +5674,8 @@ static void alc260_auto_init_analog_input(struct hda_codec *codec)
 		hda_nid_t nid = spec->autocfg.input_pins[i];
 		if (nid >= 0x12) {
 			alc_set_input_pin(codec, nid, i);
-			if (nid != ALC260_PIN_CD_NID)
+			if (nid != ALC260_PIN_CD_NID &&
+			    (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP))
 				snd_hda_codec_write(codec, nid, 0,
 						    AC_VERB_SET_AMP_GAIN_MUTE,
 						    AMP_OUT_MUTE);
@@ -9153,7 +9155,8 @@ static void alc883_auto_init_analog_input(struct hda_codec *codec)
 		hda_nid_t nid = spec->autocfg.input_pins[i];
 		if (alc883_is_input_pin(nid)) {
 			alc_set_input_pin(codec, nid, i);
-			if (nid != ALC883_PIN_CD_NID)
+			if (nid != ALC883_PIN_CD_NID &&
+			    (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP))
 				snd_hda_codec_write(codec, nid, 0,
 						    AC_VERB_SET_AMP_GAIN_MUTE,
 						    AMP_OUT_MUTE);
@@ -14880,7 +14883,8 @@ static void alc861vd_auto_init_analog_input(struct hda_codec *codec)
 		hda_nid_t nid = spec->autocfg.input_pins[i];
 		if (alc861vd_is_input_pin(nid)) {
 			alc_set_input_pin(codec, nid, i);
-			if (nid != ALC861VD_PIN_CD_NID)
+			if (nid != ALC861VD_PIN_CD_NID &&
+			    (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP))
 				snd_hda_codec_write(codec, nid, 0,
 						AC_VERB_SET_AMP_GAIN_MUTE,
 						AMP_OUT_MUTE);
@@ -16750,12 +16754,6 @@ static int alc662_is_input_pin(struct hda_codec *codec, hda_nid_t nid)
 	return (pincap & AC_PINCAP_IN) != 0;
 }
 
-static int alc662_is_output_pin(struct hda_codec *codec, hda_nid_t nid)
-{
-	unsigned int pincap = snd_hda_query_pin_caps(codec, nid);
-	return (pincap & AC_PINCAP_OUT) != 0;
-}
-
 /* create playback/capture controls for input pins */
 static int alc662_auto_create_analog_input_ctls(struct hda_codec *codec,
 						const struct auto_pin_cfg *cfg)
@@ -16844,7 +16842,7 @@ static void alc662_auto_init_analog_input(struct hda_codec *codec)
 		if (alc662_is_input_pin(codec, nid)) {
 			alc_set_input_pin(codec, nid, i);
 			if (nid != ALC662_PIN_CD_NID &&
-			    alc662_is_output_pin(codec, nid))
+			    (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP))
 				snd_hda_codec_write(codec, nid, 0,
 						    AC_VERB_SET_AMP_GAIN_MUTE,
 						    AMP_OUT_MUTE);
-- 
cgit v1.2.3


From a23b688f4d5c2490a50677b30011a677d8edf3d0 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 23 Mar 2009 15:21:36 +0100
Subject: ALSA: hda - Don't create empty/single-item input source

In patch_realtek.c, don't create empty or single-item "Input Source"
control elements that are simply superfluous.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 47 ++++++++++++++++++++++++++++++++-----------
 1 file changed, 35 insertions(+), 12 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 8dcbb04e57b..7a3c6db6d5b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1595,8 +1595,7 @@ static int alc_cap_sw_put(struct snd_kcontrol *kcontrol,
 				     snd_hda_mixer_amp_switch_put);
 }
 
-#define DEFINE_CAPMIX(num) \
-static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \
+#define _DEFINE_CAPMIX(num) \
 	{ \
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
 		.name = "Capture Switch", \
@@ -1617,7 +1616,9 @@ static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \
 		.get = alc_cap_vol_get, \
 		.put = alc_cap_vol_put, \
 		.tlv = { .c = alc_cap_vol_tlv }, \
-	}, \
+	}
+
+#define _DEFINE_CAPSRC(num) \
 	{ \
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
 		/* .name = "Capture Source", */ \
@@ -1626,15 +1627,28 @@ static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \
 		.info = alc_mux_enum_info, \
 		.get = alc_mux_enum_get, \
 		.put = alc_mux_enum_put, \
-	}, \
-	{ } /* end */ \
+	}
+
+#define DEFINE_CAPMIX(num) \
+static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \
+	_DEFINE_CAPMIX(num),				      \
+	_DEFINE_CAPSRC(num),				      \
+	{ } /* end */					      \
+}
+
+#define DEFINE_CAPMIX_NOSRC(num) \
+static struct snd_kcontrol_new alc_capture_mixer_nosrc ## num[] = { \
+	_DEFINE_CAPMIX(num),					    \
+	{ } /* end */						    \
 }
 
 /* up to three ADCs */
 DEFINE_CAPMIX(1);
 DEFINE_CAPMIX(2);
 DEFINE_CAPMIX(3);
-
+DEFINE_CAPMIX_NOSRC(1);
+DEFINE_CAPMIX_NOSRC(2);
+DEFINE_CAPMIX_NOSRC(3);
 
 /*
  * ALC880 5-stack model
@@ -4298,13 +4312,22 @@ static void alc880_auto_init(struct hda_codec *codec)
 
 static void set_capture_mixer(struct alc_spec *spec)
 {
-	static struct snd_kcontrol_new *caps[3] = {
-		alc_capture_mixer1,
-		alc_capture_mixer2,
-		alc_capture_mixer3,
+	static struct snd_kcontrol_new *caps[2][3] = {
+		{ alc_capture_mixer_nosrc1,
+		  alc_capture_mixer_nosrc2,
+		  alc_capture_mixer_nosrc3 },
+		{ alc_capture_mixer1,
+		  alc_capture_mixer2,
+		  alc_capture_mixer3 },
 	};
-	if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3)
-		spec->cap_mixer = caps[spec->num_adc_nids - 1];
+	if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) {
+		int mux;
+		if (spec->input_mux && spec->input_mux->num_items > 1)
+			mux = 1;
+		else
+			mux = 0;
+		spec->cap_mixer = caps[mux][spec->num_adc_nids - 1];
+	}
 }
 
 #define set_beep_amp(spec, nid, idx, dir) \
-- 
cgit v1.2.3


From 14bafe3278e5da952a6586a5a9a9d286566049ed Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 23 Mar 2009 16:35:39 +0100
Subject: ALSA: hda - Use cached calls to get widget caps and pin caps

Replace with the standard function calls to use caches for reading
the widget caps and pin caps.

hda_proc.c is still using the direct verbs to get raw values as
much as possible.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c      | 3 +--
 sound/pci/hda/patch_sigmatel.c | 3 +--
 2 files changed, 2 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 0f70d2d102e..a4e5e595211 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2321,8 +2321,7 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
 				 * don't power down the widget if it controls
 				 * eapd and EAPD_BTLENABLE is set.
 				 */
-				pincap = snd_hda_param_read(codec, nid,
-							    AC_PAR_PIN_CAP);
+				pincap = snd_hda_query_pin_caps(codec, nid);
 				if (pincap & AC_PINCAP_EAPD) {
 					int eapd = snd_hda_codec_read(codec,
 						nid, 0,
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index b1c180a9e9b..b5e108aa8f6 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2869,8 +2869,7 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid)
 	conn_len = snd_hda_get_connections(codec, nid, conn,
 					   HDA_MAX_CONNECTIONS);
 	for (j = 0; j < conn_len; j++) {
-		wcaps = snd_hda_param_read(codec, conn[j],
-					   AC_PAR_AUDIO_WIDGET_CAP);
+		wcaps = get_wcaps(codec, conn[j]);
 		wtype = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
 		/* we check only analog outputs */
 		if (wtype != AC_WID_AUD_OUT || (wcaps & AC_WCAP_DIGITAL))
-- 
cgit v1.2.3


From 9b6682ff4c69484b6955f89f7902e3dde2481bed Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 23 Mar 2009 22:50:52 +0100
Subject: ALSA: hda - Add quirk for Acer Ferrari 5000

Add a quirk model=acer-aspire for Acer Ferrari 5000 with ALC883 codec.
Note that model=auto doesn't work for this laptop because of broken BIOS
(that doesn't set the subsystem id properly).

Tested-by: Russ Dill <russ.dill@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7a3c6db6d5b..82097790f6f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8677,6 +8677,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC883_3ST_6ch_DIG),
 	SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0x0090, "Acer Aspire", ALC883_ACER_ASPIRE),
+	SND_PCI_QUIRK(0x1025, 0x010a, "Acer Ferrari 5000", ALC883_ACER_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE),
-- 
cgit v1.2.3