From 7e6c3989af5baee999ef9a4424e85938cba8d34a Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Wed, 4 Nov 2009 21:03:46 -0500 Subject: ALSA: intel8x0: Mute External Amplifier by default for another Sony model BugLink: https://bugs.launchpad.net/bugs/474972 This Sony model needs External Amplifier muted for audible playback, so make sure we set the inv_eapd quirk. Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/intel8x0.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 754867ed478..aac20fb4aad 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -1948,6 +1948,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { .name = "HP xw4200", /* AD1981B*/ .type = AC97_TUNE_HP_ONLY }, + { + .subvendor = 0x104d, + .subdevice = 0x8144, + .name = "Sony", + .type = AC97_TUNE_INV_EAPD + }, { .subvendor = 0x104d, .subdevice = 0x8197, -- cgit v1.2.3 From 06fe9fb4182177fb046e6d934f80254dd90956ea Mon Sep 17 00:00:00 2001 From: Dirk Hohndel Date: Mon, 28 Sep 2009 21:43:57 -0400 Subject: tree-wide: fix a very frequent spelling mistake something-bility is spelled as something-blity so a grep for 'blit' would find these lines this is so trivial that I didn't split it by subsystem / copy additional maintainers - all changes are to comments The only purpose is to get fewer false positives when grepping around the kernel sources. Signed-off-by: Dirk Hohndel Signed-off-by: Jiri Kosina --- sound/pci/ice1712/juli.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index fd948bfd9ae..9c0f78ea2c4 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -380,7 +380,7 @@ static struct snd_kcontrol_new juli_mute_controls[] __devinitdata = { * inputs) are fed from Xilinx. * * I even checked traces on board and coded a support in driver for - * an alternative possiblity - the unused I2S ICE output channels + * an alternative possibility - the unused I2S ICE output channels * switched to HW-IN/SPDIF-IN and providing the monitoring signal to * the DAC - to no avail. The I2S outputs seem to be unconnected. * -- cgit v1.2.3 From b71a8eb0fa64ec6d00175f479e3ef851703568af Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Tue, 6 Oct 2009 12:42:51 +0200 Subject: tree-wide: fix typos "selct" + "slect" -> "select" MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This patch was generated by git grep -E -i -l 's(le|el)ct' | xargs -r perl -p -i -e 's/([Ss])(le|el)ct/$1elect/ with only skipping net/netfilter/xt_SECMARK.c and include/linux/netfilter/xt_SECMARK.h which have a struct member called selctx. Signed-off-by: Uwe Kleine-König Signed-off-by: Jiri Kosina --- sound/pci/hda/patch_cirrus.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 8ba306856d3..7b0446fa600 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -947,7 +947,7 @@ static void init_input(struct hda_codec *codec) coef |= 0x0500; /* DMIC2 enable 2 channels, disable GPIO1 */ if (is_active_pin(codec, CS_DMIC1_PIN_NID)) coef |= 0x1800; /* DMIC1 enable 2 channels, disable GPIO0 - * No effect if SPDIF_OUT2 is slected in + * No effect if SPDIF_OUT2 is selected in * IDX_SPDIF_CTL. */ cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); -- cgit v1.2.3 From 401de8184a4d94688962b9258fe10ab309ffda9c Mon Sep 17 00:00:00 2001 From: Akinobu Mita Date: Fri, 13 Nov 2009 16:02:56 +0900 Subject: ALSA: ice1712: Use bitrev8 Signed-off-by: Akinobu Mita Signed-off-by: Takashi Iwai --- sound/pci/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 75c602b5b13..351654cf7b0 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -570,6 +570,7 @@ config SND_ICE1712 tristate "ICEnsemble ICE1712 (Envy24)" select SND_MPU401_UART select SND_AC97_CODEC + select BITREVERSE help Say Y here to include support for soundcards based on the ICE1712 (Envy24) chip. -- cgit v1.2.3 From 50d40f187f9182ee8caa1b83f80a0e11e2226baa Mon Sep 17 00:00:00 2001 From: Aleksey Kunitskiy Date: Sat, 14 Nov 2009 15:18:54 +0200 Subject: ALSA: ice1724 - Patch for suspend/resume for ESI Juli@ Add proper suspend/resume code for Juli@ cards. Based on ice1724 suspend/resume work of Igor Chernyshev. Fixes bug https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4413 Tested on linux-2.6.31.6 Signed-off-by: Aleksey Kunitskiy Signed-off-by: Takashi Iwai --- sound/pci/ice1712/juli.c | 32 ++++++++++++++++++++++++++++++++ 1 file changed, 32 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index fd948bfd9ae..f5020ad99a1 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -503,6 +503,31 @@ static int __devinit juli_add_controls(struct snd_ice1712 *ice) return 0; } +/* + * suspend/resume + * */ + +#ifdef CONFIG_PM +static int juli_resume(struct snd_ice1712 *ice) +{ + struct snd_akm4xxx *ak = ice->akm; + struct juli_spec *spec = ice->spec; + /* akm4358 un-reset, un-mute */ + snd_akm4xxx_reset(ak, 0); + /* reinit ak4114 */ + snd_ak4114_reinit(spec->ak4114); + return 0; +} + +static int juli_suspend(struct snd_ice1712 *ice) +{ + struct snd_akm4xxx *ak = ice->akm; + /* akm4358 reset and soft-mute */ + snd_akm4xxx_reset(ak, 1); + return 0; +} +#endif + /* * initialize the chip */ @@ -646,6 +671,13 @@ static int __devinit juli_init(struct snd_ice1712 *ice) ice->set_spdif_clock = juli_set_spdif_clock; ice->spdif.ops.open = juli_spdif_in_open; + +#ifdef CONFIG_PM + ice->pm_resume = juli_resume; + ice->pm_suspend = juli_suspend; + ice->pm_suspend_enabled = 1; +#endif + return 0; } -- cgit v1.2.3 From 5e08fe570c2dbabb5015c37049eb9a451e55c890 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 14 Nov 2009 14:37:19 +0100 Subject: ALSA: ice1724 - Fix section mismatch in prodigy_hd2_resume() Remove invlid __devinit prefix from the suspend callback. Signed-off-by: Takashi Iwai --- sound/pci/ice1712/prodigy_hifi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index c75515f5be6..6a9fee3ee78 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -1100,7 +1100,7 @@ static void ak4396_init(struct snd_ice1712 *ice) } #ifdef CONFIG_PM -static int __devinit prodigy_hd2_resume(struct snd_ice1712 *ice) +static int prodigy_hd2_resume(struct snd_ice1712 *ice) { /* initialize ak4396 codec and restore previous mixer volumes */ struct prodigy_hifi_spec *spec = ice->spec; -- cgit v1.2.3 From bf97402052483c125a9ea7bf13df0dd9b4134078 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Mon, 16 Nov 2009 11:07:17 +0200 Subject: ALSA: ice1724 - make some bitfields unsigned This is a clean up and doesn't change the behavior. Bit fields should always be unsigned. Otherwise pm_suspend_enabled will be -1 when you want it to be 1. The other bad thing is that the sparse checker will complain 36 times if they aren't unsigned. The other bitfields in that struct are unsigned already. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1712.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index 9da2dae64c5..d063149e704 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -382,8 +382,8 @@ struct snd_ice1712 { #ifdef CONFIG_PM int (*pm_suspend)(struct snd_ice1712 *); int (*pm_resume)(struct snd_ice1712 *); - int pm_suspend_enabled:1; - int pm_saved_is_spdif_master:1; + unsigned int pm_suspend_enabled:1; + unsigned int pm_saved_is_spdif_master:1; unsigned int pm_saved_spdif_ctrl; unsigned char pm_saved_spdif_cfg; unsigned int pm_saved_route; -- cgit v1.2.3 From bbb3c644bd9967753ce8c214c5e64b27c361d2a4 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Tue, 24 Nov 2009 22:51:05 -0500 Subject: ALSA: intel8x0: Mute External Amplifier by default for Gateway 4525GZ BugLink: https://bugs.launchpad.net/bugs/487884 This Gateway model needs External Amplifier muted for audible playback, so set the inv_eapd quirk for it. Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/intel8x0.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index aac20fb4aad..b990143636f 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -2062,6 +2062,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { .name = "MSI P4 ATX 645 Ultra", .type = AC97_TUNE_HP_ONLY }, + { + .subvendor = 0x161f, + .subdevice = 0x203a, + .name = "Gateway 4525GZ", /* AD1981B */ + .type = AC97_TUNE_INV_EAPD + }, { .subvendor = 0x1734, .subdevice = 0x0088, -- cgit v1.2.3 From fb716c0b7bed36064cd41d800c8f339f41adf084 Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Fri, 27 Nov 2009 18:18:33 +0100 Subject: snd-fm801: autodetect SF64-PCR (tuner-only) card MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit When primary AC97 is not found, don't fail with tons of AC97 errors. Assume that the card is SF64-PCR (tuner-only). This makes the SF64-PCR radio card work "out of the box". Also fixes a bug that can cause an oops here:         if (tea575x_tuner > 0 && (tea575x_tuner & 0x000f) < 4) { when tea575x_tuner == 16, it passes this check and causes problems a couple lines below:         chip->tea.ops = &snd_fm801_tea_ops[(tea575x_tuner & 0x000f) - 1]; Tested with SF64-PCR, but I don't have any of those sound or sound+radio cards to test if I didn't break anything. Signed-off-by: Ondrej Zary Signed-off-by: Takashi Iwai --- sound/pci/fm801.c | 40 +++++++++++++++++++++++++++------------- 1 file changed, 27 insertions(+), 13 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 60cdb9e0b68..83508b3964f 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -55,7 +55,7 @@ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card * * 1 = MediaForte 256-PCS * 2 = MediaForte 256-PCPR * 3 = MediaForte 64-PCR - * 16 = setup tuner only (this is additional bit), i.e. SF-64-PCR FM card + * 16 = setup tuner only (this is additional bit), i.e. SF64-PCR FM card * High 16-bits are video (radio) device number + 1 */ static int tea575x_tuner[SNDRV_CARDS]; @@ -67,7 +67,10 @@ MODULE_PARM_DESC(id, "ID string for the FM801 soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable FM801 soundcard."); module_param_array(tea575x_tuner, int, NULL, 0444); -MODULE_PARM_DESC(tea575x_tuner, "Enable TEA575x tuner."); +MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (1 = SF256-PCS, 2=SF256-PCPR, 3=SF64-PCR, +16=tuner-only)."); + +#define TUNER_ONLY (1<<4) +#define TUNER_TYPE_MASK (~TUNER_ONLY & 0xFFFF) /* * Direct registers @@ -160,7 +163,7 @@ struct fm801 { unsigned int multichannel: 1, /* multichannel support */ secondary: 1; /* secondary codec */ unsigned char secondary_addr; /* address of the secondary codec */ - unsigned int tea575x_tuner; /* tuner flags */ + unsigned int tea575x_tuner; /* tuner access method & flags */ unsigned short ply_ctrl; /* playback control */ unsigned short cap_ctrl; /* capture control */ @@ -1287,7 +1290,7 @@ static int snd_fm801_chip_init(struct fm801 *chip, int resume) { unsigned short cmdw; - if (chip->tea575x_tuner & 0x0010) + if (chip->tea575x_tuner & TUNER_ONLY) goto __ac97_ok; /* codec cold reset + AC'97 warm reset */ @@ -1296,11 +1299,13 @@ static int snd_fm801_chip_init(struct fm801 *chip, int resume) udelay(100); outw(0, FM801_REG(chip, CODEC_CTRL)); - if (wait_for_codec(chip, 0, AC97_RESET, msecs_to_jiffies(750)) < 0) { - snd_printk(KERN_ERR "Primary AC'97 codec not found\n"); - if (! resume) - return -EIO; - } + if (wait_for_codec(chip, 0, AC97_RESET, msecs_to_jiffies(750)) < 0) + if (!resume) { + snd_printk(KERN_INFO "Primary AC'97 codec not found, " + "assume SF64-PCR (tuner-only)\n"); + chip->tea575x_tuner = 3 | TUNER_ONLY; + goto __ac97_ok; + } if (chip->multichannel) { if (chip->secondary_addr) { @@ -1414,7 +1419,7 @@ static int __devinit snd_fm801_create(struct snd_card *card, return err; } chip->port = pci_resource_start(pci, 0); - if ((tea575x_tuner & 0x0010) == 0) { + if ((tea575x_tuner & TUNER_ONLY) == 0) { if (request_irq(pci->irq, snd_fm801_interrupt, IRQF_SHARED, "FM801", chip)) { snd_printk(KERN_ERR "unable to grab IRQ %d\n", chip->irq); @@ -1429,6 +1434,14 @@ static int __devinit snd_fm801_create(struct snd_card *card, chip->multichannel = 1; snd_fm801_chip_init(chip, 0); + /* init might set tuner access method */ + tea575x_tuner = chip->tea575x_tuner; + + if (chip->irq >= 0 && (tea575x_tuner & TUNER_ONLY)) { + pci_clear_master(pci); + free_irq(chip->irq, chip); + chip->irq = -1; + } if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { snd_fm801_free(chip); @@ -1438,12 +1451,13 @@ static int __devinit snd_fm801_create(struct snd_card *card, snd_card_set_dev(card, &pci->dev); #ifdef TEA575X_RADIO - if (tea575x_tuner > 0 && (tea575x_tuner & 0x000f) < 4) { + if ((tea575x_tuner & TUNER_TYPE_MASK) > 0 && + (tea575x_tuner & TUNER_TYPE_MASK) < 4) { chip->tea.dev_nr = tea575x_tuner >> 16; chip->tea.card = card; chip->tea.freq_fixup = 10700; chip->tea.private_data = chip; - chip->tea.ops = &snd_fm801_tea_ops[(tea575x_tuner & 0x000f) - 1]; + chip->tea.ops = &snd_fm801_tea_ops[(tea575x_tuner & TUNER_TYPE_MASK) - 1]; snd_tea575x_init(&chip->tea); } #endif @@ -1483,7 +1497,7 @@ static int __devinit snd_card_fm801_probe(struct pci_dev *pci, sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, chip->port, chip->irq); - if (tea575x_tuner[dev] & 0x0010) + if (chip->tea575x_tuner & TUNER_ONLY) goto __fm801_tuner_only; if ((err = snd_fm801_pcm(chip, 0, NULL)) < 0) { -- cgit v1.2.3 From af901ca181d92aac3a7dc265144a9081a86d8f39 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Andr=C3=A9=20Goddard=20Rosa?= Date: Sat, 14 Nov 2009 13:09:05 -0200 Subject: tree-wide: fix assorted typos all over the place MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit That is "success", "unknown", "through", "performance", "[re|un]mapping" , "access", "default", "reasonable", "[con]currently", "temperature" , "channel", "[un]used", "application", "example","hierarchy", "therefore" , "[over|under]flow", "contiguous", "threshold", "enough" and others. Signed-off-by: André Goddard Rosa Signed-off-by: Jiri Kosina --- sound/pci/ca0106/ca0106_proc.c | 2 +- sound/pci/cs46xx/imgs/cwcdma.asp | 9 +++++---- sound/pci/emu10k1/emu10k1x.c | 2 +- sound/pci/hda/patch_cmedia.c | 2 +- sound/pci/hda/patch_realtek.c | 2 +- sound/pci/rme9652/hdspm.c | 4 ++-- 6 files changed, 11 insertions(+), 10 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c index c62b7d10ec6..8d13092300d 100644 --- a/sound/pci/ca0106/ca0106_proc.c +++ b/sound/pci/ca0106/ca0106_proc.c @@ -233,7 +233,7 @@ static void snd_ca0106_proc_dump_iec958( struct snd_info_buffer *buffer, u32 val snd_iprintf(buffer, "user-defined\n"); break; default: - snd_iprintf(buffer, "unkown\n"); + snd_iprintf(buffer, "unknown\n"); break; } snd_iprintf(buffer, "Sample Bits: "); diff --git a/sound/pci/cs46xx/imgs/cwcdma.asp b/sound/pci/cs46xx/imgs/cwcdma.asp index 09d24c76f03..a65e1193c89 100644 --- a/sound/pci/cs46xx/imgs/cwcdma.asp +++ b/sound/pci/cs46xx/imgs/cwcdma.asp @@ -26,10 +26,11 @@ // // // The purpose of this code is very simple: make it possible to tranfser -// the samples 'as they are' with no alteration from a PCMreader SCB (DMA from host) -// to any other SCB. This is useful for AC3 throug SPDIF. SRC (source rate converters) -// task always alters the samples in some how, however it's from 48khz -> 48khz. The -// alterations are not audible, but AC3 wont work. +// the samples 'as they are' with no alteration from a PCMreader +// SCB (DMA from host) to any other SCB. This is useful for AC3 through SPDIF. +// SRC (source rate converters) task always alters the samples in somehow, +// however it's from 48khz -> 48khz. +// The alterations are not audible, but AC3 wont work. // // ... // | diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 36e08bd2b3c..360e3809a60 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -184,7 +184,7 @@ MODULE_PARM_DESC(enable, "Enable the EMU10K1X soundcard."); * The hardware has 3 channels for playback and 1 for capture. * - channel 0 is the front channel * - channel 1 is the rear channel - * - channel 2 is the center/lfe chanel + * - channel 2 is the center/lfe channel * Volume is controlled by the AC97 for the front and rear channels by * the PCM Playback Volume, Sigmatel Surround Playback Volume and * Surround Playback Volume. The Sigmatel 4-Speaker Stereo switch affects diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 780e1a72114..8917071d5b6 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -66,7 +66,7 @@ struct cmi_spec { struct hda_pcm pcm_rec[2]; /* PCM information */ - /* pin deafault configuration */ + /* pin default configuration */ hda_nid_t pin_nid[NUM_PINS]; unsigned int def_conf[NUM_PINS]; unsigned int pin_def_confs; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ff20048504b..872731eb49e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6619,7 +6619,7 @@ static struct hda_input_mux alc889A_mb31_capture_source = { /* Front Mic (0x01) unused */ { "Line", 0x2 }, /* Line 2 (0x03) unused */ - /* CD (0x04) unsused? */ + /* CD (0x04) unused? */ }, }; diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 0dce331a2a3..a1b10d1a384 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -3017,7 +3017,7 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry, insel = "Coaxial"; break; default: - insel = "Unkown"; + insel = "Unknown"; } switch (hdspm->control_register & HDSPM_SyncRefMask) { @@ -3028,7 +3028,7 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry, syncref = "MADI"; break; default: - syncref = "Unkown"; + syncref = "Unknown"; } snd_iprintf(buffer, "Inputsel = %s, SyncRef = %s\n", insel, syncref); -- cgit v1.2.3 From 482e46d4b7c9bfbb2edc047fafa85cee1b0fc1e1 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 9 Dec 2009 12:43:44 +0100 Subject: ALSA: ice1724 - aureon - modify WM8770 Master & DAC volume The volume levels in original implementation are incorrect and does not match the dB scale. The real range is linear (in the sense of the dB scale) from 0dB to -100dB. Remove logaritmic table and make all volumes from range 0dB..100dB. The tests are in RedHat's bugzilla #540817. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/pci/ice1712/aureon.c | 31 +++++++------------------------ 1 file changed, 7 insertions(+), 24 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c index 110d16e5273..765d7bd4c3d 100644 --- a/sound/pci/ice1712/aureon.c +++ b/sound/pci/ice1712/aureon.c @@ -689,32 +689,14 @@ static int aureon_ac97_mmute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e return change; } -static const DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -12700, 100, 1); +static const DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -10000, 100, 1); static const DECLARE_TLV_DB_SCALE(db_scale_wm_pcm, -6400, 50, 1); static const DECLARE_TLV_DB_SCALE(db_scale_wm_adc, -1200, 100, 0); static const DECLARE_TLV_DB_SCALE(db_scale_ac97_master, -4650, 150, 0); static const DECLARE_TLV_DB_SCALE(db_scale_ac97_gain, -3450, 150, 0); -/* - * Logarithmic volume values for WM8770 - * Computed as 20 * Log10(255 / x) - */ -static const unsigned char wm_vol[256] = { - 127, 48, 42, 39, 36, 34, 33, 31, 30, 29, 28, 27, 27, 26, 25, 25, 24, 24, 23, - 23, 22, 22, 21, 21, 21, 20, 20, 20, 19, 19, 19, 18, 18, 18, 18, 17, 17, 17, - 17, 16, 16, 16, 16, 15, 15, 15, 15, 15, 15, 14, 14, 14, 14, 14, 13, 13, 13, - 13, 13, 13, 13, 12, 12, 12, 12, 12, 12, 12, 11, 11, 11, 11, 11, 11, 11, 11, - 11, 10, 10, 10, 10, 10, 10, 10, 10, 10, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 8, 8, - 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 6, 6, 6, - 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, - 5, 5, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 3, 3, 3, 3, 3, - 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, - 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, - 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, - 0, 0 -}; - -#define WM_VOL_MAX (sizeof(wm_vol) - 1) +#define WM_VOL_MAX 100 +#define WM_VOL_CNT 101 /* 0dB .. -100dB */ #define WM_VOL_MUTE 0x8000 static void wm_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned short vol, unsigned short master) @@ -724,7 +706,8 @@ static void wm_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned sho if ((master & WM_VOL_MUTE) || (vol & WM_VOL_MUTE)) nvol = 0; else - nvol = 127 - wm_vol[(((vol & ~WM_VOL_MUTE) * (master & ~WM_VOL_MUTE)) / 127) & WM_VOL_MAX]; + nvol = ((vol % WM_VOL_CNT) * (master % WM_VOL_CNT)) / + WM_VOL_MAX; wm_put(ice, index, nvol); wm_put_nocache(ice, index, 0x180 | nvol); @@ -820,7 +803,7 @@ static int wm_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info * uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = voices; uinfo->value.integer.min = 0; /* mute (-101dB) */ - uinfo->value.integer.max = 0x7F; /* 0dB */ + uinfo->value.integer.max = WM_VOL_MAX; /* 0dB */ return 0; } @@ -850,7 +833,7 @@ static int wm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value * snd_ice1712_save_gpio_status(ice); for (i = 0; i < voices; i++) { unsigned int vol = ucontrol->value.integer.value[i]; - if (vol > 0x7f) + if (vol > WM_VOL_MAX) continue; vol |= spec->vol[ofs+i]; if (vol != spec->vol[ofs+i]) { -- cgit v1.2.3 From f74890277a196949e4004fe2955e1d4fb3930f98 Mon Sep 17 00:00:00 2001 From: Steve Soule Date: Mon, 14 Dec 2009 11:06:03 -0700 Subject: ALSA: ac97_codec - increase timeout for analog sections to 5 second I have a Soundblaster 16PCI. For many years, alsa has had a bug where not all of the card's controls are detected (many alsa versions, many kernel versions). In particular, Master Playback Volume is usually not detected, and so I get no sound or extremely faint sound. The problem has always been inconsistent: sometimes all of the controls are detected correctly, and sometimes a partial set is detected. It works correctly about 10% of the time. Finally, I got around to tracking down the problem. When the driver fails, it prints the kernel message "AC'97 0 analog subsections not ready". This message is generated from the function snd_ac97_mixer() in ac97_codec.c. The message indicates that the card failed to come back after reset within the time limit. The time limit is 120 milliseconds. I tried increasing the time limit to 1 second, and found that this made the driver work about 70% of the time. I tried increasing it to 5 seconds, and it now seems to work 100% of the time. I expect that this change would be completely harmless for existing cards that work, and would only introduce additional delay for cards that do not work. ALSA bug#4032. Signed-off-by: Steve Soule Signed-off-by: Jaroslav Kysela --- 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 20cb60afb20..c1192062300 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -2122,7 +2122,7 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, } /* nothing should be in powerdown mode */ snd_ac97_write_cache(ac97, AC97_GENERAL_PURPOSE, 0); - end_time = jiffies + msecs_to_jiffies(120); + end_time = jiffies + msecs_to_jiffies(5000); do { if ((snd_ac97_read(ac97, AC97_POWERDOWN) & 0x0f) == 0x0f) goto __ready_ok; -- cgit v1.2.3 From 440b004cf953bec2bc8cd91c64ae707fd7e25327 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Sun, 20 Dec 2009 12:04:08 +0100 Subject: ALSA: hda/realtek: Remove extra .capsrc_nids initialization for ALC889_INTEL Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8b375771b3a..2d3f4f893ef 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9238,8 +9238,6 @@ static struct alc_config_preset alc882_presets[] = { .dac_nids = alc883_dac_nids, .num_adc_nids = ARRAY_SIZE(alc889_adc_nids), .adc_nids = alc889_adc_nids, - .capsrc_nids = alc889_capsrc_nids, - .capsrc_nids = alc889_capsrc_nids, .dig_out_nid = ALC883_DIGOUT_NID, .dig_in_nid = ALC883_DIGIN_NID, .slave_dig_outs = alc883_slave_dig_outs, -- cgit v1.2.3