From f67d8176ba9a3dbc33454cd67057184b2ef5ee31 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 4 Feb 2009 23:30:19 +0100 Subject: ALSA: hda - Add quirk for FSC Amilo Xi2550 Added model=fujisu-pi2515 for FSC Amilo Xi2550 with ALC883 codec. Refernece: Novell bnc#450979 https://bugzilla.novell.com/show_bug.cgi?id=450979 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0040101f615..a3baa33aedf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8515,6 +8515,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1558, 0, "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", + ALC883_FUJITSU_PI2515), SND_PCI_QUIRK(0x1734, 0x1108, "Fujitsu AMILO Pi2515", ALC883_FUJITSU_PI2515), SND_PCI_QUIRK(0x1734, 0x113d, "Fujitsu AMILO Xa3530", ALC888_FUJITSU_XA3530), -- cgit v1.2.3 From e8c0ee5d77ec0f144c753a622c67dd96fa195d50 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Feb 2009 07:34:28 +0100 Subject: ALSA: hda - Fix misc workqueue issues Some fixes regarding snd-hda-intel workqueue: - Use create_singlethread_workqueue() instead of create_workqueue() as per-CPU work isn't required. - Allocate workq name string properly - Renamed the workq name to "hd-audio*" to be more obvious. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 9 +++++---- sound/pci/hda/hda_codec.h | 1 + 2 files changed, 6 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b7bba7dc7cf..0b708134d12 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -487,7 +487,6 @@ int /*__devinit*/ snd_hda_bus_new(struct snd_card *card, { struct hda_bus *bus; int err; - char qname[8]; static struct snd_device_ops dev_ops = { .dev_register = snd_hda_bus_dev_register, .dev_free = snd_hda_bus_dev_free, @@ -517,10 +516,12 @@ int /*__devinit*/ snd_hda_bus_new(struct snd_card *card, mutex_init(&bus->cmd_mutex); INIT_LIST_HEAD(&bus->codec_list); - snprintf(qname, sizeof(qname), "hda%d", card->number); - bus->workq = create_workqueue(qname); + snprintf(bus->workq_name, sizeof(bus->workq_name), + "hd-audio%d", card->number); + bus->workq = create_singlethread_workqueue(bus->workq_name); if (!bus->workq) { - snd_printk(KERN_ERR "cannot create workqueue %s\n", qname); + snd_printk(KERN_ERR "cannot create workqueue %s\n", + bus->workq_name); kfree(bus); return -ENOMEM; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 5810ef58840..09a332ada0c 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -614,6 +614,7 @@ struct hda_bus { /* unsolicited event queue */ struct hda_bus_unsolicited *unsol; + char workq_name[16]; struct workqueue_struct *workq; /* common workqueue for codecs */ /* assigned PCMs */ -- cgit v1.2.3 From 894dcd78782842924527598b0b764c9b4e679e21 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 6 Feb 2009 08:13:07 +0100 Subject: sound: usb-audio: handle wMaxPacketSize for FIXED_ENDPOINT devices For audio devices that do not have proper audio descriptors (e.g., Edirol UA-20), we use hardcoded parameters from our quirks list. However, we must still read the maximum packet size from the standard endpoint descriptor; otherwise, we might use packets that are too big and therefore rejected by the USB core. Signed-off-by: Clemens Ladisch Cc: Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index c709b956322..2ab83129d9b 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2966,6 +2966,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, return -EINVAL; } alts = &iface->altsetting[fp->altset_idx]; + fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); usb_set_interface(chip->dev, fp->iface, 0); init_usb_pitch(chip->dev, fp->iface, alts, fp); init_usb_sample_rate(chip->dev, fp->iface, alts, fp, fp->rate_max); -- cgit v1.2.3 From 4453dba54de7e517b0cd6f5e4a3f4af3b34f9e79 Mon Sep 17 00:00:00 2001 From: Eero Nurkkala Date: Fri, 6 Feb 2009 12:01:04 +0200 Subject: ASoC: TLV320AIC3X: Fix kcontrol's private value use in put callback Function snd_soc_dapm_put_volsw_aic3x misuses the kcontrol's private value by still accessing it as bitfields even SOC_SINGLE_VALUE constructs it as a pointer into struct soc_mixer_control after the commit 4eaa9819dc08d7bfd1065ce530e31b18a119dcaf. This was causing arbitrary register writes when touching the controls defined with SOC_DAPM_SINGLE_AIC3X. Signed-off-by: Eero Nurkkala Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index b47a749c5ea..aea0cb72d80 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -165,10 +165,13 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0x0f; - int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0x01; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + int max = mc->max; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; unsigned short val, val_mask; int ret; struct snd_soc_dapm_path *path; -- cgit v1.2.3 From 397d5aeeb5a2c9ca6108899a04b35a51cd904503 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 6 Feb 2009 12:01:05 +0200 Subject: ASoC: WM8990: Fix kcontrol's private value use in put callback Function wm899x_outpga_put_volsw_vu misuses the kcontrol's private value by still accessing it as bitfields even SOC_SINGLE_VALUE constructs it as a pointer into struct soc_mixer_control after the commit 4eaa9819dc08d7bfd1065ce530e31b18a119dcaf. This is very similar fix than fix to TLV320AIC3X codec made by Eero Nurkkala . This fix is compile tested only. Signed-off-by: Jarkko Nikula Cc: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8990.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 1cbb7b9b51c..a5731faa150 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -176,7 +176,9 @@ static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int reg = mc->reg; int ret; u16 val; -- cgit v1.2.3 From c6e8f2daadc6d61a32b7486a1058c8f1f9baa499 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Feb 2009 12:45:52 +0100 Subject: ALSA: hda - Add missing initialization for ALC272 ALC272 needs EAPD for speaker outputs as well as other similar ALC codecs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a3baa33aedf..ac1a6e72843 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1037,6 +1037,7 @@ do_sku: case 0x10ec0267: case 0x10ec0268: case 0x10ec0269: + case 0x10ec0272: case 0x10ec0660: case 0x10ec0662: case 0x10ec0663: -- cgit v1.2.3 From 4a5a4c56b443a213fa9c2ad27984a8681a3d7087 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Feb 2009 12:46:59 +0100 Subject: ALSA: hda - Add missing COEF initialization for ALC887 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ac1a6e72843..ae5c8a0d147 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1066,6 +1066,7 @@ do_sku: case 0x10ec0882: case 0x10ec0883: case 0x10ec0885: + case 0x10ec0887: case 0x10ec0889: snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 7); -- cgit v1.2.3 From f6f35bbe7c6494e66590cf519e21da2dd8d59e01 Mon Sep 17 00:00:00 2001 From: Roel Kluin Date: Sun, 8 Feb 2009 15:22:25 +0100 Subject: [ARM] AACI: timeout will reach -1 With a postfix decrement the timeout will reach -1 rather than 0, so the warning will not be issued. Signed-off-by: Roel Kluin Signed-off-by: Russell King --- sound/arm/aaci.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 89096e811a4..772901e41ec 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -90,7 +90,7 @@ static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg, */ do { v = readl(aaci->base + AACI_SLFR); - } while ((v & (SLFR_1TXB|SLFR_2TXB)) && timeout--); + } while ((v & (SLFR_1TXB|SLFR_2TXB)) && --timeout); if (!timeout) dev_err(&aaci->dev->dev, @@ -126,7 +126,7 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg) */ do { v = readl(aaci->base + AACI_SLFR); - } while ((v & SLFR_1TXB) && timeout--); + } while ((v & SLFR_1TXB) && --timeout); if (!timeout) { dev_err(&aaci->dev->dev, "timeout on slot 1 TX busy\n"); @@ -147,7 +147,7 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg) do { cond_resched(); v = readl(aaci->base + AACI_SLFR) & (SLFR_1RXV|SLFR_2RXV); - } while ((v != (SLFR_1RXV|SLFR_2RXV)) && timeout--); + } while ((v != (SLFR_1RXV|SLFR_2RXV)) && --timeout); if (!timeout) { dev_err(&aaci->dev->dev, "timeout on RX valid\n"); -- cgit v1.2.3 From 44a678d04babaa751c0ee98e006ede9576fa9e00 Mon Sep 17 00:00:00 2001 From: Mackenzie Morgan Date: Tue, 10 Feb 2009 17:13:43 +0100 Subject: ALSA: hda - Add quirk for Asus z37e (1043:8284) Added a quirk for Asus Z37E for fixing suspend/hibernation problem. Reference: https://bugs.edge.launchpad.net/ubuntu/+source/linux/+bug/25896 http://launchpadlibrarian.net/17053575/0001-Add-quirk-for-ASUS-Z37E-to-make-sound-audible-afte.patch https://bugtrack.alsa-project.org/alsa-bug/bug_view_page.php?bug_id=4282 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ae5c8a0d147..ed8fcbd6000 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8478,6 +8478,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2a66, "HP Acacia", 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), SND_PCI_QUIRK(0x1043, 0x82fe, "Asus P5Q-EM HDMI", ALC1200_ASUS_P5Q), SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601), SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG), -- cgit v1.2.3 From 272edb00493af32c609f43bdf1d75141756fd999 Mon Sep 17 00:00:00 2001 From: "Lopez Cruz, Misael" Date: Tue, 10 Feb 2009 02:05:07 -0600 Subject: ASoC: Update SDP3430 machine driver for snd_soc_card This patch replaces "snd_soc_machine" structure by "snd_soc_card" in SP3430 driver. This change is needed in SDP3430 driver to reflect changes introduced by "ASoC: Rename snd_soc_card to snd_soc_machine" patch (875065491fba8eb13219f16c36e79a6fb4e15c68). Signed-off-by: Misael Lopez Cruz Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/sdp3430.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index ad97836818b..e226fa75669 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -91,7 +91,7 @@ static struct snd_soc_dai_link sdp3430_dai = { }; /* Audio machine driver */ -static struct snd_soc_machine snd_soc_machine_sdp3430 = { +static struct snd_soc_card snd_soc_sdp3430 = { .name = "SDP3430", .platform = &omap_soc_platform, .dai_link = &sdp3430_dai, @@ -100,7 +100,7 @@ static struct snd_soc_machine snd_soc_machine_sdp3430 = { /* Audio subsystem */ static struct snd_soc_device sdp3430_snd_devdata = { - .machine = &snd_soc_machine_sdp3430, + .card = &snd_soc_sdp3430, .codec_dev = &soc_codec_dev_twl4030, }; -- cgit v1.2.3 From a1667e4eea0a7085815d1532d7630bb4611271d0 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Wed, 11 Feb 2009 15:22:28 +0800 Subject: ALSA: hda - allow multi-channel HDMI audio playback when ELD is not present The YAMAHA AV-X1800 requires audio infoframe to include speaker-channel mapping to play >2 channel HDMI audio. In theory that mapping should be derived from its speaker configurations contained in its ELD. However we currently cannot get ELD in console before the KMS functionalities are ready. This is a more or less general issue at least in the near future. As a workaround, we propose to allow playback of mult-channel audio when ELD is not available. Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_intelhdmi.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index 3564f4e4b74..a8643509e2a 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -419,13 +419,17 @@ static int hdmi_setup_channel_allocation(struct hda_codec *codec, /* * CA defaults to 0 for basic stereo audio */ - if (!eld->eld_ver) - return 0; - if (!eld->spk_alloc) - return 0; if (channels <= 2) return 0; + /* + * HDMI sink's ELD info cannot always be retrieved for now, e.g. + * in console or for audio devices. Assume the highest speakers + * configuration, to _not_ prohibit multi-channel audio playback. + */ + if (!eld->spk_alloc) + eld->spk_alloc = 0xffff; + /* * expand ELD's speaker allocation mask * -- cgit v1.2.3 From 606c0cee695bbd0c2bf32132999e35cff5a6dd9e Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Wed, 11 Feb 2009 15:22:29 +0800 Subject: ALSA: hda - enable HDMI audio pin out at module loading time We found that enabling/disabling HDMI audio pin out at stream start/stop time will kill the leading 500ms or so sound samples. Avoid this by enabling pin out once and for ever at module loading time. The leading ~500ms audio samples will still be lost when switching from X-channel playback to Y-channel playback where X != Y. However there's no much we can do about it: the audio infoframe has to change and it looks like either G45 or YAMAHA requires some time to switch the configuration. Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_intelhdmi.c | 42 +++++++++++++++++++---------------------- 1 file changed, 19 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index a8643509e2a..f2610d67e18 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -49,11 +49,6 @@ static struct hda_verb pinout_enable_verb[] = { {} /* terminator */ }; -static struct hda_verb pinout_disable_verb[] = { - {PIN_NID, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00}, - {} -}; - static struct hda_verb unsolicited_response_verb[] = { {PIN_NID, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | INTEL_HDMI_EVENT_TAG}, @@ -248,10 +243,6 @@ static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t nid, static void hdmi_enable_output(struct hda_codec *codec) { - /* Enable Audio InfoFrame Transmission */ - hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0); - snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT, - AC_DIPXMIT_BEST); /* Unmute */ if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP) snd_hda_codec_write(codec, PIN_NID, 0, @@ -260,17 +251,24 @@ static void hdmi_enable_output(struct hda_codec *codec) snd_hda_sequence_write(codec, pinout_enable_verb); } -static void hdmi_disable_output(struct hda_codec *codec) +/* + * Enable Audio InfoFrame Transmission + */ +static void hdmi_start_infoframe_trans(struct hda_codec *codec) { - snd_hda_sequence_write(codec, pinout_disable_verb); - if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP) - snd_hda_codec_write(codec, PIN_NID, 0, - AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0); + snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT, + AC_DIPXMIT_BEST); +} - /* - * FIXME: noises may arise when playing music after reloading the - * kernel module, until the next X restart or monitor repower. - */ +/* + * Disable Audio InfoFrame Transmission + */ +static void hdmi_stop_infoframe_trans(struct hda_codec *codec) +{ + hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0); + snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT, + AC_DIPXMIT_DISABLE); } static int hdmi_get_channel_count(struct hda_codec *codec) @@ -489,6 +487,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hdmi_setup_channel_mapping(codec, &ai); hdmi_fill_audio_infoframe(codec, &ai); + hdmi_start_infoframe_trans(codec); } @@ -566,7 +565,7 @@ static int intel_hdmi_playback_pcm_close(struct hda_pcm_stream *hinfo, { struct intel_hdmi_spec *spec = codec->spec; - hdmi_disable_output(codec); + hdmi_stop_infoframe_trans(codec); return snd_hda_multi_out_dig_close(codec, &spec->multiout); } @@ -586,8 +585,6 @@ static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, hdmi_setup_audio_infoframe(codec, substream); - hdmi_enable_output(codec); - return 0; } @@ -632,8 +629,7 @@ static int intel_hdmi_build_controls(struct hda_codec *codec) static int intel_hdmi_init(struct hda_codec *codec) { - /* disable audio output as early as possible */ - hdmi_disable_output(codec); + hdmi_enable_output(codec); snd_hda_sequence_write(codec, unsolicited_response_verb); -- cgit v1.2.3 From 9a957a24e3b4008d84e204cdf25849ae4d5592a2 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Wed, 11 Feb 2009 15:22:30 +0800 Subject: ALSA: hda - compute checksum in HDMI audio infoframe Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_intelhdmi.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index f2610d67e18..90b11374a0a 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -366,11 +366,16 @@ static void hdmi_fill_audio_infoframe(struct hda_codec *codec, struct hdmi_audio_infoframe *ai) { u8 *params = (u8 *)ai; + u8 sum = 0; int i; hdmi_debug_dip_size(codec); hdmi_clear_dip_buffers(codec); /* be paranoid */ + for (i = 0; i < sizeof(ai); i++) + sum += params[i]; + ai->checksum = - sum; + hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0); for (i = 0; i < sizeof(ai); i++) hdmi_write_dip_byte(codec, PIN_NID, params[i]); -- cgit v1.2.3 From a57c0eb65576c810c408f0a086afac179242a21c Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Wed, 11 Feb 2009 15:22:31 +0800 Subject: ALSA: hda - add id for Intel IbexPeak integrated HDMI codec Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_intelhdmi.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index 90b11374a0a..fcc77fec448 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -684,6 +684,7 @@ static struct hda_codec_preset snd_hda_preset_intelhdmi[] = { { .id = 0x80862801, .name = "G45 DEVBLC", .patch = patch_intel_hdmi }, { .id = 0x80862802, .name = "G45 DEVCTG", .patch = patch_intel_hdmi }, { .id = 0x80862803, .name = "G45 DEVELK", .patch = patch_intel_hdmi }, + { .id = 0x80862804, .name = "G45 DEVIBX", .patch = patch_intel_hdmi }, { .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_intel_hdmi }, {} /* terminator */ }; @@ -692,6 +693,7 @@ MODULE_ALIAS("snd-hda-codec-id:808629fb"); MODULE_ALIAS("snd-hda-codec-id:80862801"); MODULE_ALIAS("snd-hda-codec-id:80862802"); MODULE_ALIAS("snd-hda-codec-id:80862803"); +MODULE_ALIAS("snd-hda-codec-id:80862804"); MODULE_ALIAS("snd-hda-codec-id:10951392"); MODULE_LICENSE("GPL"); -- cgit v1.2.3 From 32cf9a16f4af01573ddec1eb073111fc20a9d7d4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Feb 2009 00:06:42 +0100 Subject: ALSA: mtpav - Fix initial value for input hwport Fix the initial value for input hwport. The old value (-1) may cause Oops when an realtime MIDI byte is received before the input port is explicitly given. Instead, now it's set to the broadcasting as default. Tested-by: Holger Dehnhardt Cc: Signed-off-by: Takashi Iwai --- sound/drivers/mtpav.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c index 5b89c0883d6..48b64e6b267 100644 --- a/sound/drivers/mtpav.c +++ b/sound/drivers/mtpav.c @@ -706,7 +706,6 @@ static int __devinit snd_mtpav_probe(struct platform_device *dev) mtp_card->card = card; mtp_card->irq = -1; mtp_card->share_irq = 0; - mtp_card->inmidiport = 0xffffffff; mtp_card->inmidistate = 0; mtp_card->outmidihwport = 0xffffffff; init_timer(&mtp_card->timer); @@ -719,6 +718,8 @@ static int __devinit snd_mtpav_probe(struct platform_device *dev) if (err < 0) goto __error; + mtp_card->inmidiport = mtp_card->num_ports + MTPAV_PIDX_BROADCAST; + err = snd_mtpav_get_ISA(mtp_card); if (err < 0) goto __error; -- cgit v1.2.3 From 26a74f1f61c5bba1c0b46e67e91e921e941f76d7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Feb 2009 00:13:19 +0100 Subject: ALSA: hda - Register (new) devices at reconfig The devices that have been newly added during reconfig must be registered. Otherwise they won't be visible to user-space. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_hwdep.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 300ab407cf4..482fb0304ca 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -175,7 +175,7 @@ static int reconfig_codec(struct hda_codec *codec) err = snd_hda_codec_build_controls(codec); if (err < 0) return err; - return 0; + return snd_card_register(codec->bus->card); } /* -- cgit v1.2.3 From 92258a3ed2f583c8720ef570f5c62b28e6c58d71 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Thu, 12 Feb 2009 17:27:27 -0200 Subject: ALSA: hda - Change HP dv7 (103c:30f4) quirk from hp-m4 to hp-dv5 model Change HP dv7 quirk: although reported to work with hp-m4 model (https://bugzilla.novell.com/show_bug.cgi?id=445321), the original report doesn't contain info about testing of internal microphone. Recently I received a report about internal mic not working (https://qa.mandriva.com/show_bug.cgi?id=44855#c193), this must be related with the forced line in on pin 0x0e done with hp-m4 model. Thus change the current quirk from STAC_HP_M4 to STAC_HP_DV5, later reported to be fixed on a provided kernel with this change (https://qa.mandriva.com/show_bug.cgi?id=44855#c196). Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 38428e22428..aa814a3c2d8 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1799,7 +1799,7 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { 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_M4), + "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, 0x30fc, -- cgit v1.2.3 From 3a08e30de2facffe8e1a25bf4fa62cbc920fbaf6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Feb 2009 11:37:08 +0100 Subject: ALSA: hda - Add missing terminator in slave dig-out array Added the missing terminator for ad1989b_slave_dig_outs[]. Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 2e7371ec2e2..7006d62ca6c 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1885,8 +1885,8 @@ static hda_nid_t ad1988_capsrc_nids[3] = { #define AD1988_SPDIF_OUT_HDMI 0x0b #define AD1988_SPDIF_IN 0x07 -static hda_nid_t ad1989b_slave_dig_outs[2] = { - AD1988_SPDIF_OUT, AD1988_SPDIF_OUT_HDMI +static hda_nid_t ad1989b_slave_dig_outs[] = { + AD1988_SPDIF_OUT, AD1988_SPDIF_OUT_HDMI, 0 }; static struct hda_input_mux ad1988_6stack_capture_source = { -- cgit v1.2.3 From 9411e21cd0cc4fd046b4f448417b0e103e80951c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Feb 2009 11:32:28 +0100 Subject: ALSA: hda - Add snd_hda_multi_out_dig_cleanup() Added the helper function snd_hda_multi_out_dig_cleanup() to clean up the digital outputs with multi setup. This call is needed in cases the codec supports multiple digital outputs as slaves. Otherwise the slave widgets aren't properly cleaned up. For a single digital output (e.g. in patch_conexant.c), this call isn't needed. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 10 ++++++++++ sound/pci/hda/hda_local.h | 2 ++ sound/pci/hda/patch_analog.c | 11 ++++++++++- sound/pci/hda/patch_sigmatel.c | 11 ++++++++++- 4 files changed, 32 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0b708134d12..d03f99298be 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3088,6 +3088,16 @@ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_prepare); +int snd_hda_multi_out_dig_cleanup(struct hda_codec *codec, + struct hda_multi_out *mout) +{ + mutex_lock(&codec->spdif_mutex); + cleanup_dig_out_stream(codec, mout->dig_out_nid); + mutex_unlock(&codec->spdif_mutex); + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_cleanup); + /* * release the digital out */ diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 1dd8716c387..44f189cb97a 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -251,6 +251,8 @@ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, unsigned int stream_tag, unsigned int format, struct snd_pcm_substream *substream); +int snd_hda_multi_out_dig_cleanup(struct hda_codec *codec, + struct hda_multi_out *mout); int snd_hda_multi_out_analog_open(struct hda_codec *codec, struct hda_multi_out *mout, struct snd_pcm_substream *substream, diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 7006d62ca6c..e48612323aa 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -275,6 +275,14 @@ static int ad198x_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, format, substream); } +static int ad198x_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ad198x_spec *spec = codec->spec; + return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout); +} + /* * Analog capture */ @@ -333,7 +341,8 @@ static struct hda_pcm_stream ad198x_pcm_digital_playback = { .ops = { .open = ad198x_dig_playback_pcm_open, .close = ad198x_dig_playback_pcm_close, - .prepare = ad198x_dig_playback_pcm_prepare + .prepare = ad198x_dig_playback_pcm_prepare, + .cleanup = ad198x_dig_playback_pcm_cleanup }, }; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index aa814a3c2d8..8027edf3c8f 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2442,6 +2442,14 @@ static int stac92xx_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, stream_tag, format, substream); } +static int stac92xx_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct sigmatel_spec *spec = codec->spec; + return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout); +} + /* * Analog capture callbacks @@ -2486,7 +2494,8 @@ static struct hda_pcm_stream stac92xx_pcm_digital_playback = { .ops = { .open = stac92xx_dig_playback_pcm_open, .close = stac92xx_dig_playback_pcm_close, - .prepare = stac92xx_dig_playback_pcm_prepare + .prepare = stac92xx_dig_playback_pcm_prepare, + .cleanup = stac92xx_dig_playback_pcm_cleanup }, }; -- cgit v1.2.3 From 14fa43f53ff3a9c3d8b9662574b7369812a31a97 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 12 Feb 2009 19:33:19 +0000 Subject: ASoC: Only register AC97 bus if it's not done already ASoC supports both explicit codec drivers for AC97 devices and a simple driver which uses the standard ALSA AC97 framework for codec support. When used with the generic AC97 codec support that will provide the ad hoc AC97 device for drivers like touchscreens to attach to so the core shouldn't do so. Reported-by: Manuel Lauss Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 55fdb4abb17..ec3f8bb4b51 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1385,7 +1385,10 @@ int snd_soc_init_card(struct snd_soc_device *socdev) mutex_lock(&codec->mutex); #ifdef CONFIG_SND_SOC_AC97_BUS - if (ac97) { + /* Only instantiate AC97 if not already done by the adaptor + * for the generic AC97 subsystem. + */ + if (ac97 && strcmp(codec->name, "AC97") != 0) { ret = soc_ac97_dev_register(codec); if (ret < 0) { printk(KERN_ERR "asoc: AC97 device register failed\n"); -- cgit v1.2.3 From d14a7e0bfc4aed6452a436c9836903fbd1a5d6ad Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 Feb 2009 10:13:03 +0100 Subject: Revert "Sound: hda - Restore PCI configuration space with interrupts off" This reverts commit 32e176c14d7a425b681ef003c9061001ddb7fc7b. That commit caused a regression with suspend on Thinkpad SL300. Reference: kernel bug#12711 http://bugzilla.kernel.org/show_bug.cgi?id=12711 Tested-by: Alexandre Rostovtsev Acked-by: Rafael J. Wysocki Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 11e791b965f..c8d9178f47e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1947,16 +1947,13 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state) return 0; } -static int azx_resume_early(struct pci_dev *pci) -{ - return pci_restore_state(pci); -} - static int azx_resume(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); struct azx *chip = card->private_data; + pci_set_power_state(pci, PCI_D0); + pci_restore_state(pci); if (pci_enable_device(pci) < 0) { printk(KERN_ERR "hda-intel: pci_enable_device failed, " "disabling device\n"); @@ -2468,7 +2465,6 @@ static struct pci_driver driver = { .remove = __devexit_p(azx_remove), #ifdef CONFIG_PM .suspend = azx_suspend, - .resume_early = azx_resume_early, .resume = azx_resume, #endif }; -- cgit v1.2.3 From e156ac4c571e3be741bc411e58820b74a9295c72 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 16 Feb 2009 15:22:39 +0100 Subject: sound: usb-audio: fix uninitialized variable with M-Audio MIDI interfaces Fix the snd_usbmidi_create_endpoints_midiman() function, which forgot to set the out_interval member of the endpoint info structure for Midiman/ M-Audio devices. Since kernel 2.6.24, any non-zero value makes the driver use interrupt transfers instead of bulk transfers. With EHCI controllers, these random interval values result in unbearably large latencies for output MIDI transfers. Signed-off-by: Clemens Ladisch Reported-by: David Tested-by: David Cc: Signed-off-by: Takashi Iwai --- sound/usb/usbmidi.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 320641ab5be..26bad373fe6 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -1625,6 +1625,7 @@ static int snd_usbmidi_create_endpoints_midiman(struct snd_usb_midi* umidi, } ep_info.out_ep = get_endpoint(hostif, 2)->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK; + ep_info.out_interval = 0; ep_info.out_cables = endpoint->out_cables & 0x5555; err = snd_usbmidi_out_endpoint_create(umidi, &ep_info, &umidi->endpoints[0]); if (err < 0) -- cgit v1.2.3 From 0412558c873f716efe902b397af0653a550f7341 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 Feb 2009 22:48:12 +0100 Subject: ALSA: usb-audio - Fix non-continuous rate detection The detection of non-continuous rates (given via rate tables) isn't processed properly (e.g. for type II). This patch fixes and simplifies the detection code. Tested-by: Joris van Rantwijk Cc: Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 2ab83129d9b..80863093d2c 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2524,7 +2524,6 @@ static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioform * build the rate table and bitmap flags */ int r, idx; - unsigned int nonzero_rates = 0; fp->rate_table = kmalloc(sizeof(int) * nr_rates, GFP_KERNEL); if (fp->rate_table == NULL) { @@ -2532,24 +2531,26 @@ static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioform return -1; } - fp->nr_rates = nr_rates; - fp->rate_min = fp->rate_max = combine_triple(&fmt[8]); + fp->nr_rates = 0; + fp->rate_min = fp->rate_max = 0; for (r = 0, idx = offset + 1; r < nr_rates; r++, idx += 3) { unsigned int rate = combine_triple(&fmt[idx]); + if (!rate) + continue; /* C-Media CM6501 mislabels its 96 kHz altsetting */ if (rate == 48000 && nr_rates == 1 && chip->usb_id == USB_ID(0x0d8c, 0x0201) && fp->altsetting == 5 && fp->maxpacksize == 392) rate = 96000; - fp->rate_table[r] = rate; - nonzero_rates |= rate; - if (rate < fp->rate_min) + fp->rate_table[fp->nr_rates] = rate; + if (!fp->rate_min || rate < fp->rate_min) fp->rate_min = rate; - else if (rate > fp->rate_max) + if (!fp->rate_max || rate > fp->rate_max) fp->rate_max = rate; fp->rates |= snd_pcm_rate_to_rate_bit(rate); + fp->nr_rates++; } - if (!nonzero_rates) { + if (!fp->nr_rates) { hwc_debug("All rates were zero. Skipping format!\n"); return -1; } -- cgit v1.2.3 From 3b03cc5b86e2052295b9b484f37226ee15c87924 Mon Sep 17 00:00:00 2001 From: Joris van Rantwijk Date: Mon, 16 Feb 2009 22:58:23 +0100 Subject: ALSA: usb-audio - Workaround for misdetected sample rate with CM6207 The CM6207 incorrectly advertises its 96 kHz playback setting as 48 kHz in its USB device descriptor. This patch extends an existing workaround in usbaudio.c to also cover the CM6207. This resolves issue 0004249 in the ALSA bug tracker. Signed-off-by: Joris van Rantwijk Cc: Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 80863093d2c..19e37451c21 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2539,7 +2539,8 @@ static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioform continue; /* C-Media CM6501 mislabels its 96 kHz altsetting */ if (rate == 48000 && nr_rates == 1 && - chip->usb_id == USB_ID(0x0d8c, 0x0201) && + (chip->usb_id == USB_ID(0x0d8c, 0x0201) || + chip->usb_id == USB_ID(0x0d8c, 0x0102)) && fp->altsetting == 5 && fp->maxpacksize == 392) rate = 96000; fp->rate_table[fp->nr_rates] = rate; -- cgit v1.2.3 From 2678f60d2bc05a12580b93eb36f089f0e55693e0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 18 Feb 2009 16:46:27 +0100 Subject: ALSA: jack - Use card->shortname for input name Currently the jack layer refers to card->longname as a part of its input device name string. However, longname is often really long and way too ugly as an identifier, such as, "HDA Intel at 0xf8400000 irq 21". This patch changes the code to use card->shortname instead. The shortname string contains usually the h/w vendor and product names but without messy I/O port or IRQ numbers. Signed-off-by: Takashi Iwai --- sound/core/jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/jack.c b/sound/core/jack.c index dd4a12dc09a..077a85262c1 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -47,7 +47,7 @@ static int snd_jack_dev_register(struct snd_device *device) int err; snprintf(jack->name, sizeof(jack->name), "%s %s", - card->longname, jack->id); + card->shortname, jack->id); jack->input_dev->name = jack->name; /* Default to the sound card device. */ -- cgit v1.2.3 From 6ce6c473a7fd742fdb0db95841e2c4c6b37337c5 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 17 Feb 2009 09:50:30 +0100 Subject: sound: virtuoso: revert "do not overwrite EEPROM on Xonar D2/D2X" This reverts commit 7e86c0e6850504ec9516b953f316a47277825e33 ("do not overwrite EEPROM on Xonar D2/D2X") because it did not actually help with the problem. More user reports show that the overwriting of the EEPROM is not triggered by using this driver but by installing Linux, and that the installation of any other operating system (even one without any CMI8788 driver) has the same effect. In other words, the presence of this driver does not have any effect on the occurrence of the error. (So far, the available evidence seems to point to a BIOS bug.) Furthermore, it turns out that the EEPROM chip is protected against stray write commands by the command format and by requiring a separate write-enable command, so the error scenario in the previous commit (that SPI writes can be misinterpreted as an EEPROM write command) is not even theoretically possible. The mixer control that was removed as a consequence of the previous commit can only be partially emulated in userspace, which also means it cannot be seen be the in-kernel OSS API emulation, so it is better to revert that change. Signed-off-by: Clemens Ladisch Cc: Signed-off-by: Takashi Iwai --- sound/pci/oxygen/virtuoso.c | 17 ++++------------- 1 file changed, 4 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 18c7c91786b..6c870c12a17 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -26,7 +26,7 @@ * SPI 0 -> 1st PCM1796 (front) * SPI 1 -> 2nd PCM1796 (surround) * SPI 2 -> 3rd PCM1796 (center/LFE) - * SPI 4 -> 4th PCM1796 (back) and EEPROM self-destruct (do not use!) + * SPI 4 -> 4th PCM1796 (back) * * GPIO 2 -> M0 of CS5381 * GPIO 3 -> M1 of CS5381 @@ -207,12 +207,6 @@ static void xonar_gpio_changed(struct oxygen *chip); static inline void pcm1796_write_spi(struct oxygen *chip, unsigned int codec, u8 reg, u8 value) { - /* - * We don't want to do writes on SPI 4 because the EEPROM, which shares - * the same pin, might get confused and broken. We'd better take care - * that the driver works with the default register values ... - */ -#if 0 /* maps ALSA channel pair number to SPI output */ static const u8 codec_map[4] = { 0, 1, 2, 4 @@ -223,7 +217,6 @@ static inline void pcm1796_write_spi(struct oxygen *chip, unsigned int codec, (codec_map[codec] << OXYGEN_SPI_CODEC_SHIFT) | OXYGEN_SPI_CEN_LATCH_CLOCK_HI, (reg << 8) | value); -#endif } static inline void pcm1796_write_i2c(struct oxygen *chip, unsigned int codec, @@ -757,9 +750,6 @@ static const DECLARE_TLV_DB_SCALE(cs4362a_db_scale, -12700, 100, 0); static int xonar_d2_control_filter(struct snd_kcontrol_new *template) { - if (!strncmp(template->name, "Master Playback ", 16)) - /* disable volume/mute because they would require SPI writes */ - return 1; if (!strncmp(template->name, "CD Capture ", 11)) /* CD in is actually connected to the video in pin */ template->private_value ^= AC97_CD ^ AC97_VIDEO; @@ -850,8 +840,9 @@ static const struct oxygen_model model_xonar_d2 = { .dac_volume_min = 0x0f, .dac_volume_max = 0xff, .misc_flags = OXYGEN_MISC_MIDI, - .function_flags = OXYGEN_FUNCTION_SPI, - .dac_i2s_format = OXYGEN_I2S_FORMAT_I2S, + .function_flags = OXYGEN_FUNCTION_SPI | + OXYGEN_FUNCTION_ENABLE_SPI_4_5, + .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; -- cgit v1.2.3 From e32740d9786b8a6c54f6e3d670567d9ef57b3b8c Mon Sep 17 00:00:00 2001 From: Harvey Harrison Date: Thu, 19 Feb 2009 11:58:37 -0800 Subject: ALSA: pcxhr.h replace signed one-bit bitfields The usage and comments make it clear values of 1/0 were intended rather than -1/0 Noticed by sparse: sound/pci/pcxhr/pcxhr.h:100:20: error: dubious one-bit signed bitfield sound/pci/pcxhr/pcxhr.h:101:22: error: dubious one-bit signed bitfield sound/pci/pcxhr/pcxhr.h:102:24: error: dubious one-bit signed bitfield sound/pci/pcxhr/pcxhr.h:103:21: error: dubious one-bit signed bitfield sound/pci/pcxhr/pcxhr.h:104:25: error: dubious one-bit signed bitfield sound/pci/pcxhr/pcxhr.h:105:20: error: dubious one-bit signed bitfield Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai --- sound/pci/pcxhr/pcxhr.h | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/pcxhr/pcxhr.h b/sound/pci/pcxhr/pcxhr.h index 84131a916c9..69d87dee699 100644 --- a/sound/pci/pcxhr/pcxhr.h +++ b/sound/pci/pcxhr/pcxhr.h @@ -97,12 +97,12 @@ struct pcxhr_mgr { int capture_chips; int fw_file_set; int firmware_num; - int is_hr_stereo:1; - int board_has_aes1:1; /* if 1 board has AES1 plug and SRC */ - int board_has_analog:1; /* if 0 the board is digital only */ - int board_has_mic:1; /* if 1 the board has microphone input */ - int board_aes_in_192k:1;/* if 1 the aes input plugs do support 192kHz */ - int mono_capture:1; /* if 1 the board does mono capture */ + unsigned int is_hr_stereo:1; + unsigned int board_has_aes1:1; /* if 1 board has AES1 plug and SRC */ + unsigned int board_has_analog:1; /* if 0 the board is digital only */ + unsigned int board_has_mic:1; /* if 1 the board has microphone input */ + unsigned int board_aes_in_192k:1;/* if 1 the aes input plugs do support 192kHz */ + unsigned int mono_capture:1; /* if 1 the board does mono capture */ struct snd_dma_buffer hostport; -- cgit v1.2.3 From 55290e1932102f57ea17e7cff895914c2dbdb4c4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Feb 2009 15:59:01 +0100 Subject: ALSA: hda - Fix parse of init_verbs sysfs entry Fixed the parse of init_verbs hwdep sysfs entry. Simplieied using sscanf. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_hwdep.c | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 482fb0304ca..4ae51dcb81a 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -277,18 +277,19 @@ static ssize_t init_verbs_store(struct device *dev, { struct snd_hwdep *hwdep = dev_get_drvdata(dev); struct hda_codec *codec = hwdep->private_data; - char *p; - struct hda_verb verb, *v; + struct hda_verb *v; + int nid, verb, param; - verb.nid = simple_strtoul(buf, &p, 0); - verb.verb = simple_strtoul(p, &p, 0); - verb.param = simple_strtoul(p, &p, 0); - if (!verb.nid || !verb.verb || !verb.param) + if (sscanf(buf, "%i %i %i", &nid, &verb, ¶m) != 3) + return -EINVAL; + if (!nid || !verb) return -EINVAL; v = snd_array_new(&codec->init_verbs); if (!v) return -ENOMEM; - *v = verb; + v->nid = nid; + v->verb = verb; + v->param = param; return count; } -- cgit v1.2.3 From 3d92e8f3ae9ba21cac30370eb254ed9dc20df043 Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Sun, 22 Feb 2009 09:38:47 +0100 Subject: m68k: atari - Rename "mfp" to "st_mfp" http://kisskb.ellerman.id.au/kisskb/buildresult/72115/: | net/mac80211/ieee80211_i.h:327: error: syntax error before 'volatile' | net/mac80211/ieee80211_i.h:350: error: syntax error before '}' token | net/mac80211/ieee80211_i.h:455: error: field 'sta' has incomplete type | distcc[19430] ERROR: compile net/mac80211/main.c on sprygo/32 failed This is caused by | # define mfp ((*(volatile struct MFP*)MFP_BAS)) in arch/m68k/include/asm/atarihw.h, which conflicts with the new "mfp" enum in net/mac80211/ieee80211_i.h. Rename "mfp" to "st_mfp", as it's a way too generic name for a global #define. Signed-off-by: Geert Uytterhoeven Signed-off-by: Linus Torvalds --- sound/oss/dmasound/dmasound_atari.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c index 57d9f154c88..38931f2f696 100644 --- a/sound/oss/dmasound/dmasound_atari.c +++ b/sound/oss/dmasound/dmasound_atari.c @@ -847,23 +847,23 @@ static int __init AtaIrqInit(void) of events. So all we need to keep the music playing is to provide the sound hardware with new data upon an interrupt from timer A. */ - mfp.tim_ct_a = 0; /* ++roman: Stop timer before programming! */ - mfp.tim_dt_a = 1; /* Cause interrupt after first event. */ - mfp.tim_ct_a = 8; /* Turn on event counting. */ + st_mfp.tim_ct_a = 0; /* ++roman: Stop timer before programming! */ + st_mfp.tim_dt_a = 1; /* Cause interrupt after first event. */ + st_mfp.tim_ct_a = 8; /* Turn on event counting. */ /* Register interrupt handler. */ if (request_irq(IRQ_MFP_TIMA, AtaInterrupt, IRQ_TYPE_SLOW, "DMA sound", AtaInterrupt)) return 0; - mfp.int_en_a |= 0x20; /* Turn interrupt on. */ - mfp.int_mk_a |= 0x20; + st_mfp.int_en_a |= 0x20; /* Turn interrupt on. */ + st_mfp.int_mk_a |= 0x20; return 1; } #ifdef MODULE static void AtaIrqCleanUp(void) { - mfp.tim_ct_a = 0; /* stop timer */ - mfp.int_en_a &= ~0x20; /* turn interrupt off */ + st_mfp.tim_ct_a = 0; /* stop timer */ + st_mfp.int_en_a &= ~0x20; /* turn interrupt off */ free_irq(IRQ_MFP_TIMA, AtaInterrupt); } #endif /* MODULE */ @@ -1599,7 +1599,7 @@ static int __init dmasound_atari_init(void) is_falcon = 0; } else return -ENODEV; - if ((mfp.int_en_a & mfp.int_mk_a & 0x20) == 0) + if ((st_mfp.int_en_a & st_mfp.int_mk_a & 0x20) == 0) return dmasound_init(); else { printk("DMA sound driver: Timer A interrupt already in use\n"); -- cgit v1.2.3 From e8bf069c419c1dc0657e02636441fe1179a9db14 Mon Sep 17 00:00:00 2001 From: Anssi Hannula Date: Sun, 22 Feb 2009 14:42:54 +0200 Subject: ALSA: aw2: do not grab every saa7146 based device Audiowerk2 driver snd-aw2 is bound to any saa7146 device as it does not check subsystem ids. Many DVB devices are saa7146 based, so aw2 driver grabs them as well. According to http://lkml.org/lkml/2008/10/15/311 aw2 devices have the subsystem ids set to 0, the saa7146 default. Fix conflicts with DVB devices by checking for subsystem ids = 0 specifically. Signed-off-by: Anssi Hannula Signed-off-by: Takashi Iwai --- sound/pci/aw2/aw2-alsa.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c index 3f00ddf450f..c7c54e7748e 100644 --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -165,7 +165,7 @@ module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Audiowerk2 soundcard."); static struct pci_device_id snd_aw2_ids[] = { - {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, PCI_ANY_ID, PCI_ANY_ID, + {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, 0, 0, 0, 0, 0}, {0} }; -- cgit v1.2.3 From 5370d96f85962769ea3df3a81cc885f257c51589 Mon Sep 17 00:00:00 2001 From: Steve Chen Date: Sat, 21 Feb 2009 08:05:04 -0600 Subject: ALSA: fix excessive background noise introduced by OSS emulation rate shrink Incorrect variable was used to get the next sample which caused S2 to be stuck with the same value resulting in loud background noise. Signed-off-by: Steve Chen Cc: Signed-off-by: Takashi Iwai --- sound/core/oss/rate.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c index a466443c4a2..2fa9299a440 100644 --- a/sound/core/oss/rate.c +++ b/sound/core/oss/rate.c @@ -157,7 +157,7 @@ static void resample_shrink(struct snd_pcm_plugin *plugin, while (dst_frames1 > 0) { S1 = S2; if (src_frames1-- > 0) { - S1 = *src; + S2 = *src; src += src_step; } if (pos & ~R_MASK) { -- cgit v1.2.3 From 2d4663816064fabb68935f920bbd7ccdc7f9392d Mon Sep 17 00:00:00 2001 From: Luke Yelavich Date: Mon, 23 Feb 2009 13:00:33 +1100 Subject: ALSA: hda - add another MacBook Pro 3,1 SSID Reference: Ubuntu bug #33245 https://bugs.launchpad.net/bugs/332456 Signed-off-by: Luke Yelavich Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ed8fcbd6000..f6571224b34 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7017,6 +7017,7 @@ static int patch_alc882(struct hda_codec *codec) case 0x106b3e00: /* iMac 24 Aluminium */ board_config = ALC885_IMAC24; break; + case 0x106b00a0: /* MacBookPro3,1 - Another revision */ case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */ case 0x106b00a4: /* MacbookPro4,1 */ case 0x106b2c00: /* Macbook Pro rev3 */ -- cgit v1.2.3 From cc374c477c9bf95f409fed16426856d86a97394f Mon Sep 17 00:00:00 2001 From: Juan Jesus Garcia de Soria Date: Mon, 23 Feb 2009 08:11:59 +0100 Subject: ALSA: hda - Quirk for Acer Aspire 6530G The Acer Aspire 6530G needs the 4930G "model" for the front mic to work properly. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f6571224b34..a680be0d453 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8470,6 +8470,8 @@ 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, 0x0166, "Acer Aspire 6530G", + ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), /* default Acer */ SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL), SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG), -- cgit v1.2.3 From 1f9da5544073d38e05139f8ce9da24e78653c73e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 24 Feb 2009 15:31:02 +0100 Subject: ALSA: emu10k1 - Fix digital/analog switch on audigy2 ZS Fix the inverted logic of shared spdif switch. Reference: Novell bnc#478496 https://bugzilla.novell.com/show_bug.cgi?id=478496 Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_main.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 7958006a1d6..101a1c13a20 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1528,6 +1528,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .ca0151_chip = 1, .spk71 = 1, .spdif_bug = 1, + .invert_shared_spdif = 1, /* digital/analog switch swapped */ .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10021102, .driver = "Audigy2", .name = "SB Audigy 2 Platinum [SB0240P]", -- cgit v1.2.3 From ea18aa464452c3e6550320d247c0306aaa2d156f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Feb 2009 17:36:33 +0100 Subject: ALSA: hda - Fix digital mic on dell-m4-1 and dell-m4-3 Fix num_dmuxes initialization for dell-m4-1 and dell-m4-3 models of IDT 92HD71bxx codec, which was wrongly set to zero. Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 8027edf3c8f..3bc427645da 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4989,7 +4989,7 @@ again: case STAC_DELL_M4_3: spec->num_dmics = 1; spec->num_smuxes = 0; - spec->num_dmuxes = 0; + spec->num_dmuxes = 1; break; default: spec->num_dmics = STAC92HD71BXX_NUM_DMICS; -- cgit v1.2.3 From bb543c969467f33c3a1a0ccfcfcd9a508cd81c54 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Feb 2009 17:44:07 +0100 Subject: ALSA: hda - Add quirk for new HP xw series Added model=hp-bpc for new HP xw series (103c:170b). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a680be0d453..6c26afcb826 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10557,6 +10557,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { 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(0x103c, 0x170b, "HP xw*", 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 38f1df27e3191d76e983cb9c6b4392582fd32fda Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 1 Mar 2009 10:55:44 +0100 Subject: ALSA: hda - Add probe_mask default for Toshiba laptop with ALC268 Some Toshiba laptops have another ALC268 codec on slot#3 that conflicts with the primary codec. The codec#3 is for the digital I/O, and should be fixed by the driver, but it'd need a bunch of changes. So, let's fix the probe problem temporarily by setting the default probe_mask value. Reference: kernel bugzilla #12735 http://bugzilla.kernel.org/show_bug.cgi?id=12735 Tested-by: Alexey Dobriyan Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c8d9178f47e..5e909e0da04 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2095,6 +2095,8 @@ 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), + /* conflict of ALC268 in slot#3 (digital I/O); a temporary fix */ + SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba laptop", 0x03), {} }; -- cgit v1.2.3 From 14b97595e0e1f47b6f809e180e5bcd8dcd995690 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Mar 2009 09:42:07 +0100 Subject: ALSA: hda - Fix typos in slave controls in patch_sigmatel.c "Headphone Playback ..." appears twice in slave_vols[] and slave_sws[]. They should be "Headphone Playback2 ..." Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3bc427645da..995b413078f 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1207,7 +1207,7 @@ static const char *slave_vols[] = { "LFE Playback Volume", "Side Playback Volume", "Headphone Playback Volume", - "Headphone Playback Volume", + "Headphone2 Playback Volume", "Speaker Playback Volume", "External Speaker Playback Volume", "Speaker2 Playback Volume", @@ -1221,7 +1221,7 @@ static const char *slave_sws[] = { "LFE Playback Switch", "Side Playback Switch", "Headphone Playback Switch", - "Headphone Playback Switch", + "Headphone2 Playback Switch", "Speaker Playback Switch", "External Speaker Playback Switch", "Speaker2 Playback Switch", -- cgit v1.2.3 From c50ff7c04225c945b13d410d50fde6ff6c59d7ee Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Mar 2009 09:43:58 +0100 Subject: ALSA: hda - Fix headphone-detect regression with multiple HP jacks The recent changes over the DAC detection mechanism in patch_sigmatel.c breaks the HP detection on the machines with multiple HP jacks. It's basically because of the workaround to support the multi-channel output. Since the HP detection is more important feature, disable the HP-swap workaroud temporarily. Reference: Novell bnc#482052 https://bugzilla.novell.com/show_bug.cgi?id=482052 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 995b413078f..6094344fb22 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3516,6 +3516,7 @@ 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. */ @@ -3535,6 +3536,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out spec->autocfg.line_out_type = AUTO_PIN_HP_OUT; spec->autocfg.hp_outs = 0; } +#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); -- cgit v1.2.3 From dde332b660cf0bc2baaba678b52768a0fb6e6da2 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Mon, 16 Mar 2009 21:32:25 +0100 Subject: ALSA: opl3sa2 - Fix NULL dereference when suspending snd_opl3sa2 Fix the OOPS during a opl3sa2 card suspend and resume if the driver is loaded but the card is not found. Signed-off-by: Krzysztof Helt Cc: Signed-off-by: Takashi Iwai --- sound/isa/opl3sa2.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index 58c972b2af0..b848d100186 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -550,21 +550,27 @@ static int __devinit snd_opl3sa2_mixer(struct snd_card *card) #ifdef CONFIG_PM static int snd_opl3sa2_suspend(struct snd_card *card, pm_message_t state) { - struct snd_opl3sa2 *chip = card->private_data; + if (card) { + struct snd_opl3sa2 *chip = card->private_data; - snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - chip->wss->suspend(chip->wss); - /* power down */ - snd_opl3sa2_write(chip, OPL3SA2_PM_CTRL, OPL3SA2_PM_D3); + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + chip->wss->suspend(chip->wss); + /* power down */ + snd_opl3sa2_write(chip, OPL3SA2_PM_CTRL, OPL3SA2_PM_D3); + } return 0; } static int snd_opl3sa2_resume(struct snd_card *card) { - struct snd_opl3sa2 *chip = card->private_data; + struct snd_opl3sa2 *chip; int i; + if (!card) + return 0; + + chip = card->private_data; /* power up */ snd_opl3sa2_write(chip, OPL3SA2_PM_CTRL, OPL3SA2_PM_D0); -- cgit v1.2.3 From 09240cf429505891d6123ce14a29f58f2a60121e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Mar 2009 07:47:18 +0100 Subject: ALSA: hda - Fix DMA mask for ATI controllers ATI controllers (at least some SB0600 models) appear buggy to handle 64bit DMA. As a workaround, reset GCAP bit0 and let the driver to use only 32bit DMA on these controllers. Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 5e909e0da04..643f0e49929 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2210,9 +2210,17 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, gcap = azx_readw(chip, GCAP); snd_printdd("chipset global capabilities = 0x%x\n", gcap); + /* ATI chips seems buggy about 64bit DMA addresses */ + if (chip->driver_type == AZX_DRIVER_ATI) + gcap &= ~0x01; + /* allow 64bit DMA address if supported by H/W */ if ((gcap & 0x01) && !pci_set_dma_mask(pci, DMA_64BIT_MASK)) pci_set_consistent_dma_mask(pci, DMA_64BIT_MASK); + else { + pci_set_dma_mask(pci, DMA_32BIT_MASK); + pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK); + } /* read number of streams from GCAP register instead of using * hardcoded value -- cgit v1.2.3 From c673ba1c23941173c16ff24c7cb34199e826c8b5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Mar 2009 07:49:14 +0100 Subject: ALSA: hda - Workaround for buggy DMA position on ATI controllers The position-buffer on ATI controllers are unreliable as well as on VIA chips, thus the same workaround for DMA position reading as VIA is useful for ATI. Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 31 ++++++++++++++++++------------- 1 file changed, 18 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 643f0e49929..f3b5723c285 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2059,26 +2059,31 @@ static int __devinit check_position_fix(struct azx *chip, int fix) { const struct snd_pci_quirk *q; - /* Check VIA HD Audio Controller exist */ - if (chip->pci->vendor == PCI_VENDOR_ID_VIA && - chip->pci->device == VIA_HDAC_DEVICE_ID) { + switch (fix) { + case POS_FIX_LPIB: + case POS_FIX_POSBUF: + return fix; + } + + /* Check VIA/ATI HD Audio Controller exist */ + switch (chip->driver_type) { + case AZX_DRIVER_VIA: + case AZX_DRIVER_ATI: chip->via_dmapos_patch = 1; /* Use link position directly, avoid any transfer problem. */ return POS_FIX_LPIB; } chip->via_dmapos_patch = 0; - if (fix == POS_FIX_AUTO) { - q = snd_pci_quirk_lookup(chip->pci, position_fix_list); - if (q) { - printk(KERN_INFO - "hda_intel: position_fix set to %d " - "for device %04x:%04x\n", - q->value, q->subvendor, q->subdevice); - return q->value; - } + q = snd_pci_quirk_lookup(chip->pci, position_fix_list); + if (q) { + printk(KERN_INFO + "hda_intel: position_fix set to %d " + "for device %04x:%04x\n", + q->value, q->subvendor, q->subdevice); + return q->value; } - return fix; + return POS_FIX_AUTO; } /* -- cgit v1.2.3 From 36c7b833e5d2501142a371e4e75281d3a29fbd6b Mon Sep 17 00:00:00 2001 From: Viral Mehta Date: Tue, 10 Mar 2009 15:43:18 +0100 Subject: ALSA: oss-mixer - Fixes recording gain control At the time of initialization, SNDRV_MIXER_OSS_PRESENT_PVOLUME bit is not set for MIC (slot 7). So, the same should not be checked when an application tries to do gain control for audio recording devices. Just check slot->present for SNDRV_MIXER_OSS_PRESENT_CVOLUME independently. Verified with a simple application which opens /dev/dsp for recording and /dev/mixer for volume control. Have tested two usb audio mic devices. Signed-off-by: Viral Mehta Signed-off-by: Takashi Iwai --- sound/core/oss/mixer_oss.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 4690b8b5681..e570649184e 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -692,6 +692,9 @@ static int snd_mixer_oss_put_volume1(struct snd_mixer_oss_file *fmixer, snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PVOLUME], left, right); if (slot->present & SNDRV_MIXER_OSS_PRESENT_CVOLUME) snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CVOLUME], left, right); + } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_CVOLUME) { + snd_mixer_oss_put_volume1_vol(fmixer, pslot, + slot->numid[SNDRV_MIXER_OSS_ITEM_CVOLUME], left, right); } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GVOLUME) { snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GVOLUME], left, right); } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GLOBAL) { -- cgit v1.2.3 From 91054598f794fb5d8a0b1e747ff8e2e8fc2115b3 Mon Sep 17 00:00:00 2001 From: Jiri Slaby Date: Wed, 11 Mar 2009 20:11:40 +0100 Subject: ALSA: pcm_oss, fix locking typo s/mutex_lock/mutex_unlock/ on 2 fail paths in snd_pcm_oss_proc_write. Probably a typo, lock should be unlocked when leaving the function. Signed-off-by: Jiri Slaby Cc: Signed-off-by: Takashi Iwai --- sound/core/oss/pcm_oss.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 0a1798eafb0..699d2890535 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2872,7 +2872,7 @@ static void snd_pcm_oss_proc_write(struct snd_info_entry *entry, setup = kmalloc(sizeof(*setup), GFP_KERNEL); if (! setup) { buffer->error = -ENOMEM; - mutex_lock(&pstr->oss.setup_mutex); + mutex_unlock(&pstr->oss.setup_mutex); return; } if (pstr->oss.setup_list == NULL) @@ -2886,7 +2886,7 @@ static void snd_pcm_oss_proc_write(struct snd_info_entry *entry, if (! template.task_name) { kfree(setup); buffer->error = -ENOMEM; - mutex_lock(&pstr->oss.setup_mutex); + mutex_unlock(&pstr->oss.setup_mutex); return; } } -- cgit v1.2.3 From 82f5d57163abed2e5ff271d03217b6f90c616eb8 Mon Sep 17 00:00:00 2001 From: Jiri Slaby Date: Wed, 11 Mar 2009 20:11:41 +0100 Subject: ALSA: mixart, fix lock imbalance There is an omitted unlock in one snd_mixart_hw_params fail path. Fix it. Signed-off-by: Jiri Slaby Cc: Signed-off-by: Takashi Iwai --- sound/pci/mixart/mixart.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index f23a73577c2..bb162507fe6 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -607,6 +607,7 @@ static int snd_mixart_hw_params(struct snd_pcm_substream *subs, /* set the format to the board */ err = mixart_set_format(stream, format); if(err < 0) { + mutex_unlock(&mgr->setup_mutex); return err; } -- cgit v1.2.3 From 6af845e4eb36fb91b322aaf77ec1cab2220a48ad Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Mar 2009 14:00:06 +0100 Subject: ALSA: Fix vunmap and free order in snd_free_sgbuf_pages() In snd_free_sgbuf_pags(), vunmap() is called after releasing the SG pages, and it causes errors on Xen as Xen manages the pages differently. Although no significant errors have been reported on the actual hardware, this order should be fixed other way round, first vunmap() then free pages. Cc: Jan Beulich Cc: Signed-off-by: Takashi Iwai --- sound/core/sgbuf.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c index d4564edd61d..4e7ec2b4987 100644 --- a/sound/core/sgbuf.c +++ b/sound/core/sgbuf.c @@ -38,6 +38,10 @@ int snd_free_sgbuf_pages(struct snd_dma_buffer *dmab) if (! sgbuf) return -EINVAL; + if (dmab->area) + vunmap(dmab->area); + dmab->area = NULL; + tmpb.dev.type = SNDRV_DMA_TYPE_DEV; tmpb.dev.dev = sgbuf->dev; for (i = 0; i < sgbuf->pages; i++) { @@ -48,9 +52,6 @@ int snd_free_sgbuf_pages(struct snd_dma_buffer *dmab) tmpb.bytes = (sgbuf->table[i].addr & ~PAGE_MASK) << PAGE_SHIFT; snd_dma_free_pages(&tmpb); } - if (dmab->area) - vunmap(dmab->area); - dmab->area = NULL; kfree(sgbuf->table); kfree(sgbuf->page_table); -- cgit v1.2.3