April 2011 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: HDA: Fix automute for Gateway NV79
by David Henningsson 29 Apr '11

29 Apr '11

The PCI SSID is 1025:031c and the codec SSID is 1025:031d, so the driver mistakes this for a SKU value, but looking at the numbers, this is obviously wrong. Cc: stable(a)kernel.org (2.6.38+) BugLink: http://bugs.launchpad.net/bugs/761861 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6c94c4a..35efee6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -19486,6 +19486,7 @@ enum { ALC272_FIXUP_MARIO, ALC662_FIXUP_CZC_P10T, ALC662_FIXUP_GIGABYTE, + ALC662_FIXUP_SKU_IGNORE, }; static const struct alc_fixup alc662_fixups[] = { @@ -19521,10 +19522,15 @@ static const struct alc_fixup alc662_fixups[] = { { } } }, + [ALC662_FIXUP_SKU_IGNORE] = { + .type = ALC_FIXUP_SKU, + .v.sku = ALC_FIXUP_SKU_IGNORE, + }, }; static struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE), + SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte", ALC662_FIXUP_GIGABYTE), -- 1.7.4.1 --------------080108060303000102000501--

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Re: [alsa-devel] 2.6.38.4 broke my audio
by Takashi Iwai 29 Apr '11

29 Apr '11

At Fri, 29 Apr 2011 09:25:08 +0100, Andrew Clayton wrote: > > On Thu, 28 Apr 2011 08:16:17 +0200, Takashi Iwai wrote: > > > OK, but now the patch assumes that it's a mutable output, so the > > auto-mute shall happen. > > > > Actually other OSes seem to mute the line-out as well, so we may > > follow the standard scenario. > > OK, so I guess the bit in the distribution that saves and restores > mixer settings at shutdown/startup needs updating to take this new > mixer into account? as the speaker mixer is always muted at startup > now. Right, it's the user-space stuff. The recent alsactl should do it right (adjust to the certain level for new control elements), I thought. Anyway, I'm inclined to revert the change once after figuring out what caused the issue on Emmanuel's machine. thanks, Takashi

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[alsa-devel] Decibel range / value from snd_hctl_elem_t
by Sebastian H. 29 Apr '11

29 Apr '11

Hello list I'm stuck with the hctl interface. What is the best way to get 1. the decibel range from a snd_hctl_elem_t 2. the decibel value from a snd_hctl_elem_t In the "Simple Mixer Interface" there are snd_mixer_selem_get_playback_dB, snd_mixer_selem_get_capture_dB, snd_mixer_selem_get_playback_dB_range and snd_mixer_selem_get_capture_dB_range But I can't find anything like that in the "High level Control Interface". Possibly related: What does TLV stand for? http://de.wikipedia.org/wiki/Type-Length-Value ? Sebastian H.

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[alsa-devel] [PATCH] ALSA: hda: add beep quirk for Realtek 0x1043831a
by Daniel Cordero 29 Apr '11

29 Apr '11

PC Beep was not being reported as enabled on my EeePC 901: SKU: enable_pcbeep=0x0 Signed-off-by: Daniel Cordero <danielcordero(a)lavabit.com> --- Broken since commit da00c24493bf6ae3772dfe7343dca033ebc75955 Author: Kailang Yang <kailang(a)realtek.com> Date: Fri Mar 19 11:23:45 2010 +0100 ALSA: hda - Add parse customize define function for Realtek codecs when the "if (spec->cdefine.enable_beep)" (now "has_cdefine_beep") check was added. Researching also found an EeePC 1000H with the same subsystem ID with a beep: http://www.alsa-project.org/db/?f=25d72fb31eb3bac7bec2390f64df5678dd7119fa but I don't know if that has stopped working on recent kernels. sound/pci/hda/patch_realtek.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index da7cdca..fe03d0d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5714,6 +5714,7 @@ static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids, static struct snd_pci_quirk beep_white_list[] = { SND_PCI_QUIRK(0x1043, 0x829f, "ASUS", 1), SND_PCI_QUIRK(0x1043, 0x83ce, "EeePC", 1), + SND_PCI_QUIRK(0x1043, 0x831a, "EeePC", 1), SND_PCI_QUIRK(0x8086, 0xd613, "Intel", 1), {} }; -- 1.7.3.4

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[alsa-devel] [PATCH 1/7] ASoC: Fix cards getting stuck in a powered state.
by Lars-Peter Clausen 29 Apr '11

29 Apr '11

Commit 52ba67bf(ASoC: Force all DAPM contexts into the same bias state) changed dapm_power_widgets to power up all DAPM contexts if at least one context is powered. As a side-effect of this change it is possible, that a card is not powered down again although non of it's DAPM contexts should be powered. This problem can arise if the card contains at least one widget-less DAPM context. As opposed to other DAPM contexts the power state for widget-less contexts is not reset at the beginning of dapm_power_widgets, so once such a context is powered it stays powered and keeps the whole card in a powered state with it. For widget-less codec DAPM contexts it is possible to recover from this situation by sending a dapm_stream_event. Since non-codec DAPM contexts (for example card contexts) do not receive stream events, cards containing such contexts will get stuck in powered state indefinitely. This is currently the case for most cards where the card's DAPM context itself is widget-less. If there is more then one widget-less context the card will stay powered indefinitely as well, since it is not possible to send stream events to two DAPM contexts at once. This patch modifies dapm_power_widgets to iterate over all widget-less DAPM contexts, instead of just looking at the current DAPM context, and if that context has a codec assigned it looks at the codec state. If the codec is active and not suspended it is assumed that the context should be powered. Otherwise it is assumed that the context does not have to be powered. As before, if at least one DAPM context should be powered all other contexts are power as well, to keep all context in the same bias state. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 51 +++++++++++++------------------------------------ 1 files changed, 14 insertions(+), 37 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 378f08a..3f2e360 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1025,13 +1025,10 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) LIST_HEAD(down_list); LIST_HEAD(async_domain); int power; + int power_card = 0; trace_snd_soc_dapm_start(card); - list_for_each_entry(d, &card->dapm_list, list) - if (d->n_widgets) - d->dev_power = 0; - /* Check which widgets we need to power and store them in * lists indicating if they should be powered up or down. */ @@ -1053,7 +1050,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) else power = 1; if (power) - w->dapm->dev_power = 1; + power_card = 1; if (w->power == power) continue; @@ -1070,45 +1067,25 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) } } - /* If there are no DAPM widgets then try to figure out power from the - * event type. + /* If we have not found a DAPM context yet, which should be powered, + * iterate over all widget-less DAPM codec contexts and try to figure + * out whether it should be powered from the codec state. */ - if (!dapm->n_widgets) { - switch (event) { - case SND_SOC_DAPM_STREAM_START: - case SND_SOC_DAPM_STREAM_RESUME: - dapm->dev_power = 1; - break; - case SND_SOC_DAPM_STREAM_STOP: - dapm->dev_power = !!dapm->codec->active; - break; - case SND_SOC_DAPM_STREAM_SUSPEND: - dapm->dev_power = 0; - break; - case SND_SOC_DAPM_STREAM_NOP: - switch (dapm->bias_level) { - case SND_SOC_BIAS_STANDBY: - case SND_SOC_BIAS_OFF: - dapm->dev_power = 0; - break; - default: - dapm->dev_power = 1; - break; + if (!power_card) { + list_for_each_entry(d, &card->dapm_list, list) { + if (d->n_widgets || !d->codec) + continue; + + if (d->codec->active && !d->codec->suspended) { + power_card = 1; + break; } - break; - default: - break; } } /* Force all contexts in the card to the same bias state */ - power = 0; list_for_each_entry(d, &card->dapm_list, list) - if (d->dev_power) - power = 1; - list_for_each_entry(d, &card->dapm_list, list) - d->dev_power = power; - + d->dev_power = power_card; /* Run all the bias changes in parallel */ list_for_each_entry(d, &dapm->card->dapm_list, list) -- 1.7.2.5

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Re: [alsa-devel] [PATCH] audio: tlv320aic26: fix PLL register configuration
by Mark Brown 28 Apr '11

28 Apr '11

On Thu, Apr 28, 2011 at 05:32:57PM -0400, Michael Williamson wrote: > - jval = (fsref == 44100) ? 7 : 8; > - dval = (fsref == 44100) ? 5264 : 1920; > + jval = fsref / (aic26->mclk / 2048); > + dval = fsref - jval * (aic26->mclk / 2048); I'd like some more brackets on this line, especially given the similarity with the previous line. Otherwise this looks fine if a bit magic: Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com>

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[alsa-devel] [PATCH - usb 1/1] Terratec Aureon 7.1 USB ID added for C-Media cm6206 quirks
by wbreyha＠gmx.net 28 Apr '11

28 Apr '11

From: Wolfgang Breyha <wbreyha(a)gmx.net> Hi! This patch adds support for the Terratec Aureon 7.1 USB which uses a C-Media cm6206 and needs all the quirks already found in the past. Greetings, Wolfgang Breyha Signed-off-by: Wolfgang Breyha <wbreyha(a)gmx.net> diff --git a/sound/usb/format.c b/sound/usb/format.c index 5b792d2..f079b5e 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -176,9 +176,11 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof if (!rate) continue; /* C-Media CM6501 mislabels its 96 kHz altsetting */ + /* Terratec Aureon 7.1 USB C-Media 6206, too */ if (rate == 48000 && nr_rates == 1 && (chip->usb_id == USB_ID(0x0d8c, 0x0201) || - chip->usb_id == USB_ID(0x0d8c, 0x0102)) && + chip->usb_id == USB_ID(0x0d8c, 0x0102) || + chip->usb_id == USB_ID(0x0ccd, 0x00b1)) && fp->altsetting == 5 && fp->maxpacksize == 392) rate = 96000; /* Creative VF0470 Live Cam reports 16 kHz instead of 8kHz */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 355759b..916b6a4 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -533,6 +533,7 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, case USB_ID(0x0d8c, 0x0102): /* C-Media CM6206 / CM106-Like Sound Device */ + case USB_ID(0x0ccd, 0x00b1): /* Terratec Aureon 7.1 USB */ return snd_usb_cm6206_boot_quirk(dev); case USB_ID(0x133e, 0x0815): -- 1.7.4.4

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[alsa-devel] [PATCH] ASoC: Work around allmodconfig failure
by Mark Brown 28 Apr '11

28 Apr '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 3 +++ sound/soc/soc-core.c | 4 ++++ sound/soc/soc-utils.c | 6 ++---- 3 files changed, 9 insertions(+), 4 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index cb6b18b..6ce3e57 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -888,6 +888,9 @@ static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) INIT_LIST_HEAD(&card->dapm_list); } +int snd_soc_util_init(void); +void snd_soc_util_exit(void); + #include <sound/soc-dai.h> #ifdef CONFIG_DEBUG_FS diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 16be3e5..a823654 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3868,12 +3868,16 @@ static int __init snd_soc_init(void) pr_warn("ASoC: Failed to create platform list debugfs file\n"); #endif + snd_soc_util_init(); + return platform_driver_register(&soc_driver); } module_init(snd_soc_init); static void __exit snd_soc_exit(void) { + snd_soc_util_exit(); + #ifdef CONFIG_DEBUG_FS debugfs_remove_recursive(snd_soc_debugfs_root); #endif diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 2865791..ec921ec 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -82,7 +82,7 @@ static struct platform_driver soc_dummy_driver = { static struct platform_device *soc_dummy_dev; -static int __init snd_soc_util_init(void) +int __init snd_soc_util_init(void) { int ret; @@ -102,11 +102,9 @@ static int __init snd_soc_util_init(void) return ret; } -module_init(snd_soc_util_init); -static void __exit snd_soc_util_exit(void) +void __exit snd_soc_util_exit(void) { platform_device_unregister(soc_dummy_dev); platform_driver_unregister(&soc_dummy_driver); } -module_exit(snd_soc_util_exit); -- 1.7.4.1

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[alsa-devel] dB range of emu10k1 hda-intel
by Raymond Yau 28 Apr '11

28 Apr '11

2011/4/26 Takashi Iwai <tiwai(a)suse.de> > > > > Is it a bug for PA server to add TLV_DB_GAIN_MUTE - infinite dB range of > > "Front" and "Surround" and those finite dB range of "Master" and "PCM" ? > > Hrm, not sure. Could you elaborate? > > Should the application add the dB range of Virtual Master and the dB range of the slave for HDA-Intel ? Correct me if I am wrong PA server seem use -168.00 dB to 0.00 dB. for hda-intel For ad1988 spec->vmaster_nid = 0x04; "Virtual Master" + "Front" + "PCM Softvol" = 58.5 + 58.5 + 51 = 168 state.Intel { control.1 { iface MIXER name 'Front Playback Volume' value.0 39 value.1 39 comment { access 'read write' type INTEGER count 2 range '0 - 39' dbmin -5850 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } D: alsa-mixer.c: Activating path analog-output D: alsa-mixer.c: Path analog-output (Analog Output), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=39, min_dB=-168, max_dB=0 D: alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes D: alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no D: alsa-mixer.c: Element Front, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=yes D: alsa-mixer.c: Element Surround, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x60, n_channels=2, override_map=yes D: alsa-mixer.c: Element Side, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0xc00, n_channels=2, override_map=yes D: alsa-mixer.c: Element Center, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x4900000000018, n_channels=1, override_map=yes D: alsa-mixer.c: Element LFE, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x80, n_channels=1, override_map=yes D: alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes I: alsa-sink.c: Hardware volume ranges from -168.00 dB to 0.00 dB. The dB range of emu10k1 calculated by PA server is (-100081dB to +12 dB) "AC97 Master" + "AC97 PCM" + "Front" = 46.5 + 46.5 + 99999.99 = 100093 D: alsa-mixer.c: Activating path analog-output D: alsa-mixer.c: Path analog-output (Analog Output), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=31, min_dB=-100081, max_dB=12 D: alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes D: alsa-mixer.c: Element Surround, direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x60, n_channels=2, override_map=yes D: alsa-mixer.c: Element PCM, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes D: alsa-mixer.c: Element External Amplifier, direction=1, switch=4, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no D: alsa-mixer.c: Option on (output-amplifier-on/Amplifier) index=1, priority=10 D: alsa-mixer.c: Option off (output-amplifier-off/No Amplifier) index=0, priority=0 D: alsa-mixer.c: Setting output-amplifier-on (Amplifier) priority=10 D: alsa-mixer.c: Setting output-amplifier-off (No Amplifier) priority=0 I: alsa-sink.c: Hardware volume ranges from -100080.99 dB to 12.00 dB.

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[alsa-devel] [PATCH] ASoC: prevent compilers from optimising pll calculation into __aeabi__uldivmod
by Barry Song 28 Apr '11

28 Apr '11

From: Barry Song <21cnbao(a)gmail.com> The newest compiliers can optimize pll calculation in several codecs into __aeabi__uldivmod which doesn't exist in kernel. Then the link will fail: ERROR: "__aeabi_uldivmod" [sound/soc/codecs/snd-soc-wm8974.ko] undefined! ERROR: "__aeabi_uldivmod" [sound/soc/codecs/snd-soc-wm8940.ko] undefined! ERROR: "__aeabi_uldivmod" [sound/soc/codecs/snd-soc-wm8510.ko] undefined! This patch prevent the optimizaton by insert ASM. Signed-off-by: Barry Song <21cnbao(a)gmail.com> Cc: Zhiwu Song <zhiwu.song(a)csr.com> Cc: Binghua Duan <binghua.duan(a)csr.com> --- sound/soc/codecs/wm8510.c | 6 ++++++ sound/soc/codecs/wm8940.c | 6 ++++++ sound/soc/codecs/wm8974.c | 6 ++++++ 3 files changed, 18 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index db0dced..a636b15 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -258,6 +258,12 @@ static void pll_factors(unsigned int target, unsigned int source) Nmod = target % source; Kpart = FIXED_PLL_SIZE * (long long)Nmod; + /* + * The following asm() prevents the compiler from optimising + * into a standard EABI function __aeabi__uldivmod() + */ + asm("" : "+r"(source)); + do_div(Kpart, source); K = Kpart & 0xFFFFFFFF; diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 25580e3..f6148af 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -520,6 +520,12 @@ static void pll_factors(unsigned int target, unsigned int source) Nmod = target % source; Kpart = FIXED_PLL_SIZE * (long long)Nmod; + /* + * The following asm() prevents the compiler from optimising + * into a standard EABI function __aeabi__uldivmod() + */ + asm("" : "+r"(source)); + do_div(Kpart, source); K = Kpart & 0xFFFFFFFF; diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index ca646a8..c9036d9 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -317,6 +317,12 @@ static void pll_factors(struct pll_ *pll_div, Nmod = target % source; Kpart = FIXED_PLL_SIZE * (long long)Nmod; + /* + * The following asm() prevents the compiler from optimising + * into a standard EABI function __aeabi__uldivmod() + */ + asm("" : "+r"(source)); + do_div(Kpart, source); K = Kpart & 0xFFFFFFFF; -- 1.7.1 Member of the CSR plc group of companies. CSR plc registered in England and Wales, registered number 4187346, registered office Churchill House, Cambridge Business Park, Cowley Road, Cambridge, CB4 0WZ, United Kingdom

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