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

[alsa-devel] HDSPM - Problems with SyncIn
by Andre Schramm 19 Oct '12

19 Oct '12

Hello, I did some tests with the SyncIn-Feature of the HDSPe cards. [root@localhost ~]# lspci |grep Mult 01:00.0 Multimedia audio controller: Xilinx Corporation RME Hammerfall DSP MADI (rev d2) 02:00.0 Multimedia audio controller: Xilinx Corporation RME Hammerfall DSP MADI (rev d2) The SyncOut of card 2:00 was plugged to the SyncIn of 1:00 and 1:00 was put to AutoSync-mode automatically upon boot. Problem is that the amixer output seems to be inconsistent and differs from the proc output as well, so I'm not sure which one to trust. [root@localhost ~]# amixer -c0 contents numid=5,iface=MIXER,name='AutoSync Reference' ; type=ENUMERATED,access=r-------,values=1,items=5 ; Item #0 'Word Clock' ; Item #1 'MADI' ; Item #2 'TCO' ; Item #3 'Sync In' ; Item #4 'None' : values=3 numid=2,iface=MIXER,name='Internal Clock' ; type=ENUMERATED,access=rw------,values=1,items=9 ; Item #0 '32 kHz' ; Item #1 '44.1 kHz' ; Item #2 '48 kHz' ; Item #3 '64 kHz' ; Item #4 '88.2 kHz' ; Item #5 '96 kHz' ; Item #6 '128 kHz' ; Item #7 '176.4 kHz' ; Item #8 '192 kHz' : values=2 numid=4,iface=MIXER,name='Preferred Sync Reference' ; type=ENUMERATED,access=rw------,values=1,items=3 ; Item #0 'Word Clock' ; Item #1 'MADI' ; Item #2 'Sync In' : values=2 numid=10,iface=MIXER,name='SYNC IN SyncCheck' ; type=ENUMERATED,access=r-------,values=1,items=4 ; Item #0 'No Lock' ; Item #1 'Lock' ; Item #2 'Sync' ; Item #3 'N/A' : values=0 numid=3,iface=MIXER,name='System Clock Mode' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Master' ; Item #1 'AutoSync' : values=1 numid=6,iface=MIXER,name='System Sample Rate' ; type=INTEGER,access=r-------,values=1,min=27000,max=207000,step=1 : values=47999 'Preferred Sync Reference' is SyncIn, 'SYNC IN SyncCheck' says 'No Lock', meaning no signal at all, but the 'System Sample Rate' coming from the master card is detected with 1Hz offset. When putting the master to 32000Hz, 31999Hz are shown and so on. [root@localhost ~]# cat /proc/asound/card0/hdspm RME MADI (Card #1) Rev.d2 Status2first3bits: 1 HW Serial: 0x0000019196d4 IRQ: 16 Registers bus: 0xfe9f0000 VM: 0xf8920000 --- System --- IRQ Pending: Audio=0, MIDI0=0, MIDI1=0, IRQcount=0 HW pointer: id = 0, rawptr = 0 (0->16320) estimated= 0 (bytes) MIDI FIFO: Out1=0x0, Out2=0x0, In1=0x0, In2=0x0 MIDIoverMADI FIFO: In=0x0, Out=0x0 Register: ctrl1=0x10344ca, ctrl2=0x0, status1=0xc30004, status2=0xc0020361 No TCO module detected. --- Settings --- Size (Latency): 2048 samples (2 periods of 8192 bytes) Line out: on ClearTrackMarker = off, Transmit in 64 Channel Mode, Auto Input off AutoSync Reference: AutoSync Preferred Sync Reference: TCO System Clock Frequency: 48000 --- Status: Inputs MADI=NoLock, WordClock=NoLock AutoSync: Reference= Sync In, Freq=0 (MADI = 3, Word = 3) Input: Coax, Mode=56 channels Now the proc-output reports 'No TCO module detected', which is correct, as no expansion modules where plugged to the cards. But it also says 'Preferred Sync Reference: TCO' and 'Freq=0', where amixer said SyncIn and 47999? Output of the master card is below. Regards, Andre [root@localhost ~]# amixer -c1 contents numid=5,iface=MIXER,name='AutoSync Reference' ; type=ENUMERATED,access=r-------,values=1,items=5 ; Item #0 'Word Clock' ; Item #1 'MADI' ; Item #2 'TCO' ; Item #3 'Sync In' ; Item #4 'None' : values=4 numid=15,iface=MIXER,name='Input Select' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'optical' ; Item #1 'coaxial' : values=0 numid=2,iface=MIXER,name='Internal Clock' ; type=ENUMERATED,access=rw------,values=1,items=9 ; Item #0 '32 kHz' ; Item #1 '44.1 kHz' ; Item #2 '48 kHz' ; Item #3 '64 kHz' ; Item #4 '88.2 kHz' ; Item #5 '96 kHz' ; Item #6 '128 kHz' ; Item #7 '176.4 kHz' ; Item #8 '192 kHz' : values=2 numid=4,iface=MIXER,name='Preferred Sync Reference' ; type=ENUMERATED,access=rw------,values=1,items=3 ; Item #0 'Word Clock' ; Item #1 'MADI' ; Item #2 'Sync In' : values=0 numid=10,iface=MIXER,name='SYNC IN SyncCheck' ; type=ENUMERATED,access=r-------,values=1,items=4 ; Item #0 'No Lock' ; Item #1 'Lock' ; Item #2 'Sync' ; Item #3 'N/A' : values=0 numid=3,iface=MIXER,name='System Clock Mode' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Master' ; Item #1 'AutoSync' : values=0 numid=6,iface=MIXER,name='System Sample Rate' ; type=INTEGER,access=r-------,values=1,min=27000,max=207000,step=1 : values=48000 [root@localhost ~]# cat /proc/asound/card1/hdspm RME MADI (Card #2) Rev.d2 Status2first3bits: 1 HW Serial: 0x000001b1c606 IRQ: 17 Registers bus: 0xfeaf0000 VM: 0xf9280000 --- System --- IRQ Pending: Audio=0, MIDI0=0, MIDI1=0, IRQcount=0 HW pointer: id = 0, rawptr = 0 (0->16320) estimated= 0 (bytes) MIDI FIFO: Out1=0x0, Out2=0x0, In1=0x0, In2=0x0 MIDIoverMADI FIFO: In=0x0, Out=0x0 Register: ctrl1=0x10004da, ctrl2=0x0, status1=0x2800000, status2=0xc0020701 No TCO module detected. --- Settings --- Size (Latency): 2048 samples (2 periods of 8192 bytes) Line out: on ClearTrackMarker = off, Transmit in 64 Channel Mode, Auto Input off AutoSync Reference: Master Preferred Sync Reference: Word Clock System Clock Frequency: 48000 --- Status: Inputs MADI=NoLock, WordClock=NoLock AutoSync: Reference= Input not valid, Freq=0 (MADI = 10, Word = 0) Input: Optical, Mode=56 channels

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Re: [alsa-devel] [PATCH 2.6.39-rc3] hda_intel: ALSA HD Audio patch for Intel Panther Point DeviceIDs
by Takashi Iwai 09 Oct '12

09 Oct '12

At Wed, 20 Apr 2011 10:59:57 -0700, Seth Heasley wrote: > > This patch adds the HD Audio Controller DeviceIDs for the Intel Panther Point PCH. > > Signed-off-by: Seth Heasley <seth.heasley(a)intel.com> Applied now. Thanks. Takashi > --- > --- linux-2.6.39-rc3/sound/pci/hda/hda_intel.c.orig 2011-04-11 17:21:51.000000000 -0700 > +++ linux-2.6.39-rc3/sound/pci/hda/hda_intel.c 2011-04-14 11:44:40.000000000 -0700 > @@ -126,6 +126,7 @@ > "{Intel, ICH10}," > "{Intel, PCH}," > "{Intel, CPT}," > + "{Intel, PPT}," > "{Intel, PBG}," > "{Intel, SCH}," > "{ATI, SB450}," > @@ -2759,6 +2760,8 @@ > { PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_PCH }, > /* PBG */ > { PCI_DEVICE(0x8086, 0x1d20), .driver_data = AZX_DRIVER_PCH }, > + /* Panther Point */ > + { PCI_DEVICE(0x8086, 0x1e20), .driver_data = AZX_DRIVER_PCH }, > /* SCH */ > { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH }, > /* Generic Intel */ >

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[alsa-devel] [PATCH] snd-usb-us122l: Fix missing NULL checks
by Karsten Wiese 27 Apr '12

27 Apr '12

Fix missing NULL checks in usb_stream_hwdep_poll() and usb_stream_hwdep_ioctl(). Wake up poll waiters before returning from usb_stream_hwdep_ioctl(). Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de> Cc: stable(a)kernel.org --- sound/usb/usx2y/us122l.c | 41 ++++++++++++++++++++--------------------- 1 files changed, 20 insertions(+), 21 deletions(-) diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index 91bb296..7abda8d 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -269,29 +269,26 @@ static unsigned int usb_stream_hwdep_poll(struct snd_hwdep *hw, struct file *file, poll_table *wait) { struct us122l *us122l = hw->private_data; - struct usb_stream *s = us122l->sk.s; unsigned *polled; unsigned int mask; poll_wait(file, &us122l->sk.sleep, wait); - switch (s->state) { - case usb_stream_ready: - if (us122l->first == file) - polled = &s->periods_polled; - else - polled = &us122l->second_periods_polled; - if (*polled != s->periods_done) { - *polled = s->periods_done; - mask = POLLIN | POLLOUT | POLLWRNORM; - break; + mask = POLLIN | POLLOUT | POLLWRNORM | POLLERR; + if (mutex_trylock(&us122l->mutex)) { + struct usb_stream *s = us122l->sk.s; + if (s && s->state == usb_stream_ready) { + if (us122l->first == file) + polled = &s->periods_polled; + else + polled = &us122l->second_periods_polled; + if (*polled != s->periods_done) { + *polled = s->periods_done; + mask = POLLIN | POLLOUT | POLLWRNORM; + } else + mask = 0; } - /* Fall through */ - mask = 0; - break; - default: - mask = POLLIN | POLLOUT | POLLWRNORM | POLLERR; - break; + mutex_unlock(&us122l->mutex); } return mask; } @@ -377,6 +374,7 @@ static int usb_stream_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, { struct usb_stream_config *cfg; struct us122l *us122l = hw->private_data; + struct usb_stream *s; unsigned min_period_frames; int err = 0; bool high_speed; @@ -422,18 +420,18 @@ static int usb_stream_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, snd_power_wait(hw->card, SNDRV_CTL_POWER_D0); mutex_lock(&us122l->mutex); + s = us122l->sk.s; if (!us122l->master) us122l->master = file; else if (us122l->master != file) { - if (memcmp(cfg, &us122l->sk.s->cfg, sizeof(*cfg))) { + if (!s || memcmp(cfg, &s->cfg, sizeof(*cfg))) { err = -EIO; goto unlock; } us122l->slave = file; } - if (!us122l->sk.s || - memcmp(cfg, &us122l->sk.s->cfg, sizeof(*cfg)) || - us122l->sk.s->state == usb_stream_xrun) { + if (!s || memcmp(cfg, &s->cfg, sizeof(*cfg)) || + s->state == usb_stream_xrun) { us122l_stop(us122l); if (!us122l_start(us122l, cfg->sample_rate, cfg->period_frames)) err = -EIO; @@ -444,6 +442,7 @@ unlock: mutex_unlock(&us122l->mutex); free: kfree(cfg); + wake_up_all(&us122l->sk.sleep); return err; } -- 1.7.2.3

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[alsa-devel] [PATCH] ALSA: HDA: Remove unconnected PCM devices for Intel HDMI
by David Henningsson 01 Feb '12

01 Feb '12

Since I'm a little new in HDMI land and the patch is non-trivial, I wouldn't mind some comments on this one. Also thanks to Jaroslav who cleared a few things out about the logical and physical devices which helped me figure out what I should do about it :-) Anyway, here's the patch. Some newer chips have more than one HDMI output, but usually not all of them are exposed as physical jacks. Removing the unused PCM devices (as indicated by BIOS in the pin config default) will reduce user confusion as they currently have to choose between several HDMI devices, some of them not working anyway. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[alsa-devel] Clevo P170HM / Sager NP8170 audio
by Andrew Mahone 04 Dec '11

04 Dec '11

This laptop is detected as ALC892 audio (and by the codec ID this seems to be valid), but none of the configurations for it quite match this device. I've been working on documenting the connections with HDA Analyzer. alsa-info output: http://www.alsa-project.org/db/?f=b86e4c6447867fbd129a36d8a63e2f5caa836c52 The laptop has built-in 5.1, and 5.1 or 7.1 output via jacks. DAC NIDs: 0x02 front 0x03 rear 0x04 clfe 0x05 side internal mixers: 0x0c front 0x0d rear 0x0e clfe 0x0f side internal speaker pins: 0x14 front 0x15 "rear" (front of laptop) 0x16 clfe external ports: fixed route: 0x17 side can select any output: 0x18 mic / output 0x1a line-in / output 0x1b headphone I've been experimenting with patch_realtek.c to add the extra DAC NID, and have had some success with a 6-channel mode for the built-in speakers and an 8-channel mode that mutes all internal speakers and sets the jacks up for output. I still need to manually connect pins 0x18, 0x1a, 0x1b to the correct internal mixers for 7.1 to work, and headphone plugin does not mute internal speakers in 5.1 mode. I have not added a plain stereo mode for internal speakers, although it appears that i can probably configure the codec to mix the front channels to center and LFE. Are there any suggestions for what to do next? My current state as a patch against the linux-3.0.y repository is at http://pastie.org/2400649 -- Andrew Mahone andrew DOT mahone AT gmail DOT com Q: How many voters does it take to change a lightbulb? A: None - if voting could change anything they'd make it illegal

3 29

Re: [alsa-devel] [patch][saa7134] do not change mute state for capturing audio
by Mauro Carvalho Chehab 03 Dec '11

03 Dec '11

Em 24-07-2011 14:45, Stas Sergeev escreveu: > 23.07.2011 19:09, Mauro Carvalho Chehab wrote: >> > In this case, it will not be autounmuted after tuning. >> Hard to tell about your solution without seeing a patch. > OK, it turns out the automute code is already there, > but it doesn't work. The driver for some reasons > starts the scan on initialization, finds the carrier: > --- > saa7134[0]/audio: found PAL main sound carrier @ 6.000 MHz [3969/324] > --- > and, because of that, disables the automute. If the > real mute is not enabled at that point, you get the > white noise right away. The automute code works fine. Maybe you have a strong interference at the default tuning frequency, leading into saa7134 miss-detection. For saa7134 driver: 1) There is an audio carrier; 2) an application wants to listen audio; 3) the device is at automute mode. So, when there's an audio carrier, it will unmute the audio at streaming start. The driver is doing the _right_ thing by letting the audio to flow to PA, when it starts the capture. Unfortunately, on your specific device, starting audio capture also enables audio at the audio output pin. So, you're noticing a problem. People with a saa7134 device without alsa stream won't notice it (old devices). People with alsa stream without anything connected to the LINE OUT people aren't noticing it, if PA is not copying the saa7134 PCM IN stream to the sound card PCM out device (the default, for PA). So, only people that has saa7134 with alsa stream that opted to wire the saa7134 device to the sound card, and with a strong interference at the default tunning frequency (400 MHz) would notice a problem. > Since I have no idea why it finds some carrier, I can't > fix that in any way. Or, maybe, not to call the scan > on driver init? What will that break? Analog tuners need to be tuned at the device init on a high frequency according with their datasheets, otherwise the PLL may fail. > Anyway, as long as the automute code is broken, > we should either start fixing it, or fix PA, or fix mplayer... As I always said, the fix should be at PA, as it makes no sense to start capturing at saa7134 without first configuring it. So, or PA should be converted into a V4L2-aware application, in order to properly init the device (with seems to be the wrong approach) or it _SHOULD_NOT_ automatically enable capture on those devices, as this may cause undesired side effects, on the devices that have the capture pin directly wired to the output pin, witch seems to be the case of your device. PS.: it seems that you've removed Lennart/alsa people from the c/c. I'm re-adding them, as I'm expecting a fix from their side. > Dunno. I wonder how come so many bugs left unfixed > for so long, resulting in a white noise to people... You're the first one that reported it, and the code is there for _years_. So, this is not a commonly noticed problem at all. Mauro.

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Re: [alsa-devel] max channels of 3 jacks 6 channels motherboard using model=auto realtek codec
by Takashi Iwai 20 Oct '11

20 Oct '11

At Sun, 10 Jul 2011 08:47:46 +0200, Takashi Iwai wrote: > > At Sun, 10 Jul 2011 11:31:44 +0800, > Raymond Yau wrote: > > > > 2011/7/6 Takashi Iwai <tiwai(a)suse.de>: > > > At Wed, 6 Jul 2011 11:53:47 +0800, > > > Raymond Yau wrote: > > >> > > >> Refer to commit ce764ab22e40a046065c9417ee5f04ad2a816ac1 > > >> > > >> ALSA: hda - Add channel-mode support to Realtek auto-parser > > >> > > >> It seem that max channel is six instead of two when the channel mode > > >> is 2ch for those 3 jacks motherboard > > >> > > >> using hda-emu and alsa-info from > > >> http://thread.gmane.org/gmane.linux.alsa.devel/86816 > > >> > > >> > PCM 0 p 44100 2 16 > > >> Open PCM ALC887 Analog for play > > >> Available PCM parameters: > > >> channels: 2/6 > > >> formats: S16_LE S32_LE > > >> rates: 44100 48000 96000 192000 > > >> Prepare PCM, rate=44100, channels=2, format=16 bits > > >> > > >> > > >> because > > >> > > >> spec->multiout.max_channels = spec->multiout.num_dacs * 2; > > >> > > >> is assigned after alc_auto_add_multi_channel_mode() increase > > >> spec->multiout.num_dacs from 1 to 3 > > >> > > >> spec->multiout.num_dacs = num_pins + 1; > > > > > > Right, and this is intentional. Since there should be enough amount > > > of DACs, the driver keeps using the surround DACs statically. > > > In that way, apps can keep opening the device in the surround mode > > > while user can switch the outputs on the fly via channel-mode mixer > > > enum. > > > > > > > > > > How can the driver support 6 channels when the input jacks have not > > been retasked as output ? > > Even if the pins aren't retasked, DACs can receive data. And then > you can change the pins and routing on the fly. > > You can open a two-channel PCM stream even if the master or (any > other) volume is muted. A similar reason. > > > This behaviour seem different from model=3stack-6ch or > > model=3stack-6ch-dig or those via codec with smart51 "OFF" The smart51 case is a bit different (we prefer DAC unused because of independent HP control), but the 3stack-6ch case is a regression, indeed. This was just because of dropped code snippet. Now I fixed with the patch below. thanks, Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Re-add need_dac_fix check for multi-io jacks of Realtek codecs During the rewrite, the check of spec->need_dac_fix and the corresponding num_dacs change was dropped from the channel-mode control. This patch re-adds it, and also enables need_dac_fix for ALC880 as default, as this feature was originally introduced to fix h/w bugs of this chip. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 124c63f..52ce075 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3196,6 +3196,8 @@ static int alc_auto_ch_mode_put(struct snd_kcontrol *kcontrol, for (i = 0; i < spec->multi_ios; i++) alc_set_multi_io(codec, i, i < ch); spec->multiout.max_channels = spec->ext_channel_count; + if (spec->need_dac_fix && !spec->const_channel_count) + spec->multiout.num_dacs = spec->multiout.max_channels / 2; return 1; } @@ -3642,6 +3644,7 @@ static int patch_alc880(struct hda_codec *codec) codec->spec = spec; spec->mixer_nid = 0x0b; + spec->need_dac_fix = 1; board_config = alc_board_config(codec, ALC880_MODEL_LAST, alc880_models, alc880_cfg_tbl); -- 1.7.6

2 1

[alsa-devel] [PATCH] ASoC: WM8903: Implement DMIC support
by Stephen Warren 14 Oct '11

14 Oct '11

In addition to the currently supported analog capture path, the WM8903 also supports digital mics. The analog and digital capture paths are exclusive; a mux is present to select the capture source. Logically, the mux exists to select the decimator's input, from either the ADC or DMIC block outputs. However, the ADC power domain also includes the DMIC interface. Consequently, this change represents the mux as existing immediately before the ADC, and selecting between the Input PGA and DMIC block outputs. An alternative might be to represent the mux in its correct location, and associate the ADC power enable controls with both the real ADC, and a fake ADC for the DMIC? Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/codecs/wm8903.c | 23 +++++++++++++++++++++-- 1 files changed, 21 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index d53f206..f3cab84 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -634,6 +634,13 @@ static const struct soc_enum lsidetone_enum = static const struct soc_enum rsidetone_enum = SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 0, 3, sidetone_text); +static const char *adcinput_text[] = { + "ADC", "DMIC" +}; + +static const struct soc_enum adcinput_enum = + SOC_ENUM_SINGLE(WM8903_CLOCK_RATE_TEST_4, 9, 2, adcinput_text); + static const char *aif_text[] = { "Left", "Right" }; @@ -767,6 +774,9 @@ static const struct snd_kcontrol_new lsidetone_mux = static const struct snd_kcontrol_new rsidetone_mux = SOC_DAPM_ENUM("DACR Sidetone Mux", rsidetone_enum); +static const struct snd_kcontrol_new adcinput_mux = + SOC_DAPM_ENUM("ADC Input", adcinput_enum); + static const struct snd_kcontrol_new lcapture_mux = SOC_DAPM_ENUM("Left Capture Mux", lcapture_enum); @@ -817,6 +827,7 @@ SND_SOC_DAPM_INPUT("IN2L"), SND_SOC_DAPM_INPUT("IN2R"), SND_SOC_DAPM_INPUT("IN3L"), SND_SOC_DAPM_INPUT("IN3R"), +SND_SOC_DAPM_INPUT("DMICDAT"), SND_SOC_DAPM_OUTPUT("HPOUTL"), SND_SOC_DAPM_OUTPUT("HPOUTR"), @@ -842,6 +853,9 @@ SND_SOC_DAPM_MUX("Right Input Mode Mux", SND_SOC_NOPM, 0, 0, &rinput_mode_mux), SND_SOC_DAPM_PGA("Left Input PGA", WM8903_POWER_MANAGEMENT_0, 1, 0, NULL, 0), SND_SOC_DAPM_PGA("Right Input PGA", WM8903_POWER_MANAGEMENT_0, 0, 0, NULL, 0), +SND_SOC_DAPM_MUX("Left ADC Input", SND_SOC_NOPM, 0, 0, &adcinput_mux), +SND_SOC_DAPM_MUX("Right ADC Input", SND_SOC_NOPM, 0, 0, &adcinput_mux), + SND_SOC_DAPM_ADC("ADCL", NULL, WM8903_POWER_MANAGEMENT_6, 1, 0), SND_SOC_DAPM_ADC("ADCR", NULL, WM8903_POWER_MANAGEMENT_6, 0, 0), @@ -979,6 +993,11 @@ static const struct snd_soc_dapm_route wm8903_intercon[] = { { "Left Input PGA", NULL, "Left Input Mode Mux" }, { "Right Input PGA", NULL, "Right Input Mode Mux" }, + { "Left ADC Input", "ADC", "Left Input PGA" }, + { "Left ADC Input", "DMIC", "DMICDAT" }, + { "Right ADC Input", "ADC", "Right Input PGA" }, + { "Right ADC Input", "DMIC", "DMICDAT" }, + { "Left Capture Mux", "Left", "ADCL" }, { "Left Capture Mux", "Right", "ADCR" }, @@ -988,9 +1007,9 @@ static const struct snd_soc_dapm_route wm8903_intercon[] = { { "AIFTXL", NULL, "Left Capture Mux" }, { "AIFTXR", NULL, "Right Capture Mux" }, - { "ADCL", NULL, "Left Input PGA" }, + { "ADCL", NULL, "Left ADC Input" }, { "ADCL", NULL, "CLK_DSP" }, - { "ADCR", NULL, "Right Input PGA" }, + { "ADCR", NULL, "Right ADC Input" }, { "ADCR", NULL, "CLK_DSP" }, { "Left Playback Mux", "Left", "AIFRXL" }, -- 1.7.1

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[alsa-devel] Report by PulseAudio: `alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail.`
by Paul Menzel 29 Sep '11

29 Sep '11

Dear ALSA developers, I got the following messages in `/var/log/syslog`. Aug 24 09:29:05 doe pulseaudio[2735]: core-util.c: Home directory /etc/timidity not ours. Aug 24 09:29:05 doe pulseaudio[2735]: lock-autospawn.c: Cannot access autospawn lock. Aug 24 09:29:05 doe pulseaudio[2735]: main.c: Failed to acquire autospawn lock Aug 24 13:00:33 doe pulseaudio[3118]: alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! Aug 24 13:00:33 doe pulseaudio[3118]: alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Aug 24 13:00:33 doe pulseaudio[3118]: alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Additionally I am seeing the a lot of the following messages in the kernel ring buffer (`dmesg`) [14293.422591] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422612] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422633] hda-intel: spurious response 0xe0160:0x0, last cmd=0x1f0004 [14293.422653] hda-intel: spurious response 0x1:0x0, last cmd=0x1f0004 [14293.422675] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422695] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422716] hda-intel: spurious response 0x20025:0x0, last cmd=0x1f0004 but I do not know if this is related. I am using the Debian Linux kernel 2.6.32 which includes all longterm patches up to 2.6.32.40. So only the ALSA drivers version 1.0.21 are used. Is there a commit that addresses the above behavior and which could be backported? Please find the output of `alsa-info.sh` [1] at the end. Thanks, Paul [1] http://alsa-project.org/main/index.php/Help_To_Debug !!################################ !!ALSA Information Script v 0.4.60 !!################################ !!Script ran on: Wed Aug 24 11:24:45 UTC 2011 !!Linux Distribution !!------------------ Debian GNU/Linux wheezy/sid \n \l !!DMI Information !!--------------- Manufacturer: System manufacturer Product Name: System Product Name Product Version: System Version !!Kernel Information !!------------------ Kernel release: 2.6.32-5-amd64 Operating System: GNU/Linux Architecture: x86_64 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.21 Library version: 1.0.24.1 Utilities version: 1.0.24.2 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes ESound Daemon: Installed - Yes (/usr/bin/esd) Running - No aRts: Installed - Yes (/usr/bin/artsd) Running - No Jack: Installed - Yes (/usr/bin/jackd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [VT82xx ]: HDA-Intel - HDA VIA VT82xx HDA VIA VT82xx at 0xfbffc000 irq 17 !!PCI Soundcards installed in the system !!-------------------------------------- 20:01.0 Audio device: VIA Technologies, Inc. VT1708/A [Azalia HDAC] (VIA High Definition Audio Controller) (rev 10) !!Advanced information - PCI Vendor/Device/Subsystem ID's !!-------------------------------------------------------- 20:01.0 0403: 1106:3288 (rev 10) Subsystem: 1043:8290 !!Modprobe options (Sound related) !!-------------------------------- snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-pcsp: index=-2 snd-usb-audio: index=-2 !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel bdl_pos_adj : 32,-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,-1,-1,-1,-1,-1 enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : 0 id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : -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,-1,-1,-1,-1,-1,-1 model : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> patch : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> position_fix : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -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,-1,-1,-1,-1,-1,-1 probe_only : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC662 rev1 Address: 0 Function Id: 0x1 Vendor Id: 0x10ec0662 Subsystem Id: 0x10438290 Revision Id: 0x100101 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A Invalid AFG subtree --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 9 Aug 24 09:27 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 8 Aug 24 09:27 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 7 Aug 24 09:27 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 6 Aug 24 13:07 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 5 Aug 24 09:27 /dev/snd/pcmC0D1c crw-rw----+ 1 root audio 116, 4 Aug 24 09:27 /dev/snd/pcmC0D1p crw-rw----+ 1 root audio 116, 3 Aug 24 09:27 /dev/snd/seq crw-rw----+ 1 root audio 116, 2 Aug 24 09:27 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 60 Aug 24 09:27 . drwxr-xr-x 3 root root 220 Aug 24 09:27 .. lrwxrwxrwx 1 root root 12 Aug 24 09:27 pci-0000:20:01.0 -> ../controlC0 !!ALSA configuration files !!------------------------ !!System wide config file (/etc/asound.conf) pcm.pulse { type pulse } ctl.pulse { type pulse } pcm.!default { type pulse } ctl.!default { type pulse } !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: VT82xx [HDA VIA VT82xx], device 0: ALC662 rev1 Analog [ALC662 rev1 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: VT82xx [HDA VIA VT82xx], device 1: ALC662 rev1 Digital [ALC662 rev1 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: VT82xx [HDA VIA VT82xx], device 0: ALC662 rev1 Analog [ALC662 rev1 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: VT82xx [HDA VIA VT82xx], device 1: ALC662 rev1 Digital [ALC662 rev1 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [VT82xx] Card hw:0 'VT82xx'/'HDA VIA VT82xx at 0xfbffc000 irq 17' Mixer name : 'Realtek ALC662 rev1' Components : 'HDA:10ec0662,10438290,00100101' Controls : 37 Simple ctrls : 20 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 63 [98%] [-1.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pswitch penum Playback channels: Front Left - Front Right Mono: Front Left: Playback [on] Front Right: Playback [on] Simple mixer control 'PCM',0 Capabilities: pvolume penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 254 [100%] [0.20dB] Front Right: Playback 254 [100%] [0.20dB] Simple mixer control 'Front',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 64 [100%] [0.00dB] [on] Front Right: Playback 64 [100%] [0.00dB] [on] Simple mixer control 'Front Mic',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 20 [65%] [-4.50dB] [on] Front Right: Playback 20 [65%] [-4.50dB] [on] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 64 [100%] [0.00dB] [on] Front Right: Playback 64 [100%] [0.00dB] [on] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 64 [100%] [0.00dB] [on] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 64 [100%] [0.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 16 [52%] [-10.50dB] [on] Front Right: Playback 16 [52%] [-10.50dB] [on] Simple mixer control 'CD',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 17 [55%] [-9.00dB] [on] Front Right: Playback 17 [55%] [-9.00dB] [on] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 17 [55%] [-9.00dB] [on] Front Right: Playback 17 [55%] [-9.00dB] [on] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined cswitch cswitch-joined penum Playback channels: Mono Capture channels: Mono Mono: Playback [on] Capture [off] Simple mixer control 'IEC958 Default PCM',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Beep',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 24 [77%] [22.50dB] [on] Front Right: Capture 24 [77%] [22.50dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 9 [29%] [0.00dB] [off] Front Right: Capture 9 [29%] [0.00dB] [off] Simple mixer control 'Channel Mode',0 Capabilities: enum Items: '2ch' '6ch' Item0: '2ch' Simple mixer control 'Digital',0 Capabilities: cvolume penum Capture channels: Front Left - Front Right Limits: Capture 0 - 120 Front Left: Capture 66 [55%] [3.00dB] Front Right: Capture 66 [55%] [3.00dB] Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Mic' Simple mixer control 'Input Source',1 Capabilities: cenum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Mic' !!Alsactl output !!------------- --startcollapse-- state.VT82xx { control.1 { iface MIXER name 'Front Playback Volume' value.0 64 value.1 64 comment { access 'read write' type INTEGER count 2 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } control.2 { iface MIXER name 'Front Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.3 { iface MIXER name 'Surround Playback Volume' value.0 64 value.1 64 comment { access 'read write' type INTEGER count 2 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } control.4 { iface MIXER name 'Surround Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.5 { iface MIXER name 'Center Playback Volume' value 64 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 0 } } control.6 { iface MIXER name 'LFE Playback Volume' value 64 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 0 } } control.7 { iface MIXER name 'Center Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.8 { iface MIXER name 'LFE Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.9 { iface MIXER name 'Headphone Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.10 { iface MIXER name 'CD Playback Volume' value.0 17 value.1 17 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -900 dbvalue.1 -900 } } control.11 { iface MIXER name 'CD Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.12 { iface MIXER name 'Line Playback Volume' value.0 16 value.1 16 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -1050 dbvalue.1 -1050 } } control.13 { iface MIXER name 'Line Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.14 { iface MIXER name 'Mic Playback Volume' value.0 17 value.1 17 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -900 dbvalue.1 -900 } } control.15 { iface MIXER name 'Mic Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.16 { iface MIXER name 'Front Mic Playback Volume' value.0 20 value.1 20 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -450 dbvalue.1 -450 } } control.17 { iface MIXER name 'Front Mic Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.18 { iface MIXER name 'Channel Mode' value '2ch' comment { access 'read write' type ENUMERATED count 1 item.0 '2ch' item.1 '6ch' } } control.19 { iface MIXER name 'Capture Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.20 { iface MIXER name 'Capture Switch' index 1 value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.21 { iface MIXER name 'Capture Volume' value.0 24 value.1 24 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -1350 dbmax 3300 dbvalue.0 2250 dbvalue.1 2250 } } control.22 { iface MIXER name 'Capture Volume' index 1 value.0 9 value.1 9 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -1350 dbmax 3300 dbvalue.0 0 dbvalue.1 0 } } control.23 { iface MIXER name 'Input Source' value Mic comment { access 'read write' type ENUMERATED count 1 item.0 Mic item.1 'Front Mic' item.2 Line item.3 CD } } control.24 { iface MIXER name 'Input Source' index 1 value Mic comment { access 'read write' type ENUMERATED count 1 item.0 Mic item.1 'Front Mic' item.2 Line item.3 CD } } control.25 { iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.26 { iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.27 { iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.28 { iface MIXER name 'IEC958 Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.29 { iface MIXER name 'IEC958 Default PCM Playback Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.30 { iface MIXER name 'IEC958 Capture Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.31 { iface MIXER name 'IEC958 Capture Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.32 { iface MIXER name 'Beep Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.33 { iface MIXER name 'Beep Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.34 { iface MIXER name 'Master Playback Volume' value 63 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 -100 } } control.35 { iface MIXER name 'Master Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.36 { iface MIXER name 'PCM Playback Volume' value.0 254 value.1 254 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 -20 dbvalue.1 -20 } } control.37 { iface MIXER name 'Digital Capture Volume' value.0 66 value.1 66 comment { access 'read write user' type INTEGER count 2 range '0 - 120' tlv '0000000100000008fffff44800000032' dbmin -3000 dbmax 3000 dbvalue.0 300 dbvalue.1 300 } } } --endcollapse-- !!All Loaded Modules !!------------------ Module binfmt_misc via drm ip6table_filter ip6_tables iptable_filter ip_tables x_tables powernow_k8 cpufreq_conservative cpufreq_userspace cpufreq_stats cpufreq_powersave reiserfs xfs exportfs fuse kvm_amd kvm loop firewire_sbp2 firewire_core crc_itu_t snd_hda_codec_realtek snd_hda_intel snd_hda_codec snd_hwdep snd_pcm_oss snd_mixer_oss snd_pcm snd_seq_midi snd_rawmidi snd_seq_midi_event snd_seq snd_timer snd_seq_device amd64_edac_mod snd psmouse evdev edac_core parport_pc soundcore i2c_viapro parport snd_page_alloc k8temp asus_atk0110 shpchp edac_mce_amd serio_raw i2c_core pci_hotplug processor button ext3 jbd mbcache sha256_generic aes_x86_64 aes_generic cbc dm_crypt dm_mod raid1 md_mod sg usbhid hid sr_mod cdrom sd_mod crc_t10dif ata_generic uhci_hcd pata_via ehci_hcd sata_via libata thermal thermal_sys via_rhine mii usbcore nls_base scsi_mod !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x14 0x01014010 0x15 0x411111f0 0x16 0x411111f0 0x18 0x01a19840 0x19 0x02a19950 0x1a 0x0181304f 0x1b 0x02214120 0x1c 0x593301f0 0x1d 0x4005c603 0x1e 0x99430130 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: !!ALSA/HDA dmesg !!------------------ [ 38.078771] alloc kstat_irqs on node 0 [ 38.078780] HDA Intel 0000:20:01.0: PCI INT A -> GSI 17 (level, low) -> IRQ 17 [ 38.078848] HDA Intel 0000:20:01.0: setting latency timer to 64 [ 38.078852] HDA Intel 0000:20:01.0: PCI: Disallowing DAC for device [ 38.165465] input: HDA Digital PCBeep as /devices/pci0000:20/0000:20:01.0/input/input5 [ 38.229088] input: ImPS/2 Logitech Wheel Mouse as /devices/platform/i8042/serio1/input/input6 -- [ 146.194407] ioctl32(Xorg:2515): Unknown cmd fd(11) cmd(c0246400){t:'d';sz:36} arg(08665f10) on /dev/dri/card0 [ 185.172076] hda-intel: IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj. [ 327.111398] eth0: link up, 100Mbps, full-duplex, lpa 0x45E1 [ 337.652026] eth0: no IPv6 routers present [14293.421429] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421447] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421467] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421488] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421509] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421530] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421550] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421571] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421592] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421613] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421634] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421655] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421677] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421695] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421716] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421737] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421758] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421778] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421799] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421820] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421841] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421861] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421882] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421903] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421924] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421945] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421965] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421987] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422007] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422028] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422049] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422070] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422090] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422112] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422132] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422153] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422174] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422195] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422216] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422236] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422257] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422280] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422300] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422321] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422342] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422363] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422384] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422405] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422425] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422446] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422467] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422488] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422507] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422528] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422549] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422570] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422591] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422612] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422633] hda-intel: spurious response 0xe0160:0x0, last cmd=0x1f0004 [14293.422653] hda-intel: spurious response 0x1:0x0, last cmd=0x1f0004 [14293.422675] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422695] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422716] hda-intel: spurious response 0x20025:0x0, last cmd=0x1f0004

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[alsa-devel] [PATCH] ASoC: sdp4430: Fix string for FM input name
by Peter Ujfalusi 26 Sep '11

26 Sep '11

The name contains invalid valid character (/), which causes problems when trying to create the debugfs directory structure: ASoC: Failed to create Aux/FM Stereo In debugfs file Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/sdp4430.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index a951774..d462f1a 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -88,7 +88,7 @@ static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = { SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_HP("Headset Stereophone", NULL), SND_SOC_DAPM_SPK("Earphone Spk", NULL), - SND_SOC_DAPM_INPUT("Aux/FM Stereo In"), + SND_SOC_DAPM_INPUT("FM Stereo In"), }; static const struct snd_soc_dapm_route audio_map[] = { @@ -113,8 +113,8 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Earphone Spk", NULL, "EP"}, /* Aux/FM Stereo In: AFML, AFMR */ - {"AFML", NULL, "Aux/FM Stereo In"}, - {"AFMR", NULL, "Aux/FM Stereo In"}, + {"AFML", NULL, "FM Stereo In"}, + {"AFMR", NULL, "FM Stereo In"}, }; static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) -- 1.7.6

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