December 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: Remove unused aic3x_i2c_init and aic3x_i2c_exit functions
by Axel Lin 02 Dec '10

02 Dec '10

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/tlv320aic3x.c | 15 --------------- 1 files changed, 0 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index fc5abdf..8cd4cf5 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1545,21 +1545,6 @@ static struct i2c_driver aic3x_i2c_driver = { .remove = aic3x_i2c_remove, .id_table = aic3x_i2c_id, }; - -static inline void aic3x_i2c_init(void) -{ - int ret; - - ret = i2c_add_driver(&aic3x_i2c_driver); - if (ret) - printk(KERN_ERR "%s: error regsitering i2c driver, %d\n", - __func__, ret); -} - -static inline void aic3x_i2c_exit(void) -{ - i2c_del_driver(&aic3x_i2c_driver); -} #endif static int __init aic3x_modinit(void) -- 1.7.2

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[alsa-devel] [PATCH] ASoC: omap: N810: Don't select CONFIG_OMAP_MUX but make it as dependency
by Jarkko Nikula 02 Dec '10

02 Dec '10

Not all omap boards use kernel based pin multiplexing so CONFIG_SND_OMAP_SOC_N810 should not select it by default as it can make harm to other boards in multi-board kernels. Therefore put CONFIG_OMAP_MUX as a dependency to N810 ASoC machine driver. Thanks to Tony Lindgren <tony(a)atomide.com> for noticing. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Cc: Tony Lindgren <tony(a)atomide.com> --- sound/soc/omap/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index d542ea2..a088db6 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -12,8 +12,8 @@ config SND_OMAP_SOC_MCPDM config SND_OMAP_SOC_N810 tristate "SoC Audio support for Nokia N810" depends on SND_OMAP_SOC && MACH_NOKIA_N810 && I2C + depends on OMAP_MUX select SND_OMAP_SOC_MCBSP - select OMAP_MUX select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on Nokia N810. -- 1.7.2.3

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[alsa-devel] [PATCH v2] ASoC: tpa6130a2: Replace DAPM code with direct interface
by Peter Ujfalusi 02 Dec '10

02 Dec '10

The use of DAPM widgets, and extra routing can cause ordering problems in the system. Machine drivers should use the exported direct interface with SND_SOC_DAPM_HP's event callback to manage the state of the amplifier. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- Hello, Changes since v1: - DAPM routing removed from the driver - tpa6130a2_stereo_enable takes snd_soc_codec as argument Peter sound/soc/codecs/tpa6130a2.c | 69 +++++++----------------------------------- sound/soc/codecs/tpa6130a2.h | 1 + 2 files changed, 12 insertions(+), 58 deletions(-) diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index c97badf..0a99f31 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -317,65 +317,24 @@ static void tpa6130a2_channel_enable(u8 channel, int enable) } } -static int tpa6130a2_pga_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - switch (event) { - case SND_SOC_DAPM_POST_PMU: - tpa6130a2_channel_enable(w->shift, 1); - break; - case SND_SOC_DAPM_POST_PMD: - tpa6130a2_channel_enable(w->shift, 0); - break; - } - return 0; -} - -static int tpa6130a2_supply_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +int tpa6130a2_stereo_enable(struct snd_soc_codec *codec, int enable) { int ret = 0; - - switch (event) { - case SND_SOC_DAPM_POST_PMU: + if (enable) { ret = tpa6130a2_power(1); - break; - case SND_SOC_DAPM_POST_PMD: + if (ret < 0) + return ret; + tpa6130a2_channel_enable(TPA6130A2_HP_EN_R | TPA6130A2_HP_EN_L, + 1); + } else { + tpa6130a2_channel_enable(TPA6130A2_HP_EN_R | TPA6130A2_HP_EN_L, + 0); ret = tpa6130a2_power(0); - break; } + return ret; } - -static const struct snd_soc_dapm_widget tpa6130a2_dapm_widgets[] = { - SND_SOC_DAPM_PGA_E("TPA6130A2 Left", SND_SOC_NOPM, - TPA6130A2_HP_EN_L, 0, NULL, 0, tpa6130a2_pga_event, - SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_PGA_E("TPA6130A2 Right", SND_SOC_NOPM, - TPA6130A2_HP_EN_R, 0, NULL, 0, tpa6130a2_pga_event, - SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_PGA_E("TPA6130A2 Stereo", SND_SOC_NOPM, - TPA6130A2_HP_EN_L | TPA6130A2_HP_EN_R, 0, NULL, 0, - tpa6130a2_pga_event, - SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_SUPPLY("TPA6130A2 Enable", SND_SOC_NOPM, - 0, 0, tpa6130a2_supply_event, - SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), - /* Outputs */ - SND_SOC_DAPM_OUTPUT("TPA6130A2 Headphone Left"), - SND_SOC_DAPM_OUTPUT("TPA6130A2 Headphone Right"), - SND_SOC_DAPM_OUTPUT("TPA6130A2 Headphone Stereo"), -}; - -static const struct snd_soc_dapm_route audio_map[] = { - {"TPA6130A2 Headphone Left", NULL, "TPA6130A2 Left"}, - {"TPA6130A2 Headphone Right", NULL, "TPA6130A2 Right"}, - {"TPA6130A2 Headphone Stereo", NULL, "TPA6130A2 Stereo"}, - - {"TPA6130A2 Headphone Left", NULL, "TPA6130A2 Enable"}, - {"TPA6130A2 Headphone Right", NULL, "TPA6130A2 Enable"}, - {"TPA6130A2 Headphone Stereo", NULL, "TPA6130A2 Enable"}, -}; +EXPORT_SYMBOL_GPL(tpa6130a2_stereo_enable); int tpa6130a2_add_controls(struct snd_soc_codec *codec) { @@ -387,18 +346,12 @@ int tpa6130a2_add_controls(struct snd_soc_codec *codec) data = i2c_get_clientdata(tpa6130a2_client); - snd_soc_dapm_new_controls(dapm, tpa6130a2_dapm_widgets, - ARRAY_SIZE(tpa6130a2_dapm_widgets)); - - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - if (data->id == TPA6140A2) return snd_soc_add_controls(codec, tpa6140a2_controls, ARRAY_SIZE(tpa6140a2_controls)); else return snd_soc_add_controls(codec, tpa6130a2_controls, ARRAY_SIZE(tpa6130a2_controls)); - } EXPORT_SYMBOL_GPL(tpa6130a2_add_controls); diff --git a/sound/soc/codecs/tpa6130a2.h b/sound/soc/codecs/tpa6130a2.h index 57e867f..5df49c8 100644 --- a/sound/soc/codecs/tpa6130a2.h +++ b/sound/soc/codecs/tpa6130a2.h @@ -57,5 +57,6 @@ #define TPA6130A2_VERSION_MASK (0x0f) extern int tpa6130a2_add_controls(struct snd_soc_codec *codec); +extern int tpa6130a2_stereo_enable(struct snd_soc_codec *codec, int enable); #endif /* __TPA6130A2_H__ */ -- 1.7.3.2

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[alsa-devel] asoc device probing and dai question
by Koul, Vinod 02 Dec '10

02 Dec '10

Hi, In our driver, the machine probe creates the soc-audio device. I am adding the codec and platform device in my soc-device. But the probe for the codec doesn't get called unless we add the cpu and codec dais in the machine probe. I saw the soc-core code and looks like it checks if cpu and codec dais are registerd if so then only proceeds and calls the codec probe. So should I continue to register the dai for both cpu and codec in machine probe. I didn't see other driver in soc/ doing this, so did I miss something. Code: In m/c driver... (the m/c driver probe is called by kernel when it finds this device, and I register it as a platform device) /* Audio machine driver */ static struct snd_soc_card snd_soc_intelmid_card = { .name = "intel_mid_sound_card", .platform = &intelmid_soc_platform, .dai_link = intelmid_msic_dai, .num_links = 1, }; /* Audio subsystem */ static struct snd_soc_device intelmid_snd_devdata = { .card = &snd_soc_intelmid_card, .codec_dev = &intel_msic_codec, }; int __devinit snd_intelmid_mc_probe(struct platform_device *pdev) { int ret_val; snd_printk(KERN_DEBUG "snd_intelmad_probe called\n"); ret_val = snd_soc_register_dais(mid_bsp_dai, ARRAY_SIZE(mid_bsp_dai)); ret_val = snd_soc_register_dais(intel_msic_dais, ARRAY_SIZE(intel_msic_dais)); intelmid_snd_device = platform_device_alloc("soc-audio", -1); if (!intelmid_snd_device) { snd_printk(KERN_ERR "Platform device allocation failed\n"); return -ENOMEM; } platform_set_drvdata(intelmid_snd_device, &intelmid_snd_devdata); intelmid_snd_devdata.dev = &intelmid_snd_device->dev; ret_val = platform_device_add(intelmid_snd_device); if (ret_val) { snd_printk(KERN_ERR "Unable to add platform device\n"); platform_device_put(intelmid_snd_device); } return ret_val; } In platform init (note I don't have probe for platform) static int __init intel_mid_soc_platform_init(void) { snd_printk(KERN_DEBUG "intel_mid_soc_platform_init called\n"); return snd_soc_register_platform(&intelmid_soc_platform); } Thanks Vinod

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[alsa-devel] Hardware events
by Rolf Theunissen 02 Dec '10

02 Dec '10

Hello, For some hardware devices (for instance the ThinkPad EC) the volume can be changed by (only) hardware buttons. Changes in the the hardware mixers are notified in the driver via the function "snd_ctl_notify" with the mask SNDRV_CLT_EVENT_MASK_VALUE. For notification of volume changes to the users of the system, an OSD can be used. To implement such a OSD, it must be possible to respond to changes done by the hardware. This raises the following question: is it possible to detect the events originating from the hardware in the functions that handle the callbacks (set via snd_mixer_elem_set_callback/snd_hctl_elem_set_callback)? This would allow to implement an OSD that only responds to the hardware buttons and that would ignore events originating from "software". Thanks

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[alsa-devel] [PATCH 0/5] ASoC: tpa6130a2: pop removal (mostly) changes
by Peter Ujfalusi 02 Dec '10

02 Dec '10

Hello, The following series improves the tpa driver to have better pop filtering capabilities. New stereo DAPM route can be used to reduce left/right power time differences. Machine drivers can opt to not use the built in DAPM routes, and use direct call to enable the stereo path. --- Peter Ujfalusi (5): ASoC: tpa6130a2: Simplify power state management ASoC: tpa6130a2: Defer SW enable from power enable ASoC: tpa6130a2: Use one event handler for PGA_E ASoC: tpa6130a2: Add stereo DAPM path ASoC: tpa6130a2: Make DAPM registration optional, and direct interface sound/soc/codecs/tpa6130a2.c | 77 ++++++++++++++++++++++++------------------ sound/soc/codecs/tpa6130a2.h | 3 +- 2 files changed, 46 insertions(+), 34 deletions(-) -- 1.7.3.2

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[alsa-devel] [PATCH] ALSA: hda: Use "alienware" model quirk for another SSID
by Daniel Chen 02 Dec '10

02 Dec '10

Hi, Attached is a patch against current sound-2.6 master HEAD (also applicable to all supported/maintained stable kernels inclusive of 2.6.32 and newer) that adds a bug reporter's SSID to the appropriate quirk table so that headphones and mics are usable. Please apply. Best, -Dan

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[alsa-devel] SND-SOC - how to test drivers ?
by Vaclav Peroutka 02 Dec '10

02 Dec '10

Hello all, I am trying to develop ALSA driver for audio subsystem for one Android mobile phone. I already have some messages in the kernel log - please see below. It means that I have s3c-i2s connected to MAX9880 (source code was not IMHO tested). And then the output from MAX9880 goes into unknown switching matrix ( for which I even don't have a datasheet - only register list and names of bits in registers - I can guess what purpose they serve to). What I want is to make some testbench driver and an application which can set switching matrix registers through IOCTL. In MAX9880 I want to preset microphone path to output and switch matrix until I hear my voice from microphone in speaker or headphones. Or, is there any better method how to do that ? AFAIK there is no alsamixer application for Android. What do you use for path and gain settings in this case ? And last question so far - where to look to fix this message: "Failed to add route LMICIN->Mic Input" ? All other routes were added so far. Thank you very much in advance, Vaclav <6>[ 2.932154] Advanced Linux Sound Architecture Driver Version 1.0.18a. <4>[ 2.953782] No device for DAI MAX9880 <4>[ 2.962009] No device for DAI s3c-i2s <4>[ 3.003635] I2S off <4>[ 3.019939] I2S off <4>[ 3.027948] Maxim MAX9880 Audio CODEC 0.12 <6>[ 3.054919] max9880: max9880_init <4>[ 3.063481] Entered s3c24xx_pcm_new <4>[ 3.071566] Entered s3c24xx_pcm_preallocate_dma_buffer <4>[ 3.080354] Entered s3c24xx_pcm_preallocate_dma_buffer <6>[ 3.088901] asoc: MAX9880 <-> s3c-i2s mapping ok <6>[ 3.097441] max9880: max9880_dapm_event 769 <6>[ 3.124311] max9880: max9880_dapm_event 0 <6>[ 3.140593] max9880: max9880_add_widgets <3>[ 3.156166] Failed to add route LMICIN->Mic Input <6>[ 3.164204] max9880: max9880_add_controls <6>[ 3.203234] ALSA device list: <6>[ 3.210962] #0: S3C6410 MAX9880 (MAX9880)

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[alsa-devel] poor sound from ALC 850/880 in alsa 1.0.21 and 1.0.23
by Colin Kincaid Williams 02 Dec '10

02 Dec '10

From: Colin Kincaid Williams <discord(a)uw.edu> Date: Wed, Dec 1, 2010 at 3:47 PM Subject: Re: [alsa-devel] poor sound from ALC 850/880 in alsa 1.0.21 and 1.0.23 To: Raymond Yau <superquad.vortex2(a)gmail.com> how do I mute them manually, without hdaanalyser? On Wed, Dec 1, 2010 at 1:35 AM, Raymond Yau <superquad.vortex2(a)gmail.com> wrote: > 2010/12/1 Colin Kincaid Williams <discord(a)uw.edu> > >> I selected node 0x0c in hdaanalyzer and see audio mixer [OxOb] in the >> Connection list, but I cannot select the radio button. >> > > According to the block diagram of alc880 datasheet, it look like two mute > switches for node 0x02 and 0x0b or checkbox in hda_analyzer > >> Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out >> Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-In vals: [0x00 0x00] [0x00 0x00] > > Unfortunately hda_analyzer seem fail to parse your codec -info > > File "./hda_analyzer.py", line 483, in <module> > sys.exit(main(sys.argv)) > File "./hda_analyzer.py", line 460, in main > if read_nodes(sys.argv[1:]) == 0: > File "./hda_analyzer.py", line 110, in read_nodes > read_nodes3(card, idx, i) > File "./hda_analyzer.py", line 77, in read_nodes3 > c = HDACodecProc(card, codec, proc_file) > File "/home/raymond/hda-analyzer/hda_proc.py", line 474, in __init__ > self.parse(proc_file) > File "/home/raymond/hda-analyzer/hda_proc.py", line 678, in parse > node.add_ampvals(line[15:], HDA_INPUT) > File "/home/raymond/hda-analyzer/hda_proc.py", line 353, in add_ampvals > self.wrongfile('amp vals ]') > File "/home/raymond/hda-analyzer/hda_proc.py", line 143, in wrongfile > raise ValueError, "wrong proc file format (%s)" % msg > ValueError: wrong proc file format (amp vals ]) > > > > >> >> On Sun, Nov 28, 2010 at 11:40 PM, Raymond Yau >> <superquad.vortex2(a)gmail.com> wrote: >> > 2010/11/28 Colin Kincaid Williams <discord(a)uw.edu> >> > >> >> no, the motherboard has 6 plugs, but no digital outputs on the back. >> >> Furthermore, I muted the CD channel, the line in was already muted, >> >> but I still have crap sound compared to my $20 mp3 player. Something >> >> else is wrong! >> >> >> >> >> > Take a look at the block diagram of alc880 datasheet >> > >> > does it help when you use hda-analyzer to mute the amp-in value from node >> > 0x0b in 0x0c ? >> > >> > >> > Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out >> > Control: name="Front Playback Volume", index=0, device=0 >> > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 >> > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> > Amp-In vals: [0x00 0x00] [0x00 0x00] >> > Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 >> > Amp-Out vals: [0x33 0x33] >> > Connection: 2 >> > 0x02 0x0b >> > >> > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >

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[alsa-devel] MAYA44E
by c0 01 Dec '10

01 Dec '10

Hi. My name is Jose. The maya44 (pci) seems to be supported in snd-ice17xx drivers. But the new maya44e (pci express) is "only" recognized in my system as: root@debian:/home/co# lspci -vvv -s 04:00.0 04:00.0 Multimedia audio controller: Device 19fe:7000 (rev 03) Subsystem: Device 0e51:0001 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 (40000ns max) Interrupt: pin A routed to IRQ 7 Region 0: Memory at fd9fc000 (32-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- I would like to help to make this card work. Best regards. Jose.

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