February 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/2] ASoC: compress - add support for metadata apis
by Vinod Koul 22 Mar '13

22 Mar '13

From: Jeeja KP <jeeja.kp(a)intel.com> Compress core added metadata apis in 9727b4, so add same in ASoC Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- Mark, Takashi: How do you guys wnat to take this one, Ideally this should go to mark's tree but the dependent apis are in takashi's tree, either way is fine with me :) --- sound/soc/soc-compress.c | 27 +++++++++++++++++++++++++++ 1 files changed, 27 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 3ea7956..4730789 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -302,11 +302,38 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream, return 0; } +static int sst_compr_set_metadata(struct snd_compr_stream *cstream, + struct snd_compr_metadata *metadata) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + int ret = 0; + + if (platform->driver->compr_ops && platform->driver->compr_ops->pointer) + ret = platform->driver->compr_ops->set_metadata(cstream, metadata); + + return ret; +} + +static int sst_compr_get_metadata(struct snd_compr_stream *cstream, + struct snd_compr_metadata *metadata) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + int ret = 0; + + if (platform->driver->compr_ops && platform->driver->compr_ops->pointer) + ret = platform->driver->compr_ops->get_metadata(cstream, metadata); + + return ret; +} /* ASoC Compress operations */ static struct snd_compr_ops soc_compr_ops = { .open = soc_compr_open, .free = soc_compr_free, .set_params = soc_compr_set_params, + .set_metadata = sst_compr_set_metadata, + .get_metadata = sst_compr_get_metadata, .get_params = soc_compr_get_params, .trigger = soc_compr_trigger, .pointer = soc_compr_pointer, -- 1.7.0.4

2 7

[alsa-devel] [PATCH] ASoC: core: Add support for platform and CODEC drivers on same device
by Charles Keepax 15 Mar '13

15 Mar '13

Currently DAI playback and capture widgets are created during soc_probe_codec and soc_probe_platform, using the device associated with the DAI to check which widgets should be created. If a device registers both a CODEC and platform driver this leads the CODEC playback and capture widgets being overwritten by the widgets created by the platform probe. It is more sensible to retain the CODEC widgets as the most common use case for registering both a CODEC and platform driver on the same chip is a CODEC which contains a DSP for compressed playback. In this situation it is more sensible to attach the routing information to the CODEC and add a thin platform driver interface to link into the compressed API. So this patch will check for existing widgets during soc_probe_platform and only create new widgets if no existing ones exist. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 04af2a6..53efe1d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1159,7 +1159,8 @@ static int soc_probe_platform(struct snd_soc_card *card, /* Create DAPM widgets for each DAI stream */ list_for_each_entry(dai, &dai_list, list) { - if (dai->dev != platform->dev) + if (dai->dev != platform->dev || + dai->playback_widget || dai->capture_widget) continue; snd_soc_dapm_new_dai_widgets(&platform->dapm, dai); -- 1.7.2.5

2 6

[alsa-devel] davici-mcasp: "tx-num-evt" confusion with number of serializers
by Michal Bachraty 11 Mar '13

11 Mar '13

Hi, I'm working on multichannel version of davinci-mcasp and also davinci-pcm. I have first version running and now I want to refine code. I found one confusion in davinci-mcasp with using of DT property "tx-num-evt". In DT binding documentation "tx-num-evt" is defined as "FIFO levels", but in Mcasp src, there is code, which mixes tx-num-evt with number of serializers (i2s data lines) that are enabled for data playback and receive (dev->txnumevt * tx_ser) mcasp_mod_bits(dev->base + MCASP_VER3_WFIFOCTL, (((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK); >From dacinci pcm, DMA data tranfer use txnumevt as number of serializers and also for data prefetching. I undestand definition "FIFO levels" as how many prefetched data are in FIFO. Prefetched data are for me frame data / 2 ( all left or right channels for one sampling time). This results me from AM335x FIFO documantation. What is original purpose for "tx-num-evt" parameter? Thanks, Michal

5 12

[alsa-devel] HDA regression on Fujitsu S7020 laptop (ALC260 codec)
by Jonathan Woithe 07 Mar '13

07 Mar '13

Having been running kernel 3.3.5 for a while, I decided to upgrade to 3.7.1 tonight. Unfortunately there have been a number of HDA-related regressions between these two kernels when running on a Fujitsu S7020 laptop which are clearly related to the HDA rewrite which has been going on recently. This laptop utilises the Realtek ALC260 codec. Testing was initially done with alsa-lib 1.0.24 and alsa-utils 1.0.24. I upgraded to version 1.0.26 of both but the behaviour remained the same under 3.3.5 and 3.7.1. Under 3.3.5, sound works perfectly. Under 3.7.1 there are a number of regressions: - there is no alsamixer control for the internal speaker. I have a "PCM" volume control which appears to do nothing. Previously there was a speaker control, the PCM control affected every output (from memory), as did the master volume. - no audio comes out of the internal speaker regardless of the "auto mute" setting. - there are no pin mode (re-tasking) controls for the mic or headphone jack in alsamixer. On this laptop the mic jack should have various bias options as well as line-in. The headphone jack should include headphone (aka output amplifier on), line-out (output amplifier off), line-in and mic options. The second ADC does show up in alsamixer as a second capture device, but probably due to the lack of retasking support it is not possible to select the headphone jack as an input option. Audio does come out of the headphone jack. However, as noted above it is not possible to switch this from headphone to line-out. The volume of the headphone output is affected by the "master" and "headphone" volume controls. The inability to set the pin modes is a showstopper for me - I need to be able to do this and thus will be stuck on older kernels until this is fixed. At the end of this post I have included the /proc/asound/card0/codec#0 output under 3.3.5 (which works well) and 3.7.1 (which is broken as described above). What needs to be done to restore the functionality for the S7020 that has been working for years? It seems that some form of model quirk needs to be ported into the new HDA framework from earlier kernels, but since this sort of thing appears to have been removed with the rewrite (at least for realtek codecs) it's difficult to see where to begin. Regards jonathan Output from /proc/asound/card0/codec#0 under kernel 3.3.5: Codec: Realtek ALC260 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0260 Subsystem Id: 0x10cf0000 Revision Id: 0x100400 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=4, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x11: Stereo Device: name="ALC260 Analog", type="Audio", device=0 Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x04 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Input Source", index=0, device=0 Control: name="Capture Switch", index=0, device=0 Control: name="Capture Volume", index=0, device=0 Device: name="ALC260 Analog", type="Audio", device=0 Amp-In caps: ofs=0x00, nsteps=0x23, stepsize=0x03, mute=1 Amp-In vals: [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 7 0x12* 0x13 0x14 0x15 0x16 0x0f 0x10 Node 0x05 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Input Source", index=1, device=0 Control: name="Capture Switch", index=1, device=0 Control: name="Capture Volume", index=1, device=0 Device: name="ALC260 Analog", type="Audio", device=2 Amp-In caps: ofs=0x00, nsteps=0x23, stepsize=0x03, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 8 0x12* 0x13 0x14 0x15 0x16 0x07 0x0f 0x10 Node 0x06 [Audio Input] wcaps 0x100391: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Unsolicited: tag=00, enabled=0 Connection: 1 0x19 Node 0x07 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="CD Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Control: name="CD Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Control: name="Mic/Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic/Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x23, nsteps=0x41, stepsize=0x03, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x03 0x03] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 8 0x12 0x13 0x14 0x15 0x16 0x17 0x0f 0x10 Node 0x08 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Headphone 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: [0x27 0x27] Connection: 2 0x02 0x07 Node 0x09 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Speaker 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: [0x80 0x80] [0x80 0x80] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x02 0x07 Node 0x0a [Audio Mixer] wcaps 0x20010e: Mono Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80] [0x80] Amp-Out caps: ofs=0x23, nsteps=0x41, stepsize=0x03, mute=0 Amp-Out vals: [0x00] Connection: 2 0x02 0x07 Node 0x0b [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x08* 0x09 Node 0x0c [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x08* 0x09 Node 0x0d [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x08* 0x09 Node 0x0e [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x08* 0x09 Node 0x0f [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001003f: IN OUT HP EAPD Detect Trigger ImpSense EAPD 0x2: EAPD Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x08 Node 0x10 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001003f: IN OUT HP EAPD Detect Trigger ImpSense EAPD 0x2: EAPD Pin Default 0xe4011110: [Both] Line Out at Sep Right Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0xc0: OUT HP Unsolicited: tag=00, enabled=0 Connection: 1 0x09 Node 0x11 [Pin Complex] wcaps 0x40010c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Connection: 1 0x0a Node 0x12 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Mic/Line Jack Mode", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x03a11820: [Jack] Mic at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x0b Node 0x13 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x24811121: [Jack] Line In at Sep Right Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x1 Misc = NO_PRESENCE Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x0c Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Headphone Jack Mode", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x0321101f: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0x40: OUT VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x0d Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x0e Node 0x16 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x88331122: [Fixed] CD at Ext Drive Bar Conn = ATAPI, Color = Black DefAssociation = 0x2, Sequence = 0x2 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x17 [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0xb7931123: [Fixed] Aux at Oth Mobile-In Conn = ATAPI, Color = Black DefAssociation = 0x2, Sequence = 0x3 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x18 [Pin Complex] wcaps 0x400380: Mono Digital Pincap 0x00000014: OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x03 Node 0x19 [Pin Complex] wcaps 0x400280: Mono Digital Pincap 0x00000024: IN Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Node 0x1a [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=13 Node 0x1b [Volume Knob Widget] wcaps 0x600080: Mono Volume-Knob: delta=0, steps=64, direct=0, val=0 Unsolicited: tag=00, enabled=0 Connection: 0 Output from /proc/asound/card0/codec#0 under kernel 3.7.1: Codec: Realtek ALC260 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0260 Subsystem Id: 0x10cf0000 Revision Id: 0x100400 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=4, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x11: Stereo Device: name="ALC260 Analog", type="Audio", device=0 Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x04 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Input Source", index=0, device=0 Control: name="Capture Switch", index=0, device=0 Control: name="Capture Volume", index=0, device=0 Device: name="ALC260 Analog", type="Audio", device=0 Amp-In caps: ofs=0x00, nsteps=0x23, stepsize=0x03, mute=1 Amp-In vals: [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 7 0x12* 0x13 0x14 0x15 0x16 0x0f 0x10 Node 0x05 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Input Source", index=1, device=0 Control: name="Capture Switch", index=1, device=0 Control: name="Capture Volume", index=1, device=0 Device: name="ALC260 Analog", type="Audio", device=2 Amp-In caps: ofs=0x00, nsteps=0x23, stepsize=0x03, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 8 0x12* 0x13 0x14 0x15 0x16 0x07 0x0f 0x10 Node 0x06 [Audio Input] wcaps 0x100391: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Unsolicited: tag=00, enabled=0 Connection: 1 0x19 Node 0x07 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="CD Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Control: name="CD Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Amp-In caps: ofs=0x23, nsteps=0x41, stepsize=0x03, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x03 0x03] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 8 0x12 0x13 0x14 0x15 0x16 0x17 0x0f 0x10 Node 0x08 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Headphone 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: [0x27 0x27] Connection: 2 0x02 0x07 Node 0x09 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="PCM 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 0x07 Node 0x0a [Audio Mixer] wcaps 0x20010e: Mono Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00] [0x80] Amp-Out caps: ofs=0x23, nsteps=0x41, stepsize=0x03, mute=0 Amp-Out vals: [0x23] Connection: 2 0x02 0x07 Node 0x0b [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x08* 0x09 Node 0x0c [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x08* 0x09 Node 0x0d [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x08* 0x09 Node 0x0e [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x08* 0x09 Node 0x0f [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001003f: IN OUT HP EAPD Detect Trigger ImpSense EAPD 0x2: EAPD Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 1 0x08 Node 0x10 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="PCM Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Phantom Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001003f: IN OUT HP EAPD Detect Trigger ImpSense EAPD 0x2: EAPD Pin Default 0xe4011110: [Both] Line Out at Sep Right Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x09 Node 0x11 [Pin Complex] wcaps 0x40010c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x0a Node 0x12 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Mic Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x03a11820: [Jack] Mic at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=03, enabled=1 Connection: 1 0x0b Node 0x13 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Line Phantom Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x24811121: [Jack] Line In at Sep Right Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x1 Misc = NO_PRESENCE Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x0c Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x0321101f: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=01, enabled=1 Connection: 1 0x0d Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x0e Node 0x16 [Pin Complex] wcaps 0x400001: Stereo Control: name="CD Phantom Jack", index=0, device=0 Pincap 0x00000020: IN Pin Default 0x88331122: [Fixed] CD at Ext Drive Bar Conn = ATAPI, Color = Black DefAssociation = 0x2, Sequence = 0x2 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x17 [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0xb7931123: [Fixed] Aux at Oth Mobile-In Conn = ATAPI, Color = Black DefAssociation = 0x2, Sequence = 0x3 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x18 [Pin Complex] wcaps 0x400380: Mono Digital Pincap 0x00000014: OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x03 Node 0x19 [Pin Complex] wcaps 0x400280: Mono Digital Pincap 0x00000024: IN Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Node 0x1a [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=13 Node 0x1b [Volume Knob Widget] wcaps 0x600080: Mono Volume-Knob: delta=0, steps=64, direct=0, val=0 Unsolicited: tag=00, enabled=0 Connection: 0

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[alsa-devel] ASoC: no-pcm (backend) error propagation
by Frkuska, Joshua 05 Mar '13

05 Mar '13

Hello, I am working off of the 3.5.7 branch of the kernel using a frontend-backend approach for my driver. No additional patches have been added for ASoC/ALSA from the 3.5.7 kernel branch. My question relates to the way the no-pcm (backend) propagates errors to the front end. My FE(frontend) consists of a cpu-dai, a pcm platform driver, and a dummy-codec. My BE(backend) consists of another cpu-dai, a non-pcm platform driver, and a real codec. The non-pcm operates with the dmaengine but just in an autonomous fashion external to the PCM's knowledge. I have the pcm-platform driver passing data to the FE cpu-dai and the backend platform then takes the data from the frontend's cpu-dai and passes it on to the BE cpu-dai, which then configures the codec-dai to get output. In the frontend if/whenever there is an underrun error in the cpu-dai, I can directly call snd_pcm_stop to stop the stream and report the error state. I can do this because the substream->ops have properly been defined in the frontend pcm. However, in the backend, the substream->ops never get initialized and rightfully so since there isn't a valid pcm for the backend. My question is, what is the correct/any way to let the frontend PCM stream know that an underrun has occurred in the backend? Is there a mechanism to handle this in ASoC? Looking through the code, I cannot find such a mechanism. I have a feeling that there should be something like snd_soc_pcm_stop to check to see if we are in a FE/BE configuration and act accordingly to stop the FE PCM stream and report errors. Again, perhaps such a mechanism exists and if so any help would be appreciated. Thanks for your time and any comments/suggestions, Joshua

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[alsa-devel] HDA Beep Device for Asus EeeBox EB1501P
by Harald Becker 05 Mar '13

05 Mar '13

Hi, I own an Asus EeeBox EB1501P with an ALC662 codec and had no HDA mixer channel for the speaker beep device. After digging somewhat in the kernel source I found that I only need to add a single line in the beep_white_list in patch_realtek.c. May be this entry can be included in the mainstream drivers: SND_PCI_QUIRK(0x1043, 0x8376, "EeePC", 1), After recompiling the driver with this entry, the mixer device for the beep channel appears and the digital beep sound is emitted on the connected speakers. ... with one problem: If I use echo -e "\a" in a shell or run the beep command utility the sound appears only on second and following invocations. Whenever the frequency changes with the beep command the first invocation stays silent or just produce some click due to ALSA driver power saving. On all further invocations with same frequency the requested tone is generated on external speakers. There is no difference in behavior if the commands are called in a terminal emulator or directly on a virtual text console. Does this happen only on my machine or is this a known problem? Is there anything I can do to fix this behavior? The kernel I used for testing is 3.8.0 from Gentoo Sources. Feel free to ask for any information you may need for debugging. Thx for any help ahead Harald

2 5

[alsa-devel] Issue with Dmix
by mateen 04 Mar '13

04 Mar '13

HI We have enabled DMIX plugin for sample rate conversion and mixing of 2 pcm streams in our alsa package (Alsa lib 1.0.22 and alsa driver 1.0.22.1) to 48K output Sample rate.However we are seeing a noise issue with 44100 Sample rate input streams or Unsigned Stereo 48K stream. This is how our current design is: We call snd_pcm_lib_malloc_page() in which alsa driver internally allocates a dma buffer and map it when Dmix is invoked. We currently reads from that mmaped buffer to our hardware buffer inside a transfer_ack_begin() function which is called from snd_pcm_period_elapsed(). However we are getting a problem in understanding how do we know how much data has been written by dmix lib in mapped buffer? As we see no ioctl or corresponding hardware function call where dmix may pass information on Size of data being written to our driver. We do see appl_ptr and hw_ptr in runtime->mmap->control and status struct. Out architecture doesn’t support defining and mapping of these status and control structure at hardware level.Thus Dmix lib allocates them and copy them to runtime substream via ioctl. As far as I could understand both appl_ptr and hw_ptr are initialised to 0 and then appl_ptr is the one adjusted by lib after every write. Should we tap this value to update hw_ptr value in status information? Will difference of hw_ptr and appl_ptr will let us know amount of frames been written? Can anyone please help us to understand criteria and concept of data copy from intermediate to hardware buffer? Any information on this would be of immense help. Regards, Shaikh Mateen

1 1

[alsa-devel] [PATCH 09/12] ASoC: dmaengine_pcm: support use of generic DMA helper
by Shawn Guo 04 Mar '13

04 Mar '13

With generic DMA device tree binding and helper function dma_request_slave_channel() in place, dmaengine_pcm should support that in requesting DMA channel for users that support generic DMA device tree binding. Instead of inventing a new API, it defines the parameters needed by dma_request_slave_channel() into struct snd_dma_channel_params, interprets filter_data into snd_dma_channel_params, and calls the helper in case that dmaengine_pcm users pass in a NULL filter_fn. Then, dmaengine_pcm users can call snd_dmaengine_pcm_open() with NULL filter_fn and snd_dma_channel_params being filter_data to direct the API to request DMA channel using generic DMA helper. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: alsa-devel(a)alsa-project.org --- include/sound/dmaengine_pcm.h | 5 +++++ sound/soc/soc-dmaengine-pcm.c | 19 +++++++++++++------ 2 files changed, 18 insertions(+), 6 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index b877334..6c0f795 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -18,6 +18,11 @@ #include <sound/pcm.h> #include <linux/dmaengine.h> +struct snd_dma_channel_params { + struct device *dev; + char *name; +}; + /** * snd_pcm_substream_to_dma_direction - Get dma_transfer_direction for a PCM * substream diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c index 111b7d92..7ef8034 100644 --- a/sound/soc/soc-dmaengine-pcm.c +++ b/sound/soc/soc-dmaengine-pcm.c @@ -247,12 +247,19 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer); static int dmaengine_pcm_request_channel(struct dmaengine_pcm_runtime_data *prtd, dma_filter_fn filter_fn, void *filter_data) { - dma_cap_mask_t mask; - - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); - dma_cap_set(DMA_CYCLIC, mask); - prtd->dma_chan = dma_request_channel(mask, filter_fn, filter_data); + if (filter_fn) { + dma_cap_mask_t mask; + + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + dma_cap_set(DMA_CYCLIC, mask); + prtd->dma_chan = dma_request_channel(mask, filter_fn, + filter_data); + } else { + struct snd_dma_channel_params *params = filter_data; + prtd->dma_chan = dma_request_slave_channel(params->dev, + params->name); + } if (!prtd->dma_chan) return -ENXIO; -- 1.7.9.5

3 4

[alsa-devel] [PATCH 1/2] ASoC: max98090: Fix checkpatch errors related to spacing
by Sachin Kamat 02 Mar '13

02 Mar '13

Fixes the following type of checkpatch errors: ERROR: "foo * bar" should be "foo *bar" Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/codecs/max98090.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 1cf017f..89f83f8 100755 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -507,16 +507,16 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, return 0; } -static const char * max98090_perf_pwr_text[] = +static const char *max98090_perf_pwr_text[] = { "High Performance", "Low Power" }; -static const char * max98090_pwr_perf_text[] = +static const char *max98090_pwr_perf_text[] = { "Low Power", "High Performance" }; static const struct soc_enum max98090_vcmbandgap_enum = SOC_ENUM_SINGLE(M98090_REG_BIAS_CONTROL, M98090_VCM_MODE_SHIFT, ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text); -static const char * max98090_osr128_text[] = { "64*fs", "128*fs" }; +static const char *max98090_osr128_text[] = { "64*fs", "128*fs" }; static const struct soc_enum max98090_osr128_enum = SOC_ENUM_SINGLE(M98090_REG_ADC_CONTROL, M98090_OSR128_SHIFT, @@ -533,28 +533,28 @@ static const struct soc_enum max98090_filter_dmic34mode_enum = M98090_FLT_DMIC34MODE_SHIFT, ARRAY_SIZE(max98090_mode_text), max98090_mode_text); -static const char * max98090_drcatk_text[] = +static const char *max98090_drcatk_text[] = { "0.5ms", "1ms", "5ms", "10ms", "25ms", "50ms", "100ms", "200ms" }; static const struct soc_enum max98090_drcatk_enum = SOC_ENUM_SINGLE(M98090_REG_DRC_TIMING, M98090_DRCATK_SHIFT, ARRAY_SIZE(max98090_drcatk_text), max98090_drcatk_text); -static const char * max98090_drcrls_text[] = +static const char *max98090_drcrls_text[] = { "8s", "4s", "2s", "1s", "0.5s", "0.25s", "0.125s", "0.0625s" }; static const struct soc_enum max98090_drcrls_enum = SOC_ENUM_SINGLE(M98090_REG_DRC_TIMING, M98090_DRCRLS_SHIFT, ARRAY_SIZE(max98090_drcrls_text), max98090_drcrls_text); -static const char * max98090_alccmp_text[] = +static const char *max98090_alccmp_text[] = { "1:1", "1:1.5", "1:2", "1:4", "1:INF" }; static const struct soc_enum max98090_alccmp_enum = SOC_ENUM_SINGLE(M98090_REG_DRC_COMPRESSOR, M98090_DRCCMP_SHIFT, ARRAY_SIZE(max98090_alccmp_text), max98090_alccmp_text); -static const char * max98090_drcexp_text[] = { "1:1", "2:1", "3:1" }; +static const char *max98090_drcexp_text[] = { "1:1", "2:1", "3:1" }; static const struct soc_enum max98090_drcexp_enum = SOC_ENUM_SINGLE(M98090_REG_DRC_EXPANDER, M98090_DRCEXP_SHIFT, @@ -857,7 +857,7 @@ static const struct soc_enum mic2_mux_enum = static const struct snd_kcontrol_new max98090_mic2_mux = SOC_DAPM_ENUM("MIC2 Mux", mic2_mux_enum); -static const char * max98090_micpre_text[] = { "Off", "On" }; +static const char *max98090_micpre_text[] = { "Off", "On" }; static const struct soc_enum max98090_pa1en_enum = SOC_ENUM_SINGLE(M98090_REG_MIC1_INPUT_LEVEL, M98090_MIC_PA1EN_SHIFT, -- 1.7.4.1

2 2

[alsa-devel] [PATCH] ASoC: soc-core: use snd_soc_register_dais() on codec
by Kuninori Morimoto 02 Mar '13

02 Mar '13

snd_soc_register_dais() considers dai counts inside. snd_soc_register_codec() does not need to care for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ceac893..5f44d22 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4047,11 +4047,10 @@ int snd_soc_register_codec(struct device *dev, mutex_unlock(&client_mutex); /* register any DAIs */ - if (num_dai) { - ret = snd_soc_register_dais(dev, dai_drv, num_dai); - if (ret < 0) - dev_err(codec->dev, "ASoC: Failed to regster" - " DAIs: %d\n", ret); + ret = snd_soc_register_dais(dev, dai_drv, num_dai); + if (ret < 0) { + dev_err(codec->dev, "ASoC: Failed to regster DAIs: %d\n", ret); + goto fail_codec_name; } dev_dbg(codec->dev, "ASoC: Registered codec '%s'\n", codec->name); @@ -4073,7 +4072,6 @@ EXPORT_SYMBOL_GPL(snd_soc_register_codec); void snd_soc_unregister_codec(struct device *dev) { struct snd_soc_codec *codec; - int i; list_for_each_entry(codec, &codec_list, list) { if (dev == codec->dev) @@ -4082,9 +4080,7 @@ void snd_soc_unregister_codec(struct device *dev) return; found: - if (codec->num_dai) - for (i = 0; i < codec->num_dai; i++) - snd_soc_unregister_dai(dev); + snd_soc_unregister_dais(dev, codec->num_dai); mutex_lock(&client_mutex); list_del(&codec->list); -- 1.7.9.5

2 1