May 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: ASoC: Fix WM9713 mic 2 preamp volume
by Matt Reimer 17 May '14

17 May '14

Fix WM9713's mic 2 preamp volume. The previous definition specified the Mic A pre-amp input select bits (aka "Mic A Source") instead. Signed-off-by: Matt Reimer <mreimer(a)sdgsystems.com> --- sound/soc/codecs/wm9713.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index acea892..5f69b52 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -134,7 +134,7 @@ SOC_DOUBLE_TLV("PCM Playback Volume", AC97_PHONE, 8, 0, 31, 1, main_tlv), SOC_SINGLE_TLV("Mic 1 Volume", AC97_MIC, 8, 31, 1, main_tlv), SOC_SINGLE_TLV("Mic 2 Volume", AC97_MIC, 0, 31, 1, main_tlv), SOC_SINGLE_TLV("Mic 1 Preamp Volume", AC97_3D_CONTROL, 10, 3, 0, mic_tlv), -SOC_SINGLE_TLV("Mic 2 Preamp Volume", AC97_3D_CONTROL, 12, 3, 0, mic_tlv), +SOC_SINGLE_TLV("Mic 2 Preamp Volume", AC97_3D_CONTROL, 8, 3, 0, mic_tlv), SOC_SINGLE("Mic Boost (+20dB) Switch", AC97_LINE, 5, 1, 0), SOC_SINGLE("Mic Headphone Mixer Volume", AC97_LINE, 0, 7, 1), -- 1.7.5.4

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[alsa-devel] [PATCH 0/7] ASoC: max98090: Couple improvements and ACPI probing support
by Jarkko Nikula 16 May '14

16 May '14

Here are couple improvements to MAX98090 codec driver. I think only 1/7 is a real fix and rest 2-7/7 are regular updates that make codec resume working and add ACPI probing support on Intel Baytrail platform. Jarkko Nikula (2): ASoC: max98090: Move microphone bias voltage setting to probe function ASoC: max98090: Add ACPI probing support Liam Girdwood (5): ASoC: max98090: Fix digital sidetone gain TLV ASoC: max98090: Fix reset at resume time. ASoC: max98090: Reset codec on resume ASoC: max98090: Mark cache as dirty prior to restoring ASoC: max98090: Make sure we configure BCLK in one place sound/soc/codecs/max98090.c | 77 +++++++++++++++++++++++++++++++++++++-------- sound/soc/codecs/max98090.h | 1 + 2 files changed, 65 insertions(+), 13 deletions(-) -- 2.0.0.rc0

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[alsa-devel] [PATCH v3 0/3] ASoC: Intel: rework the mfld platform driver
by Vinod Koul 16 May '14

16 May '14

This is the remaining bits of the v2 which were removed last time around. One patch was wrongly applied so reverting that and adding two other missed in the series. Also added Liam's ack Vinod Koul (3): Revert "ASoC: Intel: rename pcm dias to media dai" ASoC: Intel: add drain_notify support ASoC: Intel: remove codec memeber from codec structs sound/soc/intel/sst-mfld-dsp.h | 4 -- sound/soc/intel/sst-mfld-platform-compress.c | 13 +++++++- sound/soc/intel/sst-mfld-platform-pcm.c | 39 ++++++++++++------------- sound/soc/intel/sst-mfld-platform.h | 2 + 4 files changed, 32 insertions(+), 26 deletions(-)

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[alsa-devel] [PATCH 1/2] ALSA: hda - Add fixup_forced flag
by David Henningsson 16 May '14

16 May '14

The "fixup_forced" flag will indicate whether a specific fixup (or nofixup) has been set by the user, to override the driver's default. This flag will help future patches. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 9 ++++++--- sound/pci/hda/hda_codec.h | 1 + 2 files changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 90d2fda..36961ab 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -852,15 +852,17 @@ void snd_hda_pick_fixup(struct hda_codec *codec, if (codec->modelname && !strcmp(codec->modelname, "nofixup")) { codec->fixup_list = NULL; codec->fixup_id = -1; + codec->fixup_forced = 1; return; } if (codec->modelname && models) { while (models->name) { if (!strcmp(codec->modelname, models->name)) { - id = models->id; - name = models->name; - break; + codec->fixup_id = models->id; + codec->fixup_name = models->name; + codec->fixup_forced = 1; + return; } models++; } @@ -889,6 +891,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec, } } + codec->fixup_forced = 0; codec->fixup_id = id; if (id >= 0) { codec->fixup_list = fixlist; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index a423313..5825aa1 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -402,6 +402,7 @@ struct hda_codec { /* fix-up list */ int fixup_id; + unsigned int fixup_forced:1; /* fixup explicitly set by user */ const struct hda_fixup *fixup_list; const char *fixup_name; -- 1.9.1

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[alsa-devel] poor i4600 HDMI output under 1.0.27.2
by Dan MacDonald 15 May '14

15 May '14

Hi list I have got HDMI audio out to kinda work but the sound quality is awful and not really listenable. Its very choppy and is slightly more high-pitched than it should be. This is when using hdmi:CARD=HDMI,DEV=1 @48khz outputting via ALSA 1.0.27.2 / jackd 0.124.2 under Deb Jessie amd64 with kernel 3.13.10-1. Its a System76 Gazelle Pro laptop FWIW. $ lspci -v: 00:03.0 Audio device: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor HD Audio Controller (rev 06) Subsystem: CLEVO/KAPOK Computer Device 0655 Flags: bus master, fast devsel, latency 0, IRQ 49 Memory at f7e14000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel 00:1b.0 Audio device: Intel Corporation 8 Series/C220 Series Chipset High Definition Audio Controller (rev 05) Subsystem: CLEVO/KAPOK Computer Device 0655 Flags: bus master, fast devsel, latency 0, IRQ 48 Memory at f7e10000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel $ cat /proc/asound/card0/codec#0 Codec: Intel Haswell HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x80862807 Subsystem Id: 0x80860101 Revision Id: 0x100000 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 State of AFG node 0x01: Power states: D0 D3 CLKSTOP EPSS Power: setting=D0, actual=D0, Clock-stop-OK GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D3, actual=D3 Node 0x04 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D3, actual=D3 Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="HDMI/DP,pcm=3 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="ELD", index=0, device=3 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=01, enabled=1 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 In-driver Connection: 3 0x02 0x03 0x04 Node 0x06 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="HDMI/DP,pcm=7 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=1, device=0 Control: name="IEC958 Playback Pro Mask", index=1, device=0 Control: name="IEC958 Playback Default", index=1, device=0 Control: name="IEC958 Playback Switch", index=1, device=0 Control: name="ELD", index=0, device=7 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=02, enabled=1 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 In-driver Connection: 3 0x02 0x03 0x04 Node 0x07 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="HDMI/DP,pcm=8 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=2, device=0 Control: name="IEC958 Playback Pro Mask", index=2, device=0 Control: name="IEC958 Playback Default", index=2, device=0 Control: name="IEC958 Playback Switch", index=2, device=0 Control: name="ELD", index=0, device=8 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=03, enabled=1 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 In-driver Connection: 3 0x02 0x03 0x04 Node 0x08 [Vendor Defined Widget] wcaps 0xf00000: Mono

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[alsa-devel] [PATCH] ak4642 codec updates
by Sascha Hauer 15 May '14

15 May '14

This series has some updates to the ak4642 codec driver like ALC controls, improved sysclk support for the ak4648. Sascha ---------------------------------------------------------------- Sascha Hauer (6): ASoC: ak4642: Fix typo zoro -> zero ASoC: ak4642: Add ALC controls ASoC: ak4642: Add driver data and driver private struct ASoC: ak4642: Add support for extended sysclk frequencies of the ak4648 ASoC: ak4642: Implement Microphone in pathes ASoC: ak4642: Add enable gpio support Documentation/devicetree/bindings/sound/ak4642.txt | 3 + sound/soc/codecs/ak4642.c | 230 +++++++++++++++++++-- 2 files changed, 221 insertions(+), 12 deletions(-)

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[alsa-devel] [PATCH] ASoC: rsnd: Fix warnings due to improper printk formats
by Laurent Pinchart 15 May '14

15 May '14

Use the %pap printk specifier to print resource_size_t variables. This fixes warnings on platforms where resource_size_t has a different size than int. Signed-off-by: Laurent Pinchart <laurent.pinchart+renesas(a)ideasonboard.com> --- sound/soc/sh/rcar/gen.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) Would it be possible to get this patch in v3.16 ? diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 50a1ef3..1d5c025 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -252,13 +252,13 @@ static int rsnd_gen2_probe(struct platform_device *pdev, return ret; dev_dbg(dev, "Gen2 device probed\n"); - dev_dbg(dev, "SCU : %08x => %p\n", scu_res->start, + dev_dbg(dev, "SCU : %pap => %p\n", &scu_res->start, gen->base[RSND_GEN2_SCU]); - dev_dbg(dev, "ADG : %08x => %p\n", adg_res->start, + dev_dbg(dev, "ADG : %pap => %p\n", &adg_res->start, gen->base[RSND_GEN2_ADG]); - dev_dbg(dev, "SSIU : %08x => %p\n", ssiu_res->start, + dev_dbg(dev, "SSIU : %pap => %p\n", &ssiu_res->start, gen->base[RSND_GEN2_SSIU]); - dev_dbg(dev, "SSI : %08x => %p\n", ssi_res->start, + dev_dbg(dev, "SSI : %pap => %p\n", &ssi_res->start, gen->base[RSND_GEN2_SSI]); return 0; @@ -345,11 +345,11 @@ static int rsnd_gen1_probe(struct platform_device *pdev, return ret; dev_dbg(dev, "Gen1 device probed\n"); - dev_dbg(dev, "SRU : %08x => %p\n", sru_res->start, + dev_dbg(dev, "SRU : %pap => %p\n", &sru_res->start, gen->base[RSND_GEN1_SRU]); - dev_dbg(dev, "ADG : %08x => %p\n", adg_res->start, + dev_dbg(dev, "ADG : %pap => %p\n", &adg_res->start, gen->base[RSND_GEN1_ADG]); - dev_dbg(dev, "SSI : %08x => %p\n", ssi_res->start, + dev_dbg(dev, "SSI : %pap => %p\n", &ssi_res->start, gen->base[RSND_GEN1_SSI]); return 0; -- Regards, Laurent Pinchart

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[alsa-devel] [PATCH] ASoC: Intel: Fix simultaneous Baytrail SST capture and playback
by Jarkko Nikula 15 May '14

15 May '14

I managed to drop a change to stream ID setting from commit 49fee1781634 ("ASoC: Intel: Only export one Baytrail DAI") leading to non-working simultaneous capture-playback since after one DAI conversion rtd->cpu_dai->id + 1 will be the same for both playback and capture. Use substream->stream + 1 like it was in original Liam's patch. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/sst-baytrail-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c index 23a25c0627bd..6242ccce2bb7 100644 --- a/sound/soc/intel/sst-baytrail-pcm.c +++ b/sound/soc/intel/sst-baytrail-pcm.c @@ -254,7 +254,7 @@ static int sst_byt_pcm_open(struct snd_pcm_substream *substream) snd_soc_set_runtime_hwparams(substream, &sst_byt_pcm_hardware); - pcm_data->stream = sst_byt_stream_new(byt, rtd->cpu_dai->id + 1, + pcm_data->stream = sst_byt_stream_new(byt, substream->stream + 1, byt_notify_pointer, pcm_data); if (pcm_data->stream == NULL) { dev_err(rtd->dev, "failed to create stream\n"); -- 2.0.0.rc0

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[alsa-devel] [Resend] Add the stream report feature. Could you review my idea?
by JongHo Kim 15 May '14

15 May '14

(Add the alsa-devel(a)alsa-project.org) Hi I am work on ASoC part and developing the smartphone(Android). we have been doing the stress and long runtime test. audio output is no problem but I sometimes face the "playback write error(DMA or IRQ trouble?)" error message. I know the meaning of the message and I can solve the error. the problem may be caused by AP(clk) or I2S or DMA or Codec. I would like to trace the hardware error at the printed message. Could you review my idea? Thanks --- include/sound/pcm.h | 2 ++ include/sound/soc-dai.h | 2 ++ include/sound/soc.h | 1 + sound/core/pcm_lib.c | 3 +++ sound/core/pcm_native.c | 51 +++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/soc-pcm.c | 26 +++++++++++++++++++++++++ 6 files changed, 85 insertions(+) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index b4d6697..cd3093a 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -82,6 +82,7 @@ struct snd_pcm_ops { unsigned long offset); int (*mmap)(struct snd_pcm_substream *substream, struct vm_area_struct *vma); int (*ack)(struct snd_pcm_substream *substream); + void (*report)(struct snd_pcm_substream *substream); }; /* @@ -502,6 +503,7 @@ int snd_pcm_status(struct snd_pcm_substream *substream, int snd_pcm_start(struct snd_pcm_substream *substream); int snd_pcm_stop(struct snd_pcm_substream *substream, snd_pcm_state_t status); int snd_pcm_drain_done(struct snd_pcm_substream *substream); +void snd_pcm_report(struct snd_pcm_substream *substream); #ifdef CONFIG_PM int snd_pcm_suspend(struct snd_pcm_substream *substream); int snd_pcm_suspend_all(struct snd_pcm *pcm); diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index fad7676..c032393 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -184,6 +184,8 @@ struct snd_soc_dai_ops { struct snd_soc_dai *); int (*bespoke_trigger)(struct snd_pcm_substream *, int, struct snd_soc_dai *); + void (*report)(struct snd_pcm_substream *, + struct snd_soc_dai *); /* * For hardware based FIFO caused delay reporting. * Optional. diff --git a/include/sound/soc.h b/include/sound/soc.h index 0b83168..7685839 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -641,6 +641,7 @@ struct snd_soc_ops { int (*hw_free)(struct snd_pcm_substream *); int (*prepare)(struct snd_pcm_substream *); int (*trigger)(struct snd_pcm_substream *, int); + void (*report)(struct snd_pcm_substream *); }; struct snd_soc_compr_ops { diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index ce83def..0d6c9a3 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1944,6 +1944,9 @@ static int wait_for_avail(struct snd_pcm_substream *substream, "%s write error (DMA or IRQ trouble?)\n", is_playback ? "playback" : "capture"); err = -EIO; + snd_pcm_stream_unlock_irq(substream); + snd_pcm_report(substream); + snd_pcm_stream_lock_irq(substream); break; } } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index b653ab0..0ccdc10 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -978,6 +978,57 @@ int snd_pcm_drain_done(struct snd_pcm_substream *substream) SNDRV_PCM_STATE_SETUP); } +void snd_pcm_report(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + if (!runtime) { + snd_printd("runtime disappear\n"); + return; + } + + mutex_lock(&substream->pcm->open_mutex); + snd_printd("access: %i\n", runtime->access); + snd_printd("format: %i\n", runtime->format); + snd_printd("subformat: %d\n", runtime->subformat); + snd_printd("channels: %u\n", runtime->channels); + snd_printd("rate: %u (%u/%u)\n", runtime->rate, runtime->rate_num, + runtime->rate_den); + snd_printd("period_size: %lu\n", runtime->period_size); + snd_printd("buffer_size: %lu\n", runtime->buffer_size); + snd_printd("tstamp_mode: %i\n", runtime->tstamp_mode); + snd_printd("period_step: %u\n", runtime->period_step); + snd_printd("avail_min: %lu\n", runtime->control->avail_min); + snd_printd("start_threshold: %lu\n", runtime->start_threshold); + snd_printd("stop_threshold: %lu\n", runtime->stop_threshold); + snd_printd("silence_threshold: %lu\n", runtime->silence_threshold); + snd_printd("silence_size: %lu\n", runtime->silence_size); + snd_printd("boundary: %lu\n", runtime->boundary); + snd_printd("state: %i\n", runtime->status->state); + snd_printd("owner_pid: %d\n", pid_vnr(substream->pid)); + snd_printd("trigger_time: %ld.%09ld\n", + runtime->trigger_tstamp.tv_sec, + runtime->trigger_tstamp.tv_nsec); + snd_printd("tstamp: %ld.%09ld\n", + runtime->status->tstamp.tv_sec, + runtime->status->tstamp.tv_nsec); + snd_printd("delay: %ld\n", runtime->delay); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_printd("avail: %ld\n", snd_pcm_playback_avail(runtime)); + else + snd_printd("avail: %ld\n", snd_pcm_capture_avail(runtime)); + snd_printd("avail_max: %ld\n", runtime->avail_max); + snd_printd("hw_ptr: %ld\n", runtime->status->hw_ptr); + snd_printd("appl_ptr: %ld\n", runtime->control->appl_ptr); + + mutex_unlock(&substream->pcm->open_mutex); + + if (substream->ops->report) + substream->ops->report(substream); + + return; +} + /* * pause callbacks */ diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2cedf09..98a47b3 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -883,6 +883,30 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) return offset; } +static void soc_pcm_report(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + /* report the audio subsystem */ + if (rtd->dai_link->ops && rtd->dai_link->ops->report) + rtd->dai_link->ops->report(substream); + + if (cpu_dai->driver->ops->report) + cpu_dai->driver->ops->report(substream, cpu_dai); + + if (platform->driver->ops && platform->driver->ops->report) + platform->driver->ops->report(substream); + + if (codec_dai->driver->ops->report) + codec_dai->driver->ops->report(substream, codec_dai); + + mutex_unlock(&rtd->pcm_mutex); +} + /* connect a FE and BE */ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) @@ -2267,6 +2291,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) rtd->ops.close = dpcm_fe_dai_close; rtd->ops.pointer = soc_pcm_pointer; rtd->ops.ioctl = soc_pcm_ioctl; + rtd->ops.report = soc_pcm_report; } else { rtd->ops.open = soc_pcm_open; rtd->ops.hw_params = soc_pcm_hw_params; @@ -2276,6 +2301,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) rtd->ops.close = soc_pcm_close; rtd->ops.pointer = soc_pcm_pointer; rtd->ops.ioctl = soc_pcm_ioctl; + rtd->ops.report = soc_pcm_report; } if (platform->driver->ops) { -- 1.8.1.2

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[alsa-devel] [PATCH 0/2] ASoC: Intel: Export Baytrail PCM streams under one DAI
by Jarkko Nikula 14 May '14

14 May '14

It wasn't necessary to export Baytrail SST PCM streams under separate DAI links and complicated needlessly basic ALSA command usage (for instance arecord). This set exports PCM playback/capture streams under one DAI. Change should be safe now since known users are already converted and to my understanding no other distribution is using this yet (mainly due missing firmware). Liam Girdwood (2): ASoC: Intel: Make Baytrail PCM data per stream rather than per DAI device ASoC: Intel: Only export one Baytrail DAI sound/soc/intel/byt-rt5640.c | 14 +------------- sound/soc/intel/sst-baytrail-pcm.c | 23 ++++++++++------------- 2 files changed, 11 insertions(+), 26 deletions(-) -- 2.0.0.rc0

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