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

[alsa-devel] [PATCH 0/7] ASoC: Intel: Baytrail SST suspend/resume support
by Jarkko Nikula 12 May '14

12 May '14

This set adds suspend/resume support to Baytrail SST. This is implemented by putting the DSP into reset and by releasing driver resources before suspend. In resume path the DSP is rebooted in normal boot state. Active streams are restored by restarting them from last known buffer position before suspend. Patches 1-5 are preparing for the suspend/resume and patch 6 implements the core logic that takes care of stopping the DSP before suspend and restarting it again after resume. Patch 7 removes ignore_suspend flag from byt-rt5640 machine driver since it doesn't have use at the moment and adds suspend/resume callbacks that are hooked to ASoC core. This takes care that active stream will be paused and resumed over suspend/resume cycle on byt-rt5640 machine. Jarkko Nikula (5): ASoC: Intel: Sample Baytrail DSP DMA pointer only after each period ASoC: Intel: Simplify Baytrail stream control IPC construction ASoC: Intel: Pass stream start position to sst_byt_stream_start() ASoC: Intel: Move Baytrail extended fw address saving to sst_byt_boot() ASoC: Intel: Allow byt-5640 machine driver and SST core go to suspend Liam Girdwood (2): ASoC: Intel: Allow Rx/Tx message list can be cleared prior to suspend ASoC: Intel: Add Baytrail suspend/resume support sound/soc/intel/byt-rt5640.c | 14 +++- sound/soc/intel/sst-baytrail-dsp.c | 14 ++-- sound/soc/intel/sst-baytrail-ipc.c | 124 ++++++++++++++++++++++++++++------ sound/soc/intel/sst-baytrail-ipc.h | 7 +- sound/soc/intel/sst-baytrail-pcm.c | 135 ++++++++++++++++++++++++++++++++----- 5 files changed, 249 insertions(+), 45 deletions(-) -- 2.0.0.rc0

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[alsa-devel] [PATCH] ASoC: dapm: Handle SND_SOC_DAPM_REG() generically
by Lars-Peter Clausen 12 May '14

12 May '14

Commit commit de9ba98b6d ("ASoC: dapm: Make widget power register settings more flexible") added generic support for on_val/off_val in the DAPM core. With this in place there is no need anymore for having a special event callback for SND_SOC_DAPM_REG() widgets. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc-dapm.h | 7 ++----- sound/soc/soc-dapm.c | 20 -------------------- 2 files changed, 2 insertions(+), 25 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 75020f5..5932d66 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -256,9 +256,8 @@ struct device; /* generic widgets */ #define SND_SOC_DAPM_REG(wid, wname, wreg, wshift, wmask, won_val, woff_val) \ { .id = wid, .name = wname, .kcontrol_news = NULL, .num_kcontrols = 0, \ - .reg = -((wreg) + 1), .shift = wshift, .mask = wmask, \ - .on_val = won_val, .off_val = woff_val, .event = dapm_reg_event, \ - .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD} + .reg = wreg, .shift = wshift, .mask = wmask, \ + .on_val = won_val, .off_val = woff_val, } #define SND_SOC_DAPM_SUPPLY(wname, wreg, wshift, winvert, wevent, wflags) \ { .id = snd_soc_dapm_supply, .name = wname, \ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ @@ -362,8 +361,6 @@ struct regulator; struct snd_soc_dapm_widget_list; struct snd_soc_dapm_update; -int dapm_reg_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event); int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); int dapm_clock_event(struct snd_soc_dapm_widget *w, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index fb6c7b7..56fcd6f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1059,26 +1059,6 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, } /* - * Handler for generic register modifier widget. - */ -int dapm_reg_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - unsigned int val; - - if (SND_SOC_DAPM_EVENT_ON(event)) - val = w->on_val; - else - val = w->off_val; - - soc_widget_update_bits(w, -(w->reg + 1), - w->mask << w->shift, val << w->shift); - - return 0; -} -EXPORT_SYMBOL_GPL(dapm_reg_event); - -/* * Handler for regulator supply widget. */ int dapm_regulator_event(struct snd_soc_dapm_widget *w, -- 1.8.0

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[alsa-devel] [PATCH] sound: soc: intel: remove unneeded dependency from Makefile
by Lad Prabhakar 12 May '14

12 May '14

From: "Lad, Prabhakar" <prabhakar.csengg(a)gmail.com> this patch removes drops the entries form Makefile, fixing following issues when make clean was run, scripts/Makefile.clean:17: sound/soc/intel/board/Makefile: No such file or directory scripts/Makefile.clean:17: sound/soc/intel/sst/Makefile: No such file or directory Signed-off-by: Lad, Prabhakar <prabhakar.csengg(a)gmail.com> --- Found this issue on linux-next not sure if this is fixed in sound subsystem tree. sound/soc/intel/Makefile | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 57242c4..26544c5 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -32,9 +32,3 @@ PLATFORM_LIBS = platform-libs/controls_v2_dpcm.o snd-soc-sst-platform-objs := pcm.o compress.o $(PLATFORM_LIBS) obj-$(CONFIG_SND_SST_PLATFORM) += snd-soc-sst-platform.o - -# Relevant Machine driver -obj-$(CONFIG_SND_SST_MACHINE) += board/ - -# IPC driver -obj-$(CONFIG_SND_INTEL_SST) += sst/ -- 1.7.9.5

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[alsa-devel] [PATCH v2 1/5] ASoC: dapm: Skip CODEC<->CODEC links in connect_dai_link_widgets()
by Lars-Peter Clausen 12 May '14

12 May '14

For CODEC to CODEC DAI links the paths are created in snd_soc_dapm_new_pcm(). Also for CODEC to CODEC links the widgets are connected cross-over via a DAI link widget, meaning that the capture widget of one CODEC will be connected to the playback widget of the other and vice versa. Whereas snd_soc_dapm_connect_dai_link_widgets() directly connects the playback widget of the CPU DAI to the playback widget of the CODEC DAI and the capture widget of the CPU DAI to the capture widget of the CODEC DAI. So not skipping CODEC<->CODEC links in snd_soc_dapm_connect_dai_link_widgets() will create incorrect connections between the two CODECs which will cause DAPM to detect active paths where there are none and unnecessarily power up widgets. Fixes: b893ea5 ("ASoC: sapm: Automatically connect DAI link widgets in DAPM graph.") Cc: <stable(a)vger.kernel.org> (for 3.14+) Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index fb6c7b7..142a738 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3410,8 +3410,11 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) cpu_dai = rtd->cpu_dai; codec_dai = rtd->codec_dai; - /* dynamic FE links have no fixed DAI mapping */ - if (rtd->dai_link->dynamic) + /* + * dynamic FE links have no fixed DAI mapping. + * CODEC<->CODEC links have no direct connection. + */ + if (rtd->dai_link->dynamic || rtd->dai_link->params) continue; /* there is no point in connecting BE DAI links with dummies */ -- 1.8.0

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[alsa-devel] [PATCH] ASoC: Fix incorrect condition check for case SNDRV_PCM_TRIGGER_SUSPEND
by Nicolin Chen 12 May '14

12 May '14

The regular state before we execute SNDRV_PCM_TRIGGER_SUSPEND should be SNDRV_PCM_TRIGGER_START, not SNDRV_PCM_TRIGGER_STOP. Thus fix it. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 88230ea..49930bd 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1666,7 +1666,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; break; case SNDRV_PCM_TRIGGER_SUSPEND: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) + if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) continue; if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) -- 1.8.4

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[alsa-devel] [PATCH RFT] ASoC: cs42l56: Fix update mute register bits in cs42l56_digital_mute
by Axel Lin 12 May '14

12 May '14

The new value in snd_soc_update_bits() needs to match the mask bit fields. Use CS42L56_MUTE/CS42L56_UNMUTE only works when the mask is 0x01. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- For example, *Both* below calls actually *clear* CS42L56_LO_MUTE_MASK bit (0x80). snd_soc_update_bits(codec, CS42L56_LOB_VOLUME, CS42L56_LO_MUTE_MASK, CS42L56_MUTE); snd_soc_update_bits(codec, CS42L56_LOB_VOLUME, CS42L56_LO_MUTE_MASK, CS42L56_UNMUTE); So I think use CS42L56_UNMUTE/CS42L56_MUTE is not good here. Hi Brian, I don't have this h/w, so this patch needs test. I'd appreciate if you can review and test this patch. Regards, Axel sound/soc/codecs/cs42l56.c | 65 ++++++++++++++++++---------------------------- sound/soc/codecs/cs42l56.h | 2 -- 2 files changed, 25 insertions(+), 42 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 5bb134b..ecb2fed 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -839,62 +839,47 @@ static int cs42l56_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) static int cs42l56_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; + unsigned int mask; if (mute) { /* Hit the DSP Mixer first */ - snd_soc_update_bits(codec, CS42L56_DSP_MUTE_CTL, - CS42L56_ADCAMIX_MUTE_MASK | - CS42L56_ADCBMIX_MUTE_MASK | - CS42L56_PCMAMIX_MUTE_MASK | - CS42L56_PCMBMIX_MUTE_MASK | - CS42L56_MSTB_MUTE_MASK | - CS42L56_MSTA_MUTE_MASK, - CS42L56_MUTE); + mask = CS42L56_ADCAMIX_MUTE_MASK | CS42L56_ADCBMIX_MUTE_MASK | + CS42L56_PCMAMIX_MUTE_MASK | CS42L56_PCMBMIX_MUTE_MASK | + CS42L56_MSTB_MUTE_MASK | CS42L56_MSTA_MUTE_MASK; + snd_soc_update_bits(codec, CS42L56_DSP_MUTE_CTL, mask, mask); + /* Mute ADC's */ - snd_soc_update_bits(codec, CS42L56_MISC_ADC_CTL, - CS42L56_ADCA_MUTE_MASK | - CS42L56_ADCB_MUTE_MASK, - CS42L56_MUTE); + mask = CS42L56_ADCA_MUTE_MASK | CS42L56_ADCB_MUTE_MASK; + snd_soc_update_bits(codec, CS42L56_MISC_ADC_CTL, mask, mask); + /* HP And LO */ snd_soc_update_bits(codec, CS42L56_HPA_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_MUTE); + CS42L56_HP_MUTE_MASK, CS42L56_HP_MUTE_MASK); snd_soc_update_bits(codec, CS42L56_HPB_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_MUTE); + CS42L56_HP_MUTE_MASK, CS42L56_HP_MUTE_MASK); snd_soc_update_bits(codec, CS42L56_LOA_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_MUTE); + CS42L56_LO_MUTE_MASK, CS42L56_LO_MUTE_MASK); snd_soc_update_bits(codec, CS42L56_LOB_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_MUTE); + CS42L56_LO_MUTE_MASK, CS42L56_LO_MUTE_MASK); } else { - snd_soc_update_bits(codec, CS42L56_DSP_MUTE_CTL, - CS42L56_ADCAMIX_MUTE_MASK | - CS42L56_ADCBMIX_MUTE_MASK | - CS42L56_PCMAMIX_MUTE_MASK | - CS42L56_PCMBMIX_MUTE_MASK | - CS42L56_MSTB_MUTE_MASK | - CS42L56_MSTA_MUTE_MASK, - CS42L56_UNMUTE); - snd_soc_update_bits(codec, CS42L56_MISC_ADC_CTL, - CS42L56_ADCA_MUTE_MASK | - CS42L56_ADCB_MUTE_MASK, - CS42L56_UNMUTE); + mask = CS42L56_ADCAMIX_MUTE_MASK | CS42L56_ADCBMIX_MUTE_MASK | + CS42L56_PCMAMIX_MUTE_MASK | CS42L56_PCMBMIX_MUTE_MASK | + CS42L56_MSTB_MUTE_MASK | CS42L56_MSTA_MUTE_MASK; + snd_soc_update_bits(codec, CS42L56_DSP_MUTE_CTL, mask , 0); + + mask = CS42L56_ADCA_MUTE_MASK | CS42L56_ADCB_MUTE_MASK; + snd_soc_update_bits(codec, CS42L56_MISC_ADC_CTL, mask, 0); + snd_soc_update_bits(codec, CS42L56_HPA_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_HP_MUTE_MASK, 0); snd_soc_update_bits(codec, CS42L56_HPB_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_HP_MUTE_MASK, 0); snd_soc_update_bits(codec, CS42L56_LOA_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_LO_MUTE_MASK, 0); snd_soc_update_bits(codec, CS42L56_LOB_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_LO_MUTE_MASK, 0); } return 0; } diff --git a/sound/soc/codecs/cs42l56.h b/sound/soc/codecs/cs42l56.h index ad2b50a..4997587 100644 --- a/sound/soc/codecs/cs42l56.h +++ b/sound/soc/codecs/cs42l56.h @@ -116,8 +116,6 @@ #define CS42L56_DEEMPH_MASK 0x40 #define CS42L56_PLYBCK_GANG_MASK 0x10 #define CS42L56_PCM_INV_MASK 0x0c -#define CS42L56_MUTE 1 -#define CS42L56_UNMUTE 0 #define CS42L56_ADCAMIX_MUTE_MASK 0x40 #define CS42L56_ADCBMIX_MUTE_MASK 0x80 #define CS42L56_PCMAMIX_MUTE_MASK 0x10 -- 1.8.3.2

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[alsa-devel] [REGRESSION] alsa PM regression in 3.15*, hda intel, realtek codec
by Knut Petersen 10 May '14

10 May '14

S3 suspend / resume cycle works on kernels up to 3.14.x as expected. With kernels 3.15.0-rc2 it's necessary to reload the alsa modules as sound stays silent otherwise. Hardware: AOpen i915GMm-hfs mobo, hda intel sound with realtek codec. 00:1b.0 Audio device: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) High Definition Audio Controller (rev 04) Subsystem: AOPEN Inc. Device 0560 Flags: bus master, fast devsel, latency 0, IRQ 46 Memory at d5440000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+ Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 Capabilities: [100] Virtual Channel Capabilities: [130] Root Complex Link Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel Is this regression known or should I bisect? cu, Knut

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[alsa-devel] [patch] ASoC: wm0010: set error code on allocation failure
by Dan Carpenter 09 May '14

09 May '14

We should return -ENOMEM if the kzalloc() fails instead of returning -ENOMEM. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 71ce315..8dac6c5 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -682,6 +682,7 @@ static int wm0010_boot(struct snd_soc_codec *codec) if (!img_swap) { dev_err(codec->dev, "Failed to allocate image buffer\n"); + ret = -ENOMEM; goto abort; }

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Re: [alsa-devel] [PATCH 1/2] mfd: mc13xxx: Do not use platform data with devicetree
by Lee Jones 09 May '14

09 May '14

On Fri, 09 May 2014, Alexander Shiyan wrote: > Fri, 9 May 2014 12:30:30 +0100 от Lee Jones <lee.jones(a)linaro.org>: > > > This patch removes platform data usage for codec and touchscreen > > > when driver is used with devicetree. > > > This fixes possible "null pointer dereference" error if DTS uses > > > "fsl,mc13xxx-uses-codec" or "fsl,mc13xxx-uses-touch" options. > > > > > > Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> > > > --- > > > drivers/mfd/mc13xxx-core.c | 18 ++++++++++-------- > > > 1 file changed, 10 insertions(+), 8 deletions(-) > > > > Applied, thanks. > > This patch has been written as a way to separate MFD and ASoC changes of > the patch: (ASoC: mc13783: Add devicetree support) > https://git.kernel.org/cgit/linux/kernel/git/next/linux-next.git/commit/?id… > > So, it should produce conflict. Exactly. Mark, You need to revert (or remove) the original one and apply: "ASoC: mc13783: Add devicetree support" ... instead. -- Lee Jones Linaro STMicroelectronics Landing Team Lead Linaro.org │ Open source software for ARM SoCs Follow Linaro: Facebook | Twitter | Blog

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Re: [alsa-devel] Sound issue in omap-twl4030
by Chaitral Purkar 09 May '14

09 May '14

I wrongly configured McBSP channel. Now, I am getting an error at the end which is 'aplay: pcm_write:1710: write error: Input/output error'. Can anybody provide some pointers in this regard. I am copying the aplay output. ubuntu@arm:~$ aplay -v test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Plug PCM: Hardware PCM card 0 'Zoom2' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 27560 period_size : 5512 period_time : 124988 tstamp_mode : NONE period_step : 1 avail_min : 5512 period_event : 0 start_threshold : 27560 stop_threshold : 27560 silence_threshold: 0 silence_size : 0 boundary : 1806172160 appl_ptr : 0 hw_ptr : 0 aplay: pcm_write:1710: write error: Input/output error Regards, Chaitral Purkar DISCLAIMER ========== This e-mail may contain privileged and confidential information which is the property of Persistent Systems Ltd. It is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you are not authorized to read, retain, copy, print, distribute or use this message. If you have received this communication in error, please notify the sender and delete all copies of this message. Persistent Systems Ltd. does not accept any liability for virus infected mails.

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