January 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2] ASoC: pcm5102a: Provide a menu selection text
by Jörg Krause 07 Jan '17

07 Jan '17

Provide a menu selection text so that users can enable, disable or mark it as module in menuconfig. This is necessary for using the codec with the simple-card framework. Signed-off-by: Jörg Krause <joerg.krause(a)embedded.rocks> --- Changes v2: - improve commit log (suggested by Takashi Iwai) --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9e1718a8cb1c..534c8c2b5da1 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -642,7 +642,7 @@ config SND_SOC_PCM3168A_SPI select REGMAP_SPI config SND_SOC_PCM5102A - tristate + tristate "Texas Instruments PCM5102A CODEC" config SND_SOC_PCM512x tristate -- 2.11.0

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[alsa-devel] ASoC: Using mxs-saif with simple-card
by Jörg Krause 07 Jan '17

07 Jan '17

Hi, I am trying to use the simple-card framework on a custom i.MX28 board. There are two SAIF modules (saif0 and saif1) instantiated on the cpu. A reference for the platform driver is mxs-sgtl5000. The two cpu dais saif0 and saif1 are connected to the audio codec sgtl5000. saif0 is used to playback and saif1 is used to capture. Unfortunately, I do not have a sgtl5000 codec, but a pcm5102a, a wm8524 and wm8731 to play with. For all those codec I copied the mxs-sgtl5000 driver, e.g. mxs- pcm5102a, mxs-wm8524, mxs-wm8731. I will refer to these drivers as mxs- <codec> later. The only difference was in setting the mclk. That's why I'm interested in using the simple-card driver for this platform. With the help of Mans Rullgard I came up with these parts of the device tree node when using the wm8731 codec: sound { compatible = "simple-audio-card"; simple-audio-card,name = "hbm10-wm8731"; simple-audio-card,widgets = "Headphone", "Headphone Jack", "Microphone", "Microphone Jack", "Line", "Line Jack"; simple-audio-card,routing = "Headphone Jack", "RHPOUT", "Headphone Jack", "LHPOUT", "LLINEIN", "Line Jack", "MICIN", "Mic Bias", "Mic Bias", "Microphone Jack"; simple-audio-card,dai-link@0 { format = "i2s"; bitclock-master = <&dai0_master>; frame-master = <&dai0_master>; mclk-fs = <256>; dai0_master: cpu { sound-dai = <&saif0>; }; codec { sound-dai = <&wm8731>; clocks = <&saif0>; }; }; simple-audio-card,dai-link@1 { format = "i2s"; bitclock-master = <&dai1_master>; frame-master = <&dai1_master>; mclk-fs = <256>; dai1_master: cpu { sound-dai = <&saif1>; }; codec { sound-dai = <&wm8731>; }; }; }; &saif0 { #sound-dai-cells = <0>; pinctrl-names = "default"; pinctrl-0 = <&saif0_pins_a>; assigned-clocks = <&clks 53>; assigned-clock-rates = <12288000>; status = "okay"; }; &saif1 { #sound-dai-cells = <0>; pinctrl-names = "default"; pinctrl-0 = <&saif1_pins_a>; fsl,saif-master = <&saif0>; status = "okay"; }; &i2c0 { pinctrl-names = "default"; pinctrl-0 = <&i2c0_pins_a>; status = "okay"; wm8731: wm8731@1a { #sound-dai-cells = <0>; compatible = "wlf,wm8731"; reg = <0x1a>; }; }; >From my understanding of the simple-card framework this should connect saif0 (cpu dai) <-> wm8731 (codec) and saif1 (cpu dai) <-> wm8731 (codec) as it is done in the probe function of the mxs-<codec> driver, right? However, this gives me a warning while bringing up the board: sysfs: cannot create duplicate filename '/devices/soc0/sound/mxs-saif- wm8731-hifi'. Note, that the simple-card framework works fine for mxs-saif if using only the dai0 link (playback) and leaving the dai1 link (capture). Am I doing something wrong here? Is it possible to map the two dai links using the current simple-card framework? Is anyone using the simple-card framework on mxs-saif using both playback and capture? Best regards, Jörg Krause

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[alsa-devel] [PATCH resend v3] ASoC: sun4i-codec: Add "Right Mixer" to "Line Out Mono Diff." route
by Chen-Yu Tsai 07 Jan '17

07 Jan '17

The mono differential output for "Line Out" downmixes the stereo audio from the mixer, instead of just taking the left channel. Add a route from the "Right Mixer" to "Line Out Source Playback Route" through the "Mono Differential" path, so DAPM doesn't shut down everything if the left channel is muted. Fixes: 0f909f98d7cb ("ASoC: sun4i-codec: Add support for A31 Line Out playback") Signed-off-by: Chen-Yu Tsai <wens(a)csie.org> Acked-by: Maxime Ripard <maxime.ripard(a)free-electrons.com> --- Last sent 2016/11/24. Resending with Maxime's ack. --- sound/soc/sunxi/sun4i-codec.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 848af01692a0..c3aab10fa085 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1058,6 +1058,7 @@ static const struct snd_soc_dapm_route sun6i_codec_codec_dapm_routes[] = { { "Line Out Source Playback Route", "Stereo", "Left Mixer" }, { "Line Out Source Playback Route", "Stereo", "Right Mixer" }, { "Line Out Source Playback Route", "Mono Differential", "Left Mixer" }, + { "Line Out Source Playback Route", "Mono Differential", "Right Mixer" }, { "LINEOUT", NULL, "Line Out Source Playback Route" }, /* ADC Routes */ -- 2.11.0

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[alsa-devel] [PATCH 2/6 v3] alsa-lib: Fix for sync issue on xrun recover
by sutar.mounesh＠gmail.com 07 Jan '17

07 Jan '17

From: Andreas Pape <apape(a)de.adit-jv.com> If using very short periods, DSHARE/DSNOOP/DMIX may report underruns while in status 'prepared'. This prohibits correct recovery. Now slave xrun conditions for DSHARE/DSNOOP/DMIX are being handled properly. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Joshua Frkuska <joshua_frkuska(a)mentor.com> Signed-off-by: Mounesh Sutar <mounesh_sutar(a)mentor.com> diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 0770abc..4e7fda8 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -550,6 +550,101 @@ int snd_pcm_direct_timer_stop(snd_pcm_direct_t *dmix) return 0; } +/* + * Recover slave on XRUN. + * Even if direct plugins disable xrun detection, there might be an xrun + * raised directly by some drivers. + * The first client recovers slave pcm. + * Each client needs to execute sw xrun handling afterwards + */ +int snd_pcm_direct_slave_recover(snd_pcm_direct_t *direct) +{ + int ret; + int semerr; + + semerr = snd_pcm_direct_semaphore_down(direct, + DIRECT_IPC_SEM_CLIENT); + if (semerr > 0) { + SNDERR("SEMDOWN FAILED with err %d", semerr); + return semerr; + } + + if (snd_pcm_state(direct->spcm) != SND_PCM_STATE_XRUN) { + /* ignore... someone else already did recovery */ + semerr = snd_pcm_direct_semaphore_up(direct, + DIRECT_IPC_SEM_CLIENT); + if (semerr > 0) { + SNDERR("SEMUP FAILED with err %d", semerr); + return semerr; + } + return ret; + } + + ret = snd_pcm_prepare(direct->spcm); + if (ret < 0) { + SNDERR("recover: unable to prepare slave"); + semerr = snd_pcm_direct_semaphore_up(direct, + DIRECT_IPC_SEM_CLIENT); + if (semerr > 0) { + SNDERR("SEMUP FAILED with err %d", semerr); + return semerr; + } + return ret; + } + + if (direct->type == SND_PCM_TYPE_DSHARE) { + const snd_pcm_channel_area_t *dst_areas; + dst_areas = snd_pcm_mmap_areas(direct->spcm); + snd_pcm_areas_silence(dst_areas, 0, direct->spcm->channels, + direct->spcm->buffer_size, + direct->spcm->format); + } + + ret = snd_pcm_start(direct->spcm); + if (ret < 0) { + SNDERR("recover: unable to start slave"); + semerr = snd_pcm_direct_semaphore_up(direct, + DIRECT_IPC_SEM_CLIENT); + if (semerr > 0) { + SNDERR("SEMUP FAILED with err %d", semerr); + return semerr; + } + return ret; + } + direct->shmptr->recoveries++; + semerr = snd_pcm_direct_semaphore_up(direct, + DIRECT_IPC_SEM_CLIENT); + if (semerr > 0) { + SNDERR("SEMUP FAILED with err %d", semerr); + return semerr; + } + return 0; +} + +/* + * enter xrun state, if slave xrun occured + * @return: 0 - no xrun >0: xrun happened + */ +int snd_pcm_direct_client_chk_xrun(snd_pcm_direct_t *direct, snd_pcm_t *pcm) +{ + if (direct->shmptr->recoveries != direct->recoveries) { + /* no matter how many xruns we missed - + * so don't increment but just update to actual counter + */ + direct->recoveries = direct->shmptr->recoveries; + pcm->fast_ops->drop(pcm); + /* trigger_tstamp update is missing in drop callbacks */ + gettimestamp(&direct->trigger_tstamp, pcm->tstamp_type); + /* no timer clear: + * if slave already entered xrun again the event is lost. + * snd_pcm_direct_clear_timer_queue(direct); + */ + direct->state = SND_PCM_STATE_XRUN; + return 1; + } + return 0; +} + int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int nfds, unsigned short *revents) { snd_pcm_direct_t *dmix = pcm->private_data; @@ -572,6 +667,12 @@ int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned in } switch (snd_pcm_state(dmix->spcm)) { case SND_PCM_STATE_XRUN: + /* recover slave and update client state to xrun + * before returning POLLERR + */ + snd_pcm_direct_slave_recover(dmix); + snd_pcm_direct_client_chk_xrun(dmix, pcm); + /* fallthrough */ case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_SETUP: events |= POLLERR; @@ -1382,6 +1483,7 @@ int snd_pcm_direct_open_secondary_client(snd_pcm_t **spcmp, snd_pcm_direct_t *dm dmix->slave_buffer_size = spcm->buffer_size; dmix->slave_period_size = dmix->shmptr->s.period_size; dmix->slave_boundary = spcm->boundary; + dmix->recoveries = dmix->shmptr->recoveries; ret = snd_pcm_mmap(spcm); if (ret < 0) { diff --git a/src/pcm/pcm_direct.h b/src/pcm/pcm_direct.h index 91e816c..569d2be 100644 --- a/src/pcm/pcm_direct.h +++ b/src/pcm/pcm_direct.h @@ -66,6 +66,7 @@ typedef struct { char socket_name[256]; /* name of communication socket */ snd_pcm_type_t type; /* PCM type (currently only hw) */ int use_server; + unsigned int recoveries; /* no of executed recoveries on slave*/ struct { unsigned int format; snd_interval_t rate; @@ -157,6 +158,7 @@ struct snd_pcm_direct { int var_periodsize; /* allow variable period size if max_periods is != -1*/ unsigned int channels; /* client's channels */ unsigned int *bindings; + unsigned int recoveries; /* mirror of executed recoveries on slave */ union { struct { int shmid_sum; /* IPC global sum ring buffer memory identification */ @@ -318,7 +320,8 @@ int snd_pcm_direct_open_secondary_client(snd_pcm_t **spcmp, snd_pcm_direct_t *dm snd_pcm_chmap_query_t **snd_pcm_direct_query_chmaps(snd_pcm_t *pcm); snd_pcm_chmap_t *snd_pcm_direct_get_chmap(snd_pcm_t *pcm); int snd_pcm_direct_set_chmap(snd_pcm_t *pcm, const snd_pcm_chmap_t *map); - +int snd_pcm_direct_slave_recover(snd_pcm_direct_t *direct); +int snd_pcm_direct_client_chk_xrun(snd_pcm_direct_t *direct, snd_pcm_t *pcm); int snd_timer_async(snd_timer_t *timer, int sig, pid_t pid); struct timespec snd_pcm_hw_fast_tstamp(snd_pcm_t *pcm); diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index 0ab7323..258d4de 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -434,15 +434,21 @@ static int snd_pcm_dmix_sync_ptr0(snd_pcm_t *pcm, snd_pcm_uframes_t slave_hw_ptr static int snd_pcm_dmix_sync_ptr(snd_pcm_t *pcm) { snd_pcm_direct_t *dmix = pcm->private_data; + int err; switch (snd_pcm_state(dmix->spcm)) { case SND_PCM_STATE_DISCONNECTED: dmix->state = SND_PCM_STATE_DISCONNECTED; return -ENODEV; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dmix)) <0) + return err; + break; default: break; } - + if (snd_pcm_direct_client_chk_xrun(dmix, pcm)) + return -EPIPE; if (dmix->slowptr) snd_pcm_hwsync(dmix->spcm); @@ -828,12 +834,16 @@ static snd_pcm_sframes_t snd_pcm_dmix_mmap_commit(snd_pcm_t *pcm, switch (snd_pcm_state(dmix->spcm)) { case SND_PCM_STATE_XRUN: - return -EPIPE; + if ((err = snd_pcm_direct_slave_recover(dmix)) < 0) + return err; + break; case SND_PCM_STATE_SUSPENDED: return -ESTRPIPE; default: break; } + if (snd_pcm_direct_client_chk_xrun(dmix, pcm)) + return -EPIPE; if (! size) return 0; snd_pcm_mmap_appl_forward(pcm, size); @@ -841,8 +851,10 @@ static snd_pcm_sframes_t snd_pcm_dmix_mmap_commit(snd_pcm_t *pcm, if ((err = snd_pcm_dmix_start_timer(pcm, dmix)) < 0) return err; } else if (dmix->state == SND_PCM_STATE_RUNNING || - dmix->state == SND_PCM_STATE_DRAINING) - snd_pcm_dmix_sync_ptr(pcm); + dmix->state == SND_PCM_STATE_DRAINING) { + if ((err = snd_pcm_dmix_sync_ptr(pcm)) < 0) + return err; + } if (dmix->state == SND_PCM_STATE_RUNNING || dmix->state == SND_PCM_STATE_DRAINING) { /* ok, we commit the changes after the validation of area */ @@ -858,10 +870,13 @@ static snd_pcm_sframes_t snd_pcm_dmix_mmap_commit(snd_pcm_t *pcm, static snd_pcm_sframes_t snd_pcm_dmix_avail_update(snd_pcm_t *pcm) { snd_pcm_direct_t *dmix = pcm->private_data; + int err; if (dmix->state == SND_PCM_STATE_RUNNING || - dmix->state == SND_PCM_STATE_DRAINING) - snd_pcm_dmix_sync_ptr(pcm); + dmix->state == SND_PCM_STATE_DRAINING) { + if ((err = snd_pcm_dmix_sync_ptr(pcm)) < 0) + return err; + } return snd_pcm_mmap_playback_avail(pcm); } diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index a1fed5d..570f180 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -162,7 +162,7 @@ static int snd_pcm_dshare_sync_ptr0(snd_pcm_t *pcm, snd_pcm_uframes_t slave_hw_p snd_pcm_direct_t *dshare = pcm->private_data; snd_pcm_uframes_t old_slave_hw_ptr, avail; snd_pcm_sframes_t diff; - + old_slave_hw_ptr = dshare->slave_hw_ptr; dshare->slave_hw_ptr = slave_hw_ptr; diff = slave_hw_ptr - old_slave_hw_ptr; @@ -202,15 +202,21 @@ static int snd_pcm_dshare_sync_ptr0(snd_pcm_t *pcm, snd_pcm_uframes_t slave_hw_p static int snd_pcm_dshare_sync_ptr(snd_pcm_t *pcm) { snd_pcm_direct_t *dshare = pcm->private_data; + int err; switch (snd_pcm_state(dshare->spcm)) { case SND_PCM_STATE_DISCONNECTED: dshare->state = SNDRV_PCM_STATE_DISCONNECTED; return -ENODEV; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dshare)) <0) + return err; + break; default: break; } - + if (snd_pcm_direct_client_chk_xrun(dshare, pcm)) + return -EPIPE; if (dshare->slowptr) snd_pcm_hwsync(dshare->spcm); @@ -516,12 +522,16 @@ static snd_pcm_sframes_t snd_pcm_dshare_mmap_commit(snd_pcm_t *pcm, switch (snd_pcm_state(dshare->spcm)) { case SND_PCM_STATE_XRUN: - return -EPIPE; + if ((err = snd_pcm_direct_slave_recover(dshare)) < 0) + return err; + break; case SND_PCM_STATE_SUSPENDED: return -ESTRPIPE; default: break; } + if (snd_pcm_direct_client_chk_xrun(dshare, pcm)) + return -EPIPE; if (! size) return 0; snd_pcm_mmap_appl_forward(pcm, size); @@ -529,8 +539,10 @@ static snd_pcm_sframes_t snd_pcm_dshare_mmap_commit(snd_pcm_t *pcm, if ((err = snd_pcm_dshare_start_timer(dshare)) < 0) return err; } else if (dshare->state == SND_PCM_STATE_RUNNING || - dshare->state == SND_PCM_STATE_DRAINING) - snd_pcm_dshare_sync_ptr(pcm); + dshare->state == SND_PCM_STATE_DRAINING) { + if ((err = snd_pcm_dshare_sync_ptr(pcm)) < 0) + return err; + } if (dshare->state == SND_PCM_STATE_RUNNING || dshare->state == SND_PCM_STATE_DRAINING) { /* ok, we commit the changes after the validation of area */ @@ -546,10 +558,13 @@ static snd_pcm_sframes_t snd_pcm_dshare_mmap_commit(snd_pcm_t *pcm, static snd_pcm_sframes_t snd_pcm_dshare_avail_update(snd_pcm_t *pcm) { snd_pcm_direct_t *dshare = pcm->private_data; + int err; if (dshare->state == SND_PCM_STATE_RUNNING || - dshare->state == SND_PCM_STATE_DRAINING) - snd_pcm_dshare_sync_ptr(pcm); + dshare->state == SND_PCM_STATE_DRAINING) { + if ((err = snd_pcm_dshare_sync_ptr(pcm)) < 0) + return err; + } return snd_pcm_mmap_playback_avail(pcm); } diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 85f0ff4..6b721ec 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -132,14 +132,21 @@ static int snd_pcm_dsnoop_sync_ptr(snd_pcm_t *pcm) snd_pcm_direct_t *dsnoop = pcm->private_data; snd_pcm_uframes_t slave_hw_ptr, old_slave_hw_ptr, avail; snd_pcm_sframes_t diff; - + int err; + switch (snd_pcm_state(dsnoop->spcm)) { case SND_PCM_STATE_DISCONNECTED: dsnoop->state = SNDRV_PCM_STATE_DISCONNECTED; return -ENODEV; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dsnoop)) <0) + return err; + break; default: break; } + if (snd_pcm_direct_client_chk_xrun(dsnoop, pcm)) + return -EPIPE; if (dsnoop->slowptr) snd_pcm_hwsync(dsnoop->spcm); old_slave_hw_ptr = dsnoop->slave_hw_ptr; @@ -410,12 +417,16 @@ static snd_pcm_sframes_t snd_pcm_dsnoop_mmap_commit(snd_pcm_t *pcm, switch (snd_pcm_state(dsnoop->spcm)) { case SND_PCM_STATE_XRUN: - return -EPIPE; + if ((err = snd_pcm_direct_slave_recover(dsnoop)) <0) + return err; + break; case SND_PCM_STATE_SUSPENDED: return -ESTRPIPE; default: break; } + if (snd_pcm_direct_client_chk_xrun(dsnoop, pcm)) + return -EPIPE; if (dsnoop->state == SND_PCM_STATE_RUNNING) { err = snd_pcm_dsnoop_sync_ptr(pcm); if (err < 0) -- 2.7.4

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[alsa-devel] what is exactly pcm.boundary?
by Ricardo Biehl 07 Jan '17

07 Jan '17

Hello. After some thoughts in ALSA internals (hw_ptr, appl_ptr, mmap'ed buffer, ioctls) I got me in many doubts. I would love if you help me understanding this. Well, in src/pcm/pcm_local.h line ~469: 1 static inline snd_pcm_uframes_t snd_pcm_mmap_playback_avail(snd_pcm_t *pcm) 2 { 3 snd_pcm_sframes_t avail; 4 avail = *pcm->hw.ptr + pcm->buffer_size - *pcm->appl.ptr; 5 if (avail < 0) 6 avail += pcm->boundary; 7 else if ((snd_pcm_uframes_t) avail >= pcm->boundary) 8 avail -= pcm->boundary; 9 return avail; 10 } First, I assumed that both hw_ptr and appl_ptr always points to addresses inside a REGION of memory that is exactly the size of buffer = (period_size * periods). I've also tried to imagine line 4 as: The forward distance between appl_ptr and hw_ptr where our application can write new samples: |s0|s1|s2|s3|s4|s5|s6|s7|s8|s9| buffer_size = 10 For example, if hw_ptr is on 's1' and appl_ptr is on 's7' we'll have ( 1 + 10 - 7 = 4 ) [s8,s9,s0,s1] available to write, right? hw_ptr = Last sample that sound device has eaten up in ring buffer. appl_ptr = Last sample that application has written in ring buffer. Ok, the problem comes, for example, when hw_ptr is on 's9' and appl_ptr is on 's4', so the sum is ( 9 + 10 - 4 = 15 ) -- a number greater than buffer_size. Although I expected pcm->boundary is equal buffer_size and then the test in 7th line would correct it: (9 + 10 - 4 = 15) (15 >= 10) 7th line condition test (15 - buffer_size = __5__) resulting in five [s5,s6,s7,s8,s9] samples available. But it's not. Why? And what is exactly 'boundary' ? I read this <http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg29891.html> but doesn't help me Another question ... can I thrust in hw_ptr value after a HWSYNC? will it be the exactly value of the current processed sample? And to end ... is the latency of HWSYNC operation insignificant, or it could block or something that increases latency may happen? Thank you very much! Cheers :-) -- Ricardo Biehl Pasquali

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[alsa-devel] [PATCH V2 0/7] Add support for Legacy HDMI audio drivers
by Jerome Anand 07 Jan '17

07 Jan '17

Legacy (CherryTrail/ Baytrail) HDMI audio drivers added Legacy hdmi audio-Gfx interaction/ interfacing is updated to use irq chip framework Jerome Anand (7): drm/i915: setup bridge for HDMI LPE audio driver drm/i915: Add support for audio driver notifications ALSA: add shell for Intel HDMI LPE audio driver ALSA: x86: hdmi: Add audio support for BYT and CHT ALSA: x86: hdmi: Improve position reporting ALSA: x86: hdmi: Fixup some monitor ALSA: x86: hdmi: continue playback even when display resolution changes Documentation/gpu/i915.rst | 9 + drivers/gpu/drm/i915/Makefile | 3 + drivers/gpu/drm/i915/i915_drv.c | 8 +- drivers/gpu/drm/i915/i915_drv.h | 18 + drivers/gpu/drm/i915/i915_irq.c | 16 + drivers/gpu/drm/i915/i915_reg.h | 3 + drivers/gpu/drm/i915/intel_audio.c | 8 + drivers/gpu/drm/i915/intel_hdmi.c | 1 + drivers/gpu/drm/i915/intel_lpe_audio.c | 401 +++++++ include/drm/intel_lpe_audio.h | 46 + sound/Kconfig | 2 + sound/Makefile | 2 +- sound/x86/Kconfig | 16 + sound/x86/Makefile | 8 + sound/x86/intel_hdmi_audio.c | 1928 ++++++++++++++++++++++++++++++++ sound/x86/intel_hdmi_audio.h | 201 ++++ sound/x86/intel_hdmi_audio_if.c | 551 +++++++++ sound/x86/intel_hdmi_lpe_audio.c | 627 +++++++++++ sound/x86/intel_hdmi_lpe_audio.h | 685 ++++++++++++ 19 files changed, 4530 insertions(+), 3 deletions(-) create mode 100644 drivers/gpu/drm/i915/intel_lpe_audio.c create mode 100644 include/drm/intel_lpe_audio.h create mode 100644 sound/x86/Kconfig create mode 100644 sound/x86/Makefile create mode 100644 sound/x86/intel_hdmi_audio.c create mode 100644 sound/x86/intel_hdmi_audio.h create mode 100644 sound/x86/intel_hdmi_audio_if.c create mode 100644 sound/x86/intel_hdmi_lpe_audio.c create mode 100644 sound/x86/intel_hdmi_lpe_audio.h -- 2.9.3 base-commit: 362e5ebefe40f1f00a7075ad2e472ef44a36a7da

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[alsa-devel] [PATCH 1/7] ASoC: wm9705: Remove unused DAI ID defines
by Lars-Peter Clausen 06 Jan '17

06 Jan '17

The DAI ID defines are back from the time when DAIs were referenced by a numerical ID. These days a string is used for matching instead and the defines are unused. The last user of these defines was removed in commit f0fba2ad1b6b ("ASoC: multi-component - ASoC Multi-Component Support"). So remove the defines as well. This also means the wm9705.h file no longer has any content and can be removed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/wm9705.c | 2 -- sound/soc/codecs/wm9705.h | 11 ----------- sound/soc/pxa/e740_wm9705.c | 1 - sound/soc/pxa/e750_wm9705.c | 1 - 4 files changed, 15 deletions(-) delete mode 100644 sound/soc/codecs/wm9705.h diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index dcdd055..266b6ec 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -20,8 +20,6 @@ #include <sound/initval.h> #include <sound/soc.h> -#include "wm9705.h" - #define WM9705_VENDOR_ID 0x574d4c05 #define WM9705_VENDOR_ID_MASK 0xffffffff diff --git a/sound/soc/codecs/wm9705.h b/sound/soc/codecs/wm9705.h deleted file mode 100644 index 23ea9ce..0000000 --- a/sound/soc/codecs/wm9705.h +++ /dev/null @@ -1,11 +0,0 @@ -/* - * wm9705.h -- WM9705 Soc Audio driver - */ - -#ifndef _WM9705_H -#define _WM9705_H - -#define WM9705_DAI_AC97_HIFI 0 -#define WM9705_DAI_AC97_AUX 1 - -#endif diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c index 1de8765..086c37a 100644 --- a/sound/soc/pxa/e740_wm9705.c +++ b/sound/soc/pxa/e740_wm9705.c @@ -22,7 +22,6 @@ #include <asm/mach-types.h> -#include "../codecs/wm9705.h" #include "pxa2xx-ac97.h" diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c index b7eb7cd..7823278 100644 --- a/sound/soc/pxa/e750_wm9705.c +++ b/sound/soc/pxa/e750_wm9705.c @@ -22,7 +22,6 @@ #include <asm/mach-types.h> -#include "../codecs/wm9705.h" #include "pxa2xx-ac97.h" static int e750_spk_amp_event(struct snd_soc_dapm_widget *w, -- 2.1.4

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[alsa-devel] [PATCH 1/2] ASoC: Intel: bytcr-rt5640: fix settings in internal clock mode
by Pierre-Louis Bossart 06 Jan '17

06 Jan '17

Frequency value of zero did not make sense, use same 24.576MHz setting and only change the clock source in idle mode Suggested-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index e33e477..8d2fb2d 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -142,7 +142,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, * for Jack detection and button press */ ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_RCCLK, - 0, + 48000 * 512, SND_SOC_CLOCK_IN); if (!ret) { if ((byt_rt5640_quirk & BYT_RT5640_MCLK_EN) && priv->mclk) -- 2.7.4

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[alsa-devel] [PATCH] ASoC: wm_adsp: Add mechanism to preload firmware on a core
by Charles Keepax 06 Jan '17

06 Jan '17

As requirements to bring up audio paths are continuous getting tighter and the DSP download to most ADSP devices happens over an external bus it can become an important factor in the path bring up time. As such sometimes it is a reasonable trade off to download the firmware ahead of when it will be required and take a small hit on power consumption for keeping the core powered up. This "preloading" adds an additional control for each DSP core "DSPx Preload Switch" that when set to true will power up the DSP core and download the firmware currently selected in the "DSPx Firmware" control. Whilst the core is preloaded the current firmware can not be changed and the CODEC will be kept powered up and SYSCLK held on. Although future improvements may allow the SYSCLK to be powered down as well because the hardware only requires SYSCLK whilst the download is actually taking place, but this is not covered in this series. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.h | 1 + sound/soc/codecs/cs47l24.c | 3 +++ sound/soc/codecs/wm5102.c | 2 ++ sound/soc/codecs/wm5110.c | 5 +++++ sound/soc/codecs/wm_adsp.c | 45 ++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm_adsp.h | 11 +++++++++++ 6 files changed, 66 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 5670786..1822e3b 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -192,6 +192,7 @@ extern unsigned int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; #define ARIZONA_DSP_ROUTES(name) \ { name, NULL, name " Preloader"}, \ { name " Preloader", NULL, "SYSCLK" }, \ + { name " Preload", NULL, name " Preloader"}, \ { name, NULL, name " Aux 1" }, \ { name, NULL, name " Aux 2" }, \ { name, NULL, name " Aux 3" }, \ diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index 73559ae..0dd721e 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -173,6 +173,9 @@ SOC_ENUM("ISRC2 FSH", arizona_isrc_fsh[1]), SOC_ENUM("ISRC3 FSH", arizona_isrc_fsh[2]), SOC_ENUM("ASRC RATE 1", arizona_asrc_rate1), +WM_ADSP2_PRELOAD_SWITCH("DSP2", 2), +WM_ADSP2_PRELOAD_SWITCH("DSP3", 3), + ARIZONA_MIXER_CONTROLS("DSP2L", ARIZONA_DSP2LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP2R", ARIZONA_DSP2RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP3L", ARIZONA_DSP3LMIX_INPUT_1_SOURCE), diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index e7ab37d..9edd239 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -855,6 +855,8 @@ ARIZONA_LHPF_CONTROL("LHPF2 Coefficients", ARIZONA_HPLPF2_2), ARIZONA_LHPF_CONTROL("LHPF3 Coefficients", ARIZONA_HPLPF3_2), ARIZONA_LHPF_CONTROL("LHPF4 Coefficients", ARIZONA_HPLPF4_2), +WM_ADSP2_PRELOAD_SWITCH("DSP1", 1), + ARIZONA_MIXER_CONTROLS("DSP1L", ARIZONA_DSP1LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP1R", ARIZONA_DSP1RMIX_INPUT_1_SOURCE), diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 585fc70..01b5b6c 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -778,6 +778,11 @@ SOC_ENUM("ISRC2 FSH", arizona_isrc_fsh[1]), SOC_ENUM("ISRC3 FSH", arizona_isrc_fsh[2]), SOC_ENUM("ASRC RATE 1", arizona_asrc_rate1), +WM_ADSP2_PRELOAD_SWITCH("DSP1", 1), +WM_ADSP2_PRELOAD_SWITCH("DSP2", 2), +WM_ADSP2_PRELOAD_SWITCH("DSP3", 3), +WM_ADSP2_PRELOAD_SWITCH("DSP4", 4), + ARIZONA_MIXER_CONTROLS("DSP1L", ARIZONA_DSP1LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP1R", ARIZONA_DSP1RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP2L", ARIZONA_DSP2LMIX_INPUT_1_SOURCE), diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index d72ccef..73dbe4c 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2523,6 +2523,43 @@ static void wm_adsp2_set_dspclk(struct wm_adsp *dsp, unsigned int freq) adsp_err(dsp, "Failed to set clock rate: %d\n", ret); } +int wm_adsp2_preloader_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct wm_adsp *dsp = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = dsp->preloaded; + + return 0; +} +EXPORT_SYMBOL_GPL(wm_adsp2_preloader_get); + +int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct wm_adsp *dsp = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + char preload[32]; + + snprintf(preload, ARRAY_SIZE(preload), "DSP%d Preload", mc->shift); + + dsp->preloaded = ucontrol->value.integer.value[0]; + + if (ucontrol->value.integer.value[0]) + snd_soc_dapm_force_enable_pin(dapm, preload); + else + snd_soc_dapm_disable_pin(dapm, preload); + + snd_soc_dapm_sync(dapm); + + return 0; +} +EXPORT_SYMBOL_GPL(wm_adsp2_preloader_put); + int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event, unsigned int freq) @@ -2654,10 +2691,16 @@ EXPORT_SYMBOL_GPL(wm_adsp2_event); int wm_adsp2_codec_probe(struct wm_adsp *dsp, struct snd_soc_codec *codec) { - dsp->codec = codec; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + char preload[32]; + + snprintf(preload, ARRAY_SIZE(preload), "DSP%d Preload", dsp->num); + snd_soc_dapm_disable_pin(dapm, preload); wm_adsp2_init_debugfs(dsp, codec); + dsp->codec = codec; + return snd_soc_add_codec_controls(codec, &wm_adsp_fw_controls[dsp->num - 1], 1); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 411d062..3706b11 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -62,6 +62,7 @@ struct wm_adsp { int fw; int fw_ver; + bool preloaded; bool booted; bool running; @@ -86,7 +87,12 @@ struct wm_adsp { SND_SOC_DAPM_PGA_E(wname, SND_SOC_NOPM, num, 0, NULL, 0, \ wm_adsp1_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD) +#define WM_ADSP2_PRELOAD_SWITCH(wname, num) \ + SOC_SINGLE_EXT(wname " Preload Switch", SND_SOC_NOPM, num, 1, 0, \ + wm_adsp2_preloader_get, wm_adsp2_preloader_put) + #define WM_ADSP2(wname, num, event_fn) \ + SND_SOC_DAPM_SPK(wname " Preload", NULL), \ { .id = snd_soc_dapm_supply, .name = wname " Preloader", \ .reg = SND_SOC_NOPM, .shift = num, .event = event_fn, \ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD, \ @@ -110,6 +116,11 @@ int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, int wm_adsp2_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); +int wm_adsp2_preloader_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream); int wm_adsp_compr_free(struct snd_compr_stream *stream); int wm_adsp_compr_set_params(struct snd_compr_stream *stream, -- 2.1.4

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[alsa-devel] [PATCH v3 00/13] ASoC: Intel: Skylake: Driver updates
by jeeja.kp＠intel.com 06 Jan '17

06 Jan '17

From: Jeeja KP <jeeja.kp(a)intel.com> This patch series includes driver updates for the handling scenarios during System S0/S3 transitions when audio stream is stopped during system. o move DMA configuration in DSP Gateway Module widget event handler to ensure DMA is configured after resume. o Enable SSP MLCK using Supply widget. This will ensure the MCLK is enabled when the widget is power on. o Check DMA decouple register state before coupling/decoupling the DMA. o In case of HDMI pass-through pipe, reset only the FE Pipe when stream in in XRUN state. o Resume the stream from HW renderer position. o HDMI Codec pin and converter state need to be restored when stream is resumed. Changes in v2: - Addressed Mark's comment on patch 1, split this into 2 as this had additional changes other than configuring the DMA. - Addressed Takashi's comment on the usage of snd_hdac_updatel() Changes in v3: - Addressed Takashi's comment on superfluous shit used in reading the value of PPCTL register and also cleaned the condition for updating the reg value. Jeeja KP (13): ASoC: Intel: Skylake: Update link_index and format in pipe params ASoC: Intel: Skylake: Add helper function to setup host/link dma ASoC: Intel: Skylake: Configure DMA in PRE_PMD handler of Mixer ASoC: Intel: Skylake: Removed unused skl_get_format() ALSA: hda: check stream decoupled register state ASoC: Intel: Skylake: Add set_tristate DAI ops to enable SSP MCLK ASoC: Intel: Skylake: Remove unused SSP BE prepare DAI ops ASoC: Intel: Skylake: Add supply widget as non DSP widget ASoC: Intel: Skylake: Add supply widget in skl_nau_max machine ASoC: Intel: Skylake: Add supply widget in bxt_da_max machine ASoC: Intel: Skylake: Don't reset pass-through pipe in BE prepare ASoC: Intel: Skylake: set the resume point to LPIB ASoC: hdac_hdmi: Enable pin and converter in prepare sound/hda/ext/hdac_ext_stream.c | 15 +- sound/soc/codecs/hdac_hdmi.c | 28 ++-- sound/soc/intel/boards/bxt_da7219_max98357a.c | 34 ++++ sound/soc/intel/boards/skl_nau88l25_max98357a.c | 34 ++++ sound/soc/intel/skylake/skl-pcm.c | 210 +++++++++++++----------- sound/soc/intel/skylake/skl-topology.c | 27 ++- sound/soc/intel/skylake/skl-topology.h | 6 + 7 files changed, 240 insertions(+), 114 deletions(-) -- 2.5.0

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