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

[alsa-devel] [PATCH 1/1] alsa-plugins: enhancing configuration support for alsa plugins
by sutar.mounesh＠gmail.com 17 Feb '17

17 Feb '17

From: Mounesh Sutar <mounesh_sutar(a)mentor.com> Following plugins are made configurable in alsa-plugins: * oss * mix * usbstream * arcamav * speexdsp Signed-off-by: Mikhail Durnev <mikhail_durnev(a)mentor.com> Signed-off-by: Mounesh Sutar <mounesh_sutar(a)mentor.com> diff --git a/configure.ac b/configure.ac index a55c735..190d935 100644 --- a/configure.ac +++ b/configure.ac @@ -20,6 +20,30 @@ PKG_CHECK_MODULES(ALSA, alsa >= 1.0.11) AC_CHECK_LIB(asound, snd_pcm_ioplug_create,, AC_ERROR([*** libasound has no external plugin SDK]), -ldl) +AC_ARG_ENABLE([oss], + AS_HELP_STRING([--disable-oss], [Disable building of OSS plugin])) +AS_IF([test "x$enable_oss" != "xno"], + [AM_CONDITIONAL(HAVE_OSS, true)], + [AM_CONDITIONAL(HAVE_OSS, false)]) + +AC_ARG_ENABLE([mix], + AS_HELP_STRING([--disable-mix], [Disable building of upmix and vdownmix plugins])) +AS_IF([test "x$enable_mix" != "xno"], + [AM_CONDITIONAL(HAVE_MIX, true)], + [AM_CONDITIONAL(HAVE_MIX, false)]) + +AC_ARG_ENABLE([usbstream], + AS_HELP_STRING([--disable-usbstream], [Disable building of usb_stream plugin])) +AS_IF([test "x$enable_usbstream" != "xno"], + [AM_CONDITIONAL(HAVE_USBSTREAM, true)], + [AM_CONDITIONAL(HAVE_USBSTREAM, false)]) + +AC_ARG_ENABLE([arcamav], + AS_HELP_STRING([--disable-arcamav], [Disable building of Arcam AV control plugin])) +AS_IF([test "x$enable_arcamav" != "xno"], + [AM_CONDITIONAL(HAVE_ARCAMAV, true)], + [AM_CONDITIONAL(HAVE_ARCAMAV, false)]) + AC_ARG_ENABLE([jack], AS_HELP_STRING([--disable-jack], [Disable building of JACK plugin])) @@ -108,7 +132,12 @@ AC_SUBST(AVCODEC_CFLAGS) AC_SUBST(AVCODEC_LIBS) AC_SUBST(AVCODEC_HEADER) -PKG_CHECK_MODULES(speexdsp, [speexdsp >= 1.2], [HAVE_SPEEXDSP="yes"], [HAVE_SPEEXDSP=""]) +AC_ARG_ENABLE([speexdsp], + AS_HELP_STRING([--disable-speexdsp], [Disable building of speexdsp plugin])) + +if test "x$enable_speexdsp" != "xno"; then + PKG_CHECK_MODULES(speexdsp, [speexdsp >= 1.2rc2], [HAVE_SPEEXDSP="yes"], [HAVE_SPEEXDSP=""]) +fi AM_CONDITIONAL(HAVE_SPEEXDSP, test "$HAVE_SPEEXDSP" = "yes") AC_ARG_WITH([speex], -- 2.7.4

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[alsa-devel] Help with midi player application
by amidi＠LEFerguson.com 17 Feb '17

17 Feb '17

I am hoping someone has a bit of patience to help. I apologize in advance if this is not appropriate use of this mailing list. I am trying to incorporate ALSA's sequencer within an application I am writing to play MIDI pieces, and provide controls (start at measure, tempo changes, etc), while output is sent to a piano player system with midi i/o (specifically it's a Pianodisc system). The result (if it ever works) will be open source. The good news is I can get it to work. The bad news is I am having some kind of timing or buffering issue, and am at a bit of a loss where to look. I am convinced the problem is in my code, or environment, as I can get pmidi (1.7.1) to work (including build it from source, so same compile environment). The actual application is in a QT5.8 application, so it is conceivable that some aspect of QT's environment is at issue. I have embedded the calls to the ALSA sequencer inside of a separate QThread so as to isolate it from QT's event loop. Since the code fails only in context, it is not practical to post the whole code. And yes I'm sure it is related to my code, I get that, am hoping for some trouble shooting ideas, or pointers to tools. The symptoms are on a fast, long piece play hangs for a long period of time (many seconds). I put into the loop queries as to queue info/state/etc, and what I see is the queue tick advancing correctly during this pause, but the events in the queue are steady at about 80%full and not emptying. I have set this up to run through qmidinet which outputs each notes as ALSA hands it off, and that pauses also (qmidinet is not the cause; writing direct to the midi port also pauses). My hope with this email is to get some suggestions how to find out what is happening when it pauses inappropriately. "It" here is whatever processes are draining the queue; the thread making the snd_seq calls continues to run, I can query the queue and see its status, etc. I have tried this with no self-written buffer management, just letting ALSA pause the thread; I also have it running managing the queue myself looking for it getting near full and waiting until it gets more empty (staying above the "room" size). Neither work, both do approximately the same, though trying to manage the buffer seems to work better and gives me event-count visibility in the queue. Allocating a much larger queue buffer (snd_seq_set_client_pool_output) delays the pause significantly but does not solve the problem. The queue is not full (but also is not empty) when the pauses occur. My initial thoughts were the tick in the events was incorrect and so waiting, but that is not the case as best I can tell (also, that would not cause a larger buffer to pause less often). Something is stopping the queue from processing outbound for longish periods. Essentially what I do is: snd_seq_open to get the handle send_seq_alloc_queue snd_seq_client and snd_seq_create_simple_port to get the client port snd_seq_connect_to to connect snd_seq_set_queue_tempo (initially with default snd_deq_start_queue snd_seq_drain_output (until ret = 0) Then I loop through all the midi events. Each event has: snd_seq_ev_schedule_tick called with parameter 2 = 1 for relative ticks. snd_seq_event_output snd_seq_drain_output (until ret = 0) (every 10 events) check queue events and if getting full pause for 10ms I have tried it with and without the drain after each send, neither one works, with the drain seems to work a bit better (and with the drain I can call the queue status and get more info). The environment is a HyperV x64 Ubuntu 16.04.01 install with QT5.8 built from source, and ALSA from the distro version (not built from source). The same thing occurs on a Raspberry Pi 3B (which is my eventual target) running Ubuntu 16.04 Mate. Both environments will run pmidi with the same song correctly. Yes, I know it's my code, not an ALSA bug. What I'm hoping is some suggestion as to how to get more visibility as to what is happening when it pauses. Some kind of trace or debug mode, hopefully without trying to build and manually instrument ALSA. Conversely.... maybe I am using ALSA inappropriate to send just to the sequencer and on to a midi device? I just wanted its scheduler to deal with timing, there is no near term plan to use a synthesizer or other connection instead. Any suggestions or insights appreciated. I started with too little hair, and fear even more has been pulled out now. Linwood Ferguson

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[alsa-devel] [PATCH 5/6] pcm:extplug: refinement of masks in extplug
by sutar.mounesh＠gmail.com 17 Feb '17

17 Feb '17

From: Awais Belal <awais_belal(a)mentor.com> It should be possible to use empty mask format with extplug. The refinement of mask via extplug is now modified, to accept empty masks as well to work properly. Signed-off-by: Awais Belal <awais_belal(a)mentor.com> Signed-off-by: Mounesh Sutar <sutar.mounesh(a)gmail.com> diff --git a/src/pcm/pcm_extplug.c b/src/pcm/pcm_extplug.c index a04f826..1004f54 100644 --- a/src/pcm/pcm_extplug.c +++ b/src/pcm/pcm_extplug.c @@ -172,6 +172,8 @@ int snd_ext_parm_mask_refine(snd_mask_t *mask, struct snd_ext_parm *parm, int ty unsigned int i; parm += type; + if (!parm->active) + return 0; memset(&bits, 0, sizeof(bits)); for (i = 0; i < parm->num_list; i++) bits.bits[parm->list[i] / 32] |= 1U << (parm->list[i] % 32); -- 2.7.4

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[alsa-devel] [PATCH 4/6] pcm: status dump fix timestamp formatting
by sutar.mounesh＠gmail.com 17 Feb '17

17 Feb '17

From: Andreas Pape <apape(a)de.adit-jv.com> nanosecond part formatted with %06 will give incorrect/confusing results: trigger_time: 154.9748287 trigger_time: 154.60109090 trigger_time: 154.110425257 time seems to run backwards... This patch converts to us before printing which gives the correct/expected result: trigger_time: 154.009748 trigger_time: 154.060109 trigger_time: 154.110425 Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Mounesh Sutar <sutar.mounesh(a)gmail.com> diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 493e903..a16fd86 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -2194,9 +2194,10 @@ int snd_pcm_status_dump(snd_pcm_status_t *status, snd_output_t *out) assert(status); snd_output_printf(out, " state : %s\n", snd_pcm_state_name((snd_pcm_state_t) status->state)); snd_output_printf(out, " trigger_time: %ld.%06ld\n", - status->trigger_tstamp.tv_sec, status->trigger_tstamp.tv_nsec); + status->trigger_tstamp.tv_sec, + status->trigger_tstamp.tv_nsec / 1000); snd_output_printf(out, " tstamp : %ld.%06ld\n", - status->tstamp.tv_sec, status->tstamp.tv_nsec); + status->tstamp.tv_sec, status->tstamp.tv_nsec / 1000); snd_output_printf(out, " delay : %ld\n", (long)status->delay); snd_output_printf(out, " avail : %ld\n", (long)status->avail); snd_output_printf(out, " avail_max : %ld\n", (long)status->avail_max); -- 2.7.4

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[alsa-devel] [PATCH 3/6] pcm:file: Enable file writing for capture path
by sutar.mounesh＠gmail.com 17 Feb '17

17 Feb '17

From: Timo Wischer <twischer(a)de.adit-jv.com> This commit reverts parts of commit 4081be0b87ab9fa53a8906e66bc240f18a7a9a54, because it is realy useful to use the file plugin in a capture path for debugging. Also it fixes the truncate issue mentioned in above commit. Additionally following MMAP access issue is considered: $ arecord -D teeraw -M -d5 arecord.wav Recording WAVE 'arecord.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono ALSA lib pcm/pcm_file.c:358:(snd_pcm_file_write_bytes) write failed: Bad file descriptor ALSA lib pcm/pcm_file.c:358:(snd_pcm_file_write_bytes) write failed: Bad file descriptor arecord: pcm/pcm_file.c:397: snd_pcm_file_add_frames: Assertion `file->wbuf_used_bytes < file->wbuf_size_bytes' failed. Aborted by signal Aborted... Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> Signed-off-by: Mounesh Sutar <sutar.mounesh(a)gmail.com> diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index 6d119d6..0363f84 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -544,6 +544,7 @@ static snd_pcm_sframes_t snd_pcm_file_writen(snd_pcm_t *pcm, void **bufs, snd_pc static snd_pcm_sframes_t snd_pcm_file_readi(snd_pcm_t *pcm, void *buffer, snd_pcm_uframes_t size) { snd_pcm_file_t *file = pcm->private_data; + snd_pcm_channel_area_t areas[pcm->channels]; snd_pcm_sframes_t n; n = _snd_pcm_readi(file->gen.slave, buffer, size); @@ -555,8 +556,10 @@ static snd_pcm_sframes_t snd_pcm_file_readi(snd_pcm_t *pcm, void *buffer, snd_pc __snd_pcm_unlock(pcm); if (n < 0) return n; - return n * 8 / pcm->frame_bits; + n = n * 8 / pcm->frame_bits; } + snd_pcm_areas_from_buf(pcm, areas, buffer); + snd_pcm_file_add_frames(pcm, areas, 0, n); return n; } @@ -564,6 +567,7 @@ static snd_pcm_sframes_t snd_pcm_file_readi(snd_pcm_t *pcm, void *buffer, snd_pc static snd_pcm_sframes_t snd_pcm_file_readn(snd_pcm_t *pcm, void **bufs, snd_pcm_uframes_t size) { snd_pcm_file_t *file = pcm->private_data; + snd_pcm_channel_area_t areas[pcm->channels]; snd_pcm_sframes_t n; if (file->ifd >= 0) { @@ -572,6 +576,10 @@ static snd_pcm_sframes_t snd_pcm_file_readn(snd_pcm_t *pcm, void **bufs, snd_pcm } n = _snd_pcm_readn(file->gen.slave, bufs, size); + if (n > 0) { + snd_pcm_areas_from_bufs(pcm, areas, bufs); + snd_pcm_file_add_frames(pcm, areas, 0, n); + } return n; } @@ -635,7 +643,7 @@ static int snd_pcm_file_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t * params) a->first = slave->sample_bits * channel; a->step = slave->frame_bits; } - if ((file->fd < 0) && (pcm->stream == SND_PCM_STREAM_PLAYBACK)) { + if (file->fd < 0) { err = snd_pcm_file_open_output_file(file); if (err < 0) { SYSERR("failed opening output file %s", file->fname); -- 2.7.4

2 1

[alsa-devel] [PATCH 2/6] pcm:direct: fix race on clearing timer events
by sutar.mounesh＠gmail.com 17 Feb '17

17 Feb '17

From: Andreas Pape <apape(a)de.adit-jv.com> snd_timer handling is racy: plugins clear timer queue if avail_min is not reached to force a sleep on timer. The race can happen if the expected event arrives in between the avail check and the clearing of pending events. If this race happens, the user will unnecessarily wait for one more timer event. On low latency/realtime streams this can lead to xruns and must be avoided. As a fix we recheck avail after having cleared poll events. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Signed-off-by: Mounesh Sutar <sutar.mounesh(a)gmail.com> diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 364191b..8a75c42 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -515,10 +515,12 @@ int snd_pcm_direct_async(snd_pcm_t *pcm, int sig, pid_t pid) } /* empty the timer read queue */ -void snd_pcm_direct_clear_timer_queue(snd_pcm_direct_t *dmix) +int snd_pcm_direct_clear_timer_queue(snd_pcm_direct_t *dmix) { + int changed = 0; if (dmix->timer_need_poll) { while (poll(&dmix->timer_fd, 1, 0) > 0) { + changed++; /* we don't need the value */ if (dmix->tread) { snd_timer_tread_t rbuf[4]; @@ -533,15 +535,17 @@ void snd_pcm_direct_clear_timer_queue(snd_pcm_direct_t *dmix) snd_timer_tread_t rbuf[4]; int len; while ((len = snd_timer_read(dmix->timer, rbuf, - sizeof(rbuf))) > 0 && + sizeof(rbuf))) > 0 + && (++changed) && len != sizeof(rbuf[0])) ; } else { snd_timer_read_t rbuf; while (snd_timer_read(dmix->timer, &rbuf, sizeof(rbuf)) > 0) - ; + changed++; } } + return changed; } int snd_pcm_direct_timer_stop(snd_pcm_direct_t *dmix) @@ -693,6 +697,8 @@ int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned in int empty = 0; assert(pfds && nfds == 1 && revents); + +timer_changed: events = pfds[0].revents; if (events & POLLIN) { snd_pcm_uframes_t avail; @@ -720,7 +726,16 @@ int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned in break; default: if (empty) { - snd_pcm_direct_clear_timer_queue(dmix); + /* here we have a race condition: + * if period event arrived after the avail_update call + * above we might clear this event with the following + * clear_timer_queue. + * There is no way to do this in atomic manner, so we + * need to recheck avail_update if we successfully + * cleared a poll event. + */ + if (snd_pcm_direct_clear_timer_queue(dmix)) + goto timer_changed; events &= ~(POLLOUT|POLLIN); /* additional check */ switch (__snd_pcm_state(pcm)) { diff --git a/src/pcm/pcm_direct.h b/src/pcm/pcm_direct.h index fba55fd..24e85f0 100644 --- a/src/pcm/pcm_direct.h +++ b/src/pcm/pcm_direct.h @@ -322,7 +322,7 @@ int snd_pcm_direct_munmap(snd_pcm_t *pcm); int snd_pcm_direct_prepare(snd_pcm_t *pcm); int snd_pcm_direct_resume(snd_pcm_t *pcm); int snd_pcm_direct_timer_stop(snd_pcm_direct_t *dmix); -void snd_pcm_direct_clear_timer_queue(snd_pcm_direct_t *dmix); +int snd_pcm_direct_clear_timer_queue(snd_pcm_direct_t *dmix); int snd_pcm_direct_set_timer_params(snd_pcm_direct_t *dmix); int snd_pcm_direct_open_secondary_client(snd_pcm_t **spcmp, snd_pcm_direct_t *dmix, const char *client_name); -- 2.7.4

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[alsa-devel] [PATCH 1/6] pcm: dmix_rewind corrupts application pointer fix
by sutar.mounesh＠gmail.com 17 Feb '17

17 Feb '17

From: Timo Wischer <twischer(a)de.adit-jv.com> sometimes pulseaudio stops with the following assertion in libasound.so: alsa-lib-1.0.29/src/pcm/pcm.c:2761: snd_pcm_area_copy: Assertion `dst < src || dst >= src + bytes' failed. Application pointer is handled properly, in cases of rewind operations. Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> Signed-off-by: Ravikiran Polepalli <ravikiran_polepalli(a)mentor.com> Signed-off-by: Mikhail Durnev <mikhail_durnev(a)mentor.com> Signed-off-by: Mounesh Sutar <sutar.mounesh(a)gmail.com> diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index dd0356e..4b763e2 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -706,7 +706,7 @@ static snd_pcm_sframes_t snd_pcm_dmix_rewind(snd_pcm_t *pcm, snd_pcm_uframes_t f { snd_pcm_direct_t *dmix = pcm->private_data; snd_pcm_uframes_t slave_appl_ptr, slave_size; - snd_pcm_uframes_t appl_ptr, size, transfer, result; + snd_pcm_uframes_t appl_ptr, size, transfer, result, frames_to_remix; int err; const snd_pcm_channel_area_t *src_areas, *dst_areas; @@ -717,6 +717,11 @@ static snd_pcm_sframes_t snd_pcm_dmix_rewind(snd_pcm_t *pcm, snd_pcm_uframes_t f return err; } + /* (appl_ptr - last_appl_ptr) indicates the frames which are not already mixed + (last_appl_ptr - hw_ptr) indicates the frames which are already mixed but not played yet. + So they can be remixed. + */ + if (dmix->last_appl_ptr < dmix->appl_ptr) size = dmix->appl_ptr - dmix->last_appl_ptr; else @@ -729,6 +734,9 @@ static snd_pcm_sframes_t snd_pcm_dmix_rewind(snd_pcm_t *pcm, snd_pcm_uframes_t f return size; result = size; + /* Always at this point last_appl_ptr == appl_ptr + So (appl_ptr - hw_ptr) indicates the frames which can be remixed + */ if (dmix->hw_ptr < dmix->appl_ptr) size = dmix->appl_ptr - dmix->hw_ptr; else @@ -741,9 +749,12 @@ static snd_pcm_sframes_t snd_pcm_dmix_rewind(snd_pcm_t *pcm, snd_pcm_uframes_t f slave_size = dmix->slave_appl_ptr + (pcm->boundary - dmix->slave_hw_ptr); if (slave_size < size) size = slave_size; - frames -= size; - result += size; - + + /* frames which should be remixed will be saved + to also backward the appl pointer on success + */ + frames_to_remix = size; + /* add sample areas here */ src_areas = snd_pcm_mmap_areas(pcm); dst_areas = snd_pcm_mmap_areas(dmix->spcm); @@ -769,15 +780,15 @@ static snd_pcm_sframes_t snd_pcm_dmix_rewind(snd_pcm_t *pcm, snd_pcm_uframes_t f appl_ptr += transfer; appl_ptr %= pcm->buffer_size; } - dmix->last_appl_ptr -= frames; - dmix->last_appl_ptr %= pcm->boundary; - dmix->slave_appl_ptr -= frames; - dmix->slave_appl_ptr %= dmix->slave_boundary; dmix_up_sem(dmix); - snd_pcm_mmap_appl_backward(pcm, frames); + snd_pcm_mmap_appl_backward(pcm, frames_to_remix); + result += frames_to_remix; + /* At this point last_appl_ptr and appl_ptr has to indicate the + * position of the first not mixed frame + */ - return result + frames; + return result; } static snd_pcm_sframes_t snd_pcm_dmix_forwardable(snd_pcm_t *pcm) -- 2.7.4

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[alsa-devel] Applied "ASoC: nau8825: automatic BCLK and LRC divde in master mode" to the asoc tree
by Mark Brown 17 Feb '17

17 Feb '17

The patch ASoC: nau8825: automatic BCLK and LRC divde in master mode has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 1e561f6166bacb9c12d6fa1d23df07999674573e Mon Sep 17 00:00:00 2001 From: John Hsu <KCHSU0(a)nuvoton.com> Date: Fri, 17 Feb 2017 09:55:33 +0800 Subject: [PATCH] ASoC: nau8825: automatic BCLK and LRC divde in master mode configurable LRC and BCLK divide. The driver will make configurations of LRC and BCLK automatically according to BCLK and FS information in master mode. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/nau8825.c | 20 +++++++++++++++++++- 1 file changed, 19 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 4576f987a4a5..97fbeba9498f 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1231,7 +1231,7 @@ static int nau8825_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); - unsigned int val_len = 0, osr; + unsigned int val_len = 0, osr, ctrl_val, bclk_fs, bclk_div; nau8825_sema_acquire(nau8825, 3 * HZ); @@ -1261,6 +1261,24 @@ static int nau8825_hw_params(struct snd_pcm_substream *substream, osr_adc_sel[osr].clk_src << NAU8825_CLK_ADC_SRC_SFT); } + /* make BCLK and LRC divde configuration if the codec as master. */ + regmap_read(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, &ctrl_val); + if (ctrl_val & NAU8825_I2S_MS_MASTER) { + /* get the bclk and fs ratio */ + bclk_fs = snd_soc_params_to_bclk(params) / params_rate(params); + if (bclk_fs <= 32) + bclk_div = 2; + else if (bclk_fs <= 64) + bclk_div = 1; + else if (bclk_fs <= 128) + bclk_div = 0; + else + return -EINVAL; + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_LRC_DIV_MASK | NAU8825_I2S_BLK_DIV_MASK, + ((bclk_div + 1) << NAU8825_I2S_LRC_DIV_SFT) | bclk_div); + } + switch (params_width(params)) { case 16: val_len |= NAU8825_I2S_DL_16; -- 2.11.0

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[alsa-devel] [PATCH v2] ASoC: Intel: bxt: Add jack port initialize in bxt_rt298 machine
by jeeja.kp＠intel.com 17 Feb '17

17 Feb '17

From: Jeeja KP <jeeja.kp(a)intel.com> After the pcm jack is created, create and initialize the pin switch widget for each port. Pin switch is to enable/disable the pin when monitor is connected/disconnected. Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> --- Change in v2: Add the SOB. sound/soc/intel/boards/bxt_rt298.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index d5f53a6..176c080 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -459,10 +459,12 @@ static int bxt_card_late_probe(struct snd_soc_card *card) { struct bxt_rt286_private *ctx = snd_soc_card_get_drvdata(card); struct bxt_hdmi_pcm *pcm; + struct snd_soc_codec *codec = NULL; int err, i = 0; char jack_name[NAME_SIZE]; list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + codec = pcm->codec_dai->codec; snprintf(jack_name, sizeof(jack_name), "HDMI/DP, pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, @@ -480,7 +482,10 @@ static int bxt_card_late_probe(struct snd_soc_card *card) i++; } - return 0; + if (!codec) + return -EINVAL; + + return hdac_hdmi_jack_port_init(codec, &card->dapm); } -- 2.5.0

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[alsa-devel] [PATCH v2] ASoC: hdac_hdmi: Enable MST audio
by jeeja.kp＠intel.com 17 Feb '17

17 Feb '17

From: Jeeja KP <jeeja.kp(a)intel.com> With the patch series "ASoC: hdmi: Preparatory work to support MST audio", the design is extended from pin to port mapping to handle MST usecases. A pin is MST capable if pipe is valid value (i.e. not -1) in the notify callback. If pipe = -1, default port is 0 else port = pipe. A MST capable pin widget is programmed by first sending a verb to select the device/port and then the actual verb is sent to program the widget. This patch series enables MST Audio by configuring MST capable pins and extended the jack handling to report event for ports. Now the jack events are based on port connect/disconnect. The design is also extended to support mapping of multiple port to a single converter. Now with the introduction of multiple ports with a single pin, a pin widget is created to enable/disable path based on device connection. Required machine driver changes are done to handle this change. Also use the ASoC jack instead of snd_jack framework to creates jacks in the machine driver. Note: This patch series has dependency on the series "ASoC: hdmi: Preparatory work to support MST audio" Change v2: - Fixed the SOB missing in the patch. - Removed already applied patches. Jeeja KP (1): ASoC: Intel: bxt: Add jack port initialize in bxt_rt298 machine sound/soc/intel/boards/bxt_rt298.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) -- 2.5.0

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