October 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2] ASoC: core: Allow topology to override machine driver FE DAI link config.
by Liam Girdwood 03 May '19

03 May '19

Machine drivers statically define a number of DAI links that currently cannot be changed or removed by topology. This means PCMs and platform components cannot be changed by topology at runtime AND machine drivers are tightly coupled to topology. This patch allows topology to override the machine driver DAI link config in order to reuse machine drivers with different topologies and platform components. The patch supports :- 1) create new FE PCMs with a topology defined PCM ID. 2) destroy existing static FE PCMs 3) change the platform component driver. 4) assign any new HW params fixups. 5) assign a new card name prefix to differentiate this topology to userspace. The patch requires no changes to the machine drivers, but does add some platform component flags that the platform component driver can assign before loading topologies. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Changes since V1. o Now iterate over component_list to fix crash with DT enumerated cards. o Make sure name prefix is only added once with deferred probiing. include/sound/soc.h | 13 ++++++ sound/soc/soc-core.c | 101 +++++++++++++++++++++++++++++++++++++++++-- sound/soc/soc-pcm.c | 12 +++++ 3 files changed, 123 insertions(+), 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index f7579f82cc00..b1d65fcb8756 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -806,6 +806,14 @@ struct snd_soc_component_driver { unsigned int use_pmdown_time:1; /* care pmdown_time at stop */ unsigned int endianness:1; unsigned int non_legacy_dai_naming:1; + + /* this component uses topology and ignore machine driver FEs */ + const char *ignore_machine; + const char *topology_name_prefix; + int (*be_hw_params_fixup)(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params); + bool use_dai_pcm_id; /* use the DAI link PCM ID as PCM device number */ + int be_pcm_base; /* base device ID for all BE PCMs */ }; struct snd_soc_component { @@ -963,6 +971,9 @@ struct snd_soc_dai_link { /* pmdown_time is ignored at stop */ unsigned int ignore_pmdown_time:1; + /* Do not create a PCM for this DAI link (Backend link) */ + unsigned int ignore:1; + struct list_head list; /* DAI link list of the soc card */ struct snd_soc_dobj dobj; /* For topology */ }; @@ -1002,6 +1013,7 @@ struct snd_soc_card { const char *long_name; const char *driver_name; char dmi_longname[80]; + char topology_shortname[32]; struct device *dev; struct snd_card *snd_card; @@ -1011,6 +1023,7 @@ struct snd_soc_card { struct mutex dapm_mutex; bool instantiated; + bool topology_shortname_created; int (*probe)(struct snd_soc_card *card); int (*late_probe)(struct snd_soc_card *card); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4663de3cf495..38bf7d01894b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -852,6 +852,9 @@ static int soc_bind_dai_link(struct snd_soc_card *card, const char *platform_name; int i; + if (dai_link->ignore) + return 0; + dev_dbg(card->dev, "ASoC: binding %s\n", dai_link->name); if (soc_is_dai_link_bound(card, dai_link)) { @@ -1461,7 +1464,9 @@ static int soc_probe_link_dais(struct snd_soc_card *card, { struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int i, ret; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_component *component; + int i, ret, num; dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n", card->name, rtd->num, order); @@ -1507,9 +1512,28 @@ static int soc_probe_link_dais(struct snd_soc_card *card, soc_dpcm_debugfs_add(rtd); #endif + num = rtd->num; + + /* + * most drivers will register their PCMs using DAI link ordering but + * topology based drivers can use the DAI link id field to set PCM + * device number and then use rtd + a base offset of the BEs. + */ + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + if (!component->driver->use_dai_pcm_id) + continue; + + if (rtd->dai_link->no_pcm) + num += component->driver->be_pcm_base; + else + num = rtd->dai_link->id; + } + if (cpu_dai->driver->compress_new) { /*create compress_device"*/ - ret = cpu_dai->driver->compress_new(rtd, rtd->num); + ret = cpu_dai->driver->compress_new(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create compress %s\n", dai_link->stream_name); @@ -1519,7 +1543,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, if (!dai_link->params) { /* create the pcm */ - ret = soc_new_pcm(rtd, rtd->num); + ret = soc_new_pcm(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", dai_link->stream_name, ret); @@ -1846,6 +1870,74 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) EXPORT_SYMBOL_GPL(snd_soc_set_dmi_name); #endif /* CONFIG_DMI */ +static void soc_check_tplg_fes(struct snd_soc_card *card) +{ + struct snd_soc_component *component; + const struct snd_soc_component_driver *comp_drv; + struct snd_soc_dai_link *dai_link; + int i; + + list_for_each_entry(component, &component_list, list) { + + /* does this component override FEs ? */ + if (!component->driver->ignore_machine) + continue; + + /* for this machine ? */ + if (strcmp(component->driver->ignore_machine, + card->dev->driver->name)) + continue; + + /* machine matches, so override the rtd data */ + for (i = 0; i < card->num_links; i++) { + + dai_link = &card->dai_link[i]; + + /* ignore this FE */ + if (dai_link->dynamic) { + dai_link->ignore = true; + continue; + } + + dev_info(card->dev, "info: override FE DAI link %s\n", + card->dai_link[i].name); + + /* override platform component */ + dai_link->platform_name = component->name; + + /* convert non BE into BE */ + dai_link->no_pcm = 1; + + /* override any BE fixups */ + dai_link->be_hw_params_fixup = + component->driver->be_hw_params_fixup; + + /* most BE links don't set stream name, so set it to + * dai link name if it's NULL to help bind widgets. + */ + if (!dai_link->stream_name) + dai_link->stream_name = dai_link->name; + } + + /* Inform userspace we are using alternate topology */ + if (component->driver->topology_name_prefix) { + + /* topology shortname created ? */ + if (!card->topology_shortname_created) { + comp_drv = component->driver; + + snprintf(card->topology_shortname, 32, "%s-%s", + comp_drv->topology_name_prefix, + card->name); + card->topology_shortname_created = true; + } + + /* use topology shortname */ + card->name = card->topology_shortname; + } + } +} + static int snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; @@ -1855,6 +1947,9 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) mutex_lock(&client_mutex); mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); + /* check whether any platform is ignore machine FE and using topology */ + soc_check_tplg_fes(card); + /* bind DAIs */ for (i = 0; i < card->num_links; i++) { ret = soc_bind_dai_link(card, &card->dai_link[i]); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 153218aa4909..db51849ae9bd 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -865,8 +865,20 @@ int soc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; int ret; + /* perform any topology hw_params fixups before DAI */ + if (rtd->dai_link->be_hw_params_fixup) { + ret = rtd->dai_link->be_hw_params_fixup(rtd, params); + if (ret < 0) { + dev_err(rtd->dev, + "ASoC: hw_params topology fixup failed %d\n", + ret); + return ret; + } + } + if (dai->driver->ops->hw_params) { ret = dai->driver->ops->hw_params(substream, params, dai); if (ret < 0) { -- 2.17.1

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[alsa-devel] snd-usb-audio Buffer Sizes and Round Trip Latency
by Jonathan Liu 30 Apr '19

30 Apr '19

Hi, I want to start a discussion regarding round trip latency for class compliant USB audio interfaces on Linux. In particular, I am noticing with my USB 2.0 RME Babyface Pro audio interface that the round trip latency is considerably higher on Linux than on macOS High Sierra and Windows 10. I tested the round trip latency using a loopback audio cable and the ReaInsert plugin included with Reaper DAW (www.reaper.fm) that can be downloaded for Windows/macOS/Linux to calculate the additional delay. Here are the results for 48000 Hz, 24-bit on my RME Babyface Pro: === block_size/periods block_size*periods + additional_delay ~ round_trip_latency round_trip_latency = (block_size*periods + additional_delay) / 48000 * 1000 Linux 4.17.14, Class Compliant Mode (snd-usb-audio, ALSA backend): 16/2 32 + 80 ~ 2.333 ms 16/3 48 + 109 ~ 3.271 ms 32/2 64 + 129 ~ 4.021 ms 32/3 96 + 166 ~ 5.458 ms 64/2 128 + 205 ~ 6.938 ms 64/3 192 + 242 ~ 9.042 ms 128/2 256 + 352 ~ 12.667 ms 128/3 384 + 496 ~ 18.334 ms 256/2 512 + 650 ~ 24.208 ms 256/3 768 + 650 ~ 29.542 ms 512/2 1024 + 634 ~ 34.542 ms 512/3 1536 + 634 ~ 45.208 ms 1024/2 2048 + 650 ~ 56.208 ms 1024/3 3072 + 650 ~ 77.542 ms 2048/2 4096 + 633 ~ 98.521 ms 2048/3 6144 + 633 ~ 141.188 ms macOS High Sierra, Class Compliant Mode (Apple Driver): 16/2 32 + 205 ~ 4.938 ms 32/2 64 + 205 ~ 5.604 ms 64/2 128 + 205 ~ 6.938 ms 128/2 256 + 205 ~ 9.604 ms 256/2 512 + 205 ~ 14.938 ms 512/2 1024 + 205 ~ 25.604 ms 1024/2 2048 + 205 ~ 46.938 ms 2048/2 4096 + 205 ~ 89.604 ms macOS High Sierra, PC Mode (RME Driver v3.08): 16/2 32 + 59 ~ 1.896 ms 32/2 64 + 59 ~ 2.563 ms 64/2 128 + 59 ~ 3.896 ms 128/2 256 + 59 ~ 6.563 ms 256/2 512 + 59 ~ 11.596 ms 512/2 1024 + 59 ~ 22.563 ms 1024/2 2048 + 59 ~ 43.896 ms 2048/2 4096 + 59 ~ 86.563 ms Windows 10, PC Mode (RME Driver 1.099): 48/2 96 + 63 ~ 3.313 ms 64/2 128 + 63 ~ 3.979 ms 96/2 192 + 63 ~ 5.313 ms 128/2 256 + 63 ~ 6.646 ms 256/2 512 + 63 ~ 11.979 ms 512/2 1024 + 63 ~ 22.646 ms 1024/2 2048 + 63 ~ 43.979 ms 2048/2 4096 + 63 ~ 86.646 ms === Some things in particular I noticed on Linux: - additional_delay varies a bit if I close and open the audio device again - additional_delay seems to increase as the block_size increases. I can make the additional_delay stay about the same rather than increasing by setting MAX_PACKS and MAX_PACKS_HS to 1 in sound/usb/card.h. In Linux versions before 3.13 there was a nrpacks parameter for snd-usb-audio to control this but it was removed with commit https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?… - additional_delay is not constant as block_size is increased like on macOS and Windows I made a patch to snd-usb-audio to expose the snd-usb-audio constants as runtime adjustable module parameters (/sys/module/snd_usb_audio/parameters/) for testing (takes effect when the device is disconnected+reconnected and logs the parameter values to dmesg): https://aur.archlinux.org/cgit/aur.git/plain/parameters.patch?h=snd-usb-aud… The patch is used in my Arch Linux AUR package for convenience (using DKMS to avoid having to recompile entire kernel): https://aur.archlinux.org/packages/snd-usb-audio-lowlatency-dkms/ Can snd-usb-audio be improved so the additional_delay is always the same when closing/opening/reconfiguring the audio device and does not increase as the block_size increases? I noticed using USB audio on Linux at lower latencies (block_size <= 128) is more prone to audio dropouts under load compared to macOS and Windows, even with CPU power management disabled (writing 0 to /dev/cpu_dma_latency). What can be done about this? Thanks. Regards, Jonathan

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[alsa-devel] [PATCH] ASoC: rockchip: add missing INTERLEAVED PCM attribute
by Katsuhiro Suzuki 24 Apr '19

24 Apr '19

This patch adds SNDRV_PCM_INFO_INTERLEAVED into PCM hardware info. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- sound/soc/rockchip/rockchip_pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/rockchip/rockchip_pcm.c b/sound/soc/rockchip/rockchip_pcm.c index f77538319221..9e7b5fa4cf59 100644 --- a/sound/soc/rockchip/rockchip_pcm.c +++ b/sound/soc/rockchip/rockchip_pcm.c @@ -21,7 +21,8 @@ static const struct snd_pcm_hardware snd_rockchip_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_INTERLEAVED, .period_bytes_min = 32, .period_bytes_max = 8192, .periods_min = 1, -- 2.18.0

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[alsa-devel] [PATCH RFC 00/26] Kill set_fs() in ALSA codes
by Takashi Iwai 27 Mar '19

27 Mar '19

Hi, this is a slightly lengthy patchset, basically targeted for 4.13, thus it's still in RFC. The main purpose of this patchset is to get rid of get_fs() / set_fs() usages in ALSA codes. The patchset starts from a patch that looks fairly irrelevant with the goal (the conversion of HD-audio bind-mute code), but it's a step for the long trail. The biggest change in the patchset is the conversion of copy & silence PCM ops into the new copy_silence ops. It helped for reducing the code size and for handling the in-kernel buffer copies. The current patches are found in topic/kill-set_fs branch of my sound.git tree. thanks, Takashi === Takashi Iwai (26): ALSA: hda - Simplify bound-beep mute control for ALC268 ALSA: hda - Move bind-mixer switch codes to generic parser ALSA: hda - Remove the generic bind ctl helpers ALSA: hda - Remove the use of set_fs() ALSA: hda - Fix a typo in comment ALSA: hda - Remove superfluous header inclusions ALSA: opl3: Kill unused set_fs() ALSA: emu10k1: Get rid of set_fs() usage ALSA: pcm: Remove set_fs() in PCM core code ALSA: pcm: Introduce copy_silence PCM ops ALSA: Update document about copy_silence PCM ops ALSA: dummy: Convert to copy_silence ops ALSA: es1938: Convert to copy_silence ops ALSA: korg1212: Convert to copy_silence ops ALSA: nm256: Convert to copy_silence ops ALSA: rme32: Convert to copy_silence ops ALSA: rme96: Convert to copy_silence ops ALSA: rme9652: Convert to copy_silence ops ALSA: hdsp: Convert to copy_silence ops ALSA: gus: Convert to copy_silence ops ALSA: sb: Convert to copy_silence ops ALSA: sh: Convert to copy_silence ops ASoC: blackfin: Convert to copy_silence ops [media] solo6x10: Convert to copy_silence ops ALSA: pcm: Drop the old copy and silence ops ALSA: pcm: Kill set_fs() usage in OSS layer and USB gadget .../sound/kernel-api/writing-an-alsa-driver.rst | 110 +++++---- drivers/media/pci/solo6x10/solo6x10-g723.c | 11 +- drivers/usb/gadget/function/u_uac1.c | 8 +- include/sound/pcm.h | 19 +- sound/core/oss/pcm_oss.c | 62 ++--- sound/core/pcm_compat.c | 22 +- sound/core/pcm_lib.c | 174 +++++++++----- sound/core/pcm_native.c | 122 ++++++---- sound/drivers/dummy.c | 13 +- sound/drivers/opl3/opl3_oss.c | 14 -- sound/isa/gus/gus_pcm.c | 43 +--- sound/isa/sb/emu8000_pcm.c | 99 +++----- sound/pci/emu10k1/emufx.c | 127 ++++++----- sound/pci/es1938.c | 11 +- sound/pci/hda/hda_codec.c | 251 +++------------------ sound/pci/hda/hda_generic.c | 46 +++- sound/pci/hda/hda_local.h | 61 ----- sound/pci/hda/patch_realtek.c | 37 ++- sound/pci/korg1212/korg1212.c | 128 +++-------- sound/pci/nm256/nm256.c | 35 ++- sound/pci/rme32.c | 49 ++-- sound/pci/rme96.c | 52 ++--- sound/pci/rme9652/hdsp.c | 44 ++-- sound/pci/rme9652/rme9652.c | 46 ++-- sound/sh/sh_dac_audio.c | 40 +--- sound/soc/blackfin/bf5xx-ac97-pcm.c | 6 +- sound/soc/blackfin/bf5xx-ac97.c | 18 +- sound/soc/blackfin/bf5xx-i2s-pcm.c | 46 ++-- sound/soc/soc-pcm.c | 3 +- 29 files changed, 696 insertions(+), 1001 deletions(-) -- 2.12.2

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[alsa-devel] [PATCH v4] ALSA: core: Allow drivers to set R/W wait time.
by Liam Girdwood 26 Mar '19

26 Mar '19

Currently ALSA core blocks userspace for about 10 seconds for PCM R/W IO. This needs to be configurable for modern hardware like DSPs where no pointer update in milliseconds can indicate terminal DSP errors. Add a substream variable to set the wait time in ms. This allows userspace and drivers to recover more quickly from terminal DSP errors. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Changes since V1 :- o Remove API method and alow drivers to set directly. o Validate time is driver supplied times. V2 :- o Max wait calc now in ms. o checkpatch. v3:- o Remove driver supplied time validation. include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 17 ++++++++++++----- 2 files changed, 13 insertions(+), 5 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e054c583d3b3..fcdf358a25f0 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -462,6 +462,7 @@ struct snd_pcm_substream { /* -- timer section -- */ struct snd_timer *timer; /* timer */ unsigned timer_running: 1; /* time is running */ + long wait_time; /* time in ms for R/W to wait for avail */ /* -- next substream -- */ struct snd_pcm_substream *next; /* -- linked substreams -- */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 44b5ae833082..eecbfdc7a2c9 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1832,12 +1832,19 @@ static int wait_for_avail(struct snd_pcm_substream *substream, if (runtime->no_period_wakeup) wait_time = MAX_SCHEDULE_TIMEOUT; else { - wait_time = 10; - if (runtime->rate) { - long t = runtime->period_size * 2 / runtime->rate; - wait_time = max(t, wait_time); + /* use wait time from substream if available */ + if (substream->wait_time) { + wait_time = substream->wait_time; + } else { + wait_time = 10; + + if (runtime->rate) { + long t = runtime->period_size * 2 / + runtime->rate; + wait_time = max(t, wait_time); + } + wait_time = msecs_to_jiffies(wait_time * 1000); } - wait_time = msecs_to_jiffies(wait_time * 1000); } for (;;) { -- 2.17.1

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[alsa-devel] snd_hda_codec_hdmi: `hdaudio hdaudioC0D2: Unable to bind the codec`
by Paul Menzel 18 Feb '19

18 Feb '19

Dear Linux folks, With drm-tip (v4.19-rc3-946-g09b295662edd) Linux prints `snd_hda_codec_hdmi hdaudioC0D2: No i915 binding for Intel HDMI/DP codec` to the log. ``` […] [ 12.481788] snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC671: line_outs=1 (0x21/0x0/0x0/0x0/0x0) type:line [ 12.482536] snd_hda_codec_realtek hdaudioC0D0: speaker_outs=1 (0x17/0x0/0x0/0x0/0x0) [ 12.483184] snd_hda_codec_realtek hdaudioC0D0: hp_outs=1 (0x14/0x0/0x0/0x0/0x0) [ 12.483737] snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 [ 12.484198] snd_hda_codec_realtek hdaudioC0D0: inputs: [ 12.484582] snd_hda_codec_realtek hdaudioC0D0: Front Mic=0x19 [ 12.485100] snd_hda_codec_realtek hdaudioC0D0: Rear Mic=0x18 [ 12.485519] snd_hda_codec_realtek hdaudioC0D0: Line=0x1a [ 12.497685] snd_hda_codec_hdmi hdaudioC0D2: No i915 binding for Intel HDMI/DP codec [ 12.498311] hdaudio hdaudioC0D2: Unable to bind the codec [ 12.498789] input: HDA Intel PCH Front Mic as /devices/pci0000:00/0000:00:1f.3/sound/card0/input6 [ 12.499567] input: HDA Intel PCH Rear Mic as /devices/pci0000:00/0000:00:1f.3/sound/card0/input7 [ 12.500268] input: HDA Intel PCH Line as /devices/pci0000:00/0000:00:1f.3/sound/card0/input8 [ 12.501039] input: HDA Intel PCH Line Out as /devices/pci0000:00/0000:00:1f.3/sound/card0/input9 [ 12.501683] input: HDA Intel PCH Front Headphone as /devices/pci0000:00/0000:00:1f.3/sound/card0/input10 [ 17.743338] [drm:intel_pch_type [i915]] Found SunrisePoint PCH […] ``` Is that an error or just a notice? Kind regards, Paul

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[alsa-devel] pcm_dshare calculates size incorrectly on buffer rollover
by Brendan Shanks 11 Feb '19

11 Feb '19

I'm working on an embedded system based on an Ambarella H1 SoC (32-bit ARM Cortex A9). Audio playback through ALSA (and GStreamer) works fine when playing to the raw hw device. When playing through dshare (my normal configuration), playback stops after exactly 7 hours 16 minutes, for about 3 minutes. During the 3 minutes, the playback thread consumes an entire CPU core. After the 3 minutes, GStreamer reports an xrun, recovers from it, and playback goes back to normal. After some debugging, the problem seems to be in snd_pcm_dshare_sync_area(). When dshare->appl_ptr rolls over to 0, 'size' becomes huge, 3036676576. 'slave_size' is also much bigger than it should be, 1258291680. This is what 'size' is set to when the for loop starts, and I believe the for loop then spends ~3 minutes copying a huge amount of samples. This also explains the 7h16m time, it's linked to the PCM boundary which is 1258291200. At 48 kHz, 1258291200 samples takes 7h16m54s. I'm not sure what the fix should be though. Is this really a bug in dshare, or are bad values being set somewhere else? GStreamer? Or maybe the period/buffer size (480/9600) is causing problems? Any advice is appreciated, my modified snd_pcm_dshare_sync_area() and the output are below. Brendan Shanks $ more /proc/asound/card0/pcm1p/sub0/hw_params access: MMAP_INTERLEAVED format: S16_LE subformat: STD channels: 4 rate: 48000 (48000/1) period_size: 480 buffer_size: 9600 $ more /proc/asound/card0/pcm1p/sub0/sw_params tstamp_mode: ENABLE period_step: 1 avail_min: 480 start_threshold: 1 stop_threshold: 1258291200 silence_threshold: 0 silence_size: 1258291200 boundary: 1258291200 --- output from running 'gst-launch-1.0 -e audiotestsrc ! alsasink' with prints included as below snd_pcm_dshare_sync_area size 3036676576, dshare->appl_ptr 0 dshare->last_appl_ptr 1258290720 slave_hw_ptr1 1258282080 slave_period_size 480 slave_hw_ptr2 1258282080 slave_buffer_size 9600 slave_hw_ptr3 1258291680 slave_boundary 1258291200 slave_hw_ptr4 1258291680 slave_appl_ptr 0 slave_size1 1258291680 size 1258291680 appl_ptr 9120 size 1258291680 boundary 1258291200 last_appl_ptr 0 slave_appl_ptr 0 dshare->slave_appl_ptr 480 exiting snd_pcm_dshare_sync_area size 3036676576, dshare->appl_ptr 0 dshare->last_appl_ptr 1258290720 slave_hw_ptr1 8965920 slave_period_size 480 slave_hw_ptr2 8965920 slave_buffer_size 9600 slave_hw_ptr3 8975520 slave_boundary 1258291200 slave_hw_ptr4 8975520 slave_appl_ptr 8975040 slave_size1 480 size 480 appl_ptr 9120 size 480 boundary 1258291200 last_appl_ptr 0 slave_appl_ptr 8640 dshare->slave_appl_ptr 8975520 exiting --- snd_pcm_dshare_sync_area() from alsa-lib-1.1.6 with prints added: static void snd_pcm_dshare_sync_area(snd_pcm_t *pcm) { snd_pcm_direct_t *dshare = pcm->private_data; snd_pcm_uframes_t slave_hw_ptr, slave_appl_ptr, slave_size; snd_pcm_uframes_t appl_ptr, size; const snd_pcm_channel_area_t *src_areas, *dst_areas; int print = 0; /* calculate the size to transfer */ size = dshare->appl_ptr - dshare->last_appl_ptr; if (! size) return; if (size > 9600) { printf("snd_pcm_dshare_sync_area size %lu, dshare->appl_ptr %lu dshare->last_appl_ptr %lu\n", size, dshare->appl_ptr, dshare->last_appl_ptr); print = 1; } slave_hw_ptr = dshare->slave_hw_ptr; if (print) printf("slave_hw_ptr1 %lu slave_period_size %lu", slave_hw_ptr, dshare->slave_period_size); /* don't write on the last active period - this area may be cleared * by the driver during write operation... */ slave_hw_ptr -= slave_hw_ptr % dshare->slave_period_size; if (print) printf("slave_hw_ptr2 %lu slave_buffer_size %lu", slave_hw_ptr, dshare->slave_buffer_size); slave_hw_ptr += dshare->slave_buffer_size; if (print) printf("slave_hw_ptr3 %lu slave_boundary %lu", slave_hw_ptr, dshare->slave_boundary); if (dshare->slave_hw_ptr > dshare->slave_boundary) slave_hw_ptr -= dshare->slave_boundary; if (print) printf("slave_hw_ptr4 %lu slave_appl_ptr %lu", slave_hw_ptr, dshare->slave_appl_ptr); if (slave_hw_ptr < dshare->slave_appl_ptr) slave_size = slave_hw_ptr + (dshare->slave_boundary - dshare->slave_appl_ptr); else slave_size = slave_hw_ptr - dshare->slave_appl_ptr; if (print) printf("slave_size1 %lu", slave_size); if (slave_size < size) size = slave_size; if (! size) return; if (print) printf("size %lu", size); /* add sample areas here */ src_areas = snd_pcm_mmap_areas(pcm); dst_areas = snd_pcm_mmap_areas(dshare->spcm); appl_ptr = dshare->last_appl_ptr % pcm->buffer_size; if (print) printf("appl_ptr %lu size %lu boundary %lu", appl_ptr, size, pcm->boundary); dshare->last_appl_ptr += size; dshare->last_appl_ptr %= pcm->boundary; if (print) printf("last_appl_ptr %lu", dshare->last_appl_ptr); slave_appl_ptr = dshare->slave_appl_ptr % dshare->slave_buffer_size; if (print) printf("slave_appl_ptr %lu", slave_appl_ptr); dshare->slave_appl_ptr += size; dshare->slave_appl_ptr %= dshare->slave_boundary; if (print) printf("dshare->slave_appl_ptr %lu", dshare->slave_appl_ptr); for (;;) { snd_pcm_uframes_t transfer = size; if (appl_ptr + transfer > pcm->buffer_size) transfer = pcm->buffer_size - appl_ptr; if (slave_appl_ptr + transfer > dshare->slave_buffer_size) transfer = dshare->slave_buffer_size - slave_appl_ptr; share_areas(dshare, src_areas, dst_areas, appl_ptr, slave_appl_ptr, transfer); size -= transfer; if (! size) break; slave_appl_ptr += transfer; slave_appl_ptr %= dshare->slave_buffer_size; appl_ptr += transfer; appl_ptr %= pcm->buffer_size; } if (print) printf("exiting");

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[alsa-devel] [PATCH v1] ASoC: Intel: Skylake: Switch to modern UUID API
by Andy Shevchenko 21 Jan '19

21 Jan '19

Switch the driver to use modern UUID API, i.e. guid_t type and accompanying functions, such as guid_equal(). Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/skylake/skl-pcm.c | 12 ++++++------ sound/soc/intel/skylake/skl-sst-dsp.h | 6 +++--- sound/soc/intel/skylake/skl-sst-utils.c | 23 +++++++---------------- sound/soc/intel/skylake/skl-sst.c | 4 ++-- sound/soc/intel/skylake/skl-topology.c | 24 ++++++++++++------------ sound/soc/intel/skylake/skl-topology.h | 6 +++--- 6 files changed, 33 insertions(+), 42 deletions(-) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 823e39103edd..86f8ebc77a32 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1268,12 +1268,12 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig) { struct skl_sst *ctx = skl->skl_sst; struct skl_module_inst_id *pin_id; - uuid_le *uuid_mod, *uuid_tplg; + guid_t *uuid_mod, *uuid_tplg; struct skl_module *skl_module; struct uuid_module *module; int i, ret = -EIO; - uuid_mod = (uuid_le *)mconfig->guid; + uuid_mod = (guid_t *)mconfig->guid; if (list_empty(&ctx->uuid_list)) { dev_err(ctx->dev, "Module list is empty\n"); @@ -1281,7 +1281,7 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig) } list_for_each_entry(module, &ctx->uuid_list, list) { - if (uuid_le_cmp(*uuid_mod, module->uuid) == 0) { + if (guid_equal(uuid_mod, &module->uuid)) { mconfig->id.module_id = module->id; if (mconfig->module) mconfig->module->loadable = module->is_loadable; @@ -1298,7 +1298,7 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig) for (i = 0; i < skl->nr_modules; i++) { skl_module = skl->modules[i]; uuid_tplg = &skl_module->uuid; - if (!uuid_le_cmp(*uuid_mod, *uuid_tplg)) { + if (guid_equal(uuid_mod, uuid_tplg)) { mconfig->module = skl_module; ret = 0; break; @@ -1310,13 +1310,13 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig) list_for_each_entry(module, &ctx->uuid_list, list) { for (i = 0; i < MAX_IN_QUEUE; i++) { pin_id = &mconfig->m_in_pin[i].id; - if (!uuid_le_cmp(pin_id->mod_uuid, module->uuid)) + if (guid_equal(&pin_id->mod_uuid, &module->uuid)) pin_id->module_id = module->id; } for (i = 0; i < MAX_OUT_QUEUE; i++) { pin_id = &mconfig->m_out_pin[i].id; - if (!uuid_le_cmp(pin_id->mod_uuid, module->uuid)) + if (guid_equal(&pin_id->mod_uuid, &module->uuid)) pin_id->module_id = module->id; } } diff --git a/sound/soc/intel/skylake/skl-sst-dsp.h b/sound/soc/intel/skylake/skl-sst-dsp.h index e1d6f6719f7e..cbc7a93d56c2 100644 --- a/sound/soc/intel/skylake/skl-sst-dsp.h +++ b/sound/soc/intel/skylake/skl-sst-dsp.h @@ -177,7 +177,7 @@ struct skl_dsp_loader_ops { #define MAX_INSTANCE_BUFF 2 struct uuid_module { - uuid_le uuid; + guid_t uuid; int id; int is_loadable; int max_instance; @@ -241,8 +241,8 @@ void bxt_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx); int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, unsigned int offset, int index); -int skl_get_pvt_id(struct skl_sst *ctx, uuid_le *uuid_mod, int instance_id); -int skl_put_pvt_id(struct skl_sst *ctx, uuid_le *uuid_mod, int *pvt_id); +int skl_get_pvt_id(struct skl_sst *ctx, guid_t *uuid_mod, int instance_id); +int skl_put_pvt_id(struct skl_sst *ctx, guid_t *uuid_mod, int *pvt_id); int skl_get_pvt_instance_id_map(struct skl_sst *ctx, int module_id, int instance_id); void skl_freeup_uuid_list(struct skl_sst *ctx); diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c index 2ae405617876..85551321c35b 100644 --- a/sound/soc/intel/skylake/skl-sst-utils.c +++ b/sound/soc/intel/skylake/skl-sst-utils.c @@ -21,17 +21,11 @@ #include "../common/sst-dsp-priv.h" #include "skl-sst-ipc.h" - -#define UUID_STR_SIZE 37 #define DEFAULT_HASH_SHA256_LEN 32 /* FW Extended Manifest Header id = $AE1 */ #define SKL_EXT_MANIFEST_HEADER_MAGIC 0x31454124 -struct UUID { - u8 id[16]; -}; - union seg_flags { u32 ul; struct { @@ -65,7 +59,7 @@ struct module_type { struct adsp_module_entry { u32 struct_id; u8 name[8]; - struct UUID uuid; + u8 uuid[16]; struct module_type type; u8 hash1[DEFAULT_HASH_SHA256_LEN]; u32 entry_point; @@ -184,13 +178,13 @@ static inline int skl_pvtid_128(struct uuid_module *module) * This generates a 128 bit private unique id for a module TYPE so that * module instance is unique */ -int skl_get_pvt_id(struct skl_sst *ctx, uuid_le *uuid_mod, int instance_id) +int skl_get_pvt_id(struct skl_sst *ctx, guid_t *uuid_mod, int instance_id) { struct uuid_module *module; int pvt_id; list_for_each_entry(module, &ctx->uuid_list, list) { - if (uuid_le_cmp(*uuid_mod, module->uuid) == 0) { + if (guid_equal(uuid_mod, &module->uuid)) { pvt_id = skl_pvtid_128(module); if (pvt_id >= 0) { @@ -214,13 +208,13 @@ EXPORT_SYMBOL_GPL(skl_get_pvt_id); * * This frees a 128 bit private unique id previously generated */ -int skl_put_pvt_id(struct skl_sst *ctx, uuid_le *uuid_mod, int *pvt_id) +int skl_put_pvt_id(struct skl_sst *ctx, guid_t *uuid_mod, int *pvt_id) { int i; struct uuid_module *module; list_for_each_entry(module, &ctx->uuid_list, list) { - if (uuid_le_cmp(*uuid_mod, module->uuid) == 0) { + if (guid_equal(uuid_mod, &module->uuid)) { if (*pvt_id != 0) i = (*pvt_id) / 64; @@ -247,7 +241,6 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, struct adsp_fw_hdr *adsp_hdr; struct adsp_module_entry *mod_entry; int i, num_entry, size; - uuid_le *uuid_bin; const char *buf; struct skl_sst *skl = ctx->thread_context; struct uuid_module *module; @@ -279,8 +272,7 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, return -EINVAL; } - mod_entry = (struct adsp_module_entry *) - (buf + offset + adsp_hdr->len); + mod_entry = (struct adsp_module_entry *)(buf + offset + adsp_hdr->len); num_entry = adsp_hdr->num_modules; @@ -307,8 +299,7 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, goto free_uuid_list; } - uuid_bin = (uuid_le *)mod_entry->uuid.id; - memcpy(&module->uuid, uuid_bin, sizeof(module->uuid)); + guid_copy(&module->uuid, (guid_t *)&mod_entry->uuid); module->id = (i | (index << 12)); module->is_loadable = mod_entry->type.load_type; diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index 5951bbdf1f1a..92806119a755 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -420,9 +420,9 @@ static int skl_load_module(struct sst_dsp *ctx, u16 mod_id, u8 *guid) struct skl_module_table *module_entry = NULL; int ret = 0; char mod_name[64]; /* guid str = 32 chars + 4 hyphens */ - uuid_le *uuid_mod; + guid_t *uuid_mod; - uuid_mod = (uuid_le *)guid; + uuid_mod = (guid_t *)guid; snprintf(mod_name, sizeof(mod_name), "%s%pUL%s", "intel/dsp_fw_", uuid_mod, ".bin"); diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 76dde12cc2bb..160a5c864b3f 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -577,7 +577,7 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe) int ret = 0; list_for_each_entry(w_module, &pipe->w_list, node) { - uuid_le *uuid_mod; + guid_t *uuid_mod; w = w_module->w; mconfig = w->priv; @@ -585,7 +585,7 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe) if (mconfig->id.module_id < 0) { dev_err(skl->skl_sst->dev, "module %pUL id not populated\n", - (uuid_le *)mconfig->guid); + (guid_t *)mconfig->guid); return -EIO; } @@ -619,7 +619,7 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe) * FE/BE params */ skl_tplg_update_module_params(w, ctx); - uuid_mod = (uuid_le *)mconfig->guid; + uuid_mod = (guid_t *)mconfig->guid; mconfig->id.pvt_id = skl_get_pvt_id(ctx, uuid_mod, mconfig->id.instance_id); if (mconfig->id.pvt_id < 0) @@ -658,9 +658,9 @@ static int skl_tplg_unload_pipe_modules(struct skl_sst *ctx, struct skl_module_cfg *mconfig = NULL; list_for_each_entry(w_module, &pipe->w_list, node) { - uuid_le *uuid_mod; + guid_t *uuid_mod; mconfig = w_module->w->priv; - uuid_mod = (uuid_le *)mconfig->guid; + uuid_mod = (guid_t *)mconfig->guid; if (mconfig->module->loadable && ctx->dsp->fw_ops.unload_mod && mconfig->m_state > SKL_MODULE_UNINIT) { @@ -916,12 +916,12 @@ static int skl_tplg_set_module_bind_params(struct snd_soc_dapm_widget *w, return 0; } -static int skl_get_module_id(struct skl_sst *ctx, uuid_le *uuid) +static int skl_get_module_id(struct skl_sst *ctx, guid_t *uuid) { struct uuid_module *module; list_for_each_entry(module, &ctx->uuid_list, list) { - if (uuid_le_cmp(*uuid, module->uuid) == 0) + if (guid_equal(uuid, &module->uuid)) return module->id; } @@ -2121,11 +2121,11 @@ static int skl_tplg_add_pipe(struct device *dev, return 0; } -static int skl_tplg_get_uuid(struct device *dev, u8 *guid, +static int skl_tplg_get_uuid(struct device *dev, guid_t *guid, struct snd_soc_tplg_vendor_uuid_elem *uuid_tkn) { if (uuid_tkn->token == SKL_TKN_UUID) { - memcpy(guid, &uuid_tkn->uuid, 16); + guid_copy(guid, (guid_t *)&uuid_tkn->uuid); return 0; } @@ -2151,7 +2151,7 @@ static int skl_tplg_fill_pin(struct device *dev, break; case SKL_TKN_UUID: - ret = skl_tplg_get_uuid(dev, m_pin[pin_index].id.mod_uuid.b, + ret = skl_tplg_get_uuid(dev, &m_pin[pin_index].id.mod_uuid, (struct snd_soc_tplg_vendor_uuid_elem *)tkn_elem); if (ret < 0) return ret; @@ -2666,7 +2666,7 @@ static int skl_tplg_get_tokens(struct device *dev, case SND_SOC_TPLG_TUPLE_TYPE_UUID: if (is_module_guid) { - ret = skl_tplg_get_uuid(dev, mconfig->guid, + ret = skl_tplg_get_uuid(dev, (guid_t *)mconfig->guid, array->uuid); is_module_guid = false; } else { @@ -3484,7 +3484,7 @@ static int skl_tplg_get_manifest_uuid(struct device *dev, if (uuid_tkn->token == SKL_TKN_UUID) { mod = skl->modules[ref_count]; - memcpy(&mod->uuid, &uuid_tkn->uuid, sizeof(uuid_tkn->uuid)); + guid_copy(&mod->uuid, (guid_t *)&uuid_tkn->uuid); ref_count++; } else { dev_err(dev, "Not an UUID token tkn %d\n", uuid_tkn->token); diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index 82282cac9751..9311e248b6f9 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -224,7 +224,7 @@ struct skl_mod_inst_map { struct skl_uuid_inst_map { u16 inst_id; u16 reserved; - uuid_le mod_uuid; + guid_t mod_uuid; } __packed; struct skl_kpb_params { @@ -236,7 +236,7 @@ struct skl_kpb_params { }; struct skl_module_inst_id { - uuid_le mod_uuid; + guid_t mod_uuid; int module_id; u32 instance_id; int pvt_id; @@ -369,7 +369,7 @@ struct skl_module_res { }; struct skl_module { - uuid_le uuid; + guid_t uuid; u8 loadable; u8 input_pin_type; u8 output_pin_type; -- 2.18.0

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[alsa-devel] [PATCH v1 0/3] ASoC: last minute fixes
by Marcel Ziswiler 15 Jan '19

15 Jan '19

This series comes with some last minutes fixes and clean-up. Changes in v1: - Split from the Tegra series as suggested by Mark. - Fix issue in soc-core rather than working around it in tegra_sgtl5000. Marcel Ziswiler (3): ASoC: tegra_sgtl5000: fix device_node refcounting ASoC: soc-core: fix trivial checkpatch issues ASoC: soc-core: fix platform name vs. of_node assignement sound/soc/soc-core.c | 162 ++++++++++++++++++++++++--------------- sound/soc/tegra/tegra_sgtl5000.c | 17 +++- 2 files changed, 116 insertions(+), 63 deletions(-) -- 2.14.4

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[alsa-devel] [PATCH v2] staging: bcm2835-audio: interpolate audio delay
by Mike Brady 01 Jan '19

01 Jan '19

When the BCM2835 audio output is used, userspace sees a jitter up to 10ms in the audio position, aka "delay" -- the number of frames that must be output before a new frame would be played. Make this a bit nicer for userspace by interpolating the position using the CPU clock. The overhead is small -- an extra ktime_get() every time a GPU message is sent -- and another call and a few calculations whenever the delay is sought from userland. At 48,000 frames per second, i.e. approximately 20 microseconds per frame, it would take a clock inaccuracy of 20 microseconds in 10 milliseconds -- 2,000 parts per million -- to result in an inaccurate estimate, whereas crystal- or resonator-based clocks typically have an inaccuracy of 10s to 100s of parts per million. Signed-off-by: Mike Brady <mikebrady(a)eircom.net> --- Changes in v2 -- remove inappropriate addition of SNDRV_PCM_INFO_BATCH flag .../vc04_services/bcm2835-audio/bcm2835-pcm.c | 20 +++++++++++++++++++ .../vc04_services/bcm2835-audio/bcm2835.h | 1 + 2 files changed, 21 insertions(+) diff --git a/drivers/staging/vc04_services/bcm2835-audio/bcm2835-pcm.c b/drivers/staging/vc04_services/bcm2835-audio/bcm2835-pcm.c index e66da11af5cf..9053b996cada 100644 --- a/drivers/staging/vc04_services/bcm2835-audio/bcm2835-pcm.c +++ b/drivers/staging/vc04_services/bcm2835-audio/bcm2835-pcm.c @@ -74,6 +74,7 @@ void bcm2835_playback_fifo(struct bcm2835_alsa_stream *alsa_stream, atomic_set(&alsa_stream->pos, pos); alsa_stream->period_offset += bytes; + alsa_stream->interpolate_start = ktime_get(); if (alsa_stream->period_offset >= alsa_stream->period_size) { alsa_stream->period_offset %= alsa_stream->period_size; snd_pcm_period_elapsed(substream); @@ -243,6 +244,7 @@ static int snd_bcm2835_pcm_prepare(struct snd_pcm_substream *substream) atomic_set(&alsa_stream->pos, 0); alsa_stream->period_offset = 0; alsa_stream->draining = false; + alsa_stream->interpolate_start = ktime_get(); return 0; } @@ -292,6 +294,24 @@ snd_bcm2835_pcm_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct bcm2835_alsa_stream *alsa_stream = runtime->private_data; + ktime_t now = ktime_get(); + + /* Give userspace better delay reporting by interpolating between GPU + * notifications, assuming audio speed is close enough to the clock + * used for ktime + */ + + if ((ktime_to_ns(alsa_stream->interpolate_start)) && + (ktime_compare(alsa_stream->interpolate_start, now) < 0)) { + u64 interval = + (ktime_to_ns(ktime_sub(now, + alsa_stream->interpolate_start))); + u64 frames_output_in_interval = + div_u64((interval * runtime->rate), 1000000000); + snd_pcm_sframes_t frames_output_in_interval_sized = + -frames_output_in_interval; + runtime->delay = frames_output_in_interval_sized; + } return snd_pcm_indirect_playback_pointer(substream, &alsa_stream->pcm_indirect, diff --git a/drivers/staging/vc04_services/bcm2835-audio/bcm2835.h b/drivers/staging/vc04_services/bcm2835-audio/bcm2835.h index e13435d1c205..595ad584243f 100644 --- a/drivers/staging/vc04_services/bcm2835-audio/bcm2835.h +++ b/drivers/staging/vc04_services/bcm2835-audio/bcm2835.h @@ -78,6 +78,7 @@ struct bcm2835_alsa_stream { unsigned int period_offset; unsigned int buffer_size; unsigned int period_size; + ktime_t interpolate_start; struct bcm2835_audio_instance *instance; int idx; -- 2.17.1

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