January 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: usb-audio: fix PCM device order
by Alexander Tsoy 27 Jul '19

27 Jul '19

Some cards have alternate setting with non-PCM format as the first altsetting in the interface descriptors. This confuses userspace, since alsa-lib uses device 0 by default. So lets parse interfaces in two steps: 1. Parse altsettings with PCM formats. 2. Parse altsettings with non-PCM formats. This fixes at least following cards: - Audinst HUD-mx2 - Audinst HUD-mini Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/stream.c | 27 ++++++++++++++++++++++++++- 1 file changed, 26 insertions(+), 1 deletion(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 8e9548bc1f1a..b2ebeae91396 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -477,7 +477,8 @@ static struct uac2_output_terminal_descriptor * return NULL; } -int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) +static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no, + bool *has_non_pcm, bool non_pcm) { struct usb_device *dev; struct usb_interface *iface; @@ -634,6 +635,10 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) iface_no, altno); continue; } + if (fmt->bFormatType != UAC_FORMAT_TYPE_I) + *has_non_pcm = true; + if ((fmt->bFormatType == UAC_FORMAT_TYPE_I) == non_pcm) + continue; if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8)) || ((protocol == UAC_VERSION_2) && (fmt->bLength < 6))) { dev_err(&dev->dev, @@ -740,3 +745,23 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) return 0; } +int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) +{ + int err; + bool has_non_pcm = false; + + /* parse PCM formats */ + err = __snd_usb_parse_audio_interface(chip, iface_no, &has_non_pcm, false); + if (err < 0) + return err; + + if (has_non_pcm) { + /* parse non-PCM formats */ + err = __snd_usb_parse_audio_interface(chip, iface_no, &has_non_pcm, true); + if (err < 0) + return err; + } + + return 0; +} + -- 2.13.0

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[alsa-devel] alsa-lib-1.1.6.tar.bz2: some files are missing in the tarball
by Lars Wendler 13 May '19

13 May '19

Hello list, please have a look at this Gentoo bug report [1] where a user reported that he cannot compile alsa-lib-1.1.6 with --enable-alisp configure option. The error message is: make[3]: *** No rule to make target 'SI7018/sndoc-mixer.alisp', needed by 'all-am'. Stop. This happens because the directory src/conf/cards/SI7018 is missing as well as the following files: src/conf/cards/aliases.alisp src/conf/smixer.conf src/conf/sndo-mixer.alisp First I thought those files were removed intentionally but checking the alsa-lib git tree I can see all files still being available. So at Gentoo we fixed this by re-adding the files through a patch [2]. Any chance this was an accidental oversight? Kind regards Lars [1] https://bugs.gentoo.org/652422 [2] https://gitweb.gentoo.org/repo/gentoo.git/tree/media-libs/alsa-lib/files/al… P.S.: Please CC me on replies as I am not subscribed to the list. -- Lars Wendler Gentoo package maintainer GPG: 21CC CF02 4586 0A07 ED93 9F68 498F E765 960E 9B39

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[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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Re: [alsa-devel] RME Fireface alsa driver
by Takashi Sakamoto 23 Apr '19

23 Apr '19

Hi, (C.C.ed to alsa-devel) At first, I apologize that ALSA fireface driver since Linux kernel v4.12 includes some bugs for its packet handling for PCM capture substream[1]. This bug can brings disorder of jackd process. Furthermore, even if handling rx packets correctly, Fireface 400 cannot generate sounds because audio data frame is not picked up from isochronous packets to DSP[2]. I've already posted some patches to fix these bugs. The patches were already merged to ALSA upstream and they'll be going to mainline in next merge window for v4.21 kernel. They were also posted for stable kernel and later you can receive patched kernel image if you use any stable kernels. If you'll test these patches in current your environment, please get recent code from my remote repository for backport to Linux v4.17 or later: https://github.com/takaswie/snd-firewire-improve Your test will be great help to me because I've already left Fireface 400. (I rented it just for two weeks and I'll leave Fireface 800 soon as well.) On Fri, 14 Dec 2018, thet wrote: > OK I have the FF400 set up now. In my recent work for Fireface 800[3][4], I ensure that Fireface 800 can generate sounds even if the HOST LED turns on. This is the same experience with Fireface 400. Thus you can ignore whether the LED turn off or not. > What action should I take to create an error that is meaningful? > .. > when I start audacious from the terminal using the alsa output plugin > and the FF400 selected I get: Any GUI tools are not necessarily good for debugging. Please use 'aplay' insted, like: $ aplay -v -D hw:Fireface400 -r 44100 -c 18 -f S32_LE /dev/urandom $ aplay -v -D hw:Fireface400 -r 88200 -c 14 -f S32_LE /dev/urandom $ aplay -v -D hw:Fireface400 -r 192000 -c 10 -f S32_LE /dev/urandom In the above cases, I expect your unit can generate noisy sounds with the latest code of driver. Anyway, I note you not to drop list address (alsa-devel(a)alsa-project.org) from CC list. It's valuable to share experiences of errors/failures because potentially they are good information for developers to seek bugs in software and be going to fix them. [1] [alsa-devel] [PATCH 1/3] ALSA: firewire-lib: fix wrong handling payload_length as payload_quadlet http://mailman.alsa-project.org/pipermail/alsa-devel/2018-December/143147.h… [2] [alsa-devel] [PATCH] ALSA: fireface: fix for state to fetch PCM frames http://mailman.alsa-project.org/pipermail/alsa-devel/2018-December/143150.h… [3] [alsa-devel] [PATCH 0/8] ALSA: fireface: add support for Fireface 800 with MIDI functionality only http://mailman.alsa-project.org/pipermail/alsa-devel/2018-December/142826.h… [4] [alsa-devel] [PATCH 0/5] ALSA: fireface: add support for Fireface 800 with streaming functionality http://mailman.alsa-project.org/pipermail/alsa-devel/2018-December/143154.h… Regards Takashi Sakamoto

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[alsa-devel] [RFC 0/2] Add MICFIL DAI support
by Cosmin Samoila 27 Mar '19

27 Mar '19

PDM is a popular way to deliver audio from microphones to the processor in several applications, such as mobile telephones. However, current digital-audio systems use multibit audio signal (also known as multibit PCM) to represent the signal. For this purpose a set of FIR, CIC or/and Half Band filters are usually implemented on DSPs or software. This block implements the required digital interface to provide a 16-bit audio signal from a PDM microphone bitstream in a configurable output sampling rate. The implementation of this digital interface is based on the application of digital signal processing techniques in hardware. The PDM Microphone Interface architecture was designed to gate saving and minimal power consumption. It implements a bunch of filters to transform a 1-bit PDM bitstream to a 16-bit PCM signal in the audio band. To avoid aliasing frequencies in passband, the overall filter has 80 dB stopband attenuation and passband ripple less than 0.2dB. The whole module is implemented to work in a multichannel mode. All channels have the same configuration but each input channel could be turned on/off independently. Cosmin-Gabriel Samoila (2): ASoC: micfil: Add bindings for MICFIL DAI ASoC: Add MICFIL SoC Digital Audio Interface driver. .../devicetree/bindings/sound/fsl,micfil.txt | 38 + sound/soc/fsl/Kconfig | 9 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_micfil.c | 2451 ++++++++++++++++++++ sound/soc/fsl/fsl_micfil.h | 317 +++ 5 files changed, 2817 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,micfil.txt create mode 100644 sound/soc/fsl/fsl_micfil.c create mode 100644 sound/soc/fsl/fsl_micfil.h -- 2.7.4

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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 0/3] Add NXP AUDMIX device and machine drivers
by Viorel Suman 26 Mar '19

26 Mar '19

The patchset adds NXP Audio Mixer (AUDMIX) device and machine drivers and related DT bindings documentation. Changes since V3: 1. Removed machine driver DT bindings documentation. 2. Trigger machine driver probe from device driver as suggested by Nicolin. Changes since V2: 1. Moved "dais" node from machine driver DTS node to device driver DTS node as suggested by Rob. Changes since V1: 1. Original patch split into distinct patches for the device driver and DT binding documentation. 2. Replaced AMIX with AUDMIX in both code and file names as it looks more RM-compliant. 3. Removed polarity control from CPU DAI driver as suggested by Nicolin. 4. Added machine driver and related DT binding documentation. Viorel Suman (3): ASoC: fsl: Add Audio Mixer CPU DAI driver ASoC: add fsl_audmix DT binding documentation ASoC: fsl: Add Audio Mixer machine driver .../devicetree/bindings/sound/fsl,audmix.txt | 54 ++ sound/soc/fsl/Kconfig | 16 + sound/soc/fsl/Makefile | 5 + sound/soc/fsl/fsl_audmix.c | 576 +++++++++++++++++++++ sound/soc/fsl/fsl_audmix.h | 102 ++++ sound/soc/fsl/imx-audmix.c | 327 ++++++++++++ 6 files changed, 1080 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,audmix.txt create mode 100644 sound/soc/fsl/fsl_audmix.c create mode 100644 sound/soc/fsl/fsl_audmix.h create mode 100644 sound/soc/fsl/imx-audmix.c -- 2.7.4

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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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