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

[alsa-devel] [PATCH - AAF PCM plugin 0/7] Follow-up improvements
by Andre Guedes 10 Dec '18

10 Dec '18

Hi all, This patch series provides some follow-up improvements to the AVTP Audio Format (AAF) plugin. The highlight of this series is the implementation of a transmission offload mechanism which improves the plugin performance considerably. Details are provided in the following paragraphs. Currently, the AVTPDU transmission interval is controlled by the AAF plugin, in software. At every timer expiration, the plugin transmits one AVTPDU. This has some implications in terms of task scheduling. For instance, in class A streams, AVTPDUs are transmitted at every 125 us which means the application task should be scheduled-in at every 125 us. In a general purpose Linux system, such scheduling interval can be hard to cope with in the long-run. To mitigate that issue, this series introduces a transmission offload mechanism that leverages the SO_TXTIME sockopt and ETF qdisc features recently introduced to kernel 4.19. Instead of sending one AVTPDU at every timer expiration, the plugin sends several AVTPDUs at once to the kernel, configuring their Tx time so the transmission interval is maintained. The kernel can then offload packet transmission to the hardware (if the network controller supports it), providing more transmission time accuracy. This offloading mechanism enables the application to sleep for longer times, easing on task scheduling. To illustrate the improvements provided by this series, I ran a before-after experiment. The experiment setup consisted in 2 PCs with Intel i210 card connected back-to-back running an up-to-date Archlinux with kernel 4.19.4. I ran 'aplay' for 5 minutes on one PC and captured the AVTPDUs on the other PC. The metric under evaluation is the transmission interval and it is measured by checking the 'time_delta' information from ethernet frames captured at the receiving side. If you're interested in reproducing the experiments, let me know and I can share my helper scripts. The table below shows the experiment outcome for a Class A, stereo, 16-bit sample, 48 kHz stream. The unit is nanoseconds. | Mean | Stdev | Min | Max | Range | -------+--------+---------+---------+---------+---------+ Before | 125000 | 1154 | 75311 | 172144 | 96833 | After | 125000 | 18 | 124960 | 125048 | 88 | Before this patchset, the transmission interval mean is equal to the optimal value (Class A stream -> 125 us interval), and it is kept the same after the patchset. However, the dispersion measurements had improved considerably, meaning the system is consistently transmitting AVTPDUs at the correct interval. Finally, to help the review process, here follows a quick summary of the patches within this series: * PATCH 1: fixes a type in the plugin documentation. * PATCH 2: enables the user to configure the presentation time tolerance. * PATCH 3-5: refactor the code in order to land the offload mechanism code smoothly. * PATCH 6-7: implement the transmission offload mechanism. This series can also be found in my alsa-plugins tree in github [1]. Regards, Andre [1] https://github.com/aguedes/alsa-plugins Andre Guedes (7): doc: Fix typo in AAF doc aaf: Add presentation time tolerance aaf: Refactor AVTPDU transmission routines aaf: Refactor AVTPDU reception routines aaf: Refactor timeout routines aaf: Tx multiple AVTPDUs per media clock tick aaf: AVTPDU transmission periodicity aaf/pcm_aaf.c | 600 +++++++++++++++++++++++++++++++++----------------- configure.ac | 2 +- doc/aaf.txt | 40 +++- 3 files changed, 429 insertions(+), 213 deletions(-) -- 2.19.1

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[alsa-devel] [PATCH v1] ASoC: Intel: kbl_da7219_max98927: fix the audio jack button remapping
by Zhuohao Lee 10 Dec '18

10 Dec '18

>From the da7219 spec, the button A, B, C and D are remapped to 0, 1, 2 and 3 respectively where button A is KEY_PLAYPAUSE, B is KEY_VOLUMEUP, C is KEY_VOLUMEDOWN and D is KEY_VOICECOMMAND Change-Id: I67f929f226883bd289cb7beb0a7b23f40e7121e2 Signed-off-by: Zhuohao Lee <zhuohao(a)chromium.org> Signed-off-by: Max Chang <changmax(a)chromium.org> --- sound/soc/intel/boards/kbl_da7219_max98927.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 3fa1c3ca6d37..d3531f0eb357 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -262,9 +262,9 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) jack = &ctx->kabylake_headset; snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); - snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); - snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); - snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); da7219_aad_jack_det(component, &ctx->kabylake_headset); -- 2.20.0.rc2.403.gdbc3b29805-goog

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[alsa-devel] [PATCH 0/2] ALSA: HDAudio: enable dynamic selection between legacy and Skylake drivers
by Pierre-Louis Bossart 10 Dec '18

10 Dec '18

This patchset depends on the "ASoC: Intel: Skylake: probe and Kconfig simplifications" series. Add the same logic based on PCI class/subclass/prog-if in the HDaudio legacy probe, and add a module parameter to bypass or force the automatic detection should it ever fail. This patchset was tested on a SKL device with the DSP disabled and a WHL device with the DSP enabled. All 6 combinations of module parameters provided the expected behavior. The SOF driver will use exactly the same mechanism, and add a mutual exclusion with the Skylake/SST driver to avoid having 3 drivers for the same PCI ID. Two is complicated enough. Pierre-Louis Bossart (2): ALSA: HD-Audio: SKL+: abort probe if DSP is present and Skylake driver selected ALSA: HD-Audio: SKL+: force HDaudio legacy or SKL+ driver selection include/sound/hdaudio.h | 6 ++++ sound/pci/hda/Kconfig | 46 +++++++++++++++++++++++++++++ sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 53 +++++++++++++++++++++++++++++----- sound/soc/intel/Kconfig | 6 ++++ sound/soc/intel/skylake/skl.c | 44 +++++++++++++++++++--------- 6 files changed, 135 insertions(+), 22 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH] ASoC: Intel: kbl_da7219_max98927: fix the audio jack button remapping
by Zhuohao Lee 10 Dec '18

10 Dec '18

>From the da7219 spec, the button A, B, C and D are remapped to 0, 1, 2 and 3 respectively where button A is KEY_PLAYPAUSE, B is KEY_VOLUMEUP, C is KEY_VOLUMEDOWN and D is KEY_VOICECOMMAND. Signed-off-by: Zhuohao Lee <zhuohao(a)chromium.org> Signed-off-by: Max Chang <changmax(a)chromium.org> --- sound/soc/intel/boards/kbl_da7219_max98927.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 3fa1c3ca6d37..d3531f0eb357 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -262,9 +262,9 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) jack = &ctx->kabylake_headset; snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); - snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); - snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); - snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); da7219_aad_jack_det(component, &ctx->kabylake_headset); -- 2.20.0.rc2.403.gdbc3b29805-goog

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[alsa-devel] [PATCH - snd_pcm_extplug_set_param_link() 1/1] pcm: extplug: Keep format and channels the same if requested
by twischer＠de.adit-jv.com 10 Dec '18

10 Dec '18

From: Timo Wischer <twischer(a)de.adit-jv.com> Without this patch it is not possible to link the channel and format parameter if snd_pcm_extplug_set_param_*() or snd_pcm_extplug_set_slave_param_*() is called. Therefore the client and slave parameter can differ. So the extplug has to implement conversion. To avoid this the new snd_pcm_extplug_set_param_link() function can be called. As a reproduction sceanrio the following extplug source code can be used: static snd_pcm_sframes_t my_transfer(snd_pcm_extplug_t *e, const snd_pcm_channel_area_t *da, snd_pcm_uframes_t dof, const snd_pcm_channel_area_t *sa, snd_pcm_uframes_t sof, snd_pcm_uframes_t s) { return s; } static const snd_pcm_extplug_callback_t my_own_callback = { .transfer = my_transfer }; SND_PCM_PLUGIN_DEFINE_FUNC(my_plug) { snd_config_iterator_t i, next; snd_config_t *slave = NULL; snd_pcm_extplug_t *myplug; snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); const char *id; if (snd_config_get_id(n, &id) < 0) continue; if (strcmp(id, "comment") == 0 || strcmp(id, "type") == 0) continue; if (strcmp(id, "slave") == 0) { slave = n; continue; } return -EINVAL; } myplug = calloc(1, sizeof(*myplug)); myplug->version = SND_PCM_EXTPLUG_VERSION; myplug->callback = &my_own_callback; snd_pcm_extplug_create(myplug, name, root, slave, stream, mode); snd_pcm_extplug_set_param_minmax(myplug, SND_PCM_EXTPLUG_HW_CHANNELS, 1, 16); // snd_pcm_extplug_set_param_link(myplug, SND_PCM_EXTPLUG_HW_CHANNELS, 1); *pcmp = myplug->pcm; return 0; } SND_PCM_PLUGIN_SYMBOL(my_plug); To use this plugin the following ALSA configuration is required: pcm.myplug { type my_plug slave.pcm hw:Dummy } With this configuration without this patch snd_pcm_hw_params_get_channels_max() will always return 16 channels independent of the supported channels of the dummy device. Due to that for example the start up of JACK would fail: $ modprobe snd_dummy $ jackd -d alsa -P myplug ALSA: cannot set channel count to 16 for playback ALSA: cannot configure playback channel Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/include/pcm_extplug.h b/include/pcm_extplug.h index ced934f..e5c02d4 100644 --- a/include/pcm_extplug.h +++ b/include/pcm_extplug.h @@ -184,6 +184,8 @@ int snd_pcm_extplug_set_param_list(snd_pcm_extplug_t *extplug, int type, unsigne int snd_pcm_extplug_set_param_minmax(snd_pcm_extplug_t *extplug, int type, unsigned int min, unsigned int max); int snd_pcm_extplug_set_slave_param_list(snd_pcm_extplug_t *extplug, int type, unsigned int num_list, const unsigned int *list); int snd_pcm_extplug_set_slave_param_minmax(snd_pcm_extplug_t *extplug, int type, unsigned int min, unsigned int max); +int snd_pcm_extplug_set_param_link(snd_pcm_extplug_t *extplug, int type, + int keep_link); /** * set the parameter constraint with a single value diff --git a/src/pcm/pcm_ext_parm.h b/src/pcm/pcm_ext_parm.h index d25f2ab..7b99bef 100644 --- a/src/pcm/pcm_ext_parm.h +++ b/src/pcm/pcm_ext_parm.h @@ -5,6 +5,7 @@ struct snd_ext_parm { unsigned int *list; unsigned int active: 1; unsigned int integer: 1; + unsigned int keep_link: 1; }; static inline snd_mask_t *hw_param_mask(snd_pcm_hw_params_t *params, diff --git a/src/pcm/pcm_extplug.c b/src/pcm/pcm_extplug.c index 1f887c5..94002dc 100644 --- a/src/pcm/pcm_extplug.c +++ b/src/pcm/pcm_extplug.c @@ -249,7 +249,7 @@ static unsigned int get_links(struct snd_ext_parm *params) SND_PCM_HW_PARBIT_TICK_TIME); for (i = 0; i < SND_PCM_EXTPLUG_HW_PARAMS; i++) { - if (params[i].active) + if (params[i].active && !params[i].keep_link) links &= ~excl_parbits[i]; } return links; @@ -642,6 +642,17 @@ as former functions. To clear the parameter constraints, call #snd_pcm_extplug_params_reset() function. +When using snd_pcm_extplug_set_param_*() or snd_pcm_extplug_set_slave_param_*() +for any parameter. This parameter is no longer linked between the client and +slave PCM. Therefore it could differ and the extplug has to support conversion +between all valid parameter configurations. To keep the client and slave +parameter linked #snd_pcm_extplug_set_param_link() can be used for the +corresponding parameter. For example if the extplug does not support channel nor +format conversion the supported client parameters can be limited with +snd_pcm_extplug_set_param_*() and afterwards +#snd_pcm_extplug_set_param_link(ext, SND_PCM_EXTPLUG_HW_FORMAT, 1) and +#snd_pcm_extplug_set_param_link(ext, SND_PCM_EXTPLUG_HW_CHANNELS, 1) should be +called to keep the client and slave parameters the same. */ /** @@ -849,3 +860,26 @@ int snd_pcm_extplug_set_param_minmax(snd_pcm_extplug_t *extplug, int type, unsig return snd_ext_parm_set_minmax(&ext->params[type], min, max); } +/** + * @brief Keep the client and slave format/channels the same if requested. This + * is for example useful if this extplug does not support any channel + * conversion. + * @param extplug the extplug handle + * @param type parameter type + * @param keep_link if 1 the parameter identified by type will be kept the same + * for the client and slave PCM of this extplug + * @return 0 if successful, or a negative error code + */ +int snd_pcm_extplug_set_param_link(snd_pcm_extplug_t *extplug, int type, + int keep_link) +{ + extplug_priv_t *ext = extplug->pcm->private_data; + + if (type < 0 || type >= SND_PCM_EXTPLUG_HW_PARAMS) { + SNDERR("EXTPLUG: invalid parameter type %d", type); + return -EINVAL; + } + ext->params[type].keep_link = keep_link ? 1 : 0; + ext->sparams[type].keep_link = keep_link ? 1 : 0; + return 0; +} -- 2.7.4

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Re: [alsa-devel] [PATCH 1/3] ALSA: hda/realtek: ALC294 mic and headset-mode fixups for ASUS X542UN
by Jian-Hong Pan 10 Dec '18

10 Dec '18

> Kailang <kailang(a)realtek.com> 於 2018年12月5日週三下午4:36寫道： > > > > Hi Jian-Hong, > > > > Could you test to change the model to ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC? > > > > .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE ==> change to ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC We do not have ASUS X542UN in hand right now, but we have ASUS X542UQ which goes with the same quirk as X542UN. I tested with ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, and it also works. So, I am going to send new patch set using ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC for ALC294. Thanks, Jian-Hong Pan

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[alsa-devel] Patch for ALC294 Realtek soundcard in ASUS Zenbook UX433F?
by Anthony Ettinger 10 Dec '18

10 Dec '18

I saw something come across the mailing list earlier today, that supposedly fixes the no sound issues with this asus zenbook. Does anyone know when it will make it to the kernel master? I tried 4.20-rc6 but sound still doesn't work. -- Anthony Ettingers http://linkedin.com/in/chovy

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[alsa-devel] [RFC PATCH v8 0/4] Media Device Allocator API
by shuah＠kernel.org 09 Dec '18

09 Dec '18

From: Shuah Khan <shuah(a)kernel.org> This patch series has been on the ice for the last couple of years. Mauro asked me to restart the discussion on this and see if we can make progress. I rebased the series to Linux 4.19. It has been surprisingly well preserved with one small merge conflict resolved in 0004. Media Device Allocator API to allows multiple drivers share a media device. Using this API, drivers can allocate a media device with the shared struct device as the key. Once the media device is allocated by a driver, other drivers can get a reference to it. The media device is released when all the references are released. - No changes to 0001,0002 code since the v7 referenced below. - 0003 is a new patch to enable ALSA defines that have been disabled for kernel between 4.9 and 4.19. - Minor merge conflict resolution in 0004. - Added SPDX to new files. References: https://www.mail-archive.com/linux-media@vger.kernel.org/msg105854.html Please review. I am sending this as RFC even though it has been tested several times prior to this non-event rebased v8. I ran sanity tests. Shuah Khan (4): media: Media Device Allocator API media: change au0828 to use Media Device Allocator API media: media.h: Enable ALSA MEDIA_INTF_T* interface types sound/usb: Use Media Controller API to share media resources Documentation/media/kapi/mc-core.rst | 37 +++ drivers/media/Makefile | 3 +- drivers/media/media-dev-allocator.c | 132 ++++++++++ drivers/media/usb/au0828/au0828-core.c | 12 +- drivers/media/usb/au0828/au0828.h | 1 + include/media/media-dev-allocator.h | 53 ++++ include/uapi/linux/media.h | 25 +- sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 ++ sound/usb/card.h | 3 + sound/usb/media.c | 320 +++++++++++++++++++++++++ sound/usb/media.h | 73 ++++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 29 ++- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + 18 files changed, 692 insertions(+), 28 deletions(-) create mode 100644 drivers/media/media-dev-allocator.c create mode 100644 include/media/media-dev-allocator.h create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.17.0

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[alsa-devel] [PATCH] ALSA: fireface: fix reference to wrong register for clock configuration
by Takashi Sakamoto 09 Dec '18

09 Dec '18

In an initial commit, 'SYNC_STATUS' register is referred to get clock configuration, however this is wrong, according to my local note at hand for reverse-engineering about packet dump. It should be 'CLOCK_CONFIG' register. Actually, ff400_dump_clock_config() is correctly programmed. This commit fixes the bug. Cc: <stable(a)vger.kernel.org> # v4.12+ Fixes: 76fdb3a9e13a ('ALSA: fireface: add support for Fireface 400') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/fireface/ff-protocol-ff400.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/fireface/ff-protocol-ff400.c b/sound/firewire/fireface/ff-protocol-ff400.c index 64c3cb0fb926..654a50319198 100644 --- a/sound/firewire/fireface/ff-protocol-ff400.c +++ b/sound/firewire/fireface/ff-protocol-ff400.c @@ -30,7 +30,7 @@ static int ff400_get_clock(struct snd_ff *ff, unsigned int *rate, int err; err = snd_fw_transaction(ff->unit, TCODE_READ_QUADLET_REQUEST, - FF400_SYNC_STATUS, &reg, sizeof(reg), 0); + FF400_CLOCK_CONFIG, &reg, sizeof(reg), 0); if (err < 0) return err; data = le32_to_cpu(reg); -- 2.19.1

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[alsa-devel] [PATCH] ALSA: hda/realtek - Fix the mute LED regresion on Lenovo X1 Carbon
by Hui Wang 09 Dec '18

09 Dec '18

Users reported a mute LED regression on Lenovo X1 Carbon, the root cause is we applied the fixup of ALC285_FIXUP_LENOVO_HEADPHONE_NOISE to this machine, then the machine can't apply the fixup of ALC269_FIXUP_THINKPAD_ACPI anymore. To fix it, we chain two fixup together. Fixes: c4cfcf6f4297 ("ALSA: hda/realtek - fix the pop noise on headphone for lenovo laptops") Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 56882d9b1e43..5b13d9e8d6c7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6450,6 +6450,8 @@ static const struct hda_fixup alc269_fixups[] = { [ALC285_FIXUP_LENOVO_HEADPHONE_NOISE] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_invalidate_dacs, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI }, [ALC295_FIXUP_HP_AUTO_MUTE] = { .type = HDA_FIXUP_FUNC, -- 2.17.1

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