January 2020 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] arecord: buffer-size dependency on period-size and other params
by Rajwa, Marcin 15 Jan '20

15 Jan '20

Hello, I have spotted strange behavior of alsa-lib when I try to set buffer-size in arecord. It looks like the size of the buffer is very dependent on other parameters like period-size and rate for example. At first it sounds like normal behaviour, after all buffer size smaller than period size doesn’t make much sense. However what if we go in the reverse direction and set buffer-size much larger than period size? Then I see no reason to forbid such request (at least on the user space level). So to be specific I issue such command: arecod -Dplughw:0,8 -r 16000 -c 2 -f S24_LE --buffer_size 1280000 tmp.wav -vvv. The arecord response for this is buffer size of 31992 frames, 255936 bytes. Now let me also tell the buffer size of 1280000 bytes is the maximum buffer size our driver allows, likewise max period is 268800 bytes. Now if inside our driver I double max period size limit I get buffer size accordingly bigger. Similarly, if I change the format in the command line request from S24_LE to i.e S16_LE then again I will get buffer size closer to my request (bigger). So below are my questions: 1. Why alsa-lib behaves in a way I described above? Why I can not control buffer size independently from other parameters (providing it has physical sense). 2. It looks like alsa-lib first calculates initial buffer-size base on other input params like mentioned period size or sampling rate and then ask driver to refine it according to its capabilities. Is that the case? 3. I also tried to dump hw capabilities by adding –dump-hw-param flag to my command line request and here yet another surprise – in a row dedicated for buffer size it says “BUFFER_SIZE [12 4294967294]”. Where is this “4294967294” comes from, and on what basics is it calculated? I would expect at the bare minimum it should ask driver but in the snd_pcm_hw_param_dump() function I don’t see any communication with the driver. Kind Regards, Marcin Rajwa

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[alsa-devel] Applied "regulator fix for "regulator: core: Add regulator_is_equal() helper"" to the asoc tree
by Mark Brown 15 Jan '20

15 Jan '20

The patch regulator fix for "regulator: core: Add regulator_is_equal() helper" has been applied to the asoc tree at https://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 0468e667a5bead9c1b7ded92861b5a98d8d78745 Mon Sep 17 00:00:00 2001 From: Stephen Rothwell <sfr(a)canb.auug.org.au> Date: Wed, 15 Jan 2020 12:02:58 +1100 Subject: [PATCH] regulator fix for "regulator: core: Add regulator_is_equal() helper" Signed-off-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Link: https://lore.kernel.org/r/20200115120258.0e535fcb@canb.auug.org.au Acked-by: Marek Vasut <marex(a)denx.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/linux/regulator/consumer.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/linux/regulator/consumer.h b/include/linux/regulator/consumer.h index 2c89d886595c..6a92fd3105a3 100644 --- a/include/linux/regulator/consumer.h +++ b/include/linux/regulator/consumer.h @@ -596,7 +596,7 @@ regulator_bulk_set_supply_names(struct regulator_bulk_data *consumers, } static inline bool -regulator_is_equal(struct regulator *reg1, struct regulator *reg2); +regulator_is_equal(struct regulator *reg1, struct regulator *reg2) { return false; } -- 2.20.1

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[alsa-devel] Bug report for changes in ucm2 with chtrt5645 card on Lenovo Miix 320
by Tim Schumacher 15 Jan '20

15 Jan '20

Hi alsa devs, I'm on Arch Linux on a rather uncommon laptop (Lenovo Miix 320) and my sound stopped working on the upgrade from alsa-lib 1.1.9 to 1.2.1. wabbits on IRC helped me track down the problem to a change in the /usr/share/alsa/ucm (now /usr/share/alsa/ucm2) files. It seems like /usr/share/alsa/ucm2/chtrt5645/HiFi-dmic2.conf and a symlink at /usr/share/alsa/ucm2/chtrt5645/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216.conf were specifically added for this laptop but in the contrary they break the sound for me. After the upgrade to alsa-lib 1.2.1 the sound is not working. The commands play no sound: pasuspender -- speaker-test --nloops=1 --channels=2 --test=wav --device=hw:0,0 pasuspender -- speaker-test --nloops=1 --channels=2 --test=wav --device=hw:0,1 And this is the alsa-info.sh output http://alsa-project.org/db/?f=f883910a5c5101b4b1ea4202d1fe84ccd139f796 After deleting the /usr/share/alsa/ucm2/chtrt5645/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216.conf symlink both commands from above play sound and this is the alsa-info.sh output http://alsa-project.org/db/?f=e759eb9118a191b6c3b8c021fed58abc9cf95076 Thanks. - Tim

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[alsa-devel] [PATCH] ALSA: hda/analog - Minor optimization for SPDIF mux connections
by Takashi Iwai 15 Jan '20

15 Jan '20

AD HD-audio codec driver has a few code lines invoking snd_get_num_conns() and using its return value as the array index without checking. This is basically safe in all those places; at the second and later calls snd_get_num_conns() returns the value cached from the first invocation, hence the value is always consistent. However, it looks a bit confusing as if a lack of the proper check. This patch introduces a new field num_smux_conns in ad198x_spec for simplifying the code. Now we store and refer to the value more locally without invoking the extra function at each time. Reported-by: Colin King <colin.king(a)canonical.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_analog.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 88c46b051d14..2132b2acec4d 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -28,6 +28,7 @@ struct ad198x_spec { hda_nid_t eapd_nid; unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */ + int num_smux_conns; }; @@ -453,8 +454,7 @@ static int ad1983_auto_smux_enum_info(struct snd_kcontrol *kcontrol, struct ad198x_spec *spec = codec->spec; static const char * const texts2[] = { "PCM", "ADC" }; static const char * const texts3[] = { "PCM", "ADC1", "ADC2" }; - hda_nid_t dig_out = spec->gen.multiout.dig_out_nid; - int num_conns = snd_hda_get_num_conns(codec, dig_out); + int num_conns = spec->num_smux_conns; if (num_conns == 2) return snd_hda_enum_helper_info(kcontrol, uinfo, 2, texts2); @@ -481,7 +481,7 @@ static int ad1983_auto_smux_enum_put(struct snd_kcontrol *kcontrol, struct ad198x_spec *spec = codec->spec; unsigned int val = ucontrol->value.enumerated.item[0]; hda_nid_t dig_out = spec->gen.multiout.dig_out_nid; - int num_conns = snd_hda_get_num_conns(codec, dig_out); + int num_conns = spec->num_smux_conns; if (val >= num_conns) return -EINVAL; @@ -512,6 +512,7 @@ static int ad1983_add_spdif_mux_ctl(struct hda_codec *codec) num_conns = snd_hda_get_num_conns(codec, dig_out); if (num_conns != 2 && num_conns != 3) return 0; + spec->num_smux_conns = num_conns; if (!snd_hda_gen_add_kctl(&spec->gen, NULL, &ad1983_auto_smux_mixer)) return -ENOMEM; return 0; @@ -730,10 +731,12 @@ static int ad1988_auto_smux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ad198x_spec *spec = codec->spec; static const char * const texts[] = { "PCM", "ADC1", "ADC2", "ADC3", }; - int num_conns = snd_hda_get_num_conns(codec, 0x0b) + 1; + int num_conns = spec->num_smux_conns; + if (num_conns > 4) num_conns = 4; return snd_hda_enum_helper_info(kcontrol, uinfo, num_conns, texts); @@ -756,7 +759,7 @@ static int ad1988_auto_smux_enum_put(struct snd_kcontrol *kcontrol, struct ad198x_spec *spec = codec->spec; unsigned int val = ucontrol->value.enumerated.item[0]; struct nid_path *path; - int num_conns = snd_hda_get_num_conns(codec, 0x0b) + 1; + int num_conns = spec->num_smux_conns; if (val >= num_conns) return -EINVAL; @@ -847,6 +850,7 @@ static int ad1988_add_spdif_mux_ctl(struct hda_codec *codec) num_conns = snd_hda_get_num_conns(codec, 0x0b) + 1; if (num_conns != 3 && num_conns != 4) return 0; + spec->num_smux_conns = num_conns; for (i = 0; i < num_conns; i++) { struct nid_path *path = snd_array_new(&spec->gen.paths); -- 2.16.4

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[alsa-devel] ABI breakage in alsa-lib 1.2
by Jordi Mallach 15 Jan '20

15 Jan '20

[Please Cc me on replies, I'm not currently subscribed to alsa-devel] Hello, I'm one of the ALSA Debian maintainers, and some weeks ago I looked into updating our ALSA packages to 1.2.1. I quickly noticed there's been some fundamental change to libasound2, one that appears to break ABI and should have warranted an soname bump at least. With the split of the snd_tplg API to the new libatopology2, libasound2 has been left missing that part of the API that was part of the asound.h header, etc. This could have been a somewhat lesser issue if libasound2 linked to libatopology, but unfortunately this goes in the other direction, so the ABI and API breakage is not avoided. I cannot update alsa-lib in this state. I could hack around and patch it to reinstate the missing api in libasound, but this sounds like pain and suffering for everyone involved. How should we move forward with this? I am sorry I only come up with this 2 months after the release when I had identified the problem many weeks ago, I've been extremely busy. Thanks, Jordi -- Jordi Mallach <jordi(a)debian.org> Debian Project

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[alsa-devel] [PATCH RFC 0/3] ASoC: Add Multi CPU DAI support
by Bard liao 15 Jan '20

15 Jan '20

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> As discussed in [1], ASoC core supports multi codec DAIs on a DAI link. However it does not do so for CPU DAIs. So, add support for multi CPU DAIs on a DAI Link by adding multi CPU DAI in Card instantiation, suspend and resume functions, PCM ops, stream handling functions and DAPM. [1]: https://www.spinics.net/lists/alsa-devel/msg71369.html Shreyas NC (3): ASoC: Add initial support for multiple CPU DAIs ASoC: Add multiple CPU DAI support for PCM ops ASoC: Add multiple CPU DAI support in DAPM include/sound/soc.h | 15 ++ sound/soc/soc-core.c | 183 ++++++++++-------- sound/soc/soc-dapm.c | 131 +++++++------ sound/soc/soc-pcm.c | 432 ++++++++++++++++++++++++++++--------------- 4 files changed, 483 insertions(+), 278 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH] ALSA: hda - fix out of bounds read on spec->smux_paths
by Colin King 14 Jan '20

14 Jan '20

From: Colin Ian King <colin.king(a)canonical.com> It is possible for the call to snd_hda_get_num_conns to fail and return a negative error code that gets assigned to num_conns. In that specific case, the check of very large values of val against num_conns will not fail the -EINVAL check and later on an out of bounds array read on spec->smux_paths will occur. Fix this by sanity checking for an error return from the call to snd_hda_get_num_conns. Addresses-Coverity: ("Out-of-bounds read") Fixes: 272f3ea31776 ("ALSA: hda - Add SPDIF mux control to AD codec auto-parser") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/hda/patch_analog.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 88c46b051d14..399561369495 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -756,9 +756,11 @@ static int ad1988_auto_smux_enum_put(struct snd_kcontrol *kcontrol, struct ad198x_spec *spec = codec->spec; unsigned int val = ucontrol->value.enumerated.item[0]; struct nid_path *path; - int num_conns = snd_hda_get_num_conns(codec, 0x0b) + 1; + int num_conns = snd_hda_get_num_conns(codec, 0x0b); - if (val >= num_conns) + if (num_conns < 0) + return num_conns; + if (val >= num_conns + 1) return -EINVAL; if (spec->cur_smux == val) return 0; -- 2.24.0

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[alsa-devel] [PATCH 0/5] soundwire: intel: add DAI callbacks
by Pierre-Louis Bossart 14 Jan '20

14 Jan '20

The existing mainline code is missing most of the DAI callbacks needed for a functional implementation, and the existing ones need to be modified to provide the relevant information to SOF drivers. As suggested by Vinod, these patches are shared first - with the risk that they are separated from the actual DAI enablement, so reviewers might wonder why they are needed in the first place. For reference, the complete set of 100+ patches required for SoundWire on Intel platforms is available here: https://github.com/thesofproject/linux/pull/1692 Pierre-Louis Bossart (2): soundwire: intel: rename res field as link_res soundwire: intel: free all resources on hw_free() Rander Wang (3): soundwire: intel: add prepare support in sdw dai driver soundwire: intel: add trigger support in sdw dai driver soundwire: intel: add sdw_stream_setup helper for .startup callback drivers/soundwire/intel.c | 198 ++++++++++++++++++++++++++++++++++---- 1 file changed, 178 insertions(+), 20 deletions(-) base-commit: b637124800a157c4df3699d1137d8533394f7678 -- 2.20.1

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[alsa-devel] how to use DPCM to alter audio path between headphones and speaker
by Tim Harvey 14 Jan '20

14 Jan '20

Greetings, I'm working with a board that has a TI TLV320AIC3105 codec connected to an IMX6 SSI with a headphone jack and a TS3A227E headphone detect chip (HPROUT/HPLOUT), mono speaker amp output (RIGHT_LOP/RIGHT_LOM), and mic in (MIC1L). It seems that Dynamic PCM allows for the automatic routing of audio to change between headphone out and speaker out depending on jack insertion/removal event but I'm not clear how to configure that. Can this be done via simple-audio-card bindings or does it require a modified machine driver? Any pointers or references are greatly appreciated! Best Regards, Tim

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[alsa-devel] [PATCH 0/4] ASoC: msm8916-wcd-analog: MIC BIAS fixes/additions
by Stephan Gerhold 14 Jan '20

14 Jan '20

This patch series fixes some bugs in the MIC BIAS implementation in msm8916-wcd-analog. Finally, it adds support for MIC BIAS Internal3, which is needed for msm8916-longcheer-l8150. Stephan Gerhold (4): ASoC: msm8916-wcd-analog: Fix selected events for MIC BIAS External1 ASoC: msm8916-wcd-analog: Fix MIC BIAS Internal1 ASoC: msm8916-wcd-analog: Simplify MIC BIAS Internal ASoC: msm8916-wcd-analog: Add MIC BIAS Internal3 sound/soc/codecs/msm8916-wcd-analog.c | 59 ++++++++++----------------- 1 file changed, 21 insertions(+), 38 deletions(-) -- 2.24.1

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