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

[alsa-devel] utils/alsa.m4: fix atopology detection without alsatest
by GitHub pull_request - opened 04 Dec '19

04 Dec '19

alsa-project/alsa-lib pull request #16 was opened from ffontaine: Since commit 75d393a563efb578c79364a277087c6326267f52, alsa-utils fails to build with --disable-alsatest because atopology is not detected Fixes: - http://autobuild.buildroot.org/results/d0fb760669b02b813115af04adcf24530d35… Signed-off-by: Fabrice Fontaine <fontaine.fabrice(a)gmail.com> Request URL : https://github.com/alsa-project/alsa-lib/pull/16 Patch URL : https://github.com/alsa-project/alsa-lib/pull/16.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[alsa-devel] [PATCH] ASoC: snd-sof-pci - add model parameter and pass it to HDA codec driver
by Jaroslav Kysela 03 Dec '19

03 Dec '19

It may be useful to pass the specific model to the generic HDA codec routines like the legacy HDA driver (snd-hda-intel) allows. The model name "sofbus" is tricky anyway. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> --- include/sound/sof.h | 1 + sound/soc/sof/intel/hda.c | 2 +- sound/soc/sof/sof-pci-dev.c | 5 +++++ 3 files changed, 7 insertions(+), 1 deletion(-) diff --git a/include/sound/sof.h b/include/sound/sof.h index 479101736ee0..a62686baa95d 100644 --- a/include/sound/sof.h +++ b/include/sound/sof.h @@ -25,6 +25,7 @@ struct snd_sof_pdata { const char *drv_name; const char *name; const char *platform; + const char *modelname; struct device *dev; diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 91bd88fddac7..ccb640bacc99 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -271,7 +271,7 @@ static int hda_init(struct snd_sof_dev *sdev) mutex_init(&hbus->prepare_mutex); hbus->pci = pci; hbus->mixer_assigned = -1; - hbus->modelname = "sofbus"; + hbus->modelname = sdev->pdata->modelname; /* initialise hdac bus */ bus->addr = pci_resource_start(pci, 0); diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index bbeffd932de7..c01ad85aad2a 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -34,6 +34,10 @@ static int sof_pci_debug; module_param_named(sof_pci_debug, sof_pci_debug, int, 0444); MODULE_PARM_DESC(sof_pci_debug, "SOF PCI debug options (0x0 all off)"); +static char *model; +module_param(model, charp, 0444); +MODULE_PARM_DESC(model, "Use the given HDA board model."); + #define SOF_PCI_DISABLE_PM_RUNTIME BIT(0) #if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) @@ -326,6 +330,7 @@ static int sof_pci_probe(struct pci_dev *pci, sof_pdata->desc = (struct sof_dev_desc *)pci_id->driver_data; sof_pdata->dev = dev; sof_pdata->platform = dev_name(dev); + sof_pdata->modelname = model; /* alternate fw and tplg filenames ? */ if (fw_path) -- 2.20.1

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[alsa-devel] Integrating Texes Instruments PCM1860 ADC
by Umar Dockrat 03 Dec '19

03 Dec '19

Good Day, I have been struggling with the integration of a PCM1860 ADC for audio recording. I have created a "sound card" board that contains both a PCM5102A (DAC) and PCM1860 (ADC). The main processor is an Omega Onion 2 which is based on MediaTek MT7688 SoC device. I am struggling with creating the device tree as the Kernel currently has reference to the PCM186X where X is 2,3,4 or 5 but not the more simple PCM1860. The difference between the PCM1860 and the PCM186X is that the PCM1860 does not have I2C control and is HW configured using pull up/down. https://www.kernel.org/doc/Documentation/devicetree/bindings/sound/pcm186x.… I have tried multiple device tree setup's and have only got the DAC working when configured in isolation, as soon as I add any reference to the ADC then both DAC and ADC are not recognised. This DTSI allows the DAC to function https://pastebin.com/z9H67DHd There are my attempts at including the DAC and ADC: https://pastebin.com/RLsKNsiR https://pastebin.com/rmJVcCz1 Please can someone assist with the correct way to integrate this ADC device? Any suggestions on how to properly debug this further is also really appreciated. Thanks, UFD

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[alsa-devel] a52 plugin + ffmpeg with libswresample
by Giovanni Harting 03 Dec '19

03 Dec '19

Hello, I'm working on restoring functionality to the a52 plugin. ALSA build with ffmpeg without (deprecated) libavresample does not build this plugin, since libavresample is missing. Archlinux for example does not build with libavresample anymore [1], therefore a52 is missing in alsa-plugins builds since that change happened. I ported rate-lav to libswresample, which seems to work well after a month of daily use. The oddity is that a config file (asoundrc, [2]), that worked fine with alsa-lib 1.1.6, does not work with 1.1.7+. I could not find a change that causes this. I traced it down to _snd_pcm_rate_open() in pcm/pcm_rate.c, where keywords from within pcm {} are passed into, where it fails with "Unknown field card" (for example). I worked around it by filtering out all keywords [3], but that feels a little unclean. Is there a proper way to fix this? I'm not familiar with ALSA's config parsing. I'll submit patches for both once I figured out how to fix this. [1] https://bugs.archlinux.org/task/60586 [2] pcm.a52 { @args [CARD] @args.CARD { type string } type rate slave { pcm { type a52 bitrate 640 channels 6 card $CARD } rate 48000 format S16_LE } } [3] if (strcmp(id, "bitrate") == 0 || strcmp(id, "rate") == 0 || strcmp(id, "card") == 0 || strcmp(id, "channels") == 0 || strcmp(id, "format") == 0) { continue; }

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[alsa-devel] [PATCH alsa-lib] ucm: Do not fail to parse configs on cards with an empty CardComponents lists
by Hans de Goede 03 Dec '19

03 Dec '19

Since the UCM profiles for all Bay- and Cherry-Trail SST cards have been moved over to UCM2, parsing them fails with: ALSA lib ucm_subs.c:220:(uc_mgr_get_substituted_value) variable '${CardComponents}' is not defined in this context! This completely breaks audio support on all Bay- and Cherry-Trail devices. This is caused by these non-SOF ASoC using cards having an empty CardComponents list. Which in itself is fine, but is rejected by the ucm_subs.c code. This commit changes the ucm_subs code to accept an empty string as a valid value for CardComponents restoring audio functionality on these boards. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- src/ucm/ucm_subs.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) diff --git a/src/ucm/ucm_subs.c b/src/ucm/ucm_subs.c index 00afa9e3..90e395f0 100644 --- a/src/ucm/ucm_subs.c +++ b/src/ucm/ucm_subs.c @@ -25,6 +25,7 @@ */ #include "ucm_local.h" +#include <stdbool.h> #include <sys/stat.h> #include <limits.h> @@ -145,10 +146,11 @@ static char *rval_sysfs(snd_use_case_mgr_t *uc_mgr ATTRIBUTE_UNUSED, const char return strdup(path); } -#define MATCH_VARIABLE(name, id, fcn) \ +#define MATCH_VARIABLE(name, id, fcn, empty_ok) \ if (strncmp((name), (id), sizeof(id) - 1) == 0) { \ rval = fcn(uc_mgr); \ idsize = sizeof(id) - 1; \ + allow_empty = (empty_ok); \ goto __rval; \ } @@ -189,12 +191,14 @@ int uc_mgr_get_substituted_value(snd_use_case_mgr_t *uc_mgr, while (*value) { if (*value == '$' && *(value+1) == '{') { - MATCH_VARIABLE(value, "${ConfName}", rval_conf_name); - MATCH_VARIABLE(value, "${CardId}", rval_card_id); - MATCH_VARIABLE(value, "${CardDriver}", rval_card_driver); - MATCH_VARIABLE(value, "${CardName}", rval_card_name); - MATCH_VARIABLE(value, "${CardLongName}", rval_card_longname); - MATCH_VARIABLE(value, "${CardComponents}", rval_card_components); + bool allow_empty = false; + + MATCH_VARIABLE(value, "${ConfName}", rval_conf_name, false); + MATCH_VARIABLE(value, "${CardId}", rval_card_id, false); + MATCH_VARIABLE(value, "${CardDriver}", rval_card_driver, false); + MATCH_VARIABLE(value, "${CardName}", rval_card_name, false); + MATCH_VARIABLE(value, "${CardLongName}", rval_card_longname, false); + MATCH_VARIABLE(value, "${CardComponents}", rval_card_components, true); MATCH_VARIABLE2(value, "${env:", rval_env); MATCH_VARIABLE2(value, "${sys:", rval_sysfs); err = -EINVAL; @@ -208,7 +212,7 @@ int uc_mgr_get_substituted_value(snd_use_case_mgr_t *uc_mgr, } goto __error; __rval: - if (rval == NULL || rval[0] == '\0') { + if (rval == NULL || (!allow_empty && rval[0] == '\0')) { free(rval); strncpy(r, value, idsize); r[idsize] = '\0'; -- 2.23.0

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[alsa-devel] Build failure without alsa-topology
by GitHub issues - opened 03 Dec '19

03 Dec '19

alsa-project/alsa-utils issue #33 was opened from ffontaine: Commit c8fdd38c74de2e8b7b2b5a4576787d5e9b4ae807 breaks build of alsa-utils on buildroot because $(ALSA_TOPOLOGY_LIBS) is set to an empty value by https://github.com/alsa-project/alsa-lib/commits/master/utils/alsa.m4 if --enable-alsa-topology is not passed: ``` /home/br-user/autobuild/run/instance-0/output/host/bin/arm-linux-gnueabihf-gcc -D_LARGEFILE_SOURCE -D_LARGEFILE64_SOURCE -D_FILE_OFFSET_BITS=64 -O2 -o alsatplg topology.o -lasound -lm -ldl -lpthread topology.o: In function `main': topology.c:(.text.startup+0x184): undefined reference to `snd_tplg_new' topology.c:(.text.startup+0x194): undefined reference to `snd_tplg_verbose' topology.c:(.text.startup+0x1a4): undefined reference to `snd_tplg_build_file' topology.c:(.text.startup+0x1b4): undefined reference to `snd_tplg_free' topology.c:(.text.startup+0x218): undefined reference to `snd_tplg_free' ``` More information can be found here: http://autobuild.buildroot.org/results/d0f/d0fb760669b02b813115af04adcf2453… Issue URL : https://github.com/alsa-project/alsa-utils/issues/33 Repository URL: https://github.com/alsa-project/alsa-utils

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[alsa-devel] [PATCH 0/3] Add NAU8812 support
by Marco Felsch 03 Dec '19

03 Dec '19

Hi, this small series adds the documentation and support for the nau8812 devices. The exisiting nau8810 driver can handle both devices so we only need to add the dt-bindings and inform the user by adapting the Kconfig entry. Futhermore the series adds the support for the aux-port control. @John Hsu Please can you verify that register accesses to the aux-port control bits don't affect the nau8810 devices? I don't have such a device but the documentations says that those register fields are just '0'ed. So I assume the nau8810 devices ignore such writes. Marco Felsch (2): dt-bindings: ASoC: add nau8812 documentation ASoC: nau8810: add support for nau8812 Thomas Fehrenbacher (1): ASoC: nau8810: add aux input control, available on NAU8812 Documentation/devicetree/bindings/sound/nau8810.txt | 2 +- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/nau8810.c | 9 +++++++++ sound/soc/codecs/nau8810.h | 2 ++ 4 files changed, 13 insertions(+), 2 deletions(-) -- 2.20.1

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[alsa-devel] src/ucm/main.c: fix build without mixer
by GitHub pull_request - opened 03 Dec '19

03 Dec '19

alsa-project/alsa-lib pull request #15 was opened from ffontaine: Commit 4ce38a5ff466d18039b2606938f866ea3a6c9f3c breaks the build without mixer on: CCLD libasound.la /home/buildroot/autobuild/instance-1/output-1/host/lib/gcc/xtensa-buildroot-linux-uclibc/8.3.0/../../../../xtensa-buildroot-linux-uclibc/bin/ld: ucm/.libs/libucm.a(main.o): in function `snd_use_case_set': main.c:(.text+0x185c): undefined reference to `snd_mixer_selem_id_parse' Fixes: - http://autobuild.buildroot.org/results/4d91c9f82a2a61c50c457a851073b85cc09e… Signed-off-by: Fabrice Fontaine <fontaine.fabrice(a)gmail.com> Request URL : https://github.com/alsa-project/alsa-lib/pull/15 Patch URL : https://github.com/alsa-project/alsa-lib/pull/15.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[alsa-devel] [PATCH] ASoC: core: Init pcm runtime work early to avoid warnings
by Curtis Malainey 03 Dec '19

03 Dec '19

There are cases where we fail before we reach soc_new_pcm which would init the workqueue. When we fail we attempt to flush the queue which generates warnings from the workqueue subsystem when we have not inited the queue. Solution is to use a proxy function to get around this issue. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Cc: Takashi Iwai <tiwai(a)suse.de> --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 10 ++++++++++ sound/soc/soc-pcm.c | 11 ++++------- 3 files changed, 15 insertions(+), 7 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index c28a1ed5e8df9..2628967998264 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1150,6 +1150,7 @@ struct snd_soc_pcm_runtime { unsigned int num_codecs; struct delayed_work delayed_work; + void (*close_delayed_work_func)(struct snd_soc_pcm_runtime *rtd); #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dpcm_root; #endif diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 062653ab03a37..0e2e628302f1d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -435,6 +435,15 @@ static void soc_free_pcm_runtime(struct snd_soc_pcm_runtime *rtd) device_unregister(rtd->dev); } +static void close_delayed_work(struct work_struct *work) { + struct snd_soc_pcm_runtime *rtd = + container_of(work, struct snd_soc_pcm_runtime, + delayed_work.work); + + if (rtd->close_delayed_work_func) + rtd->close_delayed_work_func(rtd); +} + static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { @@ -470,6 +479,7 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( rtd->dev = dev; dev_set_drvdata(dev, rtd); + INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); /* * for rtd->codec_dais diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 01eb8700c3de5..b78f6ff2b1d3f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -637,10 +637,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) * This is to ensure there are no pops or clicks in between any music tracks * due to DAPM power cycling. */ -static void close_delayed_work(struct work_struct *work) +static void close_delayed_work(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_pcm_runtime *rtd = - container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); struct snd_soc_dai *codec_dai = rtd->codec_dais[0]; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); @@ -660,7 +658,7 @@ static void close_delayed_work(struct work_struct *work) mutex_unlock(&rtd->card->pcm_mutex); } -static void codec2codec_close_delayed_work(struct work_struct *work) +static void codec2codec_close_delayed_work(struct snd_soc_pcm_runtime *rtd) { /* * Currently nothing to do for c2c links @@ -2974,10 +2972,9 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) /* DAPM dai link stream work */ if (rtd->dai_link->params) - INIT_DELAYED_WORK(&rtd->delayed_work, - codec2codec_close_delayed_work); + rtd->close_delayed_work_func = codec2codec_close_delayed_work; else - INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); + rtd->close_delayed_work_func = close_delayed_work; pcm->nonatomic = rtd->dai_link->nonatomic; rtd->pcm = pcm; -- 2.24.0.393.g34dc348eaf-goog

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[alsa-devel] [PATCH] ASoC: da7219: remove SRM lock check retry
by Brent Lu 03 Dec '19

03 Dec '19

For platforms not able to provide WCLK in the PREPARED runtime state, it takes 400ms for codec driver to print the message "SRM failed to lock" in the da7219_dai_event() function which is called when DAPM widgets are powering up. The latency penalty to audio input/output is too much so the retry (8 times) and delay (50ms each retry) are removed. Another reason is current Cold output latency requirement in Android CDD is 500ms but will be reduced to 200ms for 2021 platforms. With the 400ms latency it would be difficult to pass the Android CTS test. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/codecs/da7219.c | 3 ++- sound/soc/codecs/da7219.h | 2 +- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index f83a6ea..042e701 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -833,7 +833,8 @@ static int da7219_dai_event(struct snd_soc_dapm_widget *w, srm_lock = true; } else { ++i; - msleep(50); + if (i < DA7219_SRM_CHECK_RETRIES) + msleep(50); } } while ((i < DA7219_SRM_CHECK_RETRIES) && (!srm_lock)); diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h index 88b67fe..3149986 100644 --- a/sound/soc/codecs/da7219.h +++ b/sound/soc/codecs/da7219.h @@ -770,7 +770,7 @@ #define DA7219_PLL_INDIV_36_TO_54_MHZ_VAL 16 /* SRM */ -#define DA7219_SRM_CHECK_RETRIES 8 +#define DA7219_SRM_CHECK_RETRIES 1 /* System Controller */ #define DA7219_SYS_STAT_CHECK_RETRIES 6 -- 2.7.4

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