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[PATCH v3 0/3] ALSA: pcm: add snd_pcm_period_elapsed() variant without acquiring lock of PCM substream
by Takashi Sakamoto 10 Jun '21

10 Jun '21

Hi, This patchset is revised version of my previous one: * https://lore.kernel.org/alsa-devel/20210609143145.146680-1-o-takashi@sakamo… All of drivers in ALSA firewire stack have two chances to process isochronous packets of any isochronous context; in software IRQ context for 1394 OHCI, and in process context of ALSA PCM application. In the process context, callbacks of .pointer and .ack are utilized. The callbacks are done by ALSA PCM core under acquiring lock of PCM substream, In design of ALSA PCM core, call of snd_pcm_period_elapsed() is used for drivers to awaken user processes from waiting for available frames. The function voluntarily acquires lock of PCM substream, therefore it is not called in the process context since it causes dead lock. As a workaround to avoid the dead lock, all of drivers in ALSA firewire stack use workqueue to delegate the call. This patchset is my attempt for the issue. A variant of 'snd_pcm_period_elapsed()' without lock acquisition is going to be added, named 'snd_pcm_period_elapsed_under_stream_lock()'. The call is available in callbacks of .pointer and .ack of snd_pcm_ops structure. Changes from v2: * delete context section from kernel API documentation Takashi Sakamoto (3): ALSA: pcm: add snd_pcm_period_elapsed() variant without acquiring lock of PCM substream ALSA: firewire-lib: operate for period elapse event in process context ALSA: firewire-lib: obsolete workqueue for period update include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 64 +++++++++++++++++++++++++++-------- sound/firewire/amdtp-stream.c | 46 ++++++++----------------- sound/firewire/amdtp-stream.h | 1 - 4 files changed, 64 insertions(+), 48 deletions(-) -- 2.27.0

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[Bug] Duplication in Array Element not Handled in Parsing conf
by GitHub issues - opened 10 Jun '21

10 Jun '21

alsa-project/alsa-lib issue #151 was opened from aiChaoSONG: When parsing array element in conf files, an auto-increase number is used as config id(`snd_config_t->id`), and array element as the config value(`snd_config_t->u.string/integer/integer64/real`), duplication is not handle in `parse_array_defs/parse_array_def/parse_value` functions. Suppose we have a conf: ``` array [ "a" "b" "a" ] ``` the parsed result will be: ``` parent | ---- snd_config_t {id: "0", u.string: "a"} | ---- snd_config_t {id: "1", u.string: "b"} | ---- snd_config_t {id: "2", u.string: "a"} ``` The duplication in "a" may be desired. But please consider this condition: suppose we have four conf files A, B, C and D, where A is the conf above, let B include A, C include A, D include B and C, so D include A twice indirectly. File A will be processed twice when parsing D, because duplication is not handled, here comes the problem, the parsed result of A in D is: ``` parent | ---- snd_config_t {id: "0", u.string: "a"} | ---- snd_config_t {id: "1", u.string: "b"} | ---- snd_config_t {id: "2", u.string: "a"} | ---- snd_config_t {id: "3", u.string: "a"} | ---- snd_config_t {id: "4", u.string: "b"} | ---- snd_config_t {id: "5", u.string: "a"} ``` This is obviously wrong. I think handle duplication in parse_value is not the solution, because element duplication in an array is wanted. The key problem here is that we parse the same conf more than once. Issue URL : https://github.com/alsa-project/alsa-lib/issues/151 Repository URL: https://github.com/alsa-project/alsa-lib

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[bug report] ASoC: rsnd: add null CLOCKIN support
by Dan Carpenter 10 Jun '21

10 Jun '21

Hello Kuninori Morimoto, The patch d6956a7dde6f: "ASoC: rsnd: add null CLOCKIN support" from May 24, 2021, leads to the following static checker warning: sound/soc/sh/rcar/adg.c:435 rsnd_adg_get_clkin() warn: 'clk' isn't an ERR_PTR sound/soc/sh/rcar/adg.c 423 static int rsnd_adg_get_clkin(struct rsnd_priv *priv) 424 { 425 struct rsnd_adg *adg = priv->adg; 426 struct device *dev = rsnd_priv_to_dev(priv); 427 struct clk *clk; 428 int i; 429 430 for (i = 0; i < CLKMAX; i++) { 431 clk = devm_clk_get(dev, clk_name[i]); If devm_clk_get() returns NULL that's not an error. 432 433 if (IS_ERR(clk)) 434 clk = rsnd_adg_null_clk_get(priv); But I think if rsnd_adg_null_clk_get() returns an error then it is an error. 435 if (IS_ERR(clk)) This is impossible. 436 goto err; So I think it should be: if (IS_ERR(clk)) { clk = rsnd_adg_null_clk_get(priv); if (!clk) goto err; } 437 438 adg->clk[i] = clk; 439 } 440 441 return 0; 442 443 err: 444 dev_err(dev, "adg clock IN get failed\n"); 445 446 rsnd_adg_null_clk_clean(priv); 447 448 return -EIO; 449 } regards, dan carpenter

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[PATCH v3 0/3] ALSA: pcm: add snd_pcm_period_elapsed() variant without acquiring lock of PCM substream
by Takashi Sakamoto 10 Jun '21

10 Jun '21

Hi, This patchset is revised version of my previous one: * https://lore.kernel.org/alsa-devel/20210609143145.146680-1-o-takashi@sakamo… All of drivers in ALSA firewire stack have two chances to process isochronous packets of any isochronous context; in software IRQ context for 1394 OHCI, and in process context of ALSA PCM application. In the process context, callbacks of .pointer and .ack are utilized. The callbacks are done by ALSA PCM core under acquiring lock of PCM substream, In design of ALSA PCM core, call of snd_pcm_period_elapsed() is used for drivers to awaken user processes from waiting for available frames. The function voluntarily acquires lock of PCM substream, therefore it is not called in the process context since it causes dead lock. As a workaround to avoid the dead lock, all of drivers in ALSA firewire stack use workqueue to delegate the call. This patchset is my attempt for the issue. A variant of 'snd_pcm_period_elapsed()' without lock acquisition is going to be added, named 'snd_pcm_period_elapsed_under_stream_lock()'. The call is available in callbacks of .pointer and .ack of snd_pcm_ops structure. Changes from v2: * delete context section from kernel API documentation Takashi Sakamoto (3): ALSA: pcm: add snd_pcm_period_elapsed() variant without acquiring lock of PCM substream ALSA: firewire-lib: operate for period elapse event in process context ALSA: firewire-lib: obsolete workqueue for period update include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 64 +++++++++++++++++++++++++++-------- sound/firewire/amdtp-stream.c | 46 ++++++++----------------- sound/firewire/amdtp-stream.h | 1 - 4 files changed, 64 insertions(+), 48 deletions(-) -- 2.27.0

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aplay change to use 16-bit samples by default is not backwards compatible and is not documented
by GitHub issues - opened 10 Jun '21

10 Jun '21

alsa-project/alsa-utils issue #96 was opened from nwoeanhinnogaehr: The change introduced in https://github.com/alsa-project/alsa-utils/commit/0c5948e98a6a8535c89b7bcab… breaks a lot of scripts and other programs I have which invoke `aplay` and expect the default sample format to be U8. While I think it's somewhat reasonable to make this change given that U8 is a rather low quality sample format, it's worth acknowledging that aplay has had this behavior for a very long time and many people probably don't expect it to change. Given that the commit log seems to only talk about the case of `arecord`, I wonder if we could reach a compromise and keep the format as U8 for `aplay` and switch it for `arecord`, since users are unlikely to complain about poor sound quality when *playing raw audio* (surely they would take the time to get the sample format set up correctly before attempting to play raw audio). Also, the man page has not been updated to reflect this change and still says that U8 is the default. Issue URL : https://github.com/alsa-project/alsa-utils/issues/96 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH v2 0/8] ASoC: tidyup snd_soc_of_parse_daifmt()
by Kuninori Morimoto 10 Jun '21

10 Jun '21

Hi Mark These are v2 of parsing for daifmt. I want to add new audio-graph-card2 sound card driver, and this is last part of necessary soc-core cleanup for it. Current some drivers are using DT, and then, snd_soc_of_parse_daifmt() parses daifmt, but bitclock/frame provider parsing part is one of headache, because we are assuming below both cases. A) node { bitclock-master; frame-master; ... }; B) link { bitclock-master = <&xxx>; frame-master = <&xxx>; ... }; The original was style A), and style B) was added later. snd_soc_of_parse_daifmt() parses A) style as original style, and user need to update to B) style for clock_provider part if needed. To handle it more flexibile, this patch-set adds new functions which separates snd_soc_of_parse_daifmt() helper function. snd_soc_daifmt_parse_format() : format part snd_soc_daifmt_parse_clock_provider_as_flag() : clock part for style A) snd_soc_daifmt_parse_clock_provider_as_phandl() : clock part for style B) snd_soc_daifmt_parse_clock_provider_as_bitmap() : clock part use with _from_bitmap v1 -> v2 - tidyup parse_clock_provider functions to _as_flag/phandle/bitmap() - don't exchange code style on each drivers. Link: https://lore.kernel.org/r/875yypdxlm.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (8): ASoC: soc-core: add snd_soc_daifmt_clock_provider_from_bitmap() ASoC: soc-core: add snd_soc_daifmt_clock_provider_fliped() ASoC: soc-core: add snd_soc_daifmt_parse_format/clock_provider() ASoC: atmel: switch to use snd_soc_daifmt_parse_format/clock_provider() ASoC: fsl: switch to use snd_soc_daifmt_parse_format/clock_provider() ASoC: meson: switch to use snd_soc_daifmt_parse_format/clock_provider() ASoC: simple-card-utils: switch to use snd_soc_daifmt_parse_format/clock_provider() ASoC: soc-core: remove snd_soc_of_parse_daifmt() include/sound/soc.h | 21 ++++-- sound/soc/atmel/mikroe-proto.c | 16 +++-- sound/soc/fsl/fsl-asoc-card.c | 14 ++-- sound/soc/generic/simple-card-utils.c | 19 ++--- sound/soc/meson/meson-card-utils.c | 13 ++-- sound/soc/soc-core.c | 99 +++++++++++++++++---------- 6 files changed, 110 insertions(+), 72 deletions(-) -- 2.25.1

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[PATCH 0/5] ALSA: More abstracition of memory alloc helpers
by Takashi Iwai 09 Jun '21

09 Jun '21

Hi, this is a patch set for the code cleanup and refactoring around the ALSA core memory allocation helpers. It begins with a cleanup of the non-used code, then followed by the refacotoring with the new ops, finally moved the memory-specific mmap handling code from PCM to the core memalloc. Takashi === Takashi Iwai (5): ALSA: trident: Drop shadow TLB pointer table ALSA: core: Drop snd_sgbuf_get_ptr() ALSA: core: Abstract memory alloc helpers ALSA: core: Move mmap handler into memalloc ops ALSA: core: Add continuous and vmalloc mmap ops include/sound/memalloc.h | 83 +----- include/sound/pcm.h | 19 -- sound/core/memalloc.c | 444 +++++++++++++++++++---------- sound/core/memalloc_local.h | 19 ++ sound/core/pcm_local.h | 5 - sound/core/pcm_memory.c | 21 -- sound/core/pcm_native.c | 39 +-- sound/core/sgbuf.c | 90 ++++-- sound/pci/trident/trident.h | 1 - sound/pci/trident/trident_main.c | 11 +- sound/pci/trident/trident_memory.c | 53 ++-- 11 files changed, 405 insertions(+), 380 deletions(-) create mode 100644 sound/core/memalloc_local.h -- 2.26.2

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Current version not working with Focusrite Scarlett 18i20
by GitHub issues - opened 09 Jun '21

09 Jun '21

alsa-project/alsa-lib issue #150 was opened from TheBitFighter: Hello, I hope I am in the right repository for this. My arch system updated alsa-card-profiles and alsa-utils today which caused my interface, a Focusrite Scarlett 18i20 1st gen to stop working. The card was still showing up in pulse and showed output, however no sound was actually played. Input was still working fine. It sounded like the output stage of the interface had been disabled, and the problem could only be fixed by booting Windows and setting new routing settings in Scarlett MixControl. The interface had exhibited some resetting and popping behavior for a few versions now (when stopping playback, changing application and playing something else there), but continued to work (given a few instances where sample rate played mismatched sample rate sent which caused weird audio slowdowns). It only stopped working today. What I did initially was reinstall the older versions of these two packages, which did not fix the problem. I now downgraded alsa-libs (1.2.5-2 -> 1.2.4-2), alsa-card-profiles (1:0.3.29-2 -> 1:0.3.28-1), lib32-alsa-lib (1.2.5-2 -> 1.2.4-1), alsa-topology-conf (1.2.5-1 -> 1.2.4-1), and alsa-ucm-conf (1.2.5-1 -> 1.2.4-1) which fixed the problem. (alsa-utils is at 1.2.5-2) `cat /etc/asound/version` now reports `Advanced Linux Sound Architecture Driver Version k5.12.9-arch1-1.` This also fixed the popping/resetting issues I had before. Thanks for the help! If you need any further information I am happy to help. Issue URL : https://github.com/alsa-project/alsa-lib/issues/150 Repository URL: https://github.com/alsa-project/alsa-lib

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opensuse tumbleweed no sound after alsa lib update (tumbleweed 2010605)
by GitHub issues - opened 09 Jun '21

09 Jun '21

alsa-project/alsa-lib issue #149 was opened from pawanyadav: sound is not working alsa (never installed pipewire and pulseaudio) asound.conf defaults.pcm.card 1 defaults.pcm.device 1 aplay -l **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Generic [HD-Audio Generic], device 0: ALC887-VD Analog [ALC887-VD Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Generic [HD-Audio Generic], device 1: ALC887-VD Digital [ALC887-VD Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 cat /proc/asound/cards 0 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xfcf40000 irq 67 1 [Generic ]: HDA-Intel - HD-Audio Generic HD-Audio Generic at 0xfce00000 irq 69 speaker-test 1.2.4 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise ALSA lib confmisc.c:855:(parse_card) cannot find card '1dmix:1' ALSA lib conf.c:5091:(_snd_config_evaluate) function snd_func_card_id returned error: No such device ALSA lib confmisc.c:422:(snd_func_concat) error evaluating strings ALSA lib conf.c:5091:(_snd_config_evaluate) function snd_func_concat returned error: No such device ALSA lib confmisc.c:1334:(snd_func_refer) error evaluating name ALSA lib conf.c:5091:(_snd_config_evaluate) function snd_func_refer returned error: No such device ALSA lib conf.c:5579:(snd_config_expand) Evaluate error: No such device ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM dmix:1dmix:1 Playback open error: -19,No such device Issue URL : https://github.com/alsa-project/alsa-lib/issues/149 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH v2] ASoC: qcom: Add four speaker support on MI2S secondary
by Srinivasa Rao Mandadapu 09 Jun '21

09 Jun '21

Add four speaker support on MI2S secondary block by using I2S SD1 line on gpio52 pin, and add channel map control support in the lpass-cpu audio driver. Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- Changes Since V1: -- removed set_channel_map/get_channel_map implementation as default kcontrols added in pcm_new API. sound/soc/qcom/lpass-cpu.c | 32 ++++++++++++++++++++++++++++++++ sound/soc/qcom/lpass-sc7180.c | 1 + sound/soc/qcom/lpass.h | 2 ++ 3 files changed, 35 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index c62d2612e8f5..15d69e311b38 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -29,6 +29,14 @@ #define LPASS_CPU_I2S_SD0_1_2_MASK GENMASK(2, 0) #define LPASS_CPU_I2S_SD0_1_2_3_MASK GENMASK(3, 0) +/* + * Channel maps for Quad channel playbacks on MI2S Secondary + */ +static struct snd_pcm_chmap_elem lpass_quad_chmaps = { + .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_RL, + SNDRV_CHMAP_FR, SNDRV_CHMAP_RR } +}; static int lpass_cpu_init_i2sctl_bitfields(struct device *dev, struct lpaif_i2sctl *i2sctl, struct regmap *map) { @@ -324,6 +332,25 @@ const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); +int lpass_cpu_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + int ret; + struct snd_soc_dai_driver *drv = dai->driver; + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + + if (drvdata->mi2s_playback_sd_mode[dai->id] == LPAIF_I2SCTL_MODE_QUAD01) { + ret = snd_pcm_add_chmap_ctls(rtd->pcm, SNDRV_PCM_STREAM_PLAYBACK, + &lpass_quad_chmaps, drv->playback.channels_max, 0, + NULL); + if (ret < 0) + return ret; + } + + return 0; +} +EXPORT_SYMBOL_GPL(lpass_cpu_pcm_new); + int asoc_qcom_lpass_cpu_dai_probe(struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); @@ -856,6 +883,11 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) PTR_ERR(drvdata->mi2s_bit_clk[dai_id])); return PTR_ERR(drvdata->mi2s_bit_clk[dai_id]); } + if (drvdata->mi2s_playback_sd_mode[dai_id] == + LPAIF_I2SCTL_MODE_QUAD01) { + variant->dai_driver[dai_id].playback.channels_min = 4; + variant->dai_driver[dai_id].playback.channels_max = 4; + } } /* Allocation for i2sctl regmap fields */ diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index 8c168d3c589e..77a556b27cf0 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -58,6 +58,7 @@ static struct snd_soc_dai_driver sc7180_lpass_cpu_dai_driver[] = { }, .probe = &asoc_qcom_lpass_cpu_dai_probe, .ops = &asoc_qcom_lpass_cpu_dai_ops, + .pcm_new = lpass_cpu_pcm_new, }, { .id = LPASS_DP_RX, .name = "Hdmi", diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 83b2e08ade06..623ddccdafff 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -259,5 +259,7 @@ void asoc_qcom_lpass_cpu_platform_shutdown(struct platform_device *pdev); int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev); int asoc_qcom_lpass_cpu_dai_probe(struct snd_soc_dai *dai); extern const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops; +int lpass_cpu_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai); #endif /* __LPASS_H__ */ -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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