June 2021 - Alsa-devel - mailman.alsa-project.org

Logitech C922x microphone intermittently working
by GitHub issues - opened 10 Jun '21

10 Jun '21

alsa-project/alsa-lib issue #152 was opened from sambo57u: I have a Logitech C922X (detected as C922) USB webcam. Running latest updates on Fedora 34. The microphone of this device only works about 1 out of 5 reboots. When not working if I install another microphone, activate that one, the unplug and plug C922 back and switch to it, it starts to work. Some kind of a timing issue that is very annoying. Don't have this problem with Logitech C270. The camera always works fine. Issue URL : https://github.com/alsa-project/alsa-lib/issues/152 Repository URL: https://github.com/alsa-project/alsa-lib

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Undeclarated constant
by GitHub issues - opened 10 Jun '21

10 Jun '21

alsa-project/alsa-plugins issue #24 was opened from unknown2045: Hi, I am trying to install the alsa-plugins, when I do the make command I get the following errors: ![Captura de pantalla de 2021-06-10 13-52-37](https://user-images.githubusercontent.com/85682900/121520376-229e… How can I solve it? Thanks in advance! Issue URL : https://github.com/alsa-project/alsa-plugins/issues/24 Repository URL: https://github.com/alsa-project/alsa-plugins

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[sound:for-next 164/168] include/sound/memalloc.h:63:17: error: use of undeclared identifier 'PAGE_SIZE'
by kernel test robot 10 Jun '21

10 Jun '21

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 83fbcaed24d797528de00c024674fd58e8f1634f commit: 37af81c5998f4b0f23fb452cffa4b8a1c00ce95b [164/168] ALSA: core: Abstract memory alloc helpers config: riscv-randconfig-r002-20210610 (attached as .config) compiler: clang version 13.0.0 (https://github.com/llvm/llvm-project d2012d965d60c3258b3a69d024491698f8aec386) reproduce (this is a W=1 build): wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross # install riscv cross compiling tool for clang build # apt-get install binutils-riscv64-linux-gnu # https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?id=… git remote add sound https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git git fetch --no-tags sound for-next git checkout 37af81c5998f4b0f23fb452cffa4b8a1c00ce95b # save the attached .config to linux build tree COMPILER_INSTALL_PATH=$HOME/0day COMPILER=clang make.cross ARCH=riscv If you fix the issue, kindly add following tag as appropriate Reported-by: kernel test robot <lkp(a)intel.com> All errors (new ones prefixed by >>): In file included from sound/core/pcm_iec958.c:8: In file included from include/sound/pcm.h:12: >> include/sound/memalloc.h:63:17: error: use of undeclared identifier 'PAGE_SIZE' return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; ^ >> include/sound/memalloc.h:63:35: error: use of undeclared identifier 'PAGE_SHIFT' return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; ^ 2 errors generated. vim +/PAGE_SIZE +63 include/sound/memalloc.h ^1da177e4c3f41 Linus Torvalds 2005-04-16 57 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 58 /* 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 59 * return the pages matching with the given byte size 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 60 */ 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 61 static inline unsigned int snd_sgbuf_aligned_pages(size_t size) 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 62 { 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 @63 return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 64 } 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 65 :::::: The code at line 63 was first introduced by commit :::::: 4cae99d9b5305ab8cccc839fccceb81ec9e5abda ALSA: memalloc: declare snd_sgbuf_aligned_pages() unconditionally :::::: TO: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

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[PATCH] ALSA: core: Fix build error due to missing PAGE_SIZE
by Takashi Iwai 10 Jun '21

10 Jun '21

The recent refactoring of memalloc stuff removed the inclusion of asm/page.h for simplicity, but it turned out this caused a compile error due the lack of PAGE_SIZE definition on some architectures. Do a partial revert for recovering from that. Fixes: 37af81c5998f ("ALSA: core: Abstract memory alloc helpers") Reported-by: kernel test robot <lkp(a)intel.com> Link: https://lore.kernel.org/r/202106101858.PfXMMuAa-lkp@intel.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/memalloc.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index 1918c60f1f35..44d87775b352 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -9,8 +9,9 @@ #ifndef __SOUND_MEMALLOC_H #define __SOUND_MEMALLOC_H +#include <asm/page.h> + struct device; -struct page; struct vm_area_struct; /* -- 2.26.2

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[sound:for-next 164/167] include/sound/memalloc.h:63:17: error: 'PAGE_SIZE' undeclared
by kernel test robot 10 Jun '21

10 Jun '21

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 7fdabab8f56239466275adea5d2f0a6fff27527b commit: 37af81c5998f4b0f23fb452cffa4b8a1c00ce95b [164/167] ALSA: core: Abstract memory alloc helpers config: h8300-randconfig-r006-20210610 (attached as .config) compiler: h8300-linux-gcc (GCC) 9.3.0 reproduce (this is a W=1 build): wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross # https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?id=… git remote add sound https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git git fetch --no-tags sound for-next git checkout 37af81c5998f4b0f23fb452cffa4b8a1c00ce95b # save the attached .config to linux build tree COMPILER_INSTALL_PATH=$HOME/0day COMPILER=gcc-9.3.0 make.cross ARCH=h8300 If you fix the issue, kindly add following tag as appropriate Reported-by: kernel test robot <lkp(a)intel.com> All errors (new ones prefixed by >>): In file included from include/sound/pcm.h:12, from sound/core/pcm_iec958.c:8: include/sound/memalloc.h: In function 'snd_sgbuf_aligned_pages': >> include/sound/memalloc.h:63:17: error: 'PAGE_SIZE' undeclared (first use in this function) 63 | return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; | ^~~~~~~~~ include/sound/memalloc.h:63:17: note: each undeclared identifier is reported only once for each function it appears in >> include/sound/memalloc.h:63:35: error: 'PAGE_SHIFT' undeclared (first use in this function) 63 | return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; | ^~~~~~~~~~ vim +/PAGE_SIZE +63 include/sound/memalloc.h ^1da177e4c3f41 Linus Torvalds 2005-04-16 57 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 58 /* 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 59 * return the pages matching with the given byte size 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 60 */ 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 61 static inline unsigned int snd_sgbuf_aligned_pages(size_t size) 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 62 { 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 @63 return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 64 } 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 65 :::::: The code at line 63 was first introduced by commit :::::: 4cae99d9b5305ab8cccc839fccceb81ec9e5abda ALSA: memalloc: declare snd_sgbuf_aligned_pages() unconditionally :::::: TO: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

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[PATCH 0/1 v2] usb-audio: Add support for Denon DN-X1600
by Damien Zammit 10 Jun '21

10 Jun '21

Hi Takashi, You are correct, the patch still works without the hunk referring to the PITCH control. I have removed the hunk and was tested with this patch and it still functioned as expected. Regards, Damien Z

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[PATCH] Docs: hda: Fix spelling mistakes
by huangjianghui 10 Jun '21

10 Jun '21

correct Reatek to Realtek Signed-off-by: huangjianghui <huangjianghui(a)uniontech.com> --- Documentation/sound/hd-audio/controls.rst | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/sound/hd-audio/controls.rst b/Documentation/sound/hd-audio/controls.rst index f2ebc4f79b44..dbe6483f4ff4 100644 --- a/Documentation/sound/hd-audio/controls.rst +++ b/Documentation/sound/hd-audio/controls.rst @@ -102,7 +102,7 @@ Conexant codecs --------------- Auto-Mute Mode - See Reatek codecs. + See Realtek codecs. Analog codecs -- 2.20.1

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[PATCH] sound/arm: Remove unnecessary variables
by zuoqilin1＠163.com 10 Jun '21

10 Jun '21

From: zuoqilin <zuoqilin(a)yulong.com> There is no need to define the variable "ret" to receive. Signed-off-by: zuoqilin <zuoqilin(a)yulong.com> --- sound/arm/pxa2xx-ac97.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 6322e639..a67e668 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -47,9 +47,7 @@ static unsigned short pxa2xx_ac97_legacy_read(struct snd_ac97 *ac97, static void pxa2xx_ac97_legacy_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { - int __always_unused ret; - - ret = pxa2xx_ac97_write(ac97->num, reg, val); + pxa2xx_ac97_write(ac97->num, reg, val); } static const struct snd_ac97_bus_ops pxa2xx_ac97_ops = { -- 1.9.1

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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.

2 2

[PATCH] usb-audio: Add support for Denon DN-X1600
by Damien Zammit 09 Jun '21

09 Jun '21

This patch applies clean to tiwai/sound.git repo on hash 82b954a96c1550bae2cb1f1f4f084425ab237cd7 and was tested on this version. This provides support for Denon DN-X1600 hardware mixer. The device itself supports 44100, 48000 and 96000 (Hz) sample rates, but switching rates via software is currently not working. Therefore, this patch hardcodes the sample rate to 48000Hz which enables all 8 channels to function correctly when the correct sample rate is selected on the hardware itself. MIDI also tested and works. Signed-off-by: Damien Zammit <damien(a)zamaudio.com> Tested-by: xalmoxis(a)gmail.com --- sound/usb/clock.c | 7 +++++ sound/usb/pcm.c | 4 +++ sound/usb/quirks-table.h | 70 ++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 81 insertions(+) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 0afae839d..52de52288 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -534,6 +534,13 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, * rate. */ clock = snd_usb_clock_find_source(chip, fmt, false); + + /* Denon DN-X1600 hardcoded + * Sample rate seems to be set on the hardware itself + */ + if (chip->usb_id == USB_ID(0x154e, 0x500e)) + return 0; + if (clock < 0) return clock; } diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index c66831ee1..6913a5b02 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -197,6 +197,10 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, if (!(fmt->attributes & UAC_EP_CS_ATTR_PITCH_CONTROL)) return 0; + /* Denon DN-X1600 doesn't properly handle this, bail out too */ + if (chip->usb_id == USB_ID(0x154e, 0x500e)) + return 0; + usb_audio_dbg(chip, "enable PITCH for EP 0x%x\n", fmt->endpoint); switch (fmt->protocol) { diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index bdba37d0f..19bb499c1 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3044,6 +3044,76 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } }, +/* Denon DN-X1600 */ +{ + USB_AUDIO_DEVICE(0x154e, 0x500e), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Denon", + .product_name = "DN-X1600", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]){ + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE, + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0x0, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ADAPTIVE, + .maxpacksize = 0x138, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0x0, + .endpoint = 0x85, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ADAPTIVE, + .maxpacksize = 0x138, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + { + .ifnum = 4, + .type = QUIRK_MIDI_STANDARD_INTERFACE, + }, + { + .ifnum = -1 + } + } + } +}, + /* Microsoft XboxLive Headset/Xbox Communicator */ { USB_DEVICE(0x045e, 0x0283), -- 2.13.1

2 1

[PATCH 0/3] ALSA: pcm:firewire: allow to operate for period elapse event in process context
by Takashi Sakamoto 09 Jun '21

09 Jun '21

Hi, This patchset comes from my former RFC: [RFC][PATCH 0/3] ALSA: pcm/firewire: allow to queue period elapse event in process context * https://lore.kernel.org/alsa-devel/20210606091838.80812-1-o-takashi@sakamoc… 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 RFC: * dismiss inlining * rename function name * improve function comments I tested the patchset with ALSA OXFW driver and ftrace, covering axfer and jackd cases since pulseaudio and pipewire are programmed with timer-based scheduling model and ALSA runtime expects drivers not to call snd_pcm_period_elapsed(). The configuration of tracer and filters is: ``` $ sudo trace-cmd record -p function_graph \ -l :mod:snd_firewire_lib \ -l fw_iso_context_queue:mod:firewire_core \ -l snd_pcm_ioctl:mod:snd_pcm \ -l snd_pcm_period_elapsed*:mod:snd_pcm ``` The runtime of axfer without '-M' option can often call .pointer and .ack. Below is a sample about .pointer case via SNDRV_PCM_IOCT_HWSYNC. The new snd_pcm_period_elapsed_under_stream_lock() should be called. ``` 3929.769359: funcgraph_entry: | snd_pcm_ioctl() { 3929.769360: funcgraph_entry: | amdtp_domain_stream_pcm_pointer() { 3929.769361: funcgraph_entry: | irq_target_callback() { 3929.769361: funcgraph_entry: | process_rx_packets() { 3929.769362: funcgraph_entry: | process_ctx_payloads() { 3929.769363: funcgraph_entry: 0.391 us | process_it_ctx_payloads(); 3929.769363: funcgraph_entry: | snd_pcm_period_elapsed_under_stream_lock() { 3929.769364: funcgraph_entry: 0.411 us | amdtp_domain_stream_pcm_pointer(); 3929.769365: funcgraph_exit: 1.723 us | } 3929.769365: funcgraph_exit: 3.106 us | } 3929.769366: funcgraph_entry: | queue_packet() { 3929.769366: funcgraph_entry: 0.561 us | fw_iso_context_queue(); 3929.769367: funcgraph_exit: 1.122 us | } 3929.769367: funcgraph_exit: 5.731 us | } 3929.769367: funcgraph_entry: | process_ctxs_in_domain() { 3929.769368: funcgraph_entry: | process_tx_packets() { 3929.769369: funcgraph_entry: | process_ctx_payloads() { 3929.769369: funcgraph_entry: 0.321 us | process_ir_ctx_payloads(); 3929.769369: funcgraph_exit: 0.962 us | } 3929.769370: funcgraph_entry: 0.491 us | fw_iso_context_queue(); 3929.769371: funcgraph_exit: 2.364 us | } 3929.769371: funcgraph_exit: 3.427 us | } 3929.769371: funcgraph_exit: 10.038 us | } 3929.769372: funcgraph_exit: 11.271 us | } 3929.769372: funcgraph_exit: 13.606 us | } ``` The runtime of jackd heavily relies on period wakeup scheduled in invocation of interrupt handler. Below is a sample about software IRQ case of 1394 OHCI. The snd_pcm_period_elapsed() should be called. ``` 5318.980502: funcgraph_entry: | irq_target_callback() { 5318.980503: funcgraph_entry: | process_rx_packets() { 5318.980503: funcgraph_entry: | process_ctx_payloads() { 5318.980503: funcgraph_entry: 0.531 us | process_it_ctx_payloads(); 5318.980505: funcgraph_entry: | snd_pcm_period_elapsed() { 5318.980505: funcgraph_entry: | snd_pcm_period_elapsed_under_stream_lock() { 5318.980505: funcgraph_entry: 0.180 us | amdtp_domain_stream_pcm_pointer(); 5318.980508: funcgraph_exit: 3.106 us | } 5318.980508: funcgraph_exit: 3.497 us | } 5318.980508: funcgraph_exit: 4.990 us | } 5318.980508: funcgraph_entry: | queue_packet() { 5318.980509: funcgraph_entry: 0.350 us | fw_iso_context_queue(); 5318.980509: funcgraph_exit: 0.702 us | } 5318.980509: funcgraph_entry: | queue_packet() { 5318.980509: funcgraph_entry: 0.291 us | fw_iso_context_queue(); 5318.980510: funcgraph_exit: 0.651 us | } 5318.980510: funcgraph_entry: | queue_packet() { 5318.980510: funcgraph_entry: 0.280 us | fw_iso_context_queue(); 5318.980511: funcgraph_exit: 0.641 us | } 5318.980511: funcgraph_exit: 8.416 us | } 5318.980511: funcgraph_entry: | process_ctxs_in_domain() { 5318.980512: funcgraph_entry: | process_tx_packets() { 5318.980512: funcgraph_entry: | process_ctx_payloads() { 5318.980512: funcgraph_entry: 0.531 us | process_ir_ctx_payloads(); 5318.980513: funcgraph_entry: | snd_pcm_period_elapsed() { 5318.980513: funcgraph_entry: | snd_pcm_period_elapsed_under_stream_lock() { 5318.980513: funcgraph_entry: 0.170 us | amdtp_domain_stream_pcm_pointer(); 5318.980514: funcgraph_exit: 0.842 us | } 5318.980514: funcgraph_exit: 1.242 us | } 5318.980514: funcgraph_exit: 2.335 us | } 5318.980515: funcgraph_entry: 0.301 us | fw_iso_context_queue(); 5318.980515: funcgraph_entry: 0.291 us | fw_iso_context_queue(); 5318.980516: funcgraph_entry: 0.290 us | fw_iso_context_queue(); 5318.980516: funcgraph_exit: 4.198 us | } 5318.980516: funcgraph_exit: 5.119 us | } 5318.980516: funcgraph_exit: 14.077 us | } ``` ALSA OXFW driver works well to select appropriate kernel API by distinguishing running context. I also tested for error cases that the sequence of tx packets is invalid or packet queueing fails. In both cases, the error doesn't bring system corruption. 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 | 68 +++++++++++++++++++++++++++-------- sound/firewire/amdtp-stream.c | 46 ++++++++---------------- sound/firewire/amdtp-stream.h | 1 - 4 files changed, 68 insertions(+), 48 deletions(-) -- 2.27.0

2 5

[PATCH] 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> --- sound/soc/qcom/lpass-cpu.c | 106 +++++++++++++++++++++++++++++++++- sound/soc/qcom/lpass-sc7180.c | 1 + sound/soc/qcom/lpass.h | 5 ++ 3 files changed, 111 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index c62d2612e8f5..88a038a107a7 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -28,6 +28,19 @@ #define LPASS_CPU_I2S_SD2_3_MASK GENMASK(3, 2) #define LPASS_CPU_I2S_SD0_1_2_MASK GENMASK(2, 0) #define LPASS_CPU_I2S_SD0_1_2_3_MASK GENMASK(3, 0) +#define LPASS_CHMAP_IDX_UNKNOWN -1 + +/* + * Channel maps for multi-channel playbacks on MI2S Secondary, up to 4 n_ch + */ +static struct snd_pcm_chmap_elem lpass_chmaps[] = { + { .channels = 2, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { } +}; static int lpass_cpu_init_i2sctl_bitfields(struct device *dev, struct lpaif_i2sctl *i2sctl, struct regmap *map) @@ -86,6 +99,9 @@ static int lpass_cpu_daiops_startup(struct snd_pcm_substream *substream, clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); return ret; } + if (drvdata->chmap_info && drvdata->chmap_info->max_channels == 4) + drvdata->chmap_info->chmap = lpass_chmaps; + return 0; } @@ -96,6 +112,7 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); + drvdata->chmap_idx = LPASS_CHMAP_IDX_UNKNOWN; } static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, @@ -224,9 +241,14 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, ret); return ret; } - if (channels >= 2) + if (channels >= 2) { ret = regmap_fields_write(i2sctl->spkmono, id, LPAIF_I2SCTL_SPKMONO_STEREO); + if (channels == 4) + drvdata->chmap_idx = 1; + else + drvdata->chmap_idx = 0; + } else ret = regmap_fields_write(i2sctl->spkmono, id, LPAIF_I2SCTL_SPKMONO_MONO); @@ -324,6 +346,84 @@ const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); +static int lpass_cpu_chmap_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + unsigned const char *map; + unsigned int i; + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + struct lpass_data *drvdata = info->private_data; + + if (info->max_channels == 2) + map = &lpass_chmaps[0].map[0]; + else if (info->max_channels == 4) + map = &lpass_chmaps[1].map[0]; + + for (i = 0; i < info->max_channels; i++) { + if (drvdata->chmap_idx == LPASS_CHMAP_IDX_UNKNOWN) + ucontrol->value.integer.value[i] = 0; + else + ucontrol->value.integer.value[i] = map[i]; + } + + return 0; +} + +static int lpass_cpu_chmap_ctl_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + unsigned int i; + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + struct lpass_data *drvdata = info->private_data; + unsigned char *map; + + if (info->max_channels == 2) + map = &lpass_chmaps[0].map[0]; + else if (info->max_channels == 4) + map = &lpass_chmaps[1].map[0]; + + for (i = 0; i < info->max_channels; i++) { + if (drvdata->chmap_idx == LPASS_CHMAP_IDX_UNKNOWN) + map[i] = 0; + else + map[i] = ucontrol->value.integer.value[i]; + } + + return 0; +} + +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); + + ret = snd_pcm_add_chmap_ctls(rtd->pcm, SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_alt_chmaps, drv->playback.channels_max, 0, + &drvdata->chmap_info); + if (ret < 0) + return ret; + + + /* + * override handlers + */ + drvdata->chmap_info->private_data = drvdata; + drvdata->chmap_info->kctl->get = lpass_cpu_chmap_ctl_get; + drvdata->chmap_info->kctl->put = lpass_cpu_chmap_ctl_set; + + /* + * default chmap supported is stereo + */ + drvdata->chmap_info->chmap = lpass_chmaps; + drvdata->chmap_idx = LPASS_CHMAP_IDX_UNKNOWN; + + 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 +956,10 @@ 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..fb82591d550c 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -76,6 +76,9 @@ struct lpass_data { /* regmap backed by the low-power audio interface (LPAIF) registers */ struct regmap *lpaif_map; struct regmap *hdmiif_map; + struct snd_pcm_chmap *chmap_info; + + int chmap_idx; /* interrupts from the low-power audio interface (LPAIF) */ int lpaif_irq; @@ -259,5 +262,7 @@ 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.

2 2

[PATCH 0/9] ASoC: tidyup snd_soc_of_parse_daifmt()
by Kuninori Morimoto 09 Jun '21

09 Jun '21

Hi Mark 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. In such case, user need to re-parse it, like below. daifmt = snd_soc_of_parse_daifmt(..., &bitclkmaster, &framemaster); daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK; if (codec == bitclkmaster) daifmt |= (codec == framemaster) ? SND_SOC_DAIFMT_CBM_CFM : SND_SOC_DAIFMT_CBM_CFS; else daifmt |= (codec == framemaster) ? SND_SOC_DAIFMT_CBS_CFM : SND_SOC_DAIFMT_CBS_CFS; This patch-set adds new functions, and handle these more simply. Unfortunately, there are too many use-case, do it by 1 function was implessible. style A) bit_frame = snd_soc_daifmt_parse_clock_provider(); daifmt = snd_soc_daifmt_parse_format(...) | /* format part */ snd_soc_daifmt_clock_provider_pickup(bit_frame); /* clock part */ style B) snd_soc_daifmt_parse_clock_provider(..., &bit, &frame); daifmt = snd_soc_daifmt_parse_format(...) | /* format part */ snd_soc_daifmt_clock_provider_pickup( /* clock part */ ((codec == bit) << 4) + (codec == frame)); Kuninori Morimoto (9): ASoC: soc-core: don't use discriminatory terms on snd_soc_runtime_get_dai_fmt() ASoC: soc-core: add snd_soc_daifmt_clock_provider_pickup() 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 | 13 +++- sound/soc/atmel/mikroe-proto.c | 18 ++--- sound/soc/fsl/fsl-asoc-card.c | 16 +--- sound/soc/generic/simple-card-utils.c | 19 ++--- sound/soc/meson/meson-card-utils.c | 15 ++-- sound/soc/soc-core.c | 103 ++++++++++++++++---------- 6 files changed, 99 insertions(+), 85 deletions(-) -- 2.25.1

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[bug report] media: sound/usb: Use Media Controller API to share media resources
by Dan Carpenter 08 Jun '21

08 Jun '21

Hello Shuah Khan, The patch 66354f18fe5f: "media: sound/usb: Use Media Controller API to share media resources" from Apr 1, 2019, leads to the following static checker warning: sound/usb/media.c:287 snd_media_device_create() warn: 'mdev' can also be NULL sound/usb/media.c 270 271 mdev = media_device_usb_allocate(usbdev, KBUILD_MODNAME, THIS_MODULE); ^^^^ If CONFIG_MEDIA_CONTROLLER is disabled then "mdev" is NULL. 272 if (IS_ERR(mdev)) 273 return -ENOMEM; 274 275 /* save media device - avoid lookups */ 276 chip->media_dev = mdev; 277 278 snd_mixer_init: 279 /* Create media entities for mixer and control dev */ 280 ret = snd_media_mixer_init(chip); 281 /* media_device might be registered, print error and continue */ 282 if (ret) 283 dev_err(&usbdev->dev, 284 "Couldn't create media mixer entities. Error: %d\n", 285 ret); 286 287 if (!media_devnode_is_registered(mdev->devnode)) { ^^^^^^^^^^^^^ dereferenced without checking here 288 /* dont'register if snd_media_mixer_init() failed */ 289 if (ret) 290 goto create_fail; 291 292 /* register media_device */ 293 ret = media_device_register(mdev); 294 create_fail: 295 if (ret) { 296 snd_media_mixer_delete(chip); 297 media_device_delete(mdev, KBUILD_MODNAME, THIS_MODULE); 298 /* clear saved media_dev */ 299 chip->media_dev = NULL; 300 dev_err(&usbdev->dev, 301 "Couldn't register media device. Error: %d\n", 302 ret); 303 return ret; 304 } 305 } 306 307 return ret; 308 } regards, dan carpenter

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alsamixer: use background color instead of COLOR_BLACK
by GitHub pull_request - opened 08 Jun '21

08 Jun '21

alsa-project/alsa-utils pull request #77 was opened from r-burns: Hi! I have a cosmetic "improvement" I'd like to share. Scare quotes because this is possibly opinionated, but I'd still like to know what you think. I think on typical terminals, COLOR_BLACK is the same or quite similar to the background color, so this does not make much of a difference. But on "solarized" color schemes and similar, the COLOR_BLACK may be quite different from the background color, so in my opinion it looks odd to have the alsamixer panel stand out this way. For example, I'm using the "Nord" color scheme on a pitch black #000000 background, and here's how the mixer looks with/without this patch (blank terminal, htop, and ncmpcpp also provided for comparison): ![image](https://user-images.githubusercontent.com/52847440/107131140-dbf46500-6888-11eb-938a-9a3258bf7184.png) As you can see, the current alsamixer in the center looks a bit strange and sticks out against my chosen background color and the other TUIs I'm using. Htop and ncmpcpp agree on using the builtin `-1` background color for their own backgrounds, but alsamixer uses COLOR_BLACK which stands out from the actual background. The alsamixer on the bottom right is using this patch, and blends in seamlessly. Again, this is just my opinion! Feel free to close if this is wrong or superfluous. Nonetheless, I thought I'd share this patch, in case anyone else feels the same way. Request URL : https://github.com/alsa-project/alsa-utils/pull/77 Patch URL : https://github.com/alsa-project/alsa-utils/pull/77.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH 00/15] ASoC: fixes for SoundWire codec drivers
by Pierre-Louis Bossart 08 Jun '21

08 Jun '21

The SOF CI exposed a set of issues with suspend/resume, error handling, register access and mixer values. These fixes were indentified with a tag so that they can be applied by linux-stable and distributions. Thanks to the Realtek and Maxim teams for their help. Bard Liao (3): ASoC: rt5682-sdw: set regcache_cache_only false before reading RT5682_DEVICE_ID ASoC: rt711-sdca-sdw: add readable for SDW_SDCA_CTL() registers ASoC: rt711-sdca: handle mbq_regmap in rt711_sdca_io_init Jack Yu (1): ASoC: rt715-sdca: fix clock stop prepare timeout issue Oder Chiou (1): ASoC: rt5682: Fix a problem with error handling in the io init function of the soundwire Pierre-Louis Bossart (10): ASoC: max98373-sdw: add missing memory allocation check ASoC: max98373-sdw: use first_hw_init flag on resume ASoC: rt1308-sdw: use first_hw_init flag on resume ASoC: rt1316-sdw: use first_hw_init flag on resume ASoC: rt5682-sdw: use first_hw_init flag on resume ASoC: rt700-sdw: use first_hw_init flag on resume ASoC: rt711-sdca-sdw: use first_hw_init flag on resume ASoC: rt711-sdw: use first_hw_init flag on resume ASoC: rt715-sdca-sdw: use first_hw_init flag on resume ASoC: rt715-sdw: use first_hw_init flag on resume sound/soc/codecs/max98373-sdw.c | 14 ++++++++------ sound/soc/codecs/max98373.h | 2 +- sound/soc/codecs/rt1308-sdw.c | 2 +- sound/soc/codecs/rt1316-sdw.c | 2 +- sound/soc/codecs/rt5682-sdw.c | 19 +++++++++++-------- sound/soc/codecs/rt700-sdw.c | 2 +- sound/soc/codecs/rt711-sdca-sdw.c | 12 +++++++++++- sound/soc/codecs/rt711-sdca.c | 4 ++++ sound/soc/codecs/rt711-sdw.c | 2 +- sound/soc/codecs/rt715-sdca-sdw.c | 3 ++- sound/soc/codecs/rt715-sdca-sdw.h | 1 + sound/soc/codecs/rt715-sdca.c | 9 ++++++--- sound/soc/codecs/rt715-sdca.h | 3 ++- sound/soc/codecs/rt715-sdw.c | 2 +- 14 files changed, 51 insertions(+), 26 deletions(-) -- 2.25.1

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