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[PATCH v2] ASoC: dt-bindings: renesas,rsnd: add post-init-providers property
by Kuninori Morimoto 09 Sep '24

09 Sep '24

At least if rsnd is using DPCM connection with Audio-Graph-Card2, fw_devlink might doesn't have enough information to break the cycle (Same problem might occur with Multi-CPU/Codec or Codec2Codec). In such case, rsnd driver will not be probed. Add post-init-providers support to break the link cycle. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - remove basic description/type - add detail description - update git-log Documentation/devicetree/bindings/sound/renesas,rsnd.yaml | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index 07ec6247d9def..458877ce4aa42 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -112,6 +112,13 @@ properties: description: List of necessary clock names. # details are defined below + post-init-providers: + description: | + At least if rsnd is using DPCM connection with Audio-Graph-Card2, fw_devlink might doesn't + have enough information to break the cycle. rsnd driver will not be probed in such case. + Add post-init-providers property to indicate which link in the cycle to break. + Same problem might occur with Multi-CPU/Codec or Codec2Codec. + # ports is below port: $ref: audio-graph-port.yaml#/definitions/port-base -- 2.43.0

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How to handle complex Codec2Codec
by Kuninori Morimoto 09 Sep '24

09 Sep '24

Hi ALSA ML ALSA SoC is supporting Codec2Codec connection, and Audio Graph Card2 is supporting it. I think its image is... +------+ +--- |Codec1| <-- | +------+ | +------+ +--> |Codec2| --> +------+ This is OK, but how to handle below ? +----------------------+ |CPU | +------+ | +--[rate convert]<--| <-- |Codec1| <-- | | | +------+ | | | +------+ | +------------------>| --> |Codec2| --> +----------------------+ +------+ I think there are 2 issues. (1) How to handle [CPU] part on Codec2Codec ? (2) How to handle [rate convert] on Codec2Codec ? I guess (1) can be handled if [CPU] was worked as platform, but do you have any other idea/solution for it ? I think (2) [rate convert] needs DPCM connection, but how to use it with Codec2Codec ? Or, can we handle Codec2Codec via DPCM somehow ?? FE BE +------+ +------+ --> |Codec1| -> |Codec2| -> +------+ +------+ If we can do this, both (1) and (2) can be solved, like below ? In such case, [Codec1] need to work as [capture] when playback. But I'm not sure how to implement it... FE BE +------+ +-----------------+ +-----+ +------+ --> |Codec1|-|CPU[rate convert]| -> |dummy|-|Codec2| -> +------+ +-----------------+ +-----+ +------+ Thank you for your help !! Best regards --- Kuninori Morimoto

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Re: [PATCH] ALSA: Add logitech Audio profile quirk
by Joshua Pius 06 Sep '24

06 Sep '24

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[PATCH] ASoC: codecs: Use devm_clk_get_enabled() helpers
by ying zuxin 06 Sep '24

06 Sep '24

From: ying zuxin <11154159(a)vivo.com> The devm_clk_get_enabled() helpers: - call devm_clk_get() - call clk_prepare_enable() and register what is needed in order to call clk_disable_unprepare() when needed, as a managed resource. This simplifies the code and avoids the calls to clk_disable_unprepare(). Signed-off-by: ying zuxin <11154159(a)vivo.com> --- sound/soc/codecs/peb2466.c | 14 +------------- 1 file changed, 1 insertion(+), 13 deletions(-) diff --git a/sound/soc/codecs/peb2466.c b/sound/soc/codecs/peb2466.c index 76ee7e3f4d9b..b67cfad4fc32 100644 --- a/sound/soc/codecs/peb2466.c +++ b/sound/soc/codecs/peb2466.c @@ -1975,12 +1975,9 @@ static int peb2466_spi_probe(struct spi_device *spi) if (IS_ERR(peb2466->reset_gpio)) return PTR_ERR(peb2466->reset_gpio); - peb2466->mclk = devm_clk_get(&peb2466->spi->dev, "mclk"); + peb2466->mclk = devm_clk_get_enabled(&peb2466->spi->dev, "mclk"); if (IS_ERR(peb2466->mclk)) return PTR_ERR(peb2466->mclk); - ret = clk_prepare_enable(peb2466->mclk); - if (ret) - return ret; if (peb2466->reset_gpio) { gpiod_set_value_cansleep(peb2466->reset_gpio, 1); @@ -2031,17 +2028,9 @@ static int peb2466_spi_probe(struct spi_device *spi) return 0; failed: - clk_disable_unprepare(peb2466->mclk); return ret; } -static void peb2466_spi_remove(struct spi_device *spi) -{ - struct peb2466 *peb2466 = spi_get_drvdata(spi); - - clk_disable_unprepare(peb2466->mclk); -} - static const struct of_device_id peb2466_of_match[] = { { .compatible = "infineon,peb2466", }, { } @@ -2061,7 +2050,6 @@ static struct spi_driver peb2466_spi_driver = { }, .id_table = peb2466_id_table, .probe = peb2466_spi_probe, - .remove = peb2466_spi_remove, }; module_spi_driver(peb2466_spi_driver); -- 2.39.0

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[PATCH] ASoC: makes rtd->initialized bit field
by Kuninori Morimoto 06 Sep '24

06 Sep '24

rtd->initialized is used to know whether soc_init_pcm_runtime() was correctly fined, and used to call snd_soc_link_exit(). We don't need to have it as bool, let's make it bit-field same as other flags. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> --- include/sound/soc.h | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 81ef380b2e06..e6e359c1a2ac 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1207,8 +1207,7 @@ struct snd_soc_pcm_runtime { /* bit field */ unsigned int pop_wait:1; unsigned int fe_compr:1; /* for Dynamic PCM */ - - bool initialized; + unsigned int initialized:1; /* CPU/Codec/Platform */ int num_components; -- 2.43.0

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[PATCH 1/2] pcm: dmix: Fix resume with multiple instances
by Takashi Iwai 06 Sep '24

06 Sep '24

The fix for PCM dmix suspend/resume checks spcm->info bit of SND_PCM_INFO_RESUME for applying a workaround for drivers with the full resume support. This assumed that scpm->info is exposed from the underlying slave PCM device. The above is true for the first opened instance, but for the second opened instance, it's a copy from the saved data in shmem. And, we dropped SND_PCM_INFO_RESUME bit there to assure not to expose the full resume capability to applications. This resulted in the inconsistencies, and when the second instance is resumed at first, it misses the snd_pcm_resume() call, hence the driver doesn't react properly any longer. For addressing it, we keep SND_PCM_INFO_RESUME bit in shmptr->s.info bits as is, while dropping the bit exposed to apps in snd_pcm_direct_hw_refine() and *_hw_params() callbacks. Fixes: 6d1d620eadf3 ("pcm: dmix: resume workaround for buggy driver") Reported-and-tested-by: Chancel Liu <chancel.liu(a)nxp.com> Closes: https://lore.kernel.org/DB9PR04MB94988752ED7C43B399E0BC00E3942@DB9PR04MB949… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm_direct.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index e53e59238119..17e677f60b19 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -1018,6 +1018,7 @@ int snd_pcm_direct_hw_refine(snd_pcm_t *pcm, snd_pcm_hw_params_t *params) } dshare->timer_ticks = hw_param_interval(params, SND_PCM_HW_PARAM_PERIOD_SIZE)->max / dshare->slave_period_size; params->info = dshare->shmptr->s.info; + params->info &= ~SND_PCM_INFO_RESUME; #ifdef REFINE_DEBUG snd_output_puts(log, "DMIX REFINE (end):\n"); snd_pcm_hw_params_dump(params, log); @@ -1031,6 +1032,7 @@ int snd_pcm_direct_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t * params) snd_pcm_direct_t *dmix = pcm->private_data; params->info = dmix->shmptr->s.info; + params->info &= ~SND_PCM_INFO_RESUME; params->rate_num = dmix->shmptr->s.rate; params->rate_den = 1; params->fifo_size = 0; @@ -1183,8 +1185,6 @@ static void save_slave_setting(snd_pcm_direct_t *dmix, snd_pcm_t *spcm) COPY_SLAVE(buffer_time); COPY_SLAVE(sample_bits); COPY_SLAVE(frame_bits); - - dmix->shmptr->s.info &= ~SND_PCM_INFO_RESUME; } #undef COPY_SLAVE -- 2.43.0

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Suspend/resume Issue on pcm_dmix.c in alsa-lib
by Chancel Liu 06 Sep '24

06 Sep '24

Hi Takashi Iwai, Jaroslav Kysela We found an issue on dmix in alsa-lib when do suspend and resume. It can be easily reproduced by following steps: 1. Run two dmix clients in parallel. (Only one client doesn't has such issue) ~# aplay xxx1.wav & ~# aplay xxx2.wav & Here I attach the asound.conf we're using. ~# cat /etc/asound.conf defaults.pcm.rate_converter "linear" pcm.dmix_44100{ type dmix ipc_key 5678293 ipc_key_add_uid yes slave{ pcm "hw:0,0" period_time 40000 format S16_LE rate 44100 } } pcm.asymed{ type asym playback.pcm "dmix_44100" capture.pcm "dsnoop_44100" } pcm.!default{ type plug route_policy "average" slave.pcm "asymed" } 2. Let linux enter into suspend and then resume(Repeat this step if not reproduced) 3. After resume, aplay will get stuck in snd_pcm_wait(). The GDB shows: (gdb) bt #0 0x0000fffff7da9264 in __GI___poll (fds=fds@entry=0xfffffffff480, nfds=nfds@entry=1, timeout=timeout@entry=240) at /usr/src/debug/glibc/2.39+git/sysdeps/unix/sysv/linux/poll.c:41 #1 0x0000fffff7edf468 in poll (__timeout=240, __nfds=1, __fds=0xfffffffff480) #2 snd1_pcm_wait_nocheck (pcm=pcm@entry=0xaaaaaaad2cb0, timeout=240, timeout@entry=-10001) at pcm.c:2993 #3 0x0000fffff7ee54a0 in snd1_pcm_write_areas (pcm=pcm@entry=0xaaaaaaad2cb0, areas=areas@entry=0xfffffffff560, offset=<optimized out>, offset@entry=0, size=<optimized out>, size@entry=1768, func=func@entry=0xfffff7ef5190 <snd_pcm_plugin_write_areas>) at pcm.c:7699 #4 0x0000fffff7ef5020 in snd_pcm_plugin_writei (pcm=0xaaaaaaad2cb0, buffer=<optimized out>, size=1768) at pcm_plugin.c:354 It seems that sometimes after suspend and resume there's no available space for data written into buffer. Then aplay keeps stuck in snd_pcm_wait(). I checked the hw_ptr of dmix and found that hw_ptr is always 0 after resume. I don't have a solution now so I turn to you for help. The version of alsa-lib is v1.2.11. Could you please help check it? Regards, Chancel Liu

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[GIT PULL] ASoC fixes for v6.11-rc6
by Mark Brown 06 Sep '24

06 Sep '24

The following changes since commit bb4485562f5907708f1c218b5d70dce04165d1e1: ASoC: cs-amp-lib: Ignore empty UEFI calibration entries (2024-08-22 17:52:26 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.11-rc6 for you to fetch changes up to 77212f300bfd6fb3edaabd1daf863cabb521854a: ASoC: codecs: lpass-va-macro: set the default codec version for sm8250 (2024-08-30 13:54:24 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.11 A larger set of fixes than I'd like at this point, but mainly due to people working on fixing module autoloading by adding missing exports of ID tables rather than anything particularly concerning. There are some other runtime fixes and quirks, and a tweak to the ABI definition for SOF which ensures that a struct layout doesn't vary depending on the architecture of the host. ---------------------------------------------------------------- Chen-Yu Tsai (1): ASoc: SOF: topology: Clear SOF link platform name upon unload Hans de Goede (2): ASoC: Intel: Boards: Fix NULL pointer deref in BYT/CHT boards harder ASoC: Intel: soc-acpi-cht: Make Lenovo Yoga Tab 3 X90F DMI match less strict Laurentiu Mihalcea (1): ASoC: SOF: ipc: replace "enum sof_comp_type" field with "uint32_t" Liao Chen (4): ASoC: intel: fix module autoloading ASoC: google: fix module autoloading ASoC: tda7419: fix module autoloading ASoC: fix module autoloading Mark Brown (1): ASoC: fix module autoloading Markuss Broks (1): ASoC: amd: yc: Add a quirk for MSI Bravo 17 (D7VEK) Matteo Martelli (1): ASoC: sunxi: sun4i-i2s: fix LRCLK polarity in i2s mode Mohan Kumar (1): ASoC: tegra: Fix CBB error during probe() Srinivas Kandagatla (1): ASoC: codecs: lpass-va-macro: set the default codec version for sm8250 Zhang Yi (1): ASoC: mediatek: mt8188-mt6359: Modify key robelin (1): ASoC: dapm: Fix UAF for snd_soc_pcm_runtime object include/sound/sof/topology.h | 2 +- include/uapi/sound/sof/abi.h | 2 +- sound/soc/amd/yc/acp6x-mach.c | 7 ++ sound/soc/codecs/chv3-codec.c | 1 + sound/soc/codecs/lpass-va-macro.c | 11 +- sound/soc/codecs/tda7419.c | 1 + sound/soc/google/chv3-i2s.c | 1 + sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/bytcht_cx2072x.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 2 +- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/bytcr_wm5102.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 1 - sound/soc/intel/keembay/kmb_platform.c | 1 + sound/soc/mediatek/mt8188/mt8188-mt6359.c | 17 ++- sound/soc/soc-dapm.c | 1 + sound/soc/sof/topology.c | 2 + sound/soc/sunxi/sun4i-i2s.c | 143 +++++++++++----------- sound/soc/tegra/tegra210_ahub.c | 10 +- 23 files changed, 127 insertions(+), 91 deletions(-)

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[PATCH] ASoC: Intel: skl_hda_dsp_generic: convert comma to semicolon
by Chen Ni 05 Sep '24

05 Sep '24

Replace comma between expressions with semicolons. Using a ',' in place of a ';' can have unintended side effects. Although that is not the case here, it is seems best to use ';' unless ',' is intended. Found by inspection. No functional change intended. Compile tested only. Signed-off-by: Chen Ni <nichen(a)iscas.ac.cn> --- sound/soc/intel/boards/skl_hda_dsp_generic.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 225867bb3310..8e104874d58c 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -217,14 +217,14 @@ static int skl_hda_audio_probe(struct platform_device *pdev) return -ENOMEM; card = &ctx->card; - card->name = "hda-dsp", - card->owner = THIS_MODULE, - card->dai_link = skl_hda_be_dai_links, - card->dapm_widgets = skl_hda_widgets, - card->dapm_routes = skl_hda_map, - card->add_dai_link = skl_hda_add_dai_link, - card->fully_routed = true, - card->late_probe = skl_hda_card_late_probe, + card->name = "hda-dsp"; + card->owner = THIS_MODULE; + card->dai_link = skl_hda_be_dai_links; + card->dapm_widgets = skl_hda_widgets; + card->dapm_routes = skl_hda_map; + card->add_dai_link = skl_hda_add_dai_link; + card->fully_routed = true; + card->late_probe = skl_hda_card_late_probe; snd_soc_card_set_drvdata(card, ctx); -- 2.25.1

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[PATCH 1/2 4.19.y] ALSA: usb-audio: Sanity checks for each pipe and EP types
by Takashi Iwai 05 Sep '24

05 Sep '24

[ Upstream commit 801ebf1043ae7b182588554cc9b9ad3c14bc2ab5 ] The recent USB core code performs sanity checks for the given pipe and EP types, and it can be hit by manipulated USB descriptors by syzbot. For making syzbot happier, this patch introduces a local helper for a sanity check in the driver side and calls it at each place before the message handling, so that we can avoid the WARNING splats. Reported-by: syzbot+d952e5e28f5fb7718d23(a)syzkaller.appspotmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- sound/usb/helper.c | 17 +++++++++++++++++ sound/usb/helper.h | 1 + sound/usb/quirks.c | 14 +++++++++++--- 3 files changed, 29 insertions(+), 3 deletions(-) diff --git a/sound/usb/helper.c b/sound/usb/helper.c index 7712e2b84183..b1cc9499c57e 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -76,6 +76,20 @@ void *snd_usb_find_csint_desc(void *buffer, int buflen, void *after, u8 dsubtype return NULL; } +/* check the validity of pipe and EP types */ +int snd_usb_pipe_sanity_check(struct usb_device *dev, unsigned int pipe) +{ + static const int pipetypes[4] = { + PIPE_CONTROL, PIPE_ISOCHRONOUS, PIPE_BULK, PIPE_INTERRUPT + }; + struct usb_host_endpoint *ep; + + ep = usb_pipe_endpoint(dev, pipe); + if (usb_pipetype(pipe) != pipetypes[usb_endpoint_type(&ep->desc)]) + return -EINVAL; + return 0; +} + /* * Wrapper for usb_control_msg(). * Allocates a temp buffer to prevent dmaing from/to the stack. @@ -88,6 +102,9 @@ int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, void *buf = NULL; int timeout; + if (snd_usb_pipe_sanity_check(dev, pipe)) + return -EINVAL; + if (size > 0) { buf = kmemdup(data, size, GFP_KERNEL); if (!buf) diff --git a/sound/usb/helper.h b/sound/usb/helper.h index f5b4c6647e4d..5e8a18b4e7b9 100644 --- a/sound/usb/helper.h +++ b/sound/usb/helper.h @@ -7,6 +7,7 @@ unsigned int snd_usb_combine_bytes(unsigned char *bytes, int size); void *snd_usb_find_desc(void *descstart, int desclen, void *after, u8 dtype); void *snd_usb_find_csint_desc(void *descstart, int desclen, void *after, u8 dsubtype); +int snd_usb_pipe_sanity_check(struct usb_device *dev, unsigned int pipe); int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, __u8 requesttype, __u16 value, __u16 index, void *data, __u16 size); diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 43cbaaff163f..087eef5e249d 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -743,11 +743,13 @@ static int snd_usb_novation_boot_quirk(struct usb_device *dev) static int snd_usb_accessmusic_boot_quirk(struct usb_device *dev) { int err, actual_length; - /* "midi send" enable */ static const u8 seq[] = { 0x4e, 0x73, 0x52, 0x01 }; + void *buf; - void *buf = kmemdup(seq, ARRAY_SIZE(seq), GFP_KERNEL); + if (snd_usb_pipe_sanity_check(dev, usb_sndintpipe(dev, 0x05))) + return -EINVAL; + buf = kmemdup(seq, ARRAY_SIZE(seq), GFP_KERNEL); if (!buf) return -ENOMEM; err = usb_interrupt_msg(dev, usb_sndintpipe(dev, 0x05), buf, @@ -772,7 +774,11 @@ static int snd_usb_accessmusic_boot_quirk(struct usb_device *dev) static int snd_usb_nativeinstruments_boot_quirk(struct usb_device *dev) { - int ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), + int ret; + + if (snd_usb_pipe_sanity_check(dev, usb_sndctrlpipe(dev, 0))) + return -EINVAL; + ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), 0xaf, USB_TYPE_VENDOR | USB_RECIP_DEVICE, 1, 0, NULL, 0, 1000); @@ -879,6 +885,8 @@ static int snd_usb_axefx3_boot_quirk(struct usb_device *dev) dev_dbg(&dev->dev, "Waiting for Axe-Fx III to boot up...\n"); + if (snd_usb_pipe_sanity_check(dev, usb_sndctrlpipe(dev, 0))) + return -EINVAL; /* If the Axe-Fx III has not fully booted, it will timeout when trying * to enable the audio streaming interface. A more generous timeout is * used here to detect when the Axe-Fx III has finished booting as the -- 2.45.2

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