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

Re: [alsa-devel] INFO: rcu detected stall in __do_softirq
by Srinivas Kandagatla 17 Aug '19

17 Aug '19

On 16/08/2019 01:10, syzbot wrote: > syzbot has bisected this bug to: > > commit 2aeac95d1a4cc85aae57ab842d5c3340df0f817f > Author: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> > Date: Tue Jun 11 10:40:41 2019 +0000 > > soundwire: add module_sdw_driver helper macro Not sure how adding a macro with no users triggers this rcu stall. BTW this was in mainline since rc1. --srini > > bisection log: https://syzkaller.appspot.com/x/bisect.txt?x=114b45ee600000 > start commit: 882e8691 Add linux-next specific files for 20190801 > git tree: linux-next > final crash: https://syzkaller.appspot.com/x/report.txt?x=134b45ee600000 > console output: https://syzkaller.appspot.com/x/log.txt?x=154b45ee600000 > kernel config: https://syzkaller.appspot.com/x/.config?x=466b331af3f34e94 > dashboard link: > https://syzkaller.appspot.com/bug?extid=6593c6b8c8b66a07cd98 > syz repro: https://syzkaller.appspot.com/x/repro.syz?x=16b216b2600000 > > Reported-by: syzbot+6593c6b8c8b66a07cd98(a)syzkaller.appspotmail.com > Fixes: 2aeac95d1a4c ("soundwire: add module_sdw_driver helper macro") > > For information about bisection process see: > https://goo.gl/tpsmEJ#bisection

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[alsa-devel] ALSA HW PARAMS GET BUFFER TIME ERROR
by 李松鞠 17 Aug '19

17 Aug '19

Hello! Dear Developers, I am from HNU University (Changsha, China), a fresh man in ALSA sound developing. Recently, a alsa bug( snd_pcm_hw_params_get_buffer_time snd_pcm_hw_params_get_period_time ) occured to me which made me confused! Firstly (Step 1) I set hardware params normally(rate: 44100, channel: 2, format: S16_LE, access: raw_interleaved, period size: not set, buffer size: not set), It was OK. Laterly (Step 2) I get hw_params and print it, it was not surprising that they were the same as what I set (buffer_time == period_time * 1024, buffer_size == period_size * 1024). But! In another code ( Add Step 3)I used snd_pcm_format_name((snd_pcm_format_t)val), snd_pcm_format_description( (snd_pcm_format_t)val) to display the format information, It got wrong buffer size(buffer_time == period_time, buffer_size == period_size * 1024 ). the Attachments are code pcm_setting.cpp(With Step 1, 2), pcm_open_seeting.cpp(With Step 1, 2, 3). My alsa-lib version is about 1.1.0 . pcm_setting.cpp running result: get rate: 44100 get channles: 1 period time: 46439 buffer time is 47554467 period size: 2048 frames buffer size default: 2097152 frames periods per buffer = 1024 frames pcm_open_setting running result: PCM handle name = 'default' PCM state = PREPARED access type = RW_INTERLEAVED format(length: 2) = 'S16_LE' (Signed 16 bit Little Endian) channels = 1 rate = 44100 bps period time = 46438 us buffer time = 46438 us period size = 2048 frames buffer size = 2097152 frames periods per buffer = 1024 frames

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Re: [alsa-devel] [PATCH] ASoC: imx-audmux: Add driver suspend and resume to support MEGA Fast
by S.j. Wang 17 Aug '19

17 Aug '19

Hi Mark > > On Fri, Aug 16, 2019 at 01:03:14AM -0400, Shengjiu Wang wrote: > > > + for (i = 0; i < reg_max; i++) > > + regcache[i] = readl(audmux_base + i * 4); > > If only there were some framework which provided a register cache! 😝 Yes, next step I can refine this driver to use the regmap. Best regards Wang shengjiu

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[alsa-devel] [PATCH] ASoC: imx-audmux: Add driver suspend and resume to support MEGA Fast
by Shengjiu Wang 16 Aug '19

16 Aug '19

For i.MX6 SoloX, there is a mode of the SoC to shutdown all power source of modules during system suspend and resume procedure. Thus, AUDMUX needs to save all the values of registers before the system suspend and restore them after the system resume. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/imx-audmux.c | 54 +++++++++++++++++++++++++++++++++++++- 1 file changed, 53 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 7595f24a006e..3ce85a43e08f 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -23,6 +23,8 @@ static struct clk *audmux_clk; static void __iomem *audmux_base; +static u32 *regcache; +static u32 reg_max; #define IMX_AUDMUX_V2_PTCR(x) ((x) * 8) #define IMX_AUDMUX_V2_PDCR(x) ((x) * 8 + 4) @@ -315,8 +317,23 @@ static int imx_audmux_probe(struct platform_device *pdev) if (of_id) pdev->id_entry = of_id->data; audmux_type = pdev->id_entry->driver_data; - if (audmux_type == IMX31_AUDMUX) + + switch (audmux_type) { + case IMX31_AUDMUX: audmux_debugfs_init(); + reg_max = 14; + break; + case IMX21_AUDMUX: + reg_max = 6; + break; + default: + dev_err(&pdev->dev, "unsupported version!\n"); + return -EINVAL; + } + + regcache = devm_kzalloc(&pdev->dev, sizeof(u32) * reg_max, GFP_KERNEL); + if (!regcache) + return -ENOMEM; if (of_id) imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node); @@ -332,12 +349,47 @@ static int imx_audmux_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM_SLEEP +static int imx_audmux_suspend(struct device *dev) +{ + int i; + + clk_prepare_enable(audmux_clk); + + for (i = 0; i < reg_max; i++) + regcache[i] = readl(audmux_base + i * 4); + + clk_disable_unprepare(audmux_clk); + + return 0; +} + +static int imx_audmux_resume(struct device *dev) +{ + int i; + + clk_prepare_enable(audmux_clk); + + for (i = 0; i < reg_max; i++) + writel(regcache[i], audmux_base + i * 4); + + clk_disable_unprepare(audmux_clk); + + return 0; +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops imx_audmux_pm = { + SET_SYSTEM_SLEEP_PM_OPS(imx_audmux_suspend, imx_audmux_resume) +}; + static struct platform_driver imx_audmux_driver = { .probe = imx_audmux_probe, .remove = imx_audmux_remove, .id_table = imx_audmux_ids, .driver = { .name = DRIVER_NAME, + .pm = &imx_audmux_pm, .of_match_table = imx_audmux_dt_ids, } }; -- 2.21.0

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[alsa-devel] [PATCH 0/4] ASoC: SOF: i.MX8 SAI and Intel ALH/SoundWire DAI support
by Pierre-Louis Bossart 16 Aug '19

16 Aug '19

This patchset adds the basic functionality to extract DAI information from the topology and send the information to firmware. The i.MX8 SAI/ESAI definitions are minimal placeholders for now, they are included to keep the alignment with the firmware/topology. For the Intel ALH, there is no specific configuration in topology file, all the programming takes place based on the dai index and the related ALH stream ID. The renaming from SoundWire to ALH (Audio Link Hub) is a better representation of the hardware, the ALH acts as a proxy between the DMA transfers and the Cadence IP. Bard liao (1): ASoC: SOF: rename SOUNDWIRE to ALH Daniel Baluta (1): ASoC: SOF: topology: Add dummy support for i.MX8 DAIs Pierre-Louis Bossart (2): ASoC: SOF: topology: initial support for Intel ALH DAI type ASoC: SOF: pcm: add ALH support include/sound/sof/dai.h | 4 +- include/uapi/sound/sof/tokens.h | 8 ++++ sound/soc/sof/pcm.c | 3 ++ sound/soc/sof/topology.c | 72 ++++++++++++++++++++++++++++++++- 4 files changed, 85 insertions(+), 2 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH v2 0/3] soundwire: debugfs support for 5.4
by Pierre-Louis Bossart 16 Aug '19

16 Aug '19

This patchset enables debugfs support and corrects all the feedback provided on an earlier RFC ('soundwire: updates for 5.4') There is one remaining hard-coded value in intel.c that will need to be fixed in a follow-up patchset not specific to debugfs: we need to remove hard-coded Intel-specific configurations from cadence_master.c (PDI offsets, etc). Changes since v1 (Feedback from GKH) Handle debugfs in a more self-contained way (no dentry as return or parameter) Used CONFIG_DEBUG_FS in structures and code to make it easier to remove if need be. No functional change for register dumps. Changes since RFC (Feedback from GKH, Vinod, Guennadi, Cezary, Sanyog): removed error checks used DEFINE_SHOW_ATTRIBUTE and seq_file fixed copyright dates fixed SPDX license info to use GPL2.0 only fixed Makefile to include debugfs only if CONFIG_DEBUG_FS is selected used static inlines for fallback compilation removed intermediate variables removed hard-coded constants in loops (used registers offsets and hardware capabilities) squashed patch 3 Pierre-Louis Bossart (3): soundwire: add debugfs support soundwire: cadence_master: add debugfs register dump soundwire: intel: add debugfs register dump drivers/soundwire/Makefile | 4 + drivers/soundwire/bus.c | 6 ++ drivers/soundwire/bus.h | 16 +++ drivers/soundwire/bus_type.c | 3 + drivers/soundwire/cadence_master.c | 107 ++++++++++++++++++++ drivers/soundwire/cadence_master.h | 4 + drivers/soundwire/debugfs.c | 151 +++++++++++++++++++++++++++++ drivers/soundwire/intel.c | 121 +++++++++++++++++++++++ drivers/soundwire/slave.c | 1 + include/linux/soundwire/sdw.h | 8 ++ 10 files changed, 421 insertions(+) create mode 100644 drivers/soundwire/debugfs.c -- 2.20.1

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[alsa-devel] [PATCH] ASoC: core: Move pcm_mutex up to card level from snd_soc_pcm_runtime
by Peter Ujfalusi 16 Aug '19

16 Aug '19

The pcm_mutex is used to prevent concurrent execution of snd_pcm_ops callbacks. This works fine most of the cases but it can not handle setups when the same DAI is used by different rtd, for example: pcm3168a have two DAIs: one for Playback and one for Capture. If the codec is connected to a single CPU DAI we need to have two dai_link to support both playback and capture. In this case the snd_pcm_ops callbacks can be executed in parallel causing unexpected races in DAI drivers. By moving the pcm_mutex up to card level this can be solved while - hopefully - not breaking other setups. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, I have tested the patch on several boards. it fixes the issue with single card with multiple dai_links where the CPU side (mcasp) is the same. However I can not test with anything which use DPCM. It would be great if the patch would gather few tested-by from folks having access to more complicated setups. The backround of this patch: https://mailman.alsa-project.org/pipermail/alsa-devel/2019-August/153864.ht… Regards, Peter include/sound/soc.h | 6 +++-- sound/soc/soc-compress.c | 48 ++++++++++++++++++++-------------------- sound/soc/soc-core.c | 2 +- sound/soc/soc-pcm.c | 36 +++++++++++++++--------------- 4 files changed, 47 insertions(+), 45 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 6ac6481b4882..ec2dbe0ac2aa 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -990,6 +990,10 @@ struct snd_soc_card { struct mutex mutex; struct mutex dapm_mutex; + /* Mutex for PCM operations */ + struct mutex pcm_mutex; + enum snd_soc_pcm_subclass pcm_subclass; + spinlock_t dpcm_lock; bool instantiated; @@ -1107,8 +1111,6 @@ struct snd_soc_pcm_runtime { struct device *dev; struct snd_soc_card *card; struct snd_soc_dai_link *dai_link; - struct mutex pcm_mutex; - enum snd_soc_pcm_subclass pcm_subclass; struct snd_pcm_ops ops; unsigned int params_select; /* currently selected param for dai link */ diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 289211069a1e..9e54d8ae6d2c 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -80,7 +80,7 @@ static int soc_compr_open(struct snd_compr_stream *cstream) struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); if (cpu_dai->driver->cops && cpu_dai->driver->cops->startup) { ret = cpu_dai->driver->cops->startup(cstream, cpu_dai); @@ -108,7 +108,7 @@ static int soc_compr_open(struct snd_compr_stream *cstream) snd_soc_runtime_activate(rtd, cstream->direction); - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return 0; @@ -118,7 +118,7 @@ static int soc_compr_open(struct snd_compr_stream *cstream) if (cpu_dai->driver->cops && cpu_dai->driver->cops->shutdown) cpu_dai->driver->cops->shutdown(cstream, cpu_dai); out: - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -224,7 +224,7 @@ static void close_delayed_work(struct work_struct *work) container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); struct snd_soc_dai *codec_dai = rtd->codec_dai; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); dev_dbg(rtd->dev, "Compress ASoC: pop wq checking: %s status: %s waiting: %s\n", @@ -239,7 +239,7 @@ static void close_delayed_work(struct work_struct *work) SND_SOC_DAPM_STREAM_STOP); } - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); } static int soc_compr_free(struct snd_compr_stream *cstream) @@ -249,7 +249,7 @@ static int soc_compr_free(struct snd_compr_stream *cstream) struct snd_soc_dai *codec_dai = rtd->codec_dai; int stream; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); if (cstream->direction == SND_COMPRESS_PLAYBACK) stream = SNDRV_PCM_STREAM_PLAYBACK; @@ -292,7 +292,7 @@ static int soc_compr_free(struct snd_compr_stream *cstream) SND_SOC_DAPM_STREAM_STOP); } - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return 0; } @@ -375,7 +375,7 @@ static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd) struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); ret = soc_compr_components_trigger(cstream, cmd); if (ret < 0) @@ -394,7 +394,7 @@ static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd) } out: - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -480,7 +480,7 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); /* * First we call set_params for the CPU DAI, then the component @@ -514,14 +514,14 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, /* cancel any delayed stream shutdown that is pending */ rtd->pop_wait = 0; - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); cancel_delayed_work_sync(&rtd->delayed_work); return 0; err: - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -593,7 +593,7 @@ static int soc_compr_get_params(struct snd_compr_stream *cstream, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret = 0; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); if (cpu_dai->driver->cops && cpu_dai->driver->cops->get_params) { ret = cpu_dai->driver->cops->get_params(cstream, params, cpu_dai); @@ -613,7 +613,7 @@ static int soc_compr_get_params(struct snd_compr_stream *cstream, } err: - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -625,7 +625,7 @@ static int soc_compr_get_caps(struct snd_compr_stream *cstream, struct snd_soc_rtdcom_list *rtdcom; int ret = 0; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -638,7 +638,7 @@ static int soc_compr_get_caps(struct snd_compr_stream *cstream, break; } - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -650,7 +650,7 @@ static int soc_compr_get_codec_caps(struct snd_compr_stream *cstream, struct snd_soc_rtdcom_list *rtdcom; int ret = 0; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -664,7 +664,7 @@ static int soc_compr_get_codec_caps(struct snd_compr_stream *cstream, break; } - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -676,7 +676,7 @@ static int soc_compr_ack(struct snd_compr_stream *cstream, size_t bytes) struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret = 0; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); if (cpu_dai->driver->cops && cpu_dai->driver->cops->ack) { ret = cpu_dai->driver->cops->ack(cstream, bytes, cpu_dai); @@ -697,7 +697,7 @@ static int soc_compr_ack(struct snd_compr_stream *cstream, size_t bytes) } err: - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -710,7 +710,7 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream, int ret = 0; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); if (cpu_dai->driver->cops && cpu_dai->driver->cops->pointer) cpu_dai->driver->cops->pointer(cstream, tstamp, cpu_dai); @@ -726,7 +726,7 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream, break; } - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -738,7 +738,7 @@ static int soc_compr_copy(struct snd_compr_stream *cstream, struct snd_soc_rtdcom_list *rtdcom; int ret = 0; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -751,7 +751,7 @@ static int soc_compr_copy(struct snd_compr_stream *cstream, break; } - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bb1e9e2c4ff4..7f2064889a23 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1344,7 +1344,6 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, rtd->dev->groups = soc_dev_attr_groups; dev_set_name(rtd->dev, "%s", name); dev_set_drvdata(rtd->dev, rtd); - mutex_init(&rtd->pcm_mutex); INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients); INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients); INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].fe_clients); @@ -2395,6 +2394,7 @@ int snd_soc_register_card(struct snd_soc_card *card) card->instantiated = 0; mutex_init(&card->mutex); mutex_init(&card->dapm_mutex); + mutex_init(&card->pcm_mutex); spin_lock_init(&card->dpcm_lock); return snd_soc_bind_card(card); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 77c986fe08d0..93bb4a347779 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -36,7 +36,7 @@ * Increments the active count for all the DAIs and components attached to a PCM * runtime. Should typically be called when a stream is opened. * - * Must be called with the rtd->pcm_mutex being held + * Must be called with the rtd->card->pcm_mutex being held */ void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream) { @@ -44,7 +44,7 @@ void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream) struct snd_soc_dai *codec_dai; int i; - lockdep_assert_held(&rtd->pcm_mutex); + lockdep_assert_held(&rtd->card->pcm_mutex); if (stream == SNDRV_PCM_STREAM_PLAYBACK) { cpu_dai->playback_active++; @@ -72,7 +72,7 @@ void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream) * Decrements the active count for all the DAIs and components attached to a PCM * runtime. Should typically be called when a stream is closed. * - * Must be called with the rtd->pcm_mutex being held + * Must be called with the rtd->card->pcm_mutex being held */ void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream) { @@ -80,7 +80,7 @@ void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream) struct snd_soc_dai *codec_dai; int i; - lockdep_assert_held(&rtd->pcm_mutex); + lockdep_assert_held(&rtd->card->pcm_mutex); if (stream == SNDRV_PCM_STREAM_PLAYBACK) { cpu_dai->playback_active--; @@ -506,7 +506,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) pm_runtime_get_sync(component->dev); } - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); /* startup the audio subsystem */ ret = snd_soc_dai_startup(cpu_dai, substream); @@ -604,7 +604,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) snd_soc_runtime_activate(rtd, substream->stream); - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return 0; config_err: @@ -623,7 +623,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) snd_soc_dai_shutdown(cpu_dai, substream); out: - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -653,7 +653,7 @@ static void close_delayed_work(struct work_struct *work) 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->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); dev_dbg(rtd->dev, "ASoC: pop wq checking: %s status: %s waiting: %s\n", codec_dai->driver->playback.stream_name, @@ -667,7 +667,7 @@ static void close_delayed_work(struct work_struct *work) SND_SOC_DAPM_STREAM_STOP); } - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); } static void codec2codec_close_delayed_work(struct work_struct *work) @@ -694,7 +694,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai; int i; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); snd_soc_runtime_deactivate(rtd, substream->stream); @@ -738,7 +738,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) SND_SOC_DAPM_STREAM_STOP); } - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -771,7 +771,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai; int i, ret = 0; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); if (rtd->dai_link->ops->prepare) { ret = rtd->dai_link->ops->prepare(substream); @@ -826,7 +826,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) snd_soc_dai_digital_mute(cpu_dai, 0, substream->stream); out: - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -876,7 +876,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai; int i, ret = 0; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); if (rtd->dai_link->ops->hw_params) { ret = rtd->dai_link->ops->hw_params(substream, params); if (ret < 0) { @@ -962,7 +962,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (ret) goto component_err; out: - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; component_err: @@ -986,7 +986,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (rtd->dai_link->ops->hw_free) rtd->dai_link->ops->hw_free(substream); - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -1001,7 +1001,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; int i; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); /* clear the corresponding DAIs parameters when going to be inactive */ if (cpu_dai->active == 1) { @@ -1043,7 +1043,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) snd_soc_dai_hw_free(cpu_dai, substream); - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return 0; } -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [GIT PULL] soundwire fixes for v5.4-rc5
by Vinod Koul 16 Aug '19

16 Aug '19

Hi Greg, We have couple of fixes queued up, please pull. Some more are in review, will send them later. These fixes are in linux-next as well. The following changes since commit 5f9e832c137075045d15cd6899ab0505cfb2ca4b: Linus 5.3-rc1 (2019-07-21 14:05:38 -0700) are available in the Git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire.git tags/soundwire-5.3-rc5 for you to fetch changes up to 8676b3ca4673517650fd509d7fa586aff87b3c28: soundwire: fix regmap dependencies and align with other serial links (2019-08-09 10:20:40 +0530) ---------------------------------------------------------------- soundwire fixes for v5.3-rc5 Pierre sent fixes which are queued now for v5.3-rc5 are: - regmap dependecy - cadence register definitions ---------------------------------------------------------------- Pierre-Louis Bossart (3): soundwire: cadence_master: fix register definition for SLAVE_STATE soundwire: cadence_master: fix definitions for INTSTAT0/1 soundwire: fix regmap dependencies and align with other serial links drivers/base/regmap/Kconfig | 2 +- drivers/soundwire/Kconfig | 7 +------ drivers/soundwire/Makefile | 2 +- drivers/soundwire/cadence_master.c | 8 ++++---- 4 files changed, 7 insertions(+), 12 deletions(-) Thanks -- ~Vinod

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[alsa-devel] [PATCH v3 1/2] ALSA: hda - Expand pin_match function to match upcoming new tbls
by Hui Wang 16 Aug '19

16 Aug '19

With the existing pintbl, we already have many entries in it. it is better to figure out a new way to reduce the size of the pintbl. We plan to define a new tbl which will match more machines with a single tbl, To do that, this function doesn't need to match all valid pins between machine and tbl, it just needs to match all pins defined in the tbl with the machine. And the plan is to move some tbls from pin_fixup_tbl to fallback_pin_fixup_tbl gradually. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 12 ++++++++---- sound/pci/hda/hda_local.h | 3 ++- sound/pci/hda/patch_realtek.c | 4 ++-- 3 files changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 92390d457567..8a134f750bae 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -884,7 +884,8 @@ EXPORT_SYMBOL_GPL(snd_hda_apply_fixup); #define IGNORE_SEQ_ASSOC (~(AC_DEFCFG_SEQUENCE | AC_DEFCFG_DEF_ASSOC)) static bool pin_config_match(struct hda_codec *codec, - const struct hda_pintbl *pins) + const struct hda_pintbl *pins, + bool match_all_pins) { const struct hda_pincfg *pin; int i; @@ -908,7 +909,8 @@ static bool pin_config_match(struct hda_codec *codec, return false; } } - if (!found && (cfg & 0xf0000000) != 0x40000000) + if (match_all_pins && + !found && (cfg & 0xf0000000) != 0x40000000) return false; } @@ -920,10 +922,12 @@ static bool pin_config_match(struct hda_codec *codec, * @codec: the HDA codec * @pin_quirk: zero-terminated pin quirk list * @fixlist: the fixup list + * @match_all_pins: all valid pins must match with the table entries */ void snd_hda_pick_pin_fixup(struct hda_codec *codec, const struct snd_hda_pin_quirk *pin_quirk, - const struct hda_fixup *fixlist) + const struct hda_fixup *fixlist, + bool match_all_pins) { const struct snd_hda_pin_quirk *pq; @@ -935,7 +939,7 @@ void snd_hda_pick_pin_fixup(struct hda_codec *codec, continue; if (codec->core.vendor_id != pq->codec) continue; - if (pin_config_match(codec, pq->pins)) { + if (pin_config_match(codec, pq->pins, match_all_pins)) { codec->fixup_id = pq->value; #ifdef CONFIG_SND_DEBUG_VERBOSE codec->fixup_name = pq->name; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 349a8312d06a..3942e1b528d8 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -361,7 +361,8 @@ void snd_hda_pick_fixup(struct hda_codec *codec, const struct hda_fixup *fixlist); void snd_hda_pick_pin_fixup(struct hda_codec *codec, const struct snd_hda_pin_quirk *pin_quirk, - const struct hda_fixup *fixlist); + const struct hda_fixup *fixlist, + bool match_all_pins); /* helper macros to retrieve pin default-config values */ #define get_defcfg_connect(cfg) \ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ac7177ce67eb..2fa03fb6bd9e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7950,7 +7950,7 @@ static int patch_alc269(struct hda_codec *codec) snd_hda_pick_fixup(codec, alc269_fixup_models, alc269_fixup_tbl, alc269_fixups); - snd_hda_pick_pin_fixup(codec, alc269_pin_fixup_tbl, alc269_fixups); + snd_hda_pick_pin_fixup(codec, alc269_pin_fixup_tbl, alc269_fixups, true); snd_hda_pick_fixup(codec, NULL, alc269_fixup_vendor_tbl, alc269_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -8935,7 +8935,7 @@ static int patch_alc662(struct hda_codec *codec) snd_hda_pick_fixup(codec, alc662_fixup_models, alc662_fixup_tbl, alc662_fixups); - snd_hda_pick_pin_fixup(codec, alc662_pin_fixup_tbl, alc662_fixups); + snd_hda_pick_pin_fixup(codec, alc662_pin_fixup_tbl, alc662_fixups, true); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); alc_auto_parse_customize_define(codec); -- 2.17.1

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[alsa-devel] [PATCH] Fix a stack buffer overflow bug check_input_term
by Hui Peng 15 Aug '19

15 Aug '19

`check_input_term` recursively calls itself with input from device side (e.g., uac_input_terminal_descriptor.bCSourceID) as argument (id). In `check_input_term`, if `check_input_term` is called with the same `id` argument as the caller, it triggers endless recursive call, resulting kernel space stack overflow. This patch fixes the bug by adding a bitmap to `struct mixer_build` to keep track of the checked ids by `check_input_term` and stop the execution if some id has been checked (similar to how parse_audio_unit handles unitid argument). Reported-by: Hui Peng <benquike(a)gmail.com> Reported-by: Mathias Payer <mathias.payer(a)nebelwelt.net> Signed-off-by: Hui Peng <benquike(a)gmail.com> --- sound/usb/mixer.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index ea487378be17..1f6c8213df82 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -68,6 +68,7 @@ struct mixer_build { unsigned char *buffer; unsigned int buflen; DECLARE_BITMAP(unitbitmap, MAX_ID_ELEMS); + DECLARE_BITMAP(termbitmap, MAX_ID_ELEMS); struct usb_audio_term oterm; const struct usbmix_name_map *map; const struct usbmix_selector_map *selector_map; @@ -782,6 +783,8 @@ static int check_input_term(struct mixer_build *state, int id, int err; void *p1; + if (test_and_set_bit(id, state->termbitmap)) + return 0; memset(term, 0, sizeof(*term)); while ((p1 = find_audio_control_unit(state, id)) != NULL) { unsigned char *hdr = p1; -- 2.22.1

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