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

[PATCH] ALSA: bebob: remove an unnecessary condition in hwdep_read()
by Takashi Sakamoto 26 Jan '21

26 Jan '21

Drivers in ALSA firewire stack supports eventing to userspace applications via ALSA hwdep interface. All of the drivers supports stream lock events. Some of them supports their unique events according to specification of target device. ALSA bebob driver supports the stream lock event only. In the case, it's enough to check condition only in loop with process blocking. However, current implementation check it again after breaking the loop. This commit removes the redundant check. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Reported-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob_hwdep.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/firewire/bebob/bebob_hwdep.c b/sound/firewire/bebob/bebob_hwdep.c index 8677e3ec8d14..6f9331655d43 100644 --- a/sound/firewire/bebob/bebob_hwdep.c +++ b/sound/firewire/bebob/bebob_hwdep.c @@ -37,11 +37,9 @@ hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count, memset(&event, 0, sizeof(event)); count = min_t(long, count, sizeof(event.lock_status)); - if (bebob->dev_lock_changed) { - event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS; - event.lock_status.status = (bebob->dev_lock_count > 0); - bebob->dev_lock_changed = false; - } + event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS; + event.lock_status.status = (bebob->dev_lock_count > 0); + bebob->dev_lock_changed = false; spin_unlock_irq(&bebob->lock); -- 2.27.0

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[PATCH] ASoC: dmaengine_pcm: add peripheral configuration
by Shengjiu Wang 26 Jan '21

26 Jan '21

The commit e7bbb7acabf4 ("dmaengine: add peripheral configuration") adds peripheral configuration for dma_slave_config. This configuration is useful for some audio peripherals, for example, the peripheral supports multi fifos, we can let the DMA know which fifos are selected. So also add this configuration for snd_dmaengine_dai_dma_data. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- include/sound/dmaengine_pcm.h | 5 +++++ sound/core/pcm_dmaengine.c | 2 ++ 2 files changed, 7 insertions(+) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index 8c5e38180fb0..96666efddb39 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -66,6 +66,9 @@ struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream) * @chan_name: Custom channel name to use when requesting DMA channel. * @fifo_size: FIFO size of the DAI controller in bytes * @flags: PCM_DAI flags, only SND_DMAENGINE_PCM_DAI_FLAG_PACK for now + * @peripheral_config: peripheral configuration for programming peripheral + * for dmaengine transfer + * @peripheral_size: peripheral configuration buffer size */ struct snd_dmaengine_dai_dma_data { dma_addr_t addr; @@ -76,6 +79,8 @@ struct snd_dmaengine_dai_dma_data { const char *chan_name; unsigned int fifo_size; unsigned int flags; + void *peripheral_config; + size_t peripheral_size; }; void snd_dmaengine_pcm_set_config_from_dai_data( diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 4d0e8fe535a1..1fc2fa077574 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -125,6 +125,8 @@ void snd_dmaengine_pcm_set_config_from_dai_data( } slave_config->slave_id = dma_data->slave_id; + slave_config->peripheral_config = dma_data->peripheral_config; + slave_config->peripheral_size = dma_data->peripheral_size; } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_set_config_from_dai_data); -- 2.27.0

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bug report: ALSA woke us up to write new data to the device, but there was actually nothing to write. Most likely this is a bug in the ALSA driver 'snd_usb_audio'
by lotus2＠tutanota.com 26 Jan '21

26 Jan '21

```$ journalctl -p3 -b --no-hostname -- Journal begins at Sun 2020-10-18 22:41:56 CEST, ends at Tue 2021-01-26 03:39:21 CET. -- Jan 26 01:29:10 pulseaudio[882]: GetManagedObjects() failed: org.freedesktop.DBus.Error.TimedOut: Failed to activate service 'org.bluez': timed out (service_start_timeout=25000ms) Jan 26 02:41:14 pulseaudio[882]: ALSA woke us up to write new data to the device, but there was actually nothing to write. Jan 26 02:41:14 pulseaudio[882]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jan 26 02:41:14 pulseaudio[882]: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. $ ``` I reported the `alsa-info_alsa-info.sh` output automatically and it is attached here Operating System: Manjaro Linux KDE Plasma Version: 5.20.5 KDE Frameworks Version: 5.78.0 Qt Version: 5.15.2 Kernel Version: 5.10.10-1-MANJARO OS Type: 64-bit

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[PATCH] ASoC: Intel: sof_sdw: set proper flags for Dell TGL-H SKU 0A5E
by Kai Vehmanen 25 Jan '21

25 Jan '21

From: Libin Yang <libin.yang(a)intel.com> Add flag "SOF_RT711_JD_SRC_JD2", flag "SOF_RT715_DAI_ID_FIX" and "SOF_SDW_FOUR_SPK" to the Dell TGL-H based SKU "0A5E". Signed-off-by: Libin Yang <libin.yang(a)intel.com> Co-developed-by: Hui Wang <hui.wang(a)canonical.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/intel/boards/sof_sdw.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index ca968901ac96..6d0d6ef711e0 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -67,6 +67,16 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { .driver_data = (void *)(SOF_RT711_JD_SRC_JD2 | SOF_RT715_DAI_ID_FIX), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0A5E") + }, + .driver_data = (void *)(SOF_RT711_JD_SRC_JD2 | + SOF_RT715_DAI_ID_FIX | + SOF_SDW_FOUR_SPK), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.29.2

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[PATCH v2] ASoC: qcom: lpass: Fix out-of-bounds DAI ID lookup
by Stephan Gerhold 25 Jan '21

25 Jan '21

The "dai_id" given into LPAIF_INTFDMA_REG(...) is already the real DAI ID, not an index into v->dai_driver. Looking it up again seems entirely redundant. For IPQ806x (and SC7180 since commit 09a4f6f5d21c ("ASoC: dt-bindings: lpass: Fix and common up lpass dai ids") this is now often an out-of-bounds read because the indexes in the "dai_driver" array no longer match the actual DAI ID. Cc: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Fixes: 7cb37b7bd0d3 ("ASoC: qcom: Add support for lpass hdmi driver") Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- Changes in v2: - Extracted from https://lore.kernel.org/alsa-devel/20210114094615.58191-2-stephan@gerhold.n… - Change commit message to clarify that this is usually not just redundant now but actually a broken out-of-bounds lookup. --- sound/soc/qcom/lpass-lpaif-reg.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-lpaif-reg.h b/sound/soc/qcom/lpass-lpaif-reg.h index 405542832e99..baf72f124ea9 100644 --- a/sound/soc/qcom/lpass-lpaif-reg.h +++ b/sound/soc/qcom/lpass-lpaif-reg.h @@ -133,7 +133,7 @@ #define LPAIF_WRDMAPERCNT_REG(v, chan) LPAIF_WRDMA_REG_ADDR(v, 0x14, (chan)) #define LPAIF_INTFDMA_REG(v, chan, reg, dai_id) \ - ((v->dai_driver[dai_id].id == LPASS_DP_RX) ? \ + ((dai_id == LPASS_DP_RX) ? \ LPAIF_HDMI_RDMA##reg##_REG(v, chan) : \ LPAIF_RDMA##reg##_REG(v, chan)) -- 2.30.0

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[PATCH v2] ASoC: mediatek: mt8192-mt6359: add format constraints for RT5682
by Tzung-Bi Shih 25 Jan '21

25 Jan '21

MT8192 determines the I2S clock rates according to the sampling rates. There is only 1 set of I2S in between MT8192 and RT5682. If playing and capturing via RT5682 in different sampling rates, the I2S data will be corrupted. Adds format constraints to the corresponding DAI links to make sure the sampling rates are symmetric. Fixes: 18b13ff23fab ("ASoC: mediatek: mt8192: add machine driver with mt6359, rt1015 and rt5682") Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- Changes from previous version: - Separated from the series [1] per [2]. - Updated commit message for more details. [1]: https://patchwork.kernel.org/project/alsa-devel/list/?series=419769 [2]: https://patchwork.kernel.org/project/alsa-devel/patch/20210122100742.369912… .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 49 +++++++++++++++++++ 1 file changed, 49 insertions(+) diff --git a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c index cc0fc72305d2..c368236d8a3a 100644 --- a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c +++ b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c @@ -410,6 +410,53 @@ static const struct snd_soc_ops mt8192_mt6359_rt1015_rt5682_capture1_ops = { .startup = mt8192_mt6359_rt1015_rt5682_cap1_startup, }; +static int +mt8192_mt6359_rt5682_startup(struct snd_pcm_substream *substream) +{ + static const unsigned int channels[] = { + 1, 2 + }; + static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, + }; + static const unsigned int rates[] = { + 48000 + }; + static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, + }; + + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + int ret; + + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + if (ret < 0) { + dev_err(rtd->dev, "hw_constraint_list channels failed\n"); + return ret; + } + + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + if (ret < 0) { + dev_err(rtd->dev, "hw_constraint_list rate failed\n"); + return ret; + } + + return 0; +} + +static const struct snd_soc_ops mt8192_mt6359_rt5682_ops = { + .startup = mt8192_mt6359_rt5682_startup, +}; + /* FE */ SND_SOC_DAILINK_DEFS(playback1, DAILINK_COMP_ARRAY(COMP_CPU("DL1")), @@ -657,6 +704,7 @@ static struct snd_soc_dai_link mt8192_mt6359_dai_links[] = { SND_SOC_DPCM_TRIGGER_PRE}, .dynamic = 1, .dpcm_playback = 1, + .ops = &mt8192_mt6359_rt5682_ops, SND_SOC_DAILINK_REG(playback3), }, { @@ -730,6 +778,7 @@ static struct snd_soc_dai_link mt8192_mt6359_dai_links[] = { SND_SOC_DPCM_TRIGGER_PRE}, .dynamic = 1, .dpcm_capture = 1, + .ops = &mt8192_mt6359_rt5682_ops, SND_SOC_DAILINK_REG(capture2), }, { -- 2.30.0.280.ga3ce27912f-goog

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[PATCH 0/2] ASoC: SOF: partial fix to Kconfig issues
by Pierre-Louis Bossart 25 Jan '21

25 Jan '21

We've had several reports of broken dependencies. The 'right' fix is to revisit the module dependencies as suggested by Arnd Bergmann. This is WIP at https://github.com/thesofproject/linux/pull/2683. Since this is taking longer than expected, I am only sharing quick fixes for now. Pierre-Louis Bossart (2): ASoC: SOF: Intel: soundwire: fix select/depend unmet dependencies ASoC: SOF: SND_INTEL_DSP_CONFIG dependency sound/soc/sof/intel/Kconfig | 3 ++- sound/soc/sof/sof-acpi-dev.c | 11 ++++++----- sound/soc/sof/sof-pci-dev.c | 10 ++++++---- 3 files changed, 14 insertions(+), 10 deletions(-) -- 2.25.1

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ASoC: simple-card, fsl-sai, ak4458, imx-ak4458
by Eliot Blennerhassett 25 Jan '21

25 Jan '21

Greetings, late last year I worked on a project using $SUBJECT. That has been suspended at least for now, so I'm writing these notes to my future self, or anyone else who may want to tread the same path. I'm happy to expand on explanation, or work on patches if anyone is interested. Background: Normally, fsl-sai is used with the platform-specific imx-ak4458 (afaik only in vendor tree). However, our project wished to be able use pcm1789 which has no imx specific driver so I started trying to use simple-card with the generic ak4458 driver. I encountered a some of problems doing this: ------------------------------------------------------------------------ 1) Reset polarity of ak4458. When imx-ak4458 is used, the platform driver handles the codec reset specified in DT compatible = "fsl,imx-audio-ak4458"; ak4458,pdn-gpio = <&gpio4 20 GPIO_ACTIVE_LOW>; Used here. Afaics gpio_set_value sets the raw value given, ignoring the polarity specified by the DT? gpio_set_value_cansleep(priv->pdn_gpio, 0); usleep_range(1000, 2000); gpio_set_value_cansleep(priv->pdn_gpio, 1); usleep_range(1000, 2000); The codec driver reset code is not used. When simple-card is used, the codec driver handles reset, specified like: compatible = "asahi-kasei,ak4458"; reset-gpios = <&gpio4 20 GPIO_ACTIVE_LOW>; and used here (inverse for power on) static void ak4458_power_off(struct ak4458_priv *ak4458) { if (ak4458->reset_gpiod) { gpiod_set_value_cansleep(ak4458->reset_gpiod, 0); usleep_range(1000, 2000); } } My understanding of gpiod functions is that value being set is the *logical* value of the GPIO. I.e. setting an active low gpio to 0 will set the hardware pin high. So it appears that the power_off and power_on functions are doing the opposite of what is intended. This is borne out by my hardware working correctly when the ------------------------------------------------------------------------ 2) Clock rate setting with simple-card When simple-card is used and DT specifies mclk fs: simple-audio-card,mclk-fs = <256>; asoc_simple_hw_params() calls snd_soc_dai_set_sysclk(..., clk_id=0, ...) The hard-coded clk_id=0 doesn't work with fsl-sai, which requires clk_id==1. For my testing purposes I changed the hard-coded value, but I think the proper solution could be to add a DT property to specify the clk_id (default=0) ? ------------------------------------------------------------------------ 3) Memory mapped stream access by aplay does not work. This precludes use of alsa plugins e.g. dmix I have found no reason or solution for this so far ------------------------------------------------------------------------ 4) Unable to get multiple source clocks working with fsl-sai With a single assigned-clock, switching between 48k and 44k1 clock rate families is accomplished by reparenting the root clock to the appropriate audio pll clock. &sai2 { assigned-clock-parents = <&clk IMX8MM_AUDIO_PLL1_OUT>; assigned-clocks = <&clk IMX8MM_CLK_SAI2>; assigned-clock-rates = <12288000>; ... }; However if two of the sai mclks could be set to 48k and 44k1 derived rates respectively, the clock reparenting would not be required, and fsl_sai_set_bclk() would search the mclks and choose the appropriate mclk for the requested rate. DT would be something like this: &sai2 { assigned-clock-parents = <&clk IMX8MM_AUDIO_PLL1_OUT>, <&clk IMX8MM_AUDIO_PLL2_OUT>; assigned-clocks = <&clk IMX8MM_CLK_SAI2>, <&clk IMX8MM_CLK_SAI1>; assigned-clock-rates = <12288000>, <11289600>; }; This setup doesn't work as I hoped it would, don't know why not. regards -- Eliot Blennerhassett

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[PATCH] ALSA: fireface: fix info leak in hwdep_read()
by Dan Carpenter 25 Jan '21

25 Jan '21

If "ff->dev_lock_changed" has not changed and "count" is too large then this will copy data beyond the end of the struct to user space. Fixes: f656edd5fb33 ("ALSA: fireface: add hwdep interface") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/firewire/fireface/ff-hwdep.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/fireface/ff-hwdep.c b/sound/firewire/fireface/ff-hwdep.c index 4b2e0dff5ddb..b84dde609a03 100644 --- a/sound/firewire/fireface/ff-hwdep.c +++ b/sound/firewire/fireface/ff-hwdep.c @@ -35,12 +35,12 @@ static long hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count, } memset(&event, 0, sizeof(event)); + count = min_t(long, count, sizeof(event.lock_status)); if (ff->dev_lock_changed) { event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS; event.lock_status.status = (ff->dev_lock_count > 0); ff->dev_lock_changed = false; - count = min_t(long, count, sizeof(event.lock_status)); } spin_unlock_irq(&ff->lock); -- 2.29.2

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[PATCH] ALSA: hda: intel-dsp-config: add PCI id for TGL-H
by Kai Vehmanen 25 Jan '21

25 Jan '21

From: Bard Liao <bard.liao(a)intel.com> Adding PCI id for TGL-H. Like for other TGL platforms, SOF is used if Soundwire codecs or PCH-DMIC is detected. Signed-off-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Xiuli Pan <xiuli.pan(a)intel.com> Reviewed-by: Libin Yang <libin.yang(a)intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/hda/intel-dsp-config.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index 6a0d070c60c9..c45686172517 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -307,6 +307,10 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0xa0c8, }, + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, + .device = 0x43c8, + }, #endif /* Elkhart Lake */ -- 2.29.2

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