December 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH v0] ASoC: rsnd: core: Check convert rate in rsnd_hw_params
by mdurnev＠gmail.com 03 Dec '20

03 Dec '20

From: Mikhail Durnev <mikhail_durnev(a)mentor.com> snd_pcm_hw_params_set_rate_near can return incorrect sample rate in some cases, e.g. when the backend output rate is set to some value higher than 48000 Hz and the input rate is 8000 Hz. So passing the value returned by snd_pcm_hw_params_set_rate_near to snd_pcm_hw_params will result in "FSO/FSI ratio error" and playing no audio at all while the userland is not properly notified about the issue. If SRC is unable to convert the requested sample rate to the sample rate the backend is using, then the requested sample rate should be adjusted in rsnd_hw_params. The userland will be notified about that change in the returned hw_params structure. Signed-off-by: Mikhail Durnev <mikhail_durnev(a)mentor.com> --- sound/soc/sh/rcar/core.c | 67 +++++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 66 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 6e670b3..8ca3fb7 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1428,8 +1428,73 @@ static int rsnd_hw_params(struct snd_soc_component *component, } if (io->converted_chan) dev_dbg(dev, "convert channels = %d\n", io->converted_chan); - if (io->converted_rate) + if (io->converted_rate) { dev_dbg(dev, "convert rate = %d\n", io->converted_rate); + + /* + * SRC supports convert rates from params_rate(hw_params)/k_down + * to params_rate(hw_params)*k_up, where k_up is always 6, and + * k_down depends on number of channels and SRC unit. + * So all SRC units can upsample audio up to 6 times regardless + * its number of channels. And all SRC units can downsample + * 2 channel audio up to 6 times too. + */ + int k_up = 6; + int k_down = 6; + int channel; + struct rsnd_mod *src_mod = rsnd_io_to_mod_src(io); + + channel = io->converted_chan ? io->converted_chan : fe_channel; + + switch (rsnd_mod_id(src_mod)) { + /* + * SRC0 can downsample 4, 6 and 8 channel audio up to 4 times. + * SRC1, SRC3 and SRC4 can downsample 4 channel audio + * up to 4 times. + * SRC1, SRC3 and SRC4 can downsample 6 and 8 channel audio + * no more than twice. + */ + case 1: + case 3: + case 4: + if (channel > 4) { + k_down = 2; + break; + } + case 0: + if (channel > 2) + k_down = 4; + break; + + /* Other SRC units do not support more than 2 channels */ + default: + if (channel > 2) + return -EINVAL; + } + + if (params_rate(hw_params) > io->converted_rate * k_down) { + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->min = + io->converted_rate * k_down; + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->max = + io->converted_rate * k_down; + hw_params->cmask |= SNDRV_PCM_HW_PARAM_RATE; + } else if (params_rate(hw_params) * k_up < io->converted_rate) { + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->min = + (io->converted_rate + k_up - 1) / k_up; + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->max = + (io->converted_rate + k_up - 1) / k_up; + hw_params->cmask |= SNDRV_PCM_HW_PARAM_RATE; + } + + /* + * TBD: Max SRC input and output rates also depend on number + * of channels and SRC unit: + * SRC1, SRC3 and SRC4 do not support more than 128kHz + * for 6 channel and 96kHz for 8 channel audio. + * Perhaps this function should return EINVAL if the input or + * the output rate exceeds the limitation. + */ + } } return rsnd_dai_call(hw_params, io, substream, hw_params); -- 2.7.4

4 4

[PATCH v2 1/1] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 03 Dec '20

03 Dec '20

If stop by underrun and DPCM BE disconnection is run simultaneously, data abort can be occurred by the sequence below. /* In core X, running dpcm_be_dai_trigger() */ for_each_dpcm_be(fe, stream, dpcm) { /* In core Y, running dpcm_be_disconnect() */ spin_lock_irqsave(&fe->card->dpcm_lock, flags); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); kfree(dpcm); /* In core X, running dpcm_be_dai_trigger() */ struct snd_soc_pcm_runtime *be = dpcm->be; <== Accessing freed memory To prevent this situation, dpcm_lock should be acquired during iteration of dpcm list in dpcm_be_dai_trigger(). Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Cc: stable(a)vger.kernel.org --- sound/soc/soc-pcm.c | 62 ++++++++++++++++++++++++++++++++------------- 1 file changed, 44 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index dcab9527ba3d..7c5d950a8628 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2073,6 +2073,9 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, return ret; } +static int dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); + static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, struct snd_pcm_substream *substream, int cmd) { @@ -2092,8 +2095,10 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) { struct snd_soc_dpcm *dpcm; + unsigned long flags; int ret = 0; + spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; @@ -2113,7 +2118,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2123,7 +2128,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2133,7 +2138,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2142,12 +2147,12 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; break; @@ -2155,12 +2160,12 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_SUSPEND; break; @@ -2168,17 +2173,20 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; break; } + if (ret) + break; } + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); return ret; } @@ -2916,10 +2924,9 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, struct snd_soc_dpcm *dpcm; int state; int ret = 1; - unsigned long flags; int i; - spin_lock_irqsave(&fe->card->dpcm_lock, flags); + lockdep_assert_held(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -2933,17 +2940,12 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, } } } - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); /* it's safe to do this BE DAI */ return ret; } -/* - * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE - * are not running, paused or suspended for the specified stream direction. - */ -int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, +static int dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { const enum snd_soc_dpcm_state state[] = { @@ -2954,6 +2956,23 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } + +/* + * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE + * are not running, paused or suspended for the specified stream direction. + */ +int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + unsigned long flags; + int ret; + + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + ret = dpcm_can_be_free_stop(fe, be, stream); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; +} EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); /* @@ -2963,6 +2982,9 @@ EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { + unsigned long flags; + int ret; + const enum snd_soc_dpcm_state state[] = { SND_SOC_DPCM_STATE_START, SND_SOC_DPCM_STATE_PAUSED, @@ -2970,6 +2992,10 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, SND_SOC_DPCM_STATE_PREPARE, }; - return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + ret = snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); -- 2.21.0

1 0

[PATCH 1/2] ASoC: dt-bindings: imx-hdmi: Add binding doc for hdmi machine driver
by Shengjiu Wang 03 Dec '20

03 Dec '20

Imx-hdmi is a new added machine driver for supporting hdmi devices on i.MX platforms. There is HDMI IP or external HDMI modules connect with SAI or AUD2HTX interface. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- .../bindings/sound/imx-audio-hdmi.yaml | 52 +++++++++++++++++++ 1 file changed, 52 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml diff --git a/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml b/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml new file mode 100644 index 000000000000..d5474f83ac2c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml @@ -0,0 +1,52 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/imx-audio-hdmi.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP i.MX audio complex with HDMI + +maintainers: + - Shengjiu Wang <shengjiu.wang(a)nxp.com> + +properties: + compatible: + enum: + - fsl,imx-audio-hdmi + - fsl,imx-audio-sii902x + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + audio-cpu: + description: The phandle of an CPU DAI controller + + hdmi-out: + description: | + This is a boolean property. If present, the transmitting function + of HDMI will be enabled, indicating there's a physical HDMI out + connector or jack on the board or it's connecting to some other IP + block, such as an HDMI encoder or display-controller. + + hdmi-in: + description: | + This is a boolean property. If present, the receiving function of + HDMI will be enabled, indicating there is a physical HDMI in + connector/jack on the board. + +required: + - compatible + - model + - audio-cpu + +additionalProperties: false + +examples: + - | + sound-hdmi { + compatible = "fsl,imx-audio-hdmi"; + model = "audio-hdmi"; + audio-cpu = <&aud2htx>; + hdmi-out; + }; -- 2.27.0

3 3

[PATCH v2 0/5] regmap/SoundWire/ASoC: Add SoundWire SDCA support
by Bard Liao 03 Dec '20

03 Dec '20

The MIPI SoundWire Device Class standard will define audio functionality beyond the scope of the existing SoundWire 1.2 standard, which is limited to the bus and interface. The description is inspired by the USB Audio Class, with "functions", "entities", "control selectors", "audio clusters". The main difference with the USB Audio class is that the devices are typically on a motherboard and descriptors stored in platform firmware instead of being retrieved from the device. The current set of devices managed in this patchset are conformant with the SDCA 0.6 specification and require dedicated drivers since the descriptors and platform firmware specification is not complete at this time. They do however rely on the hierarchical addressing required by the SDCA standard. Future devices conformant with SDCA 1.0 should rely on a class driver. This series adds support for the hierarchical SDCA addressing and extends regmap. It then provides 3 codecs for RT711-sdca headset codec, RT1316 amplifier and RT715-scda microphone codec. Note that the release of this code before the formal adoption of the SDCA 1.0 specification was formally endorsed by the MIPI Board to make sure there is no delay for Linux-based support of this specification. v2: - rt715-sdca: Use rt715_sdca prefix to avoid compiling issue. - rt715-sdca: Merge multiple mute/volume operation into single mute/volume operation - rt711-sdca: Initial ret = 0 as it could be used uninitialized. Jack Yu (1): ASoC/SoundWire: rt715-sdca: First version of rt715 sdw sdca codec driver Pierre-Louis Bossart (2): soundwire: SDCA: add helper macro to access controls regmap/SoundWire: sdw: add support for SoundWire 1.2 MBQ Shuming Fan (2): ASoC/SoundWire: rt1316: Add RT1316 SDCA vendor-specific driver ASoC/SoundWire: rt711-sdca: Add RT711 SDCA vendor-specific driver drivers/base/regmap/Kconfig | 6 +- drivers/base/regmap/Makefile | 1 + drivers/base/regmap/regmap-sdw-mbq.c | 101 ++ include/linux/regmap.h | 35 + include/linux/soundwire/sdw_registers.h | 32 + sound/soc/codecs/Kconfig | 20 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/rt1316-sdw.c | 756 ++++++++++++ sound/soc/codecs/rt1316-sdw.h | 115 ++ sound/soc/codecs/rt711-sdca-sdw.c | 424 +++++++ sound/soc/codecs/rt711-sdca-sdw.h | 101 ++ sound/soc/codecs/rt711-sdca.c | 1482 +++++++++++++++++++++++ sound/soc/codecs/rt711-sdca.h | 246 ++++ sound/soc/codecs/rt715-sdca-sdw.c | 278 +++++ sound/soc/codecs/rt715-sdca-sdw.h | 170 +++ sound/soc/codecs/rt715-sdca.c | 947 +++++++++++++++ sound/soc/codecs/rt715-sdca.h | 124 ++ 17 files changed, 4843 insertions(+), 1 deletion(-) create mode 100644 drivers/base/regmap/regmap-sdw-mbq.c create mode 100644 sound/soc/codecs/rt1316-sdw.c create mode 100644 sound/soc/codecs/rt1316-sdw.h create mode 100644 sound/soc/codecs/rt711-sdca-sdw.c create mode 100644 sound/soc/codecs/rt711-sdca-sdw.h create mode 100644 sound/soc/codecs/rt711-sdca.c create mode 100644 sound/soc/codecs/rt711-sdca.h create mode 100644 sound/soc/codecs/rt715-sdca-sdw.c create mode 100644 sound/soc/codecs/rt715-sdca-sdw.h create mode 100644 sound/soc/codecs/rt715-sdca.c create mode 100644 sound/soc/codecs/rt715-sdca.h -- 2.17.1

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[PATCH] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 03 Dec '20

03 Dec '20

If stop by underrun and DPCM BE disconnection is run simultaneously, data abort can be occurred by the sequence below. /* In core X, running dpcm_be_dai_trigger() */ for_each_dpcm_be(fe, stream, dpcm) { /* In core Y, running dpcm_be_disconnect() */ spin_lock_irqsave(&fe->card->dpcm_lock, flags); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); kfree(dpcm); /* In core X, running dpcm_be_dai_trigger() */ struct snd_soc_pcm_runtime *be = dpcm->be; <== Accessing freed memory To prevent this situation, dpcm_lock should be acquired during iteration of dpcm list in dpcm_do_trigger(). Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Cc: stable(a)vger.kernel.org --- sound/soc/soc-pcm.c | 62 ++++++++++++++++++++++++++++++++------------- 1 file changed, 44 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index dcab9527ba3d..7c5d950a8628 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2073,6 +2073,9 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, return ret; } +static int dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); + static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, struct snd_pcm_substream *substream, int cmd) { @@ -2092,8 +2095,10 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) { struct snd_soc_dpcm *dpcm; + unsigned long flags; int ret = 0; + spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; @@ -2113,7 +2118,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2123,7 +2128,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2133,7 +2138,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2142,12 +2147,12 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; break; @@ -2155,12 +2160,12 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_SUSPEND; break; @@ -2168,17 +2173,20 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; break; } + if (ret) + break; } + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); return ret; } @@ -2916,10 +2924,9 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, struct snd_soc_dpcm *dpcm; int state; int ret = 1; - unsigned long flags; int i; - spin_lock_irqsave(&fe->card->dpcm_lock, flags); + lockdep_assert_held(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -2933,17 +2940,12 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, } } } - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); /* it's safe to do this BE DAI */ return ret; } -/* - * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE - * are not running, paused or suspended for the specified stream direction. - */ -int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, +static int dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { const enum snd_soc_dpcm_state state[] = { @@ -2954,6 +2956,23 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } + +/* + * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE + * are not running, paused or suspended for the specified stream direction. + */ +int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + unsigned long flags; + int ret; + + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + ret = dpcm_can_be_free_stop(fe, be, stream); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; +} EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); /* @@ -2963,6 +2982,9 @@ EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { + unsigned long flags; + int ret; + const enum snd_soc_dpcm_state state[] = { SND_SOC_DPCM_STATE_START, SND_SOC_DPCM_STATE_PAUSED, @@ -2970,6 +2992,10 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, SND_SOC_DPCM_STATE_PREPARE, }; - return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + ret = snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); base-commit: fa02fcd94b0c8dff6cc65714510cf25ad194b90d -- 2.21.0

2 2

[PATCH] ASoC: codecs: lpass-va-macro: remove some dead code
by Dan Carpenter 02 Dec '20

02 Dec '20

The "decimator" variable is in the 0-7 range and it's unsigned so there is no need to check for negative values. Fixes: 908e6b1df26e ("ASoC: codecs: lpass-va-macro: Add support to VA Macro") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/codecs/lpass-va-macro.c | 15 ++++----------- 1 file changed, 4 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index 3e6bbef26dcb..91e6890d6efc 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -844,17 +844,10 @@ static int va_macro_hw_params(struct snd_pcm_substream *substream, for_each_set_bit(decimator, &va->active_ch_mask[dai->id], VA_MACRO_DEC_MAX) { - if (decimator >= 0) { - tx_fs_reg = CDC_VA_TX0_TX_PATH_CTL + - VA_MACRO_TX_PATH_OFFSET * decimator; - snd_soc_component_update_bits(component, tx_fs_reg, - 0x0F, tx_fs_rate); - } else { - dev_err(va_dev, - "%s: ERROR: Invalid decimator: %d\n", - __func__, decimator); - return -EINVAL; - } + tx_fs_reg = CDC_VA_TX0_TX_PATH_CTL + + VA_MACRO_TX_PATH_OFFSET * decimator; + snd_soc_component_update_bits(component, tx_fs_reg, 0x0F, + tx_fs_rate); } return 0; } -- 2.29.2

3 3

[PATCH] ASoC: rt715: remove unused parameter
by jack.yu＠realtek.com 02 Dec '20

02 Dec '20

From: Jack Yu <jack.yu(a)realtek.com> Remove unused parameter in rt715.h. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt715.h | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt715.h b/sound/soc/codecs/rt715.h index d0d0fd2a6fdb..009a8266f606 100644 --- a/sound/soc/codecs/rt715.h +++ b/sound/soc/codecs/rt715.h @@ -207,7 +207,6 @@ struct sdw_stream_data { enum { RT715_AIF1, RT715_AIF2, - RT715_AIFS, }; #define RT715_POWER_UP_DELAY_MS 400 -- 2.29.0

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[PATCH] ASoC: Intel: Boards: tgl_max98373: update TDM slot_width
by Ranjani Sridharan 02 Dec '20

02 Dec '20

From: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Speaker amp's SSP bclk configuration was changed in the topology file to be based on 12.288MHz and dai_ops->hw_params is based on s32le format. But, the TDM slot size remained set to 24 bits. This inconsistency created audible noises and needs to be corrected. This patch updates TDM slot width to 32. Fixes: bc7477fc2ab4 ("ASoC: Intel: Boards: tgl_max98373: Update TDM configuration in hw_params") Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/intel/boards/sof_maxim_common.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index b6e63ea13d64..c2a9757181fe 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -49,11 +49,11 @@ static int max98373_hw_params(struct snd_pcm_substream *substream, for_each_rtd_codec_dais(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name, MAX_98373_DEV0_NAME)) { /* DEV0 tdm slot configuration */ - snd_soc_dai_set_tdm_slot(codec_dai, 0x03, 3, 8, 24); + snd_soc_dai_set_tdm_slot(codec_dai, 0x03, 3, 8, 32); } if (!strcmp(codec_dai->component->name, MAX_98373_DEV1_NAME)) { /* DEV1 tdm slot configuration */ - snd_soc_dai_set_tdm_slot(codec_dai, 0x0C, 3, 8, 24); + snd_soc_dai_set_tdm_slot(codec_dai, 0x0C, 3, 8, 32); } } return 0; -- 2.25.1

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[RESEND PATCH 0/6] ASoC: mediatek: mt8192: support new machine rt1015p_rt5682
by Tzung-Bi Shih 02 Dec '20

02 Dec '20

The series supports new machines with rt1015p and rt5682. Reuses most of the code in mt8192-mt6359-rt1015-rt5682.c. The first 3 patches refactor the existing mt8192-mt6359-rt1015-rt5682.c. For easier to support newly added components later. The 4th patch fixes a typo in DT bindings document. The 5th patch proposes a new compatible string "mt8192_mt6359_rt1015p_rt5682" for machines with rt1015p and rt5682. The 6th patch reuses the existing machine driver to support machines with rt1015p and rt5682. Tzung-Bi Shih (6): ASoC: mediatek: mt8192: rename common symbols ASoC: mediatek: mt8192: extract rt1015_rt5682 specific DAI link ASoC: mediatek: mt8192: move rt1015_rt5682 specific data ASoC: dt-bindings: mt8192-mt6359: fix typo in the example ASoC: dt-bindings: mt8192-mt6359: add new compatible for using rt1015p ASoC: mediatek: mt8192: support rt1015p_rt5682 .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 6 +- sound/soc/mediatek/Kconfig | 1 + .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 171 +++++++++++++----- 3 files changed, 129 insertions(+), 49 deletions(-) -- 2.29.2.454.gaff20da3a2-goog

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How to interperet snd_pcm_htimestamp available frames?
by Eric Buddington 02 Dec '20

02 Dec '20

I am recording from multiple (cheap USB) mics, and trying to synchronize them. I am assuming that there will be a common minimum latency (reflecting an uncontested USB bus), with occasional spikes up - but not down. In my loop, I am calling snd_pcm_mmap_begin(), snd_pcm_htimestamp(), and snd__mmap_commit() in that order. My question is about the 'avail' value returned from htimestamp. It is documented as "Number of available frames when timestamp was grabbed". Does this mean frames since the last frame consumed with _mmap_commit()? Is there a possibility of there being no timestamp, or one where 'avail' refers to some older state? Thanks for any insight. -Eric

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