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

[PATCH] ASoC: amd: add missed checks for clk_get()
by Chuhong Yuan 03 Dec '20

03 Dec '20

cz_da7219_init() does not check the return value of clk_get(), while da7219_clk_enable() calls clk_set_rate() to dereference the pointers. Add checks to fix the problems. Fixes: bb24a31ed584 ("ASoC: AMD: Configure wclk and bclk of master codec") Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> --- sound/soc/amd/acp-da7219-max98357a.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index a7702e64ec51..d5185cd3106b 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -74,7 +74,12 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) } da7219_dai_wclk = clk_get(component->dev, "da7219-dai-wclk"); + if (IS_ERR(da7219_dai_wclk)) + return PTR_ERR(da7219_dai_wclk); + da7219_dai_bclk = clk_get(component->dev, "da7219-dai-bclk"); + if (IS_ERR(da7219_dai_bclk)) + return PTR_ERR(da7219_dai_bclk); ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_LINEOUT | -- 2.26.2

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[PATCH] ASoC: fsl_audmix: Remove unneeded data field
by Fabio Estevam 03 Dec '20

03 Dec '20

The .data field is only used to pass the string name to platform_device_register_data(). Pass the string name directly to make the code simpler. Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- sound/soc/fsl/fsl_audmix.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c index 7ad5925772e8..8dc44dec7956 100644 --- a/sound/soc/fsl/fsl_audmix.c +++ b/sound/soc/fsl/fsl_audmix.c @@ -455,7 +455,6 @@ static const struct regmap_config fsl_audmix_regmap_config = { static const struct of_device_id fsl_audmix_ids[] = { { .compatible = "fsl,imx8qm-audmix", - .data = "imx-audmix", }, { /* sentinel */ } }; @@ -465,17 +464,9 @@ static int fsl_audmix_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; struct fsl_audmix *priv; - const char *mdrv; - const struct of_device_id *of_id; void __iomem *regs; int ret; - of_id = of_match_device(fsl_audmix_ids, dev); - if (!of_id || !of_id->data) - return -EINVAL; - - mdrv = of_id->data; - priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) return -ENOMEM; @@ -510,10 +501,10 @@ static int fsl_audmix_probe(struct platform_device *pdev) goto err_disable_pm; } - priv->pdev = platform_device_register_data(dev, mdrv, 0, NULL, 0); + priv->pdev = platform_device_register_data(dev, "imx-audmix", 0, NULL, 0); if (IS_ERR(priv->pdev)) { ret = PTR_ERR(priv->pdev); - dev_err(dev, "failed to register platform %s: %d\n", mdrv, ret); + dev_err(dev, "failed to register platform: %d\n", ret); goto err_disable_pm; } -- 2.17.1

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[PATCH 1/2] ASoC: fsl-asoc-card: Add support for si476x codec
by Shengjiu Wang 03 Dec '20

03 Dec '20

The si476x codec is used for FM radio function on i.MX6 auto board, it only supports recording function. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl-asoc-card.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index a2dd3b6b7fec..f62f81ceab0d 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -131,6 +131,13 @@ static const struct snd_soc_dapm_route audio_map_tx[] = { {"CPU-Playback", NULL, "ASRC-Playback"}, }; +static const struct snd_soc_dapm_route audio_map_rx[] = { + /* 1st half -- Normal DAPM routes */ + {"CPU-Capture", NULL, "Capture"}, + /* 2nd half -- ASRC DAPM routes */ + {"ASRC-Capture", NULL, "CPU-Capture"}, +}; + /* Add all possible widgets into here without being redundant */ static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = { SND_SOC_DAPM_LINE("Line Out Jack", NULL), @@ -653,6 +660,11 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.slot_width = 32; priv->card.dapm_routes = audio_map_tx; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx); + } else if (of_device_is_compatible(np, "fsl,imx-audio-si476x")) { + codec_dai_name = "si476x-codec"; + priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + priv->card.dapm_routes = audio_map_rx; + priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_rx); } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); ret = -EINVAL; @@ -869,6 +881,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-wm8960", }, { .compatible = "fsl,imx-audio-mqs", }, { .compatible = "fsl,imx-audio-wm8524", }, + { .compatible = "fsl,imx-audio-si476x", }, {} }; MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids); -- 2.27.0

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[PATCH 0/4] ASoC/SoundWire: add Realtek SoundWire SDCA codec
by Bard Liao 03 Dec '20

03 Dec '20

Add Realtek rt1316 and rt711-sdca vender-specific drivers plus some fixup for rt715-sdca driver. Jack Yu (2): ASoC/SoundWire: rt715-sdca: use rt715_sdca prefix for symbols ASoC/SoundWire: rt715-sdca: merge mute/volume operations Shuming Fan (2): ASoC/SoundWire: rt1316: Add RT1316 SDCA vendor-specific driver ASoC/SoundWire: rt711-sdca: Add RT711 SDCA vendor-specific driver sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 4 + 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 | 12 +- sound/soc/codecs/rt715-sdca.c | 323 ++++--- sound/soc/codecs/rt715-sdca.h | 4 +- 11 files changed, 3316 insertions(+), 164 deletions(-) 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 -- 2.17.1

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[PATCH v3 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. CPU0 CPU1 dpcm_be_dai_trigger(): dpcm_be_disconnect(): for_each_dpcm_be(fe, stream, dpcm) { 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); 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

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[PATCH v3 0/1] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 03 Dec '20

03 Dec '20

>Hi Gyeongtaek > >Thank you for your help > >> I tried to find out some change from the history. >> However, I saw that dpcm_lock never be held in dpcm_be_dai_trigger() from the beginning. >> So, I just fixed some miss from the commit comment and resend. >> If you have another suggestion or request, just let me know. > >OK, thanks. >I just thought it was related to recent (especially mine :) patch. > >BYW, your posted v2 still doesn't have alignment on log. >Is it mailer issue ? Sorry. I didn't fully understand what alignment means. I fixed the log by looking at some previous commits. Thank you so much. Gyeongtaek Lee > >> And I have a question. >> This patch can?t be applied to stable branches directly by conflict. >> I found out following commits are needed to apply this patch into the 5.4 stable. >> ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state() >> ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code >> In this case, what should I do when stable maintainer requests resolving conflict to me? >> Should I send those patches to the maintainer? >> Or rewrite the patch on the latest of the each stable? > >Mark, please help him > >Thank you for your help !! > >Best regards >--- >Kuninori Morimoto > Gyeongtaek Lee (1): ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger() sound/soc/soc-pcm.c | 62 ++++++++++++++++++++++++++++++++------------- 1 file changed, 44 insertions(+), 18 deletions(-) base-commit: fa02fcd94b0c8dff6cc65714510cf25ad194b90d -- 2.21.0

1 0

[PATCH v2 0/1] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 03 Dec '20

03 Dec '20

On 03 Dec 2020 07:33:03 +0900, Kuninori Morimoto wrote: >Hi Gyeongtaek > >Thank you for your patch > >> 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 > >It is easy to read/understand if this code has alignment. > >> 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 >> --- > >Is this bugfix patch for some recent commit ? >If so, having Fixes: tag is helpful. > >Thank you for your help !! I tried to find out some change from the history. However, I saw that dpcm_lock never be held in dpcm_be_dai_trigger() from the beginning. So, I just fixed some miss from the commit comment and resend. If you have another suggestion or request, just let me know. And I have a question. This patch can’t be applied to stable branches directly by conflict. I found out following commits are needed to apply this patch into the 5.4 stable. ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state() ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code In this case, what should I do when stable maintainer requests resolving conflict to me? Should I send those patches to the maintainer? Or rewrite the patch on the latest of the each stable? Thank you! Gyeongtaek Lee > >Best regards >--- >Kuninori Morimoto > Gyeongtaek Lee (1): ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger() sound/soc/soc-pcm.c | 62 ++++++++++++++++++++++++++++++++------------- 1 file changed, 44 insertions(+), 18 deletions(-) base-commit: fa02fcd94b0c8dff6cc65714510cf25ad194b90d -- 2.21.0

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[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

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[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

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[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

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