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[PATCH AUTOSEL 6.2 07/45] ASoC: Intel: avs: nau8825: Adjust clock control
by Sasha Levin 22 Mar '23

22 Mar '23

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit 6206b2e787da2ed567922c37bb588a44f6fb6705 ] Internal clock shall be adjusted also in cases when DAPM event other than 'ON' is triggered. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Link: https://lore.kernel.org/r/20230303134854.2277146-6-amadeuszx.slawinski@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/avs/boards/nau8825.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c index 6731d8a490767..49438a67a77c6 100644 --- a/sound/soc/intel/avs/boards/nau8825.c +++ b/sound/soc/intel/avs/boards/nau8825.c @@ -33,15 +33,15 @@ avs_nau8825_clock_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *co return -EINVAL; } - if (!SND_SOC_DAPM_EVENT_ON(event)) { + if (SND_SOC_DAPM_EVENT_ON(event)) + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_MCLK, 24000000, + SND_SOC_CLOCK_IN); + else ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "set sysclk err = %d\n", ret); - return ret; - } - } + if (ret < 0) + dev_err(card->dev, "Set sysclk failed: %d\n", ret); - return 0; + return ret; } static const struct snd_kcontrol_new card_controls[] = { -- 2.39.2

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[PATCH AUTOSEL 6.2 06/45] ASoC: Intel: avs: ssm4567: Remove nau8825 bits
by Sasha Levin 22 Mar '23

22 Mar '23

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit 933de2d127281731166cf2880fa1e23c5a0f7faa ] Some of the nau8825 clock control got into the ssm4567, remove it. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Link: https://lore.kernel.org/r/20230303134854.2277146-5-amadeuszx.slawinski@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/avs/boards/ssm4567.c | 31 ---------------------------- 1 file changed, 31 deletions(-) diff --git a/sound/soc/intel/avs/boards/ssm4567.c b/sound/soc/intel/avs/boards/ssm4567.c index c5db696127624..2b7f5ad92aca7 100644 --- a/sound/soc/intel/avs/boards/ssm4567.c +++ b/sound/soc/intel/avs/boards/ssm4567.c @@ -15,7 +15,6 @@ #include <sound/soc-acpi.h> #include "../../../codecs/nau8825.h" -#define SKL_NUVOTON_CODEC_DAI "nau8825-hifi" #define SKL_SSM_CODEC_DAI "ssm4567-hifi" static struct snd_soc_codec_conf card_codec_conf[] = { @@ -34,41 +33,11 @@ static const struct snd_kcontrol_new card_controls[] = { SOC_DAPM_PIN_SWITCH("Right Speaker"), }; -static int -platform_clock_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *control, int event) -{ - struct snd_soc_dapm_context *dapm = w->dapm; - struct snd_soc_card *card = dapm->card; - struct snd_soc_dai *codec_dai; - int ret; - - codec_dai = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI); - if (!codec_dai) { - dev_err(card->dev, "Codec dai not found\n"); - return -EINVAL; - } - - if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_MCLK, 24000000, - SND_SOC_CLOCK_IN); - if (ret < 0) - dev_err(card->dev, "set sysclk err = %d\n", ret); - } else { - ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); - if (ret < 0) - dev_err(card->dev, "set sysclk err = %d\n", ret); - } - - return ret; -} - static const struct snd_soc_dapm_widget card_widgets[] = { SND_SOC_DAPM_SPK("Left Speaker", NULL), SND_SOC_DAPM_SPK("Right Speaker", NULL), SND_SOC_DAPM_SPK("DP1", NULL), SND_SOC_DAPM_SPK("DP2", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, platform_clock_control, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route card_base_routes[] = { -- 2.39.2

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[PATCH AUTOSEL 6.2 05/45] ASoC: Intel: avs: rt5682: Explicitly define codec format
by Sasha Levin 22 Mar '23

22 Mar '23

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit d24dbc865c2bd5946bef62bb862a65df092dfc79 ] rt5682 is headset codec configured in 48000/2/S24_LE format regardless of front end format, so force it to be so. Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Link: https://lore.kernel.org/r/20230303134854.2277146-4-amadeuszx.slawinski@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/avs/boards/rt5682.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/soc/intel/avs/boards/rt5682.c b/sound/soc/intel/avs/boards/rt5682.c index 473e9fe5d0bf7..b2c2ba93dcb56 100644 --- a/sound/soc/intel/avs/boards/rt5682.c +++ b/sound/soc/intel/avs/boards/rt5682.c @@ -169,6 +169,27 @@ static const struct snd_soc_ops avs_rt5682_ops = { .hw_params = avs_rt5682_hw_params, }; +static int +avs_rt5682_be_fixup(struct snd_soc_pcm_runtime *runtime, struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate, *channels; + struct snd_mask *fmt; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSPN to 24 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, struct snd_soc_dai_link **dai_link) { @@ -201,6 +222,7 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in dl->id = 0; dl->init = avs_rt5682_codec_init; dl->exit = avs_rt5682_codec_exit; + dl->be_hw_params_fixup = avs_rt5682_be_fixup; dl->ops = &avs_rt5682_ops; dl->nonatomic = 1; dl->no_pcm = 1; -- 2.39.2

1 0

[PATCH AUTOSEL 6.2 04/45] ASoC: Intel: avs: da7219: Explicitly define codec format
by Sasha Levin 22 Mar '23

22 Mar '23

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit 61f368624fe4d0c25c6e9c917574b8ace51d776e ] da7219 is headset codec configured in 48000/2/S24_LE format regardless of front end format, so force it to be so. Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Link: https://lore.kernel.org/r/20230303134854.2277146-3-amadeuszx.slawinski@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/avs/boards/da7219.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c index acd43b6108e99..1a1d572cc1d02 100644 --- a/sound/soc/intel/avs/boards/da7219.c +++ b/sound/soc/intel/avs/boards/da7219.c @@ -117,6 +117,26 @@ static void avs_da7219_codec_exit(struct snd_soc_pcm_runtime *rtd) snd_soc_component_set_jack(asoc_rtd_to_codec(rtd, 0)->component, NULL, NULL); } +static int +avs_da7219_be_fixup(struct snd_soc_pcm_runtime *runrime, struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate, *channels; + struct snd_mask *fmt; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 24 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, struct snd_soc_dai_link **dai_link) { @@ -148,6 +168,7 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in dl->num_platforms = 1; dl->id = 0; dl->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; + dl->be_hw_params_fixup = avs_da7219_be_fixup; dl->init = avs_da7219_codec_init; dl->exit = avs_da7219_codec_exit; dl->nonatomic = 1; -- 2.39.2

1 0

[PATCH AUTOSEL 6.2 03/45] ASoC: Intel: avs: max98357a: Explicitly define codec format
by Sasha Levin 22 Mar '23

22 Mar '23

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit d16c893425d07ada1fdd817ec06d322efcf69480 ] max98357a is speaker codec configured in 48000/2/S16_LE format regardless of front end format, so force it to be so. Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Link: https://lore.kernel.org/r/20230303134854.2277146-2-amadeuszx.slawinski@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/avs/boards/max98357a.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/soc/intel/avs/boards/max98357a.c b/sound/soc/intel/avs/boards/max98357a.c index 921f42caf7e09..183123d08c5a3 100644 --- a/sound/soc/intel/avs/boards/max98357a.c +++ b/sound/soc/intel/avs/boards/max98357a.c @@ -8,6 +8,7 @@ #include <linux/module.h> #include <linux/platform_device.h> +#include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-acpi.h> #include <sound/soc-dapm.h> @@ -24,6 +25,26 @@ static const struct snd_soc_dapm_route card_base_routes[] = { { "Spk", NULL, "Speaker" }, }; +static int +avs_max98357a_be_fixup(struct snd_soc_pcm_runtime *runrime, struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate, *channels; + struct snd_mask *fmt; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 16 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); + return 0; +} + static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, struct snd_soc_dai_link **dai_link) { @@ -55,6 +76,7 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in dl->num_platforms = 1; dl->id = 0; dl->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; + dl->be_hw_params_fixup = avs_max98357a_be_fixup; dl->nonatomic = 1; dl->no_pcm = 1; dl->dpcm_playback = 1; -- 2.39.2

1 0

[PATCH AUTOSEL 6.2 02/45] ASoC: codecs: tx-macro: Fix for KASAN: slab-out-of-bounds
by Sasha Levin 22 Mar '23

22 Mar '23

From: Ravulapati Vishnu Vardhan Rao <quic_visr(a)quicinc.com> [ Upstream commit e5e7e398f6bb7918dab0612eb6991f7bae95520d ] When we run syzkaller we get below Out of Bound. "KASAN: slab-out-of-bounds Read in regcache_flat_read" Below is the backtrace of the issue: dump_backtrace+0x0/0x4c8 show_stack+0x34/0x44 dump_stack_lvl+0xd8/0x118 print_address_description+0x30/0x2d8 kasan_report+0x158/0x198 __asan_report_load4_noabort+0x44/0x50 regcache_flat_read+0x10c/0x110 regcache_read+0xf4/0x180 _regmap_read+0xc4/0x278 _regmap_update_bits+0x130/0x290 regmap_update_bits_base+0xc0/0x15c snd_soc_component_update_bits+0xa8/0x22c snd_soc_component_write_field+0x68/0xd4 tx_macro_digital_mute+0xec/0x140 Actually There is no need to have decimator with 32 bits. By limiting the variable with short type u8 issue is resolved. Signed-off-by: Ravulapati Vishnu Vardhan Rao <quic_visr(a)quicinc.com> Link: https://lore.kernel.org/r/20230304080702.609-1-quic_visr@quicinc.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/lpass-tx-macro.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index 2449a2df66df0..8facdb922f076 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -242,7 +242,7 @@ enum { struct tx_mute_work { struct tx_macro *tx; - u32 decimator; + u8 decimator; struct delayed_work dwork; }; @@ -635,7 +635,7 @@ static int tx_macro_mclk_enable(struct tx_macro *tx, return 0; } -static bool is_amic_enabled(struct snd_soc_component *component, int decimator) +static bool is_amic_enabled(struct snd_soc_component *component, u8 decimator) { u16 adc_mux_reg, adc_reg, adc_n; @@ -849,7 +849,7 @@ static int tx_macro_enable_dec(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); - unsigned int decimator; + u8 decimator; u16 tx_vol_ctl_reg, dec_cfg_reg, hpf_gate_reg, tx_gain_ctl_reg; u8 hpf_cut_off_freq; int hpf_delay = TX_MACRO_DMIC_HPF_DELAY_MS; @@ -1064,7 +1064,8 @@ static int tx_macro_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_component *component = dai->component; - u32 decimator, sample_rate; + u32 sample_rate; + u8 decimator; int tx_fs_rate; struct tx_macro *tx = snd_soc_component_get_drvdata(component); @@ -1128,7 +1129,7 @@ static int tx_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream) { struct snd_soc_component *component = dai->component; struct tx_macro *tx = snd_soc_component_get_drvdata(component); - u16 decimator; + u8 decimator; /* active decimator not set yet */ if (tx->active_decimator[dai->id] == -1) -- 2.39.2

1 0

[PATCH] ASoC: SOF: ipc4: Ensure DSP is in D0I0 during sof_ipc4_set_get_data()
by Peter Ujfalusi 22 Mar '23

22 Mar '23

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> The set_get_data() IPC op bypasses the check for the no_pm flag as done with the regular IPC tx_msg op. Since set_get_data should be performed when the DSP is in D0I0, set the DSP power state to D0I0 before sending the IPC's in sof_ipc4_set_get_data(). Fixes: ceb89acc4dc8 ("ASoC: SOF: ipc4: Add support for mandatory message handling functionality") Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/ipc4.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/sof/ipc4.c b/sound/soc/sof/ipc4.c index 8ede4b952997..246b56d24a6f 100644 --- a/sound/soc/sof/ipc4.c +++ b/sound/soc/sof/ipc4.c @@ -405,6 +405,9 @@ static int sof_ipc4_tx_msg(struct snd_sof_dev *sdev, void *msg_data, size_t msg_ static int sof_ipc4_set_get_data(struct snd_sof_dev *sdev, void *data, size_t payload_bytes, bool set) { + const struct sof_dsp_power_state target_state = { + .state = SOF_DSP_PM_D0, + }; size_t payload_limit = sdev->ipc->max_payload_size; struct sof_ipc4_msg *ipc4_msg = data; struct sof_ipc4_msg tx = {{ 0 }}; @@ -435,6 +438,11 @@ static int sof_ipc4_set_get_data(struct snd_sof_dev *sdev, void *data, tx.extension |= SOF_IPC4_MOD_EXT_MSG_FIRST_BLOCK(1); + /* ensure the DSP is in D0i0 before sending IPC */ + ret = snd_sof_dsp_set_power_state(sdev, &target_state); + if (ret < 0) + return ret; + /* Serialise IPC TX */ mutex_lock(&sdev->ipc->tx_mutex); -- 2.40.0

2 1

[PATCH v2] ASoC: simple-card: add comment to indicate don't remove platforms
by Kuninori Morimoto 22 Mar '23

22 Mar '23

Basically CPU and Platform are different Component, but if CPU is using soc-generic-dmaengine-pcm, same dev will be shared between CPU and Platform, and Simple Card had been supporting it. When we focus to clean up Simple Card driver, we tend to remove platforms if no Platform was selected, but it is wrong because of above reasons. This patch adds comment why we shouldn't remove platforms. In case of CPU is not using soc-generic-dmaengine-pcm, CPU and Platform will be duplicated, but it will be ignored by snd_soc_rtd_add_component(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - add +1 comment on asoc_simple_canonicalize_platform() sound/soc/generic/audio-graph-card.c | 13 ++++++++++++- sound/soc/generic/audio-graph-card2.c | 17 +++++++++++++++++ sound/soc/generic/simple-card-utils.c | 11 ++++++++++- sound/soc/generic/simple-card.c | 22 +++++++++++++++++++++- 4 files changed, 60 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index d788f5f23a8a..4e85536a1b26 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -613,10 +613,16 @@ static int graph_count_noml(struct asoc_simple_priv *priv, return -EINVAL; } + /* + * DON'T REMOVE platforms + * see + * simple-card.c :: simple_count_noml() + */ li->num[li->link].cpus = 1; - li->num[li->link].codecs = 1; li->num[li->link].platforms = 1; + li->num[li->link].codecs = 1; + li->link += 1; /* 1xCPU-Codec */ dev_dbg(dev, "Count As Normal\n"); @@ -637,6 +643,11 @@ static int graph_count_dpcm(struct asoc_simple_priv *priv, } if (li->cpu) { + /* + * DON'T REMOVE platforms + * see + * simple-card.c :: simple_count_noml() + */ li->num[li->link].cpus = 1; li->num[li->link].platforms = 1; diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 259544f64df9..d145b74d41e4 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -1046,8 +1046,14 @@ static int graph_count_normal(struct asoc_simple_priv *priv, * => lnk: port { endpoint { .. }; }; * }; */ + /* + * DON'T REMOVE platforms + * see + * simple-card.c :: simple_count_noml() + */ li->num[li->link].cpus = li->num[li->link].platforms = graph_counter(cpu_port); + li->num[li->link].codecs = graph_counter(codec_port); of_node_put(cpu_ep); @@ -1079,6 +1085,11 @@ static int graph_count_dpcm(struct asoc_simple_priv *priv, */ if (asoc_graph_is_ports0(lnk)) { + /* + * DON'T REMOVE platforms + * see + * simple-card.c :: simple_count_noml() + */ li->num[li->link].cpus = graph_counter(rport); /* FE */ li->num[li->link].platforms = graph_counter(rport); } else { @@ -1113,8 +1124,14 @@ static int graph_count_c2c(struct asoc_simple_priv *priv, * }; * }; */ + /* + * DON'T REMOVE platforms + * see + * simple-card.c :: simple_count_noml() + */ li->num[li->link].cpus = li->num[li->link].platforms = graph_counter(codec0); + li->num[li->link].codecs = graph_counter(codec1); of_node_put(ports); diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 56552a616f21..34a9b91e676f 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -638,7 +638,16 @@ EXPORT_SYMBOL_GPL(asoc_simple_dai_init); void asoc_simple_canonicalize_platform(struct snd_soc_dai_link_component *platforms, struct snd_soc_dai_link_component *cpus) { - /* Assumes platform == cpu */ + /* + * Assumes Platform == CPU + * + * Some CPU might be using soc-generic-dmaengine-pcm. This means CPU and Platform + * are different Component, but are sharing same component->dev. + * + * Let's assume Platform is same as CPU if it doesn't identify Platform on DT. + * see + * simple-card.c :: simple_count_noml() + */ if (!platforms->of_node) platforms->of_node = cpus->of_node; } diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index e98932c16754..fcd6c44f89f6 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -509,10 +509,25 @@ static int simple_count_noml(struct asoc_simple_priv *priv, return -EINVAL; } + /* + * DON'T REMOVE platforms + * + * Some CPU might be using soc-generic-dmaengine-pcm. This means CPU and Platform + * are different Component, but are sharing same component->dev. + * Simple Card had been supported it without special Platform selection. + * We need platforms here. + * + * In case of no Platform, it will be Platform == CPU, but Platform will be + * ignored by snd_soc_rtd_add_component(). + * + * see + * simple-card-utils.c :: asoc_simple_canonicalize_platform() + */ li->num[li->link].cpus = 1; - li->num[li->link].codecs = 1; li->num[li->link].platforms = 1; + li->num[li->link].codecs = 1; + li->link += 1; return 0; @@ -531,6 +546,11 @@ static int simple_count_dpcm(struct asoc_simple_priv *priv, } if (li->cpu) { + /* + * DON'T REMOVE platforms + * see + * simple_count_noml() + */ li->num[li->link].cpus = 1; li->num[li->link].platforms = 1; -- 2.25.1

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[PATCH] ASoC: Intel: sof_rt5682: Remove conditional dpcm_capture setting
by Peter Ujfalusi 22 Mar '23

22 Mar '23

From: Yong Zhi <yong.zhi(a)intel.com> The dpcm_capture is set unconditionally, we can drop the conditional setting of it. Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/intel/boards/sof_rt5682.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 2eec32846078..76ee2e35e1ef 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -793,7 +793,6 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, } links[id].init = max_98390_spk_codec_init; links[id].ops = &max_98390_ops; - links[id].dpcm_capture = 1; } else { max_98357a_dai_link(&links[id]); -- 2.40.0

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[PATCH 0/2] ASoC: Intel: sof_rt5682: Updates for MTL BT offload
by Peter Ujfalusi 22 Mar '23

22 Mar '23

Hi, Enable BT offload for Rex variants and RVP. Regards, Peter --- Uday M Bhat (2): ASoC: Intel: sof_rt5682: Update BT offload config for Rex ASoC: Intel: sof_rt5682: Update BT offload config for MTL RVP sound/soc/intel/boards/sof_rt5682.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) -- 2.40.0

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