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[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

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[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

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[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

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[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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[PATCH] ASoC: tegra: remove unneeded semicolon
by Jiapeng Chong 22 Mar '23

22 Mar '23

./sound/soc/tegra/tegra_asoc_machine.c:206:3-4: Unneeded semicolon. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Link: https://bugzilla.openanolis.cn/show_bug.cgi?id=4583 Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/soc/tegra/tegra_asoc_machine.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra_asoc_machine.c b/sound/soc/tegra/tegra_asoc_machine.c index c2aaa496468c..f5092b410926 100644 --- a/sound/soc/tegra/tegra_asoc_machine.c +++ b/sound/soc/tegra/tegra_asoc_machine.c @@ -203,7 +203,7 @@ int tegra_asoc_machine_init(struct snd_soc_pcm_runtime *rtd) "nvidia,coupled-mic-hp-det")) { tegra_machine_mic_jack_gpio.desc = machine->gpiod_hp_det; tegra_machine_mic_jack_gpio.jack_status_check = coupled_mic_hp_check; - }; + } err = snd_soc_jack_add_gpios(&tegra_machine_mic_jack, 1, &tegra_machine_mic_jack_gpio); -- 2.20.1.7.g153144c

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[PATCH] ALSA: hda/realtek: Fix support for Dell Precision 3260
by Jaroslav Kysela 22 Mar '23

22 Mar '23

Unfortunately, in commit 5911d78fabbb a wrong codec patch was selected. The model=alc283-dac-wcaps is equivalent to ALC283_FIXUP_CHROME_BOOK not ALC295_FIXUP_CHROME_BOOK. Fixes: 5911d78fabbb ("ALSA: hda/realtek: Improve support for Dell Precision 3260") Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f09a1d7c1b18..3a304010bc9f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9260,7 +9260,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a62, "Dell Precision 5560", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0a9d, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0a9e, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0ac9, "Dell Precision 3260", ALC295_FIXUP_CHROME_BOOK), + SND_PCI_QUIRK(0x1028, 0x0ac9, "Dell Precision 3260", ALC283_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x1028, 0x0b19, "Dell XPS 15 9520", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0b1a, "Dell Precision 5570", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0b37, "Dell Inspiron 16 Plus 7620 2-in-1", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), -- 2.39.2

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[PATCH v2 0/3] ASoC: SOF: ipc4/intel: Support for ChainDMA
by Peter Ujfalusi 22 Mar '23

22 Mar '23

Hi, Changes since v1: - Fix missed mutex unlocking in sof_ipc4_widget_free() Reported by Dan Carpenter (thank you!) On a platform when the DSP is in use, we cannot select individual links to use or not use the DSP, it is either all or none. On some audio endpoint, like HDMI/DP, it is preferred to not use any processing in DSP to reduce the latency and to allow bytestream pass-through (DTS, DD, etc) IPC4 introduces a new type of end-to-end connection within the DSP which is using the host DMA and link DMA in a single buffer, working back-to-back, passing the received data without looking at it or trying to understand the format, content. This mode reduces the latency and allows non PCM streams to be sent from userspace. The feature is enabled per PCM bases, signalled in topology. Regards, Peter --- *** BLURB HERE *** Jyri Sarha (2): ASoC: SOF: ipc4: Add macros for chain-dma message bits ASoC: SOF: ipc4/intel: Add support for chained DMA Ranjani Sridharan (1): ASoC: SOF: topology: Set pipeline widget before updating IPC structures include/sound/sof/ipc4/header.h | 29 +++++++ include/uapi/sound/sof/tokens.h | 1 + sound/soc/sof/intel/hda-dai-ops.c | 18 ++++- sound/soc/sof/ipc4-pcm.c | 122 +++++++++++++++++++++++++++++- sound/soc/sof/ipc4-topology.c | 122 +++++++++++++++++++++++++++++- sound/soc/sof/ipc4-topology.h | 2 + sound/soc/sof/topology.c | 48 ++++++------ 7 files changed, 310 insertions(+), 32 deletions(-) -- 2.40.0

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USB-Audio: Add Sony inzone H7/H9 headset default/HiFi usecase
by GitHub pull_request - opened 22 Mar '23

22 Mar '23

alsa-project/alsa-ucm-conf pull request #293 was opened from laurensmiers: The Sony inzone H7/H9 headset has two usb audio outputs and one usb audio input. One pair of audio output and input is the 'chat' interface which is normally connected to a chat client. The remaining audio ouput is the 'game' interface, which is normally connected to the actual game you're playing. The 'game' interface lists as a separate USB subdevice in alsa. This commit adds the HiFi usecase for the Sony headset which is as follows: - One chat device - One audio output - One audio input - One game device - One audio output Tested with pipewire and pulseaudio, verified with pavucontrol, both devices are listed and audio can be played on both. The audio is mixed in the headset itself. Control wise there isn't much, the only thing that is reported in alsamixer is a volume control for the CAPTURE channel. However, changing the volume of this does not seem to affect the actual audio that is recorded, so I don't see a need to change this when enabling the device. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/293 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/293.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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