April 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH 00/11] ASoC: SOF: Miscellaneous fixes for IPC and trace
by Ranjani Sridharan 05 Apr '22

05 Apr '22

This series includes some miscellaenous fixes pertaining to IPC, DMA buffer and DMA trace. Libin Yang (2): ASoC: SOF: extend the interface to stop DMA trace ASoC: SOF: disable dma trace in s0ix Peter Ujfalusi (8): ASoC: SOF: Intel: hda-loader: Use local snd_dma_buffer ASoC: SOF: Remove dmab and dmab_bdl from snd_sof_dev struct ASoC: SOF: ipc: Add max_payload_size field in struct snd_sof_ipc ASoC: SOF: ipc: Use ipc->max_payload_size for message/reply size checking ASoC: SOF: Drop support for mapped control data ASoC: SOF: ipc: Use msg->reply_data directly in snd_sof_ipc_get_reply() ASoC: SOF: Drop 'header' parameter from tx_message() API ASoC: SOF: Remove header from struct snd_sof_ipc_msg tangmeng (1): ASoC: SOF: Remove redundant return statements sound/soc/sof/compress.c | 8 ++-- sound/soc/sof/debug.c | 2 +- sound/soc/sof/intel/hda-dsp.c | 8 ++-- sound/soc/sof/intel/hda-loader.c | 12 ++--- sound/soc/sof/intel/tgl.c | 6 +-- sound/soc/sof/ipc.c | 76 ++++++++++++-------------------- sound/soc/sof/ipc3-pcm.c | 8 ++-- sound/soc/sof/ipc3-topology.c | 25 +++++------ sound/soc/sof/ipc3.c | 4 +- sound/soc/sof/pcm.c | 5 +-- sound/soc/sof/pm.c | 20 ++++++--- sound/soc/sof/sof-audio.c | 3 -- sound/soc/sof/sof-audio.h | 1 - sound/soc/sof/sof-client.c | 2 +- sound/soc/sof/sof-priv.h | 26 ++++++----- sound/soc/sof/trace.c | 61 ++++++++++++++++--------- 16 files changed, 131 insertions(+), 136 deletions(-) -- 2.25.1

2 12

[PATCH v1 0/7] ASoC: wm8731: Overhaul of the driver
by Mark Brown 05 Apr '22

05 Apr '22

This series contains a bunch of fairly minor changes which overhaul and modernise the WM8731 driver, there should be no impact on the driver's functionality. Mark Brown (7): ASoC: wm8731: Update to modern DAI terminology ASoC: wm8731: Factor component init out of bus code ASoC: wm8731: Move regulator request into wm8731_init() ASoC: wm8731: Factor our MCLK and mutex initialisation ASoC: wm8731: Factor out the I2C and SPI bus code into separate modules ASoC: wm8731: Convert DT bindings to YAML format ARM: configs: Update multi_v5_defconfig for WM8731 bus refactoring .../devicetree/bindings/sound/wlf,wm8731.yaml | 97 +++++++ .../devicetree/bindings/sound/wm8731.txt | 27 -- arch/arm/configs/multi_v5_defconfig | 3 +- sound/soc/atmel/Kconfig | 4 +- sound/soc/au1x/Kconfig | 2 +- sound/soc/codecs/Kconfig | 18 +- sound/soc/codecs/Makefile | 4 + sound/soc/codecs/wm8731-i2c.c | 69 +++++ sound/soc/codecs/wm8731-spi.c | 59 ++++ sound/soc/codecs/wm8731.c | 266 +++--------------- sound/soc/codecs/wm8731.h | 27 ++ sound/soc/pxa/Kconfig | 4 +- 12 files changed, 323 insertions(+), 257 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8731.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wm8731.txt create mode 100644 sound/soc/codecs/wm8731-i2c.c create mode 100644 sound/soc/codecs/wm8731-spi.c base-commit: 0b88a659002151e354272a13ae29d8b795ef1b46 -- 2.30.2

4 18

[PATCH] ASoC: soc-pcm: use GFP_KERNEL when the code is sleepable
by Christophe JAILLET 05 Apr '22

05 Apr '22

At the kzalloc() call in dpcm_be_connect(), there is no spin lock involved. It's merely protected by card->pcm_mutex, instead. The spinlock is applied at the later call with snd_soc_pcm_stream_lock_irq() only for the list manipulations. (See it's *_irq(), not *_irqsave(); that means the context being sleepable at that point.) So, we can use GFP_KERNEL safely there. This patch revert commit d8a9c6e1f676 ("ASoC: soc-pcm: use GFP_ATOMIC for dpcm structure") which is no longer needed since commit b7898396f4bb ("ASoC: soc-pcm: Fix and cleanup DPCM locking"). Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- See discussion in the thread at: https://lore.kernel.org/all/ed322b8821fa787907c1a4cce879564d1281b69d.164233… --- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 9a954680d492..11c9853e9e80 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1214,7 +1214,7 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, be_substream->pcm->nonatomic = 1; } - dpcm = kzalloc(sizeof(struct snd_soc_dpcm), GFP_ATOMIC); + dpcm = kzalloc(sizeof(struct snd_soc_dpcm), GFP_KERNEL); if (!dpcm) return -ENOMEM; -- 2.32.0

2 1

[PATCH v2 0/6] ASRC support on Tegra186 and later
by Sameer Pujar 05 Apr '22

05 Apr '22

This series adds support for Asynchronous Sample Rate Converter (ASRC) module on Tegra186 and later generations of SoCs. ASRC is a client of AHUB. The driver and DT support is added to make it work with Tegra audio graph card. The module can be plugged into audio path using ALSA mixer controls. ASRC supports two modes of operation, where it gets the ratio info from SW and ratio detector module. Presently the support is added for SW mode. Changelog: ========== v1 -> v2: --------- * Updatated RPM resume call in ASRC driver. Specific register sequence programming is done before regcache sync. * Fixed return values of mixer control put() callbacks in ASRC driver. * No changes in other patches. Sameer Pujar (6): ASoC: tegra: Add binding doc for ASRC module ASoC: tegra: Add Tegra186 based ASRC driver ASoC: tegra: AHUB routes for ASRC module arm64: defconfig: Build Tegra ASRC module arm64: tegra: Add ASRC device on Tegra186 and later arm64: tegra: Enable ASRC on various platforms .../bindings/sound/nvidia,tegra186-asrc.yaml | 81 ++ .../bindings/sound/nvidia,tegra210-ahub.yaml | 4 + arch/arm64/boot/dts/nvidia/tegra186-p2771-0000.dts | 223 +++++ arch/arm64/boot/dts/nvidia/tegra186.dtsi | 7 + arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 223 +++++ .../arm64/boot/dts/nvidia/tegra194-p3509-0000.dtsi | 223 +++++ arch/arm64/boot/dts/nvidia/tegra194.dtsi | 8 + .../dts/nvidia/tegra234-p3737-0000+p3701-0000.dts | 223 +++++ arch/arm64/boot/dts/nvidia/tegra234.dtsi | 8 + arch/arm64/configs/defconfig | 1 + sound/soc/tegra/Kconfig | 12 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra186_asrc.c | 1046 ++++++++++++++++++++ sound/soc/tegra/tegra186_asrc.h | 112 +++ sound/soc/tegra/tegra210_ahub.c | 82 +- 15 files changed, 2254 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra186-asrc.yaml create mode 100644 sound/soc/tegra/tegra186_asrc.c create mode 100644 sound/soc/tegra/tegra186_asrc.h -- 2.7.4

2 7

[PATCH] ASoC: samsung: Constify snd_soc_dapm_{route,widget} structs
by Rikard Falkeborn 05 Apr '22

05 Apr '22

These structs only have their address assigned to the dapm_{routes,widget} fields in the snd_soc_card struct, both which are pointers to const data. Make them const to allow the compiler to put them in read-only memory. Signed-off-by: Rikard Falkeborn <rikard.falkeborn(a)gmail.com> --- sound/soc/samsung/bells.c | 4 ++-- sound/soc/samsung/littlemill.c | 4 ++-- sound/soc/samsung/lowland.c | 4 ++-- sound/soc/samsung/speyside.c | 4 ++-- sound/soc/samsung/tobermory.c | 4 ++-- 5 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c index 8b83f39c3ac9..76998a4a4cad 100644 --- a/sound/soc/samsung/bells.c +++ b/sound/soc/samsung/bells.c @@ -386,11 +386,11 @@ static struct snd_soc_codec_conf bells_codec_conf[] = { }, }; -static struct snd_soc_dapm_widget bells_widgets[] = { +static const struct snd_soc_dapm_widget bells_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), }; -static struct snd_soc_dapm_route bells_routes[] = { +static const struct snd_soc_dapm_route bells_routes[] = { { "Sub CLK_SYS", NULL, "OPCLK" }, { "CLKIN", NULL, "OPCLK" }, diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index 34067cc314ff..26c42892c059 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -228,7 +228,7 @@ static const struct snd_kcontrol_new controls[] = { SOC_DAPM_PIN_SWITCH("WM1250 Output"), }; -static struct snd_soc_dapm_widget widgets[] = { +static const struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("AMIC", NULL), @@ -239,7 +239,7 @@ static struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), }; -static struct snd_soc_dapm_route audio_paths[] = { +static const struct snd_soc_dapm_route audio_paths[] = { { "Headphone", NULL, "HPOUT1L" }, { "Headphone", NULL, "HPOUT1R" }, diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c index 7b12ccd2a9b2..8e4579fdcd7b 100644 --- a/sound/soc/samsung/lowland.c +++ b/sound/soc/samsung/lowland.c @@ -140,7 +140,7 @@ static const struct snd_kcontrol_new controls[] = { SOC_DAPM_PIN_SWITCH("Headphone"), }; -static struct snd_soc_dapm_widget widgets[] = { +static const struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -150,7 +150,7 @@ static struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_MIC("Main DMIC", NULL), }; -static struct snd_soc_dapm_route audio_paths[] = { +static const struct snd_soc_dapm_route audio_paths[] = { { "Sub IN1", NULL, "HPOUT2L" }, { "Sub IN2", NULL, "HPOUT2R" }, diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 37b1f4f60b21..1ae1a5aae72f 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -261,7 +261,7 @@ static const struct snd_kcontrol_new controls[] = { SOC_DAPM_PIN_SWITCH("Headphone"), }; -static struct snd_soc_dapm_widget widgets[] = { +static const struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -271,7 +271,7 @@ static struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_MIC("Main DMIC", NULL), }; -static struct snd_soc_dapm_route audio_paths[] = { +static const struct snd_soc_dapm_route audio_paths[] = { { "IN1RN", NULL, "MICB1" }, { "IN1RP", NULL, "MICB1" }, { "IN1RN", NULL, "MICB2" }, diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c index 8d3149a47a4c..4a56abfc243e 100644 --- a/sound/soc/samsung/tobermory.c +++ b/sound/soc/samsung/tobermory.c @@ -130,7 +130,7 @@ static const struct snd_kcontrol_new controls[] = { SOC_DAPM_PIN_SWITCH("DMIC"), }; -static struct snd_soc_dapm_widget widgets[] = { +static const struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -140,7 +140,7 @@ static struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_SPK("Main Speaker", NULL), }; -static struct snd_soc_dapm_route audio_paths[] = { +static const struct snd_soc_dapm_route audio_paths[] = { { "Headphone", NULL, "HPOUTL" }, { "Headphone", NULL, "HPOUTR" }, -- 2.35.1

2 1

[PATCH 0/6] ASoC: topology: Cleanup patches
by Amadeusz Sławiński 05 Apr '22

05 Apr '22

Following patches contain minor changes, cleaning up code to be easier to read. Clean up few loops, to be simpler or altogether remove them. Rename some things to make code easier to understand. Amadeusz Sławiński (6): ASoC: topology: Use for loop instead of while ASoC: topology: Remove unnecessary looping ASoC: topology: Return bool instead of int ASoC: topology: Rename SOC_TPLG_PASS_MIXER to _CONTROL ASoC: topology: Correct error message ASoC: topology: Rename soc_tplg_init_kcontrol() function sound/soc/soc-topology.c | 455 ++++++++++++++++++--------------------- 1 file changed, 208 insertions(+), 247 deletions(-) -- 2.25.1

4 9

[PATCH v2] soc: soc-dapm: fix two incorrect uses of list iterator
by Xiaomeng Tong 05 Apr '22

05 Apr '22

These two bug are here: list_for_each_entry_safe_continue(w, n, list, power_list); list_for_each_entry_safe_continue(w, n, list, power_list); After the list_for_each_entry_safe_continue() exits, the list iterator will always be a bogus pointer which point to an invalid struct objdect containing HEAD member. The funciton poniter 'w->event' will be a invalid value which can lead to a control-flow hijack if the 'w' can be controlled. The original intention was to continue the outer list_for_each_entry_safe() loop with the same entry if w->event is NULL, but misunderstanding the meaning of list_for_each_entry_safe_continue(). So just add a 'continue;' to fix the bug. Cc: stable(a)vger.kernel.org Fixes: 163cac061c973 ("ASoC: Factor out DAPM sequence execution") Signed-off-by: Xiaomeng Tong <xiam0nd.tong(a)gmail.com> --- changes since v1: - use continue statement instead (Mark Brown) v1:https://lore.kernel.org/lkml/20220327082138.13696-1-xiam0nd.tong@gmail.c… --- sound/soc/soc-dapm.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b06c5682445c..fb43b331a36e 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1687,8 +1687,7 @@ static void dapm_seq_run(struct snd_soc_card *card, switch (w->id) { case snd_soc_dapm_pre: if (!w->event) - list_for_each_entry_safe_continue(w, n, list, - power_list); + continue; if (event == SND_SOC_DAPM_STREAM_START) ret = w->event(w, @@ -1700,8 +1699,7 @@ static void dapm_seq_run(struct snd_soc_card *card, case snd_soc_dapm_post: if (!w->event) - list_for_each_entry_safe_continue(w, n, list, - power_list); + continue; if (event == SND_SOC_DAPM_STREAM_START) ret = w->event(w, -- 2.17.1

2 1

[PATCH] codecs: rt5682s: fix an incorrect NULL check on list iterator
by Xiaomeng Tong 05 Apr '22

05 Apr '22

The bug is here: if (!dai) { The list iterator value 'dai' will *always* be set and non-NULL by for_each_component_dais(), so it is incorrect to assume that the iterator value will be NULL if the list is empty or no element is found (In fact, it will be a bogus pointer to an invalid struct object containing the HEAD). Otherwise it will bypass the check 'if (!dai) {' (never call dev_err() and never return -ENODEV;) and lead to invalid memory access lately when calling 'rt5682s_set_bclk1_ratio(dai, factor);'. To fix the bug, just return rt5682s_set_bclk1_ratio(dai, factor); when found the 'dai', otherwise dev_err() and return -ENODEV; Cc: stable(a)vger.kernel.org Fixes: bdd229ab26be9 ("ASoC: rt5682s: Add driver for ALC5682I-VS codec") Signed-off-by: Xiaomeng Tong <xiam0nd.tong(a)gmail.com> --- sound/soc/codecs/rt5682s.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index 1e662d1be2b3..c890e51ff953 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -2686,14 +2686,11 @@ static int rt5682s_bclk_set_rate(struct clk_hw *hw, unsigned long rate, for_each_component_dais(component, dai) if (dai->id == RT5682S_AIF1) - break; - if (!dai) { - dev_err(component->dev, "dai %d not found in component\n", - RT5682S_AIF1); - return -ENODEV; - } + return rt5682s_set_bclk1_ratio(dai, factor); - return rt5682s_set_bclk1_ratio(dai, factor); + dev_err(component->dev, "dai %d not found in component\n", + RT5682S_AIF1); + return -ENODEV; } static const struct clk_ops rt5682s_dai_clk_ops[RT5682S_DAI_NUM_CLKS] = { -- 2.17.1

2 1

[PATCH] codecs: rt5682: fix an incorrect NULL check on list iterator
by Xiaomeng Tong 05 Apr '22

05 Apr '22

The bug is here: if (!dai) { The list iterator value 'dai' will *always* be set and non-NULL by for_each_component_dais(), so it is incorrect to assume that the iterator value will be NULL if the list is empty or no element is found (In fact, it will be a bogus pointer to an invalid struct object containing the HEAD). Otherwise it will bypass the check 'if (!dai) {' (never call dev_err() and never return -ENODEV;) and lead to invalid memory access lately when calling 'rt5682_set_bclk1_ratio(dai, factor);'. To fix the bug, just return rt5682_set_bclk1_ratio(dai, factor); when found the 'dai', otherwise dev_err() and return -ENODEV; Cc: stable(a)vger.kernel.org Fixes: ebbfabc16d23d ("ASoC: rt5682: Add CCF usage for providing I2S clks") Signed-off-by: Xiaomeng Tong <xiam0nd.tong(a)gmail.com> --- sound/soc/codecs/rt5682.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index be68d573a490..c9ff9c89adf7 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2822,14 +2822,11 @@ static int rt5682_bclk_set_rate(struct clk_hw *hw, unsigned long rate, for_each_component_dais(component, dai) if (dai->id == RT5682_AIF1) - break; - if (!dai) { - dev_err(rt5682->i2c_dev, "dai %d not found in component\n", - RT5682_AIF1); - return -ENODEV; - } + return rt5682_set_bclk1_ratio(dai, factor); - return rt5682_set_bclk1_ratio(dai, factor); + dev_err(rt5682->i2c_dev, "dai %d not found in component\n", + RT5682_AIF1); + return -ENODEV; } static const struct clk_ops rt5682_dai_clk_ops[RT5682_DAI_NUM_CLKS] = { -- 2.17.1

2 1

[PATCH v3 1/2] ASoC: cs35l41: Add one more variable in the debug log
by Hui Wang 05 Apr '22

05 Apr '22

otp_map[].size is a key variable to compute the value of otp_val and to update the bit_offset, it is helpful to debug if could put it in the debug log. Reviewed-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/soc/codecs/cs35l41-lib.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs35l41-lib.c b/sound/soc/codecs/cs35l41-lib.c index e5a56bcbb223..810643be13a4 100644 --- a/sound/soc/codecs/cs35l41-lib.c +++ b/sound/soc/codecs/cs35l41-lib.c @@ -822,8 +822,8 @@ int cs35l41_otp_unpack(struct device *dev, struct regmap *regmap) word_offset = otp_map_match->word_offset; for (i = 0; i < otp_map_match->num_elements; i++) { - dev_dbg(dev, "bitoffset= %d, word_offset=%d, bit_sum mod 32=%d\n", - bit_offset, word_offset, bit_sum % 32); + dev_dbg(dev, "bitoffset= %d, word_offset=%d, bit_sum mod 32=%d, otp_map[i].size = %u\n", + bit_offset, word_offset, bit_sum % 32, otp_map[i].size); if (bit_offset + otp_map[i].size - 1 >= 32) { otp_val = (otp_mem[word_offset] & GENMASK(31, bit_offset)) >> bit_offset; -- 2.25.1

3 3