May 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH AUTOSEL 5.12 34/37] ALSA: hda: generic: change the DAC ctl name for LO+SPK or LO+HP
by Sasha Levin 12 May '21

12 May '21

From: Hui Wang <hui.wang(a)canonical.com> [ Upstream commit f48652bbe3ae62ba2835a396b7e01f063e51c4cd ] Without this change, the DAC ctl's name could be changed only when the machine has both Speaker and Headphone, but we met some machines which only has Lineout and Headhpone, and the Lineout and Headphone share the Audio Mixer0 and DAC0, the ctl's name is set to "Front". On most of machines, the "Front" is used for Speaker only or Lineout only, but on this machine it is shared by Lineout and Headphone, This introduces an issue in the pipewire and pulseaudio, suppose users want the Headphone to be on and the Speaker/Lineout to be off, they could turn off the "Front", this works on most of the machines, but on this machine, the "Front" couldn't be turned off otherwise the headphone will be off too. Here we do some change to let the ctl's name change to "Headphone+LO" on this machine, and pipewire and pulseaudio already could handle "Headphone+LO" and "Speaker+LO". (https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/747) BugLink: http://bugs.launchpad.net/bugs/804178 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Link: https://lore.kernel.org/r/20210504073917.22406-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_generic.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index f5cba7afd1c6..ff0fb2d16d82 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -1202,11 +1202,17 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch, *index = ch; return "Headphone"; case AUTO_PIN_LINE_OUT: - /* This deals with the case where we have two DACs and - * one LO, one HP and one Speaker */ - if (!ch && cfg->speaker_outs && cfg->hp_outs) { - bool hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type); - bool spk_lo_shared = !path_has_mixer(codec, spec->speaker_paths[0], ctl_type); + /* This deals with the case where one HP or one Speaker or + * one HP + one Speaker need to share the DAC with LO + */ + if (!ch) { + bool hp_lo_shared = false, spk_lo_shared = false; + + if (cfg->speaker_outs) + spk_lo_shared = !path_has_mixer(codec, + spec->speaker_paths[0], ctl_type); + if (cfg->hp_outs) + hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type); if (hp_lo_shared && spk_lo_shared) return spec->vmaster_mute.hook ? "PCM" : "Master"; if (hp_lo_shared) -- 2.30.2

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[PATCH] ASoC: wm_adsp: mark more data structures with the const qualifier
by Simon Trimmer 12 May '21

12 May '21

The callback structures and memory region type table can be marked as const as they will not change during use. Fix checkpatch warning against wm_adsp_find_region function by moving const keyword to form the 'static const struct' pattern. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 18 +++++++++--------- sound/soc/codecs/wm_adsp.h | 2 +- 2 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 3dc119daf2f6..37aa020f23f6 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -303,9 +303,9 @@ #define HALO_MPU_VIO_ERR_SRC_MASK 0x00007fff #define HALO_MPU_VIO_ERR_SRC_SHIFT 0 -static struct wm_adsp_ops wm_adsp1_ops; -static struct wm_adsp_ops wm_adsp2_ops[]; -static struct wm_adsp_ops wm_halo_ops; +static const struct wm_adsp_ops wm_adsp1_ops; +static const struct wm_adsp_ops wm_adsp2_ops[]; +static const struct wm_adsp_ops wm_halo_ops; struct wm_adsp_buf { struct list_head list; @@ -824,7 +824,7 @@ const struct soc_enum wm_adsp_fw_enum[] = { }; EXPORT_SYMBOL_GPL(wm_adsp_fw_enum); -static struct wm_adsp_region const *wm_adsp_find_region(struct wm_adsp *dsp, +static const struct wm_adsp_region *wm_adsp_find_region(struct wm_adsp *dsp, int type) { int i; @@ -2240,7 +2240,7 @@ static void wmfw_v3_parse_id_header(struct wm_adsp *dsp, } static int wm_adsp_create_regions(struct wm_adsp *dsp, __be32 id, int nregions, - int *type, __be32 *base) + const int *type, __be32 *base) { struct wm_adsp_alg_region *alg_region; int i; @@ -2487,7 +2487,7 @@ static int wm_adsp2_setup_algs(struct wm_adsp *dsp) static int wm_halo_create_regions(struct wm_adsp *dsp, __be32 id, __be32 xm_base, __be32 ym_base) { - int types[] = { + static const int types[] = { WMFW_ADSP2_XM, WMFW_HALO_XM_PACKED, WMFW_ADSP2_YM, WMFW_HALO_YM_PACKED }; @@ -4500,13 +4500,13 @@ irqreturn_t wm_halo_wdt_expire(int irq, void *data) } EXPORT_SYMBOL_GPL(wm_halo_wdt_expire); -static struct wm_adsp_ops wm_adsp1_ops = { +static const struct wm_adsp_ops wm_adsp1_ops = { .validate_version = wm_adsp_validate_version, .parse_sizes = wm_adsp1_parse_sizes, .region_to_reg = wm_adsp_region_to_reg, }; -static struct wm_adsp_ops wm_adsp2_ops[] = { +static const struct wm_adsp_ops wm_adsp2_ops[] = { { .sys_config_size = sizeof(struct wm_adsp_system_config_xm_hdr), .parse_sizes = wm_adsp2_parse_sizes, @@ -4567,7 +4567,7 @@ static struct wm_adsp_ops wm_adsp2_ops[] = { }, }; -static struct wm_adsp_ops wm_halo_ops = { +static const struct wm_adsp_ops wm_halo_ops = { .sys_config_size = sizeof(struct wm_halo_system_config_xm_hdr), .parse_sizes = wm_adsp2_parse_sizes, .validate_version = wm_halo_validate_version, diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 1996350b817e..f22131d9cc29 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -64,7 +64,7 @@ struct wm_adsp { struct regmap *regmap; struct snd_soc_component *component; - struct wm_adsp_ops *ops; + const struct wm_adsp_ops *ops; unsigned int base; unsigned int base_sysinfo; -- 2.31.1

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[PATCH] Revert "ASoC: q6dsp: q6afe: remove unneeded dead-store initialization"
by Nathan Chancellor 12 May '21

12 May '21

This reverts commit 5f1b95d08de712327e452d082a50fded435ec884. The warnings that commit 5f1b95d08de7 ("ASoC: q6dsp: q6afe: remove unneeded dead-store initialization") was trying to fix were already fixed in commit 12900bacb4f3 ("ASoC: qcom: q6afe: remove useless assignments"). With both commits in the tree, port_id is uninitialized, as pointed out by clang: sound/soc/qcom/qdsp6/q6afe.c:1213:18: warning: variable 'port_id' is uninitialized when used here [-Wuninitialized] stop->port_id = port_id; ^~~~~~~ sound/soc/qcom/qdsp6/q6afe.c:1186:13: note: initialize the variable 'port_id' to silence this warning int port_id; ^ = 0 1 warning generated. Bring back the initialization so that everything works as intended. Fixes: 5f1b95d08de7 ("ASoC: q6dsp: q6afe: remove unneeded dead-store initialization") Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Nathan Chancellor <nathan(a)kernel.org> --- sound/soc/qcom/qdsp6/q6afe.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index c5c1818a6f75..729d27da0447 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -1183,7 +1183,7 @@ int q6afe_port_stop(struct q6afe_port *port) struct afe_port_cmd_device_stop *stop; struct q6afe *afe = port->afe; struct apr_pkt *pkt; - int port_id; + int port_id = port->id; int ret = 0; int index, pkt_size; void *p; base-commit: 4ac9b48adf4d561d0e33419d548278f205dd70b5 -- 2.31.1.527.g2d677e5b15

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[PATCH] ASoC: cs42l42: make HSBIAS_SENSE_EN optional
by Vitaly Rodionov 12 May '21

12 May '21

HSBIAS_SENSE_EN configures HSBIAS output current sense through the external 2.21-k resistor. HSBIAS_SENSE is hardware feature to reduce the potential pop noise during the headset plug out slowly. But on some platforms ESD voltage will affect it causing test to fail, especially with CTIA headset type. For different hardware setups, a designer might want to tweak default behavior. Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- Documentation/devicetree/bindings/sound/cs42l42.txt | 7 +++++++ sound/soc/codecs/cs42l42.c | 7 ++++++- sound/soc/codecs/cs42l42.h | 1 + 3 files changed, 14 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/cs42l42.txt b/Documentation/devicetree/bindings/sound/cs42l42.txt index 7dfaa2ab906f..5d416fdaf023 100644 --- a/Documentation/devicetree/bindings/sound/cs42l42.txt +++ b/Documentation/devicetree/bindings/sound/cs42l42.txt @@ -81,6 +81,13 @@ Optional properties: < x1 x2 x3 x4 > Default = < 15 8 4 1> + - cirrus,hs-bias-sense-disable: This is boolean property. If present the + HSBIAS sense is disabled. Configures HSBIAS output current sense through + the external 2.21-k resistor. HSBIAS_SENSE is hardware feature to reduce + the potential pop noise during the headset plug out slowly. But on some + platforms ESD voltage will affect it causing test to fail, especially + with CTIA headset type. For different hardware setups, a designer might + want to tweak default behavior. Example: diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index d7fb6b38fd7c..5087c5b781f8 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -1033,7 +1033,7 @@ static void cs42l42_process_hs_type_detect(struct cs42l42_private *cs42l42) CS42L42_AUTO_HSBIAS_HIZ_MASK | CS42L42_TIP_SENSE_EN_MASK | CS42L42_HSBIAS_SENSE_TRIP_MASK, - (1 << CS42L42_HSBIAS_SENSE_EN_SHIFT) | + (cs42l42->hs_bias_sense_en << CS42L42_HSBIAS_SENSE_EN_SHIFT) | (1 << CS42L42_AUTO_HSBIAS_HIZ_SHIFT) | (0 << CS42L42_TIP_SENSE_EN_SHIFT) | (3 << CS42L42_HSBIAS_SENSE_TRIP_SHIFT)); @@ -1808,6 +1808,11 @@ static int cs42l42_handle_device_data(struct device *dev, (cs42l42->hs_bias_ramp_rate << CS42L42_HSBIAS_RAMP_SHIFT)); + if (device_property_read_bool(dev, "cirrus,hs-bias-sense-disable")) + cs42l42->hs_bias_sense_en = 0; + else + cs42l42->hs_bias_sense_en = 1; + return 0; } diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 2e0d3836bd7e..24f7be228d5f 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -787,6 +787,7 @@ struct cs42l42_private { u8 bias_thresholds[CS42L42_NUM_BIASES]; u8 hs_bias_ramp_rate; u8 hs_bias_ramp_time; + u8 hs_bias_sense_en; u8 stream_use; }; -- 2.25.1

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[PATCH 0/9] ASoC: SOF/Intel: fix and remove Sparse warnings
by Pierre-Louis Bossart 12 May '21

12 May '21

We updated our SOF CI settings last week (see below) to use more options for Sparse, and sure enough it detected one nasty single-character bug in one of my previous patches, along with more trivial issues with string/integer sizes and signed/unsigned confusions. export ARCH=x86_64 export CF="-Wsparse-error -Wsparse-all -Wno-bitwise-pointer -Wno-pointer-arith -Wno-typesign -Wnoshadow" make -k sound/soc/sof/ C=2 make -k sound/soc/intel/common/ C=2 make -k sound/soc/intel/boards/ C=2 make -k drivers/soundwire/ C=2 Pierre-Louis Bossart (9): ASoC: SOF: Intel: hda: fix index used in inner loop ASoC: Intel: sof_rt5682: shrink platform_id names below 20 characters ASoC: Intel: sof_cs42l42: shrink platform id below 20 characters ASoC: Intel: bxt_da7219_max98357a: shrink platform_id below 20 characters ASoC: Intel: glk_rt5682_max98357a: shrink platform_id below 20 characters ASoC: Intel: kbl_da7219_max98357a: shrink platform_id below 20 characters ASoC: Intel: sof_da7219_max98373: shrink platform_id below 20 characters ASoC: Intel: sof_sdw: fix signed/unsigned warning ASoC: Intel: soc-acpi: add ull suffix for SoundWire _ADR values sound/soc/intel/boards/bxt_da7219_max98357a.c | 12 +++--- sound/soc/intel/boards/glk_rt5682_max98357a.c | 4 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 4 +- sound/soc/intel/boards/sof_cs42l42.c | 2 +- sound/soc/intel/boards/sof_da7219_max98373.c | 8 ++-- sound/soc/intel/boards/sof_rt5682.c | 20 +++++----- sound/soc/intel/boards/sof_sdw.c | 2 +- .../intel/common/soc-acpi-intel-adl-match.c | 28 +++++++------- .../intel/common/soc-acpi-intel-bxt-match.c | 2 +- .../intel/common/soc-acpi-intel-cml-match.c | 22 +++++------ .../intel/common/soc-acpi-intel-cnl-match.c | 2 +- .../intel/common/soc-acpi-intel-glk-match.c | 6 +-- .../intel/common/soc-acpi-intel-icl-match.c | 12 +++--- .../intel/common/soc-acpi-intel-jsl-match.c | 6 +-- .../intel/common/soc-acpi-intel-kbl-match.c | 2 +- .../intel/common/soc-acpi-intel-tgl-match.c | 38 +++++++++---------- sound/soc/sof/intel/hda.c | 2 +- 17 files changed, 86 insertions(+), 86 deletions(-) -- 2.25.1

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[PATCH] ASoC: soc-dai.h: Align the word of comment for SND_SOC_DAIFMT_CBC_CFC
by Kuninori Morimoto 12 May '21

12 May '21

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Let's use "consumer" instead of "follower". Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark I think this is just typo. include/sound/soc-dai.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 1358a0ceb4d0..0bc29c4516e7 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -81,7 +81,7 @@ struct snd_compr_stream; #define SND_SOC_DAIFMT_CBP_CFP (1 << 12) /* codec clk provider & frame provider */ #define SND_SOC_DAIFMT_CBC_CFP (2 << 12) /* codec clk consumer & frame provider */ #define SND_SOC_DAIFMT_CBP_CFC (3 << 12) /* codec clk provider & frame consumer */ -#define SND_SOC_DAIFMT_CBC_CFC (4 << 12) /* codec clk consumer & frame follower */ +#define SND_SOC_DAIFMT_CBC_CFC (4 << 12) /* codec clk consumer & frame consumer */ /* previous definitions kept for backwards-compatibility, do not use in new contributions */ #define SND_SOC_DAIFMT_CBM_CFM SND_SOC_DAIFMT_CBP_CFP -- 2.25.1

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[PATCH -next] ASoC: sti-sas: add missing MODULE_DEVICE_TABLE
by Zou Wei 12 May '21

12 May '21

This patch adds missing MODULE_DEVICE_TABLE definition which generates correct modalias for automatic loading of this driver when it is built as an external module. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Zou Wei <zou_wei(a)huawei.com> --- sound/soc/codecs/sti-sas.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c index ffdf7e5..82a24e3 100644 --- a/sound/soc/codecs/sti-sas.c +++ b/sound/soc/codecs/sti-sas.c @@ -408,6 +408,7 @@ static const struct of_device_id sti_sas_dev_match[] = { }, {}, }; +MODULE_DEVICE_TABLE(of, sti_sas_dev_match); static int sti_sas_driver_probe(struct platform_device *pdev) { -- 2.6.2

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[PATCH] ASoC: wm2200: remove include of wmfw.h
by Simon Trimmer 12 May '21

12 May '21

We want all wm_adsp clients to use the wm_adsp.h header as they shouldn't need to include internal sub-headers. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- sound/soc/codecs/wm2200.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index b0a6d31299bb..c35673e7f420 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -30,7 +30,6 @@ #include <sound/wm2200.h> #include "wm2200.h" -#include "wmfw.h" #include "wm_adsp.h" #define WM2200_DSP_CONTROL_1 0x00 -- 2.31.1

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[PATCH] ALSA: pcm: Need to check whether runtime is valid or not
by Chanho Park 12 May '21

12 May '21

From: eunwoo kim <ew.kim(a)samsung.com> Since snd_pcm_ioctl_sw_params_compat has no runtime variable check, if application call the ioctl after close, it can make kernel crash. So, snd_pcm_ioctl_sw_params_compat needs to check the runtime variable at the beginning of the function. Signed-off-by: eunwoo kim <ew.kim(a)samsung.com> Signed-off-by: Chanho Park <chanho61.park(a)samsung.com> --- sound/core/pcm_compat.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 590a46a9e78d..ff0de4252ff4 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -90,6 +90,9 @@ static int snd_pcm_ioctl_sw_params_compat(struct snd_pcm_substream *substream, snd_pcm_uframes_t boundary; int err; + if (!substream->runtime) + return -ENOTTY; + memset(&params, 0, sizeof(params)); if (get_user(params.tstamp_mode, &src->tstamp_mode) || get_user(params.period_step, &src->period_step) || -- 2.31.1

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[PATCH 1/5] ASoC: cs53l30: Add missing regmap use_single config
by Charles Keepax 11 May '21

11 May '21

This device requires single register transactions, this will definely cause problems with the new device ID parsing which uses regmap_bulk_read but might also show up in the cache sync sometimes. Add the missing flags to the regmap_config. Fixes: 4fc81bc88ad9 ("ASoC: cs53l30: Minor error paths fixups") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/cs53l30.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index bd33dd048c7ce..f2087bd38dbc8 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -913,6 +913,9 @@ static struct regmap_config cs53l30_regmap = { .writeable_reg = cs53l30_writeable_register, .readable_reg = cs53l30_readable_register, .cache_type = REGCACHE_RBTREE, + + .use_single_read = true, + .use_single_write = true, }; static int cs53l30_i2c_probe(struct i2c_client *client, -- 2.11.0

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