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

[PATCH 0/3] ASoC: mediatek:mt8186: fix both the speaker and hdmi
by Jiaxin Yu 29 Nov '22

29 Nov '22

The speaker and hdmi of mt8186 platform are shared the same port of I2S, when connect the external display, use build-in speakers to play audio, they both playback at the same time. So we want to manage the playback device through DAPM events. Jiaxin Yu (3): drm: bridge: it6505: bridge to hdmi-codec ASoC: hdmi-codec: Add event handler for hdmi TX ASoC: mediatek: mt8186: correct the HDMI widgets drivers/gpu/drm/bridge/ite-it6505.c | 36 +++++++++++++++--- include/sound/hdmi-codec.h | 6 +++ sound/soc/codecs/hdmi-codec.c | 37 +++++++++++++++++-- .../mt8186/mt8186-mt6366-da7219-max98357.c | 2 +- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 2 +- 5 files changed, 71 insertions(+), 12 deletions(-) -- 2.18.0

2 4

[PATCH 00/11] ASoC: Intel: avs: Refactor jack handling
by Cezary Rojewski 29 Nov '22

29 Nov '22

For all the boards included in this patchset, a similar set of changes is proposed: 1) Move jack unassignment from platform_device->remove() to dai_link->exit(). This is done to make jack init and deinit flows symmetric 2) Remove platform_device->remove() function 3) Simplify card->suspend_pre() and card->resume_post() by making use of snd_soc_card_get_codec_dai() helper While bdw_rt286 board - which is utilized by the catpt-driver - is definitely not part of "avs", same treatment applies. And thus decided to make it part of this series instead of sending it separately. Amadeusz Sławiński (10): ASoC: Intel: avs: da7219: Refactor jack handling ASoC: Intel: avs: nau8825: Refactor jack handling ASoC: Intel: avs: rt274: Refer to DAI name through a constant ASoC: Intel: avs: rt274: Refactor jack handling ASoC: Intel: avs: rt286: Add define for codec DAI name ASoC: Intel: avs: rt286: Refactor jack handling ASoC: Intel: avs: rt298: Add define for codec DAI name ASoC: Intel: avs: rt298: Refactor jack handling ASoC: Intel: avs: rt5682: Add define for codec DAI name ASoC: Intel: avs: rt5682: Refactor jack handling Cezary Rojewski (1): ASoC: Intel: bdw_rt286: Refactor jack handling sound/soc/intel/avs/boards/da7219.c | 34 +++++++---------- sound/soc/intel/avs/boards/nau8825.c | 35 ++++++------------ sound/soc/intel/avs/boards/rt274.c | 30 ++++++--------- sound/soc/intel/avs/boards/rt286.c | 37 ++++++++----------- sound/soc/intel/avs/boards/rt298.c | 37 ++++++++----------- sound/soc/intel/avs/boards/rt5682.c | 31 +++++++--------- sound/soc/intel/boards/bdw_rt286.c | 55 ++++++++-------------------- 7 files changed, 95 insertions(+), 164 deletions(-) -- 2.25.1

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[PATCH v2] ASoC: qcom: lpass-sc7180: Add maybe_unused tag for system PM ops
by Srinivasa Rao Mandadapu 29 Nov '22

29 Nov '22

Add __maybe_unused tag for system PM ops suspend and resume. This is required to fix allmodconfig compilation issue. Fixes: 2d68148f8f85 ("ASoC: qcom: lpass-sc7180: Add system suspend/resume PM ops") Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> --- Changes since v1: -- Updated Fixes tag. sound/soc/qcom/lpass-sc7180.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index b96b85a..41db661 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -163,14 +163,14 @@ static int sc7180_lpass_exit(struct platform_device *pdev) return 0; } -static int sc7180_lpass_dev_resume(struct device *dev) +static int __maybe_unused sc7180_lpass_dev_resume(struct device *dev) { struct lpass_data *drvdata = dev_get_drvdata(dev); return clk_bulk_prepare_enable(drvdata->num_clks, drvdata->clks); } -static int sc7180_lpass_dev_suspend(struct device *dev) +static int __maybe_unused sc7180_lpass_dev_suspend(struct device *dev) { struct lpass_data *drvdata = dev_get_drvdata(dev); -- 2.7.4

1 0

[PATCH v2 1/1] ASoC: SOF: Add DAI configuration support for AMD platforms.
by V sujith kumar Reddy 29 Nov '22

29 Nov '22

From: V sujith kumar Reddy <Vsujithkumar.Reddy(a)amd.com> Add support for configuring sp and hs DAI from topology. Signed-off-by: V sujith kumar Reddy <Vsujithkumar.Reddy(a)amd.com> Changes since v1 -- Apply on latest broonie-git for-next --- include/sound/sof/dai-amd.h | 1 + include/sound/sof/dai.h | 2 ++ include/uapi/sound/sof/tokens.h | 5 +++++ sound/soc/sof/ipc3-pcm.c | 2 ++ sound/soc/sof/ipc3-topology.c | 36 +++++++++++++++++++++++++-------- sound/soc/sof/sof-audio.h | 1 + sound/soc/sof/topology.c | 10 +++++++++ 7 files changed, 49 insertions(+), 8 deletions(-) diff --git a/include/sound/sof/dai-amd.h b/include/sound/sof/dai-amd.h index 92f45c180b7c..9df7ac824efe 100644 --- a/include/sound/sof/dai-amd.h +++ b/include/sound/sof/dai-amd.h @@ -17,6 +17,7 @@ struct sof_ipc_dai_acp_params { uint32_t fsync_rate; /* FSYNC frequency in Hz */ uint32_t tdm_slots; + uint32_t tdm_mode; } __packed; /* ACPDMIC Configuration Request - SOF_IPC_DAI_AMD_CONFIG */ diff --git a/include/sound/sof/dai.h b/include/sound/sof/dai.h index 9fbd3832bcdc..3041f5805b7b 100644 --- a/include/sound/sof/dai.h +++ b/include/sound/sof/dai.h @@ -86,6 +86,8 @@ enum sof_ipc_dai_type { SOF_DAI_AMD_DMIC, /**< AMD ACP DMIC */ SOF_DAI_MEDIATEK_AFE, /**< Mediatek AFE */ SOF_DAI_AMD_HS, /**< Amd HS */ + SOF_DAI_AMD_SP_VIRTUAL, /**< AMD ACP SP VIRTUAL */ + SOF_DAI_AMD_HS_VIRTUAL, /**< AMD ACP HS VIRTUAL */ }; /* general purpose DAI configuration */ diff --git a/include/uapi/sound/sof/tokens.h b/include/uapi/sound/sof/tokens.h index f187dfbd9325..bacaf8a6317e 100644 --- a/include/uapi/sound/sof/tokens.h +++ b/include/uapi/sound/sof/tokens.h @@ -198,4 +198,9 @@ /* COPIER */ #define SOF_TKN_INTEL_COPIER_NODE_TYPE 1980 +/* ACP I2S */ +#define SOF_TKN_AMD_ACPI2S_RATE 1700 +#define SOF_TKN_AMD_ACPI2S_CH 1701 +#define SOF_TKN_AMD_ACPI2S_TDM_MODE 1702 + #endif diff --git a/sound/soc/sof/ipc3-pcm.c b/sound/soc/sof/ipc3-pcm.c index dad57bef38f6..f10bfc9bd5cb 100644 --- a/sound/soc/sof/ipc3-pcm.c +++ b/sound/soc/sof/ipc3-pcm.c @@ -336,6 +336,7 @@ static int sof_ipc3_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, channels->min, channels->max); break; case SOF_DAI_AMD_SP: + case SOF_DAI_AMD_SP_VIRTUAL: rate->min = private->dai_config->acpsp.fsync_rate; rate->max = private->dai_config->acpsp.fsync_rate; channels->min = private->dai_config->acpsp.tdm_slots; @@ -347,6 +348,7 @@ static int sof_ipc3_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, channels->min, channels->max); break; case SOF_DAI_AMD_HS: + case SOF_DAI_AMD_HS_VIRTUAL: rate->min = private->dai_config->acphs.fsync_rate; rate->max = private->dai_config->acphs.fsync_rate; channels->min = private->dai_config->acphs.tdm_slots; diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index 0720e1eae084..b94cc40485ed 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -276,6 +276,16 @@ static const struct sof_topology_token acpdmic_tokens[] = { offsetof(struct sof_ipc_dai_acpdmic_params, pdm_ch)}, }; +/* ACPI2S */ +static const struct sof_topology_token acpi2s_tokens[] = { + {SOF_TKN_AMD_ACPI2S_RATE, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_dai_acp_params, fsync_rate)}, + {SOF_TKN_AMD_ACPI2S_CH, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_dai_acp_params, tdm_slots)}, + {SOF_TKN_AMD_ACPI2S_TDM_MODE, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_dai_acp_params, tdm_mode)}, +}; + /* Core tokens */ static const struct sof_topology_token core_tokens[] = { {SOF_TKN_COMP_CORE_ID, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, @@ -311,6 +321,7 @@ static const struct sof_token_info ipc3_token_list[SOF_TOKEN_COUNT] = { [SOF_SAI_TOKENS] = {"SAI tokens", sai_tokens, ARRAY_SIZE(sai_tokens)}, [SOF_AFE_TOKENS] = {"AFE tokens", afe_tokens, ARRAY_SIZE(afe_tokens)}, [SOF_ACPDMIC_TOKENS] = {"ACPDMIC tokens", acpdmic_tokens, ARRAY_SIZE(acpdmic_tokens)}, + [SOF_ACPI2S_TOKENS] = {"ACPI2S tokens", acpi2s_tokens, ARRAY_SIZE(acpi2s_tokens)}, }; /** @@ -1193,6 +1204,7 @@ static int sof_link_acp_sp_load(struct snd_soc_component *scomp, struct snd_sof_ struct snd_soc_tplg_hw_config *hw_config = slink->hw_configs; struct sof_dai_private_data *private = dai->private; u32 size = sizeof(*config); + int ret; /* handle master/slave and inverted clocks */ sof_dai_set_format(hw_config, config); @@ -1201,12 +1213,15 @@ static int sof_link_acp_sp_load(struct snd_soc_component *scomp, struct snd_sof_ memset(&config->acpsp, 0, sizeof(config->acpsp)); config->hdr.size = size; - config->acpsp.fsync_rate = le32_to_cpu(hw_config->fsync_rate); - config->acpsp.tdm_slots = le32_to_cpu(hw_config->tdm_slots); + ret = sof_update_ipc_object(scomp, &config->acpsp, SOF_ACPI2S_TOKENS, slink->tuples, + slink->num_tuples, size, slink->num_hw_configs); + if (ret < 0) + return ret; - dev_info(scomp->dev, "ACP_SP config ACP%d channel %d rate %d\n", + + dev_info(scomp->dev, "ACP_SP config ACP%d channel %d rate %d tdm_mode %d\n", config->dai_index, config->acpsp.tdm_slots, - config->acpsp.fsync_rate); + config->acpsp.fsync_rate, config->acpsp.tdm_mode); dai->number_configs = 1; dai->current_config = 0; @@ -1223,6 +1238,7 @@ static int sof_link_acp_hs_load(struct snd_soc_component *scomp, struct snd_sof_ struct snd_soc_tplg_hw_config *hw_config = slink->hw_configs; struct sof_dai_private_data *private = dai->private; u32 size = sizeof(*config); + int ret; /* Configures the DAI hardware format and inverted clocks */ sof_dai_set_format(hw_config, config); @@ -1231,12 +1247,14 @@ static int sof_link_acp_hs_load(struct snd_soc_component *scomp, struct snd_sof_ memset(&config->acphs, 0, sizeof(config->acphs)); config->hdr.size = size; - config->acphs.fsync_rate = le32_to_cpu(hw_config->fsync_rate); - config->acphs.tdm_slots = le32_to_cpu(hw_config->tdm_slots); + ret = sof_update_ipc_object(scomp, &config->acphs, SOF_ACPI2S_TOKENS, slink->tuples, + slink->num_tuples, size, slink->num_hw_configs); + if (ret < 0) + return ret; - dev_info(scomp->dev, "ACP_HS config ACP%d channel %d rate %d\n", + dev_info(scomp->dev, "ACP_HS config ACP%d channel %d rate %d tdm_mode %d\n", config->dai_index, config->acphs.tdm_slots, - config->acphs.fsync_rate); + config->acphs.fsync_rate, config->acphs.tdm_mode); dai->number_configs = 1; dai->current_config = 0; @@ -1545,9 +1563,11 @@ static int sof_ipc3_widget_setup_comp_dai(struct snd_sof_widget *swidget) ret = sof_link_acp_bt_load(scomp, slink, config, dai); break; case SOF_DAI_AMD_SP: + case SOF_DAI_AMD_SP_VIRTUAL: ret = sof_link_acp_sp_load(scomp, slink, config, dai); break; case SOF_DAI_AMD_HS: + case SOF_DAI_AMD_HS_VIRTUAL: ret = sof_link_acp_hs_load(scomp, slink, config, dai); break; case SOF_DAI_AMD_DMIC: diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index 1b5b3ea53a6e..29cf951e3526 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -248,6 +248,7 @@ enum sof_tokens { SOF_COPIER_FORMAT_TOKENS, SOF_GAIN_TOKENS, SOF_ACPDMIC_TOKENS, + SOF_ACPI2S_TOKENS, /* this should be the last */ SOF_TOKEN_COUNT, diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 9d9fcaa2a948..c668bd9d21ec 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -289,6 +289,9 @@ static const struct sof_dai_types sof_dais[] = { {"ACPDMIC", SOF_DAI_AMD_DMIC}, {"ACPHS", SOF_DAI_AMD_HS}, {"AFE", SOF_DAI_MEDIATEK_AFE}, + {"ACPSP_VIRTUAL", SOF_DAI_AMD_SP_VIRTUAL}, + {"ACPHS_VIRTUAL", SOF_DAI_AMD_HS_VIRTUAL}, + }; static enum sof_ipc_dai_type find_dai(const char *name) @@ -1895,6 +1898,13 @@ static int sof_link_load(struct snd_soc_component *scomp, int index, struct snd_ token_id = SOF_ACPDMIC_TOKENS; num_tuples += token_list[SOF_ACPDMIC_TOKENS].count; break; + case SOF_DAI_AMD_SP: + case SOF_DAI_AMD_HS: + case SOF_DAI_AMD_SP_VIRTUAL: + case SOF_DAI_AMD_HS_VIRTUAL: + token_id = SOF_ACPI2S_TOKENS; + num_tuples += token_list[SOF_ACPI2S_TOKENS].count; + break; default: break; } -- 2.25.1

2 2

[PATCH] selftests: alsa - move shared library configuration code to conf.c
by Jaroslav Kysela 29 Nov '22

29 Nov '22

The minimal alsa-lib configuration code is similar in both mixer and pcm tests. Move this code to the shared conf.c source file. Also, fix the build rules inspired by rseq tests. Build libatest.so which is linked to the both test utilities dynamically. Also, set the TEST_FILES variable for lib.mk. Cc: linux-kselftest(a)vger.kernel.org Cc: Shuah Khan <shuah(a)kernel.org> Reported-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- tools/testing/selftests/alsa/Makefile | 13 ++++- tools/testing/selftests/alsa/alsa-local.h | 2 + tools/testing/selftests/alsa/conf.c | 69 ++++++++++++++++++++++ tools/testing/selftests/alsa/mixer-test.c | 53 +---------------- tools/testing/selftests/alsa/pcm-test.c | 70 ----------------------- 5 files changed, 85 insertions(+), 122 deletions(-) diff --git a/tools/testing/selftests/alsa/Makefile b/tools/testing/selftests/alsa/Makefile index 8ac22d2eb2ba..a8c0383878d3 100644 --- a/tools/testing/selftests/alsa/Makefile +++ b/tools/testing/selftests/alsa/Makefile @@ -6,9 +6,20 @@ LDLIBS += $(shell pkg-config --libs alsa) ifeq ($(LDLIBS),) LDLIBS += -lasound endif +CFLAGS += -L$(OUTPUT) -Wl,-rpath=./ + +OVERRIDE_TARGETS = 1 TEST_GEN_PROGS := mixer-test pcm-test -pcm-test: pcm-test.c conf.c +TEST_GEN_PROGS_EXTENDED := libatest.so + +TEST_FILES := conf.d include ../lib.mk + +$(OUTPUT)/libatest.so: conf.c alsa-local.h + $(CC) $(CFLAGS) -shared -fPIC $< $(LDLIBS) -o $@ + +$(OUTPUT)/%: %.c $(TEST_GEN_PROGS_EXTENDED) alsa-local.h + $(CC) $(CFLAGS) $< $(LDLIBS) -latest -o $@ diff --git a/tools/testing/selftests/alsa/alsa-local.h b/tools/testing/selftests/alsa/alsa-local.h index fd0771ac600d..65f197ea9773 100644 --- a/tools/testing/selftests/alsa/alsa-local.h +++ b/tools/testing/selftests/alsa/alsa-local.h @@ -10,6 +10,8 @@ #include <alsa/asoundlib.h> +snd_config_t *get_alsalib_config(void); + void conf_load(void); void conf_free(void); snd_config_t *conf_by_card(int card); diff --git a/tools/testing/selftests/alsa/conf.c b/tools/testing/selftests/alsa/conf.c index dd6aa90deef9..c7ffc8f04195 100644 --- a/tools/testing/selftests/alsa/conf.c +++ b/tools/testing/selftests/alsa/conf.c @@ -28,6 +28,75 @@ struct card_data { static struct card_data *conf_cards; +static const char *alsa_config = +"ctl.hw {\n" +" @args [ CARD ]\n" +" @args.CARD.type string\n" +" type hw\n" +" card $CARD\n" +"}\n" +"pcm.hw {\n" +" @args [ CARD DEV SUBDEV ]\n" +" @args.CARD.type string\n" +" @args.DEV.type integer\n" +" @args.SUBDEV.type integer\n" +" type hw\n" +" card $CARD\n" +" device $DEV\n" +" subdevice $SUBDEV\n" +"}\n" +; + +#ifdef SND_LIB_VER +#if SND_LIB_VERSION >= SND_LIB_VER(1, 2, 6) +#define LIB_HAS_LOAD_STRING +#endif +#endif + +#ifndef LIB_HAS_LOAD_STRING +static int snd_config_load_string(snd_config_t **config, const char *s, + size_t size) +{ + snd_input_t *input; + snd_config_t *dst; + int err; + + assert(config && s); + if (size == 0) + size = strlen(s); + err = snd_input_buffer_open(&input, s, size); + if (err < 0) + return err; + err = snd_config_top(&dst); + if (err < 0) { + snd_input_close(input); + return err; + } + err = snd_config_load(dst, input); + snd_input_close(input); + if (err < 0) { + snd_config_delete(dst); + return err; + } + *config = dst; + return 0; +} +#endif + +snd_config_t *get_alsalib_config(void) +{ + snd_config_t *config; + int err; + + err = snd_config_load_string(&config, alsa_config, strlen(alsa_config)); + if (err < 0) { + ksft_print_msg("Unable to parse custom alsa-lib configuration: %s\n", + snd_strerror(err)); + ksft_exit_fail(); + } + return config; +} + static struct card_data *conf_data_by_card(int card, bool msg) { struct card_data *conf; diff --git a/tools/testing/selftests/alsa/mixer-test.c b/tools/testing/selftests/alsa/mixer-test.c index a38b89c28030..05f1749ae19d 100644 --- a/tools/testing/selftests/alsa/mixer-test.c +++ b/tools/testing/selftests/alsa/mixer-test.c @@ -26,6 +26,7 @@ #include <stdint.h> #include "../kselftest.h" +#include "alsa-local.h" #define TESTS_PER_CONTROL 7 @@ -50,56 +51,11 @@ struct ctl_data { struct ctl_data *next; }; -static const char *alsa_config = -"ctl.hw {\n" -" @args [ CARD ]\n" -" @args.CARD.type string\n" -" type hw\n" -" card $CARD\n" -"}\n" -; - int num_cards = 0; int num_controls = 0; struct card_data *card_list = NULL; struct ctl_data *ctl_list = NULL; -#ifdef SND_LIB_VER -#if SND_LIB_VERSION >= SND_LIB_VER(1, 2, 6) -#define LIB_HAS_LOAD_STRING -#endif -#endif - -#ifndef LIB_HAS_LOAD_STRING -static int snd_config_load_string(snd_config_t **config, const char *s, - size_t size) -{ - snd_input_t *input; - snd_config_t *dst; - int err; - - assert(config && s); - if (size == 0) - size = strlen(s); - err = snd_input_buffer_open(&input, s, size); - if (err < 0) - return err; - err = snd_config_top(&dst); - if (err < 0) { - snd_input_close(input); - return err; - } - err = snd_config_load(dst, input); - snd_input_close(input); - if (err < 0) { - snd_config_delete(dst); - return err; - } - *config = dst; - return 0; -} -#endif - static void find_controls(void) { char name[32]; @@ -112,12 +68,7 @@ static void find_controls(void) if (snd_card_next(&card) < 0 || card < 0) return; - err = snd_config_load_string(&config, alsa_config, strlen(alsa_config)); - if (err < 0) { - ksft_print_msg("Unable to parse custom alsa-lib configuration: %s\n", - snd_strerror(err)); - ksft_exit_fail(); - } + config = get_alsalib_config(); while (card >= 0) { sprintf(name, "hw:%d", card); diff --git a/tools/testing/selftests/alsa/pcm-test.c b/tools/testing/selftests/alsa/pcm-test.c index 2814d8f74f82..6e7dfc395b98 100644 --- a/tools/testing/selftests/alsa/pcm-test.c +++ b/tools/testing/selftests/alsa/pcm-test.c @@ -31,26 +31,6 @@ struct pcm_data { struct pcm_data *next; }; -static const char *alsa_config = -"ctl.hw {\n" -" @args [ CARD ]\n" -" @args.CARD.type string\n" -" type hw\n" -" card $CARD\n" -"}\n" -"pcm.hw {\n" -" @args [ CARD DEV SUBDEV ]\n" -" @args.CARD.type string\n" -" @args.DEV.type integer\n" -" @args.SUBDEV.type integer\n" -" type hw\n" -" card $CARD\n" -" device $DEV\n" -" subdevice $SUBDEV\n" -"}\n" - -; - int num_pcms = 0; struct pcm_data *pcm_list = NULL; @@ -77,56 +57,6 @@ long long timestamp_diff_ms(timestamp_t *tstamp) return (diff.tv_sec * 1000) + ((diff.tv_nsec + 500000L) / 1000000L); } -#ifdef SND_LIB_VER -#if SND_LIB_VERSION >= SND_LIB_VER(1, 2, 6) -#define LIB_HAS_LOAD_STRING -#endif -#endif - -#ifndef LIB_HAS_LOAD_STRING -static int snd_config_load_string(snd_config_t **config, const char *s, - size_t size) -{ - snd_input_t *input; - snd_config_t *dst; - int err; - - assert(config && s); - if (size == 0) - size = strlen(s); - err = snd_input_buffer_open(&input, s, size); - if (err < 0) - return err; - err = snd_config_top(&dst); - if (err < 0) { - snd_input_close(input); - return err; - } - err = snd_config_load(dst, input); - snd_input_close(input); - if (err < 0) { - snd_config_delete(dst); - return err; - } - *config = dst; - return 0; -} -#endif - -static snd_config_t *get_alsalib_config(void) -{ - snd_config_t *config; - int err; - - err = snd_config_load_string(&config, alsa_config, strlen(alsa_config)); - if (err < 0) { - ksft_print_msg("Unable to parse custom alsa-lib configuration: %s\n", - snd_strerror(err)); - ksft_exit_fail(); - } - return config; -} - static long device_from_id(snd_config_t *node) { const char *id; -- 2.35.3

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[PATCH v2] ALSA: usb-audio: Add quirk for Tascam Model 12
by John Keeping 29 Nov '22

29 Nov '22

Tascam's Model 12 is a mixer which can also operate as a USB audio interface. The audio interface uses explicit feedback but it seems that it does not correctly handle missing isochronous frames. When injecting an xrun (or doing anything else that pauses the playback stream) the feedback rate climbs (for example, at 44,100Hz nominal, I see a stable rate around 44,099 but xrun injection sees this peak at around 44,135 in most cases) and glitches are heard in the audio stream for several seconds - this is significantly worse than the single glitch expected for an underrun. While the stream does normally recover and the feedback rate returns to a stable value, I have seen some occurrences where this does not happen and the rate continues to increase while no audio is heard from the output. I have not found a solid reproduction for this. This misbehaviour can be avoided by totally resetting the stream state by switching the interface to alt 0 and back before restarting the playback stream. Add a new quirk flag which forces the endpoint and interface to be reconfigured whenever the stream is stopped, and use this for the Tascam Model 12. Separate interfaces are used for the playback and capture endpoints, so resetting the playback interface here will not affect the capture stream if it is running. While there are two endpoints on the interface, these are the OUT data endpoint and the IN explicit feedback endpoint corresponding to it and these are always stopped and started together. Signed-off-by: John Keeping <john(a)metanate.com> --- v2: - Set ep->need_prepare not ep->need_setup; this was missed when forward porting the patch as need_prepare is new in 6.1 - Add more details to the commit message about why it is safe to reset the interface here sound/usb/endpoint.c | 7 +++++++ sound/usb/quirks.c | 2 ++ sound/usb/usbaudio.h | 4 ++++ 3 files changed, 13 insertions(+) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 310cd6fb0038..4aaf0784940b 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -1673,6 +1673,13 @@ void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, bool keep_pending) stop_urbs(ep, false, keep_pending); if (ep->clock_ref) atomic_dec(&ep->clock_ref->locked); + + if (ep->chip->quirk_flags & QUIRK_FLAG_FORCE_IFACE_RESET && + usb_pipeout(ep->pipe)) { + ep->need_prepare = true; + if (ep->iface_ref) + ep->iface_ref->need_setup = true; + } } } diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 0f4dd3503a6a..58b37bfc885c 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2044,6 +2044,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { DEVICE_FLG(0x0644, 0x804a, /* TEAC UD-301 */ QUIRK_FLAG_ITF_USB_DSD_DAC | QUIRK_FLAG_CTL_MSG_DELAY | QUIRK_FLAG_IFACE_DELAY), + DEVICE_FLG(0x0644, 0x805f, /* TEAC Model 12 */ + QUIRK_FLAG_FORCE_IFACE_RESET), DEVICE_FLG(0x06f8, 0xb000, /* Hercules DJ Console (Windows Edition) */ QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x06f8, 0xd002, /* Hercules DJ Console (Macintosh Edition) */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index e97141ef730a..2aba508a4831 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -172,6 +172,9 @@ extern bool snd_usb_skip_validation; * Don't apply implicit feedback sync mode * QUIRK_FLAG_IFACE_SKIP_CLOSE * Don't closed interface during setting sample rate + * QUIRK_FLAG_FORCE_IFACE_RESET + * Force an interface reset whenever stopping & restarting a stream + * (e.g. after xrun) */ #define QUIRK_FLAG_GET_SAMPLE_RATE (1U << 0) @@ -194,5 +197,6 @@ extern bool snd_usb_skip_validation; #define QUIRK_FLAG_GENERIC_IMPLICIT_FB (1U << 17) #define QUIRK_FLAG_SKIP_IMPLICIT_FB (1U << 18) #define QUIRK_FLAG_IFACE_SKIP_CLOSE (1U << 19) +#define QUIRK_FLAG_FORCE_IFACE_RESET (1U << 20) #endif /* __USBAUDIO_H */ -- 2.38.1

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stop drivers from passing GFP_COMP to dma_alloc_coherent
by Christoph Hellwig 29 Nov '22

29 Nov '22

Hi all, this series fixes up various drivers that either try to get a compound page from dma_alloc_coherent or more often just did a bit of cargo cult copy and paste, and then warns about this flag passed in the DMA layer instead of silently clearing it like done by ARM and dma-iommu before. Diffstat: arch/arm/mm/dma-mapping.c | 17 ----------------- drivers/infiniband/hw/hfi1/init.c | 21 +++------------------ drivers/infiniband/hw/qib/qib_iba6120.c | 2 +- drivers/infiniband/hw/qib/qib_init.c | 21 ++++----------------- drivers/iommu/dma-iommu.c | 3 --- drivers/media/v4l2-core/videobuf-dma-contig.c | 22 ++++++++-------------- drivers/net/ethernet/broadcom/cnic.c | 6 ++---- drivers/s390/net/ism_drv.c | 3 ++- kernel/dma/mapping.c | 8 ++++++++ sound/core/memalloc.c | 5 ++--- 10 files changed, 30 insertions(+), 78 deletions(-)

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[PATCH v2] ASoC: qcom: lpass-sc7280: Add system suspend/resume PM ops
by Srinivasa Rao Mandadapu 29 Nov '22

29 Nov '22

Update lpass sc7280 platform driver with PM ops, such as system supend and resume callbacks. This update is required to disable clocks during supend and avoid XO shutdown issue. Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Tested-by: Rahul Ajmeriya <quic_rajmeriy(a)quicinc.com> --- Changes since v1: -- Add __maybe_unused flag for suspend/resume APIs. sound/soc/qcom/lpass-sc7280.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/sound/soc/qcom/lpass-sc7280.c b/sound/soc/qcom/lpass-sc7280.c index 70c4df8..d43f480 100644 --- a/sound/soc/qcom/lpass-sc7280.c +++ b/sound/soc/qcom/lpass-sc7280.c @@ -232,10 +232,28 @@ static int sc7280_lpass_exit(struct platform_device *pdev) struct lpass_data *drvdata = platform_get_drvdata(pdev); clk_bulk_disable_unprepare(drvdata->num_clks, drvdata->clks); + return 0; +} + +static int __maybe_unused sc7280_lpass_dev_resume(struct device *dev) +{ + struct lpass_data *drvdata = dev_get_drvdata(dev); + + return clk_bulk_prepare_enable(drvdata->num_clks, drvdata->clks); +} +static int __maybe_unused sc7280_lpass_dev_suspend(struct device *dev) +{ + struct lpass_data *drvdata = dev_get_drvdata(dev); + + clk_bulk_disable_unprepare(drvdata->num_clks, drvdata->clks); return 0; } +static const struct dev_pm_ops sc7280_lpass_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(sc7280_lpass_dev_suspend, sc7280_lpass_dev_resume) +}; + static struct lpass_variant sc7280_data = { .i2sctrl_reg_base = 0x1000, .i2sctrl_reg_stride = 0x1000, @@ -426,6 +444,7 @@ static struct platform_driver sc7280_lpass_cpu_platform_driver = { .driver = { .name = "sc7280-lpass-cpu", .of_match_table = of_match_ptr(sc7280_lpass_cpu_device_id), + .pm = &sc7280_lpass_pm_ops, }, .probe = asoc_qcom_lpass_cpu_platform_probe, .remove = asoc_qcom_lpass_cpu_platform_remove, -- 2.7.4

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[PATCH] ASoC: qcom: lpass-sc7280: Add system suspend/resume PM ops
by Srinivasa Rao Mandadapu 29 Nov '22

29 Nov '22

Update lpass sc7280 platform driver with PM ops, such as system supend and resume callbacks. This update is required to disable clocks during supend and avoid XO shutdown issue. Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Tested-by: Rahul Ajmeriya <quic_rajmeriy(a)quicinc.com> --- sound/soc/qcom/lpass-sc7280.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/qcom/lpass-sc7280.c b/sound/soc/qcom/lpass-sc7280.c index 70c4df8..65a04d3 100644 --- a/sound/soc/qcom/lpass-sc7280.c +++ b/sound/soc/qcom/lpass-sc7280.c @@ -232,10 +232,27 @@ static int sc7280_lpass_exit(struct platform_device *pdev) struct lpass_data *drvdata = platform_get_drvdata(pdev); clk_bulk_disable_unprepare(drvdata->num_clks, drvdata->clks); + return 0; +} + +static int sc7280_lpass_dev_resume(struct device *dev) +{ + struct lpass_data *drvdata = dev_get_drvdata(dev); + + return clk_bulk_prepare_enable(drvdata->num_clks, drvdata->clks); +} + +static int sc7280_lpass_dev_suspend(struct device *dev) +{ + struct lpass_data *drvdata = dev_get_drvdata(dev); + clk_bulk_disable_unprepare(drvdata->num_clks, drvdata->clks); return 0; } +static const struct dev_pm_ops sc7280_lpass_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(sc7280_lpass_dev_suspend, sc7280_lpass_dev_resume) +}; static struct lpass_variant sc7280_data = { .i2sctrl_reg_base = 0x1000, .i2sctrl_reg_stride = 0x1000, @@ -426,6 +443,7 @@ static struct platform_driver sc7280_lpass_cpu_platform_driver = { .driver = { .name = "sc7280-lpass-cpu", .of_match_table = of_match_ptr(sc7280_lpass_cpu_device_id), + .pm = &sc7280_lpass_pm_ops, }, .probe = asoc_qcom_lpass_cpu_platform_probe, .remove = asoc_qcom_lpass_cpu_platform_remove, -- 2.7.4

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[PATCH v2] ASoC: qcom: lpass-sc7180: Delete redundant error log from _resume()
by Matthias Kaehlcke 29 Nov '22

29 Nov '22

sc7180_lpass_dev_resume() logs an error if clk_bulk_prepare_enable() fails. The clock framework already generates error logs if anything goes wrong, so the logging in _resume() is redundant, drop it. Signed-off-by: Matthias Kaehlcke <mka(a)chromium.org> --- Changes in v2: - removed the log in the error path (requested by Doug) - updated subject (was "ASoC: qcom: lpass-sc7180: Return 0 instead of 'ret' at the end of _resume()") and commit message sound/soc/qcom/lpass-sc7180.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index 6ad1c5b5f2de..db3be15cf14c 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -162,15 +162,9 @@ static int sc7180_lpass_exit(struct platform_device *pdev) static int sc7180_lpass_dev_resume(struct device *dev) { - int ret = 0; struct lpass_data *drvdata = dev_get_drvdata(dev); - ret = clk_bulk_prepare_enable(drvdata->num_clks, drvdata->clks); - if (ret) { - dev_err(dev, "sc7180 clk prepare and enable failed\n"); - return ret; - } - return ret; + return clk_bulk_prepare_enable(drvdata->num_clks, drvdata->clks); } static int sc7180_lpass_dev_suspend(struct device *dev) -- 2.38.1.584.g0f3c55d4c2-goog

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