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

[PATCH -next] ASoC: wm2000: fix missing clk_disable_unprepare() on error in wm2000_anc_transition()
by Yang Yingliang 20 May '22

20 May '22

Fix the missing clk_disable_unprepare() before return from wm2000_anc_transition() in the error handling case. Fixes: 514cfd6dd725 ("ASoC: wm2000: Integrate with clock API") Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/codecs/wm2000.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 72e165cc6443..97ece3114b3d 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -536,7 +536,7 @@ static int wm2000_anc_transition(struct wm2000_priv *wm2000, { struct i2c_client *i2c = wm2000->i2c; int i, j; - int ret; + int ret = 0; if (wm2000->anc_mode == mode) return 0; @@ -566,13 +566,13 @@ static int wm2000_anc_transition(struct wm2000_priv *wm2000, ret = anc_transitions[i].step[j](i2c, anc_transitions[i].analogue); if (ret != 0) - return ret; + break; } if (anc_transitions[i].dest == ANC_OFF) clk_disable_unprepare(wm2000->mclk); - return 0; + return ret; } static int wm2000_anc_set_mode(struct wm2000_priv *wm2000) -- 2.25.1

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[PATCH 0/2] ASoC: SOF: mediatek: add debug dump
by Pierre-Louis Bossart 20 May '22

20 May '22

YC Hung (2): ASoC: SOF: mediatek: Add mediatek common debug dump ASoC: SOF: mediatek: Add mt8195 debug dump sound/soc/sof/mediatek/Makefile | 1 + sound/soc/sof/mediatek/mt8195/mt8195.c | 29 ++++++++ sound/soc/sof/mediatek/mtk-adsp-common.c | 84 ++++++++++++++++++++++++ sound/soc/sof/mediatek/mtk-adsp-common.h | 10 +++ 4 files changed, 124 insertions(+) create mode 100644 sound/soc/sof/mediatek/mtk-adsp-common.c create mode 100644 sound/soc/sof/mediatek/mtk-adsp-common.h -- 2.30.2

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[PATCH RESEND] ASoC: SOF: sof-client-ipc-flood-test: use pm_runtime_resume_and_get()
by Pierre-Louis Bossart 20 May '22

20 May '22

Use pm_runtime_resume_and_get() to replace the pm_runtime_get_sync() and pm_runtime_put_noidle() pattern. No functional changes. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- This patch was sent as part of the "[PATCH 0/4] ASoC: SOF: use pm_runtime_resume_and_get()" series, but somehow not applied. The other 3 patches were selected. sound/soc/sof/sof-client-ipc-flood-test.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sof/sof-client-ipc-flood-test.c b/sound/soc/sof/sof-client-ipc-flood-test.c index db3a052c5dd2b..4bdecd80248ac 100644 --- a/sound/soc/sof/sof-client-ipc-flood-test.c +++ b/sound/soc/sof/sof-client-ipc-flood-test.c @@ -217,10 +217,9 @@ static ssize_t sof_ipc_flood_dfs_write(struct file *file, const char __user *buf ipc_count = MAX_IPC_FLOOD_COUNT; } - ret = pm_runtime_get_sync(dev); + ret = pm_runtime_resume_and_get(dev); if (ret < 0 && ret != -EACCES) { dev_err_ratelimited(dev, "debugfs write failed to resume %d\n", ret); - pm_runtime_put_noidle(dev); goto out; } -- 2.30.2

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[PATCH 0/2] ASoC: remove two unnecessary gpiolib dependencies
by Pierre-Louis Bossart 20 May '22

20 May '22

Remove two dependencies - issues reported by Intel kernel test bot. Pierre-Louis Bossart (2): ASoC: max98357a: remove dependency on GPIOLIB ASoC: rt1015p: remove dependency on GPIOLIB sound/soc/codecs/Kconfig | 2 -- 1 file changed, 2 deletions(-) -- 2.30.2

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[PATCH v2 1/2] Revert "ASoC: amd: acp: Set gpio_spkr_en to None for max speaker amplifer in machine driver"
by V sujith kumar Reddy 20 May '22

20 May '22

ASoC: amd : acp : Set Speaker enable/disable pin through rt1019 codec driver. RT1019 codec has two ways of controlling the en_spkr. one way is controlling through gpio pin method the another way is through codec register update through driver. Now Speaker enable/disable is controlled through codec register updated by codec driver. This patch reverts gpio logic. This reverts commit 7fa5c33d043160eba3be9fb8e21588dff2a467c7. Signed-off-by: V sujith kumar Reddy <Vsujithkumar.Reddy(a)amd.com> --- sound/soc/amd/acp/acp-mach.h | 1 + sound/soc/amd/acp/acp-sof-mach.c | 4 ++-- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/acp/acp-mach.h b/sound/soc/amd/acp/acp-mach.h index c855f50d6b34..fd6299844ebe 100644 --- a/sound/soc/amd/acp/acp-mach.h +++ b/sound/soc/amd/acp/acp-mach.h @@ -21,6 +21,7 @@ #include <linux/gpio/consumer.h> #define EN_SPKR_GPIO_GB 0x11F +#define EN_SPKR_GPIO_NK 0x146 #define EN_SPKR_GPIO_NONE -EINVAL enum be_id { diff --git a/sound/soc/amd/acp/acp-sof-mach.c b/sound/soc/amd/acp/acp-sof-mach.c index bf61a1726f0e..8243765d490f 100644 --- a/sound/soc/amd/acp/acp-sof-mach.c +++ b/sound/soc/amd/acp/acp-sof-mach.c @@ -37,7 +37,7 @@ static struct acp_card_drvdata sof_rt5682_max_data = { .hs_codec_id = RT5682, .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, - .gpio_spkr_en = EN_SPKR_GPIO_NONE, + .gpio_spkr_en = EN_SPKR_GPIO_NK, }; static struct acp_card_drvdata sof_rt5682s_rt1019_data = { @@ -56,7 +56,7 @@ static struct acp_card_drvdata sof_rt5682s_max_data = { .hs_codec_id = RT5682S, .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, - .gpio_spkr_en = EN_SPKR_GPIO_NONE, + .gpio_spkr_en = EN_SPKR_GPIO_NK, }; static const struct snd_kcontrol_new acp_controls[] = { -- 2.25.1

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[PATCH -next] ASoC: codecs: lpass: Fix passing zero to 'PTR_ERR'
by YueHaibing 20 May '22

20 May '22

sound/soc/codecs/lpass-macro-common.c:28 lpass_macro_pds_init() warn: passing zero to 'PTR_ERR' sound/soc/codecs/lpass-macro-common.c:38 lpass_macro_pds_init() warn: passing zero to 'PTR_ERR' sound/soc/codecs/lpass-macro-common.c:54 lpass_macro_pds_init() warn: passing zero to 'ERR_PTR' dev_pm_domain_attach_by_name() may return NULL, set 'ret' as -ENODATA to fix this warning. Fixes: 1a8ee4cf8418 ("ASoC: codecs: Fix error handling in power domain init and exit handlers") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/lpass-macro-common.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/lpass-macro-common.c b/sound/soc/codecs/lpass-macro-common.c index 3c661fd61173..1b9082d237c1 100644 --- a/sound/soc/codecs/lpass-macro-common.c +++ b/sound/soc/codecs/lpass-macro-common.c @@ -25,7 +25,7 @@ struct lpass_macro *lpass_macro_pds_init(struct device *dev) l_pds->macro_pd = dev_pm_domain_attach_by_name(dev, "macro"); if (IS_ERR_OR_NULL(l_pds->macro_pd)) { - ret = PTR_ERR(l_pds->macro_pd); + ret = l_pds->macro_pd ? PTR_ERR(l_pds->macro_pd) : -ENODATA; goto macro_err; } @@ -35,7 +35,7 @@ struct lpass_macro *lpass_macro_pds_init(struct device *dev) l_pds->dcodec_pd = dev_pm_domain_attach_by_name(dev, "dcodec"); if (IS_ERR_OR_NULL(l_pds->dcodec_pd)) { - ret = PTR_ERR(l_pds->dcodec_pd); + ret = l_pds->dcodec_pd ? PTR_ERR(l_pds->dcodec_pd) : -ENODATA; goto dcodec_err; } -- 2.17.1

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topology: plugins: nhlt: fix intel dmic mode calculation
by GitHub pull_request - opened 20 May '22

20 May '22

alsa-project/alsa-utils pull request #148 was opened from juimonen: Dmic find_modes function was used with same sampling rate for both fifo a and b parameter calculations. This bug was found in testing two simultaneous dmic dais in topology. Fix this by introducing a sampling rate argument for find_modes. Signed-off-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-utils/pull/148 Patch URL : https://github.com/alsa-project/alsa-utils/pull/148.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH] ASoC: ops: Shift tested values in snd_soc_put_volsw() by +min
by Marek Vasut 19 May '22

19 May '22

While the $val/$val2 values passed in from userspace are always >= 0 integers, the limits of the control can be signed integers and the $min can be non-zero and less than zero. To correctly validate $val/$val2 against platform_max, add the $min offset to val first. Fixes: 817f7c9335ec0 ("ASoC: ops: Reject out of bounds values in snd_soc_put_volsw()") Signed-off-by: Marek Vasut <marex(a)denx.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: stable(a)vger.kernel.org --- sound/soc/soc-ops.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index f24f7354f46fe..6389a512c4dc6 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -317,7 +317,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, mask = BIT(sign_bit + 1) - 1; val = ucontrol->value.integer.value[0]; - if (mc->platform_max && val > mc->platform_max) + if (mc->platform_max && ((int)val + min) > mc->platform_max) return -EINVAL; if (val > max - min) return -EINVAL; @@ -330,7 +330,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, val = val << shift; if (snd_soc_volsw_is_stereo(mc)) { val2 = ucontrol->value.integer.value[1]; - if (mc->platform_max && val2 > mc->platform_max) + if (mc->platform_max && ((int)val2 + min) > mc->platform_max) return -EINVAL; if (val2 > max - min) return -EINVAL; -- 2.34.1

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[PATCH v1 0/2] ASOC : amd: acp :Add Machine driver changes for
by V sujith kumar Reddy 19 May '22

19 May '22

Add Machine driver support for nau8825, max98560 and rt5682s,rt1019 combination This patch set depends on: --gpio patch --https://patchwork.kernel.org/project/alsa-devel/patch/20220516160619.17832-1-Vsujithkumar.Reddy@amd.com/ --https://patchwork.kernel.org/project/alsa-devel/patch/20220516160619.17832-2-Vsujithkumar.Reddy@amd.com/ Signed-off-by: V sujith kumar Reddy <Vsujithkumar.Reddy(a)amd.com> V sujith kumar Reddy (2): ASoC: amd: acp: Add support for nau8825 and max98360 card ASoC: amd: acp: Add support for rt5682s and rt1019 card with hs instance sound/soc/amd/acp-config.c | 18 +++ sound/soc/amd/acp/Kconfig | 1 + sound/soc/amd/acp/acp-mach-common.c | 228 +++++++++++++++++++++++++--- sound/soc/amd/acp/acp-mach.h | 3 + sound/soc/amd/acp/acp-sof-mach.c | 30 ++++ 5 files changed, 263 insertions(+), 17 deletions(-) -- 2.25.1

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[v7] FROMLIST: ASoC: Intel: sof_cs42l42: adding support for ADL configuration and BT offload audio
by Terry Chen 18 May '22

18 May '22

To be able to do driver data for adl_mx98360a_cs4242 which supports two max98360a speaker amplifiers on SSP1 and cs42l42 headphone codec on SSP0 running on ADL platform. Also add the capability to machine driver of creating DAI Link for BT offload. Although BT offload always uses SSP2 port but we reserve the flexibility to assign the port number in macro. Signed-off-by: Terry Chen <terry_chen(a)wistron.corp-partner.google.com> --- sound/soc/intel/boards/sof_cs42l42.c | 86 ++++++++++++++++++- .../intel/common/soc-acpi-intel-adl-match.c | 7 ++ 2 files changed, 89 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index ce78c18798876..56bae2185904b 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -41,8 +41,13 @@ #define SOF_CS42L42_DAILINK_MASK (GENMASK(24, 10)) #define SOF_CS42L42_DAILINK(link1, link2, link3, link4, link5) \ ((((link1) | ((link2) << 3) | ((link3) << 6) | ((link4) << 9) | ((link5) << 12)) << SOF_CS42L42_DAILINK_SHIFT) & SOF_CS42L42_DAILINK_MASK) -#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(25) -#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(26) +#define SOF_BT_OFFLOAD_PRESENT BIT(25) +#define SOF_CS42L42_SSP_BT_SHIFT 26 +#define SOF_CS42L42_SSP_BT_MASK (GENMASK(28, 26)) +#define SOF_CS42L42_SSP_BT(quirk) \ + (((quirk) << SOF_CS42L42_SSP_BT_SHIFT) & SOF_CS42L42_SSP_BT_MASK) +#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(29) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(30) enum { LINK_NONE = 0, @@ -50,6 +55,7 @@ enum { LINK_SPK = 2, LINK_DMIC = 3, LINK_HDMI = 4, + LINK_BT = 5, }; /* Default: SSP2 */ @@ -278,6 +284,13 @@ static struct snd_soc_dai_link_component dmic_component[] = { } }; +static struct snd_soc_dai_link_component dummy_component[] = { + { + .name = "snd-soc-dummy", + .dai_name = "snd-soc-dummy-dai", + } +}; + static int create_spk_amp_dai_links(struct device *dev, struct snd_soc_dai_link *links, struct snd_soc_dai_link_component *cpus, @@ -467,9 +480,50 @@ static int create_hdmi_dai_links(struct device *dev, return -ENOMEM; } +static int create_bt_offload_dai_links(struct device *dev, + struct snd_soc_dai_link *links, + struct snd_soc_dai_link_component *cpus, + int *id, int ssp_bt) +{ + /* bt offload */ + if (!(sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT)) + return 0; + + links[*id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", + ssp_bt); + if (!links[*id].name) + goto devm_err; + + links[*id].id = *id; + links[*id].codecs = dummy_component; + links[*id].num_codecs = ARRAY_SIZE(dummy_component); + links[*id].platforms = platform_component; + links[*id].num_platforms = ARRAY_SIZE(platform_component); + + links[*id].dpcm_playback = 1; + links[*id].dpcm_capture = 1; + links[*id].no_pcm = 1; + links[*id].cpus = &cpus[*id]; + links[*id].num_cpus = 1; + + links[*id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ssp_bt); + if (!links[*id].cpus->dai_name) + goto devm_err; + + (*id)++; + + return 0; + +devm_err: + return -ENOMEM; +} + static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, int ssp_codec, int ssp_amp, + int ssp_bt, int dmic_be_num, int hdmi_num) { @@ -522,6 +576,14 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, goto devm_err; } break; + case LINK_BT: + ret = create_bt_offload_dai_links(dev, links, cpus, &id, ssp_bt); + if (ret < 0) { + dev_err(dev, "fail to create bt offload dai links, ret %d\n", + ret); + goto devm_err; + } + break; case LINK_NONE: /* caught here if it's not used as terminator in macro */ default: @@ -543,7 +605,7 @@ static int sof_audio_probe(struct platform_device *pdev) struct snd_soc_acpi_mach *mach; struct sof_card_private *ctx; int dmic_be_num, hdmi_num; - int ret, ssp_amp, ssp_codec; + int ret, ssp_bt, ssp_amp, ssp_codec; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -568,6 +630,9 @@ static int sof_audio_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "sof_cs42l42_quirk = %lx\n", sof_cs42l42_quirk); + ssp_bt = (sof_cs42l42_quirk & SOF_CS42L42_SSP_BT_MASK) >> + SOF_CS42L42_SSP_BT_SHIFT; + ssp_amp = (sof_cs42l42_quirk & SOF_CS42L42_SSP_AMP_MASK) >> SOF_CS42L42_SSP_AMP_SHIFT; @@ -578,9 +643,11 @@ static int sof_audio_probe(struct platform_device *pdev) if (sof_cs42l42_quirk & SOF_SPEAKER_AMP_PRESENT) sof_audio_card_cs42l42.num_links++; + if (sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT) + sof_audio_card_cs42l42.num_links++; dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, - dmic_be_num, hdmi_num); + ssp_bt, dmic_be_num, hdmi_num); if (!dai_links) return -ENOMEM; @@ -621,6 +688,17 @@ static const struct platform_device_id board_ids[] = { SOF_CS42L42_SSP_AMP(1)) | SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_NONE), }, + { + .name = "adl_mx98360a_cs4242", + .driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_CS42L42_SSP_AMP(1) | + SOF_CS42L42_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_PRESENT | + SOF_CS42L42_SSP_BT(2)) | + SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_BT), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index 7c8cd00457f81..311adeb3afbc4 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -384,6 +384,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-cs35l41.tplg", }, + { + .id = "10134242", + .drv_name = "adl_mx98360a_cs4242", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98360a_amp, + .sof_tplg_filename = "sof-adl-max98360a-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); -- 2.25.1

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