diff options
author | Mark Brown <broonie@kernel.org> | 2020-07-31 19:54:01 +0100 |
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committer | Mark Brown <broonie@kernel.org> | 2020-07-31 19:54:01 +0100 |
commit | c8f7dbdbaa15c700ea02abf92b8d9bda2e91050b (patch) | |
tree | 4f433b76dd8f3f0a0268fc89dc4ac4f607414e45 /sound/soc | |
parent | 92ed301919932f777713b9172e525674157e983d (diff) | |
parent | 5aef1ff2397d021f93d874b57dff032fdfac73de (diff) | |
download | lwn-c8f7dbdbaa15c700ea02abf92b8d9bda2e91050b.tar.gz lwn-c8f7dbdbaa15c700ea02abf92b8d9bda2e91050b.zip |
Merge remote-tracking branch 'asoc/for-5.8' into asoc-linus
Diffstat (limited to 'sound/soc')
-rw-r--r-- | sound/soc/codecs/max98357a.c | 50 | ||||
-rw-r--r-- | sound/soc/codecs/max98390.c | 2 | ||||
-rw-r--r-- | sound/soc/fsl/fsl_sai.c | 5 | ||||
-rw-r--r-- | sound/soc/fsl/fsl_sai.h | 2 | ||||
-rw-r--r-- | sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 41 | ||||
-rw-r--r-- | sound/soc/intel/boards/skl_hda_dsp_common.h | 1 | ||||
-rw-r--r-- | sound/soc/intel/boards/skl_hda_dsp_generic.c | 17 | ||||
-rw-r--r-- | sound/soc/intel/common/soc-acpi-intel-ehl-match.c | 2 | ||||
-rw-r--r-- | sound/soc/meson/axg-card.c | 20 | ||||
-rw-r--r-- | sound/soc/meson/axg-tdm-formatter.c | 11 | ||||
-rw-r--r-- | sound/soc/meson/axg-tdm-formatter.h | 1 | ||||
-rw-r--r-- | sound/soc/meson/axg-tdm-interface.c | 26 | ||||
-rw-r--r-- | sound/soc/meson/axg-tdmin.c | 16 | ||||
-rw-r--r-- | sound/soc/meson/axg-tdmout.c | 3 | ||||
-rw-r--r-- | sound/soc/meson/gx-card.c | 18 | ||||
-rw-r--r-- | sound/soc/meson/meson-card-utils.c | 4 | ||||
-rw-r--r-- | sound/soc/soc-core.c | 5 | ||||
-rw-r--r-- | sound/soc/soc-dai.c | 16 | ||||
-rw-r--r-- | sound/soc/soc-pcm.c | 42 |
19 files changed, 185 insertions, 97 deletions
diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index a8bd793a7867..151f05a68435 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -23,36 +23,61 @@ struct max98357a_priv { struct gpio_desc *sdmode; unsigned int sdmode_delay; + int sdmode_switch; }; -static int max98357a_sdmode_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) { - struct snd_soc_component *component = - snd_soc_dapm_to_component(w->dapm); + struct snd_soc_component *component = dai->component; struct max98357a_priv *max98357a = snd_soc_component_get_drvdata(component); if (!max98357a->sdmode) return 0; - if (event & SND_SOC_DAPM_POST_PMU) { - msleep(max98357a->sdmode_delay); - gpiod_set_value(max98357a->sdmode, 1); - dev_dbg(component->dev, "set sdmode to 1"); - } else if (event & SND_SOC_DAPM_PRE_PMD) { + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + mdelay(max98357a->sdmode_delay); + if (max98357a->sdmode_switch) { + gpiod_set_value(max98357a->sdmode, 1); + dev_dbg(component->dev, "set sdmode to 1"); + } + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: gpiod_set_value(max98357a->sdmode, 0); dev_dbg(component->dev, "set sdmode to 0"); + break; } return 0; } +static int max98357a_sdmode_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct max98357a_priv *max98357a = + snd_soc_component_get_drvdata(component); + + if (event & SND_SOC_DAPM_POST_PMU) + max98357a->sdmode_switch = 1; + else if (event & SND_SOC_DAPM_POST_PMD) + max98357a->sdmode_switch = 0; + + return 0; +} + static const struct snd_soc_dapm_widget max98357a_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("Speaker"), SND_SOC_DAPM_OUT_DRV_E("SD_MODE", SND_SOC_NOPM, 0, 0, NULL, 0, max98357a_sdmode_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route max98357a_dapm_routes[] = { @@ -71,6 +96,10 @@ static const struct snd_soc_component_driver max98357a_component_driver = { .non_legacy_dai_naming = 1, }; +static const struct snd_soc_dai_ops max98357a_dai_ops = { + .trigger = max98357a_daiops_trigger, +}; + static struct snd_soc_dai_driver max98357a_dai_driver = { .name = "HiFi", .playback = { @@ -90,6 +119,7 @@ static struct snd_soc_dai_driver max98357a_dai_driver = { .channels_min = 1, .channels_max = 2, }, + .ops = &max98357a_dai_ops, }; static int max98357a_platform_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index e6613b52bd78..9859a133b90c 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -678,7 +678,7 @@ static const struct snd_kcontrol_new max98390_dai_controls = static const struct snd_soc_dapm_widget max98390_dapm_widgets[] = { SND_SOC_DAPM_DAC_E("Amp Enable", "HiFi Playback", - MAX98390_R203A_AMP_EN, 0, 0, max98390_dac_event, + SND_SOC_NOPM, 0, 0, max98390_dac_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MUX("DAI Sel Mux", SND_SOC_NOPM, 0, 0, &max98390_dai_controls), diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 9d436b0c5718..7031869a023a 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -680,10 +680,11 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), 0); regmap_update_bits(sai->regmap, FSL_SAI_TCR1(ofs), - FSL_SAI_CR1_RFW_MASK, + FSL_SAI_CR1_RFW_MASK(sai->soc_data->fifo_depth), sai->soc_data->fifo_depth - FSL_SAI_MAXBURST_TX); regmap_update_bits(sai->regmap, FSL_SAI_RCR1(ofs), - FSL_SAI_CR1_RFW_MASK, FSL_SAI_MAXBURST_RX - 1); + FSL_SAI_CR1_RFW_MASK(sai->soc_data->fifo_depth), + FSL_SAI_MAXBURST_RX - 1); snd_soc_dai_init_dma_data(cpu_dai, &sai->dma_params_tx, &sai->dma_params_rx); diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 76b15deea80c..6aba7d28f5f3 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -94,7 +94,7 @@ #define FSL_SAI_CSR_FRDE BIT(0) /* SAI Transmit and Receive Configuration 1 Register */ -#define FSL_SAI_CR1_RFW_MASK 0x1f +#define FSL_SAI_CR1_RFW_MASK(x) ((x) - 1) /* SAI Transmit and Receive Configuration 2 Register */ #define FSL_SAI_CR2_SYNC BIT(30) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index b34cf6cf1139..2985f8bf30b2 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -336,22 +336,45 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_soc_dpcm *dpcm = container_of( - params, struct snd_soc_dpcm, hw_params); - struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; - struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; + struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL; + + /* + * The following loop will be called only for playback stream + * In this platform, there is only one playback device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) { + rtd_dpcm = dpcm; + break; + } + + /* + * This following loop will be called only for capture stream + * In this platform, there is only one capture device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) { + rtd_dpcm = dpcm; + break; + } + + if (!rtd_dpcm) + return -EINVAL; + + /* + * The above 2 loops are mutually exclusive based on the stream direction, + * thus rtd_dpcm variable will never be overwritten + */ /* * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ - if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || - !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || - !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { + if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) { rate->min = rate->max = 48000; chan->min = chan->max = 2; snd_mask_none(fmt); snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); - } else if (!strcmp(fe_dai_link->name, "Kbl Audio DMIC cap")) { + } else if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio DMIC cap")) { if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2) chan->min = chan->max = 2; @@ -362,7 +385,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, * The speaker on the SSP0 supports S16_LE and not S24_LE. * thus changing the mask here */ - if (!strcmp(be_dai_link->name, "SSP0-Codec")) + if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec")) snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); return 0; diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.h b/sound/soc/intel/boards/skl_hda_dsp_common.h index 507750ef67f3..4b0b3959182e 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_common.h +++ b/sound/soc/intel/boards/skl_hda_dsp_common.h @@ -33,6 +33,7 @@ struct skl_hda_private { int dai_index; const char *platform_name; bool common_hdmi_codec_drv; + bool idisp_codec; }; extern struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS]; diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 79c8947f840b..ca4900036ead 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -79,6 +79,9 @@ skl_hda_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link) link->platforms->name = ctx->platform_name; link->nonatomic = 1; + if (!ctx->idisp_codec) + return 0; + if (strstr(link->name, "HDMI")) { ret = skl_hda_hdmi_add_pcm(card, ctx->pcm_count); @@ -118,19 +121,20 @@ static char hda_soc_components[30]; static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params) { struct snd_soc_card *card = &hda_soc_card; + struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card); struct snd_soc_dai_link *dai_link; - u32 codec_count, codec_mask, idisp_mask; + u32 codec_count, codec_mask; int i, num_links, num_route; codec_mask = mach_params->codec_mask; codec_count = hweight_long(codec_mask); - idisp_mask = codec_mask & IDISP_CODEC_MASK; + ctx->idisp_codec = !!(codec_mask & IDISP_CODEC_MASK); if (!codec_count || codec_count > 2 || - (codec_count == 2 && !idisp_mask)) + (codec_count == 2 && !ctx->idisp_codec)) return -EINVAL; - if (codec_mask == idisp_mask) { + if (codec_mask == IDISP_CODEC_MASK) { /* topology with iDisp as the only HDA codec */ num_links = IDISP_DAI_COUNT + DMIC_DAI_COUNT; num_route = IDISP_ROUTE_COUNT; @@ -152,7 +156,7 @@ static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params) num_route = ARRAY_SIZE(skl_hda_map); card->dapm_widgets = skl_hda_widgets; card->num_dapm_widgets = ARRAY_SIZE(skl_hda_widgets); - if (!idisp_mask) { + if (!ctx->idisp_codec) { for (i = 0; i < IDISP_DAI_COUNT; i++) { skl_hda_be_dai_links[i].codecs = dummy_codec; skl_hda_be_dai_links[i].num_codecs = @@ -211,6 +215,8 @@ static int skl_hda_audio_probe(struct platform_device *pdev) if (!mach) return -EINVAL; + snd_soc_card_set_drvdata(&hda_soc_card, ctx); + ret = skl_hda_fill_card_info(&mach->mach_params); if (ret < 0) { dev_err(&pdev->dev, "Unsupported HDAudio/iDisp configuration found\n"); @@ -223,7 +229,6 @@ static int skl_hda_audio_probe(struct platform_device *pdev) ctx->common_hdmi_codec_drv = mach->mach_params.common_hdmi_codec_drv; hda_soc_card.dev = &pdev->dev; - snd_soc_card_set_drvdata(&hda_soc_card, ctx); if (mach->mach_params.dmic_num > 0) { snprintf(hda_soc_components, sizeof(hda_soc_components), diff --git a/sound/soc/intel/common/soc-acpi-intel-ehl-match.c b/sound/soc/intel/common/soc-acpi-intel-ehl-match.c index 45e07d886013..badafc1d54d2 100644 --- a/sound/soc/intel/common/soc-acpi-intel-ehl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-ehl-match.c @@ -12,7 +12,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_ehl_machines[] = { { - .id = "INTC1027", + .id = "10EC5660", .drv_name = "ehl_rt5660", .sof_fw_filename = "sof-ehl.ri", .sof_tplg_filename = "sof-ehl-rt5660.tplg", diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 89f7f64747cd..33058518c3da 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -116,7 +116,7 @@ static int axg_card_add_tdm_loopback(struct snd_soc_card *card, lb = &card->dai_link[*index + 1]; - lb->name = kasprintf(GFP_KERNEL, "%s-lb", pad->name); + lb->name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-lb", pad->name); if (!lb->name) return -ENOMEM; @@ -327,20 +327,22 @@ static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np, return ret; if (axg_card_cpu_is_playback_fe(dai_link->cpus->of_node)) - ret = meson_card_set_fe_link(card, dai_link, np, true); + return meson_card_set_fe_link(card, dai_link, np, true); else if (axg_card_cpu_is_capture_fe(dai_link->cpus->of_node)) - ret = meson_card_set_fe_link(card, dai_link, np, false); - else - ret = meson_card_set_be_link(card, dai_link, np); + return meson_card_set_fe_link(card, dai_link, np, false); + + ret = meson_card_set_be_link(card, dai_link, np); if (ret) return ret; - if (axg_card_cpu_is_tdm_iface(dai_link->cpus->of_node)) - ret = axg_card_parse_tdm(card, np, index); - else if (axg_card_cpu_is_codec(dai_link->cpus->of_node)) { + if (axg_card_cpu_is_codec(dai_link->cpus->of_node)) { dai_link->params = &codec_params; - dai_link->no_pcm = 0; /* link is not a DPCM BE */ + } else { + dai_link->no_pcm = 1; + snd_soc_dai_link_set_capabilities(dai_link); + if (axg_card_cpu_is_tdm_iface(dai_link->cpus->of_node)) + ret = axg_card_parse_tdm(card, np, index); } return ret; diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c index 358c8c0d861c..f7e8e9da68a0 100644 --- a/sound/soc/meson/axg-tdm-formatter.c +++ b/sound/soc/meson/axg-tdm-formatter.c @@ -70,7 +70,7 @@ EXPORT_SYMBOL_GPL(axg_tdm_formatter_set_channel_masks); static int axg_tdm_formatter_enable(struct axg_tdm_formatter *formatter) { struct axg_tdm_stream *ts = formatter->stream; - bool invert = formatter->drv->quirks->invert_sclk; + bool invert; int ret; /* Do nothing if the formatter is already enabled */ @@ -96,11 +96,12 @@ static int axg_tdm_formatter_enable(struct axg_tdm_formatter *formatter) return ret; /* - * If sclk is inverted, invert it back and provide the inversion - * required by the formatter + * If sclk is inverted, it means the bit should latched on the + * rising edge which is what our HW expects. If not, we need to + * invert it before the formatter. */ - invert ^= axg_tdm_sclk_invert(ts->iface->fmt); - ret = clk_set_phase(formatter->sclk, invert ? 180 : 0); + invert = axg_tdm_sclk_invert(ts->iface->fmt); + ret = clk_set_phase(formatter->sclk, invert ? 0 : 180); if (ret) return ret; diff --git a/sound/soc/meson/axg-tdm-formatter.h b/sound/soc/meson/axg-tdm-formatter.h index 9ef98e955cb2..a1f0dcc0ff13 100644 --- a/sound/soc/meson/axg-tdm-formatter.h +++ b/sound/soc/meson/axg-tdm-formatter.h @@ -16,7 +16,6 @@ struct snd_kcontrol; struct axg_tdm_formatter_hw { unsigned int skew_offset; - bool invert_sclk; }; struct axg_tdm_formatter_ops { diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index 6de27238e9df..36df30915378 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -119,18 +119,25 @@ static int axg_tdm_iface_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); - /* These modes are not supported */ - if (fmt & (SND_SOC_DAIFMT_CBS_CFM | SND_SOC_DAIFMT_CBM_CFS)) { + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + if (!iface->mclk) { + dev_err(dai->dev, "cpu clock master: mclk missing\n"); + return -ENODEV; + } + break; + + case SND_SOC_DAIFMT_CBM_CFM: + break; + + case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFS: dev_err(dai->dev, "only CBS_CFS and CBM_CFM are supported\n"); + /* Fall-through */ + default: return -EINVAL; } - /* If the TDM interface is the clock master, it requires mclk */ - if (!iface->mclk && (fmt & SND_SOC_DAIFMT_CBS_CFS)) { - dev_err(dai->dev, "cpu clock master: mclk missing\n"); - return -ENODEV; - } - iface->fmt = fmt; return 0; } @@ -319,7 +326,8 @@ static int axg_tdm_iface_hw_params(struct snd_pcm_substream *substream, if (ret) return ret; - if (iface->fmt & SND_SOC_DAIFMT_CBS_CFS) { + if ((iface->fmt & SND_SOC_DAIFMT_MASTER_MASK) == + SND_SOC_DAIFMT_CBS_CFS) { ret = axg_tdm_iface_set_sclk(dai, params); if (ret) return ret; diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c index 973d4c02ef8d..88ed95ae886b 100644 --- a/sound/soc/meson/axg-tdmin.c +++ b/sound/soc/meson/axg-tdmin.c @@ -228,15 +228,29 @@ static const struct axg_tdm_formatter_driver axg_tdmin_drv = { .regmap_cfg = &axg_tdmin_regmap_cfg, .ops = &axg_tdmin_ops, .quirks = &(const struct axg_tdm_formatter_hw) { - .invert_sclk = false, .skew_offset = 2, }, }; +static const struct axg_tdm_formatter_driver g12a_tdmin_drv = { + .component_drv = &axg_tdmin_component_drv, + .regmap_cfg = &axg_tdmin_regmap_cfg, + .ops = &axg_tdmin_ops, + .quirks = &(const struct axg_tdm_formatter_hw) { + .skew_offset = 3, + }, +}; + static const struct of_device_id axg_tdmin_of_match[] = { { .compatible = "amlogic,axg-tdmin", .data = &axg_tdmin_drv, + }, { + .compatible = "amlogic,g12a-tdmin", + .data = &g12a_tdmin_drv, + }, { + .compatible = "amlogic,sm1-tdmin", + .data = &g12a_tdmin_drv, }, {} }; MODULE_DEVICE_TABLE(of, axg_tdmin_of_match); diff --git a/sound/soc/meson/axg-tdmout.c b/sound/soc/meson/axg-tdmout.c index 418ec314b37d..3ceabddae629 100644 --- a/sound/soc/meson/axg-tdmout.c +++ b/sound/soc/meson/axg-tdmout.c @@ -238,7 +238,6 @@ static const struct axg_tdm_formatter_driver axg_tdmout_drv = { .regmap_cfg = &axg_tdmout_regmap_cfg, .ops = &axg_tdmout_ops, .quirks = &(const struct axg_tdm_formatter_hw) { - .invert_sclk = true, .skew_offset = 1, }, }; @@ -248,7 +247,6 @@ static const struct axg_tdm_formatter_driver g12a_tdmout_drv = { .regmap_cfg = &axg_tdmout_regmap_cfg, .ops = &axg_tdmout_ops, .quirks = &(const struct axg_tdm_formatter_hw) { - .invert_sclk = true, .skew_offset = 2, }, }; @@ -309,7 +307,6 @@ static const struct axg_tdm_formatter_driver sm1_tdmout_drv = { .regmap_cfg = &axg_tdmout_regmap_cfg, .ops = &axg_tdmout_ops, .quirks = &(const struct axg_tdm_formatter_hw) { - .invert_sclk = true, .skew_offset = 2, }, }; diff --git a/sound/soc/meson/gx-card.c b/sound/soc/meson/gx-card.c index 4abf7efb7eac..fdd2d5303b2a 100644 --- a/sound/soc/meson/gx-card.c +++ b/sound/soc/meson/gx-card.c @@ -96,21 +96,21 @@ static int gx_card_add_link(struct snd_soc_card *card, struct device_node *np, return ret; if (gx_card_cpu_identify(dai_link->cpus, "FIFO")) - ret = meson_card_set_fe_link(card, dai_link, np, true); - else - ret = meson_card_set_be_link(card, dai_link, np); + return meson_card_set_fe_link(card, dai_link, np, true); + ret = meson_card_set_be_link(card, dai_link, np); if (ret) return ret; - /* Check if the cpu is the i2s encoder and parse i2s data */ - if (gx_card_cpu_identify(dai_link->cpus, "I2S Encoder")) - ret = gx_card_parse_i2s(card, np, index); - /* Or apply codec to codec params if necessary */ - else if (gx_card_cpu_identify(dai_link->cpus, "CODEC CTRL")) { + if (gx_card_cpu_identify(dai_link->cpus, "CODEC CTRL")) { dai_link->params = &codec_params; - dai_link->no_pcm = 0; /* link is not a DPCM BE */ + } else { + dai_link->no_pcm = 1; + snd_soc_dai_link_set_capabilities(dai_link); + /* Check if the cpu is the i2s encoder and parse i2s data */ + if (gx_card_cpu_identify(dai_link->cpus, "I2S Encoder")) + ret = gx_card_parse_i2s(card, np, index); } return ret; diff --git a/sound/soc/meson/meson-card-utils.c b/sound/soc/meson/meson-card-utils.c index 5a4a91c88734..c734131ff0d6 100644 --- a/sound/soc/meson/meson-card-utils.c +++ b/sound/soc/meson/meson-card-utils.c @@ -147,10 +147,6 @@ int meson_card_set_be_link(struct snd_soc_card *card, struct device_node *np; int ret, num_codecs; - link->no_pcm = 1; - link->dpcm_playback = 1; - link->dpcm_capture = 1; - num_codecs = of_get_child_count(node); if (!num_codecs) { dev_err(card->dev, "be link %s has no codec\n", diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2b8abf88ec60..f1d641cd48da 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -446,7 +446,6 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( dev->parent = card->dev; dev->release = soc_release_rtd_dev; - dev->groups = soc_dev_attr_groups; dev_set_name(dev, "%s", dai_link->name); @@ -503,6 +502,10 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( /* see for_each_card_rtds */ list_add_tail(&rtd->list, &card->rtd_list); + ret = device_add_groups(dev, soc_dev_attr_groups); + if (ret < 0) + goto free_rtd; + return rtd; free_rtd: diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 457159975b01..cecbbed2de9d 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -400,28 +400,30 @@ void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link) struct snd_soc_dai_link_component *codec; struct snd_soc_dai *dai; bool supported[SNDRV_PCM_STREAM_LAST + 1]; + bool supported_cpu; + bool supported_codec; int direction; int i; for_each_pcm_streams(direction) { - supported[direction] = true; + supported_cpu = false; + supported_codec = false; for_each_link_cpus(dai_link, i, cpu) { dai = snd_soc_find_dai(cpu); - if (!dai || !snd_soc_dai_stream_valid(dai, direction)) { - supported[direction] = false; + if (dai && snd_soc_dai_stream_valid(dai, direction)) { + supported_cpu = true; break; } } - if (!supported[direction]) - continue; for_each_link_codecs(dai_link, i, codec) { dai = snd_soc_find_dai(codec); - if (!dai || !snd_soc_dai_stream_valid(dai, direction)) { - supported[direction] = false; + if (dai && snd_soc_dai_stream_valid(dai, direction)) { + supported_codec = true; break; } } + supported[direction] = supported_cpu && supported_codec; } dai_link->dpcm_playback = supported[SNDRV_PCM_STREAM_PLAYBACK]; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index c517064f5391..74baf1fce053 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2802,30 +2802,36 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (rtd->dai_link->dpcm_playback) { stream = SNDRV_PCM_STREAM_PLAYBACK; - for_each_rtd_cpu_dais(rtd, i, cpu_dai) - if (!snd_soc_dai_stream_valid(cpu_dai, - stream)) { - dev_err(rtd->card->dev, - "CPU DAI %s for rtd %s does not support playback\n", - cpu_dai->name, - rtd->dai_link->stream_name); - return -EINVAL; + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { + if (snd_soc_dai_stream_valid(cpu_dai, stream)) { + playback = 1; + break; } - playback = 1; + } + + if (!playback) { + dev_err(rtd->card->dev, + "No CPU DAIs support playback for stream %s\n", + rtd->dai_link->stream_name); + return -EINVAL; + } } if (rtd->dai_link->dpcm_capture) { stream = SNDRV_PCM_STREAM_CAPTURE; - for_each_rtd_cpu_dais(rtd, i, cpu_dai) - if (!snd_soc_dai_stream_valid(cpu_dai, - stream)) { - dev_err(rtd->card->dev, - "CPU DAI %s for rtd %s does not support capture\n", - cpu_dai->name, - rtd->dai_link->stream_name); - return -EINVAL; + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { + if (snd_soc_dai_stream_valid(cpu_dai, stream)) { + capture = 1; + break; } - capture = 1; + } + + if (!capture) { + dev_err(rtd->card->dev, + "No CPU DAIs support capture for stream %s\n", + rtd->dai_link->stream_name); + return -EINVAL; + } } } else { /* Adapt stream for codec2codec links */ |