From a613b63db233b6e7d46ec72f009c1cbb7db8be66 Mon Sep 17 00:00:00 2001 From: Elinor Montmasson Date: Thu, 27 Jun 2024 10:30:58 +0200 Subject: ASoC: fsl-asoc-card: add support for dai links with multiple codecs Add support for dai links using multiple codecs for multi-codec use cases. Co-developed-by: Philip-Dylan Gleonec Signed-off-by: Philip-Dylan Gleonec Signed-off-by: Elinor Montmasson Link: https://patch.msgid.link/20240627083104.123357-2-elinor.montmasson@savoirfairelinux.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl-asoc-card.c | 19 +++++++++++-------- 1 file changed, 11 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index eb67689dcd6e..e8003fbc8092 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -542,6 +542,7 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card) static int fsl_asoc_card_probe(struct platform_device *pdev) { struct device_node *cpu_np, *codec_np, *asrc_np; + struct snd_soc_dai_link_component *codec_comp; struct device_node *np = pdev->dev.of_node; struct platform_device *asrc_pdev = NULL; struct device_node *bitclkprovider = NULL; @@ -552,6 +553,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) const char *codec_dai_name; const char *codec_dev_name; u32 asrc_fmt = 0; + int codec_idx; u32 width; int ret; @@ -816,10 +818,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Normal DAI Link */ priv->dai_link[0].cpus->of_node = cpu_np; - priv->dai_link[0].codecs->dai_name = codec_dai_name; + priv->dai_link[0].codecs[0].dai_name = codec_dai_name; if (!fsl_asoc_card_is_ac97(priv)) - priv->dai_link[0].codecs->of_node = codec_np; + priv->dai_link[0].codecs[0].of_node = codec_np; else { u32 idx; @@ -830,11 +832,11 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) goto asrc_fail; } - priv->dai_link[0].codecs->name = + priv->dai_link[0].codecs[0].name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "ac97-codec.%u", (unsigned int)idx); - if (!priv->dai_link[0].codecs->name) { + if (!priv->dai_link[0].codecs[0].name) { ret = -ENOMEM; goto asrc_fail; } @@ -848,10 +850,11 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* DPCM DAI Links only if ASRC exists */ priv->dai_link[1].cpus->of_node = asrc_np; priv->dai_link[1].platforms->of_node = asrc_np; - priv->dai_link[2].codecs->dai_name = codec_dai_name; - priv->dai_link[2].codecs->of_node = codec_np; - priv->dai_link[2].codecs->name = - priv->dai_link[0].codecs->name; + for_each_link_codecs((&(priv->dai_link[2])), codec_idx, codec_comp) { + codec_comp->dai_name = priv->dai_link[0].codecs[codec_idx].dai_name; + codec_comp->of_node = priv->dai_link[0].codecs[codec_idx].of_node; + codec_comp->name = priv->dai_link[0].codecs[codec_idx].name; + } priv->dai_link[2].cpus->of_node = cpu_np; priv->dai_link[2].dai_fmt = priv->dai_fmt; priv->card.num_links = 3; -- cgit From c68fa0d9b0f8cc7c3ae7d29c02adbc97622a73f5 Mon Sep 17 00:00:00 2001 From: Elinor Montmasson Date: Thu, 27 Jun 2024 10:30:59 +0200 Subject: ASoC: fsl-asoc-card: add second dai link component for codecs Add a second dai link component for codecs that will be used for use cases with 2 codecs. It is needed for future integration of the SPDIF support, which will use spdif_receiver and spdif_transmitter drivers. To prevent deferring in use cases using only one codec, also set by default the number of codecs to 1 for the relevant dai links. Co-developed-by: Philip-Dylan Gleonec Signed-off-by: Philip-Dylan Gleonec Signed-off-by: Elinor Montmasson Link: https://patch.msgid.link/20240627083104.123357-3-elinor.montmasson@savoirfairelinux.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl-asoc-card.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index e8003fbc8092..805e2030bde4 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -296,7 +296,7 @@ static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, SND_SOC_DAILINK_DEFS(hifi, DAILINK_COMP_ARRAY(COMP_EMPTY()), - DAILINK_COMP_ARRAY(COMP_EMPTY()), + DAILINK_COMP_ARRAY(COMP_EMPTY(), COMP_EMPTY()), DAILINK_COMP_ARRAY(COMP_EMPTY())); SND_SOC_DAILINK_DEFS(hifi_fe, @@ -306,7 +306,7 @@ SND_SOC_DAILINK_DEFS(hifi_fe, SND_SOC_DAILINK_DEFS(hifi_be, DAILINK_COMP_ARRAY(COMP_EMPTY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); + DAILINK_COMP_ARRAY(COMP_EMPTY(), COMP_EMPTY())); static const struct snd_soc_dai_link fsl_asoc_card_dai[] = { /* Default ASoC DAI Link*/ @@ -622,6 +622,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) memcpy(priv->dai_link, fsl_asoc_card_dai, sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link)); + priv->dai_link[0].num_codecs = 1; + priv->dai_link[2].num_codecs = 1; priv->card.dapm_routes = audio_map; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map); -- cgit From fcc6ace84f1f6ce2211af25c3c8fb30a0fb2bb2c Mon Sep 17 00:00:00 2001 From: Elinor Montmasson Date: Thu, 27 Jun 2024 10:31:00 +0200 Subject: ASoC: fsl-asoc-card: add compatibility to use 2 codecs in dai-links Adapt the driver to work with configurations using two codecs or more. Modify fsl_asoc_card_probe() to handle use cases where 2 codecs are given in the device tree. This will be needed to add support for the SPDIF. Use cases using one codec will ignore any given codecs other than the first. Co-developed-by: Philip-Dylan Gleonec Signed-off-by: Philip-Dylan Gleonec Signed-off-by: Elinor Montmasson Link: https://patch.msgid.link/20240627083104.123357-4-elinor.montmasson@savoirfairelinux.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl-asoc-card.c | 279 ++++++++++++++++++++++++------------------ 1 file changed, 161 insertions(+), 118 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 805e2030bde4..87329731e02d 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -99,7 +99,7 @@ struct fsl_asoc_card_priv { struct simple_util_jack hp_jack; struct simple_util_jack mic_jack; struct platform_device *pdev; - struct codec_priv codec_priv; + struct codec_priv codec_priv[2]; struct cpu_priv cpu_priv; struct snd_soc_card card; u8 streams; @@ -172,10 +172,12 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - struct codec_priv *codec_priv = &priv->codec_priv; + struct codec_priv *codec_priv; + struct snd_soc_dai *codec_dai; struct cpu_priv *cpu_priv = &priv->cpu_priv; struct device *dev = rtd->card->dev; unsigned int pll_out; + int codec_idx; int ret; priv->sample_rate = params_rate(params); @@ -208,28 +210,32 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, } /* Specific configuration for PLL */ - if (codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) { - if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) - pll_out = priv->sample_rate * 384; - else - pll_out = priv->sample_rate * 256; + for_each_rtd_codec_dais(rtd, codec_idx, codec_dai) { + codec_priv = &priv->codec_priv[codec_idx]; - ret = snd_soc_dai_set_pll(snd_soc_rtd_to_codec(rtd, 0), - codec_priv->pll_id, - codec_priv->mclk_id, - codec_priv->mclk_freq, pll_out); - if (ret) { - dev_err(dev, "failed to start FLL: %d\n", ret); - goto fail; - } + if (codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) { + if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = priv->sample_rate * 384; + else + pll_out = priv->sample_rate * 256; - ret = snd_soc_dai_set_sysclk(snd_soc_rtd_to_codec(rtd, 0), - codec_priv->fll_id, - pll_out, SND_SOC_CLOCK_IN); + ret = snd_soc_dai_set_pll(codec_dai, + codec_priv->pll_id, + codec_priv->mclk_id, + codec_priv->mclk_freq, pll_out); + if (ret) { + dev_err(dev, "failed to start FLL: %d\n", ret); + goto fail; + } - if (ret && ret != -ENOTSUPP) { - dev_err(dev, "failed to set SYSCLK: %d\n", ret); - goto fail; + ret = snd_soc_dai_set_sysclk(codec_dai, + codec_priv->fll_id, + pll_out, SND_SOC_CLOCK_IN); + + if (ret && ret != -ENOTSUPP) { + dev_err(dev, "failed to set SYSCLK: %d\n", ret); + goto fail; + } } } @@ -244,28 +250,34 @@ static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct codec_priv *codec_priv = &priv->codec_priv; + struct codec_priv *codec_priv; + struct snd_soc_dai *codec_dai; struct device *dev = rtd->card->dev; + int codec_idx; int ret; priv->streams &= ~BIT(substream->stream); - if (!priv->streams && codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) { - /* Force freq to be free_freq to avoid error message in codec */ - ret = snd_soc_dai_set_sysclk(snd_soc_rtd_to_codec(rtd, 0), - codec_priv->mclk_id, - codec_priv->free_freq, - SND_SOC_CLOCK_IN); - if (ret) { - dev_err(dev, "failed to switch away from FLL: %d\n", ret); - return ret; - } + for_each_rtd_codec_dais(rtd, codec_idx, codec_dai) { + codec_priv = &priv->codec_priv[codec_idx]; - ret = snd_soc_dai_set_pll(snd_soc_rtd_to_codec(rtd, 0), - codec_priv->pll_id, 0, 0, 0); - if (ret && ret != -ENOTSUPP) { - dev_err(dev, "failed to stop FLL: %d\n", ret); - return ret; + if (!priv->streams && codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) { + /* Force freq to be free_freq to avoid error message in codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, + codec_priv->mclk_id, + codec_priv->free_freq, + SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to switch away from FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, + codec_priv->pll_id, 0, 0, 0); + if (ret && ret != -ENOTSUPP) { + dev_err(dev, "failed to stop FLL: %d\n", ret); + return ret; + } } } @@ -504,9 +516,10 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card) struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); struct snd_soc_pcm_runtime *rtd = list_first_entry( &card->rtd_list, struct snd_soc_pcm_runtime, list); - struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); - struct codec_priv *codec_priv = &priv->codec_priv; + struct snd_soc_dai *codec_dai; + struct codec_priv *codec_priv; struct device *dev = card->dev; + int codec_idx; int ret; if (fsl_asoc_card_is_ac97(priv)) { @@ -526,32 +539,37 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card) return 0; } - ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, - codec_priv->mclk_freq, SND_SOC_CLOCK_IN); - if (ret && ret != -ENOTSUPP) { - dev_err(dev, "failed to set sysclk in %s\n", __func__); - return ret; - } + for_each_rtd_codec_dais(rtd, codec_idx, codec_dai) { + codec_priv = &priv->codec_priv[codec_idx]; - if (!IS_ERR_OR_NULL(codec_priv->mclk)) - clk_prepare_enable(codec_priv->mclk); + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, + codec_priv->mclk_freq, SND_SOC_CLOCK_IN); + if (ret && ret != -ENOTSUPP) { + dev_err(dev, "failed to set sysclk in %s\n", __func__); + return ret; + } + + if (!IS_ERR_OR_NULL(codec_priv->mclk)) + clk_prepare_enable(codec_priv->mclk); + } return 0; } static int fsl_asoc_card_probe(struct platform_device *pdev) { - struct device_node *cpu_np, *codec_np, *asrc_np; + struct device_node *cpu_np, *asrc_np; struct snd_soc_dai_link_component *codec_comp; + struct device_node *codec_np[2]; struct device_node *np = pdev->dev.of_node; struct platform_device *asrc_pdev = NULL; struct device_node *bitclkprovider = NULL; struct device_node *frameprovider = NULL; struct platform_device *cpu_pdev; struct fsl_asoc_card_priv *priv; - struct device *codec_dev = NULL; - const char *codec_dai_name; - const char *codec_dev_name; + struct device *codec_dev[2] = { NULL, NULL }; + const char *codec_dai_name[2]; + const char *codec_dev_name[2]; u32 asrc_fmt = 0; int codec_idx; u32 width; @@ -580,21 +598,25 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) goto fail; } - codec_np = of_parse_phandle(np, "audio-codec", 0); - if (codec_np) { - struct platform_device *codec_pdev; - struct i2c_client *codec_i2c; + codec_np[0] = of_parse_phandle(np, "audio-codec", 0); + codec_np[1] = of_parse_phandle(np, "audio-codec", 1); - codec_i2c = of_find_i2c_device_by_node(codec_np); - if (codec_i2c) { - codec_dev = &codec_i2c->dev; - codec_dev_name = codec_i2c->name; - } - if (!codec_dev) { - codec_pdev = of_find_device_by_node(codec_np); - if (codec_pdev) { - codec_dev = &codec_pdev->dev; - codec_dev_name = codec_pdev->name; + for (codec_idx = 0; codec_idx < 2; codec_idx++) { + if (codec_np[codec_idx]) { + struct platform_device *codec_pdev; + struct i2c_client *codec_i2c; + + codec_i2c = of_find_i2c_device_by_node(codec_np[codec_idx]); + if (codec_i2c) { + codec_dev[codec_idx] = &codec_i2c->dev; + codec_dev_name[codec_idx] = codec_i2c->name; + } + if (!codec_dev[codec_idx]) { + codec_pdev = of_find_device_by_node(codec_np[codec_idx]); + if (codec_pdev) { + codec_dev[codec_idx] = &codec_pdev->dev; + codec_dev_name[codec_idx] = codec_pdev->name; + } } } } @@ -604,12 +626,14 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_pdev = of_find_device_by_node(asrc_np); /* Get the MCLK rate only, and leave it controlled by CODEC drivers */ - if (codec_dev) { - struct clk *codec_clk = clk_get(codec_dev, NULL); + for (codec_idx = 0; codec_idx < 2; codec_idx++) { + if (codec_dev[codec_idx]) { + struct clk *codec_clk = clk_get(codec_dev[codec_idx], NULL); - if (!IS_ERR(codec_clk)) { - priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); - clk_put(codec_clk); + if (!IS_ERR(codec_clk)) { + priv->codec_priv[codec_idx].mclk_freq = clk_get_rate(codec_clk); + clk_put(codec_clk); + } } } @@ -629,31 +653,33 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->card.num_dapm_routes = ARRAY_SIZE(audio_map); priv->card.driver_name = DRIVER_NAME; - priv->codec_priv.fll_id = -1; - priv->codec_priv.pll_id = -1; + for (codec_idx = 0; codec_idx < 2; codec_idx++) { + priv->codec_priv[codec_idx].fll_id = -1; + priv->codec_priv[codec_idx].pll_id = -1; + } /* Diversify the card configurations */ if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { - codec_dai_name = "cs42888"; - priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq; - priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; + codec_dai_name[0] = "cs42888"; + priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv[0].mclk_freq; + priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv[0].mclk_freq; priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT; priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT; priv->cpu_priv.slot_width = 32; priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC; } else if (of_device_is_compatible(np, "fsl,imx-audio-cs427x")) { - codec_dai_name = "cs4271-hifi"; - priv->codec_priv.mclk_id = CS427x_SYSCLK_MCLK; + codec_dai_name[0] = "cs4271-hifi"; + priv->codec_priv[0].mclk_id = CS427x_SYSCLK_MCLK; priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) { - codec_dai_name = "sgtl5000"; - priv->codec_priv.mclk_id = SGTL5000_SYSCLK; + codec_dai_name[0] = "sgtl5000"; + priv->codec_priv[0].mclk_id = SGTL5000_SYSCLK; priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; } else if (of_device_is_compatible(np, "fsl,imx-audio-tlv320aic32x4")) { - codec_dai_name = "tlv320aic32x4-hifi"; + codec_dai_name[0] = "tlv320aic32x4-hifi"; priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; } else if (of_device_is_compatible(np, "fsl,imx-audio-tlv320aic31xx")) { - codec_dai_name = "tlv320dac31xx-hifi"; + codec_dai_name[0] = "tlv320dac31xx-hifi"; priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; priv->dai_link[1].dpcm_capture = 0; priv->dai_link[2].dpcm_capture = 0; @@ -662,23 +688,23 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->card.dapm_routes = audio_map_tx; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx); } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) { - codec_dai_name = "wm8962"; - priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK; - priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; - priv->codec_priv.pll_id = WM8962_FLL; + codec_dai_name[0] = "wm8962"; + priv->codec_priv[0].mclk_id = WM8962_SYSCLK_MCLK; + priv->codec_priv[0].fll_id = WM8962_SYSCLK_FLL; + priv->codec_priv[0].pll_id = WM8962_FLL; priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) { - codec_dai_name = "wm8960-hifi"; - priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; - priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; + codec_dai_name[0] = "wm8960-hifi"; + priv->codec_priv[0].fll_id = WM8960_SYSCLK_AUTO; + priv->codec_priv[0].pll_id = WM8960_SYSCLK_AUTO; priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { - codec_dai_name = "ac97-hifi"; + codec_dai_name[0] = "ac97-hifi"; priv->dai_fmt = SND_SOC_DAIFMT_AC97; priv->card.dapm_routes = audio_map_ac97; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_ac97); } else if (of_device_is_compatible(np, "fsl,imx-audio-mqs")) { - codec_dai_name = "fsl-mqs-dai"; + codec_dai_name[0] = "fsl-mqs-dai"; priv->dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBC_CFC | SND_SOC_DAIFMT_NB_NF; @@ -687,7 +713,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->card.dapm_routes = audio_map_tx; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx); } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8524")) { - codec_dai_name = "wm8524-hifi"; + codec_dai_name[0] = "wm8524-hifi"; priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC; priv->dai_link[1].dpcm_capture = 0; priv->dai_link[2].dpcm_capture = 0; @@ -695,32 +721,32 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->card.dapm_routes = audio_map_tx; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx); } else if (of_device_is_compatible(np, "fsl,imx-audio-si476x")) { - codec_dai_name = "si476x-codec"; + codec_dai_name[0] = "si476x-codec"; priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC; priv->card.dapm_routes = audio_map_rx; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_rx); } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8958")) { - codec_dai_name = "wm8994-aif1"; + codec_dai_name[0] = "wm8994-aif1"; priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; - priv->codec_priv.mclk_id = WM8994_FLL_SRC_MCLK1; - priv->codec_priv.fll_id = WM8994_SYSCLK_FLL1; - priv->codec_priv.pll_id = WM8994_FLL1; - priv->codec_priv.free_freq = priv->codec_priv.mclk_freq; + priv->codec_priv[0].mclk_id = WM8994_FLL_SRC_MCLK1; + priv->codec_priv[0].fll_id = WM8994_SYSCLK_FLL1; + priv->codec_priv[0].pll_id = WM8994_FLL1; + priv->codec_priv[0].free_freq = priv->codec_priv[0].mclk_freq; priv->card.dapm_routes = NULL; priv->card.num_dapm_routes = 0; } else if (of_device_is_compatible(np, "fsl,imx-audio-nau8822")) { - codec_dai_name = "nau8822-hifi"; - priv->codec_priv.mclk_id = NAU8822_CLK_MCLK; - priv->codec_priv.fll_id = NAU8822_CLK_PLL; - priv->codec_priv.pll_id = NAU8822_CLK_PLL; + codec_dai_name[0] = "nau8822-hifi"; + priv->codec_priv[0].mclk_id = NAU8822_CLK_MCLK; + priv->codec_priv[0].fll_id = NAU8822_CLK_PLL; + priv->codec_priv[0].pll_id = NAU8822_CLK_PLL; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; - if (codec_dev) - priv->codec_priv.mclk = devm_clk_get(codec_dev, NULL); + if (codec_dev[0]) + priv->codec_priv[0].mclk = devm_clk_get(codec_dev[0], NULL); } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8904")) { - codec_dai_name = "wm8904-hifi"; - priv->codec_priv.mclk_id = WM8904_FLL_MCLK; - priv->codec_priv.fll_id = WM8904_CLK_FLL; - priv->codec_priv.pll_id = WM8904_FLL_MCLK; + codec_dai_name[0] = "wm8904-hifi"; + priv->codec_priv[0].mclk_id = WM8904_FLL_MCLK; + priv->codec_priv[0].fll_id = WM8904_CLK_FLL; + priv->codec_priv[0].pll_id = WM8904_FLL_MCLK; priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); @@ -732,18 +758,30 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) * Allow setting mclk-id from the device-tree node. Otherwise, the * default value for each card configuration is used. */ - of_property_read_u32(np, "mclk-id", &priv->codec_priv.mclk_id); + for_each_link_codecs((&(priv->dai_link[0])), codec_idx, codec_comp) { + of_property_read_u32_index(np, "mclk-id", codec_idx, + &priv->codec_priv[codec_idx].mclk_id); + } /* Format info from DT is optional. */ snd_soc_daifmt_parse_clock_provider_as_phandle(np, NULL, &bitclkprovider, &frameprovider); if (bitclkprovider || frameprovider) { unsigned int daifmt = snd_soc_daifmt_parse_format(np, NULL); + bool codec_bitclkprovider = false; + bool codec_frameprovider = false; + + for_each_link_codecs((&(priv->dai_link[0])), codec_idx, codec_comp) { + if (bitclkprovider && codec_np[codec_idx] == bitclkprovider) + codec_bitclkprovider = true; + if (frameprovider && codec_np[codec_idx] == frameprovider) + codec_frameprovider = true; + } - if (codec_np == bitclkprovider) - daifmt |= (codec_np == frameprovider) ? + if (codec_bitclkprovider) + daifmt |= (codec_frameprovider) ? SND_SOC_DAIFMT_CBP_CFP : SND_SOC_DAIFMT_CBP_CFC; else - daifmt |= (codec_np == frameprovider) ? + daifmt |= (codec_frameprovider) ? SND_SOC_DAIFMT_CBC_CFP : SND_SOC_DAIFMT_CBC_CFC; /* Override dai_fmt with value from DT */ @@ -759,7 +797,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) of_node_put(bitclkprovider); of_node_put(frameprovider); - if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) { + if (!fsl_asoc_card_is_ac97(priv) && !codec_dev[0]) { dev_dbg(&pdev->dev, "failed to find codec device\n"); ret = -EPROBE_DEFER; goto asrc_fail; @@ -798,7 +836,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) ret = snd_soc_of_parse_card_name(&priv->card, "model"); if (ret) { snprintf(priv->name, sizeof(priv->name), "%s-audio", - fsl_asoc_card_is_ac97(priv) ? "ac97" : codec_dev_name); + fsl_asoc_card_is_ac97(priv) ? "ac97" : codec_dev_name[0]); priv->card.name = priv->name; } priv->card.dai_link = priv->dai_link; @@ -820,11 +858,15 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Normal DAI Link */ priv->dai_link[0].cpus->of_node = cpu_np; - priv->dai_link[0].codecs[0].dai_name = codec_dai_name; + for_each_link_codecs((&(priv->dai_link[0])), codec_idx, codec_comp) { + codec_comp->dai_name = codec_dai_name[codec_idx]; + } - if (!fsl_asoc_card_is_ac97(priv)) - priv->dai_link[0].codecs[0].of_node = codec_np; - else { + if (!fsl_asoc_card_is_ac97(priv)) { + for_each_link_codecs((&(priv->dai_link[0])), codec_idx, codec_comp) { + codec_comp->of_node = codec_np[codec_idx]; + } + } else { u32 idx; ret = of_property_read_u32(cpu_np, "cell-index", &idx); @@ -926,7 +968,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_fail: of_node_put(asrc_np); - of_node_put(codec_np); + of_node_put(codec_np[0]); + of_node_put(codec_np[1]); put_device(&cpu_pdev->dev); fail: of_node_put(cpu_np); -- cgit From 6d174cc4f22461ad3fe383570527e86bf1948a2e Mon Sep 17 00:00:00 2001 From: Elinor Montmasson Date: Thu, 27 Jun 2024 10:31:01 +0200 Subject: ASoC: fsl-asoc-card: merge spdif support from imx-spdif.c The imx-spdif machine driver creates audio card to directly use an S/PDIF device. However, it doesn't support interacting with an ASRC. fsl-asoc-card already has the support to create audio card which can use the ASRC. Merge the S/PDIF support from imx-spdif into driver fsl-asoc-card to extend the support of S/PDIF audio card with the use of ASRC devices. fsl-asoc-card uses slightly different DT properties than imx-spdif: * the "spdif-controller" property from imx-spdif is named "audio-cpu" in fsl-asoc-card. * fsl-asoc-card uses codecs explicitly declared in DT with "audio-codec". With an SPDIF, codec drivers spdif_transmitter and spdif_receiver should be used. Driver imx-spdif used instead the dummy codec and a pair of boolean properties, "spdif-in" and "spdif-out". To keep backward compatibility, support for "spdif-controller", "spdif-in" and "spdif-out" is also added to fsl-asoc-card. However, it is recommended to use the new properties if possible. It is better to declare transmitter and/or receiver in DT than using the dummy codec. DTs using compatible "fsl,imx-audio-spdif" are still compatible, and fsl-asoc-card will behave the same as imx-spdif for these DTs. Signed-off-by: Elinor Montmasson Link: https://patch.msgid.link/20240627083104.123357-5-elinor.montmasson@savoirfairelinux.com Signed-off-by: Mark Brown --- arch/arm/configs/imx_v6_v7_defconfig | 1 - arch/arm64/configs/defconfig | 1 - sound/soc/fsl/Kconfig | 10 +--- sound/soc/fsl/Makefile | 2 - sound/soc/fsl/fsl-asoc-card.c | 85 ++++++++++++++++++++++++++++- sound/soc/fsl/imx-spdif.c | 103 ----------------------------------- 6 files changed, 84 insertions(+), 118 deletions(-) delete mode 100644 sound/soc/fsl/imx-spdif.c diff --git a/arch/arm/configs/imx_v6_v7_defconfig b/arch/arm/configs/imx_v6_v7_defconfig index cf2480dce285..ac5ae621b2af 100644 --- a/arch/arm/configs/imx_v6_v7_defconfig +++ b/arch/arm/configs/imx_v6_v7_defconfig @@ -311,7 +311,6 @@ CONFIG_SND_IMX_SOC=y CONFIG_SND_SOC_EUKREA_TLV320=y CONFIG_SND_SOC_IMX_ES8328=y CONFIG_SND_SOC_IMX_SGTL5000=y -CONFIG_SND_SOC_IMX_SPDIF=y CONFIG_SND_SOC_FSL_ASOC_CARD=y CONFIG_SND_SOC_AC97_CODEC=y CONFIG_SND_SOC_CS42XX8_I2C=y diff --git a/arch/arm64/configs/defconfig b/arch/arm64/configs/defconfig index 57a9abe78ee4..a6c9688fee0e 100644 --- a/arch/arm64/configs/defconfig +++ b/arch/arm64/configs/defconfig @@ -940,7 +940,6 @@ CONFIG_SND_SOC_FSL_MICFIL=m CONFIG_SND_SOC_FSL_EASRC=m CONFIG_SND_IMX_SOC=m CONFIG_SND_SOC_IMX_SGTL5000=m -CONFIG_SND_SOC_IMX_SPDIF=m CONFIG_SND_SOC_FSL_ASOC_CARD=m CONFIG_SND_SOC_IMX_AUDMIX=m CONFIG_SND_SOC_MT8183=m diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 8b7088623b6a..e283751abfef 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -303,15 +303,6 @@ config SND_SOC_IMX_SGTL5000 SND_SOC_FSL_ASOC_CARD and SND_SOC_SGTL5000 to use the newer driver. -config SND_SOC_IMX_SPDIF - tristate "SoC Audio support for i.MX boards with S/PDIF" - select SND_SOC_IMX_PCM_DMA - select SND_SOC_FSL_SPDIF - help - SoC Audio support for i.MX boards with S/PDIF - Say Y if you want to add support for SoC audio on an i.MX board with - a S/DPDIF. - config SND_SOC_FSL_ASOC_CARD tristate "Generic ASoC Sound Card with ASRC support" depends on OF && I2C @@ -323,6 +314,7 @@ config SND_SOC_FSL_ASOC_CARD select SND_SOC_FSL_ESAI select SND_SOC_FSL_SAI select SND_SOC_FSL_SSI + select SND_SOC_FSL_SPDIF select SND_SOC_TLV320AIC31XX select SND_SOC_WM8994 select MFD_WM8994 diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 2a61e2f96438..ad97244b5cc3 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -67,7 +67,6 @@ obj-$(CONFIG_SND_SOC_IMX_PCM_RPMSG) += imx-pcm-rpmsg.o snd-soc-eukrea-tlv320-y := eukrea-tlv320.o snd-soc-imx-es8328-y := imx-es8328.o snd-soc-imx-sgtl5000-y := imx-sgtl5000.o -snd-soc-imx-spdif-y := imx-spdif.o snd-soc-imx-audmix-y := imx-audmix.o snd-soc-imx-hdmi-y := imx-hdmi.o snd-soc-imx-rpmsg-y := imx-rpmsg.o @@ -76,7 +75,6 @@ snd-soc-imx-card-y := imx-card.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o -obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_AUDMIX) += snd-soc-imx-audmix.o obj-$(CONFIG_SND_SOC_IMX_HDMI) += snd-soc-imx-hdmi.o obj-$(CONFIG_SND_SOC_IMX_RPMSG) += snd-soc-imx-rpmsg.o diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 87329731e02d..82df887b3af5 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -477,6 +477,75 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, return 0; } +static int fsl_asoc_card_spdif_init(struct device_node *codec_np[], + struct device_node *cpu_np, + const char *codec_dai_name[], + struct fsl_asoc_card_priv *priv) +{ + struct device *dev = &priv->pdev->dev; + struct device_node *np = dev->of_node; + + if (!of_node_name_eq(cpu_np, "spdif")) { + dev_err(dev, "CPU phandle invalid, should be an SPDIF device\n"); + return -EINVAL; + } + + priv->dai_link[0].playback_only = true; + priv->dai_link[0].capture_only = true; + + for (int i = 0; i < 2; i++) { + if (!codec_np[i]) + break; + + if (of_device_is_compatible(codec_np[i], "linux,spdif-dit")) { + priv->dai_link[0].capture_only = false; + codec_dai_name[i] = "dit-hifi"; + } else if (of_device_is_compatible(codec_np[i], "linux,spdif-dir")) { + priv->dai_link[0].playback_only = false; + codec_dai_name[i] = "dir-hifi"; + } + } + + // Old SPDIF DT binding + if (!codec_np[0]) { + codec_dai_name[0] = snd_soc_dummy_dlc.dai_name; + if (of_property_read_bool(np, "spdif-out")) + priv->dai_link[0].capture_only = false; + if (of_property_read_bool(np, "spdif-in")) + priv->dai_link[0].playback_only = false; + } + + if (priv->dai_link[0].playback_only && priv->dai_link[0].capture_only) { + dev_err(dev, "no enabled S/PDIF DAI link\n"); + return -EINVAL; + } + + if (priv->dai_link[0].playback_only) { + priv->dai_link[1].dpcm_capture = false; + priv->dai_link[2].dpcm_capture = false; + priv->card.dapm_routes = audio_map_tx; + priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx); + } else if (priv->dai_link[0].capture_only) { + priv->dai_link[1].dpcm_playback = false; + priv->dai_link[2].dpcm_playback = false; + priv->card.dapm_routes = audio_map_rx; + priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_rx); + } + + // No DAPM routes with old bindings and dummy codec + if (!codec_np[0]) { + priv->card.dapm_routes = NULL; + priv->card.num_dapm_routes = 0; + } + + if (codec_np[0] && codec_np[1]) { + priv->dai_link[0].num_codecs = 2; + priv->dai_link[2].num_codecs = 2; + } + + return 0; +} + static int hp_jack_event(struct notifier_block *nb, unsigned long event, void *data) { @@ -582,9 +651,11 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->pdev = pdev; cpu_np = of_parse_phandle(np, "audio-cpu", 0); - /* Give a chance to old DT binding */ + /* Give a chance to old DT bindings */ if (!cpu_np) cpu_np = of_parse_phandle(np, "ssi-controller", 0); + if (!cpu_np) + cpu_np = of_parse_phandle(np, "spdif-controller", 0); if (!cpu_np) { dev_err(&pdev->dev, "CPU phandle missing or invalid\n"); ret = -EINVAL; @@ -748,6 +819,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->codec_priv[0].fll_id = WM8904_CLK_FLL; priv->codec_priv[0].pll_id = WM8904_FLL_MCLK; priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; + } else if (of_device_is_compatible(np, "fsl,imx-audio-spdif")) { + ret = fsl_asoc_card_spdif_init(codec_np, cpu_np, codec_dai_name, priv); + if (ret) + goto asrc_fail; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); ret = -EINVAL; @@ -797,7 +872,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) of_node_put(bitclkprovider); of_node_put(frameprovider); - if (!fsl_asoc_card_is_ac97(priv) && !codec_dev[0]) { + if (!fsl_asoc_card_is_ac97(priv) && !codec_dev[0] + && codec_dai_name[0] != snd_soc_dummy_dlc.dai_name) { dev_dbg(&pdev->dev, "failed to find codec device\n"); ret = -EPROBE_DEFER; goto asrc_fail; @@ -862,6 +938,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) codec_comp->dai_name = codec_dai_name[codec_idx]; } + // Old SPDIF DT binding support + if (codec_dai_name[0] == snd_soc_dummy_dlc.dai_name) + priv->dai_link[0].codecs[0].name = snd_soc_dummy_dlc.name; + if (!fsl_asoc_card_is_ac97(priv)) { for_each_link_codecs((&(priv->dai_link[0])), codec_idx, codec_comp) { codec_comp->of_node = codec_np[codec_idx]; @@ -992,6 +1072,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-wm8958", }, { .compatible = "fsl,imx-audio-nau8822", }, { .compatible = "fsl,imx-audio-wm8904", }, + { .compatible = "fsl,imx-audio-spdif", }, {} }; MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids); diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c deleted file mode 100644 index 1e57939a7e29..000000000000 --- a/sound/soc/fsl/imx-spdif.c +++ /dev/null @@ -1,103 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0+ -// -// Copyright (C) 2013 Freescale Semiconductor, Inc. - -#include -#include -#include - -struct imx_spdif_data { - struct snd_soc_dai_link dai; - struct snd_soc_card card; -}; - -static int imx_spdif_audio_probe(struct platform_device *pdev) -{ - struct device_node *spdif_np, *np = pdev->dev.of_node; - struct imx_spdif_data *data; - struct snd_soc_dai_link_component *comp; - int ret = 0; - - spdif_np = of_parse_phandle(np, "spdif-controller", 0); - if (!spdif_np) { - dev_err(&pdev->dev, "failed to find spdif-controller\n"); - ret = -EINVAL; - goto end; - } - - data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); - comp = devm_kzalloc(&pdev->dev, sizeof(*comp), GFP_KERNEL); - if (!data || !comp) { - ret = -ENOMEM; - goto end; - } - - /* - * CPU == Platform - * platform is using soc-generic-dmaengine-pcm - */ - data->dai.cpus = - data->dai.platforms = comp; - data->dai.codecs = &snd_soc_dummy_dlc; - - data->dai.num_cpus = 1; - data->dai.num_codecs = 1; - data->dai.num_platforms = 1; - - data->dai.name = "S/PDIF PCM"; - data->dai.stream_name = "S/PDIF PCM"; - data->dai.cpus->of_node = spdif_np; - data->dai.playback_only = true; - data->dai.capture_only = true; - - if (of_property_read_bool(np, "spdif-out")) - data->dai.capture_only = false; - - if (of_property_read_bool(np, "spdif-in")) - data->dai.playback_only = false; - - if (data->dai.playback_only && data->dai.capture_only) { - dev_err(&pdev->dev, "no enabled S/PDIF DAI link\n"); - goto end; - } - - data->card.dev = &pdev->dev; - data->card.dai_link = &data->dai; - data->card.num_links = 1; - data->card.owner = THIS_MODULE; - - ret = snd_soc_of_parse_card_name(&data->card, "model"); - if (ret) - goto end; - - ret = devm_snd_soc_register_card(&pdev->dev, &data->card); - if (ret) - dev_err_probe(&pdev->dev, ret, "snd_soc_register_card failed\n"); - -end: - of_node_put(spdif_np); - - return ret; -} - -static const struct of_device_id imx_spdif_dt_ids[] = { - { .compatible = "fsl,imx-audio-spdif", }, - { /* sentinel */ } -}; -MODULE_DEVICE_TABLE(of, imx_spdif_dt_ids); - -static struct platform_driver imx_spdif_driver = { - .driver = { - .name = "imx-spdif", - .pm = &snd_soc_pm_ops, - .of_match_table = imx_spdif_dt_ids, - }, - .probe = imx_spdif_audio_probe, -}; - -module_platform_driver(imx_spdif_driver); - -MODULE_AUTHOR("Freescale Semiconductor, Inc."); -MODULE_DESCRIPTION("Freescale i.MX S/PDIF machine driver"); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:imx-spdif"); -- cgit From 4359caadd17fbde64d656c10bd6f2dc91b675a11 Mon Sep 17 00:00:00 2001 From: Elinor Montmasson Date: Thu, 27 Jun 2024 10:31:02 +0200 Subject: ASoC: dt-bindings: update fsl-asoc-card bindings after imx-spdif merge The S/PDIF audio card support with compatible "fsl,imx-audio-spdif" was merged from imx-spdif into the fsl-asoc-card driver. It makes possible to use an S/PDIF with an ASRC. This merge introduces new DT bindings to use with compatible "fsl,imx-audio-spdif" to follow the way fsl-asoc-card works: * the "spdif-controller" property from imx-spdif is named "audio-cpu" in fsl-asoc-card. * fsl-asoc-card uses codecs explicitly declared in DT with "audio-codec". With an SPDIF, codec drivers spdif_transmitter and spdif_receiver should be used. Driver imx-spdif used instead the dummy codec and a pair of boolean properties, "spdif-in" and "spdif-out". In an upcoming commit, in-tree DTs will be modified to follow these new properties: * Property "spdif-controller" will be renamed "audio-cpu". * spdif_transmitter and spdif_receiver nodes will be declared and linked to the fsl-asoc-card node with the property "audio-codec". To keep backward compatibility with other DTs, support for "spdif-controller", "spdif-in" and "spdif-out" properties is kept. However, it is recommended to use the new properties if possible. It is better to declare transmitter and/or receiver in DT than using the dummy codec. DTs using compatible "fsl,imx-audio-spdif" are still supported, and fsl-asoc-card will behave the same as imx-spdif for these DTs. Signed-off-by: Elinor Montmasson Reviewed-by: Rob Herring (Arm) Link: https://patch.msgid.link/20240627083104.123357-6-elinor.montmasson@savoirfairelinux.com Signed-off-by: Mark Brown --- .../bindings/sound/fsl,imx-audio-spdif.yaml | 66 ---------------------- .../devicetree/bindings/sound/fsl-asoc-card.yaml | 53 +++++++++++++++-- 2 files changed, 49 insertions(+), 70 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml diff --git a/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml deleted file mode 100644 index 5fc543d02ecb..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml +++ /dev/null @@ -1,66 +0,0 @@ -# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) -%YAML 1.2 ---- -$id: http://devicetree.org/schemas/sound/fsl,imx-audio-spdif.yaml# -$schema: http://devicetree.org/meta-schemas/core.yaml# - -title: Freescale i.MX audio complex with S/PDIF transceiver - -maintainers: - - Shengjiu Wang - -properties: - compatible: - oneOf: - - items: - - enum: - - fsl,imx-sabreauto-spdif - - fsl,imx6sx-sdb-spdif - - const: fsl,imx-audio-spdif - - enum: - - fsl,imx-audio-spdif - - model: - $ref: /schemas/types.yaml#/definitions/string - description: User specified audio sound card name - - spdif-controller: - $ref: /schemas/types.yaml#/definitions/phandle - description: The phandle of the i.MX S/PDIF controller - - spdif-out: - type: boolean - description: - If present, the transmitting function of S/PDIF will be enabled, - indicating there's a physical S/PDIF out connector or jack on the - board or it's connecting to some other IP block, such as an HDMI - encoder or display-controller. - - spdif-in: - type: boolean - description: - If present, the receiving function of S/PDIF will be enabled, - indicating there is a physical S/PDIF in connector/jack on the board. - -required: - - compatible - - model - - spdif-controller - -anyOf: - - required: - - spdif-in - - required: - - spdif-out - -additionalProperties: false - -examples: - - | - sound-spdif { - compatible = "fsl,imx-audio-spdif"; - model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-out; - spdif-in; - }; diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml index 9922664d5ccc..92aa47ec72c7 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml @@ -65,6 +65,11 @@ properties: - fsl,imx-audio-sgtl5000 - fsl,imx-audio-wm8960 - fsl,imx-audio-wm8962 + - items: + - enum: + - fsl,imx-sabreauto-spdif + - fsl,imx6sx-sdb-spdif + - const: fsl,imx-audio-spdif - items: - enum: - fsl,imx-audio-ac97 @@ -81,6 +86,7 @@ properties: - fsl,imx-audio-wm8960 - fsl,imx-audio-wm8962 - fsl,imx-audio-wm8958 + - fsl,imx-audio-spdif model: $ref: /schemas/types.yaml#/definitions/string @@ -93,8 +99,15 @@ properties: need to add ASRC support via DPCM. audio-codec: - $ref: /schemas/types.yaml#/definitions/phandle - description: The phandle of an audio codec + $ref: /schemas/types.yaml#/definitions/phandle-array + description: | + The phandle of an audio codec. + With "fsl,imx-audio-spdif", either SPDIF audio codec spdif_transmitter, + spdif_receiver or both. + minItems: 1 + maxItems: 2 + items: + maxItems: 1 audio-cpu: $ref: /schemas/types.yaml#/definitions/phandle @@ -150,8 +163,10 @@ properties: description: dai-link uses bit clock inversion. mclk-id: - $ref: /schemas/types.yaml#/definitions/uint32 - description: main clock id, specific for each card configuration. + $ref: /schemas/types.yaml#/definitions/uint32-array + description: Main clock id for each codec, specific for each card configuration. + minItems: 1 + maxItems: 2 mux-int-port: $ref: /schemas/types.yaml#/definitions/uint32 @@ -167,6 +182,27 @@ properties: $ref: /schemas/types.yaml#/definitions/phandle description: The phandle of an CPU DAI controller + spdif-controller: + $ref: /schemas/types.yaml#/definitions/phandle + deprecated: true + description: The phandle of an S/PDIF CPU DAI controller. + + spdif-out: + type: boolean + deprecated: true + description: | + If present, the transmitting function of S/PDIF will be enabled, + indicating there's a physical S/PDIF out connector or jack on the + board or it's connecting to some other IP block, such as an HDMI + encoder or display-controller. + + spdif-in: + type: boolean + deprecated: true + description: | + If present, the receiving function of S/PDIF will be enabled, + indicating there is a physical S/PDIF in connector/jack on the board. + required: - compatible - model @@ -195,3 +231,12 @@ examples: "AIN2L", "Line In Jack", "AIN2R", "Line In Jack"; }; + + - | + sound-spdif-asrc { + compatible = "fsl,imx-audio-spdif"; + model = "spdif-asrc-audio"; + audio-cpu = <&spdif>; + audio-asrc = <&easrc>; + audio-codec = <&spdifdit>, <&spdifdir>; + }; -- cgit