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no-senders; approved; emergency; loop; banned-address; member-moderation; header-match-alsa-devel.alsa-project.org-0; header-match-alsa-devel.alsa-project.org-1; nonmember-moderation; administrivia; implicit-dest; max-recipients; max-size; news-moderation; no-subject; digests; suspicious-header X-Mailman-Version: 3.3.9 Precedence: list List-Id: "Alsa-devel mailing list for ALSA developers - http://www.alsa-project.org" Archived-At: List-Archive: List-Help: List-Owner: List-Post: List-Subscribe: List-Unsubscribe: priv->pdev pointer was set after being used in fsl_asoc_card_audmux_init(). Move this assignment at the start of the probe function, so sub-functions can correctly use pdev through priv. fsl_asoc_card_audmux_init() dereferences priv->pdev to get access to the dev struct, used with dev_err macros. As priv is zero-initialised, there would be a NULL pointer dereference. Note that if priv->dev is dereferenced before assignment but never used, for example if there is no error to be printed, the driver won't crash probably due to compiler optimisations. Fixes: 708b4351f08c ("ASoC: fsl: Add Freescale Generic ASoC Sound Card with ASRC support") Signed-off-by: Elinor Montmasson --- sound/soc/fsl/fsl-asoc-card.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 5ddc0c2fe53f..eb67689dcd6e 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -559,6 +559,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) if (!priv) return -ENOMEM; + priv->pdev = pdev; + cpu_np = of_parse_phandle(np, "audio-cpu", 0); /* Give a chance to old DT binding */ if (!cpu_np) @@ -787,7 +789,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) } /* Initialize sound card */ - priv->pdev = pdev; priv->card.dev = &pdev->dev; priv->card.owner = THIS_MODULE; ret = snd_soc_of_parse_card_name(&priv->card, "model"); From patchwork Thu Jun 20 13:25:04 2024 Content-Type: text/plain; charset="utf-8" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit X-Patchwork-Submitter: Elinor Montmasson X-Patchwork-Id: 806018 Return-Path: X-Spam-Checker-Version: SpamAssassin 3.4.0 (2014-02-07) on aws-us-west-2-korg-lkml-1.web.codeaurora.org Received: from alsa0.perex.cz (alsa0.perex.cz [77.48.224.243]) (using TLSv1.2 with cipher ECDHE-RSA-AES256-GCM-SHA384 (256/256 bits)) (No client certificate requested) by smtp.lore.kernel.org (Postfix) with ESMTPS id DFD03C27C79 for ; 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Thu, 20 Jun 2024 09:26:26 -0400 (EDT) Received: from gerard.rennes.sfl (lmontsouris-657-1-69-118.w80-15.abo.wanadoo.fr [80.15.101.118]) by mail.savoirfairelinux.com (Postfix) with ESMTPSA id 34E9A9C5625; Thu, 20 Jun 2024 09:26:24 -0400 (EDT) From: Elinor Montmasson To: Liam Girdwood , Mark Brown , Rob Herring , Krzysztof Kozlowski , Conor Dooley , Shawn Guo , Sascha Hauer , Fabio Estevam , Russell King , Catalin Marinas , Will Deacon , Jaroslav Kysela , Takashi Iwai , Shengjiu Wang , Xiubo Li , Nicolin Chen Cc: Pengutronix Kernel Team , linux-sound@vger.kernel.org, devicetree@vger.kernel.org, imx@lists.linux.dev, linux-arm-kernel@lists.infradead.org, linux-kernel@vger.kernel.org, alsa-devel@alsa-project.org, linuxppc-dev@lists.ozlabs.org, Elinor Montmasson , Philip-Dylan Subject: [PATCHv5 2/9] ASoC: fsl-asoc-card: add support for dai links with multiple codecs Date: Thu, 20 Jun 2024 15:25:04 +0200 Message-Id: <20240620132511.4291-3-elinor.montmasson@savoirfairelinux.com> X-Mailer: git-send-email 2.34.1 In-Reply-To: <20240620132511.4291-1-elinor.montmasson@savoirfairelinux.com> References: <20240620132511.4291-1-elinor.montmasson@savoirfairelinux.com> MIME-Version: 1.0 Message-ID-Hash: K5FVODGLSTWUG3VK32HUVE2EJ4LFSNBP X-Message-ID-Hash: K5FVODGLSTWUG3VK32HUVE2EJ4LFSNBP X-MailFrom: elinor.montmasson@savoirfairelinux.com X-Mailman-Rule-Misses: dmarc-mitigation; no-senders; approved; emergency; loop; banned-address; member-moderation; header-match-alsa-devel.alsa-project.org-0; header-match-alsa-devel.alsa-project.org-1; nonmember-moderation; administrivia; implicit-dest; max-recipients; max-size; news-moderation; no-subject; digests; suspicious-header X-Mailman-Version: 3.3.9 Precedence: list List-Id: "Alsa-devel mailing list for ALSA developers - http://www.alsa-project.org" Archived-At: List-Archive: List-Help: List-Owner: List-Post: List-Subscribe: List-Unsubscribe: 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 --- 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; From patchwork Thu Jun 20 13:25:05 2024 Content-Type: text/plain; charset="utf-8" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit X-Patchwork-Submitter: Elinor Montmasson X-Patchwork-Id: 806734 Return-Path: X-Spam-Checker-Version: SpamAssassin 3.4.0 (2014-02-07) on aws-us-west-2-korg-lkml-1.web.codeaurora.org Received: from alsa0.perex.cz (alsa0.perex.cz [77.48.224.243]) (using TLSv1.2 with cipher ECDHE-RSA-AES256-GCM-SHA384 (256/256 bits)) (No client certificate requested) by smtp.lore.kernel.org (Postfix) with ESMTPS id 83F87C2BA18 for ; Thu, 20 Jun 2024 13:27:41 +0000 (UTC) Received: from alsa1.perex.cz (alsa1.perex.cz [207.180.221.201]) (using TLSv1.2 with cipher ADH-AES256-GCM-SHA384 (256/256 bits)) (No client certificate requested) by alsa0.perex.cz (Postfix) with ESMTPS id 4927AE66; 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no-senders; approved; emergency; loop; banned-address; member-moderation; header-match-alsa-devel.alsa-project.org-0; header-match-alsa-devel.alsa-project.org-1; nonmember-moderation; administrivia; implicit-dest; max-recipients; max-size; news-moderation; no-subject; digests; suspicious-header X-Mailman-Version: 3.3.9 Precedence: list List-Id: "Alsa-devel mailing list for ALSA developers - http://www.alsa-project.org" Archived-At: List-Archive: List-Help: List-Owner: List-Post: List-Subscribe: List-Unsubscribe: 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 --- 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); From patchwork Thu Jun 20 13:25:06 2024 Content-Type: text/plain; charset="utf-8" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit X-Patchwork-Submitter: Elinor Montmasson X-Patchwork-Id: 806017 Return-Path: X-Spam-Checker-Version: SpamAssassin 3.4.0 (2014-02-07) on aws-us-west-2-korg-lkml-1.web.codeaurora.org Received: from alsa0.perex.cz (alsa0.perex.cz [77.48.224.243]) (using TLSv1.2 with cipher ECDHE-RSA-AES256-GCM-SHA384 (256/256 bits)) (No client certificate requested) by smtp.lore.kernel.org (Postfix) with ESMTPS id B3EECC2BA18 for ; Thu, 20 Jun 2024 13:28:00 +0000 (UTC) Received: from alsa1.perex.cz (alsa1.perex.cz [207.180.221.201]) (using TLSv1.2 with cipher ADH-AES256-GCM-SHA384 (256/256 bits)) (No client certificate requested) by alsa0.perex.cz (Postfix) with ESMTPS id DB2D1E64; Thu, 20 Jun 2024 15:27:48 +0200 (CEST) DKIM-Filter: OpenDKIM Filter v2.11.0 alsa0.perex.cz DB2D1E64 DKIM-Signature: v=1; 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no-senders; approved; emergency; loop; banned-address; member-moderation; header-match-alsa-devel.alsa-project.org-0; header-match-alsa-devel.alsa-project.org-1; nonmember-moderation; administrivia; implicit-dest; max-recipients; max-size; news-moderation; no-subject; digests; suspicious-header X-Mailman-Version: 3.3.9 Precedence: list List-Id: "Alsa-devel mailing list for ALSA developers - http://www.alsa-project.org" Archived-At: List-Archive: List-Help: List-Owner: List-Post: List-Subscribe: List-Unsubscribe: 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 --- 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); 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no-senders; approved; emergency; loop; banned-address; member-moderation; header-match-alsa-devel.alsa-project.org-0; header-match-alsa-devel.alsa-project.org-1; nonmember-moderation; administrivia; implicit-dest; max-recipients; max-size; news-moderation; no-subject; digests; suspicious-header X-Mailman-Version: 3.3.9 Precedence: list List-Id: "Alsa-devel mailing list for ALSA developers - http://www.alsa-project.org" Archived-At: List-Archive: List-Help: List-Owner: List-Post: List-Subscribe: List-Unsubscribe: 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 fsl-asoc-card to extend the support of S/PDIF audio card with the use of ASRC devices. It also substitutes the use of the dummy codec in imx-spdif with the existing spdif_transmitter and spdif_receiver codec drivers. Signed-off-by: Elinor Montmasson --- 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 | 58 +++++++++++++++ sound/soc/fsl/imx-spdif.c | 103 --------------------------- 6 files changed, 59 insertions(+), 116 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 9a371d4496c2..e3b2bfb016c8 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -296,15 +296,6 @@ 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 @@ -316,6 +307,7 @@ 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 2fe78eed3a48..1ae181b24a88 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -65,7 +65,6 @@ obj-$(CONFIG_SND_SOC_IMX_PCM_RPMSG) += imx-pcm-rpmsg.o snd-soc-eukrea-tlv320-y := eukrea-tlv320.o snd-soc-imx-es8328-y := imx-es8328.o snd-soc-imx-sgtl5000-y := imx-sgtl5000.o -snd-soc-imx-spdif-y := imx-spdif.o snd-soc-imx-audmix-y := imx-audmix.o snd-soc-imx-hdmi-y := imx-hdmi.o snd-soc-imx-rpmsg-y := imx-rpmsg.o @@ -74,7 +73,6 @@ snd-soc-imx-card-y := imx-card.o obj-$(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..c0e600525680 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -477,6 +477,59 @@ 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; + + 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"; + } + } + + 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); + } + + 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) { @@ -748,6 +801,10 @@ 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; @@ -992,6 +1049,7 @@ 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"); 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no-senders; approved; emergency; loop; banned-address; member-moderation; header-match-alsa-devel.alsa-project.org-0; header-match-alsa-devel.alsa-project.org-1; nonmember-moderation; administrivia; implicit-dest; max-recipients; max-size; news-moderation; no-subject; digests; suspicious-header X-Mailman-Version: 3.3.9 Precedence: list List-Id: "Alsa-devel mailing list for ALSA developers - http://www.alsa-project.org" Archived-At: List-Archive: List-Help: List-Owner: List-Post: List-Subscribe: List-Unsubscribe: The S/PDIF audio card support was merged from imx-spdif into the fsl-asoc-card driver, making it possible to use an S/PDIF with an ASRC. Add the new compatible and update properties. Signed-off-by: Elinor Montmasson --- .../bindings/sound/fsl-asoc-card.yaml | 30 ++++++++++++++++--- 1 file changed, 26 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml index 9922664d5ccc..f2e28b32808e 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml @@ -33,6 +33,7 @@ properties: - items: - enum: - fsl,imx-sgtl5000 + - fsl,imx-sabreauto-spdif - fsl,imx25-pdk-sgtl5000 - fsl,imx53-cpuvo-sgtl5000 - fsl,imx51-babbage-sgtl5000 @@ -54,6 +55,7 @@ properties: - fsl,imx6q-ventana-sgtl5000 - fsl,imx6sl-evk-wm8962 - fsl,imx6sx-sdb-mqs + - fsl,imx6sx-sdb-spdif - fsl,imx6sx-sdb-wm8962 - fsl,imx7d-evk-wm8960 - karo,tx53-audio-sgtl5000 @@ -65,6 +67,7 @@ properties: - fsl,imx-audio-sgtl5000 - fsl,imx-audio-wm8960 - fsl,imx-audio-wm8962 + - fsl,imx-audio-spdif - items: - enum: - fsl,imx-audio-ac97 @@ -81,6 +84,7 @@ 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 +97,15 @@ 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 +161,10 @@ 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: $ref: /schemas/types.yaml#/definitions/uint32 @@ -195,3 +208,12 @@ examples: "AIN2L", "Line In Jack", "AIN2R", "Line In Jack"; 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no-senders; approved; emergency; loop; banned-address; member-moderation; header-match-alsa-devel.alsa-project.org-0; header-match-alsa-devel.alsa-project.org-1; nonmember-moderation; administrivia; implicit-dest; max-recipients; max-size; news-moderation; no-subject; digests; suspicious-header X-Mailman-Version: 3.3.9 Precedence: list List-Id: "Alsa-devel mailing list for ALSA developers - http://www.alsa-project.org" Archived-At: List-Archive: List-Help: List-Owner: List-Post: List-Subscribe: List-Unsubscribe: imx-audio-spdif was merged into the fsl-asoc-card driver, and therefore removed. Signed-off-by: Elinor Montmasson --- .../bindings/sound/fsl,imx-audio-spdif.yaml | 66 ------------------- 1 file changed, 66 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml diff --git a/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml deleted file mode 100644 index 5fc543d02ecb..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml +++ /dev/null @@ -1,66 +0,0 @@ -# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) -%YAML 1.2 ---- -$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"; 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no-senders; approved; emergency; loop; banned-address; member-moderation; header-match-alsa-devel.alsa-project.org-0; header-match-alsa-devel.alsa-project.org-1; nonmember-moderation; administrivia; implicit-dest; max-recipients; max-size; news-moderation; no-subject; digests; suspicious-header X-Mailman-Version: 3.3.9 Precedence: list List-Id: "Alsa-devel mailing list for ALSA developers - http://www.alsa-project.org" Archived-At: List-Archive: List-Help: List-Owner: List-Post: List-Subscribe: List-Unsubscribe: Following merge of imx-spdif driver into fsl-asoc-card: * update properties to match those used by fsl-asoc-card. * S/PDIF in/out dummy codecs must now be declared explicitly, add and use them. These modifications were tested only on an imx8mn-evk board. Signed-off-by: Elinor Montmasson --- arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi | 15 +++++++++--- arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi | 15 +++++++++--- arch/arm64/boot/dts/freescale/imx8mq-evk.dts | 24 +++++++++++++++---- 3 files changed, 43 insertions(+), 11 deletions(-) diff --git a/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi b/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi index 90d1901df2b1..348855a41852 100644 --- a/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi +++ b/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi @@ -180,12 +180,21 @@ cpu { }; }; + spdif_out: spdif-out { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dit"; + }; + + spdif_in: spdif-in { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dir"; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif1>; - spdif-out; - spdif-in; + audio-cpu = <&spdif1>; + audio-codec = <&spdif_out>, <&spdif_in>; }; }; diff --git a/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi b/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi index 9e0259ddf4bc..6a47e09703a7 100644 --- a/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi +++ b/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi @@ -124,12 +124,21 @@ sound-wm8524 { "Line Out Jack", "LINEVOUTR"; }; + spdif_out: spdif-out { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dit"; + }; + + spdif_in: spdif-in { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dir"; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif1>; - spdif-out; - spdif-in; + audio-cpu = <&spdif1>; + audio-codec = <&spdif_out>, <&spdif_in>; }; sound-micfil { diff --git a/arch/arm64/boot/dts/freescale/imx8mq-evk.dts b/arch/arm64/boot/dts/freescale/imx8mq-evk.dts index 7507548cdb16..b953865f0b46 100644 --- a/arch/arm64/boot/dts/freescale/imx8mq-evk.dts +++ b/arch/arm64/boot/dts/freescale/imx8mq-evk.dts @@ -125,19 +125,33 @@ link_codec: simple-audio-card,codec { }; }; + spdif_out: spdif-out { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dit"; + }; + + spdif_in: spdif-in { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dir"; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif1>; - spdif-out; 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no-senders; approved; emergency; loop; banned-address; member-moderation; header-match-alsa-devel.alsa-project.org-0; header-match-alsa-devel.alsa-project.org-1; nonmember-moderation; administrivia; implicit-dest; max-recipients; max-size; news-moderation; no-subject; digests; suspicious-header X-Mailman-Version: 3.3.9 Precedence: list List-Id: "Alsa-devel mailing list for ALSA developers - http://www.alsa-project.org" Archived-At: List-Archive: List-Help: List-Owner: List-Post: List-Subscribe: List-Unsubscribe: Following merge of imx-spdif driver into fsl-asoc-card: * update properties to match those used by fsl-asoc-card. * S/PDIF in/out dummy codecs must now be declared explicitly, add and use them. These modifications were tested only on an imx8mn-evk board. Signed-off-by: Elinor Montmasson --- arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts | 15 ++++++++++++--- arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts | 15 ++++++++++++--- arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi | 15 ++++++++++++--- arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6qdl-colibri.dtsi | 15 ++++++++++++--- arch/arm/boot/dts/nxp/imx/imx6qdl-cubox-i.dtsi | 9 +++++++-- .../boot/dts/nxp/imx/imx6qdl-hummingboard.dtsi | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6qdl-sabreauto.dtsi | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6qdl-wandboard.dtsi | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6sx-sabreauto.dts | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi | 9 +++++++-- 12 files changed, 104 insertions(+), 28 deletions(-) diff --git a/arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts b/arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts index 95b49fc83f7b..5c664c0f2169 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts +++ b/arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts @@ -127,12 +127,21 @@ simple-audio-card,codec { }; }; + spdif_out: spdif-out { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dit"; + }; + + spdif_in: spdif-in { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dir"; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-out; - spdif-in; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>, <&spdif_in>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts b/arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts index a7d5693c5ab7..8491d656ef17 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts +++ b/arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts @@ -111,12 +111,21 @@ simple-audio-card,codec { }; }; + spdif_out: spdif-out { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dit"; + }; + + spdif_in: spdif-in { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dir"; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-in; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>, <&spdif_in>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts b/arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts index 7c298d9aa21e..ea9a98887c7b 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts +++ b/arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts @@ -90,11 +90,16 @@ sound-sgtl5000 { ssi-controller = <&ssi1>; }; + spdif_out: spdif-out { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dit"; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "On-board SPDIF"; - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi index ea40623d12e5..6f4546c59d38 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi @@ -197,11 +197,20 @@ sound { ssi-controller = <&ssi1>; }; + spdif_out: spdif-out { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dit"; + }; + + spdif_in: spdif-in { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dir"; + }; + sound_spdif: sound-spdif { compatible = "fsl,imx-audio-spdif"; - spdif-controller = <&spdif>; - spdif-in; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>, <&spdif_in>; model = "imx-spdif"; status = "disabled"; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi index 3a46ade3b6bd..6aa6b152c3ae 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi @@ -121,11 +121,16 @@ sound { mux-ext-port = <3>; }; + spdif_out: spdif-out { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dit"; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-colibri.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-colibri.dtsi index d3a7a6eeb8e0..07f15726f203 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-colibri.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-colibri.dtsi @@ -142,12 +142,21 @@ sound { ssi-controller = <&ssi1>; }; + spdif_out: spdif-out { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dit"; + }; + + spdif_in: spdif-in { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dir"; + }; + /* Optional S/PDIF in on SODIMM 88 and out on SODIMM 90, 137 or 168 */ sound_spdif: sound-spdif { compatible = "fsl,imx-audio-spdif"; - spdif-controller = <&spdif>; - spdif-in; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>, <&spdif_in>; model = "imx-spdif"; status = "disabled"; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-cubox-i.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-cubox-i.dtsi index 761566ae3cf5..28afa8a0188b 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-cubox-i.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-cubox-i.dtsi @@ -100,12 +100,17 @@ v_usb1: regulator-v-usb1 { vin-supply = <&v_5v0>; }; + spdif_out: spdif-out { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dit"; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "Integrated SPDIF"; /* IMX6 doesn't implement this yet */ - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; gpio-keys { diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-hummingboard.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-hummingboard.dtsi index a955c77cd499..67f2a007a592 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-hummingboard.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-hummingboard.dtsi @@ -140,12 +140,17 @@ sound_codec: simple-audio-card,codec { }; }; + spdif_out: spdif-out { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dit"; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "On-board SPDIF"; /* IMX6 doesn't implement this yet */ - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-sabreauto.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-sabreauto.dtsi index 6656e2e762a1..48dfd8151150 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-sabreauto.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-sabreauto.dtsi @@ -143,12 +143,17 @@ sound-cs42888 { "AIN2R", "Line In Jack"; }; + spdif_in: spdif-in { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dir"; + }; + sound-spdif { compatible = "fsl,imx-sabreauto-spdif", "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-in; + audio-cpu = <&spdif>; + audio-codec = <&spdif_in>; }; backlight { diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-wandboard.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-wandboard.dtsi index 38abb6b50f6c..5a4b9ced297a 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-wandboard.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-wandboard.dtsi @@ -26,11 +26,16 @@ sound { mux-ext-port = <3>; }; + spdif_out: spdif-out { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dit"; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; reg_1p5v: regulator-1p5v { diff --git a/arch/arm/boot/dts/nxp/imx/imx6sx-sabreauto.dts b/arch/arm/boot/dts/nxp/imx/imx6sx-sabreauto.dts index b0c27b9b0244..d2cc8b4e8b00 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6sx-sabreauto.dts +++ b/arch/arm/boot/dts/nxp/imx/imx6sx-sabreauto.dts @@ -97,11 +97,16 @@ sound-cs42888 { "AIN2R", "Line In Jack"; }; + spdif_in: spdif-in { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dir"; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-in; + audio-cpu = <&spdif>; + audio-codec = <&spdif_in>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi b/arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi index 7d4170c27732..a8c1fc02eddb 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi @@ -183,12 +183,17 @@ panel_in: endpoint { }; }; + spdif_out: spdif-out { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dit"; + }; + sound-spdif { compatible = "fsl,imx6sx-sdb-spdif", "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; };