From patchwork Thu Jul 30 09:47:02 2020 Content-Type: text/plain; charset="utf-8" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit X-Patchwork-Submitter: Shengjiu Wang X-Patchwork-Id: 1338730 Return-Path: X-Original-To: patchwork-incoming@ozlabs.org Delivered-To: patchwork-incoming@ozlabs.org Received: from lists.ozlabs.org (lists.ozlabs.org [IPv6:2401:3900:2:1::3]) (using TLSv1.3 with cipher TLS_AES_256_GCM_SHA384 (256/256 bits) key-exchange X25519 server-signature RSA-PSS (4096 bits)) (No client certificate requested) by ozlabs.org (Postfix) with ESMTPS id 4BHQh55RtMz9sRN for ; Thu, 30 Jul 2020 19:52:53 +1000 (AEST) Authentication-Results: ozlabs.org; dmarc=fail (p=none dis=none) header.from=nxp.com Received: from bilbo.ozlabs.org (lists.ozlabs.org [IPv6:2401:3900:2:1::3]) by lists.ozlabs.org (Postfix) with ESMTP id 4BHQh54Q75zDqwb for ; Thu, 30 Jul 2020 19:52:53 +1000 (AEST) X-Original-To: linuxppc-dev@lists.ozlabs.org Delivered-To: linuxppc-dev@lists.ozlabs.org Authentication-Results: lists.ozlabs.org; spf=pass (sender SPF authorized) smtp.mailfrom=nxp.com (client-ip=92.121.34.13; helo=inva020.nxp.com; envelope-from=shengjiu.wang@nxp.com; receiver=) Authentication-Results: lists.ozlabs.org; dmarc=pass (p=none dis=none) header.from=nxp.com Received: from inva020.nxp.com (inva020.nxp.com [92.121.34.13]) (using TLSv1.2 with cipher ECDHE-RSA-AES256-GCM-SHA384 (256/256 bits)) (No client certificate requested) by lists.ozlabs.org (Postfix) with ESMTPS id 4BHQfY2JgszDqsH for ; Thu, 30 Jul 2020 19:51:31 +1000 (AEST) Received: from inva020.nxp.com (localhost [127.0.0.1]) by inva020.eu-rdc02.nxp.com (Postfix) with ESMTP id F3FD01A03AA; Thu, 30 Jul 2020 11:51:26 +0200 (CEST) Received: from invc005.ap-rdc01.nxp.com (invc005.ap-rdc01.nxp.com [165.114.16.14]) by inva020.eu-rdc02.nxp.com (Postfix) with ESMTP id 5BEF61A04C3; Thu, 30 Jul 2020 11:51:22 +0200 (CEST) Received: from localhost.localdomain (shlinux2.ap.freescale.net [10.192.224.44]) by invc005.ap-rdc01.nxp.com (Postfix) with ESMTP id 8FD26402AA; Thu, 30 Jul 2020 11:51:16 +0200 (CEST) From: Shengjiu Wang To: timur@kernel.org, nicoleotsuka@gmail.com, Xiubo.Lee@gmail.com, festevam@gmail.com, lgirdwood@gmail.com, broonie@kernel.org, perex@perex.cz, tiwai@suse.com, alsa-devel@alsa-project.org, linuxppc-dev@lists.ozlabs.org, linux-kernel@vger.kernel.org Subject: [PATCH v2] ASoC: fsl-asoc-card: Remove fsl_asoc_card_set_bias_level function Date: Thu, 30 Jul 2020 17:47:02 +0800 Message-Id: <1596102422-14010-1-git-send-email-shengjiu.wang@nxp.com> X-Mailer: git-send-email 2.7.4 X-Virus-Scanned: ClamAV using ClamSMTP X-BeenThere: linuxppc-dev@lists.ozlabs.org X-Mailman-Version: 2.1.29 Precedence: list List-Id: Linux on PowerPC Developers Mail List List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Errors-To: linuxppc-dev-bounces+patchwork-incoming=ozlabs.org@lists.ozlabs.org Sender: "Linuxppc-dev" With this case: aplay -Dhw:x 16khz.wav 24khz.wav There is sound distortion for 24khz.wav. The reason is that setting PLL of WM8962 with set_bias_level function, the bias level is not changed when 24khz.wav is played, then the PLL won't be reset, the clock is not correct, so distortion happens. The resolution of this issue is to remove fsl_asoc_card_set_bias_level. Move PLL configuration to hw_params and hw_free. After removing fsl_asoc_card_set_bias_level, also test WM8960 case, it can work. Fixes: 708b4351f08c ("ASoC: fsl: Add Freescale Generic ASoC Sound Card with ASRC support") Signed-off-by: Shengjiu Wang --- changes in v2 - replace is_stream_in_use with streams - add "out" error handler in hw_params() sound/soc/fsl/fsl-asoc-card.c | 155 ++++++++++++++++------------------ 1 file changed, 71 insertions(+), 84 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 4848ba61d083..b6a2a527dc04 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -73,6 +73,7 @@ struct cpu_priv { * @codec_priv: CODEC private data * @cpu_priv: CPU private data * @card: ASoC card structure + * @streams: Mask of current active streams * @sample_rate: Current sample rate * @sample_format: Current sample format * @asrc_rate: ASRC sample rate used by Back-Ends @@ -89,6 +90,7 @@ struct fsl_asoc_card_priv { struct codec_priv codec_priv; struct cpu_priv cpu_priv; struct snd_soc_card card; + u8 streams; u32 sample_rate; snd_pcm_format_t sample_format; u32 asrc_rate; @@ -151,21 +153,17 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; 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 cpu_priv *cpu_priv = &priv->cpu_priv; struct device *dev = rtd->card->dev; + unsigned int pll_out; int ret; priv->sample_rate = params_rate(params); priv->sample_format = params_format(params); + priv->streams |= BIT(substream->stream); - /* - * If codec-dai is DAI Master and all configurations are already in the - * set_bias_level(), bypass the remaining settings in hw_params(). - * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS. - */ - if ((priv->card.set_bias_level && - priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) || - fsl_asoc_card_is_ac97(priv)) + if (fsl_asoc_card_is_ac97(priv)) return 0; /* Specific configurations of DAIs starts from here */ @@ -174,7 +172,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, cpu_priv->sysclk_dir[tx]); if (ret && ret != -ENOTSUPP) { dev_err(dev, "failed to set sysclk for cpu dai\n"); - return ret; + goto out; } if (cpu_priv->slot_width) { @@ -182,6 +180,69 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, cpu_priv->slot_width); if (ret && ret != -ENOTSUPP) { dev_err(dev, "failed to set TDM slot for cpu dai\n"); + goto out; + } + } + + /* Specific configuration for PLL */ + if (codec_priv->pll_id && codec_priv->fll_id) { + 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_pll(asoc_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 out; + } + + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), + 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 out; + } + } + + return 0; + +out: + priv->streams &= ~BIT(substream->stream); + return ret; +} + +static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct codec_priv *codec_priv = &priv->codec_priv; + struct device *dev = rtd->card->dev; + int ret; + + priv->streams &= ~BIT(substream->stream); + + if (!priv->streams && codec_priv->pll_id && + codec_priv->fll_id) { + /* Force freq to be 0 to avoid error message in codec */ + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), + codec_priv->mclk_id, + 0, + 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(asoc_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; } } @@ -191,6 +252,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, static const struct snd_soc_ops fsl_asoc_card_ops = { .hw_params = fsl_asoc_card_hw_params, + .hw_free = fsl_asoc_card_hw_free, }; static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, @@ -254,75 +316,6 @@ static struct snd_soc_dai_link fsl_asoc_card_dai[] = { }, }; -static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); - struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai *codec_dai; - struct codec_priv *codec_priv = &priv->codec_priv; - struct device *dev = card->dev; - unsigned int pll_out; - int ret; - - rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - codec_dai = asoc_rtd_to_codec(rtd, 0); - if (dapm->dev != codec_dai->dev) - return 0; - - switch (level) { - case SND_SOC_BIAS_PREPARE: - if (dapm->bias_level != SND_SOC_BIAS_STANDBY) - break; - - 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_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); - return ret; - } - - 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); - return ret; - } - break; - - case SND_SOC_BIAS_STANDBY: - if (dapm->bias_level != SND_SOC_BIAS_PREPARE) - break; - - 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 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) { - dev_err(dev, "failed to stop FLL: %d\n", ret); - return ret; - } - break; - - default: - break; - } - - return 0; -} - static int fsl_asoc_card_audmux_init(struct device_node *np, struct fsl_asoc_card_priv *priv) { @@ -611,7 +604,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Diversify the card configurations */ if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { codec_dai_name = "cs42888"; - priv->card.set_bias_level = NULL; priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq; priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT; @@ -628,26 +620,22 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) { codec_dai_name = "wm8962"; - priv->card.set_bias_level = fsl_asoc_card_set_bias_level; priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK; priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; priv->codec_priv.pll_id = WM8962_FLL; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) { codec_dai_name = "wm8960-hifi"; - priv->card.set_bias_level = fsl_asoc_card_set_bias_level; priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { codec_dai_name = "ac97-hifi"; - priv->card.set_bias_level = NULL; 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"; - priv->card.set_bias_level = NULL; priv->dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_NB_NF; @@ -657,7 +645,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) 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"; - priv->card.set_bias_level = NULL; priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; priv->dai_link[1].dpcm_capture = 0; priv->dai_link[2].dpcm_capture = 0;