From mboxrd@z Thu Jan 1 00:00:00 1970 From: Mark Brown Subject: Applied "ASoC: cs42xx8: replace codec to component" to the asoc tree Date: Mon, 12 Feb 2018 12:44:02 +0000 Message-ID: References: <87d11t1fpo.wl%kuninori.morimoto.gx@renesas.com> Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit Return-path: Received: from heliosphere.sirena.org.uk (heliosphere.sirena.org.uk [172.104.155.198]) by alsa0.perex.cz (Postfix) with ESMTP id 2FDEB267A10 for ; Mon, 12 Feb 2018 13:44:05 +0100 (CET) In-Reply-To: <87d11t1fpo.wl%kuninori.morimoto.gx@renesas.com> List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Errors-To: alsa-devel-bounces@alsa-project.org Sender: alsa-devel-bounces@alsa-project.org To: Kuninori Morimoto Cc: alsa-devel@alsa-project.org, Mark Brown List-Id: alsa-devel@alsa-project.org The patch ASoC: cs42xx8: replace codec to component has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >>From 99a9f452093e40f82e51a2f49d95ee3c04ad298b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 29 Jan 2018 03:56:58 +0000 Subject: [PATCH] ASoC: cs42xx8: replace codec to component Now we can replace Codec to Component. Let's do it. Note: xxx_codec_xxx() -> xxx_component_xxx() .idle_bias_off = 1 -> .idle_bias_on = 0 .ignore_pmdown_time = 0 -> .use_pmdown_time = 1 - -> .endianness = 1 - -> .non_legacy_dai_naming = 1 Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/cs42xx8-i2c.c | 1 - sound/soc/codecs/cs42xx8.c | 57 +++++++++++++++++++++--------------------- 2 files changed, 28 insertions(+), 30 deletions(-) diff --git a/sound/soc/codecs/cs42xx8-i2c.c b/sound/soc/codecs/cs42xx8-i2c.c index 800c1d549347..0214e3ab9da0 100644 --- a/sound/soc/codecs/cs42xx8-i2c.c +++ b/sound/soc/codecs/cs42xx8-i2c.c @@ -33,7 +33,6 @@ static int cs42xx8_i2c_probe(struct i2c_client *i2c, static int cs42xx8_i2c_remove(struct i2c_client *i2c) { - snd_soc_unregister_codec(&i2c->dev); pm_runtime_disable(&i2c->dev); return 0; diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index c1785bd4ff19..ebb9e0cf8364 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -194,8 +194,8 @@ static const struct cs42xx8_ratios cs42xx8_ratios[] = { static int cs42xx8_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { - struct snd_soc_codec *codec = codec_dai->codec; - struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = codec_dai->component; + struct cs42xx8_priv *cs42xx8 = snd_soc_component_get_drvdata(component); cs42xx8->sysclk = freq; @@ -205,8 +205,8 @@ static int cs42xx8_set_dai_sysclk(struct snd_soc_dai *codec_dai, static int cs42xx8_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int format) { - struct snd_soc_codec *codec = codec_dai->codec; - struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = codec_dai->component; + struct cs42xx8_priv *cs42xx8 = snd_soc_component_get_drvdata(component); u32 val; /* Set DAI format */ @@ -224,7 +224,7 @@ static int cs42xx8_set_dai_fmt(struct snd_soc_dai *codec_dai, val = CS42XX8_INTF_DAC_DIF_TDM | CS42XX8_INTF_ADC_DIF_TDM; break; default: - dev_err(codec->dev, "unsupported dai format\n"); + dev_err(component->dev, "unsupported dai format\n"); return -EINVAL; } @@ -241,7 +241,7 @@ static int cs42xx8_set_dai_fmt(struct snd_soc_dai *codec_dai, cs42xx8->slave_mode = false; break; default: - dev_err(codec->dev, "unsupported master/slave mode\n"); + dev_err(component->dev, "unsupported master/slave mode\n"); return -EINVAL; } @@ -252,8 +252,8 @@ static int cs42xx8_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_codec *codec = dai->codec; - struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = dai->component; + struct cs42xx8_priv *cs42xx8 = snd_soc_component_get_drvdata(component); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u32 ratio = cs42xx8->sysclk / params_rate(params); u32 i, fm, val, mask; @@ -267,7 +267,7 @@ static int cs42xx8_hw_params(struct snd_pcm_substream *substream, } if (i == ARRAY_SIZE(cs42xx8_ratios)) { - dev_err(codec->dev, "unsupported sysclk ratio\n"); + dev_err(component->dev, "unsupported sysclk ratio\n"); return -EINVAL; } @@ -285,8 +285,8 @@ static int cs42xx8_hw_params(struct snd_pcm_substream *substream, static int cs42xx8_digital_mute(struct snd_soc_dai *dai, int mute) { - struct snd_soc_codec *codec = dai->codec; - struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = dai->component; + struct cs42xx8_priv *cs42xx8 = snd_soc_component_get_drvdata(component); u8 dac_unmute = cs42xx8->tx_channels ? ~((0x1 << cs42xx8->tx_channels) - 1) : 0; @@ -382,14 +382,14 @@ const struct regmap_config cs42xx8_regmap_config = { }; EXPORT_SYMBOL_GPL(cs42xx8_regmap_config); -static int cs42xx8_codec_probe(struct snd_soc_codec *codec) +static int cs42xx8_component_probe(struct snd_soc_component *component) { - struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs42xx8_priv *cs42xx8 = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); switch (cs42xx8->drvdata->num_adcs) { case 3: - snd_soc_add_codec_controls(codec, cs42xx8_adc3_snd_controls, + snd_soc_add_component_controls(component, cs42xx8_adc3_snd_controls, ARRAY_SIZE(cs42xx8_adc3_snd_controls)); snd_soc_dapm_new_controls(dapm, cs42xx8_adc3_dapm_widgets, ARRAY_SIZE(cs42xx8_adc3_dapm_widgets)); @@ -406,18 +406,17 @@ static int cs42xx8_codec_probe(struct snd_soc_codec *codec) return 0; } -static const struct snd_soc_codec_driver cs42xx8_driver = { - .probe = cs42xx8_codec_probe, - .idle_bias_off = true, - - .component_driver = { - .controls = cs42xx8_snd_controls, - .num_controls = ARRAY_SIZE(cs42xx8_snd_controls), - .dapm_widgets = cs42xx8_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(cs42xx8_dapm_widgets), - .dapm_routes = cs42xx8_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(cs42xx8_dapm_routes), - }, +static const struct snd_soc_component_driver cs42xx8_driver = { + .probe = cs42xx8_component_probe, + .controls = cs42xx8_snd_controls, + .num_controls = ARRAY_SIZE(cs42xx8_snd_controls), + .dapm_widgets = cs42xx8_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42xx8_dapm_widgets), + .dapm_routes = cs42xx8_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(cs42xx8_dapm_routes), + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, }; const struct cs42xx8_driver_data cs42448_data = { @@ -520,9 +519,9 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap) /* Each adc supports stereo input */ cs42xx8_dai.capture.channels_max = cs42xx8->drvdata->num_adcs * 2; - ret = snd_soc_register_codec(dev, &cs42xx8_driver, &cs42xx8_dai, 1); + ret = devm_snd_soc_register_component(dev, &cs42xx8_driver, &cs42xx8_dai, 1); if (ret) { - dev_err(dev, "failed to register codec:%d\n", ret); + dev_err(dev, "failed to register component:%d\n", ret); goto err_enable; } -- 2.16.1