From patchwork Wed Oct 9 07:52:26 2024 Content-Type: text/plain; charset="utf-8" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit X-Patchwork-Submitter: Binbin Zhou X-Patchwork-Id: 13827669 Received: from mail.loongson.cn (mail.loongson.cn [114.242.206.163]) by smtp.subspace.kernel.org (Postfix) with ESMTP id 177D51865EE; Wed, 9 Oct 2024 07:52:35 +0000 (UTC) Authentication-Results: smtp.subspace.kernel.org; arc=none smtp.client-ip=114.242.206.163 ARC-Seal: i=1; a=rsa-sha256; d=subspace.kernel.org; s=arc-20240116; t=1728460358; cv=none; b=qsa2iBHCIBMBj9f7+deAmaIHtSxEmbltKAXQf+G9heBGNb4V9RXlPhHoL6qX71idF0RKysIspR5AYg42aFDqZJoIADCIfpyzo+AFMfB3NS2dPS9/T3fqAalmS1RFg1u88jRpFc6XU/+YZWmVSFTVLlp80pFr91YMbId3GkQdIaM= ARC-Message-Signature: i=1; a=rsa-sha256; d=subspace.kernel.org; s=arc-20240116; t=1728460358; c=relaxed/simple; bh=DoIcpuckAzmvscnPXH0/Iltye/RPP0oXDRodBPfpBlA=; h=From:To:Cc:Subject:Date:Message-ID:In-Reply-To:References: MIME-Version; b=FXbcXv4K1cEBXGQmBsdliILhqT0VW0Au9pxN3SF8Z5K7vzs/j0OU23FgkasVkxYhNrw06sI3VFqUezoBE3mi4leRJkb5cqlk/37ARKxJ9SFLH/iOwRVfsTx7QRixEtgjeJWh1vul7KiL22zuK/LZH7iozxIFKMcVZraq0xePbV0= ARC-Authentication-Results: i=1; smtp.subspace.kernel.org; dmarc=none (p=none dis=none) header.from=loongson.cn; spf=pass smtp.mailfrom=loongson.cn; arc=none smtp.client-ip=114.242.206.163 Authentication-Results: smtp.subspace.kernel.org; dmarc=none (p=none dis=none) header.from=loongson.cn Authentication-Results: smtp.subspace.kernel.org; spf=pass smtp.mailfrom=loongson.cn Received: from loongson.cn (unknown [223.64.68.38]) by gateway (Coremail) with SMTP id _____8AxjmtCNgZnVbIQAA--.23500S3; Wed, 09 Oct 2024 15:52:34 +0800 (CST) Received: from localhost.localdomain (unknown [223.64.68.38]) by front1 (Coremail) with SMTP id qMiowMBxn+Q9NgZniRYhAA--.37100S2; Wed, 09 Oct 2024 15:52:30 +0800 (CST) From: Binbin Zhou To: Binbin Zhou , Huacai Chen , Liam Girdwood , Mark Brown , Jaroslav Kysela , Takashi Iwai , Rob Herring , Krzysztof Kozlowski , Conor Dooley Cc: Huacai Chen , linux-sound@vger.kernel.org, devicetree@vger.kernel.org, Xuerui Wang , loongarch@lists.linux.dev, Neil Armstrong , Pierre-Louis Bossart , Richard Fitzgerald , Luca Ceresoli , Weidong Wang , Prasad Kumpatla , Herve Codina , Masahiro Yamada , Shuming Fan , Binbin Zhou Subject: [PATCH v3 4/9] ASoC: codecs: Add uda1342 codec driver Date: Wed, 9 Oct 2024 15:52:26 +0800 Message-ID: <927e46b48ca84865a216ce08e7c53df59c2a8c0b.1728459624.git.zhoubinbin@loongson.cn> X-Mailer: git-send-email 2.43.5 In-Reply-To: References: Precedence: bulk X-Mailing-List: linux-sound@vger.kernel.org List-Id: List-Subscribe: List-Unsubscribe: MIME-Version: 1.0 X-CM-TRANSID: qMiowMBxn+Q9NgZniRYhAA--.37100S2 X-CM-SenderInfo: p2kr3uplqex0o6or00hjvr0hdfq/ X-Coremail-Antispam: 1Uk129KBj9fXoW3uFyrZr4fuFW7KF18Cw4fXrc_yoW8XryfXo W3tFnYvw1rXryxuFW5X3WkWrWUZF15CayxJw1DZ3ykJ34rGa1DWrWDGr1Uua43tFZYgFWj yFySvwn3ArW2vryDl-sFpf9Il3svdjkaLaAFLSUrUUUUob8apTn2vfkv8UJUUUU8wcxFpf 9Il3svdxBIdaVrn0xqx4xG64xvF2IEw4CE5I8CrVC2j2Jv73VFW2AGmfu7bjvjm3AaLaJ3 UjIYCTnIWjp_UUUYE7kC6x804xWl14x267AKxVWUJVW8JwAFc2x0x2IEx4CE42xK8VAvwI 8IcIk0rVWrJVCq3wAFIxvE14AKwVWUAVWUZwA2ocxC64kIII0Yj41l84x0c7CEw4AK67xG Y2AK021l84ACjcxK6xIIjxv20xvE14v26F1j6w1UM28EF7xvwVC0I7IYx2IY6xkF7I0E14 v26r4UJVWxJr1l84ACjcxK6I8E87Iv67AKxVW8Jr0_Cr1UM28EF7xvwVC2z280aVCY1x02 67AKxVW8Jr0_Cr1UM2kKe7AKxVWUtVW8ZwAS0I0E0xvYzxvE52x082IY62kv0487Mc804V CY07AIYIkI8VC2zVCFFI0UMc02F40EFcxC0VAKzVAqx4xG6I80ewAv7VC0I7IYx2IY67AK xVW3AVW8Xw1lYx0Ex4A2jsIE14v26F4j6r4UJwAm72CE4IkC6x0Yz7v_Jr0_Gr1lF7xvr2 IYc2Ij64vIr41lc7CjxVAaw2AFwI0_GFv_Wryl42xK82IYc2Ij64vIr41l4I8I3I0E4IkC 6x0Yz7v_Jr0_Gr1l4IxYO2xFxVAFwI0_Jw0_GFylx2IqxVAqx4xG67AKxVWUJVWUGwC20s 026x8GjcxK67AKxVWUGVWUWwC2zVAF1VAY17CE14v26r4a6rW5MIIYrxkI7VAKI48JMIIF 0xvE2Ix0cI8IcVAFwI0_Ar0_tr1lIxAIcVC0I7IYx2IY6xkF7I0E14v26r4UJVWxJr1lIx AIcVCF04k26cxKx2IYs7xG6r1j6r1xMIIF0xvEx4A2jsIE14v26F4j6r4UJwCI42IY6I8E 87Iv6xkF7I0E14v26r4UJVWxJrUvcSsGvfC2KfnxnUUI43ZEXa7IUe6OJUUUUUU== The UDA1342 is an NXP audio codec, support 2x Stereo audio ADC (4x PGA mic inputs), stereo audio DAC, with basic audio processing. Signed-off-by: Binbin Zhou --- sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/uda1342.c | 347 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/uda1342.h | 78 +++++++++ 4 files changed, 435 insertions(+) create mode 100644 sound/soc/codecs/uda1342.c create mode 100644 sound/soc/codecs/uda1342.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6480f1bd43f4..6a6125e94d2d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -283,6 +283,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TWL4030 imply SND_SOC_TWL6040 imply SND_SOC_UDA1334 + imply SND_SOC_UDA1342 imply SND_SOC_UDA1380 imply SND_SOC_WCD9335 imply SND_SOC_WCD934X @@ -2131,6 +2132,13 @@ config SND_SOC_UDA1334 and has basic features such as de-emphasis (at 44.1 kHz sampling rate) and mute. +config SND_SOC_UDA1342 + tristate "NXP UDA1342 CODEC" + depends on I2C + help + The UDA1342 is an NXP audio codec, support 2x Stereo audio ADC (4x PGA + mic inputs), stereo audio DAC, with basic audio processing. + config SND_SOC_UDA1380 tristate depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 029fa42ce5c0..ac7d8b71b32b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -325,6 +325,7 @@ snd-soc-ts3a227e-y := ts3a227e.o snd-soc-twl4030-y := twl4030.o snd-soc-twl6040-y := twl6040.o snd-soc-uda1334-y := uda1334.o +snd-soc-uda1342-y := uda1342.o snd-soc-uda1380-y := uda1380.o snd-soc-wcd-classh-y := wcd-clsh-v2.o snd-soc-wcd-mbhc-y := wcd-mbhc-v2.o @@ -735,6 +736,7 @@ obj-$(CONFIG_SND_SOC_TS3A227E) += snd-soc-ts3a227e.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o obj-$(CONFIG_SND_SOC_UDA1334) += snd-soc-uda1334.o +obj-$(CONFIG_SND_SOC_UDA1342) += snd-soc-uda1342.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WCD_CLASSH) += snd-soc-wcd-classh.o obj-$(CONFIG_SND_SOC_WCD_MBHC) += snd-soc-wcd-mbhc.o diff --git a/sound/soc/codecs/uda1342.c b/sound/soc/codecs/uda1342.c new file mode 100644 index 000000000000..3d49a7869948 --- /dev/null +++ b/sound/soc/codecs/uda1342.c @@ -0,0 +1,347 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// uda1342.c -- UDA1342 ALSA SoC Codec driver +// Based on the WM87xx drivers by Liam Girdwood and Richard Purdie +// +// Copyright 2007 Dension Audio Systems Ltd. +// Copyright 2024 Loongson Technology Co.,Ltd. +// +// Modifications by Christian Pellegrin +// Further cleanup and restructuring by: +// Binbin Zhou + +#include +#include +#include +#include +#include +#include +#include +#include + +#include "uda1342.h" + +#define UDA134X_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE) + +struct uda1342_priv { + int sysclk; + int dai_fmt; + + struct snd_pcm_substream *provider_substream; + struct snd_pcm_substream *consumer_substream; + + struct regmap *regmap; + struct i2c_client *i2c; +}; + +static const struct reg_default uda1342_reg_defaults[] = { + { 0x00, 0x1042 }, + { 0x01, 0x0000 }, + { 0x10, 0x0088 }, + { 0x11, 0x0000 }, + { 0x12, 0x0000 }, + { 0x20, 0x0080 }, + { 0x21, 0x0080 }, +}; + +static int uda1342_mute(struct snd_soc_dai *dai, int mute, int direction) +{ + struct snd_soc_component *component = dai->component; + struct uda1342_priv *uda1342 = snd_soc_component_get_drvdata(component); + unsigned int mask; + unsigned int val = 0; + + /* Master mute */ + mask = BIT(5); + if (mute) + val = mask; + + return regmap_update_bits(uda1342->regmap, 0x10, mask, val); +} + +static int uda1342_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct uda1342_priv *uda1342 = snd_soc_component_get_drvdata(component); + struct snd_pcm_runtime *provider_runtime; + + if (uda1342->provider_substream) { + provider_runtime = uda1342->provider_substream->runtime; + + snd_pcm_hw_constraint_single(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, provider_runtime->rate); + snd_pcm_hw_constraint_single(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + provider_runtime->sample_bits); + + uda1342->consumer_substream = substream; + } else { + uda1342->provider_substream = substream; + } + + return 0; +} + +static void uda1342_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct uda1342_priv *uda1342 = snd_soc_component_get_drvdata(component); + + if (uda1342->provider_substream == substream) + uda1342->provider_substream = uda1342->consumer_substream; + + uda1342->consumer_substream = NULL; +} + +static int uda1342_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct uda1342_priv *uda1342 = snd_soc_component_get_drvdata(component); + struct device *dev = &uda1342->i2c->dev; + unsigned int hw_params = 0; + + if (substream == uda1342->consumer_substream) + return 0; + + /* set SYSCLK / fs ratio */ + switch (uda1342->sysclk / params_rate(params)) { + case 512: + break; + case 384: + hw_params |= BIT(4); + break; + case 256: + hw_params |= BIT(5); + break; + default: + dev_err(dev, "unsupported frequency\n"); + return -EINVAL; + } + + /* set DAI format and word length */ + switch (uda1342->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + switch (params_width(params)) { + case 16: + hw_params |= BIT(1); + break; + case 18: + hw_params |= BIT(2); + break; + case 20: + hw_params |= BIT(2) | BIT(1); + break; + default: + dev_err(dev, "unsupported format (right)\n"); + return -EINVAL; + } + break; + case SND_SOC_DAIFMT_LEFT_J: + hw_params |= BIT(3); + break; + default: + dev_err(dev, "unsupported format\n"); + return -EINVAL; + } + + return regmap_update_bits(uda1342->regmap, 0x0, + STATUS0_DAIFMT_MASK | STATUS0_SYSCLK_MASK, hw_params); +} + +static int uda1342_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *component = codec_dai->component; + struct uda1342_priv *uda1342 = snd_soc_component_get_drvdata(component); + struct device *dev = &uda1342->i2c->dev; + + /* + * Anything between 256fs*8Khz and 512fs*48Khz should be acceptable + * because the codec is slave. Of course limitations of the clock + * master (the IIS controller) apply. + * We'll error out on set_hw_params if it's not OK + */ + if ((freq >= (256 * 8000)) && (freq <= (512 * 48000))) { + uda1342->sysclk = freq; + return 0; + } + + dev_err(dev, "unsupported sysclk\n"); + + return -EINVAL; +} + +static int uda1342_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + struct uda1342_priv *uda1342 = snd_soc_component_get_drvdata(component); + + /* codec supports only full consumer mode */ + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_BC_FC) { + dev_err(&uda1342->i2c->dev, "unsupported consumer mode.\n"); + return -EINVAL; + } + + /* We can't setup DAI format here as it depends on the word bit num */ + /* so let's just store the value for later */ + uda1342->dai_fmt = fmt; + + return 0; +} + +static const struct snd_kcontrol_new uda1342_snd_controls[] = { + SOC_SINGLE("Master Playback Volume", 0x11, 0, 0x3F, 1), + SOC_SINGLE("Analog1 Volume", 0x12, 0, 0x1F, 1), +}; + +/* Common DAPM widgets */ +static const struct snd_soc_dapm_widget uda1342_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("VINL1"), + SND_SOC_DAPM_INPUT("VINR1"), + SND_SOC_DAPM_INPUT("VINL2"), + SND_SOC_DAPM_INPUT("VINR2"), + + SND_SOC_DAPM_DAC("DAC", "Playback", 0, 1, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", 0, 9, 0), + + SND_SOC_DAPM_OUTPUT("VOUTL"), + SND_SOC_DAPM_OUTPUT("VOUTR"), +}; + +static const struct snd_soc_dapm_route uda1342_dapm_routes[] = { + { "ADC", NULL, "VINL1" }, + { "ADC", NULL, "VINR1" }, + { "ADC", NULL, "VINL2" }, + { "ADC", NULL, "VINR2" }, + { "VOUTL", NULL, "DAC" }, + { "VOUTR", NULL, "DAC" }, +}; + +static const struct snd_soc_dai_ops uda1342_dai_ops = { + .startup = uda1342_startup, + .shutdown = uda1342_shutdown, + .hw_params = uda1342_hw_params, + .mute_stream = uda1342_mute, + .set_sysclk = uda1342_set_dai_sysclk, + .set_fmt = uda1342_set_dai_fmt, +}; + +static struct snd_soc_dai_driver uda1342_dai = { + .name = "uda1342-hifi", + /* playback capabilities */ + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = UDA134X_FORMATS, + }, + /* capture capabilities */ + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = UDA134X_FORMATS, + }, + /* pcm operations */ + .ops = &uda1342_dai_ops, +}; + +static const struct snd_soc_component_driver soc_component_dev_uda1342 = { + .controls = uda1342_snd_controls, + .num_controls = ARRAY_SIZE(uda1342_snd_controls), + .dapm_widgets = uda1342_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(uda1342_dapm_widgets), + .dapm_routes = uda1342_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(uda1342_dapm_routes), + .suspend_bias_off = 1, + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, +}; + +static const struct regmap_config uda1342_regmap = { + .reg_bits = 8, + .val_bits = 16, + .max_register = 0x21, + .reg_defaults = uda1342_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(uda1342_reg_defaults), + .cache_type = REGCACHE_MAPLE, +}; + +static int uda1342_i2c_probe(struct i2c_client *i2c) +{ + struct uda1342_priv *uda1342; + + uda1342 = devm_kzalloc(&i2c->dev, sizeof(*uda1342), GFP_KERNEL); + if (!uda1342) + return -ENOMEM; + + uda1342->regmap = devm_regmap_init_i2c(i2c, &uda1342_regmap); + if (IS_ERR(uda1342->regmap)) + return PTR_ERR(uda1342->regmap); + + i2c_set_clientdata(i2c, uda1342); + uda1342->i2c = i2c; + + return devm_snd_soc_register_component(&i2c->dev, + &soc_component_dev_uda1342, + &uda1342_dai, 1); +} + +static int uda1342_suspend(struct device *dev) +{ + struct uda1342_priv *uda1342 = dev_get_drvdata(dev); + + regcache_cache_only(uda1342->regmap, true); + + return 0; +} + +static int uda1342_resume(struct device *dev) +{ + struct uda1342_priv *uda1342 = dev_get_drvdata(dev); + + regcache_mark_dirty(uda1342->regmap); + regcache_sync(uda1342->regmap); + + return 0; +} + +static DEFINE_RUNTIME_DEV_PM_OPS(uda1342_pm_ops, + uda1342_suspend, uda1342_resume, NULL); + +static const struct i2c_device_id uda1342_i2c_id[] = { + { "uda1342", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, uda1342_i2c_id); + +static const struct of_device_id uda1342_of_match[] = { + { .compatible = "nxp,uda1342" }, + { } +}; +MODULE_DEVICE_TABLE(of, uda1342_of_match); + +static struct i2c_driver uda1342_i2c_driver = { + .driver = { + .name = "uda1342", + .of_match_table = uda1342_of_match, + .pm = pm_sleep_ptr(&uda1342_pm_ops), + }, + .probe = uda1342_i2c_probe, + .id_table = uda1342_i2c_id, +}; +module_i2c_driver(uda1342_i2c_driver); + +MODULE_DESCRIPTION("UDA1342 ALSA soc codec driver"); +MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin "); +MODULE_AUTHOR("Binbin Zhou "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/uda1342.h b/sound/soc/codecs/uda1342.h new file mode 100644 index 000000000000..ff6aea0a8b01 --- /dev/null +++ b/sound/soc/codecs/uda1342.h @@ -0,0 +1,78 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Audio support for NXP UDA1342 + * + * Copyright (c) 2005 Giorgio Padrin + * Copyright (c) 2024 Binbin Zhou + */ + +#ifndef _UDA1342_H +#define _UDA1342_H + +#define UDA1342_CLK 0x00 +#define UDA1342_IFACE 0x01 +#define UDA1342_PM 0x02 +#define UDA1342_AMIX 0x03 +#define UDA1342_HP 0x04 +#define UDA1342_MVOL 0x11 +#define UDA1342_MIXVOL 0x12 +#define UDA1342_MODE 0x12 +#define UDA1342_DEEMP 0x13 +#define UDA1342_MIXER 0x14 +#define UDA1342_INTSTAT 0x18 +#define UDA1342_DEC 0x20 +#define UDA1342_PGA 0x21 +#define UDA1342_ADC 0x22 +#define UDA1342_AGC 0x23 +#define UDA1342_DECSTAT 0x28 +#define UDA1342_RESET 0x7f + +/* Register flags */ +#define R00_EN_ADC 0x0800 +#define R00_EN_DEC 0x0400 +#define R00_EN_DAC 0x0200 +#define R00_EN_INT 0x0100 +#define R00_DAC_CLK 0x0010 +#define R01_SFORI_I2S 0x0000 +#define R01_SFORI_LSB16 0x0100 +#define R01_SFORI_LSB18 0x0200 +#define R01_SFORI_LSB20 0x0300 +#define R01_SFORI_MSB 0x0500 +#define R01_SFORI_MASK 0x0700 +#define R01_SFORO_I2S 0x0000 +#define R01_SFORO_LSB16 0x0001 +#define R01_SFORO_LSB18 0x0002 +#define R01_SFORO_LSB20 0x0003 +#define R01_SFORO_LSB24 0x0004 +#define R01_SFORO_MSB 0x0005 +#define R01_SFORO_MASK 0x0007 +#define R01_SEL_SOURCE 0x0040 +#define R01_SIM 0x0010 +#define R02_PON_PLL 0x8000 +#define R02_PON_HP 0x2000 +#define R02_PON_DAC 0x0400 +#define R02_PON_BIAS 0x0100 +#define R02_EN_AVC 0x0080 +#define R02_PON_AVC 0x0040 +#define R02_PON_LNA 0x0010 +#define R02_PON_PGAL 0x0008 +#define R02_PON_ADCL 0x0004 +#define R02_PON_PGAR 0x0002 +#define R02_PON_ADCR 0x0001 +#define R13_MTM 0x4000 +#define R14_SILENCE 0x0080 +#define R14_SDET_ON 0x0040 +#define R21_MT_ADC 0x8000 +#define R22_SEL_LNA 0x0008 +#define R22_SEL_MIC 0x0004 +#define R22_SKIP_DCFIL 0x0002 +#define R23_AGC_EN 0x0001 + +#define UDA1342_DAI_DUPLEX 0 /* playback and capture on single DAI */ +#define UDA1342_DAI_PLAYBACK 1 /* playback DAI */ +#define UDA1342_DAI_CAPTURE 2 /* capture DAI */ + +#define STATUS0_DAIFMT_MASK (~(7 << 1)) +#define STATUS0_SYSCLK_MASK (~(3 << 4)) + +#endif /* _UDA1342_H */