503 lines
14 KiB
C
503 lines
14 KiB
C
/*
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* Copyright (C) 2010-2016 Freescale Semiconductor, Inc. All Rights Reserved.
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*/
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/*
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* The code contained herein is licensed under the GNU General Public
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* License. You may obtain a copy of the GNU General Public License
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* Version 2 or later at the following locations:
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*
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* http://www.opensource.org/licenses/gpl-license.html
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* http://www.gnu.org/copyleft/gpl.html
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*/
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#include <linux/module.h>
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#include <linux/of.h>
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#include <linux/of_platform.h>
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#include <linux/slab.h>
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#include <linux/device.h>
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#include <linux/i2c.h>
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#include <linux/clk.h>
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#include <linux/delay.h>
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#include <sound/core.h>
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#include <sound/pcm.h>
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#include <sound/soc.h>
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#include <sound/initval.h>
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#include <sound/pcm_params.h>
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#include "fsl_esai.h"
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#define CODEC_CLK_EXTER_OSC 1
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#define CODEC_CLK_ESAI_HCKT 2
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#define SUPPORT_RATE_NUM 10
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struct imx_priv {
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struct clk *codec_clk;
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struct clk *esai_clk;
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unsigned int mclk_freq;
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unsigned int esai_freq;
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struct platform_device *pdev;
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struct platform_device *asrc_pdev;
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u32 asrc_rate;
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u32 asrc_format;
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bool is_codec_master;
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bool is_codec_rpmsg;
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bool is_stream_in_use[2];
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bool is_stream_tdm[2];
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};
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static struct imx_priv card_priv;
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static int imx_cs42888_surround_hw_params(struct snd_pcm_substream *substream,
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struct snd_pcm_hw_params *params)
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{
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struct snd_soc_pcm_runtime *rtd = substream->private_data;
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struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
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struct snd_soc_dai *codec_dai = rtd->codec_dai;
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struct imx_priv *priv = &card_priv;
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struct device *dev = &priv->pdev->dev;
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bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
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u32 channels = params_channels(params);
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u32 max_tdm_rate;
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u32 dai_format;
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int ret = 0;
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priv->is_stream_tdm[tx] = channels > 1 && channels % 2;
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dai_format = SND_SOC_DAIFMT_NB_NF |
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(priv->is_stream_tdm[tx] ? SND_SOC_DAIFMT_DSP_A :
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SND_SOC_DAIFMT_LEFT_J);
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priv->is_stream_in_use[tx] = true;
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if (priv->is_stream_in_use[!tx] &&
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(priv->is_stream_tdm[tx] != priv->is_stream_tdm[!tx])) {
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dev_err(dev, "Don't support different fmt for tx & rx\n");
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return -EINVAL;
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}
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priv->mclk_freq = clk_get_rate(priv->codec_clk);
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priv->esai_freq = clk_get_rate(priv->esai_clk);
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if (priv->is_codec_master) {
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/* TDM is not supported by codec in master mode */
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if (priv->is_stream_tdm[tx]) {
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dev_err(dev, "%d channels are not supported in codec master mode\n",
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channels);
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return -EINVAL;
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}
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dai_format |= SND_SOC_DAIFMT_CBM_CFM;
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if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
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ret = snd_soc_dai_set_sysclk(cpu_dai, ESAI_HCKT_EXTAL,
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priv->mclk_freq, SND_SOC_CLOCK_IN);
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else
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ret = snd_soc_dai_set_sysclk(cpu_dai, ESAI_HCKR_EXTAL,
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priv->mclk_freq, SND_SOC_CLOCK_IN);
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if (ret) {
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dev_err(dev, "failed to set cpu sysclk: %d\n", ret);
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return ret;
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}
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ret = snd_soc_dai_set_sysclk(codec_dai, 0,
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priv->mclk_freq, SND_SOC_CLOCK_OUT);
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if (ret) {
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dev_err(dev, "failed to set codec sysclk: %d\n", ret);
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return ret;
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}
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} else {
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dai_format |= SND_SOC_DAIFMT_CBS_CFS;
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if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
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ret = snd_soc_dai_set_sysclk(cpu_dai, ESAI_HCKT_EXTAL,
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priv->mclk_freq, SND_SOC_CLOCK_OUT);
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else
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ret = snd_soc_dai_set_sysclk(cpu_dai, ESAI_HCKR_EXTAL,
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priv->mclk_freq, SND_SOC_CLOCK_OUT);
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if (ret) {
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dev_err(dev, "failed to set cpu sysclk: %d\n", ret);
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return ret;
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}
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ret = snd_soc_dai_set_sysclk(codec_dai, 0,
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priv->mclk_freq, SND_SOC_CLOCK_IN);
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if (ret) {
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dev_err(dev, "failed to set codec sysclk: %d\n", ret);
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return ret;
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}
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}
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/* set cpu DAI configuration */
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ret = snd_soc_dai_set_fmt(cpu_dai, dai_format);
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if (ret) {
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dev_err(dev, "failed to set cpu dai fmt: %d\n", ret);
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return ret;
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}
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/* set i.MX active slot mask */
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if (priv->is_stream_tdm[tx]) {
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/* 2 required by ESAI BCLK divisors, 8 slots, 32 width */
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if (priv->is_codec_master)
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max_tdm_rate = priv->mclk_freq / (8*32);
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else
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max_tdm_rate = priv->esai_freq / (2*8*32);
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if (params_rate(params) > max_tdm_rate) {
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dev_err(dev,
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"maximum supported sampling rate for %d channels is %dKHz\n",
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channels, max_tdm_rate / 1000);
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return -EINVAL;
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}
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/*
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* Per datasheet, the codec expects 8 slots and 32 bits
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* for every slot in TDM mode.
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*/
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snd_soc_dai_set_tdm_slot(cpu_dai,
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BIT(channels) - 1, BIT(channels) - 1,
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8, 32);
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} else
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snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 32);
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/* set codec DAI configuration */
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ret = snd_soc_dai_set_fmt(codec_dai, dai_format);
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if (ret) {
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dev_err(dev, "failed to set codec dai fmt: %d\n", ret);
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return ret;
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}
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return 0;
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}
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static int imx_cs42888_surround_hw_free(struct snd_pcm_substream *substream)
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{
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struct imx_priv *priv = &card_priv;
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bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
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priv->is_stream_in_use[tx] = false;
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return 0;
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}
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static int imx_cs42888_surround_startup(struct snd_pcm_substream *substream)
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{
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struct snd_pcm_runtime *runtime = substream->runtime;
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static struct snd_pcm_hw_constraint_list constraint_rates;
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struct imx_priv *priv = &card_priv;
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struct device *dev = &priv->pdev->dev;
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static u32 support_rates[SUPPORT_RATE_NUM];
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int ret;
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priv->mclk_freq = clk_get_rate(priv->codec_clk);
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if (priv->mclk_freq % 12288000 == 0) {
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support_rates[0] = 48000;
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support_rates[1] = 96000;
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support_rates[2] = 192000;
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constraint_rates.list = support_rates;
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constraint_rates.count = 3;
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ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
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&constraint_rates);
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if (ret)
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return ret;
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} else
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dev_warn(dev, "mclk may be not supported %d\n", priv->mclk_freq);
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return 0;
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}
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static struct snd_soc_ops imx_cs42888_surround_ops = {
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.startup = imx_cs42888_surround_startup,
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.hw_params = imx_cs42888_surround_hw_params,
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.hw_free = imx_cs42888_surround_hw_free,
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};
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/**
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* imx_cs42888_surround_startup() is to set constrain for hw parameter, but
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* backend use same runtime as frontend, for p2p backend need to use different
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* parameter, so backend can't use the startup.
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*/
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static struct snd_soc_ops imx_cs42888_surround_ops_be = {
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.hw_params = imx_cs42888_surround_hw_params,
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.hw_free = imx_cs42888_surround_hw_free,
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};
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static const struct snd_soc_dapm_widget imx_cs42888_dapm_widgets[] = {
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SND_SOC_DAPM_LINE("Line Out Jack", NULL),
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SND_SOC_DAPM_LINE("Line In Jack", NULL),
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};
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static const struct snd_soc_dapm_route audio_map[] = {
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/* Line out jack */
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{"Line Out Jack", NULL, "AOUT1L"},
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{"Line Out Jack", NULL, "AOUT1R"},
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{"Line Out Jack", NULL, "AOUT2L"},
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{"Line Out Jack", NULL, "AOUT2R"},
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{"Line Out Jack", NULL, "AOUT3L"},
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{"Line Out Jack", NULL, "AOUT3R"},
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{"Line Out Jack", NULL, "AOUT4L"},
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{"Line Out Jack", NULL, "AOUT4R"},
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{"AIN1L", NULL, "Line In Jack"},
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{"AIN1R", NULL, "Line In Jack"},
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{"AIN2L", NULL, "Line In Jack"},
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{"AIN2R", NULL, "Line In Jack"},
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{"Playback", NULL, "CPU-Playback"},/* dai route for be and fe */
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{"CPU-Capture", NULL, "Capture"},
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{"CPU-Playback", NULL, "ASRC-Playback"},
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{"ASRC-Capture", NULL, "CPU-Capture"},
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};
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static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
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struct snd_pcm_hw_params *params) {
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struct imx_priv *priv = &card_priv;
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struct snd_interval *rate;
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struct snd_mask *mask;
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if (!priv->asrc_pdev)
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return -EINVAL;
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rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
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rate->max = rate->min = priv->asrc_rate;
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mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
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snd_mask_none(mask);
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snd_mask_set(mask, priv->asrc_format);
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return 0;
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}
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SND_SOC_DAILINK_DEFS(hifi,
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DAILINK_COMP_ARRAY(COMP_EMPTY()),
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DAILINK_COMP_ARRAY(COMP_CODEC(NULL, "cs42888")),
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DAILINK_COMP_ARRAY(COMP_EMPTY()));
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SND_SOC_DAILINK_DEFS(hifi_fe,
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DAILINK_COMP_ARRAY(COMP_EMPTY()),
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DAILINK_COMP_ARRAY(COMP_DUMMY()),
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DAILINK_COMP_ARRAY(COMP_EMPTY()));
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SND_SOC_DAILINK_DEFS(hifi_be,
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DAILINK_COMP_ARRAY(COMP_EMPTY()),
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DAILINK_COMP_ARRAY(COMP_CODEC(NULL, "cs42888")),
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DAILINK_COMP_ARRAY(COMP_DUMMY()));
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static struct snd_soc_dai_link imx_cs42888_dai[] = {
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{
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.name = "HiFi",
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.stream_name = "HiFi",
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.ops = &imx_cs42888_surround_ops,
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.ignore_pmdown_time = 1,
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SND_SOC_DAILINK_REG(hifi),
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},
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{
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.name = "HiFi-ASRC-FE",
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.stream_name = "HiFi-ASRC-FE",
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.dynamic = 1,
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.ignore_pmdown_time = 1,
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.dpcm_playback = 1,
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.dpcm_capture = 1,
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.dpcm_merged_chan = 1,
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SND_SOC_DAILINK_REG(hifi_fe),
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},
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{
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.name = "HiFi-ASRC-BE",
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.stream_name = "HiFi-ASRC-BE",
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.no_pcm = 1,
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.ignore_pmdown_time = 1,
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.dpcm_playback = 1,
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.dpcm_capture = 1,
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.ops = &imx_cs42888_surround_ops_be,
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.be_hw_params_fixup = be_hw_params_fixup,
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SND_SOC_DAILINK_REG(hifi_be),
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},
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};
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static struct snd_soc_card snd_soc_card_imx_cs42888 = {
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.name = "cs42888-audio",
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.dai_link = imx_cs42888_dai,
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.dapm_widgets = imx_cs42888_dapm_widgets,
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.num_dapm_widgets = ARRAY_SIZE(imx_cs42888_dapm_widgets),
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.dapm_routes = audio_map,
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.num_dapm_routes = ARRAY_SIZE(audio_map),
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.owner = THIS_MODULE,
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};
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/*
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* This function will register the snd_soc_pcm_link drivers.
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*/
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static int imx_cs42888_probe(struct platform_device *pdev)
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{
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struct device_node *esai_np, *codec_np;
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struct device_node *asrc_np = NULL;
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struct platform_device *esai_pdev;
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struct platform_device *asrc_pdev = NULL;
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struct imx_priv *priv = &card_priv;
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int ret;
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u32 width;
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priv->pdev = pdev;
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priv->asrc_pdev = NULL;
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if (of_property_read_bool(pdev->dev.of_node, "codec-rpmsg"))
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priv->is_codec_rpmsg = true;
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esai_np = of_parse_phandle(pdev->dev.of_node, "esai-controller", 0);
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codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0);
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if (!esai_np || !codec_np) {
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dev_err(&pdev->dev, "phandle missing or invalid\n");
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ret = -EINVAL;
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goto fail;
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}
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asrc_np = of_parse_phandle(pdev->dev.of_node, "asrc-controller", 0);
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if (asrc_np) {
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asrc_pdev = of_find_device_by_node(asrc_np);
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priv->asrc_pdev = asrc_pdev;
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}
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esai_pdev = of_find_device_by_node(esai_np);
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if (!esai_pdev) {
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dev_err(&pdev->dev, "failed to find ESAI platform device\n");
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ret = -EINVAL;
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goto fail;
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}
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if (priv->is_codec_rpmsg) {
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struct platform_device *codec_dev;
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codec_dev = of_find_device_by_node(codec_np);
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if (!codec_dev || !codec_dev->dev.driver) {
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dev_err(&pdev->dev, "failed to find codec platform device\n");
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ret = -EINVAL;
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goto fail;
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}
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priv->codec_clk = devm_clk_get(&codec_dev->dev, NULL);
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if (IS_ERR(priv->codec_clk)) {
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ret = PTR_ERR(priv->codec_clk);
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dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret);
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goto fail;
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}
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} else {
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struct i2c_client *codec_dev;
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codec_dev = of_find_i2c_device_by_node(codec_np);
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if (!codec_dev || !codec_dev->dev.driver) {
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dev_dbg(&pdev->dev, "failed to find codec platform device\n");
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ret = -EPROBE_DEFER;
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goto fail;
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}
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priv->codec_clk = devm_clk_get(&codec_dev->dev, NULL);
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if (IS_ERR(priv->codec_clk)) {
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ret = PTR_ERR(priv->codec_clk);
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dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret);
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goto fail;
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}
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}
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if (priv->is_codec_rpmsg) {
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imx_cs42888_dai[0].codecs->name = "rpmsg-audio-codec-cs42888";
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imx_cs42888_dai[0].codecs->dai_name = "cs42888";
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} else {
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imx_cs42888_dai[0].codecs->of_node = codec_np;
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}
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/*if there is no asrc controller, we only enable one device*/
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if (!asrc_pdev) {
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imx_cs42888_dai[0].cpus->dai_name = dev_name(&esai_pdev->dev);
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imx_cs42888_dai[0].platforms->of_node = esai_np;
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snd_soc_card_imx_cs42888.num_links = 1;
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snd_soc_card_imx_cs42888.num_dapm_routes =
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ARRAY_SIZE(audio_map) - 2;
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} else {
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imx_cs42888_dai[0].cpus->dai_name = dev_name(&esai_pdev->dev);
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imx_cs42888_dai[0].platforms->of_node = esai_np;
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imx_cs42888_dai[1].cpus->of_node = asrc_np;
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imx_cs42888_dai[1].platforms->of_node = asrc_np;
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imx_cs42888_dai[2].cpus->dai_name = dev_name(&esai_pdev->dev);
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snd_soc_card_imx_cs42888.num_links = 3;
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if (priv->is_codec_rpmsg) {
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imx_cs42888_dai[2].codecs->name = "rpmsg-audio-codec-cs42888";
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imx_cs42888_dai[2].codecs->dai_name = "cs42888";
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} else {
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imx_cs42888_dai[2].codecs->of_node = codec_np;
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}
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ret = of_property_read_u32(asrc_np, "fsl,asrc-rate",
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&priv->asrc_rate);
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if (ret) {
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dev_err(&pdev->dev, "failed to get output rate\n");
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ret = -EINVAL;
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goto fail;
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}
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ret = of_property_read_u32(asrc_np, "fsl,asrc-width", &width);
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if (ret) {
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dev_err(&pdev->dev, "failed to get output rate\n");
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ret = -EINVAL;
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goto fail;
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}
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if (width == 24)
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priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
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else
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priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE;
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}
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priv->esai_clk = devm_clk_get(&esai_pdev->dev, "extal");
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if (IS_ERR(priv->esai_clk)) {
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ret = PTR_ERR(priv->esai_clk);
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dev_err(&esai_pdev->dev, "failed to get cpu clk: %d\n", ret);
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goto fail;
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}
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priv->is_codec_master = false;
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if (of_property_read_bool(pdev->dev.of_node, "codec-master"))
|
|
priv->is_codec_master = true;
|
|
|
|
snd_soc_card_imx_cs42888.dev = &pdev->dev;
|
|
|
|
platform_set_drvdata(pdev, &snd_soc_card_imx_cs42888);
|
|
|
|
ret = snd_soc_register_card(&snd_soc_card_imx_cs42888);
|
|
if (ret)
|
|
dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
|
|
fail:
|
|
if (asrc_np)
|
|
of_node_put(asrc_np);
|
|
if (esai_np)
|
|
of_node_put(esai_np);
|
|
if (codec_np)
|
|
of_node_put(codec_np);
|
|
return ret;
|
|
}
|
|
|
|
static int imx_cs42888_remove(struct platform_device *pdev)
|
|
{
|
|
snd_soc_unregister_card(&snd_soc_card_imx_cs42888);
|
|
return 0;
|
|
}
|
|
|
|
static const struct of_device_id imx_cs42888_dt_ids[] = {
|
|
{ .compatible = "fsl,imx-audio-cs42888", },
|
|
{ /* sentinel */ }
|
|
};
|
|
|
|
static struct platform_driver imx_cs42888_driver = {
|
|
.probe = imx_cs42888_probe,
|
|
.remove = imx_cs42888_remove,
|
|
.driver = {
|
|
.name = "imx-cs42888",
|
|
.pm = &snd_soc_pm_ops,
|
|
.of_match_table = imx_cs42888_dt_ids,
|
|
},
|
|
};
|
|
module_platform_driver(imx_cs42888_driver);
|
|
|
|
MODULE_AUTHOR("Freescale Semiconductor, Inc.");
|
|
MODULE_DESCRIPTION("ALSA SoC cs42888 Machine Layer Driver");
|
|
MODULE_ALIAS("platform:imx-cs42888");
|
|
MODULE_LICENSE("GPL");
|