blob: 5a88426aed7ca366d960d2cd4bf9186825ec0d06 [file] [log] [blame]
/*
* Copyright (C) 2010-2016 Freescale Semiconductor, Inc. All Rights Reserved.
*/
/*
* The code contained herein is licensed under the GNU General Public
* License. You may obtain a copy of the GNU General Public License
* Version 2 or later at the following locations:
*
* http://www.opensource.org/licenses/gpl-license.html
* http://www.gnu.org/copyleft/gpl.html
*/
#include <linux/module.h>
#include <linux/of.h>
#include <linux/of_platform.h>
#include <linux/slab.h>
#include <linux/device.h>
#include <linux/i2c.h>
#include <linux/clk.h>
#include <linux/delay.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/initval.h>
#include <sound/pcm_params.h>
#include "fsl_esai.h"
#define CODEC_CLK_EXTER_OSC 1
#define CODEC_CLK_ESAI_HCKT 2
#define SUPPORT_RATE_NUM 10
struct imx_priv {
struct clk *codec_clk;
struct clk *esai_clk;
unsigned int mclk_freq;
unsigned int esai_freq;
struct platform_device *pdev;
struct platform_device *asrc_pdev;
u32 asrc_rate;
u32 asrc_format;
bool is_codec_master;
bool is_codec_rpmsg;
bool is_stream_in_use[2];
bool is_stream_tdm[2];
};
static struct imx_priv card_priv;
static int imx_cs42888_surround_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct imx_priv *priv = &card_priv;
struct device *dev = &priv->pdev->dev;
bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
u32 channels = params_channels(params);
u32 max_tdm_rate;
u32 dai_format;
int ret = 0;
priv->is_stream_tdm[tx] = channels > 1 && channels % 2;
dai_format = SND_SOC_DAIFMT_NB_NF |
(priv->is_stream_tdm[tx] ? SND_SOC_DAIFMT_DSP_A :
SND_SOC_DAIFMT_LEFT_J);
priv->is_stream_in_use[tx] = true;
if (priv->is_stream_in_use[!tx] &&
(priv->is_stream_tdm[tx] != priv->is_stream_tdm[!tx])) {
dev_err(dev, "Don't support different fmt for tx & rx\n");
return -EINVAL;
}
priv->mclk_freq = clk_get_rate(priv->codec_clk);
priv->esai_freq = clk_get_rate(priv->esai_clk);
if (priv->is_codec_master) {
/* TDM is not supported by codec in master mode */
if (priv->is_stream_tdm[tx]) {
dev_err(dev, "%d channels are not supported in codec master mode\n",
channels);
return -EINVAL;
}
dai_format |= SND_SOC_DAIFMT_CBM_CFM;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
ret = snd_soc_dai_set_sysclk(cpu_dai, ESAI_HCKT_EXTAL,
priv->mclk_freq, SND_SOC_CLOCK_IN);
else
ret = snd_soc_dai_set_sysclk(cpu_dai, ESAI_HCKR_EXTAL,
priv->mclk_freq, SND_SOC_CLOCK_IN);
if (ret) {
dev_err(dev, "failed to set cpu sysclk: %d\n", ret);
return ret;
}
ret = snd_soc_dai_set_sysclk(codec_dai, 0,
priv->mclk_freq, SND_SOC_CLOCK_OUT);
if (ret) {
dev_err(dev, "failed to set codec sysclk: %d\n", ret);
return ret;
}
} else {
dai_format |= SND_SOC_DAIFMT_CBS_CFS;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
ret = snd_soc_dai_set_sysclk(cpu_dai, ESAI_HCKT_EXTAL,
priv->mclk_freq, SND_SOC_CLOCK_OUT);
else
ret = snd_soc_dai_set_sysclk(cpu_dai, ESAI_HCKR_EXTAL,
priv->mclk_freq, SND_SOC_CLOCK_OUT);
if (ret) {
dev_err(dev, "failed to set cpu sysclk: %d\n", ret);
return ret;
}
ret = snd_soc_dai_set_sysclk(codec_dai, 0,
priv->mclk_freq, SND_SOC_CLOCK_IN);
if (ret) {
dev_err(dev, "failed to set codec sysclk: %d\n", ret);
return ret;
}
}
/* set cpu DAI configuration */
ret = snd_soc_dai_set_fmt(cpu_dai, dai_format);
if (ret) {
dev_err(dev, "failed to set cpu dai fmt: %d\n", ret);
return ret;
}
/* set i.MX active slot mask */
if (priv->is_stream_tdm[tx]) {
/* 2 required by ESAI BCLK divisors, 8 slots, 32 width */
if (priv->is_codec_master)
max_tdm_rate = priv->mclk_freq / (8*32);
else
max_tdm_rate = priv->esai_freq / (2*8*32);
if (params_rate(params) > max_tdm_rate) {
dev_err(dev,
"maximum supported sampling rate for %d channels is %dKHz\n",
channels, max_tdm_rate / 1000);
return -EINVAL;
}
/*
* Per datasheet, the codec expects 8 slots and 32 bits
* for every slot in TDM mode.
*/
snd_soc_dai_set_tdm_slot(cpu_dai,
BIT(channels) - 1, BIT(channels) - 1,
8, 32);
} else
snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 32);
/* set codec DAI configuration */
ret = snd_soc_dai_set_fmt(codec_dai, dai_format);
if (ret) {
dev_err(dev, "failed to set codec dai fmt: %d\n", ret);
return ret;
}
return 0;
}
static int imx_cs42888_surround_hw_free(struct snd_pcm_substream *substream)
{
struct imx_priv *priv = &card_priv;
bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
priv->is_stream_in_use[tx] = false;
return 0;
}
static int imx_cs42888_surround_startup(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
static struct snd_pcm_hw_constraint_list constraint_rates;
struct imx_priv *priv = &card_priv;
struct device *dev = &priv->pdev->dev;
static u32 support_rates[SUPPORT_RATE_NUM];
int ret;
priv->mclk_freq = clk_get_rate(priv->codec_clk);
if (priv->mclk_freq % 12288000 == 0) {
support_rates[0] = 48000;
support_rates[1] = 96000;
support_rates[2] = 192000;
constraint_rates.list = support_rates;
constraint_rates.count = 3;
ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
&constraint_rates);
if (ret)
return ret;
} else
dev_warn(dev, "mclk may be not supported %d\n", priv->mclk_freq);
return 0;
}
static struct snd_soc_ops imx_cs42888_surround_ops = {
.startup = imx_cs42888_surround_startup,
.hw_params = imx_cs42888_surround_hw_params,
.hw_free = imx_cs42888_surround_hw_free,
};
/**
* imx_cs42888_surround_startup() is to set constrain for hw parameter, but
* backend use same runtime as frontend, for p2p backend need to use different
* parameter, so backend can't use the startup.
*/
static struct snd_soc_ops imx_cs42888_surround_ops_be = {
.hw_params = imx_cs42888_surround_hw_params,
.hw_free = imx_cs42888_surround_hw_free,
};
static const struct snd_soc_dapm_widget imx_cs42888_dapm_widgets[] = {
SND_SOC_DAPM_LINE("Line Out Jack", NULL),
SND_SOC_DAPM_LINE("Line In Jack", NULL),
};
static const struct snd_soc_dapm_route audio_map[] = {
/* Line out jack */
{"Line Out Jack", NULL, "AOUT1L"},
{"Line Out Jack", NULL, "AOUT1R"},
{"Line Out Jack", NULL, "AOUT2L"},
{"Line Out Jack", NULL, "AOUT2R"},
{"Line Out Jack", NULL, "AOUT3L"},
{"Line Out Jack", NULL, "AOUT3R"},
{"Line Out Jack", NULL, "AOUT4L"},
{"Line Out Jack", NULL, "AOUT4R"},
{"AIN1L", NULL, "Line In Jack"},
{"AIN1R", NULL, "Line In Jack"},
{"AIN2L", NULL, "Line In Jack"},
{"AIN2R", NULL, "Line In Jack"},
{"Playback", NULL, "CPU-Playback"},/* dai route for be and fe */
{"CPU-Capture", NULL, "Capture"},
{"CPU-Playback", NULL, "ASRC-Playback"},
{"ASRC-Capture", NULL, "CPU-Capture"},
};
static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params) {
struct imx_priv *priv = &card_priv;
struct snd_interval *rate;
struct snd_mask *mask;
if (!priv->asrc_pdev)
return -EINVAL;
rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
rate->max = rate->min = priv->asrc_rate;
mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
snd_mask_none(mask);
snd_mask_set(mask, priv->asrc_format);
return 0;
}
static struct snd_soc_dai_link imx_cs42888_dai[] = {
{
.name = "HiFi",
.stream_name = "HiFi",
.codec_dai_name = "cs42888",
.ops = &imx_cs42888_surround_ops,
.ignore_pmdown_time = 1,
},
{
.name = "HiFi-ASRC-FE",
.stream_name = "HiFi-ASRC-FE",
.codec_name = "snd-soc-dummy",
.codec_dai_name = "snd-soc-dummy-dai",
.dynamic = 1,
.ignore_pmdown_time = 1,
.dpcm_playback = 1,
.dpcm_capture = 1,
.dpcm_merged_chan = 1,
},
{
.name = "HiFi-ASRC-BE",
.stream_name = "HiFi-ASRC-BE",
.codec_dai_name = "cs42888",
.platform_name = "snd-soc-dummy",
.no_pcm = 1,
.ignore_pmdown_time = 1,
.dpcm_playback = 1,
.dpcm_capture = 1,
.ops = &imx_cs42888_surround_ops_be,
.be_hw_params_fixup = be_hw_params_fixup,
},
};
static struct snd_soc_card snd_soc_card_imx_cs42888 = {
.name = "cs42888-audio",
.dai_link = imx_cs42888_dai,
.dapm_widgets = imx_cs42888_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(imx_cs42888_dapm_widgets),
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
.owner = THIS_MODULE,
};
/*
* This function will register the snd_soc_pcm_link drivers.
*/
static int imx_cs42888_probe(struct platform_device *pdev)
{
struct device_node *esai_np, *codec_np;
struct device_node *asrc_np = NULL;
struct platform_device *esai_pdev;
struct platform_device *asrc_pdev = NULL;
struct imx_priv *priv = &card_priv;
int ret;
u32 width;
priv->pdev = pdev;
priv->asrc_pdev = NULL;
if (of_property_read_bool(pdev->dev.of_node, "codec-rpmsg"))
priv->is_codec_rpmsg = true;
esai_np = of_parse_phandle(pdev->dev.of_node, "esai-controller", 0);
codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0);
if (!esai_np || !codec_np) {
dev_err(&pdev->dev, "phandle missing or invalid\n");
ret = -EINVAL;
goto fail;
}
asrc_np = of_parse_phandle(pdev->dev.of_node, "asrc-controller", 0);
if (asrc_np) {
asrc_pdev = of_find_device_by_node(asrc_np);
priv->asrc_pdev = asrc_pdev;
}
esai_pdev = of_find_device_by_node(esai_np);
if (!esai_pdev) {
dev_err(&pdev->dev, "failed to find ESAI platform device\n");
ret = -EINVAL;
goto fail;
}
if (priv->is_codec_rpmsg) {
struct platform_device *codec_dev;
codec_dev = of_find_device_by_node(codec_np);
if (!codec_dev || !codec_dev->dev.driver) {
dev_err(&pdev->dev, "failed to find codec platform device\n");
ret = -EINVAL;
goto fail;
}
priv->codec_clk = devm_clk_get(&codec_dev->dev, NULL);
if (IS_ERR(priv->codec_clk)) {
ret = PTR_ERR(priv->codec_clk);
dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret);
goto fail;
}
} else {
struct i2c_client *codec_dev;
codec_dev = of_find_i2c_device_by_node(codec_np);
if (!codec_dev || !codec_dev->dev.driver) {
dev_err(&pdev->dev, "failed to find codec platform device\n");
ret = -EINVAL;
goto fail;
}
priv->codec_clk = devm_clk_get(&codec_dev->dev, NULL);
if (IS_ERR(priv->codec_clk)) {
ret = PTR_ERR(priv->codec_clk);
dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret);
goto fail;
}
}
if (priv->is_codec_rpmsg) {
imx_cs42888_dai[0].codec_name = "rpmsg-audio-codec-cs42888";
imx_cs42888_dai[0].codec_dai_name = "cs42888";
} else {
imx_cs42888_dai[0].codec_of_node = codec_np;
}
/*if there is no asrc controller, we only enable one device*/
if (!asrc_pdev) {
imx_cs42888_dai[0].cpu_dai_name = dev_name(&esai_pdev->dev);
imx_cs42888_dai[0].platform_of_node = esai_np;
snd_soc_card_imx_cs42888.num_links = 1;
snd_soc_card_imx_cs42888.num_dapm_routes =
ARRAY_SIZE(audio_map) - 2;
} else {
imx_cs42888_dai[0].cpu_dai_name = dev_name(&esai_pdev->dev);
imx_cs42888_dai[0].platform_of_node = esai_np;
imx_cs42888_dai[1].cpu_of_node = asrc_np;
imx_cs42888_dai[1].platform_of_node = asrc_np;
imx_cs42888_dai[2].cpu_dai_name = dev_name(&esai_pdev->dev);
snd_soc_card_imx_cs42888.num_links = 3;
if (priv->is_codec_rpmsg) {
imx_cs42888_dai[2].codec_name = "rpmsg-audio-codec-cs42888";
imx_cs42888_dai[2].codec_dai_name = "cs42888";
} else {
imx_cs42888_dai[2].codec_of_node = codec_np;
}
ret = of_property_read_u32(asrc_np, "fsl,asrc-rate",
&priv->asrc_rate);
if (ret) {
dev_err(&pdev->dev, "failed to get output rate\n");
ret = -EINVAL;
goto fail;
}
ret = of_property_read_u32(asrc_np, "fsl,asrc-width", &width);
if (ret) {
dev_err(&pdev->dev, "failed to get output rate\n");
ret = -EINVAL;
goto fail;
}
if (width == 24)
priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
else
priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE;
}
priv->esai_clk = devm_clk_get(&esai_pdev->dev, "extal");
if (IS_ERR(priv->esai_clk)) {
ret = PTR_ERR(priv->esai_clk);
dev_err(&esai_pdev->dev, "failed to get cpu clk: %d\n", ret);
goto fail;
}
priv->is_codec_master = false;
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");