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OpenCloudOS-Kernel/sound/soc/fsl/fsl-asoc-card.c

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// SPDX-License-Identifier: GPL-2.0
//
// Freescale Generic ASoC Sound Card driver with ASRC
//
// Copyright (C) 2014 Freescale Semiconductor, Inc.
//
// Author: Nicolin Chen <nicoleotsuka@gmail.com>
#include <linux/clk.h>
#include <linux/i2c.h>
#include <linux/module.h>
#include <linux/of_platform.h>
#if IS_ENABLED(CONFIG_SND_AC97_CODEC)
#include <sound/ac97_codec.h>
#endif
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/jack.h>
#include <sound/simple_card_utils.h>
#include "fsl_esai.h"
#include "fsl_sai.h"
#include "imx-audmux.h"
#include "../codecs/sgtl5000.h"
#include "../codecs/wm8962.h"
#include "../codecs/wm8960.h"
#include "../codecs/wm8994.h"
#define CS427x_SYSCLK_MCLK 0
#define RX 0
#define TX 1
/* Default DAI format without Master and Slave flag */
#define DAI_FMT_BASE (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF)
/**
* struct codec_priv - CODEC private data
* @mclk_freq: Clock rate of MCLK
* @free_freq: Clock rate of MCLK for hw_free()
* @mclk_id: MCLK (or main clock) id for set_sysclk()
* @fll_id: FLL (or secordary clock) id for set_sysclk()
* @pll_id: PLL id for set_pll()
*/
struct codec_priv {
unsigned long mclk_freq;
unsigned long free_freq;
u32 mclk_id;
u32 fll_id;
u32 pll_id;
};
/**
* struct cpu_priv - CPU private data
* @sysclk_freq: SYSCLK rates for set_sysclk()
* @sysclk_dir: SYSCLK directions for set_sysclk()
* @sysclk_id: SYSCLK ids for set_sysclk()
* @slot_width: Slot width of each frame
*
* Note: [1] for tx and [0] for rx
*/
struct cpu_priv {
unsigned long sysclk_freq[2];
u32 sysclk_dir[2];
u32 sysclk_id[2];
u32 slot_width;
};
/**
* struct fsl_asoc_card_priv - Freescale Generic ASOC card private data
* @dai_link: DAI link structure including normal one and DPCM link
* @hp_jack: Headphone Jack structure
* @mic_jack: Microphone Jack structure
* @pdev: platform device pointer
* @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
* @asrc_format: ASRC sample format used by Back-Ends
* @dai_fmt: DAI format between CPU and CODEC
* @name: Card name
*/
struct fsl_asoc_card_priv {
struct snd_soc_dai_link dai_link[3];
struct asoc_simple_jack hp_jack;
struct asoc_simple_jack mic_jack;
struct platform_device *pdev;
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;
snd_pcm_format_t asrc_format;
u32 dai_fmt;
char name[32];
};
/*
* This dapm route map exists for DPCM link only.
* The other routes shall go through Device Tree.
*
* Note: keep all ASRC routes in the second half
* to drop them easily for non-ASRC cases.
*/
static const struct snd_soc_dapm_route audio_map[] = {
/* 1st half -- Normal DAPM routes */
{"Playback", NULL, "CPU-Playback"},
{"CPU-Capture", NULL, "Capture"},
/* 2nd half -- ASRC DAPM routes */
{"CPU-Playback", NULL, "ASRC-Playback"},
{"ASRC-Capture", NULL, "CPU-Capture"},
};
static const struct snd_soc_dapm_route audio_map_ac97[] = {
/* 1st half -- Normal DAPM routes */
{"Playback", NULL, "AC97 Playback"},
{"AC97 Capture", NULL, "Capture"},
/* 2nd half -- ASRC DAPM routes */
{"AC97 Playback", NULL, "ASRC-Playback"},
{"ASRC-Capture", NULL, "AC97 Capture"},
};
static const struct snd_soc_dapm_route audio_map_tx[] = {
/* 1st half -- Normal DAPM routes */
{"Playback", NULL, "CPU-Playback"},
/* 2nd half -- ASRC DAPM routes */
{"CPU-Playback", NULL, "ASRC-Playback"},
};
static const struct snd_soc_dapm_route audio_map_rx[] = {
/* 1st half -- Normal DAPM routes */
{"CPU-Capture", NULL, "Capture"},
/* 2nd half -- ASRC DAPM routes */
{"ASRC-Capture", NULL, "CPU-Capture"},
};
/* Add all possible widgets into here without being redundant */
static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = {
SND_SOC_DAPM_LINE("Line Out Jack", NULL),
SND_SOC_DAPM_LINE("Line In Jack", NULL),
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_SPK("Ext Spk", NULL),
SND_SOC_DAPM_MIC("Mic Jack", NULL),
SND_SOC_DAPM_MIC("AMIC", NULL),
SND_SOC_DAPM_MIC("DMIC", NULL),
};
static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv)
{
return priv->dai_fmt == SND_SOC_DAIFMT_AC97;
}
static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
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 (fsl_asoc_card_is_ac97(priv))
return 0;
/* Specific configurations of DAIs starts from here */
ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), cpu_priv->sysclk_id[tx],
cpu_priv->sysclk_freq[tx],
cpu_priv->sysclk_dir[tx]);
if (ret && ret != -ENOTSUPP) {
dev_err(dev, "failed to set sysclk for cpu dai\n");
goto fail;
}
if (cpu_priv->slot_width) {
ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2,
cpu_priv->slot_width);
if (ret && ret != -ENOTSUPP) {
dev_err(dev, "failed to set TDM slot for cpu dai\n");
goto fail;
}
}
/* 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 fail;
}
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 fail;
}
}
return 0;
fail:
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 free_freq to avoid error message in codec */
ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0),
codec_priv->mclk_id,
codec_priv->free_freq,
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;
}
}
return 0;
}
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,
struct snd_pcm_hw_params *params)
{
struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
struct snd_interval *rate;
struct snd_mask *mask;
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_format(mask, priv->asrc_format);
return 0;
}
SND_SOC_DAILINK_DEFS(hifi,
DAILINK_COMP_ARRAY(COMP_EMPTY()),
DAILINK_COMP_ARRAY(COMP_EMPTY()),
DAILINK_COMP_ARRAY(COMP_EMPTY()));
SND_SOC_DAILINK_DEFS(hifi_fe,
DAILINK_COMP_ARRAY(COMP_EMPTY()),
DAILINK_COMP_ARRAY(COMP_DUMMY()),
DAILINK_COMP_ARRAY(COMP_EMPTY()));
SND_SOC_DAILINK_DEFS(hifi_be,
DAILINK_COMP_ARRAY(COMP_EMPTY()),
DAILINK_COMP_ARRAY(COMP_EMPTY()),
DAILINK_COMP_ARRAY(COMP_DUMMY()));
static struct snd_soc_dai_link fsl_asoc_card_dai[] = {
/* Default ASoC DAI Link*/
{
.name = "HiFi",
.stream_name = "HiFi",
.ops = &fsl_asoc_card_ops,
SND_SOC_DAILINK_REG(hifi),
},
/* DPCM Link between Front-End and Back-End (Optional) */
{
.name = "HiFi-ASRC-FE",
.stream_name = "HiFi-ASRC-FE",
.dpcm_playback = 1,
.dpcm_capture = 1,
.dynamic = 1,
SND_SOC_DAILINK_REG(hifi_fe),
},
{
.name = "HiFi-ASRC-BE",
.stream_name = "HiFi-ASRC-BE",
.be_hw_params_fixup = be_hw_params_fixup,
.ops = &fsl_asoc_card_ops,
.dpcm_playback = 1,
.dpcm_capture = 1,
.no_pcm = 1,
SND_SOC_DAILINK_REG(hifi_be),
},
};
static int fsl_asoc_card_audmux_init(struct device_node *np,
struct fsl_asoc_card_priv *priv)
{
struct device *dev = &priv->pdev->dev;
u32 int_ptcr = 0, ext_ptcr = 0;
int int_port, ext_port;
int ret;
ret = of_property_read_u32(np, "mux-int-port", &int_port);
if (ret) {
dev_err(dev, "mux-int-port missing or invalid\n");
return ret;
}
ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
if (ret) {
dev_err(dev, "mux-ext-port missing or invalid\n");
return ret;
}
/*
* The port numbering in the hardware manual starts at 1, while
* the AUDMUX API expects it starts at 0.
*/
int_port--;
ext_port--;
/*
* Use asynchronous mode (6 wires) for all cases except AC97.
* If only 4 wires are needed, just set SSI into
* synchronous mode and enable 4 PADs in IOMUX.
*/
switch (priv->dai_fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) {
case SND_SOC_DAIFMT_CBP_CFP:
int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
IMX_AUDMUX_V2_PTCR_RFSDIR |
IMX_AUDMUX_V2_PTCR_RCLKDIR |
IMX_AUDMUX_V2_PTCR_TFSDIR |
IMX_AUDMUX_V2_PTCR_TCLKDIR;
break;
case SND_SOC_DAIFMT_CBP_CFC:
int_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
IMX_AUDMUX_V2_PTCR_RCLKDIR |
IMX_AUDMUX_V2_PTCR_TCLKDIR;
ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
IMX_AUDMUX_V2_PTCR_RFSDIR |
IMX_AUDMUX_V2_PTCR_TFSDIR;
break;
case SND_SOC_DAIFMT_CBC_CFP:
int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
IMX_AUDMUX_V2_PTCR_RFSDIR |
IMX_AUDMUX_V2_PTCR_TFSDIR;
ext_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
IMX_AUDMUX_V2_PTCR_RCLKDIR |
IMX_AUDMUX_V2_PTCR_TCLKDIR;
break;
case SND_SOC_DAIFMT_CBC_CFC:
ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
IMX_AUDMUX_V2_PTCR_RFSDIR |
IMX_AUDMUX_V2_PTCR_RCLKDIR |
IMX_AUDMUX_V2_PTCR_TFSDIR |
IMX_AUDMUX_V2_PTCR_TCLKDIR;
break;
default:
if (!fsl_asoc_card_is_ac97(priv))
return -EINVAL;
}
if (fsl_asoc_card_is_ac97(priv)) {
int_ptcr = IMX_AUDMUX_V2_PTCR_SYN |
IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
IMX_AUDMUX_V2_PTCR_TCLKDIR;
ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN |
IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
IMX_AUDMUX_V2_PTCR_TFSDIR;
}
/* Asynchronous mode can not be set along with RCLKDIR */
if (!fsl_asoc_card_is_ac97(priv)) {
unsigned int pdcr =
IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port);
ret = imx_audmux_v2_configure_port(int_port, 0,
pdcr);
if (ret) {
dev_err(dev, "audmux internal port setup failed\n");
return ret;
}
}
ret = imx_audmux_v2_configure_port(int_port, int_ptcr,
IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
if (ret) {
dev_err(dev, "audmux internal port setup failed\n");
return ret;
}
if (!fsl_asoc_card_is_ac97(priv)) {
unsigned int pdcr =
IMX_AUDMUX_V2_PDCR_RXDSEL(int_port);
ret = imx_audmux_v2_configure_port(ext_port, 0,
pdcr);
if (ret) {
dev_err(dev, "audmux external port setup failed\n");
return ret;
}
}
ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr,
IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
if (ret) {
dev_err(dev, "audmux external port setup failed\n");
return ret;
}
return 0;
}
static int hp_jack_event(struct notifier_block *nb, unsigned long event,
void *data)
{
struct snd_soc_jack *jack = (struct snd_soc_jack *)data;
struct snd_soc_dapm_context *dapm = &jack->card->dapm;
if (event & SND_JACK_HEADPHONE)
/* Disable speaker if headphone is plugged in */
snd_soc_dapm_disable_pin(dapm, "Ext Spk");
else
snd_soc_dapm_enable_pin(dapm, "Ext Spk");
return 0;
}
static struct notifier_block hp_jack_nb = {
.notifier_call = hp_jack_event,
};
static int mic_jack_event(struct notifier_block *nb, unsigned long event,
void *data)
{
struct snd_soc_jack *jack = (struct snd_soc_jack *)data;
struct snd_soc_dapm_context *dapm = &jack->card->dapm;
if (event & SND_JACK_MICROPHONE)
/* Disable dmic if microphone is plugged in */
snd_soc_dapm_disable_pin(dapm, "DMIC");
else
snd_soc_dapm_enable_pin(dapm, "DMIC");
return 0;
}
static struct notifier_block mic_jack_nb = {
.notifier_call = mic_jack_event,
};
static int fsl_asoc_card_late_probe(struct snd_soc_card *card)
{
struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
struct snd_soc_pcm_runtime *rtd = list_first_entry(
&card->rtd_list, struct snd_soc_pcm_runtime, list);
struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
struct codec_priv *codec_priv = &priv->codec_priv;
struct device *dev = card->dev;
int ret;
if (fsl_asoc_card_is_ac97(priv)) {
#if IS_ENABLED(CONFIG_SND_AC97_CODEC)
struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
struct snd_ac97 *ac97 = snd_soc_component_get_drvdata(component);
/*
* Use slots 3/4 for S/PDIF so SSI won't try to enable
* other slots and send some samples there
* due to SLOTREQ bits for S/PDIF received from codec
*/
snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS,
AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4);
#endif
return 0;
}
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 set sysclk in %s\n", __func__);
return ret;
}
return 0;
}
static int fsl_asoc_card_probe(struct platform_device *pdev)
{
struct device_node *cpu_np, *codec_np, *asrc_np;
struct device_node *np = pdev->dev.of_node;
struct platform_device *asrc_pdev = NULL;
struct device_node *bitclkprovider = NULL;
struct device_node *frameprovider = NULL;
struct platform_device *cpu_pdev;
struct fsl_asoc_card_priv *priv;
struct device *codec_dev = NULL;
const char *codec_dai_name;
const char *codec_dev_name;
u32 width;
int ret;
priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
if (!priv)
return -ENOMEM;
cpu_np = of_parse_phandle(np, "audio-cpu", 0);
/* Give a chance to old DT binding */
if (!cpu_np)
cpu_np = of_parse_phandle(np, "ssi-controller", 0);
if (!cpu_np) {
dev_err(&pdev->dev, "CPU phandle missing or invalid\n");
ret = -EINVAL;
goto fail;
}
cpu_pdev = of_find_device_by_node(cpu_np);
if (!cpu_pdev) {
dev_err(&pdev->dev, "failed to find CPU DAI device\n");
ret = -EINVAL;
goto fail;
}
codec_np = of_parse_phandle(np, "audio-codec", 0);
if (codec_np) {
struct platform_device *codec_pdev;
struct i2c_client *codec_i2c;
codec_i2c = of_find_i2c_device_by_node(codec_np);
if (codec_i2c) {
codec_dev = &codec_i2c->dev;
codec_dev_name = codec_i2c->name;
}
if (!codec_dev) {
codec_pdev = of_find_device_by_node(codec_np);
if (codec_pdev) {
codec_dev = &codec_pdev->dev;
codec_dev_name = codec_pdev->name;
}
}
}
asrc_np = of_parse_phandle(np, "audio-asrc", 0);
if (asrc_np)
asrc_pdev = of_find_device_by_node(asrc_np);
/* Get the MCLK rate only, and leave it controlled by CODEC drivers */
if (codec_dev) {
struct clk *codec_clk = clk_get(codec_dev, NULL);
if (!IS_ERR(codec_clk)) {
priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
clk_put(codec_clk);
}
}
/* Default sample rate and format, will be updated in hw_params() */
priv->sample_rate = 44100;
priv->sample_format = SNDRV_PCM_FORMAT_S16_LE;
/* Assign a default DAI format, and allow each card to overwrite it */
priv->dai_fmt = DAI_FMT_BASE;
memcpy(priv->dai_link, fsl_asoc_card_dai,
sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
priv->card.dapm_routes = audio_map;
priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
/* Diversify the card configurations */
if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
codec_dai_name = "cs42888";
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;
priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
priv->cpu_priv.slot_width = 32;
priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC;
} else if (of_device_is_compatible(np, "fsl,imx-audio-cs427x")) {
codec_dai_name = "cs4271-hifi";
priv->codec_priv.mclk_id = CS427x_SYSCLK_MCLK;
priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
} else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) {
codec_dai_name = "sgtl5000";
priv->codec_priv.mclk_id = SGTL5000_SYSCLK;
priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
} else if (of_device_is_compatible(np, "fsl,imx-audio-tlv320aic32x4")) {
codec_dai_name = "tlv320aic32x4-hifi";
priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) {
codec_dai_name = "wm8962";
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_CBP_CFP;
} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) {
codec_dai_name = "wm8960-hifi";
priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO;
priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO;
priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
} else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) {
codec_dai_name = "ac97-hifi";
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->dai_fmt = SND_SOC_DAIFMT_LEFT_J |
SND_SOC_DAIFMT_CBC_CFC |
SND_SOC_DAIFMT_NB_NF;
priv->dai_link[1].dpcm_capture = 0;
priv->dai_link[2].dpcm_capture = 0;
priv->card.dapm_routes = audio_map_tx;
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->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC;
priv->dai_link[1].dpcm_capture = 0;
priv->dai_link[2].dpcm_capture = 0;
priv->cpu_priv.slot_width = 32;
priv->card.dapm_routes = audio_map_tx;
priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
} else if (of_device_is_compatible(np, "fsl,imx-audio-si476x")) {
codec_dai_name = "si476x-codec";
priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC;
priv->card.dapm_routes = audio_map_rx;
priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_rx);
} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8958")) {
codec_dai_name = "wm8994-aif1";
priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
priv->codec_priv.mclk_id = WM8994_FLL_SRC_MCLK1;
priv->codec_priv.fll_id = WM8994_SYSCLK_FLL1;
priv->codec_priv.pll_id = WM8994_FLL1;
priv->codec_priv.free_freq = priv->codec_priv.mclk_freq;
priv->card.dapm_routes = NULL;
priv->card.num_dapm_routes = 0;
} else {
dev_err(&pdev->dev, "unknown Device Tree compatible\n");
ret = -EINVAL;
goto asrc_fail;
}
/* Format info from DT is optional. */
snd_soc_daifmt_parse_clock_provider_as_phandle(np, NULL, &bitclkprovider, &frameprovider);
if (bitclkprovider || frameprovider) {
unsigned int daifmt = snd_soc_daifmt_parse_format(np, NULL);
if (codec_np == bitclkprovider)
daifmt |= (codec_np == frameprovider) ?
SND_SOC_DAIFMT_CBP_CFP : SND_SOC_DAIFMT_CBP_CFC;
else
daifmt |= (codec_np == frameprovider) ?
SND_SOC_DAIFMT_CBC_CFP : SND_SOC_DAIFMT_CBC_CFC;
/* Override dai_fmt with value from DT */
priv->dai_fmt = daifmt;
}
/* Change direction according to format */
if (priv->dai_fmt & SND_SOC_DAIFMT_CBP_CFP) {
priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_IN;
priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_IN;
}
of_node_put(bitclkprovider);
of_node_put(frameprovider);
if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) {
dev_dbg(&pdev->dev, "failed to find codec device\n");
ret = -EPROBE_DEFER;
goto asrc_fail;
}
/* Common settings for corresponding Freescale CPU DAI driver */
if (of_node_name_eq(cpu_np, "ssi")) {
/* Only SSI needs to configure AUDMUX */
ret = fsl_asoc_card_audmux_init(np, priv);
if (ret) {
dev_err(&pdev->dev, "failed to init audmux\n");
goto asrc_fail;
}
} else if (of_node_name_eq(cpu_np, "esai")) {
struct clk *esai_clk = clk_get(&cpu_pdev->dev, "extal");
if (!IS_ERR(esai_clk)) {
priv->cpu_priv.sysclk_freq[TX] = clk_get_rate(esai_clk);
priv->cpu_priv.sysclk_freq[RX] = clk_get_rate(esai_clk);
clk_put(esai_clk);
} else if (PTR_ERR(esai_clk) == -EPROBE_DEFER) {
ret = -EPROBE_DEFER;
goto asrc_fail;
}
priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL;
priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL;
} else if (of_node_name_eq(cpu_np, "sai")) {
priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1;
priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1;
}
/* Initialize sound card */
priv->pdev = pdev;
priv->card.dev = &pdev->dev;
priv->card.owner = THIS_MODULE;
ret = snd_soc_of_parse_card_name(&priv->card, "model");
if (ret) {
snprintf(priv->name, sizeof(priv->name), "%s-audio",
fsl_asoc_card_is_ac97(priv) ? "ac97" : codec_dev_name);
priv->card.name = priv->name;
}
priv->card.dai_link = priv->dai_link;
priv->card.late_probe = fsl_asoc_card_late_probe;
priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;
priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets);
/* Drop the second half of DAPM routes -- ASRC */
if (!asrc_pdev)
priv->card.num_dapm_routes /= 2;
if (of_property_read_bool(np, "audio-routing")) {
ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing");
if (ret) {
dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret);
goto asrc_fail;
}
}
/* Normal DAI Link */
priv->dai_link[0].cpus->of_node = cpu_np;
priv->dai_link[0].codecs->dai_name = codec_dai_name;
if (!fsl_asoc_card_is_ac97(priv))
priv->dai_link[0].codecs->of_node = codec_np;
else {
u32 idx;
ret = of_property_read_u32(cpu_np, "cell-index", &idx);
if (ret) {
dev_err(&pdev->dev,
"cannot get CPU index property\n");
goto asrc_fail;
}
priv->dai_link[0].codecs->name =
devm_kasprintf(&pdev->dev, GFP_KERNEL,
"ac97-codec.%u",
(unsigned int)idx);
if (!priv->dai_link[0].codecs->name) {
ret = -ENOMEM;
goto asrc_fail;
}
}
priv->dai_link[0].platforms->of_node = cpu_np;
priv->dai_link[0].dai_fmt = priv->dai_fmt;
priv->card.num_links = 1;
if (asrc_pdev) {
/* DPCM DAI Links only if ASRC exsits */
priv->dai_link[1].cpus->of_node = asrc_np;
priv->dai_link[1].platforms->of_node = asrc_np;
priv->dai_link[2].codecs->dai_name = codec_dai_name;
priv->dai_link[2].codecs->of_node = codec_np;
priv->dai_link[2].codecs->name =
priv->dai_link[0].codecs->name;
priv->dai_link[2].cpus->of_node = cpu_np;
priv->dai_link[2].dai_fmt = priv->dai_fmt;
priv->card.num_links = 3;
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 asrc_fail;
}
ret = of_property_read_u32(asrc_np, "fsl,asrc-format",
&priv->asrc_format);
if (ret) {
/* Fallback to old binding; translate to asrc_format */
ret = of_property_read_u32(asrc_np, "fsl,asrc-width",
&width);
if (ret) {
dev_err(&pdev->dev,
"failed to decide output format\n");
goto asrc_fail;
}
if (width == 24)
priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
else
priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE;
}
}
/* Finish card registering */
platform_set_drvdata(pdev, priv);
snd_soc_card_set_drvdata(&priv->card, priv);
ret = devm_snd_soc_register_card(&pdev->dev, &priv->card);
if (ret) {
if (ret != -EPROBE_DEFER)
dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
goto asrc_fail;
}
/*
* Properties "hp-det-gpio" and "mic-det-gpio" are optional, and
* asoc_simple_init_jack uses these properties for creating
* Headphone Jack and Microphone Jack.
*
* The notifier is initialized in snd_soc_card_jack_new(), then
* snd_soc_jack_notifier_register can be called.
*/
if (of_property_read_bool(np, "hp-det-gpio")) {
ret = asoc_simple_init_jack(&priv->card, &priv->hp_jack,
1, NULL, "Headphone Jack");
if (ret)
goto asrc_fail;
snd_soc_jack_notifier_register(&priv->hp_jack.jack, &hp_jack_nb);
}
if (of_property_read_bool(np, "mic-det-gpio")) {
ret = asoc_simple_init_jack(&priv->card, &priv->mic_jack,
0, NULL, "Mic Jack");
if (ret)
goto asrc_fail;
snd_soc_jack_notifier_register(&priv->mic_jack.jack, &mic_jack_nb);
}
asrc_fail:
of_node_put(asrc_np);
of_node_put(codec_np);
put_device(&cpu_pdev->dev);
fail:
of_node_put(cpu_np);
return ret;
}
static const struct of_device_id fsl_asoc_card_dt_ids[] = {
{ .compatible = "fsl,imx-audio-ac97", },
{ .compatible = "fsl,imx-audio-cs42888", },
{ .compatible = "fsl,imx-audio-cs427x", },
{ .compatible = "fsl,imx-audio-tlv320aic32x4", },
{ .compatible = "fsl,imx-audio-sgtl5000", },
{ .compatible = "fsl,imx-audio-wm8962", },
{ .compatible = "fsl,imx-audio-wm8960", },
{ .compatible = "fsl,imx-audio-mqs", },
{ .compatible = "fsl,imx-audio-wm8524", },
{ .compatible = "fsl,imx-audio-si476x", },
{ .compatible = "fsl,imx-audio-wm8958", },
{}
};
MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids);
static struct platform_driver fsl_asoc_card_driver = {
.probe = fsl_asoc_card_probe,
.driver = {
.name = "fsl-asoc-card",
.pm = &snd_soc_pm_ops,
.of_match_table = fsl_asoc_card_dt_ids,
},
};
module_platform_driver(fsl_asoc_card_driver);
MODULE_DESCRIPTION("Freescale Generic ASoC Sound Card driver with ASRC");
MODULE_AUTHOR("Nicolin Chen <nicoleotsuka@gmail.com>");
MODULE_ALIAS("platform:fsl-asoc-card");
MODULE_LICENSE("GPL");
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