linux-stable/sound/soc/fsl/fsl-asoc-card.c
Fabio Estevam 6ab11462c6
ASoC: fsl-asoc-card: Allow passing the number of slots in use
Currently, fsl-asoc-card supports passing the width of the TDM slot, but
not the number of slots in use, as it harcodes it as two slots.

Add support for passing the number of slots in use.

Signed-off-by: Fabio Estevam <festevam@denx.de>
Link: https://lore.kernel.org/r/20230616203913.551183-1-festevam@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2023-06-21 13:42:51 +01:00

951 lines
28 KiB
C

// 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"
#include "../codecs/tlv320aic31xx.h"
#include "../codecs/nau8822.h"
#define DRIVER_NAME "fsl-asoc-card"
#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 {
struct clk *mclk;
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
* @slot_num: Number of slots 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;
u32 slot_num;
};
/**
* 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 */
{"AC97 Playback", NULL, "CPU AC97 Playback"},
{"CPU AC97 Capture", NULL, "AC97 Capture"},
/* 2nd half -- ASRC DAPM routes */
{"CPU AC97 Playback", NULL, "ASRC-Playback"},
{"ASRC-Capture", NULL, "CPU 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) {
if (!cpu_priv->slot_num)
cpu_priv->slot_num = 2;
ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3,
cpu_priv->slot_num,
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 const 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 */
return snd_soc_dapm_disable_pin(dapm, "Ext Spk");
else
return snd_soc_dapm_enable_pin(dapm, "Ext Spk");
}
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 */
return snd_soc_dapm_disable_pin(dapm, "DMIC");
else
return snd_soc_dapm_enable_pin(dapm, "DMIC");
}
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;
}
if (!IS_ERR_OR_NULL(codec_priv->mclk))
clk_prepare_enable(codec_priv->mclk);
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 asrc_fmt = 0;
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);
priv->card.driver_name = DRIVER_NAME;
/* 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-tlv320aic31xx")) {
codec_dai_name = "tlv320dac31xx-hifi";
priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
priv->dai_link[1].dpcm_capture = 0;
priv->dai_link[2].dpcm_capture = 0;
priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
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-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 if (of_device_is_compatible(np, "fsl,imx-audio-nau8822")) {
codec_dai_name = "nau8822-hifi";
priv->codec_priv.mclk_id = NAU8822_CLK_MCLK;
priv->codec_priv.fll_id = NAU8822_CLK_PLL;
priv->codec_priv.pll_id = NAU8822_CLK_PLL;
priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
if (codec_dev)
priv->codec_priv.mclk = devm_clk_get(codec_dev, NULL);
} else {
dev_err(&pdev->dev, "unknown Device Tree compatible\n");
ret = -EINVAL;
goto asrc_fail;
}
/*
* Allow setting mclk-id from the device-tree node. Otherwise, the
* default value for each card configuration is used.
*/
of_property_read_u32(np, "mclk-id", &priv->codec_priv.mclk_id);
/* 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 exists */
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", &asrc_fmt);
priv->asrc_format = (__force snd_pcm_format_t)asrc_fmt;
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) {
dev_err_probe(&pdev->dev, ret, "snd_soc_register_card failed\n");
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-tlv320aic31xx", },
{ .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", },
{ .compatible = "fsl,imx-audio-nau8822", },
{}
};
MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids);
static struct platform_driver fsl_asoc_card_driver = {
.probe = fsl_asoc_card_probe,
.driver = {
.name = DRIVER_NAME,
.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:" DRIVER_NAME);
MODULE_LICENSE("GPL");