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gbmc/linux/refs/heads/master/./sound/soc/fsl/fsl-asoc-card.c
blob: 310ee4c282a1c273ea72aec172c8546d394e2fbd [file] [edit]
// 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"
#include "../codecs/wm8904.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)
static const u32 cs42888_rates_48k[] = {
48000, 96000, 192000,
};
static const u32 cs42888_rates_44k[] = {
44100, 88200, 176400,
};
static const u32 cs42888_channels[] = {
1, 2, 4, 6, 8,
};
static const struct snd_pcm_hw_constraint_list cs42888_rate_48k_constraints = {
.list = cs42888_rates_48k,
.count = ARRAY_SIZE(cs42888_rates_48k),
};
static const struct snd_pcm_hw_constraint_list cs42888_rate_44k_constraints = {
.list = cs42888_rates_44k,
.count = ARRAY_SIZE(cs42888_rates_44k),
};
static const struct snd_pcm_hw_constraint_list cs42888_channel_constraints = {
.list = cs42888_channels,
.count = ARRAY_SIZE(cs42888_channels),
};
/**
* 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()
* @pll_ratio_s24: PLL output ratio for S24_LE format (PLL_freq = sample_rate × ratio)
* Default is 384, but some codecs (e.g., WM8904) require lower values
* to stay within PLL frequency limits
*/
struct codec_priv {
unsigned long mclk_freq;
unsigned long free_freq;
u32 mclk_id;
int fll_id;
int pll_id;
int pll_ratio_s24;
};
/**
* 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()
* @sysclk_ratio: SYSCLK ratio on sample rate
* @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 sysclk_ratio[2];
u32 slot_width;
u32 slot_num;
};
struct fsl_asoc_card_priv;
/*
* struct fsl_asoc_card_pdata - per-compatible static card description
* @sysclk_dir: initial CPU SYSCLK direction override (0 = leave default IN)
* @sysclk_ratio: SYSCLK ratio on sample rate (0 = not used)
* @slot_width: TDM slot width (0 = not TDM)
* @codec_dai_name: name of the codec DAI
* @codec_mclk_id: MCLK id passed to set_sysclk() for the codec
* @codec_fll_id: FLL id; only valid when has_pll is true
* @codec_pll_id: PLL id; only valid when has_pll is true
* @codec_pll_ratio_s24: PLL output ratio for S24_LE
* @has_pll: codec uses PLL/FLL; codec_fll_id and codec_pll_id are valid
* @dai_fmt: DAI format flags
* @playback_only: restrict card to playback direction
* @capture_only: restrict card to capture direction
* @dapm_routes: DAPM route table override
* @num_dapm_routes: number of entries in dapm_routes
* @exclude_format: PCM format bitmask excluded (for SAI + WM8960/WM8962)
* @codec_init: codec-specific init run after mclk_freq is populated
* @probe_init: optional DT-driven init run at end of probe() (e.g. SPDIF codec discovery)
*/
struct fsl_asoc_card_pdata {
u32 sysclk_dir[2];
u32 sysclk_ratio[2];
u32 slot_width;
const char *codec_dai_name;
u32 codec_mclk_id;
int codec_fll_id;
int codec_pll_id;
int codec_pll_ratio_s24;
bool has_pll;
bool playback_only;
bool capture_only;
u32 dai_fmt;
const struct snd_soc_dapm_route *dapm_routes;
int num_dapm_routes;
u64 exclude_format;
int (*codec_init)(struct fsl_asoc_card_priv *priv);
int (*probe_init)(struct device_node *codec_np[],
struct device_node *cpu_np,
const char *codec_dai_name[],
struct fsl_asoc_card_priv *priv);
};
/**
* 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
* @pdata: pointer to the per-compatible card platform data
* @codec_priv: CODEC private data
* @cpu_priv: CPU private data
* @card: ASoC card structure
* @constraint_rates: array of supported rates
* @constraint_channels: array of supported channels
* @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
* @exclude_format: excluded format;
* @name: Card name
*/
struct fsl_asoc_card_priv {
struct snd_soc_dai_link dai_link[3];
struct simple_util_jack hp_jack;
struct simple_util_jack mic_jack;
struct platform_device *pdev;
const struct fsl_asoc_card_pdata *pdata;
struct codec_priv codec_priv[2];
struct cpu_priv cpu_priv;
struct snd_soc_card card;
const struct snd_pcm_hw_constraint_list *constraint_rates;
const struct snd_pcm_hw_constraint_list *constraint_channels;
u8 streams;
u32 sample_rate;
snd_pcm_format_t sample_format;
u32 asrc_rate;
snd_pcm_format_t asrc_format;
u32 dai_fmt;
u64 exclude_format;
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 = snd_soc_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;
struct snd_soc_dai *codec_dai;
struct cpu_priv *cpu_priv = &priv->cpu_priv;
struct device *dev = rtd->card->dev;
unsigned int pll_out, sysclk_freq;
int codec_idx;
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;
if (!cpu_priv->sysclk_freq[tx] && cpu_priv->sysclk_ratio[tx])
sysclk_freq = priv->sample_rate * cpu_priv->sysclk_ratio[tx];
else
sysclk_freq = cpu_priv->sysclk_freq[tx];
/* Specific configurations of DAIs starts from here */
ret = snd_soc_dai_set_sysclk(snd_soc_rtd_to_cpu(rtd, 0), cpu_priv->sysclk_id[tx],
sysclk_freq,
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(snd_soc_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 */
for_each_rtd_codec_dais(rtd, codec_idx, codec_dai) {
codec_priv = &priv->codec_priv[codec_idx];
if (codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) {
if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
pll_out = priv->sample_rate * codec_priv->pll_ratio_s24;
else
pll_out = priv->sample_rate * 256;
ret = snd_soc_dai_set_pll(codec_dai,
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(codec_dai,
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 = snd_soc_substream_to_rtd(substream);
struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
struct codec_priv *codec_priv;
struct snd_soc_dai *codec_dai;
struct device *dev = rtd->card->dev;
int codec_idx;
int ret;
priv->streams &= ~BIT(substream->stream);
for_each_rtd_codec_dais(rtd, codec_idx, codec_dai) {
codec_priv = &priv->codec_priv[codec_idx];
if (!priv->streams && codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) {
/* Force freq to be free_freq to avoid error message in codec */
ret = snd_soc_dai_set_sysclk(codec_dai,
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(codec_dai,
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 int fsl_asoc_card_startup(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 snd_pcm_runtime *runtime = substream->runtime;
int ret;
if (priv->exclude_format && !rtd->dai_link->no_pcm) {
ret = snd_pcm_hw_constraint_mask64(runtime,
SNDRV_PCM_HW_PARAM_FORMAT,
~priv->exclude_format);
if (ret)
return ret;
}
if (priv->constraint_channels) {
ret = snd_pcm_hw_constraint_list(runtime, 0,
SNDRV_PCM_HW_PARAM_CHANNELS,
priv->constraint_channels);
if (ret)
return ret;
}
/*
* Apply rate constraints only to frontend DAI links (no_pcm = 0).
* Skip DPCM backend (no_pcm = 1) as rate is fixed by be_hw_params_fixup()
* and ASRC frontend handles rate conversion.
*/
if (priv->constraint_rates && !rtd->dai_link->no_pcm) {
ret = snd_pcm_hw_constraint_list(runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
priv->constraint_rates);
if (ret)
return ret;
}
return 0;
}
static const struct snd_soc_ops fsl_asoc_card_ops = {
.startup = fsl_asoc_card_startup,
.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;
}
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,
},
/* DPCM Link between Front-End and Back-End (Optional) */
{
.name = "HiFi-ASRC-FE",
.stream_name = "HiFi-ASRC-FE",
.dynamic = 1,
},
{
.name = "HiFi-ASRC-BE",
.stream_name = "HiFi-ASRC-BE",
.be_hw_params_fixup = be_hw_params_fixup,
.ops = &fsl_asoc_card_ops,
.no_pcm = 1,
},
};
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 fsl_asoc_card_spdif_init(struct device_node *codec_np[],
struct device_node *cpu_np,
const char *codec_dai_name[],
struct fsl_asoc_card_priv *priv)
{
struct device *dev = &priv->pdev->dev;
struct device_node *np = dev->of_node;
if (!of_node_name_eq(cpu_np, "spdif")) {
dev_err(dev, "CPU phandle invalid, should be an SPDIF device\n");
return -EINVAL;
}
priv->dai_link[0].playback_only = true;
priv->dai_link[0].capture_only = true;
for (int i = 0; i < 2; i++) {
if (!codec_np[i])
break;
if (of_device_is_compatible(codec_np[i], "linux,spdif-dit")) {
priv->dai_link[0].capture_only = false;
codec_dai_name[i] = "dit-hifi";
} else if (of_device_is_compatible(codec_np[i], "linux,spdif-dir")) {
priv->dai_link[0].playback_only = false;
codec_dai_name[i] = "dir-hifi";
}
}
// Old SPDIF DT binding
if (!codec_np[0]) {
codec_dai_name[0] = snd_soc_dummy_dlc.dai_name;
if (of_property_read_bool(np, "spdif-out"))
priv->dai_link[0].capture_only = false;
if (of_property_read_bool(np, "spdif-in"))
priv->dai_link[0].playback_only = false;
}
if (priv->dai_link[0].playback_only && priv->dai_link[0].capture_only) {
dev_err(dev, "no enabled S/PDIF DAI link\n");
return -EINVAL;
}
if (priv->dai_link[0].playback_only) {
priv->dai_link[1].playback_only = true;
priv->dai_link[2].playback_only = true;
priv->card.dapm_routes = audio_map_tx;
priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
} else if (priv->dai_link[0].capture_only) {
priv->dai_link[1].capture_only = true;
priv->dai_link[2].capture_only = true;
priv->card.dapm_routes = audio_map_rx;
priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_rx);
}
// No DAPM routes with old bindings and dummy codec
if (!codec_np[0]) {
priv->card.dapm_routes = NULL;
priv->card.num_dapm_routes = 0;
}
if (codec_np[0] && codec_np[1]) {
priv->dai_link[0].num_codecs = 2;
priv->dai_link[2].num_codecs = 2;
}
return 0;
}
static int fsl_asoc_card_cs42888_codec_init(struct fsl_asoc_card_priv *priv)
{
unsigned long mclk_freq = priv->codec_priv[0].mclk_freq;
/*
* Set CPU sysclk frequency from codec MCLK only if not already
* set by the CPU DAI init (e.g. ESAI extal clock takes precedence).
*/
if (!priv->cpu_priv.sysclk_freq[TX])
priv->cpu_priv.sysclk_freq[TX] = mclk_freq;
if (!priv->cpu_priv.sysclk_freq[RX])
priv->cpu_priv.sysclk_freq[RX] = mclk_freq;
priv->constraint_channels = &cs42888_channel_constraints;
if (mclk_freq % 12288000 == 0)
priv->constraint_rates = &cs42888_rate_48k_constraints;
else if (mclk_freq % 11289600 == 0)
priv->constraint_rates = &cs42888_rate_44k_constraints;
else
dev_warn(&priv->pdev->dev,
"Unknown MCLK frequency %lu, no rate constraints\n",
mclk_freq);
return 0;
}
static int fsl_asoc_card_wm8958_codec_init(struct fsl_asoc_card_priv *priv)
{
priv->codec_priv[0].free_freq = priv->codec_priv[0].mclk_freq;
return 0;
}
static const struct fsl_asoc_card_pdata fsl_asoc_cs42888_pdata = {
.codec_dai_name = "cs42888",
.dai_fmt = DAI_FMT_BASE | SND_SOC_DAIFMT_CBC_CFC,
.sysclk_dir = { SND_SOC_CLOCK_OUT, SND_SOC_CLOCK_OUT },
.slot_width = 32,
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
.codec_init = fsl_asoc_card_cs42888_codec_init,
};
static const struct fsl_asoc_card_pdata fsl_asoc_cs427x_pdata = {
.codec_dai_name = "cs4271-hifi",
.dai_fmt = DAI_FMT_BASE | SND_SOC_DAIFMT_CBP_CFP,
.codec_mclk_id = CS427x_SYSCLK_MCLK,
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
};
static const struct fsl_asoc_card_pdata fsl_asoc_sgtl5000_pdata = {
.codec_dai_name = "sgtl5000",
.dai_fmt = DAI_FMT_BASE | SND_SOC_DAIFMT_CBP_CFP,
.codec_mclk_id = SGTL5000_SYSCLK,
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
};
static const struct fsl_asoc_card_pdata fsl_asoc_tlv320aic32x4_pdata = {
.codec_dai_name = "tlv320aic32x4-hifi",
.dai_fmt = DAI_FMT_BASE | SND_SOC_DAIFMT_CBP_CFP,
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
};
static const struct fsl_asoc_card_pdata fsl_asoc_tlv320aic31xx_pdata = {
.codec_dai_name = "tlv320dac31xx-hifi",
.dai_fmt = DAI_FMT_BASE | SND_SOC_DAIFMT_CBC_CFC,
.sysclk_dir = { SND_SOC_CLOCK_OUT, SND_SOC_CLOCK_OUT },
.playback_only = true,
.dapm_routes = audio_map_tx,
.num_dapm_routes = ARRAY_SIZE(audio_map_tx),
};
static const struct fsl_asoc_card_pdata fsl_asoc_wm8962_pdata = {
.codec_dai_name = "wm8962",
.dai_fmt = DAI_FMT_BASE | SND_SOC_DAIFMT_CBP_CFP,
.codec_mclk_id = WM8962_SYSCLK_MCLK,
.has_pll = true,
.codec_fll_id = WM8962_SYSCLK_FLL,
.codec_pll_id = WM8962_FLL,
/*
* WM8962 has same BCLK generation limitations as WM8960.
* See WM8960 section for detailed explanation.
*/
.exclude_format = SNDRV_PCM_FMTBIT_S20_3LE,
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
};
static const struct fsl_asoc_card_pdata fsl_asoc_wm8960_pdata = {
.codec_dai_name = "wm8960-hifi",
.dai_fmt = DAI_FMT_BASE | SND_SOC_DAIFMT_CBP_CFP,
.has_pll = true,
.codec_fll_id = WM8960_SYSCLK_AUTO,
.codec_pll_id = WM8960_SYSCLK_AUTO,
/*
* WM8960 in master mode cannot generate exact 1.92 MHz BCLK
* required for S20_3LE (48kHz x 2ch x 20bit). Closest available
* is 2.048 MHz (SYSCLK/6), which causes right channel corruption.
*
* In SAI master mode, SAI derive BCLK from MCLK using integer
* dividers only. S20_3LE requires non-integer divider ratios
* with standard MCLK frequencies. For example, 48kHz stereo
* needs 1.920 MHz BCLK, which requires a divider of 6.4 from
* 12.288 MHz MCLK (not an integer).
*/
.exclude_format = SNDRV_PCM_FMTBIT_S20_3LE,
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
};
static const struct fsl_asoc_card_pdata fsl_asoc_ac97_pdata = {
.codec_dai_name = "ac97-hifi",
.dai_fmt = SND_SOC_DAIFMT_AC97,
.dapm_routes = audio_map_ac97,
.num_dapm_routes = ARRAY_SIZE(audio_map_ac97),
};
static const struct fsl_asoc_card_pdata fsl_asoc_mqs_pdata = {
.codec_dai_name = "fsl-mqs-dai",
.dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBC_CFC |
SND_SOC_DAIFMT_NB_NF,
.playback_only = true,
.dapm_routes = audio_map_tx,
.num_dapm_routes = ARRAY_SIZE(audio_map_tx),
};
static const struct fsl_asoc_card_pdata fsl_asoc_wm8524_pdata = {
.codec_dai_name = "wm8524-hifi",
.dai_fmt = DAI_FMT_BASE | SND_SOC_DAIFMT_CBC_CFC,
/* RX=0, TX=1: set TX (index 1) to CLOCK_OUT, RX stays at default IN */
.sysclk_dir = { 0, SND_SOC_CLOCK_OUT },
.sysclk_ratio = { 0, 256 },
.slot_width = 32,
.playback_only = true,
.dapm_routes = audio_map_tx,
.num_dapm_routes = ARRAY_SIZE(audio_map_tx),
};
static const struct fsl_asoc_card_pdata fsl_asoc_si476x_pdata = {
.codec_dai_name = "si476x-codec",
.dai_fmt = DAI_FMT_BASE | SND_SOC_DAIFMT_CBC_CFC,
.dapm_routes = audio_map_rx,
.num_dapm_routes = ARRAY_SIZE(audio_map_rx),
};
static const struct fsl_asoc_card_pdata fsl_asoc_wm8958_pdata = {
.codec_dai_name = "wm8994-aif1",
.dai_fmt = DAI_FMT_BASE | SND_SOC_DAIFMT_CBP_CFP,
.codec_mclk_id = WM8994_FLL_SRC_MCLK1,
.has_pll = true,
.codec_fll_id = WM8994_SYSCLK_FLL1,
.codec_pll_id = WM8994_FLL1,
.codec_init = fsl_asoc_card_wm8958_codec_init,
};
static const struct fsl_asoc_card_pdata fsl_asoc_nau8822_pdata = {
.codec_dai_name = "nau8822-hifi",
.dai_fmt = DAI_FMT_BASE | SND_SOC_DAIFMT_CBP_CFP,
.codec_mclk_id = NAU8822_CLK_MCLK,
.has_pll = true,
.codec_fll_id = NAU8822_CLK_PLL,
.codec_pll_id = NAU8822_CLK_PLL,
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
};
static const struct fsl_asoc_card_pdata fsl_asoc_wm8904_pdata = {
.codec_dai_name = "wm8904-hifi",
.dai_fmt = DAI_FMT_BASE | SND_SOC_DAIFMT_CBP_CFP,
.codec_mclk_id = WM8904_FLL_MCLK,
.has_pll = true,
.codec_fll_id = WM8904_CLK_FLL,
.codec_pll_id = WM8904_FLL_MCLK,
.codec_pll_ratio_s24 = 192,
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
};
static const struct fsl_asoc_card_pdata fsl_asoc_spdif_pdata = {
.codec_dai_name = "spdif",
.dai_fmt = DAI_FMT_BASE,
.probe_init = fsl_asoc_card_spdif_init,
};
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 = snd_soc_card_to_dapm(jack->card);
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 = snd_soc_card_to_dapm(jack->card);
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,
};
/*
* fsl_asoc_card_init_cpu - configure CPU DAI-specific settings.
*
* Called from late_probe() when the CPU DAI component is guaranteed bound.
*/
static int fsl_asoc_card_init_cpu(struct snd_soc_card *card,
struct snd_soc_pcm_runtime *rtd)
{
struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
struct device_node *np = priv->pdev->dev.of_node;
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
const char *comp_drv_name = cpu_dai->component->driver->name;
struct device *dev = card->dev;
int ret;
if (!strcmp(comp_drv_name, "fsl-ssi")) {
/* Only SSI needs to configure AUDMUX */
ret = fsl_asoc_card_audmux_init(np, priv);
if (ret) {
dev_err(dev, "failed to init audmux\n");
return ret;
}
} else if (!strcmp(comp_drv_name, "fsl-esai")) {
struct clk *esai_clk = clk_get(cpu_dai->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 {
dev_warn(dev, "failed to get ESAI extal clock: %ld\n", PTR_ERR(esai_clk));
}
priv->cpu_priv.sysclk_id[TX] = ESAI_HCKT_EXTAL;
priv->cpu_priv.sysclk_id[RX] = ESAI_HCKR_EXTAL;
} else if (!strcmp(comp_drv_name, "fsl-sai")) {
priv->cpu_priv.sysclk_id[TX] = FSL_SAI_CLK_MAST1;
priv->cpu_priv.sysclk_id[RX] = FSL_SAI_CLK_MAST1;
if (priv->pdata->exclude_format)
priv->exclude_format = priv->pdata->exclude_format;
}
return 0;
}
/*
* fsl_asoc_card_init_codecs - read codec MCLK rates and set codec sysclk.
*
* Called from late_probe() after all components are bound.
*/
static int fsl_asoc_card_init_codecs(struct snd_soc_card *card,
struct snd_soc_pcm_runtime *rtd)
{
struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
const struct fsl_asoc_card_pdata *pdata = priv->pdata;
struct snd_soc_dai *codec_dai;
struct codec_priv *codec_priv;
struct device *dev = card->dev;
int codec_idx;
int ret;
/* Read MCLK rate from each bound codec component */
for_each_rtd_codec_dais(rtd, codec_idx, codec_dai) {
struct clk *codec_clk = clk_get(codec_dai->component->dev, NULL);
codec_priv = &priv->codec_priv[codec_idx];
if (!IS_ERR(codec_clk)) {
codec_priv->mclk_freq = clk_get_rate(codec_clk);
clk_put(codec_clk);
}
}
if (pdata->codec_init) {
ret = pdata->codec_init(priv);
if (ret)
return ret;
}
for_each_rtd_codec_dais(rtd, codec_idx, codec_dai) {
codec_priv = &priv->codec_priv[codec_idx];
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 void fsl_asoc_card_free_jack(struct snd_soc_card *card)
{
struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
if (priv->hp_jack.gpio.desc) {
snd_soc_jack_notifier_unregister(&priv->hp_jack.jack, &hp_jack_nb);
snd_soc_jack_free_gpios(&priv->hp_jack.jack, 1, &priv->hp_jack.gpio);
priv->hp_jack.gpio.desc = NULL;
}
if (priv->mic_jack.gpio.desc) {
snd_soc_jack_notifier_unregister(&priv->mic_jack.jack, &mic_jack_nb);
snd_soc_jack_free_gpios(&priv->mic_jack.jack, 1, &priv->mic_jack.gpio);
priv->mic_jack.gpio.desc = NULL;
}
}
/*
* fsl_asoc_card_init_jack - register optional headphone and mic jacks.
*
* Called from late_probe() once per card bind cycle.
*/
static int fsl_asoc_card_init_jack(struct snd_soc_card *card)
{
struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
struct device_node *np = priv->pdev->dev.of_node;
int ret;
/*
* Properties "hp-det-gpios" and "mic-det-gpios" are optional.
* simple_util_init_jack() checks for the GPIO property and
* does nothing if it is absent.
*/
if (of_property_present(np, "hp-det-gpios") ||
of_property_present(np, "hp-det-gpio") /* deprecated */) {
ret = simple_util_init_jack(card, &priv->hp_jack,
1, NULL, "Headphone Jack");
if (ret)
return ret;
snd_soc_jack_notifier_register(&priv->hp_jack.jack, &hp_jack_nb);
}
if (of_property_present(np, "mic-det-gpios") ||
of_property_present(np, "mic-det-gpio") /* deprecated */) {
ret = simple_util_init_jack(card, &priv->mic_jack,
0, NULL, "Mic Jack");
if (ret)
return ret;
snd_soc_jack_notifier_register(&priv->mic_jack.jack, &mic_jack_nb);
}
return 0;
}
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;
int ret;
/* Use the first rtd which carries the CPU+codec DAIs */
rtd = list_first_entry(&card->rtd_list,
struct snd_soc_pcm_runtime, list);
ret = fsl_asoc_card_init_jack(card);
if (ret)
goto jack_fail;
ret = fsl_asoc_card_init_cpu(card, rtd);
if (ret)
goto jack_fail;
if (fsl_asoc_card_is_ac97(priv)) {
#if IS_ENABLED(CONFIG_SND_AC97_CODEC)
struct snd_soc_component *component = snd_soc_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 = fsl_asoc_card_init_codecs(card, rtd);
if (ret)
goto jack_fail;
return 0;
jack_fail:
fsl_asoc_card_free_jack(card);
return ret;
}
static int fsl_asoc_card_card_remove(struct snd_soc_card *card)
{
fsl_asoc_card_free_jack(card);
return 0;
}
static int fsl_asoc_card_probe(struct platform_device *pdev)
{
struct device_node *cpu_np, *asrc_np;
struct snd_soc_dai_link_component *codec_comp;
struct device_node *codec_np[2];
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 fsl_asoc_card_priv *priv;
const struct fsl_asoc_card_pdata *pdata;
struct snd_soc_dai_link_component *dlc;
const char *codec_dai_name[2] = { NULL, NULL };
u32 asrc_fmt = 0;
int codec_idx;
u32 width;
int ret;
priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
if (!priv)
return -ENOMEM;
priv->pdev = pdev;
pdata = of_device_get_match_data(&pdev->dev);
if (!pdata) {
dev_err(&pdev->dev, "unknown Device Tree compatible\n");
return -EINVAL;
}
priv->pdata = pdata;
cpu_np = of_parse_phandle(np, "audio-cpu", 0);
/* Give a chance to old DT bindings */
if (!cpu_np)
cpu_np = of_parse_phandle(np, "ssi-controller", 0);
if (!cpu_np)
cpu_np = of_parse_phandle(np, "spdif-controller", 0);
if (!cpu_np) {
dev_err(&pdev->dev, "CPU phandle missing or invalid\n");
ret = -EINVAL;
goto fail;
}
codec_np[0] = of_parse_phandle(np, "audio-codec", 0);
codec_np[1] = of_parse_phandle(np, "audio-codec", 1);
asrc_np = of_parse_phandle(np, "audio-asrc", 0);
if (asrc_np)
asrc_pdev = of_find_device_by_node(asrc_np);
/* 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));
/*
* "Default ASoC DAI Link": 1 cpus, 2 codecs, 1 platforms
* "DPCM Link Front-End": 1 cpus, 1 codecs (dummy), 1 platforms
* "DPCM Link Back-End": 1 cpus, 2 codecs
* totally 10 components
*/
dlc = devm_kcalloc(&pdev->dev, 10, sizeof(*dlc), GFP_KERNEL);
if (!dlc) {
ret = -ENOMEM;
goto asrc_fail;
}
priv->dai_link[0].cpus = &dlc[0];
priv->dai_link[0].num_cpus = 1;
priv->dai_link[0].codecs = &dlc[1];
priv->dai_link[0].num_codecs = 1;
priv->dai_link[0].platforms = &dlc[3];
priv->dai_link[0].num_platforms = 1;
priv->dai_link[1].cpus = &dlc[4];
priv->dai_link[1].num_cpus = 1;
priv->dai_link[1].codecs = &dlc[5];
priv->dai_link[1].num_codecs = 0; /* dummy */
priv->dai_link[1].platforms = &dlc[6];
priv->dai_link[1].num_platforms = 1;
priv->dai_link[2].cpus = &dlc[7];
priv->dai_link[2].num_cpus = 1;
priv->dai_link[2].codecs = &dlc[8];
priv->dai_link[2].num_codecs = 1;
priv->card.dapm_routes = audio_map;
priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
priv->card.driver_name = DRIVER_NAME;
for (codec_idx = 0; codec_idx < 2; codec_idx++) {
priv->codec_priv[codec_idx].fll_id = -1;
priv->codec_priv[codec_idx].pll_id = -1;
priv->codec_priv[codec_idx].pll_ratio_s24 = 384;
}
/* Diversify the card configurations */
priv->cpu_priv.sysclk_dir[TX] = pdata->sysclk_dir[TX];
priv->cpu_priv.sysclk_dir[RX] = pdata->sysclk_dir[RX];
priv->cpu_priv.sysclk_ratio[TX] = pdata->sysclk_ratio[TX];
priv->cpu_priv.sysclk_ratio[RX] = pdata->sysclk_ratio[RX];
priv->cpu_priv.slot_width = pdata->slot_width;
codec_dai_name[0] = pdata->codec_dai_name;
priv->codec_priv[0].mclk_id = pdata->codec_mclk_id;
if (pdata->has_pll) {
priv->codec_priv[0].fll_id = pdata->codec_fll_id;
priv->codec_priv[0].pll_id = pdata->codec_pll_id;
}
if (pdata->codec_pll_ratio_s24)
priv->codec_priv[0].pll_ratio_s24 = pdata->codec_pll_ratio_s24;
if (pdata->playback_only) {
priv->dai_link[1].playback_only = 1;
priv->dai_link[2].playback_only = 1;
}
if (pdata->capture_only) {
priv->dai_link[1].capture_only = 1;
priv->dai_link[2].capture_only = 1;
}
priv->dai_fmt = pdata->dai_fmt;
priv->card.dapm_routes = pdata->dapm_routes;
priv->card.num_dapm_routes = pdata->num_dapm_routes;
if (pdata->probe_init) {
ret = pdata->probe_init(codec_np, cpu_np,
codec_dai_name, priv);
if (ret)
goto asrc_fail;
}
/*
* Allow setting mclk-id from the device-tree node. Otherwise, the
* default value for each card configuration is used.
*/
for_each_link_codecs((&(priv->dai_link[0])), codec_idx, codec_comp) {
of_property_read_u32_index(np, "mclk-id", codec_idx,
&priv->codec_priv[codec_idx].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);
bool codec_bitclkprovider = false;
bool codec_frameprovider = false;
for_each_link_codecs((&(priv->dai_link[0])), codec_idx, codec_comp) {
if (bitclkprovider && codec_np[codec_idx] == bitclkprovider)
codec_bitclkprovider = true;
if (frameprovider && codec_np[codec_idx] == frameprovider)
codec_frameprovider = true;
}
if (codec_bitclkprovider)
daifmt |= (codec_frameprovider) ?
SND_SOC_DAIFMT_CBP_CFP : SND_SOC_DAIFMT_CBP_CFC;
else
daifmt |= (codec_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);
/* Initialize sound card */
priv->card.dev = &pdev->dev;
priv->card.owner = THIS_MODULE;
ret = snd_soc_of_parse_card_name(&priv->card, "model");
if (ret) {
/*
* "model" is required by the DT binding. Enforce it here so
* the driver fails with a clear message.
*/
dev_err(&pdev->dev, "Error parsing card name: %d\n", ret);
goto asrc_fail;
}
priv->card.dai_link = priv->dai_link;
priv->card.late_probe = fsl_asoc_card_late_probe;
priv->card.remove = fsl_asoc_card_card_remove;
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_present(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;
for_each_link_codecs((&(priv->dai_link[0])), codec_idx, codec_comp) {
codec_comp->dai_name = codec_dai_name[codec_idx];
}
// Old SPDIF DT binding support
if (codec_dai_name[0] == snd_soc_dummy_dlc.dai_name)
priv->dai_link[0].codecs[0].name = snd_soc_dummy_dlc.name;
if (!fsl_asoc_card_is_ac97(priv)) {
for_each_link_codecs((&(priv->dai_link[0])), codec_idx, codec_comp) {
codec_comp->of_node = codec_np[codec_idx];
}
} 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[0].name =
devm_kasprintf(&pdev->dev, GFP_KERNEL,
"ac97-codec.%u",
(unsigned int)idx);
if (!priv->dai_link[0].codecs[0].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].dpcm_merged_chan = 1;
priv->dai_link[1].ignore_pmdown_time = 1;
priv->dai_link[1].cpus->of_node = asrc_np;
priv->dai_link[1].platforms->of_node = asrc_np;
for_each_link_codecs((&(priv->dai_link[2])), codec_idx, codec_comp) {
codec_comp->dai_name = priv->dai_link[0].codecs[codec_idx].dai_name;
codec_comp->of_node = priv->dai_link[0].codecs[codec_idx].of_node;
codec_comp->name = priv->dai_link[0].codecs[codec_idx].name;
}
priv->dai_link[2].cpus->of_node = cpu_np;
priv->dai_link[2].dai_fmt = priv->dai_fmt;
priv->dai_link[2].ignore_pmdown_time = 1;
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 = 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;
}
asrc_fail:
of_node_put(asrc_np);
of_node_put(codec_np[0]);
of_node_put(codec_np[1]);
fail:
of_node_put(cpu_np);
return ret;
}
static const struct of_device_id fsl_asoc_card_dt_ids[] = {
{ .compatible = "fsl,imx-audio-ac97", .data = &fsl_asoc_ac97_pdata },
{ .compatible = "fsl,imx-audio-cs42888", .data = &fsl_asoc_cs42888_pdata },
{ .compatible = "fsl,imx-audio-cs427x", .data = &fsl_asoc_cs427x_pdata },
{ .compatible = "fsl,imx-audio-tlv320aic32x4", .data = &fsl_asoc_tlv320aic32x4_pdata },
{ .compatible = "fsl,imx-audio-tlv320aic31xx", .data = &fsl_asoc_tlv320aic31xx_pdata },
{ .compatible = "fsl,imx-audio-sgtl5000", .data = &fsl_asoc_sgtl5000_pdata },
{ .compatible = "fsl,imx-audio-wm8962", .data = &fsl_asoc_wm8962_pdata },
{ .compatible = "fsl,imx-audio-wm8960", .data = &fsl_asoc_wm8960_pdata },
{ .compatible = "fsl,imx-audio-mqs", .data = &fsl_asoc_mqs_pdata },
{ .compatible = "fsl,imx-audio-wm8524", .data = &fsl_asoc_wm8524_pdata },
{ .compatible = "fsl,imx-audio-si476x", .data = &fsl_asoc_si476x_pdata },
{ .compatible = "fsl,imx-audio-wm8958", .data = &fsl_asoc_wm8958_pdata },
{ .compatible = "fsl,imx-audio-nau8822", .data = &fsl_asoc_nau8822_pdata },
{ .compatible = "fsl,imx-audio-wm8904", .data = &fsl_asoc_wm8904_pdata },
{ .compatible = "fsl,imx-audio-spdif", .data = &fsl_asoc_spdif_pdata },
{}
};
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");