ASoC: fsl-asoc-card: Restructure to support deferrable card binding

Chancel Liu <chancel.liu@nxp.com> says:

The ASoC core has evolved over several kernel releases to support
deferrable card binding: when a component is not yet available,
devm_snd_soc_register_card() no longer propagates -EPROBE_DEFER back to
the machine driver. Instead the card is placed on an internal deferred
list and rebound automatically once the missing component registers.
As a result, registering a sound card no longer guarantees that all CPU
and codec components have already probed successfully.

This exposed two regressions in fsl-asoc-card:

1. The machine driver caches codec MCLK rate during probe(). On platforms
   where the MCLK is derived from the CPU DAI clock and its final rate is
   applied via assigned-clocks in the CPU DAI node, probing before the CPU
   DAI driver completes leaves fsl-asoc-card with a stale mclk_freq.

2. If a card defers due to a missing component, it queues the card onto
   the unbind_card_list and returns 0. The driver then proceeds to call
   simple_util_init_jack(). At this point, the snd_card pointer is NULL.

Patch 1 drops mclk management for nau8822 from this machine driver.

Patch 2 is a pure refactoring with no functional change. the large if/else
chain of of_device_is_compatible() calls in probe() is
replaced by a platform data table approach.

Patch 3 moves all component-dependent initialisation and jacks out of
probe() and into late_probe(), which is the correct place under the
deferrable binding model.

Link: https://patch.msgid.link/20260810093834.1511749-1-chancel.liu@oss.nxp.com
This commit is contained in:
Mark Brown 2026-08-13 15:49:35 +01:00
commit 802ff936c7
No known key found for this signature in database
GPG Key ID: 24D68B725D5487D0

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@ -69,7 +69,6 @@ static const struct snd_pcm_hw_constraint_list cs42888_channel_constraints = {
/**
* struct codec_priv - CODEC private data
* @mclk: Main clock of the CODEC
* @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()
@ -80,7 +79,6 @@ static const struct snd_pcm_hw_constraint_list cs42888_channel_constraints = {
* to stay within PLL frequency limits
*/
struct codec_priv {
struct clk *mclk;
unsigned long mclk_freq;
unsigned long free_freq;
u32 mclk_id;
@ -109,12 +107,58 @@ struct cpu_priv {
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
@ -135,6 +179,7 @@ struct fsl_asoc_card_priv {
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;
@ -609,6 +654,192 @@ static int fsl_asoc_card_spdif_init(struct device_node *codec_np[],
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)
{
@ -643,17 +874,173 @@ static struct notifier_block mic_jack_nb = {
.notifier_call = mic_jack_event,
};
static int fsl_asoc_card_late_probe(struct snd_soc_card *card)
/*
* 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 snd_soc_pcm_runtime *rtd = list_first_entry(
&card->rtd_list, struct snd_soc_pcm_runtime, list);
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;
@ -671,19 +1058,20 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card)
return 0;
}
for_each_rtd_codec_dais(rtd, codec_idx, codec_dai) {
codec_priv = &priv->codec_priv[codec_idx];
ret = fsl_asoc_card_init_codecs(card, rtd);
if (ret)
goto jack_fail;
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;
if (!IS_ERR_OR_NULL(codec_priv->mclk))
clk_prepare_enable(codec_priv->mclk);
}
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;
}
@ -697,12 +1085,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
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[2] = { NULL, NULL };
const struct fsl_asoc_card_pdata *pdata;
struct snd_soc_dai_link_component *dlc;
const char *codec_dai_name[2];
const char *codec_dev_name[2];
const char *codec_dai_name[2] = { NULL, NULL };
u32 asrc_fmt = 0;
int codec_idx;
u32 width;
@ -714,6 +1100,13 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
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)
@ -726,52 +1119,13 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
goto fail;
}
cpu_pdev = of_find_device_by_node(cpu_np);
if (!cpu_pdev) {
ret = dev_err_probe(&pdev->dev, -EPROBE_DEFER,
"failed to find CPU DAI device\n");
goto fail;
}
codec_np[0] = of_parse_phandle(np, "audio-codec", 0);
codec_np[1] = of_parse_phandle(np, "audio-codec", 1);
for (codec_idx = 0; codec_idx < 2; codec_idx++) {
if (codec_np[codec_idx]) {
struct platform_device *codec_pdev;
struct i2c_client *codec_i2c;
codec_i2c = of_find_i2c_device_by_node(codec_np[codec_idx]);
if (codec_i2c) {
codec_dev[codec_idx] = &codec_i2c->dev;
codec_dev_name[codec_idx] = codec_i2c->name;
}
if (!codec_dev[codec_idx]) {
codec_pdev = of_find_device_by_node(codec_np[codec_idx]);
if (codec_pdev) {
codec_dev[codec_idx] = &codec_pdev->dev;
codec_dev_name[codec_idx] = 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 */
for (codec_idx = 0; codec_idx < 2; codec_idx++) {
if (codec_dev[codec_idx]) {
struct clk *codec_clk = clk_get(codec_dev[codec_idx], NULL);
if (!IS_ERR(codec_clk)) {
priv->codec_priv[codec_idx].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;
@ -823,133 +1177,40 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
}
/* Diversify the card configurations */
if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
codec_dai_name[0] = "cs42888";
priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv[0].mclk_freq;
priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv[0].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;
priv->constraint_channels = &cs42888_channel_constraints;
if (priv->codec_priv[0].mclk_freq % 12288000 == 0)
priv->constraint_rates = &cs42888_rate_48k_constraints;
else if (priv->codec_priv[0].mclk_freq % 11289600 == 0)
priv->constraint_rates = &cs42888_rate_44k_constraints;
else
dev_warn(&pdev->dev, "Unknown MCLK frequency %lu, no rate constraints\n",
priv->codec_priv[0].mclk_freq);
} else if (of_device_is_compatible(np, "fsl,imx-audio-cs427x")) {
codec_dai_name[0] = "cs4271-hifi";
priv->codec_priv[0].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[0] = "sgtl5000";
priv->codec_priv[0].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[0] = "tlv320aic32x4-hifi";
priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
} else if (of_device_is_compatible(np, "fsl,imx-audio-tlv320aic31xx")) {
codec_dai_name[0] = "tlv320dac31xx-hifi";
priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC;
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;
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[0] = "wm8962";
priv->codec_priv[0].mclk_id = WM8962_SYSCLK_MCLK;
priv->codec_priv[0].fll_id = WM8962_SYSCLK_FLL;
priv->codec_priv[0].pll_id = WM8962_FLL;
priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
/*
* WM8962 has same BCLK generation limitations as WM8960.
* See WM8960 section for detailed explanation.
*/
if (of_node_name_eq(cpu_np, "sai"))
priv->exclude_format = SNDRV_PCM_FMTBIT_S20_3LE;
} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) {
codec_dai_name[0] = "wm8960-hifi";
priv->codec_priv[0].fll_id = WM8960_SYSCLK_AUTO;
priv->codec_priv[0].pll_id = WM8960_SYSCLK_AUTO;
priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
/*
* WM8960 in master mode cannot generate exact 1.92 MHz BCLK
* required for S20_3LE (48kHz × 2ch × 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).
*/
if (of_node_name_eq(cpu_np, "sai"))
priv->exclude_format = SNDRV_PCM_FMTBIT_S20_3LE;
} else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) {
codec_dai_name[0] = "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[0] = "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].playback_only = 1;
priv->dai_link[2].playback_only = 1;
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[0] = "wm8524-hifi";
priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC;
priv->dai_link[1].playback_only = 1;
priv->dai_link[2].playback_only = 1;
priv->cpu_priv.slot_width = 32;
priv->card.dapm_routes = audio_map_tx;
priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
priv->cpu_priv.sysclk_ratio[TX] = 256;
} else if (of_device_is_compatible(np, "fsl,imx-audio-si476x")) {
codec_dai_name[0] = "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[0] = "wm8994-aif1";
priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
priv->codec_priv[0].mclk_id = WM8994_FLL_SRC_MCLK1;
priv->codec_priv[0].fll_id = WM8994_SYSCLK_FLL1;
priv->codec_priv[0].pll_id = WM8994_FLL1;
priv->codec_priv[0].free_freq = priv->codec_priv[0].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[0] = "nau8822-hifi";
priv->codec_priv[0].mclk_id = NAU8822_CLK_MCLK;
priv->codec_priv[0].fll_id = NAU8822_CLK_PLL;
priv->codec_priv[0].pll_id = NAU8822_CLK_PLL;
priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
if (codec_dev[0])
priv->codec_priv[0].mclk = devm_clk_get(codec_dev[0], NULL);
} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8904")) {
codec_dai_name[0] = "wm8904-hifi";
priv->codec_priv[0].mclk_id = WM8904_FLL_MCLK;
priv->codec_priv[0].fll_id = WM8904_CLK_FLL;
priv->codec_priv[0].pll_id = WM8904_FLL_MCLK;
priv->codec_priv[0].pll_ratio_s24 = 192;
priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
} else if (of_device_is_compatible(np, "fsl,imx-audio-spdif")) {
ret = fsl_asoc_card_spdif_init(codec_np, cpu_np, codec_dai_name, priv);
}
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;
} else {
dev_err(&pdev->dev, "unknown Device Tree compatible\n");
ret = -EINVAL;
goto asrc_fail;
}
/*
@ -995,51 +1256,21 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
of_node_put(bitclkprovider);
of_node_put(frameprovider);
if (!fsl_asoc_card_is_ac97(priv) && !codec_dev[0]
&& codec_dai_name[0] != snd_soc_dummy_dlc.dai_name) {
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->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[0]);
priv->card.name = priv->name;
/*
* "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);
@ -1146,39 +1377,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
goto asrc_fail;
}
/*
* Properties "hp-det-gpios" and "mic-det-gpios" are optional, and
* simple_util_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_present(np, "hp-det-gpios") ||
of_property_present(np, "hp-det-gpio") /* deprecated */) {
ret = simple_util_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_present(np, "mic-det-gpios") ||
of_property_present(np, "mic-det-gpio") /* deprecated */) {
ret = simple_util_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[0]);
of_node_put(codec_np[1]);
put_device(&cpu_pdev->dev);
fail:
of_node_put(cpu_np);
@ -1186,21 +1388,21 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
}
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", },
{ .compatible = "fsl,imx-audio-wm8904", },
{ .compatible = "fsl,imx-audio-spdif", },
{ .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);