#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"
#define DAI_FMT_BASE (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF)
/**
- * CODEC private data
- *
+ * struct codec_priv - CODEC private data
* @mclk_freq: Clock rate of MCLK
* @mclk_id: MCLK (or main clock) id for set_sysclk()
* @fll_id: FLL (or secordary clock) id for set_sysclk()
};
/**
- * CPU private data
- *
- * @sysclk_freq[2]: SYSCLK rates for set_sysclk()
- * @sysclk_dir[2]: SYSCLK directions for set_sysclk()
- * @sysclk_id[2]: SYSCLK ids for set_sysclk()
+ * struct cpu_priv - CPU private data
+ * @sysclk_freq: SYSCLK rates for set_sysclk()
+ * @sysclk_dir: SYSCLK directions for set_sysclk()
+ * @sysclk_id: SYSCLK ids for set_sysclk()
* @slot_width: Slot width of each frame
*
* Note: [1] for tx and [0] for rx
};
/**
- * Freescale Generic ASOC card private data
- *
- * @dai_link[3]: DAI link structure including normal one and DPCM link
+ * 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
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;
char name[32];
};
-/**
- * This dapm route map exsits for DPCM link only.
+/*
+ * 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
{"ASRC-Capture", NULL, "AC97 Capture"},
};
+static const struct snd_soc_dapm_route audio_map_tx[] = {
+ /* 1st half -- Normal DAPM routes */
+ {"Playback", NULL, "CPU-Playback"},
+ /* 2nd half -- ASRC DAPM routes */
+ {"CPU-Playback", NULL, "ASRC-Playback"},
+};
+
/* 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),
static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_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 codec-dai is DAI Master and all configurations are already in the
- * set_bias_level(), bypass the remaining settings in hw_params().
- * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS.
- */
- if ((priv->card.set_bias_level &&
- priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) ||
- fsl_asoc_card_is_ac97(priv))
+ if (fsl_asoc_card_is_ac97(priv))
return 0;
/* Specific configurations of DAIs starts from here */
cpu_priv->sysclk_dir[tx]);
if (ret && ret != -ENOTSUPP) {
dev_err(dev, "failed to set sysclk for cpu dai\n");
- return ret;
+ goto fail;
}
if (cpu_priv->slot_width) {
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 0 to avoid error message in codec */
+ ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0),
+ codec_priv->mclk_id,
+ 0,
+ 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;
}
}
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,
},
};
-static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card,
- struct snd_soc_dapm_context *dapm,
- enum snd_soc_bias_level level)
-{
- struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
- struct snd_soc_pcm_runtime *rtd;
- struct snd_soc_dai *codec_dai;
- struct codec_priv *codec_priv = &priv->codec_priv;
- struct device *dev = card->dev;
- unsigned int pll_out;
- int ret;
-
- rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]);
- codec_dai = asoc_rtd_to_codec(rtd, 0);
- if (dapm->dev != codec_dai->dev)
- return 0;
-
- switch (level) {
- case SND_SOC_BIAS_PREPARE:
- if (dapm->bias_level != SND_SOC_BIAS_STANDBY)
- break;
-
- 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(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);
- return ret;
- }
-
- 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);
- return ret;
- }
- break;
-
- case SND_SOC_BIAS_STANDBY:
- if (dapm->bias_level != SND_SOC_BIAS_PREPARE)
- break;
-
- 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 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) {
- dev_err(dev, "failed to stop FLL: %d\n", ret);
- return ret;
- }
- break;
-
- default:
- break;
- }
-
- return 0;
-}
-
static int fsl_asoc_card_audmux_init(struct device_node *np,
struct fsl_asoc_card_priv *priv)
{
return 0;
}
+static int hp_jack_event(struct notifier_block *nb, unsigned long event,
+ void *data)
+{
+ struct snd_soc_jack *jack = (struct snd_soc_jack *)data;
+ struct snd_soc_dapm_context *dapm = &jack->card->dapm;
+
+ if (event & SND_JACK_HEADPHONE)
+ /* Disable speaker if headphone is plugged in */
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk");
+ else
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+
+ return 0;
+}
+
+static struct notifier_block hp_jack_nb = {
+ .notifier_call = hp_jack_event,
+};
+
+static int mic_jack_event(struct notifier_block *nb, unsigned long event,
+ void *data)
+{
+ struct snd_soc_jack *jack = (struct snd_soc_jack *)data;
+ struct snd_soc_dapm_context *dapm = &jack->card->dapm;
+
+ if (event & SND_JACK_MICROPHONE)
+ /* Disable dmic if microphone is plugged in */
+ snd_soc_dapm_disable_pin(dapm, "DMIC");
+ else
+ snd_soc_dapm_enable_pin(dapm, "DMIC");
+
+ return 0;
+}
+
+static struct notifier_block mic_jack_nb = {
+ .notifier_call = mic_jack_event,
+};
+
static int fsl_asoc_card_late_probe(struct snd_soc_card *card)
{
struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
struct 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 *bitclkmaster = NULL;
+ struct device_node *framemaster = NULL;
struct platform_device *cpu_pdev;
struct fsl_asoc_card_priv *priv;
- struct i2c_client *codec_dev;
+ struct device *codec_dev = NULL;
const char *codec_dai_name;
+ const char *codec_dev_name;
+ unsigned int daifmt;
u32 width;
int ret;
}
codec_np = of_parse_phandle(np, "audio-codec", 0);
- if (codec_np)
- codec_dev = of_find_i2c_device_by_node(codec_np);
- else
- codec_dev = NULL;
+ 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)
/* Get the MCLK rate only, and leave it controlled by CODEC drivers */
if (codec_dev) {
- struct clk *codec_clk = clk_get(&codec_dev->dev, NULL);
+ struct clk *codec_clk = clk_get(codec_dev, NULL);
if (!IS_ERR(codec_clk)) {
priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
/* Assign a default DAI format, and allow each card to overwrite it */
priv->dai_fmt = DAI_FMT_BASE;
+ memcpy(priv->dai_link, fsl_asoc_card_dai,
+ sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
+
+ priv->card.dapm_routes = audio_map;
+ priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
/* Diversify the card configurations */
if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
codec_dai_name = "cs42888";
- priv->card.set_bias_level = NULL;
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->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) {
codec_dai_name = "wm8962";
- priv->card.set_bias_level = fsl_asoc_card_set_bias_level;
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_CBM_CFM;
} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) {
codec_dai_name = "wm8960-hifi";
- priv->card.set_bias_level = fsl_asoc_card_set_bias_level;
priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO;
priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO;
priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
} else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) {
codec_dai_name = "ac97-hifi";
- priv->card.set_bias_level = NULL;
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_CBS_CFS |
+ 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_CBS_CFS;
+ 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 {
dev_err(&pdev->dev, "unknown Device Tree compatible\n");
ret = -EINVAL;
goto asrc_fail;
}
+ /* Format info from DT is optional. */
+ daifmt = snd_soc_of_parse_daifmt(np, NULL,
+ &bitclkmaster, &framemaster);
+ daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK;
+ if (bitclkmaster || framemaster) {
+ if (codec_np == bitclkmaster)
+ daifmt |= (codec_np == framemaster) ?
+ SND_SOC_DAIFMT_CBM_CFM : SND_SOC_DAIFMT_CBM_CFS;
+ else
+ daifmt |= (codec_np == framemaster) ?
+ SND_SOC_DAIFMT_CBS_CFM : SND_SOC_DAIFMT_CBS_CFS;
+
+ /* Override dai_fmt with value from DT */
+ priv->dai_fmt = daifmt;
+ }
+
+ /* Change direction according to format */
+ if (priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) {
+ priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_IN;
+ priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_IN;
+ }
+
+ of_node_put(bitclkmaster);
+ of_node_put(framemaster);
+
if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) {
dev_err(&pdev->dev, "failed to find codec device\n");
- ret = -EINVAL;
+ ret = -EPROBE_DEFER;
goto asrc_fail;
}
priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1;
}
- snprintf(priv->name, sizeof(priv->name), "%s-audio",
- fsl_asoc_card_is_ac97(priv) ? "ac97" :
- codec_dev->name);
-
/* Initialize sound card */
priv->pdev = pdev;
priv->card.dev = &pdev->dev;
- priv->card.name = priv->name;
+ 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.dapm_routes = fsl_asoc_card_is_ac97(priv) ?
- audio_map_ac97 : audio_map;
priv->card.late_probe = fsl_asoc_card_late_probe;
- priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;
priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets);
if (!asrc_pdev)
priv->card.num_dapm_routes /= 2;
- memcpy(priv->dai_link, fsl_asoc_card_dai,
- sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
-
- 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;
+ 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 */
goto asrc_fail;
}
- ret = of_property_read_u32(asrc_np, "fsl,asrc-width", &width);
+ ret = of_property_read_u32(asrc_np, "fsl,asrc-format",
+ &priv->asrc_format);
if (ret) {
- dev_err(&pdev->dev, "failed to get output rate\n");
- ret = -EINVAL;
- goto asrc_fail;
+ /* 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;
}
-
- if (width == 24)
- priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
- else
- priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE;
}
/* Finish card registering */
snd_soc_card_set_drvdata(&priv->card, priv);
ret = devm_snd_soc_register_card(&pdev->dev, &priv->card);
- if (ret && ret != -EPROBE_DEFER)
- dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+ if (ret) {
+ if (ret != -EPROBE_DEFER)
+ dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+ goto asrc_fail;
+ }
+
+ /*
+ * Properties "hp-det-gpio" and "mic-det-gpio" are optional, and
+ * asoc_simple_init_jack uses these properties for creating
+ * Headphone Jack and Microphone Jack.
+ *
+ * The notifier is initialized in snd_soc_card_jack_new(), then
+ * snd_soc_jack_notifier_register can be called.
+ */
+ if (of_property_read_bool(np, "hp-det-gpio")) {
+ ret = asoc_simple_init_jack(&priv->card, &priv->hp_jack,
+ 1, NULL, "Headphone Jack");
+ if (ret)
+ goto asrc_fail;
+
+ snd_soc_jack_notifier_register(&priv->hp_jack.jack, &hp_jack_nb);
+ }
+
+ if (of_property_read_bool(np, "mic-det-gpio")) {
+ ret = asoc_simple_init_jack(&priv->card, &priv->mic_jack,
+ 0, NULL, "Mic Jack");
+ if (ret)
+ goto asrc_fail;
+
+ snd_soc_jack_notifier_register(&priv->mic_jack.jack, &mic_jack_nb);
+ }
asrc_fail:
of_node_put(asrc_np);
{ .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", },
{}
};
MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids);
projects / linux-2.6-microblaze.git / blobdiff