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-rw-r--r--sound/soc/fsl/Kconfig1
-rw-r--r--sound/soc/fsl/efika-audio-fabric.c22
-rw-r--r--sound/soc/fsl/eukrea-tlv320.c19
-rw-r--r--sound/soc/fsl/fsl-asoc-card.c47
-rw-r--r--sound/soc/fsl/fsl_asrc.c103
-rw-r--r--sound/soc/fsl/fsl_esai.c141
-rw-r--r--sound/soc/fsl/fsl_sai.c54
-rw-r--r--sound/soc/fsl/fsl_ssi.c4
-rw-r--r--sound/soc/fsl/fsl_ssi.h8
-rw-r--r--sound/soc/fsl/fsl_ssi_dbg.c18
-rw-r--r--sound/soc/fsl/fsl_utils.c2
-rw-r--r--sound/soc/fsl/imx-audmix.c45
-rw-r--r--sound/soc/fsl/imx-audmux.c10
-rw-r--r--sound/soc/fsl/imx-es8328.c23
-rw-r--r--sound/soc/fsl/imx-mc13783.c10
-rw-r--r--sound/soc/fsl/imx-pcm-dma.c6
-rw-r--r--sound/soc/fsl/imx-sgtl5000.c23
-rw-r--r--sound/soc/fsl/imx-spdif.c20
-rw-r--r--sound/soc/fsl/mpc8610_hpcd.c33
-rw-r--r--sound/soc/fsl/mx27vis-aic32x4.c11
-rw-r--r--sound/soc/fsl/p1022_ds.c36
-rw-r--r--sound/soc/fsl/p1022_rdk.c35
-rw-r--r--sound/soc/fsl/pcm030-audio-fabric.c20
-rw-r--r--sound/soc/fsl/phycore-ac97.c10
-rw-r--r--sound/soc/fsl/wm1133-ev1.c10
25 files changed, 474 insertions, 237 deletions
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 55ed47c599e2..aa99c008a925 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -1,3 +1,4 @@
+# SPDX-License-Identifier: GPL-2.0-only
menu "SoC Audio for Freescale CPUs"
comment "Common SoC Audio options for Freescale CPUs:"
diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c
index 667f4215dfc0..8f6396faec9b 100644
--- a/sound/soc/fsl/efika-audio-fabric.c
+++ b/sound/soc/fsl/efika-audio-fabric.c
@@ -29,22 +29,28 @@
#define DRV_NAME "efika-audio-fabric"
+SND_SOC_DAILINK_DEFS(analog,
+ DAILINK_COMP_ARRAY(COMP_CPU("mpc5200-psc-ac97.0")),
+ DAILINK_COMP_ARRAY(COMP_CODEC("stac9766-codec",
+ "stac9766-hifi-analog")),
+ DAILINK_COMP_ARRAY(COMP_PLATFORM("mpc5200-pcm-audio")));
+
+SND_SOC_DAILINK_DEFS(iec958,
+ DAILINK_COMP_ARRAY(COMP_CPU("mpc5200-psc-ac97.1")),
+ DAILINK_COMP_ARRAY(COMP_CODEC("stac9766-codec",
+ "stac9766-hifi-IEC958")),
+ DAILINK_COMP_ARRAY(COMP_PLATFORM("mpc5200-pcm-audio")));
+
static struct snd_soc_dai_link efika_fabric_dai[] = {
{
.name = "AC97",
.stream_name = "AC97 Analog",
- .codec_dai_name = "stac9766-hifi-analog",
- .cpu_dai_name = "mpc5200-psc-ac97.0",
- .platform_name = "mpc5200-pcm-audio",
- .codec_name = "stac9766-codec",
+ SND_SOC_DAILINK_REG(analog),
},
{
.name = "AC97",
.stream_name = "AC97 IEC958",
- .codec_dai_name = "stac9766-hifi-IEC958",
- .cpu_dai_name = "mpc5200-psc-ac97.1",
- .platform_name = "mpc5200-pcm-audio",
- .codec_name = "stac9766-codec",
+ SND_SOC_DAILINK_REG(iec958),
},
};
diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c
index d648268cb454..6f3b768489f6 100644
--- a/sound/soc/fsl/eukrea-tlv320.c
+++ b/sound/soc/fsl/eukrea-tlv320.c
@@ -61,13 +61,18 @@ static const struct snd_soc_ops eukrea_tlv320_snd_ops = {
.hw_params = eukrea_tlv320_hw_params,
};
+SND_SOC_DAILINK_DEFS(hifi,
+ DAILINK_COMP_ARRAY(COMP_EMPTY()),
+ DAILINK_COMP_ARRAY(COMP_CODEC(NULL, "tlv320aic23-hifi")),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
static struct snd_soc_dai_link eukrea_tlv320_dai = {
.name = "tlv320aic23",
.stream_name = "TLV320AIC23",
- .codec_dai_name = "tlv320aic23-hifi",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM,
.ops = &eukrea_tlv320_snd_ops,
+ SND_SOC_DAILINK_REG(hifi),
};
static struct snd_soc_card eukrea_tlv320 = {
@@ -104,7 +109,7 @@ static int eukrea_tlv320_probe(struct platform_device *pdev)
codec_np = of_parse_phandle(ssi_np, "codec-handle", 0);
if (codec_np)
- eukrea_tlv320_dai.codec_of_node = codec_np;
+ eukrea_tlv320_dai.codecs->of_node = codec_np;
else
dev_err(&pdev->dev, "codec-handle node missing or invalid.\n");
@@ -128,12 +133,12 @@ static int eukrea_tlv320_probe(struct platform_device *pdev)
int_port--;
ext_port--;
- eukrea_tlv320_dai.cpu_of_node = ssi_np;
- eukrea_tlv320_dai.platform_of_node = ssi_np;
+ eukrea_tlv320_dai.cpus->of_node = ssi_np;
+ eukrea_tlv320_dai.platforms->of_node = ssi_np;
} else {
- eukrea_tlv320_dai.cpu_dai_name = "imx-ssi.0";
- eukrea_tlv320_dai.platform_name = "imx-ssi.0";
- eukrea_tlv320_dai.codec_name = "tlv320aic23-codec.0-001a";
+ eukrea_tlv320_dai.cpus->dai_name = "imx-ssi.0";
+ eukrea_tlv320_dai.platforms->name = "imx-ssi.0";
+ eukrea_tlv320_dai.codecs->name = "tlv320aic23-codec.0-001a";
eukrea_tlv320.name = "cpuimx-audio";
}
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 60f87a0d99f4..39ea9bda1394 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -200,32 +200,47 @@ static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
return 0;
}
+SND_SOC_DAILINK_DEFS(hifi,
+ DAILINK_COMP_ARRAY(COMP_EMPTY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+SND_SOC_DAILINK_DEFS(hifi_fe,
+ DAILINK_COMP_ARRAY(COMP_EMPTY()),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+SND_SOC_DAILINK_DEFS(hifi_be,
+ DAILINK_COMP_ARRAY(COMP_EMPTY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()));
+
static struct snd_soc_dai_link fsl_asoc_card_dai[] = {
/* Default ASoC DAI Link*/
{
.name = "HiFi",
.stream_name = "HiFi",
.ops = &fsl_asoc_card_ops,
+ SND_SOC_DAILINK_REG(hifi),
},
/* DPCM Link between Front-End and Back-End (Optional) */
{
.name = "HiFi-ASRC-FE",
.stream_name = "HiFi-ASRC-FE",
- .codec_name = "snd-soc-dummy",
- .codec_dai_name = "snd-soc-dummy-dai",
.dpcm_playback = 1,
.dpcm_capture = 1,
.dynamic = 1,
+ SND_SOC_DAILINK_REG(hifi_fe),
},
{
.name = "HiFi-ASRC-BE",
.stream_name = "HiFi-ASRC-BE",
- .platform_name = "snd-soc-dummy",
.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),
},
};
@@ -616,11 +631,11 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
}
/* Normal DAI Link */
- priv->dai_link[0].cpu_of_node = cpu_np;
- priv->dai_link[0].codec_dai_name = codec_dai_name;
+ 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].codec_of_node = codec_np;
+ priv->dai_link[0].codecs->of_node = codec_np;
else {
u32 idx;
@@ -631,29 +646,29 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
goto asrc_fail;
}
- priv->dai_link[0].codec_name =
+ priv->dai_link[0].codecs->name =
devm_kasprintf(&pdev->dev, GFP_KERNEL,
"ac97-codec.%u",
(unsigned int)idx);
- if (!priv->dai_link[0].codec_name) {
+ if (!priv->dai_link[0].codecs->name) {
ret = -ENOMEM;
goto asrc_fail;
}
}
- priv->dai_link[0].platform_of_node = cpu_np;
+ priv->dai_link[0].platforms->of_node = cpu_np;
priv->dai_link[0].dai_fmt = priv->dai_fmt;
priv->card.num_links = 1;
if (asrc_pdev) {
/* DPCM DAI Links only if ASRC exsits */
- priv->dai_link[1].cpu_of_node = asrc_np;
- priv->dai_link[1].platform_of_node = asrc_np;
- priv->dai_link[2].codec_dai_name = codec_dai_name;
- priv->dai_link[2].codec_of_node = codec_np;
- priv->dai_link[2].codec_name =
- priv->dai_link[0].codec_name;
- priv->dai_link[2].cpu_of_node = cpu_np;
+ 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;
diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c
index ea035c12a325..cbbf6257f08a 100644
--- a/sound/soc/fsl/fsl_asrc.c
+++ b/sound/soc/fsl/fsl_asrc.c
@@ -26,32 +26,15 @@
#define pair_dbg(fmt, ...) \
dev_dbg(&asrc_priv->pdev->dev, "Pair %c: " fmt, 'A' + index, ##__VA_ARGS__)
-/* Sample rates are aligned with that defined in pcm.h file */
-static const u8 process_option[][12][2] = {
- /* 8kHz 11.025kHz 16kHz 22.05kHz 32kHz 44.1kHz 48kHz 64kHz 88.2kHz 96kHz 176kHz 192kHz */
- {{0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 5512Hz */
- {{0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 8kHz */
- {{0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 11025Hz */
- {{1, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 16kHz */
- {{1, 2}, {1, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 22050Hz */
- {{1, 2}, {2, 1}, {2, 1}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0},}, /* 32kHz */
- {{2, 2}, {2, 2}, {2, 1}, {2, 1}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0},}, /* 44.1kHz */
- {{2, 2}, {2, 2}, {2, 1}, {2, 1}, {0, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0},}, /* 48kHz */
- {{2, 2}, {2, 2}, {2, 2}, {2, 1}, {1, 2}, {0, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0},}, /* 64kHz */
- {{2, 2}, {2, 2}, {2, 2}, {2, 2}, {1, 2}, {1, 2}, {1, 2}, {1, 1}, {1, 1}, {1, 1}, {1, 1}, {1, 1},}, /* 88.2kHz */
- {{2, 2}, {2, 2}, {2, 2}, {2, 2}, {1, 2}, {1, 2}, {1, 2}, {1, 1}, {1, 1}, {1, 1}, {1, 1}, {1, 1},}, /* 96kHz */
- {{2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 1}, {2, 1}, {2, 1}, {2, 1}, {2, 1},}, /* 176kHz */
- {{2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 1}, {2, 1}, {2, 1}, {2, 1}, {2, 1},}, /* 192kHz */
-};
-
/* Corresponding to process_option */
-static int supported_input_rate[] = {
- 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, 88200,
- 96000, 176400, 192000,
+static unsigned int supported_asrc_rate[] = {
+ 5512, 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,
+ 64000, 88200, 96000, 128000, 176400, 192000,
};
-static int supported_asrc_rate[] = {
- 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000,
+static struct snd_pcm_hw_constraint_list fsl_asrc_rate_constraints = {
+ .count = ARRAY_SIZE(supported_asrc_rate),
+ .list = supported_asrc_rate,
};
/**
@@ -80,6 +63,52 @@ static unsigned char output_clk_map_imx53[] = {
static unsigned char *clk_map[2];
/**
+ * Select the pre-processing and post-processing options
+ * Make sure to exclude following unsupported cases before
+ * calling this function:
+ * 1) inrate > 8.125 * outrate
+ * 2) inrate > 16.125 * outrate
+ *
+ * inrate: input sample rate
+ * outrate: output sample rate
+ * pre_proc: return value for pre-processing option
+ * post_proc: return value for post-processing option
+ */
+static void fsl_asrc_sel_proc(int inrate, int outrate,
+ int *pre_proc, int *post_proc)
+{
+ bool post_proc_cond2;
+ bool post_proc_cond0;
+
+ /* select pre_proc between [0, 2] */
+ if (inrate * 8 > 33 * outrate)
+ *pre_proc = 2;
+ else if (inrate * 8 > 15 * outrate) {
+ if (inrate > 152000)
+ *pre_proc = 2;
+ else
+ *pre_proc = 1;
+ } else if (inrate < 76000)
+ *pre_proc = 0;
+ else if (inrate > 152000)
+ *pre_proc = 2;
+ else
+ *pre_proc = 1;
+
+ /* Condition for selection of post-processing */
+ post_proc_cond2 = (inrate * 15 > outrate * 16 && outrate < 56000) ||
+ (inrate > 56000 && outrate < 56000);
+ post_proc_cond0 = inrate * 23 < outrate * 8;
+
+ if (post_proc_cond2)
+ *post_proc = 2;
+ else if (post_proc_cond0)
+ *post_proc = 0;
+ else
+ *post_proc = 1;
+}
+
+/**
* Request ASRC pair
*
* It assigns pair by the order of A->C->B because allocation of pair B,
@@ -239,6 +268,7 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair)
u32 inrate, outrate, indiv, outdiv;
u32 clk_index[2], div[2];
int in, out, channels;
+ int pre_proc, post_proc;
struct clk *clk;
bool ideal;
@@ -264,11 +294,11 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair)
ideal = config->inclk == INCLK_NONE;
/* Validate input and output sample rates */
- for (in = 0; in < ARRAY_SIZE(supported_input_rate); in++)
- if (inrate == supported_input_rate[in])
+ for (in = 0; in < ARRAY_SIZE(supported_asrc_rate); in++)
+ if (inrate == supported_asrc_rate[in])
break;
- if (in == ARRAY_SIZE(supported_input_rate)) {
+ if (in == ARRAY_SIZE(supported_asrc_rate)) {
pair_err("unsupported input sample rate: %dHz\n", inrate);
return -EINVAL;
}
@@ -282,7 +312,7 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair)
return -EINVAL;
}
- if ((outrate >= 8000 && outrate <= 30000) &&
+ if ((outrate >= 5512 && outrate <= 30000) &&
(outrate > 24 * inrate || inrate > 8 * outrate)) {
pair_err("exceed supported ratio range [1/24, 8] for \
inrate/outrate: %d/%d\n", inrate, outrate);
@@ -377,11 +407,13 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair)
ASRCTR_IDRi_MASK(index) | ASRCTR_USRi_MASK(index),
ASRCTR_IDR(index) | ASRCTR_USR(index));
+ fsl_asrc_sel_proc(inrate, outrate, &pre_proc, &post_proc);
+
/* Apply configurations for pre- and post-processing */
regmap_update_bits(asrc_priv->regmap, REG_ASRCFG,
ASRCFG_PREMODi_MASK(index) | ASRCFG_POSTMODi_MASK(index),
- ASRCFG_PREMOD(index, process_option[in][out][0]) |
- ASRCFG_POSTMOD(index, process_option[in][out][1]));
+ ASRCFG_PREMOD(index, pre_proc) |
+ ASRCFG_POSTMOD(index, post_proc));
return fsl_asrc_set_ideal_ratio(pair, inrate, outrate);
}
@@ -455,7 +487,9 @@ static int fsl_asrc_dai_startup(struct snd_pcm_substream *substream,
snd_pcm_hw_constraint_step(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_CHANNELS, 2);
- return 0;
+
+ return snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE, &fsl_asrc_rate_constraints);
}
static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream,
@@ -568,7 +602,6 @@ static int fsl_asrc_dai_probe(struct snd_soc_dai *dai)
return 0;
}
-#define FSL_ASRC_RATES SNDRV_PCM_RATE_8000_192000
#define FSL_ASRC_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | \
SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_S20_3LE)
@@ -579,14 +612,18 @@ static struct snd_soc_dai_driver fsl_asrc_dai = {
.stream_name = "ASRC-Playback",
.channels_min = 1,
.channels_max = 10,
- .rates = FSL_ASRC_RATES,
+ .rate_min = 5512,
+ .rate_max = 192000,
+ .rates = SNDRV_PCM_RATE_KNOT,
.formats = FSL_ASRC_FORMATS,
},
.capture = {
.stream_name = "ASRC-Capture",
.channels_min = 1,
.channels_max = 10,
- .rates = FSL_ASRC_RATES,
+ .rate_min = 5512,
+ .rate_max = 192000,
+ .rates = SNDRV_PCM_RATE_KNOT,
.formats = FSL_ASRC_FORMATS,
},
.ops = &fsl_asrc_dai_ops,
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index bad0dfed6b68..10d2210c91ef 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -9,6 +9,7 @@
#include <linux/module.h>
#include <linux/of_irq.h>
#include <linux/of_platform.h>
+#include <linux/pm_runtime.h>
#include <sound/dmaengine_pcm.h>
#include <sound/pcm_params.h>
@@ -466,30 +467,6 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
- int ret;
-
- /*
- * Some platforms might use the same bit to gate all three or two of
- * clocks, so keep all clocks open/close at the same time for safety
- */
- ret = clk_prepare_enable(esai_priv->coreclk);
- if (ret)
- return ret;
- if (!IS_ERR(esai_priv->spbaclk)) {
- ret = clk_prepare_enable(esai_priv->spbaclk);
- if (ret)
- goto err_spbaclk;
- }
- if (!IS_ERR(esai_priv->extalclk)) {
- ret = clk_prepare_enable(esai_priv->extalclk);
- if (ret)
- goto err_extalck;
- }
- if (!IS_ERR(esai_priv->fsysclk)) {
- ret = clk_prepare_enable(esai_priv->fsysclk);
- if (ret)
- goto err_fsysclk;
- }
if (!dai->active) {
/* Set synchronous mode */
@@ -506,16 +483,6 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream,
return 0;
-err_fsysclk:
- if (!IS_ERR(esai_priv->extalclk))
- clk_disable_unprepare(esai_priv->extalclk);
-err_extalck:
- if (!IS_ERR(esai_priv->spbaclk))
- clk_disable_unprepare(esai_priv->spbaclk);
-err_spbaclk:
- clk_disable_unprepare(esai_priv->coreclk);
-
- return ret;
}
static int fsl_esai_hw_params(struct snd_pcm_substream *substream,
@@ -576,20 +543,6 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream,
return 0;
}
-static void fsl_esai_shutdown(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
-
- if (!IS_ERR(esai_priv->fsysclk))
- clk_disable_unprepare(esai_priv->fsysclk);
- if (!IS_ERR(esai_priv->extalclk))
- clk_disable_unprepare(esai_priv->extalclk);
- if (!IS_ERR(esai_priv->spbaclk))
- clk_disable_unprepare(esai_priv->spbaclk);
- clk_disable_unprepare(esai_priv->coreclk);
-}
-
static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *dai)
{
@@ -658,7 +611,6 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd,
static const struct snd_soc_dai_ops fsl_esai_dai_ops = {
.startup = fsl_esai_startup,
- .shutdown = fsl_esai_shutdown,
.trigger = fsl_esai_trigger,
.hw_params = fsl_esai_hw_params,
.set_sysclk = fsl_esai_set_dai_sysclk,
@@ -947,6 +899,10 @@ static int fsl_esai_probe(struct platform_device *pdev)
return ret;
}
+ pm_runtime_enable(&pdev->dev);
+
+ regcache_cache_only(esai_priv->regmap, true);
+
ret = imx_pcm_dma_init(pdev, IMX_ESAI_DMABUF_SIZE);
if (ret)
dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret);
@@ -954,6 +910,13 @@ static int fsl_esai_probe(struct platform_device *pdev)
return ret;
}
+static int fsl_esai_remove(struct platform_device *pdev)
+{
+ pm_runtime_disable(&pdev->dev);
+
+ return 0;
+}
+
static const struct of_device_id fsl_esai_dt_ids[] = {
{ .compatible = "fsl,imx35-esai", },
{ .compatible = "fsl,vf610-esai", },
@@ -961,22 +924,35 @@ static const struct of_device_id fsl_esai_dt_ids[] = {
};
MODULE_DEVICE_TABLE(of, fsl_esai_dt_ids);
-#ifdef CONFIG_PM_SLEEP
-static int fsl_esai_suspend(struct device *dev)
-{
- struct fsl_esai *esai = dev_get_drvdata(dev);
-
- regcache_cache_only(esai->regmap, true);
- regcache_mark_dirty(esai->regmap);
-
- return 0;
-}
-
-static int fsl_esai_resume(struct device *dev)
+#ifdef CONFIG_PM
+static int fsl_esai_runtime_resume(struct device *dev)
{
struct fsl_esai *esai = dev_get_drvdata(dev);
int ret;
+ /*
+ * Some platforms might use the same bit to gate all three or two of
+ * clocks, so keep all clocks open/close at the same time for safety
+ */
+ ret = clk_prepare_enable(esai->coreclk);
+ if (ret)
+ return ret;
+ if (!IS_ERR(esai->spbaclk)) {
+ ret = clk_prepare_enable(esai->spbaclk);
+ if (ret)
+ goto err_spbaclk;
+ }
+ if (!IS_ERR(esai->extalclk)) {
+ ret = clk_prepare_enable(esai->extalclk);
+ if (ret)
+ goto err_extalclk;
+ }
+ if (!IS_ERR(esai->fsysclk)) {
+ ret = clk_prepare_enable(esai->fsysclk);
+ if (ret)
+ goto err_fsysclk;
+ }
+
regcache_cache_only(esai->regmap, false);
/* FIFO reset for safety */
@@ -987,22 +963,59 @@ static int fsl_esai_resume(struct device *dev)
ret = regcache_sync(esai->regmap);
if (ret)
- return ret;
+ goto err_regcache_sync;
/* FIFO reset done */
regmap_update_bits(esai->regmap, REG_ESAI_TFCR, ESAI_xFCR_xFR, 0);
regmap_update_bits(esai->regmap, REG_ESAI_RFCR, ESAI_xFCR_xFR, 0);
return 0;
+
+err_regcache_sync:
+ if (!IS_ERR(esai->fsysclk))
+ clk_disable_unprepare(esai->fsysclk);
+err_fsysclk:
+ if (!IS_ERR(esai->extalclk))
+ clk_disable_unprepare(esai->extalclk);
+err_extalclk:
+ if (!IS_ERR(esai->spbaclk))
+ clk_disable_unprepare(esai->spbaclk);
+err_spbaclk:
+ clk_disable_unprepare(esai->coreclk);
+
+ return ret;
+}
+
+static int fsl_esai_runtime_suspend(struct device *dev)
+{
+ struct fsl_esai *esai = dev_get_drvdata(dev);
+
+ regcache_cache_only(esai->regmap, true);
+ regcache_mark_dirty(esai->regmap);
+
+ if (!IS_ERR(esai->fsysclk))
+ clk_disable_unprepare(esai->fsysclk);
+ if (!IS_ERR(esai->extalclk))
+ clk_disable_unprepare(esai->extalclk);
+ if (!IS_ERR(esai->spbaclk))
+ clk_disable_unprepare(esai->spbaclk);
+ clk_disable_unprepare(esai->coreclk);
+
+ return 0;
}
-#endif /* CONFIG_PM_SLEEP */
+#endif /* CONFIG_PM */
static const struct dev_pm_ops fsl_esai_pm_ops = {
- SET_SYSTEM_SLEEP_PM_OPS(fsl_esai_suspend, fsl_esai_resume)
+ SET_RUNTIME_PM_OPS(fsl_esai_runtime_suspend,
+ fsl_esai_runtime_resume,
+ NULL)
+ SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend,
+ pm_runtime_force_resume)
};
static struct platform_driver fsl_esai_driver = {
.probe = fsl_esai_probe,
+ .remove = fsl_esai_remove,
.driver = {
.name = "fsl-esai-dai",
.pm = &fsl_esai_pm_ops,
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 8593269156bd..d58cc3ae90d8 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -596,15 +596,8 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream,
{
struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
- struct device *dev = &sai->pdev->dev;
int ret;
- ret = clk_prepare_enable(sai->bus_clk);
- if (ret) {
- dev_err(dev, "failed to enable bus clock: %d\n", ret);
- return ret;
- }
-
regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx), FSL_SAI_CR3_TRCE,
FSL_SAI_CR3_TRCE);
@@ -621,8 +614,6 @@ static void fsl_sai_shutdown(struct snd_pcm_substream *substream,
bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx), FSL_SAI_CR3_TRCE, 0);
-
- clk_disable_unprepare(sai->bus_clk);
}
static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = {
@@ -935,6 +926,14 @@ static int fsl_sai_runtime_suspend(struct device *dev)
{
struct fsl_sai *sai = dev_get_drvdata(dev);
+ if (sai->mclk_streams & BIT(SNDRV_PCM_STREAM_CAPTURE))
+ clk_disable_unprepare(sai->mclk_clk[sai->mclk_id[0]]);
+
+ if (sai->mclk_streams & BIT(SNDRV_PCM_STREAM_PLAYBACK))
+ clk_disable_unprepare(sai->mclk_clk[sai->mclk_id[1]]);
+
+ clk_disable_unprepare(sai->bus_clk);
+
regcache_cache_only(sai->regmap, true);
regcache_mark_dirty(sai->regmap);
@@ -944,6 +943,25 @@ static int fsl_sai_runtime_suspend(struct device *dev)
static int fsl_sai_runtime_resume(struct device *dev)
{
struct fsl_sai *sai = dev_get_drvdata(dev);
+ int ret;
+
+ ret = clk_prepare_enable(sai->bus_clk);
+ if (ret) {
+ dev_err(dev, "failed to enable bus clock: %d\n", ret);
+ return ret;
+ }
+
+ if (sai->mclk_streams & BIT(SNDRV_PCM_STREAM_PLAYBACK)) {
+ ret = clk_prepare_enable(sai->mclk_clk[sai->mclk_id[1]]);
+ if (ret)
+ goto disable_bus_clk;
+ }
+
+ if (sai->mclk_streams & BIT(SNDRV_PCM_STREAM_CAPTURE)) {
+ ret = clk_prepare_enable(sai->mclk_clk[sai->mclk_id[0]]);
+ if (ret)
+ goto disable_tx_clk;
+ }
regcache_cache_only(sai->regmap, false);
regmap_write(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_SR);
@@ -951,7 +969,23 @@ static int fsl_sai_runtime_resume(struct device *dev)
usleep_range(1000, 2000);
regmap_write(sai->regmap, FSL_SAI_TCSR, 0);
regmap_write(sai->regmap, FSL_SAI_RCSR, 0);
- return regcache_sync(sai->regmap);
+
+ ret = regcache_sync(sai->regmap);
+ if (ret)
+ goto disable_rx_clk;
+
+ return 0;
+
+disable_rx_clk:
+ if (sai->mclk_streams & BIT(SNDRV_PCM_STREAM_CAPTURE))
+ clk_disable_unprepare(sai->mclk_clk[sai->mclk_id[0]]);
+disable_tx_clk:
+ if (sai->mclk_streams & BIT(SNDRV_PCM_STREAM_PLAYBACK))
+ clk_disable_unprepare(sai->mclk_clk[sai->mclk_id[1]]);
+disable_bus_clk:
+ clk_disable_unprepare(sai->bus_clk);
+
+ return ret;
}
#endif /* CONFIG_PM */
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 09b2967befd9..fa862af25c1a 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1582,9 +1582,7 @@ static int fsl_ssi_probe(struct platform_device *pdev)
}
}
- ret = fsl_ssi_debugfs_create(&ssi->dbg_stats, dev);
- if (ret)
- goto error_asoc_register;
+ fsl_ssi_debugfs_create(&ssi->dbg_stats, dev);
/* Initially configures SSI registers */
fsl_ssi_hw_init(ssi);
diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h
index 0bdda608d414..db57cad80449 100644
--- a/sound/soc/fsl/fsl_ssi.h
+++ b/sound/soc/fsl/fsl_ssi.h
@@ -270,7 +270,6 @@ struct device;
struct fsl_ssi_dbg {
struct dentry *dbg_dir;
- struct dentry *dbg_stats;
struct {
unsigned int rfrc;
@@ -299,7 +298,7 @@ struct fsl_ssi_dbg {
void fsl_ssi_dbg_isr(struct fsl_ssi_dbg *ssi_dbg, u32 sisr);
-int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, struct device *dev);
+void fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, struct device *dev);
void fsl_ssi_debugfs_remove(struct fsl_ssi_dbg *ssi_dbg);
@@ -312,10 +311,9 @@ static inline void fsl_ssi_dbg_isr(struct fsl_ssi_dbg *stats, u32 sisr)
{
}
-static inline int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg,
- struct device *dev)
+static inline void fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg,
+ struct device *dev)
{
- return 0;
}
static inline void fsl_ssi_debugfs_remove(struct fsl_ssi_dbg *ssi_dbg)
diff --git a/sound/soc/fsl/fsl_ssi_dbg.c b/sound/soc/fsl/fsl_ssi_dbg.c
index 6f6294149476..2a20ee23dc52 100644
--- a/sound/soc/fsl/fsl_ssi_dbg.c
+++ b/sound/soc/fsl/fsl_ssi_dbg.c
@@ -126,25 +126,15 @@ static int fsl_ssi_stats_show(struct seq_file *s, void *unused)
DEFINE_SHOW_ATTRIBUTE(fsl_ssi_stats);
-int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, struct device *dev)
+void fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, struct device *dev)
{
ssi_dbg->dbg_dir = debugfs_create_dir(dev_name(dev), NULL);
- if (!ssi_dbg->dbg_dir)
- return -ENOMEM;
- ssi_dbg->dbg_stats = debugfs_create_file("stats", 0444,
- ssi_dbg->dbg_dir, ssi_dbg,
- &fsl_ssi_stats_fops);
- if (!ssi_dbg->dbg_stats) {
- debugfs_remove(ssi_dbg->dbg_dir);
- return -ENOMEM;
- }
-
- return 0;
+ debugfs_create_file("stats", 0444, ssi_dbg->dbg_dir, ssi_dbg,
+ &fsl_ssi_stats_fops);
}
void fsl_ssi_debugfs_remove(struct fsl_ssi_dbg *ssi_dbg)
{
- debugfs_remove(ssi_dbg->dbg_stats);
- debugfs_remove(ssi_dbg->dbg_dir);
+ debugfs_remove_recursive(ssi_dbg->dbg_dir);
}
diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c
index 040d06b89f00..9bab202569af 100644
--- a/sound/soc/fsl/fsl_utils.c
+++ b/sound/soc/fsl/fsl_utils.c
@@ -57,7 +57,7 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np,
of_node_put(dma_channel_np);
return ret;
}
- snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%pOFn",
+ snprintf((char *)dai->platforms->name, DAI_NAME_SIZE, "%llx.%pOFn",
(unsigned long long) res.start, dma_channel_np);
iprop = of_get_property(dma_channel_np, "cell-index", NULL);
diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c
index 9aaf3e5b45b9..9e1cb18859ce 100644
--- a/sound/soc/fsl/imx-audmix.c
+++ b/sound/soc/fsl/imx-audmix.c
@@ -205,6 +205,15 @@ static int imx_audmix_probe(struct platform_device *pdev)
return -ENOMEM;
for (i = 0; i < num_dai; i++) {
+ struct snd_soc_dai_link_component *dlc;
+
+ /* for CPU/Codec/Platform x 2 */
+ dlc = devm_kzalloc(&pdev->dev, 6 * sizeof(*dlc), GFP_KERNEL);
+ if (!dlc) {
+ dev_err(&pdev->dev, "failed to allocate dai_link\n");
+ return -ENOMEM;
+ }
+
ret = of_parse_phandle_with_args(audmix_np, "dais", NULL, i,
&args);
if (ret < 0) {
@@ -231,13 +240,21 @@ static int imx_audmix_probe(struct platform_device *pdev)
dai_name, "CPU-Capture");
}
+ priv->dai[i].cpus = &dlc[0];
+ priv->dai[i].codecs = &dlc[1];
+ priv->dai[i].platforms = &dlc[2];
+
+ priv->dai[i].num_cpus = 1;
+ priv->dai[i].num_codecs = 1;
+ priv->dai[i].num_platforms = 1;
+
priv->dai[i].name = dai_name;
priv->dai[i].stream_name = "HiFi-AUDMIX-FE";
- priv->dai[i].codec_dai_name = "snd-soc-dummy-dai";
- priv->dai[i].codec_name = "snd-soc-dummy";
- priv->dai[i].cpu_of_node = args.np;
- priv->dai[i].cpu_dai_name = dev_name(&cpu_pdev->dev);
- priv->dai[i].platform_of_node = args.np;
+ priv->dai[i].codecs->dai_name = "snd-soc-dummy-dai";
+ priv->dai[i].codecs->name = "snd-soc-dummy";
+ priv->dai[i].cpus->of_node = args.np;
+ priv->dai[i].cpus->dai_name = dev_name(&cpu_pdev->dev);
+ priv->dai[i].platforms->of_node = args.np;
priv->dai[i].dynamic = 1;
priv->dai[i].dpcm_playback = 1;
priv->dai[i].dpcm_capture = (i == 0 ? 1 : 0);
@@ -252,12 +269,20 @@ static int imx_audmix_probe(struct platform_device *pdev)
be_cp = devm_kasprintf(&pdev->dev, GFP_KERNEL,
"AUDMIX-Capture-%d", i);
+ priv->dai[num_dai + i].cpus = &dlc[3];
+ priv->dai[num_dai + i].codecs = &dlc[4];
+ priv->dai[num_dai + i].platforms = &dlc[5];
+
+ priv->dai[num_dai + i].num_cpus = 1;
+ priv->dai[num_dai + i].num_codecs = 1;
+ priv->dai[num_dai + i].num_platforms = 1;
+
priv->dai[num_dai + i].name = be_name;
- priv->dai[num_dai + i].codec_dai_name = "snd-soc-dummy-dai";
- priv->dai[num_dai + i].codec_name = "snd-soc-dummy";
- priv->dai[num_dai + i].cpu_of_node = audmix_np;
- priv->dai[num_dai + i].cpu_dai_name = be_name;
- priv->dai[num_dai + i].platform_name = "snd-soc-dummy";
+ priv->dai[num_dai + i].codecs->dai_name = "snd-soc-dummy-dai";
+ priv->dai[num_dai + i].codecs->name = "snd-soc-dummy";
+ priv->dai[num_dai + i].cpus->of_node = audmix_np;
+ priv->dai[num_dai + i].cpus->dai_name = be_name;
+ priv->dai[num_dai + i].platforms->name = "snd-soc-dummy";
priv->dai[num_dai + i].no_pcm = 1;
priv->dai[num_dai + i].dpcm_playback = 1;
priv->dai[num_dai + i].dpcm_capture = 1;
diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index 04e59e66711d..b2351cd33b0f 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -141,17 +141,11 @@ static void audmux_debugfs_init(void)
char buf[20];
audmux_debugfs_root = debugfs_create_dir("audmux", NULL);
- if (!audmux_debugfs_root) {
- pr_warning("Failed to create AUDMUX debugfs root\n");
- return;
- }
for (i = 0; i < MX31_AUDMUX_PORT7_SSI_PINS_7 + 1; i++) {
snprintf(buf, sizeof(buf), "ssi%lu", i);
- if (!debugfs_create_file(buf, 0444, audmux_debugfs_root,
- (void *)i, &audmux_debugfs_fops))
- pr_warning("Failed to create AUDMUX port %lu debugfs file\n",
- i);
+ debugfs_create_file(buf, 0444, audmux_debugfs_root,
+ (void *)i, &audmux_debugfs_fops);
}
}
diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c
index c9d8739b04a9..15a27a2cd0ca 100644
--- a/sound/soc/fsl/imx-es8328.c
+++ b/sound/soc/fsl/imx-es8328.c
@@ -74,6 +74,7 @@ static int imx_es8328_probe(struct platform_device *pdev)
struct device_node *ssi_np = NULL, *codec_np = NULL;
struct platform_device *ssi_pdev;
struct imx_es8328_data *data;
+ struct snd_soc_dai_link_component *comp;
u32 int_port, ext_port;
int ret;
struct device *dev = &pdev->dev;
@@ -147,16 +148,30 @@ static int imx_es8328_probe(struct platform_device *pdev)
goto fail;
}
+ comp = devm_kzalloc(dev, 3 * sizeof(*comp), GFP_KERNEL);
+ if (!comp) {
+ ret = -ENOMEM;
+ goto fail;
+ }
+
data->dev = dev;
data->jack_gpio = of_get_named_gpio(pdev->dev.of_node, "jack-gpio", 0);
+ data->dai.cpus = &comp[0];
+ data->dai.codecs = &comp[1];
+ data->dai.platforms = &comp[2];
+
+ data->dai.num_cpus = 1;
+ data->dai.num_codecs = 1;
+ data->dai.num_platforms = 1;
+
data->dai.name = "hifi";
data->dai.stream_name = "hifi";
- data->dai.codec_dai_name = "es8328-hifi-analog";
- data->dai.codec_of_node = codec_np;
- data->dai.cpu_of_node = ssi_np;
- data->dai.platform_of_node = ssi_np;
+ data->dai.codecs->dai_name = "es8328-hifi-analog";
+ data->dai.codecs->of_node = codec_np;
+ data->dai.cpus->of_node = ssi_np;
+ data->dai.platforms->of_node = ssi_np;
data->dai.init = &imx_es8328_dai_init;
data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM;
diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c
index 545815a27074..2b679680c93f 100644
--- a/sound/soc/fsl/imx-mc13783.c
+++ b/sound/soc/fsl/imx-mc13783.c
@@ -46,17 +46,19 @@ static const struct snd_soc_ops imx_mc13783_hifi_ops = {
.hw_params = imx_mc13783_hifi_hw_params,
};
+SND_SOC_DAILINK_DEFS(hifi,
+ DAILINK_COMP_ARRAY(COMP_CPU("imx-ssi.0")),
+ DAILINK_COMP_ARRAY(COMP_CODEC("mc13783-codec", "mc13783-hifi")),
+ DAILINK_COMP_ARRAY(COMP_PLATFORM("imx-ssi.0")));
+
static struct snd_soc_dai_link imx_mc13783_dai_mc13783[] = {
{
.name = "MC13783",
.stream_name = "Sound",
- .codec_dai_name = "mc13783-hifi",
- .codec_name = "mc13783-codec",
- .cpu_dai_name = "imx-ssi.0",
- .platform_name = "imx-ssi.0",
.ops = &imx_mc13783_hifi_ops,
.symmetric_rates = 1,
.dai_fmt = FMT_SSI,
+ SND_SOC_DAILINK_REG(hifi),
},
};
diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c
index 314814ddd2b0..04a9bc749016 100644
--- a/sound/soc/fsl/imx-pcm-dma.c
+++ b/sound/soc/fsl/imx-pcm-dma.c
@@ -1,3 +1,4 @@
+// SPDX-License-Identifier: GPL-2.0-or-later
/*
* imx-pcm-dma-mx2.c -- ALSA Soc Audio Layer
*
@@ -5,11 +6,6 @@
*
* This code is based on code copyrighted by Freescale,
* Liam Girdwood, Javier Martin and probably others.
- *
- * This program is free software; you can redistribute it and/or modify it
- * under the terms of the GNU General Public License as published by the
- * Free Software Foundation; either version 2 of the License, or (at your
- * option) any later version.
*/
#include <linux/platform_device.h>
#include <linux/dmaengine.h>
diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
index bf8597f57dce..15e8b9343c35 100644
--- a/sound/soc/fsl/imx-sgtl5000.c
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -55,6 +55,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev)
struct platform_device *ssi_pdev;
struct i2c_client *codec_dev;
struct imx_sgtl5000_data *data = NULL;
+ struct snd_soc_dai_link_component *comp;
int int_port, ext_port;
int ret;
@@ -122,6 +123,12 @@ static int imx_sgtl5000_probe(struct platform_device *pdev)
goto fail;
}
+ comp = devm_kzalloc(&pdev->dev, 3 * sizeof(*comp), GFP_KERNEL);
+ if (!comp) {
+ ret = -ENOMEM;
+ goto fail;
+ }
+
data->codec_clk = clk_get(&codec_dev->dev, NULL);
if (IS_ERR(data->codec_clk)) {
ret = PTR_ERR(data->codec_clk);
@@ -130,12 +137,20 @@ static int imx_sgtl5000_probe(struct platform_device *pdev)
data->clk_frequency = clk_get_rate(data->codec_clk);
+ data->dai.cpus = &comp[0];
+ data->dai.codecs = &comp[1];
+ data->dai.platforms = &comp[2];
+
+ data->dai.num_cpus = 1;
+ data->dai.num_codecs = 1;
+ data->dai.num_platforms = 1;
+
data->dai.name = "HiFi";
data->dai.stream_name = "HiFi";
- data->dai.codec_dai_name = "sgtl5000";
- data->dai.codec_of_node = codec_np;
- data->dai.cpu_of_node = ssi_np;
- data->dai.platform_of_node = ssi_np;
+ data->dai.codecs->dai_name = "sgtl5000";
+ data->dai.codecs->of_node = codec_np;
+ data->dai.cpus->of_node = ssi_np;
+ data->dai.platforms->of_node = ssi_np;
data->dai.init = &imx_sgtl5000_dai_init;
data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM;
diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c
index 4f7f210beb18..6c4dadf60355 100644
--- a/sound/soc/fsl/imx-spdif.c
+++ b/sound/soc/fsl/imx-spdif.c
@@ -15,6 +15,7 @@ static int imx_spdif_audio_probe(struct platform_device *pdev)
{
struct device_node *spdif_np, *np = pdev->dev.of_node;
struct imx_spdif_data *data;
+ struct snd_soc_dai_link_component *comp;
int ret = 0;
spdif_np = of_parse_phandle(np, "spdif-controller", 0);
@@ -25,17 +26,26 @@ static int imx_spdif_audio_probe(struct platform_device *pdev)
}
data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);
- if (!data) {
+ comp = devm_kzalloc(&pdev->dev, 3 * sizeof(*comp), GFP_KERNEL);
+ if (!data || !comp) {
ret = -ENOMEM;
goto end;
}
+ data->dai.cpus = &comp[0];
+ data->dai.codecs = &comp[1];
+ data->dai.platforms = &comp[2];
+
+ data->dai.num_cpus = 1;
+ data->dai.num_codecs = 1;
+ data->dai.num_platforms = 1;
+
data->dai.name = "S/PDIF PCM";
data->dai.stream_name = "S/PDIF PCM";
- data->dai.codec_dai_name = "snd-soc-dummy-dai";
- data->dai.codec_name = "snd-soc-dummy";
- data->dai.cpu_of_node = spdif_np;
- data->dai.platform_of_node = spdif_np;
+ data->dai.codecs->dai_name = "snd-soc-dummy-dai";
+ data->dai.codecs->name = "snd-soc-dummy";
+ data->dai.cpus->of_node = spdif_np;
+ data->dai.platforms->of_node = spdif_np;
data->dai.playback_only = true;
data->dai.capture_only = true;
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index f6261a3eeb0f..23617eb09ba1 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -189,6 +189,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
struct device_node *np = ssi_pdev->dev.of_node;
struct device_node *codec_np = NULL;
struct mpc8610_hpcd_data *machine_data;
+ struct snd_soc_dai_link_component *comp;
int ret = -ENODEV;
const char *sprop;
const u32 *iprop;
@@ -206,14 +207,36 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
goto error_alloc;
}
- machine_data->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev);
+ comp = devm_kzalloc(&pdev->dev, 6 * sizeof(*comp), GFP_KERNEL);
+ if (!comp) {
+ ret = -ENOMEM;
+ goto error_alloc;
+ }
+
+ machine_data->dai[0].cpus = &comp[0];
+ machine_data->dai[0].codecs = &comp[1];
+ machine_data->dai[0].platforms = &comp[2];
+
+ machine_data->dai[0].num_cpus = 1;
+ machine_data->dai[0].num_codecs = 1;
+ machine_data->dai[0].num_platforms = 1;
+
+ machine_data->dai[1].cpus = &comp[3];
+ machine_data->dai[1].codecs = &comp[4];
+ machine_data->dai[1].platforms = &comp[5];
+
+ machine_data->dai[1].num_cpus = 1;
+ machine_data->dai[1].num_codecs = 1;
+ machine_data->dai[1].num_platforms = 1;
+
+ machine_data->dai[0].cpus->dai_name = dev_name(&ssi_pdev->dev);
machine_data->dai[0].ops = &mpc8610_hpcd_ops;
/* ASoC core can match codec with device node */
- machine_data->dai[0].codec_of_node = codec_np;
+ machine_data->dai[0].codecs->of_node = codec_np;
/* The DAI name from the codec (snd_soc_dai_driver.name) */
- machine_data->dai[0].codec_dai_name = "cs4270-hifi";
+ machine_data->dai[0].codecs->dai_name = "cs4270-hifi";
/* We register two DAIs per SSI, one for playback and the other for
* capture. Currently, we only support codecs that have one DAI for
@@ -306,7 +329,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
}
/* Find the playback DMA channel to use. */
- machine_data->dai[0].platform_name = machine_data->platform_name[0];
+ machine_data->dai[0].platforms->name = machine_data->platform_name[0];
ret = fsl_asoc_get_dma_channel(np, "fsl,playback-dma",
&machine_data->dai[0],
&machine_data->dma_channel_id[0],
@@ -317,7 +340,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
}
/* Find the capture DMA channel to use. */
- machine_data->dai[1].platform_name = machine_data->platform_name[1];
+ machine_data->dai[1].platforms->name = machine_data->platform_name[1];
ret = fsl_asoc_get_dma_channel(np, "fsl,capture-dma",
&machine_data->dai[1],
&machine_data->dma_channel_id[1],
diff --git a/sound/soc/fsl/mx27vis-aic32x4.c b/sound/soc/fsl/mx27vis-aic32x4.c
index 37a4520aef62..38ac4a397742 100644
--- a/sound/soc/fsl/mx27vis-aic32x4.c
+++ b/sound/soc/fsl/mx27vis-aic32x4.c
@@ -132,16 +132,19 @@ static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = {
{"IN3_L", NULL, "Mic Bias"},
};
+SND_SOC_DAILINK_DEFS(hifi,
+ DAILINK_COMP_ARRAY(COMP_CPU("imx-ssi.0")),
+ DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic32x4.0-0018",
+ "tlv320aic32x4-hifi")),
+ DAILINK_COMP_ARRAY(COMP_PLATFORM("imx-ssi.0")));
+
static struct snd_soc_dai_link mx27vis_aic32x4_dai = {
.name = "tlv320aic32x4",
.stream_name = "TLV320AIC32X4",
- .codec_dai_name = "tlv320aic32x4-hifi",
- .platform_name = "imx-ssi.0",
- .codec_name = "tlv320aic32x4.0-0018",
- .cpu_dai_name = "imx-ssi.0",
.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM,
.ops = &mx27vis_aic32x4_snd_ops,
+ SND_SOC_DAILINK_REG(hifi),
};
static struct snd_soc_card mx27vis_aic32x4 = {
diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c
index 80384f70878d..6114b01b90f7 100644
--- a/sound/soc/fsl/p1022_ds.c
+++ b/sound/soc/fsl/p1022_ds.c
@@ -199,6 +199,7 @@ static int p1022_ds_probe(struct platform_device *pdev)
struct device_node *np = ssi_pdev->dev.of_node;
struct device_node *codec_np = NULL;
struct machine_data *mdata;
+ struct snd_soc_dai_link_component *comp;
int ret = -ENODEV;
const char *sprop;
const u32 *iprop;
@@ -216,11 +217,34 @@ static int p1022_ds_probe(struct platform_device *pdev)
goto error_put;
}
- mdata->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev);
+ comp = devm_kzalloc(&pdev->dev, 6 * sizeof(*comp), GFP_KERNEL);
+ if (!comp) {
+ ret = -ENOMEM;
+ goto error_put;
+ }
+
+ mdata->dai[0].cpus = &comp[0];
+ mdata->dai[0].codecs = &comp[1];
+ mdata->dai[0].platforms = &comp[2];
+
+ mdata->dai[0].num_cpus = 1;
+ mdata->dai[0].num_codecs = 1;
+ mdata->dai[0].num_platforms = 1;
+
+ mdata->dai[1].cpus = &comp[3];
+ mdata->dai[1].codecs = &comp[4];
+ mdata->dai[1].platforms = &comp[5];
+
+ mdata->dai[1].num_cpus = 1;
+ mdata->dai[1].num_codecs = 1;
+ mdata->dai[1].num_platforms = 1;
+
+
+ mdata->dai[0].cpus->dai_name = dev_name(&ssi_pdev->dev);
mdata->dai[0].ops = &p1022_ds_ops;
/* ASoC core can match codec with device node */
- mdata->dai[0].codec_of_node = codec_np;
+ mdata->dai[0].codecs->of_node = codec_np;
/* We register two DAIs per SSI, one for playback and the other for
* capture. We support codecs that have separate DAIs for both playback
@@ -229,8 +253,8 @@ static int p1022_ds_probe(struct platform_device *pdev)
memcpy(&mdata->dai[1], &mdata->dai[0], sizeof(struct snd_soc_dai_link));
/* The DAI names from the codec (snd_soc_dai_driver.name) */
- mdata->dai[0].codec_dai_name = "wm8776-hifi-playback";
- mdata->dai[1].codec_dai_name = "wm8776-hifi-capture";
+ mdata->dai[0].codecs->dai_name = "wm8776-hifi-playback";
+ mdata->dai[1].codecs->dai_name = "wm8776-hifi-capture";
/* Get the device ID */
iprop = of_get_property(np, "cell-index", NULL);
@@ -316,7 +340,7 @@ static int p1022_ds_probe(struct platform_device *pdev)
}
/* Find the playback DMA channel to use. */
- mdata->dai[0].platform_name = mdata->platform_name[0];
+ mdata->dai[0].platforms->name = mdata->platform_name[0];
ret = fsl_asoc_get_dma_channel(np, "fsl,playback-dma", &mdata->dai[0],
&mdata->dma_channel_id[0],
&mdata->dma_id[0]);
@@ -326,7 +350,7 @@ static int p1022_ds_probe(struct platform_device *pdev)
}
/* Find the capture DMA channel to use. */
- mdata->dai[1].platform_name = mdata->platform_name[1];
+ mdata->dai[1].platforms->name = mdata->platform_name[1];
ret = fsl_asoc_get_dma_channel(np, "fsl,capture-dma", &mdata->dai[1],
&mdata->dma_channel_id[1],
&mdata->dma_id[1]);
diff --git a/sound/soc/fsl/p1022_rdk.c b/sound/soc/fsl/p1022_rdk.c
index 1c32c2d8c6b0..72687235c0ae 100644
--- a/sound/soc/fsl/p1022_rdk.c
+++ b/sound/soc/fsl/p1022_rdk.c
@@ -203,6 +203,7 @@ static int p1022_rdk_probe(struct platform_device *pdev)
struct device_node *np = ssi_pdev->dev.of_node;
struct device_node *codec_np = NULL;
struct machine_data *mdata;
+ struct snd_soc_dai_link_component *comp;
const u32 *iprop;
int ret;
@@ -219,11 +220,33 @@ static int p1022_rdk_probe(struct platform_device *pdev)
goto error_put;
}
- mdata->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev);
+ comp = devm_kzalloc(&pdev->dev, 6 * sizeof(*comp), GFP_KERNEL);
+ if (!comp) {
+ ret = -ENOMEM;
+ goto error_put;
+ }
+
+ mdata->dai[0].cpus = &comp[0];
+ mdata->dai[0].codecs = &comp[1];
+ mdata->dai[0].platforms = &comp[2];
+
+ mdata->dai[0].num_cpus = 1;
+ mdata->dai[0].num_codecs = 1;
+ mdata->dai[0].num_platforms = 1;
+
+ mdata->dai[1].cpus = &comp[3];
+ mdata->dai[1].codecs = &comp[4];
+ mdata->dai[1].platforms = &comp[5];
+
+ mdata->dai[1].num_cpus = 1;
+ mdata->dai[1].num_codecs = 1;
+ mdata->dai[1].num_platforms = 1;
+
+ mdata->dai[0].cpus->dai_name = dev_name(&ssi_pdev->dev);
mdata->dai[0].ops = &p1022_rdk_ops;
/* ASoC core can match codec with device node */
- mdata->dai[0].codec_of_node = codec_np;
+ mdata->dai[0].codecs->of_node = codec_np;
/*
* We register two DAIs per SSI, one for playback and the other for
@@ -233,8 +256,8 @@ static int p1022_rdk_probe(struct platform_device *pdev)
memcpy(&mdata->dai[1], &mdata->dai[0], sizeof(struct snd_soc_dai_link));
/* The DAI names from the codec (snd_soc_dai_driver.name) */
- mdata->dai[0].codec_dai_name = "wm8960-hifi";
- mdata->dai[1].codec_dai_name = mdata->dai[0].codec_dai_name;
+ mdata->dai[0].codecs->dai_name = "wm8960-hifi";
+ mdata->dai[1].codecs->dai_name = mdata->dai[0].codecs->dai_name;
/*
* Configure the SSI for I2S slave mode. Older device trees have
@@ -266,7 +289,7 @@ static int p1022_rdk_probe(struct platform_device *pdev)
}
/* Find the playback DMA channel to use. */
- mdata->dai[0].platform_name = mdata->platform_name[0];
+ mdata->dai[0].platforms->name = mdata->platform_name[0];
ret = fsl_asoc_get_dma_channel(np, "fsl,playback-dma", &mdata->dai[0],
&mdata->dma_channel_id[0],
&mdata->dma_id[0]);
@@ -277,7 +300,7 @@ static int p1022_rdk_probe(struct platform_device *pdev)
}
/* Find the capture DMA channel to use. */
- mdata->dai[1].platform_name = mdata->platform_name[1];
+ mdata->dai[1].platforms->name = mdata->platform_name[1];
ret = fsl_asoc_get_dma_channel(np, "fsl,capture-dma", &mdata->dai[1],
&mdata->dma_channel_id[1],
&mdata->dma_id[1]);
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
index a7fe4ad25c52..af3c3b90c0ac 100644
--- a/sound/soc/fsl/pcm030-audio-fabric.c
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -23,20 +23,26 @@ struct pcm030_audio_data {
struct platform_device *codec_device;
};
+SND_SOC_DAILINK_DEFS(analog,
+ DAILINK_COMP_ARRAY(COMP_CPU("mpc5200-psc-ac97.0")),
+ DAILINK_COMP_ARRAY(COMP_CODEC("wm9712-codec", "wm9712-hifi")),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+SND_SOC_DAILINK_DEFS(iec958,
+ DAILINK_COMP_ARRAY(COMP_CPU("mpc5200-psc-ac97.1")),
+ DAILINK_COMP_ARRAY(COMP_CODEC("wm9712-codec", "wm9712-aux")),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
static struct snd_soc_dai_link pcm030_fabric_dai[] = {
{
.name = "AC97.0",
.stream_name = "AC97 Analog",
- .codec_dai_name = "wm9712-hifi",
- .cpu_dai_name = "mpc5200-psc-ac97.0",
- .codec_name = "wm9712-codec",
+ SND_SOC_DAILINK_REG(analog),
},
{
.name = "AC97.1",
.stream_name = "AC97 IEC958",
- .codec_dai_name = "wm9712-aux",
- .cpu_dai_name = "mpc5200-psc-ac97.1",
- .codec_name = "wm9712-codec",
+ SND_SOC_DAILINK_REG(iec958),
},
};
@@ -76,7 +82,7 @@ static int pcm030_fabric_probe(struct platform_device *op)
}
for_each_card_prelinks(card, i, dai_link)
- dai_link->platform_of_node = platform_np;
+ dai_link->platforms->of_node = platform_np;
ret = request_module("snd-soc-wm9712");
if (ret)
diff --git a/sound/soc/fsl/phycore-ac97.c b/sound/soc/fsl/phycore-ac97.c
index fe7ba6db7c96..e561f7ff1699 100644
--- a/sound/soc/fsl/phycore-ac97.c
+++ b/sound/soc/fsl/phycore-ac97.c
@@ -20,15 +20,17 @@ static struct snd_soc_card imx_phycore;
static const struct snd_soc_ops imx_phycore_hifi_ops = {
};
+SND_SOC_DAILINK_DEFS(hifi,
+ DAILINK_COMP_ARRAY(COMP_CPU("imx-ssi.0")),
+ DAILINK_COMP_ARRAY(COMP_CODEC("wm9712-codec", "wm9712-hifi")),
+ DAILINK_COMP_ARRAY(COMP_PLATFORM("imx-ssi.0")));
+
static struct snd_soc_dai_link imx_phycore_dai_ac97[] = {
{
.name = "HiFi",
.stream_name = "HiFi",
- .codec_dai_name = "wm9712-hifi",
- .codec_name = "wm9712-codec",
- .cpu_dai_name = "imx-ssi.0",
- .platform_name = "imx-ssi.0",
.ops = &imx_phycore_hifi_ops,
+ SND_SOC_DAILINK_REG(hifi),
},
};
diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c
index aad24ccbef90..52d321bede9c 100644
--- a/sound/soc/fsl/wm1133-ev1.c
+++ b/sound/soc/fsl/wm1133-ev1.c
@@ -216,18 +216,20 @@ static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd)
}
+SND_SOC_DAILINK_DEFS(ev1,
+ DAILINK_COMP_ARRAY(COMP_CPU("imx-ssi.0")),
+ DAILINK_COMP_ARRAY(COMP_CODEC("wm8350-codec.0-0x1a", "wm8350-hifi")),
+ DAILINK_COMP_ARRAY(COMP_PLATFORM("imx-ssi.0")));
+
static struct snd_soc_dai_link wm1133_ev1_dai = {
.name = "WM1133-EV1",
.stream_name = "Audio",
- .cpu_dai_name = "imx-ssi.0",
- .codec_dai_name = "wm8350-hifi",
- .platform_name = "imx-ssi.0",
- .codec_name = "wm8350-codec.0-0x1a",
.init = wm1133_ev1_init,
.ops = &wm1133_ev1_ops,
.symmetric_rates = 1,
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM,
+ SND_SOC_DAILINK_REG(ev1),
};
static struct snd_soc_card wm1133_ev1 = {
generated by cgit v1.2.3 (git 2.0.4) at 2024-11-19 18:03:37 +0000