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-rw-r--r--sound/soc/fsl/eukrea-tlv320.c5
-rw-r--r--sound/soc/fsl/fsl-asoc-card.c19
-rw-r--r--sound/soc/fsl/fsl_asrc.c28
-rw-r--r--sound/soc/fsl/fsl_esai.c14
-rw-r--r--sound/soc/fsl/fsl_ssi.c17
-rw-r--r--sound/soc/fsl/imx-sgtl5000.c6
-rw-r--r--sound/soc/fsl/imx-spdif.c3
-rw-r--r--sound/soc/fsl/imx-ssi.c2
-rw-r--r--sound/soc/fsl/imx-wm8962.c6
-rw-r--r--sound/soc/fsl/mpc5200_dma.c3
-rw-r--r--sound/soc/fsl/mpc5200_psc_ac97.c6
11 files changed, 67 insertions, 42 deletions
diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c
index eb093d5b85c..b175b0145a4 100644
--- a/sound/soc/fsl/eukrea-tlv320.c
+++ b/sound/soc/fsl/eukrea-tlv320.c
@@ -105,7 +105,7 @@ static int eukrea_tlv320_probe(struct platform_device *pdev)
int ret;
int int_port = 0, ext_port;
struct device_node *np = pdev->dev.of_node;
- struct device_node *ssi_np, *codec_np;
+ struct device_node *ssi_np = NULL, *codec_np = NULL;
eukrea_tlv320.dev = &pdev->dev;
if (np) {
@@ -217,8 +217,7 @@ static int eukrea_tlv320_probe(struct platform_device *pdev)
err:
if (ret)
dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
- if (np)
- of_node_put(ssi_np);
+ of_node_put(ssi_np);
return ret;
}
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 007c772f3ce..3f6959c8e2f 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -51,6 +51,7 @@ struct codec_priv {
* @sysclk_freq[2]: SYSCLK rates for set_sysclk()
* @sysclk_dir[2]: SYSCLK directions for set_sysclk()
* @sysclk_id[2]: SYSCLK ids for set_sysclk()
+ * @slot_width: Slot width of each frame
*
* Note: [1] for tx and [0] for rx
*/
@@ -58,6 +59,7 @@ struct cpu_priv {
unsigned long sysclk_freq[2];
u32 sysclk_dir[2];
u32 sysclk_id[2];
+ u32 slot_width;
};
/**
@@ -125,7 +127,12 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
priv->sample_rate = params_rate(params);
priv->sample_format = params_format(params);
- if (priv->card.set_bias_level)
+ /*
+ * 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)
return 0;
/* Specific configurations of DAIs starts from here */
@@ -137,6 +144,15 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
return ret;
}
+ if (cpu_priv->slot_width) {
+ ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2,
+ cpu_priv->slot_width);
+ if (ret) {
+ dev_err(dev, "failed to set TDM slot for cpu dai\n");
+ return ret;
+ }
+ }
+
return 0;
}
@@ -448,6 +464,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq;
priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
+ priv->cpu_priv.slot_width = 32;
priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
} else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) {
priv->codec_priv.mclk_id = SGTL5000_SYSCLK;
diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c
index 3b145313f93..9deabdd2b1a 100644
--- a/sound/soc/fsl/fsl_asrc.c
+++ b/sound/soc/fsl/fsl_asrc.c
@@ -684,12 +684,38 @@ static bool fsl_asrc_writeable_reg(struct device *dev, unsigned int reg)
}
}
+static struct reg_default fsl_asrc_reg[] = {
+ { REG_ASRCTR, 0x0000 }, { REG_ASRIER, 0x0000 },
+ { REG_ASRCNCR, 0x0000 }, { REG_ASRCFG, 0x0000 },
+ { REG_ASRCSR, 0x0000 }, { REG_ASRCDR1, 0x0000 },
+ { REG_ASRCDR2, 0x0000 }, { REG_ASRSTR, 0x0000 },
+ { REG_ASRRA, 0x0000 }, { REG_ASRRB, 0x0000 },
+ { REG_ASRRC, 0x0000 }, { REG_ASRPM1, 0x0000 },
+ { REG_ASRPM2, 0x0000 }, { REG_ASRPM3, 0x0000 },
+ { REG_ASRPM4, 0x0000 }, { REG_ASRPM5, 0x0000 },
+ { REG_ASRTFR1, 0x0000 }, { REG_ASRCCR, 0x0000 },
+ { REG_ASRDIA, 0x0000 }, { REG_ASRDOA, 0x0000 },
+ { REG_ASRDIB, 0x0000 }, { REG_ASRDOB, 0x0000 },
+ { REG_ASRDIC, 0x0000 }, { REG_ASRDOC, 0x0000 },
+ { REG_ASRIDRHA, 0x0000 }, { REG_ASRIDRLA, 0x0000 },
+ { REG_ASRIDRHB, 0x0000 }, { REG_ASRIDRLB, 0x0000 },
+ { REG_ASRIDRHC, 0x0000 }, { REG_ASRIDRLC, 0x0000 },
+ { REG_ASR76K, 0x0A47 }, { REG_ASR56K, 0x0DF3 },
+ { REG_ASRMCRA, 0x0000 }, { REG_ASRFSTA, 0x0000 },
+ { REG_ASRMCRB, 0x0000 }, { REG_ASRFSTB, 0x0000 },
+ { REG_ASRMCRC, 0x0000 }, { REG_ASRFSTC, 0x0000 },
+ { REG_ASRMCR1A, 0x0000 }, { REG_ASRMCR1B, 0x0000 },
+ { REG_ASRMCR1C, 0x0000 },
+};
+
static const struct regmap_config fsl_asrc_regmap_config = {
.reg_bits = 32,
.reg_stride = 4,
.val_bits = 32,
.max_register = REG_ASRMCR1C,
+ .reg_defaults = fsl_asrc_reg,
+ .num_reg_defaults = ARRAY_SIZE(fsl_asrc_reg),
.readable_reg = fsl_asrc_readable_reg,
.volatile_reg = fsl_asrc_volatile_reg,
.writeable_reg = fsl_asrc_writeable_reg,
@@ -792,7 +818,7 @@ static int fsl_asrc_probe(struct platform_device *pdev)
return -ENOMEM;
asrc_priv->pdev = pdev;
- strcpy(asrc_priv->name, np->name);
+ strncpy(asrc_priv->name, np->name, sizeof(asrc_priv->name) - 1);
/* Get the addresses and IRQ */
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index 8bcdfda09d7..ca319d59f84 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -513,10 +513,15 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream,
u32 width = snd_pcm_format_width(params_format(params));
u32 channels = params_channels(params);
u32 pins = DIV_ROUND_UP(channels, esai_priv->slots);
+ u32 slot_width = width;
u32 bclk, mask, val;
int ret;
- bclk = params_rate(params) * esai_priv->slot_width * esai_priv->slots;
+ /* Override slot_width if being specifially set */
+ if (esai_priv->slot_width)
+ slot_width = esai_priv->slot_width;
+
+ bclk = params_rate(params) * slot_width * esai_priv->slots;
ret = fsl_esai_set_bclk(dai, tx, bclk);
if (ret)
@@ -538,7 +543,7 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream,
regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), mask, val);
mask = ESAI_xCR_xSWS_MASK | (tx ? ESAI_xCR_PADC : 0);
- val = ESAI_xCR_xSWS(esai_priv->slot_width, width) | (tx ? ESAI_xCR_PADC : 0);
+ val = ESAI_xCR_xSWS(slot_width, width) | (tx ? ESAI_xCR_PADC : 0);
regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), mask, val);
@@ -734,7 +739,7 @@ static int fsl_esai_probe(struct platform_device *pdev)
return -ENOMEM;
esai_priv->pdev = pdev;
- strcpy(esai_priv->name, np->name);
+ strncpy(esai_priv->name, np->name, sizeof(esai_priv->name) - 1);
/* Get the addresses and IRQ */
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
@@ -780,9 +785,6 @@ static int fsl_esai_probe(struct platform_device *pdev)
return ret;
}
- /* Set a default slot size */
- esai_priv->slot_width = 32;
-
/* Set a default slot number */
esai_priv->slots = 2;
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index e6955170dc4..b6b0d25f6ac 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -67,8 +67,6 @@
/**
* FSLSSI_I2S_FORMATS: audio formats supported by the SSI
*
- * This driver currently only supports the SSI running in I2S slave mode.
- *
* The SSI has a limitation in that the samples must be in the same byte
* order as the host CPU. This is because when multiple bytes are written
* to the STX register, the bytes and bits must be written in the same
@@ -1099,7 +1097,7 @@ static const struct snd_soc_component_driver fsl_ssi_component = {
};
static struct snd_soc_dai_driver fsl_ssi_ac97_dai = {
- .ac97_control = 1,
+ .bus_control = true,
.playback = {
.stream_name = "AC97 Playback",
.channels_min = 2,
@@ -1363,7 +1361,7 @@ static int fsl_ssi_probe(struct platform_device *pdev)
return PTR_ERR(ssi_private->regs);
}
- ssi_private->irq = irq_of_parse_and_map(np, 0);
+ ssi_private->irq = platform_get_irq(pdev, 0);
if (!ssi_private->irq) {
dev_err(&pdev->dev, "no irq for node %s\n", np->full_name);
return -ENXIO;
@@ -1389,7 +1387,7 @@ static int fsl_ssi_probe(struct platform_device *pdev)
if (ssi_private->soc->imx) {
ret = fsl_ssi_imx_probe(pdev, ssi_private, iomem);
if (ret)
- goto error_irqmap;
+ return ret;
}
ret = snd_soc_register_component(&pdev->dev, &fsl_ssi_component,
@@ -1412,7 +1410,7 @@ static int fsl_ssi_probe(struct platform_device *pdev)
ret = fsl_ssi_debugfs_create(&ssi_private->dbg_stats, &pdev->dev);
if (ret)
- goto error_asoc_register;
+ goto error_irq;
/*
* If codec-handle property is missing from SSI node, we assume
@@ -1460,10 +1458,6 @@ error_asoc_register:
if (ssi_private->soc->imx)
fsl_ssi_imx_clean(pdev, ssi_private);
-error_irqmap:
- if (ssi_private->use_dma)
- irq_dispose_mapping(ssi_private->irq);
-
return ret;
}
@@ -1480,9 +1474,6 @@ static int fsl_ssi_remove(struct platform_device *pdev)
if (ssi_private->soc->imx)
fsl_ssi_imx_clean(pdev, ssi_private);
- if (ssi_private->use_dma)
- irq_dispose_mapping(ssi_private->irq);
-
return 0;
}
diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
index 1cb22dd034e..1dab963a59f 100644
--- a/sound/soc/fsl/imx-sgtl5000.c
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -175,10 +175,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev)
fail:
if (data && !IS_ERR(data->codec_clk))
clk_put(data->codec_clk);
- if (ssi_np)
- of_node_put(ssi_np);
- if (codec_np)
- of_node_put(codec_np);
+ of_node_put(ssi_np);
+ of_node_put(codec_np);
return ret;
}
diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c
index e1dc4014360..0c9068ebe1e 100644
--- a/sound/soc/fsl/imx-spdif.c
+++ b/sound/soc/fsl/imx-spdif.c
@@ -74,8 +74,7 @@ static int imx_spdif_audio_probe(struct platform_device *pdev)
platform_set_drvdata(pdev, data);
end:
- if (spdif_np)
- of_node_put(spdif_np);
+ of_node_put(spdif_np);
return ret;
}
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index ab2fdd76b69..60b0a5b1f1f 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -382,7 +382,7 @@ static struct snd_soc_dai_driver imx_ssi_dai = {
static struct snd_soc_dai_driver imx_ac97_dai = {
.probe = imx_ssi_dai_probe,
- .ac97_control = 1,
+ .bus_control = true,
.playback = {
.stream_name = "AC97 Playback",
.channels_min = 2,
diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c
index 3a3d17ce6ba..48179ffe154 100644
--- a/sound/soc/fsl/imx-wm8962.c
+++ b/sound/soc/fsl/imx-wm8962.c
@@ -281,10 +281,8 @@ static int imx_wm8962_probe(struct platform_device *pdev)
clk_fail:
clk_disable_unprepare(data->codec_clk);
fail:
- if (ssi_np)
- of_node_put(ssi_np);
- if (codec_np)
- of_node_put(codec_np);
+ of_node_put(ssi_np);
+ of_node_put(codec_np);
return ret;
}
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index f2b5d756b1f..0b82e209b6e 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -327,9 +327,6 @@ static int psc_dma_new(struct snd_soc_pcm_runtime *rtd)
goto capture_alloc_err;
}
- if (rtd->codec->ac97)
- rtd->codec->ac97->private_data = psc_dma;
-
return 0;
capture_alloc_err:
diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c
index 24eafa2cfbf..c6ed6ba965a 100644
--- a/sound/soc/fsl/mpc5200_psc_ac97.c
+++ b/sound/soc/fsl/mpc5200_psc_ac97.c
@@ -237,7 +237,7 @@ static const struct snd_soc_dai_ops psc_ac97_digital_ops = {
static struct snd_soc_dai_driver psc_ac97_dai[] = {
{
.name = "mpc5200-psc-ac97.0",
- .ac97_control = 1,
+ .bus_control = true,
.probe = psc_ac97_probe,
.playback = {
.stream_name = "AC97 Playback",
@@ -257,7 +257,7 @@ static struct snd_soc_dai_driver psc_ac97_dai[] = {
},
{
.name = "mpc5200-psc-ac97.1",
- .ac97_control = 1,
+ .bus_control = true,
.playback = {
.stream_name = "AC97 SPDIF",
.channels_min = 1,
@@ -282,7 +282,6 @@ static const struct snd_soc_component_driver psc_ac97_component = {
static int psc_ac97_of_probe(struct platform_device *op)
{
int rc;
- struct snd_ac97 ac97;
struct mpc52xx_psc __iomem *regs;
rc = mpc5200_audio_dma_create(op);
@@ -304,7 +303,6 @@ static int psc_ac97_of_probe(struct platform_device *op)
psc_dma = dev_get_drvdata(&op->dev);
regs = psc_dma->psc_regs;
- ac97.private_data = psc_dma;
psc_dma->imr = 0;
out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr);
generated by cgit v1.2.3-70-g09d2 (git 2.46.0) at 2024-11-25 20:11:09 +0000