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-rw-r--r--sound/soc/fsl/eukrea-tlv320.c23
-rw-r--r--sound/soc/fsl/fsl_asrc.c5
-rw-r--r--sound/soc/fsl/fsl_asrc.h3
-rw-r--r--sound/soc/fsl/fsl_esai.c2
-rw-r--r--sound/soc/fsl/fsl_sai.c2
-rw-r--r--sound/soc/fsl/fsl_spdif.c17
-rw-r--r--sound/soc/fsl/fsl_ssi.c10
-rw-r--r--sound/soc/fsl/fsl_utils.c27
-rw-r--r--sound/soc/fsl/fsl_utils.h3
-rw-r--r--sound/soc/fsl/imx-mc13783.c5
-rw-r--r--sound/soc/fsl/imx-spdif.c1
-rw-r--r--sound/soc/fsl/imx-ssi.c5
-rw-r--r--sound/soc/fsl/mx27vis-aic32x4.c12
-rw-r--r--sound/soc/fsl/wm1133-ev1.c16
14 files changed, 27 insertions, 104 deletions
diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c
index 9ce70fc67b0..e1aa3834b10 100644
--- a/sound/soc/fsl/eukrea-tlv320.c
+++ b/sound/soc/fsl/eukrea-tlv320.c
@@ -42,25 +42,6 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
int ret;
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM);
- /* fsl_ssi lacks the set_fmt ops. */
- if (ret && ret != -ENOTSUPP) {
- dev_err(cpu_dai->dev,
- "Failed to set the cpu dai format.\n");
- return ret;
- }
-
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM);
- if (ret) {
- dev_err(cpu_dai->dev,
- "Failed to set the codec format.\n");
- return ret;
- }
-
ret = snd_soc_dai_set_sysclk(codec_dai, 0,
CODEC_CLOCK, SND_SOC_CLOCK_OUT);
if (ret) {
@@ -69,7 +50,7 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream,
return ret;
}
- snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0);
+ snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 0);
ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0,
SND_SOC_CLOCK_IN);
@@ -91,6 +72,8 @@ 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,
};
diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c
index 026a8011754..c068494bae3 100644
--- a/sound/soc/fsl/fsl_asrc.c
+++ b/sound/soc/fsl/fsl_asrc.c
@@ -818,7 +818,6 @@ static int fsl_asrc_probe(struct platform_device *pdev)
return -ENOMEM;
asrc_priv->pdev = pdev;
- strncpy(asrc_priv->name, np->name, sizeof(asrc_priv->name) - 1);
/* Get the addresses and IRQ */
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
@@ -837,12 +836,12 @@ static int fsl_asrc_probe(struct platform_device *pdev)
irq = platform_get_irq(pdev, 0);
if (irq < 0) {
- dev_err(&pdev->dev, "no irq for node %s\n", np->full_name);
+ dev_err(&pdev->dev, "no irq for node %s\n", pdev->name);
return irq;
}
ret = devm_request_irq(&pdev->dev, irq, fsl_asrc_isr, 0,
- asrc_priv->name, asrc_priv);
+ dev_name(&pdev->dev), asrc_priv);
if (ret) {
dev_err(&pdev->dev, "failed to claim irq %u: %d\n", irq, ret);
return ret;
diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h
index a3f211f53c2..4aed63c4b43 100644
--- a/sound/soc/fsl/fsl_asrc.h
+++ b/sound/soc/fsl/fsl_asrc.h
@@ -433,7 +433,6 @@ struct fsl_asrc_pair {
* @channel_avail: non-occupied channel numbers
* @asrc_rate: default sample rate for ASoC Back-Ends
* @asrc_width: default sample width for ASoC Back-Ends
- * @name: driver name
*/
struct fsl_asrc {
struct snd_dmaengine_dai_dma_data dma_params_rx;
@@ -452,8 +451,6 @@ struct fsl_asrc {
int asrc_rate;
int asrc_width;
-
- char name[32];
};
extern struct snd_soc_platform_driver fsl_asrc_platform;
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index 1c08ab13637..5c7597191e3 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -774,7 +774,7 @@ static int fsl_esai_probe(struct platform_device *pdev)
irq = platform_get_irq(pdev, 0);
if (irq < 0) {
- dev_err(&pdev->dev, "no irq for node %s\n", np->full_name);
+ dev_err(&pdev->dev, "no irq for node %s\n", pdev->name);
return irq;
}
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 032d2d33619..ec79c3d5e65 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -612,7 +612,7 @@ static int fsl_sai_probe(struct platform_device *pdev)
irq = platform_get_irq(pdev, 0);
if (irq < 0) {
- dev_err(&pdev->dev, "no irq for node %s\n", np->full_name);
+ dev_err(&pdev->dev, "no irq for node %s\n", pdev->name);
return irq;
}
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index af0429421fc..75870c0ea2c 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -90,7 +90,6 @@ struct spdif_mixer_control {
* @sysclk: system clock for rx clock rate measurement
* @dma_params_tx: DMA parameters for transmit channel
* @dma_params_rx: DMA parameters for receive channel
- * @name: driver name
*/
struct fsl_spdif_priv {
struct spdif_mixer_control fsl_spdif_control;
@@ -109,12 +108,8 @@ struct fsl_spdif_priv {
struct clk *sysclk;
struct snd_dmaengine_dai_dma_data dma_params_tx;
struct snd_dmaengine_dai_dma_data dma_params_rx;
-
- /* The name space will be allocated dynamically */
- char name[0];
};
-
/* DPLL locked and lock loss interrupt handler */
static void spdif_irq_dpll_lock(struct fsl_spdif_priv *spdif_priv)
{
@@ -1169,19 +1164,15 @@ static int fsl_spdif_probe(struct platform_device *pdev)
if (!np)
return -ENODEV;
- spdif_priv = devm_kzalloc(&pdev->dev,
- sizeof(struct fsl_spdif_priv) + strlen(np->name) + 1,
- GFP_KERNEL);
+ spdif_priv = devm_kzalloc(&pdev->dev, sizeof(*spdif_priv), GFP_KERNEL);
if (!spdif_priv)
return -ENOMEM;
- strcpy(spdif_priv->name, np->name);
-
spdif_priv->pdev = pdev;
/* Initialize this copy of the CPU DAI driver structure */
memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai));
- spdif_priv->cpu_dai_drv.name = spdif_priv->name;
+ spdif_priv->cpu_dai_drv.name = dev_name(&pdev->dev);
/* Get the addresses and IRQ */
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
@@ -1198,12 +1189,12 @@ static int fsl_spdif_probe(struct platform_device *pdev)
irq = platform_get_irq(pdev, 0);
if (irq < 0) {
- dev_err(&pdev->dev, "no irq for node %s\n", np->full_name);
+ dev_err(&pdev->dev, "no irq for node %s\n", pdev->name);
return irq;
}
ret = devm_request_irq(&pdev->dev, irq, spdif_isr, 0,
- spdif_priv->name, spdif_priv);
+ dev_name(&pdev->dev), spdif_priv);
if (ret) {
dev_err(&pdev->dev, "could not claim irq %u\n", irq);
return ret;
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 059496ed9ad..2595611e8a6 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -160,7 +160,7 @@ struct fsl_ssi_soc_data {
*/
struct fsl_ssi_private {
struct regmap *regs;
- unsigned int irq;
+ int irq;
struct snd_soc_dai_driver cpu_dai_drv;
unsigned int dai_fmt;
@@ -992,8 +992,8 @@ static int fsl_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask,
regmap_update_bits(regs, CCSR_SSI_SCR, CCSR_SSI_SCR_SSIEN,
CCSR_SSI_SCR_SSIEN);
- regmap_write(regs, CCSR_SSI_STMSK, tx_mask);
- regmap_write(regs, CCSR_SSI_SRMSK, rx_mask);
+ regmap_write(regs, CCSR_SSI_STMSK, ~tx_mask);
+ regmap_write(regs, CCSR_SSI_SRMSK, ~rx_mask);
regmap_update_bits(regs, CCSR_SSI_SCR, CCSR_SSI_SCR_SSIEN, val);
@@ -1362,8 +1362,8 @@ static int fsl_ssi_probe(struct platform_device *pdev)
}
ssi_private->irq = platform_get_irq(pdev, 0);
- if (ssi_private->irq < 0) {
- dev_err(&pdev->dev, "no irq for node %s\n", np->full_name);
+ if (!ssi_private->irq) {
+ dev_err(&pdev->dev, "no irq for node %s\n", pdev->name);
return ssi_private->irq;
}
diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c
index 2ac7755da87..b9e42b503a3 100644
--- a/sound/soc/fsl/fsl_utils.c
+++ b/sound/soc/fsl/fsl_utils.c
@@ -86,33 +86,6 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np,
}
EXPORT_SYMBOL(fsl_asoc_get_dma_channel);
-/**
- * fsl_asoc_xlate_tdm_slot_mask - generate TDM slot TX/RX mask.
- *
- * @slots: Number of slots in use.
- * @tx_mask: bitmask representing active TX slots.
- * @rx_mask: bitmask representing active RX slots.
- *
- * This function used to generate the TDM slot TX/RX mask. And the TX/RX
- * mask will use a 0 bit for an active slot as default, and the default
- * active bits are at the LSB of the mask value.
- */
-int fsl_asoc_xlate_tdm_slot_mask(unsigned int slots,
- unsigned int *tx_mask,
- unsigned int *rx_mask)
-{
- if (!slots)
- return -EINVAL;
-
- if (tx_mask)
- *tx_mask = ~((1 << slots) - 1);
- if (rx_mask)
- *rx_mask = ~((1 << slots) - 1);
-
- return 0;
-}
-EXPORT_SYMBOL_GPL(fsl_asoc_xlate_tdm_slot_mask);
-
MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
MODULE_DESCRIPTION("Freescale ASoC utility code");
MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/fsl/fsl_utils.h b/sound/soc/fsl/fsl_utils.h
index df535db4031..1687b66ef18 100644
--- a/sound/soc/fsl/fsl_utils.h
+++ b/sound/soc/fsl/fsl_utils.h
@@ -22,7 +22,4 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np, const char *name,
struct snd_soc_dai_link *dai,
unsigned int *dma_channel_id,
unsigned int *dma_id);
-int fsl_asoc_xlate_tdm_slot_mask(unsigned int slots,
- unsigned int *tx_mask,
- unsigned int *rx_mask);
#endif /* _FSL_UTILS_H */
diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c
index 6bf5bce01a9..9e6493d4e7f 100644
--- a/sound/soc/fsl/imx-mc13783.c
+++ b/sound/soc/fsl/imx-mc13783.c
@@ -37,8 +37,7 @@ static int imx_mc13783_hifi_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *codec_dai = rtd->codec_dai;
int ret;
- ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xfffffffc, 0xfffffffc,
- 4, 16);
+ ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 4, 16);
if (ret)
return ret;
@@ -46,7 +45,7 @@ static int imx_mc13783_hifi_hw_params(struct snd_pcm_substream *substream,
if (ret)
return ret;
- ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x0, 0xfffffffc, 2, 16);
+ ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 16);
if (ret)
return ret;
diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c
index e94704f1b9e..33da26a1245 100644
--- a/sound/soc/fsl/imx-spdif.c
+++ b/sound/soc/fsl/imx-spdif.c
@@ -60,6 +60,7 @@ static int imx_spdif_audio_probe(struct platform_device *pdev)
data->card.dev = &pdev->dev;
data->card.dai_link = &data->dai;
data->card.num_links = 1;
+ data->card.owner = THIS_MODULE;
ret = snd_soc_of_parse_card_name(&data->card, "model");
if (ret)
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index fa801e17c51..461ce27b884 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -74,8 +74,8 @@ static int imx_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai,
sccr |= SSI_STCCR_DC(slots - 1);
writel(sccr, ssi->base + SSI_SRCCR);
- writel(tx_mask, ssi->base + SSI_STMSK);
- writel(rx_mask, ssi->base + SSI_SRMSK);
+ writel(~tx_mask, ssi->base + SSI_STMSK);
+ writel(~rx_mask, ssi->base + SSI_SRMSK);
return 0;
}
@@ -340,7 +340,6 @@ static const struct snd_soc_dai_ops imx_ssi_pcm_dai_ops = {
.set_fmt = imx_ssi_set_dai_fmt,
.set_clkdiv = imx_ssi_set_dai_clkdiv,
.set_sysclk = imx_ssi_set_dai_sysclk,
- .xlate_tdm_slot_mask = fsl_asoc_xlate_tdm_slot_mask,
.set_tdm_slot = imx_ssi_set_dai_tdm_slot,
.trigger = imx_ssi_trigger,
};
diff --git a/sound/soc/fsl/mx27vis-aic32x4.c b/sound/soc/fsl/mx27vis-aic32x4.c
index b1ced7b8d80..198eeb3f3f7 100644
--- a/sound/soc/fsl/mx27vis-aic32x4.c
+++ b/sound/soc/fsl/mx27vis-aic32x4.c
@@ -55,16 +55,6 @@ static int mx27vis_aic32x4_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
int ret;
- u32 dai_format;
-
- dai_format = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM;
-
- /* set codec DAI configuration */
- snd_soc_dai_set_fmt(codec_dai, dai_format);
-
- /* set cpu DAI configuration */
- snd_soc_dai_set_fmt(cpu_dai, dai_format);
ret = snd_soc_dai_set_sysclk(codec_dai, 0,
25000000, SND_SOC_CLOCK_OUT);
@@ -164,6 +154,8 @@ static struct snd_soc_dai_link mx27vis_aic32x4_dai = {
.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,
};
diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c
index 804749a6c61..a958937ab40 100644
--- a/sound/soc/fsl/wm1133-ev1.c
+++ b/sound/soc/fsl/wm1133-ev1.c
@@ -87,7 +87,6 @@ static int wm1133_ev1_hw_params(struct snd_pcm_substream *substream,
snd_pcm_format_t format = params_format(params);
unsigned int rate = params_rate(params);
unsigned int channels = params_channels(params);
- u32 dai_format;
/* find the correct audio parameters */
for (i = 0; i < ARRAY_SIZE(wm8350_audio); i++) {
@@ -104,22 +103,13 @@ static int wm1133_ev1_hw_params(struct snd_pcm_substream *substream,
/* codec FLL input is 14.75 MHz from MCLK */
snd_soc_dai_set_pll(codec_dai, 0, 0, 14750000, wm8350_audio[i].sysclk);
- dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM;
-
- /* set codec DAI configuration */
- snd_soc_dai_set_fmt(codec_dai, dai_format);
-
- /* set cpu DAI configuration */
- snd_soc_dai_set_fmt(cpu_dai, dai_format);
-
/* TODO: The SSI driver should figure this out for us */
switch (channels) {
case 2:
- snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0);
+ snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 0);
break;
case 1:
- snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffe, 0xffffffe, 1, 0);
+ snd_soc_dai_set_tdm_slot(cpu_dai, 0x1, 0x1, 1, 0);
break;
default:
return -EINVAL;
@@ -244,6 +234,8 @@ static struct snd_soc_dai_link wm1133_ev1_dai = {
.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,
};
static struct snd_soc_card wm1133_ev1 = {
generated by cgit v1.2.3-70-g09d2 (git 2.46.0) at 2024-12-11 20:52:57 +0000