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-rw-r--r--Documentation/devicetree/bindings/sound/fsl-sai.txt9
-rw-r--r--MAINTAINERS1
-rw-r--r--include/linux/mfd/syscon/imx6q-iomuxc-gpr.h6
-rw-r--r--include/sound/soc.h3
-rw-r--r--sound/soc/codecs/es8328.c194
-rw-r--r--sound/soc/codecs/es8328.h23
-rw-r--r--sound/soc/fsl/fsl_sai.c24
-rw-r--r--sound/soc/fsl/fsl_ssi.c80
-rw-r--r--sound/soc/soc-core.c14
9 files changed, 288 insertions, 66 deletions
diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt
index 044e5d76e2dd..740b467adf7d 100644
--- a/Documentation/devicetree/bindings/sound/fsl-sai.txt
+++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt
@@ -7,8 +7,8 @@ codec/DSP interfaces.
Required properties:
- - compatible : Compatible list, contains "fsl,vf610-sai" or
- "fsl,imx6sx-sai".
+ - compatible : Compatible list, contains "fsl,vf610-sai",
+ "fsl,imx6sx-sai" or "fsl,imx6ul-sai"
- reg : Offset and length of the register set for the device.
@@ -48,6 +48,11 @@ Required properties:
receive data by following their own bit clocks and
frame sync clocks separately.
+Optional properties (for mx6ul):
+
+ - fsl,sai-mclk-direction-output: This is a boolean property. If present,
+ indicates that SAI will output the SAI MCLK clock.
+
Note:
- If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the
default synchronous mode (sync Rx with Tx) will be used, which means both
diff --git a/MAINTAINERS b/MAINTAINERS
index a727d9959ecd..09a9cf1e0a8a 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -4661,6 +4661,7 @@ FREESCALE SOC SOUND DRIVERS
M: Timur Tabi <timur@tabi.org>
M: Nicolin Chen <nicoleotsuka@gmail.com>
M: Xiubo Li <Xiubo.Lee@gmail.com>
+R: Fabio Estevam <fabio.estevam@nxp.com>
L: alsa-devel@alsa-project.org (moderated for non-subscribers)
L: linuxppc-dev@lists.ozlabs.org
S: Maintained
diff --git a/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h b/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h
index 238c8db953eb..68353822afce 100644
--- a/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h
+++ b/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h
@@ -447,5 +447,11 @@
#define IMX6UL_GPR1_ENET2_CLK_OUTPUT (0x1 << 18)
#define IMX6UL_GPR1_ENET_CLK_DIR (0x3 << 17)
#define IMX6UL_GPR1_ENET_CLK_OUTPUT (0x3 << 17)
+#define IMX6UL_GPR1_SAI1_MCLK_DIR (0x1 << 19)
+#define IMX6UL_GPR1_SAI2_MCLK_DIR (0x1 << 20)
+#define IMX6UL_GPR1_SAI3_MCLK_DIR (0x1 << 21)
+#define IMX6UL_GPR1_SAI_MCLK_MASK (0x7 << 19)
+#define MCLK_DIR(x) (x == 1 ? IMX6UL_GPR1_SAI1_MCLK_DIR : x == 2 ? \
+ IMX6UL_GPR1_SAI2_MCLK_DIR : IMX6UL_GPR1_SAI3_MCLK_DIR)
#endif /* __LINUX_IMX6Q_IOMUXC_GPR_H */
diff --git a/include/sound/soc.h b/include/sound/soc.h
index ef25e86d51ee..fd7b58a58d6f 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -1683,6 +1683,9 @@ void snd_soc_remove_dai_link(struct snd_soc_card *card,
int snd_soc_register_dai(struct snd_soc_component *component,
struct snd_soc_dai_driver *dai_drv);
+struct snd_soc_dai *snd_soc_find_dai(
+ const struct snd_soc_dai_link_component *dlc);
+
#include <sound/soc-dai.h>
#ifdef CONFIG_DEBUG_FS
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index afa6c5db9dcc..2086d7107622 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -26,18 +26,30 @@
#include <sound/tlv.h>
#include "es8328.h"
-#define ES8328_SYSCLK_RATE_1X 11289600
-#define ES8328_SYSCLK_RATE_2X 22579200
+static const unsigned int rates_12288[] = {
+ 8000, 12000, 16000, 24000, 32000, 48000, 96000,
+};
-/* Run the codec at 22.5792 or 11.2896 MHz to support these rates */
-static struct {
- int rate;
- u8 ratio;
-} mclk_ratios[] = {
- { 8000, 9 },
- {11025, 7 },
- {22050, 4 },
- {44100, 2 },
+static const int ratios_12288[] = {
+ 10, 7, 6, 4, 3, 2, 0,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_12288 = {
+ .count = ARRAY_SIZE(rates_12288),
+ .list = rates_12288,
+};
+
+static const unsigned int rates_11289[] = {
+ 8018, 11025, 22050, 44100, 88200,
+};
+
+static const int ratios_11289[] = {
+ 9, 7, 4, 2, 0,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_11289 = {
+ .count = ARRAY_SIZE(rates_11289),
+ .list = rates_11289,
};
/* regulator supplies for sgtl5000, VDDD is an optional external supply */
@@ -57,16 +69,28 @@ static const char * const supply_names[ES8328_SUPPLY_NUM] = {
"HPVDD",
};
-#define ES8328_RATES (SNDRV_PCM_RATE_44100 | \
+#define ES8328_RATES (SNDRV_PCM_RATE_96000 | \
+ SNDRV_PCM_RATE_48000 | \
+ SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_32000 | \
SNDRV_PCM_RATE_22050 | \
- SNDRV_PCM_RATE_11025)
-#define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE)
+ SNDRV_PCM_RATE_16000 | \
+ SNDRV_PCM_RATE_11025 | \
+ SNDRV_PCM_RATE_8000)
+#define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S18_3LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE)
struct es8328_priv {
struct regmap *regmap;
struct clk *clk;
int playback_fs;
bool deemph;
+ int mclkdiv2;
+ const struct snd_pcm_hw_constraint_list *sysclk_constraints;
+ const int *mclk_ratios;
struct regulator_bulk_data supplies[ES8328_SUPPLY_NUM];
};
@@ -439,54 +463,131 @@ static int es8328_mute(struct snd_soc_dai *dai, int mute)
mute ? ES8328_DACCONTROL3_DACMUTE : 0);
}
+static int es8328_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+
+ if (es8328->sysclk_constraints)
+ snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ es8328->sysclk_constraints);
+
+ return 0;
+}
+
static int es8328_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
- int clk_rate;
int i;
int reg;
- u8 ratio;
+ int wl;
+ int ratio;
+
+ if (!es8328->sysclk_constraints) {
+ dev_err(codec->dev, "No MCLK configured\n");
+ return -EINVAL;
+ }
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
reg = ES8328_DACCONTROL2;
else
reg = ES8328_ADCCONTROL5;
- clk_rate = clk_get_rate(es8328->clk);
+ for (i = 0; i < es8328->sysclk_constraints->count; i++)
+ if (es8328->sysclk_constraints->list[i] == params_rate(params))
+ break;
- if ((clk_rate != ES8328_SYSCLK_RATE_1X) &&
- (clk_rate != ES8328_SYSCLK_RATE_2X)) {
- dev_err(codec->dev,
- "%s: clock is running at %d Hz, not %d or %d Hz\n",
- __func__, clk_rate,
- ES8328_SYSCLK_RATE_1X, ES8328_SYSCLK_RATE_2X);
+ if (i == es8328->sysclk_constraints->count) {
+ dev_err(codec->dev, "LRCLK %d unsupported with current clock\n",
+ params_rate(params));
return -EINVAL;
}
- /* find master mode MCLK to sampling frequency ratio */
- ratio = mclk_ratios[0].rate;
- for (i = 1; i < ARRAY_SIZE(mclk_ratios); i++)
- if (params_rate(params) <= mclk_ratios[i].rate)
- ratio = mclk_ratios[i].ratio;
+ ratio = es8328->mclk_ratios[i];
+ snd_soc_update_bits(codec, ES8328_MASTERMODE,
+ ES8328_MASTERMODE_MCLKDIV2,
+ es8328->mclkdiv2 ? ES8328_MASTERMODE_MCLKDIV2 : 0);
+
+ switch (params_width(params)) {
+ case 16:
+ wl = 3;
+ break;
+ case 18:
+ wl = 2;
+ break;
+ case 20:
+ wl = 1;
+ break;
+ case 24:
+ wl = 0;
+ break;
+ case 32:
+ wl = 4;
+ break;
+ default:
+ return -EINVAL;
+ }
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ snd_soc_update_bits(codec, ES8328_DACCONTROL1,
+ ES8328_DACCONTROL1_DACWL_MASK,
+ wl << ES8328_DACCONTROL1_DACWL_SHIFT);
+
es8328->playback_fs = params_rate(params);
es8328_set_deemph(codec);
- }
+ } else
+ snd_soc_update_bits(codec, ES8328_ADCCONTROL4,
+ ES8328_ADCCONTROL4_ADCWL_MASK,
+ wl << ES8328_ADCCONTROL4_ADCWL_SHIFT);
return snd_soc_update_bits(codec, reg, ES8328_RATEMASK, ratio);
}
+static int es8328_set_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+ int mclkdiv2 = 0;
+
+ switch (freq) {
+ case 0:
+ es8328->sysclk_constraints = NULL;
+ es8328->mclk_ratios = NULL;
+ break;
+ case 22579200:
+ mclkdiv2 = 1;
+ /* fallthru */
+ case 11289600:
+ es8328->sysclk_constraints = &constraints_11289;
+ es8328->mclk_ratios = ratios_11289;
+ break;
+ case 24576000:
+ mclkdiv2 = 1;
+ /* fallthru */
+ case 12288000:
+ es8328->sysclk_constraints = &constraints_12288;
+ es8328->mclk_ratios = ratios_12288;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ es8328->mclkdiv2 = mclkdiv2;
+ return 0;
+}
+
static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
- struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
- int clk_rate;
- u8 mode = ES8328_DACCONTROL1_DACWL_16;
+ u8 dac_mode = 0;
+ u8 adc_mode = 0;
/* set master/slave audio interface */
if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBM_CFM)
@@ -495,13 +596,16 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
/* interface format */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
- mode |= ES8328_DACCONTROL1_DACFORMAT_I2S;
+ dac_mode |= ES8328_DACCONTROL1_DACFORMAT_I2S;
+ adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_I2S;
break;
case SND_SOC_DAIFMT_RIGHT_J:
- mode |= ES8328_DACCONTROL1_DACFORMAT_RJUST;
+ dac_mode |= ES8328_DACCONTROL1_DACFORMAT_RJUST;
+ adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_RJUST;
break;
case SND_SOC_DAIFMT_LEFT_J:
- mode |= ES8328_DACCONTROL1_DACFORMAT_LJUST;
+ dac_mode |= ES8328_DACCONTROL1_DACFORMAT_LJUST;
+ adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_LJUST;
break;
default:
return -EINVAL;
@@ -511,18 +615,14 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF)
return -EINVAL;
- snd_soc_write(codec, ES8328_DACCONTROL1, mode);
- snd_soc_write(codec, ES8328_ADCCONTROL4, mode);
+ snd_soc_update_bits(codec, ES8328_DACCONTROL1,
+ ES8328_DACCONTROL1_DACFORMAT_MASK, dac_mode);
+ snd_soc_update_bits(codec, ES8328_ADCCONTROL4,
+ ES8328_ADCCONTROL4_ADCFORMAT_MASK, adc_mode);
/* Master serial port mode, with BCLK generated automatically */
- clk_rate = clk_get_rate(es8328->clk);
- if (clk_rate == ES8328_SYSCLK_RATE_1X)
- snd_soc_write(codec, ES8328_MASTERMODE,
- ES8328_MASTERMODE_MSC);
- else
- snd_soc_write(codec, ES8328_MASTERMODE,
- ES8328_MASTERMODE_MCLKDIV2 |
- ES8328_MASTERMODE_MSC);
+ snd_soc_update_bits(codec, ES8328_MASTERMODE,
+ ES8328_MASTERMODE_MSC, ES8328_MASTERMODE_MSC);
return 0;
}
@@ -579,8 +679,10 @@ static int es8328_set_bias_level(struct snd_soc_codec *codec,
}
static const struct snd_soc_dai_ops es8328_dai_ops = {
+ .startup = es8328_startup,
.hw_params = es8328_hw_params,
.digital_mute = es8328_mute,
+ .set_sysclk = es8328_set_sysclk,
.set_fmt = es8328_set_dai_fmt,
};
@@ -601,6 +703,7 @@ static struct snd_soc_dai_driver es8328_dai = {
.formats = ES8328_FORMATS,
},
.ops = &es8328_dai_ops,
+ .symmetric_rates = 1,
};
static int es8328_suspend(struct snd_soc_codec *codec)
@@ -708,6 +811,7 @@ const struct regmap_config es8328_regmap_config = {
.val_bits = 8,
.max_register = ES8328_REG_MAX,
.cache_type = REGCACHE_RBTREE,
+ .use_single_rw = true,
};
EXPORT_SYMBOL_GPL(es8328_regmap_config);
diff --git a/sound/soc/codecs/es8328.h b/sound/soc/codecs/es8328.h
index 156c748c89c7..1a736e72a929 100644
--- a/sound/soc/codecs/es8328.h
+++ b/sound/soc/codecs/es8328.h
@@ -22,7 +22,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
#define ES8328_CONTROL1_VMIDSEL_50k (1 << 0)
#define ES8328_CONTROL1_VMIDSEL_500k (2 << 0)
#define ES8328_CONTROL1_VMIDSEL_5k (3 << 0)
-#define ES8328_CONTROL1_VMIDSEL_MASK (7 << 0)
+#define ES8328_CONTROL1_VMIDSEL_MASK (3 << 0)
#define ES8328_CONTROL1_ENREF (1 << 2)
#define ES8328_CONTROL1_SEQEN (1 << 3)
#define ES8328_CONTROL1_SAMEFS (1 << 4)
@@ -84,7 +84,20 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
#define ES8328_ADCCONTROL1 0x09
#define ES8328_ADCCONTROL2 0x0a
#define ES8328_ADCCONTROL3 0x0b
+
#define ES8328_ADCCONTROL4 0x0c
+#define ES8328_ADCCONTROL4_ADCFORMAT_MASK (3 << 0)
+#define ES8328_ADCCONTROL4_ADCFORMAT_I2S (0 << 0)
+#define ES8328_ADCCONTROL4_ADCFORMAT_LJUST (1 << 0)
+#define ES8328_ADCCONTROL4_ADCFORMAT_RJUST (2 << 0)
+#define ES8328_ADCCONTROL4_ADCFORMAT_PCM (3 << 0)
+#define ES8328_ADCCONTROL4_ADCWL_SHIFT 2
+#define ES8328_ADCCONTROL4_ADCWL_MASK (7 << 2)
+#define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_NORMAL (0 << 5)
+#define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_INV (1 << 5)
+#define ES8328_ADCCONTROL4_ADCLRP_PCM_MSB_CLK2 (0 << 5)
+#define ES8328_ADCCONTROL4_ADCLRP_PCM_MSB_CLK1 (1 << 5)
+
#define ES8328_ADCCONTROL5 0x0d
#define ES8328_ADCCONTROL5_RATEMASK (0x1f << 0)
@@ -109,15 +122,13 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
#define ES8328_ADCCONTROL14 0x16
#define ES8328_DACCONTROL1 0x17
+#define ES8328_DACCONTROL1_DACFORMAT_MASK (3 << 1)
#define ES8328_DACCONTROL1_DACFORMAT_I2S (0 << 1)
#define ES8328_DACCONTROL1_DACFORMAT_LJUST (1 << 1)
#define ES8328_DACCONTROL1_DACFORMAT_RJUST (2 << 1)
#define ES8328_DACCONTROL1_DACFORMAT_PCM (3 << 1)
-#define ES8328_DACCONTROL1_DACWL_24 (0 << 3)
-#define ES8328_DACCONTROL1_DACWL_20 (1 << 3)
-#define ES8328_DACCONTROL1_DACWL_18 (2 << 3)
-#define ES8328_DACCONTROL1_DACWL_16 (3 << 3)
-#define ES8328_DACCONTROL1_DACWL_32 (4 << 3)
+#define ES8328_DACCONTROL1_DACWL_SHIFT 3
+#define ES8328_DACCONTROL1_DACWL_MASK (7 << 3)
#define ES8328_DACCONTROL1_DACLRP_I2S_POL_NORMAL (0 << 6)
#define ES8328_DACCONTROL1_DACLRP_I2S_POL_INV (1 << 6)
#define ES8328_DACCONTROL1_DACLRP_PCM_MSB_CLK2 (0 << 6)
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 0754df771e3b..2147994ab46f 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -21,6 +21,8 @@
#include <sound/core.h>
#include <sound/dmaengine_pcm.h>
#include <sound/pcm_params.h>
+#include <linux/mfd/syscon.h>
+#include <linux/mfd/syscon/imx6q-iomuxc-gpr.h>
#include "fsl_sai.h"
#include "imx-pcm.h"
@@ -786,10 +788,12 @@ static int fsl_sai_probe(struct platform_device *pdev)
{
struct device_node *np = pdev->dev.of_node;
struct fsl_sai *sai;
+ struct regmap *gpr;
struct resource *res;
void __iomem *base;
char tmp[8];
int irq, ret, i;
+ int index;
sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL);
if (!sai)
@@ -797,7 +801,8 @@ static int fsl_sai_probe(struct platform_device *pdev)
sai->pdev = pdev;
- if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai"))
+ if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai") ||
+ of_device_is_compatible(pdev->dev.of_node, "fsl,imx6ul-sai"))
sai->sai_on_imx = true;
sai->is_lsb_first = of_property_read_bool(np, "lsb-first");
@@ -877,6 +882,22 @@ static int fsl_sai_probe(struct platform_device *pdev)
fsl_sai_dai.symmetric_samplebits = 0;
}
+ if (of_find_property(np, "fsl,sai-mclk-direction-output", NULL) &&
+ of_device_is_compatible(pdev->dev.of_node, "fsl,imx6ul-sai")) {
+ gpr = syscon_regmap_lookup_by_compatible("fsl,imx6ul-iomuxc-gpr");
+ if (IS_ERR(gpr)) {
+ dev_err(&pdev->dev, "cannot find iomuxc registers\n");
+ return PTR_ERR(gpr);
+ }
+
+ index = of_alias_get_id(np, "sai");
+ if (index < 0)
+ return index;
+
+ regmap_update_bits(gpr, IOMUXC_GPR1, MCLK_DIR(index),
+ MCLK_DIR(index));
+ }
+
sai->dma_params_rx.addr = res->start + FSL_SAI_RDR;
sai->dma_params_tx.addr = res->start + FSL_SAI_TDR;
sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX;
@@ -898,6 +919,7 @@ static int fsl_sai_probe(struct platform_device *pdev)
static const struct of_device_id fsl_sai_ids[] = {
{ .compatible = "fsl,vf610-sai", },
{ .compatible = "fsl,imx6sx-sai", },
+ { .compatible = "fsl,imx6ul-sai", },
{ /* sentinel */ }
};
MODULE_DEVICE_TABLE(of, fsl_sai_ids);
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 08dcbbf60adb..632ecc0e3956 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -262,6 +262,7 @@ struct fsl_ssi_private {
struct fsl_ssi_dbg dbg_stats;
const struct fsl_ssi_soc_data *soc;
+ struct device *dev;
};
/*
@@ -401,6 +402,26 @@ static void fsl_ssi_rxtx_config(struct fsl_ssi_private *ssi_private,
}
/*
+ * Clear RX or TX FIFO to remove samples from the previous
+ * stream session which may be still present in the FIFO and
+ * may introduce bad samples and/or channel slipping.
+ *
+ * Note: The SOR is not documented in recent IMX datasheet, but
+ * is described in IMX51 reference manual at section 56.3.3.15.
+ */
+static void fsl_ssi_fifo_clear(struct fsl_ssi_private *ssi_private,
+ bool is_rx)
+{
+ if (is_rx) {
+ regmap_update_bits(ssi_private->regs, CCSR_SSI_SOR,
+ CCSR_SSI_SOR_RX_CLR, CCSR_SSI_SOR_RX_CLR);
+ } else {
+ regmap_update_bits(ssi_private->regs, CCSR_SSI_SOR,
+ CCSR_SSI_SOR_TX_CLR, CCSR_SSI_SOR_TX_CLR);
+ }
+}
+
+/*
* Calculate the bits that have to be disabled for the current stream that is
* getting disabled. This keeps the bits enabled that are necessary for the
* second stream to work if 'stream_active' is true.
@@ -475,9 +496,11 @@ static void fsl_ssi_config(struct fsl_ssi_private *ssi_private, bool enable,
* (online configuration)
*/
if (enable) {
- regmap_update_bits(regs, CCSR_SSI_SIER, vals->sier, vals->sier);
+ fsl_ssi_fifo_clear(ssi_private, vals->scr & CCSR_SSI_SCR_RE);
+
regmap_update_bits(regs, CCSR_SSI_SRCR, vals->srcr, vals->srcr);
regmap_update_bits(regs, CCSR_SSI_STCR, vals->stcr, vals->stcr);
+ regmap_update_bits(regs, CCSR_SSI_SIER, vals->sier, vals->sier);
} else {
u32 sier;
u32 srcr;
@@ -507,8 +530,40 @@ static void fsl_ssi_config(struct fsl_ssi_private *ssi_private, bool enable,
config_done:
/* Enabling of subunits is done after configuration */
- if (enable)
+ if (enable) {
+ if (ssi_private->use_dma && (vals->scr & CCSR_SSI_SCR_TE)) {
+ /*
+ * Be sure the Tx FIFO is filled when TE is set.
+ * Otherwise, there are some chances to start the
+ * playback with some void samples inserted first,
+ * generating a channel slip.
+ *
+ * First, SSIEN must be set, to let the FIFO be filled.
+ *
+ * Notes:
+ * - Limit this fix to the DMA case until FIQ cases can
+ * be tested.
+ * - Limit the length of the busy loop to not lock the
+ * system too long, even if 1-2 loops are sufficient
+ * in general.
+ */
+ int i;
+ int max_loop = 100;
+ regmap_update_bits(regs, CCSR_SSI_SCR,
+ CCSR_SSI_SCR_SSIEN, CCSR_SSI_SCR_SSIEN);
+ for (i = 0; i < max_loop; i++) {
+ u32 sfcsr;
+ regmap_read(regs, CCSR_SSI_SFCSR, &sfcsr);
+ if (CCSR_SSI_SFCSR_TFCNT0(sfcsr))
+ break;
+ }
+ if (i == max_loop) {
+ dev_err(ssi_private->dev,
+ "Timeout waiting TX FIFO filling\n");
+ }
+ }
regmap_update_bits(regs, CCSR_SSI_SCR, vals->scr, vals->scr);
+ }
}
@@ -671,6 +726,15 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
if (IS_ERR(ssi_private->baudclk))
return -EINVAL;
+ /*
+ * Hardware limitation: The bclk rate must be
+ * never greater than 1/5 IPG clock rate
+ */
+ if (freq * 5 > clk_get_rate(ssi_private->clk)) {
+ dev_err(cpu_dai->dev, "bitclk > ipgclk/5\n");
+ return -EINVAL;
+ }
+
baudclk_is_used = ssi_private->baudclk_streams & ~(BIT(substream->stream));
/* It should be already enough to divide clock by setting pm alone */
@@ -687,13 +751,6 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
else
clkrate = clk_round_rate(ssi_private->baudclk, tmprate);
- /*
- * Hardware limitation: The bclk rate must be
- * never greater than 1/5 IPG clock rate
- */
- if (clkrate * 5 > clk_get_rate(ssi_private->clk))
- continue;
-
clkrate /= factor;
afreq = clkrate / (i + 1);
@@ -1159,14 +1216,14 @@ static struct snd_soc_dai_driver fsl_ssi_dai_template = {
.playback = {
.stream_name = "CPU-Playback",
.channels_min = 1,
- .channels_max = 2,
+ .channels_max = 32,
.rates = FSLSSI_I2S_RATES,
.formats = FSLSSI_I2S_FORMATS,
},
.capture = {
.stream_name = "CPU-Capture",
.channels_min = 1,
- .channels_max = 2,
+ .channels_max = 32,
.rates = FSLSSI_I2S_RATES,
.formats = FSLSSI_I2S_FORMATS,
},
@@ -1403,6 +1460,7 @@ static int fsl_ssi_probe(struct platform_device *pdev)
}
ssi_private->soc = of_id->data;
+ ssi_private->dev = &pdev->dev;
sprop = of_get_property(np, "fsl,mode", NULL);
if (sprop) {
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d2e62b159610..16369cad4803 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -930,7 +930,18 @@ static struct snd_soc_component *soc_find_component(
return NULL;
}
-static struct snd_soc_dai *snd_soc_find_dai(
+/**
+ * snd_soc_find_dai - Find a registered DAI
+ *
+ * @dlc: name of the DAI and optional component info to match
+ *
+ * This function will search all regsitered components and their DAIs to
+ * find the DAI of the same name. The component's of_node and name
+ * should also match if being specified.
+ *
+ * Return: pointer of DAI, or NULL if not found.
+ */
+struct snd_soc_dai *snd_soc_find_dai(
const struct snd_soc_dai_link_component *dlc)
{
struct snd_soc_component *component;
@@ -959,6 +970,7 @@ static struct snd_soc_dai *snd_soc_find_dai(
return NULL;
}
+EXPORT_SYMBOL_GPL(snd_soc_find_dai);
static bool soc_is_dai_link_bound(struct snd_soc_card *card,
struct snd_soc_dai_link *dai_link)
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-17 18:37:12 +0000