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-rw-r--r--sound/soc/codecs/arizona.c20
-rw-r--r--sound/soc/codecs/wm2200.c3
-rw-r--r--sound/soc/codecs/wm5102.c3
-rw-r--r--sound/soc/codecs/wm5110.c3
-rw-r--r--sound/soc/codecs/wm_adsp.c20
-rw-r--r--sound/soc/davinci/davinci-mcasp.c2
-rw-r--r--sound/soc/dwc/designware_i2s.c4
-rw-r--r--sound/soc/fsl/imx-pcm-dma.c21
-rw-r--r--sound/soc/fsl/imx-pcm-fiq.c22
-rw-r--r--sound/soc/fsl/imx-pcm.c32
-rw-r--r--sound/soc/fsl/imx-pcm.h18
-rw-r--r--sound/soc/samsung/s3c24xx-i2s.c2
-rw-r--r--sound/soc/soc-dapm.c12
13 files changed, 100 insertions, 62 deletions
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 1d8bb591759..3b8e8c70b79 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -56,14 +56,14 @@
#define arizona_fll_warn(_fll, fmt, ...) \
dev_warn(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__)
#define arizona_fll_dbg(_fll, fmt, ...) \
- dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__)
+ dev_dbg(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__)
#define arizona_aif_err(_dai, fmt, ...) \
dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__)
#define arizona_aif_warn(_dai, fmt, ...) \
dev_warn(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__)
#define arizona_aif_dbg(_dai, fmt, ...) \
- dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__)
+ dev_dbg(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__)
const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = {
"None",
@@ -685,7 +685,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
}
sr_val = i;
- lrclk = snd_soc_params_to_bclk(params) / params_rate(params);
+ lrclk = rates[bclk] / params_rate(params);
arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n",
rates[bclk], rates[bclk] / lrclk);
@@ -910,7 +910,7 @@ static int arizona_calc_fll(struct arizona_fll *fll,
cfg->n = target / (ratio * Fref);
- if (target % Fref) {
+ if (target % (ratio * Fref)) {
gcd_fll = gcd(target, ratio * Fref);
arizona_fll_dbg(fll, "GCD=%u\n", gcd_fll);
@@ -922,6 +922,15 @@ static int arizona_calc_fll(struct arizona_fll *fll,
cfg->lambda = 0;
}
+ /* Round down to 16bit range with cost of accuracy lost.
+ * Denominator must be bigger than numerator so we only
+ * take care of it.
+ */
+ while (cfg->lambda >= (1 << 16)) {
+ cfg->theta >>= 1;
+ cfg->lambda >>= 1;
+ }
+
arizona_fll_dbg(fll, "N=%x THETA=%x LAMBDA=%x\n",
cfg->n, cfg->theta, cfg->lambda);
arizona_fll_dbg(fll, "FRATIO=%x(%d) OUTDIV=%x REFCLK_DIV=%x\n",
@@ -1082,6 +1091,9 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq,
id, ret);
}
+ regmap_update_bits(arizona->regmap, fll->base + 1,
+ ARIZONA_FLL1_FREERUN, 0);
+
return 0;
}
EXPORT_SYMBOL_GPL(arizona_init_fll);
diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c
index e6cefe1ac67..d8c65f57465 100644
--- a/sound/soc/codecs/wm2200.c
+++ b/sound/soc/codecs/wm2200.c
@@ -1019,8 +1019,6 @@ static const char *wm2200_mixer_texts[] = {
"EQR",
"LHPF1",
"LHPF2",
- "LHPF3",
- "LHPF4",
"DSP1.1",
"DSP1.2",
"DSP1.3",
@@ -1053,7 +1051,6 @@ static int wm2200_mixer_values[] = {
0x25,
0x50, /* EQ */
0x51,
- 0x52,
0x60, /* LHPF1 */
0x61, /* LHPF2 */
0x68, /* DSP1 */
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 7a9048dad1c..1440b3f9b7b 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -896,8 +896,7 @@ static const unsigned int wm5102_aec_loopback_values[] = {
static const struct soc_enum wm5102_aec_loopback =
SOC_VALUE_ENUM_SINGLE(ARIZONA_DAC_AEC_CONTROL_1,
- ARIZONA_AEC_LOOPBACK_SRC_SHIFT,
- ARIZONA_AEC_LOOPBACK_SRC_MASK,
+ ARIZONA_AEC_LOOPBACK_SRC_SHIFT, 0xf,
ARRAY_SIZE(wm5102_aec_loopback_texts),
wm5102_aec_loopback_texts,
wm5102_aec_loopback_values);
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index ae80c8c2853..7a090968c4f 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -344,8 +344,7 @@ static const unsigned int wm5110_aec_loopback_values[] = {
static const struct soc_enum wm5110_aec_loopback =
SOC_VALUE_ENUM_SINGLE(ARIZONA_DAC_AEC_CONTROL_1,
- ARIZONA_AEC_LOOPBACK_SRC_SHIFT,
- ARIZONA_AEC_LOOPBACK_SRC_MASK,
+ ARIZONA_AEC_LOOPBACK_SRC_SHIFT, 0xf,
ARRAY_SIZE(wm5110_aec_loopback_texts),
wm5110_aec_loopback_texts,
wm5110_aec_loopback_values);
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 7b198c38f3e..93d03bc0661 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -103,9 +103,12 @@
#define ADSP1_START_SHIFT 0 /* DSP1_START */
#define ADSP1_START_WIDTH 1 /* DSP1_START */
-#define ADSP2_CONTROL 0
-#define ADSP2_CLOCKING 1
-#define ADSP2_STATUS1 4
+#define ADSP2_CONTROL 0x0
+#define ADSP2_CLOCKING 0x1
+#define ADSP2_STATUS1 0x4
+#define ADSP2_WDMA_CONFIG_1 0x30
+#define ADSP2_WDMA_CONFIG_2 0x31
+#define ADSP2_RDMA_CONFIG_1 0x34
/*
* ADSP2 Control
@@ -324,7 +327,7 @@ static int wm_adsp_load(struct wm_adsp *dsp)
if (reg) {
buf = kmemdup(region->data, le32_to_cpu(region->len),
- GFP_KERNEL);
+ GFP_KERNEL | GFP_DMA);
if (!buf) {
adsp_err(dsp, "Out of memory\n");
return -ENOMEM;
@@ -396,7 +399,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
hdr = (void*)&firmware->data[0];
if (memcmp(hdr->magic, "WMDR", 4) != 0) {
adsp_err(dsp, "%s: invalid magic\n", file);
- return -EINVAL;
+ goto out_fw;
}
adsp_dbg(dsp, "%s: v%d.%d.%d\n", file,
@@ -439,7 +442,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
if (reg) {
buf = kmemdup(blk->data, le32_to_cpu(blk->len),
- GFP_KERNEL);
+ GFP_KERNEL | GFP_DMA);
if (!buf) {
adsp_err(dsp, "Out of memory\n");
return -ENOMEM;
@@ -642,6 +645,11 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w,
ADSP2_SYS_ENA | ADSP2_CORE_ENA |
ADSP2_START, 0);
+ /* Make sure DMAs are quiesced */
+ regmap_write(dsp->regmap, dsp->base + ADSP2_WDMA_CONFIG_1, 0);
+ regmap_write(dsp->regmap, dsp->base + ADSP2_WDMA_CONFIG_2, 0);
+ regmap_write(dsp->regmap, dsp->base + ADSP2_RDMA_CONFIG_1, 0);
+
if (dsp->dvfs) {
ret = regulator_set_voltage(dsp->dvfs, 1200000,
1800000);
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 55e2bf652be..9321e5c9d8c 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -626,7 +626,7 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev,
int word_length)
{
u32 fmt;
- u32 rotate = (32 - word_length) / 4;
+ u32 rotate = (word_length / 4) & 0x7;
u32 mask = (1ULL << word_length) - 1;
/*
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index 1aa51300c56..deb30d59965 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -210,15 +210,19 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream,
switch (config->chan_nr) {
case EIGHT_CHANNEL_SUPPORT:
ch_reg = 3;
+ break;
case SIX_CHANNEL_SUPPORT:
ch_reg = 2;
+ break;
case FOUR_CHANNEL_SUPPORT:
ch_reg = 1;
+ break;
case TWO_CHANNEL_SUPPORT:
ch_reg = 0;
break;
default:
dev_err(dev->dev, "channel not supported\n");
+ return -EINVAL;
}
i2s_disable_channels(dev, substream->stream);
diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c
index bf363d8d044..500f8ce55d7 100644
--- a/sound/soc/fsl/imx-pcm-dma.c
+++ b/sound/soc/fsl/imx-pcm-dma.c
@@ -154,26 +154,7 @@ static struct snd_soc_platform_driver imx_soc_platform_mx2 = {
.pcm_free = imx_pcm_free,
};
-static int imx_soc_platform_probe(struct platform_device *pdev)
+int imx_pcm_dma_init(struct platform_device *pdev)
{
return snd_soc_register_platform(&pdev->dev, &imx_soc_platform_mx2);
}
-
-static int imx_soc_platform_remove(struct platform_device *pdev)
-{
- snd_soc_unregister_platform(&pdev->dev);
- return 0;
-}
-
-static struct platform_driver imx_pcm_driver = {
- .driver = {
- .name = "imx-pcm-audio",
- .owner = THIS_MODULE,
- },
- .probe = imx_soc_platform_probe,
- .remove = imx_soc_platform_remove,
-};
-
-module_platform_driver(imx_pcm_driver);
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:imx-pcm-audio");
diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c
index 5ec362ae4d0..920f945cb2f 100644
--- a/sound/soc/fsl/imx-pcm-fiq.c
+++ b/sound/soc/fsl/imx-pcm-fiq.c
@@ -281,7 +281,7 @@ static struct snd_soc_platform_driver imx_soc_platform_fiq = {
.pcm_free = imx_pcm_fiq_free,
};
-static int imx_soc_platform_probe(struct platform_device *pdev)
+int imx_pcm_fiq_init(struct platform_device *pdev)
{
struct imx_ssi *ssi = platform_get_drvdata(pdev);
int ret;
@@ -314,23 +314,3 @@ failed_register:
return ret;
}
-
-static int imx_soc_platform_remove(struct platform_device *pdev)
-{
- snd_soc_unregister_platform(&pdev->dev);
- return 0;
-}
-
-static struct platform_driver imx_pcm_driver = {
- .driver = {
- .name = "imx-fiq-pcm-audio",
- .owner = THIS_MODULE,
- },
-
- .probe = imx_soc_platform_probe,
- .remove = imx_soc_platform_remove,
-};
-
-module_platform_driver(imx_pcm_driver);
-
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/imx-pcm.c b/sound/soc/fsl/imx-pcm.c
index d5cd9eff3b4..0d0625bfcb6 100644
--- a/sound/soc/fsl/imx-pcm.c
+++ b/sound/soc/fsl/imx-pcm.c
@@ -104,6 +104,38 @@ void imx_pcm_free(struct snd_pcm *pcm)
}
EXPORT_SYMBOL_GPL(imx_pcm_free);
+static int imx_pcm_probe(struct platform_device *pdev)
+{
+ if (strcmp(pdev->id_entry->name, "imx-fiq-pcm-audio") == 0)
+ return imx_pcm_fiq_init(pdev);
+
+ return imx_pcm_dma_init(pdev);
+}
+
+static int imx_pcm_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_device_id imx_pcm_devtype[] = {
+ { .name = "imx-pcm-audio", },
+ { .name = "imx-fiq-pcm-audio", },
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(platform, imx_pcm_devtype);
+
+static struct platform_driver imx_pcm_driver = {
+ .driver = {
+ .name = "imx-pcm",
+ .owner = THIS_MODULE,
+ },
+ .id_table = imx_pcm_devtype,
+ .probe = imx_pcm_probe,
+ .remove = imx_pcm_remove,
+};
+module_platform_driver(imx_pcm_driver);
+
MODULE_DESCRIPTION("Freescale i.MX PCM driver");
MODULE_AUTHOR("Sascha Hauer <s.hauer@pengutronix.de>");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h
index 83c0ed7d55c..5ae13a13a35 100644
--- a/sound/soc/fsl/imx-pcm.h
+++ b/sound/soc/fsl/imx-pcm.h
@@ -30,4 +30,22 @@ int snd_imx_pcm_mmap(struct snd_pcm_substream *substream,
int imx_pcm_new(struct snd_soc_pcm_runtime *rtd);
void imx_pcm_free(struct snd_pcm *pcm);
+#ifdef CONFIG_SND_SOC_IMX_PCM_DMA
+int imx_pcm_dma_init(struct platform_device *pdev);
+#else
+static inline int imx_pcm_dma_init(struct platform_device *pdev)
+{
+ return -ENODEV;
+}
+#endif
+
+#ifdef CONFIG_SND_SOC_IMX_PCM_FIQ
+int imx_pcm_fiq_init(struct platform_device *pdev);
+#else
+static inline int imx_pcm_fiq_init(struct platform_device *pdev)
+{
+ return -ENODEV;
+}
+#endif
+
#endif /* _IMX_PCM_H */
diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c
index ee10e8704e9..13f6dd1ceb0 100644
--- a/sound/soc/samsung/s3c24xx-i2s.c
+++ b/sound/soc/samsung/s3c24xx-i2s.c
@@ -469,7 +469,7 @@ static int s3c24xx_iis_dev_probe(struct platform_device *pdev)
{
int ret = 0;
- ret = s3c_i2sv2_register_dai(&pdev->dev, -1, &s3c2412_i2s_dai);
+ ret = snd_soc_register_dai(&pdev->dev, &s3c24xx_i2s_dai);
if (ret) {
pr_err("failed to register the dai\n");
return ret;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 1e36bc81e5a..258acadb9e7 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1023,7 +1023,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w,
if (SND_SOC_DAPM_EVENT_ON(event)) {
if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) {
- ret = regulator_allow_bypass(w->regulator, true);
+ ret = regulator_allow_bypass(w->regulator, false);
if (ret != 0)
dev_warn(w->dapm->dev,
"ASoC: Failed to bypass %s: %d\n",
@@ -1033,7 +1033,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w,
return regulator_enable(w->regulator);
} else {
if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) {
- ret = regulator_allow_bypass(w->regulator, false);
+ ret = regulator_allow_bypass(w->regulator, true);
if (ret != 0)
dev_warn(w->dapm->dev,
"ASoC: Failed to unbypass %s: %d\n",
@@ -3039,6 +3039,14 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
w->name, ret);
return NULL;
}
+
+ if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) {
+ ret = regulator_allow_bypass(w->regulator, true);
+ if (ret != 0)
+ dev_warn(w->dapm->dev,
+ "ASoC: Failed to unbypass %s: %d\n",
+ w->name, ret);
+ }
break;
case snd_soc_dapm_clock_supply:
#ifdef CONFIG_CLKDEV_LOOKUP
generated by cgit v1.2.3-70-g09d2 (git 2.46.0) at 2025-01-24 20:01:59 +0000