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authorMark Brown <broonie@linaro.org>2013-11-15 10:20:08 +0000
committerMark Brown <broonie@linaro.org>2013-11-15 10:20:08 +0000
commit514918f0d1c3e7cf78fe2acbe80f7b616065e210 (patch)
treef6a918d115acf0347ee69b2b90aada657a68ef96 /sound/soc
parent9450d92c8c647752586eeeb7927da8a50a691c96 (diff)
parentfc7dc61d9a87011aaf8a6eb3144ebf9552adf5d2 (diff)
Merge remote-tracking branch 'asoc/fix/fsl' into asoc-linus
Diffstat (limited to 'sound/soc')
-rw-r--r--sound/soc/codecs/ab8500-codec.c14
-rw-r--r--sound/soc/codecs/ak4642.c2
-rw-r--r--sound/soc/codecs/cs42l52.c93
-rw-r--r--sound/soc/codecs/cs42l52.h2
-rw-r--r--sound/soc/codecs/ml26124.c2
-rw-r--r--sound/soc/codecs/rt5640.c7
-rw-r--r--sound/soc/codecs/wm0010.c8
-rw-r--r--sound/soc/codecs/wm5110.c12
-rw-r--r--sound/soc/codecs/wm8962.c22
-rw-r--r--sound/soc/codecs/wm8996.c2
-rw-r--r--sound/soc/codecs/wm_adsp.c27
-rw-r--r--sound/soc/fsl/eukrea-tlv320.c15
-rw-r--r--sound/soc/fsl/fsl_spdif.c4
-rw-r--r--sound/soc/fsl/imx-pcm-fiq.c29
-rw-r--r--sound/soc/kirkwood/kirkwood-i2s.c2
-rw-r--r--sound/soc/samsung/i2s.c2
-rw-r--r--sound/soc/sh/rcar/core.c31
-rw-r--r--sound/soc/soc-core.c5
-rw-r--r--sound/soc/soc-pcm.c52
19 files changed, 207 insertions, 124 deletions
diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c
index a0394a8f225..21ae8d4fdbf 100644
--- a/sound/soc/codecs/ab8500-codec.c
+++ b/sound/soc/codecs/ab8500-codec.c
@@ -2300,17 +2300,17 @@ static int ab8500_codec_set_dai_tdm_slot(struct snd_soc_dai *dai,
case 0:
break;
case 1:
- slot = find_first_bit((unsigned long *)&tx_mask, 32);
+ slot = ffs(tx_mask);
snd_soc_update_bits(codec, AB8500_DASLOTCONF1, mask, slot);
snd_soc_update_bits(codec, AB8500_DASLOTCONF3, mask, slot);
snd_soc_update_bits(codec, AB8500_DASLOTCONF2, mask, slot);
snd_soc_update_bits(codec, AB8500_DASLOTCONF4, mask, slot);
break;
case 2:
- slot = find_first_bit((unsigned long *)&tx_mask, 32);
+ slot = ffs(tx_mask);
snd_soc_update_bits(codec, AB8500_DASLOTCONF1, mask, slot);
snd_soc_update_bits(codec, AB8500_DASLOTCONF3, mask, slot);
- slot = find_next_bit((unsigned long *)&tx_mask, 32, slot + 1);
+ slot = fls(tx_mask);
snd_soc_update_bits(codec, AB8500_DASLOTCONF2, mask, slot);
snd_soc_update_bits(codec, AB8500_DASLOTCONF4, mask, slot);
break;
@@ -2341,18 +2341,18 @@ static int ab8500_codec_set_dai_tdm_slot(struct snd_soc_dai *dai,
case 0:
break;
case 1:
- slot = find_first_bit((unsigned long *)&rx_mask, 32);
+ slot = ffs(rx_mask);
snd_soc_update_bits(codec, AB8500_ADSLOTSEL(slot),
AB8500_MASK_SLOT(slot),
AB8500_ADSLOTSELX_AD_OUT_TO_SLOT(AB8500_AD_OUT3, slot));
break;
case 2:
- slot = find_first_bit((unsigned long *)&rx_mask, 32);
+ slot = ffs(rx_mask);
snd_soc_update_bits(codec,
AB8500_ADSLOTSEL(slot),
AB8500_MASK_SLOT(slot),
AB8500_ADSLOTSELX_AD_OUT_TO_SLOT(AB8500_AD_OUT3, slot));
- slot = find_next_bit((unsigned long *)&rx_mask, 32, slot + 1);
+ slot = fls(rx_mask);
snd_soc_update_bits(codec,
AB8500_ADSLOTSEL(slot),
AB8500_MASK_SLOT(slot),
@@ -2575,6 +2575,8 @@ static int ab8500_codec_driver_probe(struct platform_device *pdev)
/* Create driver private-data struct */
drvdata = devm_kzalloc(&pdev->dev, sizeof(struct ab8500_codec_drvdata),
GFP_KERNEL);
+ if (!drvdata)
+ return -ENOMEM;
drvdata->sid_status = SID_UNCONFIGURED;
drvdata->anc_status = ANC_UNCONFIGURED;
dev_set_drvdata(&pdev->dev, drvdata);
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 21c35ed778c..090d499bb7e 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -257,7 +257,7 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream,
* This operation came from example code of
* "ASAHI KASEI AK4642" (japanese) manual p94.
*/
- snd_soc_write(codec, SG_SL1, PMMP | MGAIN0);
+ snd_soc_update_bits(codec, SG_SL1, PMMP | MGAIN0, PMMP | MGAIN0);
snd_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3));
snd_soc_write(codec, ALC_CTL1, ALC | LMTH0);
snd_soc_update_bits(codec, PW_MGMT1, PMADL, PMADL);
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index be2ba1b6fe4..8b427c97708 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -17,6 +17,7 @@
#include <linux/kernel.h>
#include <linux/init.h>
#include <linux/delay.h>
+#include <linux/gpio.h>
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/input.h>
@@ -1116,40 +1117,6 @@ static int cs42l52_probe(struct snd_soc_codec *codec)
cs42l52->sysclk = CS42L52_DEFAULT_CLK;
cs42l52->config.format = CS42L52_DEFAULT_FORMAT;
- /* Set Platform MICx CFG */
- snd_soc_update_bits(codec, CS42L52_MICA_CTL,
- CS42L52_MIC_CTL_TYPE_MASK,
- cs42l52->pdata.mica_cfg <<
- CS42L52_MIC_CTL_TYPE_SHIFT);
-
- snd_soc_update_bits(codec, CS42L52_MICB_CTL,
- CS42L52_MIC_CTL_TYPE_MASK,
- cs42l52->pdata.micb_cfg <<
- CS42L52_MIC_CTL_TYPE_SHIFT);
-
- /* if Single Ended, Get Mic_Select */
- if (cs42l52->pdata.mica_cfg)
- snd_soc_update_bits(codec, CS42L52_MICA_CTL,
- CS42L52_MIC_CTL_MIC_SEL_MASK,
- cs42l52->pdata.mica_sel <<
- CS42L52_MIC_CTL_MIC_SEL_SHIFT);
- if (cs42l52->pdata.micb_cfg)
- snd_soc_update_bits(codec, CS42L52_MICB_CTL,
- CS42L52_MIC_CTL_MIC_SEL_MASK,
- cs42l52->pdata.micb_sel <<
- CS42L52_MIC_CTL_MIC_SEL_SHIFT);
-
- /* Set Platform Charge Pump Freq */
- snd_soc_update_bits(codec, CS42L52_CHARGE_PUMP,
- CS42L52_CHARGE_PUMP_MASK,
- cs42l52->pdata.chgfreq <<
- CS42L52_CHARGE_PUMP_SHIFT);
-
- /* Set Platform Bias Level */
- snd_soc_update_bits(codec, CS42L52_IFACE_CTL2,
- CS42L52_IFACE_CTL2_BIAS_LVL,
- cs42l52->pdata.micbias_lvl);
-
return ret;
}
@@ -1205,6 +1172,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client,
const struct i2c_device_id *id)
{
struct cs42l52_private *cs42l52;
+ struct cs42l52_platform_data *pdata = dev_get_platdata(&i2c_client->dev);
int ret;
unsigned int devid = 0;
unsigned int reg;
@@ -1222,11 +1190,22 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client,
return ret;
}
- i2c_set_clientdata(i2c_client, cs42l52);
+ if (pdata)
+ cs42l52->pdata = *pdata;
+
+ if (cs42l52->pdata.reset_gpio) {
+ ret = gpio_request_one(cs42l52->pdata.reset_gpio,
+ GPIOF_OUT_INIT_HIGH, "CS42L52 /RST");
+ if (ret < 0) {
+ dev_err(&i2c_client->dev, "Failed to request /RST %d: %d\n",
+ cs42l52->pdata.reset_gpio, ret);
+ return ret;
+ }
+ gpio_set_value_cansleep(cs42l52->pdata.reset_gpio, 0);
+ gpio_set_value_cansleep(cs42l52->pdata.reset_gpio, 1);
+ }
- if (dev_get_platdata(&i2c_client->dev))
- memcpy(&cs42l52->pdata, dev_get_platdata(&i2c_client->dev),
- sizeof(cs42l52->pdata));
+ i2c_set_clientdata(i2c_client, cs42l52);
ret = regmap_register_patch(cs42l52->regmap, cs42l52_threshold_patch,
ARRAY_SIZE(cs42l52_threshold_patch));
@@ -1244,7 +1223,43 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client,
return ret;
}
- regcache_cache_only(cs42l52->regmap, true);
+ dev_info(&i2c_client->dev, "Cirrus Logic CS42L52, Revision: %02X\n",
+ reg & 0xFF);
+
+ /* Set Platform Data */
+ if (cs42l52->pdata.mica_cfg)
+ regmap_update_bits(cs42l52->regmap, CS42L52_MICA_CTL,
+ CS42L52_MIC_CTL_TYPE_MASK,
+ cs42l52->pdata.mica_cfg <<
+ CS42L52_MIC_CTL_TYPE_SHIFT);
+
+ if (cs42l52->pdata.micb_cfg)
+ regmap_update_bits(cs42l52->regmap, CS42L52_MICB_CTL,
+ CS42L52_MIC_CTL_TYPE_MASK,
+ cs42l52->pdata.micb_cfg <<
+ CS42L52_MIC_CTL_TYPE_SHIFT);
+
+ if (cs42l52->pdata.mica_sel)
+ regmap_update_bits(cs42l52->regmap, CS42L52_MICA_CTL,
+ CS42L52_MIC_CTL_MIC_SEL_MASK,
+ cs42l52->pdata.mica_sel <<
+ CS42L52_MIC_CTL_MIC_SEL_SHIFT);
+ if (cs42l52->pdata.micb_sel)
+ regmap_update_bits(cs42l52->regmap, CS42L52_MICB_CTL,
+ CS42L52_MIC_CTL_MIC_SEL_MASK,
+ cs42l52->pdata.micb_sel <<
+ CS42L52_MIC_CTL_MIC_SEL_SHIFT);
+
+ if (cs42l52->pdata.chgfreq)
+ regmap_update_bits(cs42l52->regmap, CS42L52_CHARGE_PUMP,
+ CS42L52_CHARGE_PUMP_MASK,
+ cs42l52->pdata.chgfreq <<
+ CS42L52_CHARGE_PUMP_SHIFT);
+
+ if (cs42l52->pdata.micbias_lvl)
+ regmap_update_bits(cs42l52->regmap, CS42L52_IFACE_CTL2,
+ CS42L52_IFACE_CTL2_BIAS_LVL,
+ cs42l52->pdata.micbias_lvl);
ret = snd_soc_register_codec(&i2c_client->dev,
&soc_codec_dev_cs42l52, &cs42l52_dai, 1);
diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h
index a935d7381af..6fb8f00f419 100644
--- a/sound/soc/codecs/cs42l52.h
+++ b/sound/soc/codecs/cs42l52.h
@@ -269,6 +269,6 @@
#define CS42L52_FIX_BITS1 0x3E
#define CS42L52_FIX_BITS2 0x47
-#define CS42L52_MAX_REGISTER 0x34
+#define CS42L52_MAX_REGISTER 0x47
#endif
diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c
index 26118828782..185fa3bc305 100644
--- a/sound/soc/codecs/ml26124.c
+++ b/sound/soc/codecs/ml26124.c
@@ -342,6 +342,8 @@ static int ml26124_hw_params(struct snd_pcm_substream *substream,
struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec);
int i = get_coeff(priv->mclk, params_rate(hw_params));
+ if (i < 0)
+ return i;
priv->substream = substream;
priv->rate = params_rate(hw_params);
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 4d041d376f3..a3fb4117963 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -1604,8 +1604,8 @@ static int rt5640_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
- unsigned int val_len = 0, val_clk, mask_clk, dai_sel;
- int pre_div, bclk_ms, frame_size;
+ unsigned int val_len = 0, val_clk, mask_clk;
+ int dai_sel, pre_div, bclk_ms, frame_size;
rt5640->lrck[dai->id] = params_rate(params);
pre_div = get_clk_info(rt5640->sysclk, rt5640->lrck[dai->id]);
@@ -1675,7 +1675,8 @@ static int rt5640_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
{
struct snd_soc_codec *codec = dai->codec;
struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
- unsigned int reg_val = 0, dai_sel;
+ unsigned int reg_val = 0;
+ int dai_sel;
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c
index d5ebcb00019..bf7804a1286 100644
--- a/sound/soc/codecs/wm0010.c
+++ b/sound/soc/codecs/wm0010.c
@@ -793,11 +793,11 @@ static int wm0010_set_sysclk(struct snd_soc_codec *codec, int source,
wm0010->max_spi_freq = 0;
} else {
for (i = 0; i < ARRAY_SIZE(pll_clock_map); i++)
- if (freq >= pll_clock_map[i].max_sysclk)
+ if (freq >= pll_clock_map[i].max_sysclk) {
+ wm0010->max_spi_freq = pll_clock_map[i].max_pll_spi_speed;
+ wm0010->pll_clkctrl1 = pll_clock_map[i].pll_clkctrl1;
break;
-
- wm0010->max_spi_freq = pll_clock_map[i].max_pll_spi_speed;
- wm0010->pll_clkctrl1 = pll_clock_map[i].pll_clkctrl1;
+ }
}
return 0;
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index b67ba33546d..f2d1094424b 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -983,24 +983,36 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = {
ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"),
ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"),
+ { "AEC Loopback", "HPOUT1L", "OUT1L" },
+ { "AEC Loopback", "HPOUT1R", "OUT1R" },
{ "HPOUT1L", NULL, "OUT1L" },
{ "HPOUT1R", NULL, "OUT1R" },
+ { "AEC Loopback", "HPOUT2L", "OUT2L" },
+ { "AEC Loopback", "HPOUT2R", "OUT2R" },
{ "HPOUT2L", NULL, "OUT2L" },
{ "HPOUT2R", NULL, "OUT2R" },
+ { "AEC Loopback", "HPOUT3L", "OUT3L" },
+ { "AEC Loopback", "HPOUT3R", "OUT3R" },
{ "HPOUT3L", NULL, "OUT3L" },
{ "HPOUT3R", NULL, "OUT3L" },
+ { "AEC Loopback", "SPKOUTL", "OUT4L" },
{ "SPKOUTLN", NULL, "OUT4L" },
{ "SPKOUTLP", NULL, "OUT4L" },
+ { "AEC Loopback", "SPKOUTR", "OUT4R" },
{ "SPKOUTRN", NULL, "OUT4R" },
{ "SPKOUTRP", NULL, "OUT4R" },
+ { "AEC Loopback", "SPKDAT1L", "OUT5L" },
+ { "AEC Loopback", "SPKDAT1R", "OUT5R" },
{ "SPKDAT1L", NULL, "OUT5L" },
{ "SPKDAT1R", NULL, "OUT5R" },
+ { "AEC Loopback", "SPKDAT2L", "OUT6L" },
+ { "AEC Loopback", "SPKDAT2R", "OUT6R" },
{ "SPKDAT2L", NULL, "OUT6L" },
{ "SPKDAT2R", NULL, "OUT6R" },
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 2bf9ee7c540..3a2f96c5442 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -1758,6 +1758,9 @@ SOC_DOUBLE_R_TLV("EQ4 Volume", WM8962_EQ3, WM8962_EQ23,
WM8962_EQL_B4_GAIN_SHIFT, 31, 0, eq_tlv),
SOC_DOUBLE_R_TLV("EQ5 Volume", WM8962_EQ3, WM8962_EQ23,
WM8962_EQL_B5_GAIN_SHIFT, 31, 0, eq_tlv),
+SND_SOC_BYTES("EQL Coefficients", WM8962_EQ4, 18),
+SND_SOC_BYTES("EQR Coefficients", WM8962_EQ24, 18),
+
SOC_SINGLE("3D Switch", WM8962_THREED1, 0, 1, 0),
SND_SOC_BYTES_MASK("3D Coefficients", WM8962_THREED1, 4, WM8962_THREED_ENA),
@@ -1775,6 +1778,11 @@ WM8962_DSP2_ENABLE("HPF2 Switch", WM8962_HPF2_ENA_SHIFT),
SND_SOC_BYTES("HPF Coefficients", WM8962_LHPF2, 1),
WM8962_DSP2_ENABLE("HD Bass Switch", WM8962_HDBASS_ENA_SHIFT),
SND_SOC_BYTES("HD Bass Coefficients", WM8962_HDBASS_AI_1, 30),
+
+SOC_DOUBLE("ALC Switch", WM8962_ALC1, WM8962_ALCL_ENA_SHIFT,
+ WM8962_ALCR_ENA_SHIFT, 1, 0),
+SND_SOC_BYTES_MASK("ALC Coefficients", WM8962_ALC1, 4,
+ WM8962_ALCL_ENA_MASK | WM8962_ALCR_ENA_MASK),
};
static const struct snd_kcontrol_new wm8962_spk_mono_controls[] = {
@@ -3616,28 +3624,28 @@ static int wm8962_i2c_probe(struct i2c_client *i2c,
0);
/* Apply static configuration for GPIOs */
- for (i = 0; i < ARRAY_SIZE(pdata->gpio_init); i++)
- if (pdata->gpio_init[i]) {
+ for (i = 0; i < ARRAY_SIZE(wm8962->pdata.gpio_init); i++)
+ if (wm8962->pdata.gpio_init[i]) {
wm8962_set_gpio_mode(wm8962, i + 1);
regmap_write(wm8962->regmap, 0x200 + i,
- pdata->gpio_init[i] & 0xffff);
+ wm8962->pdata.gpio_init[i] & 0xffff);
}
/* Put the speakers into mono mode? */
- if (pdata->spk_mono)
+ if (wm8962->pdata.spk_mono)
regmap_update_bits(wm8962->regmap, WM8962_CLASS_D_CONTROL_2,
WM8962_SPK_MONO_MASK, WM8962_SPK_MONO);
/* Micbias setup, detection enable and detection
* threasholds. */
- if (pdata->mic_cfg)
+ if (wm8962->pdata.mic_cfg)
regmap_update_bits(wm8962->regmap, WM8962_ADDITIONAL_CONTROL_4,
WM8962_MICDET_ENA |
WM8962_MICDET_THR_MASK |
WM8962_MICSHORT_THR_MASK |
WM8962_MICBIAS_LVL,
- pdata->mic_cfg);
+ wm8962->pdata.mic_cfg);
/* Latch volume update bits */
regmap_update_bits(wm8962->regmap, WM8962_LEFT_INPUT_VOLUME,
@@ -3682,7 +3690,7 @@ static int wm8962_i2c_probe(struct i2c_client *i2c,
}
if (wm8962->irq) {
- if (pdata->irq_active_low) {
+ if (wm8962->pdata.irq_active_low) {
trigger = IRQF_TRIGGER_LOW;
irq_pol = WM8962_IRQ_POL;
} else {
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index 46fe83d2b22..b70379ebd14 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -438,6 +438,8 @@ static int wm8996_get_retune_mobile_enum(struct snd_kcontrol *kcontrol,
struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec);
int block = wm8996_get_retune_mobile_block(kcontrol->id.name);
+ if (block < 0)
+ return block;
ucontrol->value.enumerated.item[0] = wm8996->retune_mobile_cfg[block];
return 0;
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index b38f3506418..53b6033658a 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -396,11 +396,12 @@ static int wm_coeff_write_control(struct snd_kcontrol *kcontrol,
ret = regmap_raw_write(adsp->regmap, reg, scratch,
ctl->len);
if (ret) {
- adsp_err(adsp, "Failed to write %zu bytes to %x\n",
- ctl->len, reg);
+ adsp_err(adsp, "Failed to write %zu bytes to %x: %d\n",
+ ctl->len, reg, ret);
kfree(scratch);
return ret;
}
+ adsp_dbg(adsp, "Wrote %zu bytes to %x\n", ctl->len, reg);
kfree(scratch);
@@ -450,11 +451,12 @@ static int wm_coeff_read_control(struct snd_kcontrol *kcontrol,
ret = regmap_raw_read(adsp->regmap, reg, scratch, ctl->len);
if (ret) {
- adsp_err(adsp, "Failed to read %zu bytes from %x\n",
- ctl->len, reg);
+ adsp_err(adsp, "Failed to read %zu bytes from %x: %d\n",
+ ctl->len, reg, ret);
kfree(scratch);
return ret;
}
+ adsp_dbg(adsp, "Read %zu bytes from %x\n", ctl->len, reg);
memcpy(buf, scratch, ctl->len);
kfree(scratch);
@@ -568,6 +570,7 @@ static int wm_adsp_load(struct wm_adsp *dsp)
file, header->ver);
goto out_fw;
}
+ adsp_info(dsp, "Firmware version: %d\n", header->ver);
if (header->core != dsp->type) {
adsp_err(dsp, "%s: invalid core %d != %d\n",
@@ -689,7 +692,8 @@ static int wm_adsp_load(struct wm_adsp *dsp)
&buf_list);
if (!buf) {
adsp_err(dsp, "Out of memory\n");
- return -ENOMEM;
+ ret = -ENOMEM;
+ goto out_fw;
}
ret = regmap_raw_write_async(regmap, reg, buf->buf,
@@ -1313,8 +1317,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
le32_to_cpu(blk->len));
if (ret != 0) {
adsp_err(dsp,
- "%s.%d: Failed to write to %x in %s\n",
- file, blocks, reg, region_name);
+ "%s.%d: Failed to write to %x in %s: %d\n",
+ file, blocks, reg, region_name, ret);
}
}
@@ -1358,6 +1362,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w,
struct snd_soc_codec *codec = w->codec;
struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec);
struct wm_adsp *dsp = &dsps[w->shift];
+ struct wm_adsp_alg_region *alg_region;
struct wm_coeff_ctl *ctl;
int ret;
int val;
@@ -1435,6 +1440,14 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w,
list_for_each_entry(ctl, &dsp->ctl_list, list)
ctl->enabled = 0;
+
+ while (!list_empty(&dsp->alg_regions)) {
+ alg_region = list_first_entry(&dsp->alg_regions,
+ struct wm_adsp_alg_region,
+ list);
+ list_del(&alg_region->list);
+ kfree(alg_region);
+ }
break;
default:
diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c
index 9a4a0ca2c1d..5983740be12 100644
--- a/sound/soc/fsl/eukrea-tlv320.c
+++ b/sound/soc/fsl/eukrea-tlv320.c
@@ -42,7 +42,8 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream,
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM);
if (ret) {
- pr_err("%s: failed set cpu dai format\n", __func__);
+ dev_err(cpu_dai->dev,
+ "Failed to set the cpu dai format.\n");
return ret;
}
@@ -50,14 +51,16 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream,
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM);
if (ret) {
- pr_err("%s: failed set codec dai format\n", __func__);
+ 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) {
- pr_err("%s: failed setting codec sysclk\n", __func__);
+ dev_err(cpu_dai->dev,
+ "Failed to set the codec sysclk.\n");
return ret;
}
snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0);
@@ -65,7 +68,8 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream,
ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0,
SND_SOC_CLOCK_IN);
if (ret) {
- pr_err("can't set CPU system clock IMX_SSP_SYS_CLK\n");
+ dev_err(cpu_dai->dev,
+ "Can't set the IMX_SSP_SYS_CLK CPU system clock.\n");
return ret;
}
@@ -155,7 +159,8 @@ static struct platform_driver eukrea_tlv320_driver = {
.owner = THIS_MODULE,
},
.probe = eukrea_tlv320_probe,
- .remove = eukrea_tlv320_remove,};
+ .remove = eukrea_tlv320_remove,
+};
module_platform_driver(eukrea_tlv320_driver);
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index ff1f34766ee..76c742a09ef 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -1107,9 +1107,9 @@ static int fsl_spdif_probe(struct platform_device *pdev)
/* Get the addresses and IRQ */
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (IS_ERR(res)) {
+ if (!res) {
dev_err(&pdev->dev, "could not determine device resources\n");
- return PTR_ERR(res);
+ return -ENXIO;
}
regs = devm_ioremap_resource(&pdev->dev, res);
diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c
index 34043c55f2a..2fc872b2def 100644
--- a/sound/soc/fsl/imx-pcm-fiq.c
+++ b/sound/soc/fsl/imx-pcm-fiq.c
@@ -44,7 +44,8 @@ struct imx_pcm_runtime_data {
struct hrtimer hrt;
int poll_time_ns;
struct snd_pcm_substream *substream;
- atomic_t running;
+ atomic_t playing;
+ atomic_t capturing;
};
static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt)
@@ -56,7 +57,7 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt)
struct pt_regs regs;
unsigned long delta;
- if (!atomic_read(&iprtd->running))
+ if (!atomic_read(&iprtd->playing) && !atomic_read(&iprtd->capturing))
return HRTIMER_NORESTART;
get_fiq_regs(&regs);
@@ -124,7 +125,6 @@ static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream)
return 0;
}
-static int fiq_enable;
static int imx_pcm_fiq;
static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
@@ -136,23 +136,27 @@ static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- atomic_set(&iprtd->running, 1);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ atomic_set(&iprtd->playing, 1);
+ else
+ atomic_set(&iprtd->capturing, 1);
hrtimer_start(&iprtd->hrt, ns_to_ktime(iprtd->poll_time_ns),
HRTIMER_MODE_REL);
- if (++fiq_enable == 1)
- enable_fiq(imx_pcm_fiq);
-
+ enable_fiq(imx_pcm_fiq);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- atomic_set(&iprtd->running, 0);
-
- if (--fiq_enable == 0)
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ atomic_set(&iprtd->playing, 0);
+ else
+ atomic_set(&iprtd->capturing, 0);
+ if (!atomic_read(&iprtd->playing) &&
+ !atomic_read(&iprtd->capturing))
disable_fiq(imx_pcm_fiq);
-
break;
+
default:
return -EINVAL;
}
@@ -200,7 +204,8 @@ static int snd_imx_open(struct snd_pcm_substream *substream)
iprtd->substream = substream;
- atomic_set(&iprtd->running, 0);
+ atomic_set(&iprtd->playing, 0);
+ atomic_set(&iprtd->capturing, 0);
hrtimer_init(&iprtd->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL);
iprtd->hrt.function = snd_hrtimer_callback;
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index 9ec38d15df9..d34d91743e3 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -568,7 +568,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
} else {
dev_info(&pdev->dev, "found external clock\n");
clk_prepare_enable(priv->extclk);
- soc_dai = &kirkwood_i2s_dai_extclk;
+ soc_dai = kirkwood_i2s_dai_extclk;
}
}
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index 2c4d2505a19..a5cbdb4f165 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -1068,7 +1068,7 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec)
dev_set_drvdata(&i2s->pdev->dev, i2s);
} else { /* Create a new platform_device for Secondary */
i2s->pdev = platform_device_alloc("samsung-i2s-sec", -1);
- if (IS_ERR(i2s->pdev))
+ if (!i2s->pdev)
return NULL;
i2s->pdev->dev.parent = &pdev->dev;
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index b234ed66307..78c35b44fc0 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -94,6 +94,7 @@
*
*/
#include <linux/pm_runtime.h>
+#include <linux/shdma-base.h>
#include "rsnd.h"
#define RSND_RATES SNDRV_PCM_RATE_8000_96000
@@ -209,13 +210,6 @@ int rsnd_dma_available(struct rsnd_dma *dma)
return !!dma->chan;
}
-static bool rsnd_dma_filter(struct dma_chan *chan, void *param)
-{
- chan->private = param;
-
- return true;
-}
-
int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma,
int is_play, int id,
int (*inquiry)(struct rsnd_dma *dma,
@@ -223,7 +217,9 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma,
int (*complete)(struct rsnd_dma *dma))
{
struct device *dev = rsnd_priv_to_dev(priv);
+ struct dma_slave_config cfg;
dma_cap_mask_t mask;
+ int ret;
if (dma->chan) {
dev_err(dev, "it already has dma channel\n");
@@ -233,15 +229,23 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma,
dma_cap_zero(mask);
dma_cap_set(DMA_SLAVE, mask);
- dma->slave.shdma_slave.slave_id = id;
-
- dma->chan = dma_request_channel(mask, rsnd_dma_filter,
- &dma->slave.shdma_slave);
+ dma->chan = dma_request_slave_channel_compat(mask, shdma_chan_filter,
+ (void *)id, dev,
+ is_play ? "tx" : "rx");
if (!dma->chan) {
dev_err(dev, "can't get dma channel\n");
return -EIO;
}
+ cfg.slave_id = id;
+ cfg.dst_addr = 0; /* use default addr when playback */
+ cfg.src_addr = 0; /* use default addr when capture */
+ cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM;
+
+ ret = dmaengine_slave_config(dma->chan, &cfg);
+ if (ret < 0)
+ goto rsnd_dma_init_err;
+
dma->dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE;
dma->priv = priv;
dma->inquiry = inquiry;
@@ -249,6 +253,11 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma,
INIT_WORK(&dma->work, rsnd_dma_do_work);
return 0;
+
+rsnd_dma_init_err:
+ rsnd_dma_quit(priv, dma);
+
+ return ret;
}
void rsnd_dma_quit(struct rsnd_priv *priv,
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index afc3fa8b747..bdc1d74eb7b 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2551,8 +2551,9 @@ int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol,
if (uinfo->value.enumerated.item > e->max - 1)
uinfo->value.enumerated.item = e->max - 1;
- strcpy(uinfo->value.enumerated.name,
- e->texts[uinfo->value.enumerated.item]);
+ strlcpy(uinfo->value.enumerated.name,
+ e->texts[uinfo->value.enumerated.item],
+ sizeof(uinfo->value.enumerated.name));
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_enum_double);
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index d4498723b37..591f0f3074c 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -190,7 +190,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
/* startup the audio subsystem */
- if (cpu_dai->driver->ops->startup) {
+ if (cpu_dai->driver->ops && cpu_dai->driver->ops->startup) {
ret = cpu_dai->driver->ops->startup(substream, cpu_dai);
if (ret < 0) {
dev_err(cpu_dai->dev, "ASoC: can't open interface"
@@ -208,7 +208,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
}
}
- if (codec_dai->driver->ops->startup) {
+ if (codec_dai->driver->ops && codec_dai->driver->ops->startup) {
ret = codec_dai->driver->ops->startup(substream, codec_dai);
if (ret < 0) {
dev_err(codec_dai->dev, "ASoC: can't open codec"
@@ -463,7 +463,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
}
}
- if (codec_dai->driver->ops->prepare) {
+ if (codec_dai->driver->ops && codec_dai->driver->ops->prepare) {
ret = codec_dai->driver->ops->prepare(substream, codec_dai);
if (ret < 0) {
dev_err(codec_dai->dev, "ASoC: DAI prepare error: %d\n",
@@ -472,7 +472,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
}
}
- if (cpu_dai->driver->ops->prepare) {
+ if (cpu_dai->driver->ops && cpu_dai->driver->ops->prepare) {
ret = cpu_dai->driver->ops->prepare(substream, cpu_dai);
if (ret < 0) {
dev_err(cpu_dai->dev, "ASoC: DAI prepare error: %d\n",
@@ -523,7 +523,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
}
}
- if (codec_dai->driver->ops->hw_params) {
+ if (codec_dai->driver->ops && codec_dai->driver->ops->hw_params) {
ret = codec_dai->driver->ops->hw_params(substream, params, codec_dai);
if (ret < 0) {
dev_err(codec_dai->dev, "ASoC: can't set %s hw params:"
@@ -532,7 +532,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
}
}
- if (cpu_dai->driver->ops->hw_params) {
+ if (cpu_dai->driver->ops && cpu_dai->driver->ops->hw_params) {
ret = cpu_dai->driver->ops->hw_params(substream, params, cpu_dai);
if (ret < 0) {
dev_err(cpu_dai->dev, "ASoC: %s hw params failed: %d\n",
@@ -559,11 +559,11 @@ out:
return ret;
platform_err:
- if (cpu_dai->driver->ops->hw_free)
+ if (cpu_dai->driver->ops && cpu_dai->driver->ops->hw_free)
cpu_dai->driver->ops->hw_free(substream, cpu_dai);
interface_err:
- if (codec_dai->driver->ops->hw_free)
+ if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free)
codec_dai->driver->ops->hw_free(substream, codec_dai);
codec_err:
@@ -600,10 +600,10 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
platform->driver->ops->hw_free(substream);
/* now free hw params for the DAIs */
- if (codec_dai->driver->ops->hw_free)
+ if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free)
codec_dai->driver->ops->hw_free(substream, codec_dai);
- if (cpu_dai->driver->ops->hw_free)
+ if (cpu_dai->driver->ops && cpu_dai->driver->ops->hw_free)
cpu_dai->driver->ops->hw_free(substream, cpu_dai);
mutex_unlock(&rtd->pcm_mutex);
@@ -618,7 +618,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
struct snd_soc_dai *codec_dai = rtd->codec_dai;
int ret;
- if (codec_dai->driver->ops->trigger) {
+ if (codec_dai->driver->ops && codec_dai->driver->ops->trigger) {
ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai);
if (ret < 0)
return ret;
@@ -630,7 +630,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
return ret;
}
- if (cpu_dai->driver->ops->trigger) {
+ if (cpu_dai->driver->ops && cpu_dai->driver->ops->trigger) {
ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai);
if (ret < 0)
return ret;
@@ -647,19 +647,20 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream,
struct snd_soc_dai *codec_dai = rtd->codec_dai;
int ret;
- if (codec_dai->driver->ops->bespoke_trigger) {
+ if (codec_dai->driver->ops &&
+ codec_dai->driver->ops->bespoke_trigger) {
ret = codec_dai->driver->ops->bespoke_trigger(substream, cmd, codec_dai);
if (ret < 0)
return ret;
}
- if (platform->driver->bespoke_trigger) {
+ if (platform->driver->ops && platform->driver->bespoke_trigger) {
ret = platform->driver->bespoke_trigger(substream, cmd);
if (ret < 0)
return ret;
}
- if (cpu_dai->driver->ops->bespoke_trigger) {
+ if (cpu_dai->driver->ops && cpu_dai->driver->ops->bespoke_trigger) {
ret = cpu_dai->driver->ops->bespoke_trigger(substream, cmd, cpu_dai);
if (ret < 0)
return ret;
@@ -684,10 +685,10 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
if (platform->driver->ops && platform->driver->ops->pointer)
offset = platform->driver->ops->pointer(substream);
- if (cpu_dai->driver->ops->delay)
+ if (cpu_dai->driver->ops && cpu_dai->driver->ops->delay)
delay += cpu_dai->driver->ops->delay(substream, cpu_dai);
- if (codec_dai->driver->ops->delay)
+ if (codec_dai->driver->ops && codec_dai->driver->ops->delay)
delay += codec_dai->driver->ops->delay(substream, codec_dai);
if (platform->driver->delay)
@@ -1037,6 +1038,12 @@ static int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream)
struct snd_pcm_substream *be_substream =
snd_soc_dpcm_get_substream(be, stream);
+ if (!be_substream) {
+ dev_err(be->dev, "ASoC: no backend %s stream\n",
+ stream ? "capture" : "playback");
+ continue;
+ }
+
/* is this op for this BE ? */
if (!snd_soc_dpcm_be_can_update(fe, be, stream))
continue;
@@ -1054,7 +1061,8 @@ static int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream)
(be->dpcm[stream].state != SND_SOC_DPCM_STATE_CLOSE))
continue;
- dev_dbg(be->dev, "ASoC: open BE %s\n", be->dai_link->name);
+ dev_dbg(be->dev, "ASoC: open %s BE %s\n",
+ stream ? "capture" : "playback", be->dai_link->name);
be_substream->runtime = be->dpcm[stream].runtime;
err = soc_pcm_open(be_substream);
@@ -1673,7 +1681,7 @@ static int soc_pcm_ioctl(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_platform *platform = rtd->platform;
- if (platform->driver->ops->ioctl)
+ if (platform->driver->ops && platform->driver->ops->ioctl)
return platform->driver->ops->ioctl(substream, cmd, arg);
return snd_pcm_lib_ioctl(substream, cmd, arg);
}
@@ -1934,8 +1942,8 @@ int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute)
dev_dbg(be->dev, "ASoC: BE digital mute %s\n", be->dai_link->name);
- if (drv->ops->digital_mute && dai->playback_active)
- drv->ops->digital_mute(dai, mute);
+ if (drv->ops && drv->ops->digital_mute && dai->playback_active)
+ drv->ops->digital_mute(dai, mute);
}
return 0;
@@ -2224,7 +2232,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params);
int snd_soc_platform_trigger(struct snd_pcm_substream *substream,
int cmd, struct snd_soc_platform *platform)
{
- if (platform->driver->ops->trigger)
+ if (platform->driver->ops && platform->driver->ops->trigger)
return platform->driver->ops->trigger(substream, cmd);
return 0;
}