Merge branch 'fix/asoc' into for-linus
diff --git a/MAINTAINERS b/MAINTAINERS
index 46e3e6b..4f555d8 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -5532,6 +5532,7 @@
SAMSUNG AUDIO (ASoC) DRIVERS
M: Jassi Brar <jassisinghbrar@gmail.com>
+M: Sangbeom Kim <sbkim73@samsung.com>
L: alsa-devel@alsa-project.org (moderated for non-subscribers)
S: Supported
F: sound/soc/samsung
diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c
index d6651c0..a118a0f 100644
--- a/sound/soc/blackfin/bf5xx-ad193x.c
+++ b/sound/soc/blackfin/bf5xx-ad193x.c
@@ -56,7 +56,7 @@
switch (params_rate(params)) {
case 48000:
- clk = 12288000;
+ clk = 24576000;
break;
}
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index 2374ca5..eedb6f5 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -27,11 +27,6 @@
int sysclk;
};
-/* ad193x register cache & default register settings */
-static const u8 ad193x_reg[AD193X_NUM_REGS] = {
- 0, 0, 0, 0, 0, 0, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0, 0, 0,
-};
-
/*
* AD193X volume/mute/de-emphasis etc. controls
*/
@@ -307,7 +302,8 @@
snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, reg);
reg = snd_soc_read(codec, AD193X_DAC_CTRL2);
- reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) | word_len;
+ reg = (reg & (~AD193X_DAC_WORD_LEN_MASK))
+ | (word_len << AD193X_DAC_WORD_LEN_SHFT);
snd_soc_write(codec, AD193X_DAC_CTRL2, reg);
reg = snd_soc_read(codec, AD193X_ADC_CTRL1);
@@ -389,9 +385,6 @@
static struct snd_soc_codec_driver soc_codec_dev_ad193x = {
.probe = ad193x_probe,
- .reg_cache_default = ad193x_reg,
- .reg_cache_size = AD193X_NUM_REGS,
- .reg_word_size = sizeof(u16),
};
#if defined(CONFIG_SPI_MASTER)
diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h
index 9747b54..cccc2e8 100644
--- a/sound/soc/codecs/ad193x.h
+++ b/sound/soc/codecs/ad193x.h
@@ -34,7 +34,8 @@
#define AD193X_DAC_LEFT_HIGH (1 << 3)
#define AD193X_DAC_BCLK_INV (1 << 7)
#define AD193X_DAC_CTRL2 0x804
-#define AD193X_DAC_WORD_LEN_MASK 0xC
+#define AD193X_DAC_WORD_LEN_SHFT 3
+#define AD193X_DAC_WORD_LEN_MASK 0x18
#define AD193X_DAC_MASTER_MUTE 1
#define AD193X_DAC_CHNL_MUTE 0x805
#define AD193X_DACL1_MUTE 0
@@ -63,7 +64,7 @@
#define AD193X_ADC_CTRL1 0x80f
#define AD193X_ADC_SERFMT_MASK 0x60
#define AD193X_ADC_SERFMT_STEREO (0 << 5)
-#define AD193X_ADC_SERFMT_TDM (1 << 2)
+#define AD193X_ADC_SERFMT_TDM (1 << 5)
#define AD193X_ADC_SERFMT_AUX (2 << 5)
#define AD193X_ADC_WORD_LEN_MASK 0x3
#define AD193X_ADC_CTRL2 0x810
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index 409d89d..fbd7eb9 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -857,6 +857,7 @@
ret = snd_soc_register_codec(&i2c->dev, &sta32x_codec, &sta32x_dai, 1);
if (ret != 0) {
dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret);
+ kfree(sta32x);
return ret;
}
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 60d740e..1725550 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2221,6 +2221,8 @@
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
if (fll) {
+ try_wait_for_completion(&wm8962->fll_lock);
+
snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1,
WM8962_FLL_ENA, WM8962_FLL_ENA);
if (wm8962->irq) {
@@ -2927,10 +2929,6 @@
WM8962_BIAS_ENA | 0x180);
msleep(5);
-
- snd_soc_update_bits(codec, WM8962_CLOCKING2,
- WM8962_CLKREG_OVD,
- WM8962_CLKREG_OVD);
}
/* VMID 2*250k */
@@ -3288,6 +3286,8 @@
snd_soc_write(codec, WM8962_FLL_CONTROL_7, fll_div.lambda);
snd_soc_write(codec, WM8962_FLL_CONTROL_8, fll_div.n);
+ try_wait_for_completion(&wm8962->fll_lock);
+
snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1,
WM8962_FLL_FRAC | WM8962_FLL_REFCLK_SRC_MASK |
WM8962_FLL_ENA, fll1);
@@ -3868,6 +3868,10 @@
*/
snd_soc_update_bits(codec, WM8962_CLOCKING2, WM8962_SYSCLK_ENA, 0);
+ /* Ensure we have soft control over all registers */
+ snd_soc_update_bits(codec, WM8962_CLOCKING2,
+ WM8962_CLKREG_OVD, WM8962_CLKREG_OVD);
+
regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies);
if (pdata) {
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index ab8e9d1..0cdb9d1 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -420,7 +420,7 @@
};
static const struct soc_enum sidetone_hpf =
- SOC_ENUM_SINGLE(WM8996_SIDETONE, 7, 6, sidetone_hpf_text);
+ SOC_ENUM_SINGLE(WM8996_SIDETONE, 7, 7, sidetone_hpf_text);
static const char *hpf_mode_text[] = {
"HiFi", "Custom", "Voice"
@@ -988,15 +988,10 @@
SND_SOC_DAPM_PGA("IN1L PGA", WM8996_POWER_MANAGEMENT_2, 5, 0, NULL, 0),
SND_SOC_DAPM_PGA("IN1R PGA", WM8996_POWER_MANAGEMENT_2, 4, 0, NULL, 0),
-SND_SOC_DAPM_MUX("IN1L Mux", SND_SOC_NOPM, 0, 0, &in1_mux),
-SND_SOC_DAPM_MUX("IN1R Mux", SND_SOC_NOPM, 0, 0, &in1_mux),
-SND_SOC_DAPM_MUX("IN2L Mux", SND_SOC_NOPM, 0, 0, &in2_mux),
-SND_SOC_DAPM_MUX("IN2R Mux", SND_SOC_NOPM, 0, 0, &in2_mux),
-
-SND_SOC_DAPM_PGA("IN1L", WM8996_POWER_MANAGEMENT_7, 2, 0, NULL, 0),
-SND_SOC_DAPM_PGA("IN1R", WM8996_POWER_MANAGEMENT_7, 3, 0, NULL, 0),
-SND_SOC_DAPM_PGA("IN2L", WM8996_POWER_MANAGEMENT_7, 6, 0, NULL, 0),
-SND_SOC_DAPM_PGA("IN2R", WM8996_POWER_MANAGEMENT_7, 7, 0, NULL, 0),
+SND_SOC_DAPM_MUX("IN1L Mux", WM8996_POWER_MANAGEMENT_7, 2, 0, &in1_mux),
+SND_SOC_DAPM_MUX("IN1R Mux", WM8996_POWER_MANAGEMENT_7, 3, 0, &in1_mux),
+SND_SOC_DAPM_MUX("IN2L Mux", WM8996_POWER_MANAGEMENT_7, 6, 0, &in2_mux),
+SND_SOC_DAPM_MUX("IN2R Mux", WM8996_POWER_MANAGEMENT_7, 7, 0, &in2_mux),
SND_SOC_DAPM_SUPPLY("DMIC2", WM8996_POWER_MANAGEMENT_7, 9, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("DMIC1", WM8996_POWER_MANAGEMENT_7, 8, 0, NULL, 0),
@@ -1213,6 +1208,16 @@
{ "AIF2RX0", NULL, "AIFCLK" },
{ "AIF2RX1", NULL, "AIFCLK" },
+ { "AIF1TX0", NULL, "AIFCLK" },
+ { "AIF1TX1", NULL, "AIFCLK" },
+ { "AIF1TX2", NULL, "AIFCLK" },
+ { "AIF1TX3", NULL, "AIFCLK" },
+ { "AIF1TX4", NULL, "AIFCLK" },
+ { "AIF1TX5", NULL, "AIFCLK" },
+
+ { "AIF2TX0", NULL, "AIFCLK" },
+ { "AIF2TX1", NULL, "AIFCLK" },
+
{ "DSP1RXL", NULL, "SYSDSPCLK" },
{ "DSP1RXR", NULL, "SYSDSPCLK" },
{ "DSP2RXL", NULL, "SYSDSPCLK" },
@@ -2106,6 +2111,9 @@
snd_soc_write(codec, WM8996_FLL_EFS_1, fll_div.lambda);
+ /* Clear any pending completions (eg, from failed startups) */
+ try_wait_for_completion(&wm8996->fll_lock);
+
snd_soc_update_bits(codec, WM8996_FLL_CONTROL_1,
WM8996_FLL_ENA, WM8996_FLL_ENA);
diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c
index 56efa0c..099614e1 100644
--- a/sound/soc/ep93xx/ep93xx-i2s.c
+++ b/sound/soc/ep93xx/ep93xx-i2s.c
@@ -385,14 +385,14 @@
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (!res) {
err = -ENODEV;
- goto fail;
+ goto fail_free_info;
}
info->mem = request_mem_region(res->start, resource_size(res),
pdev->name);
if (!info->mem) {
err = -EBUSY;
- goto fail;
+ goto fail_free_info;
}
info->regs = ioremap(info->mem->start, resource_size(info->mem));
@@ -435,6 +435,7 @@
iounmap(info->regs);
fail_release_mem:
release_mem_region(info->mem->start, resource_size(info->mem));
+fail_free_info:
kfree(info);
fail:
return err;
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 732208c..cb50598 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -879,10 +879,12 @@
* assume that device_node pointers are a valid comparison.
*/
np = of_parse_phandle(ssi_np, "fsl,playback-dma", 0);
+ of_node_put(np);
if (np == dma_channel_np)
return ssi_np;
np = of_parse_phandle(ssi_np, "fsl,capture-dma", 0);
+ of_node_put(np);
if (np == dma_channel_np)
return ssi_np;
}
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index a192979..358f0ba 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -345,8 +345,10 @@
}
machine_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL);
- if (!machine_data)
- return -ENOMEM;
+ if (!machine_data) {
+ ret = -ENOMEM;
+ goto error_alloc;
+ }
machine_data->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev);
machine_data->dai[0].ops = &mpc8610_hpcd_ops;
@@ -494,7 +496,7 @@
ret = platform_device_add(sound_device);
if (ret) {
dev_err(&pdev->dev, "platform device add failed\n");
- goto error;
+ goto error_sound;
}
dev_set_drvdata(&pdev->dev, sound_device);
@@ -502,14 +504,12 @@
return 0;
+error_sound:
+ platform_device_unregister(sound_device);
error:
- of_node_put(codec_np);
-
- if (sound_device)
- platform_device_unregister(sound_device);
-
kfree(machine_data);
-
+error_alloc:
+ of_node_put(codec_np);
return ret;
}
diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c
index 8fa4d5f..fcb862e 100644
--- a/sound/soc/fsl/p1022_ds.c
+++ b/sound/soc/fsl/p1022_ds.c
@@ -297,8 +297,10 @@
* dai->platform name should already point to an allocated buffer.
*/
ret = of_address_to_resource(dma_channel_np, 0, &res);
- if (ret)
+ if (ret) {
+ of_node_put(dma_channel_np);
return ret;
+ }
snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s",
(unsigned long long) res.start, dma_channel_np->name);
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index a33fc51..8f16cd3 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -424,7 +424,7 @@
if (!priv->mem) {
dev_err(&pdev->dev, "request_mem_region failed\n");
err = -EBUSY;
- goto error;
+ goto error_alloc;
}
priv->io = ioremap(priv->mem->start, SZ_16K);
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 30fe0d0..0aa475f 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -514,7 +514,7 @@
}
/* Set codec bias level */
- ams_delta_set_bias_level(card, SND_SOC_BIAS_STANDBY);
+ ams_delta_set_bias_level(card, dapm, SND_SOC_BIAS_STANDBY);
/* Add hook switch - can be used to control the codec from userspace
* even if line discipline fails */
@@ -649,7 +649,9 @@
ams_delta_hook_switch_gpios);
/* Keep modem power on */
- ams_delta_set_bias_level(&ams_delta_audio_card, SND_SOC_BIAS_STANDBY);
+ ams_delta_set_bias_level(&ams_delta_audio_card,
+ &ams_delta_audio_card.rtd[0].codec->dapm,
+ SND_SOC_BIAS_STANDBY);
platform_device_unregister(cx20442_platform_device);
platform_device_unregister(ams_delta_audio_platform_device);
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index b99091f..65f980e 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -185,6 +185,7 @@
select SND_SAMSUNG_I2S
select SND_SOC_WM8996
select SND_SOC_WM9081
+ select SND_SOC_WM1250_EV1
config SND_SOC_SPEYSIDE_WM8962
tristate "Audio support for Wolfson Speyside with WM8962"
diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c
index 241f55d..c6c6589 100644
--- a/sound/soc/samsung/h1940_uda1380.c
+++ b/sound/soc/samsung/h1940_uda1380.c
@@ -13,6 +13,7 @@
*
*/
+#include <linux/types.h>
#include <linux/gpio.h>
#include <sound/soc.h>
diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c
index 1e574a5..bc8c167 100644
--- a/sound/soc/samsung/rx1950_uda1380.c
+++ b/sound/soc/samsung/rx1950_uda1380.c
@@ -17,6 +17,7 @@
*
*/
+#include <linux/types.h>
#include <linux/gpio.h>
#include <sound/soc.h>
diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c
index 0b9eb5f..72535f2 100644
--- a/sound/soc/samsung/speyside_wm8962.c
+++ b/sound/soc/samsung/speyside_wm8962.c
@@ -23,6 +23,9 @@
struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
int ret;
+ if (dapm->dev != codec_dai->dev)
+ return 0;
+
switch (level) {
case SND_SOC_BIAS_PREPARE:
if (dapm->bias_level == SND_SOC_BIAS_STANDBY) {
@@ -57,6 +60,9 @@
struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
int ret;
+ if (dapm->dev != codec_dai->dev)
+ return 0;
+
switch (level) {
case SND_SOC_BIAS_STANDBY:
ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK,
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 83ad8ca..b085d8e 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1913,7 +1913,7 @@
if (prefix) {
name_len = strlen(long_name) + strlen(prefix) + 2;
- name = kmalloc(name_len, GFP_ATOMIC);
+ name = kmalloc(name_len, GFP_KERNEL);
if (!name)
return NULL;
diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c
index cca490c..a62f7dd 100644
--- a/sound/soc/soc-io.c
+++ b/sound/soc/soc-io.c
@@ -205,6 +205,25 @@
#define snd_soc_16_8_read_i2c NULL
#endif
+#if defined(CONFIG_SPI_MASTER)
+static unsigned int snd_soc_16_8_read_spi(struct snd_soc_codec *codec,
+ unsigned int r)
+{
+ struct spi_device *spi = codec->control_data;
+
+ const u16 reg = cpu_to_be16(r | 0x100);
+ u8 data;
+ int ret;
+
+ ret = spi_write_then_read(spi, ®, 2, &data, 1);
+ if (ret < 0)
+ return 0;
+ return data;
+}
+#else
+#define snd_soc_16_8_read_spi NULL
+#endif
+
static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int value)
{
@@ -295,6 +314,7 @@
int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int);
unsigned int (*read)(struct snd_soc_codec *, unsigned int);
unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int);
+ unsigned int (*spi_read)(struct snd_soc_codec *, unsigned int);
} io_types[] = {
{
.addr_bits = 4, .data_bits = 12,
@@ -318,6 +338,7 @@
.addr_bits = 16, .data_bits = 8,
.write = snd_soc_16_8_write,
.i2c_read = snd_soc_16_8_read_i2c,
+ .spi_read = snd_soc_16_8_read_spi,
},
{
.addr_bits = 16, .data_bits = 16,
@@ -383,6 +404,8 @@
#ifdef CONFIG_SPI_MASTER
codec->hw_write = do_spi_write;
#endif
+ if (io_types[i].spi_read)
+ codec->hw_read = io_types[i].spi_read;
codec->control_data = container_of(codec->dev,
struct spi_device,
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 7c17b98..38b0013 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -327,7 +327,7 @@
IRQF_TRIGGER_FALLING,
gpios[i].name,
&gpios[i]);
- if (ret)
+ if (ret < 0)
goto err;
if (gpios[i].wake) {
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index b575939..2879c88 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -290,6 +290,9 @@
codec_dai->active--;
codec->active--;
+ if (!cpu_dai->active && !codec_dai->active)
+ rtd->rate = 0;
+
/* Muting the DAC suppresses artifacts caused during digital
* shutdown, for example from stopping clocks.
*/
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 661373c..be27f1d 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -319,7 +319,7 @@
snd_soc_dapm_force_enable_pin(dapm, "Mic Bias");
/* FIXME: Calculate automatically based on DAPM routes? */
- if (!machine_is_harmony() && !machine_is_ventana())
+ if (!machine_is_harmony())
snd_soc_dapm_nc_pin(dapm, "IN1L");
if (!machine_is_seaboard() && !machine_is_aebl())
snd_soc_dapm_nc_pin(dapm, "IN1R");
@@ -395,7 +395,7 @@
platform_set_drvdata(pdev, card);
snd_soc_card_set_drvdata(card, machine);
- if (machine_is_harmony() || machine_is_ventana()) {
+ if (machine_is_harmony()) {
card->dapm_routes = harmony_audio_map;
card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map);
} else if (machine_is_seaboard()) {