summaryrefslogtreecommitdiffstats
path: root/sound/soc
diff options
context:
space:
mode:
Diffstat (limited to 'sound/soc')
-rw-r--r--sound/soc/au1x/dbdma2.c2
-rw-r--r--sound/soc/codecs/Makefile1
-rw-r--r--sound/soc/codecs/twl4030.c8
-rw-r--r--sound/soc/codecs/wm8350.c2
-rw-r--r--sound/soc/codecs/wm8580.c16
-rw-r--r--sound/soc/codecs/wm8990.c40
-rw-r--r--sound/soc/codecs/wm9705.c2
-rw-r--r--sound/soc/davinci/Kconfig7
-rw-r--r--sound/soc/davinci/davinci-evm.c63
-rw-r--r--sound/soc/davinci/davinci-i2s.c26
-rw-r--r--sound/soc/davinci/davinci-pcm.c71
-rw-r--r--sound/soc/fsl/mpc5200_psc_i2s.c3
-rw-r--r--sound/soc/omap/n810.c4
-rw-r--r--sound/soc/omap/omap-mcbsp.c12
-rw-r--r--sound/soc/omap/omap-mcbsp.h3
-rw-r--r--sound/soc/omap/omap-pcm.c5
-rw-r--r--sound/soc/omap/omap-pcm.h3
-rw-r--r--sound/soc/omap/osk5912.c4
-rw-r--r--sound/soc/pxa/palm27x.c27
-rw-r--r--sound/soc/pxa/pxa-ssp.c37
-rw-r--r--sound/soc/s3c24xx/jive_wm8750.c12
-rw-r--r--sound/soc/s3c24xx/s3c-i2s-v2.c21
-rw-r--r--sound/soc/s3c24xx/s3c2412-i2s.c2
-rw-r--r--sound/soc/sh/dma-sh7760.c3
-rw-r--r--sound/soc/soc-core.c3
25 files changed, 258 insertions, 119 deletions
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 30490a259148..594c6c5b7838 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -82,7 +82,7 @@ static struct au1xpsc_audio_dmadata *au1xpsc_audio_pcmdma[2];
/* PCM hardware DMA capabilities - platform specific */
static const struct snd_pcm_hardware au1xpsc_pcm_hardware = {
.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_INTERLEAVED,
+ SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH,
.formats = AU1XPSC_PCM_FMTS,
.period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES,
.period_bytes_max = 4096 * 1024 - 1,
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 030d2454725f..f2653803ede8 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -56,7 +56,6 @@ obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o
obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o
obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o
obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o
-obj-$(CONFIG_SND_SOC_WM8991) += snd-soc-wm8991.o
obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o
obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o
obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 921b205de28a..df7c8c281d2f 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -836,6 +836,12 @@ static DECLARE_TLV_DB_SCALE(analog_tlv, -2400, 200, 0);
static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1);
/*
+ * Gain control for earpiece amplifier
+ * 0 dB to 12 dB in 6 dB steps (mute instead of -6)
+ */
+static DECLARE_TLV_DB_SCALE(output_ear_tvl, -600, 600, 1);
+
+/*
* Capture gain after the ADCs
* from 0 dB to 31 dB in 1 dB steps
*/
@@ -900,7 +906,7 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = {
4, 3, 0, output_tvl),
SOC_SINGLE_TLV_TWL4030("Earpiece Playback Volume",
- TWL4030_REG_EAR_CTL, 4, 3, 0, output_tvl),
+ TWL4030_REG_EAR_CTL, 4, 3, 0, output_ear_tvl),
/* Common capture gain controls */
SOC_DOUBLE_R_TLV("TX1 Digital Capture Volume",
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 3b1d0993bed9..0275321ff8ab 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -968,7 +968,7 @@ static int wm8350_pcm_trigger(struct snd_pcm_substream *substream,
* required for LRC in master mode. The DACs or ADCs need a
* valid audio path i.e. pin -> ADC or DAC -> pin before
* the LRC will be enabled in master mode. */
- if (!master && cmd != SNDRV_PCM_TRIGGER_START)
+ if (!master || cmd != SNDRV_PCM_TRIGGER_START)
return 0;
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 442ea6f160fc..9f6be3d31ac0 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -268,9 +268,11 @@ static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
static int wm8580_out_vu(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- int reg = kcontrol->private_value & 0xff;
- int reg2 = (kcontrol->private_value >> 24) & 0xff;
+ unsigned int reg = mc->reg;
+ unsigned int reg2 = mc->rreg;
int ret;
u16 val;
@@ -292,15 +294,17 @@ static int wm8580_out_vu(struct snd_kcontrol *kcontrol,
return 0;
}
-#define SOC_WM8580_OUT_DOUBLE_R_TLV(xname, reg_left, reg_right, shift, max, invert, tlv_array) \
+#define SOC_WM8580_OUT_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, \
+ xinvert, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
SNDRV_CTL_ELEM_ACCESS_READWRITE, \
.tlv.p = (tlv_array), \
.info = snd_soc_info_volsw_2r, \
.get = snd_soc_get_volsw_2r, .put = wm8580_out_vu, \
- .private_value = (reg_left) | ((shift) << 8) | \
- ((max) << 12) | ((invert) << 20) | ((reg_right) << 24) }
+ .private_value = (unsigned long)&(struct soc_mixer_control) \
+ {.reg = reg_left, .rreg = reg_right, .shift = xshift, \
+ .max = xmax, .invert = xinvert} }
static const struct snd_kcontrol_new wm8580_snd_controls[] = {
SOC_WM8580_OUT_DOUBLE_R_TLV("DAC1 Playback Volume",
@@ -522,7 +526,7 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai,
reg = wm8580_read(codec, WM8580_PLLA4 + offset);
reg &= ~0x3f;
reg |= pll_div.prescale | pll_div.postscale << 1 |
- pll_div.freqmode << 4;
+ pll_div.freqmode << 3;
wm8580_write(codec, WM8580_PLLA4 + offset, reg);
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index c518c3e5aa3f..40cd274eb1ef 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -729,7 +729,7 @@ SND_SOC_DAPM_MIXER_E("INMIXL", WM8990_INTDRIVBITS, WM8990_INMIXL_PWR_BIT, 0,
inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
/* AINLMUX */
-SND_SOC_DAPM_MUX_E("AILNMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0,
+SND_SOC_DAPM_MUX_E("AINLMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0,
&wm8990_dapm_ainlmux_controls, inmixer_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
@@ -740,7 +740,7 @@ SND_SOC_DAPM_MIXER_E("INMIXR", WM8990_INTDRIVBITS, WM8990_INMIXR_PWR_BIT, 0,
inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
/* AINRMUX */
-SND_SOC_DAPM_MUX_E("AIRNMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0,
+SND_SOC_DAPM_MUX_E("AINRMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0,
&wm8990_dapm_ainrmux_controls, inmixer_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
@@ -848,40 +848,40 @@ static const struct snd_soc_dapm_route audio_map[] = {
{"LIN12 PGA", "LIN2 Switch", "LIN2"},
/* LIN34 PGA */
{"LIN34 PGA", "LIN3 Switch", "LIN3"},
- {"LIN34 PGA", "LIN4 Switch", "LIN4"},
+ {"LIN34 PGA", "LIN4 Switch", "LIN4/RXN"},
/* INMIXL */
{"INMIXL", "Record Left Volume", "LOMIX"},
{"INMIXL", "LIN2 Volume", "LIN2"},
{"INMIXL", "LINPGA12 Switch", "LIN12 PGA"},
{"INMIXL", "LINPGA34 Switch", "LIN34 PGA"},
- /* AILNMUX */
- {"AILNMUX", "INMIXL Mix", "INMIXL"},
- {"AILNMUX", "DIFFINL Mix", "LIN12PGA"},
- {"AILNMUX", "DIFFINL Mix", "LIN34PGA"},
- {"AILNMUX", "RXVOICE Mix", "LIN4/RXN"},
- {"AILNMUX", "RXVOICE Mix", "RIN4/RXP"},
+ /* AINLMUX */
+ {"AINLMUX", "INMIXL Mix", "INMIXL"},
+ {"AINLMUX", "DIFFINL Mix", "LIN12 PGA"},
+ {"AINLMUX", "DIFFINL Mix", "LIN34 PGA"},
+ {"AINLMUX", "RXVOICE Mix", "LIN4/RXN"},
+ {"AINLMUX", "RXVOICE Mix", "RIN4/RXP"},
/* ADC */
- {"Left ADC", NULL, "AILNMUX"},
+ {"Left ADC", NULL, "AINLMUX"},
/* RIN12 PGA */
{"RIN12 PGA", "RIN1 Switch", "RIN1"},
{"RIN12 PGA", "RIN2 Switch", "RIN2"},
/* RIN34 PGA */
{"RIN34 PGA", "RIN3 Switch", "RIN3"},
- {"RIN34 PGA", "RIN4 Switch", "RIN4"},
+ {"RIN34 PGA", "RIN4 Switch", "RIN4/RXP"},
/* INMIXL */
{"INMIXR", "Record Right Volume", "ROMIX"},
{"INMIXR", "RIN2 Volume", "RIN2"},
{"INMIXR", "RINPGA12 Switch", "RIN12 PGA"},
{"INMIXR", "RINPGA34 Switch", "RIN34 PGA"},
- /* AIRNMUX */
- {"AIRNMUX", "INMIXR Mix", "INMIXR"},
- {"AIRNMUX", "DIFFINR Mix", "RIN12PGA"},
- {"AIRNMUX", "DIFFINR Mix", "RIN34PGA"},
- {"AIRNMUX", "RXVOICE Mix", "RIN4/RXN"},
- {"AIRNMUX", "RXVOICE Mix", "RIN4/RXP"},
+ /* AINRMUX */
+ {"AINRMUX", "INMIXR Mix", "INMIXR"},
+ {"AINRMUX", "DIFFINR Mix", "RIN12 PGA"},
+ {"AINRMUX", "DIFFINR Mix", "RIN34 PGA"},
+ {"AINRMUX", "RXVOICE Mix", "LIN4/RXN"},
+ {"AINRMUX", "RXVOICE Mix", "RIN4/RXP"},
/* ADC */
- {"Right ADC", NULL, "AIRNMUX"},
+ {"Right ADC", NULL, "AINRMUX"},
/* LOMIX */
{"LOMIX", "LOMIX RIN3 Bypass Switch", "RIN3"},
@@ -922,7 +922,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
{"LOPMIX", "LOPMIX Left Mixer PGA Switch", "LOPGA"},
/* OUT3MIX */
- {"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXP"},
+ {"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXN"},
{"OUT3MIX", "OUT3MIX Left Out PGA Switch", "LOPGA"},
/* OUT4MIX */
@@ -949,7 +949,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
/* Output Pins */
{"LON", NULL, "LONMIX"},
{"LOP", NULL, "LOPMIX"},
- {"OUT", NULL, "OUT3MIX"},
+ {"OUT3", NULL, "OUT3MIX"},
{"LOUT", NULL, "LOUT PGA"},
{"SPKN", NULL, "SPKMIX"},
{"ROUT", NULL, "ROUT PGA"},
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
index 6e23a81dba78..c2d1a7a18fa3 100644
--- a/sound/soc/codecs/wm9705.c
+++ b/sound/soc/codecs/wm9705.c
@@ -318,7 +318,7 @@ static int wm9705_reset(struct snd_soc_codec *codec)
}
#ifdef CONFIG_PM
-static int wm9705_soc_suspend(struct platform_device *pdev)
+static int wm9705_soc_suspend(struct platform_device *pdev, pm_message_t msg)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->card->codec;
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index bd7392c9657e..411a710be660 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -10,13 +10,14 @@ config SND_DAVINCI_SOC_I2S
tristate
config SND_DAVINCI_SOC_EVM
- tristate "SoC Audio support for DaVinci EVM"
- depends on SND_DAVINCI_SOC && MACH_DAVINCI_EVM
+ tristate "SoC Audio support for DaVinci DM6446 or DM355 EVM"
+ depends on SND_DAVINCI_SOC
+ depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM
select SND_DAVINCI_SOC_I2S
select SND_SOC_TLV320AIC3X
help
Say Y if you want to add support for SoC audio on TI
- DaVinci EVM platform.
+ DaVinci DM6446 or DM355 EVM platforms.
config SND_DAVINCI_SOC_SFFSDR
tristate "SoC Audio support for SFFSDR"
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 9b90b347007c..58fd1cbedd88 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -20,7 +20,11 @@
#include <sound/soc-dapm.h>
#include <asm/dma.h>
-#include <mach/hardware.h>
+#include <asm/mach-types.h>
+
+#include <mach/asp.h>
+#include <mach/edma.h>
+#include <mach/mux.h>
#include "../codecs/tlv320aic3x.h"
#include "davinci-pcm.h"
@@ -150,7 +154,7 @@ static struct snd_soc_card snd_soc_card_evm = {
/* evm audio private data */
static struct aic3x_setup_data evm_aic3x_setup = {
- .i2c_bus = 0,
+ .i2c_bus = 1,
.i2c_address = 0x1b,
};
@@ -161,36 +165,73 @@ static struct snd_soc_device evm_snd_devdata = {
.codec_data = &evm_aic3x_setup,
};
+/* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */
static struct resource evm_snd_resources[] = {
{
- .start = DAVINCI_MCBSP_BASE,
- .end = DAVINCI_MCBSP_BASE + SZ_8K - 1,
+ .start = DAVINCI_ASP0_BASE,
+ .end = DAVINCI_ASP0_BASE + SZ_8K - 1,
.flags = IORESOURCE_MEM,
},
};
static struct evm_snd_platform_data evm_snd_data = {
- .tx_dma_ch = DM644X_DMACH_MCBSP_TX,
- .rx_dma_ch = DM644X_DMACH_MCBSP_RX,
+ .tx_dma_ch = DAVINCI_DMA_ASP0_TX,
+ .rx_dma_ch = DAVINCI_DMA_ASP0_RX,
+};
+
+/* DM335 EVM uses ASP1; line-out is a stereo mini-jack */
+static struct resource dm335evm_snd_resources[] = {
+ {
+ .start = DAVINCI_ASP1_BASE,
+ .end = DAVINCI_ASP1_BASE + SZ_8K - 1,
+ .flags = IORESOURCE_MEM,
+ },
+};
+
+static struct evm_snd_platform_data dm335evm_snd_data = {
+ .tx_dma_ch = DAVINCI_DMA_ASP1_TX,
+ .rx_dma_ch = DAVINCI_DMA_ASP1_RX,
};
static struct platform_device *evm_snd_device;
static int __init evm_init(void)
{
+ struct resource *resources;
+ unsigned num_resources;
+ struct evm_snd_platform_data *data;
+ int index;
int ret;
- evm_snd_device = platform_device_alloc("soc-audio", 0);
+ if (machine_is_davinci_evm()) {
+ davinci_cfg_reg(DM644X_MCBSP);
+
+ resources = evm_snd_resources;
+ num_resources = ARRAY_SIZE(evm_snd_resources);
+ data = &evm_snd_data;
+ index = 0;
+ } else if (machine_is_davinci_dm355_evm()) {
+ /* we don't use ASP1 IRQs, or we'd need to mux them ... */
+ davinci_cfg_reg(DM355_EVT8_ASP1_TX);
+ davinci_cfg_reg(DM355_EVT9_ASP1_RX);
+
+ resources = dm335evm_snd_resources;
+ num_resources = ARRAY_SIZE(dm335evm_snd_resources);
+ data = &dm335evm_snd_data;
+ index = 1;
+ } else
+ return -EINVAL;
+
+ evm_snd_device = platform_device_alloc("soc-audio", index);
if (!evm_snd_device)
return -ENOMEM;
platform_set_drvdata(evm_snd_device, &evm_snd_devdata);
evm_snd_devdata.dev = &evm_snd_device->dev;
- platform_device_add_data(evm_snd_device, &evm_snd_data,
- sizeof(evm_snd_data));
+ platform_device_add_data(evm_snd_device, data, sizeof(*data));
- ret = platform_device_add_resources(evm_snd_device, evm_snd_resources,
- ARRAY_SIZE(evm_snd_resources));
+ ret = platform_device_add_resources(evm_snd_device, resources,
+ num_resources);
if (ret) {
platform_device_put(evm_snd_device);
return ret;
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index ffdb9439d3d8..b1ea52fc83c7 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -24,6 +24,26 @@
#include "davinci-pcm.h"
+
+/*
+ * NOTE: terminology here is confusing.
+ *
+ * - This driver supports the "Audio Serial Port" (ASP),
+ * found on dm6446, dm355, and other DaVinci chips.
+ *
+ * - But it labels it a "Multi-channel Buffered Serial Port"
+ * (McBSP) as on older chips like the dm642 ... which was
+ * backward-compatible, possibly explaining that confusion.
+ *
+ * - OMAP chips have a controller called McBSP, which is
+ * incompatible with the DaVinci flavor of McBSP.
+ *
+ * - Newer DaVinci chips have a controller called McASP,
+ * incompatible with ASP and with either McBSP.
+ *
+ * In short: this uses ASP to implement I2S, not McBSP.
+ * And it won't be the only DaVinci implemention of I2S.
+ */
#define DAVINCI_MCBSP_DRR_REG 0x00
#define DAVINCI_MCBSP_DXR_REG 0x04
#define DAVINCI_MCBSP_SPCR_REG 0x08
@@ -421,7 +441,7 @@ static int davinci_i2s_probe(struct platform_device *pdev,
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_card *card = socdev->card;
- struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai;
+ struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai;
struct davinci_mcbsp_dev *dev;
struct resource *mem, *ioarea;
struct evm_snd_platform_data *pdata;
@@ -448,7 +468,7 @@ static int davinci_i2s_probe(struct platform_device *pdev,
cpu_dai->private_data = dev;
- dev->clk = clk_get(&pdev->dev, "McBSPCLK");
+ dev->clk = clk_get(&pdev->dev, NULL);
if (IS_ERR(dev->clk)) {
ret = -ENODEV;
goto err_free_mem;
@@ -483,7 +503,7 @@ static void davinci_i2s_remove(struct platform_device *pdev,
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_card *card = socdev->card;
- struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai;
+ struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai;
struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
struct resource *mem;
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 7af3b5b3a53d..a05996588489 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -22,6 +22,7 @@
#include <sound/soc.h>
#include <asm/dma.h>
+#include <mach/edma.h>
#include "davinci-pcm.h"
@@ -51,7 +52,7 @@ struct davinci_runtime_data {
spinlock_t lock;
int period; /* current DMA period */
int master_lch; /* Master DMA channel */
- int slave_lch; /* Slave DMA channel */
+ int slave_lch; /* linked parameter RAM reload slot */
struct davinci_pcm_dma_params *params; /* DMA params */
};
@@ -90,18 +91,18 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream)
dst_bidx = data_type;
}
- davinci_set_dma_src_params(lch, src, INCR, W8BIT);
- davinci_set_dma_dest_params(lch, dst, INCR, W8BIT);
- davinci_set_dma_src_index(lch, src_bidx, 0);
- davinci_set_dma_dest_index(lch, dst_bidx, 0);
- davinci_set_dma_transfer_params(lch, data_type, count, 1, 0, ASYNC);
+ edma_set_src(lch, src, INCR, W8BIT);
+ edma_set_dest(lch, dst, INCR, W8BIT);
+ edma_set_src_index(lch, src_bidx, 0);
+ edma_set_dest_index(lch, dst_bidx, 0);
+ edma_set_transfer_params(lch, data_type, count, 1, 0, ASYNC);
prtd->period++;
if (unlikely(prtd->period >= runtime->periods))
prtd->period = 0;
}
-static void davinci_pcm_dma_irq(int lch, u16 ch_status, void *data)
+static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data)
{
struct snd_pcm_substream *substream = data;
struct davinci_runtime_data *prtd = substream->runtime->private_data;
@@ -125,7 +126,7 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
struct davinci_runtime_data *prtd = substream->runtime->private_data;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data;
- int tcc = TCC_ANY;
+ struct edmacc_param p_ram;
int ret;
if (!dma_data)
@@ -134,22 +135,34 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
prtd->params = dma_data;
/* Request master DMA channel */
- ret = davinci_request_dma(prtd->params->channel, prtd->params->name,
+ ret = edma_alloc_channel(prtd->params->channel,
davinci_pcm_dma_irq, substream,
- &prtd->master_lch, &tcc, EVENTQ_0);
- if (ret)
+ EVENTQ_0);
+ if (ret < 0)
return ret;
+ prtd->master_lch = ret;
- /* Request slave DMA channel */
- ret = davinci_request_dma(PARAM_ANY, "Link",
- NULL, NULL, &prtd->slave_lch, &tcc, EVENTQ_0);
- if (ret) {
- davinci_free_dma(prtd->master_lch);
+ /* Request parameter RAM reload slot */
+ ret = edma_alloc_slot(EDMA_SLOT_ANY);
+ if (ret < 0) {
+ edma_free_channel(prtd->master_lch);
return ret;
}
-
- /* Link slave DMA channel in loopback */
- davinci_dma_link_lch(prtd->slave_lch, prtd->slave_lch);
+ prtd->slave_lch = ret;
+
+ /* Issue transfer completion IRQ when the channel completes a
+ * transfer, then always reload from the same slot (by a kind
+ * of loopback link). The completion IRQ handler will update
+ * the reload slot with a new buffer.
+ *
+ * REVISIT save p_ram here after setting up everything except
+ * the buffer and its length (ccnt) ... use it as a template
+ * so davinci_pcm_enqueue_dma() takes less time in IRQ.
+ */
+ edma_read_slot(prtd->slave_lch, &p_ram);
+ p_ram.opt |= TCINTEN | EDMA_TCC(prtd->master_lch);
+ p_ram.link_bcntrld = prtd->slave_lch << 5;
+ edma_write_slot(prtd->slave_lch, &p_ram);
return 0;
}
@@ -165,12 +178,12 @@ static int davinci_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:
- davinci_start_dma(prtd->master_lch);
+ edma_start(prtd->master_lch);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- davinci_stop_dma(prtd->master_lch);
+ edma_stop(prtd->master_lch);
break;
default:
ret = -EINVAL;
@@ -185,14 +198,14 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
static int davinci_pcm_prepare(struct snd_pcm_substream *substream)
{
struct davinci_runtime_data *prtd = substream->runtime->private_data;
- struct paramentry_descriptor temp;
+ struct edmacc_param temp;
prtd->period = 0;
davinci_pcm_enqueue_dma(substream);
- /* Get slave channel dma params for master channel startup */
- davinci_get_dma_params(prtd->slave_lch, &temp);
- davinci_set_dma_params(prtd->master_lch, &temp);
+ /* Copy self-linked parameter RAM entry into master channel */
+ edma_read_slot(prtd->slave_lch, &temp);
+ edma_write_slot(prtd->master_lch, &temp);
return 0;
}
@@ -208,7 +221,7 @@ davinci_pcm_pointer(struct snd_pcm_substream *substream)
spin_lock(&prtd->lock);
- davinci_dma_getposition(prtd->master_lch, &src, &dst);
+ edma_get_position(prtd->master_lch, &src, &dst);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
count = src - runtime->dma_addr;
else
@@ -253,10 +266,10 @@ static int davinci_pcm_close(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime = substream->runtime;
struct davinci_runtime_data *prtd = runtime->private_data;
- davinci_dma_unlink_lch(prtd->slave_lch, prtd->slave_lch);
+ edma_unlink(prtd->slave_lch);
- davinci_free_dma(prtd->slave_lch);
- davinci_free_dma(prtd->master_lch);
+ edma_free_slot(prtd->slave_lch);
+ edma_free_channel(prtd->master_lch);
kfree(prtd);
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 3aa729df27b5..1111c710118a 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -504,7 +504,8 @@ static struct snd_soc_dai psc_i2s_dai_template = {
static const struct snd_pcm_hardware psc_i2s_pcm_hardware = {
.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER,
+ SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_BATCH,
.formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |
SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
.rate_min = 8000,
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index a6d1178ce128..91ef17992de5 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -3,7 +3,7 @@
*
* Copyright (C) 2008 Nokia Corporation
*
- * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ * Contact: Jarkko Nikula <jhnikula@gmail.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
@@ -417,6 +417,6 @@ static void __exit n810_soc_exit(void)
module_init(n810_soc_init);
module_exit(n810_soc_exit);
-MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
+MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>");
MODULE_DESCRIPTION("ALSA SoC Nokia N810");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 9c09b94f0cf8..912614283848 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -3,7 +3,8 @@
*
* Copyright (C) 2008 Nokia Corporation
*
- * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ * Peter Ujfalusi <peter.ujfalusi@nokia.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
@@ -283,7 +284,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
break;
case SND_SOC_DAIFMT_DSP_B:
regs->srgr2 |= FPER(wlen * channels - 1);
- regs->srgr1 |= FWID(wlen * channels - 2);
+ regs->srgr1 |= FWID(0);
break;
}
@@ -302,6 +303,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
{
struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
+ unsigned int temp_fmt = fmt;
if (mcbsp_data->configured)
return 0;
@@ -328,6 +330,8 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
/* 0-bit data delay */
regs->rcr2 |= RDATDLY(0);
regs->xcr2 |= XDATDLY(0);
+ /* Invert FS polarity configuration */
+ temp_fmt ^= SND_SOC_DAIFMT_NB_IF;
break;
default:
/* Unsupported data format */
@@ -351,7 +355,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
}
/* Set bit clock (CLKX/CLKR) and FS polarities */
- switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ switch (temp_fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
/*
* Normal BCLK + FS.
@@ -529,6 +533,6 @@ static void __exit snd_omap_mcbsp_exit(void)
}
module_exit(snd_omap_mcbsp_exit);
-MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
+MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>");
MODULE_DESCRIPTION("OMAP I2S SoC Interface");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h
index df7ad13ba73d..c8147aace813 100644
--- a/sound/soc/omap/omap-mcbsp.h
+++ b/sound/soc/omap/omap-mcbsp.h
@@ -3,7 +3,8 @@
*
* Copyright (C) 2008 Nokia Corporation
*
- * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ * Peter Ujfalusi <peter.ujfalusi@nokia.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 1bdbb0427183..07cf7f46b584 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -3,7 +3,8 @@
*
* Copyright (C) 2008 Nokia Corporation
*
- * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ * Peter Ujfalusi <peter.ujfalusi@nokia.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
@@ -367,6 +368,6 @@ static void __exit omap_soc_platform_exit(void)
}
module_exit(omap_soc_platform_exit);
-MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
+MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>");
MODULE_DESCRIPTION("OMAP PCM DMA module");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h
index e4369bdfd77d..8d9d26916b05 100644
--- a/sound/soc/omap/omap-pcm.h
+++ b/sound/soc/omap/omap-pcm.h
@@ -3,7 +3,8 @@
*
* Copyright (C) 2008 Nokia Corporation
*
- * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ * Peter Ujfalusi <peter.ujfalusi@nokia.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
index a952a4eb3361..a4e149b7f0eb 100644
--- a/sound/soc/omap/osk5912.c
+++ b/sound/soc/omap/osk5912.c
@@ -62,7 +62,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream,
/* Set codec DAI configuration */
err = snd_soc_dai_set_fmt(codec_dai,
SND_SOC_DAIFMT_DSP_B |
- SND_SOC_DAIFMT_NB_IF |
+ SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM);
if (err < 0) {
printk(KERN_ERR "can't set codec DAI configuration\n");
@@ -72,7 +72,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream,
/* Set cpu DAI configuration */
err = snd_soc_dai_set_fmt(cpu_dai,
SND_SOC_DAIFMT_DSP_B |
- SND_SOC_DAIFMT_NB_IF |
+ SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM);
if (err < 0) {
printk(KERN_ERR "can't set cpu DAI configuration\n");
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index 48a73f64500b..44fcc4e01e08 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -200,7 +200,7 @@ static struct snd_soc_device palm27x_snd_devdata = {
static struct platform_device *palm27x_snd_device;
-static int __init palm27x_asoc_init(void)
+static int palm27x_asoc_probe(struct platform_device *pdev)
{
int ret;
@@ -208,6 +208,10 @@ static int __init palm27x_asoc_init(void)
machine_is_palmld()))
return -ENODEV;
+ if (pdev->dev.platform_data)
+ palm27x_ep_gpio = ((struct palm27x_asoc_info *)
+ (pdev->dev.platform_data))->jack_gpio;
+
ret = gpio_request(palm27x_ep_gpio, "Headphone Jack");
if (ret)
return ret;
@@ -245,16 +249,31 @@ err_alloc:
return ret;
}
-static void __exit palm27x_asoc_exit(void)
+static int __devexit palm27x_asoc_remove(struct platform_device *pdev)
{
free_irq(gpio_to_irq(palm27x_ep_gpio), NULL);
gpio_free(palm27x_ep_gpio);
platform_device_unregister(palm27x_snd_device);
+ return 0;
}
-void __init palm27x_asoc_set_pdata(struct palm27x_asoc_info *data)
+static struct platform_driver palm27x_wm9712_driver = {
+ .probe = palm27x_asoc_probe,
+ .remove = __devexit_p(palm27x_asoc_remove),
+ .driver = {
+ .name = "palm27x-asoc",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init palm27x_asoc_init(void)
+{
+ return platform_driver_register(&palm27x_wm9712_driver);
+}
+
+static void __exit palm27x_asoc_exit(void)
{
- palm27x_ep_gpio = data->jack_gpio;
+ platform_driver_unregister(&palm27x_wm9712_driver);
}
module_init(palm27x_asoc_init);
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 308a657928d2..286be31545df 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -280,12 +280,33 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai)
* ssp_set_clkdiv - set SSP clock divider
* @div: serial clock rate divider
*/
-static void ssp_set_scr(struct ssp_dev *dev, u32 div)
+static void ssp_set_scr(struct ssp_device *ssp, u32 div)
{
- struct ssp_device *ssp = dev->ssp;
- u32 sscr0 = ssp_read_reg(dev->ssp, SSCR0) & ~SSCR0_SCR;
+ u32 sscr0 = ssp_read_reg(ssp, SSCR0);
+
+ if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) {
+ sscr0 &= ~0x0000ff00;
+ sscr0 |= ((div - 2)/2) << 8; /* 2..512 */
+ } else {
+ sscr0 &= ~0x000fff00;
+ sscr0 |= (div - 1) << 8; /* 1..4096 */
+ }
+ ssp_write_reg(ssp, SSCR0, sscr0);
+}
+
+/**
+ * ssp_get_clkdiv - get SSP clock divider
+ */
+static u32 ssp_get_scr(struct ssp_device *ssp)
+{
+ u32 sscr0 = ssp_read_reg(ssp, SSCR0);
+ u32 div;
- ssp_write_reg(ssp, SSCR0, (sscr0 | SSCR0_SerClkDiv(div)));
+ if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP)
+ div = ((sscr0 >> 8) & 0xff) * 2 + 2;
+ else
+ div = ((sscr0 >> 8) & 0xfff) + 1;
+ return div;
}
/*
@@ -326,7 +347,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
break;
case PXA_SSP_CLK_AUDIO:
priv->sysclk = 0;
- ssp_set_scr(&priv->dev, 1);
+ ssp_set_scr(ssp, 1);
sscr0 |= SSCR0_ACS;
break;
default:
@@ -387,7 +408,7 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
ssp_write_reg(ssp, SSACD, val);
break;
case PXA_SSP_DIV_SCR:
- ssp_set_scr(&priv->dev, div);
+ ssp_set_scr(ssp, div);
break;
default:
return -ENODEV;
@@ -674,8 +695,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
case SND_SOC_DAIFMT_I2S:
sspsp = ssp_read_reg(ssp, SSPSP);
- if (((sscr0 & SSCR0_SCR) == SSCR0_SerClkDiv(4)) &&
- (width == 16)) {
+ if ((ssp_get_scr(ssp) == 4) && (width == 16)) {
/* This is a special case where the bitclk is 64fs
* and we're not dealing with 2*32 bits of audio
* samples.
@@ -806,6 +826,7 @@ static int pxa_ssp_probe(struct platform_device *pdev,
goto err_priv;
}
+ priv->dai_fmt = (unsigned int) -1;
dai->private_data = priv;
return 0;
diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c
index 32063790d95b..93e6c87b7399 100644
--- a/sound/soc/s3c24xx/jive_wm8750.c
+++ b/sound/soc/s3c24xx/jive_wm8750.c
@@ -69,8 +69,8 @@ static int jive_hw_params(struct snd_pcm_substream *substream,
break;
}
- s3c_i2sv2_calc_rate(&div, NULL, params_rate(params),
- s3c2412_get_iisclk());
+ s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params),
+ s3c2412_get_iisclk());
/* set codec DAI configuration */
ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
@@ -145,8 +145,9 @@ static struct snd_soc_dai_link jive_dai = {
};
/* jive audio machine driver */
-static struct snd_soc_machine snd_soc_machine_jive = {
+static struct snd_soc_card snd_soc_machine_jive = {
.name = "Jive",
+ .platform = &s3c24xx_soc_platform,
.dai_link = &jive_dai,
.num_links = 1,
};
@@ -157,9 +158,8 @@ static struct wm8750_setup_data jive_wm8750_setup = {
/* jive audio subsystem */
static struct snd_soc_device jive_snd_devdata = {
- .machine = &snd_soc_machine_jive,
- .platform = &s3c24xx_soc_platform,
- .codec_dev = &soc_codec_dev_wm8750_spi,
+ .card = &snd_soc_machine_jive,
+ .codec_dev = &soc_codec_dev_wm8750,
.codec_data = &jive_wm8750_setup,
};
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index 295a4c910262..ab680aac3fcb 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -473,9 +473,9 @@ static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai,
/* default table of all avaialable root fs divisors */
static unsigned int iis_fs_tab[] = { 256, 512, 384, 768 };
-int s3c2412_iis_calc_rate(struct s3c_i2sv2_rate_calc *info,
- unsigned int *fstab,
- unsigned int rate, struct clk *clk)
+int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info,
+ unsigned int *fstab,
+ unsigned int rate, struct clk *clk)
{
unsigned long clkrate = clk_get_rate(clk);
unsigned int div;
@@ -531,7 +531,7 @@ int s3c2412_iis_calc_rate(struct s3c_i2sv2_rate_calc *info,
return 0;
}
-EXPORT_SYMBOL_GPL(s3c2412_iis_calc_rate);
+EXPORT_SYMBOL_GPL(s3c_i2sv2_iis_calc_rate);
int s3c_i2sv2_probe(struct platform_device *pdev,
struct snd_soc_dai *dai,
@@ -624,15 +624,18 @@ static int s3c2412_i2s_resume(struct snd_soc_dai *dai)
int s3c_i2sv2_register_dai(struct snd_soc_dai *dai)
{
- dai->ops.trigger = s3c2412_i2s_trigger;
- dai->ops.hw_params = s3c2412_i2s_hw_params;
- dai->ops.set_fmt = s3c2412_i2s_set_fmt;
- dai->ops.set_clkdiv = s3c2412_i2s_set_clkdiv;
+ struct snd_soc_dai_ops *ops = dai->ops;
+
+ ops->trigger = s3c2412_i2s_trigger;
+ ops->hw_params = s3c2412_i2s_hw_params;
+ ops->set_fmt = s3c2412_i2s_set_fmt;
+ ops->set_clkdiv = s3c2412_i2s_set_clkdiv;
dai->suspend = s3c2412_i2s_suspend;
dai->resume = s3c2412_i2s_resume;
return snd_soc_register_dai(dai);
}
-
EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai);
+
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index 1ca3cdaa8213..b7e0b3f0bfc8 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -33,8 +33,8 @@
#include <plat/regs-s3c2412-iis.h>
-#include <plat/regs-gpio.h>
#include <plat/audio.h>
+#include <mach/regs-gpio.h>
#include <mach/dma.h>
#include "s3c24xx-pcm.h"
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 0dad3a0bb920..baddb1242c71 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -103,7 +103,8 @@ static struct snd_pcm_hardware camelot_pcm_hardware = {
.info = (SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
- SNDRV_PCM_INFO_MMAP_VALID),
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BATCH),
.formats = DMABRG_FMTS,
.rates = DMABRG_RATES,
.rate_min = 8000,
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 99712f652d0d..1cd149b9ce69 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -954,6 +954,9 @@ static int soc_remove(struct platform_device *pdev)
struct snd_soc_platform *platform = card->platform;
struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
+ if (!card->instantiated)
+ return 0;
+
run_delayed_work(&card->delayed_work);
if (platform->remove)
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-26 00:33:36 +0100