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-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/wm9705.c2
-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/magician.c2
-rw-r--r--sound/soc/pxa/palm27x.c27
-rw-r--r--sound/soc/pxa/pxa-ssp.c37
-rw-r--r--sound/soc/s3c24xx/Kconfig6
-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
21 files changed, 117 insertions, 58 deletions
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 30490a25914..594c6c5b783 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 030d2454725..f2653803ede 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 921b205de28..df7c8c281d2 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 3b1d0993bed..0275321ff8a 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 442ea6f160f..9f6be3d31ac 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/wm9705.c b/sound/soc/codecs/wm9705.c
index 6e23a81dba7..c2d1a7a18fa 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/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 3aa729df27b..1111c710118 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 a6d1178ce12..91ef17992de 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 9c09b94f0cf..91261428384 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 df7ad13ba73..c8147aace81 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 1bdbb042718..07cf7f46b58 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 e4369bdfd77..8d9d26916b0 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 a952a4eb336..a4e149b7f0e 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/magician.c b/sound/soc/pxa/magician.c
index f7c4544f785..0625c342a1c 100644
--- a/sound/soc/pxa/magician.c
+++ b/sound/soc/pxa/magician.c
@@ -27,8 +27,6 @@
#include <sound/soc.h>
#include <sound/soc-dapm.h>
-#include <mach/pxa-regs.h>
-#include <mach/hardware.h>
#include <mach/magician.h>
#include <asm/mach-types.h>
#include "../codecs/uda1380.h"
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index 48a73f64500..44fcc4e01e0 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 308a657928d..286be31545d 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/Kconfig b/sound/soc/s3c24xx/Kconfig
index 2f3a21eee05..df494d1e346 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -1,10 +1,10 @@
config SND_S3C24XX_SOC
tristate "SoC Audio for the Samsung S3CXXXX chips"
- depends on ARCH_S3C2410 || ARCH_S3C64XX
+ depends on ARCH_S3C2410
help
Say Y or M if you want to add support for codecs attached to
- the S3C24XX and S3C64XX AC97, I2S or SSP interface. You will
- also need to select the audio interfaces to support below.
+ the S3C24XX AC97 or I2S interfaces. You will also need to
+ select the audio interfaces to support below.
config SND_S3C24XX_SOC_I2S
tristate
diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c
index 32063790d95..93e6c87b739 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 295a4c91026..ab680aac3fc 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 1ca3cdaa821..b7e0b3f0bfc 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 0dad3a0bb92..baddb1242c7 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,
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-30 11:59:34 +0000