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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/wm8990.c40
-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.c5
-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/pxa/palm27x.c27
-rw-r--r--sound/soc/pxa/pxa-ssp.c36
-rw-r--r--sound/soc/s3c24xx/s3c-i2s-v2.c3
-rw-r--r--sound/soc/sh/dma-sh7760.c3
17 files changed, 109 insertions, 54 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/wm8990.c b/sound/soc/codecs/wm8990.c
index c518c3e5aa3..40cd274eb1e 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 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 90f4df7fd90..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
@@ -532,6 +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/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 de2254475d5..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.
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index 689ffcd17e1..ab680aac3fc 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -636,5 +636,6 @@ int s3c_i2sv2_register_dai(struct snd_soc_dai *dai)
return snd_soc_register_dai(dai);
}
-
EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai);
+
+MODULE_LICENSE("GPL");
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 v1.2.3-70-g09d2 (git 2.46.0) at 2024-11-08 14:40:00 +0000