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-rw-r--r--sound/soc/at32/playpaq_wm8510.c12
-rw-r--r--sound/soc/blackfin/bf5xx-i2s.c34
-rw-r--r--sound/soc/codecs/Kconfig2
-rw-r--r--sound/soc/codecs/tlv320aic23.c10
-rw-r--r--sound/soc/codecs/tlv320aic3x.c16
-rw-r--r--sound/soc/codecs/wm9713.c2
-rw-r--r--sound/soc/fsl/mpc5200_psc_i2s.c14
-rw-r--r--sound/soc/omap/omap-mcbsp.c19
-rw-r--r--sound/soc/pxa/corgi.c34
-rw-r--r--sound/soc/pxa/pxa2xx-ac97.c10
-rw-r--r--sound/soc/pxa/pxa2xx-i2s.c43
-rw-r--r--sound/soc/pxa/spitz.c46
-rw-r--r--sound/soc/soc-core.c2
-rw-r--r--sound/soc/soc-dapm.c2
14 files changed, 145 insertions, 101 deletions
diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c
index 98a2d5826a8..b1966e4dfcd 100644
--- a/sound/soc/at32/playpaq_wm8510.c
+++ b/sound/soc/at32/playpaq_wm8510.c
@@ -304,7 +304,7 @@ static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = {
-static const char *intercon[][3] = {
+static const struct snd_soc_dapm_route intercon[] = {
/* speaker connected to SPKOUT */
{"Ext Spk", NULL, "SPKOUTP"},
{"Ext Spk", NULL, "SPKOUTN"},
@@ -312,9 +312,6 @@ static const char *intercon[][3] = {
{"Mic Bias", NULL, "Int Mic"},
{"MICN", NULL, "Mic Bias"},
{"MICP", NULL, "Mic Bias"},
-
- /* Terminator */
- {NULL, NULL, NULL},
};
@@ -334,11 +331,8 @@ static int playpaq_wm8510_init(struct snd_soc_codec *codec)
/*
* Setup audio path interconnects
*/
- for (i = 0; intercon[i][0] != NULL; i++) {
- snd_soc_dapm_connect_input(codec,
- intercon[i][0],
- intercon[i][1], intercon[i][2]);
- }
+ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
/* always connected pins */
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index 827587f0818..e020c160ee4 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -70,12 +70,24 @@ static struct sport_param sport_params[2] = {
}
};
-static u16 sport_req[][7] = {
- { P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
- P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0},
- { P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS,
- P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0},
-};
+/*
+ * Setting the TFS pin selector for SPORT 0 based on whether the selected
+ * port id F or G. If the port is F then no conflict should exist for the
+ * TFS. When Port G is selected and EMAC then there is a conflict between
+ * the PHY interrupt line and TFS. Current settings prevent the conflict
+ * by ignoring the TFS pin when Port G is selected. This allows both
+ * ssm2602 using Port G and EMAC concurrently.
+ */
+#ifdef CONFIG_BF527_SPORT0_PORTF
+#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
+#else
+#define LOCAL_SPORT0_TFS (0)
+#endif
+
+static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
+ P_SPORT0_DRPRI, P_SPORT0_RSCLK, LOCAL_SPORT0_TFS, 0},
+ {P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, P_SPORT1_DRPRI,
+ P_SPORT1_RSCLK, P_SPORT1_TFS, 0} };
static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
unsigned int fmt)
@@ -98,23 +110,21 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
ret = -EINVAL;
break;
default:
+ printk(KERN_ERR "%s: Unknown DAI format type\n", __func__);
ret = -EINVAL;
break;
}
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
- case SND_SOC_DAIFMT_CBS_CFS:
- ret = -EINVAL;
- break;
- case SND_SOC_DAIFMT_CBM_CFS:
- ret = -EINVAL;
- break;
case SND_SOC_DAIFMT_CBM_CFM:
break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ case SND_SOC_DAIFMT_CBM_CFS:
case SND_SOC_DAIFMT_CBS_CFM:
ret = -EINVAL;
break;
default:
+ printk(KERN_ERR "%s: Unknown DAI master type\n", __func__);
ret = -EINVAL;
break;
}
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 4975d8573e4..38a0e3b620a 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -68,7 +68,7 @@ config SND_SOC_TLV320AIC23
depends on I2C
config SND_SOC_TLV320AIC26
- tristate "TI TLV320AIC26 Codec support"
+ tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE
depends on SPI
config SND_SOC_TLV320AIC3X
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index bac7815e00f..44308dac9e1 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -84,7 +84,7 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int value)
{
- u8 data;
+ u8 data[2];
/* TLV320AIC23 has 7 bit address and 9 bits of data
* so we need to switch one data bit into reg and rest
@@ -96,12 +96,12 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg,
return -1;
}
- data = (reg << 1) | (value >> 8 & 0x01);
+ data[0] = (reg << 1) | (value >> 8 & 0x01);
+ data[1] = value & 0xff;
tlv320aic23_write_reg_cache(codec, reg, value);
- if (codec->hw_write(codec->control_data, data,
- (value & 0xff)) == 0)
+ if (codec->hw_write(codec->control_data, data, 2) == 2)
return 0;
printk(KERN_ERR "%s cannot write %03x to register R%d\n", __func__,
@@ -674,7 +674,7 @@ static int tlv320aic23_probe(struct platform_device *pdev)
tlv320aic23_socdev = socdev;
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
- codec->hw_write = (hw_write_t) i2c_smbus_write_byte_data;
+ codec->hw_write = (hw_write_t) i2c_master_send;
codec->hw_read = NULL;
ret = i2c_add_driver(&tlv320aic23_i2c_driver);
if (ret != 0)
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 05336ed7e49..cff276ee261 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -863,17 +863,21 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
return -EINVAL;
}
- /* interface format */
- switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
- case SND_SOC_DAIFMT_I2S:
+ /*
+ * match both interface format and signal polarities since they
+ * are fixed
+ */
+ switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK |
+ SND_SOC_DAIFMT_INV_MASK)) {
+ case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF):
break;
- case SND_SOC_DAIFMT_DSP_A:
+ case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF):
iface_breg |= (0x01 << 6);
break;
- case SND_SOC_DAIFMT_RIGHT_J:
+ case (SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_NB_NF):
iface_breg |= (0x02 << 6);
break;
- case SND_SOC_DAIFMT_LEFT_J:
+ case (SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF):
iface_breg |= (0x03 << 6);
break;
default:
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index aba402b3c99..945b32ed988 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -140,7 +140,7 @@ SOC_SINGLE("Capture ADC Boost (+20dB) Switch", AC97_VIDEO, 6, 1, 0),
SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0),
SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0),
-SOC_SINGLE("ALC Decay Time ", AC97_CODEC_CLASS_REV, 4, 15, 0),
+SOC_SINGLE("ALC Decay Time", AC97_CODEC_CLASS_REV, 4, 15, 0),
SOC_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0),
SOC_ENUM("ALC Function", wm9713_enum[6]),
SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0),
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 86923299bc1..94a02eaa482 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -277,7 +277,7 @@ static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
u16 imr;
u8 psc_cmd;
- long flags;
+ unsigned long flags;
if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
s = &psc_i2s->capture;
@@ -699,9 +699,11 @@ static ssize_t psc_i2s_stat_store(struct device *dev,
return count;
}
-DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL);
-DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show, psc_i2s_stat_store);
-DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show, psc_i2s_stat_store);
+static DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL);
+static DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show,
+ psc_i2s_stat_store);
+static DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show,
+ psc_i2s_stat_store);
/* ---------------------------------------------------------------------
* OF platform bus binding code:
@@ -819,8 +821,8 @@ static int __devinit psc_i2s_of_probe(struct of_device *op,
/* Register the SYSFS files */
rc = device_create_file(psc_i2s->dev, &dev_attr_status);
- rc = device_create_file(psc_i2s->dev, &dev_attr_capture_overrun);
- rc = device_create_file(psc_i2s->dev, &dev_attr_playback_underrun);
+ rc |= device_create_file(psc_i2s->dev, &dev_attr_capture_overrun);
+ rc |= device_create_file(psc_i2s->dev, &dev_attr_playback_underrun);
if (rc)
dev_info(psc_i2s->dev, "error creating sysfs files\n");
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 0a063a98a66..8485a8a9d0f 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -43,6 +43,7 @@
struct omap_mcbsp_data {
unsigned int bus_id;
struct omap_mcbsp_reg_cfg regs;
+ unsigned int fmt;
/*
* Flags indicating is the bus already activated and configured by
* another substream
@@ -200,6 +201,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id;
+ int wlen;
unsigned long port;
if (cpu_class_is_omap1()) {
@@ -244,19 +246,29 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
/* Set word lengths */
+ wlen = 16;
regs->rcr2 |= RWDLEN2(OMAP_MCBSP_WORD_16);
regs->rcr1 |= RWDLEN1(OMAP_MCBSP_WORD_16);
regs->xcr2 |= XWDLEN2(OMAP_MCBSP_WORD_16);
regs->xcr1 |= XWDLEN1(OMAP_MCBSP_WORD_16);
- /* Set FS period and length in terms of bit clock periods */
- regs->srgr2 |= FPER(16 * 2 - 1);
- regs->srgr1 |= FWID(16 - 1);
break;
default:
/* Unsupported PCM format */
return -EINVAL;
}
+ /* Set FS period and length in terms of bit clock periods */
+ switch (mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ regs->srgr2 |= FPER(wlen * 2 - 1);
+ regs->srgr1 |= FWID(wlen - 1);
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ regs->srgr2 |= FPER(wlen * 2 - 1);
+ regs->srgr1 |= FWID(wlen * 2 - 2);
+ break;
+ }
+
omap_mcbsp_config(bus_id, &mcbsp_data->regs);
mcbsp_data->configured = 1;
@@ -276,6 +288,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
if (mcbsp_data->configured)
return 0;
+ mcbsp_data->fmt = fmt;
memset(regs, 0, sizeof(*regs));
/* Generic McBSP register settings */
regs->spcr2 |= XINTM(3) | FREE;
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index dd7fa0b329c..2718eaf7895 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -18,13 +18,13 @@
#include <linux/timer.h>
#include <linux/interrupt.h>
#include <linux/platform_device.h>
+#include <linux/gpio.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <asm/mach-types.h>
-#include <asm/hardware/scoop.h>
#include <mach/pxa-regs.h>
#include <mach/hardware.h>
#include <mach/corgi.h>
@@ -54,8 +54,8 @@ static void corgi_ext_control(struct snd_soc_codec *codec)
switch (corgi_jack_func) {
case CORGI_HP:
/* set = unmute headphone */
- set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
- set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+ gpio_set_value(CORGI_GPIO_MUTE_L, 1);
+ gpio_set_value(CORGI_GPIO_MUTE_R, 1);
snd_soc_dapm_disable_pin(codec, "Mic Jack");
snd_soc_dapm_disable_pin(codec, "Line Jack");
snd_soc_dapm_enable_pin(codec, "Headphone Jack");
@@ -63,24 +63,24 @@ static void corgi_ext_control(struct snd_soc_codec *codec)
break;
case CORGI_MIC:
/* reset = mute headphone */
- reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
- reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+ gpio_set_value(CORGI_GPIO_MUTE_L, 0);
+ gpio_set_value(CORGI_GPIO_MUTE_R, 0);
snd_soc_dapm_enable_pin(codec, "Mic Jack");
snd_soc_dapm_disable_pin(codec, "Line Jack");
snd_soc_dapm_disable_pin(codec, "Headphone Jack");
snd_soc_dapm_disable_pin(codec, "Headset Jack");
break;
case CORGI_LINE:
- reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
- reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+ gpio_set_value(CORGI_GPIO_MUTE_L, 0);
+ gpio_set_value(CORGI_GPIO_MUTE_R, 0);
snd_soc_dapm_disable_pin(codec, "Mic Jack");
snd_soc_dapm_enable_pin(codec, "Line Jack");
snd_soc_dapm_disable_pin(codec, "Headphone Jack");
snd_soc_dapm_disable_pin(codec, "Headset Jack");
break;
case CORGI_HEADSET:
- reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
- set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+ gpio_set_value(CORGI_GPIO_MUTE_L, 0);
+ gpio_set_value(CORGI_GPIO_MUTE_R, 1);
snd_soc_dapm_enable_pin(codec, "Mic Jack");
snd_soc_dapm_disable_pin(codec, "Line Jack");
snd_soc_dapm_disable_pin(codec, "Headphone Jack");
@@ -114,8 +114,8 @@ static int corgi_shutdown(struct snd_pcm_substream *substream)
struct snd_soc_codec *codec = rtd->socdev->codec;
/* set = unmute headphone */
- set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
- set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+ gpio_set_value(CORGI_GPIO_MUTE_L, 1);
+ gpio_set_value(CORGI_GPIO_MUTE_R, 1);
return 0;
}
@@ -218,22 +218,14 @@ static int corgi_set_spk(struct snd_kcontrol *kcontrol,
static int corgi_amp_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
- if (SND_SOC_DAPM_EVENT_ON(event))
- set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON);
- else
- reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON);
-
+ gpio_set_value(CORGI_GPIO_APM_ON, SND_SOC_DAPM_EVENT_ON(event));
return 0;
}
static int corgi_mic_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
- if (SND_SOC_DAPM_EVENT_ON(event))
- set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MIC_BIAS);
- else
- reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MIC_BIAS);
-
+ gpio_set_value(CORGI_GPIO_MIC_BIAS, SND_SOC_DAPM_EVENT_ON(event));
return 0;
}
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index a80ae074b09..a7a3a9c5c6f 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -49,7 +49,7 @@ struct snd_ac97_bus_ops soc_ac97_ops = {
static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_out = {
.name = "AC97 PCM Stereo out",
.dev_addr = __PREG(PCDR),
- .drcmr = &DRCMRTXPCDR,
+ .drcmr = &DRCMR(12),
.dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
DCMD_BURST32 | DCMD_WIDTH4,
};
@@ -57,7 +57,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_out = {
static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_in = {
.name = "AC97 PCM Stereo in",
.dev_addr = __PREG(PCDR),
- .drcmr = &DRCMRRXPCDR,
+ .drcmr = &DRCMR(11),
.dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
DCMD_BURST32 | DCMD_WIDTH4,
};
@@ -65,7 +65,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_in = {
static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_out = {
.name = "AC97 Aux PCM (Slot 5) Mono out",
.dev_addr = __PREG(MODR),
- .drcmr = &DRCMRTXMODR,
+ .drcmr = &DRCMR(10),
.dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
DCMD_BURST16 | DCMD_WIDTH2,
};
@@ -73,7 +73,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_out = {
static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_in = {
.name = "AC97 Aux PCM (Slot 5) Mono in",
.dev_addr = __PREG(MODR),
- .drcmr = &DRCMRRXMODR,
+ .drcmr = &DRCMR(9),
.dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
DCMD_BURST16 | DCMD_WIDTH2,
};
@@ -81,7 +81,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_in = {
static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_mic_mono_in = {
.name = "AC97 Mic PCM (Slot 6) Mono in",
.dev_addr = __PREG(MCDR),
- .drcmr = &DRCMRRXMCDR,
+ .drcmr = &DRCMR(8),
.dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
DCMD_BURST16 | DCMD_WIDTH2,
};
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 64057b1d220..e758034db5c 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -39,6 +39,45 @@ struct pxa2xx_gpio {
u32 frm;
};
+/*
+ * I2S Controller Register and Bit Definitions
+ */
+#define SACR0 __REG(0x40400000) /* Global Control Register */
+#define SACR1 __REG(0x40400004) /* Serial Audio I 2 S/MSB-Justified Control Register */
+#define SASR0 __REG(0x4040000C) /* Serial Audio I 2 S/MSB-Justified Interface and FIFO Status Register */
+#define SAIMR __REG(0x40400014) /* Serial Audio Interrupt Mask Register */
+#define SAICR __REG(0x40400018) /* Serial Audio Interrupt Clear Register */
+#define SADIV __REG(0x40400060) /* Audio Clock Divider Register. */
+#define SADR __REG(0x40400080) /* Serial Audio Data Register (TX and RX FIFO access Register). */
+
+#define SACR0_RFTH(x) ((x) << 12) /* Rx FIFO Interrupt or DMA Trigger Threshold */
+#define SACR0_TFTH(x) ((x) << 8) /* Tx FIFO Interrupt or DMA Trigger Threshold */
+#define SACR0_STRF (1 << 5) /* FIFO Select for EFWR Special Function */
+#define SACR0_EFWR (1 << 4) /* Enable EFWR Function */
+#define SACR0_RST (1 << 3) /* FIFO, i2s Register Reset */
+#define SACR0_BCKD (1 << 2) /* Bit Clock Direction */
+#define SACR0_ENB (1 << 0) /* Enable I2S Link */
+#define SACR1_ENLBF (1 << 5) /* Enable Loopback */
+#define SACR1_DRPL (1 << 4) /* Disable Replaying Function */
+#define SACR1_DREC (1 << 3) /* Disable Recording Function */
+#define SACR1_AMSL (1 << 0) /* Specify Alternate Mode */
+
+#define SASR0_I2SOFF (1 << 7) /* Controller Status */
+#define SASR0_ROR (1 << 6) /* Rx FIFO Overrun */
+#define SASR0_TUR (1 << 5) /* Tx FIFO Underrun */
+#define SASR0_RFS (1 << 4) /* Rx FIFO Service Request */
+#define SASR0_TFS (1 << 3) /* Tx FIFO Service Request */
+#define SASR0_BSY (1 << 2) /* I2S Busy */
+#define SASR0_RNE (1 << 1) /* Rx FIFO Not Empty */
+#define SASR0_TNF (1 << 0) /* Tx FIFO Not Empty */
+
+#define SAICR_ROR (1 << 6) /* Clear Rx FIFO Overrun Interrupt */
+#define SAICR_TUR (1 << 5) /* Clear Tx FIFO Underrun Interrupt */
+
+#define SAIMR_ROR (1 << 6) /* Enable Rx FIFO Overrun Condition Interrupt */
+#define SAIMR_TUR (1 << 5) /* Enable Tx FIFO Underrun Condition Interrupt */
+#define SAIMR_RFS (1 << 4) /* Enable Rx FIFO Service Interrupt */
+#define SAIMR_TFS (1 << 3) /* Enable Tx FIFO Service Interrupt */
struct pxa_i2s_port {
u32 sadiv;
@@ -54,7 +93,7 @@ static struct clk *clk_i2s;
static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_out = {
.name = "I2S PCM Stereo out",
.dev_addr = __PREG(SADR),
- .drcmr = &DRCMRTXSADR,
+ .drcmr = &DRCMR(3),
.dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
DCMD_BURST32 | DCMD_WIDTH4,
};
@@ -62,7 +101,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_out = {
static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_in = {
.name = "I2S PCM Stereo in",
.dev_addr = __PREG(SADR),
- .drcmr = &DRCMRRXSADR,
+ .drcmr = &DRCMR(2),
.dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
DCMD_BURST32 | DCMD_WIDTH4,
};
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index 8f89188e541..d307b6757e9 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -19,16 +19,15 @@
#include <linux/timer.h>
#include <linux/interrupt.h>
#include <linux/platform_device.h>
+#include <linux/gpio.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <asm/mach-types.h>
-#include <asm/hardware/scoop.h>
#include <mach/pxa-regs.h>
#include <mach/hardware.h>
-#include <mach/akita.h>
#include <mach/spitz.h>
#include "../codecs/wm8750.h"
#include "pxa2xx-pcm.h"
@@ -63,8 +62,8 @@ static void spitz_ext_control(struct snd_soc_codec *codec)
snd_soc_dapm_disable_pin(codec, "Mic Jack");
snd_soc_dapm_disable_pin(codec, "Line Jack");
snd_soc_dapm_enable_pin(codec, "Headphone Jack");
- set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
- set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
+ gpio_set_value(SPITZ_GPIO_MUTE_L, 1);
+ gpio_set_value(SPITZ_GPIO_MUTE_R, 1);
break;
case SPITZ_MIC:
/* enable mic jack and bias, mute hp */
@@ -72,8 +71,8 @@ static void spitz_ext_control(struct snd_soc_codec *codec)
snd_soc_dapm_disable_pin(codec, "Headset Jack");
snd_soc_dapm_disable_pin(codec, "Line Jack");
snd_soc_dapm_enable_pin(codec, "Mic Jack");
- reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
- reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
+ gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
+ gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
break;
case SPITZ_LINE:
/* enable line jack, disable mic bias and mute hp */
@@ -81,8 +80,8 @@ static void spitz_ext_control(struct snd_soc_codec *codec)
snd_soc_dapm_disable_pin(codec, "Headset Jack");
snd_soc_dapm_disable_pin(codec, "Mic Jack");
snd_soc_dapm_enable_pin(codec, "Line Jack");
- reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
- reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
+ gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
+ gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
break;
case SPITZ_HEADSET:
/* enable and unmute headset jack enable mic bias, mute L hp */
@@ -90,8 +89,8 @@ static void spitz_ext_control(struct snd_soc_codec *codec)
snd_soc_dapm_enable_pin(codec, "Mic Jack");
snd_soc_dapm_disable_pin(codec, "Line Jack");
snd_soc_dapm_enable_pin(codec, "Headset Jack");
- reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
- set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
+ gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
+ gpio_set_value(SPITZ_GPIO_MUTE_R, 1);
break;
case SPITZ_HP_OFF:
@@ -100,8 +99,8 @@ static void spitz_ext_control(struct snd_soc_codec *codec)
snd_soc_dapm_disable_pin(codec, "Headset Jack");
snd_soc_dapm_disable_pin(codec, "Mic Jack");
snd_soc_dapm_disable_pin(codec, "Line Jack");
- reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
- reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
+ gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
+ gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
break;
}
snd_soc_dapm_sync(codec);
@@ -215,23 +214,14 @@ static int spitz_set_spk(struct snd_kcontrol *kcontrol,
static int spitz_mic_bias(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
- if (machine_is_borzoi() || machine_is_spitz()) {
- if (SND_SOC_DAPM_EVENT_ON(event))
- set_scoop_gpio(&spitzscoop2_device.dev,
- SPITZ_SCP2_MIC_BIAS);
- else
- reset_scoop_gpio(&spitzscoop2_device.dev,
- SPITZ_SCP2_MIC_BIAS);
- }
+ if (machine_is_borzoi() || machine_is_spitz())
+ gpio_set_value(SPITZ_GPIO_MIC_BIAS,
+ SND_SOC_DAPM_EVENT_ON(event));
+
+ if (machine_is_akita())
+ gpio_set_value(AKITA_GPIO_MIC_BIAS,
+ SND_SOC_DAPM_EVENT_ON(event));
- if (machine_is_akita()) {
- if (SND_SOC_DAPM_EVENT_ON(event))
- akita_set_ioexp(&akitaioexp_device.dev,
- AKITA_IOEXP_MIC_BIAS);
- else
- akita_reset_ioexp(&akitaioexp_device.dev,
- AKITA_IOEXP_MIC_BIAS);
- }
return 0;
}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 462e635dfc7..a3adbf06b1e 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1462,7 +1462,7 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol,
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
int max = mc->max;
- unsigned int shift = mc->min;
+ unsigned int shift = mc->shift;
unsigned int rshift = mc->rshift;
if (max == 1)
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index efbd0b37810..7351db9606e 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -831,7 +831,7 @@ int snd_soc_dapm_sys_add(struct device *dev)
return ret;
asoc_debugfs = debugfs_create_dir("asoc", NULL);
- if (!IS_ERR(asoc_debugfs))
+ if (!IS_ERR(asoc_debugfs) && asoc_debugfs)
debugfs_create_u32("dapm_pop_time", 0744, asoc_debugfs,
&pop_time);
else
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-23 14:06:57 +0000