* [PATCH 01/17] ASoC: Fix for master playback/capture volume range for TWL4030 codec
@ 2008-11-18 12:38 Mark Brown
2008-11-18 12:38 ` [PATCH 02/17] ASoC: UDA134x codec driver Mark Brown
0 siblings, 1 reply; 17+ messages in thread
From: Mark Brown @ 2008-11-18 12:38 UTC (permalink / raw)
To: Takashi Iwai; +Cc: alsa-devel, Mark Brown, Peter Ujfalusi
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
FGAIN for playback is in range of 0-0x3f, while for capture GAIN it
is in the range of 0-0x1f.
The original value of 128 (0x7f) would modify the CGAIN also for
playback.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Steve Sakoman <steve@sakoman.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/codecs/twl4030.c | 4 ++--
1 files changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index c1893d2..c778eb4 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -192,10 +192,10 @@ static void twl4030_init_chip(struct snd_soc_codec *codec)
static const struct snd_kcontrol_new twl4030_snd_controls[] = {
SOC_DOUBLE_R("Master Playback Volume",
TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA,
- 0, 127, 0),
+ 0, 0x3f, 0),
SOC_DOUBLE_R("Capture Volume",
TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA,
- 0, 127, 0),
+ 0, 0x1f, 0),
};
/* add non dapm controls */
--
1.5.6.5
^ permalink raw reply related [flat|nested] 17+ messages in thread
* [PATCH 02/17] ASoC: UDA134x codec driver
2008-11-18 12:38 [PATCH 01/17] ASoC: Fix for master playback/capture volume range for TWL4030 codec Mark Brown
@ 2008-11-18 12:38 ` Mark Brown
2008-11-18 12:38 ` [PATCH 03/17] ASoC: Machine driver for for s3c24xx with uda134x Mark Brown
0 siblings, 1 reply; 17+ messages in thread
From: Mark Brown @ 2008-11-18 12:38 UTC (permalink / raw)
To: Takashi Iwai
Cc: alsa-devel, Mark Brown, Christian Pellegrin, Christian Pellegrin
From: Christian Pellegrin <chripell@gmail.com>
Signed-off-by: Christian Pellegrin <chripell@fsfe.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
include/sound/l3.h | 18 +
include/sound/uda134x.h | 26 ++
sound/soc/codecs/Kconfig | 8 +
sound/soc/codecs/Makefile | 4 +
sound/soc/codecs/l3.c | 91 ++++++
sound/soc/codecs/uda134x.c | 656 ++++++++++++++++++++++++++++++++++++++
sound/soc/codecs/uda134x_codec.h | 36 ++
7 files changed, 839 insertions(+), 0 deletions(-)
create mode 100644 include/sound/l3.h
create mode 100644 include/sound/uda134x.h
create mode 100644 sound/soc/codecs/l3.c
create mode 100644 sound/soc/codecs/uda134x.c
create mode 100644 sound/soc/codecs/uda134x_codec.h
diff --git a/include/sound/l3.h b/include/sound/l3.h
new file mode 100644
index 0000000..423a08f
--- /dev/null
+++ b/include/sound/l3.h
@@ -0,0 +1,18 @@
+#ifndef _L3_H_
+#define _L3_H_ 1
+
+struct l3_pins {
+ void (*setdat)(int);
+ void (*setclk)(int);
+ void (*setmode)(int);
+ int data_hold;
+ int data_setup;
+ int clock_high;
+ int mode_hold;
+ int mode;
+ int mode_setup;
+};
+
+int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len);
+
+#endif
diff --git a/include/sound/uda134x.h b/include/sound/uda134x.h
new file mode 100644
index 0000000..475ef8b
--- /dev/null
+++ b/include/sound/uda134x.h
@@ -0,0 +1,26 @@
+/*
+ * uda134x.h -- UDA134x ALSA SoC Codec driver
+ *
+ * Copyright 2007 Dension Audio Systems Ltd.
+ * Author: Zoltan Devai
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _UDA134X_H
+#define _UDA134X_H
+
+#include <sound/l3.h>
+
+struct uda134x_platform_data {
+ struct l3_pins l3;
+ void (*power) (int);
+ int model;
+#define UDA134X_UDA1340 1
+#define UDA134X_UDA1341 2
+#define UDA134X_UDA1344 3
+};
+
+#endif /* _UDA134X_H */
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 8a84460..04f49f5 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -10,6 +10,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_TLV320AIC26 if SPI_MASTER
select SND_SOC_TLV320AIC3X if I2C
select SND_SOC_TWL4030 if TWL4030_CORE
+ select SND_SOC_UDA134X
select SND_SOC_UDA1380 if I2C
select SND_SOC_WM8510 if (I2C || SPI_MASTER)
select SND_SOC_WM8580 if I2C
@@ -66,6 +67,9 @@ config SND_SOC_CS4270_VD33_ERRATA
bool
depends on SND_SOC_CS4270
+config SND_SOC_L3
+ tristate
+
config SND_SOC_SSM2602
tristate
@@ -85,6 +89,10 @@ config SND_SOC_TWL4030
tristate
depends on TWL4030_CORE
+config SND_SOC_UDA134X
+ tristate
+ select SND_SOC_L3
+
config SND_SOC_UDA1380
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 7ae17a6..de65723 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -3,11 +3,13 @@ snd-soc-ad1980-objs := ad1980.o
snd-soc-ad73311-objs := ad73311.o
snd-soc-ak4535-objs := ak4535.o
snd-soc-cs4270-objs := cs4270.o
+snd-soc-l3-objs := l3.o
snd-soc-ssm2602-objs := ssm2602.o
snd-soc-tlv320aic23-objs := tlv320aic23.o
snd-soc-tlv320aic26-objs := tlv320aic26.o
snd-soc-tlv320aic3x-objs := tlv320aic3x.o
snd-soc-twl4030-objs := twl4030.o
+snd-soc-uda134x-objs := uda134x.o
snd-soc-uda1380-objs := uda1380.o
snd-soc-wm8510-objs := wm8510.o
snd-soc-wm8580-objs := wm8580.o
@@ -27,11 +29,13 @@ obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o
obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
+obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o
obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o
obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o
obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o
obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o
obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o
+obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o
obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o
obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o
obj-$(CONFIG_SND_SOC_WM8580) += snd-soc-wm8580.o
diff --git a/sound/soc/codecs/l3.c b/sound/soc/codecs/l3.c
new file mode 100644
index 0000000..5353af5
--- /dev/null
+++ b/sound/soc/codecs/l3.c
@@ -0,0 +1,91 @@
+/*
+ * L3 code
+ *
+ * Copyright (C) 2008, Christian Pellegrin <chripell@evolware.org>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ *
+ * based on:
+ *
+ * L3 bus algorithm module.
+ *
+ * Copyright (C) 2001 Russell King, All Rights Reserved.
+ *
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/delay.h>
+
+#include <sound/l3.h>
+
+/*
+ * Send one byte of data to the chip. Data is latched into the chip on
+ * the rising edge of the clock.
+ */
+static void sendbyte(struct l3_pins *adap, unsigned int byte)
+{
+ int i;
+
+ for (i = 0; i < 8; i++) {
+ adap->setclk(0);
+ udelay(adap->data_hold);
+ adap->setdat(byte & 1);
+ udelay(adap->data_setup);
+ adap->setclk(1);
+ udelay(adap->clock_high);
+ byte >>= 1;
+ }
+}
+
+/*
+ * Send a set of bytes to the chip. We need to pulse the MODE line
+ * between each byte, but never at the start nor at the end of the
+ * transfer.
+ */
+static void sendbytes(struct l3_pins *adap, const u8 *buf,
+ int len)
+{
+ int i;
+
+ for (i = 0; i < len; i++) {
+ if (i) {
+ udelay(adap->mode_hold);
+ adap->setmode(0);
+ udelay(adap->mode);
+ }
+ adap->setmode(1);
+ udelay(adap->mode_setup);
+ sendbyte(adap, buf[i]);
+ }
+}
+
+int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len)
+{
+ adap->setclk(1);
+ adap->setdat(1);
+ adap->setmode(1);
+ udelay(adap->mode);
+
+ adap->setmode(0);
+ udelay(adap->mode_setup);
+ sendbyte(adap, addr);
+ udelay(adap->mode_hold);
+
+ sendbytes(adap, data, len);
+
+ adap->setclk(1);
+ adap->setdat(1);
+ adap->setmode(0);
+
+ return len;
+}
+EXPORT_SYMBOL_GPL(l3_write);
+
+MODULE_DESCRIPTION("L3 bit-banging driver");
+MODULE_AUTHOR("Christian Pellegrin <chripell@evolware.org>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
new file mode 100644
index 0000000..04b30da
--- /dev/null
+++ b/sound/soc/codecs/uda134x.c
@@ -0,0 +1,656 @@
+/*
+ * uda134x.c -- UDA134X ALSA SoC Codec driver
+ *
+ * Modifications by Christian Pellegrin <chripell@evolware.org>
+ *
+ * Copyright 2007 Dension Audio Systems Ltd.
+ * Author: Zoltan Devai
+ *
+ * Based on the WM87xx drivers by Liam Girdwood and Richard Purdie
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/delay.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include <sound/uda134x.h>
+#include <sound/l3.h>
+
+#include "uda134x_codec.h"
+
+
+#define POWER_OFF_ON_STANDBY 1
+/*
+ ALSA SOC usually puts the device in standby mode when it's not used
+ for sometime. If you define POWER_OFF_ON_STANDBY the driver will
+ turn off the ADC/DAC when this callback is invoked and turn it back
+ on when needed. Unfortunately this will result in a very light bump
+ (it can be audible only with good earphones). If this bothers you
+ just comment this line, you will have slightly higher power
+ consumption . Please note that sending the L3 command for ADC is
+ enough to make the bump, so it doesn't make difference if you
+ completely take off power from the codec.
+ */
+
+#define UDA134X_RATES SNDRV_PCM_RATE_8000_48000
+#define UDA134X_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE)
+
+struct uda134x_priv {
+ int sysclk;
+ int dai_fmt;
+
+ struct snd_pcm_substream *master_substream;
+ struct snd_pcm_substream *slave_substream;
+};
+
+/* In-data addresses are hard-coded into the reg-cache values */
+static const char uda134x_reg[UDA134X_REGS_NUM] = {
+ /* Extended address registers */
+ 0x04, 0x04, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00,
+ /* Status, data regs */
+ 0x00, 0x83, 0x00, 0x40, 0x80, 0x00,
+};
+
+/*
+ * The codec has no support for reading its registers except for peak level...
+ */
+static inline unsigned int uda134x_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u8 *cache = codec->reg_cache;
+
+ if (reg >= UDA134X_REGS_NUM)
+ return -1;
+ return cache[reg];
+}
+
+/*
+ * Write the register cache
+ */
+static inline void uda134x_write_reg_cache(struct snd_soc_codec *codec,
+ u8 reg, unsigned int value)
+{
+ u8 *cache = codec->reg_cache;
+
+ if (reg >= UDA134X_REGS_NUM)
+ return;
+ cache[reg] = value;
+}
+
+/*
+ * Write to the uda134x registers
+ *
+ */
+static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ int ret;
+ u8 addr;
+ u8 data = value;
+ struct uda134x_platform_data *pd = codec->control_data;
+
+ pr_debug("%s reg: %02X, value:%02X\n", __func__, reg, value);
+
+ if (reg >= UDA134X_REGS_NUM) {
+ printk(KERN_ERR "%s unkown register: reg: %d",
+ __func__, reg);
+ return -EINVAL;
+ }
+
+ uda134x_write_reg_cache(codec, reg, value);
+
+ switch (reg) {
+ case UDA134X_STATUS0:
+ case UDA134X_STATUS1:
+ addr = UDA134X_STATUS_ADDR;
+ break;
+ case UDA134X_DATA000:
+ case UDA134X_DATA001:
+ case UDA134X_DATA010:
+ addr = UDA134X_DATA0_ADDR;
+ break;
+ case UDA134X_DATA1:
+ addr = UDA134X_DATA1_ADDR;
+ break;
+ default:
+ /* It's an extended address register */
+ addr = (reg | UDA134X_EXTADDR_PREFIX);
+
+ ret = l3_write(&pd->l3,
+ UDA134X_DATA0_ADDR, &addr, 1);
+ if (ret != 1)
+ return -EIO;
+
+ addr = UDA134X_DATA0_ADDR;
+ data = (value | UDA134X_EXTDATA_PREFIX);
+ break;
+ }
+
+ ret = l3_write(&pd->l3,
+ addr, &data, 1);
+ if (ret != 1)
+ return -EIO;
+
+ return 0;
+}
+
+static inline void uda134x_reset(struct snd_soc_codec *codec)
+{
+ u8 reset_reg = uda134x_read_reg_cache(codec, UDA134X_STATUS0);
+ uda134x_write(codec, UDA134X_STATUS0, reset_reg | (1<<6));
+ msleep(1);
+ uda134x_write(codec, UDA134X_STATUS0, reset_reg & ~(1<<6));
+}
+
+static int uda134x_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u8 mute_reg = uda134x_read_reg_cache(codec, UDA134X_DATA010);
+
+ pr_debug("%s mute: %d\n", __func__, mute);
+
+ if (mute)
+ mute_reg |= (1<<2);
+ else
+ mute_reg &= ~(1<<2);
+
+ uda134x_write(codec, UDA134X_DATA010, mute_reg & ~(1<<2));
+
+ return 0;
+}
+
+static int uda134x_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct uda134x_priv *uda134x = codec->private_data;
+ struct snd_pcm_runtime *master_runtime;
+
+ if (uda134x->master_substream) {
+ master_runtime = uda134x->master_substream->runtime;
+
+ pr_debug("%s constraining to %d bits at %d\n", __func__,
+ master_runtime->sample_bits,
+ master_runtime->rate);
+
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE,
+ master_runtime->rate,
+ master_runtime->rate);
+
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+ master_runtime->sample_bits,
+ master_runtime->sample_bits);
+
+ uda134x->slave_substream = substream;
+ } else
+ uda134x->master_substream = substream;
+
+ return 0;
+}
+
+static void uda134x_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct uda134x_priv *uda134x = codec->private_data;
+
+ if (uda134x->master_substream == substream)
+ uda134x->master_substream = uda134x->slave_substream;
+
+ uda134x->slave_substream = NULL;
+}
+
+static int uda134x_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct uda134x_priv *uda134x = codec->private_data;
+ u8 hw_params;
+
+ if (substream == uda134x->slave_substream) {
+ pr_debug("%s ignoring hw_params for slave substream\n",
+ __func__);
+ return 0;
+ }
+
+ hw_params = uda134x_read_reg_cache(codec, UDA134X_STATUS0);
+ hw_params &= STATUS0_SYSCLK_MASK;
+ hw_params &= STATUS0_DAIFMT_MASK;
+
+ pr_debug("%s sysclk: %d, rate:%d\n", __func__,
+ uda134x->sysclk, params_rate(params));
+
+ /* set SYSCLK / fs ratio */
+ switch (uda134x->sysclk / params_rate(params)) {
+ case 512:
+ break;
+ case 384:
+ hw_params |= (1<<4);
+ break;
+ case 256:
+ hw_params |= (1<<5);
+ break;
+ default:
+ printk(KERN_ERR "%s unsupported fs\n", __func__);
+ return -EINVAL;
+ }
+
+ pr_debug("%s dai_fmt: %d, params_format:%d\n", __func__,
+ uda134x->dai_fmt, params_format(params));
+
+ /* set DAI format and word length */
+ switch (uda134x->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ hw_params |= (1<<1);
+ break;
+ case SNDRV_PCM_FORMAT_S18_3LE:
+ hw_params |= (1<<2);
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ hw_params |= ((1<<2) | (1<<1));
+ break;
+ default:
+ printk(KERN_ERR "%s unsupported format (right)\n",
+ __func__);
+ return -EINVAL;
+ }
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ hw_params |= (1<<3);
+ break;
+ default:
+ printk(KERN_ERR "%s unsupported format\n", __func__);
+ return -EINVAL;
+ }
+
+ uda134x_write(codec, UDA134X_STATUS0, hw_params);
+
+ return 0;
+}
+
+static int uda134x_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct uda134x_priv *uda134x = codec->private_data;
+
+ pr_debug("%s clk_id: %d, freq: %d, dir: %d\n", __func__,
+ clk_id, freq, dir);
+
+ /* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable
+ because the codec is slave. Of course limitations of the clock
+ master (the IIS controller) apply.
+ We'll error out on set_hw_params if it's not OK */
+ if ((freq >= (256 * 8000)) && (freq <= (512 * 48000))) {
+ uda134x->sysclk = freq;
+ return 0;
+ }
+
+ printk(KERN_ERR "%s unsupported sysclk\n", __func__);
+ return -EINVAL;
+}
+
+static int uda134x_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct uda134x_priv *uda134x = codec->private_data;
+
+ pr_debug("%s fmt: %08X\n", __func__, fmt);
+
+ /* codec supports only full slave mode */
+ if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) {
+ printk(KERN_ERR "%s unsupported slave mode\n", __func__);
+ return -EINVAL;
+ }
+
+ /* no support for clock inversion */
+ if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) {
+ printk(KERN_ERR "%s unsupported clock inversion\n", __func__);
+ return -EINVAL;
+ }
+
+ /* We can't setup DAI format here as it depends on the word bit num */
+ /* so let's just store the value for later */
+ uda134x->dai_fmt = fmt;
+
+ return 0;
+}
+
+static int uda134x_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u8 reg;
+ struct uda134x_platform_data *pd = codec->control_data;
+ int i;
+ u8 *cache = codec->reg_cache;
+
+ pr_debug("%s bias level %d\n", __func__, level);
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ /* ADC, DAC on */
+ reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1);
+ uda134x_write(codec, UDA134X_STATUS1, reg | 0x03);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ /* power on */
+ if (pd->power) {
+ pd->power(1);
+ /* Sync reg_cache with the hardware */
+ for (i = 0; i < ARRAY_SIZE(uda134x_reg); i++)
+ codec->write(codec, i, *cache++);
+ }
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ /* ADC, DAC power off */
+ reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1);
+ uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03));
+ break;
+ case SND_SOC_BIAS_OFF:
+ /* power off */
+ if (pd->power)
+ pd->power(0);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+static const char *uda134x_dsp_setting[] = {"Flat", "Minimum1",
+ "Minimum2", "Maximum"};
+static const char *uda134x_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+static const char *uda134x_mixmode[] = {"Differential", "Analog1",
+ "Analog2", "Both"};
+
+static const struct soc_enum uda134x_mixer_enum[] = {
+SOC_ENUM_SINGLE(UDA134X_DATA010, 0, 0x04, uda134x_dsp_setting),
+SOC_ENUM_SINGLE(UDA134X_DATA010, 3, 0x04, uda134x_deemph),
+SOC_ENUM_SINGLE(UDA134X_EA010, 0, 0x04, uda134x_mixmode),
+};
+
+static const struct snd_kcontrol_new uda1341_snd_controls[] = {
+SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1),
+SOC_SINGLE("Capture Volume", UDA134X_EA010, 2, 0x07, 0),
+SOC_SINGLE("Analog1 Volume", UDA134X_EA000, 0, 0x1F, 1),
+SOC_SINGLE("Analog2 Volume", UDA134X_EA001, 0, 0x1F, 1),
+
+SOC_SINGLE("Mic Sensitivity", UDA134X_EA010, 2, 7, 0),
+SOC_SINGLE("Mic Volume", UDA134X_EA101, 0, 0x1F, 0),
+
+SOC_SINGLE("Tone Control - Bass", UDA134X_DATA001, 2, 0xF, 0),
+SOC_SINGLE("Tone Control - Treble", UDA134X_DATA001, 0, 3, 0),
+
+SOC_ENUM("Sound Processing Filter", uda134x_mixer_enum[0]),
+SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]),
+SOC_ENUM("Input Mux", uda134x_mixer_enum[2]),
+
+SOC_SINGLE("AGC Switch", UDA134X_EA100, 4, 1, 0),
+SOC_SINGLE("AGC Target Volume", UDA134X_EA110, 0, 0x03, 1),
+SOC_SINGLE("AGC Timing", UDA134X_EA110, 2, 0x07, 0),
+
+SOC_SINGLE("DAC +6dB Switch", UDA134X_STATUS1, 6, 1, 0),
+SOC_SINGLE("ADC +6dB Switch", UDA134X_STATUS1, 5, 1, 0),
+SOC_SINGLE("ADC Polarity Switch", UDA134X_STATUS1, 4, 1, 0),
+SOC_SINGLE("DAC Polarity Switch", UDA134X_STATUS1, 3, 1, 0),
+SOC_SINGLE("Double Speed Playback Switch", UDA134X_STATUS1, 2, 1, 0),
+SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new uda1340_snd_controls[] = {
+SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1),
+
+SOC_SINGLE("Tone Control - Bass", UDA134X_DATA001, 2, 0xF, 0),
+SOC_SINGLE("Tone Control - Treble", UDA134X_DATA001, 0, 3, 0),
+
+SOC_ENUM("Sound Processing Filter", uda134x_mixer_enum[0]),
+SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]),
+
+SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0),
+};
+
+static int uda134x_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i, n;
+ const struct snd_kcontrol_new *ctrls;
+ struct uda134x_platform_data *pd = codec->control_data;
+
+ switch (pd->model) {
+ case UDA134X_UDA1340:
+ case UDA134X_UDA1344:
+ n = ARRAY_SIZE(uda1340_snd_controls);
+ ctrls = uda1340_snd_controls;
+ break;
+ case UDA134X_UDA1341:
+ n = ARRAY_SIZE(uda1341_snd_controls);
+ ctrls = uda1341_snd_controls;
+ break;
+ default:
+ printk(KERN_ERR "%s unkown codec type: %d",
+ __func__, pd->model);
+ return -EINVAL;
+ }
+
+ for (i = 0; i < n; i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&ctrls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+struct snd_soc_dai uda134x_dai = {
+ .name = "UDA134X",
+ /* playback capabilities */
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = UDA134X_RATES,
+ .formats = UDA134X_FORMATS,
+ },
+ /* capture capabilities */
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = UDA134X_RATES,
+ .formats = UDA134X_FORMATS,
+ },
+ /* pcm operations */
+ .ops = {
+ .startup = uda134x_startup,
+ .shutdown = uda134x_shutdown,
+ .hw_params = uda134x_hw_params,
+ },
+ /* DAI operations */
+ .dai_ops = {
+ .digital_mute = uda134x_mute,
+ .set_sysclk = uda134x_set_dai_sysclk,
+ .set_fmt = uda134x_set_dai_fmt,
+ }
+};
+EXPORT_SYMBOL(uda134x_dai);
+
+
+static int uda134x_soc_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ struct uda134x_priv *uda134x;
+ void *codec_setup_data = socdev->codec_data;
+ int ret = -ENOMEM;
+ struct uda134x_platform_data *pd;
+
+ printk(KERN_INFO "UDA134X SoC Audio Codec\n");
+
+ if (!codec_setup_data) {
+ printk(KERN_ERR "UDA134X SoC codec: "
+ "missing L3 bitbang function\n");
+ return -ENODEV;
+ }
+
+ pd = codec_setup_data;
+ switch (pd->model) {
+ case UDA134X_UDA1340:
+ case UDA134X_UDA1341:
+ case UDA134X_UDA1344:
+ break;
+ default:
+ printk(KERN_ERR "UDA134X SoC codec: "
+ "unsupported model %d\n",
+ pd->model);
+ return -EINVAL;
+ }
+
+ socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (socdev->codec == NULL)
+ return ret;
+
+ codec = socdev->codec;
+
+ uda134x = kzalloc(sizeof(struct uda134x_priv), GFP_KERNEL);
+ if (uda134x == NULL)
+ goto priv_err;
+ codec->private_data = uda134x;
+
+ codec->reg_cache = kmemdup(uda134x_reg, sizeof(uda134x_reg),
+ GFP_KERNEL);
+ if (codec->reg_cache == NULL)
+ goto reg_err;
+
+ mutex_init(&codec->mutex);
+
+ codec->reg_cache_size = sizeof(uda134x_reg);
+ codec->reg_cache_step = 1;
+
+ codec->name = "UDA134X";
+ codec->owner = THIS_MODULE;
+ codec->dai = &uda134x_dai;
+ codec->num_dai = 1;
+ codec->read = uda134x_read_reg_cache;
+ codec->write = uda134x_write;
+#ifdef POWER_OFF_ON_STANDBY
+ codec->set_bias_level = uda134x_set_bias_level;
+#endif
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->control_data = codec_setup_data;
+
+ if (pd->power)
+ pd->power(1);
+
+ uda134x_reset(codec);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "UDA134X: failed to register pcms\n");
+ goto pcm_err;
+ }
+
+ ret = uda134x_add_controls(codec);
+ if (ret < 0) {
+ printk(KERN_ERR "UDA134X: failed to register controls\n");
+ goto pcm_err;
+ }
+
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "UDA134X: failed to register card\n");
+ goto card_err;
+ }
+
+ return 0;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ kfree(codec->reg_cache);
+reg_err:
+ kfree(codec->private_data);
+priv_err:
+ kfree(codec);
+ return ret;
+}
+
+/* power down chip */
+static int uda134x_soc_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+
+ kfree(codec->private_data);
+ kfree(codec->reg_cache);
+ kfree(codec);
+
+ return 0;
+}
+
+#if defined(CONFIG_PM)
+static int uda134x_soc_suspend(struct platform_device *pdev,
+ pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int uda134x_soc_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ uda134x_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
+ uda134x_set_bias_level(codec, SND_SOC_BIAS_ON);
+ return 0;
+}
+#else
+#define uda134x_soc_suspend NULL
+#define uda134x_soc_resume NULL
+#endif /* CONFIG_PM */
+
+struct snd_soc_codec_device soc_codec_dev_uda134x = {
+ .probe = uda134x_soc_probe,
+ .remove = uda134x_soc_remove,
+ .suspend = uda134x_soc_suspend,
+ .resume = uda134x_soc_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_uda134x);
+
+MODULE_DESCRIPTION("UDA134X ALSA soc codec driver");
+MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell@evolware.org>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/uda134x_codec.h b/sound/soc/codecs/uda134x_codec.h
new file mode 100644
index 0000000..94f4404
--- /dev/null
+++ b/sound/soc/codecs/uda134x_codec.h
@@ -0,0 +1,36 @@
+#ifndef _UDA134X_CODEC_H
+#define _UDA134X_CODEC_H
+
+#define UDA134X_L3ADDR 5
+#define UDA134X_DATA0_ADDR ((UDA134X_L3ADDR << 2) | 0)
+#define UDA134X_DATA1_ADDR ((UDA134X_L3ADDR << 2) | 1)
+#define UDA134X_STATUS_ADDR ((UDA134X_L3ADDR << 2) | 2)
+
+#define UDA134X_EXTADDR_PREFIX 0xC0
+#define UDA134X_EXTDATA_PREFIX 0xE0
+
+/* UDA134X registers */
+#define UDA134X_EA000 0
+#define UDA134X_EA001 1
+#define UDA134X_EA010 2
+#define UDA134X_EA011 3
+#define UDA134X_EA100 4
+#define UDA134X_EA101 5
+#define UDA134X_EA110 6
+#define UDA134X_EA111 7
+#define UDA134X_STATUS0 8
+#define UDA134X_STATUS1 9
+#define UDA134X_DATA000 10
+#define UDA134X_DATA001 11
+#define UDA134X_DATA010 12
+#define UDA134X_DATA1 13
+
+#define UDA134X_REGS_NUM 14
+
+#define STATUS0_DAIFMT_MASK (~(7<<1))
+#define STATUS0_SYSCLK_MASK (~(3<<4))
+
+extern struct snd_soc_dai uda134x_dai;
+extern struct snd_soc_codec_device soc_codec_dev_uda134x;
+
+#endif
--
1.5.6.5
^ permalink raw reply related [flat|nested] 17+ messages in thread
* [PATCH 03/17] ASoC: Machine driver for for s3c24xx with uda134x
2008-11-18 12:38 ` [PATCH 02/17] ASoC: UDA134x codec driver Mark Brown
@ 2008-11-18 12:38 ` Mark Brown
2008-11-18 12:38 ` [PATCH 04/17] ASoC: Allow writes to uncached registers in WM8990 Mark Brown
0 siblings, 1 reply; 17+ messages in thread
From: Mark Brown @ 2008-11-18 12:38 UTC (permalink / raw)
To: Takashi Iwai
Cc: alsa-devel, Mark Brown, Christian Pellegrin, Christian Pellegrin
From: Christian Pellegrin <chripell@gmail.com>
Signed-off-by: Christian Pellegrin <chripell@fsfe.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
include/sound/s3c24xx_uda134x.h | 14 ++
sound/soc/s3c24xx/Kconfig | 5 +
sound/soc/s3c24xx/Makefile | 2 +
sound/soc/s3c24xx/s3c24xx_uda134x.c | 374 +++++++++++++++++++++++++++++++++++
4 files changed, 395 insertions(+), 0 deletions(-)
create mode 100644 include/sound/s3c24xx_uda134x.h
create mode 100644 sound/soc/s3c24xx/s3c24xx_uda134x.c
diff --git a/include/sound/s3c24xx_uda134x.h b/include/sound/s3c24xx_uda134x.h
new file mode 100644
index 0000000..33df4cb
--- /dev/null
+++ b/include/sound/s3c24xx_uda134x.h
@@ -0,0 +1,14 @@
+#ifndef _S3C24XX_UDA134X_H_
+#define _S3C24XX_UDA134X_H_ 1
+
+#include <sound/uda134x.h>
+
+struct s3c24xx_uda134x_platform_data {
+ int l3_clk;
+ int l3_mode;
+ int l3_data;
+ void (*power) (int);
+ int model;
+};
+
+#endif
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index b9f2353..fcd03ac 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -44,3 +44,8 @@ config SND_S3C24XX_SOC_LN2440SBC_ALC650
Say Y if you want to add support for SoC audio on ln2440sbc
with the ALC650.
+config SND_S3C24XX_SOC_S3C24XX_UDA134X
+ tristate "SoC I2S Audio support UDA134X wired to a S3C24XX"
+ depends on SND_S3C24XX_SOC
+ select SND_S3C24XX_SOC_I2S
+ select SND_SOC_UDA134X
diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile
index 0aa5fb0..96b3f3f 100644
--- a/sound/soc/s3c24xx/Makefile
+++ b/sound/soc/s3c24xx/Makefile
@@ -13,7 +13,9 @@ obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o
snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o
snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o
snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o
+snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o
obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o
obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o
+obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o
diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c
new file mode 100644
index 0000000..29a6813
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c
@@ -0,0 +1,374 @@
+/*
+ * Modifications by Christian Pellegrin <chripell@evolware.org>
+ *
+ * s3c24xx_uda134x.c -- S3C24XX_UDA134X ALSA SoC Audio board driver
+ *
+ * Copyright 2007 Dension Audio Systems Ltd.
+ * Author: Zoltan Devai
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/clk.h>
+#include <linux/mutex.h>
+#include <linux/gpio.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/s3c24xx_uda134x.h>
+#include <sound/uda134x.h>
+
+#include <asm/plat-s3c24xx/regs-iis.h>
+
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-i2s.h"
+#include "../codecs/uda134x_codec.h"
+
+
+/* #define ENFORCE_RATES 1 */
+/*
+ Unfortunately the S3C24XX in master mode has a limited capacity of
+ generating the clock for the codec. If you define this only rates
+ that are really available will be enforced. But be careful, most
+ user level application just want the usual sampling frequencies (8,
+ 11.025, 22.050, 44.1 kHz) and anyway resampling is a costly
+ operation for embedded systems. So if you aren't very lucky or your
+ hardware engineer wasn't very forward-looking it's better to leave
+ this undefined. If you do so an approximate value for the requested
+ sampling rate in the range -/+ 5% will be chosen. If this in not
+ possible an error will be returned.
+*/
+
+static struct clk *xtal;
+static struct clk *pclk;
+/* this is need because we don't have a place where to keep the
+ * pointers to the clocks in each substream. We get the clocks only
+ * when we are actually using them so we don't block stuff like
+ * frequency change or oscillator power-off */
+static int clk_users;
+static DEFINE_MUTEX(clk_lock);
+
+static unsigned int rates[33 * 2];
+#ifdef ENFORCE_RATES
+static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0,
+};
+#endif
+
+static struct platform_device *s3c24xx_uda134x_snd_device;
+
+int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+#ifdef ENFORCE_RATES
+ struct snd_pcm_runtime *runtime = substream->runtime;;
+#endif
+
+ mutex_lock(&clk_lock);
+ pr_debug("%s %d\n", __func__, clk_users);
+ if (clk_users == 0) {
+ xtal = clk_get(&s3c24xx_uda134x_snd_device->dev, "xtal");
+ if (!xtal) {
+ printk(KERN_ERR "%s cannot get xtal\n", __func__);
+ ret = -EBUSY;
+ } else {
+ pclk = clk_get(&s3c24xx_uda134x_snd_device->dev,
+ "pclk");
+ if (!pclk) {
+ printk(KERN_ERR "%s cannot get pclk\n",
+ __func__);
+ clk_put(xtal);
+ ret = -EBUSY;
+ }
+ }
+ if (!ret) {
+ int i, j;
+
+ for (i = 0; i < 2; i++) {
+ int fs = i ? 256 : 384;
+
+ rates[i*33] = clk_get_rate(xtal) / fs;
+ for (j = 1; j < 33; j++)
+ rates[i*33 + j] = clk_get_rate(pclk) /
+ (j * fs);
+ }
+ }
+ }
+ clk_users += 1;
+ mutex_unlock(&clk_lock);
+ if (!ret) {
+#ifdef ENFORCE_RATES
+ ret = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &hw_constraints_rates);
+ if (ret < 0)
+ printk(KERN_ERR "%s cannot set constraints\n",
+ __func__);
+#endif
+ }
+ return ret;
+}
+
+void s3c24xx_uda134x_shutdown(struct snd_pcm_substream *substream)
+{
+ mutex_lock(&clk_lock);
+ pr_debug("%s %d\n", __func__, clk_users);
+ clk_users -= 1;
+ if (clk_users == 0) {
+ clk_put(xtal);
+ xtal = NULL;
+ clk_put(pclk);
+ pclk = NULL;
+ }
+ mutex_unlock(&clk_lock);
+}
+
+static int s3c24xx_uda134x_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ unsigned int clk = 0;
+ int ret = 0;
+ int clk_source, fs_mode;
+ unsigned long rate = params_rate(params);
+ long err, cerr;
+ unsigned int div;
+ int i, bi;
+
+ err = 999999;
+ bi = 0;
+ for (i = 0; i < 2*33; i++) {
+ cerr = rates[i] - rate;
+ if (cerr < 0)
+ cerr = -cerr;
+ if (cerr < err) {
+ err = cerr;
+ bi = i;
+ }
+ }
+ if (bi / 33 == 1)
+ fs_mode = S3C2410_IISMOD_256FS;
+ else
+ fs_mode = S3C2410_IISMOD_384FS;
+ if (bi % 33 == 0) {
+ clk_source = S3C24XX_CLKSRC_MPLL;
+ div = 1;
+ } else {
+ clk_source = S3C24XX_CLKSRC_PCLK;
+ div = bi % 33;
+ }
+ pr_debug("%s desired rate %lu, %d\n", __func__, rate, bi);
+
+ clk = (fs_mode == S3C2410_IISMOD_384FS ? 384 : 256) * rate;
+ pr_debug("%s will use: %s %s %d sysclk %d err %ld\n", __func__,
+ fs_mode == S3C2410_IISMOD_384FS ? "384FS" : "256FS",
+ clk_source == S3C24XX_CLKSRC_MPLL ? "MPLLin" : "PCLK",
+ div, clk, err);
+
+ if ((err * 100 / rate) > 5) {
+ printk(KERN_ERR "S3C24XX_UDA134X: effective frequency "
+ "too different from desired (%ld%%)\n",
+ err * 100 / rate);
+ return -EINVAL;
+ }
+
+ ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, clk_source , clk,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
+ fs_mode);
+ if (ret < 0)
+ return ret;
+
+ ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK,
+ S3C2410_IISMOD_32FS);
+ if (ret < 0)
+ return ret;
+
+ ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
+ S3C24XX_PRESCALE(div, div));
+ if (ret < 0)
+ return ret;
+
+ /* set the codec system clock for DAC and ADC */
+ ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, clk,
+ SND_SOC_CLOCK_OUT);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops s3c24xx_uda134x_ops = {
+ .startup = s3c24xx_uda134x_startup,
+ .shutdown = s3c24xx_uda134x_shutdown,
+ .hw_params = s3c24xx_uda134x_hw_params,
+};
+
+static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = {
+ .name = "UDA134X",
+ .stream_name = "UDA134X",
+ .codec_dai = &uda134x_dai,
+ .cpu_dai = &s3c24xx_i2s_dai,
+ .ops = &s3c24xx_uda134x_ops,
+};
+
+static struct snd_soc_machine snd_soc_machine_s3c24xx_uda134x = {
+ .name = "S3C24XX_UDA134X",
+ .dai_link = &s3c24xx_uda134x_dai_link,
+ .num_links = 1,
+};
+
+static struct s3c24xx_uda134x_platform_data *s3c24xx_uda134x_l3_pins;
+
+static void setdat(int v)
+{
+ gpio_set_value(s3c24xx_uda134x_l3_pins->l3_data, v > 0);
+}
+
+static void setclk(int v)
+{
+ gpio_set_value(s3c24xx_uda134x_l3_pins->l3_clk, v > 0);
+}
+
+static void setmode(int v)
+{
+ gpio_set_value(s3c24xx_uda134x_l3_pins->l3_mode, v > 0);
+}
+
+static struct uda134x_platform_data s3c24xx_uda134x = {
+ .l3 = {
+ .setdat = setdat,
+ .setclk = setclk,
+ .setmode = setmode,
+ .data_hold = 1,
+ .data_setup = 1,
+ .clock_high = 1,
+ .mode_hold = 1,
+ .mode = 1,
+ .mode_setup = 1,
+ },
+};
+
+static struct snd_soc_device s3c24xx_uda134x_snd_devdata = {
+ .machine = &snd_soc_machine_s3c24xx_uda134x,
+ .platform = &s3c24xx_soc_platform,
+ .codec_dev = &soc_codec_dev_uda134x,
+ .codec_data = &s3c24xx_uda134x,
+};
+
+static int s3c24xx_uda134x_setup_pin(int pin, char *fun)
+{
+ if (gpio_request(pin, "s3c24xx_uda134x") < 0) {
+ printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: "
+ "l3 %s pin already in use", fun);
+ return -EBUSY;
+ }
+ gpio_direction_output(pin, 0);
+ return 0;
+}
+
+static int s3c24xx_uda134x_probe(struct platform_device *pdev)
+{
+ int ret;
+
+ printk(KERN_INFO "S3C24XX_UDA134X SoC Audio driver\n");
+
+ s3c24xx_uda134x_l3_pins = pdev->dev.platform_data;
+ if (s3c24xx_uda134x_l3_pins == NULL) {
+ printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: "
+ "unable to find platform data\n");
+ return -ENODEV;
+ }
+ s3c24xx_uda134x.power = s3c24xx_uda134x_l3_pins->power;
+ s3c24xx_uda134x.model = s3c24xx_uda134x_l3_pins->model;
+
+ if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_data,
+ "data") < 0)
+ return -EBUSY;
+ if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_clk,
+ "clk") < 0) {
+ gpio_free(s3c24xx_uda134x_l3_pins->l3_data);
+ return -EBUSY;
+ }
+ if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_mode,
+ "mode") < 0) {
+ gpio_free(s3c24xx_uda134x_l3_pins->l3_data);
+ gpio_free(s3c24xx_uda134x_l3_pins->l3_clk);
+ return -EBUSY;
+ }
+
+ s3c24xx_uda134x_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!s3c24xx_uda134x_snd_device) {
+ printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: "
+ "Unable to register\n");
+ return -ENOMEM;
+ }
+
+ platform_set_drvdata(s3c24xx_uda134x_snd_device,
+ &s3c24xx_uda134x_snd_devdata);
+ s3c24xx_uda134x_snd_devdata.dev = &s3c24xx_uda134x_snd_device->dev;
+ ret = platform_device_add(s3c24xx_uda134x_snd_device);
+ if (ret) {
+ printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: Unable to add\n");
+ platform_device_put(s3c24xx_uda134x_snd_device);
+ }
+
+ return ret;
+}
+
+static int s3c24xx_uda134x_remove(struct platform_device *pdev)
+{
+ platform_device_unregister(s3c24xx_uda134x_snd_device);
+ gpio_free(s3c24xx_uda134x_l3_pins->l3_data);
+ gpio_free(s3c24xx_uda134x_l3_pins->l3_clk);
+ gpio_free(s3c24xx_uda134x_l3_pins->l3_mode);
+ return 0;
+}
+
+static struct platform_driver s3c24xx_uda134x_driver = {
+ .probe = s3c24xx_uda134x_probe,
+ .remove = s3c24xx_uda134x_remove,
+ .driver = {
+ .name = "s3c24xx_uda134x",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init s3c24xx_uda134x_init(void)
+{
+ return platform_driver_register(&s3c24xx_uda134x_driver);
+}
+
+static void __exit s3c24xx_uda134x_exit(void)
+{
+ platform_driver_unregister(&s3c24xx_uda134x_driver);
+}
+
+
+module_init(s3c24xx_uda134x_init);
+module_exit(s3c24xx_uda134x_exit);
+
+MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell@evolware.org>");
+MODULE_DESCRIPTION("S3C24XX_UDA134X ALSA SoC audio driver");
+MODULE_LICENSE("GPL");
--
1.5.6.5
^ permalink raw reply related [flat|nested] 17+ messages in thread
* [PATCH 04/17] ASoC: Allow writes to uncached registers in WM8990
2008-11-18 12:38 ` [PATCH 03/17] ASoC: Machine driver for for s3c24xx with uda134x Mark Brown
@ 2008-11-18 12:38 ` Mark Brown
2008-11-18 12:38 ` [PATCH 05/17] ASoC: Enable WM8990 ADC clocking workaround Mark Brown
0 siblings, 1 reply; 17+ messages in thread
From: Mark Brown @ 2008-11-18 12:38 UTC (permalink / raw)
To: Takashi Iwai; +Cc: alsa-devel, Mark Brown
Only fully documented registers are cached in the WM8990 but additional
registers exist.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/codecs/wm8990.c | 6 +++---
1 files changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 572d22b..5c84f02 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -106,6 +106,7 @@ static const u16 wm8990_reg[] = {
0x0008, /* R60 - PLL1 */
0x0031, /* R61 - PLL2 */
0x0026, /* R62 - PLL3 */
+ 0x0000, /* R63 - Driver internal */
};
/*
@@ -126,10 +127,9 @@ static inline void wm8990_write_reg_cache(struct snd_soc_codec *codec,
unsigned int reg, unsigned int value)
{
u16 *cache = codec->reg_cache;
- BUG_ON(reg > (ARRAY_SIZE(wm8990_reg)) - 1);
- /* Reset register is uncached */
- if (reg == 0)
+ /* Reset register and reserved registers are uncached */
+ if (reg == 0 || reg > ARRAY_SIZE(wm8990_reg) - 1)
return;
cache[reg] = value;
--
1.5.6.5
^ permalink raw reply related [flat|nested] 17+ messages in thread
* [PATCH 05/17] ASoC: Enable WM8990 ADC clocking workaround
2008-11-18 12:38 ` [PATCH 04/17] ASoC: Allow writes to uncached registers in WM8990 Mark Brown
@ 2008-11-18 12:38 ` Mark Brown
2008-11-18 12:38 ` [PATCH 06/17] ASoC: Manage VMID mode for WM8990 Mark Brown
0 siblings, 1 reply; 17+ messages in thread
From: Mark Brown @ 2008-11-18 12:38 UTC (permalink / raw)
To: Takashi Iwai; +Cc: alsa-devel, Mark Brown
Enable a hardware workaround which avoids problems with the clocking of
the ADCs in certain configurations.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/codecs/wm8990.c | 6 ++++--
sound/soc/codecs/wm8990.h | 4 ++--
2 files changed, 6 insertions(+), 4 deletions(-)
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 5c84f02..938e154 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1272,9 +1272,11 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
wm8990_write(codec, WM8990_ANTIPOP2, WM8990_BUFIOEN);
- } else {
- /* ON -> standby */
+ /* Enable workaround for ADC clocking issue. */
+ wm8990_write(codec, WM8990_EXT_ACCESS_ENA, 0x2);
+ wm8990_write(codec, WM8990_EXT_CTL1, 0xa003);
+ wm8990_write(codec, WM8990_EXT_ACCESS_ENA, 0);
}
break;
diff --git a/sound/soc/codecs/wm8990.h b/sound/soc/codecs/wm8990.h
index 0e192f3..7114ddc 100644
--- a/sound/soc/codecs/wm8990.h
+++ b/sound/soc/codecs/wm8990.h
@@ -80,8 +80,8 @@
#define WM8990_PLL3 0x3E
#define WM8990_INTDRIVBITS 0x3F
-#define WM8990_REGISTER_COUNT 60
-#define WM8990_MAX_REGISTER 0x3F
+#define WM8990_EXT_ACCESS_ENA 0x75
+#define WM8990_EXT_CTL1 0x7a
/*
* Field Definitions.
--
1.5.6.5
^ permalink raw reply related [flat|nested] 17+ messages in thread
* [PATCH 06/17] ASoC: Manage VMID mode for WM8990
2008-11-18 12:38 ` [PATCH 05/17] ASoC: Enable WM8990 ADC clocking workaround Mark Brown
@ 2008-11-18 12:38 ` Mark Brown
2008-11-18 12:38 ` [PATCH 07/17] ASoC: Build tlv320aic23 cleanly Mark Brown
0 siblings, 1 reply; 17+ messages in thread
From: Mark Brown @ 2008-11-18 12:38 UTC (permalink / raw)
To: Takashi Iwai; +Cc: alsa-devel, Mark Brown
A small additional power saving can be achieved for the WM8990 by
maintaining VMID using a 2*250k divider when in standby mode.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/codecs/wm8990.c | 11 +++++++++++
1 files changed, 11 insertions(+), 0 deletions(-)
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 938e154..2d7b009 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1222,8 +1222,14 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
switch (level) {
case SND_SOC_BIAS_ON:
break;
+
case SND_SOC_BIAS_PREPARE:
+ /* VMID=2*50k */
+ val = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_1) &
+ ~WM8990_VMID_MODE_MASK;
+ wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x2);
break;
+
case SND_SOC_BIAS_STANDBY:
if (codec->bias_level == SND_SOC_BIAS_OFF) {
/* Enable all output discharge bits */
@@ -1278,6 +1284,11 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
wm8990_write(codec, WM8990_EXT_CTL1, 0xa003);
wm8990_write(codec, WM8990_EXT_ACCESS_ENA, 0);
}
+
+ /* VMID=2*250k */
+ val = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_1) &
+ ~WM8990_VMID_MODE_MASK;
+ wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x4);
break;
case SND_SOC_BIAS_OFF:
--
1.5.6.5
^ permalink raw reply related [flat|nested] 17+ messages in thread
* [PATCH 07/17] ASoC: Build tlv320aic23 cleanly
2008-11-18 12:38 ` [PATCH 06/17] ASoC: Manage VMID mode for WM8990 Mark Brown
@ 2008-11-18 12:38 ` Mark Brown
2008-11-18 12:38 ` [PATCH 08/17] ASoC: DaVinci: Fix audio stall when doing full duplex Mark Brown
0 siblings, 1 reply; 17+ messages in thread
From: Mark Brown @ 2008-11-18 12:38 UTC (permalink / raw)
To: Takashi Iwai; +Cc: alsa-devel, Mark Brown
Also merge down a couple of last minute style changes that got lost in the
shuffle.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/codecs/tlv320aic23.c | 13 +++++++++----
1 files changed, 9 insertions(+), 4 deletions(-)
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index a95b538..c903e4f 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -331,10 +331,11 @@ static int find_rate(int mclk, u32 need_adc, u32 need_dac)
adc_h = need_adc + (need_adc >> 5);
dac_l = need_dac - (need_dac >> 5);
dac_h = need_dac + (need_dac >> 5);
- for (i = 0; i < 4; i++) {
+ for (i = 0; i < ARRAY_SIZE(bosr_usb_divisor_table); i++) {
int base = mclk / bosr_usb_divisor_table[i];
int mask = sr_valid_mask[i];
- for (j = 0; j < 16; j++, mask >>= 1) {
+ for (j = 0; j < ARRAY_SIZE(sr_adc_mult_table);
+ j++, mask >>= 1) {
int adc;
int dac;
int score;
@@ -364,6 +365,7 @@ static int find_rate(int mclk, u32 need_adc, u32 need_dac)
return (best_j << 2) | best_i | (best_div << TLV320AIC23_CLKIN_SHIFT);
}
+#ifdef DEBUG
static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk,
u32 *sample_rate_adc, u32 *sample_rate_dac)
{
@@ -379,6 +381,7 @@ static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk,
*sample_rate_adc = adc;
*sample_rate_dac = dac;
}
+#endif
static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk,
u32 sample_rate_adc, u32 sample_rate_dac)
@@ -391,12 +394,14 @@ static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk,
return -EINVAL;
}
tlv320aic23_write(codec, TLV320AIC23_SRATE, data);
- if (1) {
- int adc, dac;
+#ifdef DEBUG
+ {
+ u32 adc, dac;
get_current_sample_rates(codec, mclk, &adc, &dac);
printk(KERN_DEBUG "actual samplerate = %u,%u reg=%x\n",
adc, dac, data);
}
+#endif
return 0;
}
--
1.5.6.5
^ permalink raw reply related [flat|nested] 17+ messages in thread
* [PATCH 08/17] ASoC: DaVinci: Fix audio stall when doing full duplex
2008-11-18 12:38 ` [PATCH 07/17] ASoC: Build tlv320aic23 cleanly Mark Brown
@ 2008-11-18 12:38 ` Mark Brown
2008-11-18 12:38 ` [PATCH 09/17] ASoC: Blackfin: updates Kconfig for SPORT Mark Brown
0 siblings, 1 reply; 17+ messages in thread
From: Mark Brown @ 2008-11-18 12:38 UTC (permalink / raw)
To: Takashi Iwai; +Cc: alsa-devel, Mark Brown, Steve Chen, Naresh Medisetty
From: Naresh Medisetty <naresh@ti.com>
Fix concurrent capture/playback issue.
The issue is caused by re-initialization of control registers used specifically
for capture or playback in both capture and playback operations.
Signed-off-by: Steve Chen <schen@mvista.com>
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/davinci/davinci-i2s.c | 31 +++++++++++++++++++------------
1 files changed, 19 insertions(+), 12 deletions(-)
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 8c1bf87..11c20d0 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -295,10 +295,14 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
u32 w;
/* general line settings */
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG,
- DAVINCI_MCBSP_SPCR_RINTM(3) |
- DAVINCI_MCBSP_SPCR_XINTM(3) |
- DAVINCI_MCBSP_SPCR_FREE);
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ w |= DAVINCI_MCBSP_SPCR_RINTM(3) | DAVINCI_MCBSP_SPCR_FREE;
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+ } else {
+ w |= DAVINCI_MCBSP_SPCR_XINTM(3) | DAVINCI_MCBSP_SPCR_FREE;
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+ }
i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS);
w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SRGR_REG);
@@ -329,16 +333,19 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
- w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_RCR_REG);
- MOD_REG_BIT(w, DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) |
- DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length), 1);
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, w);
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_RCR_REG);
+ MOD_REG_BIT(w, DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) |
+ DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length), 1);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, w);
- w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_XCR_REG);
- MOD_REG_BIT(w, DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) |
- DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length), 1);
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, w);
+ } else {
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_XCR_REG);
+ MOD_REG_BIT(w, DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) |
+ DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length), 1);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, w);
+ }
return 0;
}
--
1.5.6.5
^ permalink raw reply related [flat|nested] 17+ messages in thread
* [PATCH 09/17] ASoC: Blackfin: updates Kconfig for SPORT
2008-11-18 12:38 ` [PATCH 08/17] ASoC: DaVinci: Fix audio stall when doing full duplex Mark Brown
@ 2008-11-18 12:38 ` Mark Brown
2008-11-18 12:38 ` [PATCH 10/17] ASoC: AD1980 codec: add multi-channel function support Mark Brown
0 siblings, 1 reply; 17+ messages in thread
From: Mark Brown @ 2008-11-18 12:38 UTC (permalink / raw)
To: Takashi Iwai; +Cc: alsa-devel, Mark Brown, Bryan Wu, Mike Frysinger
From: Mike Frysinger <vapier.adi@gmail.com>
tweak SPORT range for non-BF54x so we get proper behavior for BF52x parts
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/blackfin/Kconfig | 2 +-
1 files changed, 1 insertions(+), 1 deletions(-)
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index dc00620..3fce187 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -80,7 +80,7 @@ config SND_BF5XX_SPORT_NUM
int "Set a SPORT for Sound chip"
depends on (SND_BF5XX_I2S || SND_BF5XX_AC97)
range 0 3 if BF54x
- range 0 1 if (BF53x || BF561)
+ range 0 1 if !BF54x
default 0
help
Set the correct SPORT for sound chip.
--
1.5.6.5
^ permalink raw reply related [flat|nested] 17+ messages in thread
* [PATCH 10/17] ASoC: AD1980 codec: add multi-channel function support
2008-11-18 12:38 ` [PATCH 09/17] ASoC: Blackfin: updates Kconfig for SPORT Mark Brown
@ 2008-11-18 12:38 ` Mark Brown
2008-11-18 12:38 ` [PATCH 11/17] ASoC: Blackfin: " Mark Brown
0 siblings, 1 reply; 17+ messages in thread
From: Mark Brown @ 2008-11-18 12:38 UTC (permalink / raw)
To: Takashi Iwai; +Cc: alsa-devel, Mark Brown, Cliff Cai, Bryan Wu
From: Cliff Cai <cliff.cai@analog.com>
We added multi-channel function to this codec driver and Blackfin ASoC driver as well.
It was tested on Blackfin hardware.
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/codecs/ad1980.c | 19 +++++++++++++++----
1 files changed, 15 insertions(+), 4 deletions(-)
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index 1397b8e..410fed9 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -85,6 +85,9 @@ SOC_DOUBLE("Line HP Swap Switch", AC97_AD_MISC, 10, 5, 1, 0),
SOC_DOUBLE("Surround Playback Volume", AC97_SURROUND_MASTER, 8, 0, 31, 1),
SOC_DOUBLE("Surround Playback Switch", AC97_SURROUND_MASTER, 15, 7, 1, 1),
+SOC_DOUBLE("Center/LFE Playback Volume", AC97_CENTER_LFE_MASTER, 8, 0, 31, 1),
+SOC_DOUBLE("Center/LFE Playback Switch", AC97_CENTER_LFE_MASTER, 15, 7, 1, 1),
+
SOC_ENUM("Capture Source", ad1980_cap_src),
SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0),
@@ -145,7 +148,7 @@ struct snd_soc_dai ad1980_dai = {
.playback = {
.stream_name = "Playback",
.channels_min = 2,
- .channels_max = 2,
+ .channels_max = 6,
.rates = SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE, },
.capture = {
@@ -192,6 +195,7 @@ static int ad1980_soc_probe(struct platform_device *pdev)
struct snd_soc_codec *codec;
int ret = 0;
u16 vendor_id2;
+ u16 ext_status;
printk(KERN_INFO "AD1980 SoC Audio Codec\n");
@@ -253,9 +257,16 @@ static int ad1980_soc_probe(struct platform_device *pdev)
"supported\n");
}
- ac97_write(codec, AC97_MASTER, 0x0000); /* unmute line out volume */
- ac97_write(codec, AC97_PCM, 0x0000); /* unmute PCM out volume */
- ac97_write(codec, AC97_REC_GAIN, 0x0000);/* unmute record volume */
+ /* unmute captures and playbacks volume */
+ ac97_write(codec, AC97_MASTER, 0x0000);
+ ac97_write(codec, AC97_PCM, 0x0000);
+ ac97_write(codec, AC97_REC_GAIN, 0x0000);
+ ac97_write(codec, AC97_CENTER_LFE_MASTER, 0x0000);
+ ac97_write(codec, AC97_SURROUND_MASTER, 0x0000);
+
+ /*power on LFE/CENTER/Surround DACs*/
+ ext_status = ac97_read(codec, AC97_EXTENDED_STATUS);
+ ac97_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800);
ad1980_add_controls(codec);
ret = snd_soc_register_card(socdev);
--
1.5.6.5
^ permalink raw reply related [flat|nested] 17+ messages in thread
* [PATCH 11/17] ASoC: Blackfin: add multi-channel function support
2008-11-18 12:38 ` [PATCH 10/17] ASoC: AD1980 codec: add multi-channel function support Mark Brown
@ 2008-11-18 12:38 ` Mark Brown
2008-11-18 12:38 ` [PATCH 12/17] ASoC: Blackfin: Fix AD1980/1 build with MMAP support disabled Mark Brown
0 siblings, 1 reply; 17+ messages in thread
From: Mark Brown @ 2008-11-18 12:38 UTC (permalink / raw)
To: Takashi Iwai; +Cc: alsa-devel, Mark Brown, Cliff Cai, Bryan Wu
From: Cliff Cai <cliff.cai@analog.com>
This patch provides a option for users to enable multi-channel function support
in Blackfin ASoC driver. Because Blackfin is without MMU, it is easy for us and
the user to enable this function at compiling stage not dynamically on the fly.
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/blackfin/Kconfig | 14 +++-
sound/soc/blackfin/bf5xx-ac97-pcm.c | 80 +++++++++++-------
sound/soc/blackfin/bf5xx-ac97.c | 156 +++++++++++++++++++++--------------
sound/soc/blackfin/bf5xx-ac97.h | 35 +++++++-
sound/soc/blackfin/bf5xx-sport.h | 2 +-
5 files changed, 184 insertions(+), 103 deletions(-)
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index 3fce187..43d89fc 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -47,7 +47,7 @@ config SND_BF5XX_AC97
properly with this driver. This driver is known to work with the
Analog Devices line of AC97 codecs.
-config SND_MMAP_SUPPORT
+config SND_BF5XX_MMAP_SUPPORT
bool "Enable MMAP Support"
depends on SND_BF5XX_AC97
default y
@@ -55,9 +55,17 @@ config SND_MMAP_SUPPORT
Say y if you want AC97 driver to support mmap mode.
We introduce an intermediate buffer to simulate mmap.
+config SND_BF5XX_MULTICHAN_SUPPORT
+ bool "Enable Multichannel Support"
+ depends on SND_BF5XX_AC97
+ default n
+ help
+ Say y if you want AC97 driver to support up to 5.1 channel audio.
+ this mode will consume much more memory for DMA.
+
config SND_BF5XX_SOC_SPORT
tristate
-
+
config SND_BF5XX_SOC_I2S
tristate
select SND_BF5XX_SOC_SPORT
@@ -90,7 +98,7 @@ config SND_BF5XX_HAVE_COLD_RESET
depends on SND_BF5XX_AC97
default y if BFIN548_EZKIT
default n if !BFIN548_EZKIT
-
+
config SND_BF5XX_RESET_GPIO_NUM
int "Set a GPIO for cold reset"
depends on SND_BF5XX_HAVE_COLD_RESET
diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c
index 25e50d2..4be1a49 100644
--- a/sound/soc/blackfin/bf5xx-ac97-pcm.c
+++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c
@@ -43,24 +43,34 @@
#include "bf5xx-ac97.h"
#include "bf5xx-sport.h"
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+static unsigned int ac97_chan_mask[] = {
+ SP_FL, /* Mono */
+ SP_STEREO, /* Stereo */
+ SP_2DOT1, /* 2.1*/
+ SP_QUAD,/*Quadraquic*/
+ SP_FL | SP_FR | SP_FC | SP_SL | SP_SR,/*5 channels */
+ SP_5DOT1, /* 5.1 */
+};
+
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
static void bf5xx_mmap_copy(struct snd_pcm_substream *substream,
snd_pcm_uframes_t count)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct sport_device *sport = runtime->private_data;
+ unsigned int chan_mask = ac97_chan_mask[runtime->channels - 1];
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- bf5xx_pcm_to_ac97(
- (struct ac97_frame *)sport->tx_dma_buf + sport->tx_pos,
- (__u32 *)runtime->dma_area + sport->tx_pos, count);
+ bf5xx_pcm_to_ac97((struct ac97_frame *)sport->tx_dma_buf +
+ sport->tx_pos, (__u16 *)runtime->dma_area + sport->tx_pos *
+ runtime->channels, count, chan_mask);
sport->tx_pos += runtime->period_size;
if (sport->tx_pos >= runtime->buffer_size)
sport->tx_pos %= runtime->buffer_size;
sport->tx_delay_pos = sport->tx_pos;
} else {
- bf5xx_ac97_to_pcm(
- (struct ac97_frame *)sport->rx_dma_buf + sport->rx_pos,
- (__u32 *)runtime->dma_area + sport->rx_pos, count);
+ bf5xx_ac97_to_pcm((struct ac97_frame *)sport->rx_dma_buf +
+ sport->rx_pos, (__u16 *)runtime->dma_area + sport->rx_pos *
+ runtime->channels, count);
sport->rx_pos += runtime->period_size;
if (sport->rx_pos >= runtime->buffer_size)
sport->rx_pos %= runtime->buffer_size;
@@ -71,7 +81,7 @@ static void bf5xx_mmap_copy(struct snd_pcm_substream *substream,
static void bf5xx_dma_irq(void *data)
{
struct snd_pcm_substream *pcm = data;
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
struct snd_pcm_runtime *runtime = pcm->runtime;
struct sport_device *sport = runtime->private_data;
bf5xx_mmap_copy(pcm, runtime->period_size);
@@ -90,7 +100,7 @@ static void bf5xx_dma_irq(void *data)
* The total rx/tx buffer is for ac97 frame to hold all pcm data
* is 0x20000 * sizeof(struct ac97_frame) / 4.
*/
-#ifdef CONFIG_SND_MMAP_SUPPORT
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
static const struct snd_pcm_hardware bf5xx_pcm_hardware = {
.info = SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_MMAP |
@@ -123,10 +133,20 @@ static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream,
static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream)
{
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
struct snd_pcm_runtime *runtime = substream->runtime;
+ struct sport_device *sport = runtime->private_data;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- memset(runtime->dma_area, 0, runtime->buffer_size);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ sport->once = 0;
+ if (runtime->dma_area)
+ memset(runtime->dma_area, 0, runtime->buffer_size);
+ memset(sport->tx_dma_buf, 0, runtime->buffer_size *
+ sizeof(struct ac97_frame));
+ } else
+ memset(sport->rx_dma_buf, 0, runtime->buffer_size *
+ sizeof(struct ac97_frame));
+#endif
snd_pcm_lib_free_pages(substream);
return 0;
}
@@ -139,7 +159,7 @@ static int bf5xx_pcm_prepare(struct snd_pcm_substream *substream)
/* An intermediate buffer is introduced for implementing mmap for
* SPORT working in TMD mode(include AC97).
*/
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
sport_set_tx_callback(sport, bf5xx_dma_irq, substream);
sport_config_tx_dma(sport, sport->tx_dma_buf, runtime->periods,
@@ -174,23 +194,21 @@ static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
case SNDRV_PCM_TRIGGER_START:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
bf5xx_mmap_copy(substream, runtime->period_size);
- snd_pcm_period_elapsed(substream);
sport->tx_delay_pos = 0;
sport_tx_start(sport);
- }
- else
+ } else
sport_rx_start(sport);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
sport->tx_pos = 0;
#endif
sport_tx_stop(sport);
} else {
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
sport->rx_pos = 0;
#endif
sport_rx_stop(sport);
@@ -208,7 +226,7 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream)
struct sport_device *sport = runtime->private_data;
unsigned int curr;
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
curr = sport->tx_delay_pos;
else
@@ -257,14 +275,16 @@ static int bf5xx_pcm_close(struct snd_pcm_substream *substream)
pr_debug("%s enter\n", __func__);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
sport->once = 0;
- memset(sport->tx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame));
+ memset(sport->tx_dma_buf, 0, runtime->buffer_size *
+ sizeof(struct ac97_frame));
} else
- memset(sport->rx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame));
+ memset(sport->rx_dma_buf, 0, runtime->buffer_size *
+ sizeof(struct ac97_frame));
return 0;
}
-#ifdef CONFIG_SND_MMAP_SUPPORT
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream,
struct vm_area_struct *vma)
{
@@ -286,13 +306,11 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel,
substream->stream ? "Capture" : "Playback", pos, count);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- bf5xx_pcm_to_ac97(
- (struct ac97_frame *)runtime->dma_area + pos,
- buf, count);
+ bf5xx_pcm_to_ac97((struct ac97_frame *)runtime->dma_area + pos,
+ (__u16 *)buf, count, chan_mask);
else
- bf5xx_ac97_to_pcm(
- (struct ac97_frame *)runtime->dma_area + pos,
- buf, count);
+ bf5xx_ac97_to_pcm((struct ac97_frame *)runtime->dma_area + pos,
+ (__u16 *)buf, count);
return 0;
}
#endif
@@ -306,7 +324,7 @@ struct snd_pcm_ops bf5xx_pcm_ac97_ops = {
.prepare = bf5xx_pcm_prepare,
.trigger = bf5xx_pcm_trigger,
.pointer = bf5xx_pcm_pointer,
-#ifdef CONFIG_SND_MMAP_SUPPORT
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
.mmap = bf5xx_pcm_mmap,
#else
.copy = bf5xx_pcm_copy,
@@ -344,7 +362,7 @@ static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
* Need to allocate local buffer when enable
* MMAP for SPORT working in TMD mode (include AC97).
*/
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
if (!sport_handle->tx_dma_buf) {
sport_handle->tx_dma_buf = dma_alloc_coherent(NULL, \
@@ -381,7 +399,7 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
struct snd_pcm_substream *substream;
struct snd_dma_buffer *buf;
int stream;
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
size_t size = bf5xx_pcm_hardware.buffer_bytes_max *
sizeof(struct ac97_frame) / 4;
#endif
@@ -395,7 +413,7 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
continue;
dma_free_coherent(NULL, buf->bytes, buf->area, 0);
buf->area = NULL;
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
if (sport_handle->tx_dma_buf)
dma_free_coherent(NULL, size, \
diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index 5e5aafb..65c162c 100644
--- a/sound/soc/blackfin/bf5xx-ac97.c
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -54,71 +54,103 @@
static int *cmd_count;
static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM;
-#if defined(CONFIG_BF54x)
+static u16 sport_req[][7] = {
+ PIN_REQ_SPORT_0,
+#ifdef PIN_REQ_SPORT_1
+ PIN_REQ_SPORT_1,
+#endif
+#ifdef PIN_REQ_SPORT_2
+ PIN_REQ_SPORT_2,
+#endif
+#ifdef PIN_REQ_SPORT_3
+ PIN_REQ_SPORT_3,
+#endif
+ };
+
static struct sport_param sport_params[4] = {
{
.dma_rx_chan = CH_SPORT0_RX,
.dma_tx_chan = CH_SPORT0_TX,
- .err_irq = IRQ_SPORT0_ERR,
+ .err_irq = IRQ_SPORT0_ERROR,
.regs = (struct sport_register *)SPORT0_TCR1,
},
+#ifdef PIN_REQ_SPORT_1
{
.dma_rx_chan = CH_SPORT1_RX,
.dma_tx_chan = CH_SPORT1_TX,
- .err_irq = IRQ_SPORT1_ERR,
+ .err_irq = IRQ_SPORT1_ERROR,
.regs = (struct sport_register *)SPORT1_TCR1,
},
+#endif
+#ifdef PIN_REQ_SPORT_2
{
.dma_rx_chan = CH_SPORT2_RX,
.dma_tx_chan = CH_SPORT2_TX,
- .err_irq = IRQ_SPORT2_ERR,
+ .err_irq = IRQ_SPORT2_ERROR,
.regs = (struct sport_register *)SPORT2_TCR1,
},
+#endif
+#ifdef PIN_REQ_SPORT_3
{
.dma_rx_chan = CH_SPORT3_RX,
.dma_tx_chan = CH_SPORT3_TX,
- .err_irq = IRQ_SPORT3_ERR,
+ .err_irq = IRQ_SPORT3_ERROR,
.regs = (struct sport_register *)SPORT3_TCR1,
}
-};
-#else
-static struct sport_param sport_params[2] = {
- {
- .dma_rx_chan = CH_SPORT0_RX,
- .dma_tx_chan = CH_SPORT0_TX,
- .err_irq = IRQ_SPORT0_ERROR,
- .regs = (struct sport_register *)SPORT0_TCR1,
- },
- {
- .dma_rx_chan = CH_SPORT1_RX,
- .dma_tx_chan = CH_SPORT1_TX,
- .err_irq = IRQ_SPORT1_ERROR,
- .regs = (struct sport_register *)SPORT1_TCR1,
- }
-};
#endif
+};
-void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u32 *src, \
- size_t count)
+void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u16 *src,
+ size_t count, unsigned int chan_mask)
{
while (count--) {
- dst->ac97_tag = TAG_VALID | TAG_PCM;
- (dst++)->ac97_pcm = *src++;
+ dst->ac97_tag = TAG_VALID;
+ if (chan_mask & SP_FL) {
+ dst->ac97_pcm_r = *src++;
+ dst->ac97_tag |= TAG_PCM_RIGHT;
+ }
+ if (chan_mask & SP_FR) {
+ dst->ac97_pcm_l = *src++;
+ dst->ac97_tag |= TAG_PCM_LEFT;
+
+ }
+#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT)
+ if (chan_mask & SP_SR) {
+ dst->ac97_sl = *src++;
+ dst->ac97_tag |= TAG_PCM_SL;
+ }
+ if (chan_mask & SP_SL) {
+ dst->ac97_sr = *src++;
+ dst->ac97_tag |= TAG_PCM_SR;
+ }
+ if (chan_mask & SP_LFE) {
+ dst->ac97_lfe = *src++;
+ dst->ac97_tag |= TAG_PCM_LFE;
+ }
+ if (chan_mask & SP_FC) {
+ dst->ac97_center = *src++;
+ dst->ac97_tag |= TAG_PCM_CENTER;
+ }
+#endif
+ dst++;
}
}
EXPORT_SYMBOL(bf5xx_pcm_to_ac97);
-void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u32 *dst, \
+void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u16 *dst,
size_t count)
{
- while (count--)
- *(dst++) = (src++)->ac97_pcm;
+ while (count--) {
+ *(dst++) = src->ac97_pcm_l;
+ *(dst++) = src->ac97_pcm_r;
+ src++;
+ }
}
EXPORT_SYMBOL(bf5xx_ac97_to_pcm);
static unsigned int sport_tx_curr_frag(struct sport_device *sport)
{
- return sport->tx_curr_frag = sport_curr_offset_tx(sport) / \
+ return sport->tx_curr_frag = sport_curr_offset_tx(sport) /
sport->tx_fragsize;
}
@@ -130,7 +162,7 @@ static void enqueue_cmd(struct snd_ac97 *ac97, __u16 addr, __u16 data)
sport_incfrag(sport, &nextfrag, 1);
- nextwrite = (struct ac97_frame *)(sport->tx_buf + \
+ nextwrite = (struct ac97_frame *)(sport->tx_buf +
nextfrag * sport->tx_fragsize);
pr_debug("sport->tx_buf:%p, nextfrag:0x%x nextwrite:%p, cmd_count:%d\n",
sport->tx_buf, nextfrag, nextwrite, cmd_count[nextfrag]);
@@ -297,20 +329,15 @@ static int bf5xx_ac97_resume(struct platform_device *pdev,
static int bf5xx_ac97_probe(struct platform_device *pdev,
struct snd_soc_dai *dai)
{
- int ret;
-#if defined(CONFIG_BF54x)
- u16 sport_req[][7] = {PIN_REQ_SPORT_0, PIN_REQ_SPORT_1,
- PIN_REQ_SPORT_2, PIN_REQ_SPORT_3};
-#else
- u16 sport_req[][7] = {PIN_REQ_SPORT_0, PIN_REQ_SPORT_1};
-#endif
+ int ret = 0;
cmd_count = (int *)get_zeroed_page(GFP_KERNEL);
if (cmd_count == NULL)
return -ENOMEM;
if (peripheral_request_list(&sport_req[sport_num][0], "soc-audio")) {
pr_err("Requesting Peripherals failed\n");
- return -EFAULT;
+ ret = -EFAULT;
+ goto peripheral_err;
}
#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
@@ -318,54 +345,52 @@ static int bf5xx_ac97_probe(struct platform_device *pdev,
if (gpio_request(CONFIG_SND_BF5XX_RESET_GPIO_NUM, "SND_AD198x RESET")) {
pr_err("Failed to request GPIO_%d for reset\n",
CONFIG_SND_BF5XX_RESET_GPIO_NUM);
- peripheral_free_list(&sport_req[sport_num][0]);
- return -1;
+ ret = -1;
+ goto gpio_err;
}
gpio_direction_output(CONFIG_SND_BF5XX_RESET_GPIO_NUM, 1);
#endif
sport_handle = sport_init(&sport_params[sport_num], 2, \
sizeof(struct ac97_frame), NULL);
if (!sport_handle) {
- peripheral_free_list(&sport_req[sport_num][0]);
-#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
- gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
-#endif
- return -ENODEV;
+ ret = -ENODEV;
+ goto sport_err;
}
/*SPORT works in TDM mode to simulate AC97 transfers*/
ret = sport_set_multichannel(sport_handle, 16, 0x1F, 1);
if (ret) {
pr_err("SPORT is busy!\n");
- kfree(sport_handle);
- peripheral_free_list(&sport_req[sport_num][0]);
-#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
- gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
-#endif
- return -EBUSY;
+ ret = -EBUSY;
+ goto sport_config_err;
}
ret = sport_config_rx(sport_handle, IRFS, 0xF, 0, (16*16-1));
if (ret) {
pr_err("SPORT is busy!\n");
- kfree(sport_handle);
- peripheral_free_list(&sport_req[sport_num][0]);
-#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
- gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
-#endif
- return -EBUSY;
+ ret = -EBUSY;
+ goto sport_config_err;
}
ret = sport_config_tx(sport_handle, ITFS, 0xF, 0, (16*16-1));
if (ret) {
pr_err("SPORT is busy!\n");
- kfree(sport_handle);
- peripheral_free_list(&sport_req[sport_num][0]);
+ ret = -EBUSY;
+ goto sport_config_err;
+ }
+
+sport_config_err:
+ kfree(sport_handle);
+sport_err:
#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
- gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
+ gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
#endif
- return -EBUSY;
- }
- return 0;
+gpio_err:
+ peripheral_free_list(&sport_req[sport_num][0]);
+peripheral_err:
+ free_page((unsigned long)cmd_count);
+ cmd_count = NULL;
+
+ return ret;
}
static void bf5xx_ac97_remove(struct platform_device *pdev,
@@ -373,6 +398,7 @@ static void bf5xx_ac97_remove(struct platform_device *pdev,
{
free_page((unsigned long)cmd_count);
cmd_count = NULL;
+ peripheral_free_list(&sport_req[sport_num][0]);
#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
#endif
@@ -389,7 +415,11 @@ struct snd_soc_dai bfin_ac97_dai = {
.playback = {
.stream_name = "AC97 Playback",
.channels_min = 2,
+#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT)
+ .channels_max = 6,
+#else
.channels_max = 2,
+#endif
.rates = SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE, },
.capture = {
diff --git a/sound/soc/blackfin/bf5xx-ac97.h b/sound/soc/blackfin/bf5xx-ac97.h
index 3f77cc5..3f2a911 100644
--- a/sound/soc/blackfin/bf5xx-ac97.h
+++ b/sound/soc/blackfin/bf5xx-ac97.h
@@ -16,21 +16,46 @@ struct ac97_frame {
u16 ac97_tag; /* slot 0 */
u16 ac97_addr; /* slot 1 */
u16 ac97_data; /* slot 2 */
- u32 ac97_pcm; /* slot 3 and 4: left and right pcm data */
+ u16 ac97_pcm_l; /*slot 3:front left*/
+ u16 ac97_pcm_r; /*slot 4:front left*/
+#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT)
+ u16 ac97_mdm_l1;
+ u16 ac97_center; /*slot 6:center*/
+ u16 ac97_sl; /*slot 7:surround left*/
+ u16 ac97_sr; /*slot 8:surround right*/
+ u16 ac97_lfe; /*slot 9:lfe*/
+#endif
} __attribute__ ((packed));
+/* Speaker location */
+#define SP_FL 0x0001
+#define SP_FR 0x0010
+#define SP_FC 0x0002
+#define SP_LFE 0x0020
+#define SP_SL 0x0004
+#define SP_SR 0x0040
+
+#define SP_STEREO (SP_FL | SP_FR)
+#define SP_2DOT1 (SP_FL | SP_FR | SP_LFE)
+#define SP_QUAD (SP_FL | SP_FR | SP_SL | SP_SR)
+#define SP_5DOT1 (SP_FL | SP_FR | SP_FC | SP_LFE | SP_SL | SP_SR)
+
#define TAG_VALID 0x8000
#define TAG_CMD 0x6000
#define TAG_PCM_LEFT 0x1000
#define TAG_PCM_RIGHT 0x0800
-#define TAG_PCM (TAG_PCM_LEFT | TAG_PCM_RIGHT)
+#define TAG_PCM_MDM_L1 0x0400
+#define TAG_PCM_CENTER 0x0200
+#define TAG_PCM_SL 0x0100
+#define TAG_PCM_SR 0x0080
+#define TAG_PCM_LFE 0x0040
extern struct snd_soc_dai bfin_ac97_dai;
-void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u32 *src, \
- size_t count);
+void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u16 *src, \
+ size_t count, unsigned int chan_mask);
-void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u32 *dst, \
+void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u16 *dst, \
size_t count);
#endif
diff --git a/sound/soc/blackfin/bf5xx-sport.h b/sound/soc/blackfin/bf5xx-sport.h
index fcadcc0..2e63dea 100644
--- a/sound/soc/blackfin/bf5xx-sport.h
+++ b/sound/soc/blackfin/bf5xx-sport.h
@@ -116,7 +116,7 @@ struct sport_device {
void *err_data;
unsigned char *tx_dma_buf;
unsigned char *rx_dma_buf;
-#ifdef CONFIG_SND_MMAP_SUPPORT
+#ifdef CONFIG_SND_BF5XX_MMAP_SUPPORT
dma_addr_t tx_dma_phy;
dma_addr_t rx_dma_phy;
int tx_pos;/*pcm sample count*/
--
1.5.6.5
^ permalink raw reply related [flat|nested] 17+ messages in thread
* [PATCH 12/17] ASoC: Blackfin: Fix AD1980/1 build with MMAP support disabled
2008-11-18 12:38 ` [PATCH 11/17] ASoC: Blackfin: " Mark Brown
@ 2008-11-18 12:38 ` Mark Brown
2008-11-18 12:38 ` [PATCH 13/17] ASoC: Fix Blackfin AC97 DAI probe function return code Mark Brown
0 siblings, 1 reply; 17+ messages in thread
From: Mark Brown @ 2008-11-18 12:38 UTC (permalink / raw)
To: Takashi Iwai; +Cc: alsa-devel, Mark Brown, Cliff Cai, Bryan Wu
From: Cliff Cai <cliff.cai@analog.com>
clean up redudent code and correct building problem in non-mmap mode
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/blackfin/bf5xx-ac97-pcm.c | 22 +++-------------------
1 files changed, 3 insertions(+), 19 deletions(-)
diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c
index 4be1a49..4d25f73 100644
--- a/sound/soc/blackfin/bf5xx-ac97-pcm.c
+++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c
@@ -193,8 +193,10 @@ static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
bf5xx_mmap_copy(substream, runtime->period_size);
sport->tx_delay_pos = 0;
+#endif
sport_tx_start(sport);
} else
sport_rx_start(sport);
@@ -267,23 +269,6 @@ static int bf5xx_pcm_open(struct snd_pcm_substream *substream)
return ret;
}
-static int bf5xx_pcm_close(struct snd_pcm_substream *substream)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct sport_device *sport = runtime->private_data;
-
- pr_debug("%s enter\n", __func__);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- sport->once = 0;
- memset(sport->tx_dma_buf, 0, runtime->buffer_size *
- sizeof(struct ac97_frame));
- } else
- memset(sport->rx_dma_buf, 0, runtime->buffer_size *
- sizeof(struct ac97_frame));
-
- return 0;
-}
-
#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream,
struct vm_area_struct *vma)
@@ -301,7 +286,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel,
void __user *buf, snd_pcm_uframes_t count)
{
struct snd_pcm_runtime *runtime = substream->runtime;
-
+ unsigned int chan_mask = ac97_chan_mask[runtime->channels - 1];
pr_debug("%s copy pos:0x%lx count:0x%lx\n",
substream->stream ? "Capture" : "Playback", pos, count);
@@ -317,7 +302,6 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel,
struct snd_pcm_ops bf5xx_pcm_ac97_ops = {
.open = bf5xx_pcm_open,
- .close = bf5xx_pcm_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = bf5xx_pcm_hw_params,
.hw_free = bf5xx_pcm_hw_free,
--
1.5.6.5
^ permalink raw reply related [flat|nested] 17+ messages in thread
* [PATCH 13/17] ASoC: Fix Blackfin AC97 DAI probe function return code
2008-11-18 12:38 ` [PATCH 12/17] ASoC: Blackfin: Fix AD1980/1 build with MMAP support disabled Mark Brown
@ 2008-11-18 12:38 ` Mark Brown
2008-11-18 12:38 ` [PATCH 14/17] ASoC: Blackfin: do not force TWI bus for ssm2602 codec Mark Brown
0 siblings, 1 reply; 17+ messages in thread
From: Mark Brown @ 2008-11-18 12:38 UTC (permalink / raw)
To: Takashi Iwai
Cc: alsa-devel, Mark Brown, Cliff Cai, Michael Hennerich, Bryan Wu
From: Michael Hennerich <michael.hennerich@analog.com>
A probe function should have a clean return 0 path.
Cc: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Michael Hennerich <michael.hennerich@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/blackfin/bf5xx-ac97.c | 2 ++
1 files changed, 2 insertions(+), 0 deletions(-)
diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index 65c162c..5dcd3f6 100644
--- a/sound/soc/blackfin/bf5xx-ac97.c
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -378,6 +378,8 @@ static int bf5xx_ac97_probe(struct platform_device *pdev,
goto sport_config_err;
}
+ return 0;
+
sport_config_err:
kfree(sport_handle);
sport_err:
--
1.5.6.5
^ permalink raw reply related [flat|nested] 17+ messages in thread
* [PATCH 14/17] ASoC: Blackfin: do not force TWI bus for ssm2602 codec
2008-11-18 12:38 ` [PATCH 13/17] ASoC: Fix Blackfin AC97 DAI probe function return code Mark Brown
@ 2008-11-18 12:38 ` Mark Brown
2008-11-18 12:38 ` [PATCH 15/17] ASoC: Blackfin: Simplify the MMAP_SUPPORT macros protected code Mark Brown
0 siblings, 1 reply; 17+ messages in thread
From: Mark Brown @ 2008-11-18 12:38 UTC (permalink / raw)
To: Takashi Iwai; +Cc: alsa-devel, Mark Brown, Cliff Cai, Mike Frysinger, Bryan Wu
From: Mike Frysinger <vapier.adi@gmail.com>
Cc: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/blackfin/Kconfig | 1 -
1 files changed, 0 insertions(+), 1 deletions(-)
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index 43d89fc..f0fd235 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -13,7 +13,6 @@ config SND_BF5XX_SOC_SSM2602
select SND_BF5XX_SOC_I2S
select SND_SOC_SSM2602
select I2C
- select I2C_BLACKFIN_TWI
help
Say Y if you want to add support for SoC audio on BF527-EZKIT.
--
1.5.6.5
^ permalink raw reply related [flat|nested] 17+ messages in thread
* [PATCH 15/17] ASoC: Blackfin: Simplify the MMAP_SUPPORT macros protected code
2008-11-18 12:38 ` [PATCH 14/17] ASoC: Blackfin: do not force TWI bus for ssm2602 codec Mark Brown
@ 2008-11-18 12:38 ` Mark Brown
2008-11-18 12:38 ` [PATCH 16/17] ASoC: Blackfin: always set a default value for that GPIO range Mark Brown
0 siblings, 1 reply; 17+ messages in thread
From: Mark Brown @ 2008-11-18 12:38 UTC (permalink / raw)
To: Takashi Iwai; +Cc: Cliff Cai, alsa-devel, Mark Brown, Bryan Wu
From: Bryan Wu <cooloney@kernel.org>
Cc: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/blackfin/bf5xx-ac97-pcm.c | 9 +++------
1 files changed, 3 insertions(+), 6 deletions(-)
diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c
index 4d25f73..d3d51bc 100644
--- a/sound/soc/blackfin/bf5xx-ac97-pcm.c
+++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c
@@ -100,17 +100,14 @@ static void bf5xx_dma_irq(void *data)
* The total rx/tx buffer is for ac97 frame to hold all pcm data
* is 0x20000 * sizeof(struct ac97_frame) / 4.
*/
-#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
static const struct snd_pcm_hardware bf5xx_pcm_hardware = {
.info = SNDRV_PCM_INFO_INTERLEAVED |
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_BLOCK_TRANSFER,
-#else
-static const struct snd_pcm_hardware bf5xx_pcm_hardware = {
- .info = SNDRV_PCM_INFO_INTERLEAVED |
- SNDRV_PCM_INFO_BLOCK_TRANSFER,
#endif
+ SNDRV_PCM_INFO_BLOCK_TRANSFER,
+
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.period_bytes_min = 32,
.period_bytes_max = 0x10000,
--
1.5.6.5
^ permalink raw reply related [flat|nested] 17+ messages in thread
* [PATCH 16/17] ASoC: Blackfin: always set a default value for that GPIO range
2008-11-18 12:38 ` [PATCH 15/17] ASoC: Blackfin: Simplify the MMAP_SUPPORT macros protected code Mark Brown
@ 2008-11-18 12:38 ` Mark Brown
2008-11-18 12:38 ` [PATCH 17/17] ASoC: Move uda134x_codec.h to uda134x.h Mark Brown
0 siblings, 1 reply; 17+ messages in thread
From: Mark Brown @ 2008-11-18 12:38 UTC (permalink / raw)
To: Takashi Iwai; +Cc: alsa-devel, Mark Brown, Cliff Cai, Mike Frysinger, Bryan Wu
From: Mike Frysinger <vapier.adi@gmail.com>
Cc: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/blackfin/Kconfig | 1 +
1 files changed, 1 insertions(+), 0 deletions(-)
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index f0fd235..e162cbb 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -104,5 +104,6 @@ config SND_BF5XX_RESET_GPIO_NUM
range 0 159
default 19 if BFIN548_EZKIT
default 5 if BFIN537_STAMP
+ default 0
help
Set the correct GPIO for RESET the sound chip.
--
1.5.6.5
^ permalink raw reply related [flat|nested] 17+ messages in thread
* [PATCH 17/17] ASoC: Move uda134x_codec.h to uda134x.h
2008-11-18 12:38 ` [PATCH 16/17] ASoC: Blackfin: always set a default value for that GPIO range Mark Brown
@ 2008-11-18 12:38 ` Mark Brown
0 siblings, 0 replies; 17+ messages in thread
From: Mark Brown @ 2008-11-18 12:38 UTC (permalink / raw)
To: Takashi Iwai; +Cc: alsa-devel, Mark Brown
For consistency with other ASoC codec drivers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/codecs/uda134x.c | 2 +-
sound/soc/codecs/uda134x.h | 36 +++++++++++++++++++++++++++++++++++
sound/soc/codecs/uda134x_codec.h | 36 -----------------------------------
sound/soc/s3c24xx/s3c24xx_uda134x.c | 2 +-
4 files changed, 38 insertions(+), 38 deletions(-)
create mode 100644 sound/soc/codecs/uda134x.h
delete mode 100644 sound/soc/codecs/uda134x_codec.h
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 04b30da..69ef521 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -24,7 +24,7 @@
#include <sound/uda134x.h>
#include <sound/l3.h>
-#include "uda134x_codec.h"
+#include "uda134x.h"
#define POWER_OFF_ON_STANDBY 1
diff --git a/sound/soc/codecs/uda134x.h b/sound/soc/codecs/uda134x.h
new file mode 100644
index 0000000..94f4404
--- /dev/null
+++ b/sound/soc/codecs/uda134x.h
@@ -0,0 +1,36 @@
+#ifndef _UDA134X_CODEC_H
+#define _UDA134X_CODEC_H
+
+#define UDA134X_L3ADDR 5
+#define UDA134X_DATA0_ADDR ((UDA134X_L3ADDR << 2) | 0)
+#define UDA134X_DATA1_ADDR ((UDA134X_L3ADDR << 2) | 1)
+#define UDA134X_STATUS_ADDR ((UDA134X_L3ADDR << 2) | 2)
+
+#define UDA134X_EXTADDR_PREFIX 0xC0
+#define UDA134X_EXTDATA_PREFIX 0xE0
+
+/* UDA134X registers */
+#define UDA134X_EA000 0
+#define UDA134X_EA001 1
+#define UDA134X_EA010 2
+#define UDA134X_EA011 3
+#define UDA134X_EA100 4
+#define UDA134X_EA101 5
+#define UDA134X_EA110 6
+#define UDA134X_EA111 7
+#define UDA134X_STATUS0 8
+#define UDA134X_STATUS1 9
+#define UDA134X_DATA000 10
+#define UDA134X_DATA001 11
+#define UDA134X_DATA010 12
+#define UDA134X_DATA1 13
+
+#define UDA134X_REGS_NUM 14
+
+#define STATUS0_DAIFMT_MASK (~(7<<1))
+#define STATUS0_SYSCLK_MASK (~(3<<4))
+
+extern struct snd_soc_dai uda134x_dai;
+extern struct snd_soc_codec_device soc_codec_dev_uda134x;
+
+#endif
diff --git a/sound/soc/codecs/uda134x_codec.h b/sound/soc/codecs/uda134x_codec.h
deleted file mode 100644
index 94f4404..0000000
--- a/sound/soc/codecs/uda134x_codec.h
+++ /dev/null
@@ -1,36 +0,0 @@
-#ifndef _UDA134X_CODEC_H
-#define _UDA134X_CODEC_H
-
-#define UDA134X_L3ADDR 5
-#define UDA134X_DATA0_ADDR ((UDA134X_L3ADDR << 2) | 0)
-#define UDA134X_DATA1_ADDR ((UDA134X_L3ADDR << 2) | 1)
-#define UDA134X_STATUS_ADDR ((UDA134X_L3ADDR << 2) | 2)
-
-#define UDA134X_EXTADDR_PREFIX 0xC0
-#define UDA134X_EXTDATA_PREFIX 0xE0
-
-/* UDA134X registers */
-#define UDA134X_EA000 0
-#define UDA134X_EA001 1
-#define UDA134X_EA010 2
-#define UDA134X_EA011 3
-#define UDA134X_EA100 4
-#define UDA134X_EA101 5
-#define UDA134X_EA110 6
-#define UDA134X_EA111 7
-#define UDA134X_STATUS0 8
-#define UDA134X_STATUS1 9
-#define UDA134X_DATA000 10
-#define UDA134X_DATA001 11
-#define UDA134X_DATA010 12
-#define UDA134X_DATA1 13
-
-#define UDA134X_REGS_NUM 14
-
-#define STATUS0_DAIFMT_MASK (~(7<<1))
-#define STATUS0_SYSCLK_MASK (~(3<<4))
-
-extern struct snd_soc_dai uda134x_dai;
-extern struct snd_soc_codec_device soc_codec_dev_uda134x;
-
-#endif
diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c
index 29a6813..487b010 100644
--- a/sound/soc/s3c24xx/s3c24xx_uda134x.c
+++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c
@@ -26,7 +26,7 @@
#include "s3c24xx-pcm.h"
#include "s3c24xx-i2s.h"
-#include "../codecs/uda134x_codec.h"
+#include "../codecs/uda134x.h"
/* #define ENFORCE_RATES 1 */
--
1.5.6.5
^ permalink raw reply related [flat|nested] 17+ messages in thread
end of thread, other threads:[~2008-11-18 12:38 UTC | newest]
Thread overview: 17+ messages (download: mbox.gz follow: Atom feed
-- links below jump to the message on this page --
2008-11-18 12:38 [PATCH 01/17] ASoC: Fix for master playback/capture volume range for TWL4030 codec Mark Brown
2008-11-18 12:38 ` [PATCH 02/17] ASoC: UDA134x codec driver Mark Brown
2008-11-18 12:38 ` [PATCH 03/17] ASoC: Machine driver for for s3c24xx with uda134x Mark Brown
2008-11-18 12:38 ` [PATCH 04/17] ASoC: Allow writes to uncached registers in WM8990 Mark Brown
2008-11-18 12:38 ` [PATCH 05/17] ASoC: Enable WM8990 ADC clocking workaround Mark Brown
2008-11-18 12:38 ` [PATCH 06/17] ASoC: Manage VMID mode for WM8990 Mark Brown
2008-11-18 12:38 ` [PATCH 07/17] ASoC: Build tlv320aic23 cleanly Mark Brown
2008-11-18 12:38 ` [PATCH 08/17] ASoC: DaVinci: Fix audio stall when doing full duplex Mark Brown
2008-11-18 12:38 ` [PATCH 09/17] ASoC: Blackfin: updates Kconfig for SPORT Mark Brown
2008-11-18 12:38 ` [PATCH 10/17] ASoC: AD1980 codec: add multi-channel function support Mark Brown
2008-11-18 12:38 ` [PATCH 11/17] ASoC: Blackfin: " Mark Brown
2008-11-18 12:38 ` [PATCH 12/17] ASoC: Blackfin: Fix AD1980/1 build with MMAP support disabled Mark Brown
2008-11-18 12:38 ` [PATCH 13/17] ASoC: Fix Blackfin AC97 DAI probe function return code Mark Brown
2008-11-18 12:38 ` [PATCH 14/17] ASoC: Blackfin: do not force TWI bus for ssm2602 codec Mark Brown
2008-11-18 12:38 ` [PATCH 15/17] ASoC: Blackfin: Simplify the MMAP_SUPPORT macros protected code Mark Brown
2008-11-18 12:38 ` [PATCH 16/17] ASoC: Blackfin: always set a default value for that GPIO range Mark Brown
2008-11-18 12:38 ` [PATCH 17/17] ASoC: Move uda134x_codec.h to uda134x.h Mark Brown
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