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Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

Browse source 
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (214 commits)
  ALSA: hda - Add pin-fix for HP dc5750
  ALSA: als4000: Fix potentially invalid DMA mode setup
  ALSA: als4000: enable burst mode
  ALSA: hda - Fix initial capsrc selection in patch_alc269()
  ASoC: TWL4030: Capture route runtime DAPM ordering fix
  ALSA: hda - Add PC-beep whitelist for an Intel board
  ALSA: hda - More relax for pending period handling
  ALSA: hda - Define AC_FMT_* constants
  ALSA: hda - Fix beep frequency on IDT 92HD73xx and 92HD71Bxx codecs
  ALSA: hda - Add support for HDMI HBR passthrough
  ALSA: hda - Set Stream Type in Stream Format according to AES0
  ALSA: hda - Fix Thinkpad X300 so SPDIF is not exposed
  ALSA: hda - FIX to not expose SPDIF on Thinkpad X301, since it does not have the ability to use SPDIF
  ASoC: wm9081: fix resource reclaim in wm9081_register error path
  ASoC: wm8978: fix a memory leak if a wm8978_register fail
  ASoC: wm8974: fix a memory leak if another WM8974 is registered
  ASoC: wm8961: fix resource reclaim in wm8961_register error path
  ASoC: wm8955: fix resource reclaim in wm8955_register error path
  ASoC: wm8940: fix a memory leak if wm8940_register return error
  ASoC: wm8904: fix resource reclaim in wm8904_register error path
  ...
This commit is contained in:
Linus Torvalds 2010-08-07 17:07:31 -07:00
commit faa38b5e0e
194 changed files with 10998 additions and 1418 deletions
Download patch file Download diff file Expand all files Collapse all files

4
sound/soc/Kconfig
View file

@ -28,9 +28,13 @@ source "sound/soc/atmel/Kconfig"
source "sound/soc/au1x/Kconfig"
source "sound/soc/blackfin/Kconfig"
source "sound/soc/davinci/Kconfig"
source "sound/soc/ep93xx/Kconfig"
source "sound/soc/fsl/Kconfig"
source "sound/soc/imx/Kconfig"
source "sound/soc/jz4740/Kconfig"
source "sound/soc/nuc900/Kconfig"
source "sound/soc/omap/Kconfig"
source "sound/soc/kirkwood/Kconfig"
source "sound/soc/pxa/Kconfig"
source "sound/soc/s3c24xx/Kconfig"
source "sound/soc/s6000/Kconfig"

4
sound/soc/Makefile
View file

@ -6,9 +6,13 @@ obj-$(CONFIG_SND_SOC) += atmel/
obj-$(CONFIG_SND_SOC) += au1x/
obj-$(CONFIG_SND_SOC) += blackfin/
obj-$(CONFIG_SND_SOC) += davinci/
obj-$(CONFIG_SND_SOC) += ep93xx/
obj-$(CONFIG_SND_SOC) += fsl/
obj-$(CONFIG_SND_SOC) += imx/
obj-$(CONFIG_SND_SOC) += jz4740/
obj-$(CONFIG_SND_SOC) += nuc900/
obj-$(CONFIG_SND_SOC) += omap/
obj-$(CONFIG_SND_SOC) += kirkwood/
obj-$(CONFIG_SND_SOC) += pxa/
obj-$(CONFIG_SND_SOC) += s3c24xx/
obj-$(CONFIG_SND_SOC) += s6000/

1
sound/soc/atmel/atmel-pcm.c
View file

@ -77,7 +77,6 @@ struct atmel_runtime_data {
size_t period_size;
dma_addr_t period_ptr; /* physical address of next period */
int periods; /* period index of period_ptr */
/* PDC register save */
u32 pdc_xpr_save;

1
sound/soc/atmel/atmel_ssc_dai.c
View file

@ -549,7 +549,6 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
printk(KERN_WARNING "atmel_ssc_dai: unsupported DAI format 0x%x\n",
ssc_p->daifmt);
return -EINVAL;
break;
}
pr_debug("atmel_ssc_hw_params: "
"RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n",

13
sound/soc/au1x/psc-ac97.c
View file

@ -375,12 +375,10 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev)
}
ret = -EBUSY;
wd->ioarea = request_mem_region(r->start, r->end - r->start + 1,
"au1xpsc_ac97");
if (!wd->ioarea)
if (!request_mem_region(r->start, resource_size(r), pdev->name))
goto out0;
wd->mmio = ioremap(r->start, 0xffff);
wd->mmio = ioremap(r->start, resource_size(r));
if (!wd->mmio)
goto out1;
@ -410,8 +408,7 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev)
snd_soc_unregister_dai(&au1xpsc_ac97_dai);
out1:
release_resource(wd->ioarea);
kfree(wd->ioarea);
release_mem_region(r->start, resource_size(r));
out0:
kfree(wd);
return ret;
@ -420,6 +417,7 @@ out0:
static int __devexit au1xpsc_ac97_drvremove(struct platform_device *pdev)
{
struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev);
struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (wd->dmapd)
au1xpsc_pcm_destroy(wd->dmapd);
@ -433,8 +431,7 @@ static int __devexit au1xpsc_ac97_drvremove(struct platform_device *pdev)
au_sync();
iounmap(wd->mmio);
release_resource(wd->ioarea);
kfree(wd->ioarea);
release_mem_region(r->start, resource_size(r));
kfree(wd);
au1xpsc_ac97_workdata = NULL; /* MDEV */

13
sound/soc/au1x/psc-i2s.c
View file

@ -321,12 +321,10 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev)
}
ret = -EBUSY;
wd->ioarea = request_mem_region(r->start, r->end - r->start + 1,
"au1xpsc_i2s");
if (!wd->ioarea)
if (!request_mem_region(r->start, resource_size(r), pdev->name))
goto out0;
wd->mmio = ioremap(r->start, 0xffff);
wd->mmio = ioremap(r->start, resource_size(r));
if (!wd->mmio)
goto out1;
@ -362,8 +360,7 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev)
snd_soc_unregister_dai(&au1xpsc_i2s_dai);
out1:
release_resource(wd->ioarea);
kfree(wd->ioarea);
release_mem_region(r->start, resource_size(r));
out0:
kfree(wd);
return ret;
@ -372,6 +369,7 @@ out0:
static int __devexit au1xpsc_i2s_drvremove(struct platform_device *pdev)
{
struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev);
struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (wd->dmapd)
au1xpsc_pcm_destroy(wd->dmapd);
@ -384,8 +382,7 @@ static int __devexit au1xpsc_i2s_drvremove(struct platform_device *pdev)
au_sync();
iounmap(wd->mmio);
release_resource(wd->ioarea);
kfree(wd->ioarea);
release_mem_region(r->start, resource_size(r));
kfree(wd);
au1xpsc_i2s_workdata = NULL; /* MDEV */

1
sound/soc/au1x/psc.h
View file

@ -32,7 +32,6 @@ struct au1xpsc_audio_data {
unsigned long rate;
unsigned long pm[2];
struct resource *ioarea;
struct mutex lock;
struct platform_device *dmapd;
};

6
sound/soc/blackfin/bf5xx-ac97.c
View file

@ -255,8 +255,7 @@ EXPORT_SYMBOL_GPL(soc_ac97_ops);
#ifdef CONFIG_PM
static int bf5xx_ac97_suspend(struct snd_soc_dai *dai)
{
struct sport_device *sport =
(struct sport_device *)dai->private_data;
struct sport_device *sport = dai->private_data;
pr_debug("%s : sport %d\n", __func__, dai->id);
if (!dai->active)
@ -271,8 +270,7 @@ static int bf5xx_ac97_suspend(struct snd_soc_dai *dai)
static int bf5xx_ac97_resume(struct snd_soc_dai *dai)
{
int ret;
struct sport_device *sport =
(struct sport_device *)dai->private_data;
struct sport_device *sport = dai->private_data;
pr_debug("%s : sport %d\n", __func__, dai->id);
if (!dai->active)

6
sound/soc/blackfin/bf5xx-tdm.c
View file

@ -210,8 +210,7 @@ static int bf5xx_tdm_set_channel_map(struct snd_soc_dai *dai,
#ifdef CONFIG_PM
static int bf5xx_tdm_suspend(struct snd_soc_dai *dai)
{
struct sport_device *sport =
(struct sport_device *)dai->private_data;
struct sport_device *sport = dai->private_data;
if (!dai->active)
return 0;
@ -225,8 +224,7 @@ static int bf5xx_tdm_suspend(struct snd_soc_dai *dai)
static int bf5xx_tdm_resume(struct snd_soc_dai *dai)
{
int ret;
struct sport_device *sport =
(struct sport_device *)dai->private_data;
struct sport_device *sport = dai->private_data;
if (!dai->active)
return 0;

20
sound/soc/codecs/Kconfig
View file

@ -22,9 +22,11 @@ config SND_SOC_ALL_CODECS
select SND_SOC_AK4642 if I2C
select SND_SOC_AK4671 if I2C
select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC
select SND_SOC_CS42L51 if I2C
select SND_SOC_CS4270 if I2C
select SND_SOC_MAX9877 if I2C
select SND_SOC_DA7210 if I2C
select SND_SOC_JZ4740 if SOC_JZ4740
select SND_SOC_MAX9877 if I2C
select SND_SOC_PCM3008
select SND_SOC_SPDIF
select SND_SOC_SSM2602 if I2C
@ -48,6 +50,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_WM8727
select SND_SOC_WM8728 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8731 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8741 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8750 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI
@ -120,13 +123,13 @@ config SND_SOC_AK4671
config SND_SOC_CQ0093VC
tristate
config SND_SOC_CS42L51
tristate
# Cirrus Logic CS4270 Codec
config SND_SOC_CS4270
tristate
config SND_SOC_DA7210
tristate
# Cirrus Logic CS4270 Codec VD = 3.3V Errata
# Select if you are affected by the errata where the part will not function
# if MCLK divide-by-1.5 is selected and VD is set to 3.3V. The driver will
@ -138,9 +141,15 @@ config SND_SOC_CS4270_VD33_ERRATA
config SND_SOC_CX20442
tristate
config SND_SOC_JZ4740_CODEC
tristate
config SND_SOC_L3
tristate
config SND_SOC_DA7210
tristate
config SND_SOC_PCM3008
tristate
@ -206,6 +215,9 @@ config SND_SOC_WM8728
config SND_SOC_WM8731
tristate
config SND_SOC_WM8741
tristate
config SND_SOC_WM8750
tristate

6
sound/soc/codecs/Makefile
View file

@ -9,6 +9,7 @@ snd-soc-ak4535-objs := ak4535.o
snd-soc-ak4642-objs := ak4642.o
snd-soc-ak4671-objs := ak4671.o
snd-soc-cq93vc-objs := cq93vc.o
snd-soc-cs42l51-objs := cs42l51.o
snd-soc-cs4270-objs := cs4270.o
snd-soc-cx20442-objs := cx20442.o
snd-soc-da7210-objs := da7210.o
@ -34,6 +35,7 @@ snd-soc-wm8711-objs := wm8711.o
snd-soc-wm8727-objs := wm8727.o
snd-soc-wm8728-objs := wm8728.o
snd-soc-wm8731-objs := wm8731.o
snd-soc-wm8741-objs := wm8741.o
snd-soc-wm8750-objs := wm8750.o
snd-soc-wm8753-objs := wm8753.o
snd-soc-wm8776-objs := wm8776.o
@ -56,6 +58,7 @@ snd-soc-wm9705-objs := wm9705.o
snd-soc-wm9712-objs := wm9712.o
snd-soc-wm9713-objs := wm9713.o
snd-soc-wm-hubs-objs := wm_hubs.o
snd-soc-jz4740-codec-objs := jz4740.o
# Amp
snd-soc-max9877-objs := max9877.o
@ -74,10 +77,12 @@ obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o
obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o
obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o
obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o
obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o
obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o
obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o
obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o
obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o
obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o
@ -99,6 +104,7 @@ obj-$(CONFIG_SND_SOC_WM8711) += snd-soc-wm8711.o
obj-$(CONFIG_SND_SOC_WM8727) += snd-soc-wm8727.o
obj-$(CONFIG_SND_SOC_WM8728) += snd-soc-wm8728.o
obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o
obj-$(CONFIG_SND_SOC_WM8741) += snd-soc-wm8741.o
obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o
obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o
obj-$(CONFIG_SND_SOC_WM8776) += snd-soc-wm8776.o

1
sound/soc/codecs/ad1836.c
View file

@ -272,6 +272,7 @@ static int ad1836_register(struct ad1836_priv *ad1836)
if (ad1836_codec) {
dev_err(codec->dev, "Another ad1836 is registered\n");
kfree(ad1836);
return -EINVAL;
}

41
sound/soc/codecs/ad193x.c
View file

@ -24,6 +24,7 @@
/* codec private data */
struct ad193x_priv {
unsigned int sysclk;
struct snd_soc_codec codec;
u8 reg_cache[AD193X_NUM_REGS];
};
@ -251,15 +252,32 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai,
return 0;
}
static int ad193x_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 ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec);
switch (freq) {
case 12288000:
case 18432000:
case 24576000:
case 36864000:
ad193x->sysclk = freq;
return 0;
}
return -EINVAL;
}
static int ad193x_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
int word_len = 0, reg = 0;
int word_len = 0, reg = 0, master_rate = 0;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->card->codec;
struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec);
/* bit size */
switch (params_format(params)) {
@ -275,6 +293,25 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream,
break;
}
switch (ad193x->sysclk) {
case 12288000:
master_rate = AD193X_PLL_INPUT_256;
break;
case 18432000:
master_rate = AD193X_PLL_INPUT_384;
break;
case 24576000:
master_rate = AD193X_PLL_INPUT_512;
break;
case 36864000:
master_rate = AD193X_PLL_INPUT_768;
break;
}
reg = snd_soc_read(codec, AD193X_PLL_CLK_CTRL0);
reg = (reg & AD193X_PLL_INPUT_MASK) | master_rate;
snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, reg);
reg = snd_soc_read(codec, AD193X_DAC_CTRL2);
reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) | word_len;
snd_soc_write(codec, AD193X_DAC_CTRL2, reg);
@ -348,6 +385,7 @@ static int ad193x_bus_probe(struct device *dev, void *ctrl_data, int bus_type)
/* pll input: mclki/xi */
snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */
snd_soc_write(codec, AD193X_PLL_CLK_CTRL1, 0x04);
ad193x->sysclk = 12288000;
ret = snd_soc_register_codec(codec);
if (ret != 0) {
@ -383,6 +421,7 @@ static struct snd_soc_dai_ops ad193x_dai_ops = {
.hw_params = ad193x_hw_params,
.digital_mute = ad193x_mute,
.set_tdm_slot = ad193x_set_tdm_slot,
.set_sysclk = ad193x_set_dai_sysclk,
.set_fmt = ad193x_set_dai_fmt,
};

5
sound/soc/codecs/ad193x.h
View file

@ -11,6 +11,11 @@
#define AD193X_PLL_CLK_CTRL0 0x800
#define AD193X_PLL_POWERDOWN 0x01
#define AD193X_PLL_INPUT_MASK (~0x6)
#define AD193X_PLL_INPUT_256 (0 << 1)
#define AD193X_PLL_INPUT_384 (1 << 1)
#define AD193X_PLL_INPUT_512 (2 << 1)
#define AD193X_PLL_INPUT_768 (3 << 1)
#define AD193X_PLL_CLK_CTRL1 0x801
#define AD193X_DAC_CTRL0 0x802
#define AD193X_DAC_POWERDOWN 0x01

36
sound/soc/codecs/ak4642.c
View file

@ -22,20 +22,13 @@
* AK4643 is tested.
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <sound/core.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/tlv.h>
#include "ak4642.h"
@ -111,6 +104,23 @@
struct snd_soc_codec_device soc_codec_dev_ak4642;
/*
* Playback Volume (table 39)
*
* max : 0x00 : +12.0 dB
* ( 0.5 dB step )
* min : 0xFE : -115.0 dB
* mute: 0xFF
*/
static const DECLARE_TLV_DB_SCALE(out_tlv, -11500, 50, 1);
static const struct snd_kcontrol_new ak4642_snd_controls[] = {
SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC,
0, 0xFF, 1, out_tlv),
};
/* codec private data */
struct ak4642_priv {
struct snd_soc_codec codec;
@ -204,7 +214,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream,
*
* PLL, Master Mode
* Audio I/F Format :MSB justified (ADC & DAC)
* Digital Volume: -8dB
* Bass Boost Level : Middle
*
* This operation came from example code of
@ -214,8 +223,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream,
ak4642_write(codec, 0x0e, 0x19);
ak4642_write(codec, 0x09, 0x91);
ak4642_write(codec, 0x0c, 0x91);
ak4642_write(codec, 0x0a, 0x28);
ak4642_write(codec, 0x0d, 0x28);
ak4642_write(codec, 0x00, 0x64);
snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP);
snd_soc_update_bits(codec, PW_MGMT2, HPMTN, HPMTN);
@ -491,8 +498,10 @@ static int ak4642_i2c_probe(struct i2c_client *i2c,
codec->control_data = i2c;
ret = ak4642_init(ak4642);
if (ret < 0)
if (ret < 0) {
printk(KERN_ERR "failed to initialise AK4642\n");
kfree(ak4642);
}
return ret;
}
@ -548,6 +557,9 @@ static int ak4642_probe(struct platform_device *pdev)
goto pcm_err;
}
snd_soc_add_controls(ak4642_codec, ak4642_snd_controls,
ARRAY_SIZE(ak4642_snd_controls));
dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION);
return ret;

763
sound/soc/codecs/cs42l51.c Normal file
View file

@ -0,0 +1,763 @@
/*
* cs42l51.c
*
* ASoC Driver for Cirrus Logic CS42L51 codecs
*
* Copyright (c) 2010 Arnaud Patard <apatard@mandriva.com>
*
* Based on cs4270.c - Copyright (c) Freescale Semiconductor
*
* 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.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* For now:
* - Only I2C is support. Not SPI
* - master mode *NOT* supported
*/
#include <linux/module.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/tlv.h>
#include <sound/initval.h>
#include <sound/pcm_params.h>
#include <sound/pcm.h>
#include <linux/i2c.h>
#include "cs42l51.h"
enum master_slave_mode {
MODE_SLAVE,
MODE_SLAVE_AUTO,
MODE_MASTER,
};
struct cs42l51_private {
unsigned int mclk;
unsigned int audio_mode; /* The mode (I2S or left-justified) */
enum master_slave_mode func;
struct snd_soc_codec codec;
u8 reg_cache[CS42L51_NUMREGS];
};
static struct snd_soc_codec *cs42l51_codec;
#define CS42L51_FORMATS ( \
SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \
SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \
SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \
SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE)
static int cs42l51_fill_cache(struct snd_soc_codec *codec)
{
u8 *cache = codec->reg_cache + 1;
struct i2c_client *i2c_client = codec->control_data;
s32 length;
length = i2c_smbus_read_i2c_block_data(i2c_client,
CS42L51_FIRSTREG | 0x80, CS42L51_NUMREGS, cache);
if (length != CS42L51_NUMREGS) {
dev_err(&i2c_client->dev,
"I2C read failure, addr=0x%x (ret=%d vs %d)\n",
i2c_client->addr, length, CS42L51_NUMREGS);
return -EIO;
}
return 0;
}
static int cs42l51_i2c_probe(struct i2c_client *i2c_client,
const struct i2c_device_id *id)
{
struct snd_soc_codec *codec;
struct cs42l51_private *cs42l51;
int ret = 0;
int reg;
if (cs42l51_codec)
return -EBUSY;
/* Verify that we have a CS42L51 */
ret = i2c_smbus_read_byte_data(i2c_client, CS42L51_CHIP_REV_ID);
if (ret < 0) {
dev_err(&i2c_client->dev, "failed to read I2C\n");
goto error;
}
if ((ret != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_A)) &&
(ret != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_B))) {
dev_err(&i2c_client->dev, "Invalid chip id\n");
ret = -ENODEV;
goto error;
}
dev_info(&i2c_client->dev, "found device cs42l51 rev %d\n",
ret & 7);
cs42l51 = kzalloc(sizeof(struct cs42l51_private), GFP_KERNEL);
if (!cs42l51) {
dev_err(&i2c_client->dev, "could not allocate codec\n");
return -ENOMEM;
}
codec = &cs42l51->codec;
mutex_init(&codec->mutex);
INIT_LIST_HEAD(&codec->dapm_widgets);
INIT_LIST_HEAD(&codec->dapm_paths);
codec->dev = &i2c_client->dev;
codec->name = "CS42L51";
codec->owner = THIS_MODULE;
codec->dai = &cs42l51_dai;
codec->num_dai = 1;
snd_soc_codec_set_drvdata(codec, cs42l51);
codec->control_data = i2c_client;
codec->reg_cache = cs42l51->reg_cache;
codec->reg_cache_size = CS42L51_NUMREGS;
i2c_set_clientdata(i2c_client, codec);
ret = cs42l51_fill_cache(codec);
if (ret < 0) {
dev_err(&i2c_client->dev, "failed to fill register cache\n");
goto error_alloc;
}
ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
if (ret < 0) {
dev_err(&i2c_client->dev, "Failed to set cache I/O: %d\n", ret);
goto error_alloc;
}
/*
* DAC configuration
* - Use signal processor
* - auto mute
* - vol changes immediate
* - no de-emphasize
*/
reg = CS42L51_DAC_CTL_DATA_SEL(1)
| CS42L51_DAC_CTL_AMUTE | CS42L51_DAC_CTL_DACSZ(0);
ret = snd_soc_write(codec, CS42L51_DAC_CTL, reg);
if (ret < 0)
goto error_alloc;
cs42l51_dai.dev = codec->dev;
cs42l51_codec = codec;
ret = snd_soc_register_codec(codec);
if (ret != 0) {
dev_err(codec->dev, "Failed to register codec: %d\n", ret);
goto error_alloc;
}
ret = snd_soc_register_dai(&cs42l51_dai);
if (ret < 0) {
dev_err(&i2c_client->dev, "failed to register DAIe\n");
goto error_reg;
}
return 0;
error_reg:
snd_soc_unregister_codec(codec);
error_alloc:
kfree(cs42l51);
error:
return ret;
}
static int cs42l51_i2c_remove(struct i2c_client *client)
{
struct cs42l51_private *cs42l51 = i2c_get_clientdata(client);
snd_soc_unregister_dai(&cs42l51_dai);
snd_soc_unregister_codec(cs42l51_codec);
cs42l51_codec = NULL;
kfree(cs42l51);
return 0;
}
static const struct i2c_device_id cs42l51_id[] = {
{"cs42l51", 0},
{}
};
MODULE_DEVICE_TABLE(i2c, cs42l51_id);
static struct i2c_driver cs42l51_i2c_driver = {
.driver = {
.name = "CS42L51 I2C",
.owner = THIS_MODULE,
},
.id_table = cs42l51_id,
.probe = cs42l51_i2c_probe,
.remove = cs42l51_i2c_remove,
};
static int cs42l51_get_chan_mix(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned long value = snd_soc_read(codec, CS42L51_PCM_MIXER)&3;
switch (value) {
default:
case 0:
ucontrol->value.integer.value[0] = 0;
break;
/* same value : (L+R)/2 and (R+L)/2 */
case 1:
case 2:
ucontrol->value.integer.value[0] = 1;
break;
case 3:
ucontrol->value.integer.value[0] = 2;
break;
}
return 0;
}
#define CHAN_MIX_NORMAL 0x00
#define CHAN_MIX_BOTH 0x55
#define CHAN_MIX_SWAP 0xFF
static int cs42l51_set_chan_mix(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned char val;
switch (ucontrol->value.integer.value[0]) {
default:
case 0:
val = CHAN_MIX_NORMAL;
break;
case 1:
val = CHAN_MIX_BOTH;
break;
case 2:
val = CHAN_MIX_SWAP;
break;
}
snd_soc_write(codec, CS42L51_PCM_MIXER, val);
return 1;
}
static const DECLARE_TLV_DB_SCALE(adc_pcm_tlv, -5150, 50, 0);
static const DECLARE_TLV_DB_SCALE(tone_tlv, -1050, 150, 0);
/* This is a lie. after -102 db, it stays at -102 */
/* maybe a range would be better */
static const DECLARE_TLV_DB_SCALE(aout_tlv, -11550, 50, 0);
static const DECLARE_TLV_DB_SCALE(boost_tlv, 1600, 1600, 0);
static const char *chan_mix[] = {
"L R",
"L+R",
"R L",
};
static const struct soc_enum cs42l51_chan_mix =
SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(chan_mix), chan_mix);
static const struct snd_kcontrol_new cs42l51_snd_controls[] = {
SOC_DOUBLE_R_SX_TLV("PCM Playback Volume",
CS42L51_PCMA_VOL, CS42L51_PCMB_VOL,
7, 0xffffff99, 0x18, adc_pcm_tlv),
SOC_DOUBLE_R("PCM Playback Switch",
CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, 7, 1, 1),
SOC_DOUBLE_R_SX_TLV("Analog Playback Volume",
CS42L51_AOUTA_VOL, CS42L51_AOUTB_VOL,
8, 0xffffff19, 0x18, aout_tlv),
SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume",
CS42L51_ADCA_VOL, CS42L51_ADCB_VOL,
7, 0xffffff99, 0x18, adc_pcm_tlv),
SOC_DOUBLE_R("ADC Mixer Switch",
CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, 7, 1, 1),
SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0),
SOC_SINGLE("Auto-Mute Switch", CS42L51_DAC_CTL, 2, 1, 0),
SOC_SINGLE("Soft Ramp Switch", CS42L51_DAC_CTL, 1, 1, 0),
SOC_SINGLE("Zero Cross Switch", CS42L51_DAC_CTL, 0, 0, 0),
SOC_DOUBLE_TLV("Mic Boost Volume",
CS42L51_MIC_CTL, 0, 1, 1, 0, boost_tlv),
SOC_SINGLE_TLV("Bass Volume", CS42L51_TONE_CTL, 0, 0xf, 1, tone_tlv),
SOC_SINGLE_TLV("Treble Volume", CS42L51_TONE_CTL, 4, 0xf, 1, tone_tlv),
SOC_ENUM_EXT("PCM channel mixer",
cs42l51_chan_mix,
cs42l51_get_chan_mix, cs42l51_set_chan_mix),
};
/*
* to power down, one must:
* 1.) Enable the PDN bit
* 2.) enable power-down for the select channels
* 3.) disable the PDN bit.
*/
static int cs42l51_pdn_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
unsigned long value;
value = snd_soc_read(w->codec, CS42L51_POWER_CTL1);
value &= ~CS42L51_POWER_CTL1_PDN;
switch (event) {
case SND_SOC_DAPM_PRE_PMD:
value |= CS42L51_POWER_CTL1_PDN;
break;
default:
case SND_SOC_DAPM_POST_PMD:
break;
}
snd_soc_update_bits(w->codec, CS42L51_POWER_CTL1,
CS42L51_POWER_CTL1_PDN, value);
return 0;
}
static const char *cs42l51_dac_names[] = {"Direct PCM",
"DSP PCM", "ADC"};
static const struct soc_enum cs42l51_dac_mux_enum =
SOC_ENUM_SINGLE(CS42L51_DAC_CTL, 6, 3, cs42l51_dac_names);
static const struct snd_kcontrol_new cs42l51_dac_mux_controls =
SOC_DAPM_ENUM("Route", cs42l51_dac_mux_enum);
static const char *cs42l51_adcl_names[] = {"AIN1 Left", "AIN2 Left",
"MIC Left", "MIC+preamp Left"};
static const struct soc_enum cs42l51_adcl_mux_enum =
SOC_ENUM_SINGLE(CS42L51_ADC_INPUT, 4, 4, cs42l51_adcl_names);
static const struct snd_kcontrol_new cs42l51_adcl_mux_controls =
SOC_DAPM_ENUM("Route", cs42l51_adcl_mux_enum);
static const char *cs42l51_adcr_names[] = {"AIN1 Right", "AIN2 Right",
"MIC Right", "MIC+preamp Right"};
static const struct soc_enum cs42l51_adcr_mux_enum =
SOC_ENUM_SINGLE(CS42L51_ADC_INPUT, 6, 4, cs42l51_adcr_names);
static const struct snd_kcontrol_new cs42l51_adcr_mux_controls =
SOC_DAPM_ENUM("Route", cs42l51_adcr_mux_enum);
static const struct snd_soc_dapm_widget cs42l51_dapm_widgets[] = {
SND_SOC_DAPM_MICBIAS("Mic Bias", CS42L51_MIC_POWER_CTL, 1, 1),
SND_SOC_DAPM_PGA_E("Left PGA", CS42L51_POWER_CTL1, 3, 1, NULL, 0,
cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD),
SND_SOC_DAPM_PGA_E("Right PGA", CS42L51_POWER_CTL1, 4, 1, NULL, 0,
cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD),
SND_SOC_DAPM_ADC_E("Left ADC", "Left HiFi Capture",
CS42L51_POWER_CTL1, 1, 1,
cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD),
SND_SOC_DAPM_ADC_E("Right ADC", "Right HiFi Capture",
CS42L51_POWER_CTL1, 2, 1,
cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD),
SND_SOC_DAPM_DAC_E("Left DAC", "Left HiFi Playback",
CS42L51_POWER_CTL1, 5, 1,
cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD),
SND_SOC_DAPM_DAC_E("Right DAC", "Right HiFi Playback",
CS42L51_POWER_CTL1, 6, 1,
cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD),
/* analog/mic */
SND_SOC_DAPM_INPUT("AIN1L"),
SND_SOC_DAPM_INPUT("AIN1R"),
SND_SOC_DAPM_INPUT("AIN2L"),
SND_SOC_DAPM_INPUT("AIN2R"),
SND_SOC_DAPM_INPUT("MICL"),
SND_SOC_DAPM_INPUT("MICR"),
SND_SOC_DAPM_MIXER("Mic Preamp Left",
CS42L51_MIC_POWER_CTL, 2, 1, NULL, 0),
SND_SOC_DAPM_MIXER("Mic Preamp Right",
CS42L51_MIC_POWER_CTL, 3, 1, NULL, 0),
/* HP */
SND_SOC_DAPM_OUTPUT("HPL"),
SND_SOC_DAPM_OUTPUT("HPR"),
/* mux */
SND_SOC_DAPM_MUX("DAC Mux", SND_SOC_NOPM, 0, 0,
&cs42l51_dac_mux_controls),
SND_SOC_DAPM_MUX("PGA-ADC Mux Left", SND_SOC_NOPM, 0, 0,
&cs42l51_adcl_mux_controls),
SND_SOC_DAPM_MUX("PGA-ADC Mux Right", SND_SOC_NOPM, 0, 0,
&cs42l51_adcr_mux_controls),
};
static const struct snd_soc_dapm_route cs42l51_routes[] = {
{"HPL", NULL, "Left DAC"},
{"HPR", NULL, "Right DAC"},
{"Left ADC", NULL, "Left PGA"},
{"Right ADC", NULL, "Right PGA"},
{"Mic Preamp Left", NULL, "MICL"},
{"Mic Preamp Right", NULL, "MICR"},
{"PGA-ADC Mux Left", "AIN1 Left", "AIN1L" },
{"PGA-ADC Mux Left", "AIN2 Left", "AIN2L" },
{"PGA-ADC Mux Left", "MIC Left", "MICL" },
{"PGA-ADC Mux Left", "MIC+preamp Left", "Mic Preamp Left" },
{"PGA-ADC Mux Right", "AIN1 Right", "AIN1R" },
{"PGA-ADC Mux Right", "AIN2 Right", "AIN2R" },
{"PGA-ADC Mux Right", "MIC Right", "MICR" },
{"PGA-ADC Mux Right", "MIC+preamp Right", "Mic Preamp Right" },
{"Left PGA", NULL, "PGA-ADC Mux Left"},
{"Right PGA", NULL, "PGA-ADC Mux Right"},
};
static int cs42l51_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int format)
{
struct snd_soc_codec *codec = codec_dai->codec;
struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec);
int ret = 0;
switch (format & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
case SND_SOC_DAIFMT_LEFT_J:
case SND_SOC_DAIFMT_RIGHT_J:
cs42l51->audio_mode = format & SND_SOC_DAIFMT_FORMAT_MASK;
break;
default:
dev_err(codec->dev, "invalid DAI format\n");
ret = -EINVAL;
}
switch (format & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
cs42l51->func = MODE_MASTER;
break;
case SND_SOC_DAIFMT_CBS_CFS:
cs42l51->func = MODE_SLAVE_AUTO;
break;
default:
ret = -EINVAL;
break;
}
return ret;
}
struct cs42l51_ratios {
unsigned int ratio;
unsigned char speed_mode;
unsigned char mclk;
};
static struct cs42l51_ratios slave_ratios[] = {
{ 512, CS42L51_QSM_MODE, 0 }, { 768, CS42L51_QSM_MODE, 0 },
{ 1024, CS42L51_QSM_MODE, 0 }, { 1536, CS42L51_QSM_MODE, 0 },
{ 2048, CS42L51_QSM_MODE, 0 }, { 3072, CS42L51_QSM_MODE, 0 },
{ 256, CS42L51_HSM_MODE, 0 }, { 384, CS42L51_HSM_MODE, 0 },
{ 512, CS42L51_HSM_MODE, 0 }, { 768, CS42L51_HSM_MODE, 0 },
{ 1024, CS42L51_HSM_MODE, 0 }, { 1536, CS42L51_HSM_MODE, 0 },
{ 128, CS42L51_SSM_MODE, 0 }, { 192, CS42L51_SSM_MODE, 0 },
{ 256, CS42L51_SSM_MODE, 0 }, { 384, CS42L51_SSM_MODE, 0 },
{ 512, CS42L51_SSM_MODE, 0 }, { 768, CS42L51_SSM_MODE, 0 },
{ 128, CS42L51_DSM_MODE, 0 }, { 192, CS42L51_DSM_MODE, 0 },
{ 256, CS42L51_DSM_MODE, 0 }, { 384, CS42L51_DSM_MODE, 0 },
};
static struct cs42l51_ratios slave_auto_ratios[] = {
{ 1024, CS42L51_QSM_MODE, 0 }, { 1536, CS42L51_QSM_MODE, 0 },
{ 2048, CS42L51_QSM_MODE, 1 }, { 3072, CS42L51_QSM_MODE, 1 },
{ 512, CS42L51_HSM_MODE, 0 }, { 768, CS42L51_HSM_MODE, 0 },
{ 1024, CS42L51_HSM_MODE, 1 }, { 1536, CS42L51_HSM_MODE, 1 },
{ 256, CS42L51_SSM_MODE, 0 }, { 384, CS42L51_SSM_MODE, 0 },
{ 512, CS42L51_SSM_MODE, 1 }, { 768, CS42L51_SSM_MODE, 1 },
{ 128, CS42L51_DSM_MODE, 0 }, { 192, CS42L51_DSM_MODE, 0 },
{ 256, CS42L51_DSM_MODE, 1 }, { 384, CS42L51_DSM_MODE, 1 },
};
static int cs42l51_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 cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec);
struct cs42l51_ratios *ratios = NULL;
int nr_ratios = 0;
unsigned int rates = 0;
unsigned int rate_min = -1;
unsigned int rate_max = 0;
int i;
cs42l51->mclk = freq;
switch (cs42l51->func) {
case MODE_MASTER:
return -EINVAL;
case MODE_SLAVE:
ratios = slave_ratios;
nr_ratios = ARRAY_SIZE(slave_ratios);
break;
case MODE_SLAVE_AUTO:
ratios = slave_auto_ratios;
nr_ratios = ARRAY_SIZE(slave_auto_ratios);
break;
}
for (i = 0; i < nr_ratios; i++) {
unsigned int rate = freq / ratios[i].ratio;
rates |= snd_pcm_rate_to_rate_bit(rate);
if (rate < rate_min)
rate_min = rate;
if (rate > rate_max)
rate_max = rate;
}
rates &= ~SNDRV_PCM_RATE_KNOT;
if (!rates) {
dev_err(codec->dev, "could not find a valid sample rate\n");
return -EINVAL;
}
codec_dai->playback.rates = rates;
codec_dai->playback.rate_min = rate_min;
codec_dai->playback.rate_max = rate_max;
codec_dai->capture.rates = rates;
codec_dai->capture.rate_min = rate_min;
codec_dai->capture.rate_max = rate_max;
return 0;
}
static int cs42l51_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->card->codec;
struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec);
int ret;
unsigned int i;
unsigned int rate;
unsigned int ratio;
struct cs42l51_ratios *ratios = NULL;
int nr_ratios = 0;
int intf_ctl, power_ctl, fmt;
switch (cs42l51->func) {
case MODE_MASTER:
return -EINVAL;
case MODE_SLAVE:
ratios = slave_ratios;
nr_ratios = ARRAY_SIZE(slave_ratios);
break;
case MODE_SLAVE_AUTO:
ratios = slave_auto_ratios;
nr_ratios = ARRAY_SIZE(slave_auto_ratios);
break;
}
/* Figure out which MCLK/LRCK ratio to use */
rate = params_rate(params); /* Sampling rate, in Hz */
ratio = cs42l51->mclk / rate; /* MCLK/LRCK ratio */
for (i = 0; i < nr_ratios; i++) {
if (ratios[i].ratio == ratio)
break;
}
if (i == nr_ratios) {
/* We did not find a matching ratio */
dev_err(codec->dev, "could not find matching ratio\n");
return -EINVAL;
}
intf_ctl = snd_soc_read(codec, CS42L51_INTF_CTL);
power_ctl = snd_soc_read(codec, CS42L51_MIC_POWER_CTL);
intf_ctl &= ~(CS42L51_INTF_CTL_MASTER | CS42L51_INTF_CTL_ADC_I2S
| CS42L51_INTF_CTL_DAC_FORMAT(7));
power_ctl &= ~(CS42L51_MIC_POWER_CTL_SPEED(3)
| CS42L51_MIC_POWER_CTL_MCLK_DIV2);
switch (cs42l51->func) {
case MODE_MASTER:
intf_ctl |= CS42L51_INTF_CTL_MASTER;
power_ctl |= CS42L51_MIC_POWER_CTL_SPEED(ratios[i].speed_mode);
break;
case MODE_SLAVE:
power_ctl |= CS42L51_MIC_POWER_CTL_SPEED(ratios[i].speed_mode);
break;
case MODE_SLAVE_AUTO:
power_ctl |= CS42L51_MIC_POWER_CTL_AUTO;
break;
}
switch (cs42l51->audio_mode) {
case SND_SOC_DAIFMT_I2S:
intf_ctl |= CS42L51_INTF_CTL_ADC_I2S;
intf_ctl |= CS42L51_INTF_CTL_DAC_FORMAT(CS42L51_DAC_DIF_I2S);
break;
case SND_SOC_DAIFMT_LEFT_J:
intf_ctl |= CS42L51_INTF_CTL_DAC_FORMAT(CS42L51_DAC_DIF_LJ24);
break;
case SND_SOC_DAIFMT_RIGHT_J:
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
case SNDRV_PCM_FORMAT_S16_BE:
fmt = CS42L51_DAC_DIF_RJ16;
break;
case SNDRV_PCM_FORMAT_S18_3LE:
case SNDRV_PCM_FORMAT_S18_3BE:
fmt = CS42L51_DAC_DIF_RJ18;
break;
case SNDRV_PCM_FORMAT_S20_3LE:
case SNDRV_PCM_FORMAT_S20_3BE:
fmt = CS42L51_DAC_DIF_RJ20;
break;
case SNDRV_PCM_FORMAT_S24_LE:
case SNDRV_PCM_FORMAT_S24_BE:
fmt = CS42L51_DAC_DIF_RJ24;
break;
default:
dev_err(codec->dev, "unknown format\n");
return -EINVAL;
}
intf_ctl |= CS42L51_INTF_CTL_DAC_FORMAT(fmt);
break;
default:
dev_err(codec->dev, "unknown format\n");
return -EINVAL;
}
if (ratios[i].mclk)
power_ctl |= CS42L51_MIC_POWER_CTL_MCLK_DIV2;
ret = snd_soc_write(codec, CS42L51_INTF_CTL, intf_ctl);
if (ret < 0)
return ret;
ret = snd_soc_write(codec, CS42L51_MIC_POWER_CTL, power_ctl);
if (ret < 0)
return ret;
return 0;
}
static int cs42l51_dai_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
int reg;
int mask = CS42L51_DAC_OUT_CTL_DACA_MUTE|CS42L51_DAC_OUT_CTL_DACB_MUTE;
reg = snd_soc_read(codec, CS42L51_DAC_OUT_CTL);
if (mute)
reg |= mask;
else
reg &= ~mask;
return snd_soc_write(codec, CS42L51_DAC_OUT_CTL, reg);
}
static struct snd_soc_dai_ops cs42l51_dai_ops = {
.hw_params = cs42l51_hw_params,
.set_sysclk = cs42l51_set_dai_sysclk,
.set_fmt = cs42l51_set_dai_fmt,
.digital_mute = cs42l51_dai_mute,
};
struct snd_soc_dai cs42l51_dai = {
.name = "CS42L51 HiFi",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_96000,
.formats = CS42L51_FORMATS,
},
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_96000,
.formats = CS42L51_FORMATS,
},
.ops = &cs42l51_dai_ops,
};
EXPORT_SYMBOL_GPL(cs42l51_dai);
static int cs42l51_probe(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec;
int ret = 0;
if (!cs42l51_codec) {
dev_err(&pdev->dev, "CS42L51 codec not yet registered\n");
return -EINVAL;
}
socdev->card->codec = cs42l51_codec;
codec = socdev->card->codec;
/* Register PCMs */
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0) {
dev_err(&pdev->dev, "failed to create PCMs\n");
return ret;
}
snd_soc_add_controls(codec, cs42l51_snd_controls,
ARRAY_SIZE(cs42l51_snd_controls));
snd_soc_dapm_new_controls(codec, cs42l51_dapm_widgets,
ARRAY_SIZE(cs42l51_dapm_widgets));
snd_soc_dapm_add_routes(codec, cs42l51_routes,
ARRAY_SIZE(cs42l51_routes));
return 0;
}
static int cs42l51_remove(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
return 0;
}
struct snd_soc_codec_device soc_codec_device_cs42l51 = {
.probe = cs42l51_probe,
.remove = cs42l51_remove
};
EXPORT_SYMBOL_GPL(soc_codec_device_cs42l51);
static int __init cs42l51_init(void)
{
int ret;
ret = i2c_add_driver(&cs42l51_i2c_driver);
if (ret != 0) {
printk(KERN_ERR "%s: can't add i2c driver\n", __func__);
return ret;
}
return 0;
}
module_init(cs42l51_init);
static void __exit cs42l51_exit(void)
{
i2c_del_driver(&cs42l51_i2c_driver);
}
module_exit(cs42l51_exit);
MODULE_AUTHOR("Arnaud Patard <[email protected]>");
MODULE_DESCRIPTION("Cirrus Logic CS42L51 ALSA SoC Codec Driver");
MODULE_LICENSE("GPL");

163
sound/soc/codecs/cs42l51.h Normal file
View file

@ -0,0 +1,163 @@
/*
* cs42l51.h
*
* ASoC Driver for Cirrus Logic CS42L51 codecs
*
* Copyright (c) 2010 Arnaud Patard <apatard@mandriva.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*/
#ifndef _CS42L51_H
#define _CS42L51_H
#define CS42L51_CHIP_ID 0x1B
#define CS42L51_CHIP_REV_A 0x00
#define CS42L51_CHIP_REV_B 0x01
#define CS42L51_CHIP_REV_ID 0x01
#define CS42L51_MK_CHIP_REV(a, b) ((a)<<3|(b))
#define CS42L51_POWER_CTL1 0x02
#define CS42L51_POWER_CTL1_PDN_DACB (1<<6)
#define CS42L51_POWER_CTL1_PDN_DACA (1<<5)
#define CS42L51_POWER_CTL1_PDN_PGAB (1<<4)
#define CS42L51_POWER_CTL1_PDN_PGAA (1<<3)
#define CS42L51_POWER_CTL1_PDN_ADCB (1<<2)
#define CS42L51_POWER_CTL1_PDN_ADCA (1<<1)
#define CS42L51_POWER_CTL1_PDN (1<<0)
#define CS42L51_MIC_POWER_CTL 0x03
#define CS42L51_MIC_POWER_CTL_AUTO (1<<7)
#define CS42L51_MIC_POWER_CTL_SPEED(x) (((x)&3)<<5)
#define CS42L51_QSM_MODE 3
#define CS42L51_HSM_MODE 2
#define CS42L51_SSM_MODE 1
#define CS42L51_DSM_MODE 0
#define CS42L51_MIC_POWER_CTL_3ST_SP (1<<4)
#define CS42L51_MIC_POWER_CTL_PDN_MICB (1<<3)
#define CS42L51_MIC_POWER_CTL_PDN_MICA (1<<2)
#define CS42L51_MIC_POWER_CTL_PDN_BIAS (1<<1)
#define CS42L51_MIC_POWER_CTL_MCLK_DIV2 (1<<0)
#define CS42L51_INTF_CTL 0x04
#define CS42L51_INTF_CTL_LOOPBACK (1<<7)
#define CS42L51_INTF_CTL_MASTER (1<<6)
#define CS42L51_INTF_CTL_DAC_FORMAT(x) (((x)&7)<<3)
#define CS42L51_DAC_DIF_LJ24 0x00
#define CS42L51_DAC_DIF_I2S 0x01
#define CS42L51_DAC_DIF_RJ24 0x02
#define CS42L51_DAC_DIF_RJ20 0x03
#define CS42L51_DAC_DIF_RJ18 0x04
#define CS42L51_DAC_DIF_RJ16 0x05
#define CS42L51_INTF_CTL_ADC_I2S (1<<2)
#define CS42L51_INTF_CTL_DIGMIX (1<<1)
#define CS42L51_INTF_CTL_MICMIX (1<<0)
#define CS42L51_MIC_CTL 0x05
#define CS42L51_MIC_CTL_ADC_SNGVOL (1<<7)
#define CS42L51_MIC_CTL_ADCD_DBOOST (1<<6)
#define CS42L51_MIC_CTL_ADCA_DBOOST (1<<5)
#define CS42L51_MIC_CTL_MICBIAS_SEL (1<<4)
#define CS42L51_MIC_CTL_MICBIAS_LVL(x) (((x)&3)<<2)
#define CS42L51_MIC_CTL_MICB_BOOST (1<<1)
#define CS42L51_MIC_CTL_MICA_BOOST (1<<0)
#define CS42L51_ADC_CTL 0x06
#define CS42L51_ADC_CTL_ADCB_HPFEN (1<<7)
#define CS42L51_ADC_CTL_ADCB_HPFRZ (1<<6)
#define CS42L51_ADC_CTL_ADCA_HPFEN (1<<5)
#define CS42L51_ADC_CTL_ADCA_HPFRZ (1<<4)
#define CS42L51_ADC_CTL_SOFTB (1<<3)
#define CS42L51_ADC_CTL_ZCROSSB (1<<2)
#define CS42L51_ADC_CTL_SOFTA (1<<1)
#define CS42L51_ADC_CTL_ZCROSSA (1<<0)
#define CS42L51_ADC_INPUT 0x07
#define CS42L51_ADC_INPUT_AINB_MUX(x) (((x)&3)<<6)
#define CS42L51_ADC_INPUT_AINA_MUX(x) (((x)&3)<<4)
#define CS42L51_ADC_INPUT_INV_ADCB (1<<3)
#define CS42L51_ADC_INPUT_INV_ADCA (1<<2)
#define CS42L51_ADC_INPUT_ADCB_MUTE (1<<1)
#define CS42L51_ADC_INPUT_ADCA_MUTE (1<<0)
#define CS42L51_DAC_OUT_CTL 0x08
#define CS42L51_DAC_OUT_CTL_HP_GAIN(x) (((x)&7)<<5)
#define CS42L51_DAC_OUT_CTL_DAC_SNGVOL (1<<4)
#define CS42L51_DAC_OUT_CTL_INV_PCMB (1<<3)
#define CS42L51_DAC_OUT_CTL_INV_PCMA (1<<2)
#define CS42L51_DAC_OUT_CTL_DACB_MUTE (1<<1)
#define CS42L51_DAC_OUT_CTL_DACA_MUTE (1<<0)
#define CS42L51_DAC_CTL 0x09
#define CS42L51_DAC_CTL_DATA_SEL(x) (((x)&3)<<6)
#define CS42L51_DAC_CTL_FREEZE (1<<5)
#define CS42L51_DAC_CTL_DEEMPH (1<<3)
#define CS42L51_DAC_CTL_AMUTE (1<<2)
#define CS42L51_DAC_CTL_DACSZ(x) (((x)&3)<<0)
#define CS42L51_ALC_PGA_CTL 0x0A
#define CS42L51_ALC_PGB_CTL 0x0B
#define CS42L51_ALC_PGX_ALCX_SRDIS (1<<7)
#define CS42L51_ALC_PGX_ALCX_ZCDIS (1<<6)
#define CS42L51_ALC_PGX_PGX_VOL(x) (((x)&0x1f)<<0)
#define CS42L51_ADCA_ATT 0x0C
#define CS42L51_ADCB_ATT 0x0D
#define CS42L51_ADCA_VOL 0x0E
#define CS42L51_ADCB_VOL 0x0F
#define CS42L51_PCMA_VOL 0x10
#define CS42L51_PCMB_VOL 0x11
#define CS42L51_MIX_MUTE_ADCMIX (1<<7)
#define CS42L51_MIX_VOLUME(x) (((x)&0x7f)<<0)
#define CS42L51_BEEP_FREQ 0x12
#define CS42L51_BEEP_VOL 0x13
#define CS42L51_BEEP_CONF 0x14
#define CS42L51_TONE_CTL 0x15
#define CS42L51_TONE_CTL_TREB(x) (((x)&0xf)<<4)
#define CS42L51_TONE_CTL_BASS(x) (((x)&0xf)<<0)
#define CS42L51_AOUTA_VOL 0x16
#define CS42L51_AOUTB_VOL 0x17
#define CS42L51_PCM_MIXER 0x18
#define CS42L51_LIMIT_THRES_DIS 0x19
#define CS42L51_LIMIT_REL 0x1A
#define CS42L51_LIMIT_ATT 0x1B
#define CS42L51_ALC_EN 0x1C
#define CS42L51_ALC_REL 0x1D
#define CS42L51_ALC_THRES 0x1E
#define CS42L51_NOISE_CONF 0x1F
#define CS42L51_STATUS 0x20
#define CS42L51_STATUS_SP_CLKERR (1<<6)
#define CS42L51_STATUS_SPEA_OVFL (1<<5)
#define CS42L51_STATUS_SPEB_OVFL (1<<4)
#define CS42L51_STATUS_PCMA_OVFL (1<<3)
#define CS42L51_STATUS_PCMB_OVFL (1<<2)
#define CS42L51_STATUS_ADCA_OVFL (1<<1)
#define CS42L51_STATUS_ADCB_OVFL (1<<0)
#define CS42L51_CHARGE_FREQ 0x21
#define CS42L51_FIRSTREG 0x01
/*
* Hack: with register 0x21, it makes 33 registers. Looks like someone in the
* i2c layer doesn't like i2c smbus block read of 33 regs. Workaround by using
* 32 regs
*/
#define CS42L51_LASTREG 0x20
#define CS42L51_NUMREGS (CS42L51_LASTREG - CS42L51_FIRSTREG + 1)
extern struct snd_soc_dai cs42l51_dai;
extern struct snd_soc_codec_device soc_codec_device_cs42l51;
#endif

48
sound/soc/codecs/da7210.c
View file

@ -15,23 +15,15 @@
* option) any later version.
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/kernel.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/tlv.h>
#include <sound/initval.h>
#include <asm/div64.h>
#include <sound/tlv.h>
#include "da7210.h"
@ -145,6 +137,29 @@
#define DA7210_VERSION "0.0.1"
/*
* Playback Volume
*
* max : 0x3F (+15.0 dB)
* (1.5 dB step)
* min : 0x11 (-54.0 dB)
* mute : 0x10
* reserved : 0x00 - 0x0F
*
* ** FIXME **
*
* Reserved area are considered as "mute".
* -> min = -79.5 dB
*/
static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1);
static const struct snd_kcontrol_new da7210_snd_controls[] = {
SOC_DOUBLE_R_TLV("HeadPhone Playback Volume",
DA7210_HP_L_VOL, DA7210_HP_R_VOL,
0, 0x3F, 0, hp_out_tlv),
};
/* Codec private data */
struct da7210_priv {
struct snd_soc_codec codec;
@ -227,10 +242,6 @@ static int da7210_startup(struct snd_pcm_substream *substream,
struct snd_soc_codec *codec = dai->codec;
if (is_play) {
/* PlayBack Volume 40 */
snd_soc_update_bits(codec, DA7210_HP_L_VOL, 0x3F, 40);
snd_soc_update_bits(codec, DA7210_HP_R_VOL, 0x3F, 40);
/* Enable Out */
snd_soc_update_bits(codec, DA7210_OUTMIX_L, 0x1F, 0x10);
snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10);
@ -488,7 +499,7 @@ static int da7210_init(struct da7210_priv *da7210)
ret = snd_soc_register_dai(&da7210_dai);
if (ret) {
dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
goto init_err;
goto codec_err;
}
/* FIXME
@ -574,6 +585,8 @@ static int da7210_init(struct da7210_priv *da7210)
return ret;
codec_err:
snd_soc_unregister_codec(codec);
init_err:
kfree(codec->reg_cache);
codec->reg_cache = NULL;
@ -601,8 +614,10 @@ static int __devinit da7210_i2c_probe(struct i2c_client *i2c,
codec->control_data = i2c;
ret = da7210_init(da7210);
if (ret < 0)
if (ret < 0) {
pr_err("Failed to initialise da7210 audio codec\n");
kfree(da7210);
}
return ret;
}
@ -656,6 +671,9 @@ static int da7210_probe(struct platform_device *pdev)
if (ret < 0)
goto pcm_err;
snd_soc_add_controls(da7210_codec, da7210_snd_controls,
ARRAY_SIZE(da7210_snd_controls));
dev_info(&pdev->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION);
pcm_err:

511
sound/soc/codecs/jz4740.c Normal file
View file

@ -0,0 +1,511 @@
/*
* Copyright (C) 2009-2010, Lars-Peter Clausen <lars@metafoo.de>
*
* 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.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 675 Mass Ave, Cambridge, MA 02139, USA.
*
*/
#include <linux/kernel.h>
#include <linux/module.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <linux/delay.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/initval.h>
#include <sound/soc-dapm.h>
#include <sound/soc.h>
#define JZ4740_REG_CODEC_1 0x0
#define JZ4740_REG_CODEC_2 0x1
#define JZ4740_CODEC_1_LINE_ENABLE BIT(29)
#define JZ4740_CODEC_1_MIC_ENABLE BIT(28)
#define JZ4740_CODEC_1_SW1_ENABLE BIT(27)
#define JZ4740_CODEC_1_ADC_ENABLE BIT(26)
#define JZ4740_CODEC_1_SW2_ENABLE BIT(25)
#define JZ4740_CODEC_1_DAC_ENABLE BIT(24)
#define JZ4740_CODEC_1_VREF_DISABLE BIT(20)
#define JZ4740_CODEC_1_VREF_AMP_DISABLE BIT(19)
#define JZ4740_CODEC_1_VREF_PULLDOWN BIT(18)
#define JZ4740_CODEC_1_VREF_LOW_CURRENT BIT(17)
#define JZ4740_CODEC_1_VREF_HIGH_CURRENT BIT(16)
#define JZ4740_CODEC_1_HEADPHONE_DISABLE BIT(14)
#define JZ4740_CODEC_1_HEADPHONE_AMP_CHANGE_ANY BIT(13)
#define JZ4740_CODEC_1_HEADPHONE_CHARGE BIT(12)
#define JZ4740_CODEC_1_HEADPHONE_PULLDOWN (BIT(11) | BIT(10))
#define JZ4740_CODEC_1_HEADPHONE_POWERDOWN_M BIT(9)
#define JZ4740_CODEC_1_HEADPHONE_POWERDOWN BIT(8)
#define JZ4740_CODEC_1_SUSPEND BIT(1)
#define JZ4740_CODEC_1_RESET BIT(0)
#define JZ4740_CODEC_1_LINE_ENABLE_OFFSET 29
#define JZ4740_CODEC_1_MIC_ENABLE_OFFSET 28
#define JZ4740_CODEC_1_SW1_ENABLE_OFFSET 27
#define JZ4740_CODEC_1_ADC_ENABLE_OFFSET 26
#define JZ4740_CODEC_1_SW2_ENABLE_OFFSET 25
#define JZ4740_CODEC_1_DAC_ENABLE_OFFSET 24
#define JZ4740_CODEC_1_HEADPHONE_DISABLE_OFFSET 14
#define JZ4740_CODEC_1_HEADPHONE_POWERDOWN_OFFSET 8
#define JZ4740_CODEC_2_INPUT_VOLUME_MASK 0x1f0000
#define JZ4740_CODEC_2_SAMPLE_RATE_MASK 0x000f00
#define JZ4740_CODEC_2_MIC_BOOST_GAIN_MASK 0x000030
#define JZ4740_CODEC_2_HEADPHONE_VOLUME_MASK 0x000003
#define JZ4740_CODEC_2_INPUT_VOLUME_OFFSET 16
#define JZ4740_CODEC_2_SAMPLE_RATE_OFFSET 8
#define JZ4740_CODEC_2_MIC_BOOST_GAIN_OFFSET 4
#define JZ4740_CODEC_2_HEADPHONE_VOLUME_OFFSET 0
static const uint32_t jz4740_codec_regs[] = {
0x021b2302, 0x00170803,
};
struct jz4740_codec {
void __iomem *base;
struct resource *mem;
uint32_t reg_cache[2];
struct snd_soc_codec codec;
};
static inline struct jz4740_codec *codec_to_jz4740(struct snd_soc_codec *codec)
{
return container_of(codec, struct jz4740_codec, codec);
}
static unsigned int jz4740_codec_read(struct snd_soc_codec *codec,
unsigned int reg)
{
struct jz4740_codec *jz4740_codec = codec_to_jz4740(codec);
return readl(jz4740_codec->base + (reg << 2));
}
static int jz4740_codec_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int val)
{
struct jz4740_codec *jz4740_codec = codec_to_jz4740(codec);
jz4740_codec->reg_cache[reg] = val;
writel(val, jz4740_codec->base + (reg << 2));
return 0;
}
static const struct snd_kcontrol_new jz4740_codec_controls[] = {
SOC_SINGLE("Master Playback Volume", JZ4740_REG_CODEC_2,
JZ4740_CODEC_2_HEADPHONE_VOLUME_OFFSET, 3, 0),
SOC_SINGLE("Master Capture Volume", JZ4740_REG_CODEC_2,
JZ4740_CODEC_2_INPUT_VOLUME_OFFSET, 31, 0),
SOC_SINGLE("Master Playback Switch", JZ4740_REG_CODEC_1,
JZ4740_CODEC_1_HEADPHONE_DISABLE_OFFSET, 1, 1),
SOC_SINGLE("Mic Capture Volume", JZ4740_REG_CODEC_2,
JZ4740_CODEC_2_MIC_BOOST_GAIN_OFFSET, 3, 0),
};
static const struct snd_kcontrol_new jz4740_codec_output_controls[] = {
SOC_DAPM_SINGLE("Bypass Switch", JZ4740_REG_CODEC_1,
JZ4740_CODEC_1_SW1_ENABLE_OFFSET, 1, 0),
SOC_DAPM_SINGLE("DAC Switch", JZ4740_REG_CODEC_1,
JZ4740_CODEC_1_SW2_ENABLE_OFFSET, 1, 0),
};
static const struct snd_kcontrol_new jz4740_codec_input_controls[] = {
SOC_DAPM_SINGLE("Line Capture Switch", JZ4740_REG_CODEC_1,
JZ4740_CODEC_1_LINE_ENABLE_OFFSET, 1, 0),
SOC_DAPM_SINGLE("Mic Capture Switch", JZ4740_REG_CODEC_1,
JZ4740_CODEC_1_MIC_ENABLE_OFFSET, 1, 0),
};
static const struct snd_soc_dapm_widget jz4740_codec_dapm_widgets[] = {
SND_SOC_DAPM_ADC("ADC", "Capture", JZ4740_REG_CODEC_1,
JZ4740_CODEC_1_ADC_ENABLE_OFFSET, 0),
SND_SOC_DAPM_DAC("DAC", "Playback", JZ4740_REG_CODEC_1,
JZ4740_CODEC_1_DAC_ENABLE_OFFSET, 0),
SND_SOC_DAPM_MIXER("Output Mixer", JZ4740_REG_CODEC_1,
JZ4740_CODEC_1_HEADPHONE_POWERDOWN_OFFSET, 1,
jz4740_codec_output_controls,
ARRAY_SIZE(jz4740_codec_output_controls)),
SND_SOC_DAPM_MIXER_NAMED_CTL("Input Mixer", SND_SOC_NOPM, 0, 0,
jz4740_codec_input_controls,
ARRAY_SIZE(jz4740_codec_input_controls)),
SND_SOC_DAPM_MIXER("Line Input", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_OUTPUT("LOUT"),
SND_SOC_DAPM_OUTPUT("ROUT"),
SND_SOC_DAPM_INPUT("MIC"),
SND_SOC_DAPM_INPUT("LIN"),
SND_SOC_DAPM_INPUT("RIN"),
};
static const struct snd_soc_dapm_route jz4740_codec_dapm_routes[] = {
{"Line Input", NULL, "LIN"},
{"Line Input", NULL, "RIN"},
{"Input Mixer", "Line Capture Switch", "Line Input"},
{"Input Mixer", "Mic Capture Switch", "MIC"},
{"ADC", NULL, "Input Mixer"},
{"Output Mixer", "Bypass Switch", "Input Mixer"},
{"Output Mixer", "DAC Switch", "DAC"},
{"LOUT", NULL, "Output Mixer"},
{"ROUT", NULL, "Output Mixer"},
};
static int jz4740_codec_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
{
uint32_t val;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->card->codec;
switch (params_rate(params)) {
case 8000:
val = 0;
break;
case 11025:
val = 1;
break;
case 12000:
val = 2;
break;
case 16000:
val = 3;
break;
case 22050:
val = 4;
break;
case 24000:
val = 5;
break;
case 32000:
val = 6;
break;
case 44100:
val = 7;
break;
case 48000:
val = 8;
break;
default:
return -EINVAL;
}
val <<= JZ4740_CODEC_2_SAMPLE_RATE_OFFSET;
snd_soc_update_bits(codec, JZ4740_REG_CODEC_2,
JZ4740_CODEC_2_SAMPLE_RATE_MASK, val);
return 0;
}
static struct snd_soc_dai_ops jz4740_codec_dai_ops = {
.hw_params = jz4740_codec_hw_params,
};
struct snd_soc_dai jz4740_codec_dai = {
.name = "jz4740",
.playback = {
.stream_name = "Playback",
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8,
},
.capture = {
.stream_name = "Capture",
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8,
},
.ops = &jz4740_codec_dai_ops,
.symmetric_rates = 1,
};
EXPORT_SYMBOL_GPL(jz4740_codec_dai);
static void jz4740_codec_wakeup(struct snd_soc_codec *codec)
{
int i;
uint32_t *cache = codec->reg_cache;
snd_soc_update_bits(codec, JZ4740_REG_CODEC_1,
JZ4740_CODEC_1_RESET, JZ4740_CODEC_1_RESET);
udelay(2);
snd_soc_update_bits(codec, JZ4740_REG_CODEC_1,
JZ4740_CODEC_1_SUSPEND | JZ4740_CODEC_1_RESET, 0);
for (i = 0; i < ARRAY_SIZE(jz4740_codec_regs); ++i)
jz4740_codec_write(codec, i, cache[i]);
}
static int jz4740_codec_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
unsigned int mask;
unsigned int value;
switch (level) {
case SND_SOC_BIAS_ON:
break;
case SND_SOC_BIAS_PREPARE:
mask = JZ4740_CODEC_1_VREF_DISABLE |
JZ4740_CODEC_1_VREF_AMP_DISABLE |
JZ4740_CODEC_1_HEADPHONE_POWERDOWN_M;
value = 0;
snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, mask, value);
break;
case SND_SOC_BIAS_STANDBY:
/* The only way to clear the suspend flag is to reset the codec */
if (codec->bias_level == SND_SOC_BIAS_OFF)
jz4740_codec_wakeup(codec);
mask = JZ4740_CODEC_1_VREF_DISABLE |
JZ4740_CODEC_1_VREF_AMP_DISABLE |
JZ4740_CODEC_1_HEADPHONE_POWERDOWN_M;
value = JZ4740_CODEC_1_VREF_DISABLE |
JZ4740_CODEC_1_VREF_AMP_DISABLE |
JZ4740_CODEC_1_HEADPHONE_POWERDOWN_M;
snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, mask, value);
break;
case SND_SOC_BIAS_OFF:
mask = JZ4740_CODEC_1_SUSPEND;
value = JZ4740_CODEC_1_SUSPEND;
snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, mask, value);
break;
default:
break;
}
codec->bias_level = level;
return 0;
}
static struct snd_soc_codec *jz4740_codec_codec;
static int jz4740_codec_dev_probe(struct platform_device *pdev)
{
int ret;
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = jz4740_codec_codec;
BUG_ON(!codec);
socdev->card->codec = codec;
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret) {
dev_err(&pdev->dev, "Failed to create pcms: %d\n", ret);
return ret;
}
snd_soc_add_controls(codec, jz4740_codec_controls,
ARRAY_SIZE(jz4740_codec_controls));
snd_soc_dapm_new_controls(codec, jz4740_codec_dapm_widgets,
ARRAY_SIZE(jz4740_codec_dapm_widgets));
snd_soc_dapm_add_routes(codec, jz4740_codec_dapm_routes,
ARRAY_SIZE(jz4740_codec_dapm_routes));
snd_soc_dapm_new_widgets(codec);
return 0;
}
static int jz4740_codec_dev_remove(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
return 0;
}
#ifdef CONFIG_PM_SLEEP
static int jz4740_codec_suspend(struct platform_device *pdev, pm_message_t state)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->card->codec;
return jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_OFF);
}
static int jz4740_codec_resume(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->card->codec;
return jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
}
#else
#define jz4740_codec_suspend NULL
#define jz4740_codec_resume NULL
#endif
struct snd_soc_codec_device soc_codec_dev_jz4740_codec = {
.probe = jz4740_codec_dev_probe,
.remove = jz4740_codec_dev_remove,
.suspend = jz4740_codec_suspend,
.resume = jz4740_codec_resume,
};
EXPORT_SYMBOL_GPL(soc_codec_dev_jz4740_codec);
static int __devinit jz4740_codec_probe(struct platform_device *pdev)
{
int ret;
struct jz4740_codec *jz4740_codec;
struct snd_soc_codec *codec;
struct resource *mem;
jz4740_codec = kzalloc(sizeof(*jz4740_codec), GFP_KERNEL);
if (!jz4740_codec)
return -ENOMEM;
mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (!mem) {
dev_err(&pdev->dev, "Failed to get mmio memory resource\n");
ret = -ENOENT;
goto err_free_codec;
}
mem = request_mem_region(mem->start, resource_size(mem), pdev->name);
if (!mem) {
dev_err(&pdev->dev, "Failed to request mmio memory region\n");
ret = -EBUSY;
goto err_free_codec;
}
jz4740_codec->base = ioremap(mem->start, resource_size(mem));
if (!jz4740_codec->base) {
dev_err(&pdev->dev, "Failed to ioremap mmio memory\n");
ret = -EBUSY;
goto err_release_mem_region;
}
jz4740_codec->mem = mem;
jz4740_codec_dai.dev = &pdev->dev;
codec = &jz4740_codec->codec;
codec->dev = &pdev->dev;
codec->name = "jz4740";
codec->owner = THIS_MODULE;
codec->read = jz4740_codec_read;
codec->write = jz4740_codec_write;
codec->set_bias_level = jz4740_codec_set_bias_level;
codec->bias_level = SND_SOC_BIAS_OFF;
codec->dai = &jz4740_codec_dai;
codec->num_dai = 1;
codec->reg_cache = jz4740_codec->reg_cache;
codec->reg_cache_size = 2;
memcpy(codec->reg_cache, jz4740_codec_regs, sizeof(jz4740_codec_regs));
mutex_init(&codec->mutex);
INIT_LIST_HEAD(&codec->dapm_widgets);
INIT_LIST_HEAD(&codec->dapm_paths);
jz4740_codec_codec = codec;
snd_soc_update_bits(codec, JZ4740_REG_CODEC_1,
JZ4740_CODEC_1_SW2_ENABLE, JZ4740_CODEC_1_SW2_ENABLE);
platform_set_drvdata(pdev, jz4740_codec);
ret = snd_soc_register_codec(codec);
if (ret) {
dev_err(&pdev->dev, "Failed to register codec\n");
goto err_iounmap;
}
ret = snd_soc_register_dai(&jz4740_codec_dai);
if (ret) {
dev_err(&pdev->dev, "Failed to register codec dai\n");
goto err_unregister_codec;
}
jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return 0;
err_unregister_codec:
snd_soc_unregister_codec(codec);
err_iounmap:
iounmap(jz4740_codec->base);
err_release_mem_region:
release_mem_region(mem->start, resource_size(mem));
err_free_codec:
kfree(jz4740_codec);
return ret;
}
static int __devexit jz4740_codec_remove(struct platform_device *pdev)
{
struct jz4740_codec *jz4740_codec = platform_get_drvdata(pdev);
struct resource *mem = jz4740_codec->mem;
snd_soc_unregister_dai(&jz4740_codec_dai);
snd_soc_unregister_codec(&jz4740_codec->codec);
iounmap(jz4740_codec->base);
release_mem_region(mem->start, resource_size(mem));
platform_set_drvdata(pdev, NULL);
kfree(jz4740_codec);
return 0;
}
static struct platform_driver jz4740_codec_driver = {
.probe = jz4740_codec_probe,
.remove = __devexit_p(jz4740_codec_remove),
.driver = {
.name = "jz4740-codec",
.owner = THIS_MODULE,
},
};
static int __init jz4740_codec_init(void)
{
return platform_driver_register(&jz4740_codec_driver);
}
module_init(jz4740_codec_init);
static void __exit jz4740_codec_exit(void)
{
platform_driver_unregister(&jz4740_codec_driver);
}
module_exit(jz4740_codec_exit);
MODULE_DESCRIPTION("JZ4740 SoC internal codec driver");
MODULE_AUTHOR("Lars-Peter Clausen <[email protected]>");
MODULE_LICENSE("GPL v2");
MODULE_ALIAS("platform:jz4740-codec");

20
sound/soc/codecs/jz4740.h Normal file
View file

@ -0,0 +1,20 @@
/*
* Copyright (C) 2009, Lars-Peter Clausen <lars@metafoo.de>
*
* 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.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 675 Mass Ave, Cambridge, MA 02139, USA.
*
*/
#ifndef __SND_SOC_CODECS_JZ4740_CODEC_H__
#define __SND_SOC_CODECS_JZ4740_CODEC_H__
extern struct snd_soc_dai jz4740_codec_dai;
extern struct snd_soc_codec_device soc_codec_dev_jz4740_codec;
#endif

94
sound/soc/codecs/spdif_transciever.c
View file

@ -16,8 +16,10 @@
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/slab.h>
#include <sound/soc.h>
#include <sound/pcm.h>
#include <sound/initval.h>
#include "spdif_transciever.h"
@ -26,6 +28,48 @@ MODULE_LICENSE("GPL");
#define STUB_RATES SNDRV_PCM_RATE_8000_96000
#define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE
static struct snd_soc_codec *spdif_dit_codec;
static int spdif_dit_codec_probe(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec;
int ret;
if (spdif_dit_codec == NULL) {
dev_err(&pdev->dev, "Codec device not registered\n");
return -ENODEV;
}
socdev->card->codec = spdif_dit_codec;
codec = spdif_dit_codec;
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0) {
dev_err(codec->dev, "failed to create pcms: %d\n", ret);
goto err_create_pcms;
}
return 0;
err_create_pcms:
return ret;
}
static int spdif_dit_codec_remove(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
snd_soc_free_pcms(socdev);
return 0;
}
struct snd_soc_codec_device soc_codec_dev_spdif_dit = {
.probe = spdif_dit_codec_probe,
.remove = spdif_dit_codec_remove,
}; EXPORT_SYMBOL_GPL(soc_codec_dev_spdif_dit);
struct snd_soc_dai dit_stub_dai = {
.name = "DIT",
.playback = {
@ -40,13 +84,61 @@ EXPORT_SYMBOL_GPL(dit_stub_dai);
static int spdif_dit_probe(struct platform_device *pdev)
{
struct snd_soc_codec *codec;
int ret;
if (spdif_dit_codec) {
dev_err(&pdev->dev, "Another Codec is registered\n");
ret = -EINVAL;
goto err_reg_codec;
}
codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
if (codec == NULL)
return -ENOMEM;
codec->dev = &pdev->dev;
mutex_init(&codec->mutex);
INIT_LIST_HEAD(&codec->dapm_widgets);
INIT_LIST_HEAD(&codec->dapm_paths);
codec->name = "spdif-dit";
codec->owner = THIS_MODULE;
codec->dai = &dit_stub_dai;
codec->num_dai = 1;
spdif_dit_codec = codec;
ret = snd_soc_register_codec(codec);
if (ret < 0) {
dev_err(codec->dev, "Failed to register codec: %d\n", ret);
goto err_reg_codec;
}
dit_stub_dai.dev = &pdev->dev;
return snd_soc_register_dai(&dit_stub_dai);
ret = snd_soc_register_dai(&dit_stub_dai);
if (ret < 0) {
dev_err(codec->dev, "Failed to register dai: %d\n", ret);
goto err_reg_dai;
}
return 0;
err_reg_dai:
snd_soc_unregister_codec(codec);
err_reg_codec:
kfree(spdif_dit_codec);
return ret;
}
static int spdif_dit_remove(struct platform_device *pdev)
{
snd_soc_unregister_dai(&dit_stub_dai);
snd_soc_unregister_codec(spdif_dit_codec);
kfree(spdif_dit_codec);
spdif_dit_codec = NULL;
return 0;
}

1
sound/soc/codecs/spdif_transciever.h
View file

@ -12,6 +12,7 @@
#ifndef CODEC_STUBS_H
#define CODEC_STUBS_H
extern struct snd_soc_codec_device soc_codec_dev_spdif_dit;
extern struct snd_soc_dai dit_stub_dai;
#endif /* CODEC_STUBS_H */

7
sound/soc/codecs/tlv320aic23.c
View file

@ -560,13 +560,16 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
switch (level) {
case SND_SOC_BIAS_ON:
/* vref/mid, osc on, dac unmute */
reg &= ~(TLV320AIC23_DEVICE_PWR_OFF | TLV320AIC23_OSC_OFF | \
TLV320AIC23_DAC_OFF);
tlv320aic23_write(codec, TLV320AIC23_PWR, reg);
break;
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
/* everything off except vref/vmid, */
tlv320aic23_write(codec, TLV320AIC23_PWR, reg | 0x0040);
tlv320aic23_write(codec, TLV320AIC23_PWR, reg | \
TLV320AIC23_CLK_OFF);
break;
case SND_SOC_BIAS_OFF:
/* everything off, dac mute, inactive */
@ -615,7 +618,6 @@ static int tlv320aic23_suspend(struct platform_device *pdev,
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->card->codec;
tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
@ -632,7 +634,6 @@ static int tlv320aic23_resume(struct platform_device *pdev)
u16 val = tlv320aic23_read_reg_cache(codec, reg);
tlv320aic23_write(codec, reg, val);
}
tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return 0;

180
sound/soc/codecs/tlv320dac33.c
View file

@ -49,8 +49,6 @@
#define NSAMPLE_MAX 5700
#define LATENCY_TIME_MS 20
#define MODE7_LTHR 10
#define MODE7_UTHR (DAC33_BUFFER_SIZE_SAMPLES - 10)
@ -62,6 +60,9 @@
#define US_TO_SAMPLES(rate, us) \
(rate / (1000000 / us))
#define UTHR_FROM_PERIOD_SIZE(samples, playrate, burstrate) \
((samples * 5000) / ((burstrate * 5000) / (burstrate - playrate)))
static void dac33_calculate_times(struct snd_pcm_substream *substream);
static int dac33_prepare_chip(struct snd_pcm_substream *substream);
@ -107,6 +108,10 @@ struct tlv320dac33_priv {
* this */
enum dac33_fifo_modes fifo_mode;/* FIFO mode selection */
unsigned int nsample; /* burst read amount from host */
int mode1_latency; /* latency caused by the i2c writes in
* us */
int auto_fifo_config; /* Configure the FIFO based on the
* period size */
u8 burst_bclkdiv; /* BCLK divider value in burst mode */
unsigned int burst_rate; /* Interface speed in Burst modes */
@ -120,6 +125,8 @@ struct tlv320dac33_priv {
* samples */
unsigned int mode7_us_to_lthr; /* Time to reach lthr from uthr */
unsigned int uthr;
enum dac33_state state;
};
@ -442,6 +449,39 @@ static int dac33_set_nsample(struct snd_kcontrol *kcontrol,
return ret;
}
static int dac33_get_uthr(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
ucontrol->value.integer.value[0] = dac33->uthr;
return 0;
}
static int dac33_set_uthr(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
int ret = 0;
if (dac33->substream)
return -EBUSY;
if (dac33->uthr == ucontrol->value.integer.value[0])
return 0;
if (ucontrol->value.integer.value[0] < (MODE7_LTHR + 10) ||
ucontrol->value.integer.value[0] > MODE7_UTHR)
ret = -EINVAL;
else
dac33->uthr = ucontrol->value.integer.value[0];
return ret;
}
static int dac33_get_fifo_mode(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
@ -503,13 +543,18 @@ static const struct snd_kcontrol_new dac33_snd_controls[] = {
DAC33_LINEL_TO_LLO_VOL, DAC33_LINER_TO_RLO_VOL, 0, 127, 1),
};
static const struct snd_kcontrol_new dac33_nsample_snd_controls[] = {
SOC_SINGLE_EXT("nSample", 0, 0, 5900, 0,
dac33_get_nsample, dac33_set_nsample),
static const struct snd_kcontrol_new dac33_mode_snd_controls[] = {
SOC_ENUM_EXT("FIFO Mode", dac33_fifo_mode_enum,
dac33_get_fifo_mode, dac33_set_fifo_mode),
};
static const struct snd_kcontrol_new dac33_fifo_snd_controls[] = {
SOC_SINGLE_EXT("nSample", 0, 0, 5900, 0,
dac33_get_nsample, dac33_set_nsample),
SOC_SINGLE_EXT("UTHR", 0, 0, MODE7_UTHR, 0,
dac33_get_uthr, dac33_set_uthr),
};
/* Analog bypass */
static const struct snd_kcontrol_new dac33_dapm_abypassl_control =
SOC_DAPM_SINGLE("Switch", DAC33_LINEL_TO_LLO_VOL, 7, 1, 1);
@ -612,7 +657,7 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33)
switch (dac33->fifo_mode) {
case DAC33_FIFO_MODE1:
dac33_write16(codec, DAC33_NSAMPLE_MSB,
DAC33_THRREG(dac33->nsample + dac33->alarm_threshold));
DAC33_THRREG(dac33->nsample));
/* Take the timestamps */
spin_lock_irq(&dac33->lock);
@ -761,6 +806,10 @@ static void dac33_shutdown(struct snd_pcm_substream *substream,
struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
dac33->substream = NULL;
/* Reset the nSample restrictions */
dac33->nsample_min = 0;
dac33->nsample_max = NSAMPLE_MAX;
}
static int dac33_hw_params(struct snd_pcm_substream *substream,
@ -985,7 +1034,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
* Configure the threshold levels, and leave 10 sample space
* at the bottom, and also at the top of the FIFO
*/
dac33_write16(codec, DAC33_UTHR_MSB, DAC33_THRREG(MODE7_UTHR));
dac33_write16(codec, DAC33_UTHR_MSB, DAC33_THRREG(dac33->uthr));
dac33_write16(codec, DAC33_LTHR_MSB, DAC33_THRREG(MODE7_LTHR));
break;
default:
@ -1003,57 +1052,71 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream)
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->card->codec;
struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
unsigned int period_size = substream->runtime->period_size;
unsigned int rate = substream->runtime->rate;
unsigned int nsample_limit;
/* In bypass mode we don't need to calculate */
if (!dac33->fifo_mode)
return;
/* Number of samples (16bit, stereo) in one period */
dac33->nsample_min = snd_pcm_lib_period_bytes(substream) / 4;
/* Number of samples (16bit, stereo) in ALSA buffer */
dac33->nsample_max = snd_pcm_lib_buffer_bytes(substream) / 4;
/* Subtract one period from the total */
dac33->nsample_max -= dac33->nsample_min;
/* Number of samples for LATENCY_TIME_MS / 2 */
dac33->alarm_threshold = substream->runtime->rate /
(1000 / (LATENCY_TIME_MS / 2));
/* Find and fix up the lowest nsmaple limit */
nsample_limit = substream->runtime->rate / (1000 / LATENCY_TIME_MS);
if (dac33->nsample_min < nsample_limit)
dac33->nsample_min = nsample_limit;
if (dac33->nsample < dac33->nsample_min)
dac33->nsample = dac33->nsample_min;
/*
* Find and fix up the highest nsmaple limit
* In order to not overflow the DAC33 buffer substract the
* alarm_threshold value from the size of the DAC33 buffer
*/
nsample_limit = DAC33_BUFFER_SIZE_SAMPLES - dac33->alarm_threshold;
if (dac33->nsample_max > nsample_limit)
dac33->nsample_max = nsample_limit;
if (dac33->nsample > dac33->nsample_max)
dac33->nsample = dac33->nsample_max;
switch (dac33->fifo_mode) {
case DAC33_FIFO_MODE1:
/* Number of samples under i2c latency */
dac33->alarm_threshold = US_TO_SAMPLES(rate,
dac33->mode1_latency);
if (dac33->auto_fifo_config) {
if (period_size <= dac33->alarm_threshold)
/*
* Configure nSamaple to number of periods,
* which covers the latency requironment.
*/
dac33->nsample = period_size *
((dac33->alarm_threshold / period_size) +
(dac33->alarm_threshold % period_size ?
1 : 0));
else
dac33->nsample = period_size;
} else {
/* nSample time shall not be shorter than i2c latency */
dac33->nsample_min = dac33->alarm_threshold;
/*
* nSample should not be bigger than alsa buffer minus
* size of one period to avoid overruns
*/
dac33->nsample_max = substream->runtime->buffer_size -
period_size;
nsample_limit = DAC33_BUFFER_SIZE_SAMPLES -
dac33->alarm_threshold;
if (dac33->nsample_max > nsample_limit)
dac33->nsample_max = nsample_limit;
/* Correct the nSample if it is outside of the ranges */
if (dac33->nsample < dac33->nsample_min)
dac33->nsample = dac33->nsample_min;
if (dac33->nsample > dac33->nsample_max)
dac33->nsample = dac33->nsample_max;
}
dac33->mode1_us_burst = SAMPLES_TO_US(dac33->burst_rate,
dac33->nsample);
dac33->t_stamp1 = 0;
dac33->t_stamp2 = 0;
break;
case DAC33_FIFO_MODE7:
if (dac33->auto_fifo_config) {
dac33->uthr = UTHR_FROM_PERIOD_SIZE(
period_size,
rate,
dac33->burst_rate) + 9;
if (dac33->uthr > MODE7_UTHR)
dac33->uthr = MODE7_UTHR;
if (dac33->uthr < (MODE7_LTHR + 10))
dac33->uthr = (MODE7_LTHR + 10);
}
dac33->mode7_us_to_lthr =
SAMPLES_TO_US(substream->runtime->rate,
MODE7_UTHR - MODE7_LTHR + 1);
SAMPLES_TO_US(substream->runtime->rate,
dac33->uthr - MODE7_LTHR + 1);
dac33->t_stamp1 = 0;
break;
default:
@ -1104,7 +1167,7 @@ static snd_pcm_sframes_t dac33_dai_delay(
struct snd_soc_codec *codec = socdev->card->codec;
struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
unsigned long long t0, t1, t_now;
unsigned int time_delta;
unsigned int time_delta, uthr;
int samples_out, samples_in, samples;
snd_pcm_sframes_t delay = 0;
@ -1182,6 +1245,7 @@ static snd_pcm_sframes_t dac33_dai_delay(
case DAC33_FIFO_MODE7:
spin_lock(&dac33->lock);
t0 = dac33->t_stamp1;
uthr = dac33->uthr;
spin_unlock(&dac33->lock);
t_now = ktime_to_us(ktime_get());
@ -1194,7 +1258,7 @@ static snd_pcm_sframes_t dac33_dai_delay(
* Either the timestamps are messed or equal. Report
* maximum delay
*/
delay = MODE7_UTHR;
delay = uthr;
goto out;
}
@ -1208,8 +1272,8 @@ static snd_pcm_sframes_t dac33_dai_delay(
substream->runtime->rate,
time_delta);
if (likely(MODE7_UTHR > samples_out))
delay = MODE7_UTHR - samples_out;
if (likely(uthr > samples_out))
delay = uthr - samples_out;
else
delay = 0;
} else {
@ -1227,8 +1291,8 @@ static snd_pcm_sframes_t dac33_dai_delay(
time_delta);
delay = MODE7_LTHR + samples_in - samples_out;
if (unlikely(delay > MODE7_UTHR))
delay = MODE7_UTHR;
if (unlikely(delay > uthr))
delay = uthr;
}
break;
default:
@ -1347,10 +1411,15 @@ static int dac33_soc_probe(struct platform_device *pdev)
snd_soc_add_controls(codec, dac33_snd_controls,
ARRAY_SIZE(dac33_snd_controls));
/* Only add the nSample controls, if we have valid IRQ number */
if (dac33->irq >= 0)
snd_soc_add_controls(codec, dac33_nsample_snd_controls,
ARRAY_SIZE(dac33_nsample_snd_controls));
/* Only add the FIFO controls, if we have valid IRQ number */
if (dac33->irq >= 0) {
snd_soc_add_controls(codec, dac33_mode_snd_controls,
ARRAY_SIZE(dac33_mode_snd_controls));
/* FIFO usage controls only, if autoio config is not selected */
if (!dac33->auto_fifo_config)
snd_soc_add_controls(codec, dac33_fifo_snd_controls,
ARRAY_SIZE(dac33_fifo_snd_controls));
}
dac33_add_widgets(codec);
@ -1481,9 +1550,14 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client,
/* Pre calculate the burst rate */
dac33->burst_rate = BURST_BASEFREQ_HZ / dac33->burst_bclkdiv / 32;
dac33->keep_bclk = pdata->keep_bclk;
dac33->auto_fifo_config = pdata->auto_fifo_config;
dac33->mode1_latency = pdata->mode1_latency;
if (!dac33->mode1_latency)
dac33->mode1_latency = 10000; /* 10ms */
dac33->irq = client->irq;
dac33->nsample = NSAMPLE_MAX;
dac33->nsample_max = NSAMPLE_MAX;
dac33->uthr = MODE7_UTHR;
/* Disable FIFO use by default */
dac33->fifo_mode = DAC33_FIFO_BYPASS;

408
sound/soc/codecs/twl4030.c
View file

@ -43,37 +43,37 @@
*/
static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = {
0x00, /* this register not used */
0x91, /* REG_CODEC_MODE (0x1) */
0xc3, /* REG_OPTION (0x2) */
0x00, /* REG_CODEC_MODE (0x1) */
0x00, /* REG_OPTION (0x2) */
0x00, /* REG_UNKNOWN (0x3) */
0x00, /* REG_MICBIAS_CTL (0x4) */
0x20, /* REG_ANAMICL (0x5) */
0x00, /* REG_ANAMICL (0x5) */
0x00, /* REG_ANAMICR (0x6) */
0x00, /* REG_AVADC_CTL (0x7) */
0x00, /* REG_ADCMICSEL (0x8) */
0x00, /* REG_DIGMIXING (0x9) */
0x0c, /* REG_ATXL1PGA (0xA) */
0x0c, /* REG_ATXR1PGA (0xB) */
0x00, /* REG_AVTXL2PGA (0xC) */
0x00, /* REG_AVTXR2PGA (0xD) */
0x0f, /* REG_ATXL1PGA (0xA) */
0x0f, /* REG_ATXR1PGA (0xB) */
0x0f, /* REG_AVTXL2PGA (0xC) */
0x0f, /* REG_AVTXR2PGA (0xD) */
0x00, /* REG_AUDIO_IF (0xE) */
0x00, /* REG_VOICE_IF (0xF) */
0x00, /* REG_ARXR1PGA (0x10) */
0x00, /* REG_ARXL1PGA (0x11) */
0x6c, /* REG_ARXR2PGA (0x12) */
0x6c, /* REG_ARXL2PGA (0x13) */
0x00, /* REG_VRXPGA (0x14) */
0x3f, /* REG_ARXR1PGA (0x10) */
0x3f, /* REG_ARXL1PGA (0x11) */
0x3f, /* REG_ARXR2PGA (0x12) */
0x3f, /* REG_ARXL2PGA (0x13) */
0x25, /* REG_VRXPGA (0x14) */
0x00, /* REG_VSTPGA (0x15) */
0x00, /* REG_VRX2ARXPGA (0x16) */
0x00, /* REG_AVDAC_CTL (0x17) */
0x00, /* REG_ARX2VTXPGA (0x18) */
0x00, /* REG_ARXL1_APGA_CTL (0x19) */
0x00, /* REG_ARXR1_APGA_CTL (0x1A) */
0x4a, /* REG_ARXL2_APGA_CTL (0x1B) */
0x4a, /* REG_ARXR2_APGA_CTL (0x1C) */
0x32, /* REG_ARXL1_APGA_CTL (0x19) */
0x32, /* REG_ARXR1_APGA_CTL (0x1A) */
0x32, /* REG_ARXL2_APGA_CTL (0x1B) */
0x32, /* REG_ARXR2_APGA_CTL (0x1C) */
0x00, /* REG_ATX2ARXPGA (0x1D) */
0x00, /* REG_BT_IF (0x1E) */
0x00, /* REG_BTPGA (0x1F) */
0x55, /* REG_BTPGA (0x1F) */
0x00, /* REG_BTSTPGA (0x20) */
0x00, /* REG_EAR_CTL (0x21) */
0x00, /* REG_HS_SEL (0x22) */
@ -85,32 +85,32 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = {
0x00, /* REG_PRECKR_CTL (0x28) */
0x00, /* REG_HFL_CTL (0x29) */
0x00, /* REG_HFR_CTL (0x2A) */
0x00, /* REG_ALC_CTL (0x2B) */
0x05, /* REG_ALC_CTL (0x2B) */
0x00, /* REG_ALC_SET1 (0x2C) */
0x00, /* REG_ALC_SET2 (0x2D) */
0x00, /* REG_BOOST_CTL (0x2E) */
0x00, /* REG_SOFTVOL_CTL (0x2F) */
0x00, /* REG_DTMF_FREQSEL (0x30) */
0x13, /* REG_DTMF_FREQSEL (0x30) */
0x00, /* REG_DTMF_TONEXT1H (0x31) */
0x00, /* REG_DTMF_TONEXT1L (0x32) */
0x00, /* REG_DTMF_TONEXT2H (0x33) */
0x00, /* REG_DTMF_TONEXT2L (0x34) */
0x00, /* REG_DTMF_TONOFF (0x35) */
0x00, /* REG_DTMF_WANONOFF (0x36) */
0x79, /* REG_DTMF_TONOFF (0x35) */
0x11, /* REG_DTMF_WANONOFF (0x36) */
0x00, /* REG_I2S_RX_SCRAMBLE_H (0x37) */
0x00, /* REG_I2S_RX_SCRAMBLE_M (0x38) */
0x00, /* REG_I2S_RX_SCRAMBLE_L (0x39) */
0x06, /* REG_APLL_CTL (0x3A) */
0x00, /* REG_DTMF_CTL (0x3B) */
0x00, /* REG_DTMF_PGA_CTL2 (0x3C) */
0x00, /* REG_DTMF_PGA_CTL1 (0x3D) */
0x44, /* REG_DTMF_PGA_CTL2 (0x3C) */
0x69, /* REG_DTMF_PGA_CTL1 (0x3D) */
0x00, /* REG_MISC_SET_1 (0x3E) */
0x00, /* REG_PCMBTMUX (0x3F) */
0x00, /* not used (0x40) */
0x00, /* not used (0x41) */
0x00, /* not used (0x42) */
0x00, /* REG_RX_PATH_SEL (0x43) */
0x00, /* REG_VDL_APGA_CTL (0x44) */
0x32, /* REG_VDL_APGA_CTL (0x44) */
0x00, /* REG_VIBRA_CTL (0x45) */
0x00, /* REG_VIBRA_SET (0x46) */
0x00, /* REG_VIBRA_PWM_SET (0x47) */
@ -143,6 +143,9 @@ struct twl4030_priv {
u8 earpiece_enabled;
u8 predrivel_enabled, predriver_enabled;
u8 carkitl_enabled, carkitr_enabled;
/* Delay needed after enabling the digimic interface */
unsigned int digimic_delay;
};
/*
@ -244,21 +247,112 @@ static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable)
udelay(10);
}
static void twl4030_init_chip(struct snd_soc_codec *codec)
static inline void twl4030_check_defaults(struct snd_soc_codec *codec)
{
u8 *cache = codec->reg_cache;
int i;
int i, difference = 0;
u8 val;
/* clear CODECPDZ prior to setting register defaults */
twl4030_codec_enable(codec, 0);
dev_dbg(codec->dev, "Checking TWL audio default configuration\n");
for (i = 1; i <= TWL4030_REG_MISC_SET_2; i++) {
twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &val, i);
if (val != twl4030_reg[i]) {
difference++;
dev_dbg(codec->dev,
"Reg 0x%02x: chip: 0x%02x driver: 0x%02x\n",
i, val, twl4030_reg[i]);
}
}
dev_dbg(codec->dev, "Found %d non maching registers. %s\n",
difference, difference ? "Not OK" : "OK");
}
static inline void twl4030_reset_registers(struct snd_soc_codec *codec)
{
int i;
/* set all audio section registers to reasonable defaults */
for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++)
if (i != TWL4030_REG_APLL_CTL)
twl4030_write(codec, i, cache[i]);
twl4030_write(codec, i, twl4030_reg[i]);
}
static void twl4030_init_chip(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct twl4030_setup_data *setup = socdev->codec_data;
struct snd_soc_codec *codec = socdev->card->codec;
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
u8 reg, byte;
int i = 0;
/* Check defaults, if instructed before anything else */
if (setup && setup->check_defaults)
twl4030_check_defaults(codec);
/* Reset registers, if no setup data or if instructed to do so */
if (!setup || (setup && setup->reset_registers))
twl4030_reset_registers(codec);
/* Refresh APLL_CTL register from HW */
twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
TWL4030_REG_APLL_CTL);
twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, byte);
/* anti-pop when changing analog gain */
reg = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1);
twl4030_write(codec, TWL4030_REG_MISC_SET_1,
reg | TWL4030_SMOOTH_ANAVOL_EN);
twl4030_write(codec, TWL4030_REG_OPTION,
TWL4030_ATXL1_EN | TWL4030_ATXR1_EN |
TWL4030_ARXL2_EN | TWL4030_ARXR2_EN);
/* REG_ARXR2_APGA_CTL reset according to the TRM: 0dB, DA_EN */
twl4030_write(codec, TWL4030_REG_ARXR2_APGA_CTL, 0x32);
/* Machine dependent setup */
if (!setup)
return;
twl4030->digimic_delay = setup->digimic_delay;
/* Configuration for headset ramp delay from setup data */
if (setup->sysclk != twl4030->sysclk)
dev_warn(codec->dev,
"Mismatch in APLL mclk: %u (configured: %u)\n",
setup->sysclk, twl4030->sysclk);
reg = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
reg &= ~TWL4030_RAMP_DELAY;
reg |= (setup->ramp_delay_value << 2);
twl4030_write_reg_cache(codec, TWL4030_REG_HS_POPN_SET, reg);
/* initiate offset cancellation */
twl4030_codec_enable(codec, 1);
reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
reg &= ~TWL4030_OFFSET_CNCL_SEL;
reg |= setup->offset_cncl_path;
twl4030_write(codec, TWL4030_REG_ANAMICL,
reg | TWL4030_CNCL_OFFSET_START);
/* wait for offset cancellation to complete */
do {
/* this takes a little while, so don't slam i2c */
udelay(2000);
twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
TWL4030_REG_ANAMICL);
} while ((i++ < 100) &&
((byte & TWL4030_CNCL_OFFSET_START) ==
TWL4030_CNCL_OFFSET_START));
/* Make sure that the reg_cache has the same value as the HW */
twl4030_write_reg_cache(codec, TWL4030_REG_ANAMICL, byte);
twl4030_codec_enable(codec, 0);
}
static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable)
{
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
@ -280,55 +374,6 @@ static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable)
twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, status);
}
static void twl4030_power_up(struct snd_soc_codec *codec)
{
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
u8 anamicl, regmisc1, byte;
int i = 0;
if (twl4030->codec_powered)
return;
/* set CODECPDZ to turn on codec */
twl4030_codec_enable(codec, 1);
/* initiate offset cancellation */
anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
twl4030_write(codec, TWL4030_REG_ANAMICL,
anamicl | TWL4030_CNCL_OFFSET_START);
/* wait for offset cancellation to complete */
do {
/* this takes a little while, so don't slam i2c */
udelay(2000);
twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
TWL4030_REG_ANAMICL);
} while ((i++ < 100) &&
((byte & TWL4030_CNCL_OFFSET_START) ==
TWL4030_CNCL_OFFSET_START));
/* Make sure that the reg_cache has the same value as the HW */
twl4030_write_reg_cache(codec, TWL4030_REG_ANAMICL, byte);
/* anti-pop when changing analog gain */
regmisc1 = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1);
twl4030_write(codec, TWL4030_REG_MISC_SET_1,
regmisc1 | TWL4030_SMOOTH_ANAVOL_EN);
/* toggle CODECPDZ as per TRM */
twl4030_codec_enable(codec, 0);
twl4030_codec_enable(codec, 1);
}
/*
* Unconditional power down
*/
static void twl4030_power_down(struct snd_soc_codec *codec)
{
/* power down */
twl4030_codec_enable(codec, 0);
}
/* Earpiece */
static const struct snd_kcontrol_new twl4030_dapm_earpiece_controls[] = {
SOC_DAPM_SINGLE("Voice", TWL4030_REG_EAR_CTL, 0, 1, 0),
@ -500,10 +545,11 @@ static const struct snd_kcontrol_new twl4030_dapm_abypassl2_control =
static const struct snd_kcontrol_new twl4030_dapm_abypassv_control =
SOC_DAPM_SINGLE("Switch", TWL4030_REG_VDL_APGA_CTL, 2, 1, 0);
/* Digital bypass gain, 0 mutes the bypass */
/* Digital bypass gain, mute instead of -30dB */
static const unsigned int twl4030_dapm_dbypass_tlv[] = {
TLV_DB_RANGE_HEAD(2),
0, 3, TLV_DB_SCALE_ITEM(-2400, 0, 1),
TLV_DB_RANGE_HEAD(3),
0, 1, TLV_DB_SCALE_ITEM(-3000, 600, 1),
2, 3, TLV_DB_SCALE_ITEM(-2400, 0, 0),
4, 7, TLV_DB_SCALE_ITEM(-1800, 600, 0),
};
@ -531,36 +577,6 @@ static const struct snd_kcontrol_new twl4030_dapm_dbypassv_control =
TWL4030_REG_VSTPGA, 0, 0x29, 0,
twl4030_dapm_dbypassv_tlv);
static int micpath_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct soc_enum *e = (struct soc_enum *)w->kcontrols->private_value;
unsigned char adcmicsel, micbias_ctl;
adcmicsel = twl4030_read_reg_cache(w->codec, TWL4030_REG_ADCMICSEL);
micbias_ctl = twl4030_read_reg_cache(w->codec, TWL4030_REG_MICBIAS_CTL);
/* Prepare the bits for the given TX path:
* shift_l == 0: TX1 microphone path
* shift_l == 2: TX2 microphone path */
if (e->shift_l) {
/* TX2 microphone path */
if (adcmicsel & TWL4030_TX2IN_SEL)
micbias_ctl |= TWL4030_MICBIAS2_CTL; /* digimic */
else
micbias_ctl &= ~TWL4030_MICBIAS2_CTL;
} else {
/* TX1 microphone path */
if (adcmicsel & TWL4030_TX1IN_SEL)
micbias_ctl |= TWL4030_MICBIAS1_CTL; /* digimic */
else
micbias_ctl &= ~TWL4030_MICBIAS1_CTL;
}
twl4030_write(w->codec, TWL4030_REG_MICBIAS_CTL, micbias_ctl);
return 0;
}
/*
* Output PGA builder:
* Handle the muting and unmuting of the given output (turning off the
@ -814,6 +830,16 @@ static int headsetrpga_event(struct snd_soc_dapm_widget *w,
return 0;
}
static int digimic_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec);
if (twl4030->digimic_delay)
mdelay(twl4030->digimic_delay);
return 0;
}
/*
* Some of the gain controls in TWL (mostly those which are associated with
* the outputs) are implemented in an interesting way:
@ -1374,14 +1400,10 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
/* Analog/Digital mic path selection.
TX1 Left/Right: either analog Left/Right or Digimic0
TX2 Left/Right: either analog Left/Right or Digimic1 */
SND_SOC_DAPM_MUX_E("TX1 Capture Route", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_micpathtx1_control, micpath_event,
SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD|
SND_SOC_DAPM_POST_REG),
SND_SOC_DAPM_MUX_E("TX2 Capture Route", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_micpathtx2_control, micpath_event,
SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD|
SND_SOC_DAPM_POST_REG),
SND_SOC_DAPM_MUX("TX1 Capture Route", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_micpathtx1_control),
SND_SOC_DAPM_MUX("TX2 Capture Route", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_micpathtx2_control),
/* Analog input mixers for the capture amplifiers */
SND_SOC_DAPM_MIXER("Analog Left",
@ -1398,10 +1420,17 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
SND_SOC_DAPM_PGA("ADC Physical Right",
TWL4030_REG_AVADC_CTL, 1, 0, NULL, 0),
SND_SOC_DAPM_PGA("Digimic0 Enable",
TWL4030_REG_ADCMICSEL, 1, 0, NULL, 0),
SND_SOC_DAPM_PGA("Digimic1 Enable",
TWL4030_REG_ADCMICSEL, 3, 0, NULL, 0),
SND_SOC_DAPM_PGA_E("Digimic0 Enable",
TWL4030_REG_ADCMICSEL, 1, 0, NULL, 0,
digimic_event, SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("Digimic1 Enable",
TWL4030_REG_ADCMICSEL, 3, 0, NULL, 0,
digimic_event, SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_SUPPLY("micbias1 select", TWL4030_REG_MICBIAS_CTL, 5, 0,
NULL, 0),
SND_SOC_DAPM_SUPPLY("micbias2 select", TWL4030_REG_MICBIAS_CTL, 6, 0,
NULL, 0),
SND_SOC_DAPM_MICBIAS("Mic Bias 1", TWL4030_REG_MICBIAS_CTL, 0, 0),
SND_SOC_DAPM_MICBIAS("Mic Bias 2", TWL4030_REG_MICBIAS_CTL, 1, 0),
@ -1419,8 +1448,11 @@ static const struct snd_soc_dapm_route intercon[] = {
/* Supply for the digital part (APLL) */
{"Digital Voice Playback Mixer", NULL, "APLL Enable"},
{"Digital R1 Playback Mixer", NULL, "AIF Enable"},
{"Digital L1 Playback Mixer", NULL, "AIF Enable"},
{"DAC Left1", NULL, "AIF Enable"},
{"DAC Right1", NULL, "AIF Enable"},
{"DAC Left2", NULL, "AIF Enable"},
{"DAC Right1", NULL, "AIF Enable"},
{"Digital R2 Playback Mixer", NULL, "AIF Enable"},
{"Digital L2 Playback Mixer", NULL, "AIF Enable"},
@ -1491,10 +1523,10 @@ static const struct snd_soc_dapm_route intercon[] = {
/* outputs */
/* Must be always connected (for AIF and APLL) */
{"Virtual HiFi OUT", NULL, "Digital L1 Playback Mixer"},
{"Virtual HiFi OUT", NULL, "Digital R1 Playback Mixer"},
{"Virtual HiFi OUT", NULL, "Digital L2 Playback Mixer"},
{"Virtual HiFi OUT", NULL, "Digital R2 Playback Mixer"},
{"Virtual HiFi OUT", NULL, "DAC Left1"},
{"Virtual HiFi OUT", NULL, "DAC Right1"},
{"Virtual HiFi OUT", NULL, "DAC Left2"},
{"Virtual HiFi OUT", NULL, "DAC Right2"},
/* Must be always connected (for APLL) */
{"Virtual Voice OUT", NULL, "Digital Voice Playback Mixer"},
/* Physical outputs */
@ -1531,6 +1563,9 @@ static const struct snd_soc_dapm_route intercon[] = {
{"Digimic0 Enable", NULL, "DIGIMIC0"},
{"Digimic1 Enable", NULL, "DIGIMIC1"},
{"DIGIMIC0", NULL, "micbias1 select"},
{"DIGIMIC1", NULL, "micbias2 select"},
/* TX1 Left capture path */
{"TX1 Capture Route", "Analog", "ADC Physical Left"},
{"TX1 Capture Route", "Digimic0", "Digimic0 Enable"},
@ -1605,10 +1640,10 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec,
break;
case SND_SOC_BIAS_STANDBY:
if (codec->bias_level == SND_SOC_BIAS_OFF)
twl4030_power_up(codec);
twl4030_codec_enable(codec, 1);
break;
case SND_SOC_BIAS_OFF:
twl4030_power_down(codec);
twl4030_codec_enable(codec, 0);
break;
}
codec->bias_level = level;
@ -1794,13 +1829,6 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
if (mode != old_mode) {
/* change rate and set CODECPDZ */
twl4030_codec_enable(codec, 0);
twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
twl4030_codec_enable(codec, 1);
}
/* sample size */
old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF);
format = old_format;
@ -1818,16 +1846,20 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
if (format != old_format) {
/* clear CODECPDZ before changing format (codec requirement) */
twl4030_codec_enable(codec, 0);
/* change format */
twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
/* set CODECPDZ afterwards */
twl4030_codec_enable(codec, 1);
if (format != old_format || mode != old_mode) {
if (twl4030->codec_powered) {
/*
* If the codec is powered, than we need to toggle the
* codec power.
*/
twl4030_codec_enable(codec, 0);
twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
twl4030_codec_enable(codec, 1);
} else {
twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
}
}
/* Store the important parameters for the DAI configuration and set
@ -1877,6 +1909,7 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
u8 old_format, format;
/* get format */
@ -1911,15 +1944,17 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
}
if (format != old_format) {
/* clear CODECPDZ before changing format (codec requirement) */
twl4030_codec_enable(codec, 0);
/* change format */
twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
/* set CODECPDZ afterwards */
twl4030_codec_enable(codec, 1);
if (twl4030->codec_powered) {
/*
* If the codec is powered, than we need to toggle the
* codec power.
*/
twl4030_codec_enable(codec, 0);
twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
twl4030_codec_enable(codec, 1);
} else {
twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
}
}
return 0;
@ -2011,6 +2046,7 @@ static int twl4030_voice_hw_params(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->card->codec;
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
u8 old_mode, mode;
/* Enable voice digital filters */
@ -2035,10 +2071,17 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream,
}
if (mode != old_mode) {
/* change rate and set CODECPDZ */
twl4030_codec_enable(codec, 0);
twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
twl4030_codec_enable(codec, 1);
if (twl4030->codec_powered) {
/*
* If the codec is powered, than we need to toggle the
* codec power.
*/
twl4030_codec_enable(codec, 0);
twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
twl4030_codec_enable(codec, 1);
} else {
twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
}
}
return 0;
@ -2068,6 +2111,7 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
u8 old_format, format;
/* get format */
@ -2099,10 +2143,17 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
}
if (format != old_format) {
/* change format and set CODECPDZ */
twl4030_codec_enable(codec, 0);
twl4030_write(codec, TWL4030_REG_VOICE_IF, format);
twl4030_codec_enable(codec, 1);
if (twl4030->codec_powered) {
/*
* If the codec is powered, than we need to toggle the
* codec power.
*/
twl4030_codec_enable(codec, 0);
twl4030_write(codec, TWL4030_REG_VOICE_IF, format);
twl4030_codec_enable(codec, 1);
} else {
twl4030_write(codec, TWL4030_REG_VOICE_IF, format);
}
}
return 0;
@ -2202,31 +2253,15 @@ static struct snd_soc_codec *twl4030_codec;
static int twl4030_soc_probe(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct twl4030_setup_data *setup = socdev->codec_data;
struct snd_soc_codec *codec;
struct twl4030_priv *twl4030;
int ret;
BUG_ON(!twl4030_codec);
codec = twl4030_codec;
twl4030 = snd_soc_codec_get_drvdata(codec);
socdev->card->codec = codec;
/* Configuration for headset ramp delay from setup data */
if (setup) {
unsigned char hs_pop;
if (setup->sysclk != twl4030->sysclk)
dev_warn(&pdev->dev,
"Mismatch in APLL mclk: %u (configured: %u)\n",
setup->sysclk, twl4030->sysclk);
hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
hs_pop &= ~TWL4030_RAMP_DELAY;
hs_pop |= (setup->ramp_delay_value << 2);
twl4030_write_reg_cache(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
}
twl4030_init_chip(pdev);
/* register pcms */
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
@ -2247,6 +2282,8 @@ static int twl4030_soc_remove(struct platform_device *pdev)
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->card->codec;
/* Reset registers to their chip default before leaving */
twl4030_reset_registers(codec);
twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF);
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
@ -2287,6 +2324,7 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev)
codec->read = twl4030_read_reg_cache;
codec->write = twl4030_write;
codec->set_bias_level = twl4030_set_bias_level;
codec->idle_bias_off = 1;
codec->dai = twl4030_dai;
codec->num_dai = ARRAY_SIZE(twl4030_dai);
codec->reg_cache_size = sizeof(twl4030_reg);
@ -2302,9 +2340,7 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev)
/* Set the defaults, and power up the codec */
twl4030->sysclk = twl4030_codec_get_mclk() / 1000;
twl4030_init_chip(codec);
codec->bias_level = SND_SOC_BIAS_OFF;
twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
ret = snd_soc_register_codec(codec);
if (ret != 0) {
@ -2322,7 +2358,7 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev)
return 0;
error_codec:
twl4030_power_down(codec);
twl4030_codec_enable(codec, 0);
kfree(codec->reg_cache);
error_cache:
kfree(twl4030);

4
sound/soc/codecs/twl4030.h
View file

@ -41,7 +41,11 @@ extern struct snd_soc_codec_device soc_codec_dev_twl4030;
struct twl4030_setup_data {
unsigned int ramp_delay_value;
unsigned int digimic_delay; /* in ms */
unsigned int sysclk;
unsigned int offset_cncl_path;
unsigned int check_defaults:1;
unsigned int reset_registers:1;
unsigned int hs_extmute:1;
void (*set_hs_extmute)(int mute);
};

58
sound/soc/codecs/twl6040.c
View file

@ -360,6 +360,13 @@ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf)
return 0;
}
static int twl6040_hs_dac_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
msleep(1);
return 0;
}
static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
@ -371,6 +378,8 @@ static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w,
else
priv->non_lp--;
msleep(1);
return 0;
}
@ -471,20 +480,6 @@ static const struct snd_kcontrol_new hfdacl_switch_controls =
static const struct snd_kcontrol_new hfdacr_switch_controls =
SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 2, 1, 0);
/* Headset driver switches */
static const struct snd_kcontrol_new hsl_driver_switch_controls =
SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSLCTL, 2, 1, 0);
static const struct snd_kcontrol_new hsr_driver_switch_controls =
SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSRCTL, 2, 1, 0);
/* Handsfree driver switches */
static const struct snd_kcontrol_new hfl_driver_switch_controls =
SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 4, 1, 0);
static const struct snd_kcontrol_new hfr_driver_switch_controls =
SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 4, 1, 0);
static const struct snd_kcontrol_new ep_driver_switch_controls =
SOC_DAPM_SINGLE("Switch", TWL6040_REG_EARCTL, 0, 1, 0);
@ -548,10 +543,14 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
TWL6040_REG_DMICBCTL, 4, 0),
/* DACs */
SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback",
TWL6040_REG_HSLCTL, 0, 0),
SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback",
TWL6040_REG_HSRCTL, 0, 0),
SND_SOC_DAPM_DAC_E("HSDAC Left", "Headset Playback",
TWL6040_REG_HSLCTL, 0, 0,
twl6040_hs_dac_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_DAC_E("HSDAC Right", "Headset Playback",
TWL6040_REG_HSRCTL, 0, 0,
twl6040_hs_dac_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_DAC_E("HFDAC Left", "Handsfree Playback",
TWL6040_REG_HFLCTL, 0, 0,
twl6040_power_mode_event,
@ -571,18 +570,19 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
SND_SOC_DAPM_SWITCH("HFDAC Right Playback",
SND_SOC_NOPM, 0, 0, &hfdacr_switch_controls),
SND_SOC_DAPM_SWITCH("Headset Left Driver",
SND_SOC_NOPM, 0, 0, &hsl_driver_switch_controls),
SND_SOC_DAPM_SWITCH("Headset Right Driver",
SND_SOC_NOPM, 0, 0, &hsr_driver_switch_controls),
SND_SOC_DAPM_SWITCH_E("Handsfree Left Driver",
SND_SOC_NOPM, 0, 0, &hfl_driver_switch_controls,
/* Analog playback drivers */
SND_SOC_DAPM_PGA_E("Handsfree Left Driver",
TWL6040_REG_HFLCTL, 4, 0, NULL, 0,
twl6040_power_mode_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_SWITCH_E("Handsfree Right Driver",
SND_SOC_NOPM, 0, 0, &hfr_driver_switch_controls,
SND_SOC_DAPM_PGA_E("Handsfree Right Driver",
TWL6040_REG_HFRCTL, 4, 0, NULL, 0,
twl6040_power_mode_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_PGA("Headset Left Driver",
TWL6040_REG_HSLCTL, 2, 0, NULL, 0),
SND_SOC_DAPM_PGA("Headset Right Driver",
TWL6040_REG_HSRCTL, 2, 0, NULL, 0),
SND_SOC_DAPM_SWITCH_E("Earphone Driver",
SND_SOC_NOPM, 0, 0, &ep_driver_switch_controls,
twl6040_power_mode_event,
@ -616,8 +616,8 @@ static const struct snd_soc_dapm_route intercon[] = {
{"HSDAC Left Playback", "Switch", "HSDAC Left"},
{"HSDAC Right Playback", "Switch", "HSDAC Right"},
{"Headset Left Driver", "Switch", "HSDAC Left Playback"},
{"Headset Right Driver", "Switch", "HSDAC Right Playback"},
{"Headset Left Driver", NULL, "HSDAC Left Playback"},
{"Headset Right Driver", NULL, "HSDAC Right Playback"},
{"HSOL", NULL, "Headset Left Driver"},
{"HSOR", NULL, "Headset Right Driver"},
@ -928,7 +928,7 @@ static int twl6040_set_dai_sysclk(struct snd_soc_dai *codec_dai,
case 19200000:
/* mclk input, pll disabled */
hppllctl |= TWL6040_MCLK_19200KHZ |
TWL6040_HPLLSQRBP |
TWL6040_HPLLSQRENA |
TWL6040_HPLLBP;
break;
case 26000000:

64
sound/soc/codecs/uda134x.c
View file

@ -28,19 +28,6 @@
#include "uda134x.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)
@ -58,7 +45,7 @@ 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,
0x00, 0x83, 0x00, 0x40, 0x80, 0xC0, 0x00,
};
/*
@ -117,6 +104,7 @@ static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg,
case UDA134X_DATA000:
case UDA134X_DATA001:
case UDA134X_DATA010:
case UDA134X_DATA011:
addr = UDA134X_DATA0_ADDR;
break;
case UDA134X_DATA1:
@ -353,8 +341,22 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec,
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);
switch (pd->model) {
case UDA134X_UDA1340:
case UDA134X_UDA1344:
case UDA134X_UDA1345:
reg = uda134x_read_reg_cache(codec, UDA134X_DATA011);
uda134x_write(codec, UDA134X_DATA011, reg | 0x03);
break;
case UDA134X_UDA1341:
reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1);
uda134x_write(codec, UDA134X_STATUS1, reg | 0x03);
break;
default:
printk(KERN_ERR "UDA134X SoC codec: "
"unsupported model %d\n", pd->model);
return -EINVAL;
}
break;
case SND_SOC_BIAS_PREPARE:
/* power on */
@ -367,8 +369,22 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec,
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));
switch (pd->model) {
case UDA134X_UDA1340:
case UDA134X_UDA1344:
case UDA134X_UDA1345:
reg = uda134x_read_reg_cache(codec, UDA134X_DATA011);
uda134x_write(codec, UDA134X_DATA011, reg & ~(0x03));
break;
case UDA134X_UDA1341:
reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1);
uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03));
break;
default:
printk(KERN_ERR "UDA134X SoC codec: "
"unsupported model %d\n", pd->model);
return -EINVAL;
}
break;
case SND_SOC_BIAS_OFF:
/* power off */
@ -531,9 +547,7 @@ static int uda134x_soc_probe(struct platform_device *pdev)
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);
@ -544,6 +558,14 @@ static int uda134x_soc_probe(struct platform_device *pdev)
uda134x_reset(codec);
if (pd->is_powered_on_standby) {
codec->set_bias_level = NULL;
uda134x_set_bias_level(codec, SND_SOC_BIAS_ON);
} else {
codec->set_bias_level = uda134x_set_bias_level;
uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
}
/* register pcms */
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0) {

5
sound/soc/codecs/uda134x.h
View file

@ -23,9 +23,10 @@
#define UDA134X_DATA000 10
#define UDA134X_DATA001 11
#define UDA134X_DATA010 12
#define UDA134X_DATA1 13
#define UDA134X_DATA011 13
#define UDA134X_DATA1 14
#define UDA134X_REGS_NUM 14
#define UDA134X_REGS_NUM 15
#define STATUS0_DAIFMT_MASK (~(7<<1))
#define STATUS0_SYSCLK_MASK (~(3<<4))

2
sound/soc/codecs/wm2000.c
View file

@ -795,6 +795,8 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c,
dev_set_drvdata(&i2c->dev, wm2000);
wm2000->anc_eng_ena = 1;
wm2000->anc_active = 1;
wm2000->spk_ena = 1;
wm2000->i2c = i2c;
wm2000_reset(wm2000);

10
sound/soc/codecs/wm8523.c
View file

@ -482,7 +482,8 @@ static int wm8523_register(struct wm8523_priv *wm8523,
if (wm8523_codec) {
dev_err(codec->dev, "Another WM8523 is registered\n");
return -EINVAL;
ret = -EINVAL;
goto err;
}
mutex_init(&codec->mutex);
@ -570,18 +571,19 @@ static int wm8523_register(struct wm8523_priv *wm8523,
ret = snd_soc_register_codec(codec);
if (ret != 0) {
dev_err(codec->dev, "Failed to register codec: %d\n", ret);
return ret;
goto err_enable;
}
ret = snd_soc_register_dai(&wm8523_dai);
if (ret != 0) {
dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
snd_soc_unregister_codec(codec);
return ret;
goto err_codec;
}
return 0;
err_codec:
snd_soc_unregister_codec(codec);
err_enable:
regulator_bulk_disable(ARRAY_SIZE(wm8523->supplies), wm8523->supplies);
err_get:

3
sound/soc/codecs/wm8711.c
View file

@ -439,7 +439,8 @@ static int wm8711_register(struct wm8711_priv *wm8711,
if (wm8711_codec) {
dev_err(codec->dev, "Another WM8711 is registered\n");
return -EINVAL;
ret = -EINVAL;
goto err;
}
mutex_init(&codec->mutex);

579
sound/soc/codecs/wm8741.c Normal file
View file

@ -0,0 +1,579 @@
/*
* wm8741.c -- WM8741 ALSA SoC Audio driver
*
* Copyright 2010 Wolfson Microelectronics plc
*
* Author: Ian Lartey <ian@opensource.wolfsonmicro.com>
*
*
* 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/moduleparam.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <linux/regulator/consumer.h>
#include <linux/slab.h>
#include <sound/core.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/tlv.h>
#include "wm8741.h"
static struct snd_soc_codec *wm8741_codec;
struct snd_soc_codec_device soc_codec_dev_wm8741;
#define WM8741_NUM_SUPPLIES 2
static const char *wm8741_supply_names[WM8741_NUM_SUPPLIES] = {
"AVDD",
"DVDD",
};
#define WM8741_NUM_RATES 4
/* codec private data */
struct wm8741_priv {
struct snd_soc_codec codec;
u16 reg_cache[WM8741_REGISTER_COUNT];
struct regulator_bulk_data supplies[WM8741_NUM_SUPPLIES];
unsigned int sysclk;
unsigned int rate_constraint_list[WM8741_NUM_RATES];
struct snd_pcm_hw_constraint_list rate_constraint;
};
static const u16 wm8741_reg_defaults[WM8741_REGISTER_COUNT] = {
0x0000, /* R0 - DACLLSB Attenuation */
0x0000, /* R1 - DACLMSB Attenuation */
0x0000, /* R2 - DACRLSB Attenuation */
0x0000, /* R3 - DACRMSB Attenuation */
0x0000, /* R4 - Volume Control */
0x000A, /* R5 - Format Control */
0x0000, /* R6 - Filter Control */
0x0000, /* R7 - Mode Control 1 */
0x0002, /* R8 - Mode Control 2 */
0x0000, /* R9 - Reset */
0x0002, /* R32 - ADDITONAL_CONTROL_1 */
};
static int wm8741_reset(struct snd_soc_codec *codec)
{
return snd_soc_write(codec, WM8741_RESET, 0);
}
static const DECLARE_TLV_DB_SCALE(dac_tlv_fine, -12700, 13, 0);
static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 400, 0);
static const struct snd_kcontrol_new wm8741_snd_controls[] = {
SOC_DOUBLE_R_TLV("Fine Playback Volume", WM8741_DACLLSB_ATTENUATION,
WM8741_DACRLSB_ATTENUATION, 1, 255, 1, dac_tlv_fine),
SOC_DOUBLE_R_TLV("Playback Volume", WM8741_DACLMSB_ATTENUATION,
WM8741_DACRMSB_ATTENUATION, 0, 511, 1, dac_tlv),
};
static const struct snd_soc_dapm_widget wm8741_dapm_widgets[] = {
SND_SOC_DAPM_DAC("DACL", "Playback", SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_DAC("DACR", "Playback", SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_OUTPUT("VOUTLP"),
SND_SOC_DAPM_OUTPUT("VOUTLN"),
SND_SOC_DAPM_OUTPUT("VOUTRP"),
SND_SOC_DAPM_OUTPUT("VOUTRN"),
};
static const struct snd_soc_dapm_route intercon[] = {
{ "VOUTLP", NULL, "DACL" },
{ "VOUTLN", NULL, "DACL" },
{ "VOUTRP", NULL, "DACR" },
{ "VOUTRN", NULL, "DACR" },
};
static int wm8741_add_widgets(struct snd_soc_codec *codec)
{
snd_soc_dapm_new_controls(codec, wm8741_dapm_widgets,
ARRAY_SIZE(wm8741_dapm_widgets));
snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
return 0;
}
static struct {
int value;
int ratio;
} lrclk_ratios[WM8741_NUM_RATES] = {
{ 1, 256 },
{ 2, 384 },
{ 3, 512 },
{ 4, 768 },
};
static int wm8741_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec);
/* The set of sample rates that can be supported depends on the
* MCLK supplied to the CODEC - enforce this.
*/
if (!wm8741->sysclk) {
dev_err(codec->dev,
"No MCLK configured, call set_sysclk() on init\n");
return -EINVAL;
}
snd_pcm_hw_constraint_list(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
&wm8741->rate_constraint);
return 0;
}
static int wm8741_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->card->codec;
struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec);
u16 iface = snd_soc_read(codec, WM8741_FORMAT_CONTROL) & 0x1FC;
int i;
/* Find a supported LRCLK ratio */
for (i = 0; i < ARRAY_SIZE(lrclk_ratios); i++) {
if (wm8741->sysclk / params_rate(params) ==
lrclk_ratios[i].ratio)
break;
}
/* Should never happen, should be handled by constraints */
if (i == ARRAY_SIZE(lrclk_ratios)) {
dev_err(codec->dev, "MCLK/fs ratio %d unsupported\n",
wm8741->sysclk / params_rate(params));
return -EINVAL;
}
/* bit size */
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
break;
case SNDRV_PCM_FORMAT_S20_3LE:
iface |= 0x0001;
break;
case SNDRV_PCM_FORMAT_S24_LE:
iface |= 0x0002;
break;
case SNDRV_PCM_FORMAT_S32_LE:
iface |= 0x0003;
break;
default:
dev_dbg(codec->dev, "wm8741_hw_params: Unsupported bit size param = %d",
params_format(params));
return -EINVAL;
}
dev_dbg(codec->dev, "wm8741_hw_params: bit size param = %d",
params_format(params));
snd_soc_write(codec, WM8741_FORMAT_CONTROL, iface);
return 0;
}
static int wm8741_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 wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec);
unsigned int val;
int i;
dev_dbg(codec->dev, "wm8741_set_dai_sysclk info: freq=%dHz\n", freq);
wm8741->sysclk = freq;
wm8741->rate_constraint.count = 0;
for (i = 0; i < ARRAY_SIZE(lrclk_ratios); i++) {
dev_dbg(codec->dev, "index = %d, ratio = %d, freq = %d",
i, lrclk_ratios[i].ratio, freq);
val = freq / lrclk_ratios[i].ratio;
/* Check that it's a standard rate since core can't
* cope with others and having the odd rates confuses
* constraint matching.
*/
switch (val) {
case 32000:
case 44100:
case 48000:
case 64000:
case 88200:
case 96000:
dev_dbg(codec->dev, "Supported sample rate: %dHz\n",
val);
wm8741->rate_constraint_list[i] = val;
wm8741->rate_constraint.count++;
break;
default:
dev_dbg(codec->dev, "Skipping sample rate: %dHz\n",
val);
}
}
/* Need at least one supported rate... */
if (wm8741->rate_constraint.count == 0)
return -EINVAL;
return 0;
}
static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
u16 iface = snd_soc_read(codec, WM8741_FORMAT_CONTROL) & 0x1C3;
/* check master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBS_CFS:
break;
default:
return -EINVAL;
}
/* interface format */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
iface |= 0x0008;
break;
case SND_SOC_DAIFMT_RIGHT_J:
break;
case SND_SOC_DAIFMT_LEFT_J:
iface |= 0x0004;
break;
case SND_SOC_DAIFMT_DSP_A:
iface |= 0x0003;
break;
case SND_SOC_DAIFMT_DSP_B:
iface |= 0x0013;
break;
default:
return -EINVAL;
}
/* clock inversion */
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
break;
case SND_SOC_DAIFMT_IB_IF:
iface |= 0x0010;
break;
case SND_SOC_DAIFMT_IB_NF:
iface |= 0x0020;
break;
case SND_SOC_DAIFMT_NB_IF:
iface |= 0x0030;
break;
default:
return -EINVAL;
}
dev_dbg(codec->dev, "wm8741_set_dai_fmt: Format=%x, Clock Inv=%x\n",
fmt & SND_SOC_DAIFMT_FORMAT_MASK,
((fmt & SND_SOC_DAIFMT_INV_MASK)));
snd_soc_write(codec, WM8741_FORMAT_CONTROL, iface);
return 0;
}
#define WM8741_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \
SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | \
SNDRV_PCM_RATE_192000)
#define WM8741_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
static struct snd_soc_dai_ops wm8741_dai_ops = {
.startup = wm8741_startup,
.hw_params = wm8741_hw_params,
.set_sysclk = wm8741_set_dai_sysclk,
.set_fmt = wm8741_set_dai_fmt,
};
struct snd_soc_dai wm8741_dai = {
.name = "WM8741",
.playback = {
.stream_name = "Playback",
.channels_min = 2, /* Mono modes not yet supported */
.channels_max = 2,
.rates = WM8741_RATES,
.formats = WM8741_FORMATS,
},
.ops = &wm8741_dai_ops,
};
EXPORT_SYMBOL_GPL(wm8741_dai);
#ifdef CONFIG_PM
static int wm8741_resume(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->card->codec;
u16 *cache = codec->reg_cache;
int i;
/* RESTORE REG Cache */
for (i = 0; i < WM8741_REGISTER_COUNT; i++) {
if (cache[i] == wm8741_reg_defaults[i] || WM8741_RESET == i)
continue;
snd_soc_write(codec, i, cache[i]);
}
return 0;
}
#else
#define wm8741_suspend NULL
#define wm8741_resume NULL
#endif
static int wm8741_probe(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec;
int ret = 0;
if (wm8741_codec == NULL) {
dev_err(&pdev->dev, "Codec device not registered\n");
return -ENODEV;
}
socdev->card->codec = wm8741_codec;
codec = wm8741_codec;
/* register pcms */
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0) {
dev_err(codec->dev, "failed to create pcms: %d\n", ret);
goto pcm_err;
}
snd_soc_add_controls(codec, wm8741_snd_controls,
ARRAY_SIZE(wm8741_snd_controls));
wm8741_add_widgets(codec);
return ret;
pcm_err:
return ret;
}
static int wm8741_remove(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
return 0;
}
struct snd_soc_codec_device soc_codec_dev_wm8741 = {
.probe = wm8741_probe,
.remove = wm8741_remove,
.resume = wm8741_resume,
};
EXPORT_SYMBOL_GPL(soc_codec_dev_wm8741);
static int wm8741_register(struct wm8741_priv *wm8741,
enum snd_soc_control_type control)
{
int ret;
struct snd_soc_codec *codec = &wm8741->codec;
int i;
if (wm8741_codec) {
dev_err(codec->dev, "Another WM8741 is registered\n");
return -EINVAL;
}
mutex_init(&codec->mutex);
INIT_LIST_HEAD(&codec->dapm_widgets);
INIT_LIST_HEAD(&codec->dapm_paths);
snd_soc_codec_set_drvdata(codec, wm8741);
codec->name = "WM8741";
codec->owner = THIS_MODULE;
codec->bias_level = SND_SOC_BIAS_OFF;
codec->set_bias_level = NULL;
codec->dai = &wm8741_dai;
codec->num_dai = 1;
codec->reg_cache_size = WM8741_REGISTER_COUNT;
codec->reg_cache = &wm8741->reg_cache;
wm8741->rate_constraint.list = &wm8741->rate_constraint_list[0];
wm8741->rate_constraint.count =
ARRAY_SIZE(wm8741->rate_constraint_list);
memcpy(codec->reg_cache, wm8741_reg_defaults,
sizeof(wm8741->reg_cache));
ret = snd_soc_codec_set_cache_io(codec, 7, 9, control);
if (ret != 0) {
dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
goto err;
}
for (i = 0; i < ARRAY_SIZE(wm8741->supplies); i++)
wm8741->supplies[i].supply = wm8741_supply_names[i];
ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8741->supplies),
wm8741->supplies);
if (ret != 0) {
dev_err(codec->dev, "Failed to request supplies: %d\n", ret);
goto err;
}
ret = regulator_bulk_enable(ARRAY_SIZE(wm8741->supplies),
wm8741->supplies);
if (ret != 0) {
dev_err(codec->dev, "Failed to enable supplies: %d\n", ret);
goto err_get;
}
ret = wm8741_reset(codec);
if (ret < 0) {
dev_err(codec->dev, "Failed to issue reset\n");
goto err_enable;
}
wm8741_dai.dev = codec->dev;
/* Change some default settings - latch VU */
wm8741->reg_cache[WM8741_DACLLSB_ATTENUATION] |= WM8741_UPDATELL;
wm8741->reg_cache[WM8741_DACLMSB_ATTENUATION] |= WM8741_UPDATELM;
wm8741->reg_cache[WM8741_DACRLSB_ATTENUATION] |= WM8741_UPDATERL;
wm8741->reg_cache[WM8741_DACRLSB_ATTENUATION] |= WM8741_UPDATERM;
wm8741_codec = codec;
ret = snd_soc_register_codec(codec);
if (ret != 0) {
dev_err(codec->dev, "Failed to register codec: %d\n", ret);
return ret;
}
ret = snd_soc_register_dai(&wm8741_dai);
if (ret != 0) {
dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
snd_soc_unregister_codec(codec);
return ret;
}
dev_dbg(codec->dev, "Successful registration\n");
return 0;
err_enable:
regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies);
err_get:
regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies);
err:
kfree(wm8741);
return ret;
}
static void wm8741_unregister(struct wm8741_priv *wm8741)
{
regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies);
snd_soc_unregister_dai(&wm8741_dai);
snd_soc_unregister_codec(&wm8741->codec);
kfree(wm8741);
wm8741_codec = NULL;
}
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
static __devinit int wm8741_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
struct wm8741_priv *wm8741;
struct snd_soc_codec *codec;
wm8741 = kzalloc(sizeof(struct wm8741_priv), GFP_KERNEL);
if (wm8741 == NULL)
return -ENOMEM;
codec = &wm8741->codec;
codec->hw_write = (hw_write_t)i2c_master_send;
i2c_set_clientdata(i2c, wm8741);
codec->control_data = i2c;
codec->dev = &i2c->dev;
return wm8741_register(wm8741, SND_SOC_I2C);
}
static __devexit int wm8741_i2c_remove(struct i2c_client *client)
{
struct wm8741_priv *wm8741 = i2c_get_clientdata(client);
wm8741_unregister(wm8741);
return 0;
}
static const struct i2c_device_id wm8741_i2c_id[] = {
{ "wm8741", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8741_i2c_id);
static struct i2c_driver wm8741_i2c_driver = {
.driver = {
.name = "WM8741",
.owner = THIS_MODULE,
},
.probe = wm8741_i2c_probe,
.remove = __devexit_p(wm8741_i2c_remove),
.id_table = wm8741_i2c_id,
};
#endif
static int __init wm8741_modinit(void)
{
int ret;
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
ret = i2c_add_driver(&wm8741_i2c_driver);
if (ret != 0) {
printk(KERN_ERR "Failed to register WM8741 I2C driver: %d\n",
ret);
}
#endif
return 0;
}
module_init(wm8741_modinit);
static void __exit wm8741_exit(void)
{
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
i2c_del_driver(&wm8741_i2c_driver);
#endif
}
module_exit(wm8741_exit);
MODULE_DESCRIPTION("ASoC WM8741 driver");
MODULE_AUTHOR("Ian Lartey <[email protected]>");
MODULE_LICENSE("GPL");

214
sound/soc/codecs/wm8741.h Normal file
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@ -0,0 +1,214 @@
/*
* wm8741.h -- WM8423 ASoC driver
*
* Copyright 2010 Wolfson Microelectronics, plc
*
* Author: Ian Lartey <ian@opensource.wolfsonmicro.com>
*
* Based on wm8753.h
*
* 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 _WM8741_H
#define _WM8741_H
/*
* Register values.
*/
#define WM8741_DACLLSB_ATTENUATION 0x00
#define WM8741_DACLMSB_ATTENUATION 0x01
#define WM8741_DACRLSB_ATTENUATION 0x02
#define WM8741_DACRMSB_ATTENUATION 0x03
#define WM8741_VOLUME_CONTROL 0x04
#define WM8741_FORMAT_CONTROL 0x05
#define WM8741_FILTER_CONTROL 0x06
#define WM8741_MODE_CONTROL_1 0x07
#define WM8741_MODE_CONTROL_2 0x08
#define WM8741_RESET 0x09
#define WM8741_ADDITIONAL_CONTROL_1 0x20
#define WM8741_REGISTER_COUNT 11
#define WM8741_MAX_REGISTER 0x20
/*
* Field Definitions.
*/
/*
* R0 (0x00) - DACLLSB_ATTENUATION
*/
#define WM8741_UPDATELL 0x0020 /* UPDATELL */
#define WM8741_UPDATELL_MASK 0x0020 /* UPDATELL */
#define WM8741_UPDATELL_SHIFT 5 /* UPDATELL */
#define WM8741_UPDATELL_WIDTH 1 /* UPDATELL */
#define WM8741_LAT_4_0_MASK 0x001F /* LAT[4:0] - [4:0] */
#define WM8741_LAT_4_0_SHIFT 0 /* LAT[4:0] - [4:0] */
#define WM8741_LAT_4_0_WIDTH 5 /* LAT[4:0] - [4:0] */
/*
* R1 (0x01) - DACLMSB_ATTENUATION
*/
#define WM8741_UPDATELM 0x0020 /* UPDATELM */
#define WM8741_UPDATELM_MASK 0x0020 /* UPDATELM */
#define WM8741_UPDATELM_SHIFT 5 /* UPDATELM */
#define WM8741_UPDATELM_WIDTH 1 /* UPDATELM */
#define WM8741_LAT_9_5_0_MASK 0x001F /* LAT[9:5] - [4:0] */
#define WM8741_LAT_9_5_0_SHIFT 0 /* LAT[9:5] - [4:0] */
#define WM8741_LAT_9_5_0_WIDTH 5 /* LAT[9:5] - [4:0] */
/*
* R2 (0x02) - DACRLSB_ATTENUATION
*/
#define WM8741_UPDATERL 0x0020 /* UPDATERL */
#define WM8741_UPDATERL_MASK 0x0020 /* UPDATERL */
#define WM8741_UPDATERL_SHIFT 5 /* UPDATERL */
#define WM8741_UPDATERL_WIDTH 1 /* UPDATERL */
#define WM8741_RAT_4_0_MASK 0x001F /* RAT[4:0] - [4:0] */
#define WM8741_RAT_4_0_SHIFT 0 /* RAT[4:0] - [4:0] */
#define WM8741_RAT_4_0_WIDTH 5 /* RAT[4:0] - [4:0] */
/*
* R3 (0x03) - DACRMSB_ATTENUATION
*/
#define WM8741_UPDATERM 0x0020 /* UPDATERM */
#define WM8741_UPDATERM_MASK 0x0020 /* UPDATERM */
#define WM8741_UPDATERM_SHIFT 5 /* UPDATERM */
#define WM8741_UPDATERM_WIDTH 1 /* UPDATERM */
#define WM8741_RAT_9_5_0_MASK 0x001F /* RAT[9:5] - [4:0] */
#define WM8741_RAT_9_5_0_SHIFT 0 /* RAT[9:5] - [4:0] */
#define WM8741_RAT_9_5_0_WIDTH 5 /* RAT[9:5] - [4:0] */
/*
* R4 (0x04) - VOLUME_CONTROL
*/
#define WM8741_AMUTE 0x0080 /* AMUTE */
#define WM8741_AMUTE_MASK 0x0080 /* AMUTE */
#define WM8741_AMUTE_SHIFT 7 /* AMUTE */
#define WM8741_AMUTE_WIDTH 1 /* AMUTE */
#define WM8741_ZFLAG_MASK 0x0060 /* ZFLAG - [6:5] */
#define WM8741_ZFLAG_SHIFT 5 /* ZFLAG - [6:5] */
#define WM8741_ZFLAG_WIDTH 2 /* ZFLAG - [6:5] */
#define WM8741_IZD 0x0010 /* IZD */
#define WM8741_IZD_MASK 0x0010 /* IZD */
#define WM8741_IZD_SHIFT 4 /* IZD */
#define WM8741_IZD_WIDTH 1 /* IZD */
#define WM8741_SOFT 0x0008 /* SOFT MUTE */
#define WM8741_SOFT_MASK 0x0008 /* SOFT MUTE */
#define WM8741_SOFT_SHIFT 3 /* SOFT MUTE */
#define WM8741_SOFT_WIDTH 1 /* SOFT MUTE */
#define WM8741_ATC 0x0004 /* ATC */
#define WM8741_ATC_MASK 0x0004 /* ATC */
#define WM8741_ATC_SHIFT 2 /* ATC */
#define WM8741_ATC_WIDTH 1 /* ATC */
#define WM8741_ATT2DB 0x0002 /* ATT2DB */
#define WM8741_ATT2DB_MASK 0x0002 /* ATT2DB */
#define WM8741_ATT2DB_SHIFT 1 /* ATT2DB */
#define WM8741_ATT2DB_WIDTH 1 /* ATT2DB */
#define WM8741_VOL_RAMP 0x0001 /* VOL_RAMP */
#define WM8741_VOL_RAMP_MASK 0x0001 /* VOL_RAMP */
#define WM8741_VOL_RAMP_SHIFT 0 /* VOL_RAMP */
#define WM8741_VOL_RAMP_WIDTH 1 /* VOL_RAMP */
/*
* R5 (0x05) - FORMAT_CONTROL
*/
#define WM8741_PWDN 0x0080 /* PWDN */
#define WM8741_PWDN_MASK 0x0080 /* PWDN */
#define WM8741_PWDN_SHIFT 7 /* PWDN */
#define WM8741_PWDN_WIDTH 1 /* PWDN */
#define WM8741_REV 0x0040 /* REV */
#define WM8741_REV_MASK 0x0040 /* REV */
#define WM8741_REV_SHIFT 6 /* REV */
#define WM8741_REV_WIDTH 1 /* REV */
#define WM8741_BCP 0x0020 /* BCP */
#define WM8741_BCP_MASK 0x0020 /* BCP */
#define WM8741_BCP_SHIFT 5 /* BCP */
#define WM8741_BCP_WIDTH 1 /* BCP */
#define WM8741_LRP 0x0010 /* LRP */
#define WM8741_LRP_MASK 0x0010 /* LRP */
#define WM8741_LRP_SHIFT 4 /* LRP */
#define WM8741_LRP_WIDTH 1 /* LRP */
#define WM8741_FMT_MASK 0x000C /* FMT - [3:2] */
#define WM8741_FMT_SHIFT 2 /* FMT - [3:2] */
#define WM8741_FMT_WIDTH 2 /* FMT - [3:2] */
#define WM8741_IWL_MASK 0x0003 /* IWL - [1:0] */
#define WM8741_IWL_SHIFT 0 /* IWL - [1:0] */
#define WM8741_IWL_WIDTH 2 /* IWL - [1:0] */
/*
* R6 (0x06) - FILTER_CONTROL
*/
#define WM8741_ZFLAG_HI 0x0080 /* ZFLAG_HI */
#define WM8741_ZFLAG_HI_MASK 0x0080 /* ZFLAG_HI */
#define WM8741_ZFLAG_HI_SHIFT 7 /* ZFLAG_HI */
#define WM8741_ZFLAG_HI_WIDTH 1 /* ZFLAG_HI */
#define WM8741_DEEMPH_MASK 0x0060 /* DEEMPH - [6:5] */
#define WM8741_DEEMPH_SHIFT 5 /* DEEMPH - [6:5] */
#define WM8741_DEEMPH_WIDTH 2 /* DEEMPH - [6:5] */
#define WM8741_DSDFILT_MASK 0x0018 /* DSDFILT - [4:3] */
#define WM8741_DSDFILT_SHIFT 3 /* DSDFILT - [4:3] */
#define WM8741_DSDFILT_WIDTH 2 /* DSDFILT - [4:3] */
#define WM8741_FIRSEL_MASK 0x0007 /* FIRSEL - [2:0] */
#define WM8741_FIRSEL_SHIFT 0 /* FIRSEL - [2:0] */
#define WM8741_FIRSEL_WIDTH 3 /* FIRSEL - [2:0] */
/*
* R7 (0x07) - MODE_CONTROL_1
*/
#define WM8741_MODE8X 0x0080 /* MODE8X */
#define WM8741_MODE8X_MASK 0x0080 /* MODE8X */
#define WM8741_MODE8X_SHIFT 7 /* MODE8X */
#define WM8741_MODE8X_WIDTH 1 /* MODE8X */
#define WM8741_OSR_MASK 0x0060 /* OSR - [6:5] */
#define WM8741_OSR_SHIFT 5 /* OSR - [6:5] */
#define WM8741_OSR_WIDTH 2 /* OSR - [6:5] */
#define WM8741_SR_MASK 0x001C /* SR - [4:2] */
#define WM8741_SR_SHIFT 2 /* SR - [4:2] */
#define WM8741_SR_WIDTH 3 /* SR - [4:2] */
#define WM8741_MODESEL_MASK 0x0003 /* MODESEL - [1:0] */
#define WM8741_MODESEL_SHIFT 0 /* MODESEL - [1:0] */
#define WM8741_MODESEL_WIDTH 2 /* MODESEL - [1:0] */
/*
* R8 (0x08) - MODE_CONTROL_2
*/
#define WM8741_DSD_GAIN 0x0040 /* DSD_GAIN */
#define WM8741_DSD_GAIN_MASK 0x0040 /* DSD_GAIN */
#define WM8741_DSD_GAIN_SHIFT 6 /* DSD_GAIN */
#define WM8741_DSD_GAIN_WIDTH 1 /* DSD_GAIN */
#define WM8741_SDOUT 0x0020 /* SDOUT */
#define WM8741_SDOUT_MASK 0x0020 /* SDOUT */
#define WM8741_SDOUT_SHIFT 5 /* SDOUT */
#define WM8741_SDOUT_WIDTH 1 /* SDOUT */
#define WM8741_DOUT 0x0010 /* DOUT */
#define WM8741_DOUT_MASK 0x0010 /* DOUT */
#define WM8741_DOUT_SHIFT 4 /* DOUT */
#define WM8741_DOUT_WIDTH 1 /* DOUT */
#define WM8741_DIFF_MASK 0x000C /* DIFF - [3:2] */
#define WM8741_DIFF_SHIFT 2 /* DIFF - [3:2] */
#define WM8741_DIFF_WIDTH 2 /* DIFF - [3:2] */
#define WM8741_DITHER_MASK 0x0003 /* DITHER - [1:0] */
#define WM8741_DITHER_SHIFT 0 /* DITHER - [1:0] */
#define WM8741_DITHER_WIDTH 2 /* DITHER - [1:0] */
/*
* R32 (0x20) - ADDITONAL_CONTROL_1
*/
#define WM8741_DSD_LEVEL 0x0002 /* DSD_LEVEL */
#define WM8741_DSD_LEVEL_MASK 0x0002 /* DSD_LEVEL */
#define WM8741_DSD_LEVEL_SHIFT 1 /* DSD_LEVEL */
#define WM8741_DSD_LEVEL_WIDTH 1 /* DSD_LEVEL */
#define WM8741_DSD_NO_NOTCH 0x0001 /* DSD_NO_NOTCH */
#define WM8741_DSD_NO_NOTCH_MASK 0x0001 /* DSD_NO_NOTCH */
#define WM8741_DSD_NO_NOTCH_SHIFT 0 /* DSD_NO_NOTCH */
#define WM8741_DSD_NO_NOTCH_WIDTH 1 /* DSD_NO_NOTCH */
#define WM8741_SYSCLK 0
extern struct snd_soc_dai wm8741_dai;
extern struct snd_soc_codec_device soc_codec_dev_wm8741;
#endif

11
sound/soc/codecs/wm8750.c
View file

@ -884,6 +884,7 @@ static int wm8750_i2c_remove(struct i2c_client *client)
static const struct i2c_device_id wm8750_i2c_id[] = {
{ "wm8750", 0 },
{ "wm8987", 0 }, /* WM8987 is register compatible with WM8750 */
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8750_i2c_id);
@ -925,14 +926,22 @@ static int __devexit wm8750_spi_remove(struct spi_device *spi)
return 0;
}
static const struct spi_device_id wm8750_spi_id[] = {
{ "wm8750", 0 },
{ "wm8987", 0 },
{ }
};
MODULE_DEVICE_TABLE(spi, wm8750_spi_id);
static struct spi_driver wm8750_spi_driver = {
.driver = {
.name = "wm8750",
.name = "WM8750 SPI Codec",
.bus = &spi_bus_type,
.owner = THIS_MODULE,
},
.probe = wm8750_spi_probe,
.remove = __devexit_p(wm8750_spi_remove),
.id_table = wm8750_spi_id,
};
#endif

13
sound/soc/codecs/wm8904.c
View file

@ -2433,7 +2433,8 @@ static int wm8904_register(struct wm8904_priv *wm8904,
if (wm8904_codec) {
dev_err(codec->dev, "Another WM8904 is registered\n");
return -EINVAL;
ret = -EINVAL;
goto err;
}
mutex_init(&codec->mutex);
@ -2462,7 +2463,8 @@ static int wm8904_register(struct wm8904_priv *wm8904,
default:
dev_err(codec->dev, "Unknown device type %d\n",
wm8904->devtype);
return -EINVAL;
ret = -EINVAL;
goto err;
}
memcpy(codec->reg_cache, wm8904_reg, sizeof(wm8904_reg));
@ -2566,18 +2568,19 @@ static int wm8904_register(struct wm8904_priv *wm8904,
ret = snd_soc_register_codec(codec);
if (ret != 0) {
dev_err(codec->dev, "Failed to register codec: %d\n", ret);
return ret;
goto err_enable;
}
ret = snd_soc_register_dai(&wm8904_dai);
if (ret != 0) {
dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
snd_soc_unregister_codec(codec);
return ret;
goto err_codec;
}
return 0;
err_codec:
snd_soc_unregister_codec(codec);
err_enable:
regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
err_get:

7
sound/soc/codecs/wm8940.c
View file

@ -845,6 +845,7 @@ static void wm8940_unregister(struct wm8940_priv *wm8940)
static int wm8940_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
int ret;
struct wm8940_priv *wm8940;
struct snd_soc_codec *codec;
@ -858,7 +859,11 @@ static int wm8940_i2c_probe(struct i2c_client *i2c,
codec->control_data = i2c;
codec->dev = &i2c->dev;
return wm8940_register(wm8940, SND_SOC_I2C);
ret = wm8940_register(wm8940, SND_SOC_I2C);
if (ret < 0)
kfree(wm8940);
return ret;
}
static int __devexit wm8940_i2c_remove(struct i2c_client *client)

10
sound/soc/codecs/wm8955.c
View file

@ -964,7 +964,8 @@ static int wm8955_register(struct wm8955_priv *wm8955,
if (wm8955_codec) {
dev_err(codec->dev, "Another WM8955 is registered\n");
return -EINVAL;
ret = -EINVAL;
goto err;
}
mutex_init(&codec->mutex);
@ -1047,18 +1048,19 @@ static int wm8955_register(struct wm8955_priv *wm8955,
ret = snd_soc_register_codec(codec);
if (ret != 0) {
dev_err(codec->dev, "Failed to register codec: %d\n", ret);
return ret;
goto err_enable;
}
ret = snd_soc_register_dai(&wm8955_dai);
if (ret != 0) {
dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
snd_soc_unregister_codec(codec);
return ret;
goto err_codec;
}
return 0;
err_codec:
snd_soc_unregister_codec(codec);
err_enable:
regulator_bulk_disable(ARRAY_SIZE(wm8955->supplies), wm8955->supplies);
err_get:

99
sound/soc/codecs/wm8960.c
View file

@ -79,12 +79,13 @@ struct wm8960_priv {
struct snd_soc_dapm_widget *lout1;
struct snd_soc_dapm_widget *rout1;
struct snd_soc_dapm_widget *out3;
bool deemph;
int playback_fs;
};
#define wm8960_reset(c) snd_soc_write(c, WM8960_RESET, 0)
/* enumerated controls */
static const char *wm8960_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
static const char *wm8960_polarity[] = {"No Inversion", "Left Inverted",
"Right Inverted", "Stereo Inversion"};
static const char *wm8960_3d_upper_cutoff[] = {"High", "Low"};
@ -93,7 +94,6 @@ static const char *wm8960_alcfunc[] = {"Off", "Right", "Left", "Stereo"};
static const char *wm8960_alcmode[] = {"ALC", "Limiter"};
static const struct soc_enum wm8960_enum[] = {
SOC_ENUM_SINGLE(WM8960_DACCTL1, 1, 4, wm8960_deemph),
SOC_ENUM_SINGLE(WM8960_DACCTL1, 5, 4, wm8960_polarity),
SOC_ENUM_SINGLE(WM8960_DACCTL2, 5, 4, wm8960_polarity),
SOC_ENUM_SINGLE(WM8960_3D, 6, 2, wm8960_3d_upper_cutoff),
@ -102,6 +102,59 @@ static const struct soc_enum wm8960_enum[] = {
SOC_ENUM_SINGLE(WM8960_ALC3, 8, 2, wm8960_alcmode),
};
static const int deemph_settings[] = { 0, 32000, 44100, 48000 };
static int wm8960_set_deemph(struct snd_soc_codec *codec)
{
struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
int val, i, best;
/* If we're using deemphasis select the nearest available sample
* rate.
*/
if (wm8960->deemph) {
best = 1;
for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) {
if (abs(deemph_settings[i] - wm8960->playback_fs) <
abs(deemph_settings[best] - wm8960->playback_fs))
best = i;
}
val = best << 1;
} else {
val = 0;
}
dev_dbg(codec->dev, "Set deemphasis %d\n", val);
return snd_soc_update_bits(codec, WM8960_DACCTL1,
0x6, val);
}
static int wm8960_get_deemph(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
return wm8960->deemph;
}
static int wm8960_put_deemph(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
int deemph = ucontrol->value.enumerated.item[0];
if (deemph > 1)
return -EINVAL;
wm8960->deemph = deemph;
return wm8960_set_deemph(codec);
}
static const DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 50, 0);
static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 50, 1);
static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0);
@ -131,23 +184,24 @@ SOC_SINGLE("Speaker DC Volume", WM8960_CLASSD3, 3, 5, 0),
SOC_SINGLE("Speaker AC Volume", WM8960_CLASSD3, 0, 5, 0),
SOC_SINGLE("PCM Playback -6dB Switch", WM8960_DACCTL1, 7, 1, 0),
SOC_ENUM("ADC Polarity", wm8960_enum[1]),
SOC_ENUM("Playback De-emphasis", wm8960_enum[0]),
SOC_ENUM("ADC Polarity", wm8960_enum[0]),
SOC_SINGLE("ADC High Pass Filter Switch", WM8960_DACCTL1, 0, 1, 0),
SOC_ENUM("DAC Polarity", wm8960_enum[2]),
SOC_SINGLE_BOOL_EXT("DAC Deemphasis Switch", 0,
wm8960_get_deemph, wm8960_put_deemph),
SOC_ENUM("3D Filter Upper Cut-Off", wm8960_enum[3]),
SOC_ENUM("3D Filter Lower Cut-Off", wm8960_enum[4]),
SOC_ENUM("3D Filter Upper Cut-Off", wm8960_enum[2]),
SOC_ENUM("3D Filter Lower Cut-Off", wm8960_enum[3]),
SOC_SINGLE("3D Volume", WM8960_3D, 1, 15, 0),
SOC_SINGLE("3D Switch", WM8960_3D, 0, 1, 0),
SOC_ENUM("ALC Function", wm8960_enum[5]),
SOC_ENUM("ALC Function", wm8960_enum[4]),
SOC_SINGLE("ALC Max Gain", WM8960_ALC1, 4, 7, 0),
SOC_SINGLE("ALC Target", WM8960_ALC1, 0, 15, 1),
SOC_SINGLE("ALC Min Gain", WM8960_ALC2, 4, 7, 0),
SOC_SINGLE("ALC Hold Time", WM8960_ALC2, 0, 15, 0),
SOC_ENUM("ALC Mode", wm8960_enum[6]),
SOC_ENUM("ALC Mode", wm8960_enum[5]),
SOC_SINGLE("ALC Decay", WM8960_ALC3, 4, 15, 0),
SOC_SINGLE("ALC Attack", WM8960_ALC3, 0, 15, 0),
@ -433,6 +487,21 @@ static int wm8960_set_dai_fmt(struct snd_soc_dai *codec_dai,
return 0;
}
static struct {
int rate;
unsigned int val;
} alc_rates[] = {
{ 48000, 0 },
{ 44100, 0 },
{ 32000, 1 },
{ 22050, 2 },
{ 24000, 2 },
{ 16000, 3 },
{ 11250, 4 },
{ 12000, 4 },
{ 8000, 5 },
};
static int wm8960_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
@ -440,7 +509,9 @@ static int wm8960_hw_params(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->card->codec;
struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
u16 iface = snd_soc_read(codec, WM8960_IFACE1) & 0xfff3;
int i;
/* bit size */
switch (params_format(params)) {
@ -454,6 +525,18 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream,
break;
}
/* Update filters for the new rate */
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
wm8960->playback_fs = params_rate(params);
wm8960_set_deemph(codec);
} else {
for (i = 0; i < ARRAY_SIZE(alc_rates); i++)
if (alc_rates[i].rate == params_rate(params))
snd_soc_update_bits(codec,
WM8960_ADDCTL3, 0x7,
alc_rates[i].val);
}
/* set iface */
snd_soc_write(codec, WM8960_IFACE1, iface);
return 0;

9
sound/soc/codecs/wm8961.c
View file

@ -1102,7 +1102,7 @@ static int wm8961_register(struct wm8961_priv *wm8961)
ret = wm8961_reset(codec);
if (ret < 0) {
dev_err(codec->dev, "Failed to issue reset\n");
return ret;
goto err;
}
/* Enable class W */
@ -1147,18 +1147,19 @@ static int wm8961_register(struct wm8961_priv *wm8961)
ret = snd_soc_register_codec(codec);
if (ret != 0) {
dev_err(codec->dev, "Failed to register codec: %d\n", ret);
return ret;
goto err;
}
ret = snd_soc_register_dai(&wm8961_dai);
if (ret != 0) {
dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
snd_soc_unregister_codec(codec);
return ret;
goto err_codec;
}
return 0;
err_codec:
snd_soc_unregister_codec(codec);
err:
kfree(wm8961);
return ret;

3
sound/soc/codecs/wm8974.c
View file

@ -670,7 +670,8 @@ static __devinit int wm8974_register(struct wm8974_priv *wm8974)
if (wm8974_codec) {
dev_err(codec->dev, "Another WM8974 is registered\n");
return -EINVAL;
ret = -EINVAL;
goto err;
}
mutex_init(&codec->mutex);

10
sound/soc/codecs/wm8978.c
View file

@ -1076,7 +1076,6 @@ static __devinit int wm8978_register(struct wm8978_priv *wm8978)
err_codec:
snd_soc_unregister_codec(codec);
err:
kfree(wm8978);
return ret;
}
@ -1085,13 +1084,13 @@ static __devexit void wm8978_unregister(struct wm8978_priv *wm8978)
wm8978_set_bias_level(&wm8978->codec, SND_SOC_BIAS_OFF);
snd_soc_unregister_dai(&wm8978_dai);
snd_soc_unregister_codec(&wm8978->codec);
kfree(wm8978);
wm8978_codec = NULL;
}
static __devinit int wm8978_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
int ret;
struct wm8978_priv *wm8978;
struct snd_soc_codec *codec;
@ -1107,13 +1106,18 @@ static __devinit int wm8978_i2c_probe(struct i2c_client *i2c,
codec->dev = &i2c->dev;
return wm8978_register(wm8978);
ret = wm8978_register(wm8978);
if (ret < 0)
kfree(wm8978);
return ret;
}
static __devexit int wm8978_i2c_remove(struct i2c_client *client)
{
struct wm8978_priv *wm8978 = i2c_get_clientdata(client);
wm8978_unregister(wm8978);
kfree(wm8978);
return 0;
}

4
sound/soc/codecs/wm8990.c
View file

@ -30,8 +30,6 @@
#include "wm8990.h"
#define WM8990_VERSION "0.2"
/* codec private data */
struct wm8990_priv {
unsigned int sysclk;
@ -1511,8 +1509,6 @@ static int wm8990_probe(struct platform_device *pdev)
struct wm8990_priv *wm8990;
int ret;
pr_info("WM8990 Audio Codec %s\n", WM8990_VERSION);
setup = socdev->codec_data;
codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
if (codec == NULL)

75
sound/soc/codecs/wm8994.c
View file

@ -1677,6 +1677,26 @@ static struct {
static int wm8994_readable(unsigned int reg)
{
switch (reg) {
case WM8994_GPIO_1:
case WM8994_GPIO_2:
case WM8994_GPIO_3:
case WM8994_GPIO_4:
case WM8994_GPIO_5:
case WM8994_GPIO_6:
case WM8994_GPIO_7:
case WM8994_GPIO_8:
case WM8994_GPIO_9:
case WM8994_GPIO_10:
case WM8994_GPIO_11:
case WM8994_INTERRUPT_STATUS_1:
case WM8994_INTERRUPT_STATUS_2:
case WM8994_INTERRUPT_RAW_STATUS_2:
return 1;
default:
break;
}
if (reg >= ARRAY_SIZE(access_masks))
return 0;
return access_masks[reg].readable != 0;
@ -2341,6 +2361,20 @@ SOC_DAPM_SINGLE("AIF2 Switch", WM8994_AIF1_ADC1_RIGHT_MIXER_ROUTING,
0, 1, 0),
};
static const struct snd_kcontrol_new aif1adc2l_mix[] = {
SOC_DAPM_SINGLE("DMIC Switch", WM8994_AIF1_ADC2_LEFT_MIXER_ROUTING,
1, 1, 0),
SOC_DAPM_SINGLE("AIF2 Switch", WM8994_AIF1_ADC2_LEFT_MIXER_ROUTING,
0, 1, 0),
};
static const struct snd_kcontrol_new aif1adc2r_mix[] = {
SOC_DAPM_SINGLE("DMIC Switch", WM8994_AIF1_ADC2_RIGHT_MIXER_ROUTING,
1, 1, 0),
SOC_DAPM_SINGLE("AIF2 Switch", WM8994_AIF1_ADC2_RIGHT_MIXER_ROUTING,
0, 1, 0),
};
static const struct snd_kcontrol_new aif2dac2l_mix[] = {
SOC_DAPM_SINGLE("Right Sidetone Switch", WM8994_DAC2_LEFT_MIXER_ROUTING,
5, 1, 0),
@ -2472,6 +2506,7 @@ static const struct snd_kcontrol_new aif3adc_mux =
static const struct snd_soc_dapm_widget wm8994_dapm_widgets[] = {
SND_SOC_DAPM_INPUT("DMIC1DAT"),
SND_SOC_DAPM_INPUT("DMIC2DAT"),
SND_SOC_DAPM_INPUT("Clock"),
SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
@ -2506,6 +2541,11 @@ SND_SOC_DAPM_MIXER("AIF1ADC1L Mixer", SND_SOC_NOPM, 0, 0,
SND_SOC_DAPM_MIXER("AIF1ADC1R Mixer", SND_SOC_NOPM, 0, 0,
aif1adc1r_mix, ARRAY_SIZE(aif1adc1r_mix)),
SND_SOC_DAPM_MIXER("AIF1ADC2L Mixer", SND_SOC_NOPM, 0, 0,
aif1adc2l_mix, ARRAY_SIZE(aif1adc2l_mix)),
SND_SOC_DAPM_MIXER("AIF1ADC2R Mixer", SND_SOC_NOPM, 0, 0,
aif1adc2r_mix, ARRAY_SIZE(aif1adc2r_mix)),
SND_SOC_DAPM_MIXER("AIF2DAC2L Mixer", SND_SOC_NOPM, 0, 0,
aif2dac2l_mix, ARRAY_SIZE(aif2dac2l_mix)),
SND_SOC_DAPM_MIXER("AIF2DAC2R Mixer", SND_SOC_NOPM, 0, 0,
@ -2668,6 +2708,14 @@ static const struct snd_soc_dapm_route intercon[] = {
{ "AIF1ADC1R Mixer", "ADC/DMIC Switch", "ADCR Mux" },
{ "AIF1ADC1R Mixer", "AIF2 Switch", "AIF2DACR" },
{ "AIF1ADC2L", NULL, "AIF1ADC2L Mixer" },
{ "AIF1ADC2L Mixer", "DMIC Switch", "DMIC2L" },
{ "AIF1ADC2L Mixer", "AIF2 Switch", "AIF2DACL" },
{ "AIF1ADC2R", NULL, "AIF1ADC2R Mixer" },
{ "AIF1ADC2R Mixer", "DMIC Switch", "DMIC2R" },
{ "AIF1ADC2R Mixer", "AIF2 Switch", "AIF2DACR" },
/* Pin level routing for AIF3 */
{ "AIF1DAC1L", NULL, "AIF1DAC Mux" },
{ "AIF1DAC1R", NULL, "AIF1DAC Mux" },
@ -2946,11 +2994,14 @@ static int wm8994_set_fll(struct snd_soc_dai *dai, int id, int src,
return 0;
}
static int opclk_divs[] = { 10, 20, 30, 40, 55, 60, 80, 120, 160 };
static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai,
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_codec *codec = dai->codec;
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
int i;
switch (dai->id) {
case 1:
@ -2988,6 +3039,25 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai,
dev_dbg(dai->dev, "AIF%d using FLL2\n", dai->id);
break;
case WM8994_SYSCLK_OPCLK:
/* Special case - a division (times 10) is given and
* no effect on main clocking.
*/
if (freq) {
for (i = 0; i < ARRAY_SIZE(opclk_divs); i++)
if (opclk_divs[i] == freq)
break;
if (i == ARRAY_SIZE(opclk_divs))
return -EINVAL;
snd_soc_update_bits(codec, WM8994_CLOCKING_2,
WM8994_OPCLK_DIV_MASK, i);
snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_2,
WM8994_OPCLK_ENA, WM8994_OPCLK_ENA);
} else {
snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_2,
WM8994_OPCLK_ENA, 0);
}
default:
return -EINVAL;
}
@ -4004,6 +4074,11 @@ static int wm8994_codec_probe(struct platform_device *pdev)
1 << WM8994_AIF2DAC_3D_GAIN_SHIFT,
1 << WM8994_AIF2DAC_3D_GAIN_SHIFT);
/* Unconditionally enable AIF1 ADC TDM mode; it only affects
* behaviour on idle TDM clock cycles. */
snd_soc_update_bits(codec, WM8994_AIF1_CONTROL_1,
WM8994_AIF1ADC_TDM, WM8994_AIF1ADC_TDM);
wm8994_update_class_w(codec);
ret = snd_soc_register_codec(codec);

3
sound/soc/codecs/wm8994.h
View file

@ -20,6 +20,9 @@ extern struct snd_soc_dai wm8994_dai[];
#define WM8994_SYSCLK_FLL1 3
#define WM8994_SYSCLK_FLL2 4
/* OPCLK is also configured with set_dai_sysclk, specify division*10 as rate. */
#define WM8994_SYSCLK_OPCLK 5
#define WM8994_FLL1 1
#define WM8994_FLL2 2

11
sound/soc/codecs/wm9081.c
View file

@ -1356,7 +1356,7 @@ static int wm9081_register(struct wm9081_priv *wm9081,
ret = snd_soc_codec_set_cache_io(codec, 8, 16, control);
if (ret != 0) {
dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
return ret;
goto err;
}
reg = snd_soc_read(codec, WM9081_SOFTWARE_RESET);
@ -1369,7 +1369,7 @@ static int wm9081_register(struct wm9081_priv *wm9081,
ret = wm9081_reset(codec);
if (ret < 0) {
dev_err(codec->dev, "Failed to issue reset\n");
return ret;
goto err;
}
wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
@ -1388,18 +1388,19 @@ static int wm9081_register(struct wm9081_priv *wm9081,
ret = snd_soc_register_codec(codec);
if (ret != 0) {
dev_err(codec->dev, "Failed to register codec: %d\n", ret);
return ret;
goto err;
}
ret = snd_soc_register_dai(&wm9081_dai);
if (ret != 0) {
dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
snd_soc_unregister_codec(codec);
return ret;
goto err_codec;
}
return 0;
err_codec:
snd_soc_unregister_codec(codec);
err:
kfree(wm9081);
return ret;

2
sound/soc/codecs/wm_hubs.c
View file

@ -410,6 +410,8 @@ static int hp_event(struct snd_soc_dapm_widget *w,
WM8993_HPOUT1L_DLY |
WM8993_HPOUT1R_DLY, 0);
snd_soc_write(codec, WM8993_DC_SERVO_0, 0);
snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1,
WM8993_HPOUT1L_ENA | WM8993_HPOUT1R_ENA,
0);

163
sound/soc/davinci/davinci-i2s.c
View file

@ -26,6 +26,7 @@
#include <mach/asp.h>
#include "davinci-pcm.h"
#include "davinci-i2s.h"
/*
@ -68,16 +69,21 @@
#define DAVINCI_MCBSP_RCR_RDATDLY(v) ((v) << 16)
#define DAVINCI_MCBSP_RCR_RFIG (1 << 18)
#define DAVINCI_MCBSP_RCR_RWDLEN2(v) ((v) << 21)
#define DAVINCI_MCBSP_RCR_RFRLEN2(v) ((v) << 24)
#define DAVINCI_MCBSP_RCR_RPHASE BIT(31)
#define DAVINCI_MCBSP_XCR_XWDLEN1(v) ((v) << 5)
#define DAVINCI_MCBSP_XCR_XFRLEN1(v) ((v) << 8)
#define DAVINCI_MCBSP_XCR_XDATDLY(v) ((v) << 16)
#define DAVINCI_MCBSP_XCR_XFIG (1 << 18)
#define DAVINCI_MCBSP_XCR_XWDLEN2(v) ((v) << 21)
#define DAVINCI_MCBSP_XCR_XFRLEN2(v) ((v) << 24)
#define DAVINCI_MCBSP_XCR_XPHASE BIT(31)
#define DAVINCI_MCBSP_SRGR_FWID(v) ((v) << 8)
#define DAVINCI_MCBSP_SRGR_FPER(v) ((v) << 16)
#define DAVINCI_MCBSP_SRGR_FSGM (1 << 28)
#define DAVINCI_MCBSP_SRGR_CLKSM BIT(29)
#define DAVINCI_MCBSP_PCR_CLKRP (1 << 0)
#define DAVINCI_MCBSP_PCR_CLKXP (1 << 1)
@ -116,6 +122,7 @@ static const unsigned char double_fmt[SNDRV_PCM_FORMAT_S32_LE + 1] = {
};
struct davinci_mcbsp_dev {
struct device *dev;
struct davinci_pcm_dma_params dma_params[2];
void __iomem *base;
#define MOD_DSP_A 0
@ -144,6 +151,11 @@ struct davinci_mcbsp_dev {
* won't end up being swapped because of the underrun.
*/
unsigned enable_channel_combine:1;
unsigned int fmt;
int clk_div;
int clk_input_pin;
bool i2s_accurate_sck;
};
static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev,
@ -254,10 +266,12 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
unsigned int pcr;
unsigned int srgr;
/* Attention srgr is updated by hw_params! */
srgr = DAVINCI_MCBSP_SRGR_FSGM |
DAVINCI_MCBSP_SRGR_FPER(DEFAULT_BITPERSAMPLE * 2 - 1) |
DAVINCI_MCBSP_SRGR_FWID(DEFAULT_BITPERSAMPLE - 1);
dev->fmt = fmt;
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBS_CFS:
@ -268,11 +282,26 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
DAVINCI_MCBSP_PCR_CLKRM;
break;
case SND_SOC_DAIFMT_CBM_CFS:
/* McBSP CLKR pin is the input for the Sample Rate Generator.
* McBSP FSR and FSX are driven by the Sample Rate Generator. */
pcr = DAVINCI_MCBSP_PCR_SCLKME |
DAVINCI_MCBSP_PCR_FSXM |
DAVINCI_MCBSP_PCR_FSRM;
pcr = DAVINCI_MCBSP_PCR_FSRM | DAVINCI_MCBSP_PCR_FSXM;
/*
* Selection of the clock input pin that is the
* input for the Sample Rate Generator.
* McBSP FSR and FSX are driven by the Sample Rate
* Generator.
*/
switch (dev->clk_input_pin) {
case MCBSP_CLKS:
pcr |= DAVINCI_MCBSP_PCR_CLKXM |
DAVINCI_MCBSP_PCR_CLKRM;
break;
case MCBSP_CLKR:
pcr |= DAVINCI_MCBSP_PCR_SCLKME;
break;
default:
dev_err(dev->dev, "bad clk_input_pin\n");
return -EINVAL;
}
break;
case SND_SOC_DAIFMT_CBM_CFM:
/* codec is master */
@ -372,6 +401,18 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
return 0;
}
static int davinci_i2s_dai_set_clkdiv(struct snd_soc_dai *cpu_dai,
int div_id, int div)
{
struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
if (div_id != DAVINCI_MCBSP_CLKGDV)
return -ENODEV;
dev->clk_div = div;
return 0;
}
static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
@ -380,8 +421,8 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
struct davinci_pcm_dma_params *dma_params =
&dev->dma_params[substream->stream];
struct snd_interval *i = NULL;
int mcbsp_word_length;
unsigned int rcr, xcr, srgr;
int mcbsp_word_length, master;
unsigned int rcr, xcr, srgr, clk_div, freq, framesize;
u32 spcr;
snd_pcm_format_t fmt;
unsigned element_cnt = 1;
@ -396,12 +437,59 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr);
}
i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS);
srgr = DAVINCI_MCBSP_SRGR_FSGM;
srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1);
master = dev->fmt & SND_SOC_DAIFMT_MASTER_MASK;
fmt = params_format(params);
mcbsp_word_length = asp_word_length[fmt];
i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS);
srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1);
switch (master) {
case SND_SOC_DAIFMT_CBS_CFS:
freq = clk_get_rate(dev->clk);
srgr = DAVINCI_MCBSP_SRGR_FSGM |
DAVINCI_MCBSP_SRGR_CLKSM;
srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length *
8 - 1);
if (dev->i2s_accurate_sck) {
clk_div = 256;
do {
framesize = (freq / (--clk_div)) /
params->rate_num *
params->rate_den;
} while (((framesize < 33) || (framesize > 4095)) &&
(clk_div));
clk_div--;
srgr |= DAVINCI_MCBSP_SRGR_FPER(framesize - 1);
} else {
/* symmetric waveforms */
clk_div = freq / (mcbsp_word_length * 16) /
params->rate_num * params->rate_den;
srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length *
16 - 1);
}
clk_div &= 0xFF;
srgr |= clk_div;
break;
case SND_SOC_DAIFMT_CBM_CFS:
srgr = DAVINCI_MCBSP_SRGR_FSGM;
clk_div = dev->clk_div - 1;
srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * 8 - 1);
srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * 16 - 1);
clk_div &= 0xFF;
srgr |= clk_div;
break;
case SND_SOC_DAIFMT_CBM_CFM:
/* Clock and frame sync given from external sources */
i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS);
srgr = DAVINCI_MCBSP_SRGR_FSGM;
srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1);
pr_debug("%s - %d FWID set: re-read srgr = %X\n",
__func__, __LINE__, snd_interval_value(i) - 1);
i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS);
srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1);
break;
default:
return -EINVAL;
}
davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr);
rcr = DAVINCI_MCBSP_RCR_RFIG;
@ -426,12 +514,41 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
element_cnt = 1;
fmt = double_fmt[fmt];
}
switch (master) {
case SND_SOC_DAIFMT_CBS_CFS:
case SND_SOC_DAIFMT_CBS_CFM:
rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(0);
xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(0);
rcr |= DAVINCI_MCBSP_RCR_RPHASE;
xcr |= DAVINCI_MCBSP_XCR_XPHASE;
break;
case SND_SOC_DAIFMT_CBM_CFM:
case SND_SOC_DAIFMT_CBM_CFS:
rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(element_cnt - 1);
xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(element_cnt - 1);
break;
default:
return -EINVAL;
}
}
dma_params->acnt = dma_params->data_type = data_type[fmt];
dma_params->fifo_level = 0;
mcbsp_word_length = asp_word_length[fmt];
rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1);
xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1);
switch (master) {
case SND_SOC_DAIFMT_CBS_CFS:
case SND_SOC_DAIFMT_CBS_CFM:
rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(0);
xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(0);
break;
case SND_SOC_DAIFMT_CBM_CFM:
case SND_SOC_DAIFMT_CBM_CFS:
rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1);
xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1);
break;
default:
return -EINVAL;
}
rcr |= DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) |
DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length);
@ -442,6 +559,10 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, xcr);
else
davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, rcr);
pr_debug("%s - %d srgr=%X\n", __func__, __LINE__, srgr);
pr_debug("%s - %d xcr=%X\n", __func__, __LINE__, xcr);
pr_debug("%s - %d rcr=%X\n", __func__, __LINE__, rcr);
return 0;
}
@ -500,6 +621,7 @@ static struct snd_soc_dai_ops davinci_i2s_dai_ops = {
.trigger = davinci_i2s_trigger,
.hw_params = davinci_i2s_hw_params,
.set_fmt = davinci_i2s_set_dai_fmt,
.set_clkdiv = davinci_i2s_dai_set_clkdiv,
};
@ -526,6 +648,8 @@ static int davinci_i2s_probe(struct platform_device *pdev)
struct snd_platform_data *pdata = pdev->dev.platform_data;
struct davinci_mcbsp_dev *dev;
struct resource *mem, *ioarea, *res;
enum dma_event_q asp_chan_q = EVENTQ_0;
enum dma_event_q ram_chan_q = EVENTQ_1;
int ret;
mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
@ -552,7 +676,17 @@ static int davinci_i2s_probe(struct platform_device *pdev)
pdata->sram_size_playback;
dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size =
pdata->sram_size_capture;
dev->clk_input_pin = pdata->clk_input_pin;
dev->i2s_accurate_sck = pdata->i2s_accurate_sck;
asp_chan_q = pdata->asp_chan_q;
ram_chan_q = pdata->ram_chan_q;
}
dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].asp_chan_q = asp_chan_q;
dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].ram_chan_q = ram_chan_q;
dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].asp_chan_q = asp_chan_q;
dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].ram_chan_q = ram_chan_q;
dev->clk = clk_get(&pdev->dev, NULL);
if (IS_ERR(dev->clk)) {
ret = -ENODEV;
@ -584,6 +718,7 @@ static int davinci_i2s_probe(struct platform_device *pdev)
goto err_free_mem;
}
dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start;
dev->dev = &pdev->dev;
davinci_i2s_dai.private_data = dev;
davinci_i2s_dai.capture.dma_data = dev->dma_params;

5
sound/soc/davinci/davinci-i2s.h
View file

@ -12,6 +12,11 @@
#ifndef _DAVINCI_I2S_H
#define _DAVINCI_I2S_H
/* McBSP dividers */
enum davinci_mcbsp_div {
DAVINCI_MCBSP_CLKGDV, /* Sample rate generator divider */
};
extern struct snd_soc_dai davinci_i2s_dai;
#endif

6
sound/soc/davinci/davinci-mcasp.c
View file

@ -890,7 +890,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
dev->rxnumevt = pdata->rxnumevt;
dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
dma_data->eventq_no = pdata->eventq_no;
dma_data->asp_chan_q = pdata->asp_chan_q;
dma_data->ram_chan_q = pdata->ram_chan_q;
dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset +
io_v2p(dev->base));
@ -904,7 +905,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
dma_data->channel = res->start;
dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE];
dma_data->eventq_no = pdata->eventq_no;
dma_data->asp_chan_q = pdata->asp_chan_q;
dma_data->ram_chan_q = pdata->ram_chan_q;
dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset +
io_v2p(dev->base));

7
sound/soc/davinci/davinci-pcm.c
View file

@ -381,7 +381,7 @@ static int request_ping_pong(struct snd_pcm_substream *substream,
/* Request ram master channel */
link = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY,
davinci_pcm_dma_irq, substream,
EVENTQ_1);
prtd->params->ram_chan_q);
if (link < 0)
goto exit1;
@ -477,7 +477,8 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
/* Request asp master DMA channel */
link = prtd->asp_channel = edma_alloc_channel(params->channel,
davinci_pcm_dma_irq, substream, EVENTQ_0);
davinci_pcm_dma_irq, substream,
prtd->params->asp_chan_q);
if (link < 0)
goto exit1;
@ -800,7 +801,7 @@ static void davinci_pcm_free(struct snd_pcm *pcm)
dma_free_writecombine(pcm->card->dev, buf->bytes,
buf->area, buf->addr);
buf->area = NULL;
iram_dma = (struct snd_dma_buffer *)buf->private_data;
iram_dma = buf->private_data;
if (iram_dma) {
sram_free(iram_dma->area, iram_dma->bytes);
kfree(iram_dma);

3
sound/soc/davinci/davinci-pcm.h
View file

@ -21,7 +21,8 @@ struct davinci_pcm_dma_params {
unsigned short acnt;
dma_addr_t dma_addr; /* device physical address for DMA */
unsigned sram_size;
enum dma_event_q eventq_no; /* event queue number */
enum dma_event_q asp_chan_q; /* event queue number for ASP channel */
enum dma_event_q ram_chan_q; /* event queue number for RAM channel */
unsigned char data_type; /* xfer data type */
unsigned char convert_mono_stereo;
unsigned int fifo_level;

2
sound/soc/davinci/davinci-vcif.c
View file

@ -203,7 +203,7 @@ static int davinci_vcif_probe(struct platform_device *pdev)
int ret;
davinci_vcif_dev = kzalloc(sizeof(struct davinci_vcif_dev), GFP_KERNEL);
if (!davinci_vc) {
if (!davinci_vcif_dev) {
dev_dbg(&pdev->dev,
"could not allocate memory for private data\n");
return -ENOMEM;

18
sound/soc/ep93xx/Kconfig Normal file
View file

@ -0,0 +1,18 @@
config SND_EP93XX_SOC
tristate "SoC Audio support for the Cirrus Logic EP93xx series"
depends on ARCH_EP93XX && SND_SOC
help
Say Y or M if you want to add support for codecs attached to
the EP93xx I2S interface.
config SND_EP93XX_SOC_I2S
tristate
config SND_EP93XX_SOC_SNAPPERCL15
tristate "SoC Audio support for Bluewater Systems Snapper CL15 module"
depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15
select SND_EP93XX_SOC_I2S
select SND_SOC_TLV320AIC23
help
Say Y or M here if you want to add support for I2S audio on the
Bluewater Systems Snapper CL15 module.

11
sound/soc/ep93xx/Makefile Normal file
View file

@ -0,0 +1,11 @@
# EP93xx Platform Support
snd-soc-ep93xx-objs := ep93xx-pcm.o
snd-soc-ep93xx-i2s-objs := ep93xx-i2s.o
obj-$(CONFIG_SND_EP93XX_SOC) += snd-soc-ep93xx.o
obj-$(CONFIG_SND_EP93XX_SOC_I2S) += snd-soc-ep93xx-i2s.o
# EP93XX Machine Support
snd-soc-snappercl15-objs := snappercl15.o
obj-$(CONFIG_SND_EP93XX_SOC_SNAPPERCL15) += snd-soc-snappercl15.o

487
sound/soc/ep93xx/ep93xx-i2s.c Normal file
View file

@ -0,0 +1,487 @@
/*
* linux/sound/soc/ep93xx-i2s.c
* EP93xx I2S driver
*
* Copyright (C) 2010 Ryan Mallon <ryan@bluewatersys.com>
*
* Based on the original driver by:
* Copyright (C) 2007 Chase Douglas <chasedouglas@gmail>
* Copyright (C) 2006 Lennert Buytenhek <buytenh@wantstofly.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.
*
*/
#include <linux/module.h>
#include <linux/init.h>
#include <linux/slab.h>
#include <linux/clk.h>
#include <linux/io.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/initval.h>
#include <sound/soc.h>
#include <mach/hardware.h>
#include <mach/ep93xx-regs.h>
#include <mach/dma.h>
#include "ep93xx-pcm.h"
#include "ep93xx-i2s.h"
#define EP93XX_I2S_TXCLKCFG 0x00
#define EP93XX_I2S_RXCLKCFG 0x04
#define EP93XX_I2S_GLCTRL 0x0C
#define EP93XX_I2S_TXLINCTRLDATA 0x28
#define EP93XX_I2S_TXCTRL 0x2C
#define EP93XX_I2S_TXWRDLEN 0x30
#define EP93XX_I2S_TX0EN 0x34
#define EP93XX_I2S_RXLINCTRLDATA 0x58
#define EP93XX_I2S_RXCTRL 0x5C
#define EP93XX_I2S_RXWRDLEN 0x60
#define EP93XX_I2S_RX0EN 0x64
#define EP93XX_I2S_WRDLEN_16 (0 << 0)
#define EP93XX_I2S_WRDLEN_24 (1 << 0)
#define EP93XX_I2S_WRDLEN_32 (2 << 0)
#define EP93XX_I2S_LINCTRLDATA_R_JUST (1 << 2) /* Right justify */
#define EP93XX_I2S_CLKCFG_LRS (1 << 0) /* lrclk polarity */
#define EP93XX_I2S_CLKCFG_CKP (1 << 1) /* Bit clock polarity */
#define EP93XX_I2S_CLKCFG_REL (1 << 2) /* First bit transition */
#define EP93XX_I2S_CLKCFG_MASTER (1 << 3) /* Master mode */
#define EP93XX_I2S_CLKCFG_NBCG (1 << 4) /* Not bit clock gating */
struct ep93xx_i2s_info {
struct clk *mclk;
struct clk *sclk;
struct clk *lrclk;
struct ep93xx_pcm_dma_params *dma_params;
struct resource *mem;
void __iomem *regs;
};
struct ep93xx_pcm_dma_params ep93xx_i2s_dma_params[] = {
[SNDRV_PCM_STREAM_PLAYBACK] = {
.name = "i2s-pcm-out",
.dma_port = EP93XX_DMA_M2P_PORT_I2S1,
},
[SNDRV_PCM_STREAM_CAPTURE] = {
.name = "i2s-pcm-in",
.dma_port = EP93XX_DMA_M2P_PORT_I2S1,
},
};
static inline void ep93xx_i2s_write_reg(struct ep93xx_i2s_info *info,
unsigned reg, unsigned val)
{
__raw_writel(val, info->regs + reg);
}
static inline unsigned ep93xx_i2s_read_reg(struct ep93xx_i2s_info *info,
unsigned reg)
{
return __raw_readl(info->regs + reg);
}
static void ep93xx_i2s_enable(struct ep93xx_i2s_info *info, int stream)
{
unsigned base_reg;
int i;
if ((ep93xx_i2s_read_reg(info, EP93XX_I2S_TX0EN) & 0x1) == 0 &&
(ep93xx_i2s_read_reg(info, EP93XX_I2S_RX0EN) & 0x1) == 0) {
/* Enable clocks */
clk_enable(info->mclk);
clk_enable(info->sclk);
clk_enable(info->lrclk);
/* Enable i2s */
ep93xx_i2s_write_reg(info, EP93XX_I2S_GLCTRL, 1);
}
/* Enable fifos */
if (stream == SNDRV_PCM_STREAM_PLAYBACK)
base_reg = EP93XX_I2S_TX0EN;
else
base_reg = EP93XX_I2S_RX0EN;
for (i = 0; i < 3; i++)
ep93xx_i2s_write_reg(info, base_reg + (i * 4), 1);
}
static void ep93xx_i2s_disable(struct ep93xx_i2s_info *info, int stream)
{
unsigned base_reg;
int i;
/* Disable fifos */
if (stream == SNDRV_PCM_STREAM_PLAYBACK)
base_reg = EP93XX_I2S_TX0EN;
else
base_reg = EP93XX_I2S_RX0EN;
for (i = 0; i < 3; i++)
ep93xx_i2s_write_reg(info, base_reg + (i * 4), 0);
if ((ep93xx_i2s_read_reg(info, EP93XX_I2S_TX0EN) & 0x1) == 0 &&
(ep93xx_i2s_read_reg(info, EP93XX_I2S_RX0EN) & 0x1) == 0) {
/* Disable i2s */
ep93xx_i2s_write_reg(info, EP93XX_I2S_GLCTRL, 0);
/* Disable clocks */
clk_disable(info->lrclk);
clk_disable(info->sclk);
clk_disable(info->mclk);
}
}
static int ep93xx_i2s_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
struct ep93xx_i2s_info *info = rtd->dai->cpu_dai->private_data;
snd_soc_dai_set_dma_data(cpu_dai, substream,
&info->dma_params[substream->stream]);
return 0;
}
static void ep93xx_i2s_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct ep93xx_i2s_info *info = rtd->dai->cpu_dai->private_data;
ep93xx_i2s_disable(info, substream->stream);
}
static int ep93xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
unsigned int fmt)
{
struct ep93xx_i2s_info *info = cpu_dai->private_data;
unsigned int clk_cfg, lin_ctrl;
clk_cfg = ep93xx_i2s_read_reg(info, EP93XX_I2S_RXCLKCFG);
lin_ctrl = ep93xx_i2s_read_reg(info, EP93XX_I2S_RXLINCTRLDATA);
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
clk_cfg |= EP93XX_I2S_CLKCFG_REL;
lin_ctrl &= ~EP93XX_I2S_LINCTRLDATA_R_JUST;
break;
case SND_SOC_DAIFMT_LEFT_J:
clk_cfg &= ~EP93XX_I2S_CLKCFG_REL;
lin_ctrl &= ~EP93XX_I2S_LINCTRLDATA_R_JUST;
break;
case SND_SOC_DAIFMT_RIGHT_J:
clk_cfg &= ~EP93XX_I2S_CLKCFG_REL;
lin_ctrl |= EP93XX_I2S_LINCTRLDATA_R_JUST;
break;
default:
return -EINVAL;
}
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBS_CFS:
/* CPU is master */
clk_cfg |= EP93XX_I2S_CLKCFG_MASTER;
break;
case SND_SOC_DAIFMT_CBM_CFM:
/* Codec is master */
clk_cfg &= ~EP93XX_I2S_CLKCFG_MASTER;
break;
default:
return -EINVAL;
}
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
/* Negative bit clock, lrclk low on left word */
clk_cfg &= ~(EP93XX_I2S_CLKCFG_CKP | EP93XX_I2S_CLKCFG_REL);
break;
case SND_SOC_DAIFMT_NB_IF:
/* Negative bit clock, lrclk low on right word */
clk_cfg &= ~EP93XX_I2S_CLKCFG_CKP;
clk_cfg |= EP93XX_I2S_CLKCFG_REL;
break;
case SND_SOC_DAIFMT_IB_NF:
/* Positive bit clock, lrclk low on left word */
clk_cfg |= EP93XX_I2S_CLKCFG_CKP;
clk_cfg &= ~EP93XX_I2S_CLKCFG_REL;
break;
case SND_SOC_DAIFMT_IB_IF:
/* Positive bit clock, lrclk low on right word */
clk_cfg |= EP93XX_I2S_CLKCFG_CKP | EP93XX_I2S_CLKCFG_REL;
break;
}
/* Write new register values */
ep93xx_i2s_write_reg(info, EP93XX_I2S_RXCLKCFG, clk_cfg);
ep93xx_i2s_write_reg(info, EP93XX_I2S_TXCLKCFG, clk_cfg);
ep93xx_i2s_write_reg(info, EP93XX_I2S_RXLINCTRLDATA, lin_ctrl);
ep93xx_i2s_write_reg(info, EP93XX_I2S_TXLINCTRLDATA, lin_ctrl);
return 0;
}
static int ep93xx_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
struct ep93xx_i2s_info *info = cpu_dai->private_data;
unsigned word_len, div, sdiv, lrdiv;
int found = 0, err;
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
word_len = EP93XX_I2S_WRDLEN_16;
break;
case SNDRV_PCM_FORMAT_S24_LE:
word_len = EP93XX_I2S_WRDLEN_24;
break;
case SNDRV_PCM_FORMAT_S32_LE:
word_len = EP93XX_I2S_WRDLEN_32;
break;
default:
return -EINVAL;
}
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
ep93xx_i2s_write_reg(info, EP93XX_I2S_TXWRDLEN, word_len);
else
ep93xx_i2s_write_reg(info, EP93XX_I2S_RXWRDLEN, word_len);
/*
* Calculate the sdiv (bit clock) and lrdiv (left/right clock) values.
* If the lrclk is pulse length is larger than the word size, then the
* bit clock will be gated for the unused bits.
*/
div = (clk_get_rate(info->mclk) / params_rate(params)) *
params_channels(params);
for (sdiv = 2; sdiv <= 4; sdiv += 2)
for (lrdiv = 32; lrdiv <= 128; lrdiv <<= 1)
if (sdiv * lrdiv == div) {
found = 1;
goto out;
}
out:
if (!found)
return -EINVAL;
err = clk_set_rate(info->sclk, clk_get_rate(info->mclk) / sdiv);
if (err)
return err;
err = clk_set_rate(info->lrclk, clk_get_rate(info->sclk) / lrdiv);
if (err)
return err;
ep93xx_i2s_enable(info, substream->stream);
return 0;
}
static int ep93xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id,
unsigned int freq, int dir)
{
struct ep93xx_i2s_info *info = cpu_dai->private_data;
if (dir == SND_SOC_CLOCK_IN || clk_id != 0)
return -EINVAL;
return clk_set_rate(info->mclk, freq);
}
#ifdef CONFIG_PM
static int ep93xx_i2s_suspend(struct snd_soc_dai *dai)
{
struct ep93xx_i2s_info *info = dai->private_data;
if (!dai->active)
return;
ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_PLAYBACK);
ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_CAPTURE);
}
static int ep93xx_i2s_resume(struct snd_soc_dai *dai)
{
struct ep93xx_i2s_info *info = dai->private_data;
if (!dai->active)
return;
ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_PLAYBACK);
ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_CAPTURE);
}
#else
#define ep93xx_i2s_suspend NULL
#define ep93xx_i2s_resume NULL
#endif
static struct snd_soc_dai_ops ep93xx_i2s_dai_ops = {
.startup = ep93xx_i2s_startup,
.shutdown = ep93xx_i2s_shutdown,
.hw_params = ep93xx_i2s_hw_params,
.set_sysclk = ep93xx_i2s_set_sysclk,
.set_fmt = ep93xx_i2s_set_dai_fmt,
};
#define EP93XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_S24_LE | \
SNDRV_PCM_FMTBIT_S32_LE)
struct snd_soc_dai ep93xx_i2s_dai = {
.name = "ep93xx-i2s",
.id = 0,
.symmetric_rates= 1,
.suspend = ep93xx_i2s_suspend,
.resume = ep93xx_i2s_resume,
.playback = {
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = EP93XX_I2S_FORMATS,
},
.capture = {
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = EP93XX_I2S_FORMATS,
},
.ops = &ep93xx_i2s_dai_ops,
};
EXPORT_SYMBOL_GPL(ep93xx_i2s_dai);
static int ep93xx_i2s_probe(struct platform_device *pdev)
{
struct ep93xx_i2s_info *info;
struct resource *res;
int err;
info = kzalloc(sizeof(struct ep93xx_i2s_info), GFP_KERNEL);
if (!info) {
err = -ENOMEM;
goto fail;
}
ep93xx_i2s_dai.dev = &pdev->dev;
ep93xx_i2s_dai.private_data = info;
info->dma_params = ep93xx_i2s_dma_params;
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (!res) {
err = -ENODEV;
goto fail;
}
info->mem = request_mem_region(res->start, resource_size(res),
pdev->name);
if (!info->mem) {
err = -EBUSY;
goto fail;
}
info->regs = ioremap(info->mem->start, resource_size(info->mem));
if (!info->regs) {
err = -ENXIO;
goto fail_release_mem;
}
info->mclk = clk_get(&pdev->dev, "mclk");
if (IS_ERR(info->mclk)) {
err = PTR_ERR(info->mclk);
goto fail_unmap_mem;
}
info->sclk = clk_get(&pdev->dev, "sclk");
if (IS_ERR(info->sclk)) {
err = PTR_ERR(info->sclk);
goto fail_put_mclk;
}
info->lrclk = clk_get(&pdev->dev, "lrclk");
if (IS_ERR(info->lrclk)) {
err = PTR_ERR(info->lrclk);
goto fail_put_sclk;
}
err = snd_soc_register_dai(&ep93xx_i2s_dai);
if (err)
goto fail_put_lrclk;
return 0;
fail_put_lrclk:
clk_put(info->lrclk);
fail_put_sclk:
clk_put(info->sclk);
fail_put_mclk:
clk_put(info->mclk);
fail_unmap_mem:
iounmap(info->regs);
fail_release_mem:
release_mem_region(info->mem->start, resource_size(info->mem));
kfree(info);
fail:
return err;
}
static int __devexit ep93xx_i2s_remove(struct platform_device *pdev)
{
struct ep93xx_i2s_info *info = ep93xx_i2s_dai.private_data;
snd_soc_unregister_dai(&ep93xx_i2s_dai);
clk_put(info->lrclk);
clk_put(info->sclk);
clk_put(info->mclk);
iounmap(info->regs);
release_mem_region(info->mem->start, resource_size(info->mem));
kfree(info);
return 0;
}
static struct platform_driver ep93xx_i2s_driver = {
.probe = ep93xx_i2s_probe,
.remove = __devexit_p(ep93xx_i2s_remove),
.driver = {
.name = "ep93xx-i2s",
.owner = THIS_MODULE,
},
};
static int __init ep93xx_i2s_init(void)
{
return platform_driver_register(&ep93xx_i2s_driver);
}
static void __exit ep93xx_i2s_exit(void)
{
platform_driver_unregister(&ep93xx_i2s_driver);
}
module_init(ep93xx_i2s_init);
module_exit(ep93xx_i2s_exit);
MODULE_ALIAS("platform:ep93xx-i2s");
MODULE_AUTHOR("Ryan Mallon <[email protected]>");
MODULE_DESCRIPTION("EP93XX I2S driver");
MODULE_LICENSE("GPL");

18
sound/soc/ep93xx/ep93xx-i2s.h Normal file
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/*
* linux/sound/soc/ep93xx-i2s.h
* EP93xx I2S driver
*
* Copyright (C) 2010 Ryan Mallon <ryan@bluewatersys.com>
*
* 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 _EP93XX_SND_SOC_I2S_H
#define _EP93XX_SND_SOC_I2S_H
extern struct snd_soc_dai ep93xx_i2s_dai;
#endif /* _EP93XX_SND_SOC_I2S_H */

319
sound/soc/ep93xx/ep93xx-pcm.c Normal file
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/*
* linux/sound/arm/ep93xx-pcm.c - EP93xx ALSA PCM interface
*
* Copyright (C) 2006 Lennert Buytenhek <buytenh@wantstofly.org>
* Copyright (C) 2006 Applied Data Systems
*
* Rewritten for the SoC audio subsystem (Based on PXA2xx code):
* Copyright (c) 2008 Ryan Mallon <ryan@bluewatersys.com>
*
* 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/init.h>
#include <linux/device.h>
#include <linux/slab.h>
#include <linux/dma-mapping.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <mach/dma.h>
#include <mach/hardware.h>
#include <mach/ep93xx-regs.h>
#include "ep93xx-pcm.h"
static const struct snd_pcm_hardware ep93xx_pcm_hardware = {
.info = (SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER),
.rates = SNDRV_PCM_RATE_8000_48000,
.rate_min = SNDRV_PCM_RATE_8000,
.rate_max = SNDRV_PCM_RATE_48000,
.formats = (SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S24_LE |
SNDRV_PCM_FMTBIT_S32_LE),
.buffer_bytes_max = 131072,
.period_bytes_min = 32,
.period_bytes_max = 32768,
.periods_min = 1,
.periods_max = 32,
.fifo_size = 32,
};
struct ep93xx_runtime_data
{
struct ep93xx_dma_m2p_client cl;
struct ep93xx_pcm_dma_params *params;
int pointer_bytes;
struct tasklet_struct period_tasklet;
int periods;
struct ep93xx_dma_buffer buf[32];
};
static void ep93xx_pcm_period_elapsed(unsigned long data)
{
struct snd_pcm_substream *substream = (struct snd_pcm_substream *)data;
snd_pcm_period_elapsed(substream);
}
static void ep93xx_pcm_buffer_started(void *cookie,
struct ep93xx_dma_buffer *buf)
{
}
static void ep93xx_pcm_buffer_finished(void *cookie,
struct ep93xx_dma_buffer *buf,
int bytes, int error)
{
struct snd_pcm_substream *substream = cookie;
struct ep93xx_runtime_data *rtd = substream->runtime->private_data;
if (buf == rtd->buf + rtd->periods - 1)
rtd->pointer_bytes = 0;
else
rtd->pointer_bytes += buf->size;
if (!error) {
ep93xx_dma_m2p_submit_recursive(&rtd->cl, buf);
tasklet_schedule(&rtd->period_tasklet);
} else {
snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
}
}
static int ep93xx_pcm_open(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *soc_rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = soc_rtd->dai->cpu_dai;
struct ep93xx_pcm_dma_params *dma_params;
struct ep93xx_runtime_data *rtd;
int ret;
dma_params = snd_soc_dai_get_dma_data(cpu_dai, substream);
snd_soc_set_runtime_hwparams(substream, &ep93xx_pcm_hardware);
rtd = kmalloc(sizeof(*rtd), GFP_KERNEL);
if (!rtd)
return -ENOMEM;
memset(&rtd->period_tasklet, 0, sizeof(rtd->period_tasklet));
rtd->period_tasklet.func = ep93xx_pcm_period_elapsed;
rtd->period_tasklet.data = (unsigned long)substream;
rtd->cl.name = dma_params->name;
rtd->cl.flags = dma_params->dma_port | EP93XX_DMA_M2P_IGNORE_ERROR |
((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
EP93XX_DMA_M2P_TX : EP93XX_DMA_M2P_RX);
rtd->cl.cookie = substream;
rtd->cl.buffer_started = ep93xx_pcm_buffer_started;
rtd->cl.buffer_finished = ep93xx_pcm_buffer_finished;
ret = ep93xx_dma_m2p_client_register(&rtd->cl);
if (ret < 0) {
kfree(rtd);
return ret;
}
substream->runtime->private_data = rtd;
return 0;
}
static int ep93xx_pcm_close(struct snd_pcm_substream *substream)
{
struct ep93xx_runtime_data *rtd = substream->runtime->private_data;
ep93xx_dma_m2p_client_unregister(&rtd->cl);
kfree(rtd);
return 0;
}
static int ep93xx_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct ep93xx_runtime_data *rtd = runtime->private_data;
size_t totsize = params_buffer_bytes(params);
size_t period = params_period_bytes(params);
int i;
snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
runtime->dma_bytes = totsize;
rtd->periods = (totsize + period - 1) / period;
for (i = 0; i < rtd->periods; i++) {
rtd->buf[i].bus_addr = runtime->dma_addr + (i * period);
rtd->buf[i].size = period;
if ((i + 1) * period > totsize)
rtd->buf[i].size = totsize - (i * period);
}
return 0;
}
static int ep93xx_pcm_hw_free(struct snd_pcm_substream *substream)
{
snd_pcm_set_runtime_buffer(substream, NULL);
return 0;
}
static int ep93xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
{
struct ep93xx_runtime_data *rtd = substream->runtime->private_data;
int ret;
int i;
ret = 0;
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
rtd->pointer_bytes = 0;
for (i = 0; i < rtd->periods; i++)
ep93xx_dma_m2p_submit(&rtd->cl, rtd->buf + i);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
ep93xx_dma_m2p_flush(&rtd->cl);
break;
default:
ret = -EINVAL;
break;
}
return ret;
}
static snd_pcm_uframes_t ep93xx_pcm_pointer(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct ep93xx_runtime_data *rtd = substream->runtime->private_data;
/* FIXME: implement this with sub-period granularity */
return bytes_to_frames(runtime, rtd->pointer_bytes);
}
static int ep93xx_pcm_mmap(struct snd_pcm_substream *substream,
struct vm_area_struct *vma)
{
struct snd_pcm_runtime *runtime = substream->runtime;
return dma_mmap_writecombine(substream->pcm->card->dev, vma,
runtime->dma_area,
runtime->dma_addr,
runtime->dma_bytes);
}
static struct snd_pcm_ops ep93xx_pcm_ops = {
.open = ep93xx_pcm_open,
.close = ep93xx_pcm_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = ep93xx_pcm_hw_params,
.hw_free = ep93xx_pcm_hw_free,
.trigger = ep93xx_pcm_trigger,
.pointer = ep93xx_pcm_pointer,
.mmap = ep93xx_pcm_mmap,
};
static int ep93xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
{
struct snd_pcm_substream *substream = pcm->streams[stream].substream;
struct snd_dma_buffer *buf = &substream->dma_buffer;
size_t size = ep93xx_pcm_hardware.buffer_bytes_max;
buf->dev.type = SNDRV_DMA_TYPE_DEV;
buf->dev.dev = pcm->card->dev;
buf->private_data = NULL;
buf->area = dma_alloc_writecombine(pcm->card->dev, size,
&buf->addr, GFP_KERNEL);
buf->bytes = size;
return (buf->area == NULL) ? -ENOMEM : 0;
}
static void ep93xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
{
struct snd_pcm_substream *substream;
struct snd_dma_buffer *buf;
int stream;
for (stream = 0; stream < 2; stream++) {
substream = pcm->streams[stream].substream;
if (!substream)
continue;
buf = &substream->dma_buffer;
if (!buf->area)
continue;
dma_free_writecombine(pcm->card->dev, buf->bytes, buf->area,
buf->addr);
buf->area = NULL;
}
}
static u64 ep93xx_pcm_dmamask = 0xffffffff;
static int ep93xx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
struct snd_pcm *pcm)
{
int ret = 0;
if (!card->dev->dma_mask)
card->dev->dma_mask = &ep93xx_pcm_dmamask;
if (!card->dev->coherent_dma_mask)
card->dev->coherent_dma_mask = 0xffffffff;
if (dai->playback.channels_min) {
ret = ep93xx_pcm_preallocate_dma_buffer(pcm,
SNDRV_PCM_STREAM_PLAYBACK);
if (ret)
return ret;
}
if (dai->capture.channels_min) {
ret = ep93xx_pcm_preallocate_dma_buffer(pcm,
SNDRV_PCM_STREAM_CAPTURE);
if (ret)
return ret;
}
return 0;
}
struct snd_soc_platform ep93xx_soc_platform = {
.name = "ep93xx-audio",
.pcm_ops = &ep93xx_pcm_ops,
.pcm_new = &ep93xx_pcm_new,
.pcm_free = &ep93xx_pcm_free_dma_buffers,
};
EXPORT_SYMBOL_GPL(ep93xx_soc_platform);
static int __init ep93xx_soc_platform_init(void)
{
return snd_soc_register_platform(&ep93xx_soc_platform);
}
static void __exit ep93xx_soc_platform_exit(void)
{
snd_soc_unregister_platform(&ep93xx_soc_platform);
}
module_init(ep93xx_soc_platform_init);
module_exit(ep93xx_soc_platform_exit);
MODULE_AUTHOR("Ryan Mallon <[email protected]>");
MODULE_DESCRIPTION("EP93xx ALSA PCM interface");
MODULE_LICENSE("GPL");

22
sound/soc/ep93xx/ep93xx-pcm.h Normal file
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/*
* sound/soc/ep93xx/ep93xx-pcm.h - EP93xx ALSA PCM interface
*
* Copyright (C) 2006 Lennert Buytenhek <buytenh@wantstofly.org>
* Copyright (C) 2006 Applied Data Systems
*
* 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 _EP93XX_SND_SOC_PCM_H
#define _EP93XX_SND_SOC_PCM_H
struct ep93xx_pcm_dma_params {
char *name;
int dma_port;
};
extern struct snd_soc_platform ep93xx_soc_platform;
#endif /* _EP93XX_SND_SOC_PCM_H */

150
sound/soc/ep93xx/snappercl15.c Normal file
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/*
* snappercl15.c -- SoC audio for Bluewater Systems Snapper CL15 module
*
* Copyright (C) 2008 Bluewater Systems Ltd
* Author: Ryan Mallon <ryan@bluewatersys.com>
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the
* Free Software Foundation; either version 2 of the License, or (at your
* option) any later version.
*
*/
#include <linux/platform_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <asm/mach-types.h>
#include <mach/hardware.h>
#include "../codecs/tlv320aic23.h"
#include "ep93xx-pcm.h"
#include "ep93xx-i2s.h"
#define CODEC_CLOCK 5644800
static int snappercl15_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;
int err;
err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_IF |
SND_SOC_DAIFMT_CBS_CFS);
err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_IF |
SND_SOC_DAIFMT_CBS_CFS);
if (err)
return err;
err = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK,
SND_SOC_CLOCK_IN);
if (err)
return err;
err = snd_soc_dai_set_sysclk(cpu_dai, 0, CODEC_CLOCK,
SND_SOC_CLOCK_OUT);
if (err)
return err;
return 0;
}
static struct snd_soc_ops snappercl15_ops = {
.hw_params = snappercl15_hw_params,
};
static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_LINE("Line In", NULL),
SND_SOC_DAPM_MIC("Mic Jack", NULL),
};
static const struct snd_soc_dapm_route audio_map[] = {
{"Headphone Jack", NULL, "LHPOUT"},
{"Headphone Jack", NULL, "RHPOUT"},
{"LLINEIN", NULL, "Line In"},
{"RLINEIN", NULL, "Line In"},
{"MICIN", NULL, "Mic Jack"},
};
static int snappercl15_tlv320aic23_init(struct snd_soc_codec *codec)
{
snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
ARRAY_SIZE(tlv320aic23_dapm_widgets));
snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
return 0;
}
static struct snd_soc_dai_link snappercl15_dai = {
.name = "tlv320aic23",
.stream_name = "AIC23",
.cpu_dai = &ep93xx_i2s_dai,
.codec_dai = &tlv320aic23_dai,
.init = snappercl15_tlv320aic23_init,
.ops = &snappercl15_ops,
};
static struct snd_soc_card snd_soc_snappercl15 = {
.name = "Snapper CL15",
.platform = &ep93xx_soc_platform,
.dai_link = &snappercl15_dai,
.num_links = 1,
};
static struct snd_soc_device snappercl15_snd_devdata = {
.card = &snd_soc_snappercl15,
.codec_dev = &soc_codec_dev_tlv320aic23,
};
static struct platform_device *snappercl15_snd_device;
static int __init snappercl15_init(void)
{
int ret;
if (!machine_is_snapper_cl15())
return -ENODEV;
ret = ep93xx_i2s_acquire(EP93XX_SYSCON_DEVCFG_I2SONAC97,
EP93XX_SYSCON_I2SCLKDIV_ORIDE |
EP93XX_SYSCON_I2SCLKDIV_SPOL);
if (ret)
return ret;
snappercl15_snd_device = platform_device_alloc("soc-audio", -1);
if (!snappercl15_snd_device)
return -ENOMEM;
platform_set_drvdata(snappercl15_snd_device, &snappercl15_snd_devdata);
snappercl15_snd_devdata.dev = &snappercl15_snd_device->dev;
ret = platform_device_add(snappercl15_snd_device);
if (ret)
platform_device_put(snappercl15_snd_device);
return ret;
}
static void __exit snappercl15_exit(void)
{
platform_device_unregister(snappercl15_snd_device);
ep93xx_i2s_release();
}
module_init(snappercl15_init);
module_exit(snappercl15_exit);
MODULE_AUTHOR("Ryan Mallon <[email protected]>");
MODULE_DESCRIPTION("ALSA SoC Snapper CL15");
MODULE_LICENSE("GPL");

1
sound/soc/fsl/mpc5200_psc_i2s.c
View file

@ -16,7 +16,6 @@
#include <asm/mpc52xx_psc.h>
#include "mpc5200_psc_i2s.h"
#include "mpc5200_dma.h"
/**

12
sound/soc/fsl/mpc5200_psc_i2s.h
View file

@ -1,12 +0,0 @@
/*
* Freescale MPC5200 PSC in I2S mode
* ALSA SoC Digital Audio Interface (DAI) driver
*
*/
#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__
#define __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__
extern struct snd_soc_dai psc_i2s_dai[];
#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__ */

19
sound/soc/imx/Kconfig
View file

@ -1,4 +1,4 @@
config SND_IMX_SOC
menuconfig SND_IMX_SOC
tristate "SoC Audio for Freescale i.MX CPUs"
depends on ARCH_MXC
select SND_PCM
@ -8,14 +8,12 @@ config SND_IMX_SOC
Say Y or M if you want to add support for codecs attached to
the i.MX SSI interface.
config SND_MXC_SOC_SSI
tristate
if SND_IMX_SOC
config SND_MXC_SOC_WM1133_EV1
tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted"
depends on SND_IMX_SOC && MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL
depends on MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL
select SND_SOC_WM8350
select SND_MXC_SOC_SSI
help
Enable support for audio on the i.MX31ADS with the WM1133-EV1
PMIC board with WM8835x fitted.
@ -23,8 +21,17 @@ config SND_MXC_SOC_WM1133_EV1
config SND_SOC_PHYCORE_AC97
tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards"
depends on MACH_PCM043 || MACH_PCA100
select SND_MXC_SOC_SSI
select SND_SOC_WM9712
help
Say Y if you want to add support for SoC audio on Phytec phyCORE
and phyCARD boards in AC97 mode
config SND_SOC_EUKREA_TLV320
tristate "Eukrea TLV320"
depends on MACH_EUKREA_MBIMX27_BASEBOARD || MACH_EUKREA_MBIMXSD_BASEBOARD
select SND_SOC_TLV320AIC23
help
Enable I2S based access to the TLV320AIC23B codec attached
to the SSI interface
endif # SND_IMX_SOC

2
sound/soc/imx/Makefile
View file

@ -8,8 +8,10 @@ endif
obj-$(CONFIG_SND_IMX_SOC) += snd-soc-imx.o
# i.MX Machine Support
snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o
snd-soc-phycore-ac97-objs := phycore-ac97.o
snd-soc-wm1133-ev1-objs := wm1133-ev1.o
obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o
obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o

137
sound/soc/imx/eukrea-tlv320.c Normal file
View file

@ -0,0 +1,137 @@
/*
* eukrea-tlv320.c -- SoC audio for eukrea_cpuimxXX in I2S mode
*
* Copyright 2010 Eric Bénard, Eukréa Electromatique <eric@eukrea.com>
*
* based on sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c
* which is Copyright 2009 Simtec Electronics
* and on sound/soc/imx/phycore-ac97.c which is
* Copyright 2009 Sascha Hauer, Pengutronix <s.hauer@pengutronix.de>
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the
* Free Software Foundation; either version 2 of the License, or (at your
* option) any later version.
*
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/device.h>
#include <linux/i2c.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <asm/mach-types.h>
#include "../codecs/tlv320aic23.h"
#include "imx-ssi.h"
#define CODEC_CLOCK 12000000
static int eukrea_tlv320_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;
int ret;
ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM);
if (ret) {
pr_err("%s: failed set cpu dai format\n", __func__);
return ret;
}
ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM);
if (ret) {
pr_err("%s: failed set codec dai format\n", __func__);
return ret;
}
ret = snd_soc_dai_set_sysclk(codec_dai, 0,
CODEC_CLOCK, SND_SOC_CLOCK_OUT);
if (ret) {
pr_err("%s: failed setting codec sysclk\n", __func__);
return ret;
}
snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0);
ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0,
SND_SOC_CLOCK_IN);
if (ret) {
pr_err("can't set CPU system clock IMX_SSP_SYS_CLK\n");
return ret;
}
return 0;
}
static struct snd_soc_ops eukrea_tlv320_snd_ops = {
.hw_params = eukrea_tlv320_hw_params,
};
static struct snd_soc_dai_link eukrea_tlv320_dai = {
.name = "tlv320aic23",
.stream_name = "TLV320AIC23",
.codec_dai = &tlv320aic23_dai,
.ops = &eukrea_tlv320_snd_ops,
};
static struct snd_soc_card eukrea_tlv320 = {
.name = "cpuimx-audio",
.platform = &imx_soc_platform,
.dai_link = &eukrea_tlv320_dai,
.num_links = 1,
};
static struct snd_soc_device eukrea_tlv320_snd_devdata = {
.card = &eukrea_tlv320,
.codec_dev = &soc_codec_dev_tlv320aic23,
};
static struct platform_device *eukrea_tlv320_snd_device;
static int __init eukrea_tlv320_init(void)
{
int ret;
if (!machine_is_eukrea_cpuimx27() && !machine_is_eukrea_cpuimx25sd()
&& !machine_is_eukrea_cpuimx35sd())
/* return happy. We might run on a totally different machine */
return 0;
eukrea_tlv320_snd_device = platform_device_alloc("soc-audio", -1);
if (!eukrea_tlv320_snd_device)
return -ENOMEM;
eukrea_tlv320_dai.cpu_dai = &imx_ssi_pcm_dai[0];
platform_set_drvdata(eukrea_tlv320_snd_device, &eukrea_tlv320_snd_devdata);
eukrea_tlv320_snd_devdata.dev = &eukrea_tlv320_snd_device->dev;
ret = platform_device_add(eukrea_tlv320_snd_device);
if (ret) {
printk(KERN_ERR "ASoC: Platform device allocation failed\n");
platform_device_put(eukrea_tlv320_snd_device);
}
return ret;
}
static void __exit eukrea_tlv320_exit(void)
{
platform_device_unregister(eukrea_tlv320_snd_device);
}
module_init(eukrea_tlv320_init);
module_exit(eukrea_tlv320_exit);
MODULE_AUTHOR("Eric Bénard <[email protected]>");
MODULE_DESCRIPTION("CPUIMX ALSA SoC driver");
MODULE_LICENSE("GPL");

6
sound/soc/imx/imx-pcm-dma-mx2.c
View file

@ -292,12 +292,16 @@ static int snd_imx_open(struct snd_pcm_substream *substream)
int ret;
iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL);
if (iprtd == NULL)
return -ENOMEM;
runtime->private_data = iprtd;
ret = snd_pcm_hw_constraint_integer(substream->runtime,
SNDRV_PCM_HW_PARAM_PERIODS);
if (ret < 0)
if (ret < 0) {
kfree(iprtd);
return ret;
}
snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware);
return 0;

6
sound/soc/imx/imx-pcm-fiq.c
View file

@ -192,6 +192,8 @@ static int snd_imx_open(struct snd_pcm_substream *substream)
int ret;
iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL);
if (iprtd == NULL)
return -ENOMEM;
runtime->private_data = iprtd;
iprtd->substream = substream;
@ -202,8 +204,10 @@ static int snd_imx_open(struct snd_pcm_substream *substream)
ret = snd_pcm_hw_constraint_integer(substream->runtime,
SNDRV_PCM_HW_PARAM_PERIODS);
if (ret < 0)
if (ret < 0) {
kfree(iprtd);
return ret;
}
snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware);
return 0;

11
sound/soc/imx/imx-ssi.c
View file

@ -83,8 +83,6 @@ static int imx_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai,
/*
* SSI DAI format configuration.
* Should only be called when port is inactive (i.e. SSIEN = 0).
* Note: We don't use the I2S modes but instead manually configure the
* SSI for I2S because the I2S mode is only a register preset.
*/
static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
{
@ -99,6 +97,10 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
/* data on rising edge of bclk, frame low 1clk before data */
strcr |= SSI_STCR_TFSI | SSI_STCR_TEFS | SSI_STCR_TXBIT0;
scr |= SSI_SCR_NET;
if (ssi->flags & IMX_SSI_USE_I2S_SLAVE) {
scr &= ~SSI_I2S_MODE_MASK;
scr |= SSI_SCR_I2S_MODE_SLAVE;
}
break;
case SND_SOC_DAIFMT_LEFT_J:
/* data on rising edge of bclk, frame high with data */
@ -143,6 +145,11 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
strcr |= SSI_STCR_TFEN0;
if (ssi->flags & IMX_SSI_NET)
scr |= SSI_SCR_NET;
if (ssi->flags & IMX_SSI_SYN)
scr |= SSI_SCR_SYN;
writel(strcr, ssi->base + SSI_STCR);
writel(strcr, ssi->base + SSI_SRCR);
writel(scr, ssi->base + SSI_SCR);

23
sound/soc/jz4740/Kconfig Normal file
View file

@ -0,0 +1,23 @@
config SND_JZ4740_SOC
tristate "SoC Audio for Ingenic JZ4740 SoC"
depends on MACH_JZ4740 && SND_SOC
help
Say Y or M if you want to add support for codecs attached to
the JZ4740 I2S interface. You will also need to select the audio
interfaces to support below.
config SND_JZ4740_SOC_I2S
depends on SND_JZ4740_SOC
tristate "SoC Audio (I2S protocol) for Ingenic JZ4740 SoC"
help
Say Y if you want to use I2S protocol and I2S codec on Ingenic JZ4740
based boards.
config SND_JZ4740_SOC_QI_LB60
tristate "SoC Audio support for Qi LB60"
depends on SND_JZ4740_SOC && JZ4740_QI_LB60
select SND_JZ4740_SOC_I2S
select SND_SOC_JZ4740_CODEC
help
Say Y if you want to add support for ASoC audio on the Qi LB60 board
a.k.a Qi Ben NanoNote.

13
sound/soc/jz4740/Makefile Normal file
View file

@ -0,0 +1,13 @@
#
# Jz4740 Platform Support
#
snd-soc-jz4740-objs := jz4740-pcm.o
snd-soc-jz4740-i2s-objs := jz4740-i2s.o
obj-$(CONFIG_SND_JZ4740_SOC) += snd-soc-jz4740.o
obj-$(CONFIG_SND_JZ4740_SOC_I2S) += snd-soc-jz4740-i2s.o
# Jz4740 Machine Support
snd-soc-qi-lb60-objs := qi_lb60.o
obj-$(CONFIG_SND_JZ4740_SOC_QI_LB60) += snd-soc-qi-lb60.o

540
sound/soc/jz4740/jz4740-i2s.c Normal file
View file

@ -0,0 +1,540 @@
/*
* Copyright (C) 2010, Lars-Peter Clausen <lars@metafoo.de>
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the
* Free Software Foundation; either version 2 of the License, or (at your
* option) any later version.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 675 Mass Ave, Cambridge, MA 02139, USA.
*
*/
#include <linux/init.h>
#include <linux/io.h>
#include <linux/kernel.h>
#include <linux/module.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <linux/clk.h>
#include <linux/delay.h>
#include <linux/dma-mapping.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/initval.h>
#include "jz4740-i2s.h"
#include "jz4740-pcm.h"
#define JZ_REG_AIC_CONF 0x00
#define JZ_REG_AIC_CTRL 0x04
#define JZ_REG_AIC_I2S_FMT 0x10
#define JZ_REG_AIC_FIFO_STATUS 0x14
#define JZ_REG_AIC_I2S_STATUS 0x1c
#define JZ_REG_AIC_CLK_DIV 0x30
#define JZ_REG_AIC_FIFO 0x34
#define JZ_AIC_CONF_FIFO_RX_THRESHOLD_MASK (0xf << 12)
#define JZ_AIC_CONF_FIFO_TX_THRESHOLD_MASK (0xf << 8)
#define JZ_AIC_CONF_OVERFLOW_PLAY_LAST BIT(6)
#define JZ_AIC_CONF_INTERNAL_CODEC BIT(5)
#define JZ_AIC_CONF_I2S BIT(4)
#define JZ_AIC_CONF_RESET BIT(3)
#define JZ_AIC_CONF_BIT_CLK_MASTER BIT(2)
#define JZ_AIC_CONF_SYNC_CLK_MASTER BIT(1)
#define JZ_AIC_CONF_ENABLE BIT(0)
#define JZ_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET 12
#define JZ_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET 8
#define JZ_AIC_CTRL_OUTPUT_SAMPLE_SIZE_MASK (0x7 << 19)
#define JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_MASK (0x7 << 16)
#define JZ_AIC_CTRL_ENABLE_RX_DMA BIT(15)
#define JZ_AIC_CTRL_ENABLE_TX_DMA BIT(14)
#define JZ_AIC_CTRL_MONO_TO_STEREO BIT(11)
#define JZ_AIC_CTRL_SWITCH_ENDIANNESS BIT(10)
#define JZ_AIC_CTRL_SIGNED_TO_UNSIGNED BIT(9)
#define JZ_AIC_CTRL_FLUSH BIT(8)
#define JZ_AIC_CTRL_ENABLE_ROR_INT BIT(6)
#define JZ_AIC_CTRL_ENABLE_TUR_INT BIT(5)
#define JZ_AIC_CTRL_ENABLE_RFS_INT BIT(4)
#define JZ_AIC_CTRL_ENABLE_TFS_INT BIT(3)
#define JZ_AIC_CTRL_ENABLE_LOOPBACK BIT(2)
#define JZ_AIC_CTRL_ENABLE_PLAYBACK BIT(1)
#define JZ_AIC_CTRL_ENABLE_CAPTURE BIT(0)
#define JZ_AIC_CTRL_OUTPUT_SAMPLE_SIZE_OFFSET 19
#define JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_OFFSET 16
#define JZ_AIC_I2S_FMT_DISABLE_BIT_CLK BIT(12)
#define JZ_AIC_I2S_FMT_ENABLE_SYS_CLK BIT(4)
#define JZ_AIC_I2S_FMT_MSB BIT(0)
#define JZ_AIC_I2S_STATUS_BUSY BIT(2)
#define JZ_AIC_CLK_DIV_MASK 0xf
struct jz4740_i2s {
struct resource *mem;
void __iomem *base;
dma_addr_t phys_base;
struct clk *clk_aic;
struct clk *clk_i2s;
struct jz4740_pcm_config pcm_config_playback;
struct jz4740_pcm_config pcm_config_capture;
};
static inline uint32_t jz4740_i2s_read(const struct jz4740_i2s *i2s,
unsigned int reg)
{
return readl(i2s->base + reg);
}
static inline void jz4740_i2s_write(const struct jz4740_i2s *i2s,
unsigned int reg, uint32_t value)
{
writel(value, i2s->base + reg);
}
static inline struct jz4740_i2s *jz4740_dai_to_i2s(struct snd_soc_dai *dai)
{
return dai->private_data;
}
static int jz4740_i2s_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai);
uint32_t conf, ctrl;
if (dai->active)
return 0;
ctrl = jz4740_i2s_read(i2s, JZ_REG_AIC_CTRL);
ctrl |= JZ_AIC_CTRL_FLUSH;
jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl);
clk_enable(i2s->clk_i2s);
conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF);
conf |= JZ_AIC_CONF_ENABLE;
jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf);
return 0;
}
static void jz4740_i2s_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai);
uint32_t conf;
if (!dai->active)
return;
conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF);
conf &= ~JZ_AIC_CONF_ENABLE;
jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf);
clk_disable(i2s->clk_i2s);
}
static int jz4740_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *dai)
{
struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai);
uint32_t ctrl;
uint32_t mask;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
mask = JZ_AIC_CTRL_ENABLE_PLAYBACK | JZ_AIC_CTRL_ENABLE_TX_DMA;
else
mask = JZ_AIC_CTRL_ENABLE_CAPTURE | JZ_AIC_CTRL_ENABLE_RX_DMA;
ctrl = jz4740_i2s_read(i2s, JZ_REG_AIC_CTRL);
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
ctrl |= mask;
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
ctrl &= ~mask;
break;
default:
return -EINVAL;
}
jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl);
return 0;
}
static int jz4740_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
{
struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai);
uint32_t format = 0;
uint32_t conf;
conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF);
conf &= ~(JZ_AIC_CONF_BIT_CLK_MASTER | JZ_AIC_CONF_SYNC_CLK_MASTER);
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBS_CFS:
conf |= JZ_AIC_CONF_BIT_CLK_MASTER | JZ_AIC_CONF_SYNC_CLK_MASTER;
format |= JZ_AIC_I2S_FMT_ENABLE_SYS_CLK;
break;
case SND_SOC_DAIFMT_CBM_CFS:
conf |= JZ_AIC_CONF_SYNC_CLK_MASTER;
break;
case SND_SOC_DAIFMT_CBS_CFM:
conf |= JZ_AIC_CONF_BIT_CLK_MASTER;
break;
case SND_SOC_DAIFMT_CBM_CFM:
break;
default:
return -EINVAL;
}
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_MSB:
format |= JZ_AIC_I2S_FMT_MSB;
break;
case SND_SOC_DAIFMT_I2S:
break;
default:
return -EINVAL;
}
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
break;
default:
return -EINVAL;
}
jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf);
jz4740_i2s_write(i2s, JZ_REG_AIC_I2S_FMT, format);
return 0;
}
static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
{
struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai);
enum jz4740_dma_width dma_width;
struct jz4740_pcm_config *pcm_config;
unsigned int sample_size;
uint32_t ctrl;
ctrl = jz4740_i2s_read(i2s, JZ_REG_AIC_CTRL);
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S8:
sample_size = 0;
dma_width = JZ4740_DMA_WIDTH_8BIT;
break;
case SNDRV_PCM_FORMAT_S16:
sample_size = 1;
dma_width = JZ4740_DMA_WIDTH_16BIT;
break;
default:
return -EINVAL;
}
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
ctrl &= ~JZ_AIC_CTRL_OUTPUT_SAMPLE_SIZE_MASK;
ctrl |= sample_size << JZ_AIC_CTRL_OUTPUT_SAMPLE_SIZE_OFFSET;
if (params_channels(params) == 1)
ctrl |= JZ_AIC_CTRL_MONO_TO_STEREO;
else
ctrl &= ~JZ_AIC_CTRL_MONO_TO_STEREO;
pcm_config = &i2s->pcm_config_playback;
pcm_config->dma_config.dst_width = dma_width;
} else {
ctrl &= ~JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_MASK;
ctrl |= sample_size << JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_OFFSET;
pcm_config = &i2s->pcm_config_capture;
pcm_config->dma_config.src_width = dma_width;
}
jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl);
snd_soc_dai_set_dma_data(dai, substream, pcm_config);
return 0;
}
static int jz4740_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id,
unsigned int freq, int dir)
{
struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai);
struct clk *parent;
int ret = 0;
switch (clk_id) {
case JZ4740_I2S_CLKSRC_EXT:
parent = clk_get(NULL, "ext");
clk_set_parent(i2s->clk_i2s, parent);
break;
case JZ4740_I2S_CLKSRC_PLL:
parent = clk_get(NULL, "pll half");
clk_set_parent(i2s->clk_i2s, parent);
ret = clk_set_rate(i2s->clk_i2s, freq);
break;
default:
return -EINVAL;
}
clk_put(parent);
return ret;
}
static int jz4740_i2s_suspend(struct snd_soc_dai *dai)
{
struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai);
uint32_t conf;
if (dai->active) {
conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF);
conf &= ~JZ_AIC_CONF_ENABLE;
jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf);
clk_disable(i2s->clk_i2s);
}
clk_disable(i2s->clk_aic);
return 0;
}
static int jz4740_i2s_resume(struct snd_soc_dai *dai)
{
struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai);
uint32_t conf;
clk_enable(i2s->clk_aic);
if (dai->active) {
clk_enable(i2s->clk_i2s);
conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF);
conf |= JZ_AIC_CONF_ENABLE;
jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf);
}
return 0;
}
static int jz4740_i2s_probe(struct platform_device *pdev, struct snd_soc_dai *dai)
{
struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai);
uint32_t conf;
conf = (7 << JZ_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET) |
(8 << JZ_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET) |
JZ_AIC_CONF_OVERFLOW_PLAY_LAST |
JZ_AIC_CONF_I2S |
JZ_AIC_CONF_INTERNAL_CODEC;
jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, JZ_AIC_CONF_RESET);
jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf);
return 0;
}
static struct snd_soc_dai_ops jz4740_i2s_dai_ops = {
.startup = jz4740_i2s_startup,
.shutdown = jz4740_i2s_shutdown,
.trigger = jz4740_i2s_trigger,
.hw_params = jz4740_i2s_hw_params,
.set_fmt = jz4740_i2s_set_fmt,
.set_sysclk = jz4740_i2s_set_sysclk,
};
#define JZ4740_I2S_FMTS (SNDRV_PCM_FMTBIT_S8 | \
SNDRV_PCM_FMTBIT_S16_LE)
struct snd_soc_dai jz4740_i2s_dai = {
.name = "jz4740-i2s",
.probe = jz4740_i2s_probe,
.playback = {
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = JZ4740_I2S_FMTS,
},
.capture = {
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = JZ4740_I2S_FMTS,
},
.symmetric_rates = 1,
.ops = &jz4740_i2s_dai_ops,
.suspend = jz4740_i2s_suspend,
.resume = jz4740_i2s_resume,
};
EXPORT_SYMBOL_GPL(jz4740_i2s_dai);
static void __devinit jz4740_i2c_init_pcm_config(struct jz4740_i2s *i2s)
{
struct jz4740_dma_config *dma_config;
/* Playback */
dma_config = &i2s->pcm_config_playback.dma_config;
dma_config->src_width = JZ4740_DMA_WIDTH_32BIT,
dma_config->transfer_size = JZ4740_DMA_TRANSFER_SIZE_16BYTE;
dma_config->request_type = JZ4740_DMA_TYPE_AIC_TRANSMIT;
dma_config->flags = JZ4740_DMA_SRC_AUTOINC;
dma_config->mode = JZ4740_DMA_MODE_SINGLE;
i2s->pcm_config_playback.fifo_addr = i2s->phys_base + JZ_REG_AIC_FIFO;
/* Capture */
dma_config = &i2s->pcm_config_capture.dma_config;
dma_config->dst_width = JZ4740_DMA_WIDTH_32BIT,
dma_config->transfer_size = JZ4740_DMA_TRANSFER_SIZE_16BYTE;
dma_config->request_type = JZ4740_DMA_TYPE_AIC_RECEIVE;
dma_config->flags = JZ4740_DMA_DST_AUTOINC;
dma_config->mode = JZ4740_DMA_MODE_SINGLE;
i2s->pcm_config_capture.fifo_addr = i2s->phys_base + JZ_REG_AIC_FIFO;
}
static int __devinit jz4740_i2s_dev_probe(struct platform_device *pdev)
{
struct jz4740_i2s *i2s;
int ret;
i2s = kzalloc(sizeof(*i2s), GFP_KERNEL);
if (!i2s)
return -ENOMEM;
i2s->mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (!i2s->mem) {
ret = -ENOENT;
goto err_free;
}
i2s->mem = request_mem_region(i2s->mem->start, resource_size(i2s->mem),
pdev->name);
if (!i2s->mem) {
ret = -EBUSY;
goto err_free;
}
i2s->base = ioremap_nocache(i2s->mem->start, resource_size(i2s->mem));
if (!i2s->base) {
ret = -EBUSY;
goto err_release_mem_region;
}
i2s->phys_base = i2s->mem->start;
i2s->clk_aic = clk_get(&pdev->dev, "aic");
if (IS_ERR(i2s->clk_aic)) {
ret = PTR_ERR(i2s->clk_aic);
goto err_iounmap;
}
i2s->clk_i2s = clk_get(&pdev->dev, "i2s");
if (IS_ERR(i2s->clk_i2s)) {
ret = PTR_ERR(i2s->clk_i2s);
goto err_clk_put_aic;
}
clk_enable(i2s->clk_aic);
jz4740_i2c_init_pcm_config(i2s);
jz4740_i2s_dai.private_data = i2s;
ret = snd_soc_register_dai(&jz4740_i2s_dai);
if (ret) {
dev_err(&pdev->dev, "Failed to register DAI\n");
goto err_clk_put_i2s;
}
platform_set_drvdata(pdev, i2s);
return 0;
err_clk_put_i2s:
clk_disable(i2s->clk_aic);
clk_put(i2s->clk_i2s);
err_clk_put_aic:
clk_put(i2s->clk_aic);
err_iounmap:
iounmap(i2s->base);
err_release_mem_region:
release_mem_region(i2s->mem->start, resource_size(i2s->mem));
err_free:
kfree(i2s);
return ret;
}
static int __devexit jz4740_i2s_dev_remove(struct platform_device *pdev)
{
struct jz4740_i2s *i2s = platform_get_drvdata(pdev);
snd_soc_unregister_dai(&jz4740_i2s_dai);
clk_disable(i2s->clk_aic);
clk_put(i2s->clk_i2s);
clk_put(i2s->clk_aic);
iounmap(i2s->base);
release_mem_region(i2s->mem->start, resource_size(i2s->mem));
platform_set_drvdata(pdev, NULL);
kfree(i2s);
return 0;
}
static struct platform_driver jz4740_i2s_driver = {
.probe = jz4740_i2s_dev_probe,
.remove = __devexit_p(jz4740_i2s_dev_remove),
.driver = {
.name = "jz4740-i2s",
.owner = THIS_MODULE,
},
};
static int __init jz4740_i2s_init(void)
{
return platform_driver_register(&jz4740_i2s_driver);
}
module_init(jz4740_i2s_init);
static void __exit jz4740_i2s_exit(void)
{
platform_driver_unregister(&jz4740_i2s_driver);
}
module_exit(jz4740_i2s_exit);
MODULE_AUTHOR("Lars-Peter Clausen, <[email protected]>");
MODULE_DESCRIPTION("Ingenic JZ4740 SoC I2S driver");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:jz4740-i2s");

18
sound/soc/jz4740/jz4740-i2s.h Normal file
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/*
* 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 _JZ4740_I2S_H
#define _JZ4740_I2S_H
/* I2S clock source */
#define JZ4740_I2S_CLKSRC_EXT 0
#define JZ4740_I2S_CLKSRC_PLL 1
#define JZ4740_I2S_BIT_CLK 0
extern struct snd_soc_dai jz4740_i2s_dai;
#endif

373
sound/soc/jz4740/jz4740-pcm.c Normal file
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/*
* Copyright (C) 2010, Lars-Peter Clausen <lars@metafoo.de>
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the
* Free Software Foundation; either version 2 of the License, or (at your
* option) any later version.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 675 Mass Ave, Cambridge, MA 02139, USA.
*
*/
#include <linux/init.h>
#include <linux/interrupt.h>
#include <linux/kernel.h>
#include <linux/module.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <linux/dma-mapping.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <asm/mach-jz4740/dma.h>
#include "jz4740-pcm.h"
struct jz4740_runtime_data {
unsigned long dma_period;
dma_addr_t dma_start;
dma_addr_t dma_pos;
dma_addr_t dma_end;
struct jz4740_dma_chan *dma;
dma_addr_t fifo_addr;
};
/* identify hardware playback capabilities */
static const struct snd_pcm_hardware jz4740_pcm_hardware = {
.info = SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER,
.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8,
.rates = SNDRV_PCM_RATE_8000_48000,
.channels_min = 1,
.channels_max = 2,
.period_bytes_min = 16,
.period_bytes_max = 2 * PAGE_SIZE,
.periods_min = 2,
.periods_max = 128,
.buffer_bytes_max = 128 * 2 * PAGE_SIZE,
.fifo_size = 32,
};
static void jz4740_pcm_start_transfer(struct jz4740_runtime_data *prtd,
struct snd_pcm_substream *substream)
{
unsigned long count;
if (prtd->dma_pos == prtd->dma_end)
prtd->dma_pos = prtd->dma_start;
if (prtd->dma_pos + prtd->dma_period > prtd->dma_end)
count = prtd->dma_end - prtd->dma_pos;
else
count = prtd->dma_period;
jz4740_dma_disable(prtd->dma);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
jz4740_dma_set_src_addr(prtd->dma, prtd->dma_pos);
jz4740_dma_set_dst_addr(prtd->dma, prtd->fifo_addr);
} else {
jz4740_dma_set_src_addr(prtd->dma, prtd->fifo_addr);
jz4740_dma_set_dst_addr(prtd->dma, prtd->dma_pos);
}
jz4740_dma_set_transfer_count(prtd->dma, count);
prtd->dma_pos += count;
jz4740_dma_enable(prtd->dma);
}
static void jz4740_pcm_dma_transfer_done(struct jz4740_dma_chan *dma, int err,
void *dev_id)
{
struct snd_pcm_substream *substream = dev_id;
struct snd_pcm_runtime *runtime = substream->runtime;
struct jz4740_runtime_data *prtd = runtime->private_data;
snd_pcm_period_elapsed(substream);
jz4740_pcm_start_transfer(prtd, substream);
}
static int jz4740_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct jz4740_runtime_data *prtd = runtime->private_data;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct jz4740_pcm_config *config;
config = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
if (!config)
return 0;
if (!prtd->dma) {
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
prtd->dma = jz4740_dma_request(substream, "PCM Capture");
else
prtd->dma = jz4740_dma_request(substream, "PCM Playback");
}
if (!prtd->dma)
return -EBUSY;
jz4740_dma_configure(prtd->dma, &config->dma_config);
prtd->fifo_addr = config->fifo_addr;
jz4740_dma_set_complete_cb(prtd->dma, jz4740_pcm_dma_transfer_done);
snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
runtime->dma_bytes = params_buffer_bytes(params);
prtd->dma_period = params_period_bytes(params);
prtd->dma_start = runtime->dma_addr;
prtd->dma_pos = prtd->dma_start;
prtd->dma_end = prtd->dma_start + runtime->dma_bytes;
return 0;
}
static int jz4740_pcm_hw_free(struct snd_pcm_substream *substream)
{
struct jz4740_runtime_data *prtd = substream->runtime->private_data;
snd_pcm_set_runtime_buffer(substream, NULL);
if (prtd->dma) {
jz4740_dma_free(prtd->dma);
prtd->dma = NULL;
}
return 0;
}
static int jz4740_pcm_prepare(struct snd_pcm_substream *substream)
{
struct jz4740_runtime_data *prtd = substream->runtime->private_data;
if (!prtd->dma)
return -EBUSY;
prtd->dma_pos = prtd->dma_start;
return 0;
}
static int jz4740_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct jz4740_runtime_data *prtd = runtime->private_data;
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
jz4740_pcm_start_transfer(prtd, substream);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
jz4740_dma_disable(prtd->dma);
break;
default:
break;
}
return 0;
}
static snd_pcm_uframes_t jz4740_pcm_pointer(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct jz4740_runtime_data *prtd = runtime->private_data;
unsigned long byte_offset;
snd_pcm_uframes_t offset;
struct jz4740_dma_chan *dma = prtd->dma;
/* prtd->dma_pos points to the end of the current transfer. So by
* subtracting prdt->dma_start we get the offset to the end of the
* current period in bytes. By subtracting the residue of the transfer
* we get the current offset in bytes. */
byte_offset = prtd->dma_pos - prtd->dma_start;
byte_offset -= jz4740_dma_get_residue(dma);
offset = bytes_to_frames(runtime, byte_offset);
if (offset >= runtime->buffer_size)
offset = 0;
return offset;
}
static int jz4740_pcm_open(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct jz4740_runtime_data *prtd;
prtd = kzalloc(sizeof(*prtd), GFP_KERNEL);
if (prtd == NULL)
return -ENOMEM;
snd_soc_set_runtime_hwparams(substream, &jz4740_pcm_hardware);
runtime->private_data = prtd;
return 0;
}
static int jz4740_pcm_close(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct jz4740_runtime_data *prtd = runtime->private_data;
kfree(prtd);
return 0;
}
static int jz4740_pcm_mmap(struct snd_pcm_substream *substream,
struct vm_area_struct *vma)
{
return remap_pfn_range(vma, vma->vm_start,
substream->dma_buffer.addr >> PAGE_SHIFT,
vma->vm_end - vma->vm_start, vma->vm_page_prot);
}
static struct snd_pcm_ops jz4740_pcm_ops = {
.open = jz4740_pcm_open,
.close = jz4740_pcm_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = jz4740_pcm_hw_params,
.hw_free = jz4740_pcm_hw_free,
.prepare = jz4740_pcm_prepare,
.trigger = jz4740_pcm_trigger,
.pointer = jz4740_pcm_pointer,
.mmap = jz4740_pcm_mmap,
};
static int jz4740_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
{
struct snd_pcm_substream *substream = pcm->streams[stream].substream;
struct snd_dma_buffer *buf = &substream->dma_buffer;
size_t size = jz4740_pcm_hardware.buffer_bytes_max;
buf->dev.type = SNDRV_DMA_TYPE_DEV;
buf->dev.dev = pcm->card->dev;
buf->private_data = NULL;
buf->area = dma_alloc_noncoherent(pcm->card->dev, size,
&buf->addr, GFP_KERNEL);
if (!buf->area)
return -ENOMEM;
buf->bytes = size;
return 0;
}
static void jz4740_pcm_free(struct snd_pcm *pcm)
{
struct snd_pcm_substream *substream;
struct snd_dma_buffer *buf;
int stream;
for (stream = 0; stream < SNDRV_PCM_STREAM_LAST; ++stream) {
substream = pcm->streams[stream].substream;
if (!substream)
continue;
buf = &substream->dma_buffer;
if (!buf->area)
continue;
dma_free_noncoherent(pcm->card->dev, buf->bytes, buf->area,
buf->addr);
buf->area = NULL;
}
}
static u64 jz4740_pcm_dmamask = DMA_BIT_MASK(32);
int jz4740_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
struct snd_pcm *pcm)
{
int ret = 0;
if (!card->dev->dma_mask)
card->dev->dma_mask = &jz4740_pcm_dmamask;
if (!card->dev->coherent_dma_mask)
card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
if (dai->playback.channels_min) {
ret = jz4740_pcm_preallocate_dma_buffer(pcm,
SNDRV_PCM_STREAM_PLAYBACK);
if (ret)
goto err;
}
if (dai->capture.channels_min) {
ret = jz4740_pcm_preallocate_dma_buffer(pcm,
SNDRV_PCM_STREAM_CAPTURE);
if (ret)
goto err;
}
err:
return ret;
}
struct snd_soc_platform jz4740_soc_platform = {
.name = "jz4740-pcm",
.pcm_ops = &jz4740_pcm_ops,
.pcm_new = jz4740_pcm_new,
.pcm_free = jz4740_pcm_free,
};
EXPORT_SYMBOL_GPL(jz4740_soc_platform);
static int __devinit jz4740_pcm_probe(struct platform_device *pdev)
{
return snd_soc_register_platform(&jz4740_soc_platform);
}
static int __devexit jz4740_pcm_remove(struct platform_device *pdev)
{
snd_soc_unregister_platform(&jz4740_soc_platform);
return 0;
}
static struct platform_driver jz4740_pcm_driver = {
.probe = jz4740_pcm_probe,
.remove = __devexit_p(jz4740_pcm_remove),
.driver = {
.name = "jz4740-pcm",
.owner = THIS_MODULE,
},
};
static int __init jz4740_soc_platform_init(void)
{
return platform_driver_register(&jz4740_pcm_driver);
}
module_init(jz4740_soc_platform_init);
static void __exit jz4740_soc_platform_exit(void)
{
return platform_driver_unregister(&jz4740_pcm_driver);
}
module_exit(jz4740_soc_platform_exit);
MODULE_AUTHOR("Lars-Peter Clausen <[email protected]>");
MODULE_DESCRIPTION("Ingenic SoC JZ4740 PCM driver");
MODULE_LICENSE("GPL");

22
sound/soc/jz4740/jz4740-pcm.h Normal file
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/*
*
* 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 _JZ4740_PCM_H
#define _JZ4740_PCM_H
#include <linux/dma-mapping.h>
#include <asm/mach-jz4740/dma.h>
/* platform data */
extern struct snd_soc_platform jz4740_soc_platform;
struct jz4740_pcm_config {
struct jz4740_dma_config dma_config;
phys_addr_t fifo_addr;
};
#endif

166
sound/soc/jz4740/qi_lb60.c Normal file
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/*
* Copyright (C) 2009, Lars-Peter Clausen <lars@metafoo.de>
*
* 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.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 675 Mass Ave, Cambridge, MA 02139, USA.
*
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/timer.h>
#include <linux/interrupt.h>
#include <linux/platform_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <linux/gpio.h>
#include "../codecs/jz4740.h"
#include "jz4740-pcm.h"
#include "jz4740-i2s.h"
#define QI_LB60_SND_GPIO JZ_GPIO_PORTB(29)
#define QI_LB60_AMP_GPIO JZ_GPIO_PORTD(4)
static int qi_lb60_spk_event(struct snd_soc_dapm_widget *widget,
struct snd_kcontrol *ctrl, int event)
{
int on = 0;
if (event & SND_SOC_DAPM_POST_PMU)
on = 1;
else if (event & SND_SOC_DAPM_PRE_PMD)
on = 0;
gpio_set_value(QI_LB60_SND_GPIO, on);
gpio_set_value(QI_LB60_AMP_GPIO, on);
return 0;
}
static const struct snd_soc_dapm_widget qi_lb60_widgets[] = {
SND_SOC_DAPM_SPK("Speaker", qi_lb60_spk_event),
SND_SOC_DAPM_MIC("Mic", NULL),
};
static const struct snd_soc_dapm_route qi_lb60_routes[] = {
{"Mic", NULL, "MIC"},
{"Speaker", NULL, "LOUT"},
{"Speaker", NULL, "ROUT"},
};
#define QI_LB60_DAIFMT (SND_SOC_DAIFMT_I2S | \
SND_SOC_DAIFMT_NB_NF | \
SND_SOC_DAIFMT_CBM_CFM)
static int qi_lb60_codec_init(struct snd_soc_codec *codec)
{
int ret;
struct snd_soc_dai *cpu_dai = codec->socdev->card->dai_link->cpu_dai;
snd_soc_dapm_nc_pin(codec, "LIN");
snd_soc_dapm_nc_pin(codec, "RIN");
ret = snd_soc_dai_set_fmt(cpu_dai, QI_LB60_DAIFMT);
if (ret < 0) {
dev_err(codec->dev, "Failed to set cpu dai format: %d\n", ret);
return ret;
}
snd_soc_dapm_new_controls(codec, qi_lb60_widgets, ARRAY_SIZE(qi_lb60_widgets));
snd_soc_dapm_add_routes(codec, qi_lb60_routes, ARRAY_SIZE(qi_lb60_routes));
snd_soc_dapm_sync(codec);
return 0;
}
static struct snd_soc_dai_link qi_lb60_dai = {
.name = "jz4740",
.stream_name = "jz4740",
.cpu_dai = &jz4740_i2s_dai,
.codec_dai = &jz4740_codec_dai,
.init = qi_lb60_codec_init,
};
static struct snd_soc_card qi_lb60 = {
.name = "QI LB60",
.dai_link = &qi_lb60_dai,
.num_links = 1,
.platform = &jz4740_soc_platform,
};
static struct snd_soc_device qi_lb60_snd_devdata = {
.card = &qi_lb60,
.codec_dev = &soc_codec_dev_jz4740_codec,
};
static struct platform_device *qi_lb60_snd_device;
static int __init qi_lb60_init(void)
{
int ret;
qi_lb60_snd_device = platform_device_alloc("soc-audio", -1);
if (!qi_lb60_snd_device)
return -ENOMEM;
ret = gpio_request(QI_LB60_SND_GPIO, "SND");
if (ret) {
pr_err("qi_lb60 snd: Failed to request SND GPIO(%d): %d\n",
QI_LB60_SND_GPIO, ret);
goto err_device_put;
}
ret = gpio_request(QI_LB60_AMP_GPIO, "AMP");
if (ret) {
pr_err("qi_lb60 snd: Failed to request AMP GPIO(%d): %d\n",
QI_LB60_AMP_GPIO, ret);
goto err_gpio_free_snd;
}
gpio_direction_output(QI_LB60_SND_GPIO, 0);
gpio_direction_output(QI_LB60_AMP_GPIO, 0);
platform_set_drvdata(qi_lb60_snd_device, &qi_lb60_snd_devdata);
qi_lb60_snd_devdata.dev = &qi_lb60_snd_device->dev;
ret = platform_device_add(qi_lb60_snd_device);
if (ret) {
pr_err("qi_lb60 snd: Failed to add snd soc device: %d\n", ret);
goto err_unset_pdata;
}
return 0;
err_unset_pdata:
platform_set_drvdata(qi_lb60_snd_device, NULL);
/*err_gpio_free_amp:*/
gpio_free(QI_LB60_AMP_GPIO);
err_gpio_free_snd:
gpio_free(QI_LB60_SND_GPIO);
err_device_put:
platform_device_put(qi_lb60_snd_device);
return ret;
}
module_init(qi_lb60_init);
static void __exit qi_lb60_exit(void)
{
gpio_free(QI_LB60_AMP_GPIO);
gpio_free(QI_LB60_SND_GPIO);
platform_device_unregister(qi_lb60_snd_device);
}
module_exit(qi_lb60_exit);
MODULE_AUTHOR("Lars-Peter Clausen <[email protected]>");
MODULE_DESCRIPTION("ALSA SoC QI LB60 Audio support");
MODULE_LICENSE("GPL v2");

20
sound/soc/kirkwood/Kconfig Normal file
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config SND_KIRKWOOD_SOC
tristate "SoC Audio for the Marvell Kirkwood chip"
depends on ARCH_KIRKWOOD
help
Say Y or M if you want to add support for codecs attached to
the Kirkwood I2S interface. You will also need to select the
audio interfaces to support below.
config SND_KIRKWOOD_SOC_I2S
tristate
config SND_KIRKWOOD_SOC_OPENRD
tristate "SoC Audio support for Kirkwood Openrd Client"
depends on SND_KIRKWOOD_SOC && MACH_OPENRD_CLIENT
select SND_KIRKWOOD_SOC_I2S
select SND_SOC_CS42L51
help
Say Y if you want to add support for SoC audio on
Openrd Client.

9
sound/soc/kirkwood/Makefile Normal file
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snd-soc-kirkwood-objs := kirkwood-dma.o
snd-soc-kirkwood-i2s-objs := kirkwood-i2s.o
obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o
obj-$(CONFIG_SND_KIRKWOOD_SOC_I2S) += snd-soc-kirkwood-i2s.o
snd-soc-openrd-objs := kirkwood-openrd.o
obj-$(CONFIG_SND_KIRKWOOD_SOC_OPENRD) += snd-soc-openrd.o

383
sound/soc/kirkwood/kirkwood-dma.c Normal file
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/*
* kirkwood-dma.c
*
* (c) 2010 Arnaud Patard <apatard@mandriva.com>
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the
* Free Software Foundation; either version 2 of the License, or (at your
* option) any later version.
*/
#include <linux/init.h>
#include <linux/module.h>
#include <linux/device.h>
#include <linux/io.h>
#include <linux/slab.h>
#include <linux/interrupt.h>
#include <linux/dma-mapping.h>
#include <linux/mbus.h>
#include <sound/soc.h>
#include "kirkwood-dma.h"
#include "kirkwood.h"
#define KIRKWOOD_RATES \
(SNDRV_PCM_RATE_44100 | \
SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
#define KIRKWOOD_FORMATS \
(SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_S24_LE | \
SNDRV_PCM_FMTBIT_S32_LE)
struct kirkwood_dma_priv {
struct snd_pcm_substream *play_stream;
struct snd_pcm_substream *rec_stream;
struct kirkwood_dma_data *data;
};
static struct snd_pcm_hardware kirkwood_dma_snd_hw = {
.info = (SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_PAUSE),
.formats = KIRKWOOD_FORMATS,
.rates = KIRKWOOD_RATES,
.rate_min = 44100,
.rate_max = 96000,
.channels_min = 1,
.channels_max = 2,
.buffer_bytes_max = KIRKWOOD_SND_MAX_PERIOD_BYTES * KIRKWOOD_SND_MAX_PERIODS,
.period_bytes_min = KIRKWOOD_SND_MIN_PERIOD_BYTES,
.period_bytes_max = KIRKWOOD_SND_MAX_PERIOD_BYTES,
.periods_min = KIRKWOOD_SND_MIN_PERIODS,
.periods_max = KIRKWOOD_SND_MAX_PERIODS,
.fifo_size = 0,
};
static u64 kirkwood_dma_dmamask = 0xFFFFFFFFUL;
static irqreturn_t kirkwood_dma_irq(int irq, void *dev_id)
{
struct kirkwood_dma_priv *prdata = dev_id;
struct kirkwood_dma_data *priv = prdata->data;
unsigned long mask, status, cause;
mask = readl(priv->io + KIRKWOOD_INT_MASK);
status = readl(priv->io + KIRKWOOD_INT_CAUSE) & mask;
cause = readl(priv->io + KIRKWOOD_ERR_CAUSE);
if (unlikely(cause)) {
printk(KERN_WARNING "%s: got err interrupt 0x%lx\n",
__func__, cause);
writel(cause, priv->io + KIRKWOOD_ERR_CAUSE);
return IRQ_HANDLED;
}
/* we've enabled only bytes interrupts ... */
if (status & ~(KIRKWOOD_INT_CAUSE_PLAY_BYTES | \
KIRKWOOD_INT_CAUSE_REC_BYTES)) {
printk(KERN_WARNING "%s: unexpected interrupt %lx\n",
__func__, status);
return IRQ_NONE;
}
/* ack int */
writel(status, priv->io + KIRKWOOD_INT_CAUSE);
if (status & KIRKWOOD_INT_CAUSE_PLAY_BYTES)
snd_pcm_period_elapsed(prdata->play_stream);
if (status & KIRKWOOD_INT_CAUSE_REC_BYTES)
snd_pcm_period_elapsed(prdata->rec_stream);
return IRQ_HANDLED;
}
static void kirkwood_dma_conf_mbus_windows(void __iomem *base, int win,
unsigned long dma,
struct mbus_dram_target_info *dram)
{
int i;
/* First disable and clear windows */
writel(0, base + KIRKWOOD_AUDIO_WIN_CTRL_REG(win));
writel(0, base + KIRKWOOD_AUDIO_WIN_BASE_REG(win));
/* try to find matching cs for current dma address */
for (i = 0; i < dram->num_cs; i++) {
struct mbus_dram_window *cs = dram->cs + i;
if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) {
writel(cs->base & 0xffff0000,
base + KIRKWOOD_AUDIO_WIN_BASE_REG(win));
writel(((cs->size - 1) & 0xffff0000) |
(cs->mbus_attr << 8) |
(dram->mbus_dram_target_id << 4) | 1,
base + KIRKWOOD_AUDIO_WIN_CTRL_REG(win));
}
}
}
static int kirkwood_dma_open(struct snd_pcm_substream *substream)
{
int err;
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
struct snd_soc_dai *cpu_dai = soc_runtime->dai->cpu_dai;
struct kirkwood_dma_data *priv;
struct kirkwood_dma_priv *prdata = cpu_dai->private_data;
unsigned long addr;
priv = snd_soc_dai_get_dma_data(cpu_dai, substream);
snd_soc_set_runtime_hwparams(substream, &kirkwood_dma_snd_hw);
/* Ensure that all constraints linked to dma burst are fullfilled */
err = snd_pcm_hw_constraint_minmax(runtime,
SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
priv->burst * 2,
KIRKWOOD_AUDIO_BUF_MAX-1);
if (err < 0)
return err;
err = snd_pcm_hw_constraint_step(runtime, 0,
SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
priv->burst);
if (err < 0)
return err;
err = snd_pcm_hw_constraint_step(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
priv->burst);
if (err < 0)
return err;
if (soc_runtime->dai->cpu_dai->private_data == NULL) {
prdata = kzalloc(sizeof(struct kirkwood_dma_priv), GFP_KERNEL);
if (prdata == NULL)
return -ENOMEM;
prdata->data = priv;
err = request_irq(priv->irq, kirkwood_dma_irq, IRQF_SHARED,
"kirkwood-i2s", prdata);
if (err) {
kfree(prdata);
return -EBUSY;
}
soc_runtime->dai->cpu_dai->private_data = prdata;
/*
* Enable Error interrupts. We're only ack'ing them but
* it's usefull for diagnostics
*/
writel((unsigned long)-1, priv->io + KIRKWOOD_ERR_MASK);
}
addr = virt_to_phys(substream->dma_buffer.area);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
prdata->play_stream = substream;
kirkwood_dma_conf_mbus_windows(priv->io,
KIRKWOOD_PLAYBACK_WIN, addr, priv->dram);
} else {
prdata->rec_stream = substream;
kirkwood_dma_conf_mbus_windows(priv->io,
KIRKWOOD_RECORD_WIN, addr, priv->dram);
}
return 0;
}
static int kirkwood_dma_close(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
struct snd_soc_dai *cpu_dai = soc_runtime->dai->cpu_dai;
struct kirkwood_dma_priv *prdata = cpu_dai->private_data;
struct kirkwood_dma_data *priv;
priv = snd_soc_dai_get_dma_data(cpu_dai, substream);
if (!prdata || !priv)
return 0;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
prdata->play_stream = NULL;
else
prdata->rec_stream = NULL;
if (!prdata->play_stream && !prdata->rec_stream) {
writel(0, priv->io + KIRKWOOD_ERR_MASK);
free_irq(priv->irq, prdata);
kfree(prdata);
soc_runtime->dai->cpu_dai->private_data = NULL;
}
return 0;
}
static int kirkwood_dma_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_pcm_runtime *runtime = substream->runtime;
snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
runtime->dma_bytes = params_buffer_bytes(params);
return 0;
}
static int kirkwood_dma_hw_free(struct snd_pcm_substream *substream)
{
snd_pcm_set_runtime_buffer(substream, NULL);
return 0;
}
static int kirkwood_dma_prepare(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
struct snd_soc_dai *cpu_dai = soc_runtime->dai->cpu_dai;
struct kirkwood_dma_data *priv;
unsigned long size, count;
priv = snd_soc_dai_get_dma_data(cpu_dai, substream);
/* compute buffer size in term of "words" as requested in specs */
size = frames_to_bytes(runtime, runtime->buffer_size);
size = (size>>2)-1;
count = snd_pcm_lib_period_bytes(substream);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
writel(count, priv->io + KIRKWOOD_PLAY_BYTE_INT_COUNT);
writel(runtime->dma_addr, priv->io + KIRKWOOD_PLAY_BUF_ADDR);
writel(size, priv->io + KIRKWOOD_PLAY_BUF_SIZE);
} else {
writel(count, priv->io + KIRKWOOD_REC_BYTE_INT_COUNT);
writel(runtime->dma_addr, priv->io + KIRKWOOD_REC_BUF_ADDR);
writel(size, priv->io + KIRKWOOD_REC_BUF_SIZE);
}
return 0;
}
static snd_pcm_uframes_t kirkwood_dma_pointer(struct snd_pcm_substream
*substream)
{
struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
struct snd_soc_dai *cpu_dai = soc_runtime->dai->cpu_dai;
struct kirkwood_dma_data *priv;
snd_pcm_uframes_t count;
priv = snd_soc_dai_get_dma_data(cpu_dai, substream);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
count = bytes_to_frames(substream->runtime,
readl(priv->io + KIRKWOOD_PLAY_BYTE_COUNT));
else
count = bytes_to_frames(substream->runtime,
readl(priv->io + KIRKWOOD_REC_BYTE_COUNT));
return count;
}
struct snd_pcm_ops kirkwood_dma_ops = {
.open = kirkwood_dma_open,
.close = kirkwood_dma_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = kirkwood_dma_hw_params,
.hw_free = kirkwood_dma_hw_free,
.prepare = kirkwood_dma_prepare,
.pointer = kirkwood_dma_pointer,
};
static int kirkwood_dma_preallocate_dma_buffer(struct snd_pcm *pcm,
int stream)
{
struct snd_pcm_substream *substream = pcm->streams[stream].substream;
struct snd_dma_buffer *buf = &substream->dma_buffer;
size_t size = kirkwood_dma_snd_hw.buffer_bytes_max;
buf->dev.type = SNDRV_DMA_TYPE_DEV;
buf->dev.dev = pcm->card->dev;
buf->area = dma_alloc_coherent(pcm->card->dev, size,
&buf->addr, GFP_KERNEL);
if (!buf->area)
return -ENOMEM;
buf->bytes = size;
buf->private_data = NULL;
return 0;
}
static int kirkwood_dma_new(struct snd_card *card,
struct snd_soc_dai *dai, struct snd_pcm *pcm)
{
int ret;
if (!card->dev->dma_mask)
card->dev->dma_mask = &kirkwood_dma_dmamask;
if (!card->dev->coherent_dma_mask)
card->dev->coherent_dma_mask = 0xffffffff;
if (dai->playback.channels_min) {
ret = kirkwood_dma_preallocate_dma_buffer(pcm,
SNDRV_PCM_STREAM_PLAYBACK);
if (ret)
return ret;
}
if (dai->capture.channels_min) {
ret = kirkwood_dma_preallocate_dma_buffer(pcm,
SNDRV_PCM_STREAM_CAPTURE);
if (ret)
return ret;
}
return 0;
}
static void kirkwood_dma_free_dma_buffers(struct snd_pcm *pcm)
{
struct snd_pcm_substream *substream;
struct snd_dma_buffer *buf;
int stream;
for (stream = 0; stream < 2; stream++) {
substream = pcm->streams[stream].substream;
if (!substream)
continue;
buf = &substream->dma_buffer;
if (!buf->area)
continue;
dma_free_coherent(pcm->card->dev, buf->bytes,
buf->area, buf->addr);
buf->area = NULL;
}
}
struct snd_soc_platform kirkwood_soc_platform = {
.name = "kirkwood-dma",
.pcm_ops = &kirkwood_dma_ops,
.pcm_new = kirkwood_dma_new,
.pcm_free = kirkwood_dma_free_dma_buffers,
};
EXPORT_SYMBOL_GPL(kirkwood_soc_platform);
static int __init kirkwood_soc_platform_init(void)
{
return snd_soc_register_platform(&kirkwood_soc_platform);
}
module_init(kirkwood_soc_platform_init);
static void __exit kirkwood_soc_platform_exit(void)
{
snd_soc_unregister_platform(&kirkwood_soc_platform);
}
module_exit(kirkwood_soc_platform_exit);
MODULE_AUTHOR("Arnaud Patard <[email protected]>");
MODULE_DESCRIPTION("Marvell Kirkwood Audio DMA module");
MODULE_LICENSE("GPL");

17
sound/soc/kirkwood/kirkwood-dma.h Normal file
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/*
* kirkwood-dma.h
*
* (c) 2010 Arnaud Patard <apatard@mandriva.com>
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the
* Free Software Foundation; either version 2 of the License, or (at your
* option) any later version.
*/
#ifndef _KIRKWOOD_DMA_H
#define _KIRKWOOD_DMA_H
extern struct snd_soc_platform kirkwood_soc_platform;
#endif

495
sound/soc/kirkwood/kirkwood-i2s.c Normal file
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/*
* kirkwood-i2s.c
*
* (c) 2010 Arnaud Patard <apatard@mandriva.com>
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the
* Free Software Foundation; either version 2 of the License, or (at your
* option) any later version.
*/
#include <linux/init.h>
#include <linux/module.h>
#include <linux/platform_device.h>
#include <linux/io.h>
#include <linux/slab.h>
#include <linux/mbus.h>
#include <linux/delay.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <plat/audio.h>
#include "kirkwood-i2s.h"
#include "kirkwood.h"
#define DRV_NAME "kirkwood-i2s"
#define KIRKWOOD_I2S_RATES \
(SNDRV_PCM_RATE_44100 | \
SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
#define KIRKWOOD_I2S_FORMATS \
(SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_S24_LE | \
SNDRV_PCM_FMTBIT_S32_LE)
struct snd_soc_dai kirkwood_i2s_dai;
static struct kirkwood_dma_data *priv;
static int kirkwood_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
unsigned int fmt)
{
unsigned long mask;
unsigned long value;
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_RIGHT_J:
mask = KIRKWOOD_I2S_CTL_RJ;
break;
case SND_SOC_DAIFMT_LEFT_J:
mask = KIRKWOOD_I2S_CTL_LJ;
break;
case SND_SOC_DAIFMT_I2S:
mask = KIRKWOOD_I2S_CTL_I2S;
break;
default:
return -EINVAL;
}
/*
* Set same format for playback and record
* This avoids some troubles.
*/
value = readl(priv->io+KIRKWOOD_I2S_PLAYCTL);
value &= ~KIRKWOOD_I2S_CTL_JUST_MASK;
value |= mask;
writel(value, priv->io+KIRKWOOD_I2S_PLAYCTL);
value = readl(priv->io+KIRKWOOD_I2S_RECCTL);
value &= ~KIRKWOOD_I2S_CTL_JUST_MASK;
value |= mask;
writel(value, priv->io+KIRKWOOD_I2S_RECCTL);
return 0;
}
static inline void kirkwood_set_dco(void __iomem *io, unsigned long rate)
{
unsigned long value;
value = KIRKWOOD_DCO_CTL_OFFSET_0;
switch (rate) {
default:
case 44100:
value |= KIRKWOOD_DCO_CTL_FREQ_11;
break;
case 48000:
value |= KIRKWOOD_DCO_CTL_FREQ_12;
break;
case 96000:
value |= KIRKWOOD_DCO_CTL_FREQ_24;
break;
}
writel(value, io + KIRKWOOD_DCO_CTL);
/* wait for dco locked */
do {
cpu_relax();
value = readl(io + KIRKWOOD_DCO_SPCR_STATUS);
value &= KIRKWOOD_DCO_SPCR_STATUS;
} while (value == 0);
}
static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
unsigned int i2s_reg, reg;
unsigned long i2s_value, value;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
i2s_reg = KIRKWOOD_I2S_PLAYCTL;
reg = KIRKWOOD_PLAYCTL;
} else {
i2s_reg = KIRKWOOD_I2S_RECCTL;
reg = KIRKWOOD_RECCTL;
}
/* set dco conf */
kirkwood_set_dco(priv->io, params_rate(params));
i2s_value = readl(priv->io+i2s_reg);
i2s_value &= ~KIRKWOOD_I2S_CTL_SIZE_MASK;
value = readl(priv->io+reg);
value &= ~KIRKWOOD_PLAYCTL_SIZE_MASK;
/*
* Size settings in play/rec i2s control regs and play/rec control
* regs must be the same.
*/
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
i2s_value |= KIRKWOOD_I2S_CTL_SIZE_16;
value |= KIRKWOOD_PLAYCTL_SIZE_16_C;
break;
/*
* doesn't work... S20_3LE != kirkwood 20bit format ?
*
case SNDRV_PCM_FORMAT_S20_3LE:
i2s_value |= KIRKWOOD_I2S_CTL_SIZE_20;
value |= KIRKWOOD_PLAYCTL_SIZE_20;
break;
*/
case SNDRV_PCM_FORMAT_S24_LE:
i2s_value |= KIRKWOOD_I2S_CTL_SIZE_24;
value |= KIRKWOOD_PLAYCTL_SIZE_24;
break;
case SNDRV_PCM_FORMAT_S32_LE:
i2s_value |= KIRKWOOD_I2S_CTL_SIZE_32;
value |= KIRKWOOD_PLAYCTL_SIZE_32;
break;
default:
return -EINVAL;
}
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
value &= ~KIRKWOOD_PLAYCTL_MONO_MASK;
if (params_channels(params) == 1)
value |= KIRKWOOD_PLAYCTL_MONO_BOTH;
else
value |= KIRKWOOD_PLAYCTL_MONO_OFF;
}
writel(i2s_value, priv->io+i2s_reg);
writel(value, priv->io+reg);
return 0;
}
static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream,
int cmd, struct snd_soc_dai *dai)
{
unsigned long value;
/*
* specs says KIRKWOOD_PLAYCTL must be read 2 times before
* changing it. So read 1 time here and 1 later.
*/
value = readl(priv->io + KIRKWOOD_PLAYCTL);
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
/* stop audio, enable interrupts */
value = readl(priv->io + KIRKWOOD_PLAYCTL);
value |= KIRKWOOD_PLAYCTL_PAUSE;
writel(value, priv->io + KIRKWOOD_PLAYCTL);
value = readl(priv->io + KIRKWOOD_INT_MASK);
value |= KIRKWOOD_INT_CAUSE_PLAY_BYTES;
writel(value, priv->io + KIRKWOOD_INT_MASK);
/* configure audio & enable i2s playback */
value = readl(priv->io + KIRKWOOD_PLAYCTL);
value &= ~KIRKWOOD_PLAYCTL_BURST_MASK;
value &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE
| KIRKWOOD_PLAYCTL_SPDIF_EN);
if (priv->burst == 32)
value |= KIRKWOOD_PLAYCTL_BURST_32;
else
value |= KIRKWOOD_PLAYCTL_BURST_128;
value |= KIRKWOOD_PLAYCTL_I2S_EN;
writel(value, priv->io + KIRKWOOD_PLAYCTL);
break;
case SNDRV_PCM_TRIGGER_STOP:
/* stop audio, disable interrupts */
value = readl(priv->io + KIRKWOOD_PLAYCTL);
value |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE;
writel(value, priv->io + KIRKWOOD_PLAYCTL);
value = readl(priv->io + KIRKWOOD_INT_MASK);
value &= ~KIRKWOOD_INT_CAUSE_PLAY_BYTES;
writel(value, priv->io + KIRKWOOD_INT_MASK);
/* disable all playbacks */
value = readl(priv->io + KIRKWOOD_PLAYCTL);
value &= ~(KIRKWOOD_PLAYCTL_I2S_EN | KIRKWOOD_PLAYCTL_SPDIF_EN);
writel(value, priv->io + KIRKWOOD_PLAYCTL);
break;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
case SNDRV_PCM_TRIGGER_SUSPEND:
value = readl(priv->io + KIRKWOOD_PLAYCTL);
value |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE;
writel(value, priv->io + KIRKWOOD_PLAYCTL);
break;
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
value = readl(priv->io + KIRKWOOD_PLAYCTL);
value &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE);
writel(value, priv->io + KIRKWOOD_PLAYCTL);
break;
default:
return -EINVAL;
}
return 0;
}
static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream,
int cmd, struct snd_soc_dai *dai)
{
unsigned long value;
value = readl(priv->io + KIRKWOOD_RECCTL);
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
/* stop audio, enable interrupts */
value = readl(priv->io + KIRKWOOD_RECCTL);
value |= KIRKWOOD_RECCTL_PAUSE;
writel(value, priv->io + KIRKWOOD_RECCTL);
value = readl(priv->io + KIRKWOOD_INT_MASK);
value |= KIRKWOOD_INT_CAUSE_REC_BYTES;
writel(value, priv->io + KIRKWOOD_INT_MASK);
/* configure audio & enable i2s record */
value = readl(priv->io + KIRKWOOD_RECCTL);
value &= ~KIRKWOOD_RECCTL_BURST_MASK;
value &= ~KIRKWOOD_RECCTL_MONO;
value &= ~(KIRKWOOD_RECCTL_PAUSE | KIRKWOOD_RECCTL_MUTE
| KIRKWOOD_RECCTL_SPDIF_EN);
if (priv->burst == 32)
value |= KIRKWOOD_RECCTL_BURST_32;
else
value |= KIRKWOOD_RECCTL_BURST_128;
value |= KIRKWOOD_RECCTL_I2S_EN;
writel(value, priv->io + KIRKWOOD_RECCTL);
break;
case SNDRV_PCM_TRIGGER_STOP:
/* stop audio, disable interrupts */
value = readl(priv->io + KIRKWOOD_RECCTL);
value |= KIRKWOOD_RECCTL_PAUSE | KIRKWOOD_RECCTL_MUTE;
writel(value, priv->io + KIRKWOOD_RECCTL);
value = readl(priv->io + KIRKWOOD_INT_MASK);
value &= ~KIRKWOOD_INT_CAUSE_REC_BYTES;
writel(value, priv->io + KIRKWOOD_INT_MASK);
/* disable all records */
value = readl(priv->io + KIRKWOOD_RECCTL);
value &= ~(KIRKWOOD_RECCTL_I2S_EN | KIRKWOOD_RECCTL_SPDIF_EN);
writel(value, priv->io + KIRKWOOD_RECCTL);
break;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
case SNDRV_PCM_TRIGGER_SUSPEND:
value = readl(priv->io + KIRKWOOD_RECCTL);
value |= KIRKWOOD_RECCTL_PAUSE | KIRKWOOD_RECCTL_MUTE;
writel(value, priv->io + KIRKWOOD_RECCTL);
break;
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
value = readl(priv->io + KIRKWOOD_RECCTL);
value &= ~(KIRKWOOD_RECCTL_PAUSE | KIRKWOOD_RECCTL_MUTE);
writel(value, priv->io + KIRKWOOD_RECCTL);
break;
default:
return -EINVAL;
}
return 0;
}
static int kirkwood_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *dai)
{
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
return kirkwood_i2s_play_trigger(substream, cmd, dai);
else
return kirkwood_i2s_rec_trigger(substream, cmd, dai);
return 0;
}
static int kirkwood_i2s_probe(struct platform_device *pdev,
struct snd_soc_dai *dai)
{
unsigned long value;
unsigned int reg_data;
/* put system in a "safe" state : */
/* disable audio interrupts */
writel(0xffffffff, priv->io + KIRKWOOD_INT_CAUSE);
writel(0, priv->io + KIRKWOOD_INT_MASK);
reg_data = readl(priv->io + 0x1200);
reg_data &= (~(0x333FF8));
reg_data |= 0x111D18;
writel(reg_data, priv->io + 0x1200);
msleep(500);
reg_data = readl(priv->io + 0x1200);
reg_data &= (~(0x333FF8));
reg_data |= 0x111D18;
writel(reg_data, priv->io + 0x1200);
/* disable playback/record */
value = readl(priv->io + KIRKWOOD_PLAYCTL);
value &= ~(KIRKWOOD_PLAYCTL_I2S_EN|KIRKWOOD_PLAYCTL_SPDIF_EN);
writel(value, priv->io + KIRKWOOD_PLAYCTL);
value = readl(priv->io + KIRKWOOD_RECCTL);
value &= ~(KIRKWOOD_RECCTL_I2S_EN | KIRKWOOD_RECCTL_SPDIF_EN);
writel(value, priv->io + KIRKWOOD_RECCTL);
return 0;
}
static void kirkwood_i2s_remove(struct platform_device *pdev,
struct snd_soc_dai *dai)
{
}
static struct snd_soc_dai_ops kirkwood_i2s_dai_ops = {
.trigger = kirkwood_i2s_trigger,
.hw_params = kirkwood_i2s_hw_params,
.set_fmt = kirkwood_i2s_set_fmt,
};
struct snd_soc_dai kirkwood_i2s_dai = {
.name = DRV_NAME,
.id = 0,
.probe = kirkwood_i2s_probe,
.remove = kirkwood_i2s_remove,
.playback = {
.channels_min = 1,
.channels_max = 2,
.rates = KIRKWOOD_I2S_RATES,
.formats = KIRKWOOD_I2S_FORMATS,},
.capture = {
.channels_min = 1,
.channels_max = 2,
.rates = KIRKWOOD_I2S_RATES,
.formats = KIRKWOOD_I2S_FORMATS,},
.ops = &kirkwood_i2s_dai_ops,
};
EXPORT_SYMBOL_GPL(kirkwood_i2s_dai);
static __devinit int kirkwood_i2s_dev_probe(struct platform_device *pdev)
{
struct resource *mem;
struct kirkwood_asoc_platform_data *data =
pdev->dev.platform_data;
int err;
priv = kzalloc(sizeof(struct kirkwood_dma_data), GFP_KERNEL);
if (!priv) {
dev_err(&pdev->dev, "allocation failed\n");
err = -ENOMEM;
goto error;
}
mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (!mem) {
dev_err(&pdev->dev, "platform_get_resource failed\n");
err = -ENXIO;
goto err_alloc;
}
priv->mem = request_mem_region(mem->start, SZ_16K, DRV_NAME);
if (!priv->mem) {
dev_err(&pdev->dev, "request_mem_region failed\n");
err = -EBUSY;
goto error;
}
priv->io = ioremap(priv->mem->start, SZ_16K);
if (!priv->io) {
dev_err(&pdev->dev, "ioremap failed\n");
err = -ENOMEM;
goto err_iomem;
}
priv->irq = platform_get_irq(pdev, 0);
if (priv->irq <= 0) {
dev_err(&pdev->dev, "platform_get_irq failed\n");
err = -ENXIO;
goto err_ioremap;
}
if (!data || !data->dram) {
dev_err(&pdev->dev, "no platform data ?!\n");
err = -EINVAL;
goto err_ioremap;
}
priv->dram = data->dram;
priv->burst = data->burst;
kirkwood_i2s_dai.capture.dma_data = priv;
kirkwood_i2s_dai.playback.dma_data = priv;
return snd_soc_register_dai(&kirkwood_i2s_dai);
err_ioremap:
iounmap(priv->io);
err_iomem:
release_mem_region(priv->mem->start, SZ_16K);
err_alloc:
kfree(priv);
error:
return err;
}
static __devexit int kirkwood_i2s_dev_remove(struct platform_device *pdev)
{
if (priv) {
iounmap(priv->io);
release_mem_region(priv->mem->start, SZ_16K);
kfree(priv);
}
snd_soc_unregister_dai(&kirkwood_i2s_dai);
return 0;
}
static struct platform_driver kirkwood_i2s_driver = {
.probe = kirkwood_i2s_dev_probe,
.remove = kirkwood_i2s_dev_remove,
.driver = {
.name = DRV_NAME,
.owner = THIS_MODULE,
},
};
static int __init kirkwood_i2s_init(void)
{
return platform_driver_register(&kirkwood_i2s_driver);
}
module_init(kirkwood_i2s_init);
static void __exit kirkwood_i2s_exit(void)
{
platform_driver_unregister(&kirkwood_i2s_driver);
}
module_exit(kirkwood_i2s_exit);
/* Module information */
MODULE_AUTHOR("Arnaud Patard, <[email protected]>");
MODULE_DESCRIPTION("Kirkwood I2S SoC Interface");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:kirkwood-i2s");

17
sound/soc/kirkwood/kirkwood-i2s.h Normal file
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/*
* kirkwood-i2s.h
*
* (c) 2010 Arnaud Patard <apatard@mandriva.com>
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the
* Free Software Foundation; either version 2 of the License, or (at your
* option) any later version.
*/
#ifndef _KIRKWOOD_I2S_H
#define _KIRKWOOD_I2S_H
extern struct snd_soc_dai kirkwood_i2s_dai;
#endif

126
sound/soc/kirkwood/kirkwood-openrd.c Normal file
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/*
* kirkwood-openrd.c
*
* (c) 2010 Arnaud Patard <apatard@mandriva.com>
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the
* Free Software Foundation; either version 2 of the License, or (at your
* option) any later version.
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/interrupt.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <sound/soc.h>
#include <mach/kirkwood.h>
#include <plat/audio.h>
#include <asm/mach-types.h>
#include "kirkwood-i2s.h"
#include "kirkwood-dma.h"
#include "../codecs/cs42l51.h"
static int openrd_client_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;
int ret;
unsigned int freq, fmt;
fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS;
ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
if (ret < 0)
return ret;
ret = snd_soc_dai_set_fmt(codec_dai, fmt);
if (ret < 0)
return ret;
switch (params_rate(params)) {
default:
case 44100:
freq = 11289600;
break;
case 48000:
freq = 12288000;
break;
case 96000:
freq = 24576000;
break;
}
return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN);
}
static struct snd_soc_ops openrd_client_ops = {
.hw_params = openrd_client_hw_params,
};
static struct snd_soc_dai_link openrd_client_dai[] = {
{
.name = "CS42L51",
.stream_name = "CS42L51 HiFi",
.cpu_dai = &kirkwood_i2s_dai,
.codec_dai = &cs42l51_dai,
.ops = &openrd_client_ops,
},
};
static struct snd_soc_card openrd_client = {
.name = "OpenRD Client",
.platform = &kirkwood_soc_platform,
.dai_link = openrd_client_dai,
.num_links = ARRAY_SIZE(openrd_client_dai),
};
static struct snd_soc_device openrd_client_snd_devdata = {
.card = &openrd_client,
.codec_dev = &soc_codec_device_cs42l51,
};
static struct platform_device *openrd_client_snd_device;
static int __init openrd_client_init(void)
{
int ret;
if (!machine_is_openrd_client())
return 0;
openrd_client_snd_device = platform_device_alloc("soc-audio", -1);
if (!openrd_client_snd_device)
return -ENOMEM;
platform_set_drvdata(openrd_client_snd_device,
&openrd_client_snd_devdata);
openrd_client_snd_devdata.dev = &openrd_client_snd_device->dev;
ret = platform_device_add(openrd_client_snd_device);
if (ret) {
printk(KERN_ERR "%s: platform_device_add failed\n", __func__);
platform_device_put(openrd_client_snd_device);
}
return ret;
}
static void __exit openrd_client_exit(void)
{
platform_device_unregister(openrd_client_snd_device);
}
module_init(openrd_client_init);
module_exit(openrd_client_exit);
/* Module information */
MODULE_AUTHOR("Arnaud Patard <[email protected]>");
MODULE_DESCRIPTION("ALSA SoC OpenRD Client");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:soc-audio");

129
sound/soc/kirkwood/kirkwood.h Normal file
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/*
* kirkwood.h
*
* (c) 2010 Arnaud Patard <apatard@mandriva.com>
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the
* Free Software Foundation; either version 2 of the License, or (at your
* option) any later version.
*/
#ifndef _KIRKWOOD_AUDIO_H
#define _KIRKWOOD_AUDIO_H
#define KIRKWOOD_RECORD_WIN 0
#define KIRKWOOD_PLAYBACK_WIN 1
#define KIRKWOOD_MAX_AUDIO_WIN 2
#define KIRKWOOD_AUDIO_WIN_BASE_REG(win) (0xA00 + ((win)<<3))
#define KIRKWOOD_AUDIO_WIN_CTRL_REG(win) (0xA04 + ((win)<<3))
#define KIRKWOOD_RECCTL 0x1000
#define KIRKWOOD_RECCTL_SPDIF_EN (1<<11)
#define KIRKWOOD_RECCTL_I2S_EN (1<<10)
#define KIRKWOOD_RECCTL_PAUSE (1<<9)
#define KIRKWOOD_RECCTL_MUTE (1<<8)
#define KIRKWOOD_RECCTL_BURST_MASK (3<<5)
#define KIRKWOOD_RECCTL_BURST_128 (2<<5)
#define KIRKWOOD_RECCTL_BURST_32 (1<<5)
#define KIRKWOOD_RECCTL_MONO (1<<4)
#define KIRKWOOD_RECCTL_MONO_CHAN_RIGHT (1<<3)
#define KIRKWOOD_RECCTL_MONO_CHAN_LEFT (0<<3)
#define KIRKWOOD_RECCTL_SIZE_MASK (7<<0)
#define KIRKWOOD_RECCTL_SIZE_16 (7<<0)
#define KIRKWOOD_RECCTL_SIZE_16_C (3<<0)
#define KIRKWOOD_RECCTL_SIZE_20 (2<<0)
#define KIRKWOOD_RECCTL_SIZE_24 (1<<0)
#define KIRKWOOD_RECCTL_SIZE_32 (0<<0)
#define KIRKWOOD_REC_BUF_ADDR 0x1004
#define KIRKWOOD_REC_BUF_SIZE 0x1008
#define KIRKWOOD_REC_BYTE_COUNT 0x100C
#define KIRKWOOD_PLAYCTL 0x1100
#define KIRKWOOD_PLAYCTL_PLAY_BUSY (1<<16)
#define KIRKWOOD_PLAYCTL_BURST_MASK (3<<11)
#define KIRKWOOD_PLAYCTL_BURST_128 (2<<11)
#define KIRKWOOD_PLAYCTL_BURST_32 (1<<11)
#define KIRKWOOD_PLAYCTL_PAUSE (1<<9)
#define KIRKWOOD_PLAYCTL_SPDIF_MUTE (1<<8)
#define KIRKWOOD_PLAYCTL_MONO_MASK (3<<5)
#define KIRKWOOD_PLAYCTL_MONO_BOTH (3<<5)
#define KIRKWOOD_PLAYCTL_MONO_OFF (0<<5)
#define KIRKWOOD_PLAYCTL_I2S_MUTE (1<<7)
#define KIRKWOOD_PLAYCTL_SPDIF_EN (1<<4)
#define KIRKWOOD_PLAYCTL_I2S_EN (1<<3)
#define KIRKWOOD_PLAYCTL_SIZE_MASK (7<<0)
#define KIRKWOOD_PLAYCTL_SIZE_16 (7<<0)
#define KIRKWOOD_PLAYCTL_SIZE_16_C (3<<0)
#define KIRKWOOD_PLAYCTL_SIZE_20 (2<<0)
#define KIRKWOOD_PLAYCTL_SIZE_24 (1<<0)
#define KIRKWOOD_PLAYCTL_SIZE_32 (0<<0)
#define KIRKWOOD_PLAY_BUF_ADDR 0x1104
#define KIRKWOOD_PLAY_BUF_SIZE 0x1108
#define KIRKWOOD_PLAY_BYTE_COUNT 0x110C
#define KIRKWOOD_DCO_CTL 0x1204
#define KIRKWOOD_DCO_CTL_OFFSET_MASK (0xFFF<<2)
#define KIRKWOOD_DCO_CTL_OFFSET_0 (0x800<<2)
#define KIRKWOOD_DCO_CTL_FREQ_MASK (3<<0)
#define KIRKWOOD_DCO_CTL_FREQ_11 (0<<0)
#define KIRKWOOD_DCO_CTL_FREQ_12 (1<<0)
#define KIRKWOOD_DCO_CTL_FREQ_24 (2<<0)
#define KIRKWOOD_DCO_SPCR_STATUS 0x120c
#define KIRKWOOD_DCO_SPCR_STATUS_DCO_LOCK (1<<16)
#define KIRKWOOD_ERR_CAUSE 0x1300
#define KIRKWOOD_ERR_MASK 0x1304
#define KIRKWOOD_INT_CAUSE 0x1308
#define KIRKWOOD_INT_MASK 0x130C
#define KIRKWOOD_INT_CAUSE_PLAY_BYTES (1<<14)
#define KIRKWOOD_INT_CAUSE_REC_BYTES (1<<13)
#define KIRKWOOD_INT_CAUSE_DMA_PLAY_END (1<<7)
#define KIRKWOOD_INT_CAUSE_DMA_PLAY_3Q (1<<6)
#define KIRKWOOD_INT_CAUSE_DMA_PLAY_HALF (1<<5)
#define KIRKWOOD_INT_CAUSE_DMA_PLAY_1Q (1<<4)
#define KIRKWOOD_INT_CAUSE_DMA_REC_END (1<<3)
#define KIRKWOOD_INT_CAUSE_DMA_REC_3Q (1<<2)
#define KIRKWOOD_INT_CAUSE_DMA_REC_HALF (1<<1)
#define KIRKWOOD_INT_CAUSE_DMA_REC_1Q (1<<0)
#define KIRKWOOD_REC_BYTE_INT_COUNT 0x1310
#define KIRKWOOD_PLAY_BYTE_INT_COUNT 0x1314
#define KIRKWOOD_BYTE_INT_COUNT_MASK 0xffffff
#define KIRKWOOD_I2S_PLAYCTL 0x2508
#define KIRKWOOD_I2S_RECCTL 0x2408
#define KIRKWOOD_I2S_CTL_JUST_MASK (0xf<<26)
#define KIRKWOOD_I2S_CTL_LJ (0<<26)
#define KIRKWOOD_I2S_CTL_I2S (5<<26)
#define KIRKWOOD_I2S_CTL_RJ (8<<26)
#define KIRKWOOD_I2S_CTL_SIZE_MASK (3<<30)
#define KIRKWOOD_I2S_CTL_SIZE_16 (3<<30)
#define KIRKWOOD_I2S_CTL_SIZE_20 (2<<30)
#define KIRKWOOD_I2S_CTL_SIZE_24 (1<<30)
#define KIRKWOOD_I2S_CTL_SIZE_32 (0<<30)
#define KIRKWOOD_AUDIO_BUF_MAX (16*1024*1024)
/* Theses values come from the marvell alsa driver */
/* need to find where they come from */
#define KIRKWOOD_SND_MIN_PERIODS 8
#define KIRKWOOD_SND_MAX_PERIODS 16
#define KIRKWOOD_SND_MIN_PERIOD_BYTES 0x4000
#define KIRKWOOD_SND_MAX_PERIOD_BYTES 0x4000
struct kirkwood_dma_data {
struct resource *mem;
void __iomem *io;
int irq;
int burst;
struct mbus_dram_target_info *dram;
};
#endif

27
sound/soc/nuc900/Kconfig Normal file
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##
## NUC900 series AC97 API
##
config SND_SOC_NUC900
tristate "SoC Audio for NUC900 series"
depends on ARCH_W90X900
help
This option enables support for AC97 mode on the NUC900 SoC.
config SND_SOC_NUC900_AC97
tristate
select AC97_BUS
select SND_AC97_CODEC
select SND_SOC_AC97_BUS
##
## Boards
##
config SND_SOC_NUC900EVB
tristate "NUC900 AC97 support for demo board"
depends on SND_SOC_NUC900
select SND_SOC_NUC900_AC97
select SND_SOC_AC97_CODEC
help
Select this option to enable audio (AC97) on the
NUC900 demoboard.

11
sound/soc/nuc900/Makefile Normal file
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# NUC900 series audio
snd-soc-nuc900-pcm-objs := nuc900-pcm.o
snd-soc-nuc900-ac97-objs := nuc900-ac97.o
obj-$(CONFIG_SND_SOC_NUC900) += snd-soc-nuc900-pcm.o
obj-$(CONFIG_SND_SOC_NUC900_AC97) += snd-soc-nuc900-ac97.o
# Boards
snd-soc-nuc900-audio-objs := nuc900-audio.o
obj-$(CONFIG_SND_SOC_NUC900EVB) += snd-soc-nuc900-audio.o

430
sound/soc/nuc900/nuc900-ac97.c Normal file
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/*
* Copyright (c) 2009-2010 Nuvoton technology corporation.
*
* Wan ZongShun <mcuos.com@gmail.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation;version 2 of the License.
*
*/
#include <linux/init.h>
#include <linux/module.h>
#include <linux/slab.h>
#include <linux/device.h>
#include <linux/delay.h>
#include <linux/mutex.h>
#include <linux/suspend.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/initval.h>
#include <sound/soc.h>
#include <linux/device.h>
#include <linux/clk.h>
#include <mach/mfp.h>
#include "nuc900-audio.h"
static DEFINE_MUTEX(ac97_mutex);
struct nuc900_audio *nuc900_ac97_data;
static int nuc900_checkready(void)
{
struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
if (!(AUDIO_READ(nuc900_audio->mmio + ACTL_ACIS0) & CODEC_READY))
return -EPERM;
return 0;
}
/* AC97 controller reads codec register */
static unsigned short nuc900_ac97_read(struct snd_ac97 *ac97,
unsigned short reg)
{
struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
unsigned long timeout = 0x10000, val;
mutex_lock(&ac97_mutex);
val = nuc900_checkready();
if (!!val) {
dev_err(nuc900_audio->dev, "AC97 codec is not ready\n");
goto out;
}
/* set the R_WB bit and write register index */
AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS1, R_WB | reg);
/* set the valid frame bit and valid slots */
val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0);
val |= (VALID_FRAME | SLOT1_VALID);
AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, val);
udelay(100);
/* polling the AC_R_FINISH */
while (!(AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON) & AC_R_FINISH)
&& timeout--)
mdelay(1);
if (!timeout) {
dev_err(nuc900_audio->dev, "AC97 read register time out !\n");
val = -EPERM;
goto out;
}
val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0) ;
val &= ~SLOT1_VALID;
AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, val);
if (AUDIO_READ(nuc900_audio->mmio + ACTL_ACIS1) >> 2 != reg) {
dev_err(nuc900_audio->dev,
"R_INDEX of REG_ACTL_ACIS1 not match!\n");
}
udelay(100);
val = (AUDIO_READ(nuc900_audio->mmio + ACTL_ACIS2) & 0xFFFF);
out:
mutex_unlock(&ac97_mutex);
return val;
}
/* AC97 controller writes to codec register */
static void nuc900_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
unsigned short val)
{
struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
unsigned long tmp, timeout = 0x10000;
mutex_lock(&ac97_mutex);
tmp = nuc900_checkready();
if (!!tmp)
dev_err(nuc900_audio->dev, "AC97 codec is not ready\n");
/* clear the R_WB bit and write register index */
AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS1, reg);
/* write register value */
AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS2, val);
/* set the valid frame bit and valid slots */
tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0);
tmp |= SLOT1_VALID | SLOT2_VALID | VALID_FRAME;
AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, tmp);
udelay(100);
/* polling the AC_W_FINISH */
while ((AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON) & AC_W_FINISH)
&& timeout--)
mdelay(1);
if (!timeout)
dev_err(nuc900_audio->dev, "AC97 write register time out !\n");
tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0);
tmp &= ~(SLOT1_VALID | SLOT2_VALID);
AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, tmp);
mutex_unlock(&ac97_mutex);
}
static void nuc900_ac97_warm_reset(struct snd_ac97 *ac97)
{
struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
unsigned long val;
mutex_lock(&ac97_mutex);
/* warm reset AC 97 */
val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON);
val |= AC_W_RES;
AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACCON, val);
udelay(100);
val = nuc900_checkready();
if (!!val)
dev_err(nuc900_audio->dev, "AC97 codec is not ready\n");
mutex_unlock(&ac97_mutex);
}
static void nuc900_ac97_cold_reset(struct snd_ac97 *ac97)
{
struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
unsigned long val;
mutex_lock(&ac97_mutex);
/* reset Audio Controller */
val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET);
val |= ACTL_RESET_BIT;
AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val);
val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET);
val &= (~ACTL_RESET_BIT);
AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val);
/* reset AC-link interface */
val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET);
val |= AC_RESET;
AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val);
val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET);
val &= ~AC_RESET;
AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val);
/* cold reset AC 97 */
val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON);
val |= AC_C_RES;
AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACCON, val);
val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON);
val &= (~AC_C_RES);
AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACCON, val);
udelay(100);
mutex_unlock(&ac97_mutex);
}
/* AC97 controller operations */
struct snd_ac97_bus_ops soc_ac97_ops = {
.read = nuc900_ac97_read,
.write = nuc900_ac97_write,
.reset = nuc900_ac97_cold_reset,
.warm_reset = nuc900_ac97_warm_reset,
}
EXPORT_SYMBOL_GPL(soc_ac97_ops);
static int nuc900_ac97_trigger(struct snd_pcm_substream *substream,
int cmd, struct snd_soc_dai *dai)
{
struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
int ret;
unsigned long val, tmp;
ret = 0;
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0);
tmp |= (SLOT3_VALID | SLOT4_VALID | VALID_FRAME);
AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, tmp);
tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_PSR);
tmp |= (P_DMA_END_IRQ | P_DMA_MIDDLE_IRQ);
AUDIO_WRITE(nuc900_audio->mmio + ACTL_PSR, tmp);
val |= AC_PLAY;
} else {
tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_RSR);
tmp |= (R_DMA_END_IRQ | R_DMA_MIDDLE_IRQ);
AUDIO_WRITE(nuc900_audio->mmio + ACTL_RSR, tmp);
val |= AC_RECORD;
}
AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0);
tmp &= ~(SLOT3_VALID | SLOT4_VALID);
AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, tmp);
AUDIO_WRITE(nuc900_audio->mmio + ACTL_PSR, RESET_PRSR);
val &= ~AC_PLAY;
} else {
AUDIO_WRITE(nuc900_audio->mmio + ACTL_RSR, RESET_PRSR);
val &= ~AC_RECORD;
}
AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val);
break;
default:
ret = -EINVAL;
}
return ret;
}
static int nuc900_ac97_probe(struct platform_device *pdev,
struct snd_soc_dai *dai)
{
struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
unsigned long val;
mutex_lock(&ac97_mutex);
/* enable unit clock */
clk_enable(nuc900_audio->clk);
/* enable audio controller and AC-link interface */
val = AUDIO_READ(nuc900_audio->mmio + ACTL_CON);
val |= (IIS_AC_PIN_SEL | ACLINK_EN);
AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val);
mutex_unlock(&ac97_mutex);
return 0;
}
static void nuc900_ac97_remove(struct platform_device *pdev,
struct snd_soc_dai *dai)
{
struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
clk_disable(nuc900_audio->clk);
}
static struct snd_soc_dai_ops nuc900_ac97_dai_ops = {
.trigger = nuc900_ac97_trigger,
};
struct snd_soc_dai nuc900_ac97_dai = {
.name = "nuc900-ac97",
.probe = nuc900_ac97_probe,
.remove = nuc900_ac97_remove,
.ac97_control = 1,
.playback = {
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.channels_min = 1,
.channels_max = 2,
},
.capture = {
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.channels_min = 1,
.channels_max = 2,
},
.ops = &nuc900_ac97_dai_ops,
}
EXPORT_SYMBOL_GPL(nuc900_ac97_dai);
static int __devinit nuc900_ac97_drvprobe(struct platform_device *pdev)
{
struct nuc900_audio *nuc900_audio;
int ret;
if (nuc900_ac97_data)
return -EBUSY;
nuc900_audio = kzalloc(sizeof(struct nuc900_audio), GFP_KERNEL);
if (!nuc900_audio)
return -ENOMEM;
spin_lock_init(&nuc900_audio->lock);
nuc900_audio->res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (!nuc900_audio->res) {
ret = -ENODEV;
goto out0;
}
if (!request_mem_region(nuc900_audio->res->start,
resource_size(nuc900_audio->res), pdev->name)) {
ret = -EBUSY;
goto out0;
}
nuc900_audio->mmio = ioremap(nuc900_audio->res->start,
resource_size(nuc900_audio->res));
if (!nuc900_audio->mmio) {
ret = -ENOMEM;
goto out1;
}
nuc900_audio->clk = clk_get(&pdev->dev, NULL);
if (IS_ERR(nuc900_audio->clk)) {
ret = PTR_ERR(nuc900_audio->clk);
goto out2;
}
nuc900_audio->irq_num = platform_get_irq(pdev, 0);
if (!nuc900_audio->irq_num) {
ret = -EBUSY;
goto out2;
}
nuc900_ac97_data = nuc900_audio;
nuc900_audio->dev = nuc900_ac97_dai.dev = &pdev->dev;
ret = snd_soc_register_dai(&nuc900_ac97_dai);
if (ret)
goto out3;
mfp_set_groupg(nuc900_audio->dev); /* enbale ac97 multifunction pin*/
return 0;
out3:
clk_put(nuc900_audio->clk);
out2:
iounmap(nuc900_audio->mmio);
out1:
release_mem_region(nuc900_audio->res->start,
resource_size(nuc900_audio->res));
out0:
kfree(nuc900_audio);
return ret;
}
static int __devexit nuc900_ac97_drvremove(struct platform_device *pdev)
{
snd_soc_unregister_dai(&nuc900_ac97_dai);
clk_put(nuc900_ac97_data->clk);
iounmap(nuc900_ac97_data->mmio);
release_mem_region(nuc900_ac97_data->res->start,
resource_size(nuc900_ac97_data->res));
nuc900_ac97_data = NULL;
return 0;
}
static struct platform_driver nuc900_ac97_driver = {
.driver = {
.name = "nuc900-audio",
.owner = THIS_MODULE,
},
.probe = nuc900_ac97_drvprobe,
.remove = __devexit_p(nuc900_ac97_drvremove),
};
static int __init nuc900_ac97_init(void)
{
return platform_driver_register(&nuc900_ac97_driver);
}
static void __exit nuc900_ac97_exit(void)
{
platform_driver_unregister(&nuc900_ac97_driver);
}
module_init(nuc900_ac97_init);
module_exit(nuc900_ac97_exit);
MODULE_AUTHOR("Wan ZongShun <[email protected]>");
MODULE_DESCRIPTION("NUC900 AC97 SoC driver!");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:nuc900-ac97");

81
sound/soc/nuc900/nuc900-audio.c Normal file
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/*
* Copyright (c) 2010 Nuvoton technology corporation.
*
* Wan ZongShun <mcuos.com@gmail.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation;version 2 of the License.
*
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/timer.h>
#include <linux/interrupt.h>
#include <linux/platform_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include "../codecs/ac97.h"
#include "nuc900-audio.h"
static struct snd_soc_dai_link nuc900evb_ac97_dai = {
.name = "AC97",
.stream_name = "AC97 HiFi",
.cpu_dai = &nuc900_ac97_dai,
.codec_dai = &ac97_dai,
};
static struct snd_soc_card nuc900evb_audio_machine = {
.name = "NUC900EVB_AC97",
.dai_link = &nuc900evb_ac97_dai,
.num_links = 1,
.platform = &nuc900_soc_platform,
};
static struct snd_soc_device nuc900evb_ac97_devdata = {
.card = &nuc900evb_audio_machine,
.codec_dev = &soc_codec_dev_ac97,
};
static struct platform_device *nuc900evb_asoc_dev;
static int __init nuc900evb_audio_init(void)
{
int ret;
ret = -ENOMEM;
nuc900evb_asoc_dev = platform_device_alloc("soc-audio", -1);
if (!nuc900evb_asoc_dev)
goto out;
/* nuc900 board audio device */
platform_set_drvdata(nuc900evb_asoc_dev, &nuc900evb_ac97_devdata);
nuc900evb_ac97_devdata.dev = &nuc900evb_asoc_dev->dev;
ret = platform_device_add(nuc900evb_asoc_dev);
if (ret) {
platform_device_put(nuc900evb_asoc_dev);
nuc900evb_asoc_dev = NULL;
}
out:
return ret;
}
static void __exit nuc900evb_audio_exit(void)
{
platform_device_unregister(nuc900evb_asoc_dev);
}
module_init(nuc900evb_audio_init);
module_exit(nuc900evb_audio_exit);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("NUC900 Series ASoC audio support");
MODULE_AUTHOR("Wan ZongShun");

117
sound/soc/nuc900/nuc900-audio.h Normal file
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/*
* Copyright (c) 2010 Nuvoton technology corporation.
*
* Wan ZongShun <mcuos.com@gmail.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation;version 2 of the License.
*
*/
#ifndef _NUC900_AUDIO_H
#define _NUC900_AUDIO_H
#include <linux/io.h>
/* Audio Control Registers */
#define ACTL_CON 0x00
#define ACTL_RESET 0x04
#define ACTL_RDSTB 0x08
#define ACTL_RDST_LENGTH 0x0C
#define ACTL_RDSTC 0x10
#define ACTL_RSR 0x14
#define ACTL_PDSTB 0x18
#define ACTL_PDST_LENGTH 0x1C
#define ACTL_PDSTC 0x20
#define ACTL_PSR 0x24
#define ACTL_IISCON 0x28
#define ACTL_ACCON 0x2C
#define ACTL_ACOS0 0x30
#define ACTL_ACOS1 0x34
#define ACTL_ACOS2 0x38
#define ACTL_ACIS0 0x3C
#define ACTL_ACIS1 0x40
#define ACTL_ACIS2 0x44
#define ACTL_COUNTER 0x48
/* bit definition of REG_ACTL_CON register */
#define R_DMA_IRQ 0x1000
#define T_DMA_IRQ 0x0800
#define IIS_AC_PIN_SEL 0x0100
#define FIFO_TH 0x0080
#define ADC_EN 0x0010
#define M80_EN 0x0008
#define ACLINK_EN 0x0004
#define IIS_EN 0x0002
/* bit definition of REG_ACTL_RESET register */
#define W5691_PLAY 0x20000
#define ACTL_RESET_BIT 0x10000
#define RECORD_RIGHT_CHNNEL 0x08000
#define RECORD_LEFT_CHNNEL 0x04000
#define PLAY_RIGHT_CHNNEL 0x02000
#define PLAY_LEFT_CHNNEL 0x01000
#define DAC_PLAY 0x00800
#define ADC_RECORD 0x00400
#define M80_PLAY 0x00200
#define AC_RECORD 0x00100
#define AC_PLAY 0x00080
#define IIS_RECORD 0x00040
#define IIS_PLAY 0x00020
#define DAC_RESET 0x00010
#define ADC_RESET 0x00008
#define M80_RESET 0x00004
#define AC_RESET 0x00002
#define IIS_RESET 0x00001
/* bit definition of REG_ACTL_ACCON register */
#define AC_BCLK_PU_EN 0x20
#define AC_R_FINISH 0x10
#define AC_W_FINISH 0x08
#define AC_W_RES 0x04
#define AC_C_RES 0x02
/* bit definition of ACTL_RSR register */
#define R_FIFO_EMPTY 0x04
#define R_DMA_END_IRQ 0x02
#define R_DMA_MIDDLE_IRQ 0x01
/* bit definition of ACTL_PSR register */
#define P_FIFO_EMPTY 0x04
#define P_DMA_END_IRQ 0x02
#define P_DMA_MIDDLE_IRQ 0x01
/* bit definition of ACTL_ACOS0 register */
#define SLOT1_VALID 0x01
#define SLOT2_VALID 0x02
#define SLOT3_VALID 0x04
#define SLOT4_VALID 0x08
#define VALID_FRAME 0x10
/* bit definition of ACTL_ACOS1 register */
#define R_WB 0x80
#define CODEC_READY 0x10
#define RESET_PRSR 0x00
#define AUDIO_WRITE(addr, val) __raw_writel(val, addr)
#define AUDIO_READ(addr) __raw_readl(addr)
struct nuc900_audio {
void __iomem *mmio;
spinlock_t lock;
dma_addr_t dma_addr[2];
unsigned long buffersize[2];
unsigned long irq_num;
struct snd_pcm_substream *substream;
struct resource *res;
struct clk *clk;
struct device *dev;
};
extern struct nuc900_audio *nuc900_ac97_data;
extern struct snd_soc_dai nuc900_ac97_dai;
extern struct snd_soc_platform nuc900_soc_platform;
#endif /*end _NUC900_AUDIO_H */

354
sound/soc/nuc900/nuc900-pcm.c Normal file
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/*
* Copyright (c) 2010 Nuvoton technology corporation.
*
* Wan ZongShun <mcuos.com@gmail.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation;version 2 of the License.
*
*/
#include <linux/module.h>
#include <linux/init.h>
#include <linux/io.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <linux/dma-mapping.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <mach/hardware.h>
#include "nuc900-audio.h"
static const struct snd_pcm_hardware nuc900_pcm_hardware = {
.info = SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_PAUSE |
SNDRV_PCM_INFO_RESUME,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.channels_min = 1,
.channels_max = 2,
.buffer_bytes_max = 4*1024,
.period_bytes_min = 1*1024,
.period_bytes_max = 4*1024,
.periods_min = 1,
.periods_max = 1024,
};
static int nuc900_dma_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct nuc900_audio *nuc900_audio = runtime->private_data;
unsigned long flags;
int ret = 0;
spin_lock_irqsave(&nuc900_audio->lock, flags);
ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
if (ret < 0)
return ret;
nuc900_audio->substream = substream;
nuc900_audio->dma_addr[substream->stream] = runtime->dma_addr;
nuc900_audio->buffersize[substream->stream] =
params_buffer_bytes(params);
spin_unlock_irqrestore(&nuc900_audio->lock, flags);
return ret;
}
static void nuc900_update_dma_register(struct snd_pcm_substream *substream,
dma_addr_t dma_addr, size_t count)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct nuc900_audio *nuc900_audio = runtime->private_data;
void __iomem *mmio_addr, *mmio_len;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
mmio_addr = nuc900_audio->mmio + ACTL_PDSTB;
mmio_len = nuc900_audio->mmio + ACTL_PDST_LENGTH;
} else {
mmio_addr = nuc900_audio->mmio + ACTL_RDSTB;
mmio_len = nuc900_audio->mmio + ACTL_RDST_LENGTH;
}
AUDIO_WRITE(mmio_addr, dma_addr);
AUDIO_WRITE(mmio_len, count);
}
static void nuc900_dma_start(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct nuc900_audio *nuc900_audio = runtime->private_data;
unsigned long val;
val = AUDIO_READ(nuc900_audio->mmio + ACTL_CON);
val |= (T_DMA_IRQ | R_DMA_IRQ);
AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val);
}
static void nuc900_dma_stop(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct nuc900_audio *nuc900_audio = runtime->private_data;
unsigned long val;
val = AUDIO_READ(nuc900_audio->mmio + ACTL_CON);
val &= ~(T_DMA_IRQ | R_DMA_IRQ);
AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val);
}
static irqreturn_t nuc900_dma_interrupt(int irq, void *dev_id)
{
struct snd_pcm_substream *substream = dev_id;
struct nuc900_audio *nuc900_audio = substream->runtime->private_data;
unsigned long val;
spin_lock(&nuc900_audio->lock);
val = AUDIO_READ(nuc900_audio->mmio + ACTL_CON);
if (val & R_DMA_IRQ) {
AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val | R_DMA_IRQ);
val = AUDIO_READ(nuc900_audio->mmio + ACTL_RSR);
if (val & R_DMA_MIDDLE_IRQ) {
val |= R_DMA_MIDDLE_IRQ;
AUDIO_WRITE(nuc900_audio->mmio + ACTL_RSR, val);
}
if (val & R_DMA_END_IRQ) {
val |= R_DMA_END_IRQ;
AUDIO_WRITE(nuc900_audio->mmio + ACTL_RSR, val);
}
} else if (val & T_DMA_IRQ) {
AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val | T_DMA_IRQ);
val = AUDIO_READ(nuc900_audio->mmio + ACTL_PSR);
if (val & P_DMA_MIDDLE_IRQ) {
val |= P_DMA_MIDDLE_IRQ;
AUDIO_WRITE(nuc900_audio->mmio + ACTL_PSR, val);
}
if (val & P_DMA_END_IRQ) {
val |= P_DMA_END_IRQ;
AUDIO_WRITE(nuc900_audio->mmio + ACTL_PSR, val);
}
} else {
dev_err(nuc900_audio->dev, "Wrong DMA interrupt status!\n");
spin_unlock(&nuc900_audio->lock);
return IRQ_HANDLED;
}
spin_unlock(&nuc900_audio->lock);
snd_pcm_period_elapsed(substream);
return IRQ_HANDLED;
}
static int nuc900_dma_hw_free(struct snd_pcm_substream *substream)
{
snd_pcm_lib_free_pages(substream);
return 0;
}
static int nuc900_dma_prepare(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct nuc900_audio *nuc900_audio = runtime->private_data;
unsigned long flags, val;
spin_lock_irqsave(&nuc900_audio->lock, flags);
nuc900_update_dma_register(substream,
nuc900_audio->dma_addr[substream->stream],
nuc900_audio->buffersize[substream->stream]);
val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET);
switch (runtime->channels) {
case 1:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
val &= ~(PLAY_LEFT_CHNNEL | PLAY_RIGHT_CHNNEL);
val |= PLAY_RIGHT_CHNNEL;
} else {
val &= ~(RECORD_LEFT_CHNNEL | RECORD_RIGHT_CHNNEL);
val |= RECORD_RIGHT_CHNNEL;
}
AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val);
break;
case 2:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
val |= (PLAY_LEFT_CHNNEL | PLAY_RIGHT_CHNNEL);
else
val |= (RECORD_LEFT_CHNNEL | RECORD_RIGHT_CHNNEL);
AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val);
break;
default:
return -EINVAL;
}
spin_unlock_irqrestore(&nuc900_audio->lock, flags);
return 0;
}
static int nuc900_dma_trigger(struct snd_pcm_substream *substream, int cmd)
{
int ret = 0;
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
nuc900_dma_start(substream);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
nuc900_dma_stop(substream);
break;
default:
ret = -EINVAL;
break;
}
return ret;
}
int nuc900_dma_getposition(struct snd_pcm_substream *substream,
dma_addr_t *src, dma_addr_t *dst)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct nuc900_audio *nuc900_audio = runtime->private_data;
if (src != NULL)
*src = AUDIO_READ(nuc900_audio->mmio + ACTL_PDSTC);
if (dst != NULL)
*dst = AUDIO_READ(nuc900_audio->mmio + ACTL_RDSTC);
return 0;
}
static snd_pcm_uframes_t nuc900_dma_pointer(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
dma_addr_t src, dst;
unsigned long res;
nuc900_dma_getposition(substream, &src, &dst);
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
res = dst - runtime->dma_addr;
else
res = src - runtime->dma_addr;
return bytes_to_frames(substream->runtime, res);
}
static int nuc900_dma_open(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct nuc900_audio *nuc900_audio;
snd_soc_set_runtime_hwparams(substream, &nuc900_pcm_hardware);
nuc900_audio = nuc900_ac97_data;
if (request_irq(nuc900_audio->irq_num, nuc900_dma_interrupt,
IRQF_DISABLED, "nuc900-dma", substream))
return -EBUSY;
runtime->private_data = nuc900_audio;
return 0;
}
static int nuc900_dma_close(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct nuc900_audio *nuc900_audio = runtime->private_data;
free_irq(nuc900_audio->irq_num, substream);
return 0;
}
static int nuc900_dma_mmap(struct snd_pcm_substream *substream,
struct vm_area_struct *vma)
{
struct snd_pcm_runtime *runtime = substream->runtime;
return dma_mmap_writecombine(substream->pcm->card->dev, vma,
runtime->dma_area,
runtime->dma_addr,
runtime->dma_bytes);
}
static struct snd_pcm_ops nuc900_dma_ops = {
.open = nuc900_dma_open,
.close = nuc900_dma_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = nuc900_dma_hw_params,
.hw_free = nuc900_dma_hw_free,
.prepare = nuc900_dma_prepare,
.trigger = nuc900_dma_trigger,
.pointer = nuc900_dma_pointer,
.mmap = nuc900_dma_mmap,
};
static void nuc900_dma_free_dma_buffers(struct snd_pcm *pcm)
{
snd_pcm_lib_preallocate_free_for_all(pcm);
}
static u64 nuc900_pcm_dmamask = DMA_BIT_MASK(32);
static int nuc900_dma_new(struct snd_card *card,
struct snd_soc_dai *dai, struct snd_pcm *pcm)
{
if (!card->dev->dma_mask)
card->dev->dma_mask = &nuc900_pcm_dmamask;
if (!card->dev->coherent_dma_mask)
card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
card->dev, 4 * 1024, (4 * 1024) - 1);
return 0;
}
struct snd_soc_platform nuc900_soc_platform = {
.name = "nuc900-dma",
.pcm_ops = &nuc900_dma_ops,
.pcm_new = nuc900_dma_new,
.pcm_free = nuc900_dma_free_dma_buffers,
}
EXPORT_SYMBOL_GPL(nuc900_soc_platform);
static int __init nuc900_soc_platform_init(void)
{
return snd_soc_register_platform(&nuc900_soc_platform);
}
static void __exit nuc900_soc_platform_exit(void)
{
snd_soc_unregister_platform(&nuc900_soc_platform);
}
module_init(nuc900_soc_platform_init);
module_exit(nuc900_soc_platform_exit);
MODULE_AUTHOR("Wan ZongShun, <[email protected]>");
MODULE_DESCRIPTION("nuc900 Audio DMA module");
MODULE_LICENSE("GPL");

173
sound/soc/omap/omap-mcbsp.c
View file

@ -59,6 +59,7 @@ struct omap_mcbsp_data {
int configured;
unsigned int in_freq;
int clk_div;
int wlen;
};
#define to_mcbsp(priv) container_of((priv), struct omap_mcbsp_data, bus_id)
@ -154,20 +155,51 @@ static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream)
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
struct omap_pcm_dma_data *dma_data;
int dma_op_mode = omap_mcbsp_get_dma_op_mode(mcbsp_data->bus_id);
int samples;
int words;
dma_data = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
/* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
if (dma_op_mode == MCBSP_DMA_MODE_THRESHOLD)
samples = snd_pcm_lib_period_bytes(substream) >> 1;
/*
* Configure McBSP threshold based on either:
* packet_size, when the sDMA is in packet mode, or
* based on the period size.
*/
if (dma_data->packet_size)
words = dma_data->packet_size;
else
words = snd_pcm_lib_period_bytes(substream) /
(mcbsp_data->wlen / 8);
else
samples = 1;
words = 1;
/* Configure McBSP internal buffer usage */
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
omap_mcbsp_set_tx_threshold(mcbsp_data->bus_id, samples - 1);
omap_mcbsp_set_tx_threshold(mcbsp_data->bus_id, words);
else
omap_mcbsp_set_rx_threshold(mcbsp_data->bus_id, samples - 1);
omap_mcbsp_set_rx_threshold(mcbsp_data->bus_id, words);
}
static int omap_mcbsp_hwrule_min_buffersize(struct snd_pcm_hw_params *params,
struct snd_pcm_hw_rule *rule)
{
struct snd_interval *buffer_size = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_BUFFER_SIZE);
struct snd_interval *channels = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
struct omap_mcbsp_data *mcbsp_data = rule->private;
struct snd_interval frames;
int size;
snd_interval_any(&frames);
size = omap_mcbsp_get_fifo_size(mcbsp_data->bus_id);
frames.min = size / channels->min;
frames.integer = 1;
return snd_interval_refine(buffer_size, &frames);
}
static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
@ -182,33 +214,35 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
if (!cpu_dai->active)
err = omap_mcbsp_request(bus_id);
/*
* OMAP3 McBSP FIFO is word structured.
* McBSP2 has 1024 + 256 = 1280 word long buffer,
* McBSP1,3,4,5 has 128 word long buffer
* This means that the size of the FIFO depends on the sample format.
* For example on McBSP3:
* 16bit samples: size is 128 * 2 = 256 bytes
* 32bit samples: size is 128 * 4 = 512 bytes
* It is simpler to place constraint for buffer and period based on
* channels.
* McBSP3 as example again (16 or 32 bit samples):
* 1 channel (mono): size is 128 frames (128 words)
* 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words)
* 4 channels: size is 128 / 4 = 32 frames (4 * 32 words)
*/
if (cpu_is_omap343x()) {
int dma_op_mode = omap_mcbsp_get_dma_op_mode(bus_id);
int max_period;
/*
* McBSP2 in OMAP3 has 1024 * 32-bit internal audio buffer.
* Set constraint for minimum buffer size to the same than FIFO
* size in order to avoid underruns in playback startup because
* HW is keeping the DMA request active until FIFO is filled.
*/
if (bus_id == 1)
snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
4096, UINT_MAX);
* Rule for the buffer size. We should not allow
* smaller buffer than the FIFO size to avoid underruns
*/
snd_pcm_hw_rule_add(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_CHANNELS,
omap_mcbsp_hwrule_min_buffersize,
mcbsp_data,
SNDRV_PCM_HW_PARAM_BUFFER_SIZE, -1);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
max_period = omap_mcbsp_get_max_tx_threshold(bus_id);
else
max_period = omap_mcbsp_get_max_rx_threshold(bus_id);
max_period++;
max_period <<= 1;
if (dma_op_mode == MCBSP_DMA_MODE_THRESHOLD)
snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
32, max_period);
/* Make sure, that the period size is always even */
snd_pcm_hw_constraint_step(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 2);
}
return err;
@ -289,11 +323,14 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
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;
struct omap_pcm_dma_data *dma_data;
int dma, bus_id = mcbsp_data->bus_id;
int wlen, channels, wpf, sync_mode = OMAP_DMA_SYNC_ELEMENT;
int pkt_size = 0;
unsigned long port;
unsigned int format, div, framesize, master;
dma_data = &omap_mcbsp_dai_dma_params[cpu_dai->id][substream->stream];
if (cpu_class_is_omap1()) {
dma = omap1_dma_reqs[bus_id][substream->stream];
port = omap1_mcbsp_port[bus_id][substream->stream];
@ -306,35 +343,74 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
} else if (cpu_is_omap343x()) {
dma = omap24xx_dma_reqs[bus_id][substream->stream];
port = omap34xx_mcbsp_port[bus_id][substream->stream];
omap_mcbsp_dai_dma_params[id][substream->stream].set_threshold =
omap_mcbsp_set_threshold;
/* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
if (omap_mcbsp_get_dma_op_mode(bus_id) ==
MCBSP_DMA_MODE_THRESHOLD)
sync_mode = OMAP_DMA_SYNC_FRAME;
} else {
return -ENODEV;
}
omap_mcbsp_dai_dma_params[id][substream->stream].name =
substream->stream ? "Audio Capture" : "Audio Playback";
omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma;
omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port;
omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode;
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
omap_mcbsp_dai_dma_params[id][substream->stream].data_type =
OMAP_DMA_DATA_TYPE_S16;
dma_data->data_type = OMAP_DMA_DATA_TYPE_S16;
wlen = 16;
break;
case SNDRV_PCM_FORMAT_S32_LE:
omap_mcbsp_dai_dma_params[id][substream->stream].data_type =
OMAP_DMA_DATA_TYPE_S32;
dma_data->data_type = OMAP_DMA_DATA_TYPE_S32;
wlen = 32;
break;
default:
return -EINVAL;
}
if (cpu_is_omap343x()) {
dma_data->set_threshold = omap_mcbsp_set_threshold;
/* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
if (omap_mcbsp_get_dma_op_mode(bus_id) ==
MCBSP_DMA_MODE_THRESHOLD) {
int period_words, max_thrsh;
snd_soc_dai_set_dma_data(cpu_dai, substream,
&omap_mcbsp_dai_dma_params[id][substream->stream]);
period_words = params_period_bytes(params) / (wlen / 8);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
max_thrsh = omap_mcbsp_get_max_tx_threshold(
mcbsp_data->bus_id);
else
max_thrsh = omap_mcbsp_get_max_rx_threshold(
mcbsp_data->bus_id);
/*
* If the period contains less or equal number of words,
* we are using the original threshold mode setup:
* McBSP threshold = sDMA frame size = period_size
* Otherwise we switch to sDMA packet mode:
* McBSP threshold = sDMA packet size
* sDMA frame size = period size
*/
if (period_words > max_thrsh) {
int divider = 0;
/*
* Look for the biggest threshold value, which
* divides the period size evenly.
*/
divider = period_words / max_thrsh;
if (period_words % max_thrsh)
divider++;
while (period_words % divider &&
divider < period_words)
divider++;
if (divider == period_words)
return -EINVAL;
pkt_size = period_words / divider;
sync_mode = OMAP_DMA_SYNC_PACKET;
} else {
sync_mode = OMAP_DMA_SYNC_FRAME;
}
}
}
dma_data->name = substream->stream ? "Audio Capture" : "Audio Playback";
dma_data->dma_req = dma;
dma_data->port_addr = port;
dma_data->sync_mode = sync_mode;
dma_data->packet_size = pkt_size;
snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
if (mcbsp_data->configured) {
/* McBSP already configured by another stream */
@ -360,7 +436,6 @@ 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);
@ -368,7 +443,6 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
break;
case SNDRV_PCM_FORMAT_S32_LE:
/* Set word lengths */
wlen = 32;
regs->rcr2 |= RWDLEN2(OMAP_MCBSP_WORD_32);
regs->rcr1 |= RWDLEN1(OMAP_MCBSP_WORD_32);
regs->xcr2 |= XWDLEN2(OMAP_MCBSP_WORD_32);
@ -409,6 +483,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
}
omap_mcbsp_config(bus_id, &mcbsp_data->regs);
mcbsp_data->wlen = wlen;
mcbsp_data->configured = 1;
return 0;

36
sound/soc/omap/omap3pandora.c
View file

@ -43,12 +43,14 @@
static struct regulator *omap3pandora_dac_reg;
static int omap3pandora_cmn_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params, unsigned int fmt)
static int omap3pandora_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;
int fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS;
int ret;
/* Set codec DAI configuration */
@ -91,24 +93,6 @@ static int omap3pandora_cmn_hw_params(struct snd_pcm_substream *substream,
return 0;
}
static int omap3pandora_out_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
return omap3pandora_cmn_hw_params(substream, params,
SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_IB_NF |
SND_SOC_DAIFMT_CBS_CFS);
}
static int omap3pandora_in_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
return omap3pandora_cmn_hw_params(substream, params,
SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS);
}
static int omap3pandora_dac_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
@ -231,12 +215,8 @@ static int omap3pandora_in_init(struct snd_soc_codec *codec)
return snd_soc_dapm_sync(codec);
}
static struct snd_soc_ops omap3pandora_out_ops = {
.hw_params = omap3pandora_out_hw_params,
};
static struct snd_soc_ops omap3pandora_in_ops = {
.hw_params = omap3pandora_in_hw_params,
static struct snd_soc_ops omap3pandora_ops = {
.hw_params = omap3pandora_hw_params,
};
/* Digital audio interface glue - connects codec <--> CPU */
@ -246,14 +226,14 @@ static struct snd_soc_dai_link omap3pandora_dai[] = {
.stream_name = "HiFi Out",
.cpu_dai = &omap_mcbsp_dai[0],
.codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
.ops = &omap3pandora_out_ops,
.ops = &omap3pandora_ops,
.init = omap3pandora_out_init,
}, {
.name = "TWL4030",
.stream_name = "Line/Mic In",
.cpu_dai = &omap_mcbsp_dai[1],
.codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
.ops = &omap3pandora_in_ops,
.ops = &omap3pandora_ops,
.init = omap3pandora_in_init,
}
};

73
sound/soc/omap/rx51.c
View file

@ -27,6 +27,7 @@
#include <linux/gpio.h>
#include <linux/platform_device.h>
#include <sound/core.h>
#include <sound/jack.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
@ -37,14 +38,22 @@
#include "omap-pcm.h"
#include "../codecs/tlv320aic3x.h"
#define RX51_TVOUT_SEL_GPIO 40
#define RX51_JACK_DETECT_GPIO 177
/*
* REVISIT: TWL4030 GPIO base in RX-51. Now statically defined to 192. This
* gpio is reserved in arch/arm/mach-omap2/board-rx51-peripherals.c
*/
#define RX51_SPEAKER_AMP_TWL_GPIO (192 + 7)
enum {
RX51_JACK_DISABLED,
RX51_JACK_TVOUT, /* tv-out */
};
static int rx51_spk_func;
static int rx51_dmic_func;
static int rx51_jack_func;
static void rx51_ext_control(struct snd_soc_codec *codec)
{
@ -57,6 +66,9 @@ static void rx51_ext_control(struct snd_soc_codec *codec)
else
snd_soc_dapm_disable_pin(codec, "DMic");
gpio_set_value(RX51_TVOUT_SEL_GPIO,
rx51_jack_func == RX51_JACK_TVOUT);
snd_soc_dapm_sync(codec);
}
@ -162,6 +174,40 @@ static int rx51_set_input(struct snd_kcontrol *kcontrol,
return 1;
}
static int rx51_get_jack(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.integer.value[0] = rx51_jack_func;
return 0;
}
static int rx51_set_jack(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
if (rx51_jack_func == ucontrol->value.integer.value[0])
return 0;
rx51_jack_func = ucontrol->value.integer.value[0];
rx51_ext_control(codec);
return 1;
}
static struct snd_soc_jack rx51_av_jack;
static struct snd_soc_jack_gpio rx51_av_jack_gpios[] = {
{
.gpio = RX51_JACK_DETECT_GPIO,
.name = "avdet-gpio",
.report = SND_JACK_VIDEOOUT,
.invert = 1,
.debounce_time = 200,
},
};
static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = {
SND_SOC_DAPM_SPK("Ext Spk", rx51_spk_event),
SND_SOC_DAPM_MIC("DMic", NULL),
@ -177,10 +223,12 @@ static const struct snd_soc_dapm_route audio_map[] = {
static const char *spk_function[] = {"Off", "On"};
static const char *input_function[] = {"ADC", "Digital Mic"};
static const char *jack_function[] = {"Off", "TV-OUT"};
static const struct soc_enum rx51_enum[] = {
SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function),
SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(input_function), input_function),
SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(jack_function), jack_function),
};
static const struct snd_kcontrol_new aic34_rx51_controls[] = {
@ -188,10 +236,13 @@ static const struct snd_kcontrol_new aic34_rx51_controls[] = {
rx51_get_spk, rx51_set_spk),
SOC_ENUM_EXT("Input Select", rx51_enum[1],
rx51_get_input, rx51_set_input),
SOC_ENUM_EXT("Jack Function", rx51_enum[2],
rx51_get_jack, rx51_set_jack),
};
static int rx51_aic34_init(struct snd_soc_codec *codec)
{
struct snd_soc_card *card = codec->socdev->card;
int err;
/* Set up NC codec pins */
@ -214,7 +265,16 @@ static int rx51_aic34_init(struct snd_soc_codec *codec)
snd_soc_dapm_sync(codec);
return 0;
/* AV jack detection */
err = snd_soc_jack_new(card, "AV Jack",
SND_JACK_VIDEOOUT, &rx51_av_jack);
if (err)
return err;
err = snd_soc_jack_add_gpios(&rx51_av_jack,
ARRAY_SIZE(rx51_av_jack_gpios),
rx51_av_jack_gpios);
return err;
}
/* Digital audio interface glue - connects codec <--> CPU */
@ -259,6 +319,11 @@ static int __init rx51_soc_init(void)
if (!machine_is_nokia_rx51())
return -ENODEV;
err = gpio_request(RX51_TVOUT_SEL_GPIO, "tvout_sel");
if (err)
goto err_gpio_tvout_sel;
gpio_direction_output(RX51_TVOUT_SEL_GPIO, 0);
rx51_snd_device = platform_device_alloc("soc-audio", -1);
if (!rx51_snd_device) {
err = -ENOMEM;
@ -277,13 +342,19 @@ static int __init rx51_soc_init(void)
err2:
platform_device_put(rx51_snd_device);
err1:
gpio_free(RX51_TVOUT_SEL_GPIO);
err_gpio_tvout_sel:
return err;
}
static void __exit rx51_soc_exit(void)
{
snd_soc_jack_free_gpios(&rx51_av_jack, ARRAY_SIZE(rx51_av_jack_gpios),
rx51_av_jack_gpios);
platform_device_unregister(rx51_snd_device);
gpio_free(RX51_TVOUT_SEL_GPIO);
}
module_init(rx51_soc_init);

10
sound/soc/s3c24xx/Kconfig
View file

@ -1,6 +1,6 @@
config SND_S3C24XX_SOC
tristate "SoC Audio for the Samsung S3CXXXX chips"
depends on ARCH_S3C2410 || ARCH_S3C64XX
depends on ARCH_S3C2410 || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210
select S3C64XX_DMA if ARCH_S3C64XX
help
Say Y or M if you want to add support for codecs attached to
@ -120,8 +120,14 @@ config SND_S3C24XX_SOC_SIMTEC_HERMES
config SND_SOC_SMDK_WM9713
tristate "SoC AC97 Audio support for SMDK with WM9713"
depends on SND_S3C24XX_SOC && MACH_SMDK6410
depends on SND_S3C24XX_SOC && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110)
select SND_SOC_WM9713
select SND_S3C_SOC_AC97
help
Sat Y if you want to add support for SoC audio on the SMDK.
config SND_S3C64XX_SOC_SMARTQ
tristate "SoC I2S Audio support for SmartQ board"
depends on SND_S3C24XX_SOC && MACH_SMARTQ
select SND_S3C64XX_SOC_I2S
select SND_SOC_WM8750

2
sound/soc/s3c24xx/Makefile
View file

@ -29,6 +29,7 @@ snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o
snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o
snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o
snd-soc-smdk-wm9713-objs := smdk_wm9713.o
snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o
obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o
obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
@ -41,3 +42,4 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o
obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o
obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o
obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o
obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o

1
sound/soc/s3c24xx/s3c-ac97.c
View file

@ -31,7 +31,6 @@
#define AC_CMD_DATA(x) (x & 0xffff)
struct s3c_ac97_info {
unsigned state;
struct clk *ac97_clk;
void __iomem *regs;
struct mutex lock;

3
sound/soc/s3c24xx/s3c-i2s-v2.c
View file

@ -32,7 +32,8 @@
#undef S3C_IIS_V2_SUPPORTED
#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413)
#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) \
|| defined(CONFIG_CPU_S5PV210)
#define S3C_IIS_V2_SUPPORTED
#endif

295
sound/soc/s3c24xx/smartq_wm8987.c Normal file
View file

@ -0,0 +1,295 @@
/* sound/soc/s3c24xx/smartq_wm8987.c
*
* Copyright 2010 Maurus Cuelenaere <mcuelenaere@gmail.com>
*
* Based on smdk6410_wm8987.c
* Copyright 2007 Wolfson Microelectronics PLC. - linux@wolfsonmicro.com
* Graeme Gregory - graeme.gregory@wolfsonmicro.com
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the
* Free Software Foundation; either version 2 of the License, or (at your
* option) any later version.
*
*/
#include <linux/module.h>
#include <linux/platform_device.h>
#include <linux/gpio.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc-dapm.h>
#include <sound/jack.h>
#include <asm/mach-types.h>
#include "s3c-dma.h"
#include "s3c64xx-i2s.h"
#include "../codecs/wm8750.h"
/*
* WM8987 is register compatible with WM8750, so using that as base driver.
*/
static struct snd_soc_card snd_soc_smartq;
static int smartq_hifi_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;
struct s3c_i2sv2_rate_calc div;
unsigned int clk = 0;
int ret;
s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params),
s3c_i2sv2_get_clock(cpu_dai));
switch (params_rate(params)) {
case 8000:
case 16000:
case 32000:
case 48000:
case 96000:
clk = 12288000;
break;
case 11025:
case 22050:
case 44100:
case 88200:
clk = 11289600;
break;
}
/* set codec DAI configuration */
ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0)
return ret;
/* set cpu DAI configuration */
ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0)
return ret;
/* set the codec system clock for DAC and ADC */
ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk,
SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
/* set MCLK division for sample rate */
ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_RCLK, div.fs_div);
if (ret < 0)
return ret;
/* set prescaler division for sample rate */
ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_PRESCALER,
div.clk_div - 1);
if (ret < 0)
return ret;
return 0;
}
/*
* SmartQ WM8987 HiFi DAI operations.
*/
static struct snd_soc_ops smartq_hifi_ops = {
.hw_params = smartq_hifi_hw_params,
};
static struct snd_soc_jack smartq_jack;
static struct snd_soc_jack_pin smartq_jack_pins[] = {
/* Disable speaker when headphone is plugged in */
{
.pin = "Internal Speaker",
.mask = SND_JACK_HEADPHONE,
},
};
static struct snd_soc_jack_gpio smartq_jack_gpios[] = {
{
.gpio = S3C64XX_GPL(12),
.name = "headphone detect",
.report = SND_JACK_HEADPHONE,
.debounce_time = 200,
},
};
static const struct snd_kcontrol_new wm8987_smartq_controls[] = {
SOC_DAPM_PIN_SWITCH("Internal Speaker"),
SOC_DAPM_PIN_SWITCH("Headphone Jack"),
SOC_DAPM_PIN_SWITCH("Internal Mic"),
};
static int smartq_speaker_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k,
int event)
{
gpio_set_value(S3C64XX_GPK(12), SND_SOC_DAPM_EVENT_OFF(event));
return 0;
}
static const struct snd_soc_dapm_widget wm8987_dapm_widgets[] = {
SND_SOC_DAPM_SPK("Internal Speaker", smartq_speaker_event),
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_MIC("Internal Mic", NULL),
};
static const struct snd_soc_dapm_route audio_map[] = {
{"Headphone Jack", NULL, "LOUT2"},
{"Headphone Jack", NULL, "ROUT2"},
{"Internal Speaker", NULL, "LOUT2"},
{"Internal Speaker", NULL, "ROUT2"},
{"Mic Bias", NULL, "Internal Mic"},
{"LINPUT2", NULL, "Mic Bias"},
};
static int smartq_wm8987_init(struct snd_soc_codec *codec)
{
int err = 0;
/* Add SmartQ specific widgets */
snd_soc_dapm_new_controls(codec, wm8987_dapm_widgets,
ARRAY_SIZE(wm8987_dapm_widgets));
/* add SmartQ specific controls */
err = snd_soc_add_controls(codec, wm8987_smartq_controls,
ARRAY_SIZE(wm8987_smartq_controls));
if (err < 0)
return err;
/* setup SmartQ specific audio path */
snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
/* set endpoints to not connected */
snd_soc_dapm_nc_pin(codec, "LINPUT1");
snd_soc_dapm_nc_pin(codec, "RINPUT1");
snd_soc_dapm_nc_pin(codec, "OUT3");
snd_soc_dapm_nc_pin(codec, "ROUT1");
/* set endpoints to default off mode */
snd_soc_dapm_enable_pin(codec, "Internal Speaker");
snd_soc_dapm_enable_pin(codec, "Internal Mic");
snd_soc_dapm_disable_pin(codec, "Headphone Jack");
err = snd_soc_dapm_sync(codec);
if (err)
return err;
/* Headphone jack detection */
err = snd_soc_jack_new(&snd_soc_smartq, "Headphone Jack",
SND_JACK_HEADPHONE, &smartq_jack);
if (err)
return err;
err = snd_soc_jack_add_pins(&smartq_jack, ARRAY_SIZE(smartq_jack_pins),
smartq_jack_pins);
if (err)
return err;
err = snd_soc_jack_add_gpios(&smartq_jack,
ARRAY_SIZE(smartq_jack_gpios),
smartq_jack_gpios);
return err;
}
static struct snd_soc_dai_link smartq_dai[] = {
{
.name = "wm8987",
.stream_name = "SmartQ Hi-Fi",
.cpu_dai = &s3c64xx_i2s_dai[0],
.codec_dai = &wm8750_dai,
.init = smartq_wm8987_init,
.ops = &smartq_hifi_ops,
},
};
static struct snd_soc_card snd_soc_smartq = {
.name = "SmartQ",
.platform = &s3c24xx_soc_platform,
.dai_link = smartq_dai,
.num_links = ARRAY_SIZE(smartq_dai),
};
static struct snd_soc_device smartq_snd_devdata = {
.card = &snd_soc_smartq,
.codec_dev = &soc_codec_dev_wm8750,
};
static struct platform_device *smartq_snd_device;
static int __init smartq_init(void)
{
int ret;
if (!machine_is_smartq7() && !machine_is_smartq5()) {
pr_info("Only SmartQ is supported by this ASoC driver\n");
return -ENODEV;
}
smartq_snd_device = platform_device_alloc("soc-audio", -1);
if (!smartq_snd_device)
return -ENOMEM;
platform_set_drvdata(smartq_snd_device, &smartq_snd_devdata);
smartq_snd_devdata.dev = &smartq_snd_device->dev;
ret = platform_device_add(smartq_snd_device);
if (ret) {
platform_device_put(smartq_snd_device);
return ret;
}
/* Initialise GPIOs used by amplifiers */
ret = gpio_request(S3C64XX_GPK(12), "amplifiers shutdown");
if (ret) {
dev_err(&smartq_snd_device->dev, "Failed to register GPK12\n");
goto err_unregister_device;
}
/* Disable amplifiers */
ret = gpio_direction_output(S3C64XX_GPK(12), 1);
if (ret) {
dev_err(&smartq_snd_device->dev, "Failed to configure GPK12\n");
goto err_free_gpio_amp_shut;
}
return 0;
err_free_gpio_amp_shut:
gpio_free(S3C64XX_GPK(12));
err_unregister_device:
platform_device_unregister(smartq_snd_device);
return ret;
}
static void __exit smartq_exit(void)
{
snd_soc_jack_free_gpios(&smartq_jack, ARRAY_SIZE(smartq_jack_gpios),
smartq_jack_gpios);
platform_device_unregister(smartq_snd_device);
}
module_init(smartq_init);
module_exit(smartq_exit);
/* Module information */
MODULE_AUTHOR("Maurus Cuelenaere <[email protected]>");
MODULE_DESCRIPTION("ALSA SoC SmartQ WM8987");
MODULE_LICENSE("GPL");

3
sound/soc/s3c24xx/smdk_wm9713.c
View file

@ -25,6 +25,9 @@ static struct snd_soc_card smdk;
* Default CFG switch settings to use this driver:
*
* SMDK6410: Set CFG1 1-3 On, CFG2 1-4 Off
* SMDKC100: Set CFG6 1-3 On, CFG7 1 On
* SMDKC110: Set CFGB10 1-2 Off, CFGB12 1-3 On
* SMDKV210: Set CFGB10 1-2 Off, CFGB12 1-3 On
*/
/*

38
sound/soc/s6000/s6000-i2s.c
View file

@ -451,16 +451,15 @@ static int __devinit s6000_i2s_probe(struct platform_device *pdev)
goto err_release_none;
}
region = request_mem_region(scbmem->start,
scbmem->end - scbmem->start + 1,
pdev->name);
region = request_mem_region(scbmem->start, resource_size(scbmem),
pdev->name);
if (!region) {
dev_err(&pdev->dev, "I2S SCB region already claimed\n");
ret = -EBUSY;
goto err_release_none;
}
mmio = ioremap(scbmem->start, scbmem->end - scbmem->start + 1);
mmio = ioremap(scbmem->start, resource_size(scbmem));
if (!mmio) {
dev_err(&pdev->dev, "can't ioremap SCB region\n");
ret = -ENOMEM;
@ -474,9 +473,8 @@ static int __devinit s6000_i2s_probe(struct platform_device *pdev)
goto err_release_map;
}
region = request_mem_region(sifmem->start,
sifmem->end - sifmem->start + 1,
pdev->name);
region = request_mem_region(sifmem->start, resource_size(sifmem),
pdev->name);
if (!region) {
dev_err(&pdev->dev, "I2S SIF region already claimed\n");
ret = -EBUSY;
@ -490,8 +488,8 @@ static int __devinit s6000_i2s_probe(struct platform_device *pdev)
goto err_release_sif;
}
region = request_mem_region(dma1->start, dma1->end - dma1->start + 1,
pdev->name);
region = request_mem_region(dma1->start, resource_size(dma1),
pdev->name);
if (!region) {
dev_err(&pdev->dev, "I2S DMA region already claimed\n");
ret = -EBUSY;
@ -500,9 +498,8 @@ static int __devinit s6000_i2s_probe(struct platform_device *pdev)
dma2 = platform_get_resource(pdev, IORESOURCE_DMA, 1);
if (dma2) {
region = request_mem_region(dma2->start,
dma2->end - dma2->start + 1,
pdev->name);
region = request_mem_region(dma2->start, resource_size(dma2),
pdev->name);
if (!region) {
dev_err(&pdev->dev,
"I2S DMA region already claimed\n");
@ -561,15 +558,15 @@ err_release_dev:
kfree(dev);
err_release_dma2:
if (dma2)
release_mem_region(dma2->start, dma2->end - dma2->start + 1);
release_mem_region(dma2->start, resource_size(dma2));
err_release_dma1:
release_mem_region(dma1->start, dma1->end - dma1->start + 1);
release_mem_region(dma1->start, resource_size(dma1));
err_release_sif:
release_mem_region(sifmem->start, (sifmem->end - sifmem->start) + 1);
release_mem_region(sifmem->start, resource_size(sifmem));
err_release_map:
iounmap(mmio);
err_release_scb:
release_mem_region(scbmem->start, (scbmem->end - scbmem->start) + 1);
release_mem_region(scbmem->start, resource_size(scbmem));
err_release_none:
return ret;
}
@ -590,19 +587,18 @@ static void __devexit s6000_i2s_remove(struct platform_device *pdev)
kfree(dev);
region = platform_get_resource(pdev, IORESOURCE_DMA, 0);
release_mem_region(region->start, region->end - region->start + 1);
release_mem_region(region->start, resource_size(region));
region = platform_get_resource(pdev, IORESOURCE_DMA, 1);
if (region)
release_mem_region(region->start,
region->end - region->start + 1);
release_mem_region(region->start, resource_size(region));
region = platform_get_resource(pdev, IORESOURCE_MEM, 0);
release_mem_region(region->start, (region->end - region->start) + 1);
release_mem_region(region->start, resource_size(region));
iounmap(mmio);
region = platform_get_resource(pdev, IORESOURCE_IO, 0);
release_mem_region(region->start, (region->end - region->start) + 1);
release_mem_region(region->start, resource_size(region));
}
static struct platform_driver s6000_i2s_driver = {

4
sound/soc/sh/Kconfig
View file

@ -48,7 +48,7 @@ config SND_SH7760_AC97
config SND_FSI_AK4642
bool "FSI-AK4642 sound support"
depends on SND_SOC_SH4_FSI
depends on SND_SOC_SH4_FSI && I2C_SH_MOBILE
select SND_SOC_AK4642
help
This option enables generic sound support for the
@ -56,7 +56,7 @@ config SND_FSI_AK4642
config SND_FSI_DA7210
bool "FSI-DA7210 sound support"
depends on SND_SOC_SH4_FSI
depends on SND_SOC_SH4_FSI && I2C_SH_MOBILE
select SND_SOC_DA7210
help
This option enables generic sound support for the

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