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Merge remote branch 'broonie-asoc/for-2.6.37' into for-2.6.37

This commit is contained in:
Mark Brown 2010-08-27 11:22:57 +01:00
parent 9745e824a8 014a27553a
commit fbd60ce791
35 changed files with 3557 additions and 108 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

57
sound/soc/atmel/atmel_ssc_dai.c
View File

@ -789,13 +789,14 @@ static struct snd_soc_dai_driver atmel_ssc_dai[NUM_SSC_DEVICES] = {
static __devinit int asoc_ssc_probe(struct platform_device *pdev)
{
return snd_soc_register_dais(&pdev->dev, atmel_ssc_dai,
ARRAY_SIZE(atmel_ssc_dai));
BUG_ON(pdev->id < 0);
BUG_ON(pdev->id >= ARRAY_SIZE(atmel_ssc_dai));
return snd_soc_register_dai(&pdev->dev, &atmel_ssc_dai[pdev->id]);
}
static int __devexit asoc_ssc_remove(struct platform_device *pdev)
{
snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(atmel_ssc_dai));
snd_soc_unregister_dai(&pdev->dev);
return 0;
}
@ -809,6 +810,56 @@ static struct platform_driver asoc_ssc_driver = {
.remove = __devexit_p(asoc_ssc_remove),
};
/**
* atmel_ssc_set_audio - Allocate the specified SSC for audio use.
*/
int atmel_ssc_set_audio(int ssc_id)
{
struct ssc_device *ssc;
static struct platform_device *dma_pdev;
struct platform_device *ssc_pdev;
int ret;
if (ssc_id < 0 || ssc_id >= ARRAY_SIZE(atmel_ssc_dai))
return -EINVAL;
/* Allocate a dummy device for DMA if we don't have one already */
if (!dma_pdev) {
dma_pdev = platform_device_alloc("atmel-pcm-audio", -1);
if (!dma_pdev)
return -ENOMEM;
ret = platform_device_add(dma_pdev);
if (ret < 0) {
platform_device_put(dma_pdev);
dma_pdev = NULL;
return ret;
}
}
ssc_pdev = platform_device_alloc("atmel-ssc-dai", ssc_id);
if (!ssc_pdev) {
ssc_free(ssc);
return -ENOMEM;
}
/* If we can grab the SSC briefly to parent the DAI device off it */
ssc = ssc_request(ssc_id);
if (IS_ERR(ssc))
pr_warn("Unable to parent ASoC SSC DAI on SSC: %ld\n",
PTR_ERR(ssc));
else
ssc_pdev->dev.parent = &(ssc->pdev->dev);
ssc_free(ssc);
ret = platform_device_add(ssc_pdev);
if (ret < 0)
platform_device_put(ssc_pdev);
return ret;
}
EXPORT_SYMBOL_GPL(atmel_ssc_set_audio);
static int __init snd_atmel_ssc_init(void)
{
return platform_driver_register(&asoc_ssc_driver);

2
sound/soc/atmel/atmel_ssc_dai.h
View File

@ -117,4 +117,6 @@ struct atmel_ssc_info {
struct atmel_ssc_state ssc_state;
};
int atmel_ssc_set_audio(int ssc);
#endif /* _AT91_SSC_DAI_H */

10
sound/soc/atmel/sam9g20_wm8731.c
View File

@ -183,8 +183,8 @@ static struct snd_soc_dai_link at91sam9g20ek_dai = {
.cpu_dai_name = "atmel-ssc-dai.0",
.codec_dai_name = "wm8731-hifi",
.init = at91sam9g20ek_wm8731_init,
.platform_name = "atmel_pcm-audio",
.codec_name = "wm8731-codec.0-001a",
.platform_name = "atmel-pcm-audio",
.codec_name = "wm8731-codec.0-001b",
.ops = &at91sam9g20ek_ops,
};
@ -205,6 +205,12 @@ static int __init at91sam9g20ek_init(void)
if (!(machine_is_at91sam9g20ek() || machine_is_at91sam9g20ek_2mmc()))
return -ENODEV;
ret = atmel_ssc_set_audio(0);
if (ret != 0) {
pr_err("Failed to set SSC 0 for audio: %d\n", ret);
return ret;
}
/*
* Codec MCLK is supplied by PCK0 - set it up.
*/

1486
sound/soc/codecs/88pm860x-codec.c Normal file
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File diff suppressed because it is too large Load Diff

97
sound/soc/codecs/88pm860x-codec.h Normal file
View File

@ -0,0 +1,97 @@
/*
* 88pm860x-codec.h -- 88PM860x ALSA SoC Audio Driver
*
* Copyright 2010 Marvell International Ltd.
* Haojian Zhuang <haojian.zhuang@marvell.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 __88PM860X_H
#define __88PM860X_H
/* The offset of these registers are 0xb0 */
#define PM860X_PCM_IFACE_1 0x00
#define PM860X_PCM_IFACE_2 0x01
#define PM860X_PCM_IFACE_3 0x02
#define PM860X_PCM_RATE 0x03
#define PM860X_EC_PATH 0x04
#define PM860X_SIDETONE_L_GAIN 0x05
#define PM860X_SIDETONE_R_GAIN 0x06
#define PM860X_SIDETONE_SHIFT 0x07
#define PM860X_ADC_OFFSET_1 0x08
#define PM860X_ADC_OFFSET_2 0x09
#define PM860X_DMIC_DELAY 0x0a
#define PM860X_I2S_IFACE_1 0x0b
#define PM860X_I2S_IFACE_2 0x0c
#define PM860X_I2S_IFACE_3 0x0d
#define PM860X_I2S_IFACE_4 0x0e
#define PM860X_EQUALIZER_N0_1 0x0f
#define PM860X_EQUALIZER_N0_2 0x10
#define PM860X_EQUALIZER_N1_1 0x11
#define PM860X_EQUALIZER_N1_2 0x12
#define PM860X_EQUALIZER_D1_1 0x13
#define PM860X_EQUALIZER_D1_2 0x14
#define PM860X_LOFI_GAIN_LEFT 0x15
#define PM860X_LOFI_GAIN_RIGHT 0x16
#define PM860X_HIFIL_GAIN_LEFT 0x17
#define PM860X_HIFIL_GAIN_RIGHT 0x18
#define PM860X_HIFIR_GAIN_LEFT 0x19
#define PM860X_HIFIR_GAIN_RIGHT 0x1a
#define PM860X_DAC_OFFSET 0x1b
#define PM860X_OFFSET_LEFT_1 0x1c
#define PM860X_OFFSET_LEFT_2 0x1d
#define PM860X_OFFSET_RIGHT_1 0x1e
#define PM860X_OFFSET_RIGHT_2 0x1f
#define PM860X_ADC_ANA_1 0x20
#define PM860X_ADC_ANA_2 0x21
#define PM860X_ADC_ANA_3 0x22
#define PM860X_ADC_ANA_4 0x23
#define PM860X_ANA_TO_ANA 0x24
#define PM860X_HS1_CTRL 0x25
#define PM860X_HS2_CTRL 0x26
#define PM860X_LO1_CTRL 0x27
#define PM860X_LO2_CTRL 0x28
#define PM860X_EAR_CTRL_1 0x29
#define PM860X_EAR_CTRL_2 0x2a
#define PM860X_AUDIO_SUPPLIES_1 0x2b
#define PM860X_AUDIO_SUPPLIES_2 0x2c
#define PM860X_ADC_EN_1 0x2d
#define PM860X_ADC_EN_2 0x2e
#define PM860X_DAC_EN_1 0x2f
#define PM860X_DAC_EN_2 0x31
#define PM860X_AUDIO_CAL_1 0x32
#define PM860X_AUDIO_CAL_2 0x33
#define PM860X_AUDIO_CAL_3 0x34
#define PM860X_AUDIO_CAL_4 0x35
#define PM860X_AUDIO_CAL_5 0x36
#define PM860X_ANA_INPUT_SEL_1 0x37
#define PM860X_ANA_INPUT_SEL_2 0x38
#define PM860X_PCM_IFACE_4 0x39
#define PM860X_I2S_IFACE_5 0x3a
#define PM860X_SHORTS 0x3b
#define PM860X_PLL_ADJ_1 0x3c
#define PM860X_PLL_ADJ_2 0x3d
/* bits definition */
#define PM860X_CLK_DIR_IN 0
#define PM860X_CLK_DIR_OUT 1
#define PM860X_DET_HEADSET (1 << 0)
#define PM860X_DET_MIC (1 << 1)
#define PM860X_DET_HOOK (1 << 2)
#define PM860X_SHORT_HEADSET (1 << 3)
#define PM860X_SHORT_LINEOUT (1 << 4)
#define PM860X_DET_MASK 0x1F
extern int pm860x_hs_jack_detect(struct snd_soc_codec *, struct snd_soc_jack *,
int, int, int, int);
extern int pm860x_mic_jack_detect(struct snd_soc_codec *, struct snd_soc_jack *,
int);
#endif /* __88PM860X_H */

8
sound/soc/codecs/Kconfig
View File

@ -10,6 +10,7 @@ config SND_SOC_I2C_AND_SPI
config SND_SOC_ALL_CODECS
tristate "Build all ASoC CODEC drivers"
select SND_SOC_88PM860X if MFD_88PM860X
select SND_SOC_L3
select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS
select SND_SOC_AD1836 if SPI_MASTER
@ -40,6 +41,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_TWL6040 if TWL4030_CORE
select SND_SOC_UDA134X
select SND_SOC_UDA1380 if I2C
select SND_SOC_WL1273 if WL1273_CORE
select SND_SOC_WM2000 if I2C
select SND_SOC_WM8350 if MFD_WM8350
select SND_SOC_WM8400 if MFD_WM8400
@ -85,6 +87,9 @@ config SND_SOC_ALL_CODECS
If unsure select "N".
config SND_SOC_88PM860X
tristate
config SND_SOC_WM_HUBS
tristate
default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y
@ -189,6 +194,9 @@ config SND_SOC_UDA134X
config SND_SOC_UDA1380
tristate
config SND_SOC_WL1273
tristate
config SND_SOC_WM8350
tristate

4
sound/soc/codecs/Makefile
View File

@ -1,3 +1,4 @@
snd-soc-88pm860x-objs := 88pm860x-codec.o
snd-soc-ac97-objs := ac97.o
snd-soc-ad1836-objs := ad1836.o
snd-soc-ad193x-objs := ad193x.o
@ -26,6 +27,7 @@ snd-soc-twl4030-objs := twl4030.o
snd-soc-twl6040-objs := twl6040.o
snd-soc-uda134x-objs := uda134x.o
snd-soc-uda1380-objs := uda1380.o
snd-soc-wl1273-objs := wl1273.o
snd-soc-wm8350-objs := wm8350.o
snd-soc-wm8400-objs := wm8400.o
snd-soc-wm8510-objs := wm8510.o
@ -67,6 +69,7 @@ snd-soc-tpa6130a2-objs := tpa6130a2.o
snd-soc-wm2000-objs := wm2000.o
snd-soc-wm9090-objs := wm9090.o
obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o
obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o
obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o
obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o
@ -96,6 +99,7 @@ obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o
obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o
obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o
obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o
obj-$(CONFIG_SND_SOC_WL1273) += snd-soc-wl1273.o
obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o
obj-$(CONFIG_SND_SOC_WM8400) += snd-soc-wm8400.o
obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o

2
sound/soc/codecs/cx20442.c
View File

@ -318,7 +318,7 @@ EXPORT_SYMBOL_GPL(v253_ops);
*/
static struct snd_soc_dai_driver cx20442_dai = {
.name = "cx20442-hifi",
.name = "cx20442-voice",
.playback = {
.stream_name = "Playback",
.channels_min = 1,

80
sound/soc/codecs/tlv320aic3x.c
View File

@ -12,11 +12,11 @@
*
* Notes:
* The AIC3X is a driver for a low power stereo audio
* codecs aic31, aic32, aic33.
* codecs aic31, aic32, aic33, aic3007.
*
* It supports full aic33 codec functionality.
* The compatibility with aic32, aic31 is as follows:
* aic32 | aic31
* The compatibility with aic32, aic31 and aic3007 is as follows:
* aic32/aic3007 | aic31
* ---------------------------------------
* MONO_LOUT -> N/A | MONO_LOUT -> N/A
* | IN1L -> LINE1L
@ -70,6 +70,10 @@ struct aic3x_priv {
unsigned int sysclk;
int master;
int gpio_reset;
#define AIC3X_MODEL_3X 0
#define AIC3X_MODEL_33 1
#define AIC3X_MODEL_3007 2
u16 model;
};
/*
@ -361,6 +365,14 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]),
};
/*
* Class-D amplifier gain. From 0 to 18 dB in 6 dB steps
*/
static DECLARE_TLV_DB_SCALE(classd_amp_tlv, 0, 600, 0);
static const struct snd_kcontrol_new aic3x_classd_amp_gain_ctrl =
SOC_DOUBLE_TLV("Class-D Amplifier Gain", CLASSD_CTRL, 6, 4, 3, 0, classd_amp_tlv);
/* Left DAC Mux */
static const struct snd_kcontrol_new aic3x_left_dac_mux_controls =
SOC_DAPM_ENUM("Route", aic3x_enum[LDAC_ENUM]);
@ -589,6 +601,15 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
SND_SOC_DAPM_INPUT("LINE2R"),
};
static const struct snd_soc_dapm_widget aic3007_dapm_widgets[] = {
/* Class-D outputs */
SND_SOC_DAPM_PGA("Left Class-D Out", CLASSD_CTRL, 3, 0, NULL, 0),
SND_SOC_DAPM_PGA("Right Class-D Out", CLASSD_CTRL, 2, 0, NULL, 0),
SND_SOC_DAPM_OUTPUT("SPOP"),
SND_SOC_DAPM_OUTPUT("SPOM"),
};
static const struct snd_soc_dapm_route intercon[] = {
/* Left Output */
{"Left DAC Mux", "DAC_L1", "Left DAC"},
@ -759,14 +780,30 @@ static const struct snd_soc_dapm_route intercon[] = {
{"GPIO1 dmic modclk", NULL, "DMic Rate 32"},
};
static const struct snd_soc_dapm_route intercon_3007[] = {
/* Class-D outputs */
{"Left Class-D Out", NULL, "Left Line Out"},
{"Right Class-D Out", NULL, "Left Line Out"},
{"SPOP", NULL, "Left Class-D Out"},
{"SPOM", NULL, "Right Class-D Out"},
};
static int aic3x_add_widgets(struct snd_soc_codec *codec)
{
struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets,
ARRAY_SIZE(aic3x_dapm_widgets));
/* set up audio path interconnects */
snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
if (aic3x->model == AIC3X_MODEL_3007) {
snd_soc_dapm_new_controls(codec, aic3007_dapm_widgets,
ARRAY_SIZE(aic3007_dapm_widgets));
snd_soc_dapm_add_routes(codec, intercon_3007, ARRAY_SIZE(intercon_3007));
}
return 0;
}
@ -1117,6 +1154,7 @@ static struct snd_soc_dai_driver aic3x_dai = {
.rates = AIC3X_RATES,
.formats = AIC3X_FORMATS,},
.ops = &aic3x_dai_ops,
.symmetric_rates = 1,
};
static int aic3x_suspend(struct snd_soc_codec *codec, pm_message_t state)
@ -1150,6 +1188,7 @@ static int aic3x_resume(struct snd_soc_codec *codec)
*/
static int aic3x_init(struct snd_soc_codec *codec)
{
struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
int reg;
aic3x_write(codec, AIC3X_PAGE_SELECT, PAGE0_SELECT);
@ -1218,6 +1257,17 @@ static int aic3x_init(struct snd_soc_codec *codec)
aic3x_write(codec, LINE2L_2_MONOLOPM_VOL, DEFAULT_VOL);
aic3x_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL);
if (aic3x->model == AIC3X_MODEL_3007) {
/* Class-D speaker driver init; datasheet p. 46 */
aic3x_write(codec, AIC3X_PAGE_SELECT, 0x0D);
aic3x_write(codec, 0xD, 0x0D);
aic3x_write(codec, 0x8, 0x5C);
aic3x_write(codec, 0x8, 0x5D);
aic3x_write(codec, 0x8, 0x5C);
aic3x_write(codec, AIC3X_PAGE_SELECT, 0x00);
aic3x_write(codec, CLASSD_CTRL, 0);
}
/* off, with power on */
aic3x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
@ -1243,6 +1293,8 @@ static int aic3x_probe(struct snd_soc_codec *codec)
snd_soc_add_controls(codec, aic3x_snd_controls,
ARRAY_SIZE(aic3x_snd_controls));
if (aic3x->model == AIC3X_MODEL_3007)
snd_soc_add_controls(codec, &aic3x_classd_amp_gain_ctrl, 1);
aic3x_add_widgets(codec);
@ -1274,6 +1326,14 @@ static struct snd_soc_codec_driver soc_codec_dev_aic3x = {
* 0x18, 0x19, 0x1A, 0x1B
*/
static const struct i2c_device_id aic3x_i2c_id[] = {
[AIC3X_MODEL_3X] = { "tlv320aic3x", 0 },
[AIC3X_MODEL_33] = { "tlv320aic33", 0 },
[AIC3X_MODEL_3007] = { "tlv320aic3007", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id);
/*
* If the i2c layer weren't so broken, we could pass this kind of data
* around
@ -1285,6 +1345,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c,
struct aic3x_setup_data *setup = pdata->setup;
struct aic3x_priv *aic3x;
int ret, i;
const struct i2c_device_id *tbl;
aic3x = kzalloc(sizeof(struct aic3x_priv), GFP_KERNEL);
if (aic3x == NULL) {
@ -1305,6 +1366,12 @@ static int aic3x_i2c_probe(struct i2c_client *i2c,
gpio_direction_output(aic3x->gpio_reset, 0);
}
for (tbl = aic3x_i2c_id; tbl->name[0]; tbl++) {
if (!strcmp(tbl->name, id->name))
break;
}
aic3x->model = tbl - aic3x_i2c_id;
for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++)
aic3x->supplies[i].supply = aic3x_supply_names[i];
@ -1359,13 +1426,6 @@ static int aic3x_i2c_remove(struct i2c_client *client)
return 0;
}
static const struct i2c_device_id aic3x_i2c_id[] = {
{ "tlv320aic3x", 0 },
{ "tlv320aic33", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id);
/* machine i2c codec control layer */
static struct i2c_driver aic3x_i2c_driver = {
.driver = {

2
sound/soc/codecs/tlv320aic3x.h
View File

@ -111,6 +111,8 @@
#define DACL1_2_MONOLOPM_VOL 75
#define DACR1_2_MONOLOPM_VOL 78
#define MONOLOPM_CTRL 79
/* Class-D speaker driver on tlv320aic3007 */
#define CLASSD_CTRL 73
/* Line Output Plus/Minus control registers */
#define LINE2L_2_LLOPM_VOL 80
#define LINE2L_2_RLOPM_VOL 87

525
sound/soc/codecs/wl1273.c Normal file
View File

@ -0,0 +1,525 @@
/*
* ALSA SoC WL1273 codec driver
*
* Author: Matti Aaltonen, <matti.j.aaltonen@nokia.com>
*
* Copyright: (C) 2010 Nokia Corporation
*
* 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.
*
* 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., 51 Franklin St, Fifth Floor, Boston, MA
* 02110-1301 USA
*
*/
#include <linux/mfd/wl1273-core.h>
#include <linux/slab.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc-dai.h>
#include <sound/soc-dapm.h>
#include <sound/initval.h>
#include "wl1273.h"
enum wl1273_mode { WL1273_MODE_BT, WL1273_MODE_FM_RX, WL1273_MODE_FM_TX };
/* codec private data */
struct wl1273_priv {
enum wl1273_mode mode;
struct wl1273_core *core;
unsigned int channels;
};
static int snd_wl1273_fm_set_i2s_mode(struct wl1273_core *core,
int rate, int width)
{
struct device *dev = &core->i2c_dev->dev;
int r = 0;
u16 mode;
dev_dbg(dev, "rate: %d\n", rate);
dev_dbg(dev, "width: %d\n", width);
mutex_lock(&core->lock);
mode = core->i2s_mode & ~WL1273_IS2_WIDTH & ~WL1273_IS2_RATE;
switch (rate) {
case 48000:
mode |= WL1273_IS2_RATE_48K;
break;
case 44100:
mode |= WL1273_IS2_RATE_44_1K;
break;
case 32000:
mode |= WL1273_IS2_RATE_32K;
break;
case 22050:
mode |= WL1273_IS2_RATE_22_05K;
break;
case 16000:
mode |= WL1273_IS2_RATE_16K;
break;
case 12000:
mode |= WL1273_IS2_RATE_12K;
break;
case 11025:
mode |= WL1273_IS2_RATE_11_025;
break;
case 8000:
mode |= WL1273_IS2_RATE_8K;
break;
default:
dev_err(dev, "Sampling rate: %d not supported\n", rate);
r = -EINVAL;
goto out;
}
switch (width) {
case 16:
mode |= WL1273_IS2_WIDTH_32;
break;
case 20:
mode |= WL1273_IS2_WIDTH_40;
break;
case 24:
mode |= WL1273_IS2_WIDTH_48;
break;
case 25:
mode |= WL1273_IS2_WIDTH_50;
break;
case 30:
mode |= WL1273_IS2_WIDTH_60;
break;
case 32:
mode |= WL1273_IS2_WIDTH_64;
break;
case 40:
mode |= WL1273_IS2_WIDTH_80;
break;
case 48:
mode |= WL1273_IS2_WIDTH_96;
break;
case 64:
mode |= WL1273_IS2_WIDTH_128;
break;
default:
dev_err(dev, "Data width: %d not supported\n", width);
r = -EINVAL;
goto out;
}
dev_dbg(dev, "WL1273_I2S_DEF_MODE: 0x%04x\n", WL1273_I2S_DEF_MODE);
dev_dbg(dev, "core->i2s_mode: 0x%04x\n", core->i2s_mode);
dev_dbg(dev, "mode: 0x%04x\n", mode);
if (core->i2s_mode != mode) {
r = wl1273_fm_write_cmd(core, WL1273_I2S_MODE_CONFIG_SET, mode);
if (r)
goto out;
core->i2s_mode = mode;
r = wl1273_fm_write_cmd(core, WL1273_AUDIO_ENABLE,
WL1273_AUDIO_ENABLE_I2S);
if (r)
goto out;
}
out:
mutex_unlock(&core->lock);
return r;
}
static int snd_wl1273_fm_set_channel_number(struct wl1273_core *core,
int channel_number)
{
struct i2c_client *client = core->i2c_dev;
struct device *dev = &client->dev;
int r = 0;
dev_dbg(dev, "%s\n", __func__);
mutex_lock(&core->lock);
if (core->channel_number == channel_number)
goto out;
if (channel_number == 1 && core->mode == WL1273_MODE_RX)
r = wl1273_fm_write_cmd(core, WL1273_MOST_MODE_SET,
WL1273_RX_MONO);
else if (channel_number == 1 && core->mode == WL1273_MODE_TX)
r = wl1273_fm_write_cmd(core, WL1273_MONO_SET,
WL1273_TX_MONO);
else if (channel_number == 2 && core->mode == WL1273_MODE_RX)
r = wl1273_fm_write_cmd(core, WL1273_MOST_MODE_SET,
WL1273_RX_STEREO);
else if (channel_number == 2 && core->mode == WL1273_MODE_TX)
r = wl1273_fm_write_cmd(core, WL1273_MONO_SET,
WL1273_TX_STEREO);
else
r = -EINVAL;
out:
mutex_unlock(&core->lock);
return r;
}
static int snd_wl1273_get_audio_route(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
ucontrol->value.integer.value[0] = wl1273->mode;
return 0;
}
static const char *wl1273_audio_route[] = { "Bt", "FmRx", "FmTx" };
static int snd_wl1273_set_audio_route(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
/* Do not allow changes while stream is running */
if (codec->active)
return -EPERM;
if (ucontrol->value.integer.value[0] < 0 ||
ucontrol->value.integer.value[0] >= ARRAY_SIZE(wl1273_audio_route))
return -EINVAL;
wl1273->mode = ucontrol->value.integer.value[0];
return 1;
}
static const struct soc_enum wl1273_enum =
SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wl1273_audio_route), wl1273_audio_route);
static int snd_wl1273_fm_audio_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
dev_dbg(codec->dev, "%s: enter.\n", __func__);
ucontrol->value.integer.value[0] = wl1273->core->audio_mode;
return 0;
}
static int snd_wl1273_fm_audio_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
int val, r = 0;
dev_dbg(codec->dev, "%s: enter.\n", __func__);
val = ucontrol->value.integer.value[0];
if (wl1273->core->audio_mode == val)
return 0;
r = wl1273_fm_set_audio(wl1273->core, val);
if (r < 0)
return r;
return 1;
}
static const char *wl1273_audio_strings[] = { "Digital", "Analog" };
static const struct soc_enum wl1273_audio_enum =
SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wl1273_audio_strings),
wl1273_audio_strings);
static int snd_wl1273_fm_volume_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
dev_dbg(codec->dev, "%s: enter.\n", __func__);
ucontrol->value.integer.value[0] = wl1273->core->volume;
return 0;
}
static int snd_wl1273_fm_volume_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
int r;
dev_dbg(codec->dev, "%s: enter.\n", __func__);
r = wl1273_fm_set_volume(wl1273->core,
ucontrol->value.integer.value[0]);
if (r)
return r;
return 1;
}
static const struct snd_kcontrol_new wl1273_controls[] = {
SOC_ENUM_EXT("Codec Mode", wl1273_enum,
snd_wl1273_get_audio_route, snd_wl1273_set_audio_route),
SOC_ENUM_EXT("Audio Switch", wl1273_audio_enum,
snd_wl1273_fm_audio_get, snd_wl1273_fm_audio_put),
SOC_SINGLE_EXT("Volume", 0, 0, WL1273_MAX_VOLUME, 0,
snd_wl1273_fm_volume_get, snd_wl1273_fm_volume_put),
};
static int wl1273_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
switch (wl1273->mode) {
case WL1273_MODE_BT:
snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_RATE,
8000, 8000);
snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_CHANNELS, 1, 1);
break;
case WL1273_MODE_FM_RX:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
pr_err("Cannot play in RX mode.\n");
return -EINVAL;
}
break;
case WL1273_MODE_FM_TX:
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
pr_err("Cannot capture in TX mode.\n");
return -EINVAL;
}
break;
default:
return -EINVAL;
break;
}
return 0;
}
static int wl1273_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 wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(rtd->codec);
struct wl1273_core *core = wl1273->core;
unsigned int rate, width, r;
if (params_format(params) != SNDRV_PCM_FORMAT_S16_LE) {
pr_err("Only SNDRV_PCM_FORMAT_S16_LE supported.\n");
return -EINVAL;
}
rate = params_rate(params);
width = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min;
if (wl1273->mode == WL1273_MODE_BT) {
if (rate != 8000) {
pr_err("Rate %d not supported.\n", params_rate(params));
return -EINVAL;
}
if (params_channels(params) != 1) {
pr_err("Only mono supported.\n");
return -EINVAL;
}
return 0;
}
if (wl1273->mode == WL1273_MODE_FM_TX &&
substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
pr_err("Only playback supported with TX.\n");
return -EINVAL;
}
if (wl1273->mode == WL1273_MODE_FM_RX &&
substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
pr_err("Only capture supported with RX.\n");
return -EINVAL;
}
if (wl1273->mode != WL1273_MODE_FM_RX &&
wl1273->mode != WL1273_MODE_FM_TX) {
pr_err("Unexpected mode: %d.\n", wl1273->mode);
return -EINVAL;
}
r = snd_wl1273_fm_set_i2s_mode(core, rate, width);
if (r)
return r;
wl1273->channels = params_channels(params);
r = snd_wl1273_fm_set_channel_number(core, wl1273->channels);
if (r)
return r;
return 0;
}
static struct snd_soc_dai_ops wl1273_dai_ops = {
.startup = wl1273_startup,
.hw_params = wl1273_hw_params,
};
static struct snd_soc_dai_driver wl1273_dai = {
.name = "wl1273-fm",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE},
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE},
.ops = &wl1273_dai_ops,
};
/* Audio interface format for the soc_card driver */
int wl1273_get_format(struct snd_soc_codec *codec, unsigned int *fmt)
{
struct wl1273_priv *wl1273;
if (codec == NULL || fmt == NULL)
return -EINVAL;
wl1273 = snd_soc_codec_get_drvdata(codec);
switch (wl1273->mode) {
case WL1273_MODE_FM_RX:
case WL1273_MODE_FM_TX:
*fmt = SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM;
break;
case WL1273_MODE_BT:
*fmt = SND_SOC_DAIFMT_DSP_A |
SND_SOC_DAIFMT_IB_NF |
SND_SOC_DAIFMT_CBM_CFM;
break;
default:
return -EINVAL;
}
return 0;
}
EXPORT_SYMBOL_GPL(wl1273_get_format);
static int wl1273_probe(struct snd_soc_codec *codec)
{
struct wl1273_core **core = codec->dev->platform_data;
struct wl1273_priv *wl1273;
int r;
dev_dbg(codec->dev, "%s.\n", __func__);
if (!core) {
dev_err(codec->dev, "Platform data is missing.\n");
return -EINVAL;
}
wl1273 = kzalloc(sizeof(struct wl1273_priv), GFP_KERNEL);
if (wl1273 == NULL) {
dev_err(codec->dev, "Cannot allocate memory.\n");
return -ENOMEM;
}
wl1273->mode = WL1273_MODE_BT;
wl1273->core = *core;
snd_soc_codec_set_drvdata(codec, wl1273);
mutex_init(&codec->mutex);
r = snd_soc_add_controls(codec, wl1273_controls,
ARRAY_SIZE(wl1273_controls));
if (r)
kfree(wl1273);
return r;
}
static int wl1273_remove(struct snd_soc_codec *codec)
{
struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
dev_dbg(codec->dev, "%s\n", __func__);
kfree(wl1273);
return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_wl1273 = {
.probe = wl1273_probe,
.remove = wl1273_remove,
};
static int __devinit wl1273_platform_probe(struct platform_device *pdev)
{
return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wl1273,
&wl1273_dai, 1);
}
static int __devexit wl1273_platform_remove(struct platform_device *pdev)
{
snd_soc_unregister_codec(&pdev->dev);
return 0;
}
MODULE_ALIAS("platform:wl1273-codec");
static struct platform_driver wl1273_platform_driver = {
.driver = {
.name = "wl1273-codec",
.owner = THIS_MODULE,
},
.probe = wl1273_platform_probe,
.remove = __devexit_p(wl1273_platform_remove),
};
static int __init wl1273_init(void)
{
return platform_driver_register(&wl1273_platform_driver);
}
module_init(wl1273_init);
static void __exit wl1273_exit(void)
{
platform_driver_unregister(&wl1273_platform_driver);
}
module_exit(wl1273_exit);
MODULE_AUTHOR("Matti Aaltonen <matti.j.aaltonen@nokia.com>");
MODULE_DESCRIPTION("ASoC WL1273 codec driver");
MODULE_LICENSE("GPL");

101
sound/soc/codecs/wl1273.h Normal file
View File

@ -0,0 +1,101 @@
/*
* sound/soc/codec/wl1273.h
*
* ALSA SoC WL1273 codec driver
*
* Copyright (C) Nokia Corporation
* Author: Matti Aaltonen <matti.j.aaltonen@nokia.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.
*
* 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.
*
* 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., 51 Franklin St, Fifth Floor, Boston, MA
* 02110-1301 USA
*
*/
#ifndef __WL1273_CODEC_H__
#define __WL1273_CODEC_H__
/* I2S protocol, left channel first, data width 16 bits */
#define WL1273_PCM_DEF_MODE 0x00
/* Rx */
#define WL1273_AUDIO_ENABLE_I2S (1 << 0)
#define WL1273_AUDIO_ENABLE_ANALOG (1 << 1)
/* Tx */
#define WL1273_AUDIO_IO_SET_ANALOG 0
#define WL1273_AUDIO_IO_SET_I2S 1
#define WL1273_POWER_SET_OFF 0
#define WL1273_POWER_SET_FM (1 << 0)
#define WL1273_POWER_SET_RDS (1 << 1)
#define WL1273_POWER_SET_RETENTION (1 << 4)
#define WL1273_PUPD_SET_OFF 0x00
#define WL1273_PUPD_SET_ON 0x01
#define WL1273_PUPD_SET_RETENTION 0x10
/* I2S mode */
#define WL1273_IS2_WIDTH_32 0x0
#define WL1273_IS2_WIDTH_40 0x1
#define WL1273_IS2_WIDTH_22_23 0x2
#define WL1273_IS2_WIDTH_23_22 0x3
#define WL1273_IS2_WIDTH_48 0x4
#define WL1273_IS2_WIDTH_50 0x5
#define WL1273_IS2_WIDTH_60 0x6
#define WL1273_IS2_WIDTH_64 0x7
#define WL1273_IS2_WIDTH_80 0x8
#define WL1273_IS2_WIDTH_96 0x9
#define WL1273_IS2_WIDTH_128 0xa
#define WL1273_IS2_WIDTH 0xf
#define WL1273_IS2_FORMAT_STD (0x0 << 4)
#define WL1273_IS2_FORMAT_LEFT (0x1 << 4)
#define WL1273_IS2_FORMAT_RIGHT (0x2 << 4)
#define WL1273_IS2_FORMAT_USER (0x3 << 4)
#define WL1273_IS2_MASTER (0x0 << 6)
#define WL1273_IS2_SLAVEW (0x1 << 6)
#define WL1273_IS2_TRI_AFTER_SENDING (0x0 << 7)
#define WL1273_IS2_TRI_ALWAYS_ACTIVE (0x1 << 7)
#define WL1273_IS2_SDOWS_RR (0x0 << 8)
#define WL1273_IS2_SDOWS_RF (0x1 << 8)
#define WL1273_IS2_SDOWS_FR (0x2 << 8)
#define WL1273_IS2_SDOWS_FF (0x3 << 8)
#define WL1273_IS2_TRI_OPT (0x0 << 10)
#define WL1273_IS2_TRI_ALWAYS (0x1 << 10)
#define WL1273_IS2_RATE_48K (0x0 << 12)
#define WL1273_IS2_RATE_44_1K (0x1 << 12)
#define WL1273_IS2_RATE_32K (0x2 << 12)
#define WL1273_IS2_RATE_22_05K (0x4 << 12)
#define WL1273_IS2_RATE_16K (0x5 << 12)
#define WL1273_IS2_RATE_12K (0x8 << 12)
#define WL1273_IS2_RATE_11_025 (0x9 << 12)
#define WL1273_IS2_RATE_8K (0xa << 12)
#define WL1273_IS2_RATE (0xf << 12)
#define WL1273_I2S_DEF_MODE (WL1273_IS2_WIDTH_32 | \
WL1273_IS2_FORMAT_STD | \
WL1273_IS2_MASTER | \
WL1273_IS2_TRI_AFTER_SENDING | \
WL1273_IS2_SDOWS_RR | \
WL1273_IS2_TRI_OPT | \
WL1273_IS2_RATE_48K)
int wl1273_get_format(struct snd_soc_codec *codec, unsigned int *fmt);
#endif /* End of __WL1273_CODEC_H__ */

2
sound/soc/codecs/wm8741.c
View File

@ -311,7 +311,7 @@ static struct snd_soc_dai_ops wm8741_dai_ops = {
};
static struct snd_soc_dai_driver wm8741_dai = {
.name = "WM8741",
.name = "wm8741",
.playback = {
.stream_name = "Playback",
.channels_min = 2, /* Mono modes not yet supported */

12
sound/soc/codecs/wm8994.c
View File

@ -3316,20 +3316,24 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream,
bclk_reg = WM8994_AIF1_BCLK;
rate_reg = WM8994_AIF1_RATE;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ||
wm8994->lrclk_shared[0])
wm8994->lrclk_shared[0]) {
lrclk_reg = WM8994_AIF1DAC_LRCLK;
else
} else {
lrclk_reg = WM8994_AIF1ADC_LRCLK;
dev_dbg(codec->dev, "AIF1 using split LRCLK\n");
}
break;
case 2:
aif1_reg = WM8994_AIF2_CONTROL_1;
bclk_reg = WM8994_AIF2_BCLK;
rate_reg = WM8994_AIF2_RATE;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ||
wm8994->lrclk_shared[1])
wm8994->lrclk_shared[1]) {
lrclk_reg = WM8994_AIF2DAC_LRCLK;
else
} else {
lrclk_reg = WM8994_AIF2ADC_LRCLK;
dev_dbg(codec->dev, "AIF2 using split LRCLK\n");
}
break;
default:
return -EINVAL;

24
sound/soc/fsl/Kconfig
View File

@ -1,24 +1,36 @@
config SND_MPC52xx_DMA
tristate
# ASoC platform support for the Freescale MPC8610 SOC. This compiles drivers
# for the SSI and the Elo DMA controller. You will still need to select
# a platform driver and a codec driver.
config SND_SOC_MPC8610
# ASoC platform support for the Freescale PowerPC SOCs that have an SSI and
# an Elo DMA controller, such as the MPC8610 and P1022. You will still need to
# select a platform driver and a codec driver.
config SND_SOC_POWERPC_SSI
tristate
depends on MPC8610
depends on FSL_SOC
config SND_SOC_MPC8610_HPCD
tristate "ALSA SoC support for the Freescale MPC8610 HPCD board"
# I2C is necessary for the CS4270 driver
depends on MPC8610_HPCD && I2C
select SND_SOC_MPC8610
select SND_SOC_POWERPC_SSI
select SND_SOC_CS4270
select SND_SOC_CS4270_VD33_ERRATA
default y if MPC8610_HPCD
help
Say Y if you want to enable audio on the Freescale MPC8610 HPCD.
config SND_SOC_P1022_DS
tristate "ALSA SoC support for the Freescale P1022 DS board"
# I2C is necessary for the WM8776 driver
depends on P1022_DS && I2C
select SND_SOC_POWERPC_SSI
select SND_SOC_WM8776
default y if P1022_DS
help
Say Y if you want to enable audio on the Freescale P1022 DS board.
This will also include the Wolfson Microelectronics WM8776 codec
driver.
config SND_SOC_MPC5200_I2S
tristate "Freescale MPC5200 PSC in I2S mode driver"
depends on PPC_MPC52xx && PPC_BESTCOMM

8
sound/soc/fsl/Makefile
View File

@ -2,10 +2,14 @@
snd-soc-mpc8610-hpcd-objs := mpc8610_hpcd.o
obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += snd-soc-mpc8610-hpcd.o
# MPC8610 Platform Support
# P1022 DS Machine Support
snd-soc-p1022-ds-objs := p1022_ds.o
obj-$(CONFIG_SND_SOC_P1022_DS) += snd-soc-p1022-ds.o
# Freescale PowerPC SSI/DMA Platform Support
snd-soc-fsl-ssi-objs := fsl_ssi.o
snd-soc-fsl-dma-objs := fsl_dma.o
obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o
obj-$(CONFIG_SND_SOC_POWERPC_SSI) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o
# MPC5200 Platform Support
obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o

122
sound/soc/fsl/fsl_dma.c
View File

@ -23,6 +23,7 @@
#include <linux/gfp.h>
#include <linux/of_platform.h>
#include <linux/list.h>
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/pcm.h>
@ -60,6 +61,7 @@ struct dma_object {
struct snd_soc_platform_driver dai;
dma_addr_t ssi_stx_phys;
dma_addr_t ssi_srx_phys;
unsigned int ssi_fifo_depth;
struct ccsr_dma_channel __iomem *channel;
unsigned int irq;
bool assigned;
@ -99,6 +101,7 @@ struct fsl_dma_private {
unsigned int irq;
struct snd_pcm_substream *substream;
dma_addr_t ssi_sxx_phys;
unsigned int ssi_fifo_depth;
dma_addr_t ld_buf_phys;
unsigned int current_link;
dma_addr_t dma_buf_phys;
@ -303,21 +306,29 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
if (!card->dev->coherent_dma_mask)
card->dev->coherent_dma_mask = fsl_dma_dmamask;
ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev,
fsl_dma_hardware.buffer_bytes_max,
&pcm->streams[0].substream->dma_buffer);
if (ret) {
dev_err(card->dev, "can't allocate playback dma buffer\n");
return ret;
/* Some codecs have separate DAIs for playback and capture, so we
* should allocate a DMA buffer only for the streams that are valid.
*/
if (dai->driver->playback.channels_min) {
ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev,
fsl_dma_hardware.buffer_bytes_max,
&pcm->streams[0].substream->dma_buffer);
if (ret) {
dev_err(card->dev, "can't alloc playback dma buffer\n");
return ret;
}
}
ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev,
fsl_dma_hardware.buffer_bytes_max,
&pcm->streams[1].substream->dma_buffer);
if (ret) {
snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
dev_err(card->dev, "can't allocate capture dma buffer\n");
return ret;
if (dai->driver->capture.channels_min) {
ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev,
fsl_dma_hardware.buffer_bytes_max,
&pcm->streams[1].substream->dma_buffer);
if (ret) {
snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
dev_err(card->dev, "can't alloc capture dma buffer\n");
return ret;
}
}
return 0;
@ -431,6 +442,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream)
else
dma_private->ssi_sxx_phys = dma->ssi_srx_phys;
dma_private->ssi_fifo_depth = dma->ssi_fifo_depth;
dma_private->dma_channel = dma->channel;
dma_private->irq = dma->irq;
dma_private->substream = substream;
@ -544,11 +556,11 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
struct device *dev = rtd->platform->dev;
/* Number of bits per sample */
unsigned int sample_size =
unsigned int sample_bits =
snd_pcm_format_physical_width(params_format(hw_params));
/* Number of bytes per frame */
unsigned int frame_size = 2 * (sample_size / 8);
unsigned int sample_bytes = sample_bits / 8;
/* Bus address of SSI STX register */
dma_addr_t ssi_sxx_phys = dma_private->ssi_sxx_phys;
@ -588,7 +600,7 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
* that offset here. While we're at it, also tell the DMA controller
* how much data to transfer per sample.
*/
switch (sample_size) {
switch (sample_bits) {
case 8:
mr |= CCSR_DMA_MR_DAHTS_1 | CCSR_DMA_MR_SAHTS_1;
ssi_sxx_phys += 3;
@ -602,22 +614,42 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
break;
default:
/* We should never get here */
dev_err(dev, "unsupported sample size %u\n", sample_size);
dev_err(dev, "unsupported sample size %u\n", sample_bits);
return -EINVAL;
}
/*
* BWC should always be a multiple of the frame size. BWC determines
* how many bytes are sent/received before the DMA controller checks the
* SSI to see if it needs to stop. For playback, the transmit FIFO can
* hold three frames, so we want to send two frames at a time. For
* capture, the receive FIFO is triggered when it contains one frame, so
* we want to receive one frame at a time.
* BWC determines how many bytes are sent/received before the DMA
* controller checks the SSI to see if it needs to stop. BWC should
* always be a multiple of the frame size, so that we always transmit
* whole frames. Each frame occupies two slots in the FIFO. The
* parameter for CCSR_DMA_MR_BWC() is rounded down the next power of two
* (MR[BWC] can only represent even powers of two).
*
* To simplify the process, we set BWC to the largest value that is
* less than or equal to the FIFO watermark. For playback, this ensures
* that we transfer the maximum amount without overrunning the FIFO.
* For capture, this ensures that we transfer the maximum amount without
* underrunning the FIFO.
*
* f = SSI FIFO depth
* w = SSI watermark value (which equals f - 2)
* b = DMA bandwidth count (in bytes)
* s = sample size (in bytes, which equals frame_size * 2)
*
* For playback, we never transmit more than the transmit FIFO
* watermark, otherwise we might write more data than the FIFO can hold.
* The watermark is equal to the FIFO depth minus two.
*
* For capture, two equations must hold:
* w > f - (b / s)
* w >= b / s
*
* So, b > 2 * s, but b must also be <= s * w. To simplify, we set
* b = s * w, which is equal to
* (dma_private->ssi_fifo_depth - 2) * sample_bytes.
*/
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
mr |= CCSR_DMA_MR_BWC(2 * frame_size);
else
mr |= CCSR_DMA_MR_BWC(frame_size);
mr |= CCSR_DMA_MR_BWC((dma_private->ssi_fifo_depth - 2) * sample_bytes);
out_be32(&dma_channel->mr, mr);
@ -864,32 +896,35 @@ static struct snd_pcm_ops fsl_dma_ops = {
.pointer = fsl_dma_pointer,
};
static int __devinit fsl_soc_dma_probe(struct of_device *of_dev,
static int __devinit fsl_soc_dma_probe(struct platform_device *pdev,
const struct of_device_id *match)
{
struct dma_object *dma;
struct device_node *np = of_dev->dev.of_node;
struct device_node *np = pdev->dev.of_node;
struct device_node *ssi_np;
struct resource res;
const uint32_t *iprop;
int ret;
/* Find the SSI node that points to us. */
ssi_np = find_ssi_node(np);
if (!ssi_np) {
dev_err(&of_dev->dev, "cannot find parent SSI node\n");
dev_err(&pdev->dev, "cannot find parent SSI node\n");
return -ENODEV;
}
ret = of_address_to_resource(ssi_np, 0, &res);
of_node_put(ssi_np);
if (ret) {
dev_err(&of_dev->dev, "could not determine device resources\n");
dev_err(&pdev->dev, "could not determine resources for %s\n",
ssi_np->full_name);
of_node_put(ssi_np);
return ret;
}
dma = kzalloc(sizeof(*dma) + strlen(np->full_name), GFP_KERNEL);
if (!dma) {
dev_err(&of_dev->dev, "could not allocate dma object\n");
dev_err(&pdev->dev, "could not allocate dma object\n");
of_node_put(ssi_np);
return -ENOMEM;
}
@ -902,9 +937,18 @@ static int __devinit fsl_soc_dma_probe(struct of_device *of_dev,
dma->ssi_stx_phys = res.start + offsetof(struct ccsr_ssi, stx0);
dma->ssi_srx_phys = res.start + offsetof(struct ccsr_ssi, srx0);
ret = snd_soc_register_platform(&of_dev->dev, &dma->dai);
iprop = of_get_property(ssi_np, "fsl,fifo-depth", NULL);
if (iprop)
dma->ssi_fifo_depth = *iprop;
else
/* Older 8610 DTs didn't have the fifo-depth property */
dma->ssi_fifo_depth = 8;
of_node_put(ssi_np);
ret = snd_soc_register_platform(&pdev->dev, &dma->dai);
if (ret) {
dev_err(&of_dev->dev, "could not register platform\n");
dev_err(&pdev->dev, "could not register platform\n");
kfree(dma);
return ret;
}
@ -912,16 +956,16 @@ static int __devinit fsl_soc_dma_probe(struct of_device *of_dev,
dma->channel = of_iomap(np, 0);
dma->irq = irq_of_parse_and_map(np, 0);
dev_set_drvdata(&of_dev->dev, dma);
dev_set_drvdata(&pdev->dev, dma);
return 0;
}
static int __devexit fsl_soc_dma_remove(struct of_device *of_dev)
static int __devexit fsl_soc_dma_remove(struct platform_device *pdev)
{
struct dma_object *dma = dev_get_drvdata(&of_dev->dev);
struct dma_object *dma = dev_get_drvdata(&pdev->dev);
snd_soc_unregister_platform(&of_dev->dev);
snd_soc_unregister_platform(&pdev->dev);
iounmap(dma->channel);
irq_dispose_mapping(dma->irq);
kfree(dma);

67
sound/soc/fsl/fsl_ssi.c
View File

@ -93,6 +93,7 @@ struct fsl_ssi_private {
unsigned int playback;
unsigned int capture;
int asynchronous;
unsigned int fifo_depth;
struct snd_soc_dai_driver cpu_dai_drv;
struct device_attribute dev_attr;
struct platform_device *pdev;
@ -337,11 +338,20 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
/*
* Set the watermark for transmit FIFI 0 and receive FIFO 0. We
* don't use FIFO 1. Since the SSI only supports stereo, the
* watermark should never be an odd number.
* don't use FIFO 1. We program the transmit water to signal a
* DMA transfer if there are only two (or fewer) elements left
* in the FIFO. Two elements equals one frame (left channel,
* right channel). This value, however, depends on the depth of
* the transmit buffer.
*
* We program the receive FIFO to notify us if at least two
* elements (one frame) have been written to the FIFO. We could
* make this value larger (and maybe we should), but this way
* data will be written to memory as soon as it's available.
*/
out_be32(&ssi->sfcsr,
CCSR_SSI_SFCSR_TFWM0(6) | CCSR_SSI_SFCSR_RFWM0(2));
CCSR_SSI_SFCSR_TFWM0(ssi_private->fifo_depth - 2) |
CCSR_SSI_SFCSR_RFWM0(ssi_private->fifo_depth - 2));
/*
* We keep the SSI disabled because if we enable it, then the
@ -614,14 +624,15 @@ static void make_lowercase(char *s)
}
}
static int __devinit fsl_ssi_probe(struct of_device *of_dev,
static int __devinit fsl_ssi_probe(struct platform_device *pdev,
const struct of_device_id *match)
{
struct fsl_ssi_private *ssi_private;
int ret = 0;
struct device_attribute *dev_attr = NULL;
struct device_node *np = of_dev->dev.of_node;
struct device_node *np = pdev->dev.of_node;
const char *p, *sprop;
const uint32_t *iprop;
struct resource res;
char name[64];
@ -634,14 +645,14 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev,
/* Check for a codec-handle property. */
if (!of_get_property(np, "codec-handle", NULL)) {
dev_err(&of_dev->dev, "missing codec-handle property\n");
dev_err(&pdev->dev, "missing codec-handle property\n");
return -ENODEV;
}
/* We only support the SSI in "I2S Slave" mode */
sprop = of_get_property(np, "fsl,mode", NULL);
if (!sprop || strcmp(sprop, "i2s-slave")) {
dev_notice(&of_dev->dev, "mode %s is unsupported\n", sprop);
dev_notice(&pdev->dev, "mode %s is unsupported\n", sprop);
return -ENODEV;
}
@ -650,7 +661,7 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev,
ssi_private = kzalloc(sizeof(struct fsl_ssi_private) + strlen(p),
GFP_KERNEL);
if (!ssi_private) {
dev_err(&of_dev->dev, "could not allocate DAI object\n");
dev_err(&pdev->dev, "could not allocate DAI object\n");
return -ENOMEM;
}
@ -664,7 +675,7 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev,
/* Get the addresses and IRQ */
ret = of_address_to_resource(np, 0, &res);
if (ret) {
dev_err(&of_dev->dev, "could not determine device resources\n");
dev_err(&pdev->dev, "could not determine device resources\n");
kfree(ssi_private);
return ret;
}
@ -678,25 +689,33 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev,
else
ssi_private->cpu_dai_drv.symmetric_rates = 1;
/* Determine the FIFO depth. */
iprop = of_get_property(np, "fsl,fifo-depth", NULL);
if (iprop)
ssi_private->fifo_depth = *iprop;
else
/* Older 8610 DTs didn't have the fifo-depth property */
ssi_private->fifo_depth = 8;
/* Initialize the the device_attribute structure */
dev_attr = &ssi_private->dev_attr;
dev_attr->attr.name = "statistics";
dev_attr->attr.mode = S_IRUGO;
dev_attr->show = fsl_sysfs_ssi_show;
ret = device_create_file(&of_dev->dev, dev_attr);
ret = device_create_file(&pdev->dev, dev_attr);
if (ret) {
dev_err(&of_dev->dev, "could not create sysfs %s file\n",
dev_err(&pdev->dev, "could not create sysfs %s file\n",
ssi_private->dev_attr.attr.name);
goto error;
}
/* Register with ASoC */
dev_set_drvdata(&of_dev->dev, ssi_private);
dev_set_drvdata(&pdev->dev, ssi_private);
ret = snd_soc_register_dai(&of_dev->dev, &ssi_private->cpu_dai_drv);
ret = snd_soc_register_dai(&pdev->dev, &ssi_private->cpu_dai_drv);
if (ret) {
dev_err(&of_dev->dev, "failed to register DAI: %d\n", ret);
dev_err(&pdev->dev, "failed to register DAI: %d\n", ret);
goto error;
}
@ -714,20 +733,20 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev,
make_lowercase(name);
ssi_private->pdev =
platform_device_register_data(&of_dev->dev, name, 0, NULL, 0);
platform_device_register_data(&pdev->dev, name, 0, NULL, 0);
if (IS_ERR(ssi_private->pdev)) {
ret = PTR_ERR(ssi_private->pdev);
dev_err(&of_dev->dev, "failed to register platform: %d\n", ret);
dev_err(&pdev->dev, "failed to register platform: %d\n", ret);
goto error;
}
return 0;
error:
snd_soc_unregister_dai(&of_dev->dev);
dev_set_drvdata(&of_dev->dev, NULL);
snd_soc_unregister_dai(&pdev->dev);
dev_set_drvdata(&pdev->dev, NULL);
if (dev_attr)
device_remove_file(&of_dev->dev, dev_attr);
device_remove_file(&pdev->dev, dev_attr);
irq_dispose_mapping(ssi_private->irq);
iounmap(ssi_private->ssi);
kfree(ssi_private);
@ -735,16 +754,16 @@ error:
return ret;
}
static int fsl_ssi_remove(struct of_device *of_dev)
static int fsl_ssi_remove(struct platform_device *pdev)
{
struct fsl_ssi_private *ssi_private = dev_get_drvdata(&of_dev->dev);
struct fsl_ssi_private *ssi_private = dev_get_drvdata(&pdev->dev);
platform_device_unregister(ssi_private->pdev);
snd_soc_unregister_dai(&of_dev->dev);
device_remove_file(&of_dev->dev, &ssi_private->dev_attr);
snd_soc_unregister_dai(&pdev->dev);
device_remove_file(&pdev->dev, &ssi_private->dev_attr);
kfree(ssi_private);
dev_set_drvdata(&of_dev->dev, NULL);
dev_set_drvdata(&pdev->dev, NULL);
return 0;
}

10
sound/soc/fsl/mpc8610_hpcd.c
View File

@ -13,6 +13,7 @@
#include <linux/module.h>
#include <linux/interrupt.h>
#include <linux/of_device.h>
#include <linux/slab.h>
#include <sound/soc.h>
#include <asm/fsl_guts.h>
@ -323,9 +324,10 @@ static int get_dma_channel(struct device_node *ssi_np,
static int mpc8610_hpcd_probe(struct platform_device *pdev)
{
struct device *dev = pdev->dev.parent;
/* of_dev is the OF device for the SSI node that probed us */
struct of_device *of_dev = container_of(dev, struct of_device, dev);
struct device_node *np = of_dev->dev.of_node;
/* ssi_pdev is the platform device for the SSI node that probed us */
struct platform_device *ssi_pdev =
container_of(dev, struct platform_device, dev);
struct device_node *np = ssi_pdev->dev.of_node;
struct device_node *codec_np = NULL;
struct platform_device *sound_device = NULL;
struct mpc8610_hpcd_data *machine_data;
@ -348,7 +350,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
if (!machine_data)
return -ENOMEM;
machine_data->dai[0].cpu_dai_name = dev_name(&of_dev->dev);
machine_data->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev);
machine_data->dai[0].ops = &mpc8610_hpcd_ops;
/* Determine the codec name, it will be used as the codec DAI name */

590
sound/soc/fsl/p1022_ds.c Normal file
View File

@ -0,0 +1,590 @@
/**
* Freescale P1022DS ALSA SoC Machine driver
*
* Author: Timur Tabi <timur@freescale.com>
*
* Copyright 2010 Freescale Semiconductor, Inc.
*
* This file is licensed under the terms of the GNU General Public License
* version 2. This program is licensed "as is" without any warranty of any
* kind, whether express or implied.
*/
#include <linux/module.h>
#include <linux/interrupt.h>
#include <linux/of_device.h>
#include <linux/slab.h>
#include <sound/soc.h>
#include <asm/fsl_guts.h>
#include "fsl_dma.h"
#include "fsl_ssi.h"
/* P1022-specific PMUXCR and DMUXCR bit definitions */
#define CCSR_GUTS_PMUXCR_UART0_I2C1_MASK 0x0001c000
#define CCSR_GUTS_PMUXCR_UART0_I2C1_UART0_SSI 0x00010000
#define CCSR_GUTS_PMUXCR_UART0_I2C1_SSI 0x00018000
#define CCSR_GUTS_PMUXCR_SSI_DMA_TDM_MASK 0x00000c00
#define CCSR_GUTS_PMUXCR_SSI_DMA_TDM_SSI 0x00000000
#define CCSR_GUTS_DMUXCR_PAD 1 /* DMA controller/channel set to pad */
#define CCSR_GUTS_DMUXCR_SSI 2 /* DMA controller/channel set to SSI */
/*
* Set the DMACR register in the GUTS
*
* The DMACR register determines the source of initiated transfers for each
* channel on each DMA controller. Rather than have a bunch of repetitive
* macros for the bit patterns, we just have a function that calculates
* them.
*
* guts: Pointer to GUTS structure
* co: The DMA controller (0 or 1)
* ch: The channel on the DMA controller (0, 1, 2, or 3)
* device: The device to set as the target (CCSR_GUTS_DMUXCR_xxx)
*/
static inline void guts_set_dmuxcr(struct ccsr_guts_85xx __iomem *guts,
unsigned int co, unsigned int ch, unsigned int device)
{
unsigned int shift = 16 + (8 * (1 - co) + 2 * (3 - ch));
clrsetbits_be32(&guts->dmuxcr, 3 << shift, device << shift);
}
/* There's only one global utilities register */
static phys_addr_t guts_phys;
#define DAI_NAME_SIZE 32
/**
* machine_data: machine-specific ASoC device data
*
* This structure contains data for a single sound platform device on an
* P1022 DS. Some of the data is taken from the device tree.
*/
struct machine_data {
struct snd_soc_dai_link dai[2];
struct snd_soc_card card;
unsigned int dai_format;
unsigned int codec_clk_direction;
unsigned int cpu_clk_direction;
unsigned int clk_frequency;
unsigned int ssi_id; /* 0 = SSI1, 1 = SSI2, etc */
unsigned int dma_id[2]; /* 0 = DMA1, 1 = DMA2, etc */
unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/
char codec_name[DAI_NAME_SIZE];
char platform_name[2][DAI_NAME_SIZE]; /* One for each DMA channel */
};
/**
* p1022_ds_machine_probe: initialize the board
*
* This function is used to initialize the board-specific hardware.
*
* Here we program the DMACR and PMUXCR registers.
*/
static int p1022_ds_machine_probe(struct platform_device *sound_device)
{
struct snd_soc_card *card = platform_get_drvdata(sound_device);
struct machine_data *mdata =
container_of(card, struct machine_data, card);
struct ccsr_guts_85xx __iomem *guts;
guts = ioremap(guts_phys, sizeof(struct ccsr_guts_85xx));
if (!guts) {
dev_err(card->dev, "could not map global utilities\n");
return -ENOMEM;
}
/* Enable SSI Tx signal */
clrsetbits_be32(&guts->pmuxcr, CCSR_GUTS_PMUXCR_UART0_I2C1_MASK,
CCSR_GUTS_PMUXCR_UART0_I2C1_UART0_SSI);
/* Enable SSI Rx signal */
clrsetbits_be32(&guts->pmuxcr, CCSR_GUTS_PMUXCR_SSI_DMA_TDM_MASK,
CCSR_GUTS_PMUXCR_SSI_DMA_TDM_SSI);
/* Enable DMA Channel for SSI */
guts_set_dmuxcr(guts, mdata->dma_id[0], mdata->dma_channel_id[0],
CCSR_GUTS_DMUXCR_SSI);
guts_set_dmuxcr(guts, mdata->dma_id[1], mdata->dma_channel_id[1],
CCSR_GUTS_DMUXCR_SSI);
iounmap(guts);
return 0;
}
/**
* p1022_ds_startup: program the board with various hardware parameters
*
* This function takes board-specific information, like clock frequencies
* and serial data formats, and passes that information to the codec and
* transport drivers.
*/
static int p1022_ds_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct machine_data *mdata =
container_of(rtd->card, struct machine_data, card);
struct device *dev = rtd->card->dev;
int ret = 0;
/* Tell the codec driver what the serial protocol is. */
ret = snd_soc_dai_set_fmt(rtd->codec_dai, mdata->dai_format);
if (ret < 0) {
dev_err(dev, "could not set codec driver audio format\n");
return ret;
}
/*
* Tell the codec driver what the MCLK frequency is, and whether it's
* a slave or master.
*/
ret = snd_soc_dai_set_sysclk(rtd->codec_dai, 0, mdata->clk_frequency,
mdata->codec_clk_direction);
if (ret < 0) {
dev_err(dev, "could not set codec driver clock params\n");
return ret;
}
return 0;
}
/**
* p1022_ds_machine_remove: Remove the sound device
*
* This function is called to remove the sound device for one SSI. We
* de-program the DMACR and PMUXCR register.
*/
static int p1022_ds_machine_remove(struct platform_device *sound_device)
{
struct snd_soc_card *card = platform_get_drvdata(sound_device);
struct machine_data *mdata =
container_of(card, struct machine_data, card);
struct ccsr_guts_85xx __iomem *guts;
guts = ioremap(guts_phys, sizeof(struct ccsr_guts_85xx));
if (!guts) {
dev_err(card->dev, "could not map global utilities\n");
return -ENOMEM;
}
/* Restore the signal routing */
clrbits32(&guts->pmuxcr, CCSR_GUTS_PMUXCR_UART0_I2C1_MASK);
clrbits32(&guts->pmuxcr, CCSR_GUTS_PMUXCR_SSI_DMA_TDM_MASK);
guts_set_dmuxcr(guts, mdata->dma_id[0], mdata->dma_channel_id[0], 0);
guts_set_dmuxcr(guts, mdata->dma_id[1], mdata->dma_channel_id[1], 0);
iounmap(guts);
return 0;
}
/**
* p1022_ds_ops: ASoC machine driver operations
*/
static struct snd_soc_ops p1022_ds_ops = {
.startup = p1022_ds_startup,
};
/**
* get_node_by_phandle_name - get a node by its phandle name
*
* This function takes a node, the name of a property in that node, and a
* compatible string. Assuming the property is a phandle to another node,
* it returns that node, (optionally) if that node is compatible.
*
* If the property is not a phandle, or the node it points to is not compatible
* with the specific string, then NULL is returned.
*/
static struct device_node *get_node_by_phandle_name(struct device_node *np,
const char *name, const char *compatible)
{
np = of_parse_phandle(np, name, 0);
if (!np)
return NULL;
if (!of_device_is_compatible(np, compatible)) {
of_node_put(np);
return NULL;
}
return np;
}
/**
* get_parent_cell_index -- return the cell-index of the parent of a node
*
* Return the value of the cell-index property of the parent of the given
* node. This is used for DMA channel nodes that need to know the DMA ID
* of the controller they are on.
*/
static int get_parent_cell_index(struct device_node *np)
{
struct device_node *parent = of_get_parent(np);
const u32 *iprop;
int ret = -1;
if (!parent)
return -1;
iprop = of_get_property(parent, "cell-index", NULL);
if (iprop)
ret = *iprop;
of_node_put(parent);
return ret;
}
/**
* codec_node_dev_name - determine the dev_name for a codec node
*
* This function determines the dev_name for an I2C node. This is the name
* that would be returned by dev_name() if this device_node were part of a
* 'struct device' It's ugly and hackish, but it works.
*
* The dev_name for such devices include the bus number and I2C address. For
* example, "cs4270-codec.0-004f".
*/
static int codec_node_dev_name(struct device_node *np, char *buf, size_t len)
{
const u32 *iprop;
int bus, addr;
char temp[DAI_NAME_SIZE];
of_modalias_node(np, temp, DAI_NAME_SIZE);
iprop = of_get_property(np, "reg", NULL);
if (!iprop)
return -EINVAL;
addr = *iprop;
bus = get_parent_cell_index(np);
if (bus < 0)
return bus;
snprintf(buf, len, "%s-codec.%u-%04x", temp, bus, addr);
return 0;
}
static int get_dma_channel(struct device_node *ssi_np,
const char *compatible,
struct snd_soc_dai_link *dai,
unsigned int *dma_channel_id,
unsigned int *dma_id)
{
struct resource res;
struct device_node *dma_channel_np;
const u32 *iprop;
int ret;
dma_channel_np = get_node_by_phandle_name(ssi_np, compatible,
"fsl,ssi-dma-channel");
if (!dma_channel_np)
return -EINVAL;
/* Determine the dev_name for the device_node. This code mimics the
* behavior of of_device_make_bus_id(). We need this because ASoC uses
* the dev_name() of the device to match the platform (DMA) device with
* the CPU (SSI) device. It's all ugly and hackish, but it works (for
* now).
*
* dai->platform name should already point to an allocated buffer.
*/
ret = of_address_to_resource(dma_channel_np, 0, &res);
if (ret)
return ret;
snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s",
(unsigned long long) res.start, dma_channel_np->name);
iprop = of_get_property(dma_channel_np, "cell-index", NULL);
if (!iprop) {
of_node_put(dma_channel_np);
return -EINVAL;
}
*dma_channel_id = *iprop;
*dma_id = get_parent_cell_index(dma_channel_np);
of_node_put(dma_channel_np);
return 0;
}
/**
* p1022_ds_probe: platform probe function for the machine driver
*
* Although this is a machine driver, the SSI node is the "master" node with
* respect to audio hardware connections. Therefore, we create a new ASoC
* device for each new SSI node that has a codec attached.
*/
static int p1022_ds_probe(struct platform_device *pdev)
{
struct device *dev = pdev->dev.parent;
/* ssi_pdev is the platform device for the SSI node that probed us */
struct platform_device *ssi_pdev =
container_of(dev, struct platform_device, dev);
struct device_node *np = ssi_pdev->dev.of_node;
struct device_node *codec_np = NULL;
struct platform_device *sound_device = NULL;
struct machine_data *mdata;
int ret = -ENODEV;
const char *sprop;
const u32 *iprop;
/* Find the codec node for this SSI. */
codec_np = of_parse_phandle(np, "codec-handle", 0);
if (!codec_np) {
dev_err(dev, "could not find codec node\n");
return -EINVAL;
}
mdata = kzalloc(sizeof(struct machine_data), GFP_KERNEL);
if (!mdata)
return -ENOMEM;
mdata->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev);
mdata->dai[0].ops = &p1022_ds_ops;
/* Determine the codec name, it will be used as the codec DAI name */
ret = codec_node_dev_name(codec_np, mdata->codec_name, DAI_NAME_SIZE);
if (ret) {
dev_err(&pdev->dev, "invalid codec node %s\n",
codec_np->full_name);
ret = -EINVAL;
goto error;
}
mdata->dai[0].codec_name = mdata->codec_name;
/* We register two DAIs per SSI, one for playback and the other for
* capture. We support codecs that have separate DAIs for both playback
* and capture.
*/
memcpy(&mdata->dai[1], &mdata->dai[0], sizeof(struct snd_soc_dai_link));
/* The DAI names from the codec (snd_soc_dai_driver.name) */
mdata->dai[0].codec_dai_name = "wm8776-hifi-playback";
mdata->dai[1].codec_dai_name = "wm8776-hifi-capture";
/* Get the device ID */
iprop = of_get_property(np, "cell-index", NULL);
if (!iprop) {
dev_err(&pdev->dev, "cell-index property not found\n");
ret = -EINVAL;
goto error;
}
mdata->ssi_id = *iprop;
/* Get the serial format and clock direction. */
sprop = of_get_property(np, "fsl,mode", NULL);
if (!sprop) {
dev_err(&pdev->dev, "fsl,mode property not found\n");
ret = -EINVAL;
goto error;
}
if (strcasecmp(sprop, "i2s-slave") == 0) {
mdata->dai_format = SND_SOC_DAIFMT_I2S;
mdata->codec_clk_direction = SND_SOC_CLOCK_OUT;
mdata->cpu_clk_direction = SND_SOC_CLOCK_IN;
/* In i2s-slave mode, the codec has its own clock source, so we
* need to get the frequency from the device tree and pass it to
* the codec driver.
*/
iprop = of_get_property(codec_np, "clock-frequency", NULL);
if (!iprop || !*iprop) {
dev_err(&pdev->dev, "codec bus-frequency "
"property is missing or invalid\n");
ret = -EINVAL;
goto error;
}
mdata->clk_frequency = *iprop;
} else if (strcasecmp(sprop, "i2s-master") == 0) {
mdata->dai_format = SND_SOC_DAIFMT_I2S;
mdata->codec_clk_direction = SND_SOC_CLOCK_IN;
mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT;
} else if (strcasecmp(sprop, "lj-slave") == 0) {
mdata->dai_format = SND_SOC_DAIFMT_LEFT_J;
mdata->codec_clk_direction = SND_SOC_CLOCK_OUT;
mdata->cpu_clk_direction = SND_SOC_CLOCK_IN;
} else if (strcasecmp(sprop, "lj-master") == 0) {
mdata->dai_format = SND_SOC_DAIFMT_LEFT_J;
mdata->codec_clk_direction = SND_SOC_CLOCK_IN;
mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT;
} else if (strcasecmp(sprop, "rj-slave") == 0) {
mdata->dai_format = SND_SOC_DAIFMT_RIGHT_J;
mdata->codec_clk_direction = SND_SOC_CLOCK_OUT;
mdata->cpu_clk_direction = SND_SOC_CLOCK_IN;
} else if (strcasecmp(sprop, "rj-master") == 0) {
mdata->dai_format = SND_SOC_DAIFMT_RIGHT_J;
mdata->codec_clk_direction = SND_SOC_CLOCK_IN;
mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT;
} else if (strcasecmp(sprop, "ac97-slave") == 0) {
mdata->dai_format = SND_SOC_DAIFMT_AC97;
mdata->codec_clk_direction = SND_SOC_CLOCK_OUT;
mdata->cpu_clk_direction = SND_SOC_CLOCK_IN;
} else if (strcasecmp(sprop, "ac97-master") == 0) {
mdata->dai_format = SND_SOC_DAIFMT_AC97;
mdata->codec_clk_direction = SND_SOC_CLOCK_IN;
mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT;
} else {
dev_err(&pdev->dev,
"unrecognized fsl,mode property '%s'\n", sprop);
ret = -EINVAL;
goto error;
}
if (!mdata->clk_frequency) {
dev_err(&pdev->dev, "unknown clock frequency\n");
ret = -EINVAL;
goto error;
}
/* Find the playback DMA channel to use. */
mdata->dai[0].platform_name = mdata->platform_name[0];
ret = get_dma_channel(np, "fsl,playback-dma", &mdata->dai[0],
&mdata->dma_channel_id[0],
&mdata->dma_id[0]);
if (ret) {
dev_err(&pdev->dev, "missing/invalid playback DMA phandle\n");
goto error;
}
/* Find the capture DMA channel to use. */
mdata->dai[1].platform_name = mdata->platform_name[1];
ret = get_dma_channel(np, "fsl,capture-dma", &mdata->dai[1],
&mdata->dma_channel_id[1],
&mdata->dma_id[1]);
if (ret) {
dev_err(&pdev->dev, "missing/invalid capture DMA phandle\n");
goto error;
}
/* Initialize our DAI data structure. */
mdata->dai[0].stream_name = "playback";
mdata->dai[1].stream_name = "capture";
mdata->dai[0].name = mdata->dai[0].stream_name;
mdata->dai[1].name = mdata->dai[1].stream_name;
mdata->card.probe = p1022_ds_machine_probe;
mdata->card.remove = p1022_ds_machine_remove;
mdata->card.name = pdev->name; /* The platform driver name */
mdata->card.num_links = 2;
mdata->card.dai_link = mdata->dai;
/* Allocate a new audio platform device structure */
sound_device = platform_device_alloc("soc-audio", -1);
if (!sound_device) {
dev_err(&pdev->dev, "platform device alloc failed\n");
ret = -ENOMEM;
goto error;
}
/* Associate the card data with the sound device */
platform_set_drvdata(sound_device, &mdata->card);
/* Register with ASoC */
ret = platform_device_add(sound_device);
if (ret) {
dev_err(&pdev->dev, "platform device add failed\n");
goto error;
}
of_node_put(codec_np);
return 0;
error:
of_node_put(codec_np);
if (sound_device)
platform_device_unregister(sound_device);
kfree(mdata);
return ret;
}
/**
* p1022_ds_remove: remove the platform device
*
* This function is called when the platform device is removed.
*/
static int __devexit p1022_ds_remove(struct platform_device *pdev)
{
struct platform_device *sound_device = dev_get_drvdata(&pdev->dev);
struct snd_soc_card *card = platform_get_drvdata(sound_device);
struct machine_data *mdata =
container_of(card, struct machine_data, card);
platform_device_unregister(sound_device);
kfree(mdata);
sound_device->dev.platform_data = NULL;
dev_set_drvdata(&pdev->dev, NULL);
return 0;
}
static struct platform_driver p1022_ds_driver = {
.probe = p1022_ds_probe,
.remove = __devexit_p(p1022_ds_remove),
.driver = {
/* The name must match the 'model' property in the device tree,
* in lowercase letters, but only the part after that last
* comma. This is because some model properties have a "fsl,"
* prefix.
*/
.name = "snd-soc-p1022",
.owner = THIS_MODULE,
},
};
/**
* p1022_ds_init: machine driver initialization.
*
* This function is called when this module is loaded.
*/
static int __init p1022_ds_init(void)
{
struct device_node *guts_np;
struct resource res;
pr_info("Freescale P1022 DS ALSA SoC machine driver\n");
/* Get the physical address of the global utilities registers */
guts_np = of_find_compatible_node(NULL, NULL, "fsl,p1022-guts");
if (of_address_to_resource(guts_np, 0, &res)) {
pr_err("p1022-ds: missing/invalid global utilities node\n");
return -EINVAL;
}
guts_phys = res.start;
of_node_put(guts_np);
return platform_driver_register(&p1022_ds_driver);
}
/**
* p1022_ds_exit: machine driver exit
*
* This function is called when this driver is unloaded.
*/
static void __exit p1022_ds_exit(void)
{
platform_driver_unregister(&p1022_ds_driver);
}
module_init(p1022_ds_init);
module_exit(p1022_ds_exit);
MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
MODULE_DESCRIPTION("Freescale P1022 DS ALSA SoC machine driver");
MODULE_LICENSE("GPL v2");

3
sound/soc/imx/imx-ssi.c
View File

@ -254,6 +254,9 @@ static int imx_ssi_hw_params(struct snd_pcm_substream *substream,
dma_data = &ssi->dma_params_rx;
}
if (ssi->flags & IMX_SSI_SYN)
reg = SSI_STCCR;
snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
sccr = readl(ssi->base + reg) & ~SSI_STCCR_WL_MASK;

2
sound/soc/omap/ams-delta.c
View File

@ -584,7 +584,7 @@ static struct snd_soc_dai_link ams_delta_dai_link = {
.name = "CX20442",
.stream_name = "CX20442",
.cpu_dai_name ="omap-mcbsp-dai.0",
.codec_dai_name = "cx20442-hifi",
.codec_dai_name = "cx20442-voice",
.init = ams_delta_cx20442_init,
.platform_name = "omap-pcm-audio",
.codec_name = "cx20442-codec",

18
sound/soc/pxa/Kconfig
View File

@ -117,6 +117,24 @@ config SND_PXA2XX_SOC_PALM27X
Say Y if you want to add support for SoC audio on
Palm T|X, T5, E2 or LifeDrive handheld computer.
config SND_SOC_SAARB
tristate "SoC Audio support for Marvell Saarb"
depends on SND_PXA2XX_SOC && MACH_SAARB
select SND_PXA_SOC_SSP
select SND_SOC_88PM860X
help
Say Y if you want to add support for SoC audio on the
Marvell Saarb reference platform.
config SND_SOC_TAVOREVB3
tristate "SoC Audio support for Marvell Tavor EVB3"
depends on SND_PXA2XX_SOC && MACH_TAVOREVB3
select SND_PXA_SOC_SSP
select SND_SOC_88PM860X
help
Say Y if you want to add support for SoC audio on the
Marvell Saarb reference platform.
config SND_SOC_ZYLONITE
tristate "SoC Audio support for Marvell Zylonite"
depends on SND_PXA2XX_SOC && MACH_ZYLONITE

4
sound/soc/pxa/Makefile
View File

@ -19,6 +19,8 @@ snd-soc-e800-objs := e800_wm9712.o
snd-soc-spitz-objs := spitz.o
snd-soc-em-x270-objs := em-x270.o
snd-soc-palm27x-objs := palm27x.o
snd-soc-saarb-objs := saarb.o
snd-soc-tavorevb3-objs := tavorevb3.o
snd-soc-zylonite-objs := zylonite.o
snd-soc-magician-objs := magician.o
snd-soc-mioa701-objs := mioa701_wm9713.o
@ -38,6 +40,8 @@ obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o
obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o
obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o
obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o
obj-$(CONFIG_SND_SOC_SAARB) += snd-soc-saarb.o
obj-$(CONFIG_SND_SOC_TAVOREVB3) += snd-soc-tavorevb3.o
obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o
obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o

3
sound/soc/pxa/e740_wm9705.c
View File

@ -198,6 +198,9 @@ free_mic_amp_gpio:
static void __exit e740_exit(void)
{
platform_device_unregister(e740_snd_device);
gpio_free(GPIO_E740_WM9705_nAVDD2);
gpio_free(GPIO_E740_AMP_ON);
gpio_free(GPIO_E740_MIC_ON);
}
module_init(e740_init);

2
sound/soc/pxa/imote2.c
View File

@ -63,7 +63,7 @@ static struct snd_soc_ops imote2_asoc_ops = {
static struct snd_soc_dai_link imote2_dai = {
.name = "WM8940",
.stream_name = "WM8940",
.cpu_dai_name = "pxa-i2s",
.cpu_dai_name = "pxa2xx-i2s",
.codec_dai_name = "wm8940-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm8940-codec.0-0034",

2
sound/soc/pxa/magician.c
View File

@ -437,7 +437,7 @@ static struct snd_soc_dai_link magician_dai[] = {
{
.name = "uda1380",
.stream_name = "UDA1380 Capture",
.cpu_dai_name = "pxa-i2s",
.cpu_dai_name = "pxa2xx-i2s",
.codec_dai_name = "uda1380-hifi-capture",
.platform_name = "pxa-pcm-audio",
.codec_name = "uda1380-codec.0-0018",

2
sound/soc/pxa/poodle.c
View File

@ -266,7 +266,7 @@ static int poodle_wm8731_init(struct snd_soc_pcm_runtime *rtd)
static struct snd_soc_dai_link poodle_dai = {
.name = "WM8731",
.stream_name = "WM8731",
.cpu_dai_name = "pxa-i2s",
.cpu_dai_name = "pxa2xx-i2s",
.codec_dai_name = "wm8731-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm8731-codec.0-001a",

1
sound/soc/pxa/pxa-ssp.c
View File

@ -758,6 +758,7 @@ static int pxa_ssp_remove(struct snd_soc_dai *dai)
struct ssp_priv *priv = snd_soc_dai_get_drvdata(dai);
pxa_ssp_free(priv->ssp);
kfree(priv);
return 0;
}

1
sound/soc/pxa/pxa2xx-ac97.c
View File

@ -24,7 +24,6 @@
#include <mach/dma.h>
#include <mach/audio.h>
#include "pxa2xx-pcm.h"
#include "pxa2xx-ac97.h"
static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97)

1
sound/soc/pxa/pxa2xx-i2s.c
View File

@ -398,3 +398,4 @@ module_exit(pxa2xx_i2s_exit);
MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
MODULE_DESCRIPTION("pxa2xx I2S SoC Interface");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:pxa2xx-i2s");

200
sound/soc/pxa/saarb.c Normal file
View File

@ -0,0 +1,200 @@
/*
* saarb.c -- SoC audio for saarb
*
* Copyright (C) 2010 Marvell International Ltd.
* Haojian Zhuang <haojian.zhuang@marvell.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/device.h>
#include <linux/clk.h>
#include <linux/i2c.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/jack.h>
#include <asm/mach-types.h>
#include "../codecs/88pm860x-codec.h"
#include "pxa-ssp.h"
static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd);
static struct platform_device *saarb_snd_device;
static struct snd_soc_jack hs_jack, mic_jack;
static struct snd_soc_jack_pin hs_jack_pins[] = {
{ .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, },
};
static struct snd_soc_jack_pin mic_jack_pins[] = {
{ .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, },
};
/* saarb machine dapm widgets */
static const struct snd_soc_dapm_widget saarb_dapm_widgets[] = {
SND_SOC_DAPM_HP("Headphone Stereophone", NULL),
SND_SOC_DAPM_LINE("Lineout Out 1", NULL),
SND_SOC_DAPM_LINE("Lineout Out 2", NULL),
SND_SOC_DAPM_SPK("Ext Speaker", NULL),
SND_SOC_DAPM_MIC("Ext Mic 1", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
SND_SOC_DAPM_MIC("Ext Mic 3", NULL),
};
/* saarb machine audio map */
static const struct snd_soc_dapm_route audio_map[] = {
{"Headset Stereophone", NULL, "HS1"},
{"Headset Stereophone", NULL, "HS2"},
{"Ext Speaker", NULL, "LSP"},
{"Ext Speaker", NULL, "LSN"},
{"Lineout Out 1", NULL, "LINEOUT1"},
{"Lineout Out 2", NULL, "LINEOUT2"},
{"MIC1P", NULL, "Mic1 Bias"},
{"MIC1N", NULL, "Mic1 Bias"},
{"Mic1 Bias", NULL, "Ext Mic 1"},
{"MIC2P", NULL, "Mic1 Bias"},
{"MIC2N", NULL, "Mic1 Bias"},
{"Mic1 Bias", NULL, "Headset Mic 2"},
{"MIC3P", NULL, "Mic3 Bias"},
{"MIC3N", NULL, "Mic3 Bias"},
{"Mic3 Bias", NULL, "Ext Mic 3"},
};
static int saarb_i2s_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->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
int width = snd_pcm_format_physical_width(params_format(params));
int ret;
ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0,
PM860X_CLK_DIR_OUT);
if (ret < 0)
return ret;
ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT);
if (ret < 0)
return 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 < 0)
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 < 0)
return ret;
ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width);
return ret;
}
static struct snd_soc_ops saarb_i2s_ops = {
.hw_params = saarb_i2s_hw_params,
};
static struct snd_soc_dai_link saarb_dai[] = {
{
.name = "88PM860x I2S",
.stream_name = "I2S Audio",
.cpu_dai_name = "pxa-ssp-dai.1",
.codec_dai_name = "88pm860x-i2s",
.platform_name = "pxa-pcm-audio",
.codec_name = "88pm860x-codec",
.init = saarb_pm860x_init,
.ops = &saarb_i2s_ops,
},
};
static struct snd_soc_card snd_soc_card_saarb = {
.name = "Saarb",
.dai_link = saarb_dai,
.num_links = ARRAY_SIZE(saarb_dai),
};
static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
int ret;
snd_soc_dapm_new_controls(codec, saarb_dapm_widgets,
ARRAY_SIZE(saarb_dapm_widgets));
snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
/* connected pins */
snd_soc_dapm_enable_pin(codec, "Ext Speaker");
snd_soc_dapm_enable_pin(codec, "Ext Mic 1");
snd_soc_dapm_enable_pin(codec, "Ext Mic 3");
snd_soc_dapm_disable_pin(codec, "Headset Mic 2");
snd_soc_dapm_disable_pin(codec, "Headset Stereophone");
ret = snd_soc_dapm_sync(codec);
if (ret)
return ret;
/* Headset jack detection */
snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE
| SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2,
&hs_jack);
snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
hs_jack_pins);
snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE,
&mic_jack);
snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
mic_jack_pins);
/* headphone, microphone detection & headset short detection */
pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE,
SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2);
pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE);
return 0;
}
static int __init saarb_init(void)
{
int ret;
if (!machine_is_saarb())
return -ENODEV;
saarb_snd_device = platform_device_alloc("soc-audio", -1);
if (!saarb_snd_device)
return -ENOMEM;
platform_set_drvdata(saarb_snd_device, &snd_soc_card_saarb);
ret = platform_device_add(saarb_snd_device);
if (ret)
platform_device_put(saarb_snd_device);
return ret;
}
static void __exit saarb_exit(void)
{
platform_device_unregister(saarb_snd_device);
}
module_init(saarb_init);
module_exit(saarb_exit);
MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang@marvell.com>");
MODULE_DESCRIPTION("ALSA SoC 88PM860x Saarb");
MODULE_LICENSE("GPL");

200
sound/soc/pxa/tavorevb3.c Normal file
View File

@ -0,0 +1,200 @@
/*
* tavorevb3.c -- SoC audio for Tavor EVB3
*
* Copyright (C) 2010 Marvell International Ltd.
* Haojian Zhuang <haojian.zhuang@marvell.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/device.h>
#include <linux/clk.h>
#include <linux/i2c.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/jack.h>
#include <asm/mach-types.h>
#include "../codecs/88pm860x-codec.h"
#include "pxa-ssp.h"
static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd);
static struct platform_device *evb3_snd_device;
static struct snd_soc_jack hs_jack, mic_jack;
static struct snd_soc_jack_pin hs_jack_pins[] = {
{ .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, },
};
static struct snd_soc_jack_pin mic_jack_pins[] = {
{ .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, },
};
/* tavorevb3 machine dapm widgets */
static const struct snd_soc_dapm_widget evb3_dapm_widgets[] = {
SND_SOC_DAPM_HP("Headset Stereophone", NULL),
SND_SOC_DAPM_LINE("Lineout Out 1", NULL),
SND_SOC_DAPM_LINE("Lineout Out 2", NULL),
SND_SOC_DAPM_SPK("Ext Speaker", NULL),
SND_SOC_DAPM_MIC("Ext Mic 1", NULL),
SND_SOC_DAPM_MIC("Headset Mic 2", NULL),
SND_SOC_DAPM_MIC("Ext Mic 3", NULL),
};
/* tavorevb3 machine audio map */
static const struct snd_soc_dapm_route audio_map[] = {
{"Headset Stereophone", NULL, "HS1"},
{"Headset Stereophone", NULL, "HS2"},
{"Ext Speaker", NULL, "LSP"},
{"Ext Speaker", NULL, "LSN"},
{"Lineout Out 1", NULL, "LINEOUT1"},
{"Lineout Out 2", NULL, "LINEOUT2"},
{"MIC1P", NULL, "Mic1 Bias"},
{"MIC1N", NULL, "Mic1 Bias"},
{"Mic1 Bias", NULL, "Ext Mic 1"},
{"MIC2P", NULL, "Mic1 Bias"},
{"MIC2N", NULL, "Mic1 Bias"},
{"Mic1 Bias", NULL, "Headset Mic 2"},
{"MIC3P", NULL, "Mic3 Bias"},
{"MIC3N", NULL, "Mic3 Bias"},
{"Mic3 Bias", NULL, "Ext Mic 3"},
};
static int evb3_i2s_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->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
int width = snd_pcm_format_physical_width(params_format(params));
int ret;
ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0,
PM860X_CLK_DIR_OUT);
if (ret < 0)
return ret;
ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT);
if (ret < 0)
return 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 < 0)
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 < 0)
return ret;
ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width);
return ret;
}
static struct snd_soc_ops evb3_i2s_ops = {
.hw_params = evb3_i2s_hw_params,
};
static struct snd_soc_dai_link evb3_dai[] = {
{
.name = "88PM860x I2S",
.stream_name = "I2S Audio",
.cpu_dai_name = "pxa-ssp-dai.1",
.codec_dai_name = "88pm860x-i2s",
.platform_name = "pxa-pcm-audio",
.codec_name = "88pm860x-codec",
.init = evb3_pm860x_init,
.ops = &evb3_i2s_ops,
},
};
static struct snd_soc_card snd_soc_card_evb3 = {
.name = "Tavor EVB3",
.dai_link = evb3_dai,
.num_links = ARRAY_SIZE(evb3_dai),
};
static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
int ret;
snd_soc_dapm_new_controls(codec, evb3_dapm_widgets,
ARRAY_SIZE(evb3_dapm_widgets));
snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
/* connected pins */
snd_soc_dapm_enable_pin(codec, "Ext Speaker");
snd_soc_dapm_enable_pin(codec, "Ext Mic 1");
snd_soc_dapm_enable_pin(codec, "Ext Mic 3");
snd_soc_dapm_disable_pin(codec, "Headset Mic 2");
snd_soc_dapm_disable_pin(codec, "Headset Stereophone");
ret = snd_soc_dapm_sync(codec);
if (ret)
return ret;
/* Headset jack detection */
snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE
| SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2,
&hs_jack);
snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
hs_jack_pins);
snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE,
&mic_jack);
snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
mic_jack_pins);
/* headphone, microphone detection & headset short detection */
pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE,
SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2);
pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE);
return 0;
}
static int __init tavorevb3_init(void)
{
int ret;
if (!machine_is_tavorevb3())
return -ENODEV;
evb3_snd_device = platform_device_alloc("soc-audio", -1);
if (!evb3_snd_device)
return -ENOMEM;
platform_set_drvdata(evb3_snd_device, &snd_soc_card_evb3);
ret = platform_device_add(evb3_snd_device);
if (ret)
platform_device_put(evb3_snd_device);
return ret;
}
static void __exit tavorevb3_exit(void)
{
platform_device_unregister(evb3_snd_device);
}
module_init(tavorevb3_init);
module_exit(tavorevb3_exit);
MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang@marvell.com>");
MODULE_DESCRIPTION("ALSA SoC 88PM860x Tavor EVB3");
MODULE_LICENSE("GPL");

2
sound/soc/pxa/z2.c
View File

@ -189,7 +189,7 @@ static struct snd_soc_ops z2_ops = {
static struct snd_soc_dai_link z2_dai = {
.name = "wm8750",
.stream_name = "WM8750",
.cpu_dai_name = "pxa-i2s",
.cpu_dai_name = "pxa2xx-i2s",
.codec_dai_name = "wm8750-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm8750-codec.0-001a",

15
sound/soc/soc-core.c
View File

@ -2916,7 +2916,7 @@ int snd_soc_register_dais(struct device *dev,
struct snd_soc_dai *dai;
int i, ret = 0;
dev_dbg(dev, "dai register %s #%d\n", dev_name(dev), count);
dev_dbg(dev, "dai register %s #%Zu\n", dev_name(dev), count);
for (i = 0; i < count; i++) {
@ -3122,10 +3122,12 @@ int snd_soc_register_codec(struct device *dev,
fixup_codec_formats(&dai_drv[i].capture);
}
/* register DAIs */
ret = snd_soc_register_dais(dev, dai_drv, num_dai);
if (ret < 0)
/* register any DAIs */
if (num_dai) {
ret = snd_soc_register_dais(dev, dai_drv, num_dai);
if (ret < 0)
goto error;
}
mutex_lock(&client_mutex);
list_add(&codec->list, &codec_list);
@ -3164,8 +3166,9 @@ void snd_soc_unregister_codec(struct device *dev)
return;
found:
for (i = 0; i < codec->num_dai; i++)
snd_soc_unregister_dai(dev);
if (codec->num_dai)
for (i = 0; i < codec->num_dai; i++)
snd_soc_unregister_dai(dev);
mutex_lock(&client_mutex);
list_del(&codec->list);