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提交 ba252af8 编写于 作者: T Takashi Iwai
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Merge branch 'topic/asoc' into for-linus 

* topic/asoc: (135 commits)
  ASoC: Apostrophe patrol
  ASoC: codec tlv320aic23 fix bogus divide by 0 message
  ASoC: fix NULL pointer dereference in soc_suspend()
  ASoC: Fix build error in twl4030.c
  ASoC: SSM2602: assign last substream to the master when shutting down
  ASoC: Blackfin: document how anomaly 05000250 is handled
  ASoC: Blackfin: set the transfer size according the ac97_frame size
  ASoC: SSM2602: remove unsupported sample rates
  ASoC: TWL4030: Check the interface format for 4 channel mode
  ASoC: TWL4030: Use reg_cache in twl4030_init_chip
  ASoC: Initialise dev for the dummy S/PDIF DAI
  ASoC: Add dummy S/PDIF codec support
  ASoC: correct print specifiers for unsigneds
  ASoC: Modify mpc5200 AC97 driver to use V9 of spin_event_timeout()
  ASoC: Switch FSL SSI DAI over to symmetric_rates
  ASoC: Mark MPC5200 AC97 as BROKEN until PowerPC merge issues are resolved
  ASoC: Fabric bindings for STAC9766 on the Efika
  ASoC: Support for AC97 on Phytec pmc030 base board.
  ASoC: AC97 driver for mpc5200
  ASoC: Main rewite of the mpc5200 audio DMA code
  ...
上级 07a2039b 74b8f955
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Showing with 12466 addition and 1646 deletion
Documentation/sound/alsa/soc/dapm.txt
浏览文件 @ ba252af8
......@@ -62,6 +62,7 @@ Audio DAPM widgets fall into a number of types:-
o Mic - Mic (and optional Jack)
o Line - Line Input/Output (and optional Jack)
o Speaker - Speaker
o Supply - Power or clock supply widget used by other widgets.
o Pre - Special PRE widget (exec before all others)
o Post - Special POST widget (exec after all others)
......
MAINTAINERS
浏览文件 @ ba252af8
......@@ -4574,7 +4574,8 @@ F: drivers/pcmcia/pxa2xx*
F: drivers/spi/pxa2xx*
F: drivers/usb/gadget/pxa2*
F: include/sound/pxa2xx-lib.h
F: sound/soc/pxa/pxa2xx*
F: sound/arm/pxa*
F: sound/soc/pxa
PXA168 SUPPORT
P: Eric Miao
......@@ -5302,11 +5303,12 @@ P: Liam Girdwood
M: lrg@slimlogic.co.uk
P: Mark Brown
M: broonie@opensource.wolfsonmicro.com
T: git git://opensource.wolfsonmicro.com/linux-2.6-asoc
T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git
L: alsa-devel@alsa-project.org (subscribers-only)
W: http://alsa-project.org/main/index.php/ASoC
S: Supported
F: sound/soc/
F: include/sound/soc*
SPARC + UltraSPARC (sparc/sparc64)
P: David S. Miller
......
arch/powerpc/include/asm/mpc52xx_psc.h
浏览文件 @ ba252af8
......@@ -28,6 +28,10 @@
#define MPC52xx_PSC_MAXNUM 6
/* Programmable Serial Controller (PSC) status register bits */
#define MPC52xx_PSC_SR_UNEX_RX 0x0001
#define MPC52xx_PSC_SR_DATA_VAL 0x0002
#define MPC52xx_PSC_SR_DATA_OVR 0x0004
#define MPC52xx_PSC_SR_CMDSEND 0x0008
#define MPC52xx_PSC_SR_CDE 0x0080
#define MPC52xx_PSC_SR_RXRDY 0x0100
#define MPC52xx_PSC_SR_RXFULL 0x0200
......@@ -61,6 +65,12 @@
#define MPC52xx_PSC_RXTX_FIFO_EMPTY 0x0001
/* PSC interrupt status/mask bits */
#define MPC52xx_PSC_IMR_UNEX_RX_SLOT 0x0001
#define MPC52xx_PSC_IMR_DATA_VALID 0x0002
#define MPC52xx_PSC_IMR_DATA_OVR 0x0004
#define MPC52xx_PSC_IMR_CMD_SEND 0x0008
#define MPC52xx_PSC_IMR_ERROR 0x0040
#define MPC52xx_PSC_IMR_DEOF 0x0080
#define MPC52xx_PSC_IMR_TXRDY 0x0100
#define MPC52xx_PSC_IMR_RXRDY 0x0200
#define MPC52xx_PSC_IMR_DB 0x0400
......@@ -117,6 +127,7 @@
#define MPC52xx_PSC_SICR_SIM_FIR (0x6 << 24)
#define MPC52xx_PSC_SICR_SIM_CODEC_24 (0x7 << 24)
#define MPC52xx_PSC_SICR_SIM_CODEC_32 (0xf << 24)
#define MPC52xx_PSC_SICR_AWR (1 << 30)
#define MPC52xx_PSC_SICR_GENCLK (1 << 23)
#define MPC52xx_PSC_SICR_I2S (1 << 22)
#define MPC52xx_PSC_SICR_CLKPOL (1 << 21)
......
include/sound/soc-dai.h
浏览文件 @ ba252af8
......@@ -44,24 +44,6 @@ struct snd_pcm_substream;
#define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */
#define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated */
/*
* DAI Left/Right Clocks.
*
* Specifies whether the DAI can support different samples for similtanious
* playback and capture. This usually requires a seperate physical frame
* clock for playback and capture.
*/
#define SND_SOC_DAIFMT_SYNC (0 << 5) /* Tx FRM = Rx FRM */
#define SND_SOC_DAIFMT_ASYNC (1 << 5) /* Tx FRM ~ Rx FRM */
/*
* TDM
*
* Time Division Multiplexing. Allows PCM data to be multplexed with other
* data on the DAI.
*/
#define SND_SOC_DAIFMT_TDM (1 << 6)
/*
* DAI hardware signal inversions.
*
......@@ -96,6 +78,10 @@ struct snd_pcm_substream;
#define SND_SOC_CLOCK_IN 0
#define SND_SOC_CLOCK_OUT 1
#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S16_LE |\
SNDRV_PCM_FMTBIT_S32_LE |\
SNDRV_PCM_FMTBIT_S32_BE)
struct snd_soc_dai_ops;
struct snd_soc_dai;
struct snd_ac97_bus_ops;
......@@ -208,6 +194,7 @@ struct snd_soc_dai {
/* DAI capabilities */
struct snd_soc_pcm_stream capture;
struct snd_soc_pcm_stream playback;
unsigned int symmetric_rates:1;
/* DAI runtime info */
struct snd_pcm_runtime *runtime;
......@@ -219,11 +206,8 @@ struct snd_soc_dai {
/* DAI private data */
void *private_data;
/* parent codec/platform */
union {
struct snd_soc_codec *codec;
struct snd_soc_platform *platform;
};
/* parent platform */
struct snd_soc_platform *platform;
struct list_head list;
};
......
include/sound/soc-dapm.h
浏览文件 @ ba252af8
......@@ -140,16 +140,30 @@
#define SND_SOC_DAPM_DAC(wname, stname, wreg, wshift, winvert) \
{ .id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \
.shift = wshift, .invert = winvert}
#define SND_SOC_DAPM_DAC_E(wname, stname, wreg, wshift, winvert, \
wevent, wflags) \
{ .id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \
.shift = wshift, .invert = winvert, \
.event = wevent, .event_flags = wflags}
#define SND_SOC_DAPM_ADC(wname, stname, wreg, wshift, winvert) \
{ .id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \
.shift = wshift, .invert = winvert}
#define SND_SOC_DAPM_ADC_E(wname, stname, wreg, wshift, winvert, \
wevent, wflags) \
{ .id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \
.shift = wshift, .invert = winvert, \
.event = wevent, .event_flags = wflags}
/* generic register modifier widget */
/* generic widgets */
#define SND_SOC_DAPM_REG(wid, wname, wreg, wshift, wmask, won_val, woff_val) \
{ .id = wid, .name = wname, .kcontrols = NULL, .num_kcontrols = 0, \
.reg = -((wreg) + 1), .shift = wshift, .mask = wmask, \
.on_val = won_val, .off_val = woff_val, .event = dapm_reg_event, \
.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD}
#define SND_SOC_DAPM_SUPPLY(wname, wreg, wshift, winvert, wevent, wflags) \
{ .id = snd_soc_dapm_supply, .name = wname, .reg = wreg, \
.shift = wshift, .invert = winvert, .event = wevent, \
.event_flags = wflags}
/* dapm kcontrol types */
#define SOC_DAPM_SINGLE(xname, reg, shift, max, invert) \
......@@ -265,8 +279,6 @@ int snd_soc_dapm_add_routes(struct snd_soc_codec *codec,
/* dapm events */
int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream,
int event);
int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev,
enum snd_soc_bias_level level);
/* dapm sys fs - used by the core */
int snd_soc_dapm_sys_add(struct device *dev);
......@@ -298,6 +310,7 @@ enum snd_soc_dapm_type {
snd_soc_dapm_vmid, /* codec bias/vmid - to minimise pops */
snd_soc_dapm_pre, /* machine specific pre widget - exec first */
snd_soc_dapm_post, /* machine specific post widget - exec last */
snd_soc_dapm_supply, /* power/clock supply */
};
/*
......@@ -357,6 +370,8 @@ struct snd_soc_dapm_widget {
unsigned char suspend:1; /* was active before suspend */
unsigned char pmdown:1; /* waiting for timeout */
int (*power_check)(struct snd_soc_dapm_widget *w);
/* external events */
unsigned short event_flags; /* flags to specify event types */
int (*event)(struct snd_soc_dapm_widget*, struct snd_kcontrol *, int);
......@@ -368,6 +383,9 @@ struct snd_soc_dapm_widget {
/* widget input and outputs */
struct list_head sources;
struct list_head sinks;
/* used during DAPM updates */
struct list_head power_list;
};
#endif
include/sound/soc.h
浏览文件 @ ba252af8
......@@ -118,6 +118,14 @@
.info = snd_soc_info_volsw, \
.get = xhandler_get, .put = xhandler_put, \
.private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) }
#define SOC_DOUBLE_EXT(xname, xreg, shift_left, shift_right, xmax, xinvert,\
xhandler_get, xhandler_put) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
.info = snd_soc_info_volsw, \
.get = xhandler_get, .put = xhandler_put, \
.private_value = (unsigned long)&(struct soc_mixer_control) \
{.reg = xreg, .shift = shift_left, .rshift = shift_right, \
.max = xmax, .invert = xinvert} }
#define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmax, xinvert,\
xhandler_get, xhandler_put, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
......@@ -206,10 +214,6 @@ void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count,
struct snd_soc_jack_gpio *gpios);
#endif
/* codec IO */
#define snd_soc_read(codec, reg) codec->read(codec, reg)
#define snd_soc_write(codec, reg, value) codec->write(codec, reg, value)
/* codec register bit access */
int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg,
unsigned short mask, unsigned short value);
......@@ -331,6 +335,7 @@ struct snd_soc_codec {
struct module *owner;
struct mutex mutex;
struct device *dev;
struct snd_soc_device *socdev;
struct list_head list;
......@@ -364,6 +369,8 @@ struct snd_soc_codec {
enum snd_soc_bias_level bias_level;
enum snd_soc_bias_level suspend_bias_level;
struct delayed_work delayed_work;
struct list_head up_list;
struct list_head down_list;
/* codec DAI's */
struct snd_soc_dai *dai;
......@@ -417,6 +424,12 @@ struct snd_soc_dai_link {
/* codec/machine specific init - e.g. add machine controls */
int (*init)(struct snd_soc_codec *codec);
/* Symmetry requirements */
unsigned int symmetric_rates:1;
/* Symmetry data - only valid if symmetry is being enforced */
unsigned int rate;
/* DAI pcm */
struct snd_pcm *pcm;
};
......@@ -490,6 +503,19 @@ struct soc_enum {
void *dapm;
};
/* codec IO */
static inline unsigned int snd_soc_read(struct snd_soc_codec *codec,
unsigned int reg)
{
return codec->read(codec, reg);
}
static inline unsigned int snd_soc_write(struct snd_soc_codec *codec,
unsigned int reg, unsigned int val)
{
return codec->write(codec, reg, val);
}
#include <sound/soc-dai.h>
#endif
include/sound/wm9081.h 0 → 100644
浏览文件 @ ba252af8
/*
* linux/sound/wm9081.h -- Platform data for WM9081
*
* Copyright 2009 Wolfson Microelectronics. PLC.
*
* 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 __LINUX_SND_WM_9081_H
#define __LINUX_SND_WM_9081_H
struct wm9081_retune_mobile_setting {
const char *name;
unsigned int rate;
u16 config[20];
};
struct wm9081_retune_mobile_config {
struct wm9081_retune_mobile_setting *configs;
int num_configs;
};
#endif
sound/aoa/fabrics/layout.c
浏览文件 @ ba252af8
......@@ -1037,7 +1037,7 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev)
}
ldev->selfptr_headphone.ptr = ldev;
ldev->selfptr_lineout.ptr = ldev;
sdev->ofdev.dev.driver_data = ldev;
dev_set_drvdata(&sdev->ofdev.dev, ldev);
list_add(&ldev->list, &layouts_list);
layouts_list_items++;
......@@ -1081,7 +1081,7 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev)
static int aoa_fabric_layout_remove(struct soundbus_dev *sdev)
{
struct layout_dev *ldev = sdev->ofdev.dev.driver_data;
struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev);
int i;
for (i=0; i<MAX_CODECS_PER_BUS; i++) {
......@@ -1114,7 +1114,7 @@ static int aoa_fabric_layout_remove(struct soundbus_dev *sdev)
#ifdef CONFIG_PM
static int aoa_fabric_layout_suspend(struct soundbus_dev *sdev, pm_message_t state)
{
struct layout_dev *ldev = sdev->ofdev.dev.driver_data;
struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev);
if (ldev->gpio.methods && ldev->gpio.methods->all_amps_off)
ldev->gpio.methods->all_amps_off(&ldev->gpio);
......@@ -1124,7 +1124,7 @@ static int aoa_fabric_layout_suspend(struct soundbus_dev *sdev, pm_message_t sta
static int aoa_fabric_layout_resume(struct soundbus_dev *sdev)
{
struct layout_dev *ldev = sdev->ofdev.dev.driver_data;
struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev);
if (ldev->gpio.methods && ldev->gpio.methods->all_amps_off)
ldev->gpio.methods->all_amps_restore(&ldev->gpio);
......
sound/aoa/soundbus/i2sbus/core.c
浏览文件 @ ba252af8
......@@ -358,14 +358,14 @@ static int i2sbus_probe(struct macio_dev* dev, const struct of_device_id *match)
return -ENODEV;
}
dev->ofdev.dev.driver_data = control;
dev_set_drvdata(&dev->ofdev.dev, control);
return 0;
}
static int i2sbus_remove(struct macio_dev* dev)
{
struct i2sbus_control *control = dev->ofdev.dev.driver_data;
struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev);
struct i2sbus_dev *i2sdev, *tmp;
list_for_each_entry_safe(i2sdev, tmp, &control->list, item)
......@@ -377,7 +377,7 @@ static int i2sbus_remove(struct macio_dev* dev)
#ifdef CONFIG_PM
static int i2sbus_suspend(struct macio_dev* dev, pm_message_t state)
{
struct i2sbus_control *control = dev->ofdev.dev.driver_data;
struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev);
struct codec_info_item *cii;
struct i2sbus_dev* i2sdev;
int err, ret = 0;
......@@ -407,7 +407,7 @@ static int i2sbus_suspend(struct macio_dev* dev, pm_message_t state)
static int i2sbus_resume(struct macio_dev* dev)
{
struct i2sbus_control *control = dev->ofdev.dev.driver_data;
struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev);
struct codec_info_item *cii;
struct i2sbus_dev* i2sdev;
int err, ret = 0;
......
sound/soc/Kconfig
浏览文件 @ ba252af8
......@@ -32,7 +32,9 @@ source "sound/soc/fsl/Kconfig"
source "sound/soc/omap/Kconfig"
source "sound/soc/pxa/Kconfig"
source "sound/soc/s3c24xx/Kconfig"
source "sound/soc/s6000/Kconfig"
source "sound/soc/sh/Kconfig"
source "sound/soc/txx9/Kconfig"
# Supported codecs
source "sound/soc/codecs/Kconfig"
......
sound/soc/Makefile
浏览文件 @ ba252af8
......@@ -10,4 +10,6 @@ obj-$(CONFIG_SND_SOC) += fsl/
obj-$(CONFIG_SND_SOC) += omap/
obj-$(CONFIG_SND_SOC) += pxa/
obj-$(CONFIG_SND_SOC) += s3c24xx/
obj-$(CONFIG_SND_SOC) += s6000/
obj-$(CONFIG_SND_SOC) += sh/
obj-$(CONFIG_SND_SOC) += txx9/
sound/soc/atmel/Kconfig
浏览文件 @ ba252af8
......@@ -41,3 +41,11 @@ config SND_AT32_SOC_PLAYPAQ_SLAVE
and FRAME signals on the PlayPaq. Unless you want to play
with the AT32 as the SSC master, you probably want to say N here,
as this will give you better sound quality.
config SND_AT91_SOC_AFEB9260
tristate "SoC Audio support for AFEB9260 board"
depends on ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC
select SND_ATMEL_SOC_SSC
select SND_SOC_TLV320AIC23
help
Say Y here to support sound on AFEB9260 board.
sound/soc/atmel/Makefile
浏览文件 @ ba252af8
......@@ -13,3 +13,4 @@ snd-soc-playpaq-objs := playpaq_wm8510.o
obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o
obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o
sound/soc/atmel/playpaq_wm8510.c
浏览文件 @ ba252af8
......@@ -117,7 +117,7 @@ static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock(
* Find actual rate, compare to requested rate
*/
actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1));
pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n",
pr_debug("playpaq_wm8510: Request rate = %u, actual rate = %u\n",
rate, actual_rate);
......
sound/soc/atmel/snd-soc-afeb9260.c 0 → 100644
浏览文件 @ ba252af8
/*
* afeb9260.c -- SoC audio for AFEB9260
*
* Copyright (C) 2009 Sergey Lapin <slapin@ossfans.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.
*
* 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/module.h>
#include <linux/moduleparam.h>
#include <linux/kernel.h>
#include <linux/clk.h>
#include <linux/platform_device.h>
#include <linux/atmel-ssc.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <asm/mach-types.h>
#include <mach/hardware.h>
#include <linux/gpio.h>
#include "../codecs/tlv320aic23.h"
#include "atmel-pcm.h"
#include "atmel_ssc_dai.h"
#define CODEC_CLOCK 12000000
static int afeb9260_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;
/* Set codec DAI configuration */
err = snd_soc_dai_set_fmt(codec_dai,
SND_SOC_DAIFMT_I2S|
SND_SOC_DAIFMT_NB_IF |
SND_SOC_DAIFMT_CBM_CFM);
if (err < 0) {
printk(KERN_ERR "can't set codec DAI configuration\n");
return err;
}
/* Set cpu DAI configuration */
err = snd_soc_dai_set_fmt(cpu_dai,
SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_IF |
SND_SOC_DAIFMT_CBM_CFM);
if (err < 0) {
printk(KERN_ERR "can't set cpu DAI configuration\n");
return err;
}
/* Set the codec system clock for DAC and ADC */
err =
snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN);
if (err < 0) {
printk(KERN_ERR "can't set codec system clock\n");
return err;
}
return err;
}
static struct snd_soc_ops afeb9260_ops = {
.hw_params = afeb9260_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 afeb9260_tlv320aic23_init(struct snd_soc_codec *codec)
{
/* Add afeb9260 specific widgets */
snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
ARRAY_SIZE(tlv320aic23_dapm_widgets));
/* Set up afeb9260 specific audio path audio_map */
snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
snd_soc_dapm_enable_pin(codec, "Headphone Jack");
snd_soc_dapm_enable_pin(codec, "Line In");
snd_soc_dapm_enable_pin(codec, "Mic Jack");
snd_soc_dapm_sync(codec);
return 0;
}
/* Digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link afeb9260_dai = {
.name = "TLV320AIC23",
.stream_name = "AIC23",
.cpu_dai = &atmel_ssc_dai[0],
.codec_dai = &tlv320aic23_dai,
.init = afeb9260_tlv320aic23_init,
.ops = &afeb9260_ops,
};
/* Audio machine driver */
static struct snd_soc_card snd_soc_machine_afeb9260 = {
.name = "AFEB9260",
.platform = &atmel_soc_platform,
.dai_link = &afeb9260_dai,
.num_links = 1,
};
/* Audio subsystem */
static struct snd_soc_device afeb9260_snd_devdata = {
.card = &snd_soc_machine_afeb9260,
.codec_dev = &soc_codec_dev_tlv320aic23,
};
static struct platform_device *afeb9260_snd_device;
static int __init afeb9260_soc_init(void)
{
int err;
struct device *dev;
struct atmel_ssc_info *ssc_p = afeb9260_dai.cpu_dai->private_data;
struct ssc_device *ssc = NULL;
if (!(machine_is_afeb9260()))
return -ENODEV;
ssc = ssc_request(0);
if (IS_ERR(ssc)) {
printk(KERN_ERR "ASoC: Failed to request SSC 0\n");
err = PTR_ERR(ssc);
ssc = NULL;
goto err_ssc;
}
ssc_p->ssc = ssc;
afeb9260_snd_device = platform_device_alloc("soc-audio", -1);
if (!afeb9260_snd_device) {
printk(KERN_ERR "ASoC: Platform device allocation failed\n");
return -ENOMEM;
}
platform_set_drvdata(afeb9260_snd_device, &afeb9260_snd_devdata);
afeb9260_snd_devdata.dev = &afeb9260_snd_device->dev;
err = platform_device_add(afeb9260_snd_device);
if (err)
goto err1;
dev = &afeb9260_snd_device->dev;
return 0;
err1:
platform_device_del(afeb9260_snd_device);
platform_device_put(afeb9260_snd_device);
err_ssc:
return err;
}
static void __exit afeb9260_soc_exit(void)
{
platform_device_unregister(afeb9260_snd_device);
}
module_init(afeb9260_soc_init);
module_exit(afeb9260_soc_exit);
MODULE_AUTHOR("Sergey Lapin <slapin@ossfans.org>");
MODULE_DESCRIPTION("ALSA SoC for AFEB9260");
MODULE_LICENSE("GPL");
sound/soc/blackfin/bf5xx-ac97.c
浏览文件 @ ba252af8
......@@ -31,6 +31,15 @@
#include "bf5xx-sport.h"
#include "bf5xx-ac97.h"
/* Anomaly notes:
* 05000250 - AD1980 is running in TDM mode and RFS/TFS are generated by SPORT
* contrtoller. But, RFSDIV and TFSDIV are always set to 16*16-1,
* while the max AC97 data size is 13*16. The DIV is always larger
* than data size. AD73311 and ad2602 are not running in TDM mode.
* AD1836 and AD73322 depend on external RFS/TFS only. So, this
* anomaly does not affect blackfin sound drivers.
*/
static int *cmd_count;
static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM;
......
sound/soc/blackfin/bf5xx-sport.c
浏览文件 @ ba252af8
......@@ -190,7 +190,7 @@ static inline int sport_hook_rx_dummy(struct sport_device *sport)
desc = get_dma_next_desc_ptr(sport->dma_rx_chan);
/* Copy the descriptor which will be damaged to backup */
temp_desc = *desc;
desc->x_count = 0xa;
desc->x_count = sport->dummy_count / 2;
desc->y_count = 0;
desc->next_desc_addr = sport->dummy_rx_desc;
local_irq_restore(flags);
......@@ -309,7 +309,7 @@ static inline int sport_hook_tx_dummy(struct sport_device *sport)
desc = get_dma_next_desc_ptr(sport->dma_tx_chan);
/* Store the descriptor which will be damaged */
temp_desc = *desc;
desc->x_count = 0xa;
desc->x_count = sport->dummy_count / 2;
desc->y_count = 0;
desc->next_desc_addr = sport->dummy_tx_desc;
local_irq_restore(flags);
......
sound/soc/codecs/Kconfig
浏览文件 @ ba252af8
......@@ -18,7 +18,9 @@ config SND_SOC_ALL_CODECS
select SND_SOC_AK4535 if I2C
select SND_SOC_CS4270 if I2C
select SND_SOC_PCM3008
select SND_SOC_SPDIF
select SND_SOC_SSM2602 if I2C
select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
select SND_SOC_TLV320AIC23 if I2C
select SND_SOC_TLV320AIC26 if SPI_MASTER
select SND_SOC_TLV320AIC3X if I2C
......@@ -35,8 +37,12 @@ config SND_SOC_ALL_CODECS
select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8900 if I2C
select SND_SOC_WM8903 if I2C
select SND_SOC_WM8940 if I2C
select SND_SOC_WM8960 if I2C
select SND_SOC_WM8971 if I2C
select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8990 if I2C
select SND_SOC_WM9081 if I2C
select SND_SOC_WM9705 if SND_SOC_AC97_BUS
select SND_SOC_WM9712 if SND_SOC_AC97_BUS
select SND_SOC_WM9713 if SND_SOC_AC97_BUS
......@@ -86,9 +92,15 @@ config SND_SOC_L3
config SND_SOC_PCM3008
tristate
config SND_SOC_SPDIF
tristate
config SND_SOC_SSM2602
tristate
config SND_SOC_STAC9766
tristate
config SND_SOC_TLV320AIC23
tristate
......@@ -138,12 +150,24 @@ config SND_SOC_WM8900
config SND_SOC_WM8903
tristate
config SND_SOC_WM8940
tristate
config SND_SOC_WM8960
tristate
config SND_SOC_WM8971
tristate
config SND_SOC_WM8988
tristate
config SND_SOC_WM8990
tristate
config SND_SOC_WM9081
tristate
config SND_SOC_WM9705
tristate
......
sound/soc/codecs/Makefile
浏览文件 @ ba252af8
......@@ -6,7 +6,9 @@ snd-soc-ak4535-objs := ak4535.o
snd-soc-cs4270-objs := cs4270.o
snd-soc-l3-objs := l3.o
snd-soc-pcm3008-objs := pcm3008.o
snd-soc-spdif-objs := spdif_transciever.o
snd-soc-ssm2602-objs := ssm2602.o
snd-soc-stac9766-objs := stac9766.o
snd-soc-tlv320aic23-objs := tlv320aic23.o
snd-soc-tlv320aic26-objs := tlv320aic26.o
snd-soc-tlv320aic3x-objs := tlv320aic3x.o
......@@ -23,8 +25,12 @@ snd-soc-wm8750-objs := wm8750.o
snd-soc-wm8753-objs := wm8753.o
snd-soc-wm8900-objs := wm8900.o
snd-soc-wm8903-objs := wm8903.o
snd-soc-wm8940-objs := wm8940.o
snd-soc-wm8960-objs := wm8960.o
snd-soc-wm8971-objs := wm8971.o
snd-soc-wm8988-objs := wm8988.o
snd-soc-wm8990-objs := wm8990.o
snd-soc-wm9081-objs := wm9081.o
snd-soc-wm9705-objs := wm9705.o
snd-soc-wm9712-objs := wm9712.o
snd-soc-wm9713-objs := wm9713.o
......@@ -37,7 +43,9 @@ obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o
obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o
obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o
obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o
obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o
obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o
obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o
obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o
......@@ -55,7 +63,11 @@ obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o
obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o
obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o
obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o
obj-$(CONFIG_SND_SOC_WM8940) += snd-soc-wm8940.o
obj-$(CONFIG_SND_SOC_WM8960) += snd-soc-wm8960.o
obj-$(CONFIG_SND_SOC_WM8988) += snd-soc-wm8988.o
obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o
obj-$(CONFIG_SND_SOC_WM9081) += snd-soc-wm9081.o
obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o
obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o
obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o
sound/soc/codecs/ac97.c
浏览文件 @ ba252af8
......@@ -53,13 +53,13 @@ struct snd_soc_dai ac97_dai = {
.channels_min = 1,
.channels_max = 2,
.rates = STD_AC97_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
.formats = SND_SOC_STD_AC97_FMTS,},
.capture = {
.stream_name = "AC97 Capture",
.channels_min = 1,
.channels_max = 2,
.rates = STD_AC97_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
.formats = SND_SOC_STD_AC97_FMTS,},
.ops = &ac97_dai_ops,
};
EXPORT_SYMBOL_GPL(ac97_dai);
......
sound/soc/codecs/ad1980.c
浏览文件 @ ba252af8
......@@ -137,13 +137,13 @@ struct snd_soc_dai ad1980_dai = {
.channels_min = 2,
.channels_max = 6,
.rates = SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE, },
.formats = SND_SOC_STD_AC97_FMTS, },
.capture = {
.stream_name = "Capture",
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE, },
.formats = SND_SOC_STD_AC97_FMTS, },
};
EXPORT_SYMBOL_GPL(ad1980_dai);
......
sound/soc/codecs/cs4270.c
浏览文件 @ ba252af8
......@@ -18,7 +18,7 @@
* - The machine driver's 'startup' function must call
* cs4270_set_dai_sysclk() with the value of MCLK.
* - Only I2S and left-justified modes are supported
* - Power management is not supported
* - Power management is supported
*/
#include <linux/module.h>
......@@ -27,6 +27,7 @@
#include <sound/soc.h>
#include <sound/initval.h>
#include <linux/i2c.h>
#include <linux/delay.h>
#include "cs4270.h"
......@@ -56,6 +57,7 @@
#define CS4270_FIRSTREG 0x01
#define CS4270_LASTREG 0x08
#define CS4270_NUMREGS (CS4270_LASTREG - CS4270_FIRSTREG + 1)
#define CS4270_I2C_INCR 0x80
/* Bit masks for the CS4270 registers */
#define CS4270_CHIPID_ID 0xF0
......@@ -64,6 +66,8 @@
#define CS4270_PWRCTL_PDN_ADC 0x20
#define CS4270_PWRCTL_PDN_DAC 0x02
#define CS4270_PWRCTL_PDN 0x01
#define CS4270_PWRCTL_PDN_ALL \
(CS4270_PWRCTL_PDN_ADC | CS4270_PWRCTL_PDN_DAC | CS4270_PWRCTL_PDN)
#define CS4270_MODE_SPEED_MASK 0x30
#define CS4270_MODE_1X 0x00
#define CS4270_MODE_2X 0x10
......@@ -109,6 +113,7 @@ struct cs4270_private {
unsigned int mclk; /* Input frequency of the MCLK pin */
unsigned int mode; /* The mode (I2S or left-justified) */
unsigned int slave_mode;
unsigned int manual_mute;
};
/**
......@@ -295,7 +300,7 @@ static int cs4270_fill_cache(struct snd_soc_codec *codec)
s32 length;
length = i2c_smbus_read_i2c_block_data(i2c_client,
CS4270_FIRSTREG | 0x80, CS4270_NUMREGS, cache);
CS4270_FIRSTREG | CS4270_I2C_INCR, CS4270_NUMREGS, cache);
if (length != CS4270_NUMREGS) {
dev_err(codec->dev, "i2c read failure, addr=0x%x\n",
......@@ -453,7 +458,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
}
/**
* cs4270_mute - enable/disable the CS4270 external mute
* cs4270_dai_mute - enable/disable the CS4270 external mute
* @dai: the SOC DAI
* @mute: 0 = disable mute, 1 = enable mute
*
......@@ -462,21 +467,52 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
* board does not have the MUTEA or MUTEB pins connected to such circuitry,
* then this function will do nothing.
*/
static int cs4270_mute(struct snd_soc_dai *dai, int mute)
static int cs4270_dai_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
struct cs4270_private *cs4270 = codec->private_data;
int reg6;
reg6 = snd_soc_read(codec, CS4270_MUTE);
if (mute)
reg6 |= CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B;
else
else {
reg6 &= ~(CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B);
reg6 |= cs4270->manual_mute;
}
return snd_soc_write(codec, CS4270_MUTE, reg6);
}
/**
* cs4270_soc_put_mute - put callback for the 'Master Playback switch'
* alsa control.
* @kcontrol: mixer control
* @ucontrol: control element information
*
* This function basically passes the arguments on to the generic
* snd_soc_put_volsw() function and saves the mute information in
* our private data structure. This is because we want to prevent
* cs4270_dai_mute() neglecting the user's decision to manually
* mute the codec's output.
*
* Returns 0 for success.
*/
static int cs4270_soc_put_mute(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct cs4270_private *cs4270 = codec->private_data;
int left = !ucontrol->value.integer.value[0];
int right = !ucontrol->value.integer.value[1];
cs4270->manual_mute = (left ? CS4270_MUTE_DAC_A : 0) |
(right ? CS4270_MUTE_DAC_B : 0);
return snd_soc_put_volsw(kcontrol, ucontrol);
}
/* A list of non-DAPM controls that the CS4270 supports */
static const struct snd_kcontrol_new cs4270_snd_controls[] = {
SOC_DOUBLE_R("Master Playback Volume",
......@@ -486,7 +522,9 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = {
SOC_SINGLE("Zero Cross Switch", CS4270_TRANS, 5, 1, 0),
SOC_SINGLE("Popguard Switch", CS4270_MODE, 0, 1, 1),
SOC_SINGLE("Auto-Mute Switch", CS4270_MUTE, 5, 1, 0),
SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 0)
SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 1),
SOC_DOUBLE_EXT("Master Playback Switch", CS4270_MUTE, 0, 1, 1, 1,
snd_soc_get_volsw, cs4270_soc_put_mute),
};
/*
......@@ -506,7 +544,7 @@ static struct snd_soc_dai_ops cs4270_dai_ops = {
.hw_params = cs4270_hw_params,
.set_sysclk = cs4270_set_dai_sysclk,
.set_fmt = cs4270_set_dai_fmt,
.digital_mute = cs4270_mute,
.digital_mute = cs4270_dai_mute,
};
struct snd_soc_dai cs4270_dai = {
......@@ -753,6 +791,57 @@ static struct i2c_device_id cs4270_id[] = {
};
MODULE_DEVICE_TABLE(i2c, cs4270_id);
#ifdef CONFIG_PM
/* This suspend/resume implementation can handle both - a simple standby
* where the codec remains powered, and a full suspend, where the voltage
* domain the codec is connected to is teared down and/or any other hardware
* reset condition is asserted.
*
* The codec's own power saving features are enabled in the suspend callback,
* and all registers are written back to the hardware when resuming.
*/
static int cs4270_i2c_suspend(struct i2c_client *client, pm_message_t mesg)
{
struct cs4270_private *cs4270 = i2c_get_clientdata(client);
struct snd_soc_codec *codec = &cs4270->codec;
int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL;
return snd_soc_write(codec, CS4270_PWRCTL, reg);
}
static int cs4270_i2c_resume(struct i2c_client *client)
{
struct cs4270_private *cs4270 = i2c_get_clientdata(client);
struct snd_soc_codec *codec = &cs4270->codec;
int reg;
/* In case the device was put to hard reset during sleep, we need to
* wait 500ns here before any I2C communication. */
ndelay(500);
/* first restore the entire register cache ... */
for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) {
u8 val = snd_soc_read(codec, reg);
if (i2c_smbus_write_byte_data(client, reg, val)) {
dev_err(codec->dev, "i2c write failed\n");
return -EIO;
}
}
/* ... then disable the power-down bits */
reg = snd_soc_read(codec, CS4270_PWRCTL);
reg &= ~CS4270_PWRCTL_PDN_ALL;
return snd_soc_write(codec, CS4270_PWRCTL, reg);
}
#else
#define cs4270_i2c_suspend NULL
#define cs4270_i2c_resume NULL
#endif /* CONFIG_PM */
/*
* cs4270_i2c_driver - I2C device identification
*
......@@ -767,6 +856,8 @@ static struct i2c_driver cs4270_i2c_driver = {
.id_table = cs4270_id,
.probe = cs4270_i2c_probe,
.remove = cs4270_i2c_remove,
.suspend = cs4270_i2c_suspend,
.resume = cs4270_i2c_resume,
};
/*
......
sound/soc/codecs/spdif_transciever.c 0 → 100644
浏览文件 @ ba252af8
/*
* ALSA SoC SPDIF DIT driver
*
* This driver is used by controllers which can operate in DIT (SPDI/F) where
* no codec is needed. This file provides stub codec that can be used
* in these configurations. TI DaVinci Audio controller uses this driver.
*
* Author: Steve Chen, <schen@mvista.com>
* Copyright: (C) 2009 MontaVista Software, Inc., <source@mvista.com>
* Copyright: (C) 2009 Texas Instruments, India
*
* 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 <sound/soc.h>
#include <sound/pcm.h>
#include "spdif_transciever.h"
#define STUB_RATES SNDRV_PCM_RATE_8000_96000
#define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE
struct snd_soc_dai dit_stub_dai = {
.name = "DIT",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 384,
.rates = STUB_RATES,
.formats = STUB_FORMATS,
},
};
static int spdif_dit_probe(struct platform_device *pdev)
{
dit_stub_dai.dev = &pdev->dev;
return snd_soc_register_dai(&dit_stub_dai);
}
static int spdif_dit_remove(struct platform_device *pdev)
{
snd_soc_unregister_dai(&dit_stub_dai);
return 0;
}
static struct platform_driver spdif_dit_driver = {
.probe = spdif_dit_probe,
.remove = spdif_dit_remove,
.driver = {
.name = "spdif-dit",
.owner = THIS_MODULE,
},
};
static int __init dit_modinit(void)
{
return platform_driver_register(&spdif_dit_driver);
}
static void __exit dit_exit(void)
{
platform_driver_unregister(&spdif_dit_driver);
}
module_init(dit_modinit);
module_exit(dit_exit);
sound/soc/codecs/spdif_transciever.h 0 → 100644
浏览文件 @ ba252af8
/*
* ALSA SoC DIT/DIR driver header
*
* Author: Steve Chen, <schen@mvista.com>
* Copyright: (C) 2008 MontaVista Software, Inc., <source@mvista.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 CODEC_STUBS_H
#define CODEC_STUBS_H
extern struct snd_soc_dai dit_stub_dai;
#endif /* CODEC_STUBS_H */
sound/soc/codecs/ssm2602.c
浏览文件 @ ba252af8
......@@ -336,15 +336,17 @@ static int ssm2602_startup(struct snd_pcm_substream *substream,
master_runtime->sample_bits,
master_runtime->rate);
snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_RATE,
master_runtime->rate,
master_runtime->rate);
snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
master_runtime->sample_bits,
master_runtime->sample_bits);
if (master_runtime->rate != 0)
snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_RATE,
master_runtime->rate,
master_runtime->rate);
if (master_runtime->sample_bits != 0)
snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
master_runtime->sample_bits,
master_runtime->sample_bits);
ssm2602->slave_substream = substream;
} else
......@@ -372,6 +374,11 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->card->codec;
struct ssm2602_priv *ssm2602 = codec->private_data;
if (ssm2602->master_substream == substream)
ssm2602->master_substream = ssm2602->slave_substream;
ssm2602->slave_substream = NULL;
/* deactivate */
if (!codec->active)
ssm2602_write(codec, SSM2602_ACTIVE, 0);
......@@ -497,11 +504,9 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
return 0;
}
#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
SNDRV_PCM_RATE_96000)
#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_32000 |\
SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
#define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
......
sound/soc/codecs/stac9766.c 0 → 100644
浏览文件 @ ba252af8
/*
* stac9766.c -- ALSA SoC STAC9766 codec support
*
* Copyright 2009 Jon Smirl, Digispeaker
* Author: Jon Smirl <jonsmirl@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; either version 2 of the License, or (at your
* option) any later version.
*
* Features:-
*
* o Support for AC97 Codec, S/PDIF
*/
#include <linux/init.h>
#include <linux/module.h>
#include <linux/device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/ac97_codec.h>
#include <sound/initval.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/tlv.h>
#include <sound/soc-of-simple.h>
#include "stac9766.h"
#define STAC9766_VERSION "0.10"
/*
* STAC9766 register cache
*/
static const u16 stac9766_reg[] = {
0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */
0x0000, 0x0000, 0x8008, 0x8008, /* e */
0x8808, 0x8808, 0x8808, 0x8808, /* 16 */
0x8808, 0x0000, 0x8000, 0x0000, /* 1e */
0x0000, 0x0000, 0x0000, 0x000f, /* 26 */
0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */
0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */
0x0000, 0x2000, 0x0000, 0x0100, /* 3e */
0x0000, 0x0000, 0x0080, 0x0000, /* 46 */
0x0000, 0x0000, 0x0003, 0xffff, /* 4e */
0x0000, 0x0000, 0x0000, 0x0000, /* 56 */
0x4000, 0x0000, 0x0000, 0x0000, /* 5e */
0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */
0x0000, 0x0000, 0x0000, 0x0000, /* 6e */
0x0000, 0x0000, 0x0000, 0x0006, /* 76 */
0x0000, 0x0000, 0x0000, 0x0000, /* 7e */
};
static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX",
"Line", "Stereo Mix", "Mono Mix", "Phone"};
static const char *stac9766_mono_mux[] = {"Mix", "Mic"};
static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"};
static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"};
static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"};
static const char *stac9766_record_all_mux[] = {"All analog",
"Analog plus DAC"};
static const char *stac9766_boost1[] = {"0dB", "10dB"};
static const char *stac9766_boost2[] = {"0dB", "20dB"};
static const char *stac9766_stereo_mic[] = {"Off", "On"};
static const struct soc_enum stac9766_record_enum =
SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux);
static const struct soc_enum stac9766_mono_enum =
SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux);
static const struct soc_enum stac9766_mic_enum =
SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux);
static const struct soc_enum stac9766_SPDIF_enum =
SOC_ENUM_SINGLE(AC97_STAC_DA_CONTROL, 1, 2, stac9766_SPDIF_mux);
static const struct soc_enum stac9766_popbypass_enum =
SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux);
static const struct soc_enum stac9766_record_all_enum =
SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2,
stac9766_record_all_mux);
static const struct soc_enum stac9766_boost1_enum =
SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */
static const struct soc_enum stac9766_boost2_enum =
SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 2, 2, stac9766_boost2); /* 0/20dB */
static const struct soc_enum stac9766_stereo_mic_enum =
SOC_ENUM_SINGLE(AC97_STAC_STEREO_MIC, 2, 1, stac9766_stereo_mic);
static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0);
static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250);
static const DECLARE_TLV_DB_LINEAR(beep_tlv, -4500, 0);
static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200);
static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = {
SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv),
SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1),
SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1,
master_tlv),
SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1),
SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1,
master_tlv),
SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1),
SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv),
SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1),
SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv),
SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1),
SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1),
SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv),
SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1),
SOC_ENUM("Mic Boost1", stac9766_boost1_enum),
SOC_ENUM("Mic Boost2", stac9766_boost2_enum),
SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv),
SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1),
SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum),
SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv),
SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1),
SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv),
SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1),
SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv),
SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1),
SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv),
SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1),
SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv),
SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1),
SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0),
SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1),
SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0),
SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum),
SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum),
SOC_ENUM("Record All Mux", stac9766_record_all_enum),
SOC_ENUM("Record Mux", stac9766_record_enum),
SOC_ENUM("Mono Mux", stac9766_mono_enum),
SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum),
};
static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int val)
{
u16 *cache = codec->reg_cache;
if (reg > AC97_STAC_PAGE0) {
stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
soc_ac97_ops.write(codec->ac97, reg, val);
stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
return 0;
}
if (reg / 2 > ARRAY_SIZE(stac9766_reg))
return -EIO;
soc_ac97_ops.write(codec->ac97, reg, val);
cache[reg / 2] = val;
return 0;
}
static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec,
unsigned int reg)
{
u16 val = 0, *cache = codec->reg_cache;
if (reg > AC97_STAC_PAGE0) {
stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
val = soc_ac97_ops.read(codec->ac97, reg - AC97_STAC_PAGE0);
stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
return val;
}
if (reg / 2 > ARRAY_SIZE(stac9766_reg))
return -EIO;
if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||
reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 ||
reg == AC97_VENDOR_ID2) {
val = soc_ac97_ops.read(codec->ac97, reg);
return val;
}
return cache[reg / 2];
}
static int ac97_analog_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct snd_pcm_runtime *runtime = substream->runtime;
unsigned short reg, vra;
vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
vra |= 0x1; /* enable variable rate audio */
stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
reg = AC97_PCM_FRONT_DAC_RATE;
else
reg = AC97_PCM_LR_ADC_RATE;
return stac9766_ac97_write(codec, reg, runtime->rate);
}
static int ac97_digital_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct snd_pcm_runtime *runtime = substream->runtime;
unsigned short reg, vra;
stac9766_ac97_write(codec, AC97_SPDIF, 0x2002);
vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
vra |= 0x5; /* Enable VRA and SPDIF out */
stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
reg = AC97_PCM_FRONT_DAC_RATE;
return stac9766_ac97_write(codec, reg, runtime->rate);
}
static int ac97_digital_trigger(struct snd_pcm_substream *substream,
int cmd, struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
unsigned short vra;
switch (cmd) {
case SNDRV_PCM_TRIGGER_STOP:
vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
vra &= !0x04;
stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
break;
}
return 0;
}
static int stac9766_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
switch (level) {
case SND_SOC_BIAS_ON: /* full On */
case SND_SOC_BIAS_PREPARE: /* partial On */
case SND_SOC_BIAS_STANDBY: /* Off, with power */
stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000);
break;
case SND_SOC_BIAS_OFF: /* Off, without power */
/* disable everything including AC link */
stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff);
break;
}
codec->bias_level = level;
return 0;
}
static int stac9766_reset(struct snd_soc_codec *codec, int try_warm)
{
if (try_warm && soc_ac97_ops.warm_reset) {
soc_ac97_ops.warm_reset(codec->ac97);
if (stac9766_ac97_read(codec, 0) == stac9766_reg[0])
return 1;
}
soc_ac97_ops.reset(codec->ac97);
if (soc_ac97_ops.warm_reset)
soc_ac97_ops.warm_reset(codec->ac97);
if (stac9766_ac97_read(codec, 0) != stac9766_reg[0])
return -EIO;
return 0;
}
static int stac9766_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;
stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static int stac9766_codec_resume(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->card->codec;
u16 id, reset;
reset = 0;
/* give the codec an AC97 warm reset to start the link */
reset:
if (reset > 5) {
printk(KERN_ERR "stac9766 failed to resume");
return -EIO;
}
codec->ac97->bus->ops->warm_reset(codec->ac97);
id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2);
if (id != 0x4c13) {
stac9766_reset(codec, 0);
reset++;
goto reset;
}
stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
stac9766_set_bias_level(codec, SND_SOC_BIAS_ON);
return 0;
}
static struct snd_soc_dai_ops stac9766_dai_ops_analog = {
.prepare = ac97_analog_prepare,
};
static struct snd_soc_dai_ops stac9766_dai_ops_digital = {
.prepare = ac97_digital_prepare,
.trigger = ac97_digital_trigger,
};
struct snd_soc_dai stac9766_dai[] = {
{
.name = "stac9766 analog",
.id = 0,
.ac97_control = 1,
/* stream cababilities */
.playback = {
.stream_name = "stac9766 analog",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = SND_SOC_STD_AC97_FMTS,
},
.capture = {
.stream_name = "stac9766 analog",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = SND_SOC_STD_AC97_FMTS,
},
/* alsa ops */
.ops = &stac9766_dai_ops_analog,
},
{
.name = "stac9766 IEC958",
.id = 1,
.ac97_control = 1,
/* stream cababilities */
.playback = {
.stream_name = "stac9766 IEC958",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_32000 | \
SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE,
},
/* alsa ops */
.ops = &stac9766_dai_ops_digital,
}
};
EXPORT_SYMBOL_GPL(stac9766_dai);
static int stac9766_codec_probe(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec;
int ret = 0;
printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION);
socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
if (socdev->card->codec == NULL)
return -ENOMEM;
codec = socdev->card->codec;
mutex_init(&codec->mutex);
codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg),
GFP_KERNEL);
if (codec->reg_cache == NULL) {
ret = -ENOMEM;
goto cache_err;
}
codec->reg_cache_size = sizeof(stac9766_reg);
codec->reg_cache_step = 2;
codec->name = "STAC9766";
codec->owner = THIS_MODULE;
codec->dai = stac9766_dai;
codec->num_dai = ARRAY_SIZE(stac9766_dai);
codec->write = stac9766_ac97_write;
codec->read = stac9766_ac97_read;
codec->set_bias_level = stac9766_set_bias_level;
INIT_LIST_HEAD(&codec->dapm_widgets);
INIT_LIST_HEAD(&codec->dapm_paths);
ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
if (ret < 0)
goto codec_err;
/* register pcms */
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0)
goto pcm_err;
/* do a cold reset for the controller and then try
* a warm reset followed by an optional cold reset for codec */
stac9766_reset(codec, 0);
ret = stac9766_reset(codec, 1);
if (ret < 0) {
printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n");
goto reset_err;
}
stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
snd_soc_add_controls(codec, stac9766_snd_ac97_controls,
ARRAY_SIZE(stac9766_snd_ac97_controls));
ret = snd_soc_init_card(socdev);
if (ret < 0)
goto reset_err;
return 0;
reset_err:
snd_soc_free_pcms(socdev);
pcm_err:
snd_soc_free_ac97_codec(codec);
codec_err:
kfree(codec->private_data);
cache_err:
kfree(socdev->card->codec);
socdev->card->codec = NULL;
return ret;
}
static int stac9766_codec_remove(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->card->codec;
if (codec == NULL)
return 0;
snd_soc_free_pcms(socdev);
snd_soc_free_ac97_codec(codec);
kfree(codec->reg_cache);
kfree(codec);
return 0;
}
struct snd_soc_codec_device soc_codec_dev_stac9766 = {
.probe = stac9766_codec_probe,
.remove = stac9766_codec_remove,
.suspend = stac9766_codec_suspend,
.resume = stac9766_codec_resume,
};
EXPORT_SYMBOL_GPL(soc_codec_dev_stac9766);
MODULE_DESCRIPTION("ASoC stac9766 driver");
MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
MODULE_LICENSE("GPL");
sound/soc/codecs/stac9766.h 0 → 100644
浏览文件 @ ba252af8
/*
* stac9766.h -- STAC9766 Soc Audio driver
*/
#ifndef _STAC9766_H
#define _STAC9766_H
#define AC97_STAC_PAGE0 0x1000
#define AC97_STAC_DA_CONTROL (AC97_STAC_PAGE0 | 0x6A)
#define AC97_STAC_ANALOG_SPECIAL (AC97_STAC_PAGE0 | 0x6E)
#define AC97_STAC_STEREO_MIC 0x78
/* STAC9766 DAI ID's */
#define STAC9766_DAI_AC97_ANALOG 0
#define STAC9766_DAI_AC97_DIGITAL 1
extern struct snd_soc_dai stac9766_dai[];
extern struct snd_soc_codec_device soc_codec_dev_stac9766;
#endif
sound/soc/codecs/tlv320aic23.c
浏览文件 @ ba252af8
......@@ -86,7 +86,7 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg,
*/
if ((reg < 0 || reg > 9) && (reg != 15)) {
printk(KERN_WARNING "%s Invalid register R%d\n", __func__, reg);
printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg);
return -1;
}
......@@ -98,7 +98,7 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg,
if (codec->hw_write(codec->control_data, data, 2) == 2)
return 0;
printk(KERN_ERR "%s cannot write %03x to register R%d\n", __func__,
printk(KERN_ERR "%s cannot write %03x to register R%u\n", __func__,
value, reg);
return -EIO;
......@@ -273,14 +273,14 @@ static const unsigned short sr_valid_mask[] = {
* Every divisor is a factor of 11*12
*/
#define SR_MULT (11*12)
#define A(x) (x) ? (SR_MULT/x) : 0
#define A(x) (SR_MULT/x)
static const unsigned char sr_adc_mult_table[] = {
A(2), A(2), A(12), A(12), A(0), A(0), A(3), A(1),
A(2), A(2), A(11), A(11), A(0), A(0), A(0), A(1)
A(2), A(2), A(12), A(12), 0, 0, A(3), A(1),
A(2), A(2), A(11), A(11), 0, 0, 0, A(1)
};
static const unsigned char sr_dac_mult_table[] = {
A(2), A(12), A(2), A(12), A(0), A(0), A(3), A(1),
A(2), A(11), A(2), A(11), A(0), A(0), A(0), A(1)
A(2), A(12), A(2), A(12), 0, 0, A(3), A(1),
A(2), A(11), A(2), A(11), 0, 0, 0, A(1)
};
static unsigned get_score(int adc, int adc_l, int adc_h, int need_adc,
......@@ -523,6 +523,8 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai,
case SND_SOC_DAIFMT_I2S:
iface_reg |= TLV320AIC23_FOR_I2S;
break;
case SND_SOC_DAIFMT_DSP_A:
iface_reg |= TLV320AIC23_LRP_ON;
case SND_SOC_DAIFMT_DSP_B:
iface_reg |= TLV320AIC23_FOR_DSP;
break;
......
sound/soc/codecs/twl4030.c
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/codecs/twl4030.h
浏览文件 @ ba252af8
......@@ -92,8 +92,9 @@
#define TWL4030_REG_VIBRA_PWM_SET 0x47
#define TWL4030_REG_ANAMIC_GAIN 0x48
#define TWL4030_REG_MISC_SET_2 0x49
#define TWL4030_REG_SW_SHADOW 0x4A
#define TWL4030_CACHEREGNUM (TWL4030_REG_MISC_SET_2 + 1)
#define TWL4030_CACHEREGNUM (TWL4030_REG_SW_SHADOW + 1)
/* Bitfield Definitions */
......@@ -110,9 +111,22 @@
#define TWL4030_APLL_RATE_44100 0x90
#define TWL4030_APLL_RATE_48000 0xA0
#define TWL4030_APLL_RATE_96000 0xE0
#define TWL4030_SEL_16K 0x04
#define TWL4030_SEL_16K 0x08
#define TWL4030_CODECPDZ 0x02
#define TWL4030_OPT_MODE 0x01
#define TWL4030_OPTION_1 (1 << 0)
#define TWL4030_OPTION_2 (0 << 0)
/* TWL4030_OPTION (0x02) Fields */
#define TWL4030_ATXL1_EN (1 << 0)
#define TWL4030_ATXR1_EN (1 << 1)
#define TWL4030_ATXL2_VTXL_EN (1 << 2)
#define TWL4030_ATXR2_VTXR_EN (1 << 3)
#define TWL4030_ARXL1_VRX_EN (1 << 4)
#define TWL4030_ARXR1_EN (1 << 5)
#define TWL4030_ARXL2_EN (1 << 6)
#define TWL4030_ARXR2_EN (1 << 7)
/* TWL4030_REG_MICBIAS_CTL (0x04) Fields */
......@@ -171,6 +185,17 @@
#define TWL4030_CLK256FS_EN 0x02
#define TWL4030_AIF_EN 0x01
/* VOICE_IF (0x0F) Fields */
#define TWL4030_VIF_SLAVE_EN 0x80
#define TWL4030_VIF_DIN_EN 0x40
#define TWL4030_VIF_DOUT_EN 0x20
#define TWL4030_VIF_SWAP 0x10
#define TWL4030_VIF_FORMAT 0x08
#define TWL4030_VIF_TRI_EN 0x04
#define TWL4030_VIF_SUB_EN 0x02
#define TWL4030_VIF_EN 0x01
/* EAR_CTL (0x21) */
#define TWL4030_EAR_GAIN 0x30
......@@ -236,7 +261,19 @@
#define TWL4030_SMOOTH_ANAVOL_EN 0x02
#define TWL4030_DIGMIC_LR_SWAP_EN 0x01
extern struct snd_soc_dai twl4030_dai;
/* TWL4030_REG_SW_SHADOW (0x4A) Fields */
#define TWL4030_HFL_EN 0x01
#define TWL4030_HFR_EN 0x02
#define TWL4030_DAI_HIFI 0
#define TWL4030_DAI_VOICE 1
extern struct snd_soc_dai twl4030_dai[2];
extern struct snd_soc_codec_device soc_codec_dev_twl4030;
struct twl4030_setup_data {
unsigned int ramp_delay_value;
unsigned int sysclk;
};
#endif /* End of __TWL4030_AUDIO_H__ */
sound/soc/codecs/uda134x.c
浏览文件 @ ba252af8
......@@ -101,7 +101,7 @@ static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg,
pr_debug("%s reg: %02X, value:%02X\n", __func__, reg, value);
if (reg >= UDA134X_REGS_NUM) {
printk(KERN_ERR "%s unkown register: reg: %d",
printk(KERN_ERR "%s unkown register: reg: %u",
__func__, reg);
return -EINVAL;
}
......@@ -296,7 +296,7 @@ static int uda134x_set_dai_sysclk(struct snd_soc_dai *codec_dai,
struct snd_soc_codec *codec = codec_dai->codec;
struct uda134x_priv *uda134x = codec->private_data;
pr_debug("%s clk_id: %d, freq: %d, dir: %d\n", __func__,
pr_debug("%s clk_id: %d, freq: %u, dir: %d\n", __func__,
clk_id, freq, dir);
/* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable
......
sound/soc/codecs/wm8350.c
浏览文件 @ ba252af8
......@@ -1108,7 +1108,7 @@ static int wm8350_set_fll(struct snd_soc_dai *codec_dai,
if (ret < 0)
return ret;
dev_dbg(wm8350->dev,
"FLL in %d FLL out %d N 0x%x K 0x%x div %d ratio %d",
"FLL in %u FLL out %u N 0x%x K 0x%x div %d ratio %d",
freq_in, freq_out, fll_div.n, fll_div.k, fll_div.div,
fll_div.ratio);
......
sound/soc/codecs/wm8350.h
浏览文件 @ ba252af8
......@@ -13,6 +13,7 @@
#define _WM8350_H
#include <sound/soc.h>
#include <linux/mfd/wm8350/audio.h>
extern struct snd_soc_dai wm8350_dai;
extern struct snd_soc_codec_device soc_codec_dev_wm8350;
......
sound/soc/codecs/wm8400.c
浏览文件 @ ba252af8
......@@ -954,7 +954,7 @@ static int fll_factors(struct wm8400_priv *wm8400, struct fll_factors *factors,
factors->outdiv *= 2;
if (factors->outdiv > 32) {
dev_err(wm8400->wm8400->dev,
"Unsupported FLL output frequency %dHz\n",
"Unsupported FLL output frequency %uHz\n",
Fout);
return -EINVAL;
}
......@@ -1003,7 +1003,7 @@ static int fll_factors(struct wm8400_priv *wm8400, struct fll_factors *factors,
factors->k = K / 10;
dev_dbg(wm8400->wm8400->dev,
"FLL: Fref=%d Fout=%d N=%x K=%x, FRATIO=%x OUTDIV=%x\n",
"FLL: Fref=%u Fout=%u N=%x K=%x, FRATIO=%x OUTDIV=%x\n",
Fref, Fout,
factors->n, factors->k, factors->fratio, factors->outdiv);
......@@ -1473,8 +1473,8 @@ static int wm8400_codec_probe(struct platform_device *dev)
codec = &priv->codec;
codec->private_data = priv;
codec->control_data = dev->dev.driver_data;
priv->wm8400 = dev->dev.driver_data;
codec->control_data = dev_get_drvdata(&dev->dev);
priv->wm8400 = dev_get_drvdata(&dev->dev);
ret = regulator_bulk_get(priv->wm8400->dev,
ARRAY_SIZE(power), &power[0]);
......
sound/soc/codecs/wm8510.c
浏览文件 @ ba252af8
......@@ -298,7 +298,7 @@ static void pll_factors(unsigned int target, unsigned int source)
if ((Ndiv < 6) || (Ndiv > 12))
printk(KERN_WARNING
"WM8510 N value %d outwith recommended range!d\n",
"WM8510 N value %u outwith recommended range!d\n",
Ndiv);
pll_div.n = Ndiv;
......
sound/soc/codecs/wm8580.c
浏览文件 @ ba252af8
......@@ -415,7 +415,7 @@ static int pll_factors(struct _pll_div *pll_div, unsigned int target,
unsigned int K, Ndiv, Nmod;
int i;
pr_debug("wm8580: PLL %dHz->%dHz\n", source, target);
pr_debug("wm8580: PLL %uHz->%uHz\n", source, target);
/* Scale the output frequency up; the PLL should run in the
* region of 90-100MHz.
......@@ -447,7 +447,7 @@ static int pll_factors(struct _pll_div *pll_div, unsigned int target,
if ((Ndiv < 5) || (Ndiv > 13)) {
printk(KERN_ERR
"WM8580 N=%d outside supported range\n", Ndiv);
"WM8580 N=%u outside supported range\n", Ndiv);
return -EINVAL;
}
......
sound/soc/codecs/wm8731.c
浏览文件 @ ba252af8
......@@ -666,14 +666,14 @@ static int __devinit wm8731_spi_probe(struct spi_device *spi)
codec->hw_write = (hw_write_t)wm8731_spi_write;
codec->dev = &spi->dev;
spi->dev.driver_data = wm8731;
dev_set_drvdata(&spi->dev, wm8731);
return wm8731_register(wm8731);
}
static int __devexit wm8731_spi_remove(struct spi_device *spi)
{
struct wm8731_priv *wm8731 = spi->dev.driver_data;
struct wm8731_priv *wm8731 = dev_get_drvdata(&spi->dev);
wm8731_unregister(wm8731);
......
sound/soc/codecs/wm8753.c
浏览文件 @ ba252af8
......@@ -703,7 +703,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target,
if ((Ndiv < 6) || (Ndiv > 12))
printk(KERN_WARNING
"wm8753: unsupported N = %d\n", Ndiv);
"wm8753: unsupported N = %u\n", Ndiv);
pll_div->n = Ndiv;
Nmod = target % source;
......@@ -1822,14 +1822,14 @@ static int __devinit wm8753_spi_probe(struct spi_device *spi)
codec->hw_write = (hw_write_t)wm8753_spi_write;
codec->dev = &spi->dev;
spi->dev.driver_data = wm8753;
dev_set_drvdata(&spi->dev, wm8753);
return wm8753_register(wm8753);
}
static int __devexit wm8753_spi_remove(struct spi_device *spi)
{
struct wm8753_priv *wm8753 = spi->dev.driver_data;
struct wm8753_priv *wm8753 = dev_get_drvdata(&spi->dev);
wm8753_unregister(wm8753);
return 0;
}
......
sound/soc/codecs/wm8900.c
浏览文件 @ ba252af8
......@@ -778,11 +778,11 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
}
if (target > 100000000)
printk(KERN_WARNING "wm8900: FLL rate %d out of range, Fref=%d"
" Fout=%d\n", target, Fref, Fout);
printk(KERN_WARNING "wm8900: FLL rate %u out of range, Fref=%u"
" Fout=%u\n", target, Fref, Fout);
if (div > 32) {
printk(KERN_ERR "wm8900: Invalid FLL division rate %u, "
"Fref=%d, Fout=%d, target=%d\n",
"Fref=%u, Fout=%u, target=%u\n",
div, Fref, Fout, target);
return -EINVAL;
}
......
sound/soc/codecs/wm8903.c
浏览文件 @ ba252af8
......@@ -217,7 +217,6 @@ struct wm8903_priv {
int sysclk;
/* Reference counts */
int charge_pump_users;
int class_w_users;
int playback_active;
int capture_active;
......@@ -373,6 +372,15 @@ static void wm8903_reset(struct snd_soc_codec *codec)
#define WM8903_OUTPUT_INT 0x2
#define WM8903_OUTPUT_IN 0x1
static int wm8903_cp_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
WARN_ON(event != SND_SOC_DAPM_POST_PMU);
mdelay(4);
return 0;
}
/*
* Event for headphone and line out amplifier power changes. Special
* power up/down sequences are required in order to maximise pop/click
......@@ -382,19 +390,20 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
struct wm8903_priv *wm8903 = codec->private_data;
struct i2c_client *i2c = codec->control_data;
u16 val;
u16 reg;
u16 dcs_reg;
u16 dcs_bit;
int shift;
u16 cp_reg = wm8903_read(codec, WM8903_CHARGE_PUMP_0);
switch (w->reg) {
case WM8903_POWER_MANAGEMENT_2:
reg = WM8903_ANALOGUE_HP_0;
dcs_bit = 0 + w->shift;
break;
case WM8903_POWER_MANAGEMENT_3:
reg = WM8903_ANALOGUE_LINEOUT_0;
dcs_bit = 2 + w->shift;
break;
default:
BUG();
......@@ -419,18 +428,6 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w,
/* Short the output */
val &= ~(WM8903_OUTPUT_SHORT << shift);
wm8903_write(codec, reg, val);
wm8903->charge_pump_users++;
dev_dbg(&i2c->dev, "Charge pump use count now %d\n",
wm8903->charge_pump_users);
if (wm8903->charge_pump_users == 1) {
dev_dbg(&i2c->dev, "Enabling charge pump\n");
wm8903_write(codec, WM8903_CHARGE_PUMP_0,
cp_reg | WM8903_CP_ENA);
mdelay(4);
}
}
if (event & SND_SOC_DAPM_POST_PMU) {
......@@ -446,6 +443,11 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w,
val |= (WM8903_OUTPUT_OUT << shift);
wm8903_write(codec, reg, val);
/* Enable the DC servo */
dcs_reg = wm8903_read(codec, WM8903_DC_SERVO_0);
dcs_reg |= dcs_bit;
wm8903_write(codec, WM8903_DC_SERVO_0, dcs_reg);
/* Remove the short */
val |= (WM8903_OUTPUT_SHORT << shift);
wm8903_write(codec, reg, val);
......@@ -458,25 +460,17 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w,
val &= ~(WM8903_OUTPUT_SHORT << shift);
wm8903_write(codec, reg, val);
/* Disable the DC servo */
dcs_reg = wm8903_read(codec, WM8903_DC_SERVO_0);
dcs_reg &= ~dcs_bit;
wm8903_write(codec, WM8903_DC_SERVO_0, dcs_reg);
/* Then disable the intermediate and output stages */
val &= ~((WM8903_OUTPUT_OUT | WM8903_OUTPUT_INT |
WM8903_OUTPUT_IN) << shift);
wm8903_write(codec, reg, val);
}
if (event & SND_SOC_DAPM_POST_PMD) {
wm8903->charge_pump_users--;
dev_dbg(&i2c->dev, "Charge pump use count now %d\n",
wm8903->charge_pump_users);
if (wm8903->charge_pump_users == 0) {
dev_dbg(&i2c->dev, "Disabling charge pump\n");
wm8903_write(codec, WM8903_CHARGE_PUMP_0,
cp_reg & ~WM8903_CP_ENA);
}
}
return 0;
}
......@@ -539,6 +533,7 @@ static int wm8903_class_w_put(struct snd_kcontrol *kcontrol,
/* ALSA can only do steps of .01dB */
static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1);
static const DECLARE_TLV_DB_SCALE(digital_sidetone_tlv, -3600, 300, 0);
static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0);
static const DECLARE_TLV_DB_SCALE(drc_tlv_thresh, 0, 75, 0);
......@@ -657,6 +652,16 @@ static const struct soc_enum rinput_inv_enum =
SOC_ENUM_SINGLE(WM8903_ANALOGUE_RIGHT_INPUT_1, 4, 3, rinput_mux_text);
static const char *sidetone_text[] = {
"None", "Left", "Right"
};
static const struct soc_enum lsidetone_enum =
SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 2, 3, sidetone_text);
static const struct soc_enum rsidetone_enum =
SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 0, 3, sidetone_text);
static const struct snd_kcontrol_new wm8903_snd_controls[] = {
/* Input PGAs - No TLV since the scale depends on PGA mode */
......@@ -700,6 +705,9 @@ SOC_DOUBLE_R_TLV("Digital Capture Volume", WM8903_ADC_DIGITAL_VOLUME_LEFT,
SOC_ENUM("ADC Companding Mode", adc_companding),
SOC_SINGLE("ADC Companding Switch", WM8903_AUDIO_INTERFACE_0, 3, 1, 0),
SOC_DOUBLE_TLV("Digital Sidetone Volume", WM8903_DAC_DIGITAL_0, 4, 8,
12, 0, digital_sidetone_tlv),
/* DAC */
SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8903_DAC_DIGITAL_VOLUME_LEFT,
WM8903_DAC_DIGITAL_VOLUME_RIGHT, 1, 120, 0, digital_tlv),
......@@ -762,6 +770,12 @@ static const struct snd_kcontrol_new rinput_mux =
static const struct snd_kcontrol_new rinput_inv_mux =
SOC_DAPM_ENUM("Right Inverting Input Mux", rinput_inv_enum);
static const struct snd_kcontrol_new lsidetone_mux =
SOC_DAPM_ENUM("DACL Sidetone Mux", lsidetone_enum);
static const struct snd_kcontrol_new rsidetone_mux =
SOC_DAPM_ENUM("DACR Sidetone Mux", rsidetone_enum);
static const struct snd_kcontrol_new left_output_mixer[] = {
SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_LEFT_MIX_0, 3, 1, 0),
SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_LEFT_MIX_0, 2, 1, 0),
......@@ -828,6 +842,9 @@ SND_SOC_DAPM_PGA("Right Input PGA", WM8903_POWER_MANAGEMENT_0, 0, 0, NULL, 0),
SND_SOC_DAPM_ADC("ADCL", "Left HiFi Capture", WM8903_POWER_MANAGEMENT_6, 1, 0),
SND_SOC_DAPM_ADC("ADCR", "Right HiFi Capture", WM8903_POWER_MANAGEMENT_6, 0, 0),
SND_SOC_DAPM_MUX("DACL Sidetone", SND_SOC_NOPM, 0, 0, &lsidetone_mux),
SND_SOC_DAPM_MUX("DACR Sidetone", SND_SOC_NOPM, 0, 0, &rsidetone_mux),
SND_SOC_DAPM_DAC("DACL", "Left Playback", WM8903_POWER_MANAGEMENT_6, 3, 0),
SND_SOC_DAPM_DAC("DACR", "Right Playback", WM8903_POWER_MANAGEMENT_6, 2, 0),
......@@ -844,26 +861,29 @@ SND_SOC_DAPM_MIXER("Right Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 0, 0,
SND_SOC_DAPM_PGA_E("Left Headphone Output PGA", WM8903_POWER_MANAGEMENT_2,
1, 0, NULL, 0, wm8903_output_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_PGA_E("Right Headphone Output PGA", WM8903_POWER_MANAGEMENT_2,
0, 0, NULL, 0, wm8903_output_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_PGA_E("Left Line Output PGA", WM8903_POWER_MANAGEMENT_3, 1, 0,
NULL, 0, wm8903_output_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_PGA_E("Right Line Output PGA", WM8903_POWER_MANAGEMENT_3, 0, 0,
NULL, 0, wm8903_output_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_PGA("Left Speaker PGA", WM8903_POWER_MANAGEMENT_5, 1, 0,
NULL, 0),
SND_SOC_DAPM_PGA("Right Speaker PGA", WM8903_POWER_MANAGEMENT_5, 0, 0,
NULL, 0),
SND_SOC_DAPM_SUPPLY("Charge Pump", WM8903_CHARGE_PUMP_0, 0, 0,
wm8903_cp_event, SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8903_CLOCK_RATES_2, 1, 0, NULL, 0),
};
static const struct snd_soc_dapm_route intercon[] = {
......@@ -909,7 +929,19 @@ static const struct snd_soc_dapm_route intercon[] = {
{ "Right Input PGA", NULL, "Right Input Mode Mux" },
{ "ADCL", NULL, "Left Input PGA" },
{ "ADCL", NULL, "CLK_DSP" },
{ "ADCR", NULL, "Right Input PGA" },
{ "ADCR", NULL, "CLK_DSP" },
{ "DACL Sidetone", "Left", "ADCL" },
{ "DACL Sidetone", "Right", "ADCR" },
{ "DACR Sidetone", "Left", "ADCL" },
{ "DACR Sidetone", "Right", "ADCR" },
{ "DACL", NULL, "DACL Sidetone" },
{ "DACL", NULL, "CLK_DSP" },
{ "DACR", NULL, "DACR Sidetone" },
{ "DACR", NULL, "CLK_DSP" },
{ "Left Output Mixer", "Left Bypass Switch", "Left Input PGA" },
{ "Left Output Mixer", "Right Bypass Switch", "Right Input PGA" },
......@@ -951,6 +983,11 @@ static const struct snd_soc_dapm_route intercon[] = {
{ "ROP", NULL, "Right Speaker PGA" },
{ "RON", NULL, "Right Speaker PGA" },
{ "Left Headphone Output PGA", NULL, "Charge Pump" },
{ "Right Headphone Output PGA", NULL, "Charge Pump" },
{ "Left Line Output PGA", NULL, "Charge Pump" },
{ "Right Line Output PGA", NULL, "Charge Pump" },
};
static int wm8903_add_widgets(struct snd_soc_codec *codec)
......@@ -985,6 +1022,11 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec,
wm8903_write(codec, WM8903_CLOCK_RATES_2,
WM8903_CLK_SYS_ENA);
/* Change DC servo dither level in startup sequence */
wm8903_write(codec, WM8903_WRITE_SEQUENCER_0, 0x11);
wm8903_write(codec, WM8903_WRITE_SEQUENCER_1, 0x1257);
wm8903_write(codec, WM8903_WRITE_SEQUENCER_2, 0x2);
wm8903_run_sequence(codec, 0);
wm8903_sync_reg_cache(codec, codec->reg_cache);
......@@ -1277,14 +1319,8 @@ static int wm8903_startup(struct snd_pcm_substream *substream,
if (wm8903->master_substream) {
master_runtime = wm8903->master_substream->runtime;
dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n",
master_runtime->sample_bits,
master_runtime->rate);
snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_RATE,
master_runtime->rate,
master_runtime->rate);
dev_dbg(&i2c->dev, "Constraining to %d bits\n",
master_runtime->sample_bits);
snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
......@@ -1523,6 +1559,7 @@ struct snd_soc_dai wm8903_dai = {
.formats = WM8903_FORMATS,
},
.ops = &wm8903_dai_ops,
.symmetric_rates = 1,
};
EXPORT_SYMBOL_GPL(wm8903_dai);
......
sound/soc/codecs/wm8940.c 0 → 100644
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/codecs/wm8940.h 0 → 100644
浏览文件 @ ba252af8
/*
* wm8940.h -- WM8940 Soc Audio driver
*
* 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 _WM8940_H
#define _WM8940_H
struct wm8940_setup_data {
/* Vref to analogue output resistance */
#define WM8940_VROI_1K 0
#define WM8940_VROI_30K 1
unsigned int vroi:1;
};
extern struct snd_soc_dai wm8940_dai;
extern struct snd_soc_codec_device soc_codec_dev_wm8940;
/* WM8940 register space */
#define WM8940_SOFTRESET 0x00
#define WM8940_POWER1 0x01
#define WM8940_POWER2 0x02
#define WM8940_POWER3 0x03
#define WM8940_IFACE 0x04
#define WM8940_COMPANDINGCTL 0x05
#define WM8940_CLOCK 0x06
#define WM8940_ADDCNTRL 0x07
#define WM8940_GPIO 0x08
#define WM8940_CTLINT 0x09
#define WM8940_DAC 0x0A
#define WM8940_DACVOL 0x0B
#define WM8940_ADC 0x0E
#define WM8940_ADCVOL 0x0F
#define WM8940_NOTCH1 0x10
#define WM8940_NOTCH2 0x11
#define WM8940_NOTCH3 0x12
#define WM8940_NOTCH4 0x13
#define WM8940_NOTCH5 0x14
#define WM8940_NOTCH6 0x15
#define WM8940_NOTCH7 0x16
#define WM8940_NOTCH8 0x17
#define WM8940_DACLIM1 0x18
#define WM8940_DACLIM2 0x19
#define WM8940_ALC1 0x20
#define WM8940_ALC2 0x21
#define WM8940_ALC3 0x22
#define WM8940_NOISEGATE 0x23
#define WM8940_PLLN 0x24
#define WM8940_PLLK1 0x25
#define WM8940_PLLK2 0x26
#define WM8940_PLLK3 0x27
#define WM8940_ALC4 0x2A
#define WM8940_INPUTCTL 0x2C
#define WM8940_PGAGAIN 0x2D
#define WM8940_ADCBOOST 0x2F
#define WM8940_OUTPUTCTL 0x31
#define WM8940_SPKMIX 0x32
#define WM8940_SPKVOL 0x36
#define WM8940_MONOMIX 0x38
#define WM8940_CACHEREGNUM 0x57
/* Clock divider Id's */
#define WM8940_BCLKDIV 0
#define WM8940_MCLKDIV 1
#define WM8940_OPCLKDIV 2
/* MCLK clock dividers */
#define WM8940_MCLKDIV_1 0
#define WM8940_MCLKDIV_1_5 1
#define WM8940_MCLKDIV_2 2
#define WM8940_MCLKDIV_3 3
#define WM8940_MCLKDIV_4 4
#define WM8940_MCLKDIV_6 5
#define WM8940_MCLKDIV_8 6
#define WM8940_MCLKDIV_12 7
/* BCLK clock dividers */
#define WM8940_BCLKDIV_1 0
#define WM8940_BCLKDIV_2 1
#define WM8940_BCLKDIV_4 2
#define WM8940_BCLKDIV_8 3
#define WM8940_BCLKDIV_16 4
#define WM8940_BCLKDIV_32 5
/* PLL Out Dividers */
#define WM8940_OPCLKDIV_1 0
#define WM8940_OPCLKDIV_2 1
#define WM8940_OPCLKDIV_3 2
#define WM8940_OPCLKDIV_4 3
#endif /* _WM8940_H */
sound/soc/codecs/wm8960.c 0 → 100644
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/codecs/wm8960.h 0 → 100644
浏览文件 @ ba252af8
/*
* wm8960.h -- WM8960 Soc Audio driver
*
* 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 _WM8960_H
#define _WM8960_H
/* WM8960 register space */
#define WM8960_CACHEREGNUM 56
#define WM8960_LINVOL 0x0
#define WM8960_RINVOL 0x1
#define WM8960_LOUT1 0x2
#define WM8960_ROUT1 0x3
#define WM8960_CLOCK1 0x4
#define WM8960_DACCTL1 0x5
#define WM8960_DACCTL2 0x6
#define WM8960_IFACE1 0x7
#define WM8960_CLOCK2 0x8
#define WM8960_IFACE2 0x9
#define WM8960_LDAC 0xa
#define WM8960_RDAC 0xb
#define WM8960_RESET 0xf
#define WM8960_3D 0x10
#define WM8960_ALC1 0x11
#define WM8960_ALC2 0x12
#define WM8960_ALC3 0x13
#define WM8960_NOISEG 0x14
#define WM8960_LADC 0x15
#define WM8960_RADC 0x16
#define WM8960_ADDCTL1 0x17
#define WM8960_ADDCTL2 0x18
#define WM8960_POWER1 0x19
#define WM8960_POWER2 0x1a
#define WM8960_ADDCTL3 0x1b
#define WM8960_APOP1 0x1c
#define WM8960_APOP2 0x1d
#define WM8960_LINPATH 0x20
#define WM8960_RINPATH 0x21
#define WM8960_LOUTMIX 0x22
#define WM8960_ROUTMIX 0x25
#define WM8960_MONOMIX1 0x26
#define WM8960_MONOMIX2 0x27
#define WM8960_LOUT2 0x28
#define WM8960_ROUT2 0x29
#define WM8960_MONO 0x2a
#define WM8960_INBMIX1 0x2b
#define WM8960_INBMIX2 0x2c
#define WM8960_BYPASS1 0x2d
#define WM8960_BYPASS2 0x2e
#define WM8960_POWER3 0x2f
#define WM8960_ADDCTL4 0x30
#define WM8960_CLASSD1 0x31
#define WM8960_CLASSD3 0x33
#define WM8960_PLL1 0x34
#define WM8960_PLL2 0x35
#define WM8960_PLL3 0x36
#define WM8960_PLL4 0x37
/*
* WM8960 Clock dividers
*/
#define WM8960_SYSCLKDIV 0
#define WM8960_DACDIV 1
#define WM8960_OPCLKDIV 2
#define WM8960_DCLKDIV 3
#define WM8960_TOCLKSEL 4
#define WM8960_SYSCLKSEL 5
#define WM8960_SYSCLK_DIV_1 (0 << 1)
#define WM8960_SYSCLK_DIV_2 (2 << 1)
#define WM8960_SYSCLK_MCLK (0 << 0)
#define WM8960_SYSCLK_PLL (1 << 0)
#define WM8960_DAC_DIV_1 (0 << 3)
#define WM8960_DAC_DIV_1_5 (1 << 3)
#define WM8960_DAC_DIV_2 (2 << 3)
#define WM8960_DAC_DIV_3 (3 << 3)
#define WM8960_DAC_DIV_4 (4 << 3)
#define WM8960_DAC_DIV_5_5 (5 << 3)
#define WM8960_DAC_DIV_6 (6 << 3)
#define WM8960_DCLK_DIV_1_5 (0 << 6)
#define WM8960_DCLK_DIV_2 (1 << 6)
#define WM8960_DCLK_DIV_3 (2 << 6)
#define WM8960_DCLK_DIV_4 (3 << 6)
#define WM8960_DCLK_DIV_6 (4 << 6)
#define WM8960_DCLK_DIV_8 (5 << 6)
#define WM8960_DCLK_DIV_12 (6 << 6)
#define WM8960_DCLK_DIV_16 (7 << 6)
#define WM8960_TOCLK_F19 (0 << 1)
#define WM8960_TOCLK_F21 (1 << 1)
#define WM8960_OPCLK_DIV_1 (0 << 0)
#define WM8960_OPCLK_DIV_2 (1 << 0)
#define WM8960_OPCLK_DIV_3 (2 << 0)
#define WM8960_OPCLK_DIV_4 (3 << 0)
#define WM8960_OPCLK_DIV_5_5 (4 << 0)
#define WM8960_OPCLK_DIV_6 (5 << 0)
extern struct snd_soc_dai wm8960_dai;
extern struct snd_soc_codec_device soc_codec_dev_wm8960;
#define WM8960_DRES_400R 0
#define WM8960_DRES_200R 1
#define WM8960_DRES_600R 2
#define WM8960_DRES_150R 3
#define WM8960_DRES_MAX 3
struct wm8960_data {
int dres;
};
#endif
sound/soc/codecs/wm8988.c 0 → 100644
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sound/soc/codecs/wm8988.h 0 → 100644
浏览文件 @ ba252af8
/*
* Copyright 2005 Openedhand Ltd.
*
* Author: Richard Purdie <richard@openedhand.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 _WM8988_H
#define _WM8988_H
/* WM8988 register space */
#define WM8988_LINVOL 0x00
#define WM8988_RINVOL 0x01
#define WM8988_LOUT1V 0x02
#define WM8988_ROUT1V 0x03
#define WM8988_ADCDAC 0x05
#define WM8988_IFACE 0x07
#define WM8988_SRATE 0x08
#define WM8988_LDAC 0x0a
#define WM8988_RDAC 0x0b
#define WM8988_BASS 0x0c
#define WM8988_TREBLE 0x0d
#define WM8988_RESET 0x0f
#define WM8988_3D 0x10
#define WM8988_ALC1 0x11
#define WM8988_ALC2 0x12
#define WM8988_ALC3 0x13
#define WM8988_NGATE 0x14
#define WM8988_LADC 0x15
#define WM8988_RADC 0x16
#define WM8988_ADCTL1 0x17
#define WM8988_ADCTL2 0x18
#define WM8988_PWR1 0x19
#define WM8988_PWR2 0x1a
#define WM8988_ADCTL3 0x1b
#define WM8988_ADCIN 0x1f
#define WM8988_LADCIN 0x20
#define WM8988_RADCIN 0x21
#define WM8988_LOUTM1 0x22
#define WM8988_LOUTM2 0x23
#define WM8988_ROUTM1 0x24
#define WM8988_ROUTM2 0x25
#define WM8988_LOUT2V 0x28
#define WM8988_ROUT2V 0x29
#define WM8988_LPPB 0x43
#define WM8988_NUM_REG 0x44
#define WM8988_SYSCLK 0
extern struct snd_soc_dai wm8988_dai;
extern struct snd_soc_codec_device soc_codec_dev_wm8988;
#endif
sound/soc/codecs/wm8990.c
浏览文件 @ ba252af8
......@@ -998,7 +998,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target,
if ((Ndiv < 6) || (Ndiv > 12))
printk(KERN_WARNING
"WM8990 N value outwith recommended range! N = %d\n", Ndiv);
"WM8990 N value outwith recommended range! N = %u\n", Ndiv);
pll_div->n = Ndiv;
Nmod = target % source;
......
sound/soc/codecs/wm9081.c 0 → 100644
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此差异已折叠。
sound/soc/codecs/wm9081.h 0 → 100644
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/codecs/wm9705.c
浏览文件 @ ba252af8
......@@ -282,14 +282,14 @@ struct snd_soc_dai wm9705_dai[] = {
.channels_min = 1,
.channels_max = 2,
.rates = WM9705_AC97_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.formats = SND_SOC_STD_AC97_FMTS,
},
.capture = {
.stream_name = "HiFi Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM9705_AC97_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.formats = SND_SOC_STD_AC97_FMTS,
},
.ops = &wm9705_dai_ops,
},
......
sound/soc/codecs/wm9712.c
浏览文件 @ ba252af8
......@@ -534,13 +534,13 @@ struct snd_soc_dai wm9712_dai[] = {
.channels_min = 1,
.channels_max = 2,
.rates = WM9712_AC97_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
.formats = SND_SOC_STD_AC97_FMTS,},
.capture = {
.stream_name = "HiFi Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM9712_AC97_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
.formats = SND_SOC_STD_AC97_FMTS,},
.ops = &wm9712_dai_ops_hifi,
},
{
......@@ -550,7 +550,7 @@ struct snd_soc_dai wm9712_dai[] = {
.channels_min = 1,
.channels_max = 1,
.rates = WM9712_AC97_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
.formats = SND_SOC_STD_AC97_FMTS,},
.ops = &wm9712_dai_ops_aux,
}
};
......@@ -585,6 +585,8 @@ static int wm9712_reset(struct snd_soc_codec *codec, int try_warm)
}
soc_ac97_ops.reset(codec->ac97);
if (soc_ac97_ops.warm_reset)
soc_ac97_ops.warm_reset(codec->ac97);
if (ac97_read(codec, 0) != wm9712_reg[0])
goto err;
return 0;
......
sound/soc/codecs/wm9713.c
浏览文件 @ ba252af8
......@@ -189,6 +189,26 @@ SOC_SINGLE("3D Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0),
SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1),
};
static int wm9713_voice_shutdown(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
u16 status, rate;
BUG_ON(event != SND_SOC_DAPM_PRE_PMD);
/* Gracefully shut down the voice interface. */
status = ac97_read(codec, AC97_EXTENDED_MID) | 0x1000;
rate = ac97_read(codec, AC97_HANDSET_RATE) & 0xF0FF;
ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0200);
schedule_timeout_interruptible(msecs_to_jiffies(1));
ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0F00);
ac97_write(codec, AC97_EXTENDED_MID, status);
return 0;
}
/* We have to create a fake left and right HP mixers because
* the codec only has a single control that is shared by both channels.
* This makes it impossible to determine the audio path using the current
......@@ -400,7 +420,8 @@ SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("HP Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("Line Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("Capture Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1),
SND_SOC_DAPM_DAC_E("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1,
wm9713_voice_shutdown, SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", AC97_EXTENDED_MID, 11, 1),
SND_SOC_DAPM_PGA("Left ADC", AC97_EXTENDED_MID, 5, 1, NULL, 0),
SND_SOC_DAPM_PGA("Right ADC", AC97_EXTENDED_MID, 4, 1, NULL, 0),
......@@ -689,7 +710,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int source)
Ndiv = target / source;
if ((Ndiv < 5) || (Ndiv > 12))
printk(KERN_WARNING
"WM9713 PLL N value %d out of recommended range!\n",
"WM9713 PLL N value %u out of recommended range!\n",
Ndiv);
pll_div->n = Ndiv;
......@@ -936,21 +957,6 @@ static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream,
return 0;
}
static void wm9713_voiceshutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
u16 status, rate;
/* Gracefully shut down the voice interface. */
status = ac97_read(codec, AC97_EXTENDED_STATUS) | 0x1000;
rate = ac97_read(codec, AC97_HANDSET_RATE) & 0xF0FF;
ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0200);
schedule_timeout_interruptible(msecs_to_jiffies(1));
ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0F00);
ac97_write(codec, AC97_EXTENDED_MID, status);
}
static int ac97_hifi_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
......@@ -1019,7 +1025,6 @@ static struct snd_soc_dai_ops wm9713_dai_ops_aux = {
static struct snd_soc_dai_ops wm9713_dai_ops_voice = {
.hw_params = wm9713_pcm_hw_params,
.shutdown = wm9713_voiceshutdown,
.set_clkdiv = wm9713_set_dai_clkdiv,
.set_pll = wm9713_set_dai_pll,
.set_fmt = wm9713_set_dai_fmt,
......@@ -1035,13 +1040,13 @@ struct snd_soc_dai wm9713_dai[] = {
.channels_min = 1,
.channels_max = 2,
.rates = WM9713_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
.formats = SND_SOC_STD_AC97_FMTS,},
.capture = {
.stream_name = "HiFi Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM9713_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
.formats = SND_SOC_STD_AC97_FMTS,},
.ops = &wm9713_dai_ops_hifi,
},
{
......@@ -1051,7 +1056,7 @@ struct snd_soc_dai wm9713_dai[] = {
.channels_min = 1,
.channels_max = 1,
.rates = WM9713_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
.formats = SND_SOC_STD_AC97_FMTS,},
.ops = &wm9713_dai_ops_aux,
},
{
......@@ -1069,6 +1074,7 @@ struct snd_soc_dai wm9713_dai[] = {
.rates = WM9713_PCM_RATES,
.formats = WM9713_PCM_FORMATS,},
.ops = &wm9713_dai_ops_voice,
.symmetric_rates = 1,
},
};
EXPORT_SYMBOL_GPL(wm9713_dai);
......
sound/soc/fsl/Kconfig
浏览文件 @ ba252af8
config SND_SOC_OF_SIMPLE
tristate
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
......@@ -22,7 +25,34 @@ config SND_SOC_MPC8610_HPCD
config SND_SOC_MPC5200_I2S
tristate "Freescale MPC5200 PSC in I2S mode driver"
depends on PPC_MPC52xx && PPC_BESTCOMM
select SND_SOC_OF_SIMPLE
select SND_MPC52xx_DMA
select PPC_BESTCOMM_GEN_BD
help
Say Y here to support the MPC5200 PSCs in I2S mode.
config SND_SOC_MPC5200_AC97
tristate "Freescale MPC5200 PSC in AC97 mode driver"
depends on PPC_MPC52xx && PPC_BESTCOMM
select AC97_BUS
select SND_MPC52xx_DMA
select PPC_BESTCOMM_GEN_BD
help
Say Y here to support the MPC5200 PSCs in AC97 mode.
config SND_MPC52xx_SOC_PCM030
tristate "SoC AC97 Audio support for Phytec pcm030 and WM9712"
depends on PPC_MPC5200_SIMPLE && BROKEN
select SND_SOC_MPC5200_AC97
select SND_SOC_WM9712
help
Say Y if you want to add support for sound on the Phytec pcm030
baseboard.
config SND_MPC52xx_SOC_EFIKA
tristate "SoC AC97 Audio support for bbplan Efika and STAC9766"
depends on PPC_EFIKA && BROKEN
select SND_SOC_MPC5200_AC97
select SND_SOC_STAC9766
help
Say Y if you want to add support for sound on the Efika.
sound/soc/fsl/Makefile
浏览文件 @ ba252af8
......@@ -10,5 +10,12 @@ snd-soc-fsl-ssi-objs := fsl_ssi.o
snd-soc-fsl-dma-objs := fsl_dma.o
obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o
# MPC5200 Platform Support
obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o
obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o
obj-$(CONFIG_SND_SOC_MPC5200_AC97) += mpc5200_psc_ac97.o
# MPC5200 Machine Support
obj-$(CONFIG_SND_MPC52xx_SOC_PCM030) += pcm030-audio-fabric.o
obj-$(CONFIG_SND_MPC52xx_SOC_EFIKA) += efika-audio-fabric.o
sound/soc/fsl/efika-audio-fabric.c 0 → 100644
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/fsl/fsl_ssi.c
浏览文件 @ ba252af8
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sound/soc/fsl/mpc5200_dma.c 0 → 100644
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/fsl/mpc5200_dma.h 0 → 100644
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/fsl/mpc5200_psc_ac97.c 0 → 100644
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/fsl/mpc5200_psc_ac97.h 0 → 100644
浏览文件 @ ba252af8
/*
* Freescale MPC5200 PSC in AC97 mode
* ALSA SoC Digital Audio Interface (DAI) driver
*
*/
#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__
#define __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__
extern struct snd_soc_dai psc_ac97_dai[];
#define MPC5200_AC97_NORMAL 0
#define MPC5200_AC97_SPDIF 1
#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ */
sound/soc/fsl/mpc5200_psc_i2s.c
浏览文件 @ ba252af8
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sound/soc/fsl/mpc5200_psc_i2s.h 0 → 100644
浏览文件 @ ba252af8
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sound/soc/fsl/pcm030-audio-fabric.c 0 → 100644
浏览文件 @ ba252af8
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sound/soc/omap/Kconfig
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/omap/Makefile
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/omap/n810.c
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/omap/omap-mcbsp.c
浏览文件 @ ba252af8
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sound/soc/omap/omap-pcm.c
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/omap/omap2evm.c
浏览文件 @ ba252af8
......@@ -86,7 +86,7 @@ static struct snd_soc_dai_link omap2evm_dai = {
.name = "TWL4030",
.stream_name = "TWL4030",
.cpu_dai = &omap_mcbsp_dai[0],
.codec_dai = &twl4030_dai,
.codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
.ops = &omap2evm_ops,
};
......
sound/soc/omap/omap3beagle.c
浏览文件 @ ba252af8
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sound/soc/omap/omap3evm.c 0 → 100644
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sound/soc/omap/omap3pandora.c
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sound/soc/omap/overo.c
浏览文件 @ ba252af8
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sound/soc/omap/sdp3430.c
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/pxa/Kconfig
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/pxa/Makefile
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/pxa/em-x270.c
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sound/soc/pxa/imote2.c 0 → 100644
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sound/soc/pxa/magician.c
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sound/soc/pxa/pxa-ssp.c
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sound/soc/pxa/pxa2xx-i2s.c
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此差异已折叠。
sound/soc/s3c24xx/s3c-i2s-v2.c
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/s3c24xx/s3c2412-i2s.c
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/s3c24xx/s3c64xx-i2s.c
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/s3c24xx/s3c64xx-i2s.h
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/s6000/Kconfig 0 → 100644
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/s6000/Makefile 0 → 100644
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/s6000/s6000-i2s.c 0 → 100644
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/s6000/s6000-i2s.h 0 → 100644
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/s6000/s6000-pcm.c 0 → 100644
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/s6000/s6000-pcm.h 0 → 100644
浏览文件 @ ba252af8
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sound/soc/s6000/s6105-ipcam.c 0 → 100644
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/sh/ssi.c
浏览文件 @ ba252af8
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sound/soc/soc-core.c
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/soc-dapm.c
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/txx9/Kconfig 0 → 100644
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/txx9/Makefile 0 → 100644
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/txx9/txx9aclc-ac97.c 0 → 100644
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/txx9/txx9aclc-generic.c 0 → 100644
浏览文件 @ ba252af8
此差异已折叠。
sound/soc/txx9/txx9aclc.c 0 → 100644
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sound/soc/txx9/txx9aclc.h 0 → 100644
浏览文件 @ ba252af8
此差异已折叠。
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