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

Browse source 
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (33 commits)
  ALSA: ASoC codec: remove unused #include <version.h>
  ALSA: ASoC: update email address for Liam Girdwood
  ALSA: hda: corrected invalid mixer values
  ALSA: hda: add mixers for analog mixer on 92hd75xx codecs
  ALSA: ASoC: Add destination and source port for DMA on OMAP1
  ALSA: ASoC: Drop device registration from GTA01 lm4857 driver
  ALSA: ASoC: Fix build of GTA01 audio driver
  ALSA: ASoC: Add widgets before setting endpoints on GTA01
  ALSA: ASoC: Fix inverted input PGA mute bits in WM8903
  ALSA: ASoC: OMAP: Set DMA stream name at runtime in McBSP DAI driver
  ALSA: ASoC: OMAP: Add support for OMAP2430 and OMAP34xx in McBSP DAI driver
  ALSA: ASoC: OMAP: Add multilink support to McBSP DAI driver
  ALSA: ASoC: Make TLV320AIC26 user-visible
  ALSA: ASoC - clean up Kconfig for TLV320AIC2
  ALSA: ASoC: Make WM8510 microphone input a DAPM mixer
  ALSA: ASoC: Implement WM8510 bias level control
  ALSA: ASoC: Remove unused AUDIO_NAME define from codec drivers
  ALSA: ASoC: tlv320aic3x: Use uniform tlv320aic naming
  ALSA: ASoC: Add WM8510 SPI support
  ALSA: ASoC: Add WM8753 SPI support
  ...
This commit is contained in:
Linus Torvalds 2008-10-13 10:06:58 -07:00
commit be3bfbba8f
56 changed files with 2038 additions and 602 deletions
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1
include/sound/soc-dapm.h
View file

@ -240,6 +240,7 @@ int snd_soc_dapm_sys_add(struct device *dev);
/* dapm audio pin control and status */
int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin);
int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin);
int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, char *pin);
int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin);
int snd_soc_dapm_sync(struct snd_soc_codec *codec);

2
sound/oss/ac97_codec.c
View file

@ -30,7 +30,7 @@
**************************************************************************
*
* History
* May 02, 2003 Liam Girdwood <liam.girdwood@wolfsonmicro.com>
* May 02, 2003 Liam Girdwood <lrg@slimlogic.co.uk>
* Removed non existant WM9700
* Added support for WM9705, WM9708, WM9709, WM9710, WM9711
* WM9712 and WM9717

2
sound/pci/ac97/ac97_patch.c
View file

@ -476,7 +476,7 @@ static int patch_yamaha_ymf753(struct snd_ac97 * ac97)
}
/*
* May 2, 2003 Liam Girdwood <liam.girdwood@wolfsonmicro.com>
* May 2, 2003 Liam Girdwood <lrg@slimlogic.co.uk>
* removed broken wolfson00 patch.
* added support for WM9705,WM9708,WM9709,WM9710,WM9711,WM9712 and WM9717.
*/

50
sound/pci/hda/patch_sigmatel.c
View file

@ -322,8 +322,8 @@ static hda_nid_t stac92hd71bxx_mux_nids[2] = {
0x1a, 0x1b
};
static hda_nid_t stac92hd71bxx_dmux_nids[1] = {
0x1c,
static hda_nid_t stac92hd71bxx_dmux_nids[2] = {
0x1c, 0x1d,
};
static hda_nid_t stac92hd71bxx_smux_nids[2] = {
@ -861,20 +861,18 @@ static struct hda_verb stac92hd71bxx_core_init[] = {
{ 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
/* connect headphone jack to dac1 */
{ 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01},
{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */
/* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */
{ 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{ 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
};
#define HD_DISABLE_PORTF 3
#define HD_DISABLE_PORTF 2
static struct hda_verb stac92hd71bxx_analog_core_init[] = {
/* start of config #1 */
/* connect port 0f to audio mixer */
{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2},
{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */
/* unmute right and left channels for node 0x0f */
{ 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* start of config #2 */
@ -883,10 +881,6 @@ static struct hda_verb stac92hd71bxx_analog_core_init[] = {
{ 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
/* connect headphone jack to dac1 */
{ 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01},
/* connect port 0d to audio mixer */
{ 0x0d, AC_VERB_SET_CONNECT_SEL, 0x2},
/* unmute dac0 input in audio mixer */
{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f},
/* unmute right and left channels for nodes 0x0a, 0xd */
{ 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
@ -1107,6 +1101,7 @@ static struct snd_kcontrol_new stac92hd83xxx_mixer[] = {
static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = {
STAC_INPUT_SOURCE(2),
STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2),
HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT),
@ -1119,8 +1114,17 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = {
HDA_CODEC_MUTE("PC Beep Switch", 0x17, 0x2, HDA_INPUT),
*/
HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT),
HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT),
HDA_CODEC_MUTE("Import0 Mux Capture Switch", 0x17, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Import0 Mux Capture Volume", 0x17, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Import1 Mux Capture Switch", 0x17, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Import1 Mux Capture Volume", 0x17, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("DAC0 Capture Switch", 0x17, 0x3, HDA_INPUT),
HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x17, 0x3, HDA_INPUT),
HDA_CODEC_MUTE("DAC1 Capture Switch", 0x17, 0x4, HDA_INPUT),
HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x17, 0x4, HDA_INPUT),
{ } /* end */
};
@ -1649,7 +1653,7 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
static unsigned int ref92hd71bxx_pin_configs[11] = {
0x02214030, 0x02a19040, 0x01a19020, 0x01014010,
0x0181302e, 0x01114010, 0x01019020, 0x90a000f0,
0x0181302e, 0x01014010, 0x01019020, 0x90a000f0,
0x90a000f0, 0x01452050, 0x01452050,
};
@ -3000,7 +3004,7 @@ static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec)
/* labels for amp mux outputs */
static const char *stac92xx_amp_labels[3] = {
"Front Microphone", "Microphone", "Line In"
"Front Microphone", "Microphone", "Line In",
};
/* create amp out controls mux on capable codecs */
@ -4327,6 +4331,16 @@ static struct hda_codec_ops stac92hd71bxx_patch_ops = {
#endif
};
static struct hda_input_mux stac92hd71bxx_dmux = {
.num_items = 4,
.items = {
{ "Analog Inputs", 0x00 },
{ "Mixer", 0x01 },
{ "Digital Mic 1", 0x02 },
{ "Digital Mic 2", 0x03 },
}
};
static int patch_stac92hd71bxx(struct hda_codec *codec)
{
struct sigmatel_spec *spec;
@ -4341,6 +4355,8 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids);
spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids);
spec->pin_nids = stac92hd71bxx_pin_nids;
memcpy(&spec->private_dimux, &stac92hd71bxx_dmux,
sizeof(stac92hd71bxx_dmux));
spec->board_config = snd_hda_check_board_config(codec,
STAC_92HD71BXX_MODELS,
stac92hd71bxx_models,
@ -4392,6 +4408,7 @@ again:
/* no output amps */
spec->num_pwrs = 0;
spec->mixer = stac92hd71bxx_analog_mixer;
spec->dinput_mux = &spec->private_dimux;
/* disable VSW */
spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF];
@ -4409,12 +4426,13 @@ again:
spec->num_pwrs = 0;
/* fallthru */
default:
spec->dinput_mux = &spec->private_dimux;
spec->mixer = stac92hd71bxx_analog_mixer;
spec->init = stac92hd71bxx_analog_core_init;
codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
}
spec->aloopback_mask = 0x20;
spec->aloopback_mask = 0x50;
spec->aloopback_shift = 0;
if (spec->board_config > STAC_92HD71BXX_REF) {
@ -4456,6 +4474,10 @@ again:
spec->multiout.num_dacs = 1;
spec->multiout.hp_nid = 0x11;
spec->multiout.dac_nids = stac92hd71bxx_dac_nids;
if (spec->dinput_mux)
spec->private_dimux.num_items +=
spec->num_dmics -
(ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1);
err = stac92xx_parse_auto_config(codec, 0x21, 0x23);
if (!err) {

17
sound/soc/at91/Kconfig
View file

@ -8,20 +8,3 @@ config SND_AT91_SOC
config SND_AT91_SOC_SSC
tristate
config SND_AT91_SOC_ETI_B1_WM8731
tristate "SoC Audio support for WM8731-based Endrelia ETI-B1 boards"
depends on SND_AT91_SOC && (MACH_ETI_B1 || MACH_ETI_C1)
select SND_AT91_SOC_SSC
select SND_SOC_WM8731
help
Say Y if you want to add support for SoC audio on WM8731-based
Endrelia Technologies Inc ETI-B1 or ETI-C1 boards.
config SND_AT91_SOC_ETI_SLAVE
bool "Run codec in slave Mode on Endrelia boards"
depends on SND_AT91_SOC_ETI_B1_WM8731
default n
help
Say Y if you want to run with the AT91 SSC generating the BCLK
and LRC signals on Endrelia boards.

5
sound/soc/at91/Makefile
View file

@ -4,8 +4,3 @@ snd-soc-at91-ssc-objs := at91-ssc.o
obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o
obj-$(CONFIG_SND_AT91_SOC_SSC) += snd-soc-at91-ssc.o
# AT91 Machine Support
snd-soc-eti-b1-wm8731-objs := eti_b1_wm8731.o
obj-$(CONFIG_SND_AT91_SOC_ETI_B1_WM8731) += snd-soc-eti-b1-wm8731.o

2
sound/soc/at91/at91-ssc.c
View file

@ -5,7 +5,7 @@
* Endrelia Technologies Inc.
*
* Based on pxa2xx Platform drivers by
* Liam Girdwood <liam.girdwood@wolfsonmicro.com>
* Liam Girdwood <lrg@slimlogic.co.uk>
*
* 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

349
sound/soc/at91/eti_b1_wm8731.c
View file

@ -1,349 +0,0 @@
/*
* eti_b1_wm8731 -- SoC audio for AT91RM9200-based Endrelia ETI_B1 board.
*
* Author: Frank Mandarino <fmandarino@endrelia.com>
* Endrelia Technologies Inc.
* Created: Mar 29, 2006
*
* Based on corgi.c by:
*
* Copyright 2005 Wolfson Microelectronics PLC.
* Copyright 2005 Openedhand Ltd.
*
* Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
* Richard Purdie <richard@openedhand.com>
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the
* Free Software Foundation; either version 2 of the License, or (at your
* option) any later version.
*
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/kernel.h>
#include <linux/clk.h>
#include <linux/timer.h>
#include <linux/interrupt.h>
#include <linux/platform_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <mach/hardware.h>
#include <mach/gpio.h>
#include "../codecs/wm8731.h"
#include "at91-pcm.h"
#include "at91-ssc.h"
#if 0
#define DBG(x...) printk(KERN_INFO "eti_b1_wm8731: " x)
#else
#define DBG(x...)
#endif
static struct clk *pck1_clk;
static struct clk *pllb_clk;
static int eti_b1_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
int ret;
/* cpu clock is the AT91 master clock sent to the SSC */
ret = snd_soc_dai_set_sysclk(cpu_dai, AT91_SYSCLK_MCK,
60000000, SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
/* codec system clock is supplied by PCK1, set to 12MHz */
ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK,
12000000, SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
/* Start PCK1 clock. */
clk_enable(pck1_clk);
DBG("pck1 started\n");
return 0;
}
static void eti_b1_shutdown(struct snd_pcm_substream *substream)
{
/* Stop PCK1 clock. */
clk_disable(pck1_clk);
DBG("pck1 stopped\n");
}
static int eti_b1_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
int ret;
#ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE
unsigned int rate;
int cmr_div, period;
/* set codec DAI configuration */
ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0)
return ret;
/* set cpu DAI configuration */
ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0)
return ret;
/*
* The SSC clock dividers depend on the sample rate. The CMR.DIV
* field divides the system master clock MCK to drive the SSC TK
* signal which provides the codec BCLK. The TCMR.PERIOD and
* RCMR.PERIOD fields further divide the BCLK signal to drive
* the SSC TF and RF signals which provide the codec DACLRC and
* ADCLRC clocks.
*
* The dividers were determined through trial and error, where a
* CMR.DIV value is chosen such that the resulting BCLK value is
* divisible, or almost divisible, by (2 * sample rate), and then
* the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1.
*/
rate = params_rate(params);
switch (rate) {
case 8000:
cmr_div = 25; /* BCLK = 60MHz/(2*25) = 1.2MHz */
period = 74; /* LRC = BCLK/(2*(74+1)) = 8000Hz */
break;
case 32000:
cmr_div = 7; /* BCLK = 60MHz/(2*7) ~= 4.28571428MHz */
period = 66; /* LRC = BCLK/(2*(66+1)) = 31982.942Hz */
break;
case 48000:
cmr_div = 13; /* BCLK = 60MHz/(2*13) ~= 2.3076923MHz */
period = 23; /* LRC = BCLK/(2*(23+1)) = 48076.923Hz */
break;
default:
printk(KERN_WARNING "unsupported rate %d on ETI-B1 board\n", rate);
return -EINVAL;
}
/* set the MCK divider for BCLK */
ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div);
if (ret < 0)
return ret;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
/* set the BCLK divider for DACLRC */
ret = snd_soc_dai_set_clkdiv(cpu_dai,
AT91SSC_TCMR_PERIOD, period);
} else {
/* set the BCLK divider for ADCLRC */
ret = snd_soc_dai_set_clkdiv(cpu_dai,
AT91SSC_RCMR_PERIOD, period);
}
if (ret < 0)
return ret;
#else /* CONFIG_SND_AT91_SOC_ETI_SLAVE */
/*
* Codec in Master Mode.
*/
/* set codec DAI configuration */
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;
/* set cpu DAI configuration */
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;
#endif /* CONFIG_SND_AT91_SOC_ETI_SLAVE */
return 0;
}
static struct snd_soc_ops eti_b1_ops = {
.startup = eti_b1_startup,
.hw_params = eti_b1_hw_params,
.shutdown = eti_b1_shutdown,
};
static const struct snd_soc_dapm_widget eti_b1_dapm_widgets[] = {
SND_SOC_DAPM_MIC("Int Mic", NULL),
SND_SOC_DAPM_SPK("Ext Spk", NULL),
};
static const struct snd_soc_dapm_route intercon[] = {
/* speaker connected to LHPOUT */
{"Ext Spk", NULL, "LHPOUT"},
/* mic is connected to Mic Jack, with WM8731 Mic Bias */
{"MICIN", NULL, "Mic Bias"},
{"Mic Bias", NULL, "Int Mic"},
};
/*
* Logic for a wm8731 as connected on a Endrelia ETI-B1 board.
*/
static int eti_b1_wm8731_init(struct snd_soc_codec *codec)
{
DBG("eti_b1_wm8731_init() called\n");
/* Add specific widgets */
snd_soc_dapm_new_controls(codec, eti_b1_dapm_widgets,
ARRAY_SIZE(eti_b1_dapm_widgets));
/* Set up specific audio path interconnects */
snd_soc_dapm_add_route(codec, intercon, ARRAY_SIZE(intercon));
/* not connected */
snd_soc_dapm_disable_pin(codec, "RLINEIN");
snd_soc_dapm_disable_pin(codec, "LLINEIN");
/* always connected */
snd_soc_dapm_enable_pin(codec, "Int Mic");
snd_soc_dapm_enable_pin(codec, "Ext Spk");
snd_soc_dapm_sync(codec);
return 0;
}
static struct snd_soc_dai_link eti_b1_dai = {
.name = "WM8731",
.stream_name = "WM8731 PCM",
.cpu_dai = &at91_ssc_dai[1],
.codec_dai = &wm8731_dai,
.init = eti_b1_wm8731_init,
.ops = &eti_b1_ops,
};
static struct snd_soc_machine snd_soc_machine_eti_b1 = {
.name = "ETI_B1_WM8731",
.dai_link = &eti_b1_dai,
.num_links = 1,
};
static struct wm8731_setup_data eti_b1_wm8731_setup = {
.i2c_bus = 0,
.i2c_address = 0x1a,
};
static struct snd_soc_device eti_b1_snd_devdata = {
.machine = &snd_soc_machine_eti_b1,
.platform = &at91_soc_platform,
.codec_dev = &soc_codec_dev_wm8731,
.codec_data = &eti_b1_wm8731_setup,
};
static struct platform_device *eti_b1_snd_device;
static int __init eti_b1_init(void)
{
int ret;
struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data;
if (!request_mem_region(AT91RM9200_BASE_SSC1, SZ_16K, "soc-audio")) {
DBG("SSC1 memory region is busy\n");
return -EBUSY;
}
ssc->base = ioremap(AT91RM9200_BASE_SSC1, SZ_16K);
if (!ssc->base) {
DBG("SSC1 memory ioremap failed\n");
ret = -ENOMEM;
goto fail_release_mem;
}
ssc->pid = AT91RM9200_ID_SSC1;
eti_b1_snd_device = platform_device_alloc("soc-audio", -1);
if (!eti_b1_snd_device) {
DBG("platform device allocation failed\n");
ret = -ENOMEM;
goto fail_io_unmap;
}
platform_set_drvdata(eti_b1_snd_device, &eti_b1_snd_devdata);
eti_b1_snd_devdata.dev = &eti_b1_snd_device->dev;
ret = platform_device_add(eti_b1_snd_device);
if (ret) {
DBG("platform device add failed\n");
platform_device_put(eti_b1_snd_device);
goto fail_io_unmap;
}
at91_set_A_periph(AT91_PIN_PB6, 0); /* TF1 */
at91_set_A_periph(AT91_PIN_PB7, 0); /* TK1 */
at91_set_A_periph(AT91_PIN_PB8, 0); /* TD1 */
at91_set_A_periph(AT91_PIN_PB9, 0); /* RD1 */
/* at91_set_A_periph(AT91_PIN_PB10, 0);*/ /* RK1 */
at91_set_A_periph(AT91_PIN_PB11, 0); /* RF1 */
/*
* Set PCK1 parent to PLLB and its rate to 12 Mhz.
*/
pllb_clk = clk_get(NULL, "pllb");
pck1_clk = clk_get(NULL, "pck1");
clk_set_parent(pck1_clk, pllb_clk);
clk_set_rate(pck1_clk, 12000000);
DBG("MCLK rate %luHz\n", clk_get_rate(pck1_clk));
/* assign the GPIO pin to PCK1 */
at91_set_B_periph(AT91_PIN_PA24, 0);
#ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE
printk(KERN_INFO "eti_b1_wm8731: Codec in Slave Mode\n");
#else
printk(KERN_INFO "eti_b1_wm8731: Codec in Master Mode\n");
#endif
return ret;
fail_io_unmap:
iounmap(ssc->base);
fail_release_mem:
release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K);
return ret;
}
static void __exit eti_b1_exit(void)
{
struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data;
clk_put(pck1_clk);
clk_put(pllb_clk);
platform_device_unregister(eti_b1_snd_device);
iounmap(ssc->base);
release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K);
}
module_init(eti_b1_init);
module_exit(eti_b1_exit);
/* Module information */
MODULE_AUTHOR("Frank Mandarino <fmandarino@endrelia.com>");
MODULE_DESCRIPTION("ALSA SoC ETI-B1-WM8731");
MODULE_LICENSE("GPL");

16
sound/soc/blackfin/Kconfig
View file

@ -17,6 +17,22 @@ config SND_BF5XX_SOC_SSM2602
help
Say Y if you want to add support for SoC audio on BF527-EZKIT.
config SND_BF5XX_SOC_AD73311
tristate "SoC AD73311 Audio support for Blackfin"
depends on SND_BF5XX_I2S
select SND_BF5XX_SOC_I2S
select SND_SOC_AD73311
help
Say Y if you want to add support for AD73311 codec on Blackfin.
config SND_BFIN_AD73311_SE
int "PF pin for AD73311L Chip Select"
depends on SND_BF5XX_SOC_AD73311
default 4
help
Enter the GPIO used to control AD73311's SE pin. Acceptable
values are 0 to 7
config SND_BF5XX_AC97
tristate "SoC AC97 Audio for the ADI BF5xx chip"
depends on BLACKFIN && SND_SOC

3
sound/soc/blackfin/Makefile
View file

@ -14,7 +14,8 @@ obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o
# Blackfin Machine Support
snd-ad1980-objs := bf5xx-ad1980.o
snd-ssm2602-objs := bf5xx-ssm2602.o
snd-ad73311-objs := bf5xx-ad73311.o
obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o
obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o
obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o

42
sound/soc/blackfin/bf5xx-ac97-pcm.c
View file

@ -56,6 +56,7 @@ static void bf5xx_mmap_copy(struct snd_pcm_substream *substream,
sport->tx_pos += runtime->period_size;
if (sport->tx_pos >= runtime->buffer_size)
sport->tx_pos %= runtime->buffer_size;
sport->tx_delay_pos = sport->tx_pos;
} else {
bf5xx_ac97_to_pcm(
(struct ac97_frame *)sport->rx_dma_buf + sport->rx_pos,
@ -72,7 +73,15 @@ static void bf5xx_dma_irq(void *data)
struct snd_pcm_substream *pcm = data;
#if defined(CONFIG_SND_MMAP_SUPPORT)
struct snd_pcm_runtime *runtime = pcm->runtime;
struct sport_device *sport = runtime->private_data;
bf5xx_mmap_copy(pcm, runtime->period_size);
if (pcm->stream == SNDRV_PCM_STREAM_PLAYBACK) {
if (sport->once == 0) {
snd_pcm_period_elapsed(pcm);
bf5xx_mmap_copy(pcm, runtime->period_size);
sport->once = 1;
}
}
#endif
snd_pcm_period_elapsed(pcm);
}
@ -114,6 +123,10 @@ static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream,
static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
memset(runtime->dma_area, 0, runtime->buffer_size);
snd_pcm_lib_free_pages(substream);
return 0;
}
@ -127,16 +140,11 @@ static int bf5xx_pcm_prepare(struct snd_pcm_substream *substream)
* SPORT working in TMD mode(include AC97).
*/
#if defined(CONFIG_SND_MMAP_SUPPORT)
size_t size = bf5xx_pcm_hardware.buffer_bytes_max
* sizeof(struct ac97_frame) / 4;
/*clean up intermediate buffer*/
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
memset(sport->tx_dma_buf, 0, size);
sport_set_tx_callback(sport, bf5xx_dma_irq, substream);
sport_config_tx_dma(sport, sport->tx_dma_buf, runtime->periods,
runtime->period_size * sizeof(struct ac97_frame));
} else {
memset(sport->rx_dma_buf, 0, size);
sport_set_rx_callback(sport, bf5xx_dma_irq, substream);
sport_config_rx_dma(sport, sport->rx_dma_buf, runtime->periods,
runtime->period_size * sizeof(struct ac97_frame));
@ -164,8 +172,12 @@ static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
pr_debug("%s enter\n", __func__);
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
bf5xx_mmap_copy(substream, runtime->period_size);
snd_pcm_period_elapsed(substream);
sport->tx_delay_pos = 0;
sport_tx_start(sport);
}
else
sport_rx_start(sport);
break;
@ -198,7 +210,7 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream)
#if defined(CONFIG_SND_MMAP_SUPPORT)
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
curr = sport->tx_pos;
curr = sport->tx_delay_pos;
else
curr = sport->rx_pos;
#else
@ -237,6 +249,21 @@ static int bf5xx_pcm_open(struct snd_pcm_substream *substream)
return ret;
}
static int bf5xx_pcm_close(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct sport_device *sport = runtime->private_data;
pr_debug("%s enter\n", __func__);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
sport->once = 0;
memset(sport->tx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame));
} else
memset(sport->rx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame));
return 0;
}
#ifdef CONFIG_SND_MMAP_SUPPORT
static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream,
struct vm_area_struct *vma)
@ -272,6 +299,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel,
struct snd_pcm_ops bf5xx_pcm_ac97_ops = {
.open = bf5xx_pcm_open,
.close = bf5xx_pcm_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = bf5xx_pcm_hw_params,
.hw_free = bf5xx_pcm_hw_free,

1
sound/soc/blackfin/bf5xx-ac97.c
View file

@ -128,7 +128,6 @@ static void enqueue_cmd(struct snd_ac97 *ac97, __u16 addr, __u16 data)
int nextfrag = sport_tx_curr_frag(sport);
struct ac97_frame *nextwrite;
sport_incfrag(sport, &nextfrag, 1);
sport_incfrag(sport, &nextfrag, 1);
nextwrite = (struct ac97_frame *)(sport->tx_buf + \

240
sound/soc/blackfin/bf5xx-ad73311.c Normal file
View file

@ -0,0 +1,240 @@
/*
* File: sound/soc/blackfin/bf5xx-ad73311.c
* Author: Cliff Cai <Cliff.Cai@analog.com>
*
* Created: Thur Sep 25 2008
* Description: Board driver for ad73311 sound chip
*
* Modified:
* Copyright 2008 Analog Devices Inc.
*
* Bugs: Enter bugs at http://blackfin.uclinux.org/
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, see the file COPYING, or 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/device.h>
#include <linux/delay.h>
#include <linux/gpio.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/pcm_params.h>
#include <asm/blackfin.h>
#include <asm/cacheflush.h>
#include <asm/irq.h>
#include <asm/dma.h>
#include <asm/portmux.h>
#include "../codecs/ad73311.h"
#include "bf5xx-sport.h"
#include "bf5xx-i2s-pcm.h"
#include "bf5xx-i2s.h"
#if CONFIG_SND_BF5XX_SPORT_NUM == 0
#define bfin_write_SPORT_TCR1 bfin_write_SPORT0_TCR1
#define bfin_read_SPORT_TCR1 bfin_read_SPORT0_TCR1
#define bfin_write_SPORT_TCR2 bfin_write_SPORT0_TCR2
#define bfin_write_SPORT_TX16 bfin_write_SPORT0_TX16
#define bfin_read_SPORT_STAT bfin_read_SPORT0_STAT
#else
#define bfin_write_SPORT_TCR1 bfin_write_SPORT1_TCR1
#define bfin_read_SPORT_TCR1 bfin_read_SPORT1_TCR1
#define bfin_write_SPORT_TCR2 bfin_write_SPORT1_TCR2
#define bfin_write_SPORT_TX16 bfin_write_SPORT1_TX16
#define bfin_read_SPORT_STAT bfin_read_SPORT1_STAT
#endif
#define GPIO_SE CONFIG_SND_BFIN_AD73311_SE
static struct snd_soc_machine bf5xx_ad73311;
static int snd_ad73311_startup(void)
{
pr_debug("%s enter\n", __func__);
/* Pull up SE pin on AD73311L */
gpio_set_value(GPIO_SE, 1);
return 0;
}
static int snd_ad73311_configure(void)
{
unsigned short ctrl_regs[6];
unsigned short status = 0;
int count = 0;
/* DMCLK = MCLK = 16.384 MHz
* SCLK = DMCLK/8 = 2.048 MHz
* Sample Rate = DMCLK/2048 = 8 KHz
*/
ctrl_regs[0] = AD_CONTROL | AD_WRITE | CTRL_REG_B | REGB_MCDIV(0) | \
REGB_SCDIV(0) | REGB_DIRATE(0);
ctrl_regs[1] = AD_CONTROL | AD_WRITE | CTRL_REG_C | REGC_PUDEV | \
REGC_PUADC | REGC_PUDAC | REGC_PUREF | REGC_REFUSE ;
ctrl_regs[2] = AD_CONTROL | AD_WRITE | CTRL_REG_D | REGD_OGS(2) | \
REGD_IGS(2);
ctrl_regs[3] = AD_CONTROL | AD_WRITE | CTRL_REG_E | REGE_DA(0x1f);
ctrl_regs[4] = AD_CONTROL | AD_WRITE | CTRL_REG_F | REGF_SEEN ;
ctrl_regs[5] = AD_CONTROL | AD_WRITE | CTRL_REG_A | REGA_MODE_DATA;
local_irq_disable();
snd_ad73311_startup();
udelay(1);
bfin_write_SPORT_TCR1(TFSR);
bfin_write_SPORT_TCR2(0xF);
SSYNC();
/* SPORT Tx Register is a 8 x 16 FIFO, all the data can be put to
* FIFO before enable SPORT to transfer the data
*/
for (count = 0; count < 6; count++)
bfin_write_SPORT_TX16(ctrl_regs[count]);
SSYNC();
bfin_write_SPORT_TCR1(bfin_read_SPORT_TCR1() | TSPEN);
SSYNC();
/* When TUVF is set, the data is already send out */
while (!(status & TUVF) && count++ < 10000) {
udelay(1);
status = bfin_read_SPORT_STAT();
SSYNC();
}
bfin_write_SPORT_TCR1(bfin_read_SPORT_TCR1() & ~TSPEN);
SSYNC();
local_irq_enable();
if (count == 10000) {
printk(KERN_ERR "ad73311: failed to configure codec\n");
return -1;
}
return 0;
}
static int bf5xx_probe(struct platform_device *pdev)
{
int err;
if (gpio_request(GPIO_SE, "AD73311_SE")) {
printk(KERN_ERR "%s: Failed ro request GPIO_%d\n", __func__, GPIO_SE);
return -EBUSY;
}
gpio_direction_output(GPIO_SE, 0);
err = snd_ad73311_configure();
if (err < 0)
return -EFAULT;
return 0;
}
static int bf5xx_ad73311_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
pr_debug("%s enter\n", __func__);
cpu_dai->private_data = sport_handle;
return 0;
}
static int bf5xx_ad73311_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 *cpu_dai = rtd->dai->cpu_dai;
int ret = 0;
pr_debug("%s rate %d format %x\n", __func__, params_rate(params),
params_format(params));
/* set cpu DAI configuration */
ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
if (ret < 0)
return ret;
return 0;
}
static struct snd_soc_ops bf5xx_ad73311_ops = {
.startup = bf5xx_ad73311_startup,
.hw_params = bf5xx_ad73311_hw_params,
};
static struct snd_soc_dai_link bf5xx_ad73311_dai = {
.name = "ad73311",
.stream_name = "AD73311",
.cpu_dai = &bf5xx_i2s_dai,
.codec_dai = &ad73311_dai,
.ops = &bf5xx_ad73311_ops,
};
static struct snd_soc_machine bf5xx_ad73311 = {
.name = "bf5xx_ad73311",
.probe = bf5xx_probe,
.dai_link = &bf5xx_ad73311_dai,
.num_links = 1,
};
static struct snd_soc_device bf5xx_ad73311_snd_devdata = {
.machine = &bf5xx_ad73311,
.platform = &bf5xx_i2s_soc_platform,
.codec_dev = &soc_codec_dev_ad73311,
};
static struct platform_device *bf52x_ad73311_snd_device;
static int __init bf5xx_ad73311_init(void)
{
int ret;
pr_debug("%s enter\n", __func__);
bf52x_ad73311_snd_device = platform_device_alloc("soc-audio", -1);
if (!bf52x_ad73311_snd_device)
return -ENOMEM;
platform_set_drvdata(bf52x_ad73311_snd_device, &bf5xx_ad73311_snd_devdata);
bf5xx_ad73311_snd_devdata.dev = &bf52x_ad73311_snd_device->dev;
ret = platform_device_add(bf52x_ad73311_snd_device);
if (ret)
platform_device_put(bf52x_ad73311_snd_device);
return ret;
}
static void __exit bf5xx_ad73311_exit(void)
{
pr_debug("%s enter\n", __func__);
platform_device_unregister(bf52x_ad73311_snd_device);
}
module_init(bf5xx_ad73311_init);
module_exit(bf5xx_ad73311_exit);
/* Module information */
MODULE_AUTHOR("Cliff Cai");
MODULE_DESCRIPTION("ALSA SoC AD73311 Blackfin");
MODULE_LICENSE("GPL");

47
sound/soc/blackfin/bf5xx-i2s.c
View file

@ -70,6 +70,13 @@ static struct sport_param sport_params[2] = {
}
};
static u16 sport_req[][7] = {
{ P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0},
{ P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS,
P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0},
};
static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
unsigned int fmt)
{
@ -78,6 +85,14 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
/* interface format:support I2S,slave mode */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
bf5xx_i2s.tcr1 |= TFSR | TCKFE;
bf5xx_i2s.rcr1 |= RFSR | RCKFE;
bf5xx_i2s.tcr2 |= TSFSE;
bf5xx_i2s.rcr2 |= RSFSE;
break;
case SND_SOC_DAIFMT_DSP_A:
bf5xx_i2s.tcr1 |= TFSR;
bf5xx_i2s.rcr1 |= RFSR;
break;
case SND_SOC_DAIFMT_LEFT_J:
ret = -EINVAL;
@ -127,14 +142,17 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream,
case SNDRV_PCM_FORMAT_S16_LE:
bf5xx_i2s.tcr2 |= 15;
bf5xx_i2s.rcr2 |= 15;
sport_handle->wdsize = 2;
break;
case SNDRV_PCM_FORMAT_S24_LE:
bf5xx_i2s.tcr2 |= 23;
bf5xx_i2s.rcr2 |= 23;
sport_handle->wdsize = 3;
break;
case SNDRV_PCM_FORMAT_S32_LE:
bf5xx_i2s.tcr2 |= 31;
bf5xx_i2s.rcr2 |= 31;
sport_handle->wdsize = 4;
break;
}
@ -145,17 +163,17 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream,
* need to configure both of them at the time when the first
* stream is opened.
*
* CPU DAI format:I2S, slave mode.
* CPU DAI:slave mode.
*/
ret = sport_config_rx(sport_handle, RFSR | RCKFE,
RSFSE|bf5xx_i2s.rcr2, 0, 0);
ret = sport_config_rx(sport_handle, bf5xx_i2s.rcr1,
bf5xx_i2s.rcr2, 0, 0);
if (ret) {
pr_err("SPORT is busy!\n");
return -EBUSY;
}
ret = sport_config_tx(sport_handle, TFSR | TCKFE,
TSFSE|bf5xx_i2s.tcr2, 0, 0);
ret = sport_config_tx(sport_handle, bf5xx_i2s.tcr1,
bf5xx_i2s.tcr2, 0, 0);
if (ret) {
pr_err("SPORT is busy!\n");
return -EBUSY;
@ -174,13 +192,6 @@ static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream)
static int bf5xx_i2s_probe(struct platform_device *pdev,
struct snd_soc_dai *dai)
{
u16 sport_req[][7] = {
{ P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0},
{ P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS,
P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0},
};
pr_debug("%s enter\n", __func__);
if (peripheral_request_list(&sport_req[sport_num][0], "soc-audio")) {
pr_err("Requesting Peripherals failed\n");
@ -198,6 +209,13 @@ static int bf5xx_i2s_probe(struct platform_device *pdev,
return 0;
}
static void bf5xx_i2s_remove(struct platform_device *pdev,
struct snd_soc_dai *dai)
{
pr_debug("%s enter\n", __func__);
peripheral_free_list(&sport_req[sport_num][0]);
}
#ifdef CONFIG_PM
static int bf5xx_i2s_suspend(struct platform_device *dev,
struct snd_soc_dai *dai)
@ -263,15 +281,16 @@ struct snd_soc_dai bf5xx_i2s_dai = {
.id = 0,
.type = SND_SOC_DAI_I2S,
.probe = bf5xx_i2s_probe,
.remove = bf5xx_i2s_remove,
.suspend = bf5xx_i2s_suspend,
.resume = bf5xx_i2s_resume,
.playback = {
.channels_min = 2,
.channels_min = 1,
.channels_max = 2,
.rates = BF5XX_I2S_RATES,
.formats = BF5XX_I2S_FORMATS,},
.capture = {
.channels_min = 2,
.channels_min = 1,
.channels_max = 2,
.rates = BF5XX_I2S_RATES,
.formats = BF5XX_I2S_FORMATS,},

2
sound/soc/blackfin/bf5xx-sport.h
View file

@ -123,6 +123,8 @@ struct sport_device {
int rx_pos;
unsigned int tx_buffer_size;
unsigned int rx_buffer_size;
int tx_delay_pos;
int once;
#endif
void *private_data;
};

11
sound/soc/codecs/Kconfig
View file

@ -3,9 +3,11 @@ config SND_SOC_ALL_CODECS
depends on I2C
select SPI
select SPI_MASTER
select SND_SOC_AD73311
select SND_SOC_AK4535
select SND_SOC_CS4270
select SND_SOC_SSM2602
select SND_SOC_TLV320AIC23
select SND_SOC_TLV320AIC26
select SND_SOC_TLV320AIC3X
select SND_SOC_UDA1380
@ -34,6 +36,9 @@ config SND_SOC_AC97_CODEC
config SND_SOC_AD1980
tristate
config SND_SOC_AD73311
tristate
config SND_SOC_AK4535
tristate
@ -58,9 +63,13 @@ config SND_SOC_CS4270_VD33_ERRATA
config SND_SOC_SSM2602
tristate
config SND_SOC_TLV320AIC23
tristate
depends on I2C
config SND_SOC_TLV320AIC26
tristate "TI TLV320AIC26 Codec support"
depends on SND_SOC && SPI
depends on SPI
config SND_SOC_TLV320AIC3X
tristate

4
sound/soc/codecs/Makefile
View file

@ -1,8 +1,10 @@
snd-soc-ac97-objs := ac97.o
snd-soc-ad1980-objs := ad1980.o
snd-soc-ad73311-objs := ad73311.o
snd-soc-ak4535-objs := ak4535.o
snd-soc-cs4270-objs := cs4270.o
snd-soc-ssm2602-objs := ssm2602.o
snd-soc-tlv320aic23-objs := tlv320aic23.o
snd-soc-tlv320aic26-objs := tlv320aic26.o
snd-soc-tlv320aic3x-objs := tlv320aic3x.o
snd-soc-uda1380-objs := uda1380.o
@ -20,9 +22,11 @@ snd-soc-wm9713-objs := wm9713.o
obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o
obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o
obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o
obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o
obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o
obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o
obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o

3
sound/soc/codecs/ac97.c
View file

@ -2,8 +2,7 @@
* ac97.c -- ALSA Soc AC97 codec support
*
* Copyright 2005 Wolfson Microelectronics PLC.
* Author: Liam Girdwood
* liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
* Author: Liam Girdwood <lrg@slimlogic.co.uk>
*
* 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

1
sound/soc/codecs/ad1980.c
View file

@ -13,7 +13,6 @@
#include <linux/init.h>
#include <linux/module.h>
#include <linux/version.h>
#include <linux/kernel.h>
#include <linux/device.h>
#include <sound/core.h>

107
sound/soc/codecs/ad73311.c Normal file
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@ -0,0 +1,107 @@
/*
* ad73311.c -- ALSA Soc AD73311 codec support
*
* Copyright: Analog Device Inc.
* Author: Cliff Cai <cliff.cai@analog.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.
*
* Revision history
* 25th Sep 2008 Initial version.
*/
#include <linux/init.h>
#include <linux/module.h>
#include <linux/version.h>
#include <linux/kernel.h>
#include <linux/device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/ac97_codec.h>
#include <sound/initval.h>
#include <sound/soc.h>
#include "ad73311.h"
struct snd_soc_dai ad73311_dai = {
.name = "AD73311",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 1,
.rates = SNDRV_PCM_RATE_8000,
.formats = SNDRV_PCM_FMTBIT_S16_LE, },
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 1,
.rates = SNDRV_PCM_RATE_8000,
.formats = SNDRV_PCM_FMTBIT_S16_LE, },
};
EXPORT_SYMBOL_GPL(ad73311_dai);
static int ad73311_soc_probe(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec;
int ret = 0;
codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
if (codec == NULL)
return -ENOMEM;
mutex_init(&codec->mutex);
codec->name = "AD73311";
codec->owner = THIS_MODULE;
codec->dai = &ad73311_dai;
codec->num_dai = 1;
socdev->codec = codec;
INIT_LIST_HEAD(&codec->dapm_widgets);
INIT_LIST_HEAD(&codec->dapm_paths);
/* register pcms */
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0) {
printk(KERN_ERR "ad73311: failed to create pcms\n");
goto pcm_err;
}
ret = snd_soc_register_card(socdev);
if (ret < 0) {
printk(KERN_ERR "ad73311: failed to register card\n");
goto register_err;
}
return ret;
register_err:
snd_soc_free_pcms(socdev);
pcm_err:
kfree(socdev->codec);
socdev->codec = NULL;
return ret;
}
static int ad73311_soc_remove(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->codec;
if (codec == NULL)
return 0;
snd_soc_free_pcms(socdev);
kfree(codec);
return 0;
}
struct snd_soc_codec_device soc_codec_dev_ad73311 = {
.probe = ad73311_soc_probe,
.remove = ad73311_soc_remove,
};
EXPORT_SYMBOL_GPL(soc_codec_dev_ad73311);
MODULE_DESCRIPTION("ASoC ad73311 driver");
MODULE_AUTHOR("Cliff Cai ");
MODULE_LICENSE("GPL");

90
sound/soc/codecs/ad73311.h Normal file
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/*
* File: sound/soc/codec/ad73311.h
* Based on:
* Author: Cliff Cai <cliff.cai@analog.com>
*
* Created: Thur Sep 25, 2008
* Description: definitions for AD73311 registers
*
*
* Modified:
* Copyright 2006 Analog Devices Inc.
*
* Bugs: Enter bugs at http://blackfin.uclinux.org/
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, see the file COPYING, or write
* to the Free Software Foundation, Inc.,
* 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef __AD73311_H__
#define __AD73311_H__
#define AD_CONTROL 0x8000
#define AD_DATA 0x0000
#define AD_READ 0x4000
#define AD_WRITE 0x0000
/* Control register A */
#define CTRL_REG_A (0 << 8)
#define REGA_MODE_PRO 0x00
#define REGA_MODE_DATA 0x01
#define REGA_MODE_MIXED 0x03
#define REGA_DLB 0x04
#define REGA_SLB 0x08
#define REGA_DEVC(x) ((x & 0x7) << 4)
#define REGA_RESET 0x80
/* Control register B */
#define CTRL_REG_B (1 << 8)
#define REGB_DIRATE(x) (x & 0x3)
#define REGB_SCDIV(x) ((x & 0x3) << 2)
#define REGB_MCDIV(x) ((x & 0x7) << 4)
#define REGB_CEE (1 << 7)
/* Control register C */
#define CTRL_REG_C (2 << 8)
#define REGC_PUDEV (1 << 0)
#define REGC_PUADC (1 << 3)
#define REGC_PUDAC (1 << 4)
#define REGC_PUREF (1 << 5)
#define REGC_REFUSE (1 << 6)
/* Control register D */
#define CTRL_REG_D (3 << 8)
#define REGD_IGS(x) (x & 0x7)
#define REGD_RMOD (1 << 3)
#define REGD_OGS(x) ((x & 0x7) << 4)
#define REGD_MUTE (x << 7)
/* Control register E */
#define CTRL_REG_E (4 << 8)
#define REGE_DA(x) (x & 0x1f)
#define REGE_IBYP (1 << 5)
/* Control register F */
#define CTRL_REG_F (5 << 8)
#define REGF_SEEN (1 << 5)
#define REGF_INV (1 << 6)
#define REGF_ALB (1 << 7)
extern struct snd_soc_dai ad73311_dai;
extern struct snd_soc_codec_device soc_codec_dev_ad73311;
#endif

1
sound/soc/codecs/ak4535.c
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@ -28,7 +28,6 @@
#include "ak4535.h"
#define AUDIO_NAME "ak4535"
#define AK4535_VERSION "0.3"
struct snd_soc_codec_device soc_codec_dev_ak4535;

1
sound/soc/codecs/ssm2602.c
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@ -42,7 +42,6 @@
#include "ssm2602.h"
#define AUDIO_NAME "ssm2602"
#define SSM2602_VERSION "0.1"
struct snd_soc_codec_device soc_codec_dev_ssm2602;

714
sound/soc/codecs/tlv320aic23.c Normal file
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/*
* ALSA SoC TLV320AIC23 codec driver
*
* Author: Arun KS, <arunks@mistralsolutions.com>
* Copyright: (C) 2008 Mistral Solutions Pvt Ltd.,
*
* Based on sound/soc/codecs/wm8731.c by Richard Purdie
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*
* Notes:
* The AIC23 is a driver for a low power stereo audio
* codec tlv320aic23
*
* The machine layer should disable unsupported inputs/outputs by
* snd_soc_dapm_disable_pin(codec, "LHPOUT"), etc.
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/tlv.h>
#include <sound/initval.h>
#include "tlv320aic23.h"
#define AIC23_VERSION "0.1"
struct tlv320aic23_srate_reg_info {
u32 sample_rate;
u8 control; /* SR3, SR2, SR1, SR0 and BOSR */
u8 divider; /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */
};
/*
* AIC23 register cache
*/
static const u16 tlv320aic23_reg[] = {
0x0097, 0x0097, 0x00F9, 0x00F9, /* 0 */
0x001A, 0x0004, 0x0007, 0x0001, /* 4 */
0x0020, 0x0000, 0x0000, 0x0000, /* 8 */
0x0000, 0x0000, 0x0000, 0x0000, /* 12 */
};
/*
* read tlv320aic23 register cache
*/
static inline unsigned int tlv320aic23_read_reg_cache(struct snd_soc_codec
*codec, unsigned int reg)
{
u16 *cache = codec->reg_cache;
if (reg >= ARRAY_SIZE(tlv320aic23_reg))
return -1;
return cache[reg];
}
/*
* write tlv320aic23 register cache
*/
static inline void tlv320aic23_write_reg_cache(struct snd_soc_codec *codec,
u8 reg, u16 value)
{
u16 *cache = codec->reg_cache;
if (reg >= ARRAY_SIZE(tlv320aic23_reg))
return;
cache[reg] = value;
}
/*
* write to the tlv320aic23 register space
*/
static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int value)
{
u8 data;
/* TLV320AIC23 has 7 bit address and 9 bits of data
* so we need to switch one data bit into reg and rest
* of data into val
*/
if ((reg < 0 || reg > 9) && (reg != 15)) {
printk(KERN_WARNING "%s Invalid register R%d\n", __func__, reg);
return -1;
}
data = (reg << 1) | (value >> 8 & 0x01);
tlv320aic23_write_reg_cache(codec, reg, value);
if (codec->hw_write(codec->control_data, data,
(value & 0xff)) == 0)
return 0;
printk(KERN_ERR "%s cannot write %03x to register R%d\n", __func__,
value, reg);
return -EIO;
}
static const char *rec_src_text[] = { "Line", "Mic" };
static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"};
static const struct soc_enum rec_src_enum =
SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text);
static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls =
SOC_DAPM_ENUM("Input Select", rec_src_enum);
static const struct soc_enum tlv320aic23_rec_src =
SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text);
static const struct soc_enum tlv320aic23_deemph =
SOC_ENUM_SINGLE(TLV320AIC23_DIGT, 1, 4, deemph_text);
static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -12100, 100, 0);
static const DECLARE_TLV_DB_SCALE(input_gain_tlv, -1725, 75, 0);
static const DECLARE_TLV_DB_SCALE(sidetone_vol_tlv, -1800, 300, 0);
static int snd_soc_tlv320aic23_put_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
u16 val, reg;
val = (ucontrol->value.integer.value[0] & 0x07);
/* linear conversion to userspace
* 000 = -6db
* 001 = -9db
* 010 = -12db
* 011 = -18db (Min)
* 100 = 0db (Max)
*/
val = (val >= 4) ? 4 : (3 - val);
reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (~0x1C0);
tlv320aic23_write(codec, TLV320AIC23_ANLG, reg | (val << 6));
return 0;
}
static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
u16 val;
val = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (0x1C0);
val = val >> 6;
val = (val >= 4) ? 4 : (3 - val);
ucontrol->value.integer.value[0] = val;
return 0;
}
#define SOC_TLV320AIC23_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
SNDRV_CTL_ELEM_ACCESS_READWRITE,\
.tlv.p = (tlv_array), \
.info = snd_soc_info_volsw, .get = snd_soc_tlv320aic23_get_volsw,\
.put = snd_soc_tlv320aic23_put_volsw, \
.private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) }
static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = {
SOC_DOUBLE_R_TLV("Digital Playback Volume", TLV320AIC23_LCHNVOL,
TLV320AIC23_RCHNVOL, 0, 127, 0, out_gain_tlv),
SOC_SINGLE("Digital Playback Switch", TLV320AIC23_DIGT, 3, 1, 1),
SOC_DOUBLE_R("Line Input Switch", TLV320AIC23_LINVOL,
TLV320AIC23_RINVOL, 7, 1, 0),
SOC_DOUBLE_R_TLV("Line Input Volume", TLV320AIC23_LINVOL,
TLV320AIC23_RINVOL, 0, 31, 0, input_gain_tlv),
SOC_SINGLE("Mic Input Switch", TLV320AIC23_ANLG, 1, 1, 1),
SOC_SINGLE("Mic Booster Switch", TLV320AIC23_ANLG, 0, 1, 0),
SOC_TLV320AIC23_SINGLE_TLV("Sidetone Volume", TLV320AIC23_ANLG,
6, 4, 0, sidetone_vol_tlv),
SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph),
};
/* add non dapm controls */
static int tlv320aic23_add_controls(struct snd_soc_codec *codec)
{
int err, i;
for (i = 0; i < ARRAY_SIZE(tlv320aic23_snd_controls); i++) {
err = snd_ctl_add(codec->card,
snd_soc_cnew(&tlv320aic23_snd_controls[i],
codec, NULL));
if (err < 0)
return err;
}
return 0;
}
/* PGA Mixer controls for Line and Mic switch */
static const struct snd_kcontrol_new tlv320aic23_output_mixer_controls[] = {
SOC_DAPM_SINGLE("Line Bypass Switch", TLV320AIC23_ANLG, 3, 1, 0),
SOC_DAPM_SINGLE("Mic Sidetone Switch", TLV320AIC23_ANLG, 5, 1, 0),
SOC_DAPM_SINGLE("Playback Switch", TLV320AIC23_ANLG, 4, 1, 0),
};
static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
SND_SOC_DAPM_DAC("DAC", "Playback", TLV320AIC23_PWR, 3, 1),
SND_SOC_DAPM_ADC("ADC", "Capture", TLV320AIC23_PWR, 2, 1),
SND_SOC_DAPM_MUX("Capture Source", SND_SOC_NOPM, 0, 0,
&tlv320aic23_rec_src_mux_controls),
SND_SOC_DAPM_MIXER("Output Mixer", TLV320AIC23_PWR, 4, 1,
&tlv320aic23_output_mixer_controls[0],
ARRAY_SIZE(tlv320aic23_output_mixer_controls)),
SND_SOC_DAPM_PGA("Line Input", TLV320AIC23_PWR, 0, 1, NULL, 0),
SND_SOC_DAPM_PGA("Mic Input", TLV320AIC23_PWR, 1, 1, NULL, 0),
SND_SOC_DAPM_OUTPUT("LHPOUT"),
SND_SOC_DAPM_OUTPUT("RHPOUT"),
SND_SOC_DAPM_OUTPUT("LOUT"),
SND_SOC_DAPM_OUTPUT("ROUT"),
SND_SOC_DAPM_INPUT("LLINEIN"),
SND_SOC_DAPM_INPUT("RLINEIN"),
SND_SOC_DAPM_INPUT("MICIN"),
};
static const struct snd_soc_dapm_route intercon[] = {
/* Output Mixer */
{"Output Mixer", "Line Bypass Switch", "Line Input"},
{"Output Mixer", "Playback Switch", "DAC"},
{"Output Mixer", "Mic Sidetone Switch", "Mic Input"},
/* Outputs */
{"RHPOUT", NULL, "Output Mixer"},
{"LHPOUT", NULL, "Output Mixer"},
{"LOUT", NULL, "Output Mixer"},
{"ROUT", NULL, "Output Mixer"},
/* Inputs */
{"Line Input", "NULL", "LLINEIN"},
{"Line Input", "NULL", "RLINEIN"},
{"Mic Input", "NULL", "MICIN"},
/* input mux */
{"Capture Source", "Line", "Line Input"},
{"Capture Source", "Mic", "Mic Input"},
{"ADC", NULL, "Capture Source"},
};
/* tlv320aic23 related */
static const struct tlv320aic23_srate_reg_info srate_reg_info[] = {
{4000, 0x06, 1}, /* 4000 */
{8000, 0x06, 0}, /* 8000 */
{16000, 0x0C, 1}, /* 16000 */
{22050, 0x11, 1}, /* 22050 */
{24000, 0x00, 1}, /* 24000 */
{32000, 0x0C, 0}, /* 32000 */
{44100, 0x11, 0}, /* 44100 */
{48000, 0x00, 0}, /* 48000 */
{88200, 0x1F, 0}, /* 88200 */
{96000, 0x0E, 0}, /* 96000 */
};
static int tlv320aic23_add_widgets(struct snd_soc_codec *codec)
{
snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
ARRAY_SIZE(tlv320aic23_dapm_widgets));
/* set up audio path interconnects */
snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
snd_soc_dapm_new_widgets(codec);
return 0;
}
static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->codec;
u16 iface_reg, data;
u8 count = 0;
iface_reg =
tlv320aic23_read_reg_cache(codec,
TLV320AIC23_DIGT_FMT) & ~(0x03 << 2);
/* Search for the right sample rate */
/* Verify what happens if the rate is not supported
* now it goes to 96Khz */
while ((srate_reg_info[count].sample_rate != params_rate(params)) &&
(count < ARRAY_SIZE(srate_reg_info))) {
count++;
}
data = (srate_reg_info[count].divider << TLV320AIC23_CLKIN_SHIFT) |
(srate_reg_info[count]. control << TLV320AIC23_BOSR_SHIFT) |
TLV320AIC23_USB_CLK_ON;
tlv320aic23_write(codec, TLV320AIC23_SRATE, data);
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
break;
case SNDRV_PCM_FORMAT_S20_3LE:
iface_reg |= (0x01 << 2);
break;
case SNDRV_PCM_FORMAT_S24_LE:
iface_reg |= (0x02 << 2);
break;
case SNDRV_PCM_FORMAT_S32_LE:
iface_reg |= (0x03 << 2);
break;
}
tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
return 0;
}
static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->codec;
/* set active */
tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0001);
return 0;
}
static void tlv320aic23_shutdown(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->codec;
/* deactivate */
if (!codec->active) {
udelay(50);
tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
}
}
static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
u16 reg;
reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT);
if (mute)
reg |= TLV320AIC23_DACM_MUTE;
else
reg &= ~TLV320AIC23_DACM_MUTE;
tlv320aic23_write(codec, TLV320AIC23_DIGT, reg);
return 0;
}
static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
u16 iface_reg;
iface_reg =
tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT_FMT) & (~0x03);
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
iface_reg |= TLV320AIC23_MS_MASTER;
break;
case SND_SOC_DAIFMT_CBS_CFS:
break;
default:
return -EINVAL;
}
/* interface format */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
iface_reg |= TLV320AIC23_FOR_I2S;
break;
case SND_SOC_DAIFMT_DSP_A:
iface_reg |= TLV320AIC23_FOR_DSP;
break;
case SND_SOC_DAIFMT_RIGHT_J:
break;
case SND_SOC_DAIFMT_LEFT_J:
iface_reg |= TLV320AIC23_FOR_LJUST;
break;
default:
return -EINVAL;
}
tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
return 0;
}
static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_codec *codec = codec_dai->codec;
switch (freq) {
case 12000000:
return 0;
}
return -EINVAL;
}
static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
u16 reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_PWR) & 0xff7f;
switch (level) {
case SND_SOC_BIAS_ON:
/* vref/mid, osc on, dac unmute */
tlv320aic23_write(codec, TLV320AIC23_PWR, reg);
break;
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
/* everything off except vref/vmid, */
tlv320aic23_write(codec, TLV320AIC23_PWR, reg | 0x0040);
break;
case SND_SOC_BIAS_OFF:
/* everything off, dac mute, inactive */
tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff);
break;
}
codec->bias_level = level;
return 0;
}
#define AIC23_RATES SNDRV_PCM_RATE_8000_96000
#define AIC23_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
struct snd_soc_dai tlv320aic23_dai = {
.name = "tlv320aic23",
.playback = {
.stream_name = "Playback",
.channels_min = 2,
.channels_max = 2,
.rates = AIC23_RATES,
.formats = AIC23_FORMATS,},
.capture = {
.stream_name = "Capture",
.channels_min = 2,
.channels_max = 2,
.rates = AIC23_RATES,
.formats = AIC23_FORMATS,},
.ops = {
.prepare = tlv320aic23_pcm_prepare,
.hw_params = tlv320aic23_hw_params,
.shutdown = tlv320aic23_shutdown,
},
.dai_ops = {
.digital_mute = tlv320aic23_mute,
.set_fmt = tlv320aic23_set_dai_fmt,
.set_sysclk = tlv320aic23_set_dai_sysclk,
}
};
EXPORT_SYMBOL_GPL(tlv320aic23_dai);
static int tlv320aic23_suspend(struct platform_device *pdev,
pm_message_t state)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->codec;
tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static int tlv320aic23_resume(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->codec;
int i;
u16 reg;
/* Sync reg_cache with the hardware */
for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) {
u16 val = tlv320aic23_read_reg_cache(codec, reg);
tlv320aic23_write(codec, reg, val);
}
tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
tlv320aic23_set_bias_level(codec, codec->suspend_bias_level);
return 0;
}
/*
* initialise the AIC23 driver
* register the mixer and dsp interfaces with the kernel
*/
static int tlv320aic23_init(struct snd_soc_device *socdev)
{
struct snd_soc_codec *codec = socdev->codec;
int ret = 0;
u16 reg;
codec->name = "tlv320aic23";
codec->owner = THIS_MODULE;
codec->read = tlv320aic23_read_reg_cache;
codec->write = tlv320aic23_write;
codec->set_bias_level = tlv320aic23_set_bias_level;
codec->dai = &tlv320aic23_dai;
codec->num_dai = 1;
codec->reg_cache_size = ARRAY_SIZE(tlv320aic23_reg);
codec->reg_cache =
kmemdup(tlv320aic23_reg, sizeof(tlv320aic23_reg), GFP_KERNEL);
if (codec->reg_cache == NULL)
return -ENOMEM;
/* Reset codec */
tlv320aic23_write(codec, TLV320AIC23_RESET, 0);
/* register pcms */
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0) {
printk(KERN_ERR "tlv320aic23: failed to create pcms\n");
goto pcm_err;
}
/* power on device */
tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
tlv320aic23_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K);
/* Unmute input */
reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_LINVOL);
tlv320aic23_write(codec, TLV320AIC23_LINVOL,
(reg & (~TLV320AIC23_LIM_MUTED)) |
(TLV320AIC23_LRS_ENABLED));
reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_RINVOL);
tlv320aic23_write(codec, TLV320AIC23_RINVOL,
(reg & (~TLV320AIC23_LIM_MUTED)) |
TLV320AIC23_LRS_ENABLED);
reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG);
tlv320aic23_write(codec, TLV320AIC23_ANLG,
(reg) & (~TLV320AIC23_BYPASS_ON) &
(~TLV320AIC23_MICM_MUTED));
/* Default output volume */
tlv320aic23_write(codec, TLV320AIC23_LCHNVOL,
TLV320AIC23_DEFAULT_OUT_VOL &
TLV320AIC23_OUT_VOL_MASK);
tlv320aic23_write(codec, TLV320AIC23_RCHNVOL,
TLV320AIC23_DEFAULT_OUT_VOL &
TLV320AIC23_OUT_VOL_MASK);
tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1);
tlv320aic23_add_controls(codec);
tlv320aic23_add_widgets(codec);
ret = snd_soc_register_card(socdev);
if (ret < 0) {
printk(KERN_ERR "tlv320aic23: failed to register card\n");
goto card_err;
}
return ret;
card_err:
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
pcm_err:
kfree(codec->reg_cache);
return ret;
}
static struct snd_soc_device *tlv320aic23_socdev;
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
/*
* If the i2c layer weren't so broken, we could pass this kind of data
* around
*/
static int tlv320aic23_codec_probe(struct i2c_client *i2c,
const struct i2c_device_id *i2c_id)
{
struct snd_soc_device *socdev = tlv320aic23_socdev;
struct snd_soc_codec *codec = socdev->codec;
int ret;
if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA))
return -EINVAL;
i2c_set_clientdata(i2c, codec);
codec->control_data = i2c;
ret = tlv320aic23_init(socdev);
if (ret < 0) {
printk(KERN_ERR "tlv320aic23: failed to initialise AIC23\n");
goto err;
}
return ret;
err:
kfree(codec);
kfree(i2c);
return ret;
}
static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c)
{
put_device(&i2c->dev);
return 0;
}
static const struct i2c_device_id tlv320aic23_id[] = {
{"tlv320aic23", 0},
{}
};
MODULE_DEVICE_TABLE(i2c, tlv320aic23_id);
static struct i2c_driver tlv320aic23_i2c_driver = {
.driver = {
.name = "tlv320aic23",
},
.probe = tlv320aic23_codec_probe,
.remove = __exit_p(tlv320aic23_i2c_remove),
.id_table = tlv320aic23_id,
};
#endif
static int tlv320aic23_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 "AIC23 Audio Codec %s\n", AIC23_VERSION);
codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
if (codec == NULL)
return -ENOMEM;
socdev->codec = codec;
mutex_init(&codec->mutex);
INIT_LIST_HEAD(&codec->dapm_widgets);
INIT_LIST_HEAD(&codec->dapm_paths);
tlv320aic23_socdev = socdev;
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
codec->hw_write = (hw_write_t) i2c_smbus_write_byte_data;
codec->hw_read = NULL;
ret = i2c_add_driver(&tlv320aic23_i2c_driver);
if (ret != 0)
printk(KERN_ERR "can't add i2c driver");
#endif
return ret;
}
static int tlv320aic23_remove(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->codec;
if (codec->control_data)
tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
i2c_del_driver(&tlv320aic23_i2c_driver);
#endif
kfree(codec->reg_cache);
kfree(codec);
return 0;
}
struct snd_soc_codec_device soc_codec_dev_tlv320aic23 = {
.probe = tlv320aic23_probe,
.remove = tlv320aic23_remove,
.suspend = tlv320aic23_suspend,
.resume = tlv320aic23_resume,
};
EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320aic23);
MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver");
MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
MODULE_LICENSE("GPL");

122
sound/soc/codecs/tlv320aic23.h Normal file
View file

@ -0,0 +1,122 @@
/*
* ALSA SoC TLV320AIC23 codec driver
*
* Author: Arun KS, <arunks@mistralsolutions.com>
* Copyright: (C) 2008 Mistral Solutions Pvt Ltd
*
* 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 _TLV320AIC23_H
#define _TLV320AIC23_H
/* Codec TLV320AIC23 */
#define TLV320AIC23_LINVOL 0x00
#define TLV320AIC23_RINVOL 0x01
#define TLV320AIC23_LCHNVOL 0x02
#define TLV320AIC23_RCHNVOL 0x03
#define TLV320AIC23_ANLG 0x04
#define TLV320AIC23_DIGT 0x05
#define TLV320AIC23_PWR 0x06
#define TLV320AIC23_DIGT_FMT 0x07
#define TLV320AIC23_SRATE 0x08
#define TLV320AIC23_ACTIVE 0x09
#define TLV320AIC23_RESET 0x0F
/* Left (right) line input volume control register */
#define TLV320AIC23_LRS_ENABLED 0x0100
#define TLV320AIC23_LIM_MUTED 0x0080
#define TLV320AIC23_LIV_DEFAULT 0x0017
#define TLV320AIC23_LIV_MAX 0x001f
#define TLV320AIC23_LIV_MIN 0x0000
/* Left (right) channel headphone volume control register */
#define TLV320AIC23_LZC_ON 0x0080
#define TLV320AIC23_LHV_DEFAULT 0x0079
#define TLV320AIC23_LHV_MAX 0x007f
#define TLV320AIC23_LHV_MIN 0x0000
/* Analog audio path control register */
#define TLV320AIC23_STA_REG(x) ((x)<<6)
#define TLV320AIC23_STE_ENABLED 0x0020
#define TLV320AIC23_DAC_SELECTED 0x0010
#define TLV320AIC23_BYPASS_ON 0x0008
#define TLV320AIC23_INSEL_MIC 0x0004
#define TLV320AIC23_MICM_MUTED 0x0002
#define TLV320AIC23_MICB_20DB 0x0001
/* Digital audio path control register */
#define TLV320AIC23_DACM_MUTE 0x0008
#define TLV320AIC23_DEEMP_32K 0x0002
#define TLV320AIC23_DEEMP_44K 0x0004
#define TLV320AIC23_DEEMP_48K 0x0006
#define TLV320AIC23_ADCHP_ON 0x0001
/* Power control down register */
#define TLV320AIC23_DEVICE_PWR_OFF 0x0080
#define TLV320AIC23_CLK_OFF 0x0040
#define TLV320AIC23_OSC_OFF 0x0020
#define TLV320AIC23_OUT_OFF 0x0010
#define TLV320AIC23_DAC_OFF 0x0008
#define TLV320AIC23_ADC_OFF 0x0004
#define TLV320AIC23_MIC_OFF 0x0002
#define TLV320AIC23_LINE_OFF 0x0001
/* Digital audio interface register */
#define TLV320AIC23_MS_MASTER 0x0040
#define TLV320AIC23_LRSWAP_ON 0x0020
#define TLV320AIC23_LRP_ON 0x0010
#define TLV320AIC23_IWL_16 0x0000
#define TLV320AIC23_IWL_20 0x0004
#define TLV320AIC23_IWL_24 0x0008
#define TLV320AIC23_IWL_32 0x000C
#define TLV320AIC23_FOR_I2S 0x0002
#define TLV320AIC23_FOR_DSP 0x0003
#define TLV320AIC23_FOR_LJUST 0x0001
/* Sample rate control register */
#define TLV320AIC23_CLKOUT_HALF 0x0080
#define TLV320AIC23_CLKIN_HALF 0x0040
#define TLV320AIC23_BOSR_384fs 0x0002 /* BOSR_272fs in USB mode */
#define TLV320AIC23_USB_CLK_ON 0x0001
#define TLV320AIC23_SR_MASK 0xf
#define TLV320AIC23_CLKOUT_SHIFT 7
#define TLV320AIC23_CLKIN_SHIFT 6
#define TLV320AIC23_SR_SHIFT 2
#define TLV320AIC23_BOSR_SHIFT 1
/* Digital interface register */
#define TLV320AIC23_ACT_ON 0x0001
/*
* AUDIO related MACROS
*/
#define TLV320AIC23_DEFAULT_OUT_VOL 0x70
#define TLV320AIC23_DEFAULT_IN_VOLUME 0x10
#define TLV320AIC23_OUT_VOL_MIN TLV320AIC23_LHV_MIN
#define TLV320AIC23_OUT_VOL_MAX TLV320AIC23_LHV_MAX
#define TLV320AIC23_OUT_VO_RANGE (TLV320AIC23_OUT_VOL_MAX - \
TLV320AIC23_OUT_VOL_MIN)
#define TLV320AIC23_OUT_VOL_MASK TLV320AIC23_OUT_VOL_MAX
#define TLV320AIC23_IN_VOL_MIN TLV320AIC23_LIV_MIN
#define TLV320AIC23_IN_VOL_MAX TLV320AIC23_LIV_MAX
#define TLV320AIC23_IN_VOL_RANGE (TLV320AIC23_IN_VOL_MAX - \
TLV320AIC23_IN_VOL_MIN)
#define TLV320AIC23_IN_VOL_MASK TLV320AIC23_IN_VOL_MAX
#define TLV320AIC23_SIDETONE_MASK 0x1c0
#define TLV320AIC23_SIDETONE_0 0x100
#define TLV320AIC23_SIDETONE_6 0x000
#define TLV320AIC23_SIDETONE_9 0x040
#define TLV320AIC23_SIDETONE_12 0x080
#define TLV320AIC23_SIDETONE_18 0x0c0
extern struct snd_soc_dai tlv320aic23_dai;
extern struct snd_soc_codec_device soc_codec_dev_tlv320aic23;
#endif /* _TLV320AIC23_H */

5
sound/soc/codecs/tlv320aic3x.c
View file

@ -48,7 +48,6 @@
#include "tlv320aic3x.h"
#define AUDIO_NAME "aic3x"
#define AIC3X_VERSION "0.2"
/* codec private data */
@ -991,7 +990,7 @@ EXPORT_SYMBOL_GPL(aic3x_headset_detected);
SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
struct snd_soc_dai aic3x_dai = {
.name = "aic3x",
.name = "tlv320aic3x",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
@ -1055,7 +1054,7 @@ static int aic3x_init(struct snd_soc_device *socdev)
struct aic3x_setup_data *setup = socdev->codec_data;
int reg, ret = 0;
codec->name = "aic3x";
codec->name = "tlv320aic3x";
codec->owner = THIS_MODULE;
codec->read = aic3x_read_reg_cache;
codec->write = aic3x_write;

1
sound/soc/codecs/uda1380.c
View file

@ -36,7 +36,6 @@
#include "uda1380.h"
#define UDA1380_VERSION "0.6"
#define AUDIO_NAME "uda1380"
/*
* uda1380 register cache

113
sound/soc/codecs/wm8510.c
View file

@ -3,7 +3,7 @@
*
* Copyright 2006 Wolfson Microelectronics PLC.
*
* Author: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
* Author: Liam Girdwood <lrg@slimlogic.co.uk>
*
* 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
@ -18,6 +18,7 @@
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <linux/spi/spi.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@ -27,7 +28,6 @@
#include "wm8510.h"
#define AUDIO_NAME "wm8510"
#define WM8510_VERSION "0.6"
struct snd_soc_codec_device soc_codec_dev_wm8510;
@ -55,6 +55,9 @@ static const u16 wm8510_reg[WM8510_CACHEREGNUM] = {
0x0001,
};
#define WM8510_POWER1_BIASEN 0x08
#define WM8510_POWER1_BUFIOEN 0x10
/*
* read wm8510 register cache
*/
@ -224,9 +227,9 @@ SND_SOC_DAPM_PGA("SpkN Out", WM8510_POWER3, 5, 0, NULL, 0),
SND_SOC_DAPM_PGA("SpkP Out", WM8510_POWER3, 6, 0, NULL, 0),
SND_SOC_DAPM_PGA("Mono Out", WM8510_POWER3, 7, 0, NULL, 0),
SND_SOC_DAPM_PGA("Mic PGA", WM8510_POWER2, 2, 0,
&wm8510_micpga_controls[0],
ARRAY_SIZE(wm8510_micpga_controls)),
SND_SOC_DAPM_MIXER("Mic PGA", WM8510_POWER2, 2, 0,
&wm8510_micpga_controls[0],
ARRAY_SIZE(wm8510_micpga_controls)),
SND_SOC_DAPM_MIXER("Boost Mixer", WM8510_POWER2, 4, 0,
&wm8510_boost_controls[0],
ARRAY_SIZE(wm8510_boost_controls)),
@ -526,23 +529,35 @@ static int wm8510_mute(struct snd_soc_dai *dai, int mute)
static int wm8510_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
u16 power1 = wm8510_read_reg_cache(codec, WM8510_POWER1) & ~0x3;
switch (level) {
case SND_SOC_BIAS_ON:
wm8510_write(codec, WM8510_POWER1, 0x1ff);
wm8510_write(codec, WM8510_POWER2, 0x1ff);
wm8510_write(codec, WM8510_POWER3, 0x1ff);
break;
case SND_SOC_BIAS_PREPARE:
case SND_SOC_BIAS_STANDBY:
power1 |= 0x1; /* VMID 50k */
wm8510_write(codec, WM8510_POWER1, power1);
break;
case SND_SOC_BIAS_STANDBY:
power1 |= WM8510_POWER1_BIASEN | WM8510_POWER1_BUFIOEN;
if (codec->bias_level == SND_SOC_BIAS_OFF) {
/* Initial cap charge at VMID 5k */
wm8510_write(codec, WM8510_POWER1, power1 | 0x3);
mdelay(100);
}
power1 |= 0x2; /* VMID 500k */
wm8510_write(codec, WM8510_POWER1, power1);
break;
case SND_SOC_BIAS_OFF:
/* everything off, dac mute, inactive */
wm8510_write(codec, WM8510_POWER1, 0x0);
wm8510_write(codec, WM8510_POWER2, 0x0);
wm8510_write(codec, WM8510_POWER3, 0x0);
wm8510_write(codec, WM8510_POWER1, 0);
wm8510_write(codec, WM8510_POWER2, 0);
wm8510_write(codec, WM8510_POWER3, 0);
break;
}
codec->bias_level = level;
return 0;
}
@ -640,6 +655,7 @@ static int wm8510_init(struct snd_soc_device *socdev)
}
/* power on device */
codec->bias_level = SND_SOC_BIAS_OFF;
wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
wm8510_add_controls(codec);
wm8510_add_widgets(codec);
@ -747,6 +763,62 @@ err_driver:
}
#endif
#if defined(CONFIG_SPI_MASTER)
static int __devinit wm8510_spi_probe(struct spi_device *spi)
{
struct snd_soc_device *socdev = wm8510_socdev;
struct snd_soc_codec *codec = socdev->codec;
int ret;
codec->control_data = spi;
ret = wm8510_init(socdev);
if (ret < 0)
dev_err(&spi->dev, "failed to initialise WM8510\n");
return ret;
}
static int __devexit wm8510_spi_remove(struct spi_device *spi)
{
return 0;
}
static struct spi_driver wm8510_spi_driver = {
.driver = {
.name = "wm8510",
.bus = &spi_bus_type,
.owner = THIS_MODULE,
},
.probe = wm8510_spi_probe,
.remove = __devexit_p(wm8510_spi_remove),
};
static int wm8510_spi_write(struct spi_device *spi, const char *data, int len)
{
struct spi_transfer t;
struct spi_message m;
u8 msg[2];
if (len <= 0)
return 0;
msg[0] = data[0];
msg[1] = data[1];
spi_message_init(&m);
memset(&t, 0, (sizeof t));
t.tx_buf = &msg[0];
t.len = len;
spi_message_add_tail(&t, &m);
spi_sync(spi, &m);
return len;
}
#endif /* CONFIG_SPI_MASTER */
static int wm8510_probe(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
@ -772,8 +844,14 @@ static int wm8510_probe(struct platform_device *pdev)
codec->hw_write = (hw_write_t)i2c_master_send;
ret = wm8510_add_i2c_device(pdev, setup);
}
#else
/* Add other interfaces here */
#endif
#if defined(CONFIG_SPI_MASTER)
if (setup->spi) {
codec->hw_write = (hw_write_t)wm8510_spi_write;
ret = spi_register_driver(&wm8510_spi_driver);
if (ret != 0)
printk(KERN_ERR "can't add spi driver");
}
#endif
if (ret != 0)
@ -795,6 +873,9 @@ static int wm8510_remove(struct platform_device *pdev)
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
i2c_unregister_device(codec->control_data);
i2c_del_driver(&wm8510_i2c_driver);
#endif
#if defined(CONFIG_SPI_MASTER)
spi_unregister_driver(&wm8510_spi_driver);
#endif
kfree(codec);

1
sound/soc/codecs/wm8510.h
View file

@ -94,6 +94,7 @@
#define WM8510_MCLKDIV_12 (7 << 5)
struct wm8510_setup_data {
int spi;
int i2c_bus;
unsigned short i2c_address;
};

2
sound/soc/codecs/wm8580.c
View file

@ -18,7 +18,6 @@
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/version.h>
#include <linux/kernel.h>
#include <linux/init.h>
#include <linux/delay.h>
@ -36,7 +35,6 @@
#include "wm8580.h"
#define AUDIO_NAME "wm8580"
#define WM8580_VERSION "0.1"
struct pll_state {

1
sound/soc/codecs/wm8731.c
View file

@ -29,7 +29,6 @@
#include "wm8731.h"
#define AUDIO_NAME "wm8731"
#define WM8731_VERSION "0.13"
struct snd_soc_codec_device soc_codec_dev_wm8731;

1
sound/soc/codecs/wm8750.c
View file

@ -29,7 +29,6 @@
#include "wm8750.h"
#define AUDIO_NAME "WM8750"
#define WM8750_VERSION "0.12"
/* codec private data */

75
sound/soc/codecs/wm8753.c
View file

@ -2,8 +2,7 @@
* wm8753.c -- WM8753 ALSA Soc Audio driver
*
* Copyright 2003 Wolfson Microelectronics PLC.
* Author: Liam Girdwood
* liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
* Author: Liam Girdwood <lrg@slimlogic.co.uk>
*
* 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
@ -40,6 +39,7 @@
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <linux/spi/spi.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@ -51,7 +51,6 @@
#include "wm8753.h"
#define AUDIO_NAME "wm8753"
#define WM8753_VERSION "0.16"
static int caps_charge = 2000;
@ -1719,6 +1718,63 @@ err_driver:
}
#endif
#if defined(CONFIG_SPI_MASTER)
static int __devinit wm8753_spi_probe(struct spi_device *spi)
{
struct snd_soc_device *socdev = wm8753_socdev;
struct snd_soc_codec *codec = socdev->codec;
int ret;
codec->control_data = spi;
ret = wm8753_init(socdev);
if (ret < 0)
dev_err(&spi->dev, "failed to initialise WM8753\n");
return ret;
}
static int __devexit wm8753_spi_remove(struct spi_device *spi)
{
return 0;
}
static struct spi_driver wm8753_spi_driver = {
.driver = {
.name = "wm8753",
.bus = &spi_bus_type,
.owner = THIS_MODULE,
},
.probe = wm8753_spi_probe,
.remove = __devexit_p(wm8753_spi_remove),
};
static int wm8753_spi_write(struct spi_device *spi, const char *data, int len)
{
struct spi_transfer t;
struct spi_message m;
u8 msg[2];
if (len <= 0)
return 0;
msg[0] = data[0];
msg[1] = data[1];
spi_message_init(&m);
memset(&t, 0, (sizeof t));
t.tx_buf = &msg[0];
t.len = len;
spi_message_add_tail(&t, &m);
spi_sync(spi, &m);
return len;
}
#endif
static int wm8753_probe(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
@ -1753,8 +1809,14 @@ static int wm8753_probe(struct platform_device *pdev)
codec->hw_write = (hw_write_t)i2c_master_send;
ret = wm8753_add_i2c_device(pdev, setup);
}
#else
/* Add other interfaces here */
#endif
#if defined(CONFIG_SPI_MASTER)
if (setup->spi) {
codec->hw_write = (hw_write_t)wm8753_spi_write;
ret = spi_register_driver(&wm8753_spi_driver);
if (ret != 0)
printk(KERN_ERR "can't add spi driver");
}
#endif
if (ret != 0) {
@ -1797,6 +1859,9 @@ static int wm8753_remove(struct platform_device *pdev)
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
i2c_unregister_device(codec->control_data);
i2c_del_driver(&wm8753_i2c_driver);
#endif
#if defined(CONFIG_SPI_MASTER)
spi_unregister_driver(&wm8753_spi_driver);
#endif
kfree(codec->private_data);
kfree(codec);

4
sound/soc/codecs/wm8753.h
View file

@ -2,8 +2,7 @@
* wm8753.h -- audio driver for WM8753
*
* Copyright 2003 Wolfson Microelectronics PLC.
* Author: Liam Girdwood
* liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
* Author: Liam Girdwood <lrg@slimlogic.co.uk>
*
* 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
@ -79,6 +78,7 @@
#define WM8753_ADCTL2 0x3f
struct wm8753_setup_data {
int spi;
int i2c_bus;
unsigned short i2c_address;
};

1
sound/soc/codecs/wm8900.c
View file

@ -18,7 +18,6 @@
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/version.h>
#include <linux/kernel.h>
#include <linux/init.h>
#include <linux/delay.h>

4
sound/soc/codecs/wm8903.c
View file

@ -653,14 +653,14 @@ static const struct snd_kcontrol_new wm8903_snd_controls[] = {
/* Input PGAs - No TLV since the scale depends on PGA mode */
SOC_SINGLE("Left Input PGA Switch", WM8903_ANALOGUE_LEFT_INPUT_0,
7, 1, 0),
7, 1, 1),
SOC_SINGLE("Left Input PGA Volume", WM8903_ANALOGUE_LEFT_INPUT_0,
0, 31, 0),
SOC_SINGLE("Left Input PGA Common Mode Switch", WM8903_ANALOGUE_LEFT_INPUT_1,
6, 1, 0),
SOC_SINGLE("Right Input PGA Switch", WM8903_ANALOGUE_RIGHT_INPUT_0,
7, 1, 0),
7, 1, 1),
SOC_SINGLE("Right Input PGA Volume", WM8903_ANALOGUE_RIGHT_INPUT_0,
0, 31, 0),
SOC_SINGLE("Right Input PGA Common Mode Switch", WM8903_ANALOGUE_RIGHT_INPUT_1,

1
sound/soc/codecs/wm8971.c
View file

@ -29,7 +29,6 @@
#include "wm8971.h"
#define AUDIO_NAME "wm8971"
#define WM8971_VERSION "0.9"
#define WM8971_REG_COUNT 43

1
sound/soc/codecs/wm8990.c
View file

@ -30,7 +30,6 @@
#include "wm8990.h"
#define AUDIO_NAME "wm8990"
#define WM8990_VERSION "0.2"
/* codec private data */

3
sound/soc/codecs/wm9712.c
View file

@ -2,8 +2,7 @@
* wm9712.c -- ALSA Soc WM9712 codec support
*
* Copyright 2006 Wolfson Microelectronics PLC.
* Author: Liam Girdwood
* liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
* Author: Liam Girdwood <lrg@slimlogic.co.uk>
*
* 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

3
sound/soc/codecs/wm9713.c
View file

@ -2,8 +2,7 @@
* wm9713.c -- ALSA Soc WM9713 codec support
*
* Copyright 2006 Wolfson Microelectronics PLC.
* Author: Liam Girdwood
* liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
* Author: Liam Girdwood <lrg@slimlogic.co.uk>
*
* 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

8
sound/soc/omap/Kconfig
View file

@ -13,3 +13,11 @@ config SND_OMAP_SOC_N810
select SND_SOC_TLV320AIC3X
help
Say Y if you want to add support for SoC audio on Nokia N810.
config SND_OMAP_SOC_OSK5912
tristate "SoC Audio support for omap osk5912"
depends on SND_OMAP_SOC && MACH_OMAP_OSK
select SND_OMAP_SOC_MCBSP
select SND_SOC_TLV320AIC23
help
Say Y if you want to add support for SoC audio on osk5912.

2
sound/soc/omap/Makefile
View file

@ -7,5 +7,7 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o
# OMAP Machine Support
snd-soc-n810-objs := n810.o
snd-soc-osk5912-objs := osk5912.o
obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o

6
sound/soc/omap/n810.c
View file

@ -247,9 +247,9 @@ static int n810_aic33_init(struct snd_soc_codec *codec)
int i, err;
/* Not connected */
snd_soc_dapm_disable_pin(codec, "MONO_LOUT");
snd_soc_dapm_disable_pin(codec, "HPLCOM");
snd_soc_dapm_disable_pin(codec, "HPRCOM");
snd_soc_dapm_nc_pin(codec, "MONO_LOUT");
snd_soc_dapm_nc_pin(codec, "HPLCOM");
snd_soc_dapm_nc_pin(codec, "HPRCOM");
/* Add N810 specific controls */
for (i = 0; i < ARRAY_SIZE(aic33_n810_controls); i++) {

181
sound/soc/omap/omap-mcbsp.c
View file

@ -59,12 +59,7 @@ static struct omap_mcbsp_data mcbsp_data[NUM_LINKS];
* Stream DMA parameters. DMA request line and port address are set runtime
* since they are different between OMAP1 and later OMAPs
*/
static struct omap_pcm_dma_data omap_mcbsp_dai_dma_params[NUM_LINKS][2] = {
{
{ .name = "I2S PCM Stereo out", },
{ .name = "I2S PCM Stereo in", },
},
};
static struct omap_pcm_dma_data omap_mcbsp_dai_dma_params[NUM_LINKS][2];
#if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX)
static const int omap1_dma_reqs[][2] = {
@ -84,11 +79,22 @@ static const unsigned long omap1_mcbsp_port[][2] = {
static const int omap1_dma_reqs[][2] = {};
static const unsigned long omap1_mcbsp_port[][2] = {};
#endif
#if defined(CONFIG_ARCH_OMAP2420)
static const int omap2420_dma_reqs[][2] = {
#if defined(CONFIG_ARCH_OMAP24XX) || defined(CONFIG_ARCH_OMAP34XX)
static const int omap24xx_dma_reqs[][2] = {
{ OMAP24XX_DMA_MCBSP1_TX, OMAP24XX_DMA_MCBSP1_RX },
{ OMAP24XX_DMA_MCBSP2_TX, OMAP24XX_DMA_MCBSP2_RX },
#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP34XX)
{ OMAP24XX_DMA_MCBSP3_TX, OMAP24XX_DMA_MCBSP3_RX },
{ OMAP24XX_DMA_MCBSP4_TX, OMAP24XX_DMA_MCBSP4_RX },
{ OMAP24XX_DMA_MCBSP5_TX, OMAP24XX_DMA_MCBSP5_RX },
#endif
};
#else
static const int omap24xx_dma_reqs[][2] = {};
#endif
#if defined(CONFIG_ARCH_OMAP2420)
static const unsigned long omap2420_mcbsp_port[][2] = {
{ OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1,
OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR1 },
@ -96,10 +102,43 @@ static const unsigned long omap2420_mcbsp_port[][2] = {
OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR1 },
};
#else
static const int omap2420_dma_reqs[][2] = {};
static const unsigned long omap2420_mcbsp_port[][2] = {};
#endif
#if defined(CONFIG_ARCH_OMAP2430)
static const unsigned long omap2430_mcbsp_port[][2] = {
{ OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR,
OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR },
{ OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR,
OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR },
{ OMAP2430_MCBSP3_BASE + OMAP_MCBSP_REG_DXR,
OMAP2430_MCBSP3_BASE + OMAP_MCBSP_REG_DRR },
{ OMAP2430_MCBSP4_BASE + OMAP_MCBSP_REG_DXR,
OMAP2430_MCBSP4_BASE + OMAP_MCBSP_REG_DRR },
{ OMAP2430_MCBSP5_BASE + OMAP_MCBSP_REG_DXR,
OMAP2430_MCBSP5_BASE + OMAP_MCBSP_REG_DRR },
};
#else
static const unsigned long omap2430_mcbsp_port[][2] = {};
#endif
#if defined(CONFIG_ARCH_OMAP34XX)
static const unsigned long omap34xx_mcbsp_port[][2] = {
{ OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR,
OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR },
{ OMAP34XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR,
OMAP34XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR },
{ OMAP34XX_MCBSP3_BASE + OMAP_MCBSP_REG_DXR,
OMAP34XX_MCBSP3_BASE + OMAP_MCBSP_REG_DRR },
{ OMAP34XX_MCBSP4_BASE + OMAP_MCBSP_REG_DXR,
OMAP34XX_MCBSP4_BASE + OMAP_MCBSP_REG_DRR },
{ OMAP34XX_MCBSP5_BASE + OMAP_MCBSP_REG_DXR,
OMAP34XX_MCBSP5_BASE + OMAP_MCBSP_REG_DRR },
};
#else
static const unsigned long omap34xx_mcbsp_port[][2] = {};
#endif
static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
@ -167,14 +206,19 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
dma = omap1_dma_reqs[bus_id][substream->stream];
port = omap1_mcbsp_port[bus_id][substream->stream];
} else if (cpu_is_omap2420()) {
dma = omap2420_dma_reqs[bus_id][substream->stream];
dma = omap24xx_dma_reqs[bus_id][substream->stream];
port = omap2420_mcbsp_port[bus_id][substream->stream];
} else if (cpu_is_omap2430()) {
dma = omap24xx_dma_reqs[bus_id][substream->stream];
port = omap2430_mcbsp_port[bus_id][substream->stream];
} else if (cpu_is_omap343x()) {
dma = omap24xx_dma_reqs[bus_id][substream->stream];
port = omap34xx_mcbsp_port[bus_id][substream->stream];
} else {
/*
* TODO: Add support for 2430 and 3430
*/
return -ENODEV;
}
omap_mcbsp_dai_dma_params[id][substream->stream].name =
substream->stream ? "Audio Capture" : "Audio Playback";
omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma;
omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port;
cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream];
@ -245,6 +289,11 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
regs->rcr2 |= RDATDLY(1);
regs->xcr2 |= XDATDLY(1);
break;
case SND_SOC_DAIFMT_DSP_A:
/* 0-bit data delay */
regs->rcr2 |= RDATDLY(0);
regs->xcr2 |= XDATDLY(0);
break;
default:
/* Unsupported data format */
return -EINVAL;
@ -310,7 +359,7 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data,
int clk_id)
{
int sel_bit;
u16 reg;
u16 reg, reg_devconf1 = OMAP243X_CONTROL_DEVCONF1;
if (cpu_class_is_omap1()) {
/* OMAP1's can use only external source clock */
@ -320,6 +369,12 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data,
return 0;
}
if (cpu_is_omap2420() && mcbsp_data->bus_id > 1)
return -EINVAL;
if (cpu_is_omap343x())
reg_devconf1 = OMAP343X_CONTROL_DEVCONF1;
switch (mcbsp_data->bus_id) {
case 0:
reg = OMAP2_CONTROL_DEVCONF0;
@ -329,20 +384,26 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data,
reg = OMAP2_CONTROL_DEVCONF0;
sel_bit = 6;
break;
/* TODO: Support for ports 3 - 5 in OMAP2430 and OMAP34xx */
case 2:
reg = reg_devconf1;
sel_bit = 0;
break;
case 3:
reg = reg_devconf1;
sel_bit = 2;
break;
case 4:
reg = reg_devconf1;
sel_bit = 4;
break;
default:
return -EINVAL;
}
if (cpu_class_is_omap2()) {
if (clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK) {
omap_ctrl_writel(omap_ctrl_readl(reg) &
~(1 << sel_bit), reg);
} else {
omap_ctrl_writel(omap_ctrl_readl(reg) |
(1 << sel_bit), reg);
}
}
if (clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK)
omap_ctrl_writel(omap_ctrl_readl(reg) & ~(1 << sel_bit), reg);
else
omap_ctrl_writel(omap_ctrl_readl(reg) | (1 << sel_bit), reg);
return 0;
}
@ -376,37 +437,49 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
return err;
}
struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS] = {
{
.name = "omap-mcbsp-dai",
.id = 0,
.type = SND_SOC_DAI_I2S,
.playback = {
.channels_min = 2,
.channels_max = 2,
.rates = OMAP_MCBSP_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.capture = {
.channels_min = 2,
.channels_max = 2,
.rates = OMAP_MCBSP_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.ops = {
.startup = omap_mcbsp_dai_startup,
.shutdown = omap_mcbsp_dai_shutdown,
.trigger = omap_mcbsp_dai_trigger,
.hw_params = omap_mcbsp_dai_hw_params,
},
.dai_ops = {
.set_fmt = omap_mcbsp_dai_set_dai_fmt,
.set_clkdiv = omap_mcbsp_dai_set_clkdiv,
.set_sysclk = omap_mcbsp_dai_set_dai_sysclk,
},
.private_data = &mcbsp_data[0].bus_id,
},
#define OMAP_MCBSP_DAI_BUILDER(link_id) \
{ \
.name = "omap-mcbsp-dai-(link_id)", \
.id = (link_id), \
.type = SND_SOC_DAI_I2S, \
.playback = { \
.channels_min = 2, \
.channels_max = 2, \
.rates = OMAP_MCBSP_RATES, \
.formats = SNDRV_PCM_FMTBIT_S16_LE, \
}, \
.capture = { \
.channels_min = 2, \
.channels_max = 2, \
.rates = OMAP_MCBSP_RATES, \
.formats = SNDRV_PCM_FMTBIT_S16_LE, \
}, \
.ops = { \
.startup = omap_mcbsp_dai_startup, \
.shutdown = omap_mcbsp_dai_shutdown, \
.trigger = omap_mcbsp_dai_trigger, \
.hw_params = omap_mcbsp_dai_hw_params, \
}, \
.dai_ops = { \
.set_fmt = omap_mcbsp_dai_set_dai_fmt, \
.set_clkdiv = omap_mcbsp_dai_set_clkdiv, \
.set_sysclk = omap_mcbsp_dai_set_dai_sysclk, \
}, \
.private_data = &mcbsp_data[(link_id)].bus_id, \
}
struct snd_soc_dai omap_mcbsp_dai[] = {
OMAP_MCBSP_DAI_BUILDER(0),
OMAP_MCBSP_DAI_BUILDER(1),
#if NUM_LINKS >= 3
OMAP_MCBSP_DAI_BUILDER(2),
#endif
#if NUM_LINKS == 5
OMAP_MCBSP_DAI_BUILDER(3),
OMAP_MCBSP_DAI_BUILDER(4),
#endif
};
EXPORT_SYMBOL_GPL(omap_mcbsp_dai);
MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");

16
sound/soc/omap/omap-mcbsp.h
View file

@ -38,11 +38,17 @@ enum omap_mcbsp_div {
OMAP_MCBSP_CLKGDV, /* Sample rate generator divider */
};
/*
* REVISIT: Preparation for the ASoC v2. Let the number of available links to
* be same than number of McBSP ports found in OMAP(s) we are compiling for.
*/
#define NUM_LINKS 1
#if defined(CONFIG_ARCH_OMAP2420)
#define NUM_LINKS 2
#endif
#if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX)
#undef NUM_LINKS
#define NUM_LINKS 3
#endif
#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP34XX)
#undef NUM_LINKS
#define NUM_LINKS 5
#endif
extern struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS];

4
sound/soc/omap/omap-pcm.c
View file

@ -97,7 +97,7 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
prtd->dma_data = dma_data;
err = omap_request_dma(dma_data->dma_req, dma_data->name,
omap_pcm_dma_irq, substream, &prtd->dma_ch);
if (!cpu_is_omap1510()) {
if (!err & !cpu_is_omap1510()) {
/*
* Link channel with itself so DMA doesn't need any
* reprogramming while looping the buffer
@ -147,12 +147,14 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
dma_params.src_or_dst_synch = OMAP_DMA_DST_SYNC;
dma_params.src_start = runtime->dma_addr;
dma_params.dst_start = dma_data->port_addr;
dma_params.dst_port = OMAP_DMA_PORT_MPUI;
} else {
dma_params.src_amode = OMAP_DMA_AMODE_CONSTANT;
dma_params.dst_amode = OMAP_DMA_AMODE_POST_INC;
dma_params.src_or_dst_synch = OMAP_DMA_SRC_SYNC;
dma_params.src_start = dma_data->port_addr;
dma_params.dst_start = runtime->dma_addr;
dma_params.src_port = OMAP_DMA_PORT_MPUI;
}
/*
* Set DMA transfer frame size equal to ALSA period size and frame

232
sound/soc/omap/osk5912.c Normal file
View file

@ -0,0 +1,232 @@
/*
* osk5912.c -- SoC audio for OSK 5912
*
* Copyright (C) 2008 Mistral Solutions
*
* Contact: Arun KS <arunks@mistralsolutions.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
*
*/
#include <linux/clk.h>
#include <linux/platform_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <asm/mach-types.h>
#include <mach/hardware.h>
#include <linux/gpio.h>
#include <mach/mcbsp.h>
#include "omap-mcbsp.h"
#include "omap-pcm.h"
#include "../codecs/tlv320aic23.h"
#define CODEC_CLOCK 12000000
static struct clk *tlv320aic23_mclk;
static int osk_startup(struct snd_pcm_substream *substream)
{
return clk_enable(tlv320aic23_mclk);
}
static void osk_shutdown(struct snd_pcm_substream *substream)
{
clk_disable(tlv320aic23_mclk);
}
static int osk_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_DSP_A |
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_DSP_A |
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 osk_ops = {
.startup = osk_startup,
.hw_params = osk_hw_params,
.shutdown = osk_shutdown,
};
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 osk_tlv320aic23_init(struct snd_soc_codec *codec)
{
/* Add osk5912 specific widgets */
snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
ARRAY_SIZE(tlv320aic23_dapm_widgets));
/* Set up osk5912 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 osk_dai = {
.name = "TLV320AIC23",
.stream_name = "AIC23",
.cpu_dai = &omap_mcbsp_dai[0],
.codec_dai = &tlv320aic23_dai,
.init = osk_tlv320aic23_init,
.ops = &osk_ops,
};
/* Audio machine driver */
static struct snd_soc_machine snd_soc_machine_osk = {
.name = "OSK5912",
.dai_link = &osk_dai,
.num_links = 1,
};
/* Audio subsystem */
static struct snd_soc_device osk_snd_devdata = {
.machine = &snd_soc_machine_osk,
.platform = &omap_soc_platform,
.codec_dev = &soc_codec_dev_tlv320aic23,
};
static struct platform_device *osk_snd_device;
static int __init osk_soc_init(void)
{
int err;
u32 curRate;
struct device *dev;
if (!(machine_is_omap_osk()))
return -ENODEV;
osk_snd_device = platform_device_alloc("soc-audio", -1);
if (!osk_snd_device)
return -ENOMEM;
platform_set_drvdata(osk_snd_device, &osk_snd_devdata);
osk_snd_devdata.dev = &osk_snd_device->dev;
*(unsigned int *)osk_dai.cpu_dai->private_data = 0; /* McBSP1 */
err = platform_device_add(osk_snd_device);
if (err)
goto err1;
dev = &osk_snd_device->dev;
tlv320aic23_mclk = clk_get(dev, "mclk");
if (IS_ERR(tlv320aic23_mclk)) {
printk(KERN_ERR "Could not get mclk clock\n");
return -ENODEV;
}
if (clk_get_usecount(tlv320aic23_mclk) > 0) {
/* MCLK is already in use */
printk(KERN_WARNING
"MCLK in use at %d Hz. We change it to %d Hz\n",
(uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK);
}
/*
* Configure 12 MHz output on MCLK.
*/
curRate = (uint) clk_get_rate(tlv320aic23_mclk);
if (curRate != CODEC_CLOCK) {
if (clk_set_rate(tlv320aic23_mclk, CODEC_CLOCK)) {
printk(KERN_ERR "Cannot set MCLK for AIC23 CODEC\n");
err = -ECANCELED;
goto err1;
}
}
printk(KERN_INFO "MCLK = %d [%d], usecount = %d\n",
(uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK,
clk_get_usecount(tlv320aic23_mclk));
return 0;
err1:
clk_put(tlv320aic23_mclk);
platform_device_del(osk_snd_device);
platform_device_put(osk_snd_device);
return err;
}
static void __exit osk_soc_exit(void)
{
platform_device_unregister(osk_snd_device);
}
module_init(osk_soc_init);
module_exit(osk_soc_exit);
MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
MODULE_DESCRIPTION("ALSA SoC OSK 5912");
MODULE_LICENSE("GPL");

6
sound/soc/pxa/corgi.c
View file

@ -4,7 +4,7 @@
* Copyright 2005 Wolfson Microelectronics PLC.
* Copyright 2005 Openedhand Ltd.
*
* Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
* Authors: Liam Girdwood <lrg@slimlogic.co.uk>
* Richard Purdie <richard@openedhand.com>
*
* This program is free software; you can redistribute it and/or modify it
@ -281,8 +281,8 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec)
{
int i, err;
snd_soc_dapm_disable_pin(codec, "LLINEIN");
snd_soc_dapm_disable_pin(codec, "RLINEIN");
snd_soc_dapm_nc_pin(codec, "LLINEIN");
snd_soc_dapm_nc_pin(codec, "RLINEIN");
/* Add corgi specific controls */
for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) {

2
sound/soc/pxa/em-x270.c
View file

@ -9,7 +9,7 @@
* Copyright 2005 Wolfson Microelectronics PLC.
* Copyright 2005 Openedhand Ltd.
*
* Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
* Authors: Liam Girdwood <lrg@slimlogic.co.uk>
* Richard Purdie <richard@openedhand.com>
*
* This program is free software; you can redistribute it and/or modify it

6
sound/soc/pxa/poodle.c
View file

@ -4,7 +4,7 @@
* Copyright 2005 Wolfson Microelectronics PLC.
* Copyright 2005 Openedhand Ltd.
*
* Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
* Authors: Liam Girdwood <lrg@slimlogic.co.uk>
* Richard Purdie <richard@openedhand.com>
*
* This program is free software; you can redistribute it and/or modify it
@ -242,8 +242,8 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec)
{
int i, err;
snd_soc_dapm_disable_pin(codec, "LLINEIN");
snd_soc_dapm_disable_pin(codec, "RLINEIN");
snd_soc_dapm_nc_pin(codec, "LLINEIN");
snd_soc_dapm_nc_pin(codec, "RLINEIN");
snd_soc_dapm_enable_pin(codec, "MICIN");
/* Add poodle specific controls */

4
sound/soc/pxa/pxa2xx-i2s.c
View file

@ -3,7 +3,7 @@
*
* Copyright 2005 Wolfson Microelectronics PLC.
* Author: Liam Girdwood
* liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
* lrg@slimlogic.co.uk
*
* 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
@ -405,6 +405,6 @@ module_init(pxa2xx_i2s_init);
module_exit(pxa2xx_i2s_exit);
/* Module information */
MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
MODULE_DESCRIPTION("pxa2xx I2S SoC Interface");
MODULE_LICENSE("GPL");

16
sound/soc/pxa/spitz.c
View file

@ -4,7 +4,7 @@
* Copyright 2005 Wolfson Microelectronics PLC.
* Copyright 2005 Openedhand Ltd.
*
* Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
* Authors: Liam Girdwood <lrg@slimlogic.co.uk>
* Richard Purdie <richard@openedhand.com>
*
* This program is free software; you can redistribute it and/or modify it
@ -281,13 +281,13 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec)
int i, err;
/* NC codec pins */
snd_soc_dapm_disable_pin(codec, "RINPUT1");
snd_soc_dapm_disable_pin(codec, "LINPUT2");
snd_soc_dapm_disable_pin(codec, "RINPUT2");
snd_soc_dapm_disable_pin(codec, "LINPUT3");
snd_soc_dapm_disable_pin(codec, "RINPUT3");
snd_soc_dapm_disable_pin(codec, "OUT3");
snd_soc_dapm_disable_pin(codec, "MONO1");
snd_soc_dapm_nc_pin(codec, "RINPUT1");
snd_soc_dapm_nc_pin(codec, "LINPUT2");
snd_soc_dapm_nc_pin(codec, "RINPUT2");
snd_soc_dapm_nc_pin(codec, "LINPUT3");
snd_soc_dapm_nc_pin(codec, "RINPUT3");
snd_soc_dapm_nc_pin(codec, "OUT3");
snd_soc_dapm_nc_pin(codec, "MONO1");
/* Add spitz specific controls */
for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) {

6
sound/soc/pxa/tosa.c
View file

@ -4,7 +4,7 @@
* Copyright 2005 Wolfson Microelectronics PLC.
* Copyright 2005 Openedhand Ltd.
*
* Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
* Authors: Liam Girdwood <lrg@slimlogic.co.uk>
* Richard Purdie <richard@openedhand.com>
*
* This program is free software; you can redistribute it and/or modify it
@ -190,8 +190,8 @@ static int tosa_ac97_init(struct snd_soc_codec *codec)
{
int i, err;
snd_soc_dapm_disable_pin(codec, "OUT3");
snd_soc_dapm_disable_pin(codec, "MONOOUT");
snd_soc_dapm_nc_pin(codec, "OUT3");
snd_soc_dapm_nc_pin(codec, "MONOOUT");
/* add tosa specific controls */
for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) {

72
sound/soc/s3c24xx/neo1973_wm8753.c
View file

@ -511,21 +511,20 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec)
DBG("Entered %s\n", __func__);
/* set up NC codec pins */
snd_soc_dapm_disable_pin(codec, "LOUT2");
snd_soc_dapm_disable_pin(codec, "ROUT2");
snd_soc_dapm_disable_pin(codec, "OUT3");
snd_soc_dapm_disable_pin(codec, "OUT4");
snd_soc_dapm_disable_pin(codec, "LINE1");
snd_soc_dapm_disable_pin(codec, "LINE2");
/* set endpoints to default mode */
set_scenario_endpoints(codec, NEO_AUDIO_OFF);
snd_soc_dapm_nc_pin(codec, "LOUT2");
snd_soc_dapm_nc_pin(codec, "ROUT2");
snd_soc_dapm_nc_pin(codec, "OUT3");
snd_soc_dapm_nc_pin(codec, "OUT4");
snd_soc_dapm_nc_pin(codec, "LINE1");
snd_soc_dapm_nc_pin(codec, "LINE2");
/* Add neo1973 specific widgets */
snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets,
ARRAY_SIZE(wm8753_dapm_widgets));
/* set endpoints to default mode */
set_scenario_endpoints(codec, NEO_AUDIO_OFF);
/* add neo1973 specific controls */
for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_controls); i++) {
err = snd_ctl_add(codec->card,
@ -603,6 +602,8 @@ static int lm4857_i2c_probe(struct i2c_client *client,
{
DBG("Entered %s\n", __func__);
i2c = client;
lm4857_write_regs();
return 0;
}
@ -611,6 +612,8 @@ static int lm4857_i2c_remove(struct i2c_client *client)
{
DBG("Entered %s\n", __func__);
i2c = NULL;
return 0;
}
@ -650,7 +653,7 @@ static void lm4857_shutdown(struct i2c_client *dev)
}
static const struct i2c_device_id lm4857_i2c_id[] = {
{ "neo1973_lm4857", 0 }
{ "neo1973_lm4857", 0 },
{ }
};
@ -668,48 +671,6 @@ static struct i2c_driver lm4857_i2c_driver = {
};
static struct platform_device *neo1973_snd_device;
static struct i2c_client *lm4857_client;
static int __init neo1973_add_lm4857_device(struct platform_device *pdev,
int i2c_bus,
unsigned short i2c_address)
{
struct i2c_board_info info;
struct i2c_adapter *adapter;
struct i2c_client *client;
int ret;
ret = i2c_add_driver(&lm4857_i2c_driver);
if (ret != 0) {
dev_err(&pdev->dev, "can't add lm4857 driver\n");
return ret;
}
memset(&info, 0, sizeof(struct i2c_board_info));
info.addr = i2c_address;
strlcpy(info.type, "neo1973_lm4857", I2C_NAME_SIZE);
adapter = i2c_get_adapter(i2c_bus);
if (!adapter) {
dev_err(&pdev->dev, "can't get i2c adapter %d\n", i2c_bus);
goto err_driver;
}
client = i2c_new_device(adapter, &info);
i2c_put_adapter(adapter);
if (!client) {
dev_err(&pdev->dev, "can't add lm4857 device at 0x%x\n",
(unsigned int)info.addr);
goto err_driver;
}
lm4857_client = client;
return 0;
err_driver:
i2c_del_driver(&lm4857_i2c_driver);
return -ENODEV;
}
static int __init neo1973_init(void)
{
@ -736,8 +697,8 @@ static int __init neo1973_init(void)
return ret;
}
ret = neo1973_add_lm4857_device(neo1973_snd_device,
neo1973_wm8753_setup, 0x7C);
ret = i2c_add_driver(&lm4857_i2c_driver);
if (ret != 0)
platform_device_unregister(neo1973_snd_device);
@ -748,7 +709,6 @@ static void __exit neo1973_exit(void)
{
DBG("Entered %s\n", __func__);
i2c_unregister_device(lm4857_client);
i2c_del_driver(&lm4857_i2c_driver);
platform_device_unregister(neo1973_snd_device);
}

5
sound/soc/soc-core.c
View file

@ -4,8 +4,7 @@
* Copyright 2005 Wolfson Microelectronics PLC.
* Copyright 2005 Openedhand Ltd.
*
* Author: Liam Girdwood
* liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
* Author: Liam Girdwood <lrg@slimlogic.co.uk>
* with code, comments and ideas from :-
* Richard Purdie <richard@openedhand.com>
*
@ -1886,7 +1885,7 @@ module_init(snd_soc_init);
module_exit(snd_soc_exit);
/* Module information */
MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
MODULE_DESCRIPTION("ALSA SoC Core");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:soc-audio");

25
sound/soc/soc-dapm.c
View file

@ -2,8 +2,7 @@
* soc-dapm.c -- ALSA SoC Dynamic Audio Power Management
*
* Copyright 2005 Wolfson Microelectronics PLC.
* Author: Liam Girdwood
* liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
* Author: Liam Girdwood <lrg@slimlogic.co.uk>
*
* 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
@ -1483,6 +1482,26 @@ int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin)
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin);
/**
* snd_soc_dapm_nc_pin - permanently disable pin.
* @codec: SoC codec
* @pin: pin name
*
* Marks the specified pin as being not connected, disabling it along
* any parent or child widgets. At present this is identical to
* snd_soc_dapm_disable_pin() but in future it will be extended to do
* additional things such as disabling controls which only affect
* paths through the pin.
*
* NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
* do any widget power switching.
*/
int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, char *pin)
{
return snd_soc_dapm_set_pin(codec, pin, 0);
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin);
/**
* snd_soc_dapm_get_pin_status - get audio pin status
* @codec: audio codec
@ -1521,6 +1540,6 @@ void snd_soc_dapm_free(struct snd_soc_device *socdev)
EXPORT_SYMBOL_GPL(snd_soc_dapm_free);
/* Module information */
MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
MODULE_DESCRIPTION("Dynamic Audio Power Management core for ALSA SoC");
MODULE_LICENSE("GPL");