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Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into next/boards

Forráskód böngészése 
The asoc branch that was already merged into v3.4 contains some
board-level changes that conflict with patches we already have
here, so pull in that branch to resolve the conflicts.

Conflicts:
	arch/arm/mach-imx/mach-imx27_visstrim_m10.c
	arch/arm/mach-omap2/board-omap4panda.c

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
[olof: Amended fix for mismerge as reported by Kevin Hilman]
Signed-off-by: Olof Johansson <olof@lixom.net>
This commit is contained in:
Arnd Bergmann
2012-03-24 11:33:59 +00:00
committed by Olof Johansson
szülő 70688056a8 22f8d05535
commit a754a87ce8
214 fájl változott, egészen pontosan 14178 új sor hozzáadva és 6320 régi sor törölve
Download Patch File Download Diff File Expand all files Collapse all files

13
sound/soc/omap/Kconfig
Fájl megtekintése

@@ -97,16 +97,19 @@ config SND_OMAP_SOC_SDP3430
Say Y if you want to add support for SoC audio on Texas Instruments
SDP3430.
config SND_OMAP_SOC_SDP4430
tristate "SoC Audio support for Texas Instruments SDP4430"
depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_4430SDP
config SND_OMAP_SOC_OMAP_ABE_TWL6040
tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec"
depends on TWL4030_CORE && SND_OMAP_SOC && ARCH_OMAP4
select SND_OMAP_SOC_DMIC
select SND_OMAP_SOC_MCPDM
select SND_SOC_TWL6040
select SND_SOC_DMIC
help
Say Y if you want to add support for SoC audio on Texas Instruments
SDP4430.
Say Y if you want to add support for SoC audio on OMAP boards using
ABE and twl6040 codec. This driver currently supports:
- SDP4430/Blaze boards
- PandaBoard (4430)
- PandaBoardES (4460)
config SND_OMAP_SOC_OMAP4_HDMI
tristate "SoC Audio support for Texas Instruments OMAP4 HDMI"

4
sound/soc/omap/Makefile
Fájl megtekintése

@@ -20,7 +20,7 @@ snd-soc-overo-objs := overo.o
snd-soc-omap3evm-objs := omap3evm.o
snd-soc-am3517evm-objs := am3517evm.o
snd-soc-sdp3430-objs := sdp3430.o
snd-soc-sdp4430-objs := sdp4430.o
snd-soc-omap-abe-twl6040-objs := omap-abe-twl6040.o
snd-soc-omap3pandora-objs := omap3pandora.o
snd-soc-omap3beagle-objs := omap3beagle.o
snd-soc-zoom2-objs := zoom2.o
@@ -36,7 +36,7 @@ obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o
obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o
obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o
obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
obj-$(CONFIG_SND_OMAP_SOC_SDP4430) += snd-soc-sdp4430.o
obj-$(CONFIG_SND_OMAP_SOC_OMAP_ABE_TWL6040) += snd-soc-omap-abe-twl6040.o
obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o
obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o

2
sound/soc/omap/ams-delta.c
Fájl megtekintése

@@ -544,7 +544,7 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
snd_soc_dapm_disable_pin(dapm, "AGCOUT");
/* Add virtual switch */
ret = snd_soc_add_controls(codec, ams_delta_audio_controls,
ret = snd_soc_add_codec_controls(codec, ams_delta_audio_controls,
ARRAY_SIZE(ams_delta_audio_controls));
if (ret)
dev_warn(card->dev,

17
sound/soc/omap/n810.c
Fájl megtekintése

@@ -55,9 +55,8 @@ static int n810_spk_func;
static int n810_jack_func;
static int n810_dmic_func;
static void n810_ext_control(struct snd_soc_codec *codec)
static void n810_ext_control(struct snd_soc_dapm_context *dapm)
{
struct snd_soc_dapm_context *dapm = &codec->dapm;
int hp = 0, line1l = 0;
switch (n810_jack_func) {
@@ -102,7 +101,7 @@ static int n810_startup(struct snd_pcm_substream *substream)
snd_pcm_hw_constraint_minmax(runtime,
SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
n810_ext_control(codec);
n810_ext_control(&codec->dapm);
return clk_enable(sys_clkout2);
}
@@ -142,13 +141,13 @@ static int n810_get_spk(struct snd_kcontrol *kcontrol,
static int n810_set_spk(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
if (n810_spk_func == ucontrol->value.integer.value[0])
return 0;
n810_spk_func = ucontrol->value.integer.value[0];
n810_ext_control(codec);
n810_ext_control(&card->dapm);
return 1;
}
@@ -164,13 +163,13 @@ static int n810_get_jack(struct snd_kcontrol *kcontrol,
static int n810_set_jack(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
if (n810_jack_func == ucontrol->value.integer.value[0])
return 0;
n810_jack_func = ucontrol->value.integer.value[0];
n810_ext_control(codec);
n810_ext_control(&card->dapm);
return 1;
}
@@ -186,13 +185,13 @@ static int n810_get_input(struct snd_kcontrol *kcontrol,
static int n810_set_input(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
if (n810_dmic_func == ucontrol->value.integer.value[0])
return 0;
n810_dmic_func = ucontrol->value.integer.value[0];
n810_ext_control(codec);
n810_ext_control(&card->dapm);
return 1;
}

349
sound/soc/omap/omap-abe-twl6040.c Normal file
Fájl megtekintése

@@ -0,0 +1,349 @@
/*
* omap-abe-twl6040.c -- SoC audio for TI OMAP based boards with ABE and
* twl6040 codec
*
* Author: Misael Lopez Cruz <misael.lopez@ti.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 <linux/mfd/twl6040.h>
#include <linux/platform_data/omap-abe-twl6040.h>
#include <linux/module.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/jack.h>
#include <asm/mach-types.h>
#include <plat/hardware.h>
#include <plat/mux.h>
#include "omap-dmic.h"
#include "omap-mcpdm.h"
#include "omap-pcm.h"
#include "../codecs/twl6040.h"
static int omap_abe_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_card *card = codec->card;
struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev);
int clk_id, freq;
int ret;
clk_id = twl6040_get_clk_id(rtd->codec);
if (clk_id == TWL6040_SYSCLK_SEL_HPPLL)
freq = pdata->mclk_freq;
else if (clk_id == TWL6040_SYSCLK_SEL_LPPLL)
freq = 32768;
else
return -EINVAL;
/* set the codec mclk */
ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, freq,
SND_SOC_CLOCK_IN);
if (ret) {
printk(KERN_ERR "can't set codec system clock\n");
return ret;
}
return ret;
}
static struct snd_soc_ops omap_abe_ops = {
.hw_params = omap_abe_hw_params,
};
static int omap_abe_dmic_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->cpu_dai;
int ret = 0;
ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_SYSCLK_PAD_CLKS,
19200000, SND_SOC_CLOCK_IN);
if (ret < 0) {
printk(KERN_ERR "can't set DMIC cpu system clock\n");
return ret;
}
ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_ABE_DMIC_CLK, 2400000,
SND_SOC_CLOCK_OUT);
if (ret < 0) {
printk(KERN_ERR "can't set DMIC output clock\n");
return ret;
}
return 0;
}
static struct snd_soc_ops omap_abe_dmic_ops = {
.hw_params = omap_abe_dmic_hw_params,
};
/* Headset jack */
static struct snd_soc_jack hs_jack;
/*Headset jack detection DAPM pins */
static struct snd_soc_jack_pin hs_jack_pins[] = {
{
.pin = "Headset Mic",
.mask = SND_JACK_MICROPHONE,
},
{
.pin = "Headset Stereophone",
.mask = SND_JACK_HEADPHONE,
},
};
/* SDP4430 machine DAPM */
static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
/* Outputs */
SND_SOC_DAPM_HP("Headset Stereophone", NULL),
SND_SOC_DAPM_SPK("Earphone Spk", NULL),
SND_SOC_DAPM_SPK("Ext Spk", NULL),
SND_SOC_DAPM_LINE("Line Out", NULL),
SND_SOC_DAPM_SPK("Vibrator", NULL),
/* Inputs */
SND_SOC_DAPM_MIC("Headset Mic", NULL),
SND_SOC_DAPM_MIC("Main Handset Mic", NULL),
SND_SOC_DAPM_MIC("Sub Handset Mic", NULL),
SND_SOC_DAPM_LINE("Line In", NULL),
};
static const struct snd_soc_dapm_route audio_map[] = {
/* Routings for outputs */
{"Headset Stereophone", NULL, "HSOL"},
{"Headset Stereophone", NULL, "HSOR"},
{"Earphone Spk", NULL, "EP"},
{"Ext Spk", NULL, "HFL"},
{"Ext Spk", NULL, "HFR"},
{"Line Out", NULL, "AUXL"},
{"Line Out", NULL, "AUXR"},
{"Vibrator", NULL, "VIBRAL"},
{"Vibrator", NULL, "VIBRAR"},
/* Routings for inputs */
{"HSMIC", NULL, "Headset Mic"},
{"Headset Mic", NULL, "Headset Mic Bias"},
{"MAINMIC", NULL, "Main Handset Mic"},
{"Main Handset Mic", NULL, "Main Mic Bias"},
{"SUBMIC", NULL, "Sub Handset Mic"},
{"Sub Handset Mic", NULL, "Main Mic Bias"},
{"AFML", NULL, "Line In"},
{"AFMR", NULL, "Line In"},
};
static inline void twl6040_disconnect_pin(struct snd_soc_dapm_context *dapm,
int connected, char *pin)
{
if (!connected)
snd_soc_dapm_disable_pin(dapm, pin);
}
static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_card *card = codec->card;
struct snd_soc_dapm_context *dapm = &codec->dapm;
struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev);
int hs_trim;
int ret = 0;
/* Disable not connected paths if not used */
twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone");
twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk");
twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk");
twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out");
twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vinrator");
twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic");
twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic");
twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic");
twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In");
/*
* Configure McPDM offset cancellation based on the HSOTRIM value from
* twl6040.
*/
hs_trim = twl6040_get_trim_value(codec, TWL6040_TRIM_HSOTRIM);
omap_mcpdm_configure_dn_offsets(rtd, TWL6040_HSF_TRIM_LEFT(hs_trim),
TWL6040_HSF_TRIM_RIGHT(hs_trim));
/* Headset jack detection only if it is supported */
if (pdata->jack_detection) {
ret = snd_soc_jack_new(codec, "Headset Jack",
SND_JACK_HEADSET, &hs_jack);
if (ret)
return ret;
ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
hs_jack_pins);
twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET);
}
return ret;
}
static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = {
SND_SOC_DAPM_MIC("Digital Mic", NULL),
};
static const struct snd_soc_dapm_route dmic_audio_map[] = {
{"DMic", NULL, "Digital Mic"},
{"Digital Mic", NULL, "Digital Mic1 Bias"},
};
static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
ret = snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets,
ARRAY_SIZE(dmic_dapm_widgets));
if (ret)
return ret;
return snd_soc_dapm_add_routes(dapm, dmic_audio_map,
ARRAY_SIZE(dmic_audio_map));
}
/* Digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link twl6040_dmic_dai[] = {
{
.name = "TWL6040",
.stream_name = "TWL6040",
.cpu_dai_name = "omap-mcpdm",
.codec_dai_name = "twl6040-legacy",
.platform_name = "omap-pcm-audio",
.codec_name = "twl6040-codec",
.init = omap_abe_twl6040_init,
.ops = &omap_abe_ops,
},
{
.name = "DMIC",
.stream_name = "DMIC Capture",
.cpu_dai_name = "omap-dmic",
.codec_dai_name = "dmic-hifi",
.platform_name = "omap-pcm-audio",
.codec_name = "dmic-codec",
.init = omap_abe_dmic_init,
.ops = &omap_abe_dmic_ops,
},
};
static struct snd_soc_dai_link twl6040_only_dai[] = {
{
.name = "TWL6040",
.stream_name = "TWL6040",
.cpu_dai_name = "omap-mcpdm",
.codec_dai_name = "twl6040-legacy",
.platform_name = "omap-pcm-audio",
.codec_name = "twl6040-codec",
.init = omap_abe_twl6040_init,
.ops = &omap_abe_ops,
},
};
/* Audio machine driver */
static struct snd_soc_card omap_abe_card = {
.owner = THIS_MODULE,
.dapm_widgets = twl6040_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets),
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
};
static __devinit int omap_abe_probe(struct platform_device *pdev)
{
struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev);
struct snd_soc_card *card = &omap_abe_card;
int ret;
card->dev = &pdev->dev;
if (!pdata) {
dev_err(&pdev->dev, "Missing pdata\n");
return -ENODEV;
}
if (pdata->card_name) {
card->name = pdata->card_name;
} else {
dev_err(&pdev->dev, "Card name is not provided\n");
return -ENODEV;
}
if (!pdata->mclk_freq) {
dev_err(&pdev->dev, "MCLK frequency missing\n");
return -ENODEV;
}
if (pdata->has_dmic) {
card->dai_link = twl6040_dmic_dai;
card->num_links = ARRAY_SIZE(twl6040_dmic_dai);
} else {
card->dai_link = twl6040_only_dai;
card->num_links = ARRAY_SIZE(twl6040_only_dai);
}
ret = snd_soc_register_card(card);
if (ret)
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
return ret;
}
static int __devexit omap_abe_remove(struct platform_device *pdev)
{
struct snd_soc_card *card = platform_get_drvdata(pdev);
snd_soc_unregister_card(card);
return 0;
}
static struct platform_driver omap_abe_driver = {
.driver = {
.name = "omap-abe-twl6040",
.owner = THIS_MODULE,
.pm = &snd_soc_pm_ops,
},
.probe = omap_abe_probe,
.remove = __devexit_p(omap_abe_remove),
};
module_platform_driver(omap_abe_driver);
MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez@ti.com>");
MODULE_DESCRIPTION("ALSA SoC for OMAP boards with ABE and twl6040 codec");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:omap-abe-twl6040");

7
sound/soc/omap/omap-dmic.c
Fájl megtekintése

@@ -113,12 +113,10 @@ static int omap_dmic_dai_startup(struct snd_pcm_substream *substream,
mutex_lock(&dmic->mutex);
if (!dai->active) {
snd_pcm_hw_constraint_msbits(substream->runtime, 0, 32, 24);
if (!dai->active)
dmic->active = 1;
} else {
else
ret = -EBUSY;
}
mutex_unlock(&dmic->mutex);
@@ -445,6 +443,7 @@ static struct snd_soc_dai_driver omap_dmic_dai = {
.channels_max = 6,
.rates = SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000,
.formats = SNDRV_PCM_FMTBIT_S32_LE,
.sig_bits = 24,
},
.ops = &omap_dmic_dai_ops,
};

8
sound/soc/omap/omap-mcbsp.c
Fájl megtekintése

@@ -744,17 +744,17 @@ static const struct snd_kcontrol_new omap_mcbsp3_st_controls[] = {
omap_mcbsp3_set_st_ch1_volume),
};
int omap_mcbsp_st_add_controls(struct snd_soc_codec *codec, int mcbsp_id)
int omap_mcbsp_st_add_controls(struct snd_soc_dai *dai)
{
if (!cpu_is_omap34xx())
return -ENODEV;
switch (mcbsp_id) {
switch (dai->id) {
case 1: /* McBSP 2 */
return snd_soc_add_controls(codec, omap_mcbsp2_st_controls,
return snd_soc_add_dai_controls(dai, omap_mcbsp2_st_controls,
ARRAY_SIZE(omap_mcbsp2_st_controls));
case 2: /* McBSP 3 */
return snd_soc_add_controls(codec, omap_mcbsp3_st_controls,
return snd_soc_add_dai_controls(dai, omap_mcbsp3_st_controls,
ARRAY_SIZE(omap_mcbsp3_st_controls));
default:
break;

2
sound/soc/omap/omap-mcbsp.h
Fájl megtekintése

@@ -59,6 +59,6 @@ enum omap_mcbsp_div {
#define NUM_LINKS 5
#endif
int omap_mcbsp_st_add_controls(struct snd_soc_codec *codec, int mcbsp_id);
int omap_mcbsp_st_add_controls(struct snd_soc_dai *dai);
#endif

2
sound/soc/omap/omap-mcpdm.c
Fájl megtekintése

@@ -419,12 +419,14 @@ static struct snd_soc_dai_driver omap_mcpdm_dai = {
.channels_max = 5,
.rates = OMAP_MCPDM_RATES,
.formats = OMAP_MCPDM_FORMATS,
.sig_bits = 24,
},
.capture = {
.channels_min = 1,
.channels_max = 3,
.rates = OMAP_MCPDM_RATES,
.formats = OMAP_MCPDM_FORMATS,
.sig_bits = 24,
},
.ops = &omap_mcpdm_dai_ops,
};

25
sound/soc/omap/rx51.c
Fájl megtekintése

@@ -59,9 +59,8 @@ static int rx51_spk_func;
static int rx51_dmic_func;
static int rx51_jack_func;
static void rx51_ext_control(struct snd_soc_codec *codec)
static void rx51_ext_control(struct snd_soc_dapm_context *dapm)
{
struct snd_soc_dapm_context *dapm = &codec->dapm;
int hp = 0, hs = 0, tvout = 0;
switch (rx51_jack_func) {
@@ -102,11 +101,11 @@ static int rx51_startup(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_card *card = rtd->card;
snd_pcm_hw_constraint_minmax(runtime,
SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
rx51_ext_control(codec);
rx51_ext_control(&card->dapm);
return 0;
}
@@ -138,13 +137,13 @@ static int rx51_get_spk(struct snd_kcontrol *kcontrol,
static int rx51_set_spk(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
if (rx51_spk_func == ucontrol->value.integer.value[0])
return 0;
rx51_spk_func = ucontrol->value.integer.value[0];
rx51_ext_control(codec);
rx51_ext_control(&card->dapm);
return 1;
}
@@ -184,13 +183,13 @@ static int rx51_get_input(struct snd_kcontrol *kcontrol,
static int rx51_set_input(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
if (rx51_dmic_func == ucontrol->value.integer.value[0])
return 0;
rx51_dmic_func = ucontrol->value.integer.value[0];
rx51_ext_control(codec);
rx51_ext_control(&card->dapm);
return 1;
}
@@ -206,13 +205,13 @@ static int rx51_get_jack(struct snd_kcontrol *kcontrol,
static int rx51_set_jack(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
if (rx51_jack_func == ucontrol->value.integer.value[0])
return 0;
rx51_jack_func = ucontrol->value.integer.value[0];
rx51_ext_control(codec);
rx51_ext_control(&card->dapm);
return 1;
}
@@ -297,7 +296,7 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd)
snd_soc_dapm_nc_pin(dapm, "LINE1R");
/* Add RX-51 specific controls */
err = snd_soc_add_controls(codec, aic34_rx51_controls,
err = snd_soc_add_card_controls(rtd->card, aic34_rx51_controls,
ARRAY_SIZE(aic34_rx51_controls));
if (err < 0)
return err;
@@ -314,7 +313,7 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd)
return err;
snd_soc_limit_volume(codec, "TPA6130A2 Headphone Playback Volume", 42);
err = omap_mcbsp_st_add_controls(codec, 1);
err = omap_mcbsp_st_add_controls(rtd->cpu_dai);
if (err < 0)
return err;
@@ -335,7 +334,7 @@ static int rx51_aic34b_init(struct snd_soc_dapm_context *dapm)
{
int err;
err = snd_soc_add_controls(dapm->codec, aic34_rx51_controlsb,
err = snd_soc_add_card_controls(dapm->card, aic34_rx51_controlsb,
ARRAY_SIZE(aic34_rx51_controlsb));
if (err < 0)
return err;

279
sound/soc/omap/sdp4430.c
Fájl megtekintése

@@ -1,279 +0,0 @@
/*
* sdp4430.c -- SoC audio for TI OMAP4430 SDP
*
* Author: Misael Lopez Cruz <x0052729@ti.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 <linux/mfd/twl6040.h>
#include <linux/module.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/jack.h>
#include <asm/mach-types.h>
#include <plat/hardware.h>
#include <plat/mux.h>
#include "omap-dmic.h"
#include "omap-mcpdm.h"
#include "omap-pcm.h"
#include "../codecs/twl6040.h"
static int sdp4430_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
int clk_id, freq;
int ret;
clk_id = twl6040_get_clk_id(rtd->codec);
if (clk_id == TWL6040_SYSCLK_SEL_HPPLL)
freq = 38400000;
else if (clk_id == TWL6040_SYSCLK_SEL_LPPLL)
freq = 32768;
else
return -EINVAL;
/* set the codec mclk */
ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, freq,
SND_SOC_CLOCK_IN);
if (ret) {
printk(KERN_ERR "can't set codec system clock\n");
return ret;
}
return ret;
}
static struct snd_soc_ops sdp4430_ops = {
.hw_params = sdp4430_hw_params,
};
static int sdp4430_dmic_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->cpu_dai;
int ret = 0;
ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_SYSCLK_PAD_CLKS,
19200000, SND_SOC_CLOCK_IN);
if (ret < 0) {
printk(KERN_ERR "can't set DMIC cpu system clock\n");
return ret;
}
ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_ABE_DMIC_CLK, 2400000,
SND_SOC_CLOCK_OUT);
if (ret < 0) {
printk(KERN_ERR "can't set DMIC output clock\n");
return ret;
}
return 0;
}
static struct snd_soc_ops sdp4430_dmic_ops = {
.hw_params = sdp4430_dmic_hw_params,
};
/* Headset jack */
static struct snd_soc_jack hs_jack;
/*Headset jack detection DAPM pins */
static struct snd_soc_jack_pin hs_jack_pins[] = {
{
.pin = "Headset Mic",
.mask = SND_JACK_MICROPHONE,
},
{
.pin = "Headset Stereophone",
.mask = SND_JACK_HEADPHONE,
},
};
/* SDP4430 machine DAPM */
static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = {
SND_SOC_DAPM_MIC("Ext Mic", NULL),
SND_SOC_DAPM_SPK("Ext Spk", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
SND_SOC_DAPM_HP("Headset Stereophone", NULL),
SND_SOC_DAPM_SPK("Earphone Spk", NULL),
SND_SOC_DAPM_INPUT("FM Stereo In"),
};
static const struct snd_soc_dapm_route audio_map[] = {
/* External Mics: MAINMIC, SUBMIC with bias*/
{"MAINMIC", NULL, "Main Mic Bias"},
{"SUBMIC", NULL, "Main Mic Bias"},
{"Main Mic Bias", NULL, "Ext Mic"},
/* External Speakers: HFL, HFR */
{"Ext Spk", NULL, "HFL"},
{"Ext Spk", NULL, "HFR"},
/* Headset Mic: HSMIC with bias */
{"HSMIC", NULL, "Headset Mic Bias"},
{"Headset Mic Bias", NULL, "Headset Mic"},
/* Headset Stereophone (Headphone): HSOL, HSOR */
{"Headset Stereophone", NULL, "HSOL"},
{"Headset Stereophone", NULL, "HSOR"},
/* Earphone speaker */
{"Earphone Spk", NULL, "EP"},
/* Aux/FM Stereo In: AFML, AFMR */
{"AFML", NULL, "FM Stereo In"},
{"AFMR", NULL, "FM Stereo In"},
};
static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
int ret, hs_trim;
/*
* Configure McPDM offset cancellation based on the HSOTRIM value from
* twl6040.
*/
hs_trim = twl6040_get_trim_value(codec, TWL6040_TRIM_HSOTRIM);
omap_mcpdm_configure_dn_offsets(rtd, TWL6040_HSF_TRIM_LEFT(hs_trim),
TWL6040_HSF_TRIM_RIGHT(hs_trim));
/* Headset jack detection */
ret = snd_soc_jack_new(codec, "Headset Jack",
SND_JACK_HEADSET, &hs_jack);
if (ret)
return ret;
ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
hs_jack_pins);
if (machine_is_omap_4430sdp())
twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET);
else
snd_soc_jack_report(&hs_jack, SND_JACK_HEADSET, SND_JACK_HEADSET);
return ret;
}
static const struct snd_soc_dapm_widget sdp4430_dmic_dapm_widgets[] = {
SND_SOC_DAPM_MIC("Digital Mic", NULL),
};
static const struct snd_soc_dapm_route dmic_audio_map[] = {
{"DMic", NULL, "Digital Mic1 Bias"},
{"Digital Mic1 Bias", NULL, "Digital Mic"},
};
static int sdp4430_dmic_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
ret = snd_soc_dapm_new_controls(dapm, sdp4430_dmic_dapm_widgets,
ARRAY_SIZE(sdp4430_dmic_dapm_widgets));
if (ret)
return ret;
return snd_soc_dapm_add_routes(dapm, dmic_audio_map,
ARRAY_SIZE(dmic_audio_map));
}
/* Digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link sdp4430_dai[] = {
{
.name = "TWL6040",
.stream_name = "TWL6040",
.cpu_dai_name = "omap-mcpdm",
.codec_dai_name = "twl6040-legacy",
.platform_name = "omap-pcm-audio",
.codec_name = "twl6040-codec",
.init = sdp4430_twl6040_init,
.ops = &sdp4430_ops,
},
{
.name = "DMIC",
.stream_name = "DMIC Capture",
.cpu_dai_name = "omap-dmic",
.codec_dai_name = "dmic-hifi",
.platform_name = "omap-pcm-audio",
.codec_name = "dmic-codec",
.init = sdp4430_dmic_init,
.ops = &sdp4430_dmic_ops,
},
};
/* Audio machine driver */
static struct snd_soc_card snd_soc_sdp4430 = {
.name = "SDP4430",
.owner = THIS_MODULE,
.dai_link = sdp4430_dai,
.num_links = ARRAY_SIZE(sdp4430_dai),
.dapm_widgets = sdp4430_twl6040_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(sdp4430_twl6040_dapm_widgets),
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
};
static struct platform_device *sdp4430_snd_device;
static int __init sdp4430_soc_init(void)
{
int ret;
if (!machine_is_omap_4430sdp())
return -ENODEV;
printk(KERN_INFO "SDP4430 SoC init\n");
sdp4430_snd_device = platform_device_alloc("soc-audio", -1);
if (!sdp4430_snd_device) {
printk(KERN_ERR "Platform device allocation failed\n");
return -ENOMEM;
}
platform_set_drvdata(sdp4430_snd_device, &snd_soc_sdp4430);
ret = platform_device_add(sdp4430_snd_device);
if (ret)
goto err;
return 0;
err:
printk(KERN_ERR "Unable to add platform device\n");
platform_device_put(sdp4430_snd_device);
return ret;
}
module_init(sdp4430_soc_init);
static void __exit sdp4430_soc_exit(void)
{
platform_device_unregister(sdp4430_snd_device);
}
module_exit(sdp4430_soc_exit);
MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>");
MODULE_DESCRIPTION("ALSA SoC SDP4430");
MODULE_LICENSE("GPL");