linux/sound/soc/codecs/cq93vc.c
Liam Girdwood ce6120cca2 ASoC: Decouple DAPM from CODECs
Decoupling Dynamic Audio Power Management (DAPM) from codec devices is
required when developing ASoC further. Such as for other ASoC components to
have DAPM widgets or when extending DAPM to handle cross-device paths.

This patch decouples DAPM related variables from struct snd_soc_codec and
moves them to new struct snd_soc_dapm_context that is used to encapsulate
DAPM context of a device. ASoC core and API of DAPM functions are modified
to use DAPM context instead of codec.

This patch does not change current functionality and a large part of changes
come because of structure and internal API changes.

Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some
minor core changes, codecs and machine driver conversions from
Jarkko Nikula <jhnikula@gmail.com>.

Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Nicolas Ferre <nicolas.ferre@atmel.com>
Cc: Manuel Lauss <manuel.lauss@googlemail.com>
Cc: Mike Frysinger <vapier.adi@gmail.com>
Cc: Cliff Cai <cliff.cai@analog.com>
Cc: Kevin Hilman <khilman@deeprootsystems.com>
Cc: Ryan Mallon <ryan@bluewatersys.com>
Cc: Timur Tabi <timur@freescale.com>
Cc: Sascha Hauer <s.hauer@pengutronix.de>
Cc: Lars-Peter Clausen <lars@metafoo.de>
Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org>
Cc: Wan ZongShun <mcuos.com@gmail.com>
Cc: Eric Miao <eric.y.miao@gmail.com>
Cc: Jassi Brar <jassi.brar@samsung.com>
Cc: Daniel Gloeckner <dg@emlix.com>
Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-06 11:28:29 -04:00

226 lines
5.5 KiB
C

/*
* ALSA SoC CQ0093 Voice Codec Driver for DaVinci platforms
*
* Copyright (C) 2010 Texas Instruments, Inc
*
* Author: Miguel Aguilar <miguel.aguilar@ridgerun.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/init.h>
#include <linux/io.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/platform_device.h>
#include <linux/device.h>
#include <linux/slab.h>
#include <linux/clk.h>
#include <linux/mfd/davinci_voicecodec.h>
#include <linux/spi/spi.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/soc-dai.h>
#include <sound/soc-dapm.h>
#include <sound/initval.h>
#include <mach/dm365.h>
static inline unsigned int cq93vc_read(struct snd_soc_codec *codec,
unsigned int reg)
{
struct davinci_vc *davinci_vc = codec->control_data;
return readl(davinci_vc->base + reg);
}
static inline int cq93vc_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int value)
{
struct davinci_vc *davinci_vc = codec->control_data;
writel(value, davinci_vc->base + reg);
return 0;
}
static const struct snd_kcontrol_new cq93vc_snd_controls[] = {
SOC_SINGLE("PGA Capture Volume", DAVINCI_VC_REG05, 0, 0x03, 0),
SOC_SINGLE("Mono DAC Playback Volume", DAVINCI_VC_REG09, 0, 0x3f, 0),
};
static int cq93vc_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
u8 reg = cq93vc_read(codec, DAVINCI_VC_REG09) & ~DAVINCI_VC_REG09_MUTE;
if (mute)
cq93vc_write(codec, DAVINCI_VC_REG09,
reg | DAVINCI_VC_REG09_MUTE);
else
cq93vc_write(codec, DAVINCI_VC_REG09, reg);
return 0;
}
static int cq93vc_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_codec *codec = codec_dai->codec;
struct davinci_vc *davinci_vc = codec->control_data;
switch (freq) {
case 22579200:
case 27000000:
case 33868800:
davinci_vc->cq93vc.sysclk = freq;
return 0;
}
return -EINVAL;
}
static int cq93vc_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
switch (level) {
case SND_SOC_BIAS_ON:
cq93vc_write(codec, DAVINCI_VC_REG12,
DAVINCI_VC_REG12_POWER_ALL_ON);
break;
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
cq93vc_write(codec, DAVINCI_VC_REG12,
DAVINCI_VC_REG12_POWER_ALL_OFF);
break;
case SND_SOC_BIAS_OFF:
/* force all power off */
cq93vc_write(codec, DAVINCI_VC_REG12,
DAVINCI_VC_REG12_POWER_ALL_OFF);
break;
}
codec->dapm.bias_level = level;
return 0;
}
#define CQ93VC_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000)
#define CQ93VC_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE)
static struct snd_soc_dai_ops cq93vc_dai_ops = {
.digital_mute = cq93vc_mute,
.set_sysclk = cq93vc_set_dai_sysclk,
};
static struct snd_soc_dai_driver cq93vc_dai = {
.name = "cq93vc-hifi",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 2,
.rates = CQ93VC_RATES,
.formats = CQ93VC_FORMATS,},
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = CQ93VC_RATES,
.formats = CQ93VC_FORMATS,},
.ops = &cq93vc_dai_ops,
};
static int cq93vc_resume(struct snd_soc_codec *codec)
{
cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return 0;
}
static int cq93vc_probe(struct snd_soc_codec *codec)
{
struct davinci_vc *davinci_vc = codec->dev->platform_data;
davinci_vc->cq93vc.codec = codec;
codec->control_data = davinci_vc;
/* Set controls */
snd_soc_add_controls(codec, cq93vc_snd_controls,
ARRAY_SIZE(cq93vc_snd_controls));
/* Off, with power on */
cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return 0;
}
static int cq93vc_remove(struct snd_soc_codec *codec)
{
cq93vc_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_cq93vc = {
.read = cq93vc_read,
.write = cq93vc_write,
.set_bias_level = cq93vc_set_bias_level,
.probe = cq93vc_probe,
.remove = cq93vc_remove,
.resume = cq93vc_resume,
};
static int cq93vc_platform_probe(struct platform_device *pdev)
{
return snd_soc_register_codec(&pdev->dev,
&soc_codec_dev_cq93vc, &cq93vc_dai, 1);
}
static int cq93vc_platform_remove(struct platform_device *pdev)
{
snd_soc_unregister_codec(&pdev->dev);
return 0;
}
static struct platform_driver cq93vc_codec_driver = {
.driver = {
.name = "cq93vc-codec",
.owner = THIS_MODULE,
},
.probe = cq93vc_platform_probe,
.remove = __devexit_p(cq93vc_platform_remove),
};
static int __init cq93vc_init(void)
{
return platform_driver_register(&cq93vc_codec_driver);
}
module_init(cq93vc_init);
static void __exit cq93vc_exit(void)
{
platform_driver_unregister(&cq93vc_codec_driver);
}
module_exit(cq93vc_exit);
MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC CQ0093 Voice Codec Driver");
MODULE_AUTHOR("Miguel Aguilar");
MODULE_LICENSE("GPL");