linux/sound/soc/pxa/z2.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

241 lines
5.4 KiB
C

/*
* linux/sound/soc/pxa/z2.c
*
* SoC Audio driver for Aeronix Zipit Z2
*
* Copyright (C) 2009 Ken McGuire <kenm@desertweyr.com>
* Copyright (C) 2010 Marek Vasut <marek.vasut@gmail.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/timer.h>
#include <linux/interrupt.h>
#include <linux/platform_device.h>
#include <linux/gpio.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/jack.h>
#include <asm/mach-types.h>
#include <mach/hardware.h>
#include <mach/audio.h>
#include <mach/z2.h>
#include "../codecs/wm8750.h"
#include "pxa2xx-i2s.h"
static struct snd_soc_card snd_soc_z2;
static int z2_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
unsigned int clk = 0;
int ret = 0;
switch (params_rate(params)) {
case 8000:
case 16000:
case 48000:
case 96000:
clk = 12288000;
break;
case 11025:
case 22050:
case 44100:
clk = 11289600;
break;
}
/* 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;
/* set the codec system clock for DAC and ADC */
ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk,
SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
/* set the I2S system clock as input (unused) */
ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
return 0;
}
static struct snd_soc_jack hs_jack;
/* Headset jack detection DAPM pins */
static struct snd_soc_jack_pin hs_jack_pins[] = {
{
.pin = "Mic Jack",
.mask = SND_JACK_MICROPHONE,
},
{
.pin = "Headphone Jack",
.mask = SND_JACK_HEADPHONE,
},
};
/* Headset jack detection gpios */
static struct snd_soc_jack_gpio hs_jack_gpios[] = {
{
.gpio = GPIO37_ZIPITZ2_HEADSET_DETECT,
.name = "hsdet-gpio",
.report = SND_JACK_HEADSET,
.debounce_time = 200,
},
};
/* z2 machine dapm widgets */
static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_MIC("Mic Jack", NULL),
SND_SOC_DAPM_SPK("Ext Spk", NULL),
/* headset is a mic and mono headphone */
SND_SOC_DAPM_HP("Headset Jack", NULL),
};
/* Z2 machine audio_map */
static const struct snd_soc_dapm_route audio_map[] = {
/* headphone connected to LOUT1, ROUT1 */
{"Headphone Jack", NULL, "LOUT1"},
{"Headphone Jack", NULL, "ROUT1"},
/* ext speaker connected to LOUT2, ROUT2 */
{"Ext Spk", NULL , "ROUT2"},
{"Ext Spk", NULL , "LOUT2"},
/* mic is connected to R input 2 - with bias */
{"RINPUT2", NULL, "Mic Bias"},
{"Mic Bias", NULL, "Mic Jack"},
};
/*
* Logic for a wm8750 as connected on a Z2 Device
*/
static int z2_wm8750_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
/* NC codec pins */
snd_soc_dapm_disable_pin(dapm, "LINPUT3");
snd_soc_dapm_disable_pin(dapm, "RINPUT3");
snd_soc_dapm_disable_pin(dapm, "OUT3");
snd_soc_dapm_disable_pin(dapm, "MONO");
/* Add z2 specific widgets */
snd_soc_dapm_new_controls(dapm, wm8750_dapm_widgets,
ARRAY_SIZE(wm8750_dapm_widgets));
/* Set up z2 specific audio paths */
snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
ret = snd_soc_dapm_sync(dapm);
if (ret)
goto err;
/* Jack detection API stuff */
ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET,
&hs_jack);
if (ret)
goto err;
ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
hs_jack_pins);
if (ret)
goto err;
ret = snd_soc_jack_add_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios),
hs_jack_gpios);
if (ret)
goto err;
return 0;
err:
return ret;
}
static struct snd_soc_ops z2_ops = {
.hw_params = z2_hw_params,
};
/* z2 digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link z2_dai = {
.name = "wm8750",
.stream_name = "WM8750",
.cpu_dai_name = "pxa2xx-i2s",
.codec_dai_name = "wm8750-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm8750-codec.0-001a",
.init = z2_wm8750_init,
.ops = &z2_ops,
};
/* z2 audio machine driver */
static struct snd_soc_card snd_soc_z2 = {
.name = "Z2",
.dai_link = &z2_dai,
.num_links = 1,
};
static struct platform_device *z2_snd_device;
static int __init z2_init(void)
{
int ret;
if (!machine_is_zipit2())
return -ENODEV;
z2_snd_device = platform_device_alloc("soc-audio", -1);
if (!z2_snd_device)
return -ENOMEM;
platform_set_drvdata(z2_snd_device, &snd_soc_z2);
ret = platform_device_add(z2_snd_device);
if (ret)
platform_device_put(z2_snd_device);
return ret;
}
static void __exit z2_exit(void)
{
platform_device_unregister(z2_snd_device);
}
module_init(z2_init);
module_exit(z2_exit);
MODULE_AUTHOR("Ken McGuire <kenm@desertweyr.com>, "
"Marek Vasut <marek.vasut@gmail.com>");
MODULE_DESCRIPTION("ALSA SoC ZipitZ2");
MODULE_LICENSE("GPL");