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linux-next/sound/soc/omap/osk5912.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

220 lines
5.3 KiB
C

/*
* 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 <plat/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->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
int err;
/* Set codec DAI configuration */
err = snd_soc_dai_set_fmt(codec_dai,
SND_SOC_DAIFMT_DSP_B |
SND_SOC_DAIFMT_NB_NF |
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_B |
SND_SOC_DAIFMT_NB_NF |
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_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
/* Add osk5912 specific widgets */
snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets,
ARRAY_SIZE(tlv320aic23_dapm_widgets));
/* Set up osk5912 specific audio path audio_map */
snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
snd_soc_dapm_enable_pin(dapm, "Line In");
snd_soc_dapm_enable_pin(dapm, "Mic Jack");
snd_soc_dapm_sync(dapm);
return 0;
}
/* Digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link osk_dai = {
.name = "TLV320AIC23",
.stream_name = "AIC23",
.cpu_dai_name = "omap-mcbsp-dai.0",
.codec_dai_name = "tlv320aic23-hifi",
.platform_name = "omap-pcm-audio",
.codec_name = "tlv320aic23-codec",
.init = osk_tlv320aic23_init,
.ops = &osk_ops,
};
/* Audio machine driver */
static struct snd_soc_card snd_soc_card_osk = {
.name = "OSK5912",
.dai_link = &osk_dai,
.num_links = 1,
};
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, &snd_soc_card_osk);
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;
}
/*
* 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]\n",
(uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK);
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");