f0fba2ad1b
This patch extends the ASoC API to allow sound cards to have more than one
CODEC and more than one platform DMA controller. This is achieved by dividing
some current ASoC structures that contain both driver data and device data into
structures that only either contain device data or driver data. i.e.
struct snd_soc_codec ---> struct snd_soc_codec (device data)
+-> struct snd_soc_codec_driver (driver data)
struct snd_soc_platform ---> struct snd_soc_platform (device data)
+-> struct snd_soc_platform_driver (driver data)
struct snd_soc_dai ---> struct snd_soc_dai (device data)
+-> struct snd_soc_dai_driver (driver data)
struct snd_soc_device ---> deleted
This now allows ASoC to be more tightly aligned with the Linux driver model and
also means that every ASoC codec, platform and (platform) DAI is a kernel
device. ASoC component private data is now stored as device private data.
The ASoC sound card struct snd_soc_card has also been updated to store lists
of it's components rather than a pointer to a codec and platform. The PCM
runtime struct soc_pcm_runtime now has pointers to all its components.
This patch adds DAPM support for ASoC multi-component and removes struct
snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec
or runtime PCM level basis rather than using snd_soc_socdev.
Other notable multi-component changes:-
* Stream operations now de-reference less structures.
* close_delayed work() now runs on a DAI basis rather than looping all DAIs
in a card.
* PM suspend()/resume() operations can now handle N CODECs and Platforms
per sound card.
* Added soc_bind_dai_link() to bind the component devices to the sound card.
* Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove
DAI link components.
* sysfs entries can now be registered per component per card.
* snd_soc_new_pcms() functionailty rolled into dai_link_probe().
* snd_soc_register_codec() now does all the codec list and mutex init.
This patch changes the probe() and remove() of the CODEC drivers as follows:-
o Make CODEC driver a platform driver
o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core.
o Removed all static codec pointers (drivers now support > 1 codec dev)
o snd_soc_register_pcms() now done by core.
o snd_soc_register_dai() folded into snd_soc_register_codec().
CS4270 portions:
Acked-by: Timur Tabi <timur@freescale.com>
Some TLV320aic23 and Cirrus platform fixes.
Signed-off-by: Ryan Mallon <ryan@bluewatersys.com>
TI CODEC and OMAP fixes
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Samsung platform and misc fixes :-
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Reviewed-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
MPC8610 and PPC fixes.
Signed-off-by: Timur Tabi <timur@freescale.com>
i.MX fixes and some core fixes.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
J4740 platform fixes:-
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
CC: Tony Lindgren <tony@atomide.com>
CC: Nicolas Ferre <nicolas.ferre@atmel.com>
CC: Kevin Hilman <khilman@deeprootsystems.com>
CC: Sascha Hauer <s.hauer@pengutronix.de>
CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
CC: Kuninori Morimoto <morimoto.kuninori@renesas.com>
CC: Daniel Gloeckner <dg@emlix.com>
CC: Manuel Lauss <mano@roarinelk.homelinux.net>
CC: Mike Frysinger <vapier.adi@gmail.com>
CC: Arnaud Patard <apatard@mandriva.com>
CC: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
312 lines
7.2 KiB
C
312 lines
7.2 KiB
C
/*
|
|
* raumfeld_audio.c -- SoC audio for Raumfeld audio devices
|
|
*
|
|
* Copyright (c) 2009 Daniel Mack <daniel@caiaq.de>
|
|
*
|
|
* based on code from:
|
|
*
|
|
* Wolfson Microelectronics PLC.
|
|
* Openedhand Ltd.
|
|
* Liam Girdwood <lrg@slimlogic.co.uk>
|
|
* 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/i2c.h>
|
|
#include <linux/delay.h>
|
|
#include <linux/gpio.h>
|
|
#include <sound/pcm.h>
|
|
#include <sound/soc.h>
|
|
#include <sound/soc-dapm.h>
|
|
|
|
#include <asm/mach-types.h>
|
|
|
|
#include "pxa-ssp.h"
|
|
|
|
#define GPIO_SPDIF_RESET (38)
|
|
#define GPIO_MCLK_RESET (111)
|
|
#define GPIO_CODEC_RESET (120)
|
|
|
|
static struct i2c_client *max9486_client;
|
|
static struct i2c_board_info max9486_hwmon_info = {
|
|
I2C_BOARD_INFO("max9485", 0x63),
|
|
};
|
|
|
|
#define MAX9485_MCLK_FREQ_112896 0x22
|
|
#define MAX9485_MCLK_FREQ_122880 0x23
|
|
#define MAX9485_MCLK_FREQ_225792 0x32
|
|
#define MAX9485_MCLK_FREQ_245760 0x33
|
|
|
|
static void set_max9485_clk(char clk)
|
|
{
|
|
i2c_master_send(max9486_client, &clk, 1);
|
|
}
|
|
|
|
static void raumfeld_enable_audio(bool en)
|
|
{
|
|
if (en) {
|
|
gpio_set_value(GPIO_MCLK_RESET, 1);
|
|
|
|
/* wait some time to let the clocks become stable */
|
|
msleep(100);
|
|
|
|
gpio_set_value(GPIO_SPDIF_RESET, 1);
|
|
gpio_set_value(GPIO_CODEC_RESET, 1);
|
|
} else {
|
|
gpio_set_value(GPIO_MCLK_RESET, 0);
|
|
gpio_set_value(GPIO_SPDIF_RESET, 0);
|
|
gpio_set_value(GPIO_CODEC_RESET, 0);
|
|
}
|
|
}
|
|
|
|
/* CS4270 */
|
|
static int raumfeld_cs4270_startup(struct snd_pcm_substream *substream)
|
|
{
|
|
struct snd_soc_pcm_runtime *rtd = substream->private_data;
|
|
struct snd_soc_dai *codec_dai = rtd->codec_dai;
|
|
|
|
/* set freq to 0 to enable all possible codec sample rates */
|
|
return snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0);
|
|
}
|
|
|
|
static void raumfeld_cs4270_shutdown(struct snd_pcm_substream *substream)
|
|
{
|
|
struct snd_soc_pcm_runtime *rtd = substream->private_data;
|
|
struct snd_soc_dai *codec_dai = rtd->codec_dai;
|
|
|
|
/* set freq to 0 to enable all possible codec sample rates */
|
|
snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0);
|
|
}
|
|
|
|
static int raumfeld_cs4270_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 fmt, clk = 0;
|
|
int ret = 0;
|
|
|
|
switch (params_rate(params)) {
|
|
case 44100:
|
|
set_max9485_clk(MAX9485_MCLK_FREQ_112896);
|
|
clk = 11289600;
|
|
break;
|
|
case 48000:
|
|
set_max9485_clk(MAX9485_MCLK_FREQ_122880);
|
|
clk = 12288000;
|
|
break;
|
|
case 88200:
|
|
set_max9485_clk(MAX9485_MCLK_FREQ_225792);
|
|
clk = 22579200;
|
|
break;
|
|
case 96000:
|
|
set_max9485_clk(MAX9485_MCLK_FREQ_245760);
|
|
clk = 24576000;
|
|
break;
|
|
default:
|
|
return -EINVAL;
|
|
}
|
|
|
|
fmt = SND_SOC_DAIFMT_I2S |
|
|
SND_SOC_DAIFMT_NB_NF |
|
|
SND_SOC_DAIFMT_CBS_CFS;
|
|
|
|
/* setup the CODEC DAI */
|
|
ret = snd_soc_dai_set_fmt(codec_dai, fmt);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, 0);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
/* setup the CPU DAI */
|
|
ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static struct snd_soc_ops raumfeld_cs4270_ops = {
|
|
.startup = raumfeld_cs4270_startup,
|
|
.shutdown = raumfeld_cs4270_shutdown,
|
|
.hw_params = raumfeld_cs4270_hw_params,
|
|
};
|
|
|
|
static int raumfeld_line_suspend(struct platform_device *pdev, pm_message_t state)
|
|
{
|
|
raumfeld_enable_audio(false);
|
|
return 0;
|
|
}
|
|
|
|
static int raumfeld_line_resume(struct platform_device *pdev)
|
|
{
|
|
raumfeld_enable_audio(true);
|
|
return 0;
|
|
}
|
|
|
|
/* AK4104 */
|
|
|
|
static int raumfeld_ak4104_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 fmt, ret = 0, clk = 0;
|
|
|
|
switch (params_rate(params)) {
|
|
case 44100:
|
|
set_max9485_clk(MAX9485_MCLK_FREQ_112896);
|
|
clk = 11289600;
|
|
break;
|
|
case 48000:
|
|
set_max9485_clk(MAX9485_MCLK_FREQ_122880);
|
|
clk = 12288000;
|
|
break;
|
|
case 88200:
|
|
set_max9485_clk(MAX9485_MCLK_FREQ_225792);
|
|
clk = 22579200;
|
|
break;
|
|
case 96000:
|
|
set_max9485_clk(MAX9485_MCLK_FREQ_245760);
|
|
clk = 24576000;
|
|
break;
|
|
default:
|
|
return -EINVAL;
|
|
}
|
|
|
|
fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF;
|
|
|
|
/* setup the CODEC DAI */
|
|
ret = snd_soc_dai_set_fmt(codec_dai, fmt | SND_SOC_DAIFMT_CBS_CFS);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
/* setup the CPU DAI */
|
|
ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
ret = snd_soc_dai_set_fmt(cpu_dai, fmt | SND_SOC_DAIFMT_CBS_CFS);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static struct snd_soc_ops raumfeld_ak4104_ops = {
|
|
.hw_params = raumfeld_ak4104_hw_params,
|
|
};
|
|
|
|
static struct snd_soc_dai_link raumfeld_dai[] = {
|
|
{
|
|
.name = "ak4104",
|
|
.stream_name = "Playback",
|
|
.cpu_dai_name = "pxa-ssp-dai.1",
|
|
.codec_dai_name = "ak4104-hifi",
|
|
.platform_name = "pxa-pcm-audio",
|
|
.ops = &raumfeld_ak4104_ops,
|
|
.codec_name = "ak4104-codec.0",
|
|
},
|
|
{
|
|
.name = "CS4270",
|
|
.stream_name = "CS4270",
|
|
.cpu_dai_name = "pxa-ssp-dai.0",
|
|
.platform_name = "pxa-pcm-audio",
|
|
.codec_dai_name = "cs4270-hifi",
|
|
.codec_name = "cs4270-codec.0-0048",
|
|
.ops = &raumfeld_cs4270_ops,
|
|
},};
|
|
|
|
static struct snd_soc_card snd_soc_raumfeld = {
|
|
.name = "Raumfeld",
|
|
.dai_link = raumfeld_dai,
|
|
.suspend_post = raumfeld_line_suspend,
|
|
.resume_pre = raumfeld_line_resume,
|
|
.num_links = ARRAY_SIZE(raumfeld_dai),
|
|
};
|
|
|
|
static struct platform_device *raumfeld_audio_device;
|
|
|
|
static int __init raumfeld_audio_init(void)
|
|
{
|
|
int ret;
|
|
|
|
if (!machine_is_raumfeld_speaker() &&
|
|
!machine_is_raumfeld_connector())
|
|
return 0;
|
|
|
|
max9486_client = i2c_new_device(i2c_get_adapter(0),
|
|
&max9486_hwmon_info);
|
|
|
|
if (!max9486_client)
|
|
return -ENOMEM;
|
|
|
|
set_max9485_clk(MAX9485_MCLK_FREQ_122880);
|
|
|
|
/* Register LINE and SPDIF */
|
|
raumfeld_audio_device = platform_device_alloc("soc-audio", 0);
|
|
if (!raumfeld_audio_device)
|
|
return -ENOMEM;
|
|
|
|
platform_set_drvdata(raumfeld_audio_device,
|
|
&snd_soc_raumfeld);
|
|
ret = platform_device_add(raumfeld_audio_device);
|
|
|
|
/* no S/PDIF on Speakers */
|
|
if (machine_is_raumfeld_speaker())
|
|
return ret;
|
|
|
|
raumfeld_enable_audio(true);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void __exit raumfeld_audio_exit(void)
|
|
{
|
|
raumfeld_enable_audio(false);
|
|
|
|
platform_device_unregister(raumfeld_audio_device);
|
|
|
|
i2c_unregister_device(max9486_client);
|
|
|
|
gpio_free(GPIO_MCLK_RESET);
|
|
gpio_free(GPIO_CODEC_RESET);
|
|
gpio_free(GPIO_SPDIF_RESET);
|
|
}
|
|
|
|
module_init(raumfeld_audio_init);
|
|
module_exit(raumfeld_audio_exit);
|
|
|
|
/* Module information */
|
|
MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
|
|
MODULE_DESCRIPTION("Raumfeld audio SoC");
|
|
MODULE_LICENSE("GPL");
|