mirror of
https://github.com/torvalds/linux.git
synced 2024-11-05 19:41:54 +00:00
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>
147 lines
3.6 KiB
C
147 lines
3.6 KiB
C
/*
|
|
* snappercl15.c -- SoC audio for Bluewater Systems Snapper CL15 module
|
|
*
|
|
* Copyright (C) 2008 Bluewater Systems Ltd
|
|
* Author: Ryan Mallon <ryan@bluewatersys.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/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 "../codecs/tlv320aic23.h"
|
|
#include "ep93xx-pcm.h"
|
|
|
|
#define CODEC_CLOCK 5644800
|
|
|
|
static int snappercl15_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;
|
|
|
|
err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
|
|
SND_SOC_DAIFMT_NB_IF |
|
|
SND_SOC_DAIFMT_CBS_CFS);
|
|
|
|
err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
|
|
SND_SOC_DAIFMT_NB_IF |
|
|
SND_SOC_DAIFMT_CBS_CFS);
|
|
if (err)
|
|
return err;
|
|
|
|
err = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK,
|
|
SND_SOC_CLOCK_IN);
|
|
if (err)
|
|
return err;
|
|
|
|
err = snd_soc_dai_set_sysclk(cpu_dai, 0, CODEC_CLOCK,
|
|
SND_SOC_CLOCK_OUT);
|
|
if (err)
|
|
return err;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static struct snd_soc_ops snappercl15_ops = {
|
|
.hw_params = snappercl15_hw_params,
|
|
};
|
|
|
|
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 snappercl15_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd)
|
|
{
|
|
struct snd_soc_codec *codec = rtd->codec;
|
|
|
|
snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
|
|
ARRAY_SIZE(tlv320aic23_dapm_widgets));
|
|
|
|
snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
|
|
return 0;
|
|
}
|
|
|
|
static struct snd_soc_dai_link snappercl15_dai = {
|
|
.name = "tlv320aic23",
|
|
.stream_name = "AIC23",
|
|
.cpu_dai_name = "ep93xx-i2s",
|
|
.codec_dai_name = "tlv320aic23-hifi",
|
|
.codec_name = "tlv320aic23-codec.0-001a",
|
|
.platform_name = "ep93xx-pcm-audio",
|
|
.init = snappercl15_tlv320aic23_init,
|
|
.ops = &snappercl15_ops,
|
|
};
|
|
|
|
static struct snd_soc_card snd_soc_snappercl15 = {
|
|
.name = "Snapper CL15",
|
|
.dai_link = &snappercl15_dai,
|
|
.num_links = 1,
|
|
};
|
|
|
|
static struct platform_device *snappercl15_snd_device;
|
|
|
|
static int __init snappercl15_init(void)
|
|
{
|
|
int ret;
|
|
|
|
if (!machine_is_snapper_cl15())
|
|
return -ENODEV;
|
|
|
|
ret = ep93xx_i2s_acquire(EP93XX_SYSCON_DEVCFG_I2SONAC97,
|
|
EP93XX_SYSCON_I2SCLKDIV_ORIDE |
|
|
EP93XX_SYSCON_I2SCLKDIV_SPOL);
|
|
if (ret)
|
|
return ret;
|
|
|
|
snappercl15_snd_device = platform_device_alloc("soc-audio", -1);
|
|
if (!snappercl15_snd_device)
|
|
return -ENOMEM;
|
|
|
|
platform_set_drvdata(snappercl15_snd_device, &snd_soc_snappercl15);
|
|
ret = platform_device_add(snappercl15_snd_device);
|
|
if (ret)
|
|
platform_device_put(snappercl15_snd_device);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void __exit snappercl15_exit(void)
|
|
{
|
|
platform_device_unregister(snappercl15_snd_device);
|
|
ep93xx_i2s_release();
|
|
}
|
|
|
|
module_init(snappercl15_init);
|
|
module_exit(snappercl15_exit);
|
|
|
|
MODULE_AUTHOR("Ryan Mallon <ryan@bluewatersys.com>");
|
|
MODULE_DESCRIPTION("ALSA SoC Snapper CL15");
|
|
MODULE_LICENSE("GPL");
|
|
|