linux_dsm_epyc7002/sound/soc/pxa/zylonite.c
Liam Girdwood f0fba2ad1b ASoC: multi-component - ASoC Multi-Component Support
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>
2010-08-12 14:00:00 +01:00

293 lines
6.9 KiB
C

/*
* zylonite.c -- SoC audio for Zylonite
*
* Copyright 2008 Wolfson Microelectronics PLC.
* Author: Mark Brown <broonie@opensource.wolfsonmicro.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/moduleparam.h>
#include <linux/device.h>
#include <linux/clk.h>
#include <linux/i2c.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include "../codecs/wm9713.h"
#include "pxa2xx-ac97.h"
#include "pxa-ssp.h"
/*
* There is a physical switch SW15 on the board which changes the MCLK
* for the WM9713 between the standard AC97 master clock and the
* output of the CLK_POUT signal from the PXA.
*/
static int clk_pout;
module_param(clk_pout, int, 0);
MODULE_PARM_DESC(clk_pout, "Use CLK_POUT as WM9713 MCLK (SW15 on board).");
static struct clk *pout;
static struct snd_soc_card zylonite;
static const struct snd_soc_dapm_widget zylonite_dapm_widgets[] = {
SND_SOC_DAPM_HP("Headphone", NULL),
SND_SOC_DAPM_MIC("Headset Microphone", NULL),
SND_SOC_DAPM_MIC("Handset Microphone", NULL),
SND_SOC_DAPM_SPK("Multiactor", NULL),
SND_SOC_DAPM_SPK("Headset Earpiece", NULL),
};
/* Currently supported audio map */
static const struct snd_soc_dapm_route audio_map[] = {
/* Headphone output connected to HPL/HPR */
{ "Headphone", NULL, "HPL" },
{ "Headphone", NULL, "HPR" },
/* On-board earpiece */
{ "Headset Earpiece", NULL, "OUT3" },
/* Headphone mic */
{ "MIC2A", NULL, "Mic Bias" },
{ "Mic Bias", NULL, "Headset Microphone" },
/* On-board mic */
{ "MIC1", NULL, "Mic Bias" },
{ "Mic Bias", NULL, "Handset Microphone" },
/* Multiactor differentially connected over SPKL/SPKR */
{ "Multiactor", NULL, "SPKL" },
{ "Multiactor", NULL, "SPKR" },
};
static int zylonite_wm9713_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
if (clk_pout)
snd_soc_dai_set_pll(rtd->codec_dai, 0, 0,
clk_get_rate(pout), 0);
snd_soc_dapm_new_controls(codec, zylonite_dapm_widgets,
ARRAY_SIZE(zylonite_dapm_widgets));
snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
/* Static setup for now */
snd_soc_dapm_enable_pin(codec, "Headphone");
snd_soc_dapm_enable_pin(codec, "Headset Earpiece");
snd_soc_dapm_sync(codec);
return 0;
}
static int zylonite_voice_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 pll_out = 0;
unsigned int wm9713_div = 0;
int ret = 0;
int rate = params_rate(params);
int width = snd_pcm_format_physical_width(params_format(params));
/* Only support ratios that we can generate neatly from the AC97
* based master clock - in particular, this excludes 44.1kHz.
* In most applications the voice DAC will be used for telephony
* data so multiples of 8kHz will be the common case.
*/
switch (rate) {
case 8000:
wm9713_div = 12;
break;
case 16000:
wm9713_div = 6;
break;
case 48000:
wm9713_div = 2;
break;
default:
/* Don't support OSS emulation */
return -EINVAL;
}
/* Add 1 to the width for the leading clock cycle */
pll_out = rate * (width + 1) * 8;
ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1);
if (ret < 0)
return ret;
ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, pll_out);
if (ret < 0)
return ret;
if (clk_pout)
ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_PLL_DIV,
WM9713_PCMDIV(wm9713_div));
else
ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_DIV,
WM9713_PCMDIV(wm9713_div));
if (ret < 0)
return ret;
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;
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;
return 0;
}
static struct snd_soc_ops zylonite_voice_ops = {
.hw_params = zylonite_voice_hw_params,
};
static struct snd_soc_dai_link zylonite_dai[] = {
{
.name = "AC97",
.stream_name = "AC97 HiFi",
.codec_name = "wm9713-codec",
.platform_name = "pxa-pcm-audio",
.cpu_dai_name = "pxa-ac97.0",
.codec_name = "wm9713-hifi",
.init = zylonite_wm9713_init,
},
{
.name = "AC97 Aux",
.stream_name = "AC97 Aux",
.codec_name = "wm9713-codec",
.platform_name = "pxa-pcm-audio",
.cpu_dai_name = "pxa-ac97.1",
.codec_name = "wm9713-aux",
},
{
.name = "WM9713 Voice",
.stream_name = "WM9713 Voice",
.codec_name = "wm9713-codec",
.platform_name = "pxa-pcm-audio",
.cpu_dai_name = "pxa-ssp-dai.2",
.codec_name = "wm9713-voice",
.ops = &zylonite_voice_ops,
},
};
static int zylonite_probe(struct platform_device *pdev)
{
int ret;
if (clk_pout) {
pout = clk_get(NULL, "CLK_POUT");
if (IS_ERR(pout)) {
dev_err(&pdev->dev, "Unable to obtain CLK_POUT: %ld\n",
PTR_ERR(pout));
return PTR_ERR(pout);
}
ret = clk_enable(pout);
if (ret != 0) {
dev_err(&pdev->dev, "Unable to enable CLK_POUT: %d\n",
ret);
clk_put(pout);
return ret;
}
dev_dbg(&pdev->dev, "MCLK enabled at %luHz\n",
clk_get_rate(pout));
}
return 0;
}
static int zylonite_remove(struct platform_device *pdev)
{
if (clk_pout) {
clk_disable(pout);
clk_put(pout);
}
return 0;
}
static int zylonite_suspend_post(struct platform_device *pdev,
pm_message_t state)
{
if (clk_pout)
clk_disable(pout);
return 0;
}
static int zylonite_resume_pre(struct platform_device *pdev)
{
int ret = 0;
if (clk_pout) {
ret = clk_enable(pout);
if (ret != 0)
dev_err(&pdev->dev, "Unable to enable CLK_POUT: %d\n",
ret);
}
return ret;
}
static struct snd_soc_card zylonite = {
.name = "Zylonite",
.probe = &zylonite_probe,
.remove = &zylonite_remove,
.suspend_post = &zylonite_suspend_post,
.resume_pre = &zylonite_resume_pre,
.dai_link = zylonite_dai,
.num_links = ARRAY_SIZE(zylonite_dai),
.owner = THIS_MODULE,
};
static struct platform_device *zylonite_snd_ac97_device;
static int __init zylonite_init(void)
{
int ret;
zylonite_snd_ac97_device = platform_device_alloc("soc-audio", -1);
if (!zylonite_snd_ac97_device)
return -ENOMEM;
platform_set_drvdata(zylonite_snd_ac97_device, &zylonite);
ret = platform_device_add(zylonite_snd_ac97_device);
if (ret != 0)
platform_device_put(zylonite_snd_ac97_device);
return ret;
}
static void __exit zylonite_exit(void)
{
platform_device_unregister(zylonite_snd_ac97_device);
}
module_init(zylonite_init);
module_exit(zylonite_exit);
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
MODULE_DESCRIPTION("ALSA SoC WM9713 Zylonite");
MODULE_LICENSE("GPL");