AuxXxilium/linux_dsm_epyc7002
1
0
Fork 0
mirror of https://github.com/AuxXxilium/linux_dsm_epyc7002.git synced 2024-12-02 22:36:41 +07:00
Code Issues Actions Packages Projects Releases Wiki Activity
2f24111a08
linux_dsm_epyc7002/sound/soc/davinci/davinci-evm.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

307 lines
8.1 KiB
C
Raw Blame History

/*
* ASoC driver for TI DAVINCI EVM platform
*
* Author: Vladimir Barinov, <vbarinov@embeddedalley.com>
* Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.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/i2c.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <asm/dma.h>
#include <asm/mach-types.h>
#include <mach/asp.h>
#include <mach/edma.h>
#include <mach/mux.h>
#include "../codecs/tlv320aic3x.h"
#include "davinci-pcm.h"
#include "davinci-i2s.h"
#include "davinci-mcasp.h"
#define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \
SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF)
static int evm_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 ret = 0;
unsigned sysclk;
/* ASP1 on DM355 EVM is clocked by an external oscillator */
if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm() ||
machine_is_davinci_dm365_evm())
sysclk = 27000000;
/* ASP0 in DM6446 EVM is clocked by U55, as configured by
* board-dm644x-evm.c using GPIOs from U18. There are six
* options; here we "know" we use a 48 KHz sample rate.
*/
else if (machine_is_davinci_evm())
sysclk = 12288000;
else if (machine_is_davinci_da830_evm() ||
machine_is_davinci_da850_evm())
sysclk = 24576000;
else
return -EINVAL;
/* set codec DAI configuration */
ret = snd_soc_dai_set_fmt(codec_dai, AUDIO_FORMAT);
if (ret < 0)
return ret;
/* set cpu DAI configuration */
ret = snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT);
if (ret < 0)
return ret;
/* set the codec system clock */
ret = snd_soc_dai_set_sysclk(codec_dai, 0, sysclk, SND_SOC_CLOCK_OUT);
if (ret < 0)
return ret;
return 0;
}
static int evm_spdif_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 *cpu_dai = rtd->cpu_dai;
/* set cpu DAI configuration */
return snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT);
}
static struct snd_soc_ops evm_ops = {
.hw_params = evm_hw_params,
};
static struct snd_soc_ops evm_spdif_ops = {
.hw_params = evm_spdif_hw_params,
};
/* davinci-evm machine dapm widgets */
static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_LINE("Line Out", NULL),
SND_SOC_DAPM_MIC("Mic Jack", NULL),
SND_SOC_DAPM_LINE("Line In", NULL),
};
/* davinci-evm machine audio_mapnections to the codec pins */
static const struct snd_soc_dapm_route audio_map[] = {
/* Headphone connected to HPLOUT, HPROUT */
{"Headphone Jack", NULL, "HPLOUT"},
{"Headphone Jack", NULL, "HPROUT"},
/* Line Out connected to LLOUT, RLOUT */
{"Line Out", NULL, "LLOUT"},
{"Line Out", NULL, "RLOUT"},
/* Mic connected to (MIC3L | MIC3R) */
{"MIC3L", NULL, "Mic Bias 2V"},
{"MIC3R", NULL, "Mic Bias 2V"},
{"Mic Bias 2V", NULL, "Mic Jack"},
/* Line In connected to (LINE1L | LINE2L), (LINE1R | LINE2R) */
{"LINE1L", NULL, "Line In"},
{"LINE2L", NULL, "Line In"},
{"LINE1R", NULL, "Line In"},
{"LINE2R", NULL, "Line In"},
};
/* Logic for a aic3x as connected on a davinci-evm */
static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
/* Add davinci-evm specific widgets */
snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets,
ARRAY_SIZE(aic3x_dapm_widgets));
/* Set up davinci-evm specific audio path audio_map */
snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
/* not connected */
snd_soc_dapm_disable_pin(codec, "MONO_LOUT");
snd_soc_dapm_disable_pin(codec, "HPLCOM");
snd_soc_dapm_disable_pin(codec, "HPRCOM");
/* always connected */
snd_soc_dapm_enable_pin(codec, "Headphone Jack");
snd_soc_dapm_enable_pin(codec, "Line Out");
snd_soc_dapm_enable_pin(codec, "Mic Jack");
snd_soc_dapm_enable_pin(codec, "Line In");
snd_soc_dapm_sync(codec);
return 0;
}
/* davinci-evm digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link evm_dai = {
.name = "TLV320AIC3X",
.stream_name = "AIC3X",
.cpu_dai_name = "davinci-mcasp.0",
.codec_dai_name = "tlv320aic3x-hifi",
.codec_name = "tlv320aic3x-codec.0-001a",
.platform_name = "davinci-pcm-audio",
.init = evm_aic3x_init,
.ops = &evm_ops,
};
static struct snd_soc_dai_link dm365_evm_dai = {
#ifdef CONFIG_SND_DM365_AIC3X_CODEC
.name = "TLV320AIC3X",
.stream_name = "AIC3X",
.cpu_dai_name = "davinci-i2s",
.codec_dai_name = "tlv320aic3x-hifi",
.init = evm_aic3x_init,
.codec_name = "tlv320aic3x-codec.0-001a",
.ops = &evm_ops,
#elif defined(CONFIG_SND_DM365_VOICE_CODEC)
.name = "Voice Codec - CQ93VC",
.stream_name = "CQ93",
.cpu_dai_name = "davinci-vcif",
.codec_dai_name = "cq93vc-hifi",
.codec_name = "cq93vc-codec",
#endif
.platform_name = "davinci-pcm-audio",
};
static struct snd_soc_dai_link dm6467_evm_dai[] = {
{
.name = "TLV320AIC3X",
.stream_name = "AIC3X",
.cpu_dai_name= "davinci-mcasp.0",
.codec_dai_name = "tlv320aic3x-hifi",
.platform_name ="davinci-pcm-audio",
.codec_name = "tlv320aic3x-codec.0-001a",
.init = evm_aic3x_init,
.ops = &evm_ops,
},
{
.name = "McASP",
.stream_name = "spdif",
.cpu_dai_name= "davinci-mcasp.1",
.codec_dai_name = "dit-hifi",
.codec_name = "spdif_dit",
.platform_name = "davinci-pcm-audio",
.ops = &evm_spdif_ops,
},
};
static struct snd_soc_dai_link da8xx_evm_dai = {
.name = "TLV320AIC3X",
.stream_name = "AIC3X",
.cpu_dai_name= "davinci-mcasp.0",
.codec_dai_name = "tlv320aic3x-hifi",
.codec_name = "tlv320aic3x-codec.0-001a",
.platform_name = "davinci-pcm-audio",
.init = evm_aic3x_init,
.ops = &evm_ops,
};
/* davinci dm6446, dm355 evm audio machine driver */
static struct snd_soc_card snd_soc_card_evm = {
.name = "DaVinci EVM",
.dai_link = &evm_dai,
.num_links = 1,
};
/* davinci dm365 evm audio machine driver */
static struct snd_soc_card dm365_snd_soc_card_evm = {
.name = "DaVinci DM365 EVM",
.dai_link = &dm365_evm_dai,
.num_links = 1,
};
/* davinci dm6467 evm audio machine driver */
static struct snd_soc_card dm6467_snd_soc_card_evm = {
.name = "DaVinci DM6467 EVM",
.dai_link = dm6467_evm_dai,
.num_links = ARRAY_SIZE(dm6467_evm_dai),
};
static struct snd_soc_card da830_snd_soc_card = {
.name = "DA830/OMAP-L137 EVM",
.dai_link = &da8xx_evm_dai,
.num_links = 1,
};
static struct snd_soc_card da850_snd_soc_card = {
.name = "DA850/OMAP-L138 EVM",
.dai_link = &da8xx_evm_dai,
.num_links = 1,
};
static struct platform_device *evm_snd_device;
static int __init evm_init(void)
{
struct snd_soc_card *evm_snd_dev_data;
int index;
int ret;
if (machine_is_davinci_evm()) {
evm_snd_dev_data = &snd_soc_card_evm;
index = 0;
} else if (machine_is_davinci_dm355_evm()) {
evm_snd_dev_data = &snd_soc_card_evm;
index = 1;
} else if (machine_is_davinci_dm365_evm()) {
evm_snd_dev_data = &dm365_snd_soc_card_evm;
index = 0;
} else if (machine_is_davinci_dm6467_evm()) {
evm_snd_dev_data = &dm6467_snd_soc_card_evm;
index = 0;
} else if (machine_is_davinci_da830_evm()) {
evm_snd_dev_data = &da830_snd_soc_card;
index = 1;
} else if (machine_is_davinci_da850_evm()) {
evm_snd_dev_data = &da850_snd_soc_card;
index = 0;
} else
return -EINVAL;
evm_snd_device = platform_device_alloc("soc-audio", index);
if (!evm_snd_device)
return -ENOMEM;
platform_set_drvdata(evm_snd_device, evm_snd_dev_data);
ret = platform_device_add(evm_snd_device);
if (ret)
platform_device_put(evm_snd_device);
return ret;
}
static void __exit evm_exit(void)
{
platform_device_unregister(evm_snd_device);
}
module_init(evm_init);
module_exit(evm_exit);
MODULE_AUTHOR("Vladimir Barinov");
MODULE_DESCRIPTION("TI DAVINCI EVM ASoC driver");
MODULE_LICENSE("GPL");
Reference in New Issue View Git Blame Copy Permalink