Techwizz/kernel-fxtec-pro1x
1
0
Fork 0
Code Issues Pull requests Projects Releases Packages Wiki Activity
kernel-fxtec-pro1x/sound/soc/imx/wm1133-ev1.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

303 lines
8.8 KiB
C
Raw Blame History

/*
* wm1133-ev1.c - Audio for WM1133-EV1 on i.MX31ADS
*
* Copyright (c) 2010 Wolfson Microelectronics plc
* Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
*
* Based on an earlier driver for the same hardware by Liam Girdwood.
*
* 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 <linux/clk.h>
#include <sound/core.h>
#include <sound/jack.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <mach/audmux.h>
#include "imx-ssi.h"
#include "../codecs/wm8350.h"
/* There is a silicon mic on the board optionally connected via a solder pad
* SP1. Define this to enable it.
*/
#undef USE_SIMIC
struct _wm8350_audio {
unsigned int channels;
snd_pcm_format_t format;
unsigned int rate;
unsigned int sysclk;
unsigned int bclkdiv;
unsigned int clkdiv;
unsigned int lr_rate;
};
/* in order of power consumption per rate (lowest first) */
static const struct _wm8350_audio wm8350_audio[] = {
/* 16bit mono modes */
{1, SNDRV_PCM_FORMAT_S16_LE, 8000, 12288000 >> 1,
WM8350_BCLK_DIV_48, WM8350_DACDIV_3, 16,},
/* 16 bit stereo modes */
{2, SNDRV_PCM_FORMAT_S16_LE, 8000, 12288000,
WM8350_BCLK_DIV_48, WM8350_DACDIV_6, 32,},
{2, SNDRV_PCM_FORMAT_S16_LE, 16000, 12288000,
WM8350_BCLK_DIV_24, WM8350_DACDIV_3, 32,},
{2, SNDRV_PCM_FORMAT_S16_LE, 32000, 12288000,
WM8350_BCLK_DIV_12, WM8350_DACDIV_1_5, 32,},
{2, SNDRV_PCM_FORMAT_S16_LE, 48000, 12288000,
WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
{2, SNDRV_PCM_FORMAT_S16_LE, 96000, 24576000,
WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
{2, SNDRV_PCM_FORMAT_S16_LE, 11025, 11289600,
WM8350_BCLK_DIV_32, WM8350_DACDIV_4, 32,},
{2, SNDRV_PCM_FORMAT_S16_LE, 22050, 11289600,
WM8350_BCLK_DIV_16, WM8350_DACDIV_2, 32,},
{2, SNDRV_PCM_FORMAT_S16_LE, 44100, 11289600,
WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
{2, SNDRV_PCM_FORMAT_S16_LE, 88200, 22579200,
WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
/* 24bit stereo modes */
{2, SNDRV_PCM_FORMAT_S24_LE, 48000, 12288000,
WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
{2, SNDRV_PCM_FORMAT_S24_LE, 96000, 24576000,
WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
{2, SNDRV_PCM_FORMAT_S24_LE, 44100, 11289600,
WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
{2, SNDRV_PCM_FORMAT_S24_LE, 88200, 22579200,
WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
};
static int wm1133_ev1_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 i, found = 0;
snd_pcm_format_t format = params_format(params);
unsigned int rate = params_rate(params);
unsigned int channels = params_channels(params);
u32 dai_format;
/* find the correct audio parameters */
for (i = 0; i < ARRAY_SIZE(wm8350_audio); i++) {
if (rate == wm8350_audio[i].rate &&
format == wm8350_audio[i].format &&
channels == wm8350_audio[i].channels) {
found = 1;
break;
}
}
if (!found)
return -EINVAL;
/* codec FLL input is 14.75 MHz from MCLK */
snd_soc_dai_set_pll(codec_dai, 0, 0, 14750000, wm8350_audio[i].sysclk);
dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM;
/* set codec DAI configuration */
snd_soc_dai_set_fmt(codec_dai, dai_format);
/* set cpu DAI configuration */
snd_soc_dai_set_fmt(cpu_dai, dai_format);
/* TODO: The SSI driver should figure this out for us */
switch (channels) {
case 2:
snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0);
break;
case 1:
snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffe, 0xffffffe, 1, 0);
break;
default:
return -EINVAL;
}
/* set MCLK as the codec system clock for DAC and ADC */
snd_soc_dai_set_sysclk(codec_dai, WM8350_MCLK_SEL_PLL_MCLK,
wm8350_audio[i].sysclk, SND_SOC_CLOCK_IN);
/* set codec BCLK division for sample rate */
snd_soc_dai_set_clkdiv(codec_dai, WM8350_BCLK_CLKDIV,
wm8350_audio[i].bclkdiv);
/* DAI is synchronous and clocked with DAC LRCLK & ADC LRC */
snd_soc_dai_set_clkdiv(codec_dai,
WM8350_DACLR_CLKDIV, wm8350_audio[i].lr_rate);
snd_soc_dai_set_clkdiv(codec_dai,
WM8350_ADCLR_CLKDIV, wm8350_audio[i].lr_rate);
/* now configure DAC and ADC clocks */
snd_soc_dai_set_clkdiv(codec_dai,
WM8350_DAC_CLKDIV, wm8350_audio[i].clkdiv);
snd_soc_dai_set_clkdiv(codec_dai,
WM8350_ADC_CLKDIV, wm8350_audio[i].clkdiv);
return 0;
}
static struct snd_soc_ops wm1133_ev1_ops = {
.hw_params = wm1133_ev1_hw_params,
};
static const struct snd_soc_dapm_widget wm1133_ev1_widgets[] = {
#ifdef USE_SIMIC
SND_SOC_DAPM_MIC("SiMIC", NULL),
#endif
SND_SOC_DAPM_MIC("Mic1 Jack", NULL),
SND_SOC_DAPM_MIC("Mic2 Jack", NULL),
SND_SOC_DAPM_LINE("Line In Jack", NULL),
SND_SOC_DAPM_LINE("Line Out Jack", NULL),
SND_SOC_DAPM_HP("Headphone Jack", NULL),
};
/* imx32ads soc_card audio map */
static const struct snd_soc_dapm_route wm1133_ev1_map[] = {
#ifdef USE_SIMIC
/* SiMIC --> IN1LN (with automatic bias) via SP1 */
{ "IN1LN", NULL, "Mic Bias" },
{ "Mic Bias", NULL, "SiMIC" },
#endif
/* Mic 1 Jack --> IN1LN and IN1LP (with automatic bias) */
{ "IN1LN", NULL, "Mic Bias" },
{ "IN1LP", NULL, "Mic1 Jack" },
{ "Mic Bias", NULL, "Mic1 Jack" },
/* Mic 2 Jack --> IN1RN and IN1RP (with automatic bias) */
{ "IN1RN", NULL, "Mic Bias" },
{ "IN1RP", NULL, "Mic2 Jack" },
{ "Mic Bias", NULL, "Mic2 Jack" },
/* Line in Jack --> AUX (L+R) */
{ "IN3R", NULL, "Line In Jack" },
{ "IN3L", NULL, "Line In Jack" },
/* Out1 --> Headphone Jack */
{ "Headphone Jack", NULL, "OUT1R" },
{ "Headphone Jack", NULL, "OUT1L" },
/* Out1 --> Line Out Jack */
{ "Line Out Jack", NULL, "OUT2R" },
{ "Line Out Jack", NULL, "OUT2L" },
};
static struct snd_soc_jack hp_jack;
static struct snd_soc_jack_pin hp_jack_pins[] = {
{ .pin = "Headphone Jack", .mask = SND_JACK_HEADPHONE },
};
static struct snd_soc_jack mic_jack;
static struct snd_soc_jack_pin mic_jack_pins[] = {
{ .pin = "Mic1 Jack", .mask = SND_JACK_MICROPHONE },
{ .pin = "Mic2 Jack", .mask = SND_JACK_MICROPHONE },
};
static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
snd_soc_dapm_new_controls(codec, wm1133_ev1_widgets,
ARRAY_SIZE(wm1133_ev1_widgets));
snd_soc_dapm_add_routes(codec, wm1133_ev1_map,
ARRAY_SIZE(wm1133_ev1_map));
/* Headphone jack detection */
snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, &hp_jack);
snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins),
hp_jack_pins);
wm8350_hp_jack_detect(codec, WM8350_JDR, &hp_jack, SND_JACK_HEADPHONE);
/* Microphone jack detection */
snd_soc_jack_new(codec, "Microphone",
SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack);
snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
mic_jack_pins);
wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE,
SND_JACK_BTN_0);
snd_soc_dapm_force_enable_pin(codec, "Mic Bias");
return 0;
}
static struct snd_soc_dai_link wm1133_ev1_dai = {
.name = "WM1133-EV1",
.stream_name = "Audio",
.cpu_dai_name = "imx-ssi-dai.0",
.codec_dai_name = "wm8350-hifi",
.platform_name = "imx-fiq-pcm-audio.0",
.codec_name = "wm8350-codec.0-0x1a",
.init = wm1133_ev1_init,
.ops = &wm1133_ev1_ops,
.symmetric_rates = 1,
};
static struct snd_soc_card wm1133_ev1 = {
.name = "WM1133-EV1",
.dai_link = &wm1133_ev1_dai,
.num_links = 1,
};
static struct platform_device *wm1133_ev1_snd_device;
static int __init wm1133_ev1_audio_init(void)
{
int ret;
unsigned int ptcr, pdcr;
/* SSI0 mastered by port 5 */
ptcr = MXC_AUDMUX_V2_PTCR_SYN |
MXC_AUDMUX_V2_PTCR_TFSDIR |
MXC_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT5_SSI_PINS_5) |
MXC_AUDMUX_V2_PTCR_TCLKDIR |
MXC_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT5_SSI_PINS_5);
pdcr = MXC_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT5_SSI_PINS_5);
mxc_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0, ptcr, pdcr);
ptcr = MXC_AUDMUX_V2_PTCR_SYN;
pdcr = MXC_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0);
mxc_audmux_v2_configure_port(MX31_AUDMUX_PORT5_SSI_PINS_5, ptcr, pdcr);
wm1133_ev1_snd_device = platform_device_alloc("soc-audio", -1);
if (!wm1133_ev1_snd_device)
return -ENOMEM;
platform_set_drvdata(wm1133_ev1_snd_device, &wm1133_ev1);
ret = platform_device_add(wm1133_ev1_snd_device);
if (ret)
platform_device_put(wm1133_ev1_snd_device);
return ret;
}
module_init(wm1133_ev1_audio_init);
static void __exit wm1133_ev1_audio_exit(void)
{
platform_device_unregister(wm1133_ev1_snd_device);
}
module_exit(wm1133_ev1_audio_exit);
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
MODULE_DESCRIPTION("Audio for WM1133-EV1 on i.MX31ADS");
MODULE_LICENSE("GPL");
Reference in a new issue View git blame Copy permalink