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linux_dsm_epyc7002/sound/soc/pxa/magician.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

559 lines
14 KiB
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/*
* SoC audio for HTC Magician
*
* Copyright (c) 2006 Philipp Zabel <philipp.zabel@gmail.com>
*
* based on spitz.c,
* Authors: 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/timer.h>
#include <linux/interrupt.h>
#include <linux/platform_device.h>
#include <linux/delay.h>
#include <linux/gpio.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 <sound/uda1380.h>
#include <mach/magician.h>
#include <asm/mach-types.h>
#include "../codecs/uda1380.h"
#include "pxa2xx-i2s.h"
#include "pxa-ssp.h"
#define MAGICIAN_MIC 0
#define MAGICIAN_MIC_EXT 1
static int magician_hp_switch;
static int magician_spk_switch = 1;
static int magician_in_sel = MAGICIAN_MIC;
static void magician_ext_control(struct snd_soc_codec *codec)
{
if (magician_spk_switch)
snd_soc_dapm_enable_pin(codec, "Speaker");
else
snd_soc_dapm_disable_pin(codec, "Speaker");
if (magician_hp_switch)
snd_soc_dapm_enable_pin(codec, "Headphone Jack");
else
snd_soc_dapm_disable_pin(codec, "Headphone Jack");
switch (magician_in_sel) {
case MAGICIAN_MIC:
snd_soc_dapm_disable_pin(codec, "Headset Mic");
snd_soc_dapm_enable_pin(codec, "Call Mic");
break;
case MAGICIAN_MIC_EXT:
snd_soc_dapm_disable_pin(codec, "Call Mic");
snd_soc_dapm_enable_pin(codec, "Headset Mic");
break;
}
snd_soc_dapm_sync(codec);
}
static int magician_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
/* check the jack status at stream startup */
magician_ext_control(codec);
return 0;
}
/*
* Magician uses SSP port for playback.
*/
static int magician_playback_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 acps, acds, width, rate;
unsigned int div4 = PXA_SSP_CLK_SCDB_4;
int ret = 0;
rate = params_rate(params);
width = snd_pcm_format_physical_width(params_format(params));
/*
* rate = SSPSCLK / (2 * width(16 or 32))
* SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1)
*/
switch (params_rate(params)) {
case 8000:
/* off by a factor of 2: bug in the PXA27x audio clock? */
acps = 32842000;
switch (width) {
case 16:
/* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */
acds = PXA_SSP_CLK_AUDIO_DIV_16;
break;
default: /* 32 */
/* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */
acds = PXA_SSP_CLK_AUDIO_DIV_8;
}
break;
case 11025:
acps = 5622000;
switch (width) {
case 16:
/* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */
acds = PXA_SSP_CLK_AUDIO_DIV_4;
break;
default: /* 32 */
/* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */
acds = PXA_SSP_CLK_AUDIO_DIV_2;
}
break;
case 22050:
acps = 5622000;
switch (width) {
case 16:
/* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */
acds = PXA_SSP_CLK_AUDIO_DIV_2;
break;
default: /* 32 */
/* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */
acds = PXA_SSP_CLK_AUDIO_DIV_1;
}
break;
case 44100:
acps = 5622000;
switch (width) {
case 16:
/* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */
acds = PXA_SSP_CLK_AUDIO_DIV_2;
break;
default: /* 32 */
/* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */
acds = PXA_SSP_CLK_AUDIO_DIV_1;
}
break;
case 48000:
acps = 12235000;
switch (width) {
case 16:
/* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */
acds = PXA_SSP_CLK_AUDIO_DIV_2;
break;
default: /* 32 */
/* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */
acds = PXA_SSP_CLK_AUDIO_DIV_1;
}
break;
case 96000:
default:
acps = 12235000;
switch (width) {
case 16:
/* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */
acds = PXA_SSP_CLK_AUDIO_DIV_1;
break;
default: /* 32 */
/* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */
acds = PXA_SSP_CLK_AUDIO_DIV_2;
div4 = PXA_SSP_CLK_SCDB_1;
break;
}
break;
}
/* set codec DAI configuration */
ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB |
SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0)
return ret;
/* set cpu DAI configuration */
ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0)
return ret;
ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width);
if (ret < 0)
return ret;
/* set audio clock as clock source */
ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0,
SND_SOC_CLOCK_OUT);
if (ret < 0)
return ret;
/* set the SSP audio system clock ACDS divider */
ret = snd_soc_dai_set_clkdiv(cpu_dai,
PXA_SSP_AUDIO_DIV_ACDS, acds);
if (ret < 0)
return ret;
/* set the SSP audio system clock SCDB divider4 */
ret = snd_soc_dai_set_clkdiv(cpu_dai,
PXA_SSP_AUDIO_DIV_SCDB, div4);
if (ret < 0)
return ret;
/* set SSP audio pll clock */
ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, acps);
if (ret < 0)
return ret;
return 0;
}
/*
* Magician uses I2S for capture.
*/
static int magician_capture_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;
/* set codec DAI configuration */
ret = snd_soc_dai_set_fmt(codec_dai,
SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0)
return ret;
/* set cpu DAI configuration */
ret = snd_soc_dai_set_fmt(cpu_dai,
SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0)
return ret;
/* set the I2S system clock as output */
ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
SND_SOC_CLOCK_OUT);
if (ret < 0)
return ret;
return 0;
}
static struct snd_soc_ops magician_capture_ops = {
.startup = magician_startup,
.hw_params = magician_capture_hw_params,
};
static struct snd_soc_ops magician_playback_ops = {
.startup = magician_startup,
.hw_params = magician_playback_hw_params,
};
static int magician_get_hp(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.integer.value[0] = magician_hp_switch;
return 0;
}
static int magician_set_hp(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
if (magician_hp_switch == ucontrol->value.integer.value[0])
return 0;
magician_hp_switch = ucontrol->value.integer.value[0];
magician_ext_control(codec);
return 1;
}
static int magician_get_spk(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.integer.value[0] = magician_spk_switch;
return 0;
}
static int magician_set_spk(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
if (magician_spk_switch == ucontrol->value.integer.value[0])
return 0;
magician_spk_switch = ucontrol->value.integer.value[0];
magician_ext_control(codec);
return 1;
}
static int magician_get_input(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.integer.value[0] = magician_in_sel;
return 0;
}
static int magician_set_input(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
if (magician_in_sel == ucontrol->value.integer.value[0])
return 0;
magician_in_sel = ucontrol->value.integer.value[0];
switch (magician_in_sel) {
case MAGICIAN_MIC:
gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 1);
break;
case MAGICIAN_MIC_EXT:
gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 0);
}
return 1;
}
static int magician_spk_power(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, SND_SOC_DAPM_EVENT_ON(event));
return 0;
}
static int magician_hp_power(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
gpio_set_value(EGPIO_MAGICIAN_EP_POWER, SND_SOC_DAPM_EVENT_ON(event));
return 0;
}
static int magician_mic_bias(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, SND_SOC_DAPM_EVENT_ON(event));
return 0;
}
/* magician machine dapm widgets */
static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
SND_SOC_DAPM_HP("Headphone Jack", magician_hp_power),
SND_SOC_DAPM_SPK("Speaker", magician_spk_power),
SND_SOC_DAPM_MIC("Call Mic", magician_mic_bias),
SND_SOC_DAPM_MIC("Headset Mic", magician_mic_bias),
};
/* magician machine audio_map */
static const struct snd_soc_dapm_route audio_map[] = {
/* Headphone connected to VOUTL, VOUTR */
{"Headphone Jack", NULL, "VOUTL"},
{"Headphone Jack", NULL, "VOUTR"},
/* Speaker connected to VOUTL, VOUTR */
{"Speaker", NULL, "VOUTL"},
{"Speaker", NULL, "VOUTR"},
/* Mics are connected to VINM */
{"VINM", NULL, "Headset Mic"},
{"VINM", NULL, "Call Mic"},
};
static const char *input_select[] = {"Call Mic", "Headset Mic"};
static const struct soc_enum magician_in_sel_enum =
SOC_ENUM_SINGLE_EXT(2, input_select);
static const struct snd_kcontrol_new uda1380_magician_controls[] = {
SOC_SINGLE_BOOL_EXT("Headphone Switch",
(unsigned long)&magician_hp_switch,
magician_get_hp, magician_set_hp),
SOC_SINGLE_BOOL_EXT("Speaker Switch",
(unsigned long)&magician_spk_switch,
magician_get_spk, magician_set_spk),
SOC_ENUM_EXT("Input Select", magician_in_sel_enum,
magician_get_input, magician_set_input),
};
/*
* Logic for a uda1380 as connected on a HTC Magician
*/
static int magician_uda1380_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
int err;
/* NC codec pins */
snd_soc_dapm_nc_pin(codec, "VOUTLHP");
snd_soc_dapm_nc_pin(codec, "VOUTRHP");
/* FIXME: is anything connected here? */
snd_soc_dapm_nc_pin(codec, "VINL");
snd_soc_dapm_nc_pin(codec, "VINR");
/* Add magician specific controls */
err = snd_soc_add_controls(codec, uda1380_magician_controls,
ARRAY_SIZE(uda1380_magician_controls));
if (err < 0)
return err;
/* Add magician specific widgets */
snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets,
ARRAY_SIZE(uda1380_dapm_widgets));
/* Set up magician specific audio path interconnects */
snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
snd_soc_dapm_sync(codec);
return 0;
}
/* magician digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link magician_dai[] = {
{
.name = "uda1380",
.stream_name = "UDA1380 Playback",
.cpu_dai_name = "pxa-ssp-dai.0",
.codec_dai_name = "uda1380-hifi-playback",
.platform_name = "pxa-pcm-audio",
.codec_name = "uda1380-codec.0-0018",
.init = magician_uda1380_init,
.ops = &magician_playback_ops,
},
{
.name = "uda1380",
.stream_name = "UDA1380 Capture",
.cpu_dai_name = "pxa-i2s",
.codec_dai_name = "uda1380-hifi-capture",
.platform_name = "pxa-pcm-audio",
.codec_name = "uda1380-codec.0-0018",
.ops = &magician_capture_ops,
}
};
/* magician audio machine driver */
static struct snd_soc_card snd_soc_card_magician = {
.name = "Magician",
.dai_link = magician_dai,
.num_links = ARRAY_SIZE(magician_dai),
};
static struct platform_device *magician_snd_device;
/*
* FIXME: move into magician board file once merged into the pxa tree
*/
static struct uda1380_platform_data uda1380_info = {
.gpio_power = EGPIO_MAGICIAN_CODEC_POWER,
.gpio_reset = EGPIO_MAGICIAN_CODEC_RESET,
.dac_clk = UDA1380_DAC_CLK_WSPLL,
};
static struct i2c_board_info i2c_board_info[] = {
{
I2C_BOARD_INFO("uda1380", 0x18),
.platform_data = &uda1380_info,
},
};
static int __init magician_init(void)
{
int ret;
struct i2c_adapter *adapter;
struct i2c_client *client;
if (!machine_is_magician())
return -ENODEV;
adapter = i2c_get_adapter(0);
if (!adapter)
return -ENODEV;
client = i2c_new_device(adapter, i2c_board_info);
i2c_put_adapter(adapter);
if (!client)
return -ENODEV;
ret = gpio_request(EGPIO_MAGICIAN_SPK_POWER, "SPK_POWER");
if (ret)
goto err_request_spk;
ret = gpio_request(EGPIO_MAGICIAN_EP_POWER, "EP_POWER");
if (ret)
goto err_request_ep;
ret = gpio_request(EGPIO_MAGICIAN_MIC_POWER, "MIC_POWER");
if (ret)
goto err_request_mic;
ret = gpio_request(EGPIO_MAGICIAN_IN_SEL0, "IN_SEL0");
if (ret)
goto err_request_in_sel0;
ret = gpio_request(EGPIO_MAGICIAN_IN_SEL1, "IN_SEL1");
if (ret)
goto err_request_in_sel1;
gpio_set_value(EGPIO_MAGICIAN_IN_SEL0, 0);
magician_snd_device = platform_device_alloc("soc-audio", -1);
if (!magician_snd_device) {
ret = -ENOMEM;
goto err_pdev;
}
platform_set_drvdata(magician_snd_device, &snd_soc_card_magician);
ret = platform_device_add(magician_snd_device);
if (ret) {
platform_device_put(magician_snd_device);
goto err_pdev;
}
return 0;
err_pdev:
gpio_free(EGPIO_MAGICIAN_IN_SEL1);
err_request_in_sel1:
gpio_free(EGPIO_MAGICIAN_IN_SEL0);
err_request_in_sel0:
gpio_free(EGPIO_MAGICIAN_MIC_POWER);
err_request_mic:
gpio_free(EGPIO_MAGICIAN_EP_POWER);
err_request_ep:
gpio_free(EGPIO_MAGICIAN_SPK_POWER);
err_request_spk:
return ret;
}
static void __exit magician_exit(void)
{
platform_device_unregister(magician_snd_device);
gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, 0);
gpio_set_value(EGPIO_MAGICIAN_EP_POWER, 0);
gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, 0);
gpio_free(EGPIO_MAGICIAN_IN_SEL1);
gpio_free(EGPIO_MAGICIAN_IN_SEL0);
gpio_free(EGPIO_MAGICIAN_MIC_POWER);
gpio_free(EGPIO_MAGICIAN_EP_POWER);
gpio_free(EGPIO_MAGICIAN_SPK_POWER);
}
module_init(magician_init);
module_exit(magician_exit);
MODULE_AUTHOR("Philipp Zabel");
MODULE_DESCRIPTION("ALSA SoC Magician");
MODULE_LICENSE("GPL");
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