renegade-project/linux-next
2
0
Fork 0
mirror of https://github.com/edk2-porting/linux-next.git synced 2024-12-26 06:04:14 +08:00
Code
9f3db423f9
linux-next/sound/soc/pxa/magician.c
Lars-Peter Clausen 92be58106e ASoC: pxa: Make static string arrays 'const 'char * const []' 
const char * const [] is the preferred type for static string arrays since
this states explicitly that the individual entries are not going to be
changed. Due to limitations in the ASoC API it was not possible to use it
for enum text arrays. Commit 87023ff74 ('ASoC: Declare const properly for
enum texts') changed this, but most drivers still use 'const char
* []' as the type for their enum text arrays.

Change these occurrences of 'static * const char * []' to 'static const
char * const []'.

The conversion was done automatically using the following coccinelle semantic
patch:
// <smpl>
@disable optional_qualifier@
identifier s;
@@
 static
-const char *
+const char * const
 s[] = ...;
// </smpl>

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
2016-11-13 11:36:59 +00:00

535 lines
13 KiB
C
Raw Blame History

/*
* 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/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_dapm_context *dapm)
{
snd_soc_dapm_mutex_lock(dapm);
if (magician_spk_switch)
snd_soc_dapm_enable_pin_unlocked(dapm, "Speaker");
else
snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker");
if (magician_hp_switch)
snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
else
snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
switch (magician_in_sel) {
case MAGICIAN_MIC:
snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Mic");
snd_soc_dapm_enable_pin_unlocked(dapm, "Call Mic");
break;
case MAGICIAN_MIC_EXT:
snd_soc_dapm_disable_pin_unlocked(dapm, "Call Mic");
snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Mic");
break;
}
snd_soc_dapm_sync_unlocked(dapm);
snd_soc_dapm_mutex_unlock(dapm);
}
static int magician_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
/* check the jack status at stream startup */
magician_ext_control(&rtd->card->dapm);
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;
unsigned int div4 = PXA_SSP_CLK_SCDB_4;
int ret = 0;
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_card *card = 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(&card->dapm);
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_card *card = 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(&card->dapm);
return 1;
}
static int magician_get_input(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.enumerated.item[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.enumerated.item[0])
return 0;
magician_in_sel = ucontrol->value.enumerated.item[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 * const 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),
};
/* 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",
.ops = &magician_playback_ops,
},
{
.name = "uda1380",
.stream_name = "UDA1380 Capture",
.cpu_dai_name = "pxa2xx-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",
.owner = THIS_MODULE,
.dai_link = magician_dai,
.num_links = ARRAY_SIZE(magician_dai),
.controls = uda1380_magician_controls,
.num_controls = ARRAY_SIZE(uda1380_magician_controls),
.dapm_widgets = uda1380_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets),
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
.fully_routed = true,
};
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");
View Git Blame Copy Permalink