linux/sound/soc/pxa/magician.c

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// SPDX-License-Identifier: GPL-2.0-or-later
/*
* 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>
*/
#include <linux/module.h>
#include <linux/timer.h>
#include <linux/interrupt.h>
#include <linux/platform_device.h>
#include <linux/delay.h>
#include <linux/gpio/consumer.h>
#include <linux/i2c.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.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 struct gpio_desc *gpiod_spk_power, *gpiod_ep_power, *gpiod_mic_power;
static struct gpio_desc *gpiod_in_sel0, *gpiod_in_sel1;
static void magician_ext_control(struct snd_soc_dapm_context *dapm)
{
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 13:53:46 +00:00
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 = asoc_substream_to_rtd(substream);
/* 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 = asoc_substream_to_rtd(substream);
struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
unsigned int width;
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_BC_FC);
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_BP_FP);
if (ret < 0)
return ret;
width = snd_pcm_format_physical_width(params_format(params));
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;
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 = asoc_substream_to_rtd(substream);
struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
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_BC_FC);
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_BP_FP);
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 const struct snd_soc_ops magician_capture_ops = {
.startup = magician_startup,
.hw_params = magician_capture_hw_params,
};
static const 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:
gpiod_set_value(gpiod_in_sel1, 1);
break;
case MAGICIAN_MIC_EXT:
gpiod_set_value(gpiod_in_sel1, 0);
}
return 1;
}
static int magician_spk_power(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
gpiod_set_value(gpiod_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)
{
gpiod_set_value(gpiod_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)
{
gpiod_set_value(gpiod_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 */
SND_SOC_DAILINK_DEFS(playback,
DAILINK_COMP_ARRAY(COMP_CPU("pxa-ssp-dai.0")),
DAILINK_COMP_ARRAY(COMP_CODEC("uda1380-codec.0-0018",
"uda1380-hifi-playback")),
DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio")));
SND_SOC_DAILINK_DEFS(capture,
DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-i2s")),
DAILINK_COMP_ARRAY(COMP_CODEC("uda1380-codec.0-0018",
"uda1380-hifi-capture")),
DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio")));
static struct snd_soc_dai_link magician_dai[] = {
{
.name = "uda1380",
.stream_name = "UDA1380 Playback",
.ops = &magician_playback_ops,
SND_SOC_DAILINK_REG(playback),
},
{
.name = "uda1380",
.stream_name = "UDA1380 Capture",
.ops = &magician_capture_ops,
SND_SOC_DAILINK_REG(capture),
}
};
/* 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 int magician_audio_probe(struct platform_device *pdev)
{
struct device *dev = &pdev->dev;
gpiod_spk_power = devm_gpiod_get(dev, "SPK_POWER", GPIOD_OUT_LOW);
if (IS_ERR(gpiod_spk_power))
return PTR_ERR(gpiod_spk_power);
gpiod_ep_power = devm_gpiod_get(dev, "EP_POWER", GPIOD_OUT_LOW);
if (IS_ERR(gpiod_ep_power))
return PTR_ERR(gpiod_ep_power);
gpiod_mic_power = devm_gpiod_get(dev, "MIC_POWER", GPIOD_OUT_LOW);
if (IS_ERR(gpiod_mic_power))
return PTR_ERR(gpiod_mic_power);
gpiod_in_sel0 = devm_gpiod_get(dev, "IN_SEL0", GPIOD_OUT_HIGH);
if (IS_ERR(gpiod_in_sel0))
return PTR_ERR(gpiod_in_sel0);
gpiod_in_sel1 = devm_gpiod_get(dev, "IN_SEL1", GPIOD_OUT_LOW);
if (IS_ERR(gpiod_in_sel1))
return PTR_ERR(gpiod_in_sel1);
snd_soc_card_magician.dev = &pdev->dev;
return devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_magician);
}
static struct platform_driver magician_audio_driver = {
.driver.name = "magician-audio",
.driver.pm = &snd_soc_pm_ops,
.probe = magician_audio_probe,
};
module_platform_driver(magician_audio_driver);
MODULE_AUTHOR("Philipp Zabel");
MODULE_DESCRIPTION("ALSA SoC Magician");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:magician-audio");