WSL2-Linux-Kernel/sound/soc/ti/rx51.c

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// SPDX-License-Identifier: GPL-2.0-only
/*
* rx51.c -- SoC audio for Nokia RX-51
*
* Copyright (C) 2008 - 2009 Nokia Corporation
*
* Contact: Peter Ujfalusi <peter.ujfalusi@ti.com>
* Eduardo Valentin <eduardo.valentin@nokia.com>
* Jarkko Nikula <jarkko.nikula@bitmer.com>
*/
#include <linux/delay.h>
#include <linux/gpio.h>
#include <linux/platform_device.h>
#include <linux/gpio/consumer.h>
#include <linux/module.h>
#include <sound/core.h>
#include <sound/jack.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <linux/platform_data/asoc-ti-mcbsp.h>
#include <asm/mach-types.h>
#include "omap-mcbsp.h"
enum {
RX51_JACK_DISABLED,
RX51_JACK_TVOUT, /* tv-out with stereo output */
RX51_JACK_HP, /* headphone: stereo output, no mic */
RX51_JACK_HS, /* headset: stereo output with mic */
};
struct rx51_audio_pdata {
struct gpio_desc *tvout_selection_gpio;
struct gpio_desc *jack_detection_gpio;
struct gpio_desc *eci_sw_gpio;
struct gpio_desc *speaker_amp_gpio;
};
static int rx51_spk_func;
static int rx51_dmic_func;
static int rx51_jack_func;
static void rx51_ext_control(struct snd_soc_dapm_context *dapm)
{
struct snd_soc_card *card = dapm->card;
struct rx51_audio_pdata *pdata = snd_soc_card_get_drvdata(card);
int hp = 0, hs = 0, tvout = 0;
switch (rx51_jack_func) {
case RX51_JACK_TVOUT:
tvout = 1;
hp = 1;
break;
case RX51_JACK_HS:
hs = 1;
/* fall through */
case RX51_JACK_HP:
hp = 1;
break;
}
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 16:53:46 +03:00
snd_soc_dapm_mutex_lock(dapm);
if (rx51_spk_func)
snd_soc_dapm_enable_pin_unlocked(dapm, "Ext Spk");
else
snd_soc_dapm_disable_pin_unlocked(dapm, "Ext Spk");
if (rx51_dmic_func)
snd_soc_dapm_enable_pin_unlocked(dapm, "DMic");
else
snd_soc_dapm_disable_pin_unlocked(dapm, "DMic");
if (hp)
snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
else
snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
if (hs)
snd_soc_dapm_enable_pin_unlocked(dapm, "HS Mic");
else
snd_soc_dapm_disable_pin_unlocked(dapm, "HS Mic");
gpiod_set_value(pdata->tvout_selection_gpio, tvout);
snd_soc_dapm_sync_unlocked(dapm);
snd_soc_dapm_mutex_unlock(dapm);
}
static int rx51_startup(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
struct snd_soc_card *card = rtd->card;
snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2);
rx51_ext_control(&card->dapm);
return 0;
}
static int rx51_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);
/* Set the codec system clock for DAC and ADC */
return snd_soc_dai_set_sysclk(codec_dai, 0, 19200000,
SND_SOC_CLOCK_IN);
}
static const struct snd_soc_ops rx51_ops = {
.startup = rx51_startup,
.hw_params = rx51_hw_params,
};
static int rx51_get_spk(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.enumerated.item[0] = rx51_spk_func;
return 0;
}
static int rx51_set_spk(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
if (rx51_spk_func == ucontrol->value.enumerated.item[0])
return 0;
rx51_spk_func = ucontrol->value.enumerated.item[0];
rx51_ext_control(&card->dapm);
return 1;
}
static int rx51_spk_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
struct snd_soc_dapm_context *dapm = w->dapm;
struct snd_soc_card *card = dapm->card;
struct rx51_audio_pdata *pdata = snd_soc_card_get_drvdata(card);
gpiod_set_raw_value_cansleep(pdata->speaker_amp_gpio,
!!SND_SOC_DAPM_EVENT_ON(event));
return 0;
}
static int rx51_get_input(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.enumerated.item[0] = rx51_dmic_func;
return 0;
}
static int rx51_set_input(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
if (rx51_dmic_func == ucontrol->value.enumerated.item[0])
return 0;
rx51_dmic_func = ucontrol->value.enumerated.item[0];
rx51_ext_control(&card->dapm);
return 1;
}
static int rx51_get_jack(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.enumerated.item[0] = rx51_jack_func;
return 0;
}
static int rx51_set_jack(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
if (rx51_jack_func == ucontrol->value.enumerated.item[0])
return 0;
rx51_jack_func = ucontrol->value.enumerated.item[0];
rx51_ext_control(&card->dapm);
return 1;
}
static struct snd_soc_jack rx51_av_jack;
static struct snd_soc_jack_gpio rx51_av_jack_gpios[] = {
{
.name = "avdet-gpio",
.report = SND_JACK_HEADSET,
.invert = 1,
.debounce_time = 200,
},
};
static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = {
SND_SOC_DAPM_SPK("Ext Spk", rx51_spk_event),
SND_SOC_DAPM_MIC("DMic", NULL),
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_MIC("HS Mic", NULL),
SND_SOC_DAPM_LINE("FM Transmitter", NULL),
SND_SOC_DAPM_SPK("Earphone", NULL),
};
static const struct snd_soc_dapm_route audio_map[] = {
{"Ext Spk", NULL, "HPLOUT"},
{"Ext Spk", NULL, "HPROUT"},
{"Ext Spk", NULL, "HPLCOM"},
{"Ext Spk", NULL, "HPRCOM"},
{"FM Transmitter", NULL, "LLOUT"},
{"FM Transmitter", NULL, "RLOUT"},
{"Headphone Jack", NULL, "TPA6130A2 HPLEFT"},
{"Headphone Jack", NULL, "TPA6130A2 HPRIGHT"},
{"TPA6130A2 LEFTIN", NULL, "LLOUT"},
{"TPA6130A2 RIGHTIN", NULL, "RLOUT"},
{"DMic Rate 64", NULL, "DMic"},
{"DMic", NULL, "Mic Bias"},
{"b LINE2R", NULL, "MONO_LOUT"},
{"Earphone", NULL, "b HPLOUT"},
{"LINE1L", NULL, "HS Mic"},
{"HS Mic", NULL, "b Mic Bias"},
};
static const char * const spk_function[] = {"Off", "On"};
static const char * const input_function[] = {"ADC", "Digital Mic"};
static const char * const jack_function[] = {
"Off", "TV-OUT", "Headphone", "Headset"
};
static const struct soc_enum rx51_enum[] = {
SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function),
SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(input_function), input_function),
SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(jack_function), jack_function),
};
static const struct snd_kcontrol_new aic34_rx51_controls[] = {
SOC_ENUM_EXT("Speaker Function", rx51_enum[0],
rx51_get_spk, rx51_set_spk),
SOC_ENUM_EXT("Input Select", rx51_enum[1],
rx51_get_input, rx51_set_input),
SOC_ENUM_EXT("Jack Function", rx51_enum[2],
rx51_get_jack, rx51_set_jack),
SOC_DAPM_PIN_SWITCH("FM Transmitter"),
SOC_DAPM_PIN_SWITCH("Earphone"),
};
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>
2010-03-17 23:15:21 +03:00
static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_card *card = rtd->card;
struct rx51_audio_pdata *pdata = snd_soc_card_get_drvdata(card);
int err;
snd_soc_limit_volume(card, "TPA6130A2 Headphone Playback Volume", 42);
err = omap_mcbsp_st_add_controls(rtd, 2);
if (err < 0) {
dev_err(card->dev, "Failed to add MCBSP controls\n");
return err;
}
/* AV jack detection */
err = snd_soc_card_jack_new(rtd->card, "AV Jack",
SND_JACK_HEADSET | SND_JACK_VIDEOOUT,
&rx51_av_jack, NULL, 0);
if (err) {
dev_err(card->dev, "Failed to add AV Jack\n");
return err;
}
/* prepare gpio for snd_soc_jack_add_gpios */
rx51_av_jack_gpios[0].gpio = desc_to_gpio(pdata->jack_detection_gpio);
devm_gpiod_put(card->dev, pdata->jack_detection_gpio);
err = snd_soc_jack_add_gpios(&rx51_av_jack,
ARRAY_SIZE(rx51_av_jack_gpios),
rx51_av_jack_gpios);
if (err) {
dev_err(card->dev, "Failed to add GPIOs\n");
return err;
}
return err;
}
/* Digital audio interface glue - connects codec <--> CPU */
SND_SOC_DAILINK_DEFS(aic34,
DAILINK_COMP_ARRAY(COMP_CPU("omap-mcbsp.2")),
DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic3x-codec.2-0018",
"tlv320aic3x-hifi")),
DAILINK_COMP_ARRAY(COMP_PLATFORM("omap-mcbsp.2")));
static struct snd_soc_dai_link rx51_dai[] = {
{
.name = "TLV320AIC34",
.stream_name = "AIC34",
.dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF |
SND_SOC_DAIFMT_CBM_CFM,
.init = rx51_aic34_init,
.ops = &rx51_ops,
SND_SOC_DAILINK_REG(aic34),
},
};
static struct snd_soc_aux_dev rx51_aux_dev[] = {
{
.dlc = COMP_AUX("tlv320aic3x-codec.2-0019"),
},
{
.dlc = COMP_AUX("tpa6130a2.2-0060"),
},
};
static struct snd_soc_codec_conf rx51_codec_conf[] = {
{
.dlc = COMP_CODEC_CONF("tlv320aic3x-codec.2-0019"),
.name_prefix = "b",
},
{
.dlc = COMP_CODEC_CONF("tpa6130a2.2-0060"),
.name_prefix = "TPA6130A2",
},
};
/* Audio card */
static struct snd_soc_card rx51_sound_card = {
.name = "RX-51",
.owner = THIS_MODULE,
.dai_link = rx51_dai,
.num_links = ARRAY_SIZE(rx51_dai),
.aux_dev = rx51_aux_dev,
.num_aux_devs = ARRAY_SIZE(rx51_aux_dev),
.codec_conf = rx51_codec_conf,
.num_configs = ARRAY_SIZE(rx51_codec_conf),
.fully_routed = true,
.controls = aic34_rx51_controls,
.num_controls = ARRAY_SIZE(aic34_rx51_controls),
.dapm_widgets = aic34_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(aic34_dapm_widgets),
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
};
static int rx51_soc_probe(struct platform_device *pdev)
{
struct rx51_audio_pdata *pdata;
struct device_node *np = pdev->dev.of_node;
struct snd_soc_card *card = &rx51_sound_card;
int err;
if (!machine_is_nokia_rx51() && !of_machine_is_compatible("nokia,omap3-n900"))
return -ENODEV;
card->dev = &pdev->dev;
if (np) {
struct device_node *dai_node;
dai_node = of_parse_phandle(np, "nokia,cpu-dai", 0);
if (!dai_node) {
dev_err(&pdev->dev, "McBSP node is not provided\n");
return -EINVAL;
}
rx51_dai[0].cpus->dai_name = NULL;
rx51_dai[0].platforms->name = NULL;
rx51_dai[0].cpus->of_node = dai_node;
rx51_dai[0].platforms->of_node = dai_node;
dai_node = of_parse_phandle(np, "nokia,audio-codec", 0);
if (!dai_node) {
dev_err(&pdev->dev, "Codec node is not provided\n");
return -EINVAL;
}
rx51_dai[0].codecs->name = NULL;
rx51_dai[0].codecs->of_node = dai_node;
dai_node = of_parse_phandle(np, "nokia,audio-codec", 1);
if (!dai_node) {
dev_err(&pdev->dev, "Auxiliary Codec node is not provided\n");
return -EINVAL;
}
rx51_aux_dev[0].dlc.name = NULL;
rx51_aux_dev[0].dlc.of_node = dai_node;
rx51_codec_conf[0].dlc.name = NULL;
rx51_codec_conf[0].dlc.of_node = dai_node;
dai_node = of_parse_phandle(np, "nokia,headphone-amplifier", 0);
if (!dai_node) {
dev_err(&pdev->dev, "Headphone amplifier node is not provided\n");
return -EINVAL;
}
rx51_aux_dev[1].dlc.name = NULL;
rx51_aux_dev[1].dlc.of_node = dai_node;
rx51_codec_conf[1].dlc.name = NULL;
rx51_codec_conf[1].dlc.of_node = dai_node;
}
pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL);
if (pdata == NULL)
return -ENOMEM;
snd_soc_card_set_drvdata(card, pdata);
pdata->tvout_selection_gpio = devm_gpiod_get(card->dev,
"tvout-selection",
GPIOD_OUT_LOW);
if (IS_ERR(pdata->tvout_selection_gpio)) {
dev_err(card->dev, "could not get tvout selection gpio\n");
return PTR_ERR(pdata->tvout_selection_gpio);
}
pdata->jack_detection_gpio = devm_gpiod_get(card->dev,
"jack-detection",
GPIOD_ASIS);
if (IS_ERR(pdata->jack_detection_gpio)) {
dev_err(card->dev, "could not get jack detection gpio\n");
return PTR_ERR(pdata->jack_detection_gpio);
}
pdata->eci_sw_gpio = devm_gpiod_get(card->dev, "eci-switch",
GPIOD_OUT_HIGH);
if (IS_ERR(pdata->eci_sw_gpio)) {
dev_err(card->dev, "could not get eci switch gpio\n");
return PTR_ERR(pdata->eci_sw_gpio);
}
pdata->speaker_amp_gpio = devm_gpiod_get(card->dev,
"speaker-amplifier",
GPIOD_OUT_LOW);
if (IS_ERR(pdata->speaker_amp_gpio)) {
dev_err(card->dev, "could not get speaker enable gpio\n");
return PTR_ERR(pdata->speaker_amp_gpio);
}
err = devm_snd_soc_register_card(card->dev, card);
if (err) {
dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", err);
return err;
}
return 0;
}
#if defined(CONFIG_OF)
static const struct of_device_id rx51_audio_of_match[] = {
{ .compatible = "nokia,n900-audio", },
{},
};
MODULE_DEVICE_TABLE(of, rx51_audio_of_match);
#endif
static struct platform_driver rx51_soc_driver = {
.driver = {
.name = "rx51-audio",
.of_match_table = of_match_ptr(rx51_audio_of_match),
},
.probe = rx51_soc_probe,
};
module_platform_driver(rx51_soc_driver);
MODULE_AUTHOR("Nokia Corporation");
MODULE_DESCRIPTION("ALSA SoC Nokia RX-51");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:rx51-audio");