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audio_codec.c
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audio_codec.c
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/*
* APBridge ALSA SoC dummy codec driver
* Copyright 2016 Google Inc.
* Copyright 2016 Linaro Ltd.
*
* Released under the GPLv2 only.
*/
#include <linux/kernel.h>
#include <linux/module.h>
#include <linux/pm_runtime.h>
#include <sound/soc.h>
#include <sound/pcm_params.h>
#include <uapi/linux/input.h>
#include "audio_codec.h"
#include "audio_apbridgea.h"
#include "audio_manager.h"
static struct gbaudio_codec_info *gbcodec;
static struct gbaudio_data_connection *
find_data(struct gbaudio_module_info *module, int id)
{
struct gbaudio_data_connection *data;
list_for_each_entry(data, &module->data_list, list) {
if (id == data->id)
return data;
}
return NULL;
}
static struct gbaudio_stream_params *
find_dai_stream_params(struct gbaudio_codec_info *codec, int id, int stream)
{
struct gbaudio_codec_dai *dai;
list_for_each_entry(dai, &codec->dai_list, list) {
if (dai->id == id)
return &dai->params[stream];
}
return NULL;
}
static int gbaudio_module_enable_tx(struct gbaudio_codec_info *codec,
struct gbaudio_module_info *module, int id)
{
int module_state, ret = 0;
uint16_t data_cport, i2s_port, cportid;
uint8_t sig_bits, channels;
uint32_t format, rate;
struct gbaudio_data_connection *data;
struct gbaudio_stream_params *params;
/* find the dai */
data = find_data(module, id);
if (!data) {
dev_err(module->dev, "%d:DATA connection missing\n", id);
return -ENODEV;
}
module_state = data->state[SNDRV_PCM_STREAM_PLAYBACK];
params = find_dai_stream_params(codec, id, SNDRV_PCM_STREAM_PLAYBACK);
if (!params) {
dev_err(codec->dev, "Failed to fetch dai_stream pointer\n");
return -EINVAL;
}
/* register cport */
if (module_state < GBAUDIO_CODEC_STARTUP) {
i2s_port = 0; /* fixed for now */
cportid = data->connection->hd_cport_id;
ret = gb_audio_apbridgea_register_cport(data->connection,
i2s_port, cportid,
AUDIO_APBRIDGEA_DIRECTION_TX);
if (ret) {
dev_err_ratelimited(module->dev,
"reg_cport failed:%d\n", ret);
return ret;
}
data->state[SNDRV_PCM_STREAM_PLAYBACK] =
GBAUDIO_CODEC_STARTUP;
dev_dbg(module->dev, "Dynamic Register %d DAI\n", cportid);
}
/* hw_params */
if (module_state < GBAUDIO_CODEC_HWPARAMS) {
format = params->format;
channels = params->channels;
rate = params->rate;
sig_bits = params->sig_bits;
data_cport = data->connection->intf_cport_id;
ret = gb_audio_gb_set_pcm(module->mgmt_connection, data_cport,
format, rate, channels, sig_bits);
if (ret) {
dev_err_ratelimited(module->dev, "set_pcm failed:%d\n",
ret);
return ret;
}
data->state[SNDRV_PCM_STREAM_PLAYBACK] =
GBAUDIO_CODEC_HWPARAMS;
dev_dbg(module->dev, "Dynamic hw_params %d DAI\n", data_cport);
}
/* prepare */
if (module_state < GBAUDIO_CODEC_PREPARE) {
data_cport = data->connection->intf_cport_id;
ret = gb_audio_gb_set_tx_data_size(module->mgmt_connection,
data_cport, 192);
if (ret) {
dev_err_ratelimited(module->dev,
"set_tx_data_size failed:%d\n",
ret);
return ret;
}
ret = gb_audio_gb_activate_tx(module->mgmt_connection,
data_cport);
if (ret) {
dev_err_ratelimited(module->dev,
"activate_tx failed:%d\n", ret);
return ret;
}
data->state[SNDRV_PCM_STREAM_PLAYBACK] =
GBAUDIO_CODEC_PREPARE;
dev_dbg(module->dev, "Dynamic prepare %d DAI\n", data_cport);
}
return 0;
}
static int gbaudio_module_disable_tx(struct gbaudio_module_info *module, int id)
{
int ret;
uint16_t data_cport, cportid, i2s_port;
int module_state;
struct gbaudio_data_connection *data;
/* find the dai */
data = find_data(module, id);
if (!data) {
dev_err(module->dev, "%d:DATA connection missing\n", id);
return -ENODEV;
}
module_state = data->state[SNDRV_PCM_STREAM_PLAYBACK];
if (module_state > GBAUDIO_CODEC_HWPARAMS) {
data_cport = data->connection->intf_cport_id;
ret = gb_audio_gb_deactivate_tx(module->mgmt_connection,
data_cport);
if (ret) {
dev_err_ratelimited(module->dev,
"deactivate_tx failed:%d\n", ret);
return ret;
}
dev_dbg(module->dev, "Dynamic deactivate %d DAI\n", data_cport);
data->state[SNDRV_PCM_STREAM_PLAYBACK] =
GBAUDIO_CODEC_HWPARAMS;
}
if (module_state > GBAUDIO_CODEC_SHUTDOWN) {
i2s_port = 0; /* fixed for now */
cportid = data->connection->hd_cport_id;
ret = gb_audio_apbridgea_unregister_cport(data->connection,
i2s_port, cportid,
AUDIO_APBRIDGEA_DIRECTION_TX);
if (ret) {
dev_err_ratelimited(module->dev,
"unregister_cport failed:%d\n",
ret);
return ret;
}
dev_dbg(module->dev, "Dynamic Unregister %d DAI\n", cportid);
data->state[SNDRV_PCM_STREAM_PLAYBACK] =
GBAUDIO_CODEC_SHUTDOWN;
}
return 0;
}
static int gbaudio_module_enable_rx(struct gbaudio_codec_info *codec,
struct gbaudio_module_info *module, int id)
{
int module_state, ret = 0;
uint16_t data_cport, i2s_port, cportid;
uint8_t sig_bits, channels;
uint32_t format, rate;
struct gbaudio_data_connection *data;
struct gbaudio_stream_params *params;
/* find the dai */
data = find_data(module, id);
if (!data) {
dev_err(module->dev, "%d:DATA connection missing\n", id);
return -ENODEV;
}
module_state = data->state[SNDRV_PCM_STREAM_CAPTURE];
params = find_dai_stream_params(codec, id, SNDRV_PCM_STREAM_CAPTURE);
if (!params) {
dev_err(codec->dev, "Failed to fetch dai_stream pointer\n");
return -EINVAL;
}
/* register cport */
if (module_state < GBAUDIO_CODEC_STARTUP) {
i2s_port = 0; /* fixed for now */
cportid = data->connection->hd_cport_id;
ret = gb_audio_apbridgea_register_cport(data->connection,
i2s_port, cportid,
AUDIO_APBRIDGEA_DIRECTION_RX);
if (ret) {
dev_err_ratelimited(module->dev,
"reg_cport failed:%d\n", ret);
return ret;
}
data->state[SNDRV_PCM_STREAM_CAPTURE] =
GBAUDIO_CODEC_STARTUP;
dev_dbg(module->dev, "Dynamic Register %d DAI\n", cportid);
}
/* hw_params */
if (module_state < GBAUDIO_CODEC_HWPARAMS) {
format = params->format;
channels = params->channels;
rate = params->rate;
sig_bits = params->sig_bits;
data_cport = data->connection->intf_cport_id;
ret = gb_audio_gb_set_pcm(module->mgmt_connection, data_cport,
format, rate, channels, sig_bits);
if (ret) {
dev_err_ratelimited(module->dev, "set_pcm failed:%d\n",
ret);
return ret;
}
data->state[SNDRV_PCM_STREAM_CAPTURE] =
GBAUDIO_CODEC_HWPARAMS;
dev_dbg(module->dev, "Dynamic hw_params %d DAI\n", data_cport);
}
/* prepare */
if (module_state < GBAUDIO_CODEC_PREPARE) {
data_cport = data->connection->intf_cport_id;
ret = gb_audio_gb_set_rx_data_size(module->mgmt_connection,
data_cport, 192);
if (ret) {
dev_err_ratelimited(module->dev,
"set_rx_data_size failed:%d\n",
ret);
return ret;
}
ret = gb_audio_gb_activate_rx(module->mgmt_connection,
data_cport);
if (ret) {
dev_err_ratelimited(module->dev,
"activate_rx failed:%d\n", ret);
return ret;
}
data->state[SNDRV_PCM_STREAM_CAPTURE] =
GBAUDIO_CODEC_PREPARE;
dev_dbg(module->dev, "Dynamic prepare %d DAI\n", data_cport);
}
return 0;
}
static int gbaudio_module_disable_rx(struct gbaudio_module_info *module, int id)
{
int ret;
uint16_t data_cport, cportid, i2s_port;
int module_state;
struct gbaudio_data_connection *data;
/* find the dai */
data = find_data(module, id);
if (!data) {
dev_err(module->dev, "%d:DATA connection missing\n", id);
return -ENODEV;
}
module_state = data->state[SNDRV_PCM_STREAM_CAPTURE];
if (module_state > GBAUDIO_CODEC_HWPARAMS) {
data_cport = data->connection->intf_cport_id;
ret = gb_audio_gb_deactivate_rx(module->mgmt_connection,
data_cport);
if (ret) {
dev_err_ratelimited(module->dev,
"deactivate_rx failed:%d\n", ret);
return ret;
}
dev_dbg(module->dev, "Dynamic deactivate %d DAI\n", data_cport);
data->state[SNDRV_PCM_STREAM_CAPTURE] =
GBAUDIO_CODEC_HWPARAMS;
}
if (module_state > GBAUDIO_CODEC_SHUTDOWN) {
i2s_port = 0; /* fixed for now */
cportid = data->connection->hd_cport_id;
ret = gb_audio_apbridgea_unregister_cport(data->connection,
i2s_port, cportid,
AUDIO_APBRIDGEA_DIRECTION_RX);
if (ret) {
dev_err_ratelimited(module->dev,
"unregister_cport failed:%d\n",
ret);
return ret;
}
dev_dbg(module->dev, "Dynamic Unregister %d DAI\n", cportid);
data->state[SNDRV_PCM_STREAM_CAPTURE] =
GBAUDIO_CODEC_SHUTDOWN;
}
return 0;
}
int gbaudio_module_update(struct gbaudio_codec_info *codec,
struct snd_soc_dapm_widget *w,
struct gbaudio_module_info *module, int enable)
{
int dai_id, ret;
char intf_name[NAME_SIZE], dir[NAME_SIZE];
dev_dbg(module->dev, "%s:Module update %s sequence\n", w->name,
enable ? "Enable":"Disable");
if ((w->id != snd_soc_dapm_aif_in) && (w->id != snd_soc_dapm_aif_out)){
dev_dbg(codec->dev, "No action required for %s\n", w->name);
return 0;
}
/* parse dai_id from AIF widget's stream_name */
ret = sscanf(w->sname, "%s %d %s", intf_name, &dai_id, dir);
if (ret < 3) {
dev_err(codec->dev, "Error while parsing dai_id for %s\n",
w->name);
return -EINVAL;
}
mutex_lock(&codec->lock);
if (w->id == snd_soc_dapm_aif_in) {
if (enable)
ret = gbaudio_module_enable_tx(codec, module, dai_id);
else
ret = gbaudio_module_disable_tx(module, dai_id);
} else if (w->id == snd_soc_dapm_aif_out) {
if (enable)
ret = gbaudio_module_enable_rx(codec, module, dai_id);
else
ret = gbaudio_module_disable_rx(module, dai_id);
}
mutex_unlock(&codec->lock);
return ret;
}
EXPORT_SYMBOL(gbaudio_module_update);
/*
* codec DAI ops
*/
static int gbcodec_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct gbaudio_codec_info *codec = dev_get_drvdata(dai->dev);
struct gbaudio_stream_params *params;
mutex_lock(&codec->lock);
if (list_empty(&codec->module_list)) {
dev_err(codec->dev, "No codec module available\n");
mutex_unlock(&codec->lock);
return -ENODEV;
}
params = find_dai_stream_params(codec, dai->id, substream->stream);
if (!params) {
dev_err(codec->dev, "Failed to fetch dai_stream pointer\n");
mutex_unlock(&codec->lock);
return -EINVAL;
}
params->state = GBAUDIO_CODEC_STARTUP;
mutex_unlock(&codec->lock);
/* to prevent suspend in case of active audio */
pm_stay_awake(dai->dev);
return 0;
}
static void gbcodec_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct gbaudio_codec_info *codec = dev_get_drvdata(dai->dev);
struct gbaudio_stream_params *params;
mutex_lock(&codec->lock);
if (list_empty(&codec->module_list))
dev_info(codec->dev, "No codec module available during shutdown\n");
params = find_dai_stream_params(codec, dai->id, substream->stream);
if (!params) {
dev_err(codec->dev, "Failed to fetch dai_stream pointer\n");
mutex_unlock(&codec->lock);
return;
}
params->state = GBAUDIO_CODEC_SHUTDOWN;
mutex_unlock(&codec->lock);
pm_relax(dai->dev);
return;
}
static int gbcodec_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *hwparams,
struct snd_soc_dai *dai)
{
int ret;
uint8_t sig_bits, channels;
uint32_t format, rate;
struct gbaudio_module_info *module;
struct gbaudio_data_connection *data;
struct gb_bundle *bundle;
struct gbaudio_codec_info *codec = dev_get_drvdata(dai->dev);
struct gbaudio_stream_params *params;
mutex_lock(&codec->lock);
if (list_empty(&codec->module_list)) {
dev_err(codec->dev, "No codec module available\n");
mutex_unlock(&codec->lock);
return -ENODEV;
}
/*
* assuming, currently only 48000 Hz, 16BIT_LE, stereo
* is supported, validate params before configuring codec
*/
if (params_channels(hwparams) != 2) {
dev_err(dai->dev, "Invalid channel count:%d\n",
params_channels(hwparams));
mutex_unlock(&codec->lock);
return -EINVAL;
}
channels = params_channels(hwparams);
if (params_rate(hwparams) != 48000) {
dev_err(dai->dev, "Invalid sampling rate:%d\n",
params_rate(hwparams));
mutex_unlock(&codec->lock);
return -EINVAL;
}
rate = GB_AUDIO_PCM_RATE_48000;
if (params_format(hwparams) != SNDRV_PCM_FORMAT_S16_LE) {
dev_err(dai->dev, "Invalid format:%d\n",
params_format(hwparams));
mutex_unlock(&codec->lock);
return -EINVAL;
}
format = GB_AUDIO_PCM_FMT_S16_LE;
/* find the data connection */
list_for_each_entry(module, &codec->module_list, list) {
data = find_data(module, dai->id);
if (data)
break;
}
if (!data) {
dev_err(dai->dev, "DATA connection missing\n");
mutex_unlock(&codec->lock);
return -EINVAL;
}
params = find_dai_stream_params(codec, dai->id, substream->stream);
if (!params) {
dev_err(codec->dev, "Failed to fetch dai_stream pointer\n");
mutex_unlock(&codec->lock);
return -EINVAL;
}
bundle = to_gb_bundle(module->dev);
ret = gb_pm_runtime_get_sync(bundle);
if (ret) {
mutex_unlock(&codec->lock);
return ret;
}
ret = gb_audio_apbridgea_set_config(data->connection, 0,
AUDIO_APBRIDGEA_PCM_FMT_16,
AUDIO_APBRIDGEA_PCM_RATE_48000,
6144000);
if (ret) {
dev_err_ratelimited(dai->dev, "%d: Error during set_config\n",
ret);
mutex_unlock(&codec->lock);
return ret;
}
gb_pm_runtime_put_noidle(bundle);
params->state = GBAUDIO_CODEC_HWPARAMS;
params->format = format;
params->rate = rate;
params->channels = channels;
params->sig_bits = sig_bits;
mutex_unlock(&codec->lock);
return 0;
}
static int gbcodec_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
int ret;
struct gbaudio_module_info *module;
struct gbaudio_data_connection *data;
struct gb_bundle *bundle;
struct gbaudio_codec_info *codec = dev_get_drvdata(dai->dev);
struct gbaudio_stream_params *params;
mutex_lock(&codec->lock);
if (list_empty(&codec->module_list)) {
dev_err(codec->dev, "No codec module available\n");
mutex_unlock(&codec->lock);
return -ENODEV;
}
list_for_each_entry(module, &codec->module_list, list) {
/* find the dai */
data = find_data(module, dai->id);
if (data)
break;
}
if (!data) {
dev_err(dai->dev, "DATA connection missing\n");
mutex_unlock(&codec->lock);
return -ENODEV;
}
params = find_dai_stream_params(codec, dai->id, substream->stream);
if (!params) {
dev_err(codec->dev, "Failed to fetch dai_stream pointer\n");
mutex_unlock(&codec->lock);
return -EINVAL;
}
bundle = to_gb_bundle(module->dev);
ret = gb_pm_runtime_get_sync(bundle);
if (ret) {
mutex_unlock(&codec->lock);
return ret;
}
switch (substream->stream) {
case SNDRV_PCM_STREAM_PLAYBACK:
ret = gb_audio_apbridgea_set_tx_data_size(data->connection, 0,
192);
break;
case SNDRV_PCM_STREAM_CAPTURE:
ret = gb_audio_apbridgea_set_rx_data_size(data->connection, 0,
192);
break;
}
if (ret) {
mutex_unlock(&codec->lock);
dev_err_ratelimited(dai->dev, "set_data_size failed:%d\n",
ret);
return ret;
}
gb_pm_runtime_put_noidle(bundle);
params->state = GBAUDIO_CODEC_PREPARE;
mutex_unlock(&codec->lock);
return 0;
}
static int gbcodec_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
{
int ret;
struct gbaudio_data_connection *data;
struct gbaudio_module_info *module;
struct gb_bundle *bundle;
struct gbaudio_codec_info *codec = dev_get_drvdata(dai->dev);
struct gbaudio_stream_params *params;
dev_dbg(dai->dev, "Mute:%d, Direction:%s\n", mute,
stream ? "CAPTURE":"PLAYBACK");
mutex_lock(&codec->lock);
params = find_dai_stream_params(codec, dai->id, stream);
if (!params) {
dev_err(codec->dev, "Failed to fetch dai_stream pointer\n");
mutex_unlock(&codec->lock);
return -EINVAL;
}
if (list_empty(&codec->module_list)) {
dev_err(codec->dev, "No codec module available\n");
if (mute) {
params->state = GBAUDIO_CODEC_STOP;
ret = 0;
} else {
ret = -ENODEV;
}
mutex_unlock(&codec->lock);
return ret;
}
list_for_each_entry(module, &codec->module_list, list) {
/* find the dai */
data = find_data(module, dai->id);
if (data)
break;
}
if (!data) {
dev_err(dai->dev, "%s:%s DATA connection missing\n",
dai->name, module->name);
mutex_unlock(&codec->lock);
return -ENODEV;
}
bundle = to_gb_bundle(module->dev);
ret = gb_pm_runtime_get_sync(bundle);
if (ret) {
mutex_unlock(&codec->lock);
return ret;
}
if (!mute && !stream) {/* start playback */
ret = gb_audio_apbridgea_prepare_tx(data->connection,
0);
if (!ret)
ret = gb_audio_apbridgea_start_tx(data->connection,
0, 0);
params->state = GBAUDIO_CODEC_START;
} else if (!mute && stream) {/* start capture */
ret = gb_audio_apbridgea_prepare_rx(data->connection,
0);
if (!ret)
ret = gb_audio_apbridgea_start_rx(data->connection,
0);
params->state = GBAUDIO_CODEC_START;
} else if (mute && !stream) {/* stop playback */
ret = gb_audio_apbridgea_stop_tx(data->connection, 0);
if (!ret)
ret = gb_audio_apbridgea_shutdown_tx(data->connection,
0);
params->state = GBAUDIO_CODEC_STOP;
} else if (mute && stream) {/* stop capture */
ret = gb_audio_apbridgea_stop_rx(data->connection, 0);
if (!ret)
ret = gb_audio_apbridgea_shutdown_rx(data->connection,
0);
params->state = GBAUDIO_CODEC_STOP;
} else
ret = -EINVAL;
if (ret)
dev_err_ratelimited(dai->dev,
"%s:Error during %s %s stream:%d\n",
module->name, mute ? "Mute" : "Unmute",
stream ? "Capture" : "Playback", ret);
gb_pm_runtime_put_noidle(bundle);
mutex_unlock(&codec->lock);
return ret;
}
static struct snd_soc_dai_ops gbcodec_dai_ops = {
.startup = gbcodec_startup,
.shutdown = gbcodec_shutdown,
.hw_params = gbcodec_hw_params,
.prepare = gbcodec_prepare,
.mute_stream = gbcodec_mute_stream,
};
static struct snd_soc_dai_driver gbaudio_dai[] = {
{
.name = "apb-i2s0",
.id = 0,
.playback = {
.stream_name = "I2S 0 Playback",
.rates = SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FORMAT_S16_LE,
.rate_max = 48000,
.rate_min = 48000,
.channels_min = 1,
.channels_max = 2,
},
.capture = {
.stream_name = "I2S 0 Capture",
.rates = SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FORMAT_S16_LE,
.rate_max = 48000,
.rate_min = 48000,
.channels_min = 1,
.channels_max = 2,
},
.ops = &gbcodec_dai_ops,
},
};
static int gbaudio_init_jack(struct gbaudio_module_info *module,
struct snd_soc_codec *codec)
{
int ret;
if (!module->jack_mask)
return 0;
snprintf(module->jack_name, NAME_SIZE, "GB %d Headset Jack",
module->dev_id);
ret = snd_soc_jack_new(codec, module->jack_name, module->jack_mask,
&module->headset_jack);
if (ret) {
dev_err(module->dev, "Failed to create new jack\n");
return ret;
}
if (!module->button_mask)
return 0;
snprintf(module->button_name, NAME_SIZE, "GB %d Button Jack",
module->dev_id);
ret = snd_soc_jack_new(codec, module->button_name, module->button_mask,
&module->button_jack);
if (ret) {
dev_err(module->dev, "Failed to create button jack\n");
return ret;
}
/*
* Currently, max 4 buttons are supported with following key mapping
* BTN_0 = KEY_MEDIA
* BTN_1 = KEY_VOICECOMMAND
* BTN_2 = KEY_VOLUMEUP
* BTN_3 = KEY_VOLUMEDOWN
*/
if (module->button_mask & SND_JACK_BTN_0) {
ret = snd_jack_set_key(module->button_jack.jack, SND_JACK_BTN_0,
KEY_MEDIA);
if (ret) {
dev_err(module->dev, "Failed to set BTN_0\n");
return ret;
}
}
if (module->button_mask & SND_JACK_BTN_1) {
ret = snd_jack_set_key(module->button_jack.jack, SND_JACK_BTN_1,
KEY_VOICECOMMAND);
if (ret) {
dev_err(module->dev, "Failed to set BTN_1\n");
return ret;
}
}
if (module->button_mask & SND_JACK_BTN_2) {
ret = snd_jack_set_key(module->button_jack.jack, SND_JACK_BTN_2,
KEY_VOLUMEUP);
if (ret) {
dev_err(module->dev, "Failed to set BTN_2\n");
return ret;
}
}
if (module->button_mask & SND_JACK_BTN_3) {
ret = snd_jack_set_key(module->button_jack.jack, SND_JACK_BTN_3,
KEY_VOLUMEDOWN);
if (ret) {
dev_err(module->dev, "Failed to set BTN_0\n");
return ret;
}
}
/* FIXME
* verify if this is really required
set_bit(INPUT_PROP_NO_DUMMY_RELEASE,
module->button_jack.jack->input_dev->propbit);
*/
return 0;
}
int gbaudio_register_module(struct gbaudio_module_info *module)
{
int ret;
struct snd_soc_codec *codec;
struct snd_card *card;
struct snd_soc_jack *jack = NULL;
if (!gbcodec) {
dev_err(module->dev, "GB Codec not yet probed\n");
return -EAGAIN;
}
codec = gbcodec->codec;
card = codec->card->snd_card;
down_write(&card->controls_rwsem);
if (module->num_dais) {
dev_err(gbcodec->dev,
"%d:DAIs not supported via gbcodec driver\n",
module->num_dais);
up_write(&card->controls_rwsem);
return -EINVAL;
}
ret = gbaudio_init_jack(module, codec);
if (ret) {
up_write(&card->controls_rwsem);
return ret;
}
if (module->dapm_widgets)
snd_soc_dapm_new_controls(&codec->dapm, module->dapm_widgets,
module->num_dapm_widgets);
if (module->controls)
snd_soc_add_codec_controls(codec, module->controls,
module->num_controls);
if (module->dapm_routes)
snd_soc_dapm_add_routes(&codec->dapm, module->dapm_routes,
module->num_dapm_routes);
/* card already instantiated, create widgets here only */
if (codec->card->instantiated) {
snd_soc_dapm_link_component_dai_widgets(codec->card,
&codec->dapm);
#ifdef CONFIG_SND_JACK
/* register jack devices for this module from codec->jack_list */
list_for_each_entry(jack, &codec->jack_list, list) {
if ((jack == &module->headset_jack)
|| (jack == &module->button_jack))
snd_device_register(codec->card->snd_card,
jack->jack);
}
#endif
}
mutex_lock(&gbcodec->lock);
list_add(&module->list, &gbcodec->module_list);
mutex_unlock(&gbcodec->lock);
if (codec->card->instantiated)
ret = snd_soc_dapm_new_widgets(&codec->dapm);
dev_dbg(codec->dev, "Registered %s module\n", module->name);
up_write(&card->controls_rwsem);
return ret;
}
EXPORT_SYMBOL(gbaudio_register_module);
static void gbaudio_codec_clean_data_tx(struct gbaudio_data_connection *data)
{
uint16_t i2s_port, cportid;
int ret;
if (list_is_singular(&gbcodec->module_list)) {
ret = gb_audio_apbridgea_stop_tx(data->connection, 0);
if (ret)
return;
ret = gb_audio_apbridgea_shutdown_tx(data->connection,
0);
if (ret)
return;
}
i2s_port = 0; /* fixed for now */
cportid = data->connection->hd_cport_id;
ret = gb_audio_apbridgea_unregister_cport(data->connection,
i2s_port, cportid,
AUDIO_APBRIDGEA_DIRECTION_TX);
data->state[0] = GBAUDIO_CODEC_SHUTDOWN;
}
static void gbaudio_codec_clean_data_rx(struct gbaudio_data_connection *data)
{
uint16_t i2s_port, cportid;
int ret;
if (list_is_singular(&gbcodec->module_list)) {
ret = gb_audio_apbridgea_stop_rx(data->connection, 0);
if (ret)
return;
ret = gb_audio_apbridgea_shutdown_rx(data->connection,
0);
if (ret)
return;
}
i2s_port = 0; /* fixed for now */
cportid = data->connection->hd_cport_id;
ret = gb_audio_apbridgea_unregister_cport(data->connection,
i2s_port, cportid,
AUDIO_APBRIDGEA_DIRECTION_RX);
data->state[1] = GBAUDIO_CODEC_SHUTDOWN;
}
static void gbaudio_codec_cleanup(struct gbaudio_module_info *module)
{
struct gbaudio_data_connection *data;
int pb_state, cap_state;
dev_dbg(gbcodec->dev, "%s: removed, cleanup APBridge\n", module->name);
list_for_each_entry(data, &module->data_list, list) {
pb_state = data->state[0];
cap_state = data->state[1];
if (pb_state > GBAUDIO_CODEC_SHUTDOWN)
gbaudio_codec_clean_data_tx(data);
if (cap_state > GBAUDIO_CODEC_SHUTDOWN)
gbaudio_codec_clean_data_rx(data);
}
}
void gbaudio_unregister_module(struct gbaudio_module_info *module)
{
struct snd_soc_codec *codec = gbcodec->codec;
struct snd_card *card = codec->card->snd_card;
struct snd_soc_jack *jack, *next_j;
int mask;
dev_dbg(codec->dev, "Unregister %s module\n", module->name);
down_write(&card->controls_rwsem);
mutex_lock(&gbcodec->lock);
gbaudio_codec_cleanup(module);
list_del(&module->list);
dev_dbg(codec->dev, "Process Unregister %s module\n", module->name);
mutex_unlock(&gbcodec->lock);
#ifdef CONFIG_SND_JACK
/* free jack devices for this module from codec->jack_list */
list_for_each_entry_safe(jack, next_j, &codec->jack_list, list) {
if (jack == &module->headset_jack)
mask = GBCODEC_JACK_MASK;
else if (jack == &module->button_jack)
mask = GBCODEC_JACK_BUTTON_MASK;
else
mask = 0;
if (mask) {
dev_dbg(module->dev, "Report %s removal\n",
jack->jack->id);
snd_soc_jack_report(jack, 0, mask);
snd_device_free(codec->card->snd_card, jack->jack);
list_del(&jack->list);
}
}
#endif
if (module->dapm_routes) {
dev_dbg(codec->dev, "Removing %d routes\n",
module->num_dapm_routes);
snd_soc_dapm_del_routes(&codec->dapm, module->dapm_routes,
module->num_dapm_routes);
}
if (module->controls) {
dev_dbg(codec->dev, "Removing %d controls\n",
module->num_controls);
snd_soc_remove_codec_controls(codec, module->controls,
module->num_controls);
}
if (module->dapm_widgets) {
dev_dbg(codec->dev, "Removing %d widgets\n",
module->num_dapm_widgets);
snd_soc_dapm_free_controls(&codec->dapm, module->dapm_widgets,
module->num_dapm_widgets);
}
dev_dbg(codec->dev, "Unregistered %s module\n", module->name);
up_write(&card->controls_rwsem);
}
EXPORT_SYMBOL(gbaudio_unregister_module);
/*
* codec driver ops
*/
static int gbcodec_probe(struct snd_soc_codec *codec)
{
int i;
struct gbaudio_codec_info *info;
struct gbaudio_codec_dai *dai;
info = devm_kzalloc(codec->dev, sizeof(*info), GFP_KERNEL);
if (!info)
return -ENOMEM;
info->dev = codec->dev;
INIT_LIST_HEAD(&info->module_list);
mutex_init(&info->lock);
INIT_LIST_HEAD(&info->dai_list);
/* init dai_list used to maintain runtime stream info */
for (i = 0; i < ARRAY_SIZE(gbaudio_dai); i++) {
dai = devm_kzalloc(codec->dev, sizeof(*dai), GFP_KERNEL);