greybus: audio: Enable audio path based on control switch state only

As per current implementation, audio data is played from each individual
SPK module connected to endo frame. This is not a valid requirement in
case of capture/headset path. So, provide a mechanism to enable
individual module path based on it's control switch state.

Signed-off-by: Vaibhav Agarwal <vaibhav.agarwal@linaro.org>
Reviewed-by: Mark Greer <mgreer@animalcreek.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@google.com>
This commit is contained in:
Vaibhav Agarwal 2016-08-04 15:14:29 +05:30 committed by Greg Kroah-Hartman
parent 90579d4b57
commit ce9413062f

View file

@ -323,11 +323,6 @@ EXPORT_SYMBOL(gbaudio_module_update);
static int gbcodec_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
int ret = 0;
__u16 i2s_port, cportid;
int state;
struct gbaudio_data_connection *data;
struct gbaudio_module_info *module;
struct gbaudio_codec_info *codec = dev_get_drvdata(dai->dev);
mutex_lock(&codec->lock);
@ -338,164 +333,26 @@ static int gbcodec_startup(struct snd_pcm_substream *substream,
return -ENODEV;
}
state = codec->stream[substream->stream].state;
list_for_each_entry(module, &codec->module_list, list) {
/* find the dai */
data = find_data(module, dai->name);
if (!data) {
dev_err(dai->dev, "%s:%s DATA connection missing\n",
dai->name, module->name);
continue;
}
/* register cport */
i2s_port = 0; /* fixed for now */
cportid = data->connection->hd_cport_id;
switch (substream->stream) {
case SNDRV_PCM_STREAM_CAPTURE:
ret = gb_audio_apbridgea_register_cport(
data->connection,
i2s_port, cportid,
AUDIO_APBRIDGEA_DIRECTION_RX);
break;
case SNDRV_PCM_STREAM_PLAYBACK:
ret = gb_audio_apbridgea_register_cport(
data->connection,
i2s_port, cportid,
AUDIO_APBRIDGEA_DIRECTION_TX);
break;
default:
dev_err(dai->dev, "Inavlid stream\n");
mutex_unlock(&codec->lock);
return -EINVAL;
}
dev_dbg(dai->dev, "Register %s:%d DAI, ret:%d\n", dai->name,
cportid, ret);
state = GBAUDIO_CODEC_STARTUP;
module->ctrlstate[substream->stream] = state;
dev_dbg(dai->dev, "%s: state:%d\n", module->name, state);
}
codec->stream[substream->stream].state = state;
codec->stream[substream->stream].state = GBAUDIO_CODEC_STARTUP;
codec->stream[substream->stream].dai_name = dai->name;
mutex_unlock(&codec->lock);
/* to prevent suspend in case of active audio */
pm_stay_awake(dai->dev);
return ret;
}
static int gbmodule_shutdown_tx(struct gbaudio_module_info *module,
struct gbaudio_data_connection *data,
int codec_state, struct device *dev)
{
int ret, module_state;
__u16 i2s_port, cportid;
module_state = module->ctrlstate[0];
if (module_state == GBAUDIO_CODEC_SHUTDOWN) {
dev_dbg(dev, "%s: module already configured\n",
module->name);
return 0;
}
/* deactivate */
cportid = data->connection->intf_cport_id;
if (module_state >= GBAUDIO_CODEC_PREPARE) {
ret = gb_audio_gb_deactivate_tx(module->mgmt_connection,
cportid);
if (ret)
return ret;
}
/* unregister cport */
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);
return ret;
}
static int gbmodule_shutdown_rx(struct gbaudio_module_info *module,
struct gbaudio_data_connection *data,
int codec_state, struct device *dev)
{
int ret, module_state;
__u16 i2s_port, cportid;
module_state = module->ctrlstate[1];
if (module_state == GBAUDIO_CODEC_SHUTDOWN) {
dev_dbg(dev, "%s: module already configured\n",
module->name);
return 0;
}
/* deactivate */
cportid = data->connection->intf_cport_id;
if (module_state >= GBAUDIO_CODEC_PREPARE) {
ret = gb_audio_gb_deactivate_rx(module->mgmt_connection,
cportid);
if (ret)
return ret;
}
/* unregister cport */
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);
return ret;
return 0;
}
static void gbcodec_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
int ret, state;
struct gbaudio_module_info *module;
struct gbaudio_data_connection *data;
struct gbaudio_codec_info *codec = dev_get_drvdata(dai->dev);
mutex_lock(&codec->lock);
if (list_empty(&codec->module_list)) {
dev_err(codec->dev, "No codec module available\n");
codec->stream[substream->stream].state = GBAUDIO_CODEC_SHUTDOWN;
codec->stream[substream->stream].dai_name = NULL;
mutex_unlock(&codec->lock);
pm_relax(dai->dev);
return;
}
if (list_empty(&codec->module_list))
dev_info(codec->dev, "No codec module available during shutdown\n");
state = codec->stream[substream->stream].state;
list_for_each_entry(module, &codec->module_list, list) {
/* find the dai */
data = find_data(module, dai->name);
if (!data) {
dev_err(dai->dev, "%s:%s DATA connection missing\n",
dai->name, module->name);
continue;
}
switch (substream->stream) {
case SNDRV_PCM_STREAM_PLAYBACK:
ret = gbmodule_shutdown_tx(module, data, state,
dai->dev);
break;
case SNDRV_PCM_STREAM_CAPTURE:
ret = gbmodule_shutdown_rx(module, data, state,
dai->dev);
break;
}
dev_dbg(dai->dev, "Unregister %s DAI, ret:%d\n", dai->name,
ret);
state = GBAUDIO_CODEC_SHUTDOWN;
module->ctrlstate[substream->stream] = state;
dev_dbg(dai->dev, "%s: state:%d\n", module->name, state);
}
codec->stream[substream->stream].state = state;
codec->stream[substream->stream].state = GBAUDIO_CODEC_SHUTDOWN;
codec->stream[substream->stream].dai_name = NULL;
mutex_unlock(&codec->lock);
pm_relax(dai->dev);
@ -509,10 +366,8 @@ static int gbcodec_hw_params(struct snd_pcm_substream *substream,
int ret;
uint8_t sig_bits, channels;
uint32_t format, rate;
uint16_t data_cport;
struct gbaudio_module_info *module;
struct gbaudio_data_connection *data;
int state;
struct gbaudio_codec_info *codec = dev_get_drvdata(dai->dev);
mutex_lock(&codec->lock);
@ -551,122 +406,37 @@ static int gbcodec_hw_params(struct snd_pcm_substream *substream,
}
format = GB_AUDIO_PCM_FMT_S16_LE;
state = codec->stream[substream->stream].state;
/* find the data connection */
list_for_each_entry(module, &codec->module_list, list) {
/* find the data connection */
data = find_data(module, dai->name);
if (!data) {
dev_err(dai->dev, "%s:%s DATA connection missing\n",
dai->name, module->name);
continue;
}
data_cport = data->connection->intf_cport_id;
/* XXX check impact of sig_bit
* it should not change ideally
*/
dev_dbg(dai->dev,
"cport:%d, rate:%d, channel %d, format %d, sig_bits:%d\n",
data_cport, rate, channels, format, sig_bits);
ret = gb_audio_gb_set_pcm(module->mgmt_connection, data_cport,
format, rate, channels, sig_bits);
if (ret) {
dev_err_ratelimited(dai->dev, "%d: Error during set_pcm\n", ret);
goto func_exit;
}
if (state < GBAUDIO_CODEC_HWPARAMS) {
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);
goto func_exit;
}
}
state = GBAUDIO_CODEC_HWPARAMS;
module->ctrlstate[substream->stream] = state;
dev_dbg(dai->dev, "%s: state:%d\n", module->name, state);
if (data)
break;
}
codec->stream[substream->stream].state = state;
if (!data) {
dev_err(dai->dev, "DATA connection missing\n");
mutex_unlock(&codec->lock);
return -EINVAL;
}
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;
}
codec->stream[substream->stream].state = GBAUDIO_CODEC_HWPARAMS;
codec->stream[substream->stream].format = format;
codec->stream[substream->stream].rate = rate;
codec->stream[substream->stream].channels = channels;
codec->stream[substream->stream].sig_bits = sig_bits;
func_exit:
mutex_unlock(&codec->lock);
return ret;
}
static int gbmodule_prepare_tx(struct gbaudio_module_info *module,
struct gbaudio_data_connection *data,
int codec_state, struct device *dev)
{
int ret;
uint16_t data_cport;
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(dev, "%d:Error during set_tx_data_size, cport:%d\n",
ret, data_cport);
return ret;
}
if (codec_state < GBAUDIO_CODEC_PREPARE) {
ret = gb_audio_apbridgea_set_tx_data_size(data->connection, 0,
192);
if (ret) {
dev_err_ratelimited(dev,
"%d:Error during apbridgea set_tx_data_size, cport\n",
ret);
return ret;
}
}
ret = gb_audio_gb_activate_tx(module->mgmt_connection,
data_cport);
if (ret)
dev_err_ratelimited(dev, "%s:Error during activate stream,%d\n",
module->name, ret);
return ret;
}
static int gbmodule_prepare_rx(struct gbaudio_module_info *module,
struct gbaudio_data_connection *data,
int codec_state, struct device *dev)
{
int ret;
uint16_t data_cport;
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(dev, "%d:Error during set_rx_data_size, cport:%d\n",
ret, data_cport);
return ret;
}
if (codec_state < GBAUDIO_CODEC_PREPARE) {
ret = gb_audio_apbridgea_set_rx_data_size(data->connection, 0,
192);
if (ret) {
dev_err_ratelimited(dev,
"%d:Error during apbridgea_set_rx_data_size\n",
ret);
return ret;
}
}
ret = gb_audio_gb_activate_rx(module->mgmt_connection,
data_cport);
if (ret)
dev_err_ratelimited(dev, "%s:Error during activate stream,%d\n",
module->name, ret);
return ret;
return 0;
}
static int gbcodec_prepare(struct snd_pcm_substream *substream,
@ -674,7 +444,6 @@ static int gbcodec_prepare(struct snd_pcm_substream *substream,
{
int ret;
struct gbaudio_module_info *module;
int state;
struct gbaudio_data_connection *data;
struct gbaudio_codec_info *codec = dev_get_drvdata(dai->dev);
@ -686,41 +455,38 @@ static int gbcodec_prepare(struct snd_pcm_substream *substream,
return -ENODEV;
}
state = codec->stream[substream->stream].state;
list_for_each_entry(module, &codec->module_list, list) {
/* find the dai */
data = find_data(module, dai->name);
if (!data) {
dev_err(dai->dev, "%s:%s DATA connection missing\n",
dai->name, module->name);
continue;
}
switch (substream->stream) {
case SNDRV_PCM_STREAM_PLAYBACK:
ret = gbmodule_prepare_tx(module, data, state,
dai->dev);
if (data)
break;
case SNDRV_PCM_STREAM_CAPTURE:
ret = gbmodule_prepare_rx(module, data, state,
dai->dev);
break;
}
if (ret == -ENODEV)
continue;
if (ret) {
goto func_exit;
}
state = GBAUDIO_CODEC_PREPARE;
module->ctrlstate[substream->stream] = state;
dev_dbg(dai->dev, "%s: state:%d\n", module->name, state);
}
codec->stream[substream->stream].state = state;
if (!data) {
dev_err(dai->dev, "DATA connection missing\n");
mutex_unlock(&codec->lock);
return -ENODEV;
}
func_exit:
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;
}
codec->stream[substream->stream].state = GBAUDIO_CODEC_PREPARE;
mutex_unlock(&codec->lock);
return ret;
return 0;
}
static int gbcodec_mute_stream(struct snd_soc_dai *dai, int mute, int stream)