mirror of https://gitee.com/openkylin/linux.git
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
Greg Kroah-Hartman
parent
90579d4b57
commit
ce9413062f
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@ -323,11 +323,6 @@ EXPORT_SYMBOL(gbaudio_module_update);
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static int gbcodec_startup(struct snd_pcm_substream *substream,
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struct snd_soc_dai *dai)
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{
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int ret = 0;
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__u16 i2s_port, cportid;
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int state;
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struct gbaudio_data_connection *data;
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struct gbaudio_module_info *module;
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struct gbaudio_codec_info *codec = dev_get_drvdata(dai->dev);
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mutex_lock(&codec->lock);
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@ -338,164 +333,26 @@ static int gbcodec_startup(struct snd_pcm_substream *substream,
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return -ENODEV;
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}
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state = codec->stream[substream->stream].state;
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list_for_each_entry(module, &codec->module_list, list) {
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/* find the dai */
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data = find_data(module, dai->name);
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if (!data) {
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dev_err(dai->dev, "%s:%s DATA connection missing\n",
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dai->name, module->name);
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continue;
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}
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/* register cport */
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i2s_port = 0; /* fixed for now */
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cportid = data->connection->hd_cport_id;
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switch (substream->stream) {
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case SNDRV_PCM_STREAM_CAPTURE:
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ret = gb_audio_apbridgea_register_cport(
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data->connection,
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i2s_port, cportid,
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AUDIO_APBRIDGEA_DIRECTION_RX);
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break;
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case SNDRV_PCM_STREAM_PLAYBACK:
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ret = gb_audio_apbridgea_register_cport(
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data->connection,
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i2s_port, cportid,
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AUDIO_APBRIDGEA_DIRECTION_TX);
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break;
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default:
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dev_err(dai->dev, "Inavlid stream\n");
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mutex_unlock(&codec->lock);
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return -EINVAL;
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}
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dev_dbg(dai->dev, "Register %s:%d DAI, ret:%d\n", dai->name,
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cportid, ret);
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state = GBAUDIO_CODEC_STARTUP;
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module->ctrlstate[substream->stream] = state;
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dev_dbg(dai->dev, "%s: state:%d\n", module->name, state);
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}
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codec->stream[substream->stream].state = state;
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codec->stream[substream->stream].state = GBAUDIO_CODEC_STARTUP;
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codec->stream[substream->stream].dai_name = dai->name;
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mutex_unlock(&codec->lock);
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/* to prevent suspend in case of active audio */
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pm_stay_awake(dai->dev);
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return ret;
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}
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static int gbmodule_shutdown_tx(struct gbaudio_module_info *module,
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struct gbaudio_data_connection *data,
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int codec_state, struct device *dev)
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{
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int ret, module_state;
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__u16 i2s_port, cportid;
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module_state = module->ctrlstate[0];
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if (module_state == GBAUDIO_CODEC_SHUTDOWN) {
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dev_dbg(dev, "%s: module already configured\n",
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module->name);
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return 0;
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}
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/* deactivate */
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cportid = data->connection->intf_cport_id;
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if (module_state >= GBAUDIO_CODEC_PREPARE) {
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ret = gb_audio_gb_deactivate_tx(module->mgmt_connection,
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cportid);
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if (ret)
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return ret;
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}
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/* unregister cport */
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i2s_port = 0; /* fixed for now */
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cportid = data->connection->hd_cport_id;
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ret = gb_audio_apbridgea_unregister_cport(data->connection, i2s_port,
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cportid,
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AUDIO_APBRIDGEA_DIRECTION_TX);
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return ret;
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}
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static int gbmodule_shutdown_rx(struct gbaudio_module_info *module,
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struct gbaudio_data_connection *data,
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int codec_state, struct device *dev)
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{
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int ret, module_state;
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__u16 i2s_port, cportid;
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module_state = module->ctrlstate[1];
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if (module_state == GBAUDIO_CODEC_SHUTDOWN) {
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dev_dbg(dev, "%s: module already configured\n",
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module->name);
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return 0;
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}
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/* deactivate */
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cportid = data->connection->intf_cport_id;
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if (module_state >= GBAUDIO_CODEC_PREPARE) {
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ret = gb_audio_gb_deactivate_rx(module->mgmt_connection,
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cportid);
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if (ret)
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return ret;
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}
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/* unregister cport */
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i2s_port = 0; /* fixed for now */
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cportid = data->connection->hd_cport_id;
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ret = gb_audio_apbridgea_unregister_cport(data->connection, i2s_port,
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cportid,
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AUDIO_APBRIDGEA_DIRECTION_RX);
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return ret;
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return 0;
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}
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static void gbcodec_shutdown(struct snd_pcm_substream *substream,
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struct snd_soc_dai *dai)
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{
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int ret, state;
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struct gbaudio_module_info *module;
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struct gbaudio_data_connection *data;
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struct gbaudio_codec_info *codec = dev_get_drvdata(dai->dev);
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mutex_lock(&codec->lock);
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if (list_empty(&codec->module_list)) {
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dev_err(codec->dev, "No codec module available\n");
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codec->stream[substream->stream].state = GBAUDIO_CODEC_SHUTDOWN;
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codec->stream[substream->stream].dai_name = NULL;
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mutex_unlock(&codec->lock);
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pm_relax(dai->dev);
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return;
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}
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if (list_empty(&codec->module_list))
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dev_info(codec->dev, "No codec module available during shutdown\n");
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state = codec->stream[substream->stream].state;
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list_for_each_entry(module, &codec->module_list, list) {
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/* find the dai */
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data = find_data(module, dai->name);
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if (!data) {
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dev_err(dai->dev, "%s:%s DATA connection missing\n",
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dai->name, module->name);
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continue;
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}
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switch (substream->stream) {
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case SNDRV_PCM_STREAM_PLAYBACK:
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ret = gbmodule_shutdown_tx(module, data, state,
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dai->dev);
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break;
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case SNDRV_PCM_STREAM_CAPTURE:
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ret = gbmodule_shutdown_rx(module, data, state,
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dai->dev);
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break;
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}
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dev_dbg(dai->dev, "Unregister %s DAI, ret:%d\n", dai->name,
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ret);
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state = GBAUDIO_CODEC_SHUTDOWN;
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module->ctrlstate[substream->stream] = state;
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dev_dbg(dai->dev, "%s: state:%d\n", module->name, state);
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}
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codec->stream[substream->stream].state = state;
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codec->stream[substream->stream].state = GBAUDIO_CODEC_SHUTDOWN;
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codec->stream[substream->stream].dai_name = NULL;
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mutex_unlock(&codec->lock);
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pm_relax(dai->dev);
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@ -509,10 +366,8 @@ static int gbcodec_hw_params(struct snd_pcm_substream *substream,
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int ret;
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uint8_t sig_bits, channels;
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uint32_t format, rate;
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uint16_t data_cport;
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struct gbaudio_module_info *module;
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struct gbaudio_data_connection *data;
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int state;
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struct gbaudio_codec_info *codec = dev_get_drvdata(dai->dev);
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mutex_lock(&codec->lock);
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@ -551,122 +406,37 @@ static int gbcodec_hw_params(struct snd_pcm_substream *substream,
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}
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format = GB_AUDIO_PCM_FMT_S16_LE;
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state = codec->stream[substream->stream].state;
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/* find the data connection */
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list_for_each_entry(module, &codec->module_list, list) {
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/* find the data connection */
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data = find_data(module, dai->name);
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if (!data) {
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dev_err(dai->dev, "%s:%s DATA connection missing\n",
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dai->name, module->name);
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continue;
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}
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data_cport = data->connection->intf_cport_id;
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/* XXX check impact of sig_bit
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* it should not change ideally
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*/
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dev_dbg(dai->dev,
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"cport:%d, rate:%d, channel %d, format %d, sig_bits:%d\n",
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data_cport, rate, channels, format, sig_bits);
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ret = gb_audio_gb_set_pcm(module->mgmt_connection, data_cport,
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format, rate, channels, sig_bits);
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if (ret) {
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dev_err_ratelimited(dai->dev, "%d: Error during set_pcm\n", ret);
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goto func_exit;
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}
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if (state < GBAUDIO_CODEC_HWPARAMS) {
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ret = gb_audio_apbridgea_set_config(data->connection, 0,
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AUDIO_APBRIDGEA_PCM_FMT_16,
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AUDIO_APBRIDGEA_PCM_RATE_48000,
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6144000);
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if (ret) {
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dev_err_ratelimited(dai->dev,
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"%d: Error during set_config\n", ret);
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goto func_exit;
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}
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}
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state = GBAUDIO_CODEC_HWPARAMS;
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module->ctrlstate[substream->stream] = state;
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dev_dbg(dai->dev, "%s: state:%d\n", module->name, state);
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if (data)
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break;
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}
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codec->stream[substream->stream].state = state;
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if (!data) {
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dev_err(dai->dev, "DATA connection missing\n");
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mutex_unlock(&codec->lock);
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return -EINVAL;
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}
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ret = gb_audio_apbridgea_set_config(data->connection, 0,
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AUDIO_APBRIDGEA_PCM_FMT_16,
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AUDIO_APBRIDGEA_PCM_RATE_48000,
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6144000);
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if (ret) {
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dev_err_ratelimited(dai->dev, "%d: Error during set_config\n",
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ret);
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mutex_unlock(&codec->lock);
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return ret;
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}
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codec->stream[substream->stream].state = GBAUDIO_CODEC_HWPARAMS;
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codec->stream[substream->stream].format = format;
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codec->stream[substream->stream].rate = rate;
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codec->stream[substream->stream].channels = channels;
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codec->stream[substream->stream].sig_bits = sig_bits;
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func_exit:
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mutex_unlock(&codec->lock);
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return ret;
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}
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static int gbmodule_prepare_tx(struct gbaudio_module_info *module,
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struct gbaudio_data_connection *data,
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int codec_state, struct device *dev)
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{
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int ret;
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uint16_t data_cport;
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data_cport = data->connection->intf_cport_id;
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ret = gb_audio_gb_set_tx_data_size(module->mgmt_connection, data_cport,
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192);
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if (ret) {
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dev_err_ratelimited(dev, "%d:Error during set_tx_data_size, cport:%d\n",
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ret, data_cport);
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return ret;
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}
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if (codec_state < GBAUDIO_CODEC_PREPARE) {
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ret = gb_audio_apbridgea_set_tx_data_size(data->connection, 0,
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192);
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if (ret) {
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dev_err_ratelimited(dev,
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"%d:Error during apbridgea set_tx_data_size, cport\n",
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ret);
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return ret;
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}
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}
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ret = gb_audio_gb_activate_tx(module->mgmt_connection,
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data_cport);
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if (ret)
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dev_err_ratelimited(dev, "%s:Error during activate stream,%d\n",
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module->name, ret);
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return ret;
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}
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static int gbmodule_prepare_rx(struct gbaudio_module_info *module,
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struct gbaudio_data_connection *data,
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int codec_state, struct device *dev)
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{
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int ret;
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uint16_t data_cport;
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data_cport = data->connection->intf_cport_id;
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ret = gb_audio_gb_set_rx_data_size(module->mgmt_connection, data_cport,
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192);
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if (ret) {
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dev_err_ratelimited(dev, "%d:Error during set_rx_data_size, cport:%d\n",
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ret, data_cport);
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return ret;
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}
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if (codec_state < GBAUDIO_CODEC_PREPARE) {
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ret = gb_audio_apbridgea_set_rx_data_size(data->connection, 0,
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192);
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if (ret) {
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dev_err_ratelimited(dev,
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"%d:Error during apbridgea_set_rx_data_size\n",
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ret);
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return ret;
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}
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}
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ret = gb_audio_gb_activate_rx(module->mgmt_connection,
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data_cport);
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if (ret)
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dev_err_ratelimited(dev, "%s:Error during activate stream,%d\n",
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module->name, ret);
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return ret;
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return 0;
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}
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static int gbcodec_prepare(struct snd_pcm_substream *substream,
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@ -674,7 +444,6 @@ static int gbcodec_prepare(struct snd_pcm_substream *substream,
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{
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int ret;
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struct gbaudio_module_info *module;
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int state;
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struct gbaudio_data_connection *data;
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struct gbaudio_codec_info *codec = dev_get_drvdata(dai->dev);
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@ -686,41 +455,38 @@ static int gbcodec_prepare(struct snd_pcm_substream *substream,
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return -ENODEV;
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}
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state = codec->stream[substream->stream].state;
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list_for_each_entry(module, &codec->module_list, list) {
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/* find the dai */
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data = find_data(module, dai->name);
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if (!data) {
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dev_err(dai->dev, "%s:%s DATA connection missing\n",
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dai->name, module->name);
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continue;
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}
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switch (substream->stream) {
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case SNDRV_PCM_STREAM_PLAYBACK:
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ret = gbmodule_prepare_tx(module, data, state,
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dai->dev);
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if (data)
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break;
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case SNDRV_PCM_STREAM_CAPTURE:
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ret = gbmodule_prepare_rx(module, data, state,
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dai->dev);
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break;
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}
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if (ret == -ENODEV)
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continue;
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if (ret) {
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goto func_exit;
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}
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state = GBAUDIO_CODEC_PREPARE;
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module->ctrlstate[substream->stream] = state;
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dev_dbg(dai->dev, "%s: state:%d\n", module->name, state);
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}
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codec->stream[substream->stream].state = state;
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if (!data) {
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dev_err(dai->dev, "DATA connection missing\n");
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mutex_unlock(&codec->lock);
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return -ENODEV;
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}
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func_exit:
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switch (substream->stream) {
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case SNDRV_PCM_STREAM_PLAYBACK:
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ret = gb_audio_apbridgea_set_tx_data_size(data->connection, 0,
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192);
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break;
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case SNDRV_PCM_STREAM_CAPTURE:
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ret = gb_audio_apbridgea_set_rx_data_size(data->connection, 0,
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192);
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break;
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}
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if (ret) {
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mutex_unlock(&codec->lock);
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dev_err_ratelimited(dai->dev, "set_data_size failed:%d\n",
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ret);
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return ret;
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}
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codec->stream[substream->stream].state = GBAUDIO_CODEC_PREPARE;
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mutex_unlock(&codec->lock);
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return ret;
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return 0;
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}
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static int gbcodec_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
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