/**
* @brief Start new A/V session. There can only be one session at the time. If you register more
* it will result in undefined behaviour.
*
* @param messenger The messenger handle.
* @param userdata The agent handling A/V session (i.e. phone).
* @param video_width Width of video frame.
* @param video_height Height of video frame.
* @return ToxAv*
* @retval NULL On error.
*/
ToxAv *toxav_new( Tox* messenger, ToxAvCodecSettings* codec_settings)
{
ToxAv *av = calloc ( sizeof(ToxAv), 1);
if (av == NULL)
return NULL;
av->messenger = (Messenger *)messenger;
av->msi_session = msi_init_session(av->messenger);
av->msi_session->agent_handler = av;
av->rtp_sessions[0] = av->rtp_sessions [1] = NULL;
/* NOTE: This should be user defined or? */
av->j_buf = create_queue(codec_settings->jbuf_capacity);
av->cs = codec_init_session(codec_settings->audio_bitrate,
codec_settings->audio_frame_duration,
codec_settings->audio_sample_rate,
codec_settings->audio_channels,
codec_settings->video_width,
codec_settings->video_height,
codec_settings->video_bitrate);
return av;
}
/**
* @brief Must be call before any RTP transmission occurs.
*
* @param av Handler.
* @return int
* @retval 0 Success.
* @retval ToxAvError On error.
*/
int toxav_prepare_transmission ( ToxAv *av, int32_t call_index, ToxAvCodecSettings *codec_settings, int support_video )
{
if ( !av->msi_session || av->msi_session->max_calls <= call_index || !av->msi_session->calls[call_index] ) {
/*fprintf(stderr, "Error while starting audio RTP session: invalid call!\n");*/
return ErrorInternal;
}
CallSpecific *call = &av->calls[call_index];
call->crtps[audio_index] =
rtp_init_session(
type_audio,
av->messenger,
av->msi_session->calls[call_index]->peers[0],
av->msi_session->calls[call_index]->key_peer,
av->msi_session->calls[call_index]->key_local,
av->msi_session->calls[call_index]->nonce_peer,
av->msi_session->calls[call_index]->nonce_local);
if ( !call->crtps[audio_index] ) {
/*fprintf(stderr, "Error while starting audio RTP session!\n");*/
return ErrorStartingAudioRtp;
}
if ( support_video ) {
call->crtps[video_index] =
rtp_init_session (
type_video,
av->messenger,
av->msi_session->calls[call_index]->peers[0],
av->msi_session->calls[call_index]->key_peer,
av->msi_session->calls[call_index]->key_local,
av->msi_session->calls[call_index]->nonce_peer,
av->msi_session->calls[call_index]->nonce_local);
if ( !call->crtps[video_index] ) {
/*fprintf(stderr, "Error while starting video RTP session!\n");*/
return ErrorStartingVideoRtp;
}
}
if ( !(call->j_buf = create_queue(codec_settings->jbuf_capacity)) ) return ErrorInternal;
call->cs = codec_init_session(codec_settings->audio_bitrate,
codec_settings->audio_frame_duration,
codec_settings->audio_sample_rate,
codec_settings->audio_channels,
codec_settings->video_width,
codec_settings->video_height,
codec_settings->video_bitrate);
return call->cs ? ErrorNone : ErrorInternal;
}
/**
* @brief Must be call before any RTP transmission occurs.
*
* @param av Handler.
* @return int
* @retval 0 Success.
* @retval ToxAvError On error.
*/
int toxav_prepare_transmission ( ToxAv *av, int32_t call_index, uint32_t jbuf_capacity, uint32_t VAD_treshold,
int support_video )
{
if ( !av->msi_session || cii(call_index, av->msi_session) ||
!av->msi_session->calls[call_index] || !av->msi_session->calls[call_index]->csettings_peer ||
av->calls[call_index].call_active) {
LOGGER_ERROR("Error while starting RTP session: invalid call!\n");
return ErrorInternal;
}
CallSpecific *call = &av->calls[call_index];
call->crtps[audio_index] =
rtp_init_session(type_audio, av->messenger, av->msi_session->calls[call_index]->peers[0]);
if ( !call->crtps[audio_index] ) {
LOGGER_ERROR("Error while starting audio RTP session!\n");
return ErrorInternal;
}
call->crtps[audio_index]->call_index = call_index;
call->crtps[audio_index]->av = av;
if ( support_video ) {
call->crtps[video_index] =
rtp_init_session(type_video, av->messenger, av->msi_session->calls[call_index]->peers[0]);
if ( !call->crtps[video_index] ) {
LOGGER_ERROR("Error while starting video RTP session!\n");
goto error;
}
call->crtps[video_index]->call_index = call_index;
call->crtps[video_index]->av = av;
call->frame_limit = 0;
call->frame_id = 0;
call->frame_outid = 0;
call->frame_buf = calloc(MAX_VIDEOFRAME_SIZE, 1);
if (!call->frame_buf) {
LOGGER_WARNING("Frame buffer allocation failed!");
goto error;
}
}
if ( !(call->j_buf = create_queue(jbuf_capacity)) ) {
LOGGER_WARNING("Jitter buffer creaton failed!");
goto error;
}
ToxAvCSettings csettings_peer = toxavcsettings_cast(&av->msi_session->calls[call_index]->csettings_peer[0]);
ToxAvCSettings csettings_local = toxavcsettings_cast(&av->msi_session->calls[call_index]->csettings_local);
LOGGER_DEBUG(
"Type: %u \n"
"Video bitrate: %u \n"
"Video height: %u \n"
"Video width: %u \n"
"Audio bitrate: %u \n"
"Audio framedur: %u \n"
"Audio sample rate: %u \n"
"Audio channels: %u \n",
csettings_peer.call_type,
csettings_peer.video_bitrate,
csettings_peer.max_video_height,
csettings_peer.max_video_width,
csettings_peer.audio_bitrate,
csettings_peer.audio_frame_duration,
csettings_peer.audio_sample_rate,
csettings_peer.audio_channels );
if ( (call->cs = codec_init_session(csettings_local.audio_bitrate,
csettings_local.audio_frame_duration,
csettings_local.audio_sample_rate,
csettings_local.audio_channels,
csettings_peer.audio_channels,
VAD_treshold,
csettings_local.max_video_width,
csettings_local.max_video_height,
csettings_local.video_bitrate) )) {
if ( pthread_mutex_init(&call->mutex, NULL) != 0 ) goto error;
//todo: add error checks
pthread_mutex_init(&call->decode_cond_mutex, NULL);
pthread_cond_init(&call->decode_cond, NULL);
void **arg = malloc(2 * sizeof(void *));
arg[0] = av;
arg[1] = call;
pthread_t temp;
pthread_attr_t attr;
pthread_attr_init(&attr);
pthread_attr_setstacksize(&attr, 1 << 18);
pthread_create(&temp, &attr, toxav_decoding, arg);
pthread_attr_destroy(&attr);
LOGGER_WARNING("Got here");
call->call_active = 1;
return ErrorNone;
}
error:
rtp_terminate_session(call->crtps[audio_index], av->messenger);
rtp_terminate_session(call->crtps[video_index], av->messenger);
free(call->frame_buf);
terminate_queue(call->j_buf);
codec_terminate_session(call->cs);
return ErrorInternal;
}
/**
* @brief Must be call before any RTP transmission occurs.
*
* @param av Handler.
* @return int
* @retval 0 Success.
* @retval ToxAvError On error.
*/
int toxav_prepare_transmission ( ToxAv *av, int32_t call_index, ToxAvCodecSettings *codec_settings, int support_video )
{
if ( !av->msi_session || cii(call_index, av->msi_session) ||
!av->msi_session->calls[call_index] || av->calls[call_index].call_active) {
LOGGER_ERROR("Error while starting RTP session: invalid call!\n");
return ErrorInternal;
}
CallSpecific *call = &av->calls[call_index];
call->crtps[audio_index] =
rtp_init_session(type_audio, av->messenger, av->msi_session->calls[call_index]->peers[0]);
if ( !call->crtps[audio_index] ) {
LOGGER_ERROR("Error while starting audio RTP session!\n");
return ErrorStartingAudioRtp;
}
if ( support_video ) {
call->crtps[video_index] =
rtp_init_session(type_video, av->messenger, av->msi_session->calls[call_index]->peers[0]);
if ( !call->crtps[video_index] ) {
LOGGER_ERROR("Error while starting video RTP session!\n");
rtp_terminate_session(call->crtps[audio_index], av->messenger);
return ErrorStartingVideoRtp;
}
call->frame_limit = 0;
call->frame_id = 0;
call->frame_outid = 0;
call->frame_buf = calloc(MAX_VIDEOFRAME_SIZE, 1);
if (!call->frame_buf) {
rtp_terminate_session(call->crtps[audio_index], av->messenger);
rtp_terminate_session(call->crtps[video_index], av->messenger);
LOGGER_WARNING("Frame buffer allocation failed!");
return ErrorInternal;
}
}
if ( !(call->j_buf = create_queue(codec_settings->jbuf_capacity)) ) {
rtp_terminate_session(call->crtps[audio_index], av->messenger);
rtp_terminate_session(call->crtps[video_index], av->messenger);
free(call->frame_buf);
LOGGER_WARNING("Jitter buffer creaton failed!");
return ErrorInternal;
}
if ( (call->cs = codec_init_session(codec_settings->audio_bitrate,
codec_settings->audio_frame_duration,
codec_settings->audio_sample_rate,
codec_settings->audio_channels,
codec_settings->audio_VAD_tolerance,
codec_settings->video_width,
codec_settings->video_height,
codec_settings->video_bitrate) )) {
call->call_active = 1;
return ErrorNone;
}
rtp_terminate_session(call->crtps[audio_index], av->messenger);
rtp_terminate_session(call->crtps[video_index], av->messenger);
free(call->frame_buf);
terminate_queue(call->j_buf);
return ErrorInternal;
}
