void ExternalOutput::writeAudioData(char* buf, int len){ RtpHeader* head = reinterpret_cast<RtpHeader*>(buf); uint16_t currentAudioSequenceNumber = head->getSeqNumber(); if (currentAudioSequenceNumber != lastAudioSequenceNumber_ + 1) { // Something screwy. We should always see sequence numbers incrementing monotonically. ELOG_DEBUG("Unexpected audio sequence number; current %d, previous %d", currentAudioSequenceNumber, lastAudioSequenceNumber_); } lastAudioSequenceNumber_ = currentAudioSequenceNumber; if (firstAudioTimestamp_ == -1) { firstAudioTimestamp_ = head->getTimestamp(); } timeval time; gettimeofday(&time, NULL); // Figure out our audio codec. if(context_->oformat->audio_codec == AV_CODEC_ID_NONE) { //We dont need any other payload at this time if(head->getPayloadType() == PCMU_8000_PT){ context_->oformat->audio_codec = AV_CODEC_ID_PCM_MULAW; } else if (head->getPayloadType() == OPUS_48000_PT) { context_->oformat->audio_codec = AV_CODEC_ID_OPUS; } } initContext(); if (audio_stream_ == NULL) { // not yet. return; } long long currentTimestamp = head->getTimestamp(); if (currentTimestamp - firstAudioTimestamp_ < 0) { // we wrapped. add 2^32 to correct this. We only handle a single wrap around since that's 13 hours of recording, minimum. currentTimestamp += 0xFFFFFFFF; } long long timestampToWrite = (currentTimestamp - firstAudioTimestamp_) / (audio_stream_->codec->time_base.den / audio_stream_->time_base.den); // Adjust for our start time offset timestampToWrite += audioOffsetMsec_ / (1000 / audio_stream_->time_base.den); // in practice, our timebase den is 1000, so this operation is a no-op. /* ELOG_DEBUG("Writing audio frame %d with timestamp %u, normalized timestamp %u, audio offset msec %u, length %d, input timebase: %d/%d, target timebase: %d/%d", */ /* head->getSeqNumber(), head->getTimestamp(), timestampToWrite, audioOffsetMsec_, ret, */ /* audio_stream_->codec->time_base.num, audio_stream_->codec->time_base.den, // timebase we requested */ /* audio_stream_->time_base.num, audio_stream_->time_base.den); // actual timebase */ AVPacket avpkt; av_init_packet(&avpkt); avpkt.data = (uint8_t*) buf + head->getHeaderLength(); avpkt.size = len - head->getHeaderLength(); avpkt.pts = timestampToWrite; avpkt.stream_index = 1; av_write_frame(context_, &avpkt); av_free_packet(&avpkt); }
void ExternalOutput::writeAudioData(char* buf, int len){ RtpHeader* head = reinterpret_cast<RtpHeader*>(buf); if (firstAudioTimestamp_ == -1) { firstAudioTimestamp_ = head->getTimestamp(); } timeval time; gettimeofday(&time, NULL); // Figure out our audio codec. if(context_->oformat->audio_codec == AV_CODEC_ID_NONE) { //We dont need any other payload at this time if(head->getPayloadType() == PCMU_8000_PT){ context_->oformat->audio_codec = AV_CODEC_ID_PCM_MULAW; } else if (head->getPayloadType() == OPUS_48000_PT) { context_->oformat->audio_codec = AV_CODEC_ID_OPUS; } } initContext(); if (audio_stream_ == NULL) { // not yet. return; } int ret = inputProcessor_->unpackageAudio(reinterpret_cast<unsigned char*>(buf), len, unpackagedAudioBuffer_); if (ret <= 0) return; long long currentTimestamp = head->getTimestamp(); if (currentTimestamp - firstAudioTimestamp_ < 0) { // we wrapped. add 2^32 to correct this. We only handle a single wrap around since that's 13 hours of recording, minimum. currentTimestamp += 0xFFFFFFFF; } long long timestampToWrite = (currentTimestamp - firstAudioTimestamp_) / (audio_stream_->codec->time_base.den / audio_stream_->time_base.den); // Adjust for our start time offset timestampToWrite += audioOffsetMsec_ / (1000 / audio_stream_->time_base.den); // in practice, our timebase den is 1000, so this operation is a no-op. /* ELOG_DEBUG("Writing audio frame %d with timestamp %u, normalized timestamp %u, audio offset msec %u, length %d, input timebase: %d/%d, target timebase: %d/%d", */ /* head->getSeqNumber(), head->getTimestamp(), timestampToWrite, audioOffsetMsec_, ret, */ /* audio_stream_->codec->time_base.num, audio_stream_->codec->time_base.den, // timebase we requested */ /* audio_stream_->time_base.num, audio_stream_->time_base.den); // actual timebase */ AVPacket avpkt; av_init_packet(&avpkt); avpkt.data = unpackagedAudioBuffer_; avpkt.size = ret; avpkt.pts = timestampToWrite; avpkt.stream_index = 1; av_write_frame(context_, &avpkt); av_free_packet(&avpkt); }
void ExternalOutput::writeAudioData(char* buf, int len) { RtpHeader* head = reinterpret_cast<RtpHeader*>(buf); uint16_t currentAudioSequenceNumber = head->getSeqNumber(); if (first_audio_timestamp_ != -1 && currentAudioSequenceNumber != lastAudioSequenceNumber_ + 1) { // Something screwy. We should always see sequence numbers incrementing monotonically. ELOG_DEBUG("Unexpected audio sequence number; current %d, previous %d", currentAudioSequenceNumber, lastAudioSequenceNumber_); } lastAudioSequenceNumber_ = currentAudioSequenceNumber; if (first_audio_timestamp_ == -1) { first_audio_timestamp_ = head->getTimestamp(); } // Figure out our audio codec. if (context_->oformat->audio_codec == AV_CODEC_ID_NONE) { // We dont need any other payload at this time if (head->getPayloadType() == PCMU_8000_PT) { context_->oformat->audio_codec = AV_CODEC_ID_PCM_MULAW; } else if (head->getPayloadType() == OPUS_48000_PT) { context_->oformat->audio_codec = AV_CODEC_ID_OPUS; } } initContext(); if (audio_stream_ == NULL) { // not yet. return; } long long currentTimestamp = head->getTimestamp(); // NOLINT if (currentTimestamp - first_audio_timestamp_ < 0) { // we wrapped. add 2^32 to correct this. We only handle a single wrap around // since that's 13 hours of recording, minimum. currentTimestamp += 0xFFFFFFFF; } long long timestampToWrite = (currentTimestamp - first_audio_timestamp_) / // NOLINT (audio_stream_->codec->sample_rate / audio_stream_->time_base.den); // generally 48000 / 1000 for the denominator portion, at least for opus // Adjust for our start time offset // in practice, our timebase den is 1000, so this operation is a no-op. timestampToWrite += audio_offset_ms_ / (1000 / audio_stream_->time_base.den); AVPacket avpkt; av_init_packet(&avpkt); avpkt.data = reinterpret_cast<uint8_t*>(buf) + head->getHeaderLength(); avpkt.size = len - head->getHeaderLength(); avpkt.pts = timestampToWrite; avpkt.stream_index = 1; av_interleaved_write_frame(context_, &avpkt); // takes ownership of the packet }
bool RtcpRrGenerator::isRetransmitOfOldPacket(std::shared_ptr<dataPacket> packet) { RtpHeader *head = reinterpret_cast<RtpHeader*>(packet->data); if (!RtpUtils::sequenceNumberLessThan(head->getSeqNumber(), rr_info_.max_seq) || rr_info_.jitter.jitter == 0) { return false; } int64_t time_diff_ms = static_cast<uint32_t>(packet->received_time_ms) - rr_info_.last_recv_ts; int64_t timestamp_diff = static_cast<int32_t>(head->getTimestamp() - rr_info_.last_rtp_ts); uint16_t clock_rate = type_ == VIDEO_PACKET ? getVideoClockRate(head->getPayloadType()) : getAudioClockRate(head->getPayloadType()); int64_t rtp_time_stamp_diff_ms = timestamp_diff / clock_rate; int64_t max_delay_ms = ((2 * rr_info_.jitter.jitter) / clock_rate); return time_diff_ms > rtp_time_stamp_diff_ms + max_delay_ms; }
bool RtcpRrGenerator::handleRtpPacket(std::shared_ptr<dataPacket> packet) { RtpHeader *head = reinterpret_cast<RtpHeader*>(packet->data); if (ssrc_ != head->getSSRC()) { ELOG_DEBUG("message: handleRtpPacket ssrc not found, ssrc: %u", head->getSSRC()); return false; } uint16_t seq_num = head->getSeqNumber(); rr_info_.packets_received++; if (rr_info_.base_seq == -1) { rr_info_.base_seq = head->getSeqNumber(); } if (rr_info_.max_seq == -1) { rr_info_.max_seq = seq_num; } else if (!RtpUtils::sequenceNumberLessThan(seq_num, rr_info_.max_seq)) { if (seq_num < rr_info_.max_seq) { rr_info_.cycle++; } rr_info_.max_seq = seq_num; } rr_info_.extended_seq = (rr_info_.cycle << 16) | rr_info_.max_seq; uint16_t clock_rate = type_ == VIDEO_PACKET ? getVideoClockRate(head->getPayloadType()) : getAudioClockRate(head->getPayloadType()); if (head->getTimestamp() != rr_info_.last_rtp_ts && !isRetransmitOfOldPacket(packet)) { int transit_time = static_cast<int>((packet->received_time_ms * clock_rate) - head->getTimestamp()); int delta = abs(transit_time - rr_info_.jitter.transit_time); if (rr_info_.jitter.transit_time != 0 && delta < MAX_DELAY) { rr_info_.jitter.jitter += (1. / 16.) * (static_cast<double>(delta) - rr_info_.jitter.jitter); } rr_info_.jitter.transit_time = transit_time; } rr_info_.last_rtp_ts = head->getTimestamp(); rr_info_.last_recv_ts = static_cast<uint32_t>(packet->received_time_ms); uint64_t now = ClockUtils::timePointToMs(clock_->now()); if (rr_info_.next_packet_ms == 0) { // Schedule the first packet uint16_t selected_interval = selectInterval(); rr_info_.next_packet_ms = now + selected_interval; return false; } if (now >= rr_info_.next_packet_ms) { ELOG_DEBUG("message: should send packet, ssrc: %u", ssrc_); return true; } return false; }
void ExternalOutput::writeVideoData(char* buf, int len) { RtpHeader* head = reinterpret_cast<RtpHeader*>(buf); uint16_t current_video_sequence_number = head->getSeqNumber(); if (current_video_sequence_number != last_video_sequence_number_ + 1) { // Something screwy. We should always see sequence numbers incrementing monotonically. ELOG_DEBUG("Unexpected video sequence number; current %d, previous %d", current_video_sequence_number, last_video_sequence_number_); // Restart the depacketizer so it looks for the start of a frame if (depacketizer_!= nullptr) { depacketizer_->reset(); } } last_video_sequence_number_ = current_video_sequence_number; if (first_video_timestamp_ == -1) { first_video_timestamp_ = head->getTimestamp(); } auto map_iterator = video_maps_.find(head->getPayloadType()); if (map_iterator != video_maps_.end()) { updateVideoCodec(map_iterator->second); if (map_iterator->second.encoding_name == "VP8" || map_iterator->second.encoding_name == "H264") { maybeWriteVideoPacket(buf, len); } } }
int InputProcessor::unpackageVideo(unsigned char* inBuff, int inBuffLen, unsigned char* outBuff, int* gotFrame) { if (videoUnpackager == 0) { ELOG_DEBUG("Unpackager not correctly initialized"); return -1; } int inBuffOffset = 0; *gotFrame = 0; RtpHeader* head = reinterpret_cast<RtpHeader*>(inBuff); if (head->getPayloadType() != 100) { return -1; } int l = inBuffLen - head->getHeaderLength(); inBuffOffset+=head->getHeaderLength(); erizo::RTPPayloadVP8* parsed = pars.parseVP8((unsigned char*) &inBuff[inBuffOffset], l); memcpy(outBuff, parsed->data, parsed->dataLength); if (head->getMarker()) { *gotFrame = 1; } int ret = parsed->dataLength; delete parsed; return ret; }
int main(int argc, const char * argv[]) { // data setup uint32_t first = 0xFFFF | 0x1FFFFFFF; uint32_t timestamp = 0x0128; uint32_t ssrc = 0x01 | 0x02 | 0x04 | 0x08 | 0x256; int somedata[4]; // The htonl() function converts the unsigned integer hostlong from host byte order to network byte order. // on the other side: // The ntohl() function converts the unsigned integer netlong from network byte order to host byte order. somedata[0] = htonl(first); somedata[1] = htonl(timestamp); somedata[2] = htonl(ssrc); somedata[2] = htonl(ssrc); somedata[3] = htonl(ssrc); RtpHeader* head = reinterpret_cast<RtpHeader*>(somedata); printf("version: %" PRIu8 "\n", head->getVersion()); printf("padding: %" PRIu8 "\n", head->hasPadding()); printf("extension: %" PRIu8 "\n", head->getExtension()); printf("marker: %" PRIu8 "\n", head->getMarker()); printf("payload type: %" PRIu8 "\n", head->getPayloadType()); printf("sequence number: %" PRIu16 "\n", head->getSeqNumber()); printf("timestamp %" PRIu32 "\n", head->getTimestamp()); printf("ssrc %" PRIu32 "\n", head->getSSRC()); printf("header length: %u\n", head->getHeaderLength()); return 0; }
void RRGenerationHandler::handleRtpPacket(std::shared_ptr<dataPacket> packet) { RtpHeader *head = reinterpret_cast<RtpHeader*>(packet->data); auto rr_packet_pair = rr_info_map_.find(head->getSSRC()); if (rr_packet_pair == rr_info_map_.end()) { ELOG_DEBUG("%s message: handleRtpPacket ssrc not found, ssrc: %u", connection_->toLog(), head->getSSRC()); return; } std::shared_ptr<RRPackets> selected_packet_info = rr_packet_pair->second; uint16_t seq_num = head->getSeqNumber(); selected_packet_info->packets_received++; if (selected_packet_info->base_seq == -1) { selected_packet_info->ssrc = head->getSSRC(); selected_packet_info->base_seq = head->getSeqNumber(); } if (selected_packet_info->max_seq == -1) { selected_packet_info->max_seq = seq_num; } else if (!rtpSequenceLessThan(seq_num, selected_packet_info->max_seq)) { if (seq_num < selected_packet_info->max_seq) { selected_packet_info->cycle++; } selected_packet_info->max_seq = seq_num; } selected_packet_info->extended_seq = (selected_packet_info->cycle << 16) | selected_packet_info->max_seq; uint16_t clock_rate = selected_packet_info->type == VIDEO_PACKET ? getVideoClockRate(head->getPayloadType()) : getAudioClockRate(head->getPayloadType()); if (head->getTimestamp() != selected_packet_info->last_rtp_ts && !isRetransmitOfOldPacket(packet, selected_packet_info)) { int transit_time = static_cast<int>((packet->received_time_ms * clock_rate) - head->getTimestamp()); int delta = abs(transit_time - selected_packet_info->jitter.transit_time); if (selected_packet_info->jitter.transit_time != 0 && delta < MAX_DELAY) { selected_packet_info->jitter.jitter += (1. / 16.) * (static_cast<double>(delta) - selected_packet_info->jitter.jitter); } selected_packet_info->jitter.transit_time = transit_time; } selected_packet_info->last_rtp_ts = head->getTimestamp(); selected_packet_info->last_recv_ts = static_cast<uint32_t>(packet->received_time_ms); uint64_t now = ClockUtils::timePointToMs(clock::now()); if (selected_packet_info->next_packet_ms == 0) { // Schedule the first packet uint16_t selected_interval = selectInterval(selected_packet_info); selected_packet_info->next_packet_ms = now + selected_interval; return; } if (now >= selected_packet_info->next_packet_ms) { sendRR(selected_packet_info); } }
void ExternalOutput::writeAudioData(char* buf, int len) { RtpHeader* head = reinterpret_cast<RtpHeader*>(buf); uint16_t current_audio_sequence_number = head->getSeqNumber(); if (first_audio_timestamp_ != -1 && current_audio_sequence_number != last_audio_sequence_number_ + 1) { // Something screwy. We should always see sequence numbers incrementing monotonically. ELOG_DEBUG("Unexpected audio sequence number; current %d, previous %d", current_audio_sequence_number, last_audio_sequence_number_); } last_audio_sequence_number_ = current_audio_sequence_number; if (first_audio_timestamp_ == -1) { first_audio_timestamp_ = head->getTimestamp(); } auto map_iterator = audio_maps_.find(head->getPayloadType()); if (map_iterator != audio_maps_.end()) { updateAudioCodec(map_iterator->second); } initContext(); if (audio_stream_ == nullptr) { // not yet. return; } long long current_timestamp = head->getTimestamp(); // NOLINT if (current_timestamp - first_audio_timestamp_ < 0) { // we wrapped. add 2^32 to correct this. We only handle a single wrap around // since that's 13 hours of recording, minimum. current_timestamp += 0xFFFFFFFF; } long long timestamp_to_write = (current_timestamp - first_audio_timestamp_) / // NOLINT (audio_stream_->codec->sample_rate / audio_stream_->time_base.den); // generally 48000 / 1000 for the denominator portion, at least for opus // Adjust for our start time offset // in practice, our timebase den is 1000, so this operation is a no-op. timestamp_to_write += audio_offset_ms_ / (1000 / audio_stream_->time_base.den); AVPacket av_packet; av_init_packet(&av_packet); av_packet.data = reinterpret_cast<uint8_t*>(buf) + head->getHeaderLength(); av_packet.size = len - head->getHeaderLength(); av_packet.pts = timestamp_to_write; av_packet.stream_index = 1; av_interleaved_write_frame(context_, &av_packet); // takes ownership of the packet }
void MediaStream::changeDeliverPayloadType(DataPacket *dp, packetType type) { RtpHeader* h = reinterpret_cast<RtpHeader*>(dp->data); RtcpHeader *chead = reinterpret_cast<RtcpHeader*>(dp->data); if (!chead->isRtcp()) { int internalPT = h->getPayloadType(); int externalPT = internalPT; if (type == AUDIO_PACKET) { externalPT = remote_sdp_->getAudioExternalPT(internalPT); } else if (type == VIDEO_PACKET) { externalPT = remote_sdp_->getVideoExternalPT(externalPT); } if (internalPT != externalPT) { h->setPayloadType(externalPT); } } }
int OneToManyTranscoder::deliverVideoData_(char* buf, int len) { memcpy(sendVideoBuffer_, buf, len); RtpHeader* theHead = reinterpret_cast<RtpHeader*>(buf); // ELOG_DEBUG("extension %d pt %u", theHead->getExtension(), // theHead->getPayloadType()); if (theHead->getPayloadType() == 100) { ip_->deliverVideoData(sendVideoBuffer_, len); } else { this->receiveRtpData((unsigned char*) buf, len); } sentPackets_++; return 0; }
// parses incoming payload type, replaces occurence in buf void MediaStream::parseIncomingPayloadType(char *buf, int len, packetType type) { RtcpHeader* chead = reinterpret_cast<RtcpHeader*>(buf); RtpHeader* h = reinterpret_cast<RtpHeader*>(buf); if (!chead->isRtcp()) { int externalPT = h->getPayloadType(); int internalPT = externalPT; if (type == AUDIO_PACKET) { internalPT = remote_sdp_->getAudioInternalPT(externalPT); } else if (type == VIDEO_PACKET) { internalPT = remote_sdp_->getVideoInternalPT(externalPT); } if (externalPT != internalPT) { h->setPayloadType(internalPT); } else { // ELOG_WARN("onTransportData did not find mapping for %i", externalPT); } } }
// parses incoming payload type, replaces occurence in buf void WebRtcConnection::parseIncomingPayloadType(char *buf, int len, packetType type) { RtcpHeader* chead = reinterpret_cast<RtcpHeader*>(buf); RtpHeader* h = reinterpret_cast<RtpHeader*>(buf); if (!chead->isRtcp()) { int externalPT = h->getPayloadType(); int internalPT = externalPT; if (type == AUDIO_PACKET) { internalPT = remoteSdp_.getAudioInternalPT(externalPT); } else if (type == VIDEO_PACKET) { internalPT = remoteSdp_.getVideoInternalPT(externalPT); } if (externalPT != internalPT) { h->setPayloadType(internalPT); // ELOG_ERROR("onTransportData mapping %i to %i", externalPT, internalPT); } else { // ELOG_ERROR("onTransportData did not find mapping for %i", externalPT); } } }
int WebRtcConnection::deliverVideoData_(char* buf, int len) { writeSsrc(buf, len, this->getVideoSinkSSRC()); if (videoTransport_ != NULL) { if (videoEnabled_ == true) { RtpHeader* h = reinterpret_cast<RtpHeader*>(buf); if (h->getPayloadType() == RED_90000_PT && !remoteSdp_.supportPayloadType(RED_90000_PT)) { // This is a RED/FEC payload, but our remote endpoint doesn't support that (most likely because it's firefox :/ ) // Let's go ahead and run this through our fec receiver to convert it to raw VP8 webrtc::RTPHeader hackyHeader; hackyHeader.headerLength = h->getHeaderLength(); hackyHeader.sequenceNumber = h->getSeqNumber(); // FEC copies memory, manages its own memory, including memory passed in callbacks (in the callback, be sure to memcpy out of webrtc's buffers if (fec_receiver_.AddReceivedRedPacket(hackyHeader, (const uint8_t*) buf, len, ULP_90000_PT) == 0) { fec_receiver_.ProcessReceivedFec(); } } else { this->queueData(0, buf, len, videoTransport_); } } } return len; }
void ExternalOutput::writeVideoData(char* buf, int len){ RtpHeader* head = reinterpret_cast<RtpHeader*>(buf); if (head->getPayloadType() == RED_90000_PT) { int totalLength = head->getHeaderLength(); int rtpHeaderLength = totalLength; RedHeader *redhead = reinterpret_cast<RedHeader*>(buf + totalLength); if (redhead->payloadtype == VP8_90000_PT) { while (redhead->follow) { totalLength += redhead->getLength() + 4; // RED header redhead = reinterpret_cast<RedHeader*>(buf + totalLength); } // Parse RED packet to VP8 packet. // Copy RTP header memcpy(deliverMediaBuffer_, buf, rtpHeaderLength); // Copy payload data memcpy(deliverMediaBuffer_ + totalLength, buf + totalLength + 1, len - totalLength - 1); // Copy payload type RtpHeader *mediahead = reinterpret_cast<RtpHeader*>(deliverMediaBuffer_); mediahead->setPayloadType(redhead->payloadtype); buf = reinterpret_cast<char*>(deliverMediaBuffer_); len = len - 1 - totalLength + rtpHeaderLength; } } if (firstVideoTimestamp_ == -1) { firstVideoTimestamp_ = head->getTimestamp(); } int gotUnpackagedFrame = false; int ret = inputProcessor_->unpackageVideo(reinterpret_cast<unsigned char*>(buf), len, unpackagedBufferpart_, &gotUnpackagedFrame); if (ret < 0){ ELOG_ERROR("Error Unpackaging Video"); return; } initContext(); if (video_stream_ == NULL) { // could not init our context yet. return; } unpackagedSize_ += ret; unpackagedBufferpart_ += ret; if (gotUnpackagedFrame) { unpackagedBufferpart_ -= unpackagedSize_; long long currentTimestamp = head->getTimestamp(); if (currentTimestamp - firstVideoTimestamp_ < 0) { // we wrapped. add 2^32 to correct this. We only handle a single wrap around since that's ~13 hours of recording, minimum. currentTimestamp += 0xFFFFFFFF; } long long timestampToWrite = (currentTimestamp - firstVideoTimestamp_) / (90000 / video_stream_->time_base.den); // All of our video offerings are using a 90khz clock. // Adjust for our start time offset timestampToWrite += videoOffsetMsec_ / (1000 / video_stream_->time_base.den); // in practice, our timebase den is 1000, so this operation is a no-op. /* ELOG_DEBUG("Writing video frame %d with timestamp %u, normalized timestamp %u, video offset msec %u, length %d, input timebase: %d/%d, target timebase: %d/%d", */ /* head->getSeqNumber(), head->getTimestamp(), timestampToWrite, videoOffsetMsec_, unpackagedSize_, */ /* video_stream_->codec->time_base.num, video_stream_->codec->time_base.den, // timebase we requested */ /* video_stream_->time_base.num, video_stream_->time_base.den); // actual timebase */ AVPacket avpkt; av_init_packet(&avpkt); avpkt.data = unpackagedBufferpart_; avpkt.size = unpackagedSize_; avpkt.pts = timestampToWrite; avpkt.stream_index = 0; av_write_frame(context_, &avpkt); av_free_packet(&avpkt); unpackagedSize_ = 0; unpackagedBufferpart_ = unpackagedBuffer_; } }
void ExternalOutput::queueData(char* buffer, int length, packetType type) { if (!recording_) { return; } RtcpHeader *head = reinterpret_cast<RtcpHeader*>(buffer); if (head->isRtcp()) { return; } if (first_data_received_ == time_point()) { first_data_received_ = clock::now(); if (getAudioSinkSSRC() == 0) { ELOG_DEBUG("No audio detected"); audio_map_ = RtpMap{0, "PCMU", 8000, AUDIO_TYPE, 1}; audio_codec_ = AV_CODEC_ID_PCM_MULAW; } } if (need_to_send_fir_ && video_source_ssrc_) { sendFirPacket(); need_to_send_fir_ = false; } if (type == VIDEO_PACKET) { RtpHeader* h = reinterpret_cast<RtpHeader*>(buffer); uint8_t payloadtype = h->getPayloadType(); if (video_offset_ms_ == -1) { video_offset_ms_ = ClockUtils::durationToMs(clock::now() - first_data_received_); ELOG_DEBUG("File %s, video offset msec: %llu", context_->filename, video_offset_ms_); video_queue_.setTimebase(video_maps_[payloadtype].clock_rate); } // If this is a red header, let's push it to our fec_receiver_, which will spit out frames in one // of our other callbacks. // Otherwise, just stick it straight into the video queue. if (payloadtype == RED_90000_PT) { // The only things AddReceivedRedPacket uses are headerLength and sequenceNumber. // Unfortunately the amount of crap // we would have to pull in from the WebRtc project to fully construct // a webrtc::RTPHeader object is obscene. So // let's just do this hacky fix. webrtc::RTPHeader hacky_header; hacky_header.headerLength = h->getHeaderLength(); hacky_header.sequenceNumber = h->getSeqNumber(); // AddReceivedRedPacket returns 0 if there's data to process if (0 == fec_receiver_->AddReceivedRedPacket(hacky_header, (const uint8_t*)buffer, length, ULP_90000_PT)) { fec_receiver_->ProcessReceivedFec(); } } else { video_queue_.pushPacket(buffer, length); } } else { if (audio_offset_ms_ == -1) { audio_offset_ms_ = ClockUtils::durationToMs(clock::now() - first_data_received_); ELOG_DEBUG("File %s, audio offset msec: %llu", context_->filename, audio_offset_ms_); // Let's also take a moment to set our audio queue timebase. RtpHeader* h = reinterpret_cast<RtpHeader*>(buffer); if (h->getPayloadType() == PCMU_8000_PT) { audio_queue_.setTimebase(8000); } else if (h->getPayloadType() == OPUS_48000_PT) { audio_queue_.setTimebase(48000); } } audio_queue_.pushPacket(buffer, length); } if (audio_queue_.hasData() || video_queue_.hasData()) { // One or both of our queues has enough data to write stuff out. Notify our writer. cond_.notify_one(); } }
void ExternalOutput::queueData(char* buffer, int length, packetType type){ if (!recording_) { return; } RtcpHeader *head = reinterpret_cast<RtcpHeader*>(buffer); if (head->isRtcp()){ return; } if (firstDataReceived_ == -1) { timeval time; gettimeofday(&time, NULL); firstDataReceived_ = (time.tv_sec * 1000) + (time.tv_usec / 1000); if (this->getAudioSinkSSRC() == 0){ ELOG_DEBUG("No audio detected"); context_->oformat->audio_codec = AV_CODEC_ID_PCM_MULAW; } } timeval time; gettimeofday(&time, NULL); unsigned long long millis = (time.tv_sec * 1000) + (time.tv_usec / 1000); if (millis -lastFullIntraFrameRequest_ >FIR_INTERVAL_MS){ this->sendFirPacket(); lastFullIntraFrameRequest_ = millis; } if (type == VIDEO_PACKET){ if(this->videoOffsetMsec_ == -1) { videoOffsetMsec_ = ((time.tv_sec * 1000) + (time.tv_usec / 1000)) - firstDataReceived_; ELOG_DEBUG("File %s, video offset msec: %llu", context_->filename, videoOffsetMsec_); } // If this is a red header, let's push it to our fec_receiver_, which will spit out frames in one of our other callbacks. // Otherwise, just stick it straight into the video queue. RtpHeader* h = reinterpret_cast<RtpHeader*>(buffer); if (h->getPayloadType() == RED_90000_PT) { // The only things AddReceivedRedPacket uses are headerLength and sequenceNumber. Unfortunately the amount of crap // we would have to pull in from the WebRtc project to fully construct a webrtc::RTPHeader object is obscene. So // let's just do this hacky fix. webrtc::RTPHeader hackyHeader; hackyHeader.headerLength = h->getHeaderLength(); hackyHeader.sequenceNumber = h->getSeqNumber(); fec_receiver_.AddReceivedRedPacket(hackyHeader, (const uint8_t*)buffer, length, ULP_90000_PT); fec_receiver_.ProcessReceivedFec(); } else { videoQueue_.pushPacket(buffer, length); } }else{ if(this->audioOffsetMsec_ == -1) { audioOffsetMsec_ = ((time.tv_sec * 1000) + (time.tv_usec / 1000)) - firstDataReceived_; ELOG_DEBUG("File %s, audio offset msec: %llu", context_->filename, audioOffsetMsec_); } audioQueue_.pushPacket(buffer, length); } if( audioQueue_.hasData() || videoQueue_.hasData()) { // One or both of our queues has enough data to write stuff out. Notify our writer. cond_.notify_one(); } }