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Cleaned up, renamed and moved a few classes for consistency

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This commit is contained in:
Paul-Louis Ageneau
2021-01-25 11:23:14 +01:00
parent 470d145a9a
commit fe4afcef85
27 changed files with 196 additions and 170 deletions
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12
examples/streamer/main.cpp
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@ -217,9 +217,9 @@ shared_ptr<ClientTrackData> addVideo(const shared_ptr<PeerConnection> pc, const
video.addSSRC(ssrc, cname, msid);
auto track = pc->addTrack(video);
// create RTP configuration
auto rtpConfig = shared_ptr<RTPPacketizationConfig>(new RTPPacketizationConfig(ssrc, cname, payloadType, H264RTPPacketizer::defaultClockRate));
auto rtpConfig = shared_ptr<RtpPacketizationConfig>(new RtpPacketizationConfig(ssrc, cname, payloadType, H264RtpPacketizer::defaultClockRate));
// create packetizer
auto packetizer = shared_ptr<H264RTPPacketizer>(new H264RTPPacketizer(rtpConfig));
auto packetizer = shared_ptr<H264RtpPacketizer>(new H264RtpPacketizer(rtpConfig));
// create H264 and RTCP SP handler
shared_ptr<H264PacketizationHandler> h264Handler(new H264PacketizationHandler(H264PacketizationHandler::Separator::Length, packetizer));
// set handler
@ -235,9 +235,9 @@ shared_ptr<ClientTrackData> addAudio(const shared_ptr<PeerConnection> pc, const
audio.addSSRC(ssrc, cname, msid);
auto track = pc->addTrack(audio);
// create RTP configuration
auto rtpConfig = shared_ptr<RTPPacketizationConfig>(new RTPPacketizationConfig(ssrc, cname, payloadType, OpusRTPPacketizer::defaultClockRate));
auto rtpConfig = shared_ptr<RtpPacketizationConfig>(new RtpPacketizationConfig(ssrc, cname, payloadType, OpusRtpPacketizer::defaultClockRate));
// create packetizer
auto packetizer = make_shared<OpusRTPPacketizer>(rtpConfig);
auto packetizer = make_shared<OpusRtpPacketizer>(rtpConfig);
// create opus and RTCP SP handler
auto opusHandler = make_shared<OpusPacketizationHandler>(packetizer);
// set handler
@ -449,8 +449,8 @@ void addToStream(shared_ptr<Client> client, bool isAddingVideo) {
auto currentTime_s = currentTime_us / (1000 * 1000);
// set start time of stream
video->sender->rtpConfig->setStartTime(currentTime_s, RTPPacketizationConfig::EpochStart::T1970);
audio->sender->rtpConfig->setStartTime(currentTime_s, RTPPacketizationConfig::EpochStart::T1970);
video->sender->rtpConfig->setStartTime(currentTime_s, RtpPacketizationConfig::EpochStart::T1970);
audio->sender->rtpConfig->setStartTime(currentTime_s, RtpPacketizationConfig::EpochStart::T1970);
// start stat recording of RTCP SR
video->sender->startRecording();