webrtc 代码学习（三十四） SetLocal 和 SetRemote

SetLocal 和 SetRemote
作者：LanPZzzz

1. 获取到 offer， SetRemoteDescription，然后CreateAnswer

-> PeerConnection::SetRemoteDescription
->PeerConnection::ApplyRemoteDescription， 这里的主要参照是 offer 中的 sdp
->PeerConnection::UpdateTransceiversAndDataChannels
-> new_session 中有 offer 信息， const ContentInfos& new_contents = new_session.description()->contents(); 中 old_local_description 和 old_remote_description 都是null
-> for (size_t i = 0; i < new_contents.size(); ++i)
-> PeerConnection::AssociateTransceiver
-> PeerConnection::GetAssociatedTransceiver 根据 mid 从 transceivers_ 获取 transceiver，mid = content.name， sdp 中 a=mid:0。（代码1），可能找不到
-> 如果上面找不到 transceiver， RtpTransceiverDirectionHasRecv 是否是有Recv的， 如果是 PeerConnection::FindAvailableTransceiverToReceive 根据 type 值找
-> 找不到，就在重新创建 transceiver
->   transceiver->internal()->set_mid(content.name); transceiver->internal()->set_mline_index(mline_index); 设置
->
->PeerConnection::UpdateTransceiverChannel 设置Channel， 如果原先的没有，创建VoiceChannel 或者 VideoChannel
->
->
->
-> PeerConnection::UpdateSessionState
-> 如果是IsUnifiedPlan， for (auto transceiver : transceivers_) （代码2）
->PeerConnection::FindMediaSectionForTransceiver
->     return sdesc->description()->GetContentByName(transceiver->internal()->mid()); 根据 transceiver 的mid 获取 ContentInfo，因为上面已经设置了可用的 transceiver 的mid，这里就是找上面可用的，不可用就不要了
->
->
-> for (auto transceiver : now_receiving_transceivers) 经过代码2，获取出可用的 transceiver
-> StatsCollector::AddTrack
-> CreateTrackReport
->
-> Conductor::OnAddTrack，回调给外面
->
->
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代码1

PeerConnection::FindAvailableTransceiverToReceive
。。。 
  for (auto transceiver : transceivers_) {
    if (transceiver->media_type() == media_type &&
        transceiver->internal()->created_by_addtrack() && !transceiver->mid() &&
        !transceiver->stopped()) {
      return transceiver;
    }
  }
  。。。。

代码2

    for (auto transceiver : transceivers_) {
      根据上面的 mid 获取 content，获取到的是可用的
      const ContentInfo* content =
          FindMediaSectionForTransceiver(transceiver, remote_description());
      if (!content) {
        continue;
      }
      const MediaContentDescription* media_desc = content->media_description();
      RtpTransceiverDirection local_direction =
          RtpTransceiverDirectionReversed(media_desc->direction());
      // From the WebRTC specification, steps 2.2.8.5/6 of section 4.4.1.6 "Set
      // the RTCSessionDescription: If direction is sendrecv or recvonly, and
      // transceiver's current direction is neither sendrecv nor recvonly,
      // process the addition of a remote track for the media description.
      
      获取 stream
      std::vector<std::string> stream_ids;
      if (!media_desc->streams().empty()) {
        // The remote description has signaled the stream IDs.
        stream_ids = media_desc->streams()[0].stream_ids();
      }
      if (RtpTransceiverDirectionHasRecv(local_direction) &&
          (!transceiver->fired_direction() ||
           !RtpTransceiverDirectionHasRecv(*transceiver->fired_direction()))) {
        RTC_LOG(LS_INFO) << "Processing the addition of a new track for MID="
                         << content->name << " (added to "
                         << GetStreamIdsString(stream_ids) << ").";

        std::vector<rtc::scoped_refptr<MediaStreamInterface>> media_streams;
        枚举stream
        for (const std::string& stream_id : stream_ids) {
          rtc::scoped_refptr<MediaStreamInterface> stream =
              remote_streams_->find(stream_id);
              没有找到，就重新创建
          if (!stream) {
            stream = MediaStreamProxy::Create(rtc::Thread::Current(),
                                              MediaStream::Create(s