Análisis de código fuente de WebRtcConnection (3)

Análisis del código fuente de WebRtcConnection, analiza el papel de cada función
// 异步加入远端candidate
boost::future<void> WebRtcConnection::addRemoteCandidate(std::string mid, int mLineIndex, std::string sdp) {
  return asyncTask([mid, mLineIndex, sdp] (std::shared_ptr<WebRtcConnection> connection) {
    connection->addRemoteCandidateSync(mid, mLineIndex, sdp);
  });
}

// 加入远端candidate实现，向remote_sdp_加入candidate信息，类似前面processRemoteSdp
bool WebRtcConnection::addRemoteCandidateSync(std::string mid, int mLineIndex, std::string sdp) {
  // TODO(pedro) Check type of transport.
  ELOG_DEBUG("%s message: Adding remote Candidate, candidate: %s, mid: %s, sdpMLine: %d",
              toLog(), sdp.c_str(), mid.c_str(), mLineIndex);
  if (video_transport_ == nullptr && audio_transport_ == nullptr) {
    ELOG_WARN("%s message: addRemoteCandidate on NULL transport", toLog());
    return false;
  }
  MediaType theType;
  std::string theMid;

  // TODO(pedro) check if this works with video+audio and no bundle
  if (mLineIndex == -1) {
    ELOG_DEBUG("%s message: All candidates received", toLog());
    if (video_transport_) {
      video_transport_->getIceConnection()->setReceivedLastCandidate(true);
    } else if (audio_transport_) {
      audio_transport_->getIceConnection()->setReceivedLastCandidate(true);
    }
    return true;
  }

  if ((!mid.compare("video")) || (mLineIndex == remote_sdp_->videoSdpMLine)) {
    theType = VIDEO_TYPE;
    theMid = "video";
  } else {
    theType = AUDIO_TYPE;
    theMid = "audio";
  }
  SdpInfo tempSdp(rtp_mappings_);
  std::string username = remote_sdp_->getUsername(theType);
  std::string password = remote_sdp_->getPassword(theType);
  tempSdp.setCredentials(username, password, OTHER);
  bool res = false;
  if (tempSdp.initWithSdp(sdp, theMid)) {
    if (theType == VIDEO_TYPE || bundle_) {
      res = video_transport_->setRemoteCandidates(tempSdp.getCandidateInfos(), bundle_);
    } else if (theType == AUDIO_TYPE) {
      res = audio_transport_->setRemoteCandidates(tempSdp.getCandidateInfos(), bundle_);
    } else {
      ELOG_ERROR("%s message: add remote candidate with no Media (video or audio), candidate: %s",
                  toLog(), sdp.c_str() );
    }
  }

  for (uint8_t it = 0; it < tempSdp.getCandidateInfos().size(); it++) {
    remote_sdp_->addCandidate(tempSdp.getCandidateInfos()[it]);
  }
  return res;
}

// 在audio_transport_与video_transport_中处理本地local_sdp_,并返回sdp信息
std::string WebRtcConnection::getLocalSdp() {
  ELOG_DEBUG("%s message: Getting Local Sdp", toLog());
  if (video_transport_ != nullptr && getCurrentState() != CONN_READY) {
    video_transport_->processLocalSdp(local_sdp_.get());
  }
  if (!bundle_ && audio_transport_ != nullptr && getCurrentState() != CONN_READY) {
    audio_transport_->processLocalSdp(local_sdp_.get());
  }
  local_sdp_->profile = remote_sdp_->profile;
  return local_sdp_->getSdp();
}

// 获取json格式candidate sdp字符串
std::string WebRtcConnection::getJSONCandidate(const std::string& mid, const std::string& sdp) {
  std::map <std::string, std::string> object;
  object["sdpMid"] = mid;
  object["candidate"] = sdp;
  object["sdpMLineIndex"] =
  std::to_string((mid.compare("video")?local_sdp_->audioSdpMLine : local_sdp_->videoSdpMLine));

  std::ostringstream theString;
  theString << "{";
  for (std::map<std::string, std::string>::const_iterator it = object.begin(); it != object.end(); ++it) {
    theString << "\"" << it->first << "\":\"" << it->second << "\"";
    if (++it != object.end()) {
      theString << ",";
    }
    --it;
  }
  theString << "}";
  return theString.str();
}

// 向上回调远端 audio或video的candidate json字符串
void WebRtcConnection::onCandidate(const CandidateInfo& cand, Transport *transport) {
  std::string sdp = local_sdp_->addCandidate(cand);
  ELOG_DEBUG("%s message: Discovered New Candidate, candidate: %s", toLog(), sdp.c_str());
  if (trickle_enabled_) {
    if (!bundle_) {
      std::string object = this->getJSONCandidate(transport->transport_name, sdp);
      maybeNotifyWebRtcConnectionEvent(CONN_CANDIDATE, object);
    } else {
      if (remote_sdp_->hasAudio) {
        std::string object = this->getJSONCandidate("audio", sdp);
        maybeNotifyWebRtcConnectionEvent(CONN_CANDIDATE, object);
      }
      if (remote_sdp_->hasVideo) {
        std::string object2 = this->getJSONCandidate("video", sdp);
        maybeNotifyWebRtcConnectionEvent(CONN_CANDIDATE, object2);
      }
    }
  }
}

// remb的流控回调
void WebRtcConnection::onREMBFromTransport(RtcpHeader *chead, Transport *transport) {
  std::vector<std::shared_ptr<MediaStream>> streams;

  for (uint8_t index = 0; index < chead->getREMBNumSSRC(); index++) {
    uint32_t ssrc_feed = chead->getREMBFeedSSRC(index);
    forEachMediaStream([ssrc_feed, &streams] (const std::shared_ptr<MediaStream> &media_stream) {
      if (media_stream->isSinkSSRC(ssrc_feed)) {
        streams.push_back(media_stream);
      }
    });
  }

  distributor_->distribute(chead->getREMBBitRate(), chead->getSSRC(), streams, transport);
}

// rtcp的数据回调，并发送给media_streams_
void WebRtcConnection::onRtcpFromTransport(std::shared_ptr<DataPacket> packet, Transport *transport) {
  RtpUtils::forEachRtcpBlock(packet, [this, packet, transport](RtcpHeader *chead) {
    uint32_t ssrc = chead->isFeedback() ? chead->getSourceSSRC() : chead->getSSRC();
    if (chead->isREMB()) {
      onREMBFromTransport(chead, transport);
      return;
    }
    std::shared_ptr<DataPacket> rtcp = std::make_shared<DataPacket>(*packet);
    rtcp->length = (ntohs(chead->length) + 1) * 4;
    std::memcpy(rtcp->data, chead, rtcp->length);
    forEachMediaStream([rtcp, transport, ssrc] (const std::shared_ptr<MediaStream> &media_stream) {
      if (media_stream->isSourceSSRC(ssrc) || media_stream->isSinkSSRC(ssrc)) {
        media_stream->onTransportData(rtcp, transport);
      }
    });
  });
}

// rtp的数据回调，并发送给media_streams_
void WebRtcConnection::onTransportData(std::shared_ptr<DataPacket> packet, Transport *transport) {
  if (getCurrentState() != CONN_READY) {
    return;
  }
  char* buf = packet->data;
  RtcpHeader *chead = reinterpret_cast<RtcpHeader*> (buf);
  if (chead->isRtcp()) {
    onRtcpFromTransport(packet, transport);
    return;
  } else {
    RtpHeader *head = reinterpret_cast<RtpHeader*> (buf);
    uint32_t ssrc = head->getSSRC();
    forEachMediaStream([packet, transport, ssrc] (const std::shared_ptr<MediaStream> &media_stream) {
      if (media_stream->isSourceSSRC(ssrc) || media_stream->isSinkSSRC(ssrc)) {
        media_stream->onTransportData(packet, transport);
      }
    });
  }
}

// 事件回调通知上层
void WebRtcConnection::maybeNotifyWebRtcConnectionEvent(const WebRTCEvent& event, const std::string& message) {
  boost::mutex::scoped_lock lock(event_listener_mutex_);
  if (!conn_event_listener_) {
      return;
  }
  conn_event_listener_->notifyEvent(event, message);
}

// 异步执行任务函数
boost::future<void> WebRtcConnection::asyncTask(
    std::function<void(std::shared_ptr<WebRtcConnection>)> f) {
  auto task_promise = std::make_shared<boost::promise<void>>();
  std::weak_ptr<WebRtcConnection> weak_this = shared_from_this();
  worker_->task([weak_this, f, task_promise] {
    if (auto this_ptr = weak_this.lock()) {
      f(this_ptr);
    }
    task_promise->set_value();
  });
  return task_promise->get_future();
}

// 更新WebRtcConnection当前状态，started、gathered、ready、failed,并回调状态至上层
void WebRtcConnection::updateState(TransportState state, Transport * transport) {
  boost::mutex::scoped_lock lock(update_state_mutex_);
  WebRTCEvent temp = global_state_;
  std::string msg = "";
  ELOG_DEBUG("%s transportName: %s, new_state: %d", toLog(), transport->transport_name.c_str(), state);
  if (video_transport_.get() == nullptr && audio_transport_.get() == nullptr) {
    ELOG_ERROR("%s message: Updating NULL transport, state: %d", toLog(), state);
    return;
  }
  if (global_state_ == CONN_FAILED) {
    // if current state is failed -> noop
    return;
  }
  switch (state) {
    case TRANSPORT_STARTED:
      if (bundle_) {
        temp = CONN_STARTED;
      } else {
        if ((!remote_sdp_->hasAudio || (audio_transport_.get() != nullptr
                  && audio_transport_->getTransportState() == TRANSPORT_STARTED)) &&
            (!remote_sdp_->hasVideo || (video_transport_.get() != nullptr
                  && video_transport_->getTransportState() == TRANSPORT_STARTED))) {
            // WebRTCConnection will be ready only when all channels are ready.
            temp = CONN_STARTED;
          }
      }
      break;
    case TRANSPORT_GATHERED:
      if (bundle_) {
        if (!remote_sdp_->getCandidateInfos().empty()) {
          // Passing now new candidates that could not be passed before
          if (remote_sdp_->hasVideo) {
            video_transport_->setRemoteCandidates(remote_sdp_->getCandidateInfos(), bundle_);
          }
          if (!bundle_ && remote_sdp_->hasAudio) {
            audio_transport_->setRemoteCandidates(remote_sdp_->getCandidateInfos(), bundle_);
          }
        }
        if (!trickle_enabled_) {
          temp = CONN_GATHERED;
          msg = this->getLocalSdp();
        }
      } else {
        if ((!local_sdp_->hasAudio || (audio_transport_.get() != nullptr
                  && audio_transport_->getTransportState() == TRANSPORT_GATHERED)) &&
            (!local_sdp_->hasVideo || (video_transport_.get() != nullptr
                  && video_transport_->getTransportState() == TRANSPORT_GATHERED))) {
            // WebRTCConnection will be ready only when all channels are ready.
            if (!trickle_enabled_) {
              temp = CONN_GATHERED;
              msg = this->getLocalSdp();
            }
          }
      }
      break;
    case TRANSPORT_READY:
      if (bundle_) {
        temp = CONN_READY;
        trackTransportInfo();
        forEachMediaStreamAsyncNoPromise([] (const std::shared_ptr<MediaStream> &media_stream) {
          media_stream->sendPLIToFeedback();
        });
      } else {
        if ((!remote_sdp_->hasAudio || (audio_transport_.get() != nullptr
                  && audio_transport_->getTransportState() == TRANSPORT_READY)) &&
            (!remote_sdp_->hasVideo || (video_transport_.get() != nullptr
                  && video_transport_->getTransportState() == TRANSPORT_READY))) {
            // WebRTCConnection will be ready only when all channels are ready.
            temp = CONN_READY;
            trackTransportInfo();
            forEachMediaStreamAsyncNoPromise([] (const std::shared_ptr<MediaStream> &media_stream) {
              media_stream->sendPLIToFeedback();
            });
          }
      }
      break;
    case TRANSPORT_FAILED:
      temp = CONN_FAILED;
      sending_ = false;
      msg = remote_sdp_->getSdp();
      ELOG_ERROR("%s message: Transport Failed, transportType: %s", toLog(), transport->transport_name.c_str() );
      cond_.notify_one();
      break;
    default:
      ELOG_DEBUG("%s message: Doing nothing on state, state %d", toLog(), state);
      break;
  }

  if (audio_transport_.get() != nullptr && video_transport_.get() != nullptr) {
    ELOG_DEBUG("%s message: %s, transportName: %s, videoTransportState: %d"
              ", audioTransportState: %d, calculatedState: %d, globalState: %d",
      toLog(),
      "Update Transport State",
      transport->transport_name.c_str(),
      static_cast<int>(audio_transport_->getTransportState()),
      static_cast<int>(video_transport_->getTransportState()),
      static_cast<int>(temp),
      static_cast<int>(global_state_));
  }

  if (global_state_ == temp) {
    return;
  }

  global_state_ = temp;

  ELOG_INFO("%s newGlobalState: %d", toLog(), temp);
  maybeNotifyWebRtcConnectionEvent(global_state_, msg);
}

// media_stream设置audio_info和video_info的candidate信息
void WebRtcConnection::trackTransportInfo() {
  CandidatePair candidate_pair;
  std::string audio_info;
  std::string video_info;
  if (video_enabled_ && video_transport_) {
    candidate_pair = video_transport_->getIceConnection()->getSelectedPair();
    video_info = candidate_pair.clientHostType;
  }

  if (audio_enabled_ && audio_transport_) {
    candidate_pair = audio_transport_->getIceConnection()->getSelectedPair();
    audio_info = candidate_pair.clientHostType;
  }

  asyncTask([audio_info, video_info] (std::shared_ptr<WebRtcConnection> connection) {
    connection->forEachMediaStreamAsyncNoPromise(
      [audio_info, video_info] (const std::shared_ptr<MediaStream> &media_stream) {
        media_stream->setTransportInfo(audio_info, video_info);
      });
  });
}

// 设置元数据
void WebRtcConnection::setMetadata(std::map<std::string, std::string> metadata) {
  setLogContext(metadata);
}

// 设置上层的回调函数
void WebRtcConnection::setWebRtcConnectionEventListener(WebRtcConnectionEventListener* listener) {
  boost::mutex::scoped_lock lock(event_listener_mutex_);
  this->conn_event_listener_ = listener;
}

// 获取当前WebRtcConnection的状态
WebRTCEvent WebRtcConnection::getCurrentState() {
  return global_state_;
}

// 异步写包功能
void WebRtcConnection::write(std::shared_ptr<DataPacket> packet) {
  asyncTask([packet] (std::shared_ptr<WebRtcConnection> connection) {
    connection->syncWrite(packet);
  });
}

// 向audio或video transport中写入包数据
void WebRtcConnection::syncWrite(std::shared_ptr<DataPacket> packet) {
  if (!sending_) {
    return;
  }
  Transport *transport = (bundle_ || packet->type == VIDEO_PACKET) ? video_transport_.get() : audio_transport_.get();
  if (transport == nullptr) {
    return;
  }
  this->extension_processor_.processRtpExtensions(packet);
  transport->write(packet->data, packet->length);
}

// 设置video_transport_,仅用于测试
void WebRtcConnection::setTransport(std::shared_ptr<Transport> transport) {  // Only for Testing purposes
  video_transport_ = std::move(transport);
  bundle_ = true;
}
Análisis de código fuente de WebRtcConnection (3)

Supongo que te gusta
