webrtc-m79-peerconnection_client-创建answer

 PeerConnection::CreateAnswer 

// Conductor 的继承关系 ///
class Conductor : public webrtc::PeerConnectionObserver,
                  public webrtc::CreateSessionDescriptionObserver,
                  public PeerConnectionClientObserver,
                  public MainWndCallback
                  
void PeerConnection::CreateAnswer(CreateSessionDescriptionObserver* observer,  observer 就是 Conductor 的指针
                                  const RTCOfferAnswerOptions& options) {
  RTC_DCHECK_RUN_ON(signaling_thread());
  // Chain this operation. If asynchronous operations are pending on the chain,
  // this operation will be queued to be invoked, otherwise the contents of the
  // lambda will execute immediately.
  operations_chain_->ChainOperation(
      [this_weak_ptr = weak_ptr_factory_.GetWeakPtr(), / weak_ptr_factory_ 的初始化是在 PeerConnection 的构造函数中 weak_ptr_factory_(this)
       observer_refptr =                                                                                                     这里的 observer 就是 Conductor 的 this 指针
           rtc::scoped_refptr<CreateSessionDescriptionObserver>(observer), ///将 Conductor 的指针包裹到 CreateSessionDescriptionObserver 类型的 observer_refptr 中
       options](std::function<void()> operations_chain_callback) {
        // Abort early if |this_weak_ptr| is no longer valid.
        if (!this_weak_ptr) {
          observer_refptr->OnFailure(RTCError(
              RTCErrorType::INTERNAL_ERROR,
              "CreateAnswer failed because the session was shut down"));
          operations_chain_callback();
          return;
        }
        // The operation completes asynchronously when the wrapper is invoked.
        rtc::scoped_refptr<CreateSessionDescriptionObserverOperationWrapper>
            observer_wrapper(new rtc::RefCountedObject<
                             CreateSessionDescriptionObserverOperationWrapper>(
                std::move(observer_refptr),  又将 observer_refptr 包裹进 CreateSessionDescriptionObserverOperationWrapper 类型的 observer_wrapper
                std::move(operations_chain_callback)));
        this_weak_ptr->DoCreateAnswer(options, observer_wrapper); /// PeerConnection::DoCreateAnswer 
      });
}


void PeerConnection::DoCreateAnswer(
    const RTCOfferAnswerOptions& options,
    rtc::scoped_refptr<CreateSessionDescriptionObserver> observer) { /// observer 就是 CreateSessionDescriptionObserverOperationWrapper
  RTC_DCHECK_RUN_ON(signaling_thread());
  TRACE_EVENT0("webrtc", "PeerConnection::DoCreateAnswer");
  if (!observer) {
    RTC_LOG(LS_ERROR) << "CreateAnswer - observer is NULL.";
    return;
  }

  // If a session error has occurred the PeerConnection is in a possibly
  // inconsistent state so fail right away.
  if (session_error() != SessionError::kNone) {
    std::string error_message = GetSessionErrorMsg();
    RTC_LOG(LS_ERROR) << "CreateAnswer: " << error_message;
    PostCreateSessionDescriptionFailure(
        observer,
        RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
    return;
  }

  if (!(signaling_state_ == kHaveRemoteOffer ||
        signaling_state_ == kHaveLocalPrAnswer)) {
    std::string error =
        "PeerConnection cannot create an answer in a state other than "
        "have-remote-offer or have-local-pranswer.";
    RTC_LOG(LS_ERROR) << error;
    PostCreateSessionDescriptionFailure(
        observer, RTCError(RTCErrorType::INVALID_STATE, std::move(error)));
    return;
  }

  // The remote description should be set if we're in the right state.
  RTC_DCHECK(remote_description());

  if (IsUnifiedPlan()) {
    if (options.offer_to_receive_audio != RTCOfferAnswerOptions::kUndefined) {
      RTC_LOG(LS_WARNING) << "CreateAnswer: offer_to_receive_audio is not "
                             "supported with Unified Plan semantics. Use the "
                             "RtpTransceiver API instead.";
    }
    if (options.offer_to_receive_video != RTCOfferAnswerOptions::kUndefined) {
      RTC_LOG(LS_WARNING) << "CreateAnswer: offer_to_receive_video is not "
                             "supported with Unified Plan semantics. Use the "
                             "RtpTransceiver API instead.";
    }
  }

  cricket::MediaSessionOptions session_options;
  GetOptionsForAnswer(options, &session_options);

  webrtc_session_desc_factory_->CreateAnswer(observer, session_options); // webrtc_session_desc_factory_ 就是 WebRtcSessionDescriptionFactory 具体见前面分析
}  所以这里就是 WebRtcSessionDescriptionFactory::CreateAnswer


void WebRtcSessionDescriptionFactory::CreateAnswer(
    CreateSessionDescriptionObserver* observer, 
    const cricket::MediaSessionOptions& session_options) {
  std::string error = "CreateAnswer";
  if (certificate_request_state_ == CERTIFICATE_FAILED) {
    error += kFailedDueToIdentityFailed;
    RTC_LOG(LS_ERROR) << error;
    PostCreateSessionDescriptionFailed(observer, error);
    return;
  }
  if (!pc_->remote_description()) {
    error += " can't be called before SetRemoteDescription.";
    RTC_LOG(LS_ERROR) << error;
    PostCreateSessionDescriptionFailed(observer, error);
    return;
  }
  if (pc_->remote_description()->GetType() != SdpType::kOffer) {
    error += " failed because remote_description is not an offer.";
    RTC_LOG(LS_ERROR) << error;
    PostCreateSessionDescriptionFailed(observer, error);
    return;
  }

  if (!ValidMediaSessionOptions(session_options)) {
    error += " called with invalid session options.";
    RTC_LOG(LS_ERROR) << error;
    PostCreateSessionDescriptionFailed(observer, error);
    return;
  }

  CreateSessionDescriptionRequest request(
      CreateSessionDescriptionRequest::kAnswer, observer, session_options); / 传递 observer 到 request 中
  if (certificate_request_state_ == CERTIFICATE_WAITING) {
    create_session_description_requests_.push(request);
  } else {
    RTC_DCHECK(certificate_request_state_ == CERTIFICATE_SUCCEEDED ||
               certificate_request_state_ == CERTIFICATE_NOT_NEEDED);
    InternalCreateAnswer(request);  
  }
}

void WebRtcSessionDescriptionFactory::InternalCreateAnswer(
    CreateSessionDescriptionRequest request) {
  if (pc_->remote_description()) {
    for (cricket::MediaDescriptionOptions& options :
         request.options.media_description_options) {
      // According to http://tools.ietf.org/html/rfc5245#section-9.2.1.1
      // an answer should also contain new ICE ufrag and password if an offer
      // has been received with new ufrag and password.
      options.transport_options.ice_restart =
          pc_->IceRestartPending(options.mid);
      // We should pass the current SSL role to the transport description
      // factory, if there is already an existing ongoing session.
      rtc::SSLRole ssl_role;
      if (pc_->GetSslRole(options.mid, &ssl_role)) {
        options.transport_options.prefer_passive_role =
            (rtc::SSL_SERVER == ssl_role);
      }
    }
  }

  std::unique_ptr<cricket::SessionDescription> desc =
      session_desc_factory_.CreateAnswer( // session_desc_factory_ 在 WebRtcSessionDescriptionFactory 构造函数中初始化
          pc_->remote_description() ? pc_->remote_description()->description() / 就是 MediaSessionDescriptionFactory::CreateAnswer
                                    : nullptr,
          request.options,
          pc_->local_description() ? pc_->local_description()->description()
                                   : nullptr);
  if (!desc) {
    PostCreateSessionDescriptionFailed(request.observer,
                                       "Failed to initialize the answer.");
    return;
  }

  // RFC 3264
  // If the answer is different from the offer in any way (different IP
  // addresses, ports, etc.), the origin line MUST be different in the answer.
  // In that case, the version number in the "o=" line of the answer is
  // unrelated to the version number in the o line of the offer.
  // Get a new version number by increasing the |session_version_answer_|.
  // The |session_version_| is a uint64_t, the wrap around should not happen.
  RTC_DCHECK(session_version_ + 1 > session_version_);
  auto answer = std::make_unique<JsepSessionDescription>(
      SdpType::kAnswer, std::move(desc), session_id_,
      rtc::ToString(session_version_++)); / 
  if (pc_->local_description()) {
    // Include all local ICE candidates in the SessionDescription unless
    // the remote peer has requested an ICE restart.
    for (const cricket::MediaDescriptionOptions& options :
         request.options.media_description_options) {
      if (!options.transport_options.ice_restart) {
        CopyCandidatesFromSessionDescription(pc_->local_description(),
                                             options.mid, answer.get());
      }
    }
  }
  PostCreateSessionDescriptionSucceeded(request.observer, std::move(answer)); / request.observer 就是 CreateSessionDescriptionObserverOperationWrapper
}  PostCreateSessionDescriptionSucceeded 见下面分析


  MediaSessionDescriptionFactory::CreateAnswer

std::unique_ptr<SessionDescription>
MediaSessionDescriptionFactory::CreateAnswer(
    const SessionDescription* offer,
    const MediaSessionOptions& session_options,
    const SessionDescription* current_description) const {
  if (!offer) {
    return nullptr;
  }

  // Must have options for exactly as many sections as in the offer.
  RTC_DCHECK_EQ(offer->contents().size(),
                session_options.media_description_options.size());

  IceCredentialsIterator ice_credentials(
      session_options.pooled_ice_credentials);

  std::vector<const ContentInfo*> current_active_contents;
  if (current_description) {
    current_active_contents =
        GetActiveContents(*current_description, session_options);
  }

  StreamParamsVec current_streams =
      GetCurrentStreamParams(current_active_contents);

  // Get list of all possible codecs that respects existing payload type
  // mappings and uses a single payload type space.
  //
  // Note that these lists may be further filtered for each m= section; this
  // step is done just to establish the payload type mappings shared by all
  // sections.
  AudioCodecs answer_audio_codecs;
  VideoCodecs answer_video_codecs;
  RtpDataCodecs answer_rtp_data_codecs;
  GetCodecsForAnswer(current_active_contents, *offer, &answer_audio_codecs,
                     &answer_video_codecs, &answer_rtp_data_codecs);

  if (!session_options.vad_enabled) {
    // If application doesn't want CN codecs in answer.
    StripCNCodecs(&answer_audio_codecs);
  }

  auto answer = std::make_unique<SessionDescription>();

  // If the offer supports BUNDLE, and we want to use it too, create a BUNDLE
  // group in the answer with the appropriate content names.
  const ContentGroup* offer_bundle = offer->GetGroupByName(GROUP_TYPE_BUNDLE);
  ContentGroup answer_bundle(GROUP_TYPE_BUNDLE);
  // Transport info shared by the bundle group.
  std::unique_ptr<TransportInfo> bundle_transport;

  answer->set_extmap_allow_mixed(offer->extmap_allow_mixed());

  // Iterate through the media description options, matching with existing
  // media descriptions in |current_description|.
  size_t msection_index = 0;
  for (const MediaDescriptionOptions& media_description_options :
       session_options.media_description_options) {
    const ContentInfo* offer_content = &offer->contents()[msection_index];
    // Media types and MIDs must match between the remote offer and the
    // MediaDescriptionOptions.
    RTC_DCHECK(
        IsMediaContentOfType(offer_content, media_description_options.type));
    RTC_DCHECK(media_description_options.mid == offer_content->name);
    const ContentInfo* current_content = nullptr;
    if (current_description &&
        msection_index < current_description->contents().size()) {
      current_content = &current_description->contents()[msection_index];
    }
    RtpHeaderExtensions header_extensions = RtpHeaderExtensionsFromCapabilities(
        UnstoppedRtpHeaderExtensionCapabilities(
            media_description_options.header_extensions));
    switch (media_description_options.type) {
      case MEDIA_TYPE_AUDIO:
        if (!AddAudioContentForAnswer(
                media_description_options, session_options, offer_content,
                offer, current_content, current_description,
                bundle_transport.get(), answer_audio_codecs, header_extensions,
                &current_streams, answer.get(), &ice_credentials)) {
          return nullptr;
        }
        break;
      case MEDIA_TYPE_VIDEO:
        if (!AddVideoContentForAnswer(
                media_description_options, session_options, offer_content,
                offer, current_content, current_description,
                bundle_transport.get(), answer_video_codecs, header_extensions,
                &current_streams, answer.get(), &ice_credentials)) {
          return nullptr;
        }
        break;
      case MEDIA_TYPE_DATA:
        if (!AddDataContentForAnswer(
                media_description_options, session_options, offer_content,
                offer, current_content, current_description,
                bundle_transport.get(), answer_rtp_data_codecs,
                &current_streams, answer.get(), &ice_credentials)) {
          return nullptr;
        }
        break;
      default:
        RTC_NOTREACHED();
    }
    ++msection_index;
    // See if we can add the newly generated m= section to the BUNDLE group in
    // the answer.
    ContentInfo& added = answer->contents().back();
    if (!added.rejected && session_options.bundle_enabled && offer_bundle &&
        offer_bundle->HasContentName(added.name)) {
      answer_bundle.AddContentName(added.name);
      bundle_transport.reset(
          new TransportInfo(*answer->GetTransportInfoByName(added.name)));
    }
  }

  // If a BUNDLE group was offered, put a BUNDLE group in the answer even if
  // it's empty. RFC5888 says:
  //
  //   A SIP entity that receives an offer that contains an "a=group" line
  //   with semantics that are understood MUST return an answer that
  //   contains an "a=group" line with the same semantics.
  if (offer_bundle) {
    answer->AddGroup(answer_bundle);
  }

  if (answer_bundle.FirstContentName()) {
    // Share the same ICE credentials and crypto params across all contents,
    // as BUNDLE requires.
    if (!UpdateTransportInfoForBundle(answer_bundle, answer.get())) {
      RTC_LOG(LS_ERROR)
          << "CreateAnswer failed to UpdateTransportInfoForBundle.";
      return NULL;
    }

    if (!UpdateCryptoParamsForBundle(answer_bundle, answer.get())) {
      RTC_LOG(LS_ERROR)
          << "CreateAnswer failed to UpdateCryptoParamsForBundle.";
      return NULL;
    }
  }

  // The following determines how to signal MSIDs to ensure compatibility with
  // older endpoints (in particular, older Plan B endpoints).
  if (is_unified_plan_) {
    // Unified Plan needs to look at what the offer included to find the most
    // compatible answer.
    if (offer->msid_signaling() == 0) {
      // We end up here in one of three cases:
      // 1. An empty offer. We'll reply with an empty answer so it doesn't
      //    matter what we pick here.
      // 2. A data channel only offer. We won't add any MSIDs to the answer so
      //    it also doesn't matter what we pick here.
      // 3. Media that's either sendonly or inactive from the remote endpoint.
      //    We don't have any information to say whether the endpoint is Plan B
      //    or Unified Plan, so be conservative and send both.
      answer->set_msid_signaling(cricket::kMsidSignalingMediaSection |
                                 cricket::kMsidSignalingSsrcAttribute);
    } else if (offer->msid_signaling() ==
               (cricket::kMsidSignalingMediaSection |
                cricket::kMsidSignalingSsrcAttribute)) {
      // If both a=msid and a=ssrc MSID signaling methods were used, we're
      // probably talking to a Unified Plan endpoint so respond with just
      // a=msid.
      answer->set_msid_signaling(cricket::kMsidSignalingMediaSection);
    } else {
      // Otherwise, it's clear which method the offerer is using so repeat that
      // back to them.
      answer->set_msid_signaling(offer->msid_signaling());
    }
  } else {
    // Plan B always signals MSID using a=ssrc lines.
    answer->set_msid_signaling(cricket::kMsidSignalingSsrcAttribute);
  }

  return answer; / 
}


/ 将创建好的 answer 发送到信令服务器，再由信令服务器转发到对端 


void WebRtcSessionDescriptionFactory::PostCreateSessionDescriptionSucceeded(
    CreateSessionDescriptionObserver* observer,  CreateSessionDescriptionObserverOperationWrapper 的父类是 CreateSessionDescriptionObserver 
    std::unique_ptr<SessionDescriptionInterface> description) { 这里的 observer 就是 CreateSessionDescriptionObserverOperationWrapper 的指针
  CreateSessionDescriptionMsg* msg =
      new CreateSessionDescriptionMsg(observer, RTCError::OK()); ///这里的 observer 就是 CreateSessionDescriptionObserverOperationWrapper 的指针
  msg->description = std::move(description); /// 重新包裹 SDP
  signaling_thread_->Post(RTC_FROM_HERE, this, // 在信号线程进行发送，会在信号线程里面执行 WebRtcSessionDescriptionFactory::OnMessage
                          MSG_CREATE_SESSIONDESCRIPTION_SUCCESS, msg);
}

void WebRtcSessionDescriptionFactory::OnMessage(rtc::Message* msg) {
  switch (msg->message_id) {
    case MSG_CREATE_SESSIONDESCRIPTION_SUCCESS: {
      CreateSessionDescriptionMsg* param =
          static_cast<CreateSessionDescriptionMsg*>(msg->pdata);
      param->observer->OnSuccess(param->description.release()); / 就是 CreateSessionDescriptionObserverOperationWrapper::OnSucess
      delete param;      // 不过 OnSuccess 是在 CreateSessionDescriptionObserverOperationWrapper 的声明中定义的
      break;
    }
    case MSG_CREATE_SESSIONDESCRIPTION_FAILED: {
      CreateSessionDescriptionMsg* param =
          static_cast<CreateSessionDescriptionMsg*>(msg->pdata);
      param->observer->OnFailure(std::move(param->error));
      delete param;
      break;
    }
    case MSG_USE_CONSTRUCTOR_CERTIFICATE: {
      rtc::ScopedRefMessageData<rtc::RTCCertificate>* param =
          static_cast<rtc::ScopedRefMessageData<rtc::RTCCertificate>*>(
              msg->pdata);
      RTC_LOG(LS_INFO) << "Using certificate supplied to the constructor.";
      SetCertificate(param->data());
      delete param;
      break;
    }
    default:
      RTC_NOTREACHED();
      break;
  }
}


/// 下面的 OnSucess 就是在 CreateSessionDescriptionObserverOperationWrapper 声明中定义的 OnSucess
  void OnSuccess(SessionDescriptionInterface* desc) override {
    RTC_DCHECK(!was_called_);
#ifdef RTC_DCHECK_IS_ON
    was_called_ = true;
#endif  // RTC_DCHECK_IS_ON
    // Completing the operation before invoking the observer allows the observer
    // to execute SetLocalDescription() without delay.
    operation_complete_callback_(); // 实际上就是 OperationsChain::CallbackHandle::OnOperationComplete()
    observer_->OnSuccess(desc); /// 这里的 observer_ 就是包含了 Conductor 指针的 CreateSessionDescriptionObserver ，所以也就是调用 Conductor::OnSuccess
  }

void Conductor::OnSuccess(webrtc::SessionDescriptionInterface* desc) {
  peer_connection_->SetLocalDescription(  实际上就是 void PeerConnection::SetLocalDescription(SetSessionDescriptionObserver* observer,SessionDescriptionInterface* desc_ptr)
      DummySetSessionDescriptionObserver::Create(), desc); / 这里会触发 dtls 的过程，具体参考下方

  std::string sdp;
  desc->ToString(&sdp);

  // For loopback test. To save some connecting delay.
  if (loopback_) {
    // Replace message type from "offer" to "answer"
    std::unique_ptr<webrtc::SessionDescriptionInterface> session_description =
        webrtc::CreateSessionDescription(webrtc::SdpType::kAnswer, sdp);
    peer_connection_->SetRemoteDescription(
        DummySetSessionDescriptionObserver::Create(),
        session_description.release());
    return;
  }

  Json::StyledWriter writer;
  Json::Value jmessage;
  jmessage[kSessionDescriptionTypeName] =
      webrtc::SdpTypeToString(desc->GetType()); / 填写 sdp 类型，这里就是 answer （与前面Conductor::OnSuccess中的分析要区别开，上面是发送offer，而这里是接收到offer后，向对端回复answer）
  jmessage[kSessionDescriptionSdpName] = sdp;
  SendMessage(writer.write(jmessage));  最后会通过 PeerConnectionClient::SendToPeer 发送出去，实际上就是通过 control_socket_ 
}