Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.

webrtc/api/webrtcsession.cc

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 1151 matching lines @@
   }
   return webrtc::GetTrackIdBySsrc(remote_desc_->description(), ssrc, track_id);
 }
 
 std::string WebRtcSession::BadStateErrMsg(State state) {
   std::ostringstream desc;
   desc << "Called in wrong state: " << GetStateString(state);
   return desc.str();
 }
 
+void WebRtcSession::SetAudioPlayout(uint32_t ssrc, bool enable) {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (!voice_channel_) {
+    LOG(LS_ERROR) << "SetAudioPlayout: No audio channel exists.";
+    return;
+  }
+  if (!voice_channel_->SetOutputVolume(ssrc, enable ? 1 : 0)) {
+    // Allow that SetOutputVolume fail if |enable| is false but assert
+    // otherwise. This in the normal case when the underlying media channel has
+    // already been deleted.
+    ASSERT(enable == false);
+  }
+}
+
+void WebRtcSession::SetAudioSend(uint32_t ssrc,
+                                 bool enable,
+                                 const cricket::AudioOptions& options,
+                                 cricket::AudioSource* source) {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (!voice_channel_) {
+    LOG(LS_ERROR) << "SetAudioSend: No audio channel exists.";
+    return;
+  }
+  if (!voice_channel_->SetAudioSend(ssrc, enable, &options, source)) {
+    LOG(LS_ERROR) << "SetAudioSend: ssrc is incorrect: " << ssrc;
+  }
+}
+
+void WebRtcSession::SetAudioPlayoutVolume(uint32_t ssrc, double volume) {
+  ASSERT(signaling_thread()->IsCurrent());
+  ASSERT(volume >= 0 && volume <= 10);
+  if (!voice_channel_) {
+    LOG(LS_ERROR) << "SetAudioPlayoutVolume: No audio channel exists.";
+    return;
+  }
+
+  if (!voice_channel_->SetOutputVolume(ssrc, volume)) {
+    ASSERT(false);
+  }
+}
+
+void WebRtcSession::SetRawAudioSink(uint32_t ssrc,
+                                    std::unique_ptr<AudioSinkInterface> sink) {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (!voice_channel_)
+    return;
+
+  voice_channel_->SetRawAudioSink(ssrc, std::move(sink));
+}
+
+RtpParameters WebRtcSession::GetAudioRtpSendParameters(uint32_t ssrc) const {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (voice_channel_) {
+    return voice_channel_->GetRtpSendParameters(ssrc);
+  }
+  return RtpParameters();
+}
+
+bool WebRtcSession::SetAudioRtpSendParameters(uint32_t ssrc,
+                                              const RtpParameters& parameters) {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (!voice_channel_) {
+    return false;
+  }
+  return voice_channel_->SetRtpSendParameters(ssrc, parameters);
+}
+
+RtpParameters WebRtcSession::GetAudioRtpReceiveParameters(uint32_t ssrc) const {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (voice_channel_) {
+    return voice_channel_->GetRtpReceiveParameters(ssrc);
+  }
+  return RtpParameters();
+}
+
+bool WebRtcSession::SetAudioRtpReceiveParameters(
+    uint32_t ssrc,
+    const RtpParameters& parameters) {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (!voice_channel_) {
+    return false;
+  }
+  return voice_channel_->SetRtpReceiveParameters(ssrc, parameters);
+}
+
+void WebRtcSession::SetVideoPlayout(
+    uint32_t ssrc,
+    bool enable,
+    rtc::VideoSinkInterface<cricket::VideoFrame>* sink) {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (!video_channel_) {
+    LOG(LS_WARNING) << "SetVideoPlayout: No video channel exists.";
+    return;
+  }
+  if (!video_channel_->SetSink(ssrc, enable ? sink : NULL)) {
+    // Allow that SetSink fail if |sink| is NULL but assert otherwise.
+    // This in the normal case when the underlying media channel has already
+    // been deleted.
+    ASSERT(sink == NULL);
+  }
+}
+
+void WebRtcSession::SetVideoSend(
+    uint32_t ssrc,
+    bool enable,
+    const cricket::VideoOptions* options,
+    rtc::VideoSourceInterface<cricket::VideoFrame>* source) {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (!video_channel_) {
+    LOG(LS_WARNING) << "SetVideoSend: No video channel exists.";
+    return;
+  }
+  if (!video_channel_->SetVideoSend(ssrc, enable, options, source)) {
+    // Allow that MuteStream fail if |enable| is false and |source| is NULL but
+    // assert otherwise. This in the normal case when the underlying media
+    // channel has already been deleted.
+    ASSERT(enable == false && source == nullptr);
+  }
+}
+
+RtpParameters WebRtcSession::GetVideoRtpSendParameters(uint32_t ssrc) const {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (video_channel_) {
+    return video_channel_->GetRtpSendParameters(ssrc);
+  }
+  return RtpParameters();
+}
+
+bool WebRtcSession::SetVideoRtpSendParameters(uint32_t ssrc,
+                                              const RtpParameters& parameters) {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (!video_channel_) {
+    return false;
+  }
+  return video_channel_->SetRtpSendParameters(ssrc, parameters);
+}
+
+RtpParameters WebRtcSession::GetVideoRtpReceiveParameters(uint32_t ssrc) const {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (video_channel_) {
+    return video_channel_->GetRtpReceiveParameters(ssrc);
+  }
+  return RtpParameters();
+}
+
+bool WebRtcSession::SetVideoRtpReceiveParameters(
+    uint32_t ssrc,
+    const RtpParameters& parameters) {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (!video_channel_) {
+    return false;
+  }
+  return video_channel_->SetRtpReceiveParameters(ssrc, parameters);
+}
+
 bool WebRtcSession::CanInsertDtmf(const std::string& track_id) {
   ASSERT(signaling_thread()->IsCurrent());
   if (!voice_channel_) {
     LOG(LS_ERROR) << "CanInsertDtmf: No audio channel exists.";
     return false;
   }
   uint32_t send_ssrc = 0;
   // The Dtmf is negotiated per channel not ssrc, so we only check if the ssrc
   // exists.
   if (!local_desc_ ||
@@ skipping 423 matching lines @@
       bundle_transport, create_rtcp_transport_channel, audio_options_));
   if (!voice_channel_) {
     return false;
   }
   if (require_rtcp_mux) {
     voice_channel_->ActivateRtcpMux();
   }
 
   voice_channel_->SignalDtlsSetupFailure.connect(
       this, &WebRtcSession::OnDtlsSetupFailure);
+  voice_channel_->SignalFirstPacketReceived.connect(
+      this, &WebRtcSession::OnChannelFirstPacketReceived);
 
   SignalVoiceChannelCreated();
   voice_channel_->SignalSentPacket.connect(this,
                                            &WebRtcSession::OnSentPacket_w);
   return true;
 }
 
 bool WebRtcSession::CreateVideoChannel(const cricket::ContentInfo* content,
                                        const std::string* bundle_transport) {
   bool require_rtcp_mux =
       rtcp_mux_policy_ == PeerConnectionInterface::kRtcpMuxPolicyRequire;
   bool create_rtcp_transport_channel = !require_rtcp_mux;
   video_channel_.reset(channel_manager_->CreateVideoChannel(
       media_controller_, transport_controller_.get(), content->name,
       bundle_transport, create_rtcp_transport_channel, video_options_));
   if (!video_channel_) {
     return false;
   }
   if (require_rtcp_mux) {
     video_channel_->ActivateRtcpMux();
   }
   video_channel_->SignalDtlsSetupFailure.connect(
       this, &WebRtcSession::OnDtlsSetupFailure);
+  video_channel_->SignalFirstPacketReceived.connect(
+      this, &WebRtcSession::OnChannelFirstPacketReceived);
 
   SignalVideoChannelCreated();
   video_channel_->SignalSentPacket.connect(this,
                                            &WebRtcSession::OnSentPacket_w);
   return true;
 }
 
 bool WebRtcSession::CreateDataChannel(const cricket::ContentInfo* content,
                                       const std::string* bundle_transport) {
   bool sctp = (data_channel_type_ == cricket::DCT_SCTP);
@@ skipping 21 matching lines @@
   SignalDataChannelCreated();
   data_channel_->SignalSentPacket.connect(this, &WebRtcSession::OnSentPacket_w);
   return true;
 }
 
 void WebRtcSession::OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp) {
   SetError(ERROR_TRANSPORT,
            rtcp ? kDtlsSetupFailureRtcp : kDtlsSetupFailureRtp);
 }
 
+void WebRtcSession::OnChannelFirstPacketReceived(
+    cricket::BaseChannel* channel) {
+  ASSERT(signaling_thread()->IsCurrent());
+
+  if (!received_first_audio_packet_ &&
+      channel->media_type() == cricket::MEDIA_TYPE_AUDIO) {
+    received_first_audio_packet_ = true;
+    SignalFirstAudioPacketReceived();
+  } else if (!received_first_video_packet_ &&
+             channel->media_type() == cricket::MEDIA_TYPE_VIDEO) {
+    received_first_video_packet_ = true;
+    SignalFirstVideoPacketReceived();
+  }
+}
+
 void WebRtcSession::OnDataChannelMessageReceived(
     cricket::DataChannel* channel,
     const cricket::ReceiveDataParams& params,
     const rtc::CopyOnWriteBuffer& payload) {
   RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
   if (params.type == cricket::DMT_CONTROL && IsOpenMessage(payload)) {
     // Received OPEN message; parse and signal that a new data channel should
     // be created.
     std::string label;
     InternalDataChannelInit config;
@@ skipping 298 matching lines @@
                                                   ssl_cipher_suite);
   }
 }
 
 void WebRtcSession::OnSentPacket_w(const rtc::SentPacket& sent_packet) {
   RTC_DCHECK(worker_thread()->IsCurrent());
   media_controller_->call_w()->OnSentPacket(sent_packet);
 }
 
 }  // namespace webrtc