Initial implementation of RtpTransportControllerReceive and related interfaces.

webrtc/voice_engine/channel.h

 /*
  *  Copyright (c) 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.
  */
 
 #ifndef WEBRTC_VOICE_ENGINE_CHANNEL_H_
 #define WEBRTC_VOICE_ENGINE_CHANNEL_H_
 
 #include <memory>
 
 #include "webrtc/api/audio/audio_mixer.h"
 #include "webrtc/api/call/audio_sink.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/optional.h"
+#include "webrtc/call/rtp_transport_controller_receive.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/acm2/codec_manager.h"
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
 #include "webrtc/modules/audio_processing/rms_level.h"
 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
@@ skipping 91 matching lines @@
   }
 
  private:
   rtc::CriticalSection lock_;
   State state_;
 };
 
 class Channel
     : public RtpData,
       public RtpFeedback,
+      public RtpPacketSinkInterface,
       public FileCallback,  // receiving notification from file player &
                             // recorder
       public Transport,
       public AudioPacketizationCallback,  // receive encoded packets from the
                                           // ACM
       public MixerParticipant,  // supplies output mixer with audio frames
       public OverheadObserver {
  public:
   friend class VoERtcpObserver;
 
@@ skipping 55 matching lines @@
                                    int max_frame_length_ms);
 
   // VoENetwork
   int32_t RegisterExternalTransport(Transport* transport);
   int32_t DeRegisterExternalTransport();
   int32_t ReceivedRTPPacket(const uint8_t* received_packet,
                             size_t length,
                             const PacketTime& packet_time);
   // TODO(nisse, solenberg): Delete when VoENetwork is deleted.
   int32_t ReceivedRTCPPacket(const uint8_t* data, size_t length);
-  void OnRtpPacket(const RtpPacketReceived& packet);
+
+  // RtpPacketSinkInterface implementation.
+  void OnRtpPacket(const RtpPacketReceived& packet) override;
 
   // VoEFile
   int StartPlayingFileLocally(const char* fileName,
                               bool loop,
                               FileFormats format,
                               int startPosition,
                               float volumeScaling,
                               int stopPosition,
                               const CodecInst* codecInst);
   int StartPlayingFileLocally(InStream* stream,
@@ skipping 44 matching lines @@
 
   // DTMF.
   int SendTelephoneEventOutband(int event, int duration_ms);
   int SetSendTelephoneEventPayloadType(int payload_type, int payload_frequency);
 
   // VoERTP_RTCP
   int SetLocalSSRC(unsigned int ssrc);
   int GetLocalSSRC(unsigned int& ssrc);
   int GetRemoteSSRC(unsigned int& ssrc);
   int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id);
-  int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id);
   void EnableSendTransportSequenceNumber(int id);
-  void EnableReceiveTransportSequenceNumber(int id);
 
   void RegisterSenderCongestionControlObjects(
       RtpPacketSender* rtp_packet_sender,
       TransportFeedbackObserver* transport_feedback_observer,
       PacketRouter* packet_router,
       RtcpBandwidthObserver* bandwidth_observer);
   void RegisterReceiverCongestionControlObjects(PacketRouter* packet_router);
   void ResetCongestionControlObjects();
 
   void SetRTCPStatus(bool enable);
@@ skipping 219 matching lines @@
   std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
 
   // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed.
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_