Unify Transport and newapi::Transport 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.
  */
 
@@ skipping 360 matching lines @@
                                 uint32_t rate) override;
     void OnIncomingSSRCChanged(uint32_t ssrc) override;
     void OnIncomingCSRCChanged(uint32_t CSRC, bool added) override;
 
     // From RtpAudioFeedback in the RTP/RTCP module
     void OnPlayTelephoneEvent(uint8_t event,
                               uint16_t lengthMs,
                               uint8_t volume) override;
 
     // From Transport (called by the RTP/RTCP module)
-    int SendPacket(const void* data, size_t len) override;
-    int SendRTCPPacket(const void* data, size_t len) override;
+    bool SendRtp(const uint8_t* data, size_t len) override;
+    bool SendRtcp(const uint8_t* data, size_t len) override;
 
     // From MixerParticipant
     int32_t GetAudioFrame(int32_t id, AudioFrame* audioFrame) override;
     int32_t NeededFrequency(int32_t id) const override;
 
     // From FileCallback
     void PlayNotification(int32_t id, uint32_t durationMs) override;
     void RecordNotification(int32_t id, uint32_t durationMs) override;
     void PlayFileEnded(int32_t id) override;
     void RecordFileEnded(int32_t id) override;
@@ skipping 65 matching lines @@
                        const RTPHeader& header, bool in_order);
     bool HandleRtxPacket(const uint8_t* packet,
                          size_t packet_length,
                          const RTPHeader& header);
     bool IsPacketInOrder(const RTPHeader& header) const;
     bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const;
     int ResendPackets(const uint16_t* sequence_numbers, int length);
     int InsertInbandDtmfTone();
     int32_t MixOrReplaceAudioWithFile(int mixingFrequency);
     int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency);
-    int32_t SendPacketRaw(const void *data, size_t len, bool RTCP);
     void UpdatePlayoutTimestamp(bool rtcp);
     void UpdatePacketDelay(uint32_t timestamp,
                            uint16_t sequenceNumber);
     void RegisterReceiveCodecsToRTPModule();
 
     int SetRedPayloadType(int red_payload_type);
     int SetSendRtpHeaderExtension(bool enable, RTPExtensionType type,
                                   unsigned char id);
 
     int32_t GetPlayoutFrequency();
@@ skipping 103 matching lines @@
     rtc::scoped_ptr<NetworkPredictor> network_predictor_;
     // An associated send channel.
     rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_;
     ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_