Unify Transport and newapi::Transport interfaces.

talk/media/webrtc/webrtcvoiceengine.h

 /*
  * libjingle
  * Copyright 2004 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 232 matching lines @@
   void OnRtcpReceived(rtc::Buffer* packet,
                       const rtc::PacketTime& packet_time) override;
   void OnReadyToSend(bool ready) override {}
   bool GetStats(VoiceMediaInfo* info) override;
   // Gets last reported error from WebRtc voice engine.  This should be only
   // called in response a failure.
   void GetLastMediaError(uint32* ssrc,
                          VoiceMediaChannel::Error* error) override;
 
   // implements Transport interface
-  int SendPacket(const void* data, size_t len) override {
+  bool SendRtp(const uint8_t* data, size_t len) override {
     rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
                        kMaxRtpPacketLen);
-    return VoiceMediaChannel::SendPacket(&packet) ? static_cast<int>(len) : -1;
+    return VoiceMediaChannel::SendPacket(&packet);
   }
 
-  int SendRTCPPacket(const void* data, size_t len) override {
+  bool SendRtcp(const uint8_t* data, size_t len) override {
     rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
                        kMaxRtpPacketLen);
-    return VoiceMediaChannel::SendRtcp(&packet) ? static_cast<int>(len) : -1;
+    return VoiceMediaChannel::SendRtcp(&packet);
   }
 
   bool FindSsrc(int channel_num, uint32* ssrc);
   void OnError(uint32 ssrc, int error);
 
   bool sending() const { return send_ != SEND_NOTHING; }
   int GetReceiveChannelNum(uint32 ssrc) const;
   int GetSendChannelNum(uint32 ssrc) const;
 
  private:
@@ skipping 99 matching lines @@
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
   // Do not lock this on the VoE media processor thread; potential for deadlock
   // exists.
   mutable rtc::CriticalSection receive_channels_cs_;
 };
 
 }  // namespace cricket
 
 #endif  // TALK_MEDIA_WEBRTCVOICEENGINE_H_