Delete Timing class, timing.h, and update all users.

webrtc/media/base/rtpdataengine.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_MEDIA_BASE_RTPDATAENGINE_H_
 #define WEBRTC_MEDIA_BASE_RTPDATAENGINE_H_
 
 #include <memory>
 #include <string>
 #include <vector>
 
-#include "webrtc/base/timing.h"
 #include "webrtc/media/base/mediachannel.h"
 #include "webrtc/media/base/mediaconstants.h"
 #include "webrtc/media/base/mediaengine.h"
 
 namespace cricket {
 
 struct DataCodec;
 
 class RtpDataEngine : public DataEngineInterface {
  public:
   RtpDataEngine();
 
   virtual DataMediaChannel* CreateChannel(DataChannelType data_channel_type);
 
   virtual const std::vector<DataCodec>& data_codecs() {
     return data_codecs_;
   }
 
-  // Mostly for testing with a fake clock.  Ownership is passed in.
-  void SetTiming(rtc::Timing* timing) {
-    timing_.reset(timing);
-  }
-
  private:
   std::vector<DataCodec> data_codecs_;
-  std::unique_ptr<rtc::Timing> timing_;
 };
 
 // Keep track of sequence number and timestamp of an RTP stream.  The
 // sequence number starts with a "random" value and increments.  The
 // timestamp starts with a "random" value and increases monotonically
 // according to the clockrate.
 class RtpClock {
  public:
   RtpClock(int clockrate, uint16_t first_seq_num, uint32_t timestamp_offset)
       : clockrate_(clockrate),
         last_seq_num_(first_seq_num),
         timestamp_offset_(timestamp_offset) {}
 
   // Given the current time (in number of seconds which must be
   // monotonically increasing), Return the next sequence number and
   // timestamp.
   void Tick(double now, int* seq_num, uint32_t* timestamp);
 
  private:
   int clockrate_;
   uint16_t last_seq_num_;
   uint32_t timestamp_offset_;
 };
 
 class RtpDataMediaChannel : public DataMediaChannel {
  public:
-  // Timing* Used for the RtpClock
-  explicit RtpDataMediaChannel(rtc::Timing* timing);
-  // Sets Timing == NULL, so you'll need to call set_timer() before
-  // using it.  This is needed by FakeMediaEngine.
   RtpDataMediaChannel();
   virtual ~RtpDataMediaChannel();
 
-  void set_timing(rtc::Timing* timing) {
-    timing_ = timing;
-  }
-
   virtual bool SetSendParameters(const DataSendParameters& params);
   virtual bool SetRecvParameters(const DataRecvParameters& params);
   virtual bool AddSendStream(const StreamParams& sp);
   virtual bool RemoveSendStream(uint32_t ssrc);
   virtual bool AddRecvStream(const StreamParams& sp);
   virtual bool RemoveRecvStream(uint32_t ssrc);
   virtual bool SetSend(bool send) {
     sending_ = send;
     return true;
   }
   virtual bool SetReceive(bool receive) {
     receiving_ = receive;
     return true;
   }
   virtual void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
                                 const rtc::PacketTime& packet_time);
   virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
                               const rtc::PacketTime& packet_time) {}
   virtual void OnReadyToSend(bool ready) {}
   virtual bool SendData(
     const SendDataParams& params,
     const rtc::CopyOnWriteBuffer& payload,
     SendDataResult* result);
 
  private:
-  void Construct(rtc::Timing* timing);
+  void Construct();
   bool SetMaxSendBandwidth(int bps);
   bool SetSendCodecs(const std::vector<DataCodec>& codecs);
   bool SetRecvCodecs(const std::vector<DataCodec>& codecs);
 
   bool sending_;
   bool receiving_;
-  rtc::Timing* timing_;
   std::vector<DataCodec> send_codecs_;
   std::vector<DataCodec> recv_codecs_;
   std::vector<StreamParams> send_streams_;
   std::vector<StreamParams> recv_streams_;
   std::map<uint32_t, RtpClock*> rtp_clock_by_send_ssrc_;
   std::unique_ptr<rtc::RateLimiter> send_limiter_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_RTPDATAENGINE_H_