Wire up PacketTime to ReceiveStreams.

webrtc/video/rampup_tests.h

 /*
  *  Copyright (c) 2014 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 84 matching lines @@
                         size_t number_of_streams,
                         bool rtx_used);
 
   virtual void SetSendStream(VideoSendStream* send_stream);
 
   virtual void OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs,
                                        unsigned int bitrate);
 
   bool SendRtp(const uint8_t* data, size_t length) override;
 
-  DeliveryStatus DeliverPacket(MediaType media_type, const uint8_t* packet,
-                               size_t length) override;
+  DeliveryStatus DeliverPacket(MediaType media_type,
+                               const uint8_t* packet,
+                               size_t length,
+                               const PacketTime& packet_time) override;
 
   bool SendRtcp(const uint8_t* packet, size_t length) override;
 
   // Produces a string similar to "1stream_nortx", depending on the values of
   // number_of_streams_ and rtx_used_;
   std::string GetModifierString();
 
   // This method defines the state machine for the ramp up-down-up test.
   void EvolveTestState(unsigned int bitrate_bps);
 
@@ skipping 34 matching lines @@
   void RunRampUpTest(size_t num_streams,
                      unsigned int start_bitrate_bps,
                      const std::string& extension_type,
                      bool rtx,
                      bool red);
 
   void RunRampUpDownUpTest(size_t number_of_streams, bool rtx, bool red);
 };
 }  // namespace webrtc
 #endif  // WEBRTC_VIDEO_RAMPUP_TESTS_H_