Wire up send-side bandwidth estimation.

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.
  */
 
 #ifndef WEBRTC_VIDEO_RAMPUP_TESTS_H_
 #define WEBRTC_VIDEO_RAMPUP_TESTS_H_
 
 #include <map>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/call.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/system_wrappers/interface/event_wrapper.h"
 #include "webrtc/test/call_test.h"
 #include "webrtc/video/transport_adapter.h"
 
 namespace webrtc {
 
 static const int kTransmissionTimeOffsetExtensionId = 6;
 static const int kAbsSendTimeExtensionId = 7;
+static const int kTransportSequenceNumberExtensionId = 8;
 static const unsigned int kSingleStreamTargetBps = 1000000;
 
 class Clock;
+class PacketRouter;
 class ReceiveStatistics;
 class RtpHeaderParser;
 class RTPPayloadRegistry;
 class RtpRtcp;
 
 class StreamObserver : public newapi::Transport, public RemoteBitrateObserver {
  public:
   typedef std::map<uint32_t, int> BytesSentMap;
   typedef std::map<uint32_t, uint32_t> SsrcMap;
   StreamObserver(const SsrcMap& rtx_media_ssrcs,
                  newapi::Transport* feedback_transport,
                  Clock* clock);
+  virtual ~StreamObserver();
 
   void set_expected_bitrate_bps(unsigned int expected_bitrate_bps);
 
   void set_start_bitrate_bps(unsigned int start_bitrate_bps);
 
   void OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs,
                                unsigned int bitrate) override;
 
   bool SendRtp(const uint8_t* packet, size_t length) override;
 
   bool SendRtcp(const uint8_t* packet, size_t length) override;
 
   EventTypeWrapper Wait();
 
   void SetRemoteBitrateEstimator(RemoteBitrateEstimator* rbe);
 
+  PacketRouter* GetPacketRouter();
+
  private:
   void ReportResult(const std::string& measurement,
                     size_t value,
                     const std::string& units);
   void TriggerTestDone() EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
   Clock* const clock_;
   const rtc::scoped_ptr<EventWrapper> test_done_;
   const rtc::scoped_ptr<RtpHeaderParser> rtp_parser_;
   rtc::scoped_ptr<RtpRtcp> rtp_rtcp_;
+  rtc::scoped_ptr<PacketRouter> packet_router_;
   internal::TransportAdapter feedback_transport_;
   const rtc::scoped_ptr<ReceiveStatistics> receive_stats_;
   const rtc::scoped_ptr<RTPPayloadRegistry> payload_registry_;
   rtc::scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
 
   rtc::CriticalSection crit_;
   unsigned int expected_bitrate_bps_ GUARDED_BY(crit_);
   unsigned int start_bitrate_bps_ GUARDED_BY(crit_);
   SsrcMap rtx_media_ssrcs_ GUARDED_BY(crit_);
   size_t total_sent_ GUARDED_BY(crit_);
@@ skipping 73 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_