More Simulation Framework features

webrtc/modules/remote_bitrate_estimator/test/packet_sender.h

 /*
  *  Copyright (c) 2015 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 22 matching lines @@
         // clock_, the clock of the PacketSender and the Source must be aligned.
         // We assume that both start at time 0.
         clock_(0) {}
   virtual ~PacketSender() {}
   // Call GiveFeedback() with the returned interval in milliseconds, provided
   // there is a new estimate available.
   // Note that changing the feedback interval affects the timing of when the
   // output of the estimators is sampled and therefore the baseline files may
   // have to be regenerated.
   virtual int GetFeedbackIntervalMs() const = 0;
+  virtual void set_choke_filter(ChokeFilter* choke_filter) {}
   void SetSenderTimestamps(Packets* in_out);
 
  protected:
   SimulatedClock clock_;
 };
 
 class VideoSender : public PacketSender, public BitrateObserver {
  public:
   VideoSender(PacketProcessorListener* listener,
               VideoSource* source,
               BandwidthEstimatorType estimator);
   virtual ~VideoSender();
 
   int GetFeedbackIntervalMs() const override;
   void RunFor(int64_t time_ms, Packets* in_out) override;
 
   virtual VideoSource* source() const { return source_; }
 
   // Implements BitrateObserver.
   void OnNetworkChanged(uint32_t target_bitrate_bps,
                         uint8_t fraction_lost,
                         int64_t rtt) override;
 
+  void Pause();
+  void Resume();
+
  protected:
   void ProcessFeedbackAndGeneratePackets(int64_t time_ms,
                                          std::list<FeedbackPacket*>* feedbacks,
                                          Packets* generated);
 
+  bool running_;
   VideoSource* source_;
   rtc::scoped_ptr<BweSender> bwe_;
   int64_t start_of_run_ms_;
   std::list<Module*> modules_;
 
  private:
   DISALLOW_COPY_AND_ASSIGN(VideoSender);
 };
 
 class PacedVideoSender : public VideoSender, public PacedSender::Callback {
@@ skipping 24 matching lines @@
 
   PacedSender pacer_;
   Packets queue_;
   Packets pacer_queue_;
 
   DISALLOW_IMPLICIT_CONSTRUCTORS(PacedVideoSender);
 };
 
 class TcpSender : public PacketSender {
  public:
-  TcpSender(PacketProcessorListener* listener, int flow_id, int64_t offset_ms)
-      : PacketSender(listener, flow_id),
-        cwnd_(10),
-        ssthresh_(std::numeric_limits<int>::max()),
-        ack_received_(false),
-        last_acked_seq_num_(0),
-        next_sequence_number_(0),
-        offset_ms_(offset_ms),
-        last_reduction_time_ms_(-1),
-        last_rtt_ms_(0) {}
-
+  TcpSender(PacketProcessorListener* listener, int flow_id, int64_t offset_ms);
+  TcpSender(PacketProcessorListener* listener,
+            int flow_id,
+            int64_t offset_ms,
+            int send_limit_bytes);
   virtual ~TcpSender() {}
 
   void RunFor(int64_t time_ms, Packets* in_out) override;
   int GetFeedbackIntervalMs() const override { return 10; }
+  void set_choke_filter(ChokeFilter* choke_filter) override;
 
  private:
   struct InFlight {
    public:
     InFlight(const MediaPacket& packet)
         : sequence_number(packet.header().sequenceNumber),
           time_ms(packet.send_time_us() / 1000) {}
 
     InFlight(uint16_t seq_num, int64_t now_ms)
         : sequence_number(seq_num), time_ms(now_ms) {}
@@ skipping 16 matching lines @@
 
   float cwnd_;
   int ssthresh_;
   std::set<InFlight> in_flight_;
   bool ack_received_;
   uint16_t last_acked_seq_num_;
   uint16_t next_sequence_number_;
   int64_t offset_ms_;
   int64_t last_reduction_time_ms_;
   int64_t last_rtt_ms_;
+  int total_sent_bytes_;
+  int send_limit_bytes_;  // Initialized by default as kNoLimit.
+  bool running_;          // Initialized by default as true.
+  ChokeFilter* choke_filter_;
+  int64_t last_generated_packets_ms_;
+  uint32_t estimated_kbps_;
+  size_t num_recent_sent_packets_;
 };
 }  // namespace bwe
 }  // namespace testing
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_PACKET_SENDER_H_