Add histogram stats for send delay for a sent video stream.

webrtc/video/send_statistics_proxy.h

 /*
  *  Copyright (c) 2013 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 19 matching lines @@
 
 namespace webrtc {
 
 class SendStatisticsProxy : public CpuOveruseMetricsObserver,
                             public RtcpStatisticsCallback,
                             public RtcpPacketTypeCounterObserver,
                             public StreamDataCountersCallback,
                             public BitrateStatisticsObserver,
                             public FrameCountObserver,
                             public VideoEncoderRateObserver,
-                            public SendSideDelayObserver {
+                            public SendSideDelayObserver,
+                            public SendPacketObserver {
  public:
   static const int kStatsTimeoutMs;
 
   SendStatisticsProxy(Clock* clock,
                       const VideoSendStream::Config& config,
                       VideoEncoderConfig::ContentType content_type);
   virtual ~SendStatisticsProxy();
 
   VideoSendStream::Stats GetStats();
 
@@ skipping 11 matching lines @@
   // how stats are collected.
   void SetContentType(VideoEncoderConfig::ContentType content_type);
 
   // Implements VideoEncoderRateObserver.
   void OnSetRates(uint32_t bitrate_bps, int framerate) override;
 
   // Implements CpuOveruseMetricsObserver.
   void OnEncodedFrameTimeMeasured(int encode_time_ms,
                                   const CpuOveruseMetrics& metrics) override;
 
+  // Called when a packet is sent (leaving socket).
+  bool OnSentPacket(int packet_id);
+
  protected:
   // From RtcpStatisticsCallback.
   void StatisticsUpdated(const RtcpStatistics& statistics,
                          uint32_t ssrc) override;
   void CNameChanged(const char* cname, uint32_t ssrc) override;
   // From RtcpPacketTypeCounterObserver.
   void RtcpPacketTypesCounterUpdated(
       uint32_t ssrc,
       const RtcpPacketTypeCounter& packet_counter) override;
   // From StreamDataCountersCallback.
   void DataCountersUpdated(const StreamDataCounters& counters,
                            uint32_t ssrc) override;
 
   // From BitrateStatisticsObserver.
   void Notify(const BitrateStatistics& total_stats,
               const BitrateStatistics& retransmit_stats,
               uint32_t ssrc) override;
 
   // From FrameCountObserver.
   void FrameCountUpdated(const FrameCounts& frame_counts,
                          uint32_t ssrc) override;
 
   void SendSideDelayUpdated(int avg_delay_ms,
                             int max_delay_ms,
                             uint32_t ssrc) override;
 
+  // From SendPacketObserver.
+  // Called when a packet is sent to the transport.
+  void OnSendPacket(uint16_t packet_id,
+                    int64_t capture_time_ms,
+                    uint32_t ssrc) override;
+
  private:
+  // Map holding sent packets (mapped by sequence number).
+  struct SequenceNumberOlderThan {
+    bool operator()(uint16_t seq1, uint16_t seq2) const {
+      return IsNewerSequenceNumber(seq2, seq1);
+    }
+  };
+  struct Packet {
+    Packet(uint32_t ssrc, int64_t capture_time_ms, int64_t send_time_ms)
+        : ssrc(ssrc),
+          capture_time_ms(capture_time_ms),
+          send_time_ms(send_time_ms) {}
+    uint32_t ssrc;
+    int64_t capture_time_ms;
+    int64_t send_time_ms;
+  };
+  typedef std::map<uint16_t, Packet, SequenceNumberOlderThan> PacketMap;
+
   class SampleCounter {
    public:
     SampleCounter() : sum(0), num_samples(0) {}
     ~SampleCounter() {}
     void Add(int sample);
     int Avg(int min_required_samples) const;
 
    private:
     int sum;
     int num_samples;
@@ skipping 16 matching lines @@
     int64_t resolution_update_ms;
     int64_t bitrate_update_ms;
   };
   struct QpCounters {
     SampleCounter vp8;  // QP range: 0-127
     SampleCounter vp9;  // QP range: 0-255
   };
   void PurgeOldStats() EXCLUSIVE_LOCKS_REQUIRED(crit_);
   VideoSendStream::StreamStats* GetStatsEntry(uint32_t ssrc)
       EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  void RemoveOld(int64_t now, PacketMap* packets)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
   Clock* const clock_;
   const VideoSendStream::Config config_;
   rtc::CriticalSection crit_;
   VideoEncoderConfig::ContentType content_type_ GUARDED_BY(crit_);
   VideoSendStream::Stats stats_ GUARDED_BY(crit_);
   uint32_t last_sent_frame_timestamp_ GUARDED_BY(crit_);
   std::map<uint32_t, StatsUpdateTimes> update_times_ GUARDED_BY(crit_);
   rtc::ExpFilter encode_time_ GUARDED_BY(crit_);
 
+  // Send delay stats.
+  PacketMap packets_ GUARDED_BY(crit_);
+  size_t num_old_packets_;
+  size_t num_skipped_packets_;
+
   // Contains stats used for UMA histograms. These stats will be reset if
   // content type changes between real-time video and screenshare, since these
   // will be reported separately.
   struct UmaSamplesContainer {
     UmaSamplesContainer(const char* prefix,
                         const VideoSendStream::Stats& start_stats,
                         Clock* clock);
     ~UmaSamplesContainer();
 
     void UpdateHistograms(const VideoSendStream::Config& config,
@@ skipping 16 matching lines @@
     SampleCounter delay_counter_;
     SampleCounter max_delay_counter_;
     rtc::RateTracker input_frame_rate_tracker_;
     rtc::RateTracker sent_frame_rate_tracker_;
     int64_t first_rtcp_stats_time_ms_;
     int64_t first_rtp_stats_time_ms_;
     ReportBlockStats report_block_stats_;
     const VideoSendStream::Stats start_stats_;
     std::map<int, QpCounters>
         qp_counters_;  // QP counters mapped by spatial idx.
+    std::map<uint32_t, SampleCounter> send_delay_counters_;  // Mapped by SSRC.
   };
 
   std::unique_ptr<UmaSamplesContainer> uma_container_ GUARDED_BY(crit_);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_VIDEO_SEND_STATISTICS_PROXY_H_