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

webrtc/video/send_delay_stats.h

Index: webrtc/video/send_delay_stats.h
diff --git a/webrtc/video/send_delay_stats.h b/webrtc/video/send_delay_stats.h
new file mode 100644
index 0000000000000000000000000000000000000000..20a97814c72e808b3acfd5a920290e01a993cc8d
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+++ b/webrtc/video/send_delay_stats.h
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+/*
+ *  Copyright (c) 2016 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_SEND_DELAY_STATS_H_
+#define WEBRTC_VIDEO_SEND_DELAY_STATS_H_
+
+#include <map>
+#include <memory>
+#include <set>
+
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/video_send_stream.h"
+
+namespace webrtc {
+
+class SendDelayStats : public SendPacketObserver {
+ public:
+  explicit SendDelayStats(Clock* clock);
+  virtual ~SendDelayStats();
+
+  // Adds the configured ssrcs for the rtp streams.
+  // Stats will be calculated for these streams.
+  void AddSsrcs(const VideoSendStream::Config& config);
+
+  // Called when a packet is sent (leaving socket).
+  bool OnSentPacket(int packet_id, int64_t time_ms);
+
+ protected:
+  // 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;
+  };
+
+  void UpdateHistograms();
+  void RemoveOld(int64_t now, PacketMap* packets)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_);
+
+  Clock* const clock_;
+  rtc::CriticalSection crit_;
+
+  PacketMap packets_ GUARDED_BY(crit_);
+  size_t num_old_packets_ GUARDED_BY(crit_);
+  size_t num_skipped_packets_ GUARDED_BY(crit_);
+
+  std::set<uint32_t> ssrcs_ GUARDED_BY(crit_);
+  std::map<uint32_t, SampleCounter> send_delay_counters_
+      GUARDED_BY(crit_);  // Mapped by SSRC.
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_VIDEO_SEND_DELAY_STATS_H_