Repo created

TMessagesProj/jni/voip/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 VIDEO_SEND_DELAY_STATS_H_
+#define VIDEO_SEND_DELAY_STATS_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <map>
+
+#include "api/units/timestamp.h"
+#include "call/video_send_stream.h"
+#include "modules/include/module_common_types_public.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/clock.h"
+#include "video/stats_counter.h"
+
+namespace webrtc {
+
+// Used to collect delay stats for video streams. The class gets callbacks
+// from more than one threads and internally uses a mutex for data access
+// synchronization.
+// TODO(bugs.webrtc.org/11993): OnSendPacket and OnSentPacket will eventually
+// be called consistently on the same thread. Once we're there, we should be
+// able to avoid locking (at least for the fast path).
+class SendDelayStats {
+ public:
+  explicit SendDelayStats(Clock* clock);
+  ~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, Timestamp time);
+
+  // Called when a packet is sent to the transport.
+  void OnSendPacket(uint16_t packet_id, Timestamp capture_time, uint32_t ssrc);
+
+ 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 {
+    AvgCounter* send_delay;
+    Timestamp capture_time;
+    Timestamp send_time;
+  };
+
+  void UpdateHistograms();
+  void RemoveOld(Timestamp now) RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  Clock* const clock_;
+  Mutex mutex_;
+
+  std::map<uint16_t, Packet, SequenceNumberOlderThan> packets_
+      RTC_GUARDED_BY(mutex_);
+  size_t num_old_packets_ RTC_GUARDED_BY(mutex_);
+  size_t num_skipped_packets_ RTC_GUARDED_BY(mutex_);
+
+  // Mapped by SSRC.
+  std::map<uint32_t, AvgCounter> send_delay_counters_ RTC_GUARDED_BY(mutex_);
+};
+
+}  // namespace webrtc
+#endif  // VIDEO_SEND_DELAY_STATS_H_