| /* |
| * 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 <map> |
| #include <memory> |
| #include <set> |
| |
| #include "call/video_send_stream.h" |
| #include "rtc_base/criticalsection.h" |
| #include "rtc_base/thread_annotations.h" |
| #include "system_wrappers/include/clock.h" |
| #include "video/stats_counter.h" |
| |
| namespace webrtc { |
| |
| class SendDelayStats : public SendPacketObserver { |
| public: |
| explicit SendDelayStats(Clock* clock); |
| ~SendDelayStats() override; |
| |
| // 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; |
| |
| void UpdateHistograms(); |
| void RemoveOld(int64_t now, PacketMap* packets) |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_); |
| AvgCounter* GetSendDelayCounter(uint32_t ssrc) |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_); |
| |
| Clock* const clock_; |
| rtc::CriticalSection crit_; |
| |
| PacketMap packets_ RTC_GUARDED_BY(crit_); |
| size_t num_old_packets_ RTC_GUARDED_BY(crit_); |
| size_t num_skipped_packets_ RTC_GUARDED_BY(crit_); |
| |
| std::set<uint32_t> ssrcs_ RTC_GUARDED_BY(crit_); |
| |
| // Mapped by SSRC. |
| std::map<uint32_t, std::unique_ptr<AvgCounter>> send_delay_counters_ |
| RTC_GUARDED_BY(crit_); |
| }; |
| |
| } // namespace webrtc |
| #endif // VIDEO_SEND_DELAY_STATS_H_ |