| /* |
| * 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. |
| */ |
| |
| #ifndef MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_ |
| #define MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_ |
| |
| #include "modules/rtp_rtcp/include/receive_statistics.h" |
| |
| #include <algorithm> |
| #include <map> |
| #include <vector> |
| |
| #include "absl/types/optional.h" |
| #include "modules/include/module_common_types_public.h" |
| #include "rtc_base/critical_section.h" |
| #include "rtc_base/rate_statistics.h" |
| #include "rtc_base/thread_annotations.h" |
| |
| namespace webrtc { |
| |
| class StreamStatisticianImpl : public StreamStatistician, |
| public RtpPacketSinkInterface { |
| public: |
| StreamStatisticianImpl(uint32_t ssrc, |
| Clock* clock, |
| int max_reordering_threshold, |
| RtcpStatisticsCallback* rtcp_callback, |
| StreamDataCountersCallback* rtp_callback); |
| ~StreamStatisticianImpl() override; |
| |
| // |reset| here and in next method restarts calculation of fraction_lost stat. |
| bool GetStatistics(RtcpStatistics* statistics, bool reset) override; |
| bool GetActiveStatisticsAndReset(RtcpStatistics* statistics); |
| void GetDataCounters(size_t* bytes_received, |
| uint32_t* packets_received) const override; |
| void GetReceiveStreamDataCounters( |
| StreamDataCounters* data_counters) const override; |
| uint32_t BitrateReceived() const override; |
| |
| // Implements RtpPacketSinkInterface |
| void OnRtpPacket(const RtpPacketReceived& packet) override; |
| |
| void FecPacketReceived(const RtpPacketReceived& packet); |
| void SetMaxReorderingThreshold(int max_reordering_threshold); |
| void EnableRetransmitDetection(bool enable); |
| |
| private: |
| bool IsRetransmitOfOldPacket(const RtpPacketReceived& packet, |
| int64_t now_ms) const |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_lock_); |
| RtcpStatistics CalculateRtcpStatistics() |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_lock_); |
| void UpdateJitter(const RtpPacketReceived& packet, int64_t receive_time_ms) |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_lock_); |
| // Updates StreamStatistician for out of order packets. |
| // Returns true if packet considered to be out of order. |
| bool UpdateOutOfOrder(const RtpPacketReceived& packet, |
| int64_t sequence_number, |
| int64_t now_ms) |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_lock_); |
| // Updates StreamStatistician for incoming packets. |
| StreamDataCounters UpdateCounters(const RtpPacketReceived& packet); |
| // Checks if this StreamStatistician received any rtp packets. |
| bool ReceivedRtpPacket() const RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_lock_) { |
| return received_seq_max_ >= 0; |
| } |
| |
| const uint32_t ssrc_; |
| Clock* const clock_; |
| rtc::CriticalSection stream_lock_; |
| RateStatistics incoming_bitrate_ RTC_GUARDED_BY(&stream_lock_); |
| // In number of packets or sequence numbers. |
| int max_reordering_threshold_ RTC_GUARDED_BY(&stream_lock_); |
| bool enable_retransmit_detection_ RTC_GUARDED_BY(&stream_lock_); |
| |
| // Stats on received RTP packets. |
| uint32_t jitter_q4_ RTC_GUARDED_BY(&stream_lock_); |
| uint32_t cumulative_loss_ RTC_GUARDED_BY(&stream_lock_); |
| |
| int64_t last_receive_time_ms_ RTC_GUARDED_BY(&stream_lock_); |
| uint32_t last_received_timestamp_ RTC_GUARDED_BY(&stream_lock_); |
| SequenceNumberUnwrapper seq_unwrapper_ RTC_GUARDED_BY(&stream_lock_); |
| int64_t received_seq_first_ RTC_GUARDED_BY(&stream_lock_); |
| int64_t received_seq_max_ RTC_GUARDED_BY(&stream_lock_); |
| // Assume that the other side restarted when there are two sequential packets |
| // with large jump from received_seq_max_. |
| absl::optional<uint16_t> received_seq_out_of_order_ |
| RTC_GUARDED_BY(&stream_lock_); |
| |
| // Current counter values. |
| StreamDataCounters receive_counters_ RTC_GUARDED_BY(&stream_lock_); |
| |
| // Counter values when we sent the last report. |
| uint32_t last_report_inorder_packets_ RTC_GUARDED_BY(&stream_lock_); |
| uint32_t last_report_old_packets_ RTC_GUARDED_BY(&stream_lock_); |
| int64_t last_report_seq_max_ RTC_GUARDED_BY(&stream_lock_); |
| RtcpStatistics last_reported_statistics_ RTC_GUARDED_BY(&stream_lock_); |
| |
| // stream_lock_ shouldn't be held when calling callbacks. |
| RtcpStatisticsCallback* const rtcp_callback_; |
| StreamDataCountersCallback* const rtp_callback_; |
| }; |
| |
| class ReceiveStatisticsImpl : public ReceiveStatistics { |
| public: |
| ReceiveStatisticsImpl(Clock* clock, |
| RtcpStatisticsCallback* rtcp_callback, |
| StreamDataCountersCallback* rtp_callback); |
| |
| ~ReceiveStatisticsImpl() override; |
| |
| // Implements ReceiveStatisticsProvider. |
| std::vector<rtcp::ReportBlock> RtcpReportBlocks(size_t max_blocks) override; |
| |
| // Implements RtpPacketSinkInterface |
| void OnRtpPacket(const RtpPacketReceived& packet) override; |
| |
| // Implements ReceiveStatistics. |
| void FecPacketReceived(const RtpPacketReceived& packet) override; |
| // Note: More specific return type for use in the implementation. |
| StreamStatisticianImpl* GetStatistician(uint32_t ssrc) const override; |
| void SetMaxReorderingThreshold(int max_reordering_threshold) override; |
| void SetMaxReorderingThreshold(uint32_t ssrc, |
| int max_reordering_threshold) override; |
| void EnableRetransmitDetection(uint32_t ssrc, bool enable) override; |
| |
| private: |
| StreamStatisticianImpl* GetOrCreateStatistician(uint32_t ssrc); |
| |
| Clock* const clock_; |
| rtc::CriticalSection receive_statistics_lock_; |
| uint32_t last_returned_ssrc_; |
| int max_reordering_threshold_ RTC_GUARDED_BY(receive_statistics_lock_); |
| std::map<uint32_t, StreamStatisticianImpl*> statisticians_ |
| RTC_GUARDED_BY(receive_statistics_lock_); |
| |
| RtcpStatisticsCallback* const rtcp_stats_callback_; |
| StreamDataCountersCallback* const rtp_stats_callback_; |
| }; |
| } // namespace webrtc |
| #endif // MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_ |