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a4d873786f
This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
58 lines
1.9 KiB
C++
58 lines
1.9 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_RTP_RTCP_SOURCE_PACKET_LOSS_STATS_H_
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#define MODULES_RTP_RTCP_SOURCE_PACKET_LOSS_STATS_H_
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#include <stdint.h>
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#include <set>
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namespace webrtc {
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// Keeps track of statistics of packet loss including whether losses are a
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// single packet or multiple packets in a row.
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class PacketLossStats {
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public:
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PacketLossStats();
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~PacketLossStats();
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// Adds a lost packet to the stats by sequence number.
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void AddLostPacket(uint16_t sequence_number);
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// Queries the number of packets that were lost by themselves, no neighboring
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// packets were lost.
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int GetSingleLossCount() const;
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// Queries the number of times that multiple packets with sequential numbers
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// were lost. This is the number of events with more than one packet lost,
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// regardless of the size of the event;
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int GetMultipleLossEventCount() const;
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// Queries the number of packets lost in multiple packet loss events. Combined
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// with the event count, this can be used to determine the average event size.
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int GetMultipleLossPacketCount() const;
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private:
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std::set<uint16_t> lost_packets_buffer_;
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std::set<uint16_t> lost_packets_wrapped_buffer_;
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int single_loss_historic_count_;
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int multiple_loss_historic_event_count_;
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int multiple_loss_historic_packet_count_;
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void ComputeLossCounts(int* out_single_loss_count,
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int* out_multiple_loss_event_count,
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int* out_multiple_loss_packet_count) const;
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void PruneBuffer();
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_PACKET_LOSS_STATS_H_
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