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}
89 lines
2.6 KiB
C++
89 lines
2.6 KiB
C++
/*
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* Copyright (c) 2016 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 VIDEO_SEND_DELAY_STATS_H_
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#define VIDEO_SEND_DELAY_STATS_H_
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#include <stddef.h>
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#include <stdint.h>
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#include <map>
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#include <memory>
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#include <set>
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#include "call/video_send_stream.h"
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#include "modules/include/module_common_types_public.h"
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#include "rtc_base/critical_section.h"
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#include "rtc_base/thread_annotations.h"
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#include "system_wrappers/include/clock.h"
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#include "video/stats_counter.h"
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namespace webrtc {
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class SendDelayStats : public SendPacketObserver {
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public:
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explicit SendDelayStats(Clock* clock);
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~SendDelayStats() override;
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// Adds the configured ssrcs for the rtp streams.
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// Stats will be calculated for these streams.
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void AddSsrcs(const VideoSendStream::Config& config);
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// Called when a packet is sent (leaving socket).
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bool OnSentPacket(int packet_id, int64_t time_ms);
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protected:
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// From SendPacketObserver.
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// Called when a packet is sent to the transport.
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void OnSendPacket(uint16_t packet_id,
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int64_t capture_time_ms,
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uint32_t ssrc) override;
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private:
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// Map holding sent packets (mapped by sequence number).
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struct SequenceNumberOlderThan {
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bool operator()(uint16_t seq1, uint16_t seq2) const {
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return IsNewerSequenceNumber(seq2, seq1);
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}
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};
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struct Packet {
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Packet(uint32_t ssrc, int64_t capture_time_ms, int64_t send_time_ms)
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: ssrc(ssrc),
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capture_time_ms(capture_time_ms),
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send_time_ms(send_time_ms) {}
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uint32_t ssrc;
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int64_t capture_time_ms;
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int64_t send_time_ms;
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};
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typedef std::map<uint16_t, Packet, SequenceNumberOlderThan> PacketMap;
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void UpdateHistograms();
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void RemoveOld(int64_t now, PacketMap* packets)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
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AvgCounter* GetSendDelayCounter(uint32_t ssrc)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
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Clock* const clock_;
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rtc::CriticalSection crit_;
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PacketMap packets_ RTC_GUARDED_BY(crit_);
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size_t num_old_packets_ RTC_GUARDED_BY(crit_);
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size_t num_skipped_packets_ RTC_GUARDED_BY(crit_);
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std::set<uint32_t> ssrcs_ RTC_GUARDED_BY(crit_);
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// Mapped by SSRC.
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std::map<uint32_t, std::unique_ptr<AvgCounter>> send_delay_counters_
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RTC_GUARDED_BY(crit_);
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};
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} // namespace webrtc
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#endif // VIDEO_SEND_DELAY_STATS_H_
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