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https://github.com/mollyim/webrtc.git
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c8152fe4a8
Bug: webrtc:10198 Change-Id: I226768c2a6bd97ffcd0638e5bc6a1c286b71815f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267704 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37435}
83 lines
3 KiB
C++
83 lines
3 KiB
C++
/*
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* Copyright (c) 2013 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_INCLUDE_RECEIVE_STATISTICS_H_
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#define MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
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#include <map>
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#include <memory>
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#include <vector>
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#include "absl/types/optional.h"
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#include "call/rtp_packet_sink_interface.h"
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#include "modules/rtp_rtcp/include/rtcp_statistics.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
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namespace webrtc {
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class Clock;
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class ReceiveStatisticsProvider {
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public:
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virtual ~ReceiveStatisticsProvider() = default;
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// Collects receive statistic in a form of rtcp report blocks.
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// Returns at most `max_blocks` report blocks.
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virtual std::vector<rtcp::ReportBlock> RtcpReportBlocks(
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size_t max_blocks) = 0;
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};
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class StreamStatistician {
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public:
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virtual ~StreamStatistician();
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virtual RtpReceiveStats GetStats() const = 0;
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// Returns average over the stream life time.
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virtual absl::optional<int> GetFractionLostInPercent() const = 0;
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// TODO(bugs.webrtc.org/10679): Delete, migrate users to the above GetStats
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// method (and extend RtpReceiveStats if needed).
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// Gets receive stream data counters.
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virtual StreamDataCounters GetReceiveStreamDataCounters() const = 0;
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virtual uint32_t BitrateReceived() const = 0;
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};
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class ReceiveStatistics : public ReceiveStatisticsProvider,
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public RtpPacketSinkInterface {
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public:
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~ReceiveStatistics() override = default;
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// Returns a thread-safe instance of ReceiveStatistics.
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// https://chromium.googlesource.com/chromium/src/+/lkgr/docs/threading_and_tasks.md#threading-lexicon
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static std::unique_ptr<ReceiveStatistics> Create(Clock* clock);
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// Returns a thread-compatible instance of ReceiveStatistics.
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static std::unique_ptr<ReceiveStatistics> CreateThreadCompatible(
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Clock* clock);
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// Returns a pointer to the statistician of an ssrc.
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virtual StreamStatistician* GetStatistician(uint32_t ssrc) const = 0;
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// TODO(bugs.webrtc.org/10669): Deprecated, delete as soon as downstream
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// projects are updated. This method sets the max reordering threshold of all
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// current and future streams.
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virtual void SetMaxReorderingThreshold(int max_reordering_threshold) = 0;
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// Sets the max reordering threshold in number of packets.
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virtual void SetMaxReorderingThreshold(uint32_t ssrc,
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int max_reordering_threshold) = 0;
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// Detect retransmissions, enabling updates of the retransmitted counters. The
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// default is false.
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virtual void EnableRetransmitDetection(uint32_t ssrc, bool enable) = 0;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
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