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