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/*
* 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_
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