1 /*
2  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
3  *
4  *  Use of this source code is governed by a BSD-style license
5  *  that can be found in the LICENSE file in the root of the source
6  *  tree. An additional intellectual property rights grant can be found
7  *  in the file PATENTS.  All contributing project authors may
8  *  be found in the AUTHORS file in the root of the source tree.
9  */
10 
11 #ifndef MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
12 #define MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
13 
14 #include <map>
15 #include <memory>
16 #include <vector>
17 
18 #include "absl/types/optional.h"
19 #include "call/rtp_packet_sink_interface.h"
20 #include "modules/include/module.h"
21 #include "modules/rtp_rtcp/include/rtcp_statistics.h"
22 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
23 #include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
24 #include "rtc_base/deprecation.h"
25 
26 namespace webrtc {
27 
28 class Clock;
29 
30 class ReceiveStatisticsProvider {
31  public:
32   virtual ~ReceiveStatisticsProvider() = default;
33   // Collects receive statistic in a form of rtcp report blocks.
34   // Returns at most |max_blocks| report blocks.
35   virtual std::vector<rtcp::ReportBlock> RtcpReportBlocks(
36       size_t max_blocks) = 0;
37 };
38 
39 class StreamStatistician {
40  public:
41   virtual ~StreamStatistician();
42 
43   virtual RtpReceiveStats GetStats() const = 0;
44 
45   // Returns average over the stream life time.
46   virtual absl::optional<int> GetFractionLostInPercent() const = 0;
47 
48   // TODO(nisse): Delete, migrate users to the above the GetStats method.
49   // Gets received stream data counters (includes reset counter values).
50   virtual StreamDataCounters GetReceiveStreamDataCounters() const = 0;
51 
52   virtual uint32_t BitrateReceived() const = 0;
53 };
54 
55 class ReceiveStatistics : public ReceiveStatisticsProvider,
56                           public RtpPacketSinkInterface {
57  public:
58   ~ReceiveStatistics() override = default;
59 
60   static std::unique_ptr<ReceiveStatistics> Create(Clock* clock);
61 
62   // Returns a pointer to the statistician of an ssrc.
63   virtual StreamStatistician* GetStatistician(uint32_t ssrc) const = 0;
64 
65   // TODO(bugs.webrtc.org/10669): Deprecated, delete as soon as downstream
66   // projects are updated. This method sets the max reordering threshold of all
67   // current and future streams.
68   virtual void SetMaxReorderingThreshold(int max_reordering_threshold) = 0;
69 
70   // Sets the max reordering threshold in number of packets.
71   virtual void SetMaxReorderingThreshold(uint32_t ssrc,
72                                          int max_reordering_threshold) = 0;
73   // Detect retransmissions, enabling updates of the retransmitted counters. The
74   // default is false.
75   virtual void EnableRetransmitDetection(uint32_t ssrc, bool enable) = 0;
76 };
77 
78 }  // namespace webrtc
79 #endif  // MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
80