1 /* 2 * Copyright (c) 2020 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 API_VOIP_VOIP_STATISTICS_H_ 12 #define API_VOIP_VOIP_STATISTICS_H_ 13 14 #include "api/neteq/neteq.h" 15 #include "api/voip/voip_base.h" 16 17 namespace webrtc { 18 19 struct IngressStatistics { 20 // Stats included from api/neteq/neteq.h. 21 NetEqLifetimeStatistics neteq_stats; 22 23 // Represents the total duration in seconds of all samples that have been 24 // received. 25 // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalsamplesduration 26 double total_duration = 0.0; 27 }; 28 29 // Remote statistics obtained via remote RTCP SR/RR report received. 30 struct RemoteRtcpStatistics { 31 // Jitter as defined in RFC 3550 [6.4.1] expressed in seconds. 32 double jitter = 0.0; 33 34 // Cumulative packets lost as defined in RFC 3550 [6.4.1] 35 int64_t packets_lost = 0; 36 37 // Fraction lost as defined in RFC 3550 [6.4.1] expressed as a floating 38 // pointer number. 39 double fraction_lost = 0.0; 40 41 // https://w3c.github.io/webrtc-stats/#dom-rtcremoteinboundrtpstreamstats-roundtriptime 42 absl::optional<double> round_trip_time; 43 44 // Last time (not RTP timestamp) when RTCP report received in milliseconds. 45 int64_t last_report_received_timestamp_ms; 46 }; 47 48 struct ChannelStatistics { 49 // https://w3c.github.io/webrtc-stats/#dom-rtcsentrtpstreamstats-packetssent 50 uint64_t packets_sent = 0; 51 52 // https://w3c.github.io/webrtc-stats/#dom-rtcsentrtpstreamstats-bytessent 53 uint64_t bytes_sent = 0; 54 55 // https://w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats-packetsreceived 56 uint64_t packets_received = 0; 57 58 // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-bytesreceived 59 uint64_t bytes_received = 0; 60 61 // https://w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats-jitter 62 double jitter = 0.0; 63 64 // https://w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats-packetslost 65 int64_t packets_lost = 0; 66 67 // SSRC from remote media endpoint as indicated either by RTP header in RFC 68 // 3550 [5.1] or RTCP SSRC of sender in RFC 3550 [6.4.1]. 69 absl::optional<uint32_t> remote_ssrc; 70 71 absl::optional<RemoteRtcpStatistics> remote_rtcp; 72 }; 73 74 // VoipStatistics interface provides the interfaces for querying metrics around 75 // the jitter buffer (NetEq) performance. 76 class VoipStatistics { 77 public: 78 // Gets the audio ingress statistics by `ingress_stats` reference. 79 // Returns following VoipResult; 80 // kOk - successfully set provided IngressStatistics reference. 81 // kInvalidArgument - `channel_id` is invalid. 82 virtual VoipResult GetIngressStatistics(ChannelId channel_id, 83 IngressStatistics& ingress_stats) = 0; 84 85 // Gets the channel statistics by `channel_stats` reference. 86 // Returns following VoipResult; 87 // kOk - successfully set provided ChannelStatistics reference. 88 // kInvalidArgument - `channel_id` is invalid. 89 virtual VoipResult GetChannelStatistics(ChannelId channel_id, 90 ChannelStatistics& channel_stats) = 0; 91 92 protected: 93 virtual ~VoipStatistics() = default; 94 }; 95 96 } // namespace webrtc 97 98 #endif // API_VOIP_VOIP_STATISTICS_H_ 99