voip_statistics.h - mozsearch

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/*

 *  Copyright (c) 2020 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 API_VOIP_VOIP_STATISTICS_H_

#define API_VOIP_VOIP_STATISTICS_H_

#include <cstdint>

#include "absl/types/optional.h"

#include "api/neteq/neteq.h"

#include "api/voip/voip_base.h"

namespace webrtc {

struct IngressStatistics {

  // Stats included from api/neteq/neteq.h.

  NetEqLifetimeStatistics neteq_stats;

  // Represents the total duration in seconds of all samples that have been

  // received.

  // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalsamplesduration

  double total_duration = 0.0;

};

// Remote statistics obtained via remote RTCP SR/RR report received.

struct RemoteRtcpStatistics {

  // Jitter as defined in RFC 3550 [6.4.1] expressed in seconds.

  double jitter = 0.0;

  // Cumulative packets lost as defined in RFC 3550 [6.4.1]

  int64_t packets_lost = 0;

  // Fraction lost as defined in RFC 3550 [6.4.1] expressed as a floating

  // pointer number.

  double fraction_lost = 0.0;

  // https://w3c.github.io/webrtc-stats/#dom-rtcremoteinboundrtpstreamstats-roundtriptime

  absl::optional<double> round_trip_time;

  // Last time (not RTP timestamp) when RTCP report received in milliseconds.

  int64_t last_report_received_timestamp_ms;

};

struct ChannelStatistics {

  // https://w3c.github.io/webrtc-stats/#dom-rtcsentrtpstreamstats-packetssent

  uint64_t packets_sent = 0;

  // https://w3c.github.io/webrtc-stats/#dom-rtcsentrtpstreamstats-bytessent

  uint64_t bytes_sent = 0;

  // https://w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats-packetsreceived

  uint64_t packets_received = 0;

  // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-bytesreceived

  uint64_t bytes_received = 0;

  // https://w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats-jitter

  double jitter = 0.0;

  // https://w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats-packetslost

  int64_t packets_lost = 0;

  // SSRC from remote media endpoint as indicated either by RTP header in RFC

  // 3550 [5.1] or RTCP SSRC of sender in RFC 3550 [6.4.1].

  absl::optional<uint32_t> remote_ssrc;

  absl::optional<RemoteRtcpStatistics> remote_rtcp;

};

// VoipStatistics interface provides the interfaces for querying metrics around

// the jitter buffer (NetEq) performance.

class VoipStatistics {

 public:

  // Gets the audio ingress statistics by `ingress_stats` reference.

  // Returns following VoipResult;

  //  kOk - successfully set provided IngressStatistics reference.

  //  kInvalidArgument - `channel_id` is invalid.

  virtual VoipResult GetIngressStatistics(ChannelId channel_id,

                                          IngressStatistics& ingress_stats) = 0;

  // Gets the channel statistics by `channel_stats` reference.

  // Returns following VoipResult;

  //  kOk - successfully set provided ChannelStatistics reference.

  //  kInvalidArgument - `channel_id` is invalid.

  virtual VoipResult GetChannelStatistics(ChannelId channel_id,

                                          ChannelStatistics& channel_stats) = 0;

 protected:

  virtual ~VoipStatistics() = default;

};

}  // namespace webrtc

#endif  // API_VOIP_VOIP_STATISTICS_H_