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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 "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_
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