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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;
};
// VoipStatistics interface provides the interfaces for querying metrics around
// the jitter buffer (NetEq) performance.
class VoipStatistics {
public:
// Gets the audio ingress statistics. Returns absl::nullopt when channel_id is
// invalid.
virtual absl::optional<IngressStatistics> GetIngressStatistics(
ChannelId channel_id) = 0;
protected:
virtual ~VoipStatistics() = default;
};
} // namespace webrtc
#endif // API_VOIP_VOIP_STATISTICS_H_