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https://github.com/mollyim/webrtc.git
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fe9009be92
The gn files are simplified from what's currently in the RingRTC repo, looking ahead to when the RingRTC Rust and Java builds for Android aren't included in the WebRTC build.
104 lines
2.9 KiB
C++
104 lines
2.9 KiB
C++
/*
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* Copyright 2019-2021 Signal Messenger, LLC
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* SPDX-License-Identifier: AGPL-3.0-only
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*/
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#ifndef RFFI_API_STATS_OBSERVER_INTF_H__
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#define RFFI_API_STATS_OBSERVER_INTF_H__
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#include "api/peer_connection_interface.h"
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#include "rffi/api/rffi_defs.h"
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/**
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* Rust friendly wrapper for creating objects that implement the
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* webrtc::StatsCollector interface.
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*
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*/
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namespace webrtc {
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namespace rffi {
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class StatsObserverRffi;
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} // namespace rffi
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} // namespace webrtc
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typedef struct {
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uint32_t ssrc;
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uint32_t packets_sent;
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uint64_t bytes_sent;
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int32_t remote_packets_lost;
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double remote_jitter;
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double remote_round_trip_time;
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double total_audio_energy;
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double echo_likelihood;
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} AudioSenderStatistics;
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typedef struct {
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uint32_t ssrc;
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uint32_t packets_sent;
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uint64_t bytes_sent;
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uint32_t frames_encoded;
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uint32_t key_frames_encoded;
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double total_encode_time;
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uint32_t frame_width;
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uint32_t frame_height;
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uint64_t retransmitted_packets_sent;
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uint64_t retransmitted_bytes_sent;
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double total_packet_send_delay;
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uint32_t nack_count;
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uint32_t pli_count;
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uint32_t quality_limitation_reason; // 0 - kNone, 1 - kCpu, 2 - kBandwidth, 3 - kOther
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uint32_t quality_limitation_resolution_changes;
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int32_t remote_packets_lost;
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double remote_jitter;
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double remote_round_trip_time;
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} VideoSenderStatistics;
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typedef struct {
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uint32_t ssrc;
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uint32_t packets_received;
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int32_t packets_lost;
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uint64_t bytes_received;
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double jitter;
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double total_audio_energy;
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} AudioReceiverStatistics;
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typedef struct {
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uint32_t ssrc;
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uint32_t packets_received;
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int32_t packets_lost;
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uint64_t bytes_received;
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uint32_t frames_decoded;
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uint32_t key_frames_decoded;
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double total_decode_time;
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uint32_t frame_width;
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uint32_t frame_height;
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} VideoReceiverStatistics;
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typedef struct {
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double current_round_trip_time;
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double available_outgoing_bitrate;
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} ConnectionStatistics;
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typedef struct {
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int64_t timestamp_us;
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uint32_t audio_sender_statistics_size;
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const AudioSenderStatistics *audio_sender_statistics;
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uint32_t video_sender_statistics_size;
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const VideoSenderStatistics *video_sender_statistics;
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uint32_t audio_receiver_statistics_size;
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const AudioReceiverStatistics *audio_receiver_statistics;
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uint32_t video_receiver_statistics_count;
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const VideoReceiverStatistics *video_receiver_statistics;
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ConnectionStatistics connection_statistics;
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} MediaStatistics;
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/* Stats Observer Callback callback function pointers */
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typedef struct {
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void (*OnStatsComplete)(void* stats_observer_borrowed, const MediaStatistics* media_statistics_borrowed);
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} StatsObserverCallbacks;
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RUSTEXPORT webrtc::rffi::StatsObserverRffi*
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Rust_createStatsObserver(void* stats_observer_borrowed,
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const StatsObserverCallbacks* stats_observer_cbs_borrowed);
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#endif /* RFFI_API_STATS_OBSERVER_INTF_H__ */
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