/* * Copyright (c) 2012 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 COMMON_TYPES_H_ #define COMMON_TYPES_H_ #include // For size_t #include namespace webrtc { // Statistics for RTCP packet types. struct RtcpPacketTypeCounter { RtcpPacketTypeCounter() : first_packet_time_ms(-1), nack_packets(0), fir_packets(0), pli_packets(0), nack_requests(0), unique_nack_requests(0) {} void Add(const RtcpPacketTypeCounter& other) { nack_packets += other.nack_packets; fir_packets += other.fir_packets; pli_packets += other.pli_packets; nack_requests += other.nack_requests; unique_nack_requests += other.unique_nack_requests; if (other.first_packet_time_ms != -1 && (other.first_packet_time_ms < first_packet_time_ms || first_packet_time_ms == -1)) { // Use oldest time. first_packet_time_ms = other.first_packet_time_ms; } } void Subtract(const RtcpPacketTypeCounter& other) { nack_packets -= other.nack_packets; fir_packets -= other.fir_packets; pli_packets -= other.pli_packets; nack_requests -= other.nack_requests; unique_nack_requests -= other.unique_nack_requests; if (other.first_packet_time_ms != -1 && (other.first_packet_time_ms > first_packet_time_ms || first_packet_time_ms == -1)) { // Use youngest time. first_packet_time_ms = other.first_packet_time_ms; } } int64_t TimeSinceFirstPacketInMs(int64_t now_ms) const { return (first_packet_time_ms == -1) ? -1 : (now_ms - first_packet_time_ms); } int UniqueNackRequestsInPercent() const { if (nack_requests == 0) { return 0; } return static_cast((unique_nack_requests * 100.0f / nack_requests) + 0.5f); } int64_t first_packet_time_ms; // Time when first packet is sent/received. uint32_t nack_packets; // Number of RTCP NACK packets. uint32_t fir_packets; // Number of RTCP FIR packets. uint32_t pli_packets; // Number of RTCP PLI packets. uint32_t nack_requests; // Number of NACKed RTP packets. uint32_t unique_nack_requests; // Number of unique NACKed RTP packets. }; class RtcpPacketTypeCounterObserver { public: virtual ~RtcpPacketTypeCounterObserver() {} virtual void RtcpPacketTypesCounterUpdated( uint32_t ssrc, const RtcpPacketTypeCounter& packet_counter) = 0; }; // Callback, used to notify an observer whenever new rates have been estimated. class BitrateStatisticsObserver { public: virtual ~BitrateStatisticsObserver() {} virtual void Notify(uint32_t total_bitrate_bps, uint32_t retransmit_bitrate_bps, uint32_t ssrc) = 0; }; struct FrameCounts { FrameCounts() : key_frames(0), delta_frames(0) {} int key_frames; int delta_frames; }; // Callback, used to notify an observer whenever frame counts have been updated. class FrameCountObserver { public: virtual ~FrameCountObserver() {} virtual void FrameCountUpdated(const FrameCounts& frame_counts, uint32_t ssrc) = 0; }; // Callback, used to notify an observer whenever the send-side delay is updated. class SendSideDelayObserver { public: virtual ~SendSideDelayObserver() {} virtual void SendSideDelayUpdated(int avg_delay_ms, int max_delay_ms, uint32_t ssrc) = 0; }; // Callback, used to notify an observer whenever a packet is sent to the // transport. // TODO(asapersson): This class will remove the need for SendSideDelayObserver. // Remove SendSideDelayObserver once possible. class SendPacketObserver { public: virtual ~SendPacketObserver() {} virtual void OnSendPacket(uint16_t packet_id, int64_t capture_time_ms, uint32_t ssrc) = 0; }; // Callback, used to notify an observer when the overhead per packet // has changed. class OverheadObserver { public: virtual ~OverheadObserver() = default; virtual void OnOverheadChanged(size_t overhead_bytes_per_packet) = 0; }; // RTP enum { kRtpCsrcSize = 15 }; // RFC 3550 page 13 // ================================================================== // Video specific types // ================================================================== // TODO(nisse): Delete, and switch to fourcc values everywhere? // Supported video types. enum class VideoType { kUnknown, kI420, kIYUV, kRGB24, kABGR, kARGB, kARGB4444, kRGB565, kARGB1555, kYUY2, kYV12, kUYVY, kMJPEG, kNV21, kNV12, kBGRA, }; // TODO(magjed): Move this and other H264 related classes out to their own file. namespace H264 { enum Profile { kProfileConstrainedBaseline, kProfileBaseline, kProfileMain, kProfileConstrainedHigh, kProfileHigh, }; } // namespace H264 struct SpatialLayer { bool operator==(const SpatialLayer& other) const; bool operator!=(const SpatialLayer& other) const { return !(*this == other); } unsigned short width; unsigned short height; float maxFramerate; // fps. unsigned char numberOfTemporalLayers; unsigned int maxBitrate; // kilobits/sec. unsigned int targetBitrate; // kilobits/sec. unsigned int minBitrate; // kilobits/sec. unsigned int qpMax; // minimum quality bool active; // encoded and sent. }; // Simulcast is when the same stream is encoded multiple times with different // settings such as resolution. typedef SpatialLayer SimulcastStream; // Bandwidth over-use detector options. These are used to drive // experimentation with bandwidth estimation parameters. // See modules/remote_bitrate_estimator/overuse_detector.h // TODO(terelius): This is only used in overuse_estimator.cc, and only in the // default constructed state. Can we move the relevant variables into that // class and delete this? See also disabled warning at line 27 struct OverUseDetectorOptions { OverUseDetectorOptions() : initial_slope(8.0 / 512.0), initial_offset(0), initial_e(), initial_process_noise(), initial_avg_noise(0.0), initial_var_noise(50) { initial_e[0][0] = 100; initial_e[1][1] = 1e-1; initial_e[0][1] = initial_e[1][0] = 0; initial_process_noise[0] = 1e-13; initial_process_noise[1] = 1e-3; } double initial_slope; double initial_offset; double initial_e[2][2]; double initial_process_noise[2]; double initial_avg_noise; double initial_var_noise; }; // Minimum and maximum playout delay values from capture to render. // These are best effort values. // // A value < 0 indicates no change from previous valid value. // // min = max = 0 indicates that the receiver should try and render // frame as soon as possible. // // min = x, max = y indicates that the receiver is free to adapt // in the range (x, y) based on network jitter. // // Note: Given that this gets embedded in a union, it is up-to the owner to // initialize these values. struct PlayoutDelay { int min_ms; int max_ms; }; } // namespace webrtc #endif // COMMON_TYPES_H_