| /* |
| * 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 <stddef.h> // For size_t |
| #include <cstdint> |
| |
| // TODO(bugs.webrtc.org/7660): Delete include once downstream code is updated. |
| #include "api/video/video_codec_type.h" |
| // TODO(bugs.webrtc.org/5876): For AudioFrameType. Delete when downstream code |
| // is updated. |
| #include "modules/audio_coding/include/audio_coding_module_typedefs.h" |
| |
| #if defined(_MSC_VER) |
| // Disable "new behavior: elements of array will be default initialized" |
| // warning. Affects OverUseDetectorOptions. |
| #pragma warning(disable : 4351) |
| #endif |
| |
| 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<int>((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_ |