| /* | 
 |  *  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 { | 
 |  | 
 | // TODO(bugs.webrtc.org/6883): This type should be split into separate types for | 
 | // audio and video, and then moved out of this file. | 
 | enum FrameTypeDeprecated { | 
 |   kEmptyFrame = 0, | 
 |   kAudioFrameSpeech = 1, | 
 |   kAudioFrameCN = 2, | 
 |   kVideoFrameKey = 3, | 
 |   kVideoFrameDelta = 4, | 
 | }; | 
 |  | 
 | // Can't use RTC_DEPRECATED until Chromium is updated. | 
 | typedef FrameTypeDeprecated FrameType; | 
 |  | 
 | using VideoFrameType = FrameTypeDeprecated; | 
 |  | 
 | // 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_ |