| /* |
| * 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 WEBRTC_COMMON_TYPES_H_ |
| #define WEBRTC_COMMON_TYPES_H_ |
| |
| #include <stddef.h> |
| #include <string.h> |
| #include <ostream> |
| #include <string> |
| #include <vector> |
| |
| #include "webrtc/api/video/video_content_type.h" |
| #include "webrtc/api/video/video_rotation.h" |
| #include "webrtc/api/video/video_timing.h" |
| #include "webrtc/rtc_base/array_view.h" |
| #include "webrtc/rtc_base/checks.h" |
| #include "webrtc/rtc_base/deprecation.h" |
| #include "webrtc/rtc_base/optional.h" |
| #include "webrtc/typedefs.h" |
| |
| #if defined(_MSC_VER) |
| // Disable "new behavior: elements of array will be default initialized" |
| // warning. Affects OverUseDetectorOptions. |
| #pragma warning(disable : 4351) |
| #endif |
| |
| #if defined(WEBRTC_EXPORT) |
| #define WEBRTC_DLLEXPORT _declspec(dllexport) |
| #elif defined(WEBRTC_DLL) |
| #define WEBRTC_DLLEXPORT _declspec(dllimport) |
| #else |
| #define WEBRTC_DLLEXPORT |
| #endif |
| |
| #ifndef NULL |
| #define NULL 0 |
| #endif |
| |
| #define RTP_PAYLOAD_NAME_SIZE 32u |
| |
| #if defined(WEBRTC_WIN) || defined(WIN32) |
| // Compares two strings without regard to case. |
| #define STR_CASE_CMP(s1, s2) ::_stricmp(s1, s2) |
| // Compares characters of two strings without regard to case. |
| #define STR_NCASE_CMP(s1, s2, n) ::_strnicmp(s1, s2, n) |
| #else |
| #define STR_CASE_CMP(s1, s2) ::strcasecmp(s1, s2) |
| #define STR_NCASE_CMP(s1, s2, n) ::strncasecmp(s1, s2, n) |
| #endif |
| |
| namespace webrtc { |
| |
| class RewindableStream { |
| public: |
| virtual ~RewindableStream() {} |
| virtual int Rewind() = 0; |
| }; |
| |
| class InStream : public RewindableStream { |
| public: |
| // Reads |len| bytes from file to |buf|. Returns the number of bytes read |
| // or -1 on error. |
| virtual int Read(void* buf, size_t len) = 0; |
| }; |
| |
| class OutStream : public RewindableStream { |
| public: |
| // Writes |len| bytes from |buf| to file. The actual writing may happen |
| // some time later. Call Flush() to force a write. |
| virtual bool Write(const void* buf, size_t len) = 0; |
| }; |
| |
| enum TraceModule { |
| kTraceUndefined = 0, |
| // not a module, triggered from the engine code |
| kTraceVoice = 0x0001, |
| // not a module, triggered from the engine code |
| kTraceVideo = 0x0002, |
| // not a module, triggered from the utility code |
| kTraceUtility = 0x0003, |
| kTraceRtpRtcp = 0x0004, |
| kTraceTransport = 0x0005, |
| kTraceSrtp = 0x0006, |
| kTraceAudioCoding = 0x0007, |
| kTraceAudioMixerServer = 0x0008, |
| kTraceAudioMixerClient = 0x0009, |
| kTraceFile = 0x000a, |
| kTraceAudioProcessing = 0x000b, |
| kTraceVideoCoding = 0x0010, |
| kTraceVideoMixer = 0x0011, |
| kTraceAudioDevice = 0x0012, |
| kTraceVideoRenderer = 0x0014, |
| kTraceVideoCapture = 0x0015, |
| kTraceRemoteBitrateEstimator = 0x0017, |
| }; |
| |
| enum TraceLevel { |
| kTraceNone = 0x0000, // no trace |
| kTraceStateInfo = 0x0001, |
| kTraceWarning = 0x0002, |
| kTraceError = 0x0004, |
| kTraceCritical = 0x0008, |
| kTraceApiCall = 0x0010, |
| kTraceDefault = 0x00ff, |
| |
| kTraceModuleCall = 0x0020, |
| kTraceMemory = 0x0100, // memory info |
| kTraceTimer = 0x0200, // timing info |
| kTraceStream = 0x0400, // "continuous" stream of data |
| |
| // used for debug purposes |
| kTraceDebug = 0x0800, // debug |
| kTraceInfo = 0x1000, // debug info |
| |
| // Non-verbose level used by LS_INFO of logging.h. Do not use directly. |
| kTraceTerseInfo = 0x2000, |
| |
| kTraceAll = 0xffff |
| }; |
| |
| // External Trace API |
| class TraceCallback { |
| public: |
| virtual void Print(TraceLevel level, const char* message, int length) = 0; |
| |
| protected: |
| virtual ~TraceCallback() {} |
| TraceCallback() {} |
| }; |
| |
| enum FileFormats { |
| kFileFormatWavFile = 1, |
| kFileFormatCompressedFile = 2, |
| kFileFormatPreencodedFile = 4, |
| kFileFormatPcm16kHzFile = 7, |
| kFileFormatPcm8kHzFile = 8, |
| kFileFormatPcm32kHzFile = 9 |
| }; |
| |
| enum FrameType { |
| kEmptyFrame = 0, |
| kAudioFrameSpeech = 1, |
| kAudioFrameCN = 2, |
| kVideoFrameKey = 3, |
| kVideoFrameDelta = 4, |
| }; |
| |
| // Statistics for an RTCP channel |
| struct RtcpStatistics { |
| RtcpStatistics() |
| : fraction_lost(0), |
| packets_lost(0), |
| extended_highest_sequence_number(0), |
| jitter(0) {} |
| |
| uint8_t fraction_lost; |
| union { |
| uint32_t packets_lost; |
| RTC_DEPRECATED uint32_t cumulative_lost; |
| }; |
| union { |
| uint32_t extended_highest_sequence_number; |
| RTC_DEPRECATED uint32_t extended_max_sequence_number; |
| }; |
| uint32_t jitter; |
| }; |
| |
| class RtcpStatisticsCallback { |
| public: |
| virtual ~RtcpStatisticsCallback() {} |
| |
| virtual void StatisticsUpdated(const RtcpStatistics& statistics, |
| uint32_t ssrc) = 0; |
| virtual void CNameChanged(const char* cname, uint32_t ssrc) = 0; |
| }; |
| |
| // 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; |
| }; |
| |
| // Rate statistics for a stream. |
| struct BitrateStatistics { |
| BitrateStatistics() : bitrate_bps(0), packet_rate(0) {} |
| |
| uint32_t bitrate_bps; // Bitrate in bits per second. |
| uint32_t packet_rate; // Packet rate in packets per second. |
| }; |
| |
| // 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; |
| }; |
| |
| // ================================================================== |
| // Voice specific types |
| // ================================================================== |
| |
| // Each codec supported can be described by this structure. |
| struct CodecInst { |
| int pltype; |
| char plname[RTP_PAYLOAD_NAME_SIZE]; |
| int plfreq; |
| int pacsize; |
| size_t channels; |
| int rate; // bits/sec unlike {start,min,max}Bitrate elsewhere in this file! |
| |
| bool operator==(const CodecInst& other) const { |
| return pltype == other.pltype && |
| (STR_CASE_CMP(plname, other.plname) == 0) && |
| plfreq == other.plfreq && pacsize == other.pacsize && |
| channels == other.channels && rate == other.rate; |
| } |
| |
| bool operator!=(const CodecInst& other) const { return !(*this == other); } |
| |
| friend std::ostream& operator<<(std::ostream& os, const CodecInst& ci) { |
| os << "{pltype: " << ci.pltype; |
| os << ", plname: " << ci.plname; |
| os << ", plfreq: " << ci.plfreq; |
| os << ", pacsize: " << ci.pacsize; |
| os << ", channels: " << ci.channels; |
| os << ", rate: " << ci.rate << "}"; |
| return os; |
| } |
| }; |
| |
| // RTP |
| enum { kRtpCsrcSize = 15 }; // RFC 3550 page 13 |
| |
| enum PayloadFrequencies { |
| kFreq8000Hz = 8000, |
| kFreq16000Hz = 16000, |
| kFreq32000Hz = 32000 |
| }; |
| |
| // Degree of bandwidth reduction. |
| enum VadModes { |
| kVadConventional = 0, // lowest reduction |
| kVadAggressiveLow, |
| kVadAggressiveMid, |
| kVadAggressiveHigh // highest reduction |
| }; |
| |
| // NETEQ statistics. |
| struct NetworkStatistics { |
| // current jitter buffer size in ms |
| uint16_t currentBufferSize; |
| // preferred (optimal) buffer size in ms |
| uint16_t preferredBufferSize; |
| // adding extra delay due to "peaky jitter" |
| bool jitterPeaksFound; |
| // Total number of audio samples received, including synthesized samples. |
| // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalsamplesreceived |
| uint64_t totalSamplesReceived; |
| // Total number of inbound audio samples that are based on synthesized data to |
| // conceal packet loss. |
| // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-concealedsamples |
| uint64_t concealedSamples; |
| // Loss rate (network + late); fraction between 0 and 1, scaled to Q14. |
| uint16_t currentPacketLossRate; |
| // Late loss rate; fraction between 0 and 1, scaled to Q14. |
| union { |
| RTC_DEPRECATED uint16_t currentDiscardRate; |
| }; |
| // fraction (of original stream) of synthesized audio inserted through |
| // expansion (in Q14) |
| uint16_t currentExpandRate; |
| // fraction (of original stream) of synthesized speech inserted through |
| // expansion (in Q14) |
| uint16_t currentSpeechExpandRate; |
| // fraction of synthesized speech inserted through pre-emptive expansion |
| // (in Q14) |
| uint16_t currentPreemptiveRate; |
| // fraction of data removed through acceleration (in Q14) |
| uint16_t currentAccelerateRate; |
| // fraction of data coming from secondary decoding (in Q14) |
| uint16_t currentSecondaryDecodedRate; |
| // Fraction of secondary data, including FEC and RED, that is discarded (in |
| // Q14). Discarding of secondary data can be caused by the reception of the |
| // primary data, obsoleting the secondary data. It can also be caused by early |
| // or late arrival of secondary data. |
| uint16_t currentSecondaryDiscardedRate; |
| // clock-drift in parts-per-million (negative or positive) |
| int32_t clockDriftPPM; |
| // average packet waiting time in the jitter buffer (ms) |
| int meanWaitingTimeMs; |
| // median packet waiting time in the jitter buffer (ms) |
| int medianWaitingTimeMs; |
| // min packet waiting time in the jitter buffer (ms) |
| int minWaitingTimeMs; |
| // max packet waiting time in the jitter buffer (ms) |
| int maxWaitingTimeMs; |
| // added samples in off mode due to packet loss |
| size_t addedSamples; |
| }; |
| |
| // Statistics for calls to AudioCodingModule::PlayoutData10Ms(). |
| struct AudioDecodingCallStats { |
| AudioDecodingCallStats() |
| : calls_to_silence_generator(0), |
| calls_to_neteq(0), |
| decoded_normal(0), |
| decoded_plc(0), |
| decoded_cng(0), |
| decoded_plc_cng(0), |
| decoded_muted_output(0) {} |
| |
| int calls_to_silence_generator; // Number of calls where silence generated, |
| // and NetEq was disengaged from decoding. |
| int calls_to_neteq; // Number of calls to NetEq. |
| int decoded_normal; // Number of calls where audio RTP packet decoded. |
| int decoded_plc; // Number of calls resulted in PLC. |
| int decoded_cng; // Number of calls where comfort noise generated due to DTX. |
| int decoded_plc_cng; // Number of calls resulted where PLC faded to CNG. |
| int decoded_muted_output; // Number of calls returning a muted state output. |
| }; |
| |
| // ================================================================== |
| // 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, |
| }; |
| |
| // Video codec |
| enum { kPayloadNameSize = 32 }; |
| enum { kMaxSimulcastStreams = 4 }; |
| enum { kMaxSpatialLayers = 5 }; |
| enum { kMaxTemporalStreams = 4 }; |
| |
| enum VideoCodecComplexity { |
| kComplexityNormal = 0, |
| kComplexityHigh = 1, |
| kComplexityHigher = 2, |
| kComplexityMax = 3 |
| }; |
| |
| enum VP8ResilienceMode { |
| kResilienceOff, // The stream produced by the encoder requires a |
| // recovery frame (typically a key frame) to be |
| // decodable after a packet loss. |
| kResilientStream, // A stream produced by the encoder is resilient to |
| // packet losses, but packets within a frame subsequent |
| // to a loss can't be decoded. |
| kResilientFrames // Same as kResilientStream but with added resilience |
| // within a frame. |
| }; |
| |
| class TemporalLayersFactory; |
| // VP8 specific |
| struct VideoCodecVP8 { |
| // TODO(nisse): Unused, delete? |
| bool pictureLossIndicationOn; |
| VideoCodecComplexity complexity; |
| VP8ResilienceMode resilience; |
| unsigned char numberOfTemporalLayers; |
| bool denoisingOn; |
| bool errorConcealmentOn; |
| bool automaticResizeOn; |
| bool frameDroppingOn; |
| int keyFrameInterval; |
| TemporalLayersFactory* tl_factory; |
| }; |
| |
| // VP9 specific. |
| struct VideoCodecVP9 { |
| VideoCodecComplexity complexity; |
| bool resilienceOn; |
| unsigned char numberOfTemporalLayers; |
| bool denoisingOn; |
| bool frameDroppingOn; |
| int keyFrameInterval; |
| bool adaptiveQpMode; |
| bool automaticResizeOn; |
| unsigned char numberOfSpatialLayers; |
| bool flexibleMode; |
| }; |
| |
| // 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 |
| |
| // H264 specific. |
| struct VideoCodecH264 { |
| bool frameDroppingOn; |
| int keyFrameInterval; |
| // These are NULL/0 if not externally negotiated. |
| const uint8_t* spsData; |
| size_t spsLen; |
| const uint8_t* ppsData; |
| size_t ppsLen; |
| H264::Profile profile; |
| }; |
| |
| // Video codec types |
| enum VideoCodecType { |
| kVideoCodecVP8, |
| kVideoCodecVP9, |
| kVideoCodecH264, |
| kVideoCodecI420, |
| kVideoCodecRED, |
| kVideoCodecULPFEC, |
| kVideoCodecFlexfec, |
| kVideoCodecGeneric, |
| kVideoCodecUnknown |
| }; |
| |
| // Translates from name of codec to codec type and vice versa. |
| const char* CodecTypeToPayloadString(VideoCodecType type); |
| VideoCodecType PayloadStringToCodecType(const std::string& name); |
| // TODO(kthelgason): Remove these methods once upstream projects |
| // have been updated. |
| rtc::Optional<const char*> CodecTypeToPayloadName(VideoCodecType type); |
| rtc::Optional<VideoCodecType> PayloadNameToCodecType(const std::string& name); |
| |
| union VideoCodecUnion { |
| VideoCodecVP8 VP8; |
| VideoCodecVP9 VP9; |
| VideoCodecH264 H264; |
| }; |
| |
| // Simulcast is when the same stream is encoded multiple times with different |
| // settings such as resolution. |
| struct SimulcastStream { |
| unsigned short width; |
| unsigned short height; |
| unsigned char numberOfTemporalLayers; |
| unsigned int maxBitrate; // kilobits/sec. |
| unsigned int targetBitrate; // kilobits/sec. |
| unsigned int minBitrate; // kilobits/sec. |
| unsigned int qpMax; // minimum quality |
| }; |
| |
| struct SpatialLayer { |
| int scaling_factor_num; |
| int scaling_factor_den; |
| int target_bitrate_bps; |
| // TODO(ivica): Add max_quantizer and min_quantizer? |
| }; |
| |
| enum VideoCodecMode { kRealtimeVideo, kScreensharing }; |
| |
| // Common video codec properties |
| class VideoCodec { |
| public: |
| VideoCodec(); |
| |
| // Public variables. TODO(hta): Make them private with accessors. |
| VideoCodecType codecType; |
| char plName[kPayloadNameSize]; |
| unsigned char plType; |
| |
| unsigned short width; |
| unsigned short height; |
| |
| unsigned int startBitrate; // kilobits/sec. |
| unsigned int maxBitrate; // kilobits/sec. |
| unsigned int minBitrate; // kilobits/sec. |
| unsigned int targetBitrate; // kilobits/sec. |
| |
| uint32_t maxFramerate; |
| |
| unsigned int qpMax; |
| unsigned char numberOfSimulcastStreams; |
| SimulcastStream simulcastStream[kMaxSimulcastStreams]; |
| SpatialLayer spatialLayers[kMaxSpatialLayers]; |
| |
| VideoCodecMode mode; |
| bool expect_encode_from_texture; |
| |
| // Timing frames configuration. There is delay of delay_ms between two |
| // consequent timing frames, excluding outliers. Frame is always made a |
| // timing frame if it's at least outlier_ratio in percent of "ideal" average |
| // frame given bitrate and framerate, i.e. if it's bigger than |
| // |outlier_ratio / 100.0 * bitrate_bps / fps| in bits. This way, timing |
| // frames will not be sent too often usually. Yet large frames will always |
| // have timing information for debug purposes because they are more likely to |
| // cause extra delays. |
| struct TimingFrameTriggerThresholds { |
| int64_t delay_ms; |
| uint16_t outlier_ratio_percent; |
| } timing_frame_thresholds; |
| |
| bool operator==(const VideoCodec& other) const = delete; |
| bool operator!=(const VideoCodec& other) const = delete; |
| |
| // Accessors for codec specific information. |
| // There is a const version of each that returns a reference, |
| // and a non-const version that returns a pointer, in order |
| // to allow modification of the parameters. |
| VideoCodecVP8* VP8(); |
| const VideoCodecVP8& VP8() const; |
| VideoCodecVP9* VP9(); |
| const VideoCodecVP9& VP9() const; |
| VideoCodecH264* H264(); |
| const VideoCodecH264& H264() const; |
| |
| private: |
| // TODO(hta): Consider replacing the union with a pointer type. |
| // This will allow removing the VideoCodec* types from this file. |
| VideoCodecUnion codec_specific_; |
| }; |
| |
| class BitrateAllocation { |
| public: |
| static const uint32_t kMaxBitrateBps; |
| BitrateAllocation(); |
| |
| bool SetBitrate(size_t spatial_index, |
| size_t temporal_index, |
| uint32_t bitrate_bps); |
| |
| uint32_t GetBitrate(size_t spatial_index, size_t temporal_index) const; |
| |
| // Get the sum of all the temporal layer for a specific spatial layer. |
| uint32_t GetSpatialLayerSum(size_t spatial_index) const; |
| |
| uint32_t get_sum_bps() const { return sum_; } // Sum of all bitrates. |
| uint32_t get_sum_kbps() const { return (sum_ + 500) / 1000; } |
| |
| inline bool operator==(const BitrateAllocation& other) const { |
| return memcmp(bitrates_, other.bitrates_, sizeof(bitrates_)) == 0; |
| } |
| inline bool operator!=(const BitrateAllocation& other) const { |
| return !(*this == other); |
| } |
| |
| // Expensive, please use only in tests. |
| std::string ToString() const; |
| std::ostream& operator<<(std::ostream& os) const; |
| |
| private: |
| uint32_t sum_; |
| uint32_t bitrates_[kMaxSpatialLayers][kMaxTemporalStreams]; |
| }; |
| |
| // 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; |
| }; |
| |
| // This structure will have the information about when packet is actually |
| // received by socket. |
| struct PacketTime { |
| PacketTime() : timestamp(-1), not_before(-1) {} |
| PacketTime(int64_t timestamp, int64_t not_before) |
| : timestamp(timestamp), not_before(not_before) {} |
| |
| int64_t timestamp; // Receive time after socket delivers the data. |
| int64_t not_before; // Earliest possible time the data could have arrived, |
| // indicating the potential error in the |timestamp| |
| // value,in case the system is busy. |
| // For example, the time of the last select() call. |
| // If unknown, this value will be set to zero. |
| }; |
| |
| // 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; |
| }; |
| |
| // Class to represent the value of RTP header extensions that are |
| // variable-length strings (e.g., RtpStreamId and RtpMid). |
| // Unlike std::string, it can be copied with memcpy and cleared with memset. |
| // |
| // Empty value represents unset header extension (use empty() to query). |
| class StringRtpHeaderExtension { |
| public: |
| // String RTP header extensions are limited to 16 bytes because it is the |
| // maximum length that can be encoded with one-byte header extensions. |
| static constexpr size_t kMaxSize = 16; |
| |
| static bool IsLegalName(rtc::ArrayView<const char> name); |
| |
| StringRtpHeaderExtension() { value_[0] = 0; } |
| explicit StringRtpHeaderExtension(rtc::ArrayView<const char> value) { |
| Set(value.data(), value.size()); |
| } |
| StringRtpHeaderExtension(const StringRtpHeaderExtension&) = default; |
| StringRtpHeaderExtension& operator=(const StringRtpHeaderExtension&) = |
| default; |
| |
| bool empty() const { return value_[0] == 0; } |
| const char* data() const { return value_; } |
| size_t size() const { return strnlen(value_, kMaxSize); } |
| |
| void Set(rtc::ArrayView<const uint8_t> value) { |
| Set(reinterpret_cast<const char*>(value.data()), value.size()); |
| } |
| void Set(const char* data, size_t size); |
| |
| friend bool operator==(const StringRtpHeaderExtension& lhs, |
| const StringRtpHeaderExtension& rhs) { |
| return strncmp(lhs.value_, rhs.value_, kMaxSize) == 0; |
| } |
| friend bool operator!=(const StringRtpHeaderExtension& lhs, |
| const StringRtpHeaderExtension& rhs) { |
| return !(lhs == rhs); |
| } |
| |
| private: |
| char value_[kMaxSize]; |
| }; |
| |
| // StreamId represents RtpStreamId which is a string. |
| typedef StringRtpHeaderExtension StreamId; |
| |
| // Mid represents RtpMid which is a string. |
| typedef StringRtpHeaderExtension Mid; |
| |
| struct RTPHeaderExtension { |
| RTPHeaderExtension(); |
| |
| bool hasTransmissionTimeOffset; |
| int32_t transmissionTimeOffset; |
| bool hasAbsoluteSendTime; |
| uint32_t absoluteSendTime; |
| bool hasTransportSequenceNumber; |
| uint16_t transportSequenceNumber; |
| |
| // Audio Level includes both level in dBov and voiced/unvoiced bit. See: |
| // https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/ |
| bool hasAudioLevel; |
| bool voiceActivity; |
| uint8_t audioLevel; |
| |
| // For Coordination of Video Orientation. See |
| // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/ |
| // ts_126114v120700p.pdf |
| bool hasVideoRotation; |
| VideoRotation videoRotation; |
| |
| // TODO(ilnik): Refactor this and one above to be rtc::Optional() and remove |
| // a corresponding bool flag. |
| bool hasVideoContentType; |
| VideoContentType videoContentType; |
| |
| bool has_video_timing; |
| VideoSendTiming video_timing; |
| |
| PlayoutDelay playout_delay = {-1, -1}; |
| |
| // For identification of a stream when ssrc is not signaled. See |
| // https://tools.ietf.org/html/draft-ietf-avtext-rid-09 |
| // TODO(danilchap): Update url from draft to release version. |
| StreamId stream_id; |
| StreamId repaired_stream_id; |
| |
| // For identifying the media section used to interpret this RTP packet. See |
| // https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-38 |
| Mid mid; |
| }; |
| |
| struct RTPHeader { |
| RTPHeader(); |
| |
| bool markerBit; |
| uint8_t payloadType; |
| uint16_t sequenceNumber; |
| uint32_t timestamp; |
| uint32_t ssrc; |
| uint8_t numCSRCs; |
| uint32_t arrOfCSRCs[kRtpCsrcSize]; |
| size_t paddingLength; |
| size_t headerLength; |
| int payload_type_frequency; |
| RTPHeaderExtension extension; |
| }; |
| |
| struct RtpPacketCounter { |
| RtpPacketCounter() |
| : header_bytes(0), payload_bytes(0), padding_bytes(0), packets(0) {} |
| |
| void Add(const RtpPacketCounter& other) { |
| header_bytes += other.header_bytes; |
| payload_bytes += other.payload_bytes; |
| padding_bytes += other.padding_bytes; |
| packets += other.packets; |
| } |
| |
| void Subtract(const RtpPacketCounter& other) { |
| RTC_DCHECK_GE(header_bytes, other.header_bytes); |
| header_bytes -= other.header_bytes; |
| RTC_DCHECK_GE(payload_bytes, other.payload_bytes); |
| payload_bytes -= other.payload_bytes; |
| RTC_DCHECK_GE(padding_bytes, other.padding_bytes); |
| padding_bytes -= other.padding_bytes; |
| RTC_DCHECK_GE(packets, other.packets); |
| packets -= other.packets; |
| } |
| |
| void AddPacket(size_t packet_length, const RTPHeader& header) { |
| ++packets; |
| header_bytes += header.headerLength; |
| padding_bytes += header.paddingLength; |
| payload_bytes += |
| packet_length - (header.headerLength + header.paddingLength); |
| } |
| |
| size_t TotalBytes() const { |
| return header_bytes + payload_bytes + padding_bytes; |
| } |
| |
| size_t header_bytes; // Number of bytes used by RTP headers. |
| size_t payload_bytes; // Payload bytes, excluding RTP headers and padding. |
| size_t padding_bytes; // Number of padding bytes. |
| uint32_t packets; // Number of packets. |
| }; |
| |
| // Data usage statistics for a (rtp) stream. |
| struct StreamDataCounters { |
| StreamDataCounters(); |
| |
| void Add(const StreamDataCounters& other) { |
| transmitted.Add(other.transmitted); |
| retransmitted.Add(other.retransmitted); |
| fec.Add(other.fec); |
| 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 StreamDataCounters& other) { |
| transmitted.Subtract(other.transmitted); |
| retransmitted.Subtract(other.retransmitted); |
| fec.Subtract(other.fec); |
| 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); |
| } |
| |
| // Returns the number of bytes corresponding to the actual media payload (i.e. |
| // RTP headers, padding, retransmissions and fec packets are excluded). |
| // Note this function does not have meaning for an RTX stream. |
| size_t MediaPayloadBytes() const { |
| return transmitted.payload_bytes - retransmitted.payload_bytes - |
| fec.payload_bytes; |
| } |
| |
| int64_t first_packet_time_ms; // Time when first packet is sent/received. |
| RtpPacketCounter transmitted; // Number of transmitted packets/bytes. |
| RtpPacketCounter retransmitted; // Number of retransmitted packets/bytes. |
| RtpPacketCounter fec; // Number of redundancy packets/bytes. |
| }; |
| |
| // Callback, called whenever byte/packet counts have been updated. |
| class StreamDataCountersCallback { |
| public: |
| virtual ~StreamDataCountersCallback() {} |
| |
| virtual void DataCountersUpdated(const StreamDataCounters& counters, |
| uint32_t ssrc) = 0; |
| }; |
| |
| // RTCP mode to use. Compound mode is described by RFC 4585 and reduced-size |
| // RTCP mode is described by RFC 5506. |
| enum class RtcpMode { kOff, kCompound, kReducedSize }; |
| |
| enum NetworkState { |
| kNetworkUp, |
| kNetworkDown, |
| }; |
| |
| struct RtpKeepAliveConfig final { |
| // If no packet has been sent for |timeout_interval_ms|, send a keep-alive |
| // packet. The keep-alive packet is an empty (no payload) RTP packet with a |
| // payload type of 20 as long as the other end has not negotiated the use of |
| // this value. If this value has already been negotiated, then some other |
| // unused static payload type from table 5 of RFC 3551 shall be used and set |
| // in |payload_type|. |
| int64_t timeout_interval_ms = -1; |
| uint8_t payload_type = 20; |
| |
| bool operator==(const RtpKeepAliveConfig& o) const { |
| return timeout_interval_ms == o.timeout_interval_ms && |
| payload_type == o.payload_type; |
| } |
| bool operator!=(const RtpKeepAliveConfig& o) const { return !(*this == o); } |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_COMMON_TYPES_H_ |