diff options
Diffstat (limited to 'chromium/third_party/webrtc/common_types.h')
| -rw-r--r-- | chromium/third_party/webrtc/common_types.h | 344 |
1 files changed, 208 insertions, 136 deletions
diff --git a/chromium/third_party/webrtc/common_types.h b/chromium/third_party/webrtc/common_types.h index 3d47b86132e..6892a83f084 100644 --- a/chromium/third_party/webrtc/common_types.h +++ b/chromium/third_party/webrtc/common_types.h @@ -11,6 +11,12 @@ #ifndef WEBRTC_COMMON_TYPES_H_ #define WEBRTC_COMMON_TYPES_H_ +#include <stddef.h> +#include <string.h> + +#include <string> +#include <vector> + #include "webrtc/typedefs.h" #if defined(_MSC_VER) @@ -33,7 +39,7 @@ #define RTP_PAYLOAD_NAME_SIZE 32 -#if defined(WEBRTC_WIN) +#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. @@ -89,7 +95,6 @@ enum TraceModule kTraceAudioDevice = 0x0012, kTraceVideoRenderer = 0x0014, kTraceVideoCapture = 0x0015, - kTraceVideoPreocessing = 0x0016, kTraceRemoteBitrateEstimator = 0x0017, }; @@ -157,71 +162,6 @@ enum FrameType kVideoFrameDelta = 4, // depends on the previus frame }; -// Interface for encrypting and decrypting regular data and rtp/rtcp packets. -// Implement this interface if you wish to provide an encryption scheme to -// the voice or video engines. -class Encryption -{ -public: - // Encrypt the given data. - // - // Args: - // channel: The channel to encrypt data for. - // in_data: The data to encrypt. This data is bytes_in bytes long. - // out_data: The buffer to write the encrypted data to. You may write more - // bytes of encrypted data than what you got as input, up to a maximum - // of webrtc::kViEMaxMtu if you are encrypting in the video engine, or - // webrtc::kVoiceEngineMaxIpPacketSizeBytes for the voice engine. - // bytes_in: The number of bytes in the input buffer. - // bytes_out: The number of bytes written in out_data. - virtual void encrypt( - int channel, - unsigned char* in_data, - unsigned char* out_data, - int bytes_in, - int* bytes_out) = 0; - - // Decrypts the given data. This should reverse the effects of encrypt(). - // - // Args: - // channel_no: The channel to decrypt data for. - // in_data: The data to decrypt. This data is bytes_in bytes long. - // out_data: The buffer to write the decrypted data to. You may write more - // bytes of decrypted data than what you got as input, up to a maximum - // of webrtc::kViEMaxMtu if you are encrypting in the video engine, or - // webrtc::kVoiceEngineMaxIpPacketSizeBytes for the voice engine. - // bytes_in: The number of bytes in the input buffer. - // bytes_out: The number of bytes written in out_data. - virtual void decrypt( - int channel, - unsigned char* in_data, - unsigned char* out_data, - int bytes_in, - int* bytes_out) = 0; - - // Encrypts a RTCP packet. Otherwise, this method has the same contract as - // encrypt(). - virtual void encrypt_rtcp( - int channel, - unsigned char* in_data, - unsigned char* out_data, - int bytes_in, - int* bytes_out) = 0; - - // Decrypts a RTCP packet. Otherwise, this method has the same contract as - // decrypt(). - virtual void decrypt_rtcp( - int channel, - unsigned char* in_data, - unsigned char* out_data, - int bytes_in, - int* bytes_out) = 0; - -protected: - virtual ~Encryption() {} - Encryption() {} -}; - // External transport callback interface class Transport { @@ -257,6 +197,24 @@ class RtcpStatisticsCallback { uint32_t ssrc) = 0; }; +// Statistics for RTCP packet types. +struct RtcpPacketTypeCounter { + RtcpPacketTypeCounter() + : nack_packets(0), + fir_packets(0), + pli_packets(0) {} + + void Add(const RtcpPacketTypeCounter& other) { + nack_packets += other.nack_packets; + fir_packets += other.fir_packets; + pli_packets += other.pli_packets; + } + + uint32_t nack_packets; + uint32_t fir_packets; + uint32_t pli_packets; +}; + // Data usage statistics for a (rtp) stream struct StreamDataCounters { StreamDataCounters() @@ -315,14 +273,26 @@ class FrameCountObserver { // ================================================================== // Each codec supported can be described by this structure. -struct CodecInst -{ - int pltype; - char plname[RTP_PAYLOAD_NAME_SIZE]; - int plfreq; - int pacsize; - int channels; - int rate; // bits/sec unlike {start,min,max}Bitrate elsewhere in this file! +struct CodecInst { + int pltype; + char plname[RTP_PAYLOAD_NAME_SIZE]; + int plfreq; + int pacsize; + int 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); + } }; // RTP @@ -498,6 +468,7 @@ enum AudioLayers kAudioLinuxPulse = 4 }; +// TODO(henrika): to be removed. enum NetEqModes // NetEQ playout configurations { // Optimized trade-off between low delay and jitter robustness for two-way @@ -514,6 +485,7 @@ enum NetEqModes // NetEQ playout configurations kNetEqOff = 3, }; +// TODO(henrika): to be removed. enum OnHoldModes // On Hold direction { kHoldSendAndPlay = 0, // Put both sending and playing in on-hold state. @@ -521,6 +493,7 @@ enum OnHoldModes // On Hold direction kHoldPlayOnly // Put only playing in on-hold state. }; +// TODO(henrika): to be removed. enum AmrMode { kRfc3267BwEfficient = 0, @@ -584,23 +557,34 @@ enum VP8ResilienceMode { }; // VP8 specific -struct VideoCodecVP8 -{ - bool pictureLossIndicationOn; - bool feedbackModeOn; - VideoCodecComplexity complexity; - VP8ResilienceMode resilience; - unsigned char numberOfTemporalLayers; - bool denoisingOn; - bool errorConcealmentOn; - bool automaticResizeOn; - bool frameDroppingOn; - int keyFrameInterval; -}; - -// Unknown specific -struct VideoCodecGeneric -{ +struct VideoCodecVP8 { + bool pictureLossIndicationOn; + bool feedbackModeOn; + VideoCodecComplexity complexity; + VP8ResilienceMode resilience; + unsigned char numberOfTemporalLayers; + bool denoisingOn; + bool errorConcealmentOn; + bool automaticResizeOn; + bool frameDroppingOn; + int keyFrameInterval; + + bool operator==(const VideoCodecVP8& other) const { + return pictureLossIndicationOn == other.pictureLossIndicationOn && + feedbackModeOn == other.feedbackModeOn && + complexity == other.complexity && + resilience == other.resilience && + numberOfTemporalLayers == other.numberOfTemporalLayers && + denoisingOn == other.denoisingOn && + errorConcealmentOn == other.errorConcealmentOn && + automaticResizeOn == other.automaticResizeOn && + frameDroppingOn == other.frameDroppingOn && + keyFrameInterval == other.keyFrameInterval; + } + + bool operator!=(const VideoCodecVP8& other) const { + return !(*this == other); + } }; // Video codec types @@ -617,21 +601,33 @@ enum VideoCodecType union VideoCodecUnion { VideoCodecVP8 VP8; - VideoCodecGeneric Generic; }; // 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 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 + + bool operator==(const SimulcastStream& other) const { + return width == other.width && + height == other.height && + numberOfTemporalLayers == other.numberOfTemporalLayers && + maxBitrate == other.maxBitrate && + targetBitrate == other.targetBitrate && + minBitrate == other.minBitrate && + qpMax == other.qpMax; + } + + bool operator!=(const SimulcastStream& other) const { + return !(*this == other); + } }; enum VideoCodecMode { @@ -640,31 +636,60 @@ enum VideoCodecMode { }; // Common video codec properties -struct VideoCodec -{ - 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 char maxFramerate; - - VideoCodecUnion codecSpecific; - - unsigned int qpMax; - unsigned char numberOfSimulcastStreams; - SimulcastStream simulcastStream[kMaxSimulcastStreams]; - - VideoCodecMode mode; +struct VideoCodec { + 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. + + unsigned char maxFramerate; + + VideoCodecUnion codecSpecific; + + unsigned int qpMax; + unsigned char numberOfSimulcastStreams; + SimulcastStream simulcastStream[kMaxSimulcastStreams]; + + VideoCodecMode mode; + + // When using an external encoder/decoder this allows to pass + // extra options without requiring webrtc to be aware of them. + Config* extra_options; + + bool operator==(const VideoCodec& other) const { + bool ret = codecType == other.codecType && + (STR_CASE_CMP(plName, other.plName) == 0) && + plType == other.plType && + width == other.width && + height == other.height && + startBitrate == other.startBitrate && + maxBitrate == other.maxBitrate && + minBitrate == other.minBitrate && + targetBitrate == other.targetBitrate && + maxFramerate == other.maxFramerate && + qpMax == other.qpMax && + numberOfSimulcastStreams == other.numberOfSimulcastStreams && + mode == other.mode; + if (ret && codecType == kVideoCodecVP8) { + ret &= (codecSpecific.VP8 == other.codecSpecific.VP8); + } + + for (unsigned char i = 0; i < other.numberOfSimulcastStreams && ret; ++i) { + ret &= (simulcastStream[i] == other.simulcastStream[i]); + } + return ret; + } - // When using an external encoder/decoder this allows to pass - // extra options without requiring webrtc to be aware of them. - Config* extra_options; + bool operator!=(const VideoCodec& other) const { + return !(*this == other); + } }; // Bandwidth over-use detector options. These are used to drive @@ -697,20 +722,67 @@ struct OverUseDetectorOptions { // This structure will have the information about when packet is actually // received by socket. struct PacketTime { - PacketTime() : timestamp(-1), max_error_us(-1) {} - PacketTime(int64_t timestamp, int64_t max_error_us) - : timestamp(timestamp), max_error_us(max_error_us) { + 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. +}; + +struct RTPHeaderExtension { + RTPHeaderExtension() + : hasTransmissionTimeOffset(false), + transmissionTimeOffset(0), + hasAbsoluteSendTime(false), + absoluteSendTime(0), + hasAudioLevel(false), + audioLevel(0) {} + + bool hasTransmissionTimeOffset; + int32_t transmissionTimeOffset; + bool hasAbsoluteSendTime; + uint32_t absoluteSendTime; + + // 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; + uint8_t audioLevel; +}; + +struct RTPHeader { + RTPHeader() + : markerBit(false), + payloadType(0), + sequenceNumber(0), + timestamp(0), + ssrc(0), + numCSRCs(0), + paddingLength(0), + headerLength(0), + payload_type_frequency(0), + extension() { + memset(&arrOfCSRCs, 0, sizeof(arrOfCSRCs)); } - int64_t timestamp; // Receive time after socket delivers the data. - int64_t max_error_us; // 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. + bool markerBit; + uint8_t payloadType; + uint16_t sequenceNumber; + uint32_t timestamp; + uint32_t ssrc; + uint8_t numCSRCs; + uint32_t arrOfCSRCs[kRtpCsrcSize]; + uint8_t paddingLength; + uint16_t headerLength; + int payload_type_frequency; + RTPHeaderExtension extension; }; } // namespace webrtc #endif // WEBRTC_COMMON_TYPES_H_ - |