| /* |
| * 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> |
| #include <string.h> |
| #include <ostream> |
| #include <string> |
| #include <vector> |
| |
| #include "api/array_view.h" |
| #include "api/optional.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/deprecation.h" |
| #include "typedefs.h" // NOLINT(build/include) |
| |
| #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; |
| }; |
| |
| // For the deprecated MediaFile module. |
| enum FileFormats { |
| kFileFormatWavFile = 1, |
| 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 { |
| int32_t packets_lost; // Defined as a 24 bit signed integer in RTCP |
| 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; |
| // Stats below correspond to similarly-named fields in the WebRTC stats spec. |
| // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats |
| uint64_t totalSamplesReceived; |
| uint64_t concealedSamples; |
| uint64_t concealmentEvents; |
| uint64_t jitterBufferDelayMs; |
| // Stats below DO NOT correspond directly to anything in the WebRTC stats |
| // 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 { 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 { |
| VideoCodecComplexity complexity; |
| VP8ResilienceMode resilience; |
| unsigned char numberOfTemporalLayers; |
| bool denoisingOn; |
| 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, |
| kVideoCodecMultiplex, |
| kVideoCodecUnknown |
| }; |
| |
| // Translates from name of codec to codec type and vice versa. |
| const char* CodecTypeToPayloadString(VideoCodecType type); |
| VideoCodecType PayloadStringToCodecType(const std::string& name); |
| |
| union VideoCodecUnion { |
| VideoCodecVP8 VP8; |
| VideoCodecVP9 VP9; |
| VideoCodecH264 H264; |
| }; |
| |
| struct SpatialLayer { |
| 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 active; // encoded and sent. |
| }; |
| |
| // Simulcast is when the same stream is encoded multiple times with different |
| // settings such as resolution. |
| typedef SpatialLayer SimulcastStream; |
| |
| enum VideoCodecMode { kRealtimeVideo, kScreensharing }; |
| |
| // Common video codec properties |
| class VideoCodec { |
| public: |
| VideoCodec(); |
| |
| // Public variables. TODO(hta): Make them private with accessors. |
| VideoCodecType codecType; |
| 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; |
| |
| // This enables/disables encoding and sending when there aren't multiple |
| // simulcast streams,by allocating 0 bitrate if inactive. |
| bool active; |
| |
| 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); |
| |
| bool HasBitrate(size_t spatial_index, size_t temporal_index) const; |
| |
| uint32_t GetBitrate(size_t spatial_index, size_t temporal_index) const; |
| |
| // Whether the specific spatial layers has the bitrate set in any of its |
| // temporal layers. |
| bool IsSpatialLayerUsed(size_t spatial_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]; |
| bool has_bitrate_[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; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // COMMON_TYPES_H_ |