| Index: voice_engine/channel.h
|
| diff --git a/voice_engine/channel.h b/voice_engine/channel.h
|
| index 5c95a64f1c68ba5291aba3e4255d2ac91d310941..48aeafe17c080e289188330e5c520ac2e3d339e1 100644
|
| --- a/voice_engine/channel.h
|
| +++ b/voice_engine/channel.h
|
| @@ -56,13 +56,37 @@ class RtpPacketReceived;
|
| class RtpRtcp;
|
| class RtpTransportControllerSendInterface;
|
| class TelephoneEventHandler;
|
| -class VoERTPObserver;
|
| class VoiceEngineObserver;
|
|
|
| -struct CallStatistics;
|
| -struct ReportBlock;
|
| struct SenderInfo;
|
|
|
| +struct CallStatistics {
|
| + unsigned short fractionLost;
|
| + unsigned int cumulativeLost;
|
| + unsigned int extendedMax;
|
| + unsigned int jitterSamples;
|
| + int64_t rttMs;
|
| + size_t bytesSent;
|
| + int packetsSent;
|
| + size_t bytesReceived;
|
| + int packetsReceived;
|
| + // The capture ntp time (in local timebase) of the first played out audio
|
| + // frame.
|
| + int64_t capture_start_ntp_time_ms_;
|
| +};
|
| +
|
| +// See section 6.4.2 in http://www.ietf.org/rfc/rfc3550.txt for details.
|
| +struct ReportBlock {
|
| + uint32_t sender_SSRC; // SSRC of sender
|
| + uint32_t source_SSRC;
|
| + uint8_t fraction_lost;
|
| + uint32_t cumulative_num_packets_lost;
|
| + uint32_t extended_highest_sequence_number;
|
| + uint32_t interarrival_jitter;
|
| + uint32_t last_SR_timestamp;
|
| + uint32_t delay_since_last_SR;
|
| +};
|
| +
|
| namespace voe {
|
|
|
| class OutputMixer;
|
| @@ -215,10 +239,8 @@ class Channel
|
| int SendTelephoneEventOutband(int event, int duration_ms);
|
| int SetSendTelephoneEventPayloadType(int payload_type, int payload_frequency);
|
|
|
| - // VoERTP_RTCP
|
| + // RTP+RTCP
|
| int SetLocalSSRC(unsigned int ssrc);
|
| - int GetLocalSSRC(unsigned int& ssrc);
|
| - int GetRemoteSSRC(unsigned int& ssrc);
|
| int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id);
|
| int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id);
|
| void EnableSendTransportSequenceNumber(int id);
|
| @@ -231,13 +253,7 @@ class Channel
|
| void ResetSenderCongestionControlObjects();
|
| void ResetReceiverCongestionControlObjects();
|
| void SetRTCPStatus(bool enable);
|
| - int GetRTCPStatus(bool& enabled);
|
| int SetRTCP_CNAME(const char cName[256]);
|
| - int GetRemoteRTCP_CNAME(char cName[256]);
|
| - int SendApplicationDefinedRTCPPacket(unsigned char subType,
|
| - unsigned int name,
|
| - const char* data,
|
| - unsigned short dataLengthInBytes);
|
| int GetRemoteRTCPReportBlocks(std::vector<ReportBlock>* report_blocks);
|
| int GetRTPStatistics(CallStatistics& stats);
|
| void SetNACKStatus(bool enable, int maxNumberOfPackets);
|
| @@ -343,6 +359,7 @@ class Channel
|
| private:
|
| class ProcessAndEncodeAudioTask;
|
|
|
| + int GetRemoteSSRC(unsigned int& ssrc);
|
| void OnUplinkPacketLossRate(float packet_loss_rate);
|
| bool InputMute() const;
|
| bool OnRtpPacketWithHeader(const uint8_t* received_packet,
|
|
|