| /* |
| * 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_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_ |
| #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_ |
| |
| #include <list> |
| #include <vector> |
| |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_payload_registry.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_receiver.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h" |
| #include "webrtc/system_wrappers/interface/scoped_ptr.h" |
| |
| #ifdef MATLAB |
| class MatlabPlot; |
| #endif |
| |
| namespace webrtc { |
| |
| class ModuleRtpRtcpImpl : public RtpRtcp { |
| public: |
| explicit ModuleRtpRtcpImpl(const RtpRtcp::Configuration& configuration); |
| |
| virtual ~ModuleRtpRtcpImpl(); |
| |
| // Returns the number of milliseconds until the module want a worker thread to |
| // call Process. |
| virtual WebRtc_Word32 TimeUntilNextProcess(); |
| |
| // Process any pending tasks such as timeouts. |
| virtual WebRtc_Word32 Process(); |
| |
| // Receiver part. |
| |
| // Configure a timeout value. |
| virtual WebRtc_Word32 SetPacketTimeout(const WebRtc_UWord32 rtp_timeout_ms, |
| const WebRtc_UWord32 rtcp_timeout_ms); |
| |
| // Set periodic dead or alive notification. |
| virtual WebRtc_Word32 SetPeriodicDeadOrAliveStatus( |
| const bool enable, |
| const WebRtc_UWord8 sample_time_seconds); |
| |
| // Get periodic dead or alive notification status. |
| virtual WebRtc_Word32 PeriodicDeadOrAliveStatus( |
| bool& enable, |
| WebRtc_UWord8& sample_time_seconds); |
| |
| virtual WebRtc_Word32 RegisterReceivePayload(const CodecInst& voice_codec); |
| |
| virtual WebRtc_Word32 RegisterReceivePayload(const VideoCodec& video_codec); |
| |
| virtual WebRtc_Word32 ReceivePayloadType(const CodecInst& voice_codec, |
| WebRtc_Word8* pl_type); |
| |
| virtual WebRtc_Word32 ReceivePayloadType(const VideoCodec& video_codec, |
| WebRtc_Word8* pl_type); |
| |
| virtual WebRtc_Word32 DeRegisterReceivePayload( |
| const WebRtc_Word8 payload_type); |
| |
| // Register RTP header extension. |
| virtual WebRtc_Word32 RegisterReceiveRtpHeaderExtension( |
| const RTPExtensionType type, |
| const WebRtc_UWord8 id); |
| |
| virtual WebRtc_Word32 DeregisterReceiveRtpHeaderExtension( |
| const RTPExtensionType type); |
| |
| // Get the currently configured SSRC filter. |
| virtual WebRtc_Word32 SSRCFilter(WebRtc_UWord32& allowed_ssrc) const; |
| |
| // Set a SSRC to be used as a filter for incoming RTP streams. |
| virtual WebRtc_Word32 SetSSRCFilter(const bool enable, |
| const WebRtc_UWord32 allowed_ssrc); |
| |
| // Get last received remote timestamp. |
| virtual WebRtc_UWord32 RemoteTimestamp() const; |
| |
| // Get the local time of the last received remote timestamp. |
| virtual int64_t LocalTimeOfRemoteTimeStamp() const; |
| |
| // Get the current estimated remote timestamp. |
| virtual WebRtc_Word32 EstimatedRemoteTimeStamp( |
| WebRtc_UWord32& timestamp) const; |
| |
| virtual WebRtc_UWord32 RemoteSSRC() const; |
| |
| virtual WebRtc_Word32 RemoteCSRCs( |
| WebRtc_UWord32 arr_of_csrc[kRtpCsrcSize]) const; |
| |
| virtual WebRtc_Word32 SetRTXReceiveStatus(const bool enable, |
| const WebRtc_UWord32 ssrc); |
| |
| virtual WebRtc_Word32 RTXReceiveStatus(bool* enable, |
| WebRtc_UWord32* ssrc) const; |
| |
| // Called by the network module when we receive a packet. |
| virtual WebRtc_Word32 IncomingPacket(const WebRtc_UWord8* incoming_packet, |
| const WebRtc_UWord16 packet_length); |
| |
| // Sender part. |
| |
| virtual WebRtc_Word32 RegisterSendPayload(const CodecInst& voice_codec); |
| |
| virtual WebRtc_Word32 RegisterSendPayload(const VideoCodec& video_codec); |
| |
| virtual WebRtc_Word32 DeRegisterSendPayload(const WebRtc_Word8 payload_type); |
| |
| virtual WebRtc_Word8 SendPayloadType() const; |
| |
| // Register RTP header extension. |
| virtual WebRtc_Word32 RegisterSendRtpHeaderExtension( |
| const RTPExtensionType type, |
| const WebRtc_UWord8 id); |
| |
| virtual WebRtc_Word32 DeregisterSendRtpHeaderExtension( |
| const RTPExtensionType type); |
| |
| // Get start timestamp. |
| virtual WebRtc_UWord32 StartTimestamp() const; |
| |
| // Configure start timestamp, default is a random number. |
| virtual WebRtc_Word32 SetStartTimestamp(const WebRtc_UWord32 timestamp); |
| |
| virtual WebRtc_UWord16 SequenceNumber() const; |
| |
| // Set SequenceNumber, default is a random number. |
| virtual WebRtc_Word32 SetSequenceNumber(const WebRtc_UWord16 seq); |
| |
| virtual WebRtc_UWord32 SSRC() const; |
| |
| // Configure SSRC, default is a random number. |
| virtual WebRtc_Word32 SetSSRC(const WebRtc_UWord32 ssrc); |
| |
| virtual WebRtc_Word32 CSRCs(WebRtc_UWord32 arr_of_csrc[kRtpCsrcSize]) const; |
| |
| virtual WebRtc_Word32 SetCSRCs(const WebRtc_UWord32 arr_of_csrc[kRtpCsrcSize], |
| const WebRtc_UWord8 arr_length); |
| |
| virtual WebRtc_Word32 SetCSRCStatus(const bool include); |
| |
| virtual WebRtc_UWord32 PacketCountSent() const; |
| |
| virtual int CurrentSendFrequencyHz() const; |
| |
| virtual WebRtc_UWord32 ByteCountSent() const; |
| |
| virtual WebRtc_Word32 SetRTXSendStatus(const RtxMode mode, |
| const bool set_ssrc, |
| const WebRtc_UWord32 ssrc); |
| |
| virtual WebRtc_Word32 RTXSendStatus(RtxMode* mode, |
| WebRtc_UWord32* ssrc) const; |
| |
| // Sends kRtcpByeCode when going from true to false. |
| virtual WebRtc_Word32 SetSendingStatus(const bool sending); |
| |
| virtual bool Sending() const; |
| |
| // Drops or relays media packets. |
| virtual WebRtc_Word32 SetSendingMediaStatus(const bool sending); |
| |
| virtual bool SendingMedia() const; |
| |
| // Used by the codec module to deliver a video or audio frame for |
| // packetization. |
| virtual WebRtc_Word32 SendOutgoingData( |
| const FrameType frame_type, |
| const WebRtc_Word8 payload_type, |
| const WebRtc_UWord32 time_stamp, |
| int64_t capture_time_ms, |
| const WebRtc_UWord8* payload_data, |
| const WebRtc_UWord32 payload_size, |
| const RTPFragmentationHeader* fragmentation = NULL, |
| const RTPVideoHeader* rtp_video_hdr = NULL); |
| |
| virtual void TimeToSendPacket(uint32_t ssrc, uint16_t sequence_number, |
| int64_t capture_time_ms); |
| // RTCP part. |
| |
| // Get RTCP status. |
| virtual RTCPMethod RTCP() const; |
| |
| // Configure RTCP status i.e on/off. |
| virtual WebRtc_Word32 SetRTCPStatus(const RTCPMethod method); |
| |
| // Set RTCP CName. |
| virtual WebRtc_Word32 SetCNAME(const char c_name[RTCP_CNAME_SIZE]); |
| |
| // Get RTCP CName. |
| virtual WebRtc_Word32 CNAME(char c_name[RTCP_CNAME_SIZE]); |
| |
| // Get remote CName. |
| virtual WebRtc_Word32 RemoteCNAME(const WebRtc_UWord32 remote_ssrc, |
| char c_name[RTCP_CNAME_SIZE]) const; |
| |
| // Get remote NTP. |
| virtual WebRtc_Word32 RemoteNTP(WebRtc_UWord32* received_ntp_secs, |
| WebRtc_UWord32* received_ntp_frac, |
| WebRtc_UWord32* rtcp_arrival_time_secs, |
| WebRtc_UWord32* rtcp_arrival_time_frac, |
| WebRtc_UWord32* rtcp_timestamp) const; |
| |
| virtual WebRtc_Word32 AddMixedCNAME(const WebRtc_UWord32 ssrc, |
| const char c_name[RTCP_CNAME_SIZE]); |
| |
| virtual WebRtc_Word32 RemoveMixedCNAME(const WebRtc_UWord32 ssrc); |
| |
| // Get RoundTripTime. |
| virtual WebRtc_Word32 RTT(const WebRtc_UWord32 remote_ssrc, |
| WebRtc_UWord16* rtt, |
| WebRtc_UWord16* avg_rtt, |
| WebRtc_UWord16* min_rtt, |
| WebRtc_UWord16* max_rtt) const; |
| |
| // Reset RoundTripTime statistics. |
| virtual WebRtc_Word32 ResetRTT(const WebRtc_UWord32 remote_ssrc); |
| |
| virtual void SetRtt(uint32_t rtt); |
| |
| // Force a send of an RTCP packet. |
| // Normal SR and RR are triggered via the process function. |
| virtual WebRtc_Word32 SendRTCP(WebRtc_UWord32 rtcp_packet_type = kRtcpReport); |
| |
| // Statistics of our locally created statistics of the received RTP stream. |
| virtual WebRtc_Word32 StatisticsRTP(WebRtc_UWord8* fraction_lost, |
| WebRtc_UWord32* cum_lost, |
| WebRtc_UWord32* ext_max, |
| WebRtc_UWord32* jitter, |
| WebRtc_UWord32* max_jitter = NULL) const; |
| |
| // Reset RTP statistics. |
| virtual WebRtc_Word32 ResetStatisticsRTP(); |
| |
| virtual WebRtc_Word32 ResetReceiveDataCountersRTP(); |
| |
| virtual WebRtc_Word32 ResetSendDataCountersRTP(); |
| |
| // Statistics of the amount of data sent and received. |
| virtual WebRtc_Word32 DataCountersRTP(WebRtc_UWord32* bytes_sent, |
| WebRtc_UWord32* packets_sent, |
| WebRtc_UWord32* bytes_received, |
| WebRtc_UWord32* packets_received) const; |
| |
| virtual WebRtc_Word32 ReportBlockStatistics( |
| WebRtc_UWord8* fraction_lost, |
| WebRtc_UWord32* cum_lost, |
| WebRtc_UWord32* ext_max, |
| WebRtc_UWord32* jitter, |
| WebRtc_UWord32* jitter_transmission_time_offset); |
| |
| // Get received RTCP report, sender info. |
| virtual WebRtc_Word32 RemoteRTCPStat(RTCPSenderInfo* sender_info); |
| |
| // Get received RTCP report, report block. |
| virtual WebRtc_Word32 RemoteRTCPStat( |
| std::vector<RTCPReportBlock>* receive_blocks) const; |
| |
| // Set received RTCP report block. |
| virtual WebRtc_Word32 AddRTCPReportBlock( |
| const WebRtc_UWord32 ssrc, const RTCPReportBlock* receive_block); |
| |
| virtual WebRtc_Word32 RemoveRTCPReportBlock(const WebRtc_UWord32 ssrc); |
| |
| // (REMB) Receiver Estimated Max Bitrate. |
| virtual bool REMB() const; |
| |
| virtual WebRtc_Word32 SetREMBStatus(const bool enable); |
| |
| virtual WebRtc_Word32 SetREMBData(const WebRtc_UWord32 bitrate, |
| const WebRtc_UWord8 number_of_ssrc, |
| const WebRtc_UWord32* ssrc); |
| |
| // (IJ) Extended jitter report. |
| virtual bool IJ() const; |
| |
| virtual WebRtc_Word32 SetIJStatus(const bool enable); |
| |
| // (TMMBR) Temporary Max Media Bit Rate. |
| virtual bool TMMBR() const; |
| |
| virtual WebRtc_Word32 SetTMMBRStatus(const bool enable); |
| |
| WebRtc_Word32 SetTMMBN(const TMMBRSet* bounding_set); |
| |
| virtual WebRtc_UWord16 MaxPayloadLength() const; |
| |
| virtual WebRtc_UWord16 MaxDataPayloadLength() const; |
| |
| virtual WebRtc_Word32 SetMaxTransferUnit(const WebRtc_UWord16 size); |
| |
| virtual WebRtc_Word32 SetTransportOverhead( |
| const bool tcp, |
| const bool ipv6, |
| const WebRtc_UWord8 authentication_overhead = 0); |
| |
| // (NACK) Negative acknowledgment part. |
| |
| // Is Negative acknowledgment requests on/off? |
| virtual NACKMethod NACK() const; |
| |
| // Turn negative acknowledgment requests on/off. |
| virtual WebRtc_Word32 SetNACKStatus(const NACKMethod method, |
| int max_reordering_threshold); |
| |
| virtual int SelectiveRetransmissions() const; |
| |
| virtual int SetSelectiveRetransmissions(uint8_t settings); |
| |
| // Send a Negative acknowledgment packet. |
| virtual WebRtc_Word32 SendNACK(const WebRtc_UWord16* nack_list, |
| const WebRtc_UWord16 size); |
| |
| // Store the sent packets, needed to answer to a negative acknowledgment |
| // requests. |
| virtual WebRtc_Word32 SetStorePacketsStatus( |
| const bool enable, const WebRtc_UWord16 number_to_store); |
| |
| // (APP) Application specific data. |
| virtual WebRtc_Word32 SetRTCPApplicationSpecificData( |
| const WebRtc_UWord8 sub_type, |
| const WebRtc_UWord32 name, |
| const WebRtc_UWord8* data, |
| const WebRtc_UWord16 length); |
| |
| // (XR) VOIP metric. |
| virtual WebRtc_Word32 SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric); |
| |
| // Audio part. |
| |
| // Set audio packet size, used to determine when it's time to send a DTMF |
| // packet in silence (CNG). |
| virtual WebRtc_Word32 SetAudioPacketSize( |
| const WebRtc_UWord16 packet_size_samples); |
| |
| // Forward DTMFs to decoder for playout. |
| virtual int SetTelephoneEventForwardToDecoder(bool forward_to_decoder); |
| |
| // Is forwarding of outband telephone events turned on/off? |
| virtual bool TelephoneEventForwardToDecoder() const; |
| |
| virtual bool SendTelephoneEventActive(WebRtc_Word8& telephone_event) const; |
| |
| // Send a TelephoneEvent tone using RFC 2833 (4733). |
| virtual WebRtc_Word32 SendTelephoneEventOutband(const WebRtc_UWord8 key, |
| const WebRtc_UWord16 time_ms, |
| const WebRtc_UWord8 level); |
| |
| // Set payload type for Redundant Audio Data RFC 2198. |
| virtual WebRtc_Word32 SetSendREDPayloadType(const WebRtc_Word8 payload_type); |
| |
| // Get payload type for Redundant Audio Data RFC 2198. |
| virtual WebRtc_Word32 SendREDPayloadType(WebRtc_Word8& payload_type) const; |
| |
| // Set status and id for header-extension-for-audio-level-indication. |
| virtual WebRtc_Word32 SetRTPAudioLevelIndicationStatus( |
| const bool enable, const WebRtc_UWord8 id); |
| |
| // Get status and id for header-extension-for-audio-level-indication. |
| virtual WebRtc_Word32 GetRTPAudioLevelIndicationStatus( |
| bool& enable, WebRtc_UWord8& id) const; |
| |
| // Store the audio level in d_bov for header-extension-for-audio-level- |
| // indication. |
| virtual WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_d_bov); |
| |
| // Video part. |
| |
| virtual RtpVideoCodecTypes ReceivedVideoCodec() const; |
| |
| virtual RtpVideoCodecTypes SendVideoCodec() const; |
| |
| virtual WebRtc_Word32 SendRTCPSliceLossIndication( |
| const WebRtc_UWord8 picture_id); |
| |
| // Set method for requestion a new key frame. |
| virtual WebRtc_Word32 SetKeyFrameRequestMethod( |
| const KeyFrameRequestMethod method); |
| |
| // Send a request for a keyframe. |
| virtual WebRtc_Word32 RequestKeyFrame(); |
| |
| virtual WebRtc_Word32 SetCameraDelay(const WebRtc_Word32 delay_ms); |
| |
| virtual void SetTargetSendBitrate(const WebRtc_UWord32 bitrate); |
| |
| virtual WebRtc_Word32 SetGenericFECStatus( |
| const bool enable, |
| const WebRtc_UWord8 payload_type_red, |
| const WebRtc_UWord8 payload_type_fec); |
| |
| virtual WebRtc_Word32 GenericFECStatus( |
| bool& enable, |
| WebRtc_UWord8& payload_type_red, |
| WebRtc_UWord8& payload_type_fec); |
| |
| virtual WebRtc_Word32 SetFecParameters( |
| const FecProtectionParams* delta_params, |
| const FecProtectionParams* key_params); |
| |
| virtual WebRtc_Word32 LastReceivedNTP(WebRtc_UWord32& NTPsecs, |
| WebRtc_UWord32& NTPfrac, |
| WebRtc_UWord32& remote_sr); |
| |
| virtual WebRtc_Word32 BoundingSet(bool& tmmbr_owner, |
| TMMBRSet*& bounding_set_rec); |
| |
| virtual void BitrateSent(WebRtc_UWord32* total_rate, |
| WebRtc_UWord32* video_rate, |
| WebRtc_UWord32* fec_rate, |
| WebRtc_UWord32* nackRate) const; |
| |
| virtual void SetRemoteSSRC(const WebRtc_UWord32 ssrc); |
| |
| virtual WebRtc_UWord32 SendTimeOfSendReport(const WebRtc_UWord32 send_report); |
| |
| // Good state of RTP receiver inform sender. |
| virtual WebRtc_Word32 SendRTCPReferencePictureSelection( |
| const WebRtc_UWord64 picture_id); |
| |
| void OnReceivedTMMBR(); |
| |
| // Bad state of RTP receiver request a keyframe. |
| void OnRequestIntraFrame(); |
| |
| // Received a request for a new SLI. |
| void OnReceivedSliceLossIndication(const WebRtc_UWord8 picture_id); |
| |
| // Received a new reference frame. |
| void OnReceivedReferencePictureSelectionIndication( |
| const WebRtc_UWord64 picture_id); |
| |
| void OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers); |
| |
| void OnRequestSendReport(); |
| |
| protected: |
| void RegisterChildModule(RtpRtcp* module); |
| |
| void DeRegisterChildModule(RtpRtcp* module); |
| |
| bool UpdateRTCPReceiveInformationTimers(); |
| |
| void ProcessDeadOrAliveTimer(); |
| |
| WebRtc_UWord32 BitrateReceivedNow() const; |
| |
| // Get remote SequenceNumber. |
| WebRtc_UWord16 RemoteSequenceNumber() const; |
| |
| // Only for internal testing. |
| WebRtc_UWord32 LastSendReport(WebRtc_UWord32& last_rtcptime); |
| |
| RTPPayloadRegistry rtp_payload_registry_; |
| |
| RTPSender rtp_sender_; |
| scoped_ptr<RTPReceiver> rtp_receiver_; |
| |
| RTCPSender rtcp_sender_; |
| RTCPReceiver rtcp_receiver_; |
| |
| Clock* clock_; |
| |
| private: |
| int64_t RtcpReportInterval(); |
| |
| RTPReceiverAudio* rtp_telephone_event_handler_; |
| |
| WebRtc_Word32 id_; |
| const bool audio_; |
| bool collision_detected_; |
| WebRtc_Word64 last_process_time_; |
| WebRtc_Word64 last_bitrate_process_time_; |
| WebRtc_Word64 last_packet_timeout_process_time_; |
| WebRtc_Word64 last_rtt_process_time_; |
| WebRtc_UWord16 packet_overhead_; |
| |
| scoped_ptr<CriticalSectionWrapper> critical_section_module_ptrs_; |
| scoped_ptr<CriticalSectionWrapper> critical_section_module_ptrs_feedback_; |
| ModuleRtpRtcpImpl* default_module_; |
| std::list<ModuleRtpRtcpImpl*> child_modules_; |
| |
| // Dead or alive. |
| bool dead_or_alive_active_; |
| WebRtc_UWord32 dead_or_alive_timeout_ms_; |
| WebRtc_Word64 dead_or_alive_last_timer_; |
| // Send side |
| NACKMethod nack_method_; |
| WebRtc_UWord32 nack_last_time_sent_full_; |
| WebRtc_UWord16 nack_last_seq_number_sent_; |
| |
| bool simulcast_; |
| VideoCodec send_video_codec_; |
| KeyFrameRequestMethod key_frame_req_method_; |
| |
| RemoteBitrateEstimator* remote_bitrate_; |
| |
| #ifdef MATLAB |
| MatlabPlot* plot1_; |
| #endif |
| |
| RtcpRttObserver* rtt_observer_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_ |
