Reformatted rtp_rtcp_impl*.
BUG= TEST=Trybots. Review URL: https://webrtc-codereview.appspot.com/1035004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3374 4adac7df-926f-26a2-2b94-8c16560cd09d
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phoglund@webrtc.org
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@ -12,13 +12,14 @@
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
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#include <list>
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#include <vector>
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#include "modules/rtp_rtcp/interface/rtp_rtcp.h"
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#include "modules/rtp_rtcp/source/rtcp_receiver.h"
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#include "modules/rtp_rtcp/source/rtcp_sender.h"
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#include "modules/rtp_rtcp/source/rtp_receiver.h"
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#include "modules/rtp_rtcp/source/rtp_sender.h"
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#include "system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_receiver.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#ifdef MATLAB
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class MatlabPlot;
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@ -28,496 +29,501 @@ namespace webrtc {
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class ModuleRtpRtcpImpl : public RtpRtcp {
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public:
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explicit ModuleRtpRtcpImpl(const RtpRtcp::Configuration& configuration);
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explicit ModuleRtpRtcpImpl(const RtpRtcp::Configuration& configuration);
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virtual ~ModuleRtpRtcpImpl();
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virtual ~ModuleRtpRtcpImpl();
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// returns the number of milliseconds until the module want a worker thread to call Process
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virtual WebRtc_Word32 TimeUntilNextProcess();
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// Returns the number of milliseconds until the module want a worker thread to
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// call Process.
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virtual WebRtc_Word32 TimeUntilNextProcess();
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// Process any pending tasks such as timeouts
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virtual WebRtc_Word32 Process();
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// Process any pending tasks such as timeouts.
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virtual WebRtc_Word32 Process();
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/**
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* Receiver
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*/
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// configure a timeout value
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virtual WebRtc_Word32 SetPacketTimeout(const WebRtc_UWord32 RTPtimeoutMS,
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const WebRtc_UWord32 RTCPtimeoutMS);
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// Receiver part.
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// Set periodic dead or alive notification
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virtual WebRtc_Word32 SetPeriodicDeadOrAliveStatus(
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const bool enable,
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const WebRtc_UWord8 sampleTimeSeconds);
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// Configure a timeout value.
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virtual WebRtc_Word32 SetPacketTimeout(const WebRtc_UWord32 rtp_timeout_ms,
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const WebRtc_UWord32 rtcp_timeout_ms);
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// Get periodic dead or alive notification status
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virtual WebRtc_Word32 PeriodicDeadOrAliveStatus(
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bool &enable,
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WebRtc_UWord8 &sampleTimeSeconds);
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// Set periodic dead or alive notification.
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virtual WebRtc_Word32 SetPeriodicDeadOrAliveStatus(
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const bool enable,
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const WebRtc_UWord8 sample_time_seconds);
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virtual WebRtc_Word32 RegisterReceivePayload(const CodecInst& voiceCodec);
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// Get periodic dead or alive notification status.
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virtual WebRtc_Word32 PeriodicDeadOrAliveStatus(
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bool& enable,
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WebRtc_UWord8& sample_time_seconds);
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virtual WebRtc_Word32 RegisterReceivePayload(const VideoCodec& videoCodec);
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virtual WebRtc_Word32 RegisterReceivePayload(const CodecInst& voice_codec);
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virtual WebRtc_Word32 ReceivePayloadType(const CodecInst& voiceCodec,
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WebRtc_Word8* plType);
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virtual WebRtc_Word32 RegisterReceivePayload(const VideoCodec& video_codec);
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virtual WebRtc_Word32 ReceivePayloadType(const VideoCodec& videoCodec,
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WebRtc_Word8* plType);
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virtual WebRtc_Word32 ReceivePayloadType(const CodecInst& voice_codec,
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WebRtc_Word8* pl_type);
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virtual WebRtc_Word32 DeRegisterReceivePayload(
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const WebRtc_Word8 payloadType);
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virtual WebRtc_Word32 ReceivePayloadType(const VideoCodec& video_codec,
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WebRtc_Word8* pl_type);
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// register RTP header extension
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virtual WebRtc_Word32 RegisterReceiveRtpHeaderExtension(
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const RTPExtensionType type,
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const WebRtc_UWord8 id);
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virtual WebRtc_Word32 DeRegisterReceivePayload(
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const WebRtc_Word8 payload_type);
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virtual WebRtc_Word32 DeregisterReceiveRtpHeaderExtension(
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const RTPExtensionType type);
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// Register RTP header extension.
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virtual WebRtc_Word32 RegisterReceiveRtpHeaderExtension(
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const RTPExtensionType type,
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const WebRtc_UWord8 id);
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// get the currently configured SSRC filter
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virtual WebRtc_Word32 SSRCFilter(WebRtc_UWord32& allowedSSRC) const;
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virtual WebRtc_Word32 DeregisterReceiveRtpHeaderExtension(
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const RTPExtensionType type);
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// set a SSRC to be used as a filter for incoming RTP streams
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virtual WebRtc_Word32 SetSSRCFilter(const bool enable, const WebRtc_UWord32 allowedSSRC);
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// Get the currently configured SSRC filter.
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virtual WebRtc_Word32 SSRCFilter(WebRtc_UWord32& allowed_ssrc) const;
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// Get last received remote timestamp
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virtual WebRtc_UWord32 RemoteTimestamp() const;
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// Set a SSRC to be used as a filter for incoming RTP streams.
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virtual WebRtc_Word32 SetSSRCFilter(const bool enable,
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const WebRtc_UWord32 allowed_ssrc);
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// Get the local time of the last received remote timestamp.
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virtual int64_t LocalTimeOfRemoteTimeStamp() const;
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// Get last received remote timestamp.
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virtual WebRtc_UWord32 RemoteTimestamp() const;
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// Get the current estimated remote timestamp
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virtual WebRtc_Word32 EstimatedRemoteTimeStamp(WebRtc_UWord32& timestamp) const;
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// Get the local time of the last received remote timestamp.
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virtual int64_t LocalTimeOfRemoteTimeStamp() const;
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virtual WebRtc_UWord32 RemoteSSRC() const;
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// Get the current estimated remote timestamp.
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virtual WebRtc_Word32 EstimatedRemoteTimeStamp(
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WebRtc_UWord32& timestamp) const;
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virtual WebRtc_Word32 RemoteCSRCs( WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const ;
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virtual WebRtc_UWord32 RemoteSSRC() const;
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virtual WebRtc_Word32 SetRTXReceiveStatus(const bool enable,
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const WebRtc_UWord32 SSRC);
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virtual WebRtc_Word32 RemoteCSRCs(
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WebRtc_UWord32 arr_of_csrc[kRtpCsrcSize]) const;
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virtual WebRtc_Word32 RTXReceiveStatus(bool* enable,
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WebRtc_UWord32* SSRC) const;
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virtual WebRtc_Word32 SetRTXReceiveStatus(const bool enable,
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const WebRtc_UWord32 ssrc);
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// called by the network module when we receive a packet
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virtual WebRtc_Word32 IncomingPacket( const WebRtc_UWord8* incomingPacket,
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const WebRtc_UWord16 packetLength);
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virtual WebRtc_Word32 RTXReceiveStatus(bool* enable,
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WebRtc_UWord32* ssrc) const;
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/**
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* Sender
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*/
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virtual WebRtc_Word32 RegisterSendPayload(const CodecInst& voiceCodec);
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// Called by the network module when we receive a packet.
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virtual WebRtc_Word32 IncomingPacket(const WebRtc_UWord8* incoming_packet,
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const WebRtc_UWord16 packet_length);
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virtual WebRtc_Word32 RegisterSendPayload(const VideoCodec& videoCodec);
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// Sender part.
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virtual WebRtc_Word32 DeRegisterSendPayload(const WebRtc_Word8 payloadType);
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virtual WebRtc_Word32 RegisterSendPayload(const CodecInst& voice_codec);
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virtual WebRtc_Word8 SendPayloadType() const;
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virtual WebRtc_Word32 RegisterSendPayload(const VideoCodec& video_codec);
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// register RTP header extension
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virtual WebRtc_Word32 RegisterSendRtpHeaderExtension(
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const RTPExtensionType type,
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const WebRtc_UWord8 id);
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virtual WebRtc_Word32 DeRegisterSendPayload(const WebRtc_Word8 payload_type);
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virtual WebRtc_Word32 DeregisterSendRtpHeaderExtension(
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const RTPExtensionType type);
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virtual WebRtc_Word8 SendPayloadType() const;
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// get start timestamp
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virtual WebRtc_UWord32 StartTimestamp() const;
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// Register RTP header extension.
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virtual WebRtc_Word32 RegisterSendRtpHeaderExtension(
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const RTPExtensionType type,
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const WebRtc_UWord8 id);
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// configure start timestamp, default is a random number
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virtual WebRtc_Word32 SetStartTimestamp(const WebRtc_UWord32 timestamp);
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virtual WebRtc_Word32 DeregisterSendRtpHeaderExtension(
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const RTPExtensionType type);
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virtual WebRtc_UWord16 SequenceNumber() const;
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// Get start timestamp.
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virtual WebRtc_UWord32 StartTimestamp() const;
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// Set SequenceNumber, default is a random number
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virtual WebRtc_Word32 SetSequenceNumber(const WebRtc_UWord16 seq);
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// Configure start timestamp, default is a random number.
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virtual WebRtc_Word32 SetStartTimestamp(const WebRtc_UWord32 timestamp);
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virtual WebRtc_UWord32 SSRC() const;
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virtual WebRtc_UWord16 SequenceNumber() const;
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// configure SSRC, default is a random number
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virtual WebRtc_Word32 SetSSRC(const WebRtc_UWord32 ssrc);
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// Set SequenceNumber, default is a random number.
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virtual WebRtc_Word32 SetSequenceNumber(const WebRtc_UWord16 seq);
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virtual WebRtc_Word32 CSRCs( WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const ;
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virtual WebRtc_UWord32 SSRC() const;
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virtual WebRtc_Word32 SetCSRCs( const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize],
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const WebRtc_UWord8 arrLength);
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// Configure SSRC, default is a random number.
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virtual WebRtc_Word32 SetSSRC(const WebRtc_UWord32 ssrc);
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virtual WebRtc_Word32 SetCSRCStatus(const bool include);
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virtual WebRtc_Word32 CSRCs(WebRtc_UWord32 arr_of_csrc[kRtpCsrcSize]) const;
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virtual WebRtc_UWord32 PacketCountSent() const;
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virtual WebRtc_Word32 SetCSRCs(const WebRtc_UWord32 arr_of_csrc[kRtpCsrcSize],
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const WebRtc_UWord8 arr_length);
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virtual int CurrentSendFrequencyHz() const;
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virtual WebRtc_Word32 SetCSRCStatus(const bool include);
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virtual WebRtc_UWord32 ByteCountSent() const;
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virtual WebRtc_UWord32 PacketCountSent() const;
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virtual WebRtc_Word32 SetRTXSendStatus(const bool enable,
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const bool setSSRC,
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const WebRtc_UWord32 SSRC);
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virtual int CurrentSendFrequencyHz() const;
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virtual WebRtc_Word32 RTXSendStatus(bool* enable,
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WebRtc_UWord32* SSRC) const;
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virtual WebRtc_UWord32 ByteCountSent() const;
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// sends kRtcpByeCode when going from true to false
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virtual WebRtc_Word32 SetSendingStatus(const bool sending);
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virtual WebRtc_Word32 SetRTXSendStatus(const bool enable,
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const bool set_ssrc,
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const WebRtc_UWord32 ssrc);
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virtual bool Sending() const;
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virtual WebRtc_Word32 RTXSendStatus(bool* enable,
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WebRtc_UWord32* ssrc) const;
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// Drops or relays media packets
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virtual WebRtc_Word32 SetSendingMediaStatus(const bool sending);
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// Sends kRtcpByeCode when going from true to false.
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virtual WebRtc_Word32 SetSendingStatus(const bool sending);
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virtual bool SendingMedia() const;
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virtual bool Sending() const;
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// Used by the codec module to deliver a video or audio frame for packetization
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virtual WebRtc_Word32 SendOutgoingData(
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const FrameType frameType,
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const WebRtc_Word8 payloadType,
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const WebRtc_UWord32 timeStamp,
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int64_t capture_time_ms,
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord32 payloadSize,
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const RTPFragmentationHeader* fragmentation = NULL,
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const RTPVideoHeader* rtpVideoHdr = NULL);
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// Drops or relays media packets.
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virtual WebRtc_Word32 SetSendingMediaStatus(const bool sending);
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virtual void TimeToSendPacket(uint32_t ssrc, uint16_t sequence_number,
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int64_t capture_time_ms);
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/*
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* RTCP
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*/
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virtual bool SendingMedia() const;
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// Get RTCP status
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virtual RTCPMethod RTCP() const;
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// Used by the codec module to deliver a video or audio frame for
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// packetization.
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virtual WebRtc_Word32 SendOutgoingData(
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const FrameType frame_type,
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const WebRtc_Word8 payload_type,
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const WebRtc_UWord32 time_stamp,
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int64_t capture_time_ms,
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const WebRtc_UWord8* payload_data,
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const WebRtc_UWord32 payload_size,
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const RTPFragmentationHeader* fragmentation = NULL,
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const RTPVideoHeader* rtp_video_hdr = NULL);
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// configure RTCP status i.e on/off
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virtual WebRtc_Word32 SetRTCPStatus(const RTCPMethod method);
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virtual void TimeToSendPacket(uint32_t ssrc, uint16_t sequence_number,
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int64_t capture_time_ms);
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// RTCP part.
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// Set RTCP CName
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virtual WebRtc_Word32 SetCNAME(const char cName[RTCP_CNAME_SIZE]);
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// Get RTCP status.
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virtual RTCPMethod RTCP() const;
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// Get RTCP CName
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virtual WebRtc_Word32 CNAME(char cName[RTCP_CNAME_SIZE]);
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// Configure RTCP status i.e on/off.
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virtual WebRtc_Word32 SetRTCPStatus(const RTCPMethod method);
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// Get remote CName
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virtual WebRtc_Word32 RemoteCNAME(const WebRtc_UWord32 remoteSSRC,
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char cName[RTCP_CNAME_SIZE]) const;
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// Set RTCP CName.
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virtual WebRtc_Word32 SetCNAME(const char c_name[RTCP_CNAME_SIZE]);
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// Get remote NTP
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virtual WebRtc_Word32 RemoteNTP(WebRtc_UWord32 *ReceivedNTPsecs,
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WebRtc_UWord32 *ReceivedNTPfrac,
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WebRtc_UWord32 *RTCPArrivalTimeSecs,
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WebRtc_UWord32 *RTCPArrivalTimeFrac,
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WebRtc_UWord32 *rtcp_timestamp) const;
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// Get RTCP CName.
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virtual WebRtc_Word32 CNAME(char c_name[RTCP_CNAME_SIZE]);
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virtual WebRtc_Word32 AddMixedCNAME(const WebRtc_UWord32 SSRC,
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const char cName[RTCP_CNAME_SIZE]);
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// Get remote CName.
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virtual WebRtc_Word32 RemoteCNAME(const WebRtc_UWord32 remote_ssrc,
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char c_name[RTCP_CNAME_SIZE]) const;
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virtual WebRtc_Word32 RemoveMixedCNAME(const WebRtc_UWord32 SSRC);
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// Get remote NTP.
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virtual WebRtc_Word32 RemoteNTP(WebRtc_UWord32* received_ntp_secs,
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WebRtc_UWord32* received_ntp_frac,
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WebRtc_UWord32* rtcp_arrival_time_secs,
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WebRtc_UWord32* rtcp_arrival_time_frac,
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WebRtc_UWord32* rtcp_timestamp) const;
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// Get RoundTripTime
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virtual WebRtc_Word32 RTT(const WebRtc_UWord32 remoteSSRC,
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WebRtc_UWord16* RTT,
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WebRtc_UWord16* avgRTT,
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WebRtc_UWord16* minRTT,
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WebRtc_UWord16* maxRTT) const;
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virtual WebRtc_Word32 AddMixedCNAME(const WebRtc_UWord32 ssrc,
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const char c_name[RTCP_CNAME_SIZE]);
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// Reset RoundTripTime statistics
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virtual WebRtc_Word32 ResetRTT(const WebRtc_UWord32 remoteSSRC);
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virtual WebRtc_Word32 RemoveMixedCNAME(const WebRtc_UWord32 ssrc);
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virtual void SetRtt(uint32_t rtt);
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// Get RoundTripTime.
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virtual WebRtc_Word32 RTT(const WebRtc_UWord32 remote_ssrc,
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WebRtc_UWord16* rtt,
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WebRtc_UWord16* avg_rtt,
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WebRtc_UWord16* min_rtt,
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WebRtc_UWord16* max_rtt) const;
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// Force a send of an RTCP packet
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// normal SR and RR are triggered via the process function
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virtual WebRtc_Word32 SendRTCP(WebRtc_UWord32 rtcpPacketType = kRtcpReport);
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// Reset RoundTripTime statistics.
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virtual WebRtc_Word32 ResetRTT(const WebRtc_UWord32 remote_ssrc);
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// statistics of our localy created statistics of the received RTP stream
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virtual WebRtc_Word32 StatisticsRTP(WebRtc_UWord8 *fraction_lost,
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WebRtc_UWord32 *cum_lost,
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WebRtc_UWord32 *ext_max,
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WebRtc_UWord32 *jitter,
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WebRtc_UWord32 *max_jitter = NULL) const;
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virtual void SetRtt(uint32_t rtt);
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// Reset RTP statistics
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virtual WebRtc_Word32 ResetStatisticsRTP();
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// Force a send of an RTCP packet.
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// Normal SR and RR are triggered via the process function.
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virtual WebRtc_Word32 SendRTCP(WebRtc_UWord32 rtcp_packet_type = kRtcpReport);
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virtual WebRtc_Word32 ResetReceiveDataCountersRTP();
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// Statistics of our locally created statistics of the received RTP stream.
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virtual WebRtc_Word32 StatisticsRTP(WebRtc_UWord8* fraction_lost,
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WebRtc_UWord32* cum_lost,
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WebRtc_UWord32* ext_max,
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WebRtc_UWord32* jitter,
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WebRtc_UWord32* max_jitter = NULL) const;
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virtual WebRtc_Word32 ResetSendDataCountersRTP();
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// Reset RTP statistics.
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virtual WebRtc_Word32 ResetStatisticsRTP();
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// statistics of the amount of data sent and received
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virtual WebRtc_Word32 DataCountersRTP(WebRtc_UWord32 *bytesSent,
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WebRtc_UWord32 *packetsSent,
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WebRtc_UWord32 *bytesReceived,
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WebRtc_UWord32 *packetsReceived) const;
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virtual WebRtc_Word32 ResetReceiveDataCountersRTP();
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virtual WebRtc_Word32 ReportBlockStatistics(
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WebRtc_UWord8 *fraction_lost,
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WebRtc_UWord32 *cum_lost,
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WebRtc_UWord32 *ext_max,
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WebRtc_UWord32 *jitter,
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WebRtc_UWord32 *jitter_transmission_time_offset);
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virtual WebRtc_Word32 ResetSendDataCountersRTP();
|
||||
|
||||
// Get received RTCP report, sender info
|
||||
virtual WebRtc_Word32 RemoteRTCPStat( RTCPSenderInfo* senderInfo);
|
||||
// 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;
|
||||
|
||||
// Get received RTCP report, report block
|
||||
virtual WebRtc_Word32 RemoteRTCPStat(
|
||||
std::vector<RTCPReportBlock>* receiveBlocks) 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);
|
||||
|
||||
// Set received RTCP report block
|
||||
virtual WebRtc_Word32 AddRTCPReportBlock(const WebRtc_UWord32 SSRC,
|
||||
const RTCPReportBlock* receiveBlock);
|
||||
// Get received RTCP report, sender info.
|
||||
virtual WebRtc_Word32 RemoteRTCPStat(RTCPSenderInfo* sender_info);
|
||||
|
||||
virtual WebRtc_Word32 RemoveRTCPReportBlock(const WebRtc_UWord32 SSRC);
|
||||
// Get received RTCP report, report block.
|
||||
virtual WebRtc_Word32 RemoteRTCPStat(
|
||||
std::vector<RTCPReportBlock>* receive_blocks) const;
|
||||
|
||||
/*
|
||||
* (REMB) Receiver Estimated Max Bitrate
|
||||
*/
|
||||
virtual bool REMB() const;
|
||||
// Set received RTCP report block.
|
||||
virtual WebRtc_Word32 AddRTCPReportBlock(
|
||||
const WebRtc_UWord32 ssrc, const RTCPReportBlock* receive_block);
|
||||
|
||||
virtual WebRtc_Word32 SetREMBStatus(const bool enable);
|
||||
virtual WebRtc_Word32 RemoveRTCPReportBlock(const WebRtc_UWord32 ssrc);
|
||||
|
||||
virtual WebRtc_Word32 SetREMBData(const WebRtc_UWord32 bitrate,
|
||||
const WebRtc_UWord8 numberOfSSRC,
|
||||
const WebRtc_UWord32* SSRC);
|
||||
// (REMB) Receiver Estimated Max Bitrate.
|
||||
virtual bool REMB() const;
|
||||
|
||||
/*
|
||||
* (IJ) Extended jitter report.
|
||||
*/
|
||||
virtual bool IJ() const;
|
||||
virtual WebRtc_Word32 SetREMBStatus(const bool enable);
|
||||
|
||||
virtual WebRtc_Word32 SetIJStatus(const bool enable);
|
||||
virtual WebRtc_Word32 SetREMBData(const WebRtc_UWord32 bitrate,
|
||||
const WebRtc_UWord8 number_of_ssrc,
|
||||
const WebRtc_UWord32* ssrc);
|
||||
|
||||
/*
|
||||
* (TMMBR) Temporary Max Media Bit Rate
|
||||
*/
|
||||
virtual bool TMMBR() const ;
|
||||
// (IJ) Extended jitter report.
|
||||
virtual bool IJ() const;
|
||||
|
||||
virtual WebRtc_Word32 SetTMMBRStatus(const bool enable);
|
||||
virtual WebRtc_Word32 SetIJStatus(const bool enable);
|
||||
|
||||
WebRtc_Word32 SetTMMBN(const TMMBRSet* boundingSet);
|
||||
// (TMMBR) Temporary Max Media Bit Rate.
|
||||
virtual bool TMMBR() const;
|
||||
|
||||
virtual WebRtc_UWord16 MaxPayloadLength() const;
|
||||
virtual WebRtc_Word32 SetTMMBRStatus(const bool enable);
|
||||
|
||||
virtual WebRtc_UWord16 MaxDataPayloadLength() const;
|
||||
WebRtc_Word32 SetTMMBN(const TMMBRSet* bounding_set);
|
||||
|
||||
virtual WebRtc_Word32 SetMaxTransferUnit(const WebRtc_UWord16 size);
|
||||
virtual WebRtc_UWord16 MaxPayloadLength() const;
|
||||
|
||||
virtual WebRtc_Word32 SetTransportOverhead(const bool TCP,
|
||||
const bool IPV6,
|
||||
const WebRtc_UWord8 authenticationOverhead = 0);
|
||||
virtual WebRtc_UWord16 MaxDataPayloadLength() const;
|
||||
|
||||
/*
|
||||
* (NACK) Negative acknowledgement
|
||||
*/
|
||||
virtual WebRtc_Word32 SetMaxTransferUnit(const WebRtc_UWord16 size);
|
||||
|
||||
// Is Negative acknowledgement requests on/off?
|
||||
virtual NACKMethod NACK() const ;
|
||||
virtual WebRtc_Word32 SetTransportOverhead(
|
||||
const bool tcp,
|
||||
const bool ipv6,
|
||||
const WebRtc_UWord8 authentication_overhead = 0);
|
||||
|
||||
// Turn negative acknowledgement requests on/off
|
||||
virtual WebRtc_Word32 SetNACKStatus(const NACKMethod method);
|
||||
// (NACK) Negative acknowledgment part.
|
||||
|
||||
virtual int SelectiveRetransmissions() const;
|
||||
// Is Negative acknowledgment requests on/off?
|
||||
virtual NACKMethod NACK() const;
|
||||
|
||||
virtual int SetSelectiveRetransmissions(uint8_t settings);
|
||||
// Turn negative acknowledgment requests on/off.
|
||||
virtual WebRtc_Word32 SetNACKStatus(const NACKMethod method);
|
||||
|
||||
// Send a Negative acknowledgement packet
|
||||
virtual WebRtc_Word32 SendNACK(const WebRtc_UWord16* nackList,
|
||||
const WebRtc_UWord16 size);
|
||||
virtual int SelectiveRetransmissions() const;
|
||||
|
||||
// Store the sent packets, needed to answer to a Negative acknowledgement requests
|
||||
virtual WebRtc_Word32 SetStorePacketsStatus(const bool enable, const WebRtc_UWord16 numberToStore = 200);
|
||||
virtual int SetSelectiveRetransmissions(uint8_t settings);
|
||||
|
||||
/*
|
||||
* (APP) Application specific data
|
||||
*/
|
||||
virtual WebRtc_Word32 SetRTCPApplicationSpecificData(const WebRtc_UWord8 subType,
|
||||
const WebRtc_UWord32 name,
|
||||
const WebRtc_UWord8* data,
|
||||
const WebRtc_UWord16 length);
|
||||
/*
|
||||
* (XR) VOIP metric
|
||||
*/
|
||||
virtual WebRtc_Word32 SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric);
|
||||
// Send a Negative acknowledgment packet.
|
||||
virtual WebRtc_Word32 SendNACK(const WebRtc_UWord16* nack_list,
|
||||
const WebRtc_UWord16 size);
|
||||
|
||||
/*
|
||||
* Audio
|
||||
*/
|
||||
// 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 = 200);
|
||||
|
||||
// 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 packetSizeSamples);
|
||||
// (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);
|
||||
|
||||
// Outband DTMF detection
|
||||
virtual WebRtc_Word32 SetTelephoneEventStatus(const bool enable,
|
||||
const bool forwardToDecoder,
|
||||
const bool detectEndOfTone = false);
|
||||
// (XR) VOIP metric.
|
||||
virtual WebRtc_Word32 SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric);
|
||||
|
||||
// Is outband DTMF turned on/off?
|
||||
virtual bool TelephoneEvent() const;
|
||||
// Audio part.
|
||||
|
||||
// Is forwarding of outband telephone events turned on/off?
|
||||
virtual bool TelephoneEventForwardToDecoder() const;
|
||||
// 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);
|
||||
|
||||
virtual bool SendTelephoneEventActive(WebRtc_Word8& telephoneEvent) const;
|
||||
// Outband DTMF detection.
|
||||
virtual WebRtc_Word32 SetTelephoneEventStatus(
|
||||
const bool enable,
|
||||
const bool forward_to_decoder,
|
||||
const bool detect_end_of_tone = false);
|
||||
|
||||
// Send a TelephoneEvent tone using RFC 2833 (4733)
|
||||
virtual WebRtc_Word32 SendTelephoneEventOutband(const WebRtc_UWord8 key,
|
||||
// Is outband DTMF turned on/off?
|
||||
virtual bool TelephoneEvent() const;
|
||||
|
||||
// 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 payloadType);
|
||||
// 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& payloadType) const;
|
||||
// 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);
|
||||
// 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;
|
||||
// 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 dBov for header-extension-for-audio-level-indication.
|
||||
virtual WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_dBov);
|
||||
// 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
|
||||
*/
|
||||
virtual RtpVideoCodecTypes ReceivedVideoCodec() const;
|
||||
// Video part.
|
||||
|
||||
virtual RtpVideoCodecTypes SendVideoCodec() const;
|
||||
virtual RtpVideoCodecTypes ReceivedVideoCodec() const;
|
||||
|
||||
virtual WebRtc_Word32 SendRTCPSliceLossIndication(const WebRtc_UWord8 pictureID);
|
||||
virtual RtpVideoCodecTypes SendVideoCodec() const;
|
||||
|
||||
// Set method for requestion a new key frame
|
||||
virtual WebRtc_Word32 SetKeyFrameRequestMethod(const KeyFrameRequestMethod method);
|
||||
virtual WebRtc_Word32 SendRTCPSliceLossIndication(
|
||||
const WebRtc_UWord8 picture_id);
|
||||
|
||||
// send a request for a keyframe
|
||||
virtual WebRtc_Word32 RequestKeyFrame();
|
||||
// Set method for requestion a new key frame.
|
||||
virtual WebRtc_Word32 SetKeyFrameRequestMethod(
|
||||
const KeyFrameRequestMethod method);
|
||||
|
||||
virtual WebRtc_Word32 SetCameraDelay(const WebRtc_Word32 delayMS);
|
||||
// Send a request for a keyframe.
|
||||
virtual WebRtc_Word32 RequestKeyFrame();
|
||||
|
||||
virtual void SetTargetSendBitrate(const WebRtc_UWord32 bitrate);
|
||||
virtual WebRtc_Word32 SetCameraDelay(const WebRtc_Word32 delay_ms);
|
||||
|
||||
virtual WebRtc_Word32 SetGenericFECStatus(const bool enable,
|
||||
const WebRtc_UWord8 payloadTypeRED,
|
||||
const WebRtc_UWord8 payloadTypeFEC);
|
||||
virtual void SetTargetSendBitrate(const WebRtc_UWord32 bitrate);
|
||||
|
||||
virtual WebRtc_Word32 GenericFECStatus(bool& enable,
|
||||
WebRtc_UWord8& payloadTypeRED,
|
||||
WebRtc_UWord8& payloadTypeFEC);
|
||||
virtual WebRtc_Word32 SetGenericFECStatus(
|
||||
const bool enable,
|
||||
const WebRtc_UWord8 payload_type_red,
|
||||
const WebRtc_UWord8 payload_type_fec);
|
||||
|
||||
virtual WebRtc_Word32 SetFecParameters(
|
||||
const FecProtectionParams* delta_params,
|
||||
const FecProtectionParams* key_params);
|
||||
virtual WebRtc_Word32 GenericFECStatus(
|
||||
bool& enable,
|
||||
WebRtc_UWord8& payload_type_red,
|
||||
WebRtc_UWord8& payload_type_fec);
|
||||
|
||||
virtual WebRtc_Word32 LastReceivedNTP(WebRtc_UWord32& NTPsecs,
|
||||
WebRtc_UWord32& NTPfrac,
|
||||
WebRtc_UWord32& remoteSR);
|
||||
virtual WebRtc_Word32 SetFecParameters(
|
||||
const FecProtectionParams* delta_params,
|
||||
const FecProtectionParams* key_params);
|
||||
|
||||
virtual WebRtc_Word32 BoundingSet(bool &tmmbrOwner,
|
||||
TMMBRSet*& boundingSetRec);
|
||||
virtual WebRtc_Word32 LastReceivedNTP(WebRtc_UWord32& NTPsecs,
|
||||
WebRtc_UWord32& NTPfrac,
|
||||
WebRtc_UWord32& remote_sr);
|
||||
|
||||
virtual void BitrateSent(WebRtc_UWord32* totalRate,
|
||||
WebRtc_UWord32* videoRate,
|
||||
WebRtc_UWord32* fecRate,
|
||||
WebRtc_UWord32* nackRate) const;
|
||||
virtual WebRtc_Word32 BoundingSet(bool& tmmbr_owner,
|
||||
TMMBRSet*& bounding_set_rec);
|
||||
|
||||
virtual int EstimatedReceiveBandwidth(
|
||||
WebRtc_UWord32* available_bandwidth) const;
|
||||
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 int EstimatedReceiveBandwidth(
|
||||
WebRtc_UWord32* available_bandwidth) const;
|
||||
|
||||
virtual WebRtc_UWord32 SendTimeOfSendReport(const WebRtc_UWord32 sendReport);
|
||||
virtual void SetRemoteSSRC(const WebRtc_UWord32 ssrc);
|
||||
|
||||
// good state of RTP receiver inform sender
|
||||
virtual WebRtc_Word32 SendRTCPReferencePictureSelection(const WebRtc_UWord64 pictureID);
|
||||
virtual WebRtc_UWord32 SendTimeOfSendReport(const WebRtc_UWord32 send_report);
|
||||
|
||||
void OnReceivedTMMBR();
|
||||
// Good state of RTP receiver inform sender.
|
||||
virtual WebRtc_Word32 SendRTCPReferencePictureSelection(
|
||||
const WebRtc_UWord64 picture_id);
|
||||
|
||||
// bad state of RTP receiver request a keyframe
|
||||
void OnRequestIntraFrame();
|
||||
void OnReceivedTMMBR();
|
||||
|
||||
// received a request for a new SLI
|
||||
void OnReceivedSliceLossIndication(const WebRtc_UWord8 pictureID);
|
||||
// Bad state of RTP receiver request a keyframe.
|
||||
void OnRequestIntraFrame();
|
||||
|
||||
// received a new refereence frame
|
||||
void OnReceivedReferencePictureSelectionIndication(
|
||||
const WebRtc_UWord64 pitureID);
|
||||
// Received a request for a new SLI.
|
||||
void OnReceivedSliceLossIndication(const WebRtc_UWord8 picture_id);
|
||||
|
||||
void OnReceivedNACK(const WebRtc_UWord16 nackSequenceNumbersLength,
|
||||
const WebRtc_UWord16* nackSequenceNumbers);
|
||||
// Received a new reference frame.
|
||||
void OnReceivedReferencePictureSelectionIndication(
|
||||
const WebRtc_UWord64 picture_id);
|
||||
|
||||
void OnRequestSendReport();
|
||||
void OnReceivedNACK(const WebRtc_UWord16 nack_sequence_numbers_length,
|
||||
const WebRtc_UWord16* nack_sequence_numbers);
|
||||
|
||||
// Following function is only called when constructing the object so no
|
||||
// need to worry about data race.
|
||||
void OwnsClock() { _owns_clock = true; }
|
||||
void OnRequestSendReport();
|
||||
|
||||
protected:
|
||||
void RegisterChildModule(RtpRtcp* module);
|
||||
// Following function is only called when constructing the object so no
|
||||
// need to worry about data race.
|
||||
void OwnsClock() {
|
||||
owns_clock_ = true;
|
||||
}
|
||||
|
||||
void DeRegisterChildModule(RtpRtcp* module);
|
||||
protected:
|
||||
void RegisterChildModule(RtpRtcp* module);
|
||||
|
||||
bool UpdateRTCPReceiveInformationTimers();
|
||||
void DeRegisterChildModule(RtpRtcp* module);
|
||||
|
||||
void ProcessDeadOrAliveTimer();
|
||||
bool UpdateRTCPReceiveInformationTimers();
|
||||
|
||||
WebRtc_UWord32 BitrateReceivedNow() const;
|
||||
void ProcessDeadOrAliveTimer();
|
||||
|
||||
// Get remote SequenceNumber
|
||||
WebRtc_UWord16 RemoteSequenceNumber() const;
|
||||
WebRtc_UWord32 BitrateReceivedNow() const;
|
||||
|
||||
// only for internal testing
|
||||
WebRtc_UWord32 LastSendReport(WebRtc_UWord32& lastRTCPTime);
|
||||
// Get remote SequenceNumber.
|
||||
WebRtc_UWord16 RemoteSequenceNumber() const;
|
||||
|
||||
RTPSender _rtpSender;
|
||||
RTPReceiver _rtpReceiver;
|
||||
// Only for internal testing.
|
||||
WebRtc_UWord32 LastSendReport(WebRtc_UWord32& last_rtcptime);
|
||||
|
||||
RTCPSender _rtcpSender;
|
||||
RTCPReceiver _rtcpReceiver;
|
||||
RTPSender rtp_sender_;
|
||||
RTPReceiver rtp_receiver_;
|
||||
|
||||
bool _owns_clock;
|
||||
RtpRtcpClock& _clock;
|
||||
private:
|
||||
int64_t RtcpReportInterval();
|
||||
RTCPSender rtcp_sender_;
|
||||
RTCPReceiver rtcp_receiver_;
|
||||
|
||||
WebRtc_Word32 _id;
|
||||
const bool _audio;
|
||||
bool _collisionDetected;
|
||||
WebRtc_Word64 _lastProcessTime;
|
||||
WebRtc_Word64 _lastBitrateProcessTime;
|
||||
WebRtc_Word64 _lastPacketTimeoutProcessTime;
|
||||
WebRtc_UWord16 _packetOverHead;
|
||||
bool owns_clock_;
|
||||
RtpRtcpClock& clock_;
|
||||
|
||||
scoped_ptr<CriticalSectionWrapper> _criticalSectionModulePtrs;
|
||||
scoped_ptr<CriticalSectionWrapper> _criticalSectionModulePtrsFeedback;
|
||||
ModuleRtpRtcpImpl* _defaultModule;
|
||||
std::list<ModuleRtpRtcpImpl*> _childModules;
|
||||
private:
|
||||
int64_t RtcpReportInterval();
|
||||
|
||||
// Dead or alive
|
||||
bool _deadOrAliveActive;
|
||||
WebRtc_UWord32 _deadOrAliveTimeoutMS;
|
||||
WebRtc_Word64 _deadOrAliveLastTimer;
|
||||
// send side
|
||||
NACKMethod _nackMethod;
|
||||
WebRtc_UWord32 _nackLastTimeSent;
|
||||
WebRtc_UWord16 _nackLastSeqNumberSent;
|
||||
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_UWord16 packet_overhead_;
|
||||
|
||||
bool _simulcast;
|
||||
VideoCodec _sendVideoCodec;
|
||||
KeyFrameRequestMethod _keyFrameReqMethod;
|
||||
scoped_ptr<CriticalSectionWrapper> critical_section_module_ptrs_;
|
||||
scoped_ptr<CriticalSectionWrapper> critical_section_module_ptrs_feedback_;
|
||||
ModuleRtpRtcpImpl* default_module_;
|
||||
std::list<ModuleRtpRtcpImpl*> child_modules_;
|
||||
|
||||
RemoteBitrateEstimator* remote_bitrate_;
|
||||
// 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_;
|
||||
WebRtc_UWord16 nack_last_seq_number_sent_;
|
||||
|
||||
RtcpRttObserver* rtt_observer_;
|
||||
bool simulcast_;
|
||||
VideoCodec send_video_codec_;
|
||||
KeyFrameRequestMethod key_frame_req_method_;
|
||||
|
||||
RemoteBitrateEstimator* remote_bitrate_;
|
||||
|
||||
RtcpRttObserver* rtt_observer_;
|
||||
|
||||
#ifdef MATLAB
|
||||
MatlabPlot* _plot1;
|
||||
MatlabPlot* plot1_;
|
||||
#endif
|
||||
};
|
||||
} // namespace webrtc
|
||||
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
|
||||
|
||||
} // namespace webrtc
|
||||
|
||||
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
|
||||
|
||||
Loading…
Reference in New Issue
Block a user