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Reformatted rtp_rtcp_impl*.

Browse Source 
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
This commit is contained in:
phoglund@webrtc.org 2013-01-16 10:27:33 +00:00
parent 77a584be1d
commit acfdd96ee3
2 changed files with 1311 additions and 1337 deletions
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1942
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
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webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
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@ -12,13 +12,14 @@
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
#include <list> #include <list>
#include <vector>
#include "modules/rtp_rtcp/interface/rtp_rtcp.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "modules/rtp_rtcp/source/rtcp_receiver.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
#include "modules/rtp_rtcp/source/rtcp_sender.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
#include "modules/rtp_rtcp/source/rtp_receiver.h" #include "webrtc/modules/rtp_rtcp/source/rtp_receiver.h"
#include "modules/rtp_rtcp/source/rtp_sender.h" #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
#include "system_wrappers/interface/scoped_ptr.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h"
#ifdef MATLAB #ifdef MATLAB
class MatlabPlot; class MatlabPlot;
@ -28,496 +29,501 @@ namespace webrtc {
class ModuleRtpRtcpImpl : public RtpRtcp { class ModuleRtpRtcpImpl : public RtpRtcp {
public: public:
explicit ModuleRtpRtcpImpl(const RtpRtcp::Configuration& configuration); explicit ModuleRtpRtcpImpl(const RtpRtcp::Configuration& configuration);
virtual ~ModuleRtpRtcpImpl(); virtual ~ModuleRtpRtcpImpl();
// returns the number of milliseconds until the module want a worker thread to call Process // Returns the number of milliseconds until the module want a worker thread to
virtual WebRtc_Word32 TimeUntilNextProcess(); // call Process.
virtual WebRtc_Word32 TimeUntilNextProcess();
// Process any pending tasks such as timeouts // Process any pending tasks such as timeouts.
virtual WebRtc_Word32 Process(); virtual WebRtc_Word32 Process();
/** // Receiver part.
* Receiver
*/
// configure a timeout value
virtual WebRtc_Word32 SetPacketTimeout(const WebRtc_UWord32 RTPtimeoutMS,
const WebRtc_UWord32 RTCPtimeoutMS);
// Set periodic dead or alive notification // Configure a timeout value.
virtual WebRtc_Word32 SetPeriodicDeadOrAliveStatus( virtual WebRtc_Word32 SetPacketTimeout(const WebRtc_UWord32 rtp_timeout_ms,
const bool enable, const WebRtc_UWord32 rtcp_timeout_ms);
const WebRtc_UWord8 sampleTimeSeconds);
// Get periodic dead or alive notification status // Set periodic dead or alive notification.
virtual WebRtc_Word32 PeriodicDeadOrAliveStatus( virtual WebRtc_Word32 SetPeriodicDeadOrAliveStatus(
bool &enable, const bool enable,
WebRtc_UWord8 &sampleTimeSeconds); const WebRtc_UWord8 sample_time_seconds);
virtual WebRtc_Word32 RegisterReceivePayload(const CodecInst& voiceCodec); // Get periodic dead or alive notification status.
virtual WebRtc_Word32 PeriodicDeadOrAliveStatus(
bool& enable,
WebRtc_UWord8& sample_time_seconds);
virtual WebRtc_Word32 RegisterReceivePayload(const VideoCodec& videoCodec); virtual WebRtc_Word32 RegisterReceivePayload(const CodecInst& voice_codec);
virtual WebRtc_Word32 ReceivePayloadType(const CodecInst& voiceCodec, virtual WebRtc_Word32 RegisterReceivePayload(const VideoCodec& video_codec);
WebRtc_Word8* plType);
virtual WebRtc_Word32 ReceivePayloadType(const VideoCodec& videoCodec, virtual WebRtc_Word32 ReceivePayloadType(const CodecInst& voice_codec,
WebRtc_Word8* plType); WebRtc_Word8* pl_type);
virtual WebRtc_Word32 DeRegisterReceivePayload( virtual WebRtc_Word32 ReceivePayloadType(const VideoCodec& video_codec,
const WebRtc_Word8 payloadType); WebRtc_Word8* pl_type);
// register RTP header extension virtual WebRtc_Word32 DeRegisterReceivePayload(
virtual WebRtc_Word32 RegisterReceiveRtpHeaderExtension( const WebRtc_Word8 payload_type);
const RTPExtensionType type,
const WebRtc_UWord8 id);
virtual WebRtc_Word32 DeregisterReceiveRtpHeaderExtension( // Register RTP header extension.
const RTPExtensionType type); virtual WebRtc_Word32 RegisterReceiveRtpHeaderExtension(
const RTPExtensionType type,
const WebRtc_UWord8 id);
// get the currently configured SSRC filter virtual WebRtc_Word32 DeregisterReceiveRtpHeaderExtension(
virtual WebRtc_Word32 SSRCFilter(WebRtc_UWord32& allowedSSRC) const; const RTPExtensionType type);
// set a SSRC to be used as a filter for incoming RTP streams // Get the currently configured SSRC filter.
virtual WebRtc_Word32 SetSSRCFilter(const bool enable, const WebRtc_UWord32 allowedSSRC); virtual WebRtc_Word32 SSRCFilter(WebRtc_UWord32& allowed_ssrc) const;
// Get last received remote timestamp // Set a SSRC to be used as a filter for incoming RTP streams.
virtual WebRtc_UWord32 RemoteTimestamp() const; virtual WebRtc_Word32 SetSSRCFilter(const bool enable,
const WebRtc_UWord32 allowed_ssrc);
// Get the local time of the last received remote timestamp. // Get last received remote timestamp.
virtual int64_t LocalTimeOfRemoteTimeStamp() const; virtual WebRtc_UWord32 RemoteTimestamp() const;
// Get the current estimated remote timestamp // Get the local time of the last received remote timestamp.
virtual WebRtc_Word32 EstimatedRemoteTimeStamp(WebRtc_UWord32& timestamp) const; virtual int64_t LocalTimeOfRemoteTimeStamp() const;
virtual WebRtc_UWord32 RemoteSSRC() const; // Get the current estimated remote timestamp.
virtual WebRtc_Word32 EstimatedRemoteTimeStamp(
WebRtc_UWord32& timestamp) const;
virtual WebRtc_Word32 RemoteCSRCs( WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const ; virtual WebRtc_UWord32 RemoteSSRC() const;
virtual WebRtc_Word32 SetRTXReceiveStatus(const bool enable, virtual WebRtc_Word32 RemoteCSRCs(
const WebRtc_UWord32 SSRC); WebRtc_UWord32 arr_of_csrc[kRtpCsrcSize]) const;
virtual WebRtc_Word32 RTXReceiveStatus(bool* enable, virtual WebRtc_Word32 SetRTXReceiveStatus(const bool enable,
WebRtc_UWord32* SSRC) const; const WebRtc_UWord32 ssrc);
// called by the network module when we receive a packet virtual WebRtc_Word32 RTXReceiveStatus(bool* enable,
virtual WebRtc_Word32 IncomingPacket( const WebRtc_UWord8* incomingPacket, WebRtc_UWord32* ssrc) const;
const WebRtc_UWord16 packetLength);
/** // Called by the network module when we receive a packet.
* Sender virtual WebRtc_Word32 IncomingPacket(const WebRtc_UWord8* incoming_packet,
*/ const WebRtc_UWord16 packet_length);
virtual WebRtc_Word32 RegisterSendPayload(const CodecInst& voiceCodec);
virtual WebRtc_Word32 RegisterSendPayload(const VideoCodec& videoCodec); // Sender part.
virtual WebRtc_Word32 DeRegisterSendPayload(const WebRtc_Word8 payloadType); virtual WebRtc_Word32 RegisterSendPayload(const CodecInst& voice_codec);
virtual WebRtc_Word8 SendPayloadType() const; virtual WebRtc_Word32 RegisterSendPayload(const VideoCodec& video_codec);
// register RTP header extension virtual WebRtc_Word32 DeRegisterSendPayload(const WebRtc_Word8 payload_type);
virtual WebRtc_Word32 RegisterSendRtpHeaderExtension(
const RTPExtensionType type,
const WebRtc_UWord8 id);
virtual WebRtc_Word32 DeregisterSendRtpHeaderExtension( virtual WebRtc_Word8 SendPayloadType() const;
const RTPExtensionType type);
// get start timestamp // Register RTP header extension.
virtual WebRtc_UWord32 StartTimestamp() const; virtual WebRtc_Word32 RegisterSendRtpHeaderExtension(
const RTPExtensionType type,
const WebRtc_UWord8 id);
// configure start timestamp, default is a random number virtual WebRtc_Word32 DeregisterSendRtpHeaderExtension(
virtual WebRtc_Word32 SetStartTimestamp(const WebRtc_UWord32 timestamp); const RTPExtensionType type);
virtual WebRtc_UWord16 SequenceNumber() const; // Get start timestamp.
virtual WebRtc_UWord32 StartTimestamp() const;
// Set SequenceNumber, default is a random number // Configure start timestamp, default is a random number.
virtual WebRtc_Word32 SetSequenceNumber(const WebRtc_UWord16 seq); virtual WebRtc_Word32 SetStartTimestamp(const WebRtc_UWord32 timestamp);
virtual WebRtc_UWord32 SSRC() const; virtual WebRtc_UWord16 SequenceNumber() const;
// configure SSRC, default is a random number // Set SequenceNumber, default is a random number.
virtual WebRtc_Word32 SetSSRC(const WebRtc_UWord32 ssrc); virtual WebRtc_Word32 SetSequenceNumber(const WebRtc_UWord16 seq);
virtual WebRtc_Word32 CSRCs( WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const ; virtual WebRtc_UWord32 SSRC() const;
virtual WebRtc_Word32 SetCSRCs( const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize], // Configure SSRC, default is a random number.
const WebRtc_UWord8 arrLength); virtual WebRtc_Word32 SetSSRC(const WebRtc_UWord32 ssrc);
virtual WebRtc_Word32 SetCSRCStatus(const bool include); virtual WebRtc_Word32 CSRCs(WebRtc_UWord32 arr_of_csrc[kRtpCsrcSize]) const;
virtual WebRtc_UWord32 PacketCountSent() const; virtual WebRtc_Word32 SetCSRCs(const WebRtc_UWord32 arr_of_csrc[kRtpCsrcSize],
const WebRtc_UWord8 arr_length);
virtual int CurrentSendFrequencyHz() const; virtual WebRtc_Word32 SetCSRCStatus(const bool include);
virtual WebRtc_UWord32 ByteCountSent() const; virtual WebRtc_UWord32 PacketCountSent() const;
virtual WebRtc_Word32 SetRTXSendStatus(const bool enable, virtual int CurrentSendFrequencyHz() const;
const bool setSSRC,
const WebRtc_UWord32 SSRC);
virtual WebRtc_Word32 RTXSendStatus(bool* enable, virtual WebRtc_UWord32 ByteCountSent() const;
WebRtc_UWord32* SSRC) const;
// sends kRtcpByeCode when going from true to false virtual WebRtc_Word32 SetRTXSendStatus(const bool enable,
virtual WebRtc_Word32 SetSendingStatus(const bool sending); const bool set_ssrc,
const WebRtc_UWord32 ssrc);
virtual bool Sending() const; virtual WebRtc_Word32 RTXSendStatus(bool* enable,
WebRtc_UWord32* ssrc) const;
// Drops or relays media packets // Sends kRtcpByeCode when going from true to false.
virtual WebRtc_Word32 SetSendingMediaStatus(const bool sending); virtual WebRtc_Word32 SetSendingStatus(const bool sending);
virtual bool SendingMedia() const; virtual bool Sending() const;
// Used by the codec module to deliver a video or audio frame for packetization // Drops or relays media packets.
virtual WebRtc_Word32 SendOutgoingData( virtual WebRtc_Word32 SetSendingMediaStatus(const bool sending);
const FrameType frameType,
const WebRtc_Word8 payloadType,
const WebRtc_UWord32 timeStamp,
int64_t capture_time_ms,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord32 payloadSize,
const RTPFragmentationHeader* fragmentation = NULL,
const RTPVideoHeader* rtpVideoHdr = NULL);
virtual void TimeToSendPacket(uint32_t ssrc, uint16_t sequence_number, virtual bool SendingMedia() const;
int64_t capture_time_ms);
/*
* RTCP
*/
// Get RTCP status // Used by the codec module to deliver a video or audio frame for
virtual RTCPMethod RTCP() const; // 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);
// configure RTCP status i.e on/off virtual void TimeToSendPacket(uint32_t ssrc, uint16_t sequence_number,
virtual WebRtc_Word32 SetRTCPStatus(const RTCPMethod method); int64_t capture_time_ms);
// RTCP part.
// Set RTCP CName // Get RTCP status.
virtual WebRtc_Word32 SetCNAME(const char cName[RTCP_CNAME_SIZE]); virtual RTCPMethod RTCP() const;
// Get RTCP CName // Configure RTCP status i.e on/off.
virtual WebRtc_Word32 CNAME(char cName[RTCP_CNAME_SIZE]); virtual WebRtc_Word32 SetRTCPStatus(const RTCPMethod method);
// Get remote CName // Set RTCP CName.
virtual WebRtc_Word32 RemoteCNAME(const WebRtc_UWord32 remoteSSRC, virtual WebRtc_Word32 SetCNAME(const char c_name[RTCP_CNAME_SIZE]);
char cName[RTCP_CNAME_SIZE]) const;
// Get remote NTP // Get RTCP CName.
virtual WebRtc_Word32 RemoteNTP(WebRtc_UWord32 *ReceivedNTPsecs, virtual WebRtc_Word32 CNAME(char c_name[RTCP_CNAME_SIZE]);
WebRtc_UWord32 *ReceivedNTPfrac,
WebRtc_UWord32 *RTCPArrivalTimeSecs,
WebRtc_UWord32 *RTCPArrivalTimeFrac,
WebRtc_UWord32 *rtcp_timestamp) const;
virtual WebRtc_Word32 AddMixedCNAME(const WebRtc_UWord32 SSRC, // Get remote CName.
const char cName[RTCP_CNAME_SIZE]); virtual WebRtc_Word32 RemoteCNAME(const WebRtc_UWord32 remote_ssrc,
char c_name[RTCP_CNAME_SIZE]) const;
virtual WebRtc_Word32 RemoveMixedCNAME(const WebRtc_UWord32 SSRC); // 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;
// Get RoundTripTime virtual WebRtc_Word32 AddMixedCNAME(const WebRtc_UWord32 ssrc,
virtual WebRtc_Word32 RTT(const WebRtc_UWord32 remoteSSRC, const char c_name[RTCP_CNAME_SIZE]);
WebRtc_UWord16* RTT,
WebRtc_UWord16* avgRTT,
WebRtc_UWord16* minRTT,
WebRtc_UWord16* maxRTT) const;
// Reset RoundTripTime statistics virtual WebRtc_Word32 RemoveMixedCNAME(const WebRtc_UWord32 ssrc);
virtual WebRtc_Word32 ResetRTT(const WebRtc_UWord32 remoteSSRC);
virtual void SetRtt(uint32_t rtt); // 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;
// Force a send of an RTCP packet // Reset RoundTripTime statistics.
// normal SR and RR are triggered via the process function virtual WebRtc_Word32 ResetRTT(const WebRtc_UWord32 remote_ssrc);
virtual WebRtc_Word32 SendRTCP(WebRtc_UWord32 rtcpPacketType = kRtcpReport);
// statistics of our localy created statistics of the received RTP stream virtual void SetRtt(uint32_t rtt);
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 // Force a send of an RTCP packet.
virtual WebRtc_Word32 ResetStatisticsRTP(); // Normal SR and RR are triggered via the process function.
virtual WebRtc_Word32 SendRTCP(WebRtc_UWord32 rtcp_packet_type = kRtcpReport);
virtual WebRtc_Word32 ResetReceiveDataCountersRTP(); // 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;
virtual WebRtc_Word32 ResetSendDataCountersRTP(); // Reset RTP statistics.
virtual WebRtc_Word32 ResetStatisticsRTP();
// statistics of the amount of data sent and received virtual WebRtc_Word32 ResetReceiveDataCountersRTP();
virtual WebRtc_Word32 DataCountersRTP(WebRtc_UWord32 *bytesSent,
WebRtc_UWord32 *packetsSent,
WebRtc_UWord32 *bytesReceived,
WebRtc_UWord32 *packetsReceived) const;
virtual WebRtc_Word32 ReportBlockStatistics( virtual WebRtc_Word32 ResetSendDataCountersRTP();
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 // Statistics of the amount of data sent and received.
virtual WebRtc_Word32 RemoteRTCPStat( RTCPSenderInfo* senderInfo); 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 ReportBlockStatistics(
virtual WebRtc_Word32 RemoteRTCPStat( WebRtc_UWord8* fraction_lost,
std::vector<RTCPReportBlock>* receiveBlocks) const; WebRtc_UWord32* cum_lost,
WebRtc_UWord32* ext_max,
WebRtc_UWord32* jitter,
WebRtc_UWord32* jitter_transmission_time_offset);
// Set received RTCP report block // Get received RTCP report, sender info.
virtual WebRtc_Word32 AddRTCPReportBlock(const WebRtc_UWord32 SSRC, virtual WebRtc_Word32 RemoteRTCPStat(RTCPSenderInfo* sender_info);
const RTCPReportBlock* receiveBlock);
virtual WebRtc_Word32 RemoveRTCPReportBlock(const WebRtc_UWord32 SSRC); // Get received RTCP report, report block.
virtual WebRtc_Word32 RemoteRTCPStat(
std::vector<RTCPReportBlock>* receive_blocks) const;
/* // Set received RTCP report block.
* (REMB) Receiver Estimated Max Bitrate virtual WebRtc_Word32 AddRTCPReportBlock(
*/ const WebRtc_UWord32 ssrc, const RTCPReportBlock* receive_block);
virtual bool REMB() const;
virtual WebRtc_Word32 SetREMBStatus(const bool enable); virtual WebRtc_Word32 RemoveRTCPReportBlock(const WebRtc_UWord32 ssrc);
virtual WebRtc_Word32 SetREMBData(const WebRtc_UWord32 bitrate, // (REMB) Receiver Estimated Max Bitrate.
const WebRtc_UWord8 numberOfSSRC, virtual bool REMB() const;
const WebRtc_UWord32* SSRC);
/* virtual WebRtc_Word32 SetREMBStatus(const bool enable);
* (IJ) Extended jitter report.
*/
virtual bool IJ() const;
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);
/* // (IJ) Extended jitter report.
* (TMMBR) Temporary Max Media Bit Rate virtual bool IJ() const;
*/
virtual bool TMMBR() 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, virtual WebRtc_UWord16 MaxDataPayloadLength() const;
const bool IPV6,
const WebRtc_UWord8 authenticationOverhead = 0);
/* virtual WebRtc_Word32 SetMaxTransferUnit(const WebRtc_UWord16 size);
* (NACK) Negative acknowledgement
*/
// Is Negative acknowledgement requests on/off? virtual WebRtc_Word32 SetTransportOverhead(
virtual NACKMethod NACK() const ; const bool tcp,
const bool ipv6,
const WebRtc_UWord8 authentication_overhead = 0);
// Turn negative acknowledgement requests on/off // (NACK) Negative acknowledgment part.
virtual WebRtc_Word32 SetNACKStatus(const NACKMethod method);
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 int SelectiveRetransmissions() const;
virtual WebRtc_Word32 SendNACK(const WebRtc_UWord16* nackList,
const WebRtc_UWord16 size);
// Store the sent packets, needed to answer to a Negative acknowledgement requests virtual int SetSelectiveRetransmissions(uint8_t settings);
virtual WebRtc_Word32 SetStorePacketsStatus(const bool enable, const WebRtc_UWord16 numberToStore = 200);
/* // Send a Negative acknowledgment packet.
* (APP) Application specific data virtual WebRtc_Word32 SendNACK(const WebRtc_UWord16* nack_list,
*/ const WebRtc_UWord16 size);
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);
/* // Store the sent packets, needed to answer to a negative acknowledgment
* Audio // 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) // (APP) Application specific data.
virtual WebRtc_Word32 SetAudioPacketSize(const WebRtc_UWord16 packetSizeSamples); virtual WebRtc_Word32 SetRTCPApplicationSpecificData(
const WebRtc_UWord8 sub_type,
const WebRtc_UWord32 name,
const WebRtc_UWord8* data,
const WebRtc_UWord16 length);
// Outband DTMF detection // (XR) VOIP metric.
virtual WebRtc_Word32 SetTelephoneEventStatus(const bool enable, virtual WebRtc_Word32 SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric);
const bool forwardToDecoder,
const bool detectEndOfTone = false);
// Is outband DTMF turned on/off? // Audio part.
virtual bool TelephoneEvent() const;
// Is forwarding of outband telephone events turned on/off? // Set audio packet size, used to determine when it's time to send a DTMF
virtual bool TelephoneEventForwardToDecoder() const; // 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) // Is outband DTMF turned on/off?
virtual WebRtc_Word32 SendTelephoneEventOutband(const WebRtc_UWord8 key, 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_UWord16 time_ms,
const WebRtc_UWord8 level); const WebRtc_UWord8 level);
// Set payload type for Redundant Audio Data RFC 2198 // Set payload type for Redundant Audio Data RFC 2198.
virtual WebRtc_Word32 SetSendREDPayloadType(const WebRtc_Word8 payloadType); virtual WebRtc_Word32 SetSendREDPayloadType(const WebRtc_Word8 payload_type);
// Get payload type for Redundant Audio Data RFC 2198 // Get payload type for Redundant Audio Data RFC 2198.
virtual WebRtc_Word32 SendREDPayloadType(WebRtc_Word8& payloadType) const; virtual WebRtc_Word32 SendREDPayloadType(WebRtc_Word8& payload_type) const;
// Set status and ID for header-extension-for-audio-level-indication. // Set status and id for header-extension-for-audio-level-indication.
virtual WebRtc_Word32 SetRTPAudioLevelIndicationStatus(const bool enable, virtual WebRtc_Word32 SetRTPAudioLevelIndicationStatus(
const WebRtc_UWord8 ID); const bool enable, const WebRtc_UWord8 id);
// Get status and ID for header-extension-for-audio-level-indication. // Get status and id for header-extension-for-audio-level-indication.
virtual WebRtc_Word32 GetRTPAudioLevelIndicationStatus(bool& enable, virtual WebRtc_Word32 GetRTPAudioLevelIndicationStatus(
WebRtc_UWord8& ID) const; bool& enable, WebRtc_UWord8& id) const;
// Store the audio level in dBov for header-extension-for-audio-level-indication. // Store the audio level in d_bov for header-extension-for-audio-level-
virtual WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_dBov); // indication.
virtual WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_d_bov);
/* // Video part.
* Video
*/
virtual RtpVideoCodecTypes ReceivedVideoCodec() const;
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 SendRTCPSliceLossIndication(
virtual WebRtc_Word32 SetKeyFrameRequestMethod(const KeyFrameRequestMethod method); const WebRtc_UWord8 picture_id);
// send a request for a keyframe // Set method for requestion a new key frame.
virtual WebRtc_Word32 RequestKeyFrame(); 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, virtual void SetTargetSendBitrate(const WebRtc_UWord32 bitrate);
const WebRtc_UWord8 payloadTypeRED,
const WebRtc_UWord8 payloadTypeFEC);
virtual WebRtc_Word32 GenericFECStatus(bool& enable, virtual WebRtc_Word32 SetGenericFECStatus(
WebRtc_UWord8& payloadTypeRED, const bool enable,
WebRtc_UWord8& payloadTypeFEC); const WebRtc_UWord8 payload_type_red,
const WebRtc_UWord8 payload_type_fec);
virtual WebRtc_Word32 SetFecParameters( virtual WebRtc_Word32 GenericFECStatus(
const FecProtectionParams* delta_params, bool& enable,
const FecProtectionParams* key_params); WebRtc_UWord8& payload_type_red,
WebRtc_UWord8& payload_type_fec);
virtual WebRtc_Word32 LastReceivedNTP(WebRtc_UWord32& NTPsecs, virtual WebRtc_Word32 SetFecParameters(
WebRtc_UWord32& NTPfrac, const FecProtectionParams* delta_params,
WebRtc_UWord32& remoteSR); const FecProtectionParams* key_params);
virtual WebRtc_Word32 BoundingSet(bool &tmmbrOwner, virtual WebRtc_Word32 LastReceivedNTP(WebRtc_UWord32& NTPsecs,
TMMBRSet*& boundingSetRec); WebRtc_UWord32& NTPfrac,
WebRtc_UWord32& remote_sr);
virtual void BitrateSent(WebRtc_UWord32* totalRate, virtual WebRtc_Word32 BoundingSet(bool& tmmbr_owner,
WebRtc_UWord32* videoRate, TMMBRSet*& bounding_set_rec);
WebRtc_UWord32* fecRate,
WebRtc_UWord32* nackRate) const;
virtual int EstimatedReceiveBandwidth( virtual void BitrateSent(WebRtc_UWord32* total_rate,
WebRtc_UWord32* available_bandwidth) const; 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_UWord32 SendTimeOfSendReport(const WebRtc_UWord32 send_report);
virtual WebRtc_Word32 SendRTCPReferencePictureSelection(const WebRtc_UWord64 pictureID);
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 OnReceivedTMMBR();
void OnRequestIntraFrame();
// received a request for a new SLI // Bad state of RTP receiver request a keyframe.
void OnReceivedSliceLossIndication(const WebRtc_UWord8 pictureID); void OnRequestIntraFrame();
// received a new refereence frame // Received a request for a new SLI.
void OnReceivedReferencePictureSelectionIndication( void OnReceivedSliceLossIndication(const WebRtc_UWord8 picture_id);
const WebRtc_UWord64 pitureID);
void OnReceivedNACK(const WebRtc_UWord16 nackSequenceNumbersLength, // Received a new reference frame.
const WebRtc_UWord16* nackSequenceNumbers); 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 void OnRequestSendReport();
// need to worry about data race.
void OwnsClock() { _owns_clock = true; }
protected: // Following function is only called when constructing the object so no
void RegisterChildModule(RtpRtcp* module); // 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_UWord32 BitrateReceivedNow() const;
WebRtc_UWord16 RemoteSequenceNumber() const;
// only for internal testing // Get remote SequenceNumber.
WebRtc_UWord32 LastSendReport(WebRtc_UWord32& lastRTCPTime); WebRtc_UWord16 RemoteSequenceNumber() const;
RTPSender _rtpSender; // Only for internal testing.
RTPReceiver _rtpReceiver; WebRtc_UWord32 LastSendReport(WebRtc_UWord32& last_rtcptime);
RTCPSender _rtcpSender; RTPSender rtp_sender_;
RTCPReceiver _rtcpReceiver; RTPReceiver rtp_receiver_;
bool _owns_clock; RTCPSender rtcp_sender_;
RtpRtcpClock& _clock; RTCPReceiver rtcp_receiver_;
private:
int64_t RtcpReportInterval();
WebRtc_Word32 _id; bool owns_clock_;
const bool _audio; RtpRtcpClock& clock_;
bool _collisionDetected;
WebRtc_Word64 _lastProcessTime;
WebRtc_Word64 _lastBitrateProcessTime;
WebRtc_Word64 _lastPacketTimeoutProcessTime;
WebRtc_UWord16 _packetOverHead;
scoped_ptr<CriticalSectionWrapper> _criticalSectionModulePtrs; private:
scoped_ptr<CriticalSectionWrapper> _criticalSectionModulePtrsFeedback; int64_t RtcpReportInterval();
ModuleRtpRtcpImpl* _defaultModule;
std::list<ModuleRtpRtcpImpl*> _childModules;
// Dead or alive WebRtc_Word32 id_;
bool _deadOrAliveActive; const bool audio_;
WebRtc_UWord32 _deadOrAliveTimeoutMS; bool collision_detected_;
WebRtc_Word64 _deadOrAliveLastTimer; WebRtc_Word64 last_process_time_;
// send side WebRtc_Word64 last_bitrate_process_time_;
NACKMethod _nackMethod; WebRtc_Word64 last_packet_timeout_process_time_;
WebRtc_UWord32 _nackLastTimeSent; WebRtc_UWord16 packet_overhead_;
WebRtc_UWord16 _nackLastSeqNumberSent;
bool _simulcast; scoped_ptr<CriticalSectionWrapper> critical_section_module_ptrs_;
VideoCodec _sendVideoCodec; scoped_ptr<CriticalSectionWrapper> critical_section_module_ptrs_feedback_;
KeyFrameRequestMethod _keyFrameReqMethod; 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 #ifdef MATLAB
MatlabPlot* _plot1; MatlabPlot* plot1_;
#endif #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_