blob: 3cdf162c308a906579359b39805b851629111482 [file] [log] [blame]
* Copyright (c) 2011 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.
#include "voe_rtp_rtcp.h"
#include "ref_count.h"
#include "shared_data.h"
namespace webrtc {
class VoERTP_RTCPImpl : public virtual voe::SharedData,
public VoERTP_RTCP,
public voe::RefCount
virtual int Release();
// Registration of observers for RTP and RTCP callbacks
virtual int RegisterRTPObserver(int channel, VoERTPObserver& observer);
virtual int DeRegisterRTPObserver(int channel);
virtual int RegisterRTCPObserver(int channel, VoERTCPObserver& observer);
virtual int DeRegisterRTCPObserver(int channel);
virtual int SetRTCPStatus(int channel, bool enable);
virtual int GetRTCPStatus(int channel, bool& enabled);
virtual int SetRTCP_CNAME(int channel, const char cName[256]);
virtual int GetRTCP_CNAME(int channel, char cName[256]);
virtual int GetRemoteRTCP_CNAME(int channel, char cName[256]);
virtual int GetRemoteRTCPData(int channel,
unsigned int& NTPHigh,
unsigned int& NTPLow,
unsigned int& timestamp,
unsigned int& playoutTimestamp,
unsigned int* jitter = NULL,
unsigned short* fractionLost = NULL);
virtual int SendApplicationDefinedRTCPPacket(
int channel,
const unsigned char subType,
unsigned int name,
const char* data,
unsigned short dataLengthInBytes);
virtual int SetLocalSSRC(int channel, unsigned int ssrc);
virtual int GetLocalSSRC(int channel, unsigned int& ssrc);
virtual int GetRemoteSSRC(int channel, unsigned int& ssrc);
// RTP Header Extension for Client-to-Mixer Audio Level Indication
virtual int SetRTPAudioLevelIndicationStatus(int channel,
bool enable,
unsigned char ID);
virtual int GetRTPAudioLevelIndicationStatus(int channel,
bool& enabled,
unsigned char& ID);
virtual int GetRemoteCSRCs(int channel, unsigned int arrCSRC[15]);
// Statistics
virtual int GetRTPStatistics(int channel,
unsigned int& averageJitterMs,
unsigned int& maxJitterMs,
unsigned int& discardedPackets);
virtual int GetRTCPStatistics(int channel, CallStatistics& stats);
// RTP keepalive mechanism (maintains NAT mappings associated to RTP flows)
virtual int SetRTPKeepaliveStatus(int channel,
bool enable,
unsigned char unknownPayloadType,
int deltaTransmitTimeSeconds = 15);
virtual int GetRTPKeepaliveStatus(int channel,
bool& enabled,
unsigned char& unknownPayloadType,
int& deltaTransmitTimeSeconds);
// FEC
virtual int SetFECStatus(int channel,
bool enable,
int redPayloadtype = -1);
virtual int GetFECStatus(int channel, bool& enabled, int& redPayloadtype);
// Store RTP and RTCP packets and dump to file (compatible with rtpplay)
virtual int StartRTPDump(int channel,
const char fileNameUTF8[1024],
RTPDirections direction = kRtpIncoming);
virtual int StopRTPDump(int channel,
RTPDirections direction = kRtpIncoming);
virtual int RTPDumpIsActive(int channel,
RTPDirections direction = kRtpIncoming);
// Insert (and transmits) extra RTP packet into active RTP audio stream
virtual int InsertExtraRTPPacket(int channel,
unsigned char payloadType,
bool markerBit,
const char* payloadData,
unsigned short payloadSize);
virtual ~VoERTP_RTCPImpl();
} // namespace webrtc