1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
|
/*
* 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_VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H
#define WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H
#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
#include "webrtc/voice_engine/shared_data.h"
namespace webrtc {
class VoERTP_RTCPImpl : public VoERTP_RTCP
{
public:
// RTCP
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 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);
// SSRC
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 SetSendAudioLevelIndicationStatus(int channel,
bool enable,
unsigned char id);
virtual int SetReceiveAudioLevelIndicationStatus(int channel,
bool enable,
unsigned char id);
// RTP Header Extension for Absolute Sender Time
virtual int SetSendAbsoluteSenderTimeStatus(int channel,
bool enable,
unsigned char id);
virtual int SetReceiveAbsoluteSenderTimeStatus(int channel,
bool enable,
unsigned char id);
// Statistics
virtual int GetRTPStatistics(int channel,
unsigned int& averageJitterMs,
unsigned int& maxJitterMs,
unsigned int& discardedPackets);
virtual int GetRTCPStatistics(int channel, CallStatistics& stats);
virtual int GetRemoteRTCPReportBlocks(
int channel, std::vector<ReportBlock>* report_blocks);
// RED
virtual int SetREDStatus(int channel,
bool enable,
int redPayloadtype = -1);
virtual int GetREDStatus(int channel, bool& enabled, int& redPayloadtype);
//NACK
virtual int SetNACKStatus(int channel,
bool enable,
int maxNoPackets);
// 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);
virtual int SetVideoEngineBWETarget(int channel, ViENetwork* vie_network,
int video_channel);
protected:
VoERTP_RTCPImpl(voe::SharedData* shared);
virtual ~VoERTP_RTCPImpl();
private:
voe::SharedData* _shared;
};
} // namespace webrtc
#endif // WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H
|
