OLD | NEW |
| (Empty) |
1 /* | |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | |
3 * | |
4 * Use of this source code is governed by a BSD-style license | |
5 * that can be found in the LICENSE file in the root of the source | |
6 * tree. An additional intellectual property rights grant can be found | |
7 * in the file PATENTS. All contributing project authors may | |
8 * be found in the AUTHORS file in the root of the source tree. | |
9 */ | |
10 | |
11 // This sub-API supports the following functionalities: | |
12 // | |
13 // - Callbacks for RTP and RTCP events such as modified SSRC or CSRC. | |
14 // - SSRC handling. | |
15 // - Transmission of RTCP sender reports. | |
16 // - Obtaining RTCP data from incoming RTCP sender reports. | |
17 // - RTP and RTCP statistics (jitter, packet loss, RTT etc.). | |
18 // - Redundant Coding (RED) | |
19 // - Writing RTP and RTCP packets to binary files for off-line analysis of | |
20 // the call quality. | |
21 // | |
22 // Usage example, omitting error checking: | |
23 // | |
24 // using namespace webrtc; | |
25 // VoiceEngine* voe = VoiceEngine::Create(); | |
26 // VoEBase* base = VoEBase::GetInterface(voe); | |
27 // VoERTP_RTCP* rtp_rtcp = VoERTP_RTCP::GetInterface(voe); | |
28 // base->Init(); | |
29 // int ch = base->CreateChannel(); | |
30 // ... | |
31 // rtp_rtcp->SetLocalSSRC(ch, 12345); | |
32 // ... | |
33 // base->DeleteChannel(ch); | |
34 // base->Terminate(); | |
35 // base->Release(); | |
36 // rtp_rtcp->Release(); | |
37 // VoiceEngine::Delete(voe); | |
38 // | |
39 #ifndef VOICE_ENGINE_VOE_RTP_RTCP_H_ | |
40 #define VOICE_ENGINE_VOE_RTP_RTCP_H_ | |
41 | |
42 #include <vector> | |
43 #include "common_types.h" // NOLINT(build/include) | |
44 | |
45 namespace webrtc { | |
46 | |
47 class VoiceEngine; | |
48 | |
49 // VoERTPObserver | |
50 class WEBRTC_DLLEXPORT VoERTPObserver { | |
51 public: | |
52 virtual void OnIncomingCSRCChanged(int channel, | |
53 unsigned int CSRC, | |
54 bool added) = 0; | |
55 | |
56 virtual void OnIncomingSSRCChanged(int channel, unsigned int SSRC) = 0; | |
57 | |
58 protected: | |
59 virtual ~VoERTPObserver() {} | |
60 }; | |
61 | |
62 // CallStatistics | |
63 struct CallStatistics { | |
64 unsigned short fractionLost; | |
65 unsigned int cumulativeLost; | |
66 unsigned int extendedMax; | |
67 unsigned int jitterSamples; | |
68 int64_t rttMs; | |
69 size_t bytesSent; | |
70 int packetsSent; | |
71 size_t bytesReceived; | |
72 int packetsReceived; | |
73 // The capture ntp time (in local timebase) of the first played out audio | |
74 // frame. | |
75 int64_t capture_start_ntp_time_ms_; | |
76 }; | |
77 | |
78 // See section 6.4.1 in http://www.ietf.org/rfc/rfc3550.txt for details. | |
79 struct SenderInfo { | |
80 uint32_t NTP_timestamp_high; | |
81 uint32_t NTP_timestamp_low; | |
82 uint32_t RTP_timestamp; | |
83 uint32_t sender_packet_count; | |
84 uint32_t sender_octet_count; | |
85 }; | |
86 | |
87 // See section 6.4.2 in http://www.ietf.org/rfc/rfc3550.txt for details. | |
88 struct ReportBlock { | |
89 uint32_t sender_SSRC; // SSRC of sender | |
90 uint32_t source_SSRC; | |
91 uint8_t fraction_lost; | |
92 uint32_t cumulative_num_packets_lost; | |
93 uint32_t extended_highest_sequence_number; | |
94 uint32_t interarrival_jitter; | |
95 uint32_t last_SR_timestamp; | |
96 uint32_t delay_since_last_SR; | |
97 }; | |
98 | |
99 // VoERTP_RTCP | |
100 class WEBRTC_DLLEXPORT VoERTP_RTCP { | |
101 public: | |
102 // Factory for the VoERTP_RTCP sub-API. Increases an internal | |
103 // reference counter if successful. Returns NULL if the API is not | |
104 // supported or if construction fails. | |
105 static VoERTP_RTCP* GetInterface(VoiceEngine* voiceEngine); | |
106 | |
107 // Releases the VoERTP_RTCP sub-API and decreases an internal | |
108 // reference counter. Returns the new reference count. This value should | |
109 // be zero for all sub-API:s before the VoiceEngine object can be safely | |
110 // deleted. | |
111 virtual int Release() = 0; | |
112 | |
113 // Sets the local RTP synchronization source identifier (SSRC) explicitly. | |
114 virtual int SetLocalSSRC(int channel, unsigned int ssrc) = 0; | |
115 | |
116 // Gets the local RTP SSRC of a specified |channel|. | |
117 virtual int GetLocalSSRC(int channel, unsigned int& ssrc) = 0; | |
118 | |
119 // Gets the SSRC of the incoming RTP packets. | |
120 virtual int GetRemoteSSRC(int channel, unsigned int& ssrc) = 0; | |
121 | |
122 // Sets the status of rtp-audio-level-indication on a specific |channel|. | |
123 virtual int SetSendAudioLevelIndicationStatus(int channel, | |
124 bool enable, | |
125 unsigned char id = 1) = 0; | |
126 | |
127 // Sets the RTCP status on a specific |channel|. | |
128 virtual int SetRTCPStatus(int channel, bool enable) = 0; | |
129 | |
130 // Gets the RTCP status on a specific |channel|. | |
131 virtual int GetRTCPStatus(int channel, bool& enabled) = 0; | |
132 | |
133 // Sets the canonical name (CNAME) parameter for RTCP reports on a | |
134 // specific |channel|. | |
135 virtual int SetRTCP_CNAME(int channel, const char cName[256]) = 0; | |
136 | |
137 // Gets the canonical name (CNAME) parameter for incoming RTCP reports | |
138 // on a specific channel. | |
139 virtual int GetRemoteRTCP_CNAME(int channel, char cName[256]) = 0; | |
140 | |
141 // Gets RTCP statistics for a specific |channel|. | |
142 virtual int GetRTCPStatistics(int channel, CallStatistics& stats) = 0; | |
143 | |
144 protected: | |
145 VoERTP_RTCP() {} | |
146 virtual ~VoERTP_RTCP() {} | |
147 }; | |
148 | |
149 } // namespace webrtc | |
150 | |
151 #endif // #ifndef VOICE_ENGINE_VOE_RTP_RTCP_H_ | |
OLD | NEW |