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Side by Side Diff: voice_engine/include/voe_rtp_rtcp.h

Issue 3006383002: Remove VoERTP_RTCP (Closed)
Patch Set: rebase+remove Created 3 years, 3 months ago
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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_
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