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Side by Side Diff: webrtc/video_receive_stream.h

Issue 1171533002: Disable reduced-size RTCP in default config. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rename tests Created 5 years, 6 months ago
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1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 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 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 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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100 std::string ToString() const; 100 std::string ToString() const;
101 101
102 // Decoders for every payload that we can receive. 102 // Decoders for every payload that we can receive.
103 std::vector<Decoder> decoders; 103 std::vector<Decoder> decoders;
104 104
105 // Receive-stream specific RTP settings. 105 // Receive-stream specific RTP settings.
106 struct Rtp { 106 struct Rtp {
107 Rtp() 107 Rtp()
108 : remote_ssrc(0), 108 : remote_ssrc(0),
109 local_ssrc(0), 109 local_ssrc(0),
110 rtcp_mode(newapi::kRtcpReducedSize), 110 rtcp_mode(newapi::kRtcpCompound),
111 remb(true) {} 111 remb(false) {}
112 std::string ToString() const; 112 std::string ToString() const;
113 113
114 // Synchronization source (stream identifier) to be received. 114 // Synchronization source (stream identifier) to be received.
115 uint32_t remote_ssrc; 115 uint32_t remote_ssrc;
116 // Sender SSRC used for sending RTCP (such as receiver reports). 116 // Sender SSRC used for sending RTCP (such as receiver reports).
117 uint32_t local_ssrc; 117 uint32_t local_ssrc;
118 118
119 // See RtcpMode for description. 119 // See RtcpMode for description.
120 newapi::RtcpMode rtcp_mode; 120 newapi::RtcpMode rtcp_mode;
121 121
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192 // TODO(pbos): Add info on currently-received codec to Stats. 192 // TODO(pbos): Add info on currently-received codec to Stats.
193 virtual Stats GetStats() const = 0; 193 virtual Stats GetStats() const = 0;
194 194
195 protected: 195 protected:
196 virtual ~VideoReceiveStream() {} 196 virtual ~VideoReceiveStream() {}
197 }; 197 };
198 198
199 } // namespace webrtc 199 } // namespace webrtc
200 200
201 #endif // WEBRTC_VIDEO_RECEIVE_STREAM_H_ 201 #endif // WEBRTC_VIDEO_RECEIVE_STREAM_H_
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