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Side by Side Diff: webrtc/video_receive_stream.h

Issue 1369263002: Unify Transport and newapi::Transport interfaces. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: self-review Created 5 years, 2 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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82 82
83 uint32_t ssrc = 0; 83 uint32_t ssrc = 0;
84 std::string c_name; 84 std::string c_name;
85 StreamDataCounters rtp_stats; 85 StreamDataCounters rtp_stats;
86 RtcpPacketTypeCounter rtcp_packet_type_counts; 86 RtcpPacketTypeCounter rtcp_packet_type_counts;
87 RtcpStatistics rtcp_stats; 87 RtcpStatistics rtcp_stats;
88 }; 88 };
89 89
90 struct Config { 90 struct Config {
91 Config() = delete; 91 Config() = delete;
92 explicit Config(newapi::Transport* rtcp_send_transport) 92 explicit Config(Transport* rtcp_send_transport)
93 : rtcp_send_transport(rtcp_send_transport) {} 93 : rtcp_send_transport(rtcp_send_transport) {}
94 94
95 std::string ToString() const; 95 std::string ToString() const;
96 96
97 // Decoders for every payload that we can receive. 97 // Decoders for every payload that we can receive.
98 std::vector<Decoder> decoders; 98 std::vector<Decoder> decoders;
99 99
100 // Receive-stream specific RTP settings. 100 // Receive-stream specific RTP settings.
101 struct Rtp { 101 struct Rtp {
102 std::string ToString() const; 102 std::string ToString() const;
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137 137
138 // Map from video RTP payload type -> RTX config. 138 // Map from video RTP payload type -> RTX config.
139 typedef std::map<int, Rtx> RtxMap; 139 typedef std::map<int, Rtx> RtxMap;
140 RtxMap rtx; 140 RtxMap rtx;
141 141
142 // RTP header extensions used for the received stream. 142 // RTP header extensions used for the received stream.
143 std::vector<RtpExtension> extensions; 143 std::vector<RtpExtension> extensions;
144 } rtp; 144 } rtp;
145 145
146 // Transport for outgoing packets (RTCP). 146 // Transport for outgoing packets (RTCP).
147 newapi::Transport* rtcp_send_transport = nullptr; 147 Transport* rtcp_send_transport = nullptr;
148 148
149 // VideoRenderer will be called for each decoded frame. 'nullptr' disables 149 // VideoRenderer will be called for each decoded frame. 'nullptr' disables
150 // rendering of this stream. 150 // rendering of this stream.
151 VideoRenderer* renderer = nullptr; 151 VideoRenderer* renderer = nullptr;
152 152
153 // Expected delay needed by the renderer, i.e. the frame will be delivered 153 // Expected delay needed by the renderer, i.e. the frame will be delivered
154 // this many milliseconds, if possible, earlier than the ideal render time. 154 // this many milliseconds, if possible, earlier than the ideal render time.
155 // Only valid if 'renderer' is set. 155 // Only valid if 'renderer' is set.
156 int render_delay_ms = 10; 156 int render_delay_ms = 10;
157 157
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175 int target_delay_ms = 0; 175 int target_delay_ms = 0;
176 }; 176 };
177 177
178 // TODO(pbos): Add info on currently-received codec to Stats. 178 // TODO(pbos): Add info on currently-received codec to Stats.
179 virtual Stats GetStats() const = 0; 179 virtual Stats GetStats() const = 0;
180 }; 180 };
181 181
182 } // namespace webrtc 182 } // namespace webrtc
183 183
184 #endif // WEBRTC_VIDEO_RECEIVE_STREAM_H_ 184 #endif // WEBRTC_VIDEO_RECEIVE_STREAM_H_
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