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Side by Side Diff: webrtc/video_receive_stream.h

Issue 1273363005: Add send transports to individual webrtc::Call streams. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase+comment Created 5 years, 3 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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79 int discarded_packets = 0; 79 int discarded_packets = 0;
80 80
81 uint32_t ssrc = 0; 81 uint32_t ssrc = 0;
82 std::string c_name; 82 std::string c_name;
83 StreamDataCounters rtp_stats; 83 StreamDataCounters rtp_stats;
84 RtcpPacketTypeCounter rtcp_packet_type_counts; 84 RtcpPacketTypeCounter rtcp_packet_type_counts;
85 RtcpStatistics rtcp_stats; 85 RtcpStatistics rtcp_stats;
86 }; 86 };
87 87
88 struct Config { 88 struct Config {
89 Config() = delete;
90 explicit Config(newapi::Transport* rtcp_send_transport)
91 : rtcp_send_transport(rtcp_send_transport) {}
92
89 std::string ToString() const; 93 std::string ToString() const;
90 94
91 // Decoders for every payload that we can receive. 95 // Decoders for every payload that we can receive.
92 std::vector<Decoder> decoders; 96 std::vector<Decoder> decoders;
93 97
94 // Receive-stream specific RTP settings. 98 // Receive-stream specific RTP settings.
95 struct Rtp { 99 struct Rtp {
96 std::string ToString() const; 100 std::string ToString() const;
97 101
98 // Synchronization source (stream identifier) to be received. 102 // Synchronization source (stream identifier) to be received.
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130 }; 134 };
131 135
132 // Map from video RTP payload type -> RTX config. 136 // Map from video RTP payload type -> RTX config.
133 typedef std::map<int, Rtx> RtxMap; 137 typedef std::map<int, Rtx> RtxMap;
134 RtxMap rtx; 138 RtxMap rtx;
135 139
136 // RTP header extensions used for the received stream. 140 // RTP header extensions used for the received stream.
137 std::vector<RtpExtension> extensions; 141 std::vector<RtpExtension> extensions;
138 } rtp; 142 } rtp;
139 143
144 // Transport for outgoing packets (RTCP).
145 newapi::Transport* rtcp_send_transport = nullptr;
146
140 // VideoRenderer will be called for each decoded frame. 'nullptr' disables 147 // VideoRenderer will be called for each decoded frame. 'nullptr' disables
141 // rendering of this stream. 148 // rendering of this stream.
142 VideoRenderer* renderer = nullptr; 149 VideoRenderer* renderer = nullptr;
143 150
144 // Expected delay needed by the renderer, i.e. the frame will be delivered 151 // Expected delay needed by the renderer, i.e. the frame will be delivered
145 // this many milliseconds, if possible, earlier than the ideal render time. 152 // this many milliseconds, if possible, earlier than the ideal render time.
146 // Only valid if 'renderer' is set. 153 // Only valid if 'renderer' is set.
147 int render_delay_ms = 10; 154 int render_delay_ms = 10;
148 155
149 // Identifier for an A/V synchronization group. Empty string to disable. 156 // Identifier for an A/V synchronization group. Empty string to disable.
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166 int target_delay_ms = 0; 173 int target_delay_ms = 0;
167 }; 174 };
168 175
169 // TODO(pbos): Add info on currently-received codec to Stats. 176 // TODO(pbos): Add info on currently-received codec to Stats.
170 virtual Stats GetStats() const = 0; 177 virtual Stats GetStats() const = 0;
171 }; 178 };
172 179
173 } // namespace webrtc 180 } // namespace webrtc
174 181
175 #endif // WEBRTC_VIDEO_RECEIVE_STREAM_H_ 182 #endif // WEBRTC_VIDEO_RECEIVE_STREAM_H_
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