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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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93 // initialized from inside the VideoSendStream. | 93 // initialized from inside the VideoSendStream. |
94 VideoEncoder* encoder = nullptr; | 94 VideoEncoder* encoder = nullptr; |
95 } encoder_settings; | 95 } encoder_settings; |
96 | 96 |
97 static const size_t kDefaultMaxPacketSize = 1500 - 40; // TCP over IPv4. | 97 static const size_t kDefaultMaxPacketSize = 1500 - 40; // TCP over IPv4. |
98 struct Rtp { | 98 struct Rtp { |
99 std::string ToString() const; | 99 std::string ToString() const; |
100 | 100 |
101 std::vector<uint32_t> ssrcs; | 101 std::vector<uint32_t> ssrcs; |
102 | 102 |
103 // See RtcpMode for description. | |
104 // TODO(pbos): Use this to configure the send stream. | |
pthatcher1
2015/10/23 20:40:01
To fix this TODO, I think all we have to do is cha
Taylor Brandstetter
2015/11/11 19:42:40
Done.
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105 RtcpMode rtcp_mode = RtcpMode::kCompound; | |
106 | |
103 // Max RTP packet size delivered to send transport from VideoEngine. | 107 // Max RTP packet size delivered to send transport from VideoEngine. |
104 size_t max_packet_size = kDefaultMaxPacketSize; | 108 size_t max_packet_size = kDefaultMaxPacketSize; |
105 | 109 |
106 // RTP header extensions to use for this send stream. | 110 // RTP header extensions to use for this send stream. |
107 std::vector<RtpExtension> extensions; | 111 std::vector<RtpExtension> extensions; |
108 | 112 |
109 // See NackConfig for description. | 113 // See NackConfig for description. |
110 NackConfig nack; | 114 NackConfig nack; |
111 | 115 |
112 // See FecConfig for description. | 116 // See FecConfig for description. |
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174 // in the config. Encoder settings are passed on to the encoder instance along | 178 // in the config. Encoder settings are passed on to the encoder instance along |
175 // with the VideoStream settings. | 179 // with the VideoStream settings. |
176 virtual bool ReconfigureVideoEncoder(const VideoEncoderConfig& config) = 0; | 180 virtual bool ReconfigureVideoEncoder(const VideoEncoderConfig& config) = 0; |
177 | 181 |
178 virtual Stats GetStats() = 0; | 182 virtual Stats GetStats() = 0; |
179 }; | 183 }; |
180 | 184 |
181 } // namespace webrtc | 185 } // namespace webrtc |
182 | 186 |
183 #endif // WEBRTC_VIDEO_SEND_STREAM_H_ | 187 #endif // WEBRTC_VIDEO_SEND_STREAM_H_ |
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