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1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 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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66 // Receive-stream specific RTP settings. | 66 // Receive-stream specific RTP settings. |
67 struct Rtp { | 67 struct Rtp { |
68 std::string ToString() const; | 68 std::string ToString() const; |
69 | 69 |
70 // Synchronization source (stream identifier) to be received. | 70 // Synchronization source (stream identifier) to be received. |
71 uint32_t remote_ssrc = 0; | 71 uint32_t remote_ssrc = 0; |
72 | 72 |
73 // Sender SSRC used for sending RTCP (such as receiver reports). | 73 // Sender SSRC used for sending RTCP (such as receiver reports). |
74 uint32_t local_ssrc = 0; | 74 uint32_t local_ssrc = 0; |
75 | 75 |
76 // Enable feedback for send side bandwidth estimation. | |
77 // See draft-holmer-rmcat-transport-wide-cc-extensions for details. | |
the sun
2015/12/22 00:14:14
nit: can you make it a url even?
stefan-webrtc
2016/01/07 13:43:41
Done.
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78 bool transport_cc = false; | |
79 | |
76 // RTP header extensions used for the received stream. | 80 // RTP header extensions used for the received stream. |
77 std::vector<RtpExtension> extensions; | 81 std::vector<RtpExtension> extensions; |
78 } rtp; | 82 } rtp; |
79 | 83 |
80 Transport* receive_transport = nullptr; | 84 Transport* receive_transport = nullptr; |
81 Transport* rtcp_send_transport = nullptr; | 85 Transport* rtcp_send_transport = nullptr; |
82 | 86 |
83 // Underlying VoiceEngine handle, used to map AudioReceiveStream to lower- | 87 // Underlying VoiceEngine handle, used to map AudioReceiveStream to lower- |
84 // level components. | 88 // level components. |
85 // TODO(solenberg): Remove when VoiceEngine channels are created outside | 89 // TODO(solenberg): Remove when VoiceEngine channels are created outside |
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109 // Only one sink can be set and passing a null sink, clears an existing one. | 113 // Only one sink can be set and passing a null sink, clears an existing one. |
110 // NOTE: Audio must still somehow be pulled through AudioTransport for audio | 114 // NOTE: Audio must still somehow be pulled through AudioTransport for audio |
111 // to stream through this sink. In practice, this happens if mixed audio | 115 // to stream through this sink. In practice, this happens if mixed audio |
112 // is being pulled+rendered and/or if audio is being pulled for the purposes | 116 // is being pulled+rendered and/or if audio is being pulled for the purposes |
113 // of feeding to the AEC. | 117 // of feeding to the AEC. |
114 virtual void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) = 0; | 118 virtual void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) = 0; |
115 }; | 119 }; |
116 } // namespace webrtc | 120 } // namespace webrtc |
117 | 121 |
118 #endif // WEBRTC_AUDIO_RECEIVE_STREAM_H_ | 122 #endif // WEBRTC_AUDIO_RECEIVE_STREAM_H_ |
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