Chromium Code Reviews Chromium Code Reviews
Issue 2826263004:
Move responsibility for RTP header extensions on video receive. (Closed)
| OLD | NEW |
|---|---|
| 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 #ifndef WEBRTC_CALL_CALL_H_ | 10 #ifndef WEBRTC_CALL_CALL_H_ |
| (...skipping 82 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
| 93 | 93 |
| 94 int send_bandwidth_bps = 0; // Estimated available send bandwidth. | 94 int send_bandwidth_bps = 0; // Estimated available send bandwidth. |
| 95 int max_padding_bitrate_bps = 0; // Cumulative configured max padding. | 95 int max_padding_bitrate_bps = 0; // Cumulative configured max padding. |
| 96 int recv_bandwidth_bps = 0; // Estimated available receive bandwidth. | 96 int recv_bandwidth_bps = 0; // Estimated available receive bandwidth. |
| 97 int64_t pacer_delay_ms = 0; | 97 int64_t pacer_delay_ms = 0; |
| 98 int64_t rtt_ms = -1; | 98 int64_t rtt_ms = -1; |
| 99 }; | 99 }; |
| 100 | 100 |
| 101 static Call* Create(const Call::Config& config); | 101 static Call* Create(const Call::Config& config); |
| 102 | 102 |
| 103 // TODO(nisse): Should move to RtpTransportController. | |
| 104 // Rtp header extensions can be renegotiated mid-call. | |
| 105 virtual void SetVideoReceiveRtpHeaderExtensions( | |
| 106 const std::vector<RtpExtension>& extensions) = 0; | |
|
pthatcher1
2017/04/29 00:25:18
It makes sense for this to be per-RtpTransport, bu
| |
| 107 | |
| 103 virtual AudioSendStream* CreateAudioSendStream( | 108 virtual AudioSendStream* CreateAudioSendStream( |
| 104 const AudioSendStream::Config& config) = 0; | 109 const AudioSendStream::Config& config) = 0; |
| 105 virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0; | 110 virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0; |
| 106 | 111 |
| 107 virtual AudioReceiveStream* CreateAudioReceiveStream( | 112 virtual AudioReceiveStream* CreateAudioReceiveStream( |
| 108 const AudioReceiveStream::Config& config) = 0; | 113 const AudioReceiveStream::Config& config) = 0; |
| 109 virtual void DestroyAudioReceiveStream( | 114 virtual void DestroyAudioReceiveStream( |
| 110 AudioReceiveStream* receive_stream) = 0; | 115 AudioReceiveStream* receive_stream) = 0; |
| 111 | 116 |
| 112 virtual VideoSendStream* CreateVideoSendStream( | 117 virtual VideoSendStream* CreateVideoSendStream( |
| (...skipping 46 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
| 159 const rtc::NetworkRoute& network_route) = 0; | 164 const rtc::NetworkRoute& network_route) = 0; |
| 160 | 165 |
| 161 virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0; | 166 virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0; |
| 162 | 167 |
| 163 virtual ~Call() {} | 168 virtual ~Call() {} |
| 164 }; | 169 }; |
| 165 | 170 |
| 166 } // namespace webrtc | 171 } // namespace webrtc |
| 167 | 172 |
| 168 #endif // WEBRTC_CALL_CALL_H_ | 173 #endif // WEBRTC_CALL_CALL_H_ |
| OLD | NEW |
