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| 1 /* | |
| 2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. | |
| 3 * | |
| 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 | |
| 6 * tree. An additional intellectual property rights grant can be found | |
| 7 * in the file PATENTS. All contributing project authors may | |
| 8 * be found in the AUTHORS file in the root of the source tree. | |
| 9 */ | |
| 10 | |
| 11 #ifndef WEBRTC_CALL_RTX_RECEIVE_STREAM_H_ | |
| 12 #define WEBRTC_CALL_RTX_RECEIVE_STREAM_H_ | |
| 13 | |
| 14 #include <map> | |
| 15 | |
| 16 #include "webrtc/call/rtp_demuxer.h" | |
| 17 | |
| 18 namespace webrtc { | |
| 19 | |
| 20 class RtxReceiveStream : public RtpPacketSinkInterface { | |
| 21 public: | |
| 22 RtxReceiveStream(RtpPacketSinkInterface* media_sink, | |
| 23 const std::map<int, int> rtx_payload_type_map, | |
|
danilchap
2017/05/17 13:43:56
remove const
(it is helpful for const&, but when p
nisse-webrtc
2017/05/18 09:13:20
Ooop, I left this in as an experiment, I would hav
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| 24 uint32_t media_ssrc); | |
| 25 | |
| 26 // RtpPacketSinkInterface. | |
| 27 void OnRtpPacket(const RtpPacketReceived& packet) override; | |
| 28 | |
| 29 private: | |
| 30 RtpPacketSinkInterface* const media_sink_; | |
| 31 // Mapping rtx_payload_type_map_[rtx] = associated. | |
| 32 const std::map<int, int> rtx_payload_type_map_; | |
| 33 // TODO(nisse): Ultimately, the media receive stream shouldn't care about the | |
| 34 // ssrc, and we should delete this. | |
| 35 const uint32_t media_ssrc_; | |
| 36 }; | |
| 37 | |
| 38 } // namespace webrtc | |
| 39 | |
| 40 #endif // WEBRTC_CALL_RTX_RECEIVE_STREAM_H_ | |
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