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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 #ifndef WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_RECEIVE_INTERFACE_H_ | |
| 11 #define WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_RECEIVE_INTERFACE_H_ | |
| 12 | |
| 13 #include <memory> | |
| 14 | |
| 15 #include "webrtc/call/rtp_packet_sink_interface.h" | |
| 16 | |
| 17 namespace webrtc { | |
| 18 | |
| 19 // An RtpTransportReceiver is responsible for the rtp-specific but | |
| 20 // media-independent state needed for receiving a stream. | |
|
pthatcher
2017/06/02 22:44:17
"stream" is ambiguous in the RTP world. Needs to
| |
| 21 // TODO(nisse): Currently, only owns the association between ssrc and | |
| 22 // the stream's RtpPacketSinkInterface. Ownership of corresponding | |
| 23 // objects from modules/rtp_rtcp/ should move to this class (or | |
| 24 // rather, the corresponding implmentation class), and we should add | |
| 25 // have methods for getting rtp receive stats. | |
| 26 class RtpTransportReceiver { | |
| 27 public: | |
| 28 virtual ~RtpTransportReceiver() {} | |
| 29 }; | |
| 30 | |
| 31 // This class acts as a factory for RtpTransportReceive objects. | |
| 32 class RtpTransportControllerReceiveInterface { | |
| 33 public: | |
| 34 virtual ~RtpTransportControllerReceiveInterface() {} | |
| 35 | |
| 36 virtual std::unique_ptr<RtpTransportReceiver> CreateReceiver( | |
| 37 uint32_t ssrc, | |
| 38 RtpPacketSinkInterface* sink) = 0; | |
| 39 // For registering additional sinks, needed for FlexFEC. | |
| 40 virtual void AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink) = 0; | |
| 41 virtual size_t RemoveSink(const RtpPacketSinkInterface* sink) = 0; | |
| 42 }; | |
| 43 | |
| 44 } // namespace webrtc | |
| 45 | |
| 46 #endif // WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_RECEIVE_INTERFACE_H_ | |
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