|  | /* | 
|  | *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved. | 
|  | * | 
|  | *  Use of this source code is governed by a BSD-style license | 
|  | *  that can be found in the LICENSE file in the root of the source | 
|  | *  tree. An additional intellectual property rights grant can be found | 
|  | *  in the file PATENTS.  All contributing project authors may | 
|  | *  be found in the AUTHORS file in the root of the source tree. | 
|  | */ | 
|  |  | 
|  | #ifndef CALL_RECEIVE_STREAM_H_ | 
|  | #define CALL_RECEIVE_STREAM_H_ | 
|  |  | 
|  | #include <cstdint> | 
|  | #include <vector> | 
|  |  | 
|  | #include "api/crypto/frame_decryptor_interface.h" | 
|  | #include "api/frame_transformer_interface.h" | 
|  | #include "api/rtp_headers.h" | 
|  | #include "api/scoped_refptr.h" | 
|  | #include "api/transport/rtp/rtp_source.h" | 
|  |  | 
|  | namespace webrtc { | 
|  |  | 
|  | // Common base interface for MediaReceiveStreamInterface based classes and | 
|  | // FlexfecReceiveStream. | 
|  | class ReceiveStreamInterface { | 
|  | public: | 
|  | // Receive-stream specific RTP settings. | 
|  | // TODO(tommi): This struct isn't needed at this level anymore. Move it closer | 
|  | // to where it's used. | 
|  | struct ReceiveStreamRtpConfig { | 
|  | // Synchronization source (stream identifier) to be received. | 
|  | // This member will not change mid-stream and can be assumed to be const | 
|  | // post initialization. | 
|  | uint32_t remote_ssrc = 0; | 
|  |  | 
|  | // Sender SSRC used for sending RTCP (such as receiver reports). | 
|  | // This value may change mid-stream and must be done on the same thread | 
|  | // that the value is read on (i.e. packet delivery). | 
|  | uint32_t local_ssrc = 0; | 
|  | }; | 
|  |  | 
|  | protected: | 
|  | virtual ~ReceiveStreamInterface() {} | 
|  | }; | 
|  |  | 
|  | // Either an audio or video receive stream. | 
|  | class MediaReceiveStreamInterface : public ReceiveStreamInterface { | 
|  | public: | 
|  | // Starts stream activity. | 
|  | // When a stream is active, it can receive, process and deliver packets. | 
|  | virtual void Start() = 0; | 
|  |  | 
|  | // Stops stream activity. Must be called to match with a previous call to | 
|  | // `Start()`. When a stream has been stopped, it won't receive, decode, | 
|  | // process or deliver packets to downstream objects such as callback pointers | 
|  | // set in the config struct. | 
|  | virtual void Stop() = 0; | 
|  |  | 
|  | virtual void SetDepacketizerToDecoderFrameTransformer( | 
|  | scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) = 0; | 
|  |  | 
|  | virtual void SetFrameDecryptor( | 
|  | scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) = 0; | 
|  |  | 
|  | virtual std::vector<RtpSource> GetSources() const = 0; | 
|  |  | 
|  | virtual void SetRtcpMode(RtcpMode mode) = 0; | 
|  | }; | 
|  |  | 
|  | }  // namespace webrtc | 
|  |  | 
|  | #endif  // CALL_RECEIVE_STREAM_H_ |