| /* | 
 |  *  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( | 
 |       rtc::scoped_refptr<webrtc::FrameTransformerInterface> | 
 |           frame_transformer) = 0; | 
 |  | 
 |   virtual void SetFrameDecryptor( | 
 |       rtc::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_ |