| /* |
| * 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 <vector> |
| |
| #include "api/crypto/frame_decryptor_interface.h" |
| #include "api/frame_transformer_interface.h" |
| #include "api/media_types.h" |
| #include "api/scoped_refptr.h" |
| #include "api/transport/rtp/rtp_source.h" |
| #include "modules/rtp_rtcp/include/rtp_header_extension_map.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; |
| |
| // Enable feedback for send side bandwidth estimation. |
| // See |
| // https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions |
| // for details. |
| // This value may change mid-stream and must be done on the same thread |
| // that the value is read on (i.e. packet delivery). |
| bool transport_cc = false; |
| |
| // RTP header extensions used for the received stream. |
| // This value may change mid-stream and must be done on the same thread |
| // that the value is read on (i.e. packet delivery). |
| std::vector<RtpExtension> extensions; |
| }; |
| |
| // Set/change the rtp header extensions. Must be called on the packet |
| // delivery thread. |
| virtual void SetRtpExtensions(std::vector<RtpExtension> extensions) = 0; |
| virtual RtpHeaderExtensionMap GetRtpExtensionMap() const = 0; |
| |
| // Returns a bool for whether feedback for send side bandwidth estimation is |
| // enabled. See |
| // https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions |
| // for details. |
| // This value may change mid-stream and must be done on the same thread |
| // that the value is read on (i.e. packet delivery). |
| virtual bool transport_cc() const = 0; |
| |
| virtual void SetTransportCc(bool transport_cc) = 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; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // CALL_RECEIVE_STREAM_H_ |