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Diffstat (limited to 'third_party/libwebrtc/call/receive_stream.h')
-rw-r--r-- | third_party/libwebrtc/call/receive_stream.h | 72 |
1 files changed, 72 insertions, 0 deletions
diff --git a/third_party/libwebrtc/call/receive_stream.h b/third_party/libwebrtc/call/receive_stream.h new file mode 100644 index 0000000000..8a99059ec5 --- /dev/null +++ b/third_party/libwebrtc/call/receive_stream.h @@ -0,0 +1,72 @@ +/* + * 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" + +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; +}; + +} // namespace webrtc + +#endif // CALL_RECEIVE_STREAM_H_ |