102 lines
3.7 KiB
C++
102 lines
3.7 KiB
C++
/*
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* Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef CALL_RECEIVE_STREAM_H_
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#define CALL_RECEIVE_STREAM_H_
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#include <vector>
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#include "api/crypto/frame_decryptor_interface.h"
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#include "api/frame_transformer_interface.h"
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#include "api/media_types.h"
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#include "api/scoped_refptr.h"
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#include "api/transport/rtp/rtp_source.h"
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#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
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namespace webrtc {
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// Common base interface for MediaReceiveStreamInterface based classes and
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// FlexfecReceiveStream.
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class ReceiveStreamInterface {
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public:
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// Receive-stream specific RTP settings.
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// TODO(tommi): This struct isn't needed at this level anymore. Move it closer
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// to where it's used.
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struct ReceiveStreamRtpConfig {
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// Synchronization source (stream identifier) to be received.
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// This member will not change mid-stream and can be assumed to be const
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// post initialization.
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uint32_t remote_ssrc = 0;
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// Sender SSRC used for sending RTCP (such as receiver reports).
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// This value may change mid-stream and must be done on the same thread
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// that the value is read on (i.e. packet delivery).
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uint32_t local_ssrc = 0;
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// Enable feedback for send side bandwidth estimation.
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// See
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// https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions
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// for details.
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// This value may change mid-stream and must be done on the same thread
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// that the value is read on (i.e. packet delivery).
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bool transport_cc = false;
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// RTP header extensions used for the received stream.
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// This value may change mid-stream and must be done on the same thread
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// that the value is read on (i.e. packet delivery).
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std::vector<RtpExtension> extensions;
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};
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// Set/change the rtp header extensions. Must be called on the packet
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// delivery thread.
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virtual void SetRtpExtensions(std::vector<RtpExtension> extensions) = 0;
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virtual RtpHeaderExtensionMap GetRtpExtensionMap() const = 0;
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// Returns a bool for whether feedback for send side bandwidth estimation is
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// enabled. See
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// https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions
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// for details.
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// This value may change mid-stream and must be done on the same thread
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// that the value is read on (i.e. packet delivery).
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virtual bool transport_cc() const = 0;
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virtual void SetTransportCc(bool transport_cc) = 0;
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protected:
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virtual ~ReceiveStreamInterface() {}
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};
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// Either an audio or video receive stream.
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class MediaReceiveStreamInterface : public ReceiveStreamInterface {
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public:
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// Starts stream activity.
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// When a stream is active, it can receive, process and deliver packets.
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virtual void Start() = 0;
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// Stops stream activity. Must be called to match with a previous call to
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// `Start()`. When a stream has been stopped, it won't receive, decode,
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// process or deliver packets to downstream objects such as callback pointers
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// set in the config struct.
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virtual void Stop() = 0;
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virtual void SetDepacketizerToDecoderFrameTransformer(
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rtc::scoped_refptr<webrtc::FrameTransformerInterface>
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frame_transformer) = 0;
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virtual void SetFrameDecryptor(
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rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) = 0;
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virtual std::vector<RtpSource> GetSources() const = 0;
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};
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} // namespace webrtc
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#endif // CALL_RECEIVE_STREAM_H_
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