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1f38a38b6f
Setting the rtp header extensions on the packet delivery thread (currently worker, soon to be network), is now possible without taking the hit of deleting and recreating the receive stream (and rtp receiver and related state). Bug: webrtc:11993 Change-Id: I9bbe306844a25d85d79cd216092ead66eaf68960 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223741 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34953}
93 lines
3.3 KiB
C++
93 lines
3.3 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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namespace webrtc {
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// Common base interface for MediaReceiveStream based classes and
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// FlexfecReceiveStream.
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class ReceiveStream {
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public:
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// Receive-stream specific RTP settings.
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struct RtpConfig {
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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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// Called on the packet delivery thread since some members of the config may
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// change mid-stream (e.g. the local ssrc). All mutation must also happen on
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// the packet delivery thread. Return value can be assumed to
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// only be used in the calling context (on the stack basically).
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virtual const RtpConfig& rtp_config() const = 0;
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protected:
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virtual ~ReceiveStream() {}
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};
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// Either an audio or video receive stream.
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class MediaReceiveStream : public ReceiveStream {
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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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