bad99ab253
reduced-size RTCP, i.e. not prefixing RTCP packets with either a sender report or receiver report has been implemented for a long time but only for video. This CL adds it for audio as well. This reduces the size of audio NACKs (16 bytes, typically one NACK per packet) sent by not prefixing it with a receiver report (32 bytes). Other packets are not affected as e.g. transport-cc feedback does not add a RR even though that is technically required. The effect on NACK can be tested by running Chromium with --disable-webrtc-encryption --force-fieldtrials=WebRTC-FakeNetworkReceiveConfig/loss_percent:5/ against this fiddle negotiating audio nack: https://jsfiddle.net/fippo/8ubtLnfx/1/ BUG=webrtc:340041654 Change-Id: I06fb94742ff1b6f9a464c404bfc53913f23498d8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350269 Commit-Queue: Philipp Hancke <phancke@meta.com> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42330}
75 lines
2.4 KiB
C++
75 lines
2.4 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 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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};
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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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virtual void SetRtcpMode(RtcpMode mode) = 0;
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};
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} // namespace webrtc
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#endif // CALL_RECEIVE_STREAM_H_
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