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The feauture was added in https://webrtc-review.googlesource.com/c/src/+/291119 in order to ensure RTX packet is part of BWE even before the RTP stream is known. However, it cause an issue if media is signaled with an SSRC that has this RTX SSRC. Since BWE is now notified about received packets before demuxing to the correct receive stream, it is not necessary to demux RTX packets before the media SSRC is known. Note that WebRTC require at least one negotiated SSRC/MID before RTCP feedback can be sent. Ie, for BWE to work, at least one media SSRC must be known after this cl. It can either be unsignaled or signaled. BWE tested with BweRampupWithInitialProbeTest. Bug: webrtc:14795, webrtc:14817, b/320258158 Change-Id: Icf2c67bedc352720bf846b9ee38d509346af36f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340141 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41785}
45 lines
1.5 KiB
C++
45 lines
1.5 KiB
C++
/*
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* Copyright (c) 2018 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_PACKET_RECEIVER_H_
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#define CALL_PACKET_RECEIVER_H_
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#include "absl/functional/any_invocable.h"
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#include "api/media_types.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/copy_on_write_buffer.h"
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namespace webrtc {
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class PacketReceiver {
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public:
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// Demux RTCP packets. Must be called on the worker thread.
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virtual void DeliverRtcpPacket(rtc::CopyOnWriteBuffer packet) = 0;
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// Invoked once when a packet is received that can not be demuxed.
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// If the method returns true, a new attempt is made to demux the packet.
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using OnUndemuxablePacketHandler =
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absl::AnyInvocable<bool(const RtpPacketReceived& parsed_packet)>;
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// Must be called on the worker thread.
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// If `media_type` is not Audio or Video, packets may be used for BWE
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// calculations but are not demuxed.
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virtual void DeliverRtpPacket(
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MediaType media_type,
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RtpPacketReceived packet,
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OnUndemuxablePacketHandler undemuxable_packet_handler) = 0;
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protected:
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virtual ~PacketReceiver() {}
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};
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} // namespace webrtc
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#endif // CALL_PACKET_RECEIVER_H_
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