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DeliverRtpPacket use a parsed RTP packet as argument where the RTP extensions are supposed to be known. This method is implemented in webrt::Call and temporary used by the exising method Call::DeliverRtp, but the idea is to instead avoid extra packet parsing by forwarding a RtpPacketReceived from RtpTransport::DemuxRtpPacket via WebrtcVideoChannel::OnPacketReceived and WebrtcVoiceChannel. DeliverRtcpPacket is also implemented in Call and is directly used in PeerConnection::InitializeRtcpCallback. Bug: webrtc:14795, webrtc:7135 Change-Id: Ib6ffe8e1229ac07fa459ee2fc9a0af8455a23bac Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290401 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39015}
61 lines
2 KiB
C++
61 lines
2 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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enum DeliveryStatus {
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DELIVERY_OK,
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DELIVERY_UNKNOWN_SSRC,
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DELIVERY_PACKET_ERROR,
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};
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virtual DeliveryStatus DeliverPacket(MediaType media_type,
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rtc::CopyOnWriteBuffer packet,
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int64_t packet_time_us) = 0;
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// Demux RTCP packets. Must be called on the worker thread.
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virtual void DeliverRtcpPacket(rtc::CopyOnWriteBuffer packet) {
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// TODO(perkj, https://bugs.webrtc.org/7135): Implement in FakeCall and
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// FakeNetworkPipe.
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RTC_CHECK_NOTREACHED();
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}
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// Invoked once when a packet 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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// Demux RTP packets. Must be called on the worker thread.
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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) {
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// TODO(perkj, https://bugs.webrtc.org/7135): Implement in FakeCall and
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// FakeNetworkPipe.
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RTC_CHECK_NOTREACHED();
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}
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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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