f2a083f262
This reverts commit 897ea04db5db2e591e28bd884191be58d9bcdc63. Reason for revert: Speculative revert as it could be the reason why perf tests started failing: https://ci.chromium.org/p/webrtc/g/perf/console?limit=200 Original change's description: > Delete PacketReceiver::DeliverPacket from all implementations > > And fix tests that still depend on extensions to be known by the receiver. > > Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3 > > Bug: webrtc:7135,webrtc:14795 > Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290996 > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#39184} Bug: webrtc:7135,webrtc:14795,b/266658815 Change-Id: I9d03f4952938d176ffee110a707acadc1846457c No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291400 Commit-Queue: Andrey Logvin <landrey@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Owners-Override: Andrey Logvin <landrey@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@google.com> Cr-Commit-Position: refs/heads/main@{#39189}
66 lines
2.1 KiB
C++
66 lines
2.1 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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// TODO(perkj, https://bugs.webrtc.org/7135): Remove this method. This method
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// is no longer used by PeerConnections. Some tests still use it.
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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) {
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RTC_CHECK_NOTREACHED();
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}
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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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