webrtc/call/packet_receiver.h
Per K 664cf14f9f Reland "Delete PacketReceiver::DeliverPacket from all implementations"
This reverts commit f2a083f262.

Reason for revert: Test problem fixed in https://webrtc-review.googlesource.com/c/src/+/291333.

Original change's description:
> Revert "Delete PacketReceiver::DeliverPacket from all implementations"
>
> This reverts commit 897ea04db5.
>
> 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}

Bug: webrtc:7135,webrtc:14795,b/266658815
Change-Id: Ia640f4342a1f42012ba5295003e17aef7613ad80
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291440
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39199}
2023-01-25 18:18:29 +00:00

43 lines
1.4 KiB
C++

/*
* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef CALL_PACKET_RECEIVER_H_
#define CALL_PACKET_RECEIVER_H_
#include "absl/functional/any_invocable.h"
#include "api/media_types.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/checks.h"
#include "rtc_base/copy_on_write_buffer.h"
namespace webrtc {
class PacketReceiver {
public:
// Demux RTCP packets. Must be called on the worker thread.
virtual void DeliverRtcpPacket(rtc::CopyOnWriteBuffer packet) = 0;
// Invoked once when a packet packet is received that can not be demuxed.
// If the method returns true, a new attempt is made to demux the packet.
using OnUndemuxablePacketHandler =
absl::AnyInvocable<bool(const RtpPacketReceived& parsed_packet)>;
// Demux RTP packets. Must be called on the worker thread.
virtual void DeliverRtpPacket(
MediaType media_type,
RtpPacketReceived packet,
OnUndemuxablePacketHandler undemuxable_packet_handler) = 0;
protected:
virtual ~PacketReceiver() {}
};
} // namespace webrtc
#endif // CALL_PACKET_RECEIVER_H_