mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 05:40:42 +01:00
93c9aa1914
with downstream fixes. Bug: webrtc:42226242 Change-Id: I88d7b5ffc1f86c01ea13948c27b4210d032f4190 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361360 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42921}
44 lines
1.5 KiB
C++
44 lines
1.5 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/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 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)>;
|
|
|
|
// Must be called on the worker thread.
|
|
// If `media_type` is not Audio or Video, packets may be used for BWE
|
|
// calculations but are not demuxed.
|
|
virtual void DeliverRtpPacket(
|
|
MediaType media_type,
|
|
RtpPacketReceived packet,
|
|
OnUndemuxablePacketHandler undemuxable_packet_handler) = 0;
|
|
|
|
protected:
|
|
virtual ~PacketReceiver() {}
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // CALL_PACKET_RECEIVER_H_
|