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/*
* 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)>;
// 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_
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