| /* |
| * 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 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_ |