Repo created

TMessagesProj/jni/voip/webrtc/call/packet_receiver.h

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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_