webrtc/call/rtcp_packet_sink_interface.h
Mirko Bonadei bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00

29 lines
1 KiB
C++

/*
* Copyright (c) 2017 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 WEBRTC_CALL_RTCP_PACKET_SINK_INTERFACE_H_
#define WEBRTC_CALL_RTCP_PACKET_SINK_INTERFACE_H_
#include "webrtc/api/array_view.h"
namespace webrtc {
// This class represents a receiver of unparsed RTCP packets.
// TODO(eladalon): Replace this by demuxing over parsed rather than raw data.
// Whether this should be over an entire RTCP packet, or over RTCP blocks,
// is still under discussion.
class RtcpPacketSinkInterface {
public:
virtual ~RtcpPacketSinkInterface() = default;
virtual void OnRtcpPacket(rtc::ArrayView<const uint8_t> packet) = 0;
};
} // namespace webrtc
#endif // WEBRTC_CALL_RTCP_PACKET_SINK_INTERFACE_H_