cb83bdf01f
1. Demux RTCP messages according to the sender-SSRC.
2. Demux RTCP messages according to the RSID (resolved to an SSRC, then compared to the sender-RTCP).
3. Allow listening in on all RTCP messages passing through the demuxer ("broadcast sinks").
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2943693003
Cr-Commit-Position: refs/heads/master@{#18763}
30 lines
1.0 KiB
C++
30 lines
1.0 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_CALL_RTCP_PACKET_SINK_INTERFACE_H_
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#define WEBRTC_CALL_RTCP_PACKET_SINK_INTERFACE_H_
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#include "webrtc/base/array_view.h"
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namespace webrtc {
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// This class represents a receiver of unparsed RTCP packets.
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// TODO(eladalon): Replace this by demuxing over parsed rather than raw data.
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// Whether this should be over an entire RTCP packet, or over RTCP blocks,
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// is still under discussion.
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class RtcpPacketSinkInterface {
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public:
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virtual ~RtcpPacketSinkInterface() = default;
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virtual void OnRtcpPacket(rtc::ArrayView<const uint8_t> packet) = 0;
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};
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} // namespace webrtc
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#endif // WEBRTC_CALL_RTCP_PACKET_SINK_INTERFACE_H_
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