810acbc93e
These new classes are intended to replace the old NETEQTEST_RTPpacket classes. The code in rtp_analyze.cc has been updated to use the new classes; other test applications will follow. BUG=2692 R=andrew@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11769004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5901 4adac7df-926f-26a2-2b94-8c16560cd09d
37 lines
1.0 KiB
C++
37 lines
1.0 KiB
C++
/*
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* Copyright (c) 2014 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_MODULES_AUDIO_CODING_NETEQ4_TOOLS_PACKET_SOURCE_H_
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#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_TOOLS_PACKET_SOURCE_H_
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#include "webrtc/system_wrappers/interface/constructor_magic.h"
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namespace webrtc {
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namespace test {
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class Packet;
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// Interface class for an object delivering RTP packets to test applications.
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class PacketSource {
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public:
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PacketSource() {}
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virtual ~PacketSource() {}
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// Returns a pointer to the next packet.
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virtual Packet* NextPacket() = 0;
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private:
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DISALLOW_COPY_AND_ASSIGN(PacketSource);
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};
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} // namespace test
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_TOOLS_PACKET_SOURCE_H_
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