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RTPStream, and NetEq as such. Also mark all other virtual overrides in the same files. This will make further changes to these classes safer by ensuring that the compile breaks if the base class changes and not all overrides are fixed. This also deletes ACMTest.cc, which existed solely to define ~ACMTest(), which was marked pure virtual in the header. (Pure virtual destructors still need a definition.) Because there is another pure virtual method in this class, the class is already abstract, so there's no benefit to making the desturctor pure. Making it non-pure allows removing the separate source file. BUG=none TEST=none R=henrik.lundin@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/29389004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7144 4adac7df-926f-26a2-2b94-8c16560cd09d
119 lines
3.3 KiB
C++
119 lines
3.3 KiB
C++
/*
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* Copyright (c) 2012 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_MAIN_TEST_RTPFILE_H_
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#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_
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#include <stdio.h>
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#include <queue>
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#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
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#include "webrtc/modules/interface/module_common_types.h"
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#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class RTPStream {
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public:
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virtual ~RTPStream() {
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}
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virtual void Write(const uint8_t payloadType, const uint32_t timeStamp,
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const int16_t seqNo, const uint8_t* payloadData,
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const uint16_t payloadSize, uint32_t frequency) = 0;
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// Returns the packet's payload size. Zero should be treated as an
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// end-of-stream (in the case that EndOfFile() is true) or an error.
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virtual uint16_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
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uint16_t payloadSize, uint32_t* offset) = 0;
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virtual bool EndOfFile() const = 0;
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protected:
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void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType, int16_t seqNo,
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uint32_t timeStamp, uint32_t ssrc);
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void ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const uint8_t* rtpHeader);
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};
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class RTPPacket {
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public:
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RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo,
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const uint8_t* payloadData, uint16_t payloadSize,
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uint32_t frequency);
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~RTPPacket();
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uint8_t payloadType;
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uint32_t timeStamp;
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int16_t seqNo;
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uint8_t* payloadData;
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uint16_t payloadSize;
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uint32_t frequency;
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};
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class RTPBuffer : public RTPStream {
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public:
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RTPBuffer();
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~RTPBuffer();
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virtual void Write(const uint8_t payloadType, const uint32_t timeStamp,
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const int16_t seqNo, const uint8_t* payloadData,
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const uint16_t payloadSize, uint32_t frequency) OVERRIDE;
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virtual uint16_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
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uint16_t payloadSize, uint32_t* offset) OVERRIDE;
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virtual bool EndOfFile() const OVERRIDE;
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private:
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RWLockWrapper* _queueRWLock;
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std::queue<RTPPacket *> _rtpQueue;
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};
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class RTPFile : public RTPStream {
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public:
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~RTPFile() {
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}
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RTPFile()
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: _rtpFile(NULL),
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_rtpEOF(false) {
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}
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void Open(const char *outFilename, const char *mode);
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void Close();
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void WriteHeader();
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void ReadHeader();
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virtual void Write(const uint8_t payloadType, const uint32_t timeStamp,
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const int16_t seqNo, const uint8_t* payloadData,
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const uint16_t payloadSize, uint32_t frequency) OVERRIDE;
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virtual uint16_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
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uint16_t payloadSize, uint32_t* offset) OVERRIDE;
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virtual bool EndOfFile() const OVERRIDE {
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return _rtpEOF;
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}
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private:
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FILE* _rtpFile;
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bool _rtpEOF;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_
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