3e6db2321c
This is the last piece of the old directory layout of the modules. Duplicated header files are left in audio_coding/main/include until downstream code is updated to the new location. They have pragma warnings added to them and identical header guards as the new headers to avoid breaking things. BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc NOTRY=True NOPRESUBMIT=True Review URL: https://codereview.webrtc.org/1481493004 Cr-Commit-Position: refs/heads/master@{#10803}
127 lines
3.3 KiB
C++
127 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_TEST_RTPFILE_H_
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#define WEBRTC_MODULES_AUDIO_CODING_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/include/audio_coding_module.h"
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#include "webrtc/modules/include/module_common_types.h"
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#include "webrtc/system_wrappers/include/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 size_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 size_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
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size_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, size_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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size_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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void Write(const uint8_t payloadType,
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const uint32_t timeStamp,
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const int16_t seqNo,
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const uint8_t* payloadData,
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const size_t payloadSize,
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uint32_t frequency) override;
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size_t Read(WebRtcRTPHeader* rtpInfo,
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uint8_t* payloadData,
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size_t payloadSize,
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uint32_t* offset) override;
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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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void Write(const uint8_t payloadType,
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const uint32_t timeStamp,
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const int16_t seqNo,
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const uint8_t* payloadData,
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const size_t payloadSize,
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uint32_t frequency) override;
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size_t Read(WebRtcRTPHeader* rtpInfo,
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uint8_t* payloadData,
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size_t payloadSize,
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uint32_t* offset) override;
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bool EndOfFile() const override { return _rtpEOF; }
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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_TEST_RTPFILE_H_
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