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Diffstat (limited to 'webrtc/modules/audio_coding/main/test/RTPFile.h')
-rw-r--r-- | webrtc/modules/audio_coding/main/test/RTPFile.h | 126 |
1 files changed, 0 insertions, 126 deletions
diff --git a/webrtc/modules/audio_coding/main/test/RTPFile.h b/webrtc/modules/audio_coding/main/test/RTPFile.h deleted file mode 100644 index c79b63e164..0000000000 --- a/webrtc/modules/audio_coding/main/test/RTPFile.h +++ /dev/null @@ -1,126 +0,0 @@ -/* - * Copyright (c) 2012 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_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_ - -#include <stdio.h> -#include <queue> - -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" -#include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/system_wrappers/include/rw_lock_wrapper.h" -#include "webrtc/typedefs.h" - -namespace webrtc { - -class RTPStream { - public: - virtual ~RTPStream() { - } - - virtual void Write(const uint8_t payloadType, const uint32_t timeStamp, - const int16_t seqNo, const uint8_t* payloadData, - const size_t payloadSize, uint32_t frequency) = 0; - - // Returns the packet's payload size. Zero should be treated as an - // end-of-stream (in the case that EndOfFile() is true) or an error. - virtual size_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData, - size_t payloadSize, uint32_t* offset) = 0; - virtual bool EndOfFile() const = 0; - - protected: - void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType, int16_t seqNo, - uint32_t timeStamp, uint32_t ssrc); - - void ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const uint8_t* rtpHeader); -}; - -class RTPPacket { - public: - RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo, - const uint8_t* payloadData, size_t payloadSize, - uint32_t frequency); - - ~RTPPacket(); - - uint8_t payloadType; - uint32_t timeStamp; - int16_t seqNo; - uint8_t* payloadData; - size_t payloadSize; - uint32_t frequency; -}; - -class RTPBuffer : public RTPStream { - public: - RTPBuffer(); - - ~RTPBuffer(); - - void Write(const uint8_t payloadType, - const uint32_t timeStamp, - const int16_t seqNo, - const uint8_t* payloadData, - const size_t payloadSize, - uint32_t frequency) override; - - size_t Read(WebRtcRTPHeader* rtpInfo, - uint8_t* payloadData, - size_t payloadSize, - uint32_t* offset) override; - - bool EndOfFile() const override; - - private: - RWLockWrapper* _queueRWLock; - std::queue<RTPPacket *> _rtpQueue; -}; - -class RTPFile : public RTPStream { - public: - ~RTPFile() { - } - - RTPFile() - : _rtpFile(NULL), - _rtpEOF(false) { - } - - void Open(const char *outFilename, const char *mode); - - void Close(); - - void WriteHeader(); - - void ReadHeader(); - - void Write(const uint8_t payloadType, - const uint32_t timeStamp, - const int16_t seqNo, - const uint8_t* payloadData, - const size_t payloadSize, - uint32_t frequency) override; - - size_t Read(WebRtcRTPHeader* rtpInfo, - uint8_t* payloadData, - size_t payloadSize, - uint32_t* offset) override; - - bool EndOfFile() const override { return _rtpEOF; } - - private: - FILE* _rtpFile; - bool _rtpEOF; -}; - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_ |