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Diffstat (limited to 'webrtc/modules/audio_coding/test/RTPFile.h')
-rw-r--r-- | webrtc/modules/audio_coding/test/RTPFile.h | 126 |
1 files changed, 126 insertions, 0 deletions
diff --git a/webrtc/modules/audio_coding/test/RTPFile.h b/webrtc/modules/audio_coding/test/RTPFile.h new file mode 100644 index 0000000000..696d41ebd2 --- /dev/null +++ b/webrtc/modules/audio_coding/test/RTPFile.h @@ -0,0 +1,126 @@ +/* + * 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_TEST_RTPFILE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_TEST_RTPFILE_H_ + +#include <stdio.h> +#include <queue> + +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" +#include "webrtc/modules/include/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_TEST_RTPFILE_H_ |