Index: webrtc/modules/audio_coding/main/test/RTPFile.h |
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 6bad755af9d390ca8330ea117898683cc14f6859..0000000000000000000000000000000000000000 |
--- 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/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_MAIN_TEST_RTPFILE_H_ |