| Index: webrtc/modules/audio_coding/main/test/utility.h
|
| diff --git a/webrtc/modules/audio_coding/main/test/utility.h b/webrtc/modules/audio_coding/main/test/utility.h
|
| deleted file mode 100644
|
| index e936ec1cddaf484dfdf523f676ce750d64a33fa6..0000000000000000000000000000000000000000
|
| --- a/webrtc/modules/audio_coding/main/test/utility.h
|
| +++ /dev/null
|
| @@ -1,139 +0,0 @@
|
| -/*
|
| - * Copyright (c) 2011 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_UTILITY_H_
|
| -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_
|
| -
|
| -#include "testing/gtest/include/gtest/gtest.h"
|
| -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
|
| -
|
| -namespace webrtc {
|
| -
|
| -//-----------------------------
|
| -#define CHECK_ERROR(f) \
|
| - do { \
|
| - EXPECT_GE(f, 0) << "Error Calling API"; \
|
| - } while(0)
|
| -
|
| -//-----------------------------
|
| -#define CHECK_PROTECTED(f) \
|
| - do { \
|
| - if (f >= 0) { \
|
| - ADD_FAILURE() << "Error Calling API"; \
|
| - } else { \
|
| - printf("An expected error is caught.\n"); \
|
| - } \
|
| - } while(0)
|
| -
|
| -//----------------------------
|
| -#define CHECK_ERROR_MT(f) \
|
| - do { \
|
| - if (f < 0) { \
|
| - fprintf(stderr, "Error Calling API in file %s at line %d \n", \
|
| - __FILE__, __LINE__); \
|
| - } \
|
| - } while(0)
|
| -
|
| -//----------------------------
|
| -#define CHECK_PROTECTED_MT(f) \
|
| - do { \
|
| - if (f >= 0) { \
|
| - fprintf(stderr, "Error Calling API in file %s at line %d \n", \
|
| - __FILE__, __LINE__); \
|
| - } else { \
|
| - printf("An expected error is caught.\n"); \
|
| - } \
|
| - } while(0)
|
| -
|
| -#define DELETE_POINTER(p) \
|
| - do { \
|
| - if (p != NULL) { \
|
| - delete p; \
|
| - p = NULL; \
|
| - } \
|
| - } while(0)
|
| -
|
| -class ACMTestTimer {
|
| - public:
|
| - ACMTestTimer();
|
| - ~ACMTestTimer();
|
| -
|
| - void Reset();
|
| - void Tick10ms();
|
| - void Tick1ms();
|
| - void Tick100ms();
|
| - void Tick1sec();
|
| - void CurrentTimeHMS(char* currTime);
|
| - void CurrentTime(unsigned long& h, unsigned char& m, unsigned char& s,
|
| - unsigned short& ms);
|
| -
|
| - private:
|
| - void Adjust();
|
| -
|
| - unsigned short _msec;
|
| - unsigned char _sec;
|
| - unsigned char _min;
|
| - unsigned long _hour;
|
| -};
|
| -
|
| -class CircularBuffer {
|
| - public:
|
| - CircularBuffer(uint32_t len);
|
| - ~CircularBuffer();
|
| -
|
| - void SetArithMean(bool enable);
|
| - void SetVariance(bool enable);
|
| -
|
| - void Update(const double newVal);
|
| - void IsBufferFull();
|
| -
|
| - int16_t Variance(double& var);
|
| - int16_t ArithMean(double& mean);
|
| -
|
| - protected:
|
| - double* _buff;
|
| - uint32_t _idx;
|
| - uint32_t _buffLen;
|
| -
|
| - bool _buffIsFull;
|
| - bool _calcAvg;
|
| - bool _calcVar;
|
| - double _sum;
|
| - double _sumSqr;
|
| -};
|
| -
|
| -int16_t ChooseCodec(CodecInst& codecInst);
|
| -
|
| -void PrintCodecs();
|
| -
|
| -bool FixedPayloadTypeCodec(const char* payloadName);
|
| -
|
| -class VADCallback : public ACMVADCallback {
|
| - public:
|
| - VADCallback();
|
| - ~VADCallback() {
|
| - }
|
| -
|
| - int32_t InFrameType(FrameType frame_type);
|
| -
|
| - void PrintFrameTypes();
|
| - void Reset();
|
| -
|
| - private:
|
| - uint32_t _numFrameTypes[5];
|
| -};
|
| -
|
| -void UseLegacyAcm(webrtc::Config* config);
|
| -
|
| -void UseNewAcm(webrtc::Config* config);
|
| -
|
| -} // namespace webrtc
|
| -
|
| -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_
|
|
|