session-android/jni/webrtc/modules/audio_coding/main/test/utility.h

153 lines
4.8 KiB
C
Raw Normal View History

/*
* 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/interface/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 DTMFDetector : public AudioCodingFeedback {
public:
DTMFDetector();
~DTMFDetector();
// used for inband DTMF detection
int32_t IncomingDtmf(const uint8_t digitDtmf, const bool toneEnded);
void PrintDetectedDigits();
private:
uint32_t _toneCntr[1000];
};
class VADCallback : public ACMVADCallback {
public:
VADCallback();
~VADCallback() {
}
int32_t InFrameType(int16_t frameType);
void PrintFrameTypes();
void Reset();
private:
uint32_t _numFrameTypes[6];
};
void UseLegacyAcm(webrtc::Config* config);
void UseNewAcm(webrtc::Config* config);
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_