mirror of
https://github.com/oxen-io/session-android.git
synced 2024-11-28 20:45:17 +00:00
153 lines
4.8 KiB
C++
153 lines
4.8 KiB
C++
/*
|
|
* 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_
|