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https://github.com/oxen-io/session-android.git
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d83a3d71bc
Merge in RedPhone // FREEBIE
153 lines
4.8 KiB
C++
153 lines
4.8 KiB
C++
/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_
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#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
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namespace webrtc {
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//-----------------------------
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#define CHECK_ERROR(f) \
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do { \
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EXPECT_GE(f, 0) << "Error Calling API"; \
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} while(0)
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//-----------------------------
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#define CHECK_PROTECTED(f) \
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do { \
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if (f >= 0) { \
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ADD_FAILURE() << "Error Calling API"; \
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} else { \
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printf("An expected error is caught.\n"); \
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} \
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} while(0)
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//----------------------------
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#define CHECK_ERROR_MT(f) \
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do { \
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if (f < 0) { \
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fprintf(stderr, "Error Calling API in file %s at line %d \n", \
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__FILE__, __LINE__); \
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} \
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} while(0)
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//----------------------------
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#define CHECK_PROTECTED_MT(f) \
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do { \
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if (f >= 0) { \
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fprintf(stderr, "Error Calling API in file %s at line %d \n", \
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__FILE__, __LINE__); \
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} else { \
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printf("An expected error is caught.\n"); \
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} \
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} while(0)
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#define DELETE_POINTER(p) \
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do { \
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if (p != NULL) { \
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delete p; \
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p = NULL; \
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} \
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} while(0)
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class ACMTestTimer {
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public:
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ACMTestTimer();
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~ACMTestTimer();
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void Reset();
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void Tick10ms();
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void Tick1ms();
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void Tick100ms();
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void Tick1sec();
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void CurrentTimeHMS(char* currTime);
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void CurrentTime(unsigned long& h, unsigned char& m, unsigned char& s,
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unsigned short& ms);
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private:
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void Adjust();
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unsigned short _msec;
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unsigned char _sec;
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unsigned char _min;
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unsigned long _hour;
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};
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class CircularBuffer {
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public:
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CircularBuffer(uint32_t len);
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~CircularBuffer();
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void SetArithMean(bool enable);
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void SetVariance(bool enable);
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void Update(const double newVal);
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void IsBufferFull();
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int16_t Variance(double& var);
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int16_t ArithMean(double& mean);
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protected:
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double* _buff;
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uint32_t _idx;
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uint32_t _buffLen;
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bool _buffIsFull;
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bool _calcAvg;
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bool _calcVar;
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double _sum;
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double _sumSqr;
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};
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int16_t ChooseCodec(CodecInst& codecInst);
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void PrintCodecs();
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bool FixedPayloadTypeCodec(const char* payloadName);
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class DTMFDetector : public AudioCodingFeedback {
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public:
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DTMFDetector();
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~DTMFDetector();
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// used for inband DTMF detection
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int32_t IncomingDtmf(const uint8_t digitDtmf, const bool toneEnded);
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void PrintDetectedDigits();
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private:
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uint32_t _toneCntr[1000];
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};
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class VADCallback : public ACMVADCallback {
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public:
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VADCallback();
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~VADCallback() {
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}
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int32_t InFrameType(int16_t frameType);
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void PrintFrameTypes();
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void Reset();
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private:
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uint32_t _numFrameTypes[6];
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};
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void UseLegacyAcm(webrtc::Config* config);
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void UseNewAcm(webrtc::Config* config);
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_
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