diff options
Diffstat (limited to 'webrtc/modules/audio_coding/test/utility.h')
| -rw-r--r-- | webrtc/modules/audio_coding/test/utility.h | 139 |
1 files changed, 139 insertions, 0 deletions
diff --git a/webrtc/modules/audio_coding/test/utility.h b/webrtc/modules/audio_coding/test/utility.h new file mode 100644 index 0000000000..23869be7ed --- /dev/null +++ b/webrtc/modules/audio_coding/test/utility.h @@ -0,0 +1,139 @@ +/* + * 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_TEST_UTILITY_H_ +#define WEBRTC_MODULES_AUDIO_CODING_TEST_UTILITY_H_ + +#include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/modules/audio_coding/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_TEST_UTILITY_H_ |