webrtc/modules/audio_coding/main/test/utility.h - Issue 1481493004: audio_coding: remove "main" directory

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Issue 1481493004: audio_coding: remove "main" directory (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Rebased Created 5 years ago

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1 /*

2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.

3 *

4 * Use of this source code is governed by a BSD-style license

5 * that can be found in the LICENSE file in the root of the source

6 * tree. An additional intellectual property rights grant can be found

7 * in the file PATENTS. All contributing project authors may

8 * be found in the AUTHORS file in the root of the source tree.

9 */

10

11 #ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_

12 #define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_

13

14 #include "testing/gtest/include/gtest/gtest.h"

15 #include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"

16

17 namespace webrtc {

18

19 //-----------------------------

20 #define CHECK_ERROR(f) \

21 do { \

22 EXPECT_GE(f, 0) << "Error Calling API"; \

23 } while(0)

24

25 //-----------------------------

26 #define CHECK_PROTECTED(f) \

27 do { \

28 if (f >= 0) { \

29 ADD_FAILURE() << "Error Calling API"; \

30 } else { \

31 printf("An expected error is caught.\n"); \

32 } \

33 } while(0)

34

35 //----------------------------

36 #define CHECK_ERROR_MT(f) \

37 do { \

38 if (f < 0) { \

39 fprintf(stderr, "Error Calling API in file %s at line %d \n", \

40 __FILE__, __LINE__); \

41 } \

42 } while(0)

43

44 //----------------------------

45 #define CHECK_PROTECTED_MT(f) \

46 do { \

47 if (f >= 0) { \

48 fprintf(stderr, "Error Calling API in file %s at line %d \n", \

49 __FILE__, __LINE__); \

50 } else { \

51 printf("An expected error is caught.\n"); \

52 } \

53 } while(0)

54

55 #define DELETE_POINTER(p) \

56 do { \

57 if (p != NULL) { \

58 delete p; \

59 p = NULL; \

60 } \

61 } while(0)

62

63 class ACMTestTimer {

64 public:

65 ACMTestTimer();

66 ~ACMTestTimer();

67

68 void Reset();

69 void Tick10ms();

70 void Tick1ms();

71 void Tick100ms();

72 void Tick1sec();

73 void CurrentTimeHMS(char* currTime);

74 void CurrentTime(unsigned long& h, unsigned char& m, unsigned char& s,

75 unsigned short& ms);

76

77 private:

78 void Adjust();

79

80 unsigned short _msec;

81 unsigned char _sec;

82 unsigned char _min;

83 unsigned long _hour;

84 };

85

86 class CircularBuffer {

87 public:

88 CircularBuffer(uint32_t len);

89 ~CircularBuffer();

90

91 void SetArithMean(bool enable);

92 void SetVariance(bool enable);

93

94 void Update(const double newVal);

95 void IsBufferFull();

96

97 int16_t Variance(double& var);

98 int16_t ArithMean(double& mean);

99

100 protected:

101 double* _buff;

102 uint32_t _idx;

103 uint32_t _buffLen;

104

105 bool _buffIsFull;

106 bool _calcAvg;

107 bool _calcVar;

108 double _sum;

109 double _sumSqr;

110 };

111

112 int16_t ChooseCodec(CodecInst& codecInst);

113

114 void PrintCodecs();

115

116 bool FixedPayloadTypeCodec(const char* payloadName);

117

118 class VADCallback : public ACMVADCallback {

119 public:

120 VADCallback();

121 ~VADCallback() {

122 }

123

124 int32_t InFrameType(FrameType frame_type);

125

126 void PrintFrameTypes();

127 void Reset();

128

129 private:

130 uint32_t _numFrameTypes[5];

131 };

132

133 void UseLegacyAcm(webrtc::Config* config);

134

135 void UseNewAcm(webrtc::Config* config);

136

137 } // namespace webrtc

138

139 #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_

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