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In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
138 lines
4.5 KiB
C++
138 lines
4.5 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 MODULES_AUDIO_CODING_TEST_UTILITY_H_
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#define MODULES_AUDIO_CODING_TEST_UTILITY_H_
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#include "modules/audio_coding/include/audio_coding_module.h"
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#include "test/gtest.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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// To avoid clashes with CircularBuffer in APM.
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namespace test {
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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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} // namespace test
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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 VADCallback : public ACMVADCallback {
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public:
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VADCallback();
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int32_t InFrameType(FrameType frame_type) override;
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void PrintFrameTypes();
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void Reset();
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private:
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uint32_t _numFrameTypes[5];
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_TEST_UTILITY_H_
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