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audio_coding: remove "main" directory

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This is the last piece of the old directory layout of the modules.

Duplicated header files are left in audio_coding/main/include until
downstream code is updated to the new location. They have pragma
warnings added to them and identical header guards as the new headers to avoid breaking things.

BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
NOTRY=True
NOPRESUBMIT=True

Review URL: https://codereview.webrtc.org/1481493004

Cr-Commit-Position: refs/heads/master@{#10803}
This commit is contained in:
kjellander
2015-11-26 04:44:54 -08:00
committed by Commit bot
parent a3c20bb9a0
commit 3e6db2321c
85 changed files with 1276 additions and 498 deletions
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139
webrtc/modules/audio_coding/test/utility.h Normal file
View File

@ -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_