blob: d0e7d5954f18f6572d2ed24957de1ca36912a02a [file] [log] [blame]
/*
* Copyright (c) 2012 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 MODULES_AUDIO_CODING_TEST_ISACTEST_H_
#define MODULES_AUDIO_CODING_TEST_ISACTEST_H_
#include <string.h>
#include <memory>
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/test/ACMTest.h"
#include "modules/audio_coding/test/Channel.h"
#include "modules/audio_coding/test/PCMFile.h"
#include "modules/audio_coding/test/utility.h"
#define MAX_FILE_NAME_LENGTH_BYTE 500
#define NO_OF_CLIENTS 15
namespace webrtc {
struct ACMTestISACConfig {
int32_t currentRateBitPerSec;
int16_t currentFrameSizeMsec;
int16_t encodingMode;
uint32_t initRateBitPerSec;
int16_t initFrameSizeInMsec;
bool enforceFrameSize;
};
class ISACTest : public ACMTest {
public:
explicit ISACTest(int testMode);
~ISACTest();
void Perform();
private:
void Setup();
void Run10ms();
void EncodeDecode(int testNr, ACMTestISACConfig& wbISACConfig,
ACMTestISACConfig& swbISACConfig);
void SwitchingSamplingRate(int testNr, int maxSampRateChange);
std::unique_ptr<AudioCodingModule> _acmA;
std::unique_ptr<AudioCodingModule> _acmB;
std::unique_ptr<Channel> _channel_A2B;
std::unique_ptr<Channel> _channel_B2A;
PCMFile _inFileA;
PCMFile _inFileB;
PCMFile _outFileA;
PCMFile _outFileB;
uint8_t _idISAC16kHz;
uint8_t _idISAC32kHz;
CodecInst _paramISAC16kHz;
CodecInst _paramISAC32kHz;
std::string file_name_swb_;
ACMTestTimer _myTimer;
int _testMode;
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_TEST_ISACTEST_H_