blob: 55227022148186397c8550f8cb98be8c802d92bc [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 ACM_ISAC_TEST_H
#define ACM_ISAC_TEST_H
#include <string.h>
#include "ACMTest.h"
#include "Channel.h"
#include "PCMFile.h"
#include "audio_coding_module.h"
#include "utility.h"
#include "common_types.h"
#define MAX_FILE_NAME_LENGTH_BYTE 500
#define NO_OF_CLIENTS 15
namespace webrtc {
struct ACMTestISACConfig {
int32_t currentRateBitPerSec;
int16_t currentFrameSizeMsec;
uint32_t maxRateBitPerSec;
int16_t maxPayloadSizeByte;
int16_t encodingMode;
uint32_t initRateBitPerSec;
int16_t initFrameSizeInMsec;
bool enforceFrameSize;
};
class ISACTest : public ACMTest {
public:
ISACTest(int testMode);
~ISACTest();
void Perform();
private:
int16_t Setup();
int16_t SetupConference();
int16_t RunConference();
void Run10ms();
void EncodeDecode(int testNr, ACMTestISACConfig& wbISACConfig,
ACMTestISACConfig& swbISACConfig);
void TestBWE(int testNr);
void SwitchingSamplingRate(int testNr, int maxSampRateChange);
AudioCodingModule* _acmA;
AudioCodingModule* _acmB;
Channel* _channel_A2B;
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;
AudioCodingModule* _defaultACM32;
AudioCodingModule* _defaultACM16;
AudioCodingModule* _confACM[NO_OF_CLIENTS];
AudioCodingModule* _clientACM[NO_OF_CLIENTS];
Channel* _conf2Client[NO_OF_CLIENTS];
Channel* _client2Conf[NO_OF_CLIENTS];
PCMFile _clientOutFile[NO_OF_CLIENTS];
};
} // namespace webrtc
#endif