mirror of
https://github.com/oxen-io/session-android.git
synced 2024-11-28 20:45:17 +00:00
d83a3d71bc
Merge in RedPhone // FREEBIE
82 lines
2.1 KiB
C++
82 lines
2.1 KiB
C++
/*
|
|
* 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 WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ISACTEST_H_
|
|
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ISACTEST_H_
|
|
|
|
#include <string.h>
|
|
|
|
#include "webrtc/common_types.h"
|
|
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
|
|
#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
|
|
#include "webrtc/modules/audio_coding/main/test/Channel.h"
|
|
#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
|
|
#include "webrtc/modules/audio_coding/main/test/utility.h"
|
|
#include "webrtc/system_wrappers/interface/scoped_ptr.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:
|
|
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);
|
|
|
|
scoped_ptr<AudioCodingModule> _acmA;
|
|
scoped_ptr<AudioCodingModule> _acmB;
|
|
|
|
scoped_ptr<Channel> _channel_A2B;
|
|
scoped_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 // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ISACTEST_H_
|