| /* |
| * 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 |