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Diffstat (limited to 'webrtc/modules/audio_coding/test/iSACTest.h')
-rw-r--r-- | webrtc/modules/audio_coding/test/iSACTest.h | 79 |
1 files changed, 79 insertions, 0 deletions
diff --git a/webrtc/modules/audio_coding/test/iSACTest.h b/webrtc/modules/audio_coding/test/iSACTest.h new file mode 100644 index 0000000000..c5bb515437 --- /dev/null +++ b/webrtc/modules/audio_coding/test/iSACTest.h @@ -0,0 +1,79 @@ +/* + * 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_TEST_ISACTEST_H_ +#define WEBRTC_MODULES_AUDIO_CODING_TEST_ISACTEST_H_ + +#include <string.h> + +#include "webrtc/base/scoped_ptr.h" +#include "webrtc/common_types.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/test/ACMTest.h" +#include "webrtc/modules/audio_coding/test/Channel.h" +#include "webrtc/modules/audio_coding/test/PCMFile.h" +#include "webrtc/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); + + rtc::scoped_ptr<AudioCodingModule> _acmA; + rtc::scoped_ptr<AudioCodingModule> _acmB; + + rtc::scoped_ptr<Channel> _channel_A2B; + rtc::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_TEST_ISACTEST_H_ |