92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
81 lines
1.9 KiB
C++
81 lines
1.9 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_CODING_TEST_ISACTEST_H_
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#define MODULES_AUDIO_CODING_TEST_ISACTEST_H_
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#include <string.h>
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#include <memory>
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#include "common_types.h"
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#include "modules/audio_coding/include/audio_coding_module.h"
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#include "modules/audio_coding/test/ACMTest.h"
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#include "modules/audio_coding/test/Channel.h"
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#include "modules/audio_coding/test/PCMFile.h"
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#include "modules/audio_coding/test/utility.h"
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#define MAX_FILE_NAME_LENGTH_BYTE 500
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#define NO_OF_CLIENTS 15
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namespace webrtc {
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struct ACMTestISACConfig {
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int32_t currentRateBitPerSec;
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int16_t currentFrameSizeMsec;
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int16_t encodingMode;
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uint32_t initRateBitPerSec;
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int16_t initFrameSizeInMsec;
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bool enforceFrameSize;
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};
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class ISACTest : public ACMTest {
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public:
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explicit ISACTest(int testMode);
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~ISACTest();
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void Perform();
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private:
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void Setup();
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void Run10ms();
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void EncodeDecode(int testNr, ACMTestISACConfig& wbISACConfig,
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ACMTestISACConfig& swbISACConfig);
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void SwitchingSamplingRate(int testNr, int maxSampRateChange);
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std::unique_ptr<AudioCodingModule> _acmA;
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std::unique_ptr<AudioCodingModule> _acmB;
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std::unique_ptr<Channel> _channel_A2B;
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std::unique_ptr<Channel> _channel_B2A;
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PCMFile _inFileA;
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PCMFile _inFileB;
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PCMFile _outFileA;
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PCMFile _outFileB;
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uint8_t _idISAC16kHz;
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uint8_t _idISAC32kHz;
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CodecInst _paramISAC16kHz;
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CodecInst _paramISAC32kHz;
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std::string file_name_swb_;
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ACMTestTimer _myTimer;
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int _testMode;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_TEST_ISACTEST_H_
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