fbeb76ab51
This reverts commit b46c4bf27ba5c417fcba7f200d80fa4634e7e1a1. Reason for revert: breaks a downstream project Original change's description: > [ACM] iSAC audio codec removed > > Note: this CL has to leave behind one part of iSAC, which is its VAD > currently used by AGC1 in APM. The target visibility has been > restricted and the VAD will be removed together with AGC1 when the > time comes. > > Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319 > > Bug: webrtc:14450 > Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421 > Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38652} Bug: webrtc:14450 Change-Id: Ice138004e84e8c5f896684e8d01133d4b2a77bb7 No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283800 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Owners-Override: Mirko Bonadei <mbonadei@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#38655}
69 lines
1.6 KiB
C++
69 lines
1.6 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 "modules/audio_coding/include/audio_coding_module.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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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 {
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public:
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ISACTest();
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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,
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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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std::string file_name_swb_;
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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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