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https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
41 lines
1.1 KiB
C++
41 lines
1.1 KiB
C++
/*
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* Copyright (c) 2016 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_PROCESSING_AGC2_DOWN_SAMPLER_H_
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#define MODULES_AUDIO_PROCESSING_AGC2_DOWN_SAMPLER_H_
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#include "api/array_view.h"
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#include "modules/audio_processing/agc2/biquad_filter.h"
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#include "rtc_base/constructormagic.h"
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namespace webrtc {
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class ApmDataDumper;
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class DownSampler {
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public:
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explicit DownSampler(ApmDataDumper* data_dumper);
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void Initialize(int sample_rate_hz);
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void DownSample(rtc::ArrayView<const float> in, rtc::ArrayView<float> out);
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private:
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ApmDataDumper* data_dumper_;
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int sample_rate_hz_;
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int down_sampling_factor_;
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BiQuadFilter low_pass_filter_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(DownSampler);
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AGC2_DOWN_SAMPLER_H_
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