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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.
56 lines
1.8 KiB
C++
56 lines
1.8 KiB
C++
/*
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* Copyright (c) 2013 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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// Modified from the Chromium original here:
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// src/media/base/sinc_resampler_unittest.cc
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#ifndef COMMON_AUDIO_RESAMPLER_SINUSOIDAL_LINEAR_CHIRP_SOURCE_H_
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#define COMMON_AUDIO_RESAMPLER_SINUSOIDAL_LINEAR_CHIRP_SOURCE_H_
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#include "common_audio/resampler/sinc_resampler.h"
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#include "rtc_base/constructormagic.h"
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namespace webrtc {
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// Fake audio source for testing the resampler. Generates a sinusoidal linear
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// chirp (http://en.wikipedia.org/wiki/Chirp) which can be tuned to stress the
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// resampler for the specific sample rate conversion being used.
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class SinusoidalLinearChirpSource : public SincResamplerCallback {
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public:
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// |delay_samples| can be used to insert a fractional sample delay into the
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// source. It will produce zeros until non-negative time is reached.
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SinusoidalLinearChirpSource(int sample_rate,
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size_t samples,
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double max_frequency,
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double delay_samples);
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~SinusoidalLinearChirpSource() override {}
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void Run(size_t frames, float* destination) override;
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double Frequency(size_t position);
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private:
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enum { kMinFrequency = 5 };
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int sample_rate_;
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size_t total_samples_;
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double max_frequency_;
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double k_;
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size_t current_index_;
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double delay_samples_;
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RTC_DISALLOW_COPY_AND_ASSIGN(SinusoidalLinearChirpSource);
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};
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} // namespace webrtc
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#endif // COMMON_AUDIO_RESAMPLER_SINUSOIDAL_LINEAR_CHIRP_SOURCE_H_
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