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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.
90 lines
2.9 KiB
C++
90 lines
2.9 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_PROCESSING_AGC_LOUDNESS_HISTOGRAM_H_
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#define MODULES_AUDIO_PROCESSING_AGC_LOUDNESS_HISTOGRAM_H_
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#include <stdint.h>
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#include <memory>
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namespace webrtc {
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// This class implements the histogram of loudness with circular buffers so that
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// the histogram tracks the last T seconds of the loudness.
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class LoudnessHistogram {
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public:
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// Create a non-sliding LoudnessHistogram.
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static LoudnessHistogram* Create();
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// Create a sliding LoudnessHistogram, i.e. the histogram represents the last
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// |window_size| samples.
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static LoudnessHistogram* Create(int window_size);
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~LoudnessHistogram();
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// Insert RMS and the corresponding activity probability.
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void Update(double rms, double activity_probability);
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// Reset the histogram, forget the past.
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void Reset();
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// Current loudness, which is actually the mean of histogram in loudness
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// domain.
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double CurrentRms() const;
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// Sum of the histogram content.
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double AudioContent() const;
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// Number of times the histogram has been updated.
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int num_updates() const { return num_updates_; }
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private:
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LoudnessHistogram();
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explicit LoudnessHistogram(int window);
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// Find the histogram bin associated with the given |rms|.
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int GetBinIndex(double rms);
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void RemoveOldestEntryAndUpdate();
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void InsertNewestEntryAndUpdate(int activity_prob_q10, int hist_index);
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void UpdateHist(int activity_prob_q10, int hist_index);
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void RemoveTransient();
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// Number of histogram bins.
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static const int kHistSize = 77;
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// Number of times the histogram is updated
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int num_updates_;
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// Audio content, this should be equal to the sum of the components of
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// |bin_count_q10_|.
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int64_t audio_content_q10_;
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// LoudnessHistogram of input RMS in Q10 with |kHistSize_| bins. In each
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// 'Update(),' we increment the associated histogram-bin with the given
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// probability. The increment is implemented in Q10 to avoid rounding errors.
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int64_t bin_count_q10_[kHistSize];
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// Circular buffer for probabilities
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std::unique_ptr<int[]> activity_probability_;
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// Circular buffer for histogram-indices of probabilities.
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std::unique_ptr<int[]> hist_bin_index_;
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// Current index of circular buffer, where the newest data will be written to,
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// therefore, pointing to the oldest data if buffer is full.
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int buffer_index_;
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// Indicating if buffer is full and we had a wrap around.
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int buffer_is_full_;
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// Size of circular buffer.
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int len_circular_buffer_;
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int len_high_activity_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AGC_LOUDNESS_HISTOGRAM_H_
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