resampler.h - mozsearch

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/*

Copyright 2018 Google Inc. All Rights Reserved.

Licensed under the Apache License, Version 2.0 (the "License");

you may not use this file except in compliance with the License.

You may obtain a copy of the License at

    http://www.apache.org/licenses/LICENSE-2.0

Unless required by applicable law or agreed to in writing, software

distributed under the License is distributed on an "AS-IS" BASIS,

WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.

See the License for the specific language governing permissions and

limitations under the License.

*/

#ifndef RESONANCE_AUDIO_DSP_RESAMPLER_H_

#define RESONANCE_AUDIO_DSP_RESAMPLER_H_

#include "base/audio_buffer.h"

namespace vraudio {

// Class that provides rational resampling of audio data.

class Resampler {

 public:

  Resampler();

  // Resamples an |AudioBuffer| of input data sampled at |source_frequency| to

  // |destination_frequency|.

//

  // @param input Input data to be resampled.

  // @param output Resampled output data.

  void Process(const AudioBuffer& input, AudioBuffer* output);

  // Returns the maximum length which the output buffer will be, given the

  // current source and destination frequencies and input length. The actual

  // output length will either be this or one less.

//

  // @param input_length Length of the input.

  // @return Maximum length of the output.

  size_t GetMaxOutputLength(size_t input_length) const;

  // Returns the next length which the output buffer will be, given the

  // current source and destination frequencies and input length.

//

  // @param input_length Length of the input.

  // @return Next length of the output.

  size_t GetNextOutputLength(size_t input_length) const;

  // Sets the source and destination sampling rate as well as the number of

  // channels. Note this method only resets the filter state number of channel

  // changes.

//

  // @param source_frequency Sampling rate of input data.

  // @param destination_frequency Desired output sampling rate.

  // @param num_channels Number of channels to process.

  void SetRateAndNumChannels(int source_frequency, int destination_frequency,

                             size_t num_channels);

  // Returns whether the sampling rates provided are supported by the resampler.

//

  // @param source Source sampling rate.

  // @param destination Destination sampling rate.

  // @return True if the sampling rate pair are supported.

  static bool AreSampleRatesSupported(int source, int destination);

  // Resets the inner state of the |Resampler| allowing its use repeatedly on

  // different data streams.

  void ResetState();

 private:

  friend class PolyphaseFilterTest;

  // Initializes the |state_| buffer. Called when sampling rate is changed or

  // the state is reset.

//

  // @param size_t old_state_num_frames Number of frames in the |state_| buffer

  //     previous to the most recent call to |GenerateInterpolatingFilter|.

  void InitializeStateBuffer(size_t old_state_num_frames);

  // Generates a windowed sinc to act as the interpolating/anti-aliasing filter.

//

  // @param sample_rate The system sampling rate.

  void GenerateInterpolatingFilter(int sample_rate);

  // Arranges the anti aliasing filter coefficients in polyphase filter format.

//

  // @param filter_length Number of frames in |filter| containing filter

  //     coefficients.

  // @param filter Vector of filter coefficients.

  void ArrangeFilterAsPolyphase(size_t filter_length,

                                const AudioBuffer::Channel& filter);

  // Generates Hann windowed sinc function anti aliasing filters.

//

  // @param cutoff_frequency Transition band (-3dB) frequency of the filter.

  // @param sample_rate The system sampling rate.

  // @param filter_length Number of frames in |buffer| containing filter

  //     coefficients.

  // @param buffer |AudioBuffer::Channel| to contain the filter coefficients.

  void GenerateSincFilter(float cutoff_frequency, float sample_rate,

                          size_t filter_length, AudioBuffer::Channel* buffer);

  // Rate of the interpolator section of the rational sampling rate converter.

  size_t up_rate_;

  // Rate of the decimator section of the rational sampling rate convereter.

  size_t down_rate_;

  // Time variable for the polyphase filter.

  size_t time_modulo_up_rate_;

  // Marks the last processed sample of the input.

  size_t last_processed_sample_;

  // Number of channels in the |AudioBuffer|s processed.

  size_t num_channels_;

  // Number of filter coefficients in each phase of the polyphase filter.

  size_t coeffs_per_phase_;

  // Filter coefficients stored in polyphase form.

  AudioBuffer transposed_filter_coeffs_;

  // Filter coefficients in planar form, used for calculating the transposed

  // filter.

  AudioBuffer temporary_filter_coeffs_;

  // Buffer holding the samples of input required between input buffers.

  AudioBuffer state_;

};

}  // namespace vraudio

#endif  // RESONANCE_AUDIO_DSP_RESAMPLER_H_