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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_