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/*
* Copyright (C) 2013 Google Inc. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are
* met:
*
* * Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* * Redistributions in binary form must reproduce the above
* copyright notice, this list of conditions and the following disclaimer
* in the documentation and/or other materials provided with the
* distribution.
* * Neither the name of Google Inc. nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
* OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
* LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
* DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
* THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
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* OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef UpSampler_h
#define UpSampler_h
#include "platform/audio/AudioArray.h"
#include "platform/audio/DirectConvolver.h"
namespace WebCore {
// UpSampler up-samples the source stream by a factor of 2x.
class PLATFORM_EXPORT UpSampler {
public:
UpSampler(size_t inputBlockSize);
// The destination buffer |destP| is of size sourceFramesToProcess * 2.
void process(const float* sourceP, float* destP, size_t sourceFramesToProcess);
void reset();
// Latency based on the source sample-rate.
size_t latencyFrames() const;
private:
enum { DefaultKernelSize = 128 };
size_t m_inputBlockSize;
// Computes ideal band-limited filter coefficients to sample in between each source sample-frame.
// This filter will be used to compute the odd sample-frames of the output.
void initializeKernel();
AudioFloatArray m_kernel;
// Computes the odd sample-frames of the output.
DirectConvolver m_convolver;
AudioFloatArray m_tempBuffer;
// Delay line for generating the even sample-frames of the output.
// The source samples are delayed exactly to match the linear phase delay of the FIR filter (convolution)
// used to generate the odd sample-frames of the output.
AudioFloatArray m_inputBuffer;
};
} // namespace WebCore
#endif // UpSampler_h