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/*
* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_COMMON_AUDIO_LAPPED_TRANSFORM_H_
#define WEBRTC_COMMON_AUDIO_LAPPED_TRANSFORM_H_
#include <complex>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_audio/blocker.h"
#include "webrtc/common_audio/real_fourier.h"
#include "webrtc/system_wrappers/interface/aligned_array.h"
namespace webrtc {
// Helper class for audio processing modules which operate on frequency domain
// input derived from the windowed time domain audio stream.
//
// The input audio chunk is sliced into possibly overlapping blocks, multiplied
// by a window and transformed with an FFT implementation. The transformed data
// is supplied to the given callback for processing. The processed output is
// then inverse transformed into the time domain and spliced back into a chunk
// which constitutes the final output of this processing module.
class LappedTransform {
public:
class Callback {
public:
virtual ~Callback() {}
virtual void ProcessAudioBlock(const std::complex<float>* const* in_block,
int num_in_channels, size_t frames,
int num_out_channels,
std::complex<float>* const* out_block) = 0;
};
// Construct a transform instance. |chunk_length| is the number of samples in
// each channel. |window| defines the window, owned by the caller (a copy is
// made internally); |window| should have length equal to |block_length|.
// |block_length| defines the length of a block, in samples.
// |shift_amount| is in samples. |callback| is the caller-owned audio
// processing function called for each block of the input chunk.
LappedTransform(int num_in_channels,
int num_out_channels,
size_t chunk_length,
const float* window,
size_t block_length,
size_t shift_amount,
Callback* callback);
~LappedTransform() {}
// Main audio processing helper method. Internally slices |in_chunk| into
// blocks, transforms them to frequency domain, calls the callback for each
// block and returns a de-blocked time domain chunk of audio through
// |out_chunk|. Both buffers are caller-owned.
void ProcessChunk(const float* const* in_chunk, float* const* out_chunk);
// Get the chunk length.
//
// The chunk length is the number of samples per channel that must be passed
// to ProcessChunk via the parameter in_chunk.
//
// Returns the same chunk_length passed to the LappedTransform constructor.
size_t chunk_length() const { return chunk_length_; }
// Get the number of input channels.
//
// This is the number of arrays that must be passed to ProcessChunk via
// in_chunk.
//
// Returns the same num_in_channels passed to the LappedTransform constructor.
int num_in_channels() const { return num_in_channels_; }
// Get the number of output channels.
//
// This is the number of arrays that must be passed to ProcessChunk via
// out_chunk.
//
// Returns the same num_out_channels passed to the LappedTransform
// constructor.
int num_out_channels() const { return num_out_channels_; }
private:
// Internal middleware callback, given to the blocker. Transforms each block
// and hands it over to the processing method given at construction time.
class BlockThunk : public BlockerCallback {
public:
explicit BlockThunk(LappedTransform* parent) : parent_(parent) {}
virtual void ProcessBlock(const float* const* input, size_t num_frames,
int num_input_channels, int num_output_channels,
float* const* output);
private:
LappedTransform* const parent_;
} blocker_callback_;
const int num_in_channels_;
const int num_out_channels_;
const size_t block_length_;
const size_t chunk_length_;
Callback* const block_processor_;
Blocker blocker_;
rtc::scoped_ptr<RealFourier> fft_;
const size_t cplx_length_;
AlignedArray<float> real_buf_;
AlignedArray<std::complex<float> > cplx_pre_;
AlignedArray<std::complex<float> > cplx_post_;
};
} // namespace webrtc
#endif // WEBRTC_COMMON_AUDIO_LAPPED_TRANSFORM_H_
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