04c50981f8
We are using the Ooura FFT in a few places: - AGC - Transient suppression - Noise suppression The optimized OpenMAX DL FFT is considerably faster, but currently does not compile everywhere, notably on iOS. This change will allow us to use Openmax when possible and otherwise fall back to Ooura. (Unfortunately, noise suppression won't be able to take advantage of it since it's not C++. Upgrade time?) R=aluebs@webrtc.org, mgraczyk@chromium.org Review URL: https://webrtc-codereview.appspot.com/45789004 Cr-Commit-Position: refs/heads/master@{#8798} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8798 4adac7df-926f-26a2-2b94-8c16560cd09d
96 lines
3.6 KiB
C++
96 lines
3.6 KiB
C++
/*
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* Copyright (c) 2014 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 WEBRTC_COMMON_AUDIO_LAPPED_TRANSFORM_H_
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#define WEBRTC_COMMON_AUDIO_LAPPED_TRANSFORM_H_
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#include <complex>
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#include "webrtc/base/checks.h"
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/common_audio/blocker.h"
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#include "webrtc/common_audio/real_fourier.h"
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#include "webrtc/system_wrappers/interface/aligned_array.h"
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namespace webrtc {
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// Helper class for audio processing modules which operate on frequency domain
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// input derived from the windowed time domain audio stream.
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//
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// The input audio chunk is sliced into possibly overlapping blocks, multiplied
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// by a window and transformed with an FFT implementation. The transformed data
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// is supplied to the given callback for processing. The processed output is
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// then inverse transformed into the time domain and spliced back into a chunk
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// which constitutes the final output of this processing module.
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class LappedTransform {
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public:
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class Callback {
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public:
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virtual ~Callback() {}
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virtual void ProcessAudioBlock(const std::complex<float>* const* in_block,
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int in_channels, int frames,
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int out_channels,
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std::complex<float>* const* out_block) = 0;
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};
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// Construct a transform instance. |chunk_length| is the number of samples in
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// each channel. |window| defines the window, owned by the caller (a copy is
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// made internally); |window| should have length equal to |block_length|.
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// |block_length| defines the length of a block, in samples.
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// |shift_amount| is in samples. |callback| is the caller-owned audio
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// processing function called for each block of the input chunk.
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LappedTransform(int in_channels, int out_channels, int chunk_length,
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const float* window, int block_length, int shift_amount,
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Callback* callback);
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~LappedTransform() {}
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// Main audio processing helper method. Internally slices |in_chunk| into
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// blocks, transforms them to frequency domain, calls the callback for each
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// block and returns a de-blocked time domain chunk of audio through
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// |out_chunk|. Both buffers are caller-owned.
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void ProcessChunk(const float* const* in_chunk, float* const* out_chunk);
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private:
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// Internal middleware callback, given to the blocker. Transforms each block
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// and hands it over to the processing method given at construction time.
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class BlockThunk : public BlockerCallback {
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public:
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explicit BlockThunk(LappedTransform* parent) : parent_(parent) {}
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virtual void ProcessBlock(const float* const* input, int num_frames,
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int num_input_channels, int num_output_channels,
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float* const* output);
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private:
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LappedTransform* const parent_;
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} blocker_callback_;
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const int in_channels_;
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const int out_channels_;
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const int block_length_;
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const int chunk_length_;
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Callback* const block_processor_;
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Blocker blocker_;
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rtc::scoped_ptr<RealFourier> fft_;
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const int cplx_length_;
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AlignedArray<float> real_buf_;
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AlignedArray<std::complex<float> > cplx_pre_;
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AlignedArray<std::complex<float> > cplx_post_;
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};
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} // namespace webrtc
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#endif // WEBRTC_COMMON_AUDIO_LAPPED_TRANSFORM_H_
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