Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(607)

Side by Side Diff: webrtc/modules/audio_processing/utility/lapped_transform.h

Issue 1846903004: Moved ring-buffer related files from common_audio to audio_processing (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase Created 4 years, 8 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
11 #ifndef WEBRTC_COMMON_AUDIO_LAPPED_TRANSFORM_H_ 11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_LAPPED_TRANSFORM_H_
12 #define WEBRTC_COMMON_AUDIO_LAPPED_TRANSFORM_H_ 12 #define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_LAPPED_TRANSFORM_H_
13 13
14 #include <complex> 14 #include <complex>
15 #include <memory> 15 #include <memory>
16 16
17 #include "webrtc/common_audio/blocker.h"
18 #include "webrtc/common_audio/real_fourier.h" 17 #include "webrtc/common_audio/real_fourier.h"
18 #include "webrtc/modules/audio_processing/utility/blocker.h"
19 #include "webrtc/system_wrappers/include/aligned_array.h" 19 #include "webrtc/system_wrappers/include/aligned_array.h"
20 20
21 namespace webrtc { 21 namespace webrtc {
22 22
23 // Helper class for audio processing modules which operate on frequency domain 23 // Helper class for audio processing modules which operate on frequency domain
24 // input derived from the windowed time domain audio stream. 24 // input derived from the windowed time domain audio stream.
25 // 25 //
26 // The input audio chunk is sliced into possibly overlapping blocks, multiplied 26 // The input audio chunk is sliced into possibly overlapping blocks, multiplied
27 // by a window and transformed with an FFT implementation. The transformed data 27 // by a window and transformed with an FFT implementation. The transformed data
28 // is supplied to the given callback for processing. The processed output is 28 // is supplied to the given callback for processing. The processed output is
(...skipping 85 matching lines...) Expand 10 before | Expand all | Expand 10 after
114 114
115 std::unique_ptr<RealFourier> fft_; 115 std::unique_ptr<RealFourier> fft_;
116 const size_t cplx_length_; 116 const size_t cplx_length_;
117 AlignedArray<float> real_buf_; 117 AlignedArray<float> real_buf_;
118 AlignedArray<std::complex<float> > cplx_pre_; 118 AlignedArray<std::complex<float> > cplx_pre_;
119 AlignedArray<std::complex<float> > cplx_post_; 119 AlignedArray<std::complex<float> > cplx_post_;
120 }; 120 };
121 121
122 } // namespace webrtc 122 } // namespace webrtc
123 123
124 #endif // WEBRTC_COMMON_AUDIO_LAPPED_TRANSFORM_H_ 124 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_LAPPED_TRANSFORM_H_
125
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698