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Side by Side Diff: webrtc/modules/audio_processing/transient/transient_suppressor.cc

Issue 1726043002: Revert of Replace scoped_ptr with unique_ptr in webrtc/common_audio/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 10 months ago
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1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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 #include "webrtc/modules/audio_processing/transient/transient_suppressor.h" 11 #include "webrtc/modules/audio_processing/transient/transient_suppressor.h"
12 12
13 #include <math.h> 13 #include <math.h>
14 #include <string.h> 14 #include <string.h>
15 #include <cmath> 15 #include <cmath>
16 #include <complex> 16 #include <complex>
17 #include <deque> 17 #include <deque>
18 #include <set> 18 #include <set>
19 19
20 #include "webrtc/base/checks.h"
21 #include "webrtc/common_audio/fft4g.h" 20 #include "webrtc/common_audio/fft4g.h"
22 #include "webrtc/common_audio/include/audio_util.h" 21 #include "webrtc/common_audio/include/audio_util.h"
23 #include "webrtc/common_audio/signal_processing/include/signal_processing_librar y.h" 22 #include "webrtc/common_audio/signal_processing/include/signal_processing_librar y.h"
24 #include "webrtc/modules/audio_processing/transient/common.h" 23 #include "webrtc/modules/audio_processing/transient/common.h"
25 #include "webrtc/modules/audio_processing/transient/transient_detector.h" 24 #include "webrtc/modules/audio_processing/transient/transient_detector.h"
26 #include "webrtc/modules/audio_processing/ns/windows_private.h" 25 #include "webrtc/modules/audio_processing/ns/windows_private.h"
27 #include "webrtc/system_wrappers/include/logging.h" 26 #include "webrtc/system_wrappers/include/logging.h"
28 #include "webrtc/typedefs.h" 27 #include "webrtc/typedefs.h"
29 28
30 namespace webrtc { 29 namespace webrtc {
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94 detection_rate_hz != ts::kSampleRate48kHz) { 93 detection_rate_hz != ts::kSampleRate48kHz) {
95 return -1; 94 return -1;
96 } 95 }
97 if (num_channels <= 0) { 96 if (num_channels <= 0) {
98 return -1; 97 return -1;
99 } 98 }
100 99
101 detector_.reset(new TransientDetector(detection_rate_hz)); 100 detector_.reset(new TransientDetector(detection_rate_hz));
102 data_length_ = sample_rate_hz * ts::kChunkSizeMs / 1000; 101 data_length_ = sample_rate_hz * ts::kChunkSizeMs / 1000;
103 if (data_length_ > analysis_length_) { 102 if (data_length_ > analysis_length_) {
104 RTC_NOTREACHED(); 103 assert(false);
105 return -1; 104 return -1;
106 } 105 }
107 buffer_delay_ = analysis_length_ - data_length_; 106 buffer_delay_ = analysis_length_ - data_length_;
108 107
109 complex_analysis_length_ = analysis_length_ / 2 + 1; 108 complex_analysis_length_ = analysis_length_ / 2 + 1;
110 RTC_DCHECK_GE(complex_analysis_length_, kMaxVoiceBin); 109 assert(complex_analysis_length_ >= kMaxVoiceBin);
111 num_channels_ = num_channels; 110 num_channels_ = num_channels;
112 in_buffer_.reset(new float[analysis_length_ * num_channels_]); 111 in_buffer_.reset(new float[analysis_length_ * num_channels_]);
113 memset(in_buffer_.get(), 112 memset(in_buffer_.get(),
114 0, 113 0,
115 analysis_length_ * num_channels_ * sizeof(in_buffer_[0])); 114 analysis_length_ * num_channels_ * sizeof(in_buffer_[0]));
116 detection_length_ = detection_rate_hz * ts::kChunkSizeMs / 1000; 115 detection_length_ = detection_rate_hz * ts::kChunkSizeMs / 1000;
117 detection_buffer_.reset(new float[detection_length_]); 116 detection_buffer_.reset(new float[detection_length_]);
118 memset(detection_buffer_.get(), 117 memset(detection_buffer_.get(),
119 0, 118 0,
120 detection_length_ * sizeof(detection_buffer_[0])); 119 detection_length_ * sizeof(detection_buffer_[0]));
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415 const float magnitude_ratio = new_magnitude / magnitudes_[i]; 414 const float magnitude_ratio = new_magnitude / magnitudes_[i];
416 415
417 fft_buffer_[i * 2] *= magnitude_ratio; 416 fft_buffer_[i * 2] *= magnitude_ratio;
418 fft_buffer_[i * 2 + 1] *= magnitude_ratio; 417 fft_buffer_[i * 2 + 1] *= magnitude_ratio;
419 magnitudes_[i] = new_magnitude; 418 magnitudes_[i] = new_magnitude;
420 } 419 }
421 } 420 }
422 } 421 }
423 422
424 } // namespace webrtc 423 } // namespace webrtc
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