Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(485)

Side by Side Diff: webrtc/modules/audio_coding/neteq/merge.cc

Issue 2810483002: Add SafeMin() and SafeMax(), which accept args of different types (Closed)
Patch Set: trigger error earlier Created 3 years, 7 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) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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_coding/neteq/merge.h" 11 #include "webrtc/modules/audio_coding/neteq/merge.h"
12 12
13 #include <assert.h> 13 #include <assert.h>
14 #include <string.h> // memmove, memcpy, memset, size_t 14 #include <string.h> // memmove, memcpy, memset, size_t
15 15
16 #include <algorithm> // min, max 16 #include <algorithm> // min, max
17 #include <memory> 17 #include <memory>
18 18
19 #include "webrtc/base/safe_conversions.h"
20 #include "webrtc/base/safe_minmax.h"
19 #include "webrtc/common_audio/signal_processing/include/signal_processing_librar y.h" 21 #include "webrtc/common_audio/signal_processing/include/signal_processing_librar y.h"
20 #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h" 22 #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
21 #include "webrtc/modules/audio_coding/neteq/cross_correlation.h" 23 #include "webrtc/modules/audio_coding/neteq/cross_correlation.h"
22 #include "webrtc/modules/audio_coding/neteq/dsp_helper.h" 24 #include "webrtc/modules/audio_coding/neteq/dsp_helper.h"
23 #include "webrtc/modules/audio_coding/neteq/expand.h" 25 #include "webrtc/modules/audio_coding/neteq/expand.h"
24 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h" 26 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
25 27
26 namespace webrtc { 28 namespace webrtc {
27 29
28 Merge::Merge(int fs_hz, 30 Merge::Merge(int fs_hz,
(...skipping 173 matching lines...) Expand 10 before | Expand all | Expand 10 after
202 // Trim the length to exactly |required_length|. 204 // Trim the length to exactly |required_length|.
203 expanded_.PopBack(expanded_.Size() - required_length); 205 expanded_.PopBack(expanded_.Size() - required_length);
204 } 206 }
205 assert(expanded_.Size() >= required_length); 207 assert(expanded_.Size() >= required_length);
206 return required_length; 208 return required_length;
207 } 209 }
208 210
209 int16_t Merge::SignalScaling(const int16_t* input, size_t input_length, 211 int16_t Merge::SignalScaling(const int16_t* input, size_t input_length,
210 const int16_t* expanded_signal) const { 212 const int16_t* expanded_signal) const {
211 // Adjust muting factor if new vector is more or less of the BGN energy. 213 // Adjust muting factor if new vector is more or less of the BGN energy.
212 const size_t mod_input_length = 214 const auto mod_input_length = rtc::SafeMin<size_t>(
213 std::min(static_cast<size_t>(64 * fs_mult_), input_length); 215 64 * rtc::dchecked_cast<size_t>(fs_mult_), input_length);
214 const int16_t expanded_max = 216 const int16_t expanded_max =
215 WebRtcSpl_MaxAbsValueW16(expanded_signal, mod_input_length); 217 WebRtcSpl_MaxAbsValueW16(expanded_signal, mod_input_length);
216 int32_t factor = (expanded_max * expanded_max) / 218 int32_t factor = (expanded_max * expanded_max) /
217 (std::numeric_limits<int32_t>::max() / 219 (std::numeric_limits<int32_t>::max() /
218 static_cast<int32_t>(mod_input_length)); 220 static_cast<int32_t>(mod_input_length));
219 const int expanded_shift = factor == 0 ? 0 : 31 - WebRtcSpl_NormW32(factor); 221 const int expanded_shift = factor == 0 ? 0 : 31 - WebRtcSpl_NormW32(factor);
220 int32_t energy_expanded = WebRtcSpl_DotProductWithScale(expanded_signal, 222 int32_t energy_expanded = WebRtcSpl_DotProductWithScale(expanded_signal,
221 expanded_signal, 223 expanded_signal,
222 mod_input_length, 224 mod_input_length,
223 expanded_shift); 225 expanded_shift);
(...skipping 145 matching lines...) Expand 10 before | Expand all | Expand 10 after
369 } 371 }
370 return best_correlation_index; 372 return best_correlation_index;
371 } 373 }
372 374
373 size_t Merge::RequiredFutureSamples() { 375 size_t Merge::RequiredFutureSamples() {
374 return fs_hz_ / 100 * num_channels_; // 10 ms. 376 return fs_hz_ / 100 * num_channels_; // 10 ms.
375 } 377 }
376 378
377 379
378 } // namespace webrtc 380 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/common_video/h264/sps_vui_rewriter.cc ('k') | webrtc/modules/audio_processing/audio_processing_unittest.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698