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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, |
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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); |
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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 |
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