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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 #include "webrtc/rtc_base/safe_conversions.h" | |
26 #include "webrtc/rtc_base/safe_minmax.h" | |
27 | 27 |
28 namespace webrtc { | 28 namespace webrtc { |
29 | 29 |
30 Merge::Merge(int fs_hz, | 30 Merge::Merge(int fs_hz, |
31 size_t num_channels, | 31 size_t num_channels, |
32 Expand* expand, | 32 Expand* expand, |
33 SyncBuffer* sync_buffer) | 33 SyncBuffer* sync_buffer) |
34 : fs_hz_(fs_hz), | 34 : fs_hz_(fs_hz), |
35 num_channels_(num_channels), | 35 num_channels_(num_channels), |
36 fs_mult_(fs_hz_ / 8000), | 36 fs_mult_(fs_hz_ / 8000), |
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372 } | 372 } |
373 return best_correlation_index; | 373 return best_correlation_index; |
374 } | 374 } |
375 | 375 |
376 size_t Merge::RequiredFutureSamples() { | 376 size_t Merge::RequiredFutureSamples() { |
377 return fs_hz_ / 100 * num_channels_; // 10 ms. | 377 return fs_hz_ / 100 * num_channels_; // 10 ms. |
378 } | 378 } |
379 | 379 |
380 | 380 |
381 } // namespace webrtc | 381 } // namespace webrtc |
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