Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(111)

Side by Side Diff: webrtc/common_audio/include/audio_util.h

Issue 1712513002: Replace scoped_ptr with unique_ptr in webrtc/common_audio/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 4 years, 10 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
« no previous file with comments | « webrtc/common_audio/fir_filter_unittest.cc ('k') | webrtc/common_audio/lapped_transform.h » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
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 #ifndef WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_ 11 #ifndef WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
12 #define WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_ 12 #define WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
13 13
14 #include <limits> 14 #include <limits>
15 #include <cstring> 15 #include <cstring>
16 16
17 #include "webrtc/base/checks.h" 17 #include "webrtc/base/checks.h"
18 #include "webrtc/base/scoped_ptr.h"
19 #include "webrtc/typedefs.h" 18 #include "webrtc/typedefs.h"
20 19
21 namespace webrtc { 20 namespace webrtc {
22 21
23 typedef std::numeric_limits<int16_t> limits_int16; 22 typedef std::numeric_limits<int16_t> limits_int16;
24 23
25 // The conversion functions use the following naming convention: 24 // The conversion functions use the following naming convention:
26 // S16: int16_t [-32768, 32767] 25 // S16: int16_t [-32768, 32767]
27 // Float: float [-1.0, 1.0] 26 // Float: float [-1.0, 1.0]
28 // FloatS16: float [-32768.0, 32767.0] 27 // FloatS16: float [-32768.0, 32767.0]
(...skipping 150 matching lines...) Expand 10 before | Expand all | Expand 10 after
179 178
180 template <> 179 template <>
181 void DownmixInterleavedToMono<int16_t>(const int16_t* interleaved, 180 void DownmixInterleavedToMono<int16_t>(const int16_t* interleaved,
182 size_t num_frames, 181 size_t num_frames,
183 int num_channels, 182 int num_channels,
184 int16_t* deinterleaved); 183 int16_t* deinterleaved);
185 184
186 } // namespace webrtc 185 } // namespace webrtc
187 186
188 #endif // WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_ 187 #endif // WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
OLDNEW
« no previous file with comments | « webrtc/common_audio/fir_filter_unittest.cc ('k') | webrtc/common_audio/lapped_transform.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698