Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(650)

Side by Side Diff: webrtc/modules/audio_processing/transient/file_utils.h

Issue 1413333002: system_wrappers: rename interface -> include (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased again! Created 5 years, 1 month 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) 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_MODULES_AUDIO_PROCESSING_TRANSIENT_FILE_UTILS_H_ 11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TRANSIENT_FILE_UTILS_H_
12 #define WEBRTC_MODULES_AUDIO_PROCESSING_TRANSIENT_FILE_UTILS_H_ 12 #define WEBRTC_MODULES_AUDIO_PROCESSING_TRANSIENT_FILE_UTILS_H_
13 13
14 #include <string.h> 14 #include <string.h>
15 15
16 #include "webrtc/system_wrappers/interface/file_wrapper.h" 16 #include "webrtc/system_wrappers/include/file_wrapper.h"
17 #include "webrtc/typedefs.h" 17 #include "webrtc/typedefs.h"
18 18
19 namespace webrtc { 19 namespace webrtc {
20 20
21 // This is a copy of the cast included in the Chromium codebase here: 21 // This is a copy of the cast included in the Chromium codebase here:
22 // http://cs.chromium.org/src/third_party/cld/base/casts.h 22 // http://cs.chromium.org/src/third_party/cld/base/casts.h
23 template <class Dest, class Source> 23 template <class Dest, class Source>
24 inline Dest bit_cast(const Source& source) { 24 inline Dest bit_cast(const Source& source) {
25 // A compile error here means your Dest and Source have different sizes. 25 // A compile error here means your Dest and Source have different sizes.
26 static_assert(sizeof(Dest) == sizeof(Source), 26 static_assert(sizeof(Dest) == sizeof(Source),
(...skipping 82 matching lines...) Expand 10 before | Expand all | Expand 10 after
109 // |file|. It flushes |file|, so after this call there are no writings pending. 109 // |file|. It flushes |file|, so after this call there are no writings pending.
110 // |file| must be previously opened. 110 // |file| must be previously opened.
111 // Returns the number of doubles written or -1 on error. 111 // Returns the number of doubles written or -1 on error.
112 size_t WriteDoubleBufferToFile(FileWrapper* file, 112 size_t WriteDoubleBufferToFile(FileWrapper* file,
113 size_t length, 113 size_t length,
114 const double* buffer); 114 const double* buffer);
115 115
116 } // namespace webrtc 116 } // namespace webrtc
117 117
118 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_TRANSIENT_FILE_UTILS_H_ 118 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_TRANSIENT_FILE_UTILS_H_
OLDNEW
« no previous file with comments | « webrtc/modules/audio_processing/transient/click_annotate.cc ('k') | webrtc/modules/audio_processing/transient/file_utils.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698