Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(3503)

Side by Side Diff: webrtc/common_audio/audio_converter.h

Issue 1335923002: Add RTC_ prefix to (D)CHECKs and related macros. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase. Created 5 years, 3 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/base/virtualsocketserver.cc ('k') | webrtc/common_audio/audio_converter.cc » ('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) 2014 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2014 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
(...skipping 31 matching lines...) Expand 10 before | Expand all | Expand 10 after
42 int src_channels() const { return src_channels_; } 42 int src_channels() const { return src_channels_; }
43 size_t src_frames() const { return src_frames_; } 43 size_t src_frames() const { return src_frames_; }
44 int dst_channels() const { return dst_channels_; } 44 int dst_channels() const { return dst_channels_; }
45 size_t dst_frames() const { return dst_frames_; } 45 size_t dst_frames() const { return dst_frames_; }
46 46
47 protected: 47 protected:
48 AudioConverter(); 48 AudioConverter();
49 AudioConverter(int src_channels, size_t src_frames, int dst_channels, 49 AudioConverter(int src_channels, size_t src_frames, int dst_channels,
50 size_t dst_frames); 50 size_t dst_frames);
51 51
52 // Helper to CHECK that inputs are correctly sized. 52 // Helper to RTC_CHECK that inputs are correctly sized.
53 void CheckSizes(size_t src_size, size_t dst_capacity) const; 53 void CheckSizes(size_t src_size, size_t dst_capacity) const;
54 54
55 private: 55 private:
56 const int src_channels_; 56 const int src_channels_;
57 const size_t src_frames_; 57 const size_t src_frames_;
58 const int dst_channels_; 58 const int dst_channels_;
59 const size_t dst_frames_; 59 const size_t dst_frames_;
60 60
61 RTC_DISALLOW_COPY_AND_ASSIGN(AudioConverter); 61 RTC_DISALLOW_COPY_AND_ASSIGN(AudioConverter);
62 }; 62 };
63 63
64 } // namespace webrtc 64 } // namespace webrtc
65 65
66 #endif // WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_ 66 #endif // WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_
OLDNEW
« no previous file with comments | « webrtc/base/virtualsocketserver.cc ('k') | webrtc/common_audio/audio_converter.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698