Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(466)

Side by Side Diff: webrtc/modules/audio_coding/codecs/opus/opus_inst.h

Issue 1446093003: Adding stddef.h to opus_inst.h. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: 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
« no previous file with comments | « no previous file | no next file » | 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
11 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_ 11 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
12 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_ 12 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
13 13
14 #include <stddef.h>
15
14 #include "opus.h" 16 #include "opus.h"
15 17
16 struct WebRtcOpusEncInst { 18 struct WebRtcOpusEncInst {
17 OpusEncoder* encoder; 19 OpusEncoder* encoder;
18 int channels; 20 int channels;
19 int in_dtx_mode; 21 int in_dtx_mode;
20 // When Opus is in DTX mode, we use |zero_counts| to count consecutive zeros 22 // When Opus is in DTX mode, we use |zero_counts| to count consecutive zeros
21 // to break long zero segment so as to prevent DTX from going wrong. We use 23 // to break long zero segment so as to prevent DTX from going wrong. We use
22 // one counter for each channel. After each encoding, |zero_counts| contain 24 // one counter for each channel. After each encoding, |zero_counts| contain
23 // the remaining zeros from the last frame. 25 // the remaining zeros from the last frame.
24 // TODO(minyue): remove this when Opus gets an internal fix to DTX. 26 // TODO(minyue): remove this when Opus gets an internal fix to DTX.
25 size_t* zero_counts; 27 size_t* zero_counts;
26 }; 28 };
27 29
28 struct WebRtcOpusDecInst { 30 struct WebRtcOpusDecInst {
29 OpusDecoder* decoder; 31 OpusDecoder* decoder;
30 int prev_decoded_samples; 32 int prev_decoded_samples;
31 int channels; 33 int channels;
32 int in_dtx_mode; 34 int in_dtx_mode;
33 }; 35 };
34 36
35 37
36 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_ 38 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
OLDNEW
« no previous file with comments | « no previous file | no next file » | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698