Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(110)

Side by Side Diff: webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h

Issue 1353803002: Simple cleanups of AudioDecoder and AudioEncoder classes (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@dmove-isac
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
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 48 matching lines...) Expand 10 before | Expand all | Expand 10 after
59 59
60 size_t SamplesPerChannel() const; 60 size_t SamplesPerChannel() const;
61 61
62 const int num_channels_; 62 const int num_channels_;
63 const int payload_type_; 63 const int payload_type_;
64 const size_t num_10ms_frames_per_packet_; 64 const size_t num_10ms_frames_per_packet_;
65 size_t num_10ms_frames_buffered_; 65 size_t num_10ms_frames_buffered_;
66 uint32_t first_timestamp_in_buffer_; 66 uint32_t first_timestamp_in_buffer_;
67 const rtc::scoped_ptr<EncoderState[]> encoders_; 67 const rtc::scoped_ptr<EncoderState[]> encoders_;
68 rtc::Buffer interleave_buffer_; 68 rtc::Buffer interleave_buffer_;
69 RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderG722);
69 }; 70 };
70 71
71 } // namespace webrtc 72 } // namespace webrtc
72 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_INCLUDE_AUDIO_ENCODER_G722_H_ 73 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_INCLUDE_AUDIO_ENCODER_G722_H_
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698