Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(433)

Side by Side Diff: webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h

Issue 1345433002: Add RTC_ prefix to contructormagic macros. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Formatting fix. 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 73 matching lines...) Expand 10 before | Expand all | Expand 10 after
84 84
85 // Have we accepted input but not yet emitted it in a packet? 85 // Have we accepted input but not yet emitted it in a packet?
86 bool packet_in_progress_ = false; 86 bool packet_in_progress_ = false;
87 87
88 // Timestamp of the first input of the currently in-progress packet. 88 // Timestamp of the first input of the currently in-progress packet.
89 uint32_t packet_timestamp_; 89 uint32_t packet_timestamp_;
90 90
91 // Timestamp of the previously encoded packet. 91 // Timestamp of the previously encoded packet.
92 uint32_t last_encoded_timestamp_; 92 uint32_t last_encoded_timestamp_;
93 93
94 DISALLOW_COPY_AND_ASSIGN(AudioEncoderIsacT); 94 RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIsacT);
95 }; 95 };
96 96
97 template <typename T> 97 template <typename T>
98 class AudioDecoderIsacT final : public AudioDecoder { 98 class AudioDecoderIsacT final : public AudioDecoder {
99 public: 99 public:
100 AudioDecoderIsacT(); 100 AudioDecoderIsacT();
101 explicit AudioDecoderIsacT(LockedIsacBandwidthInfo* bwinfo); 101 explicit AudioDecoderIsacT(LockedIsacBandwidthInfo* bwinfo);
102 ~AudioDecoderIsacT() override; 102 ~AudioDecoderIsacT() override;
103 103
104 bool HasDecodePlc() const override; 104 bool HasDecodePlc() const override;
(...skipping 10 matching lines...) Expand all
115 size_t encoded_len, 115 size_t encoded_len,
116 int sample_rate_hz, 116 int sample_rate_hz,
117 int16_t* decoded, 117 int16_t* decoded,
118 SpeechType* speech_type) override; 118 SpeechType* speech_type) override;
119 119
120 private: 120 private:
121 typename T::instance_type* isac_state_; 121 typename T::instance_type* isac_state_;
122 LockedIsacBandwidthInfo* bwinfo_; 122 LockedIsacBandwidthInfo* bwinfo_;
123 int decoder_sample_rate_hz_; 123 int decoder_sample_rate_hz_;
124 124
125 DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsacT); 125 RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsacT);
126 }; 126 };
127 127
128 } // namespace webrtc 128 } // namespace webrtc
129 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_ 129 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698