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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_COMMON_AUDIO_AUDIO_CONVERTER_H_ | 11 #ifndef WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_ |
12 #define WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_ | 12 #define WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_ |
13 | 13 |
14 #include <memory> | 14 #include <memory> |
15 | 15 |
16 #include "webrtc/rtc_base/constructormagic.h" | 16 #include "webrtc/base/constructormagic.h" |
17 | 17 |
18 namespace webrtc { | 18 namespace webrtc { |
19 | 19 |
20 // Format conversion (remixing and resampling) for audio. Only simple remixing | 20 // Format conversion (remixing and resampling) for audio. Only simple remixing |
21 // conversions are supported: downmix to mono (i.e. |dst_channels| == 1) or | 21 // conversions are supported: downmix to mono (i.e. |dst_channels| == 1) or |
22 // upmix from mono (i.e. |src_channels == 1|). | 22 // upmix from mono (i.e. |src_channels == 1|). |
23 // | 23 // |
24 // The source and destination chunks have the same duration in time; specifying | 24 // The source and destination chunks have the same duration in time; specifying |
25 // the number of frames is equivalent to specifying the sample rates. | 25 // the number of frames is equivalent to specifying the sample rates. |
26 class AudioConverter { | 26 class AudioConverter { |
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58 const size_t src_frames_; | 58 const size_t src_frames_; |
59 const size_t dst_channels_; | 59 const size_t dst_channels_; |
60 const size_t dst_frames_; | 60 const size_t dst_frames_; |
61 | 61 |
62 RTC_DISALLOW_COPY_AND_ASSIGN(AudioConverter); | 62 RTC_DISALLOW_COPY_AND_ASSIGN(AudioConverter); |
63 }; | 63 }; |
64 | 64 |
65 } // namespace webrtc | 65 } // namespace webrtc |
66 | 66 |
67 #endif // WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_ | 67 #endif // WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_ |
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