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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_ | 11 #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_ |
12 #define WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_ | 12 #define WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_ |
| 13 // The contents of this file have moved to |
| 14 // //webrtc/audio/utility. This file is deprecated. |
13 | 15 |
14 #include "webrtc/typedefs.h" | 16 // TODO(aleloi): Remove this file when clients have updated their |
15 | 17 // includes. See bugs.webrtc.org/6548. |
16 namespace webrtc { | 18 #include "webrtc/audio/utility/audio_frame_operations.h" |
17 | |
18 class AudioFrame; | |
19 | |
20 // TODO(andrew): consolidate this with utility.h and audio_frame_manipulator.h. | |
21 // Change reference parameters to pointers. Consider using a namespace rather | |
22 // than a class. | |
23 class AudioFrameOperations { | |
24 public: | |
25 // Upmixes mono |src_audio| to stereo |dst_audio|. This is an out-of-place | |
26 // operation, meaning src_audio and dst_audio must point to different | |
27 // buffers. It is the caller's responsibility to ensure that |dst_audio| is | |
28 // sufficiently large. | |
29 static void MonoToStereo(const int16_t* src_audio, size_t samples_per_channel, | |
30 int16_t* dst_audio); | |
31 // |frame.num_channels_| will be updated. This version checks for sufficient | |
32 // buffer size and that |num_channels_| is mono. | |
33 static int MonoToStereo(AudioFrame* frame); | |
34 | |
35 // Downmixes stereo |src_audio| to mono |dst_audio|. This is an in-place | |
36 // operation, meaning |src_audio| and |dst_audio| may point to the same | |
37 // buffer. | |
38 static void StereoToMono(const int16_t* src_audio, size_t samples_per_channel, | |
39 int16_t* dst_audio); | |
40 // |frame.num_channels_| will be updated. This version checks that | |
41 // |num_channels_| is stereo. | |
42 static int StereoToMono(AudioFrame* frame); | |
43 | |
44 // Swap the left and right channels of |frame|. Fails silently if |frame| is | |
45 // not stereo. | |
46 static void SwapStereoChannels(AudioFrame* frame); | |
47 | |
48 // Conditionally zero out contents of |frame| for implementing audio mute: | |
49 // |previous_frame_muted| && |current_frame_muted| - Zero out whole frame. | |
50 // |previous_frame_muted| && !|current_frame_muted| - Fade-in at frame start. | |
51 // !|previous_frame_muted| && |current_frame_muted| - Fade-out at frame end. | |
52 // !|previous_frame_muted| && !|current_frame_muted| - Leave frame untouched. | |
53 static void Mute(AudioFrame* frame, bool previous_frame_muted, | |
54 bool current_frame_muted); | |
55 | |
56 static int Scale(float left, float right, AudioFrame& frame); | |
57 | |
58 static int ScaleWithSat(float scale, AudioFrame& frame); | |
59 }; | |
60 | |
61 } // namespace webrtc | |
62 | 19 |
63 #endif // #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_ | 20 #endif // #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_ |
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