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1 /* | 1 /* |
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 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_DEVICE_FINE_AUDIO_BUFFER_H_ | 11 #ifndef WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_ |
12 #define WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_ | 12 #define WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_ |
13 | 13 |
14 #include "webrtc/base/scoped_ptr.h" | 14 #include <memory> |
| 15 |
15 #include "webrtc/typedefs.h" | 16 #include "webrtc/typedefs.h" |
16 | 17 |
17 namespace webrtc { | 18 namespace webrtc { |
18 | 19 |
19 class AudioDeviceBuffer; | 20 class AudioDeviceBuffer; |
20 | 21 |
21 // FineAudioBuffer takes an AudioDeviceBuffer (ADB) which deals with audio data | 22 // FineAudioBuffer takes an AudioDeviceBuffer (ADB) which deals with audio data |
22 // corresponding to 10ms of data. It then allows for this data to be pulled in | 23 // corresponding to 10ms of data. It then allows for this data to be pulled in |
23 // a finer or coarser granularity. I.e. interacting with this class instead of | 24 // a finer or coarser granularity. I.e. interacting with this class instead of |
24 // directly with the AudioDeviceBuffer one can ask for any number of audio data | 25 // directly with the AudioDeviceBuffer one can ask for any number of audio data |
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79 // Number of bytes delivered by GetPlayoutData() call and provided to | 80 // Number of bytes delivered by GetPlayoutData() call and provided to |
80 // DeliverRecordedData(). | 81 // DeliverRecordedData(). |
81 const size_t desired_frame_size_bytes_; | 82 const size_t desired_frame_size_bytes_; |
82 // Sample rate in Hertz. | 83 // Sample rate in Hertz. |
83 const int sample_rate_; | 84 const int sample_rate_; |
84 // Number of audio samples per 10ms. | 85 // Number of audio samples per 10ms. |
85 const size_t samples_per_10_ms_; | 86 const size_t samples_per_10_ms_; |
86 // Number of audio bytes per 10ms. | 87 // Number of audio bytes per 10ms. |
87 const size_t bytes_per_10_ms_; | 88 const size_t bytes_per_10_ms_; |
88 // Storage for output samples that are not yet asked for. | 89 // Storage for output samples that are not yet asked for. |
89 rtc::scoped_ptr<int8_t[]> playout_cache_buffer_; | 90 std::unique_ptr<int8_t[]> playout_cache_buffer_; |
90 // Location of first unread output sample. | 91 // Location of first unread output sample. |
91 size_t playout_cached_buffer_start_; | 92 size_t playout_cached_buffer_start_; |
92 // Number of bytes stored in output (contain samples to be played out) cache. | 93 // Number of bytes stored in output (contain samples to be played out) cache. |
93 size_t playout_cached_bytes_; | 94 size_t playout_cached_bytes_; |
94 // Storage for input samples that are about to be delivered to the WebRTC | 95 // Storage for input samples that are about to be delivered to the WebRTC |
95 // ADB or remains from the last successful delivery of a 10ms audio buffer. | 96 // ADB or remains from the last successful delivery of a 10ms audio buffer. |
96 rtc::scoped_ptr<int8_t[]> record_cache_buffer_; | 97 std::unique_ptr<int8_t[]> record_cache_buffer_; |
97 // Required (max) size in bytes of the |record_cache_buffer_|. | 98 // Required (max) size in bytes of the |record_cache_buffer_|. |
98 const size_t required_record_buffer_size_bytes_; | 99 const size_t required_record_buffer_size_bytes_; |
99 // Number of bytes in input (contains recorded samples) cache. | 100 // Number of bytes in input (contains recorded samples) cache. |
100 size_t record_cached_bytes_; | 101 size_t record_cached_bytes_; |
101 // Read and write pointers used in the buffering scheme on the recording side. | 102 // Read and write pointers used in the buffering scheme on the recording side. |
102 size_t record_read_pos_; | 103 size_t record_read_pos_; |
103 size_t record_write_pos_; | 104 size_t record_write_pos_; |
104 }; | 105 }; |
105 | 106 |
106 } // namespace webrtc | 107 } // namespace webrtc |
107 | 108 |
108 #endif // WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_ | 109 #endif // WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_ |
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