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1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 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_ANDROID_OPENSLES_PLAYER_H_ | 11 #ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_PLAYER_H_ |
12 #define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_PLAYER_H_ | 12 #define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_PLAYER_H_ |
13 | 13 |
| 14 #include <memory> |
| 15 |
14 #include <SLES/OpenSLES.h> | 16 #include <SLES/OpenSLES.h> |
15 #include <SLES/OpenSLES_Android.h> | 17 #include <SLES/OpenSLES_Android.h> |
16 #include <SLES/OpenSLES_AndroidConfiguration.h> | 18 #include <SLES/OpenSLES_AndroidConfiguration.h> |
17 | 19 |
18 #include "webrtc/base/scoped_ptr.h" | |
19 #include "webrtc/base/thread_checker.h" | 20 #include "webrtc/base/thread_checker.h" |
20 #include "webrtc/modules/audio_device/android/audio_common.h" | 21 #include "webrtc/modules/audio_device/android/audio_common.h" |
21 #include "webrtc/modules/audio_device/android/audio_manager.h" | 22 #include "webrtc/modules/audio_device/android/audio_manager.h" |
22 #include "webrtc/modules/audio_device/android/opensles_common.h" | 23 #include "webrtc/modules/audio_device/android/opensles_common.h" |
23 #include "webrtc/modules/audio_device/include/audio_device_defines.h" | 24 #include "webrtc/modules/audio_device/include/audio_device_defines.h" |
24 #include "webrtc/modules/audio_device/audio_device_generic.h" | 25 #include "webrtc/modules/audio_device/audio_device_generic.h" |
25 #include "webrtc/modules/utility/include/helpers_android.h" | 26 #include "webrtc/modules/utility/include/helpers_android.h" |
26 | 27 |
27 namespace webrtc { | 28 namespace webrtc { |
28 | 29 |
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143 SLDataFormat_PCM pcm_format_; | 144 SLDataFormat_PCM pcm_format_; |
144 | 145 |
145 // Number of bytes per audio buffer in each |audio_buffers_[i]|. | 146 // Number of bytes per audio buffer in each |audio_buffers_[i]|. |
146 // Typical sizes are 480 or 512 bytes corresponding to native output buffer | 147 // Typical sizes are 480 or 512 bytes corresponding to native output buffer |
147 // sizes of 240 or 256 audio frames respectively. | 148 // sizes of 240 or 256 audio frames respectively. |
148 size_t bytes_per_buffer_; | 149 size_t bytes_per_buffer_; |
149 | 150 |
150 // Queue of audio buffers to be used by the player object for rendering | 151 // Queue of audio buffers to be used by the player object for rendering |
151 // audio. They will be used in a Round-robin way and the size of each buffer | 152 // audio. They will be used in a Round-robin way and the size of each buffer |
152 // is given by FineAudioBuffer::RequiredBufferSizeBytes(). | 153 // is given by FineAudioBuffer::RequiredBufferSizeBytes(). |
153 rtc::scoped_ptr<SLint8[]> audio_buffers_[kNumOfOpenSLESBuffers]; | 154 std::unique_ptr<SLint8[]> audio_buffers_[kNumOfOpenSLESBuffers]; |
154 | 155 |
155 // FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data | 156 // FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data |
156 // in chunks of 10ms. It then allows for this data to be pulled in | 157 // in chunks of 10ms. It then allows for this data to be pulled in |
157 // a finer or coarser granularity. I.e. interacting with this class instead | 158 // a finer or coarser granularity. I.e. interacting with this class instead |
158 // of directly with the AudioDeviceBuffer one can ask for any number of | 159 // of directly with the AudioDeviceBuffer one can ask for any number of |
159 // audio data samples. | 160 // audio data samples. |
160 // Example: native buffer size is 240 audio frames at 48kHz sample rate. | 161 // Example: native buffer size is 240 audio frames at 48kHz sample rate. |
161 // WebRTC will provide 480 audio frames per 10ms but OpenSL ES asks for 240 | 162 // WebRTC will provide 480 audio frames per 10ms but OpenSL ES asks for 240 |
162 // in each callback (one every 5ms). This class can then ask for 240 and the | 163 // in each callback (one every 5ms). This class can then ask for 240 and the |
163 // FineAudioBuffer will ask WebRTC for new data only every second callback | 164 // FineAudioBuffer will ask WebRTC for new data only every second callback |
164 // and also cach non-utilized audio. | 165 // and also cach non-utilized audio. |
165 rtc::scoped_ptr<FineAudioBuffer> fine_buffer_; | 166 std::unique_ptr<FineAudioBuffer> fine_buffer_; |
166 | 167 |
167 // Keeps track of active audio buffer 'n' in the audio_buffers_[n] queue. | 168 // Keeps track of active audio buffer 'n' in the audio_buffers_[n] queue. |
168 // Example (kNumOfOpenSLESBuffers = 2): counts 0, 1, 0, 1, ... | 169 // Example (kNumOfOpenSLESBuffers = 2): counts 0, 1, 0, 1, ... |
169 int buffer_index_; | 170 int buffer_index_; |
170 | 171 |
171 // The engine object which provides the SLEngineItf interface. | 172 // The engine object which provides the SLEngineItf interface. |
172 // Created by the global Open SL ES constructor slCreateEngine(). | 173 // Created by the global Open SL ES constructor slCreateEngine(). |
173 webrtc::ScopedSLObjectItf engine_object_; | 174 webrtc::ScopedSLObjectItf engine_object_; |
174 | 175 |
175 // This interface exposes creation methods for all the OpenSL ES object types. | 176 // This interface exposes creation methods for all the OpenSL ES object types. |
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196 // properties. This interface is supported on the Audio Player object. | 197 // properties. This interface is supported on the Audio Player object. |
197 SLVolumeItf volume_; | 198 SLVolumeItf volume_; |
198 | 199 |
199 // Last time the OpenSL ES layer asked for audio data to play out. | 200 // Last time the OpenSL ES layer asked for audio data to play out. |
200 uint32_t last_play_time_; | 201 uint32_t last_play_time_; |
201 }; | 202 }; |
202 | 203 |
203 } // namespace webrtc | 204 } // namespace webrtc |
204 | 205 |
205 #endif // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_PLAYER_H_ | 206 #endif // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_PLAYER_H_ |
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