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Side by Side Diff: webrtc/modules/audio_device/android/opensles_player.h

Issue 1440623002: OpenSL ES stability improvements (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: nit Created 5 years, 1 month ago
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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
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45 // garbage collection pauses and it supports reduced audio output latency. 45 // garbage collection pauses and it supports reduced audio output latency.
46 // If the device doesn't claim this feature but supports API level 9 (Android 46 // If the device doesn't claim this feature but supports API level 9 (Android
47 // platform version 2.3) or later, then we can still use the OpenSL ES APIs but 47 // platform version 2.3) or later, then we can still use the OpenSL ES APIs but
48 // the output latency may be higher. 48 // the output latency may be higher.
49 class OpenSLESPlayer { 49 class OpenSLESPlayer {
50 public: 50 public:
51 // The lower output latency path is used only if the application requests a 51 // The lower output latency path is used only if the application requests a
52 // buffer count of 2 or more, and a buffer size and sample rate that are 52 // buffer count of 2 or more, and a buffer size and sample rate that are
53 // compatible with the device's native output configuration provided via the 53 // compatible with the device's native output configuration provided via the
54 // audio manager at construction. 54 // audio manager at construction.
55 static const int kNumOfOpenSLESBuffers = 2; 55 static const int kNumOfOpenSLESBuffers = 4;
56 56
57 // There is no need for this class to use JNI. 57 // There is no need for this class to use JNI.
58 static int32_t SetAndroidAudioDeviceObjects(void* javaVM, void* context) { 58 static int32_t SetAndroidAudioDeviceObjects(void* javaVM, void* context) {
59 return 0; 59 return 0;
60 } 60 }
61 static void ClearAndroidAudioDeviceObjects() {} 61 static void ClearAndroidAudioDeviceObjects() {}
62 62
63 explicit OpenSLESPlayer(AudioManager* audio_manager); 63 explicit OpenSLESPlayer(AudioManager* audio_manager);
64 ~OpenSLESPlayer(); 64 ~OpenSLESPlayer();
65 65
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188 SLPlayItf player_; 188 SLPlayItf player_;
189 189
190 // The Android Simple Buffer Queue interface is supported on the audio player 190 // The Android Simple Buffer Queue interface is supported on the audio player
191 // and it provides methods to send audio data from the source to the audio 191 // and it provides methods to send audio data from the source to the audio
192 // player for rendering. 192 // player for rendering.
193 SLAndroidSimpleBufferQueueItf simple_buffer_queue_; 193 SLAndroidSimpleBufferQueueItf simple_buffer_queue_;
194 194
195 // This interface exposes controls for manipulating the object’s audio volume 195 // This interface exposes controls for manipulating the object’s audio volume
196 // properties. This interface is supported on the Audio Player object. 196 // properties. This interface is supported on the Audio Player object.
197 SLVolumeItf volume_; 197 SLVolumeItf volume_;
198
199 // Last time the OpenSL ES layer asked for audio data to play out.
200 uint32_t last_play_time_;
198 }; 201 };
199 202
200 } // namespace webrtc 203 } // namespace webrtc
201 204
202 #endif // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_PLAYER_H_ 205 #endif // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_PLAYER_H_
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