Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(253)

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

Issue 3007763002: Move array_view.h to webrtc/api/ (Closed)
Patch Set: rebase Created 3 years, 3 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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 #include "webrtc/modules/audio_device/android/opensles_player.h" 11 #include "webrtc/modules/audio_device/android/opensles_player.h"
12 12
13 #include <android/log.h> 13 #include <android/log.h>
14 14
15 #include "webrtc/api/array_view.h"
15 #include "webrtc/modules/audio_device/android/audio_common.h" 16 #include "webrtc/modules/audio_device/android/audio_common.h"
16 #include "webrtc/modules/audio_device/android/audio_manager.h" 17 #include "webrtc/modules/audio_device/android/audio_manager.h"
17 #include "webrtc/modules/audio_device/fine_audio_buffer.h" 18 #include "webrtc/modules/audio_device/fine_audio_buffer.h"
18 #include "webrtc/rtc_base/array_view.h"
19 #include "webrtc/rtc_base/arraysize.h" 19 #include "webrtc/rtc_base/arraysize.h"
20 #include "webrtc/rtc_base/checks.h" 20 #include "webrtc/rtc_base/checks.h"
21 #include "webrtc/rtc_base/format_macros.h" 21 #include "webrtc/rtc_base/format_macros.h"
22 #include "webrtc/rtc_base/timeutils.h" 22 #include "webrtc/rtc_base/timeutils.h"
23 23
24 #define TAG "OpenSLESPlayer" 24 #define TAG "OpenSLESPlayer"
25 #define ALOGV(...) __android_log_print(ANDROID_LOG_VERBOSE, TAG, __VA_ARGS__) 25 #define ALOGV(...) __android_log_print(ANDROID_LOG_VERBOSE, TAG, __VA_ARGS__)
26 #define ALOGD(...) __android_log_print(ANDROID_LOG_DEBUG, TAG, __VA_ARGS__) 26 #define ALOGD(...) __android_log_print(ANDROID_LOG_DEBUG, TAG, __VA_ARGS__)
27 #define ALOGE(...) __android_log_print(ANDROID_LOG_ERROR, TAG, __VA_ARGS__) 27 #define ALOGE(...) __android_log_print(ANDROID_LOG_ERROR, TAG, __VA_ARGS__)
28 #define ALOGW(...) __android_log_print(ANDROID_LOG_WARN, TAG, __VA_ARGS__) 28 #define ALOGW(...) __android_log_print(ANDROID_LOG_WARN, TAG, __VA_ARGS__)
(...skipping 387 matching lines...) Expand 10 before | Expand all | Expand 10 after
416 RTC_DCHECK(player_); 416 RTC_DCHECK(player_);
417 SLuint32 state; 417 SLuint32 state;
418 SLresult err = (*player_)->GetPlayState(player_, &state); 418 SLresult err = (*player_)->GetPlayState(player_, &state);
419 if (SL_RESULT_SUCCESS != err) { 419 if (SL_RESULT_SUCCESS != err) {
420 ALOGE("GetPlayState failed: %d", err); 420 ALOGE("GetPlayState failed: %d", err);
421 } 421 }
422 return state; 422 return state;
423 } 423 }
424 424
425 } // namespace webrtc 425 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/modules/audio_device/BUILD.gn ('k') | webrtc/modules/audio_device/android/opensles_recorder.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698