2011-09-12 12:24:39 +00:00
|
|
|
# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
|
|
|
|
|
#
|
|
|
|
|
# Use of this source code is governed by a BSD-style license
|
|
|
|
|
# that can be found in the LICENSE file in the root of the source
|
|
|
|
|
# tree. An additional intellectual property rights grant can be found
|
|
|
|
|
# in the file PATENTS. All contributing project authors may
|
|
|
|
|
# be found in the AUTHORS file in the root of the source tree.
|
2011-07-07 08:21:25 +00:00
|
|
|
|
|
|
|
|
{
|
|
|
|
|
'targets': [
|
|
|
|
|
{
|
|
|
|
|
'target_name': 'webrtc_utility',
|
2013-01-18 23:42:21 +00:00
|
|
|
'type': 'static_library',
|
2011-07-07 08:21:25 +00:00
|
|
|
'dependencies': [
|
2011-09-12 12:24:39 +00:00
|
|
|
'audio_coding_module',
|
2013-08-05 18:45:19 +00:00
|
|
|
'media_file',
|
2013-04-30 23:43:26 +00:00
|
|
|
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
|
2015-01-14 09:30:52 +00:00
|
|
|
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
|
2011-07-07 08:21:25 +00:00
|
|
|
],
|
|
|
|
|
'sources': [
|
2015-11-04 08:31:52 +01:00
|
|
|
'include/audio_frame_operations.h',
|
|
|
|
|
'include/file_player.h',
|
|
|
|
|
'include/file_recorder.h',
|
|
|
|
|
'include/helpers_android.h',
|
|
|
|
|
'include/helpers_ios.h',
|
|
|
|
|
'include/jvm_android.h',
|
|
|
|
|
'include/process_thread.h',
|
2015-02-25 11:50:17 +00:00
|
|
|
'source/audio_frame_operations.cc',
|
|
|
|
|
'source/coder.cc',
|
|
|
|
|
'source/coder.h',
|
|
|
|
|
'source/file_player_impl.cc',
|
|
|
|
|
'source/file_player_impl.h',
|
|
|
|
|
'source/file_recorder_impl.cc',
|
|
|
|
|
'source/file_recorder_impl.h',
|
|
|
|
|
'source/helpers_android.cc',
|
2015-07-14 17:04:08 +02:00
|
|
|
'source/helpers_ios.mm',
|
Adding support for OpenSL ES output in native WebRTC
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo
Summary:
- Removes dependency of the 'enable_android_opensl' compiler flag.
Instead, OpenSL ES is always supported, and will enabled for devices that
supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.
Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.
R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51759004
Cr-Commit-Position: refs/heads/master@{#9208}
2015-05-18 16:49:16 +02:00
|
|
|
'source/jvm_android.cc',
|
2015-02-25 11:50:17 +00:00
|
|
|
'source/process_thread_impl.cc',
|
|
|
|
|
'source/process_thread_impl.h',
|
2011-08-31 17:03:54 +00:00
|
|
|
],
|
2011-07-07 08:21:25 +00:00
|
|
|
},
|
2011-11-24 07:20:00 +00:00
|
|
|
], # targets
|
2011-07-07 08:21:25 +00:00
|
|
|
}
|