| ff761fba8274d93bd73e76c8b8a1f2d0776dd840 |
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04-Nov-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
modules: more interface -> include renames
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
/external/webrtc/webrtc/modules/audio_device/include/audio_device.h
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| c14f5ff60fb0c42c97702de112a9e8f1eccba574 |
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23-Sep-2015 |
henrika <henrika@webrtc.org> |
Improving support for Android Audio Effects in WebRTC.
Now also supports AGC and NS effects and adds the possibility
to override default settings.
R=magjed@webrtc.org, pbos@webrtc.org, sophiechang@chromium.org
TBR=perkj
BUG=NONE
Review URL: https://codereview.webrtc.org/1344563002 .
Cr-Commit-Position: refs/heads/master@{#10030}
/external/webrtc/webrtc/modules/audio_device/include/audio_device.h
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| ba35d05a4918b3efa7ab88674781aadb48017ff8 |
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14-Jul-2015 |
henrika <henrika@webrtc.org> |
Cleanup of iOS AudioDevice implementation
TBR=tkchin
BUG=webrtc:4789
TEST=modules_unittests --gtest_filter=AudioDeviceTest* and AppRTCDemo
Review URL: https://codereview.webrtc.org/1206783002 .
Cr-Commit-Position: refs/heads/master@{#9578}
/external/webrtc/webrtc/modules/audio_device/include/audio_device.h
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| b26198972c1fcb4aa7abaf3895b007e301e7d5dc |
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18-May-2015 |
henrika <henrika@chromium.org> |
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}
/external/webrtc/webrtc/modules/audio_device/include/audio_device.h
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| 474d1eb22376898b36bcd04b0ce3860fa12fd984 |
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09-Mar-2015 |
henrika@webrtc.org <henrika@webrtc.org> |
Adds C++/JNI/Java unit test for audio device module on Android.
This CL adds support for unittests of the AudioDeviceModule on Android using both Java and C++. The new framework uses ::testing::TesWithParam to support both Java-based audio and OpenSL ES based audio. However, given existing issues in our OpenSL ES implementation, the list of test parameters only contains Java in this first version. Open SL ES will be enabled as soon as the backend has been refactored.
It also:
- Removes the redundant JNIEnv* argument in webrtc::VoiceEngine::SetAndroidObjects().
- Modifies usage of enable_android_opensl and the WEBRTC_ANDROID_OPENSLES define.
- Adds kAndroidJavaAudio and kAndroidOpenSLESAudio to AudioLayer enumerator.
- Fixes some bugs which were discovered when running the tests.
BUG=NONE
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40069004
Cr-Commit-Position: refs/heads/master@{#8651}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8651 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/include/audio_device.h
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| 5c32a84620154a81ef983c23b1d3e6c957ad478a |
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09-Dec-2014 |
tommi@webrtc.org <tommi@webrtc.org> |
Attempt to fix FYI bots.
The FYI bots went red after https://webrtc-codereview.appspot.com/32179004/ landed.
TBR=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7853 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/include/audio_device.h
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| a954c07ee1c93175e6ebbeb20517b347474362ae |
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09-Dec-2014 |
henrika@webrtc.org <henrika@webrtc.org> |
AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer
BUG=4034
R=andrew@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7849 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/include/audio_device.h
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| c7c432aa9b8c9f9ba6d41554917784a27b21426a |
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02-Apr-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove AudioDevice::{Microphone,Speaker}IsAvailable.
This was only used for logging, except on Mac, where the methods are
now private.
BUG=3132
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5831 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/include/audio_device.h
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| 811269df40fd8cd036b68cfe39bc04cacac0a698 |
|
11-Jul-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Include files from webrtc/.. paths in audio_device/.
BUG=1662
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1785005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4330 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/include/audio_device.h
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| 2550988baaf3a50a2eb1a595c26bc7912ad99b30 |
|
09-Apr-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
WebRtc_Word32 -> int32_t in audio_device/
BUG=314
Review URL: https://webrtc-codereview.appspot.com/1302006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3793 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/include/audio_device.h
|
| 14b43beb7ce4440b30dcea31196de5b4a529cb6b |
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22-Oct-2012 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move src/ -> webrtc/
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/915006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/include/audio_device.h
|