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/audio_device_impl.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/audio_device_impl.h
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931e6583b21d2d3d1ee8fd240f63708dc56d1a19 |
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20-May-2015 |
Tommi <tommi@webrtc.org> |
Remove unnecessary dependencies for voe when building with include_internal_audio_device==0. In particular and practical terms, this avoids pulling in AudioDeviceModuleImpl and associated classes, in Chrome. BUG= R=henrika@webrtc.org Review URL: https://webrtc-codereview.appspot.com/49999004 Cr-Commit-Position: refs/heads/master@{#9229}
/external/webrtc/webrtc/modules/audio_device/audio_device_impl.h
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68898a265283de31f16e519c1218e716e61ba508 |
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19-May-2015 |
Tommi <tommi@webrtc.org> |
Remove AudioDeviceUtility. The class doesn't do anything in almost all cases except for grabbing and releasing locks + allocate memory. There are a couple of methods there such as WaitForKey and GetTimeInMs that are used, but those methods aren't specific to audio and we have implementations of these elsewhere. The third method, StringCompare isn't used anywhere (and also isn't specific to audio). BUG= R=henrika@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50009004 Cr-Commit-Position: refs/heads/master@{#9220}
/external/webrtc/webrtc/modules/audio_device/audio_device_impl.h
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24e56e3ee8997761369784cecddac7821d89aec7 |
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19-May-2015 |
henrika <henrika@chromium.org> |
Fixes Chromium FYI build issue on Android. See https://build.chromium.org/p/chromium.webrtc.fyi/waterfall?builder=Android%20Builder%20(dbg) for details BUG= R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/47219004 Cr-Commit-Position: refs/heads/master@{#9217}
/external/webrtc/webrtc/modules/audio_device/audio_device_impl.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/audio_device_impl.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/audio_device_impl.h
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14665ff7d4024d07e58622f498b23fd980001871 |
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04-Mar-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro Clang version changed 223108:230914 Details: https://chromium.googlesource.com/chromium/src/+/e144d30..6fdb142/tools/clang/scripts/update.sh Removes the OVERRIDE macro defined in: * webrtc/base/common.h * webrtc/typedefs.h The majority of the source changes were done by running this in src/: perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h" -o -name "*.cc*" -o -name "*.mm*"` which converted all: virtual Foo() OVERRIDE functions to: Foo() override Then I manually edited: * talk/media/webrtc/fakewebrtccommon.h * webrtc/test/fake_common.h Remaining uses of OVERRIDE was fixed by search+replace. Manual edits were done to fix virtual destructors that were overriding inherited ones. Finally a build error related to the pure virtual definitions of Read, Write and Rewind in common_types.h required a bit of refactoring in: * webrtc/common_types.cc * webrtc/common_types.h * webrtc/system_wrappers/interface/file_wrapper.h * webrtc/system_wrappers/source/file_impl.cc This roll should make it possible for us to finally re-enable deadlock detection for TSan on the buildbots. BUG=4106 R=pbos@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/41069004 Cr-Commit-Position: refs/heads/master@{#8596} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/audio_device_impl.h
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4161715e3f7e744bc9ef3d3ae437da1e8e4de38d |
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29-Jan-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Remove ChangeUniqueID. This fixes a two year old TODO of deleting dead code :) In cases where the _id or id_ member variable is being used for tracing, I changed the member to at least be const. It doesn't look like id's are that useful anymore so maybe the next step is to get rid of them. BUG= R=henrika@webrtc.org, perkj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/37849004 Cr-Commit-Position: refs/heads/master@{#8201} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8201 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/audio_device_impl.h
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0b1534c52eab79372557a6d81aaf4dd9407f55d3 |
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15-Dec-2014 |
pkasting@chromium.org <pkasting@chromium.org> |
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess. This fixes a variety of MSVC warnings about value truncations when implicitly storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and removes the need for a number of explicit casts. This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack". BUG=chromium:81439 TEST=none R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/33649004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/audio_device_impl.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/audio_device_impl.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/audio_device_impl.h
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096515b0702aaa00dc561cd7cf20df8b826f97c4 |
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30-Jul-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix some chromium-style warnings in webrtc/modules/audio_device/ BUG=163 R=xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1897005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4426 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/audio_device_impl.h
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811269df40fd8cd036b68cfe39bc04cacac0a698 |
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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/audio_device_impl.h
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2550988baaf3a50a2eb1a595c26bc7912ad99b30 |
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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/audio_device_impl.h
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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/audio_device_impl.h
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