8c38e8b9b96d72317d6ce94c1442113b4e385dcb |
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26-Nov-2015 |
Peter Boström <pbos@webrtc.org> |
Clean up PlatformThread. * Move PlatformThread to rtc::. * Remove ::CreateThread factory method. * Make non-scoped_ptr from a lot of invocations. * Make Start/Stop void. * Remove rtc::Thread priorities, which were unused and would collide. * Add ::IsRunning() to PlatformThread. BUG= R=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1476453002 . Cr-Commit-Position: refs/heads/master@{#10812}
/external/webrtc/webrtc/modules/audio_device/dummy/file_audio_device.h
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12411ef40e08c5e28ccde54ab3418c96676ffcbc |
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23-Nov-2015 |
pbos <pbos@webrtc.org> |
Move ThreadWrapper to ProcessThread in base. Also removes all virtual methods. Permits using a thread from rtc_base_approved (namely event tracing). BUG=webrtc:5158 R=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1469013002 Cr-Commit-Position: refs/heads/master@{#10760}
/external/webrtc/webrtc/modules/audio_device/dummy/file_audio_device.h
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98f53510b222f71fdd8b799b2f33737ceeb28c61 |
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28-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
system_wrappers: rename interface -> include BUG=webrtc:5095 R=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1413333002 . Cr-Commit-Position: refs/heads/master@{#10438}
/external/webrtc/webrtc/modules/audio_device/dummy/file_audio_device.h
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dce40cf804019a9898b6ab8d8262466b697c56e0 |
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24-Aug-2015 |
Peter Kasting <pkasting@google.com> |
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
/external/webrtc/webrtc/modules/audio_device/dummy/file_audio_device.h
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361981faa86668cd9b20a2837d0b166fc024cd9b |
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19-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Use scoped_ptr for ThreadWrapper::CreateThread. BUG= R=henrika@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/45799004 Cr-Commit-Position: refs/heads/master@{#8794} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8794 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/dummy/file_audio_device.h
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86639737b83d8877abc4810100e30a8af863189d |
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13-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove thread id from ThreadWrapper::Start(). Removes ThreadPosix::InitParams and a corresponding wait for an event. This unblocks ThreadPosix::Start which had to wait for thread scheduling for an event to trigger on the spawned thread, giving faster Start() calls. BUG=4413 R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/43699004 Cr-Commit-Position: refs/heads/master@{#8709} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8709 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/dummy/file_audio_device.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/dummy/file_audio_device.h
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8454ad1b3ec8446e3a4882b41e78a6c1bda536b8 |
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11-Jun-2014 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Reland: Making WebRTC able to play and record audio to files for tests. By specifying the define WEBRTC_DUMMY_FILE_DEVICES (which is similar to WEBRTC_DUMMY_AUDIO_BUILD) an application will be able to tell WebRTC to play out audio to a file and feed audio in from a file. We want to do so we can better test WebRTC-using applications by recording what the audio stack outputs and feeding known audio in for quality tests. R=henrika@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19729004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6403 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/dummy/file_audio_device.h
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e08a11c4a17241c3f1232e1d6923e011860f0d88 |
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11-Jun-2014 |
minyue@webrtc.org <minyue@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 6395 "Making WebRTC able to play and record audio to file..." > Making WebRTC able to play and record audio to files for tests. > > By specifying the define WEBRTC_DUMMY_FILE_DEVICES (which is similar to > WEBRTC_DUMMY_AUDIO_BUILD) an application will be able to tell WebRTC to > play out audio to a file and feed audio in from a file. We want to do > so we can better test WebRTC-using applications by recording what the > audio stack outputs and feeding known audio in for quality tests. > > R=henrika@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/20609004 TBR=phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/18529004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6396 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/dummy/file_audio_device.h
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fa042ca15d18fc02e3b93c3278f899ba0e825f5e |
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11-Jun-2014 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Making WebRTC able to play and record audio to files for tests. By specifying the define WEBRTC_DUMMY_FILE_DEVICES (which is similar to WEBRTC_DUMMY_AUDIO_BUILD) an application will be able to tell WebRTC to play out audio to a file and feed audio in from a file. We want to do so we can better test WebRTC-using applications by recording what the audio stack outputs and feeding known audio in for quality tests. R=henrika@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20609004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6395 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/dummy/file_audio_device.h
|