| 8c38e8b9b96d72317d6ce94c1442113b4e385dcb |
|
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 |
|
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 |
|
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 |
|
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 |
|
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 |
|
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 |
|
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
|
| 8454ad1b3ec8446e3a4882b41e78a6c1bda536b8 |
|
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
|
| e08a11c4a17241c3f1232e1d6923e011860f0d88 |
|
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
|
| fa042ca15d18fc02e3b93c3278f899ba0e825f5e |
|
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
|