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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
36b29d1df3a44122621ed58d4e266ab5449b919d
/
pc
/
test
/
fakeaudiocapturemodule.h
36b29d1
Enable cpplint in pc/
by Steve Anton
· 7 years ago
6592f2c
Removes more unused ADM APIs:
by henrika
· 7 years ago
a32dd01
Reland "Remove AudioDeviceObserver and make ADM not inherit from the Module interface."
by Fredrik Solenberg
· 7 years ago
d4404c2
Revert "Remove AudioDeviceObserver and make ADM not inherit from the Module interface."
by Fredrik Solenberg
· 7 years ago
34cdd2d
Remove AudioDeviceObserver and make ADM not inherit from the Module interface.
by Fredrik Solenberg
· 7 years ago
7120742
Adding NOLINT for typedefs.h and common_types.h
by Mirko Bonadei
· 7 years ago
92ea95e
Fixing WebRTC after moving from src/webrtc to src/
by Mirko Bonadei
· 7 years ago
bb54720
Moving src/webrtc into src/.
by Mirko Bonadei
· 7 years ago
[Renamed from webrtc/pc/test/fakeaudiocapturemodule.h]
9868042
Removes unused APIs from the ADM (part II).
by henrika
· 7 years ago
ecf312e
Removes unused WaveOut APIs from ADM.
by henrika
· 7 years ago
c20978e
Rename webrtc/base -> webrtc/rtc_base
by Edward Lemur
· 7 years ago
a80c16a
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
by Henrik Kjellander
· 8 years ago
c3771cc
Update includes for webrtc/{base => rtc_base} rename (2/3)
by kjellander
· 8 years ago
7bb87ee
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
by ossu
· 8 years ago
[Renamed (98%) from webrtc/api/test/fakeaudiocapturemodule.h]
88e31a3
Fix warnings, simplify ADM.
by maxmorin
· 8 years ago
fd8be34
Remove webrtc/base/scoped_ptr.h
by kwiberg
· 9 years ago
6ab3db2
Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ )
by kwiberg
· 9 years ago
65fc62e
Remove webrtc/base/scoped_ptr.h
by kwiberg
· 9 years ago
82d7862
Change default timestamp to 64 bits in all webrtc directories.
by Honghai Zhang
· 9 years ago
ef8b61e
Enable -Winconsistent-missing-override flag.
by nisse
· 9 years ago
d1fe281
Replace scoped_ptr with unique_ptr in webrtc/api/
by kwiberg
· 9 years ago
a26ac92
Reland of move ignored return code from modules. (patchset #1 id:1 of https://codereview.webrtc.org/1736663004/ )
by pbos
· 9 years ago
da33a8a
Revert of Remove ignored return code from modules. (patchset #3 id:40001 of https://codereview.webrtc.org/1703833002/ )
by torbjorng
· 9 years ago
f14c47a
Remove ignored return code from modules.
by Peter Boström
· 9 years ago
b24317b
Fix license headers in webrtc/api.
by kjellander
· 9 years ago
15583c1
Move talk/app/webrtc to webrtc/api
by Henrik Kjellander
· 9 years ago
[Renamed (98%) from talk/app/webrtc/test/fakeaudiocapturemodule.h]
5ad935c
Remove mutable from rtc::CriticalSection members.
by pbos
· 9 years ago
0c4e06b
Use suffixed {uint,int}{8,16,32,64}_t types.
by Peter Boström
· 9 years ago
c14f5ff
Improving support for Android Audio Effects in WebRTC.
by henrika
· 9 years ago
dce40cf
Update a ton of audio code to use size_t more correctly and in general reduce
by Peter Kasting
· 9 years ago
ee8c6d3
In PeerConnectionTestWrapper, put audio input on a separate thread.
by deadbeef
· 9 years ago
f045e4d
Prepare to convert various types to size_t.
by Peter Kasting
· 10 years ago
14665ff
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
by kjellander@webrtc.org
· 10 years ago
4161715
Remove ChangeUniqueID.
by tommi@webrtc.org
· 10 years ago
5f93d0a
Update libjingle license statements at top of talk files for consistency
by jlmiller@webrtc.org
· 10 years ago
0b1534c
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
by pkasting@chromium.org
· 10 years ago
a954c07
AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer
by henrika@webrtc.org
· 10 years ago
1972ff8
Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE.
by henrik.lundin@webrtc.org
· 10 years ago
d4e598d
(Auto)update libjingle 72097588-> 72159069
by buildbot@webrtc.org
· 10 years ago
8804a29
Add CriticalSection to fakeaudiocapturemodule to protect the variables which will be accessed from process_thread_ and the main thread.
by wu@webrtc.org
· 11 years ago
28e2075
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk
by henrike@webrtc.org
· 11 years ago