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gerrit-public.fairphone.software
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platform
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external
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webrtc
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325b34542d9bdc5a38d01d1c69e61a63eaa36ab1
325b345
There was an old scaling for CNG 48 kHz in the code, from the time where Audio Coding Module didn't have full 48 kHz support. This CL removes the scaling.
by Tina le Grand
· 9 years ago
88eeac4
Adding video_processing to presubmit lint check
by mflodman
· 9 years ago
4654d20
Add test which verifies that the RTP header extensions are set correctly for FEC packets.
by Stefan Holmer
· 9 years ago
03ef053
Merge webrtc/video_engine/ into webrtc/video/
by Peter Boström
· 9 years ago
99ab944
Clang format of video_processing folder.
by mflodman
· 9 years ago
a440c6f
Roll chromium_revision 3b8be21..9dfb3a1 (363445:363718)
by kjellander
· 9 years ago
3868692
Free SCTP data channels asynchronously in PeerConnection.
by deadbeef
· 9 years ago
46ad542
Revert of "Create rtc::AtomicInt POD struct." (patchset #3 id:40001 of https://codereview.webrtc.org/1498953002/ )
by pbos
· 9 years ago
6f28cf0
Implement standalone event tracing in AppRTCDemo.
by Peter Boström
· 9 years ago
84f0970
Reland of "Create rtc::AtomicInt POD struct."
by Peter Boström
· 9 years ago
0f490a5
Add logs when stun or turn host lookup is completed.
by Honghai Zhang
· 9 years ago
cd4003f
Use @webrtc.org addresses for OWNERS.
by Peter Boström
· 9 years ago
cf890bc
Roll gtest-parallel.
by Peter Boström
· 9 years ago
0608dc5
Roll chromium_revision 4918765..3b8be21 (363393:363445)
by kjellander
· 9 years ago
5f6deaf
Remove unused RTP-header parser.
by Peter Boström
· 9 years ago
ab82cbb
Disable RtcEventLogTest.DropOldEvents on memcheck.
by Peter Boström
· 9 years ago
03671cb
Use existing parser in ReceivesAndRetransmitsNack.
by Peter Boström
· 9 years ago
fc47ed6
rtcp::Rrtr block moved into own file and got Parse function
by Danil Chapovalov
· 9 years ago
1aa420b
Remove avg encode time from CpuOveruseMetric struct and use value from OnEncodedFrame instead.
by asapersson
· 9 years ago
9d69c3f
Return a copy of the supported RTP header extensions instead of a reference.
by Stefan Holmer
· 9 years ago
b86d4e4
Prepare the AudioSendStream to be hooked up to send-side BWE.
by Stefan Holmer
· 9 years ago
03f80eb
Refactor EglBase configuration.
by nisse
· 9 years ago
a856542
Initial VideoProcessing refactoring.
by mflodman
· 9 years ago
2512f44
Roll chromium_revision 292ab9f..4918765 (363376:363393)
by kjellander
· 9 years ago
c9f1cb8
Roll chromium_revision 72c3265..292ab9f (363365:363376)
by kjellander
· 9 years ago
34a7054
Roll chromium_revision 626eecf..72c3265 (363027:363365)
by kjellander
· 9 years ago
1218d7a
Allow remote fingerprint update during a call
by Guo-wei Shieh
· 9 years ago
86aaa4b
Revert "Allow remote fingerprint update during a call"
by Guo-wei Shieh
· 9 years ago
9c38c2d
Allow remote fingerprint update during a call
by Guo-wei Shieh
· 9 years ago
381b421
Ping backup connection at a slower rate
by Honghai Zhang
· 9 years ago
45b0efd
Stop sending stun binding requests after certain amount of time.
by honghaiz
· 9 years ago
9e1b992
Clear old decoders after recreating the receiver.
by Peter Boström
· 9 years ago
97f7e13
rtcp::ReceiverReport moved into own file and got Parse function
by Danil Chapovalov
· 9 years ago
7c704b8
Use webrtc/base/logging.h in stefan@'s ownership.
by Peter Boström
· 9 years ago
b572768
- Remove calls to VoEDtmf from WVoE/MC.
by Fredrik Solenberg
· 9 years ago
fcdcf4a
Disable RtcEventLogTest.DropOldEvents on DrMemory.
by Peter Boström
· 9 years ago
66f7f4e
Roll chromium_revision d3aa9b1..626eecf (362950:363027)
by kjellander
· 9 years ago
fd59523
Add webrtc/base to deprecated APIs.
by kjellander
· 9 years ago
bc32ab4
Remove 'video_engine_core_unittests' binary.
by Peter Boström
· 9 years ago
ff24c04
Add tests for vp9 (non-flexible mode) using different spatial and temporal configurations.
by Åsa Persson
· 9 years ago
1a5cf6e
Remove the unused NullMediaEngine (and NullVoiceEngine+NullVideoEngine).
by Fredrik Solenberg
· 9 years ago
f7c5776
Refactorings to send RTCP packets directly via the RtcpPacket callback, with some simplifications enabled by this. NACK now also sent via RtcpPacket.
by Erik Språng
· 9 years ago
9cf0c3d
Removes MAYBE_ from several test case names in JsepPeerConnectionP2PTestClient.
by Ivo Creusen
· 9 years ago
d048aa0
Make the audio codecs' GN targets self-sufficient
by Henrik Lundin
· 9 years ago
b4a1ae5
Add separate send-side UMA stats for screenshare and video.
by sprang
· 9 years ago
29e3003
Bring back baremetal trybots to the default set.
by kjellander@webrtc.org
· 9 years ago
5385554
Roll chromium_revision 7461ceb..d3aa9b1 (362933:362950)
by kjellander
· 9 years ago
a4527c8
Add comments about the Audio parts of the public Call API being WIP.
by Fredrik Solenberg
· 9 years ago
74a5ffb
Roll chromium_revision f068d2f..7461ceb (362762:362933)
by kjellander
· 9 years ago
631e134
Rewrote the thread synchronization parts of the test for the locking in APM in response to a locking problem when running in a single-threaded manner.
by peah
· 9 years ago
917ba52
autoroll: Update Clang script path.
by kjellander
· 9 years ago
53047c9
Add PRESUBMIT check for native API changes.
by kjellander
· 9 years ago
c3e0fe7
Make it extra safe when deleting a turn entry.
by honghaiz
· 9 years ago
7635684
Fix Mac ObjC PeerConnection API compilation.
by tkchin
· 9 years ago
9462052
In some rare Android systems ConnectivityManager may be null.
by honghaiz
· 9 years ago
a448607
Roll chromium_revision a45c85a..f068d2f (362609:362762)
by kjellander
· 9 years ago
3c28d0d
Disable PeerConnectionEndToEndTest.Call on Mac.
by kjellander@webrtc.org
· 9 years ago
1d63dd0
- Remove cricket::VoiceChannel::PressDtmf(); AFAICT unused.
by solenberg
· 9 years ago
ee524f7
Adding Java binding for CreateSender.
by deadbeef
· 9 years ago
de0fc58
Adding two more debug macros for logging scalar values to file.
by peah
· 9 years ago
7e4e01a
Add header extension filtering for WebRtcVoiceEngine/MediaChannel.
by solenberg
· 9 years ago
2515af2
Removing some unnecessary string manipulation code from VoEBase::GetVersion().
by solenberg
· 9 years ago
d20d247
Fix VideoCaptureAndroid, drop frame when switching camera using textures.
by perkj
· 9 years ago
226a602
Fix problem when drawing to the Android Media encoder surface.
by perkj
· 9 years ago
c729032
Resolves issue with multiple calls to audio unit initialization
by henrika
· 9 years ago
40455d6
This cl change so that we use EGL14 where it is supported and EGL10 otherwise. The idea is to make this agnostic to an application and for WebRTC except in EGLBase.
by perkj
· 9 years ago
e338499
Revert of Add tests for vp9 (non-flexible mode) using different spatial and temporal configurations. (patchset #18 id:580001 of https://codereview.webrtc.org/1437463002/ )
by asapersson
· 9 years ago
43b4806
Add tests for vp9 (non-flexible mode) using different spatial and temporal configurations.
by asapersson
· 9 years ago
06104b8
Roll chromium_revision eeff895..a45c85a (362465:362609)
by kjellander
· 9 years ago
41b0798
Adding CreatePeerConnection method that uses new PC Initialize method.
by deadbeef
· 9 years ago
62a91ee
Roll chromium_revision 35f35af..eeff895 (362385:362465)
by kjellander
· 9 years ago
187db63
Remove VideoReceiveStream deregister of decoders.
by Peter Boström
· 9 years ago
04a6bb9
Roll chromium_revision f9fedae..35f35af (362322:362385)
by kjellander
· 9 years ago
f94abf7
Nuke webrtc/common_video/plane.*.
by Peter Boström
· 9 years ago
dfbb3a4
Simplify CodecManager::RegisterEncoder()
by kwiberg
· 9 years ago
46c9cc0
Provide method for returning certificate expiration time stamp.
by Torbjorn Granlund
· 9 years ago
ea07373
Enable cpplint for webrtc/audio and webrtc/call, and fix all uncovered cpplint errors.
by Fredrik Solenberg
· 9 years ago
0de97f1
WebRtcVideoCapturer: SetCaptureState(CS_STOPPED) on Stop and ensure state changes in unittest.
by hbos
· 9 years ago
ec192bd
Revert of Add _decoder CHECK to VCMGenericDecoder constructor. (patchset #2 id:20001 of https://codereview.webrtc.org/1485713002/ )
by kjellander
· 9 years ago
cb9792e
Fix VideoCapturerAndroidTest.testStartWhileCameraIsAlreadyOpen on Android M.
by perkj
· 9 years ago
9f8d39d
Add simple end to end test for video capture and encode using textures.
by perkj
· 9 years ago
021282f
Roll chromium_revision 47ce5fe..f9fedae (362117:362322)
by kjellander
· 9 years ago
14f4144
Add helper KeepRefUntilDone.
by perkj
· 9 years ago
ee69ed5
Add separate event for camera freeze.
by glaznev
· 9 years ago
70c0e29
Disable PeerConnectionEndToEndTest.Call for TSan.
by kjellander@webrtc.org
· 9 years ago
f893df3
Add third_party/libc++static to .gitignore
by kjellander@webrtc.org
· 9 years ago
a443ec1
Add _decoder CHECK to VCMGenericDecoder constructor.
by Peter Boström
· 9 years ago
7640ffa
Initialize type_preference_ in TestPort.
by pbos
· 9 years ago
f9203c6
Roll chromium_revision faa24ae..47ce5fe (362083:362117)
by kjellander
· 9 years ago
99f8566
Roll chromium_revision 0da9346..faa24ae (362069:362083)
by kjellander
· 9 years ago
bdf001a
Roll chromium_revision 8f57310..0da9346 (362067:362069)
by kjellander
· 9 years ago
90728b9
Roll chromium_revision 3e15d1a..8f57310 (362064:362067)
by kjellander
· 9 years ago
4c14254
Roll chromium_revision df4d569..3e15d1a (362055:362064)
by kjellander
· 9 years ago
df3efa8
Introduced the new locking scheme
by peah
· 9 years ago
3236b91
Roll chromium_revision c54812d..df4d569 (362052:362055)
by kjellander
· 9 years ago
535727e
Roll chromium_revision 5ac8f02..c54812d (362046:362052)
by kjellander
· 9 years ago
ae54b83
Android SurfaceViewRenderer: Add resetStatistics() method
by magjed
· 9 years ago
43f1809
Roll chromium_revision 7b99051..5ac8f02 (361977:362046)
by kjellander
· 9 years ago
2fe1cb0
Don't overwrite audio stats when they're not available.
by andrew
· 9 years ago
7e43138
-Removed the state as an input to the FilterAdaptation function.
by peah
· 9 years ago
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