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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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ceb450b51d92637f0e28baf79ec7c0a7a3dc18d8
ceb450b
Roll chromium_revision c8eec9a..5c83f4e (360565:360728)
by kjellander
· 10 years ago
17c0aff
Enable VP9 HW decoder on Exynos chips.
by Alex Glaznev
· 10 years ago
7593aad
Re-enable mistakenly disabled PEM tests. Misc cleanup and alignment fixes.
by torbjorng
· 10 years ago
7755e20
Chrome has now been updated.
by perkj
· 10 years ago
726b1f7
Removed dummy "mediastreamsignaling.h"
by perkj
· 10 years ago
191c1f9
Disable all JsepPeerConnectionP2PTestClient tests on Mac due to flakiness on Mac Debug bots.
by ivoc
· 10 years ago
12e21a0
Remove dead code (we no longer support SILK)
by kwiberg
· 10 years ago
ef45323
Android: Make classes non-final
by Magnus Jedvert
· 10 years ago
062e14e
Roll chromium_revision bb7899a..c8eec9a (360504:360565)
by kjellander
· 10 years ago
f399f21
Disable PhysicalSocketTest.TestUdpReadyToSendIPv4 on linux due to flakiness on the Linux64 Debug bot.
by ivoc
· 10 years ago
f22695c
Remove build_with_libjingle and exclude failing iOS tests from 'All' target.
by kjellander@webrtc.org
· 10 years ago
1503867
Disabled several JsepPeerConnectionP2PTestClient tests on Mac, due to flakiness on Debug Mac trybots.
by ivoc
· 10 years ago
e488a0d
Fix DTLS packet boundary handling in SSLStreamAdapterTests.
by jbauch
· 10 years ago
87097a8
Roll chromium_revision ed2e3fb..bb7899a (360379:360504)
by kjellander
· 10 years ago
b6755ab
Revert of Adding thread timeout for audio recorer thread in Java (patchset #2 id:20001 of https://codereview.webrtc.org/1444313002/ )
by henrika
· 10 years ago
488e75f
Patchset 1 yet again relands without modification https://codereview.webrtc.org/1422963003/
by Per
· 10 years ago
0969398
Revert of Remove android_rel from CQ since all of its machines are offline. (patchset #1 id:1 of https://codereview.webrtc.org/1459083002/ )
by kjellander
· 10 years ago
392d0c2
Remove android_rel from CQ since all of its machines are offline.
by Henrik Kjellander
· 10 years ago
521ed7b
Reland Convert internal representation of Srtp cryptos from string to int
by Guo-wei Shieh
· 10 years ago
318166b
Revert of Convert internal representation of Srtp cryptos from string to int. (patchset #10 id:180001 of https://codereview.webrtc.org/1416673006/ )
by guoweis
· 10 years ago
2764e10
Convert internal representation of Srtp cryptos from string to int.
by guoweis
· 10 years ago
3c652b6
modules/audio_coding: Remove some codec include dirs
by kjellander@webrtc.org
· 10 years ago
b7ce964
modules/video_coding/utility: Remove include
by kjellander@webrtc.org
· 10 years ago
1b20d81
Roll chromium_revision 64f2817..ed2e3fb (360275:360379)
by kjellander
· 10 years ago
0f59a88
modules/video_processing: refactor interface->include + more.
by Henrik Kjellander
· 10 years ago
ed7d6ec
WebRTC: Add compability header for video_coding refactoring.
by Henrik Kjellander
· 10 years ago
ad948c4
Preliminary support of VP9 HW encoder on Android.
by Alex Glaznev
· 10 years ago
2557b86
modules/video_coding refactorings
by Henrik Kjellander
· 10 years ago
4dd7a65
Temporarily disable VERIFY while bug is investigated.
by phoglund
· 10 years ago
223692a
Remove dead code
by kwiberg
· 10 years ago
e1a27d4
Move CNG/RED payload type extraction to Rent-A-Codec
by kwiberg
· 10 years ago
49a6c99
Disables BitrateEstimatorTest.SwitchesToASTThenBackToTOFForVideo on win_drmemory_full due to flakiness.
by ivoc
· 10 years ago
2446e5a
Fixed the render queue item size preallocation and verification, moved constant declarations, removed redundant queue allocation
by peah
· 10 years ago
0219c9b
rtcp::App moved into own file and got Parse function
by danilchap
· 10 years ago
2aff615
Remove spammy logging of RTCP delivery failures.
by Peter Boström
· 10 years ago
f70568c
So long and thanks for all the code reviews!
by andrew
· 10 years ago
cb50c96
Set temporal up switch bit to false for flexible mode (one temporal layer is configured currently).
by asapersson
· 10 years ago
aa45843
Roll chromium_revision a6d9f7f..64f2817 (360123:360275)
by kjellander
· 10 years ago
310b093
Fix active tcp port to 9
by Guo-wei Shieh
· 10 years ago
2935e01
Several Tick counter improvements try #2."
by thaloun
· 10 years ago
c073615
Update references to TLS1_CK_ECDHE_RSA_CHACHA20_POLY1305, etc.
by davidben
· 10 years ago
0a75749
Roll chromium_revision 04756fa..a6d9f7f (360053:360123)
by kjellander
· 10 years ago
32f3996
Re-apply change https://codereview.webrtc.org/1426673007/
by honghaiz
· 10 years ago
5c489c9
Add OpenSL ES enable setting to AppRTCDemo (part 2).
by henrika
· 10 years ago
2be7c54
Remove ViEEncoder::ScaleInputImage.
by Peter Boström
· 10 years ago
bd05f0b
Unconditionally build VP9 support.
by Peter Boström
· 10 years ago
18adf0a
Add UMA for send bwe and pacer bitrate.
by stefan
· 10 years ago
d9eec76
Trace encoding/decoding time in a generic way.
by pbos
· 10 years ago
5a71f03
Deactivate the audio session after a call ends using the AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation constant
by henrika
· 10 years ago
45e998d
Roll chromium_revision a2e8a40..04756fa (359987:360053)
by kjellander
· 10 years ago
fd614c2
Adding thread timeout for audio recorer thread in Java
by henrika
· 10 years ago
e663392
Add OpenSL ES enable setting to AppRTCDemo.
by glaznev
· 10 years ago
3c12f4d
Revert of Create rtc::AtomicInt POD struct. (patchset #12 id:220001 of https://codereview.webrtc.org/1420043008/ )
by pbos
· 10 years ago
192164e
Preparational work before introducing the locks in order to harmonize the code:
by peah
· 10 years ago
4d291f7
Applied the render queueing to the agc.
by peah
· 10 years ago
03179cd
Roll chromium_revision 6fd4bdd..a2e8a40 (359891:359987)
by kjellander
· 10 years ago
740c4f1
Remove packet initializer in RtpRtcpRtxNackTest.
by pbos
· 10 years ago
854e84c
Use webrtc/base/logging.h for video coding/processing.
by pbos
· 10 years ago
c91d173
Revert of Several Tick counter improvements. (patchset #8 id:140001 of https://codereview.webrtc.org/1415923010/ )
by thaloun
· 10 years ago
fa6228e
Introduced the render sample queue for the aec and aecm.
by peah
· 10 years ago
4c27e4b
Several Tick counter improvements.
by Tim Haloun
· 10 years ago
eb8b388
Fix VP9 support in AppRTCDemo.
by Alex Glaznev
· 10 years ago
6f8ce06
common_video: rename interface -> include
by kjellander
· 10 years ago
591cb1f
Roll chromium_revision c958aa7..6fd4bdd (359816:359891)
by kjellander
· 10 years ago
b27f590
Create rtc::AtomicInt POD struct.
by pbos
· 10 years ago
3528a27
Flesh out webrtc/.gitignore
by brucedawson
· 10 years ago
482b12e
Remove BundleFilter filtering of RTCP.
by pbos
· 10 years ago
8b85de2
Converted a bunch of error checking in Audio[Receive|Send]Stream to RTC_CHECKs instead. They should never fail.
by solenberg
· 10 years ago
9a7c838
Adding stddef.h to opus_inst.h.
by minyue
· 10 years ago
3a94154
Move some send stream configuration into webrtc::AudioSendStream.
by solenberg
· 10 years ago
633a3aa
ThreadUtils: Add joinUninterruptibly() with timeout
by magjed
· 10 years ago
e155ae6
Move CNG and RED management into the Rent-A-Codec
by kwiberg
· 10 years ago
54e9232
Revert of Do not delete the turn port entry right away when the respective connection is deleted. (patchset #5 id:260001 of https://codereview.webrtc.org/1426673007/ )
by tommi
· 10 years ago
2a654fa
Roll chromium_revision cad2987..c958aa7 (359796:359816)
by kjellander
· 10 years ago
0b9e29c
Remove include dirs from modules/{media_file,pacing}
by Henrik Kjellander
· 10 years ago
3e0f602
Android EglBase: Add support for creating EGLSurface from Surface, not SurfaceHolder
by magjed
· 10 years ago
d9b75be
Fix a data race in the thread unit tests.
by nisse
· 10 years ago
6f14be8
Add limit for minimum number of required samples before recording input and sent framerate stats.
by asapersson
· 10 years ago
3c735f4
Roll chromium_revision b77e5bb..cad2987 (359767:359796)
by kjellander
· 10 years ago
8c64860
Roll chromium_revision 3b7968d..b77e5bb (359482:359767)
by kjellander
· 10 years ago
e58fe8e
Do not delete the turn port entry right away when the respective connection is deleted.
by honghaiz
· 10 years ago
fa5d0db
cleanup: get rid of basicdefs.h include
by tfarina
· 10 years ago
a4845ef
Fix flaky tests
by honghaiz
· 10 years ago
4a41361
Android SurfaceViewRenderer: Never hold a pending frame indefinitely
by magjed
· 10 years ago
c01c254
Revert of Android MediaCodecVideoDecoder: Manage lifetime of texture frames (patchset #12 id:320001 of https://codereview.webrtc.org/1422963003/ )
by Per
· 10 years ago
f8506cb
rtcp::Ij renamed to rtcp::ExtendedJitterReport
by danilchap
· 10 years ago
cbe9f51
Revert of Remove global list of SRTP sessions. (patchset #4 id:60001 of https://codereview.webrtc.org/1416093010/ )
by phoglund
· 10 years ago
0fa9b22
Remove scoped_ptrs for VCM sender_ and receiver_.
by pbos
· 10 years ago
df948f0
rtcp::ReportBlock refactored to contain parsing
by danilchap
· 10 years ago
0a41893
Remove BitrateController dependency fromVideoReceiveStream.
by mflodman
· 10 years ago
464c087
Rename screenshare test.
by philipel
· 10 years ago
0e7e259
Move BitrateAllocator from BitrateController logic to Call.
by mflodman
· 10 years ago
69191ed
Roll chromium_revision 4771dd5..3b7968d (359351:359482)
by kjellander
· 10 years ago
faac497
Fix for scenario where m-line is revived after being set to port 0.
by deadbeef
· 10 years ago
69d0d46
Roll chromium_revision e658ee0..4771dd5 (359300:359351)
by kjellander
· 10 years ago
2cd7afe
Do not delete a connection until it has not received anything for 30 seconds.
by Honghai Zhang
· 10 years ago
8597543
Schedule a CreatePermissionRequest after the success of a previous request
by Honghai Zhang
· 10 years ago
68876f9
Introduces Android API level linting, fixes all current API lint errors.
by Patrik Höglund
· 10 years ago
56a34df
Re-add a thread check in Call::Call that was removed by mistake in a rebase.
by solenberg
· 10 years ago
9576e54
Reland "Prepare MediaCodecVideoEncoder for surface textures.""
by perkj
· 10 years ago
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