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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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28e2919cfd2c31641609dfebc173e02c1ca4d012
28e2919
Adding Android binding for RTCConfiguration::max_ipv6_networks.
by deadbeef
· 7 years ago
9eb3d19
Fix a crash in PeerConnectionFactory.SetVideoHwAccelerationOptions.
by sakal
· 7 years ago
2d4040e
Add a comment that RTCAVFoundationVideoSource is deprecated.
by sakal
· 7 years ago
571f2a6
Roll chromium_revision 1236993289..7b48b3e487 (489596:489906)
by buildbot
· 7 years ago
81f1da3
Adding missing resources to audio_codec_speed_tests.
by minyue
· 7 years ago
f5f793c
Take smaller interface for RtpRtcp::Configuration::receive_statistics
by danilchap
· 7 years ago
77415f5
Revert of Disable SeqNumUnwrapper death tests to avoid breaking downstream builds. (patchset #1 id:1 of https://codereview.chromium.org/2985083002/ )
by philipel
· 7 years ago
40e7ebd
Pin depot_tools version in DEPS to prevent breakages
by oprypin
· 7 years ago
adb58b8
Renable some Opus tests after Opus 1.2.1 update.
by minyue-webrtc
· 7 years ago
9c0914f
Do not crop DesktopFrame if the size won't change
by Zijie He
· 7 years ago
2059bb3
Adding Obj-C binding for RTCConfiguration::max_ipv6_networks.
by deadbeef
· 7 years ago
ecf3d53
Add histogram for FallbackDesktopCapturerWrapper and BlankDetectorDesktopCapturerWrapper
by Zijie He
· 7 years ago
d21eab3
Add "max_ipv6_networks" field to RTCConfiguration.
by deadbeef
· 7 years ago
3427f53
Relanding: Move "max IPv6 networks" logic to BasicPortAllocator, and fix sorting.
by deadbeef
· 7 years ago
58f1725
Add gn dependency between ana_debug_dump_proto and ana_config_proto.
by jbudorick
· 7 years ago
74544f9
Return translated position in MouseCursorMonitor
by Zijie He
· 7 years ago
54c7215
Fix issues with NetworkMonitor singleton when used by multiple clients.
by deadbeef
· 7 years ago
8e24556
Disable SeqNumUnwrapper death tests to avoid breaking downstream builds.
by philipel
· 7 years ago
7956c0f
Implemented a new sequence number unwrapper in sequence_number_util.h.
by philipel
· 7 years ago
8de1826
Reland "Allow AudioSendStream to reconfig AudioNetworkAdaptor"
by minyue-webrtc
· 7 years ago
edd6cec
Roll chromium_revision 4dc4bd62ba..1236993289 (489546:489596)
by buildbot
· 7 years ago
7df370b
Revert "Allow AudioSendStream to reconfig AudioNetworkAdaptor"
by Minyue Li
· 7 years ago
4a88120
Allow AudioSendStream to reconfig AudioNetworkAdaptor
by minyue-webrtc
· 7 years ago
abbc430
Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public.
by eladalon
· 7 years ago
22d162d
Roll chromium_revision 226fdf1148..4dc4bd62ba (489501:489546)
by buildbot
· 7 years ago
bc88531
Roll chromium_revision c92ec9bd3c..226fdf1148 (489455:489501)
by buildbot
· 7 years ago
b38f386
Update native plugin dll for turn servers and video.
by gyzhou
· 7 years ago
3b673c6
Removed file RTCCameraVideoCapturer.mm that isn't needed
by jtteh
· 7 years ago
b1c9d1d
Avoid that previous settings in APM are overwritten by WebRtcVoiceEngine
by peah
· 7 years ago
0748277
Roll chromium_revision f470dd6dfd..c92ec9bd3c (489395:489455)
by buildbot
· 7 years ago
d1d6c5a
Add jamiewalch to OWNERS.
by jamiewalch
· 7 years ago
dba4f94
Roll chromium_revision 88beb225b9..f470dd6dfd (489316:489395)
by buildbot
· 7 years ago
2f0803c
Roll chromium_revision fd8f995919..88beb225b9 (488572:489316)
by buildbot
· 7 years ago
61b0ed0
[iOS] Fix incorrectly oriented frames when rapidly switching between cameras.
by jtteh
· 7 years ago
96b69bd
Refactor composing report blocks for rtcp Sender/Receiver reports.
by danilchap
· 7 years ago
7fb11d7
Shrink critical-section scope in ReceiveStatisticsImpl::GetActiveStatisticians()
by eladalon
· 7 years ago
6209dcd
Add SetReportBlocks to rtcp Sender/Receive Report classes.
by danilchap
· 7 years ago
fb14312
Reland of Injectable Obj-C video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2980173002/ )
by kthelgason
· 7 years ago
8337727
Remove deprecated RtpRtcp::SetAudioPacketSize
by danilchap
· 7 years ago
e264a9e
Rename isolated_output to test_output and add a method to get the test_output directory.
by ilnik
· 7 years ago
d5e3d0f
Creating a more generic memcheck suppression
by mbonadei
· 7 years ago
42f44f9
Get rid of unnecessary cast of FlexfecReceiveStreamImpl to FlexfecReceiveStream
by eladalon
· 7 years ago
59cac99
Report minimum PSNR in VideoQualityTest and save corresponding frame to file
by ilnik
· 7 years ago
d3f3c34
Remove NullObjectReceiveStatistics() in rtp_rtcp module
by danilchap
· 7 years ago
a04d9c3
Remove RtpRtcp::RemoteRTCPStat(RTCPSenderInfo*) as unused
by danilchap
· 7 years ago
d0727bf
Fix NSInteger formatting warning from clang 6
by oprypin
· 7 years ago
1e64cfa
Fix autoroller in accordance to upstream change
by oprypin
· 7 years ago
ec390b5
When a track is added/removed directly to MediaStream notify observer->OnRenegotionNeeded
by korniltsev.anatoly
· 7 years ago
d083e85
Remove traces from {send,receive}_statistics_proxy.cc
by ehmaldonado
· 7 years ago
65e1f94
Throttle log message in FrameBuffer.
by philipel
· 7 years ago
c43d565
Remove setting configuration parameter to itself.
by danilchap
· 7 years ago
cc8b906
iOS AppRTCMobile: Close peerconnection when disconnecting
by magjed
· 7 years ago
e029d99
Integer overflow bug in low_cut_filter.
by Alex Loiko
· 7 years ago
fe43df1
Ignore NewApi Android Lint warning + Roll chromium_revision
by ehmaldonado
· 7 years ago
bf82021
Disable some Opus tests pending an update
by flim
· 7 years ago
f3a48ab
Delete unused field from AndroidVideoTrackSource
by korniltsev.anatoly
· 7 years ago
48e4d6d
Add zijiehe@chromium.org as OWNERS in WebRTC DesktopCapturer related logic
by Zijie He
· 7 years ago
cd66a77
Create new constructors and fields to support a better mouse cursor monitor
by Zijie He
· 7 years ago
817c8af
Revert of Move "max IPv6 networks" logic to BasicPortAllocator, and fix sorting. (patchset #2 id:20001 of https://codereview.webrtc.org/2983213002/ )
by deadbeef
· 7 years ago
d14d9f7
Use array declaration for extension URIs.
by Steve Anton
· 7 years ago
ad95614
Move "max IPv6 networks" logic to BasicPortAllocator, and fix sorting.
by deadbeef
· 7 years ago
a3251dd
Add parsing/serializing for MID RTP header extension.
by Steve Anton
· 7 years ago
3296256
Fixing lint issue
by henrika
· 7 years ago
cfccdae
Adds WebRtcAudioTrack.setAudioTrackUsageAttribute API
by henrika
· 7 years ago
e29117e
Modifies closing of AudioTrack resource on Android
by henrika
· 7 years ago
8ac955b
Set target API to 18 for MediaCodecUtils.
by sakal
· 7 years ago
cb79d23
Add common TLS extensions
by Emad Omara
· 7 years ago
3c45186
Move total audio energy and duration tracking to AudioLevel and protect with existing critial section.
by zstein
· 7 years ago
c4a5c14
Print general usage information for event_log_analyzer
by terelius
· 7 years ago
ad908f8
Roll chromium_revision 08a8c75946..1238950005 (488179:488218)
by buildbot
· 7 years ago
6d1dfa7
Roll chromium_revision d5f03e2816..08a8c75946 (488153:488179)
by buildbot
· 7 years ago
12d3084
Correct the calculation of discard rate.
by minyue-webrtc
· 7 years ago
5ed0904
Roll chromium_revision 40e1a3f583..d5f03e2816 (488128:488153)
by buildbot
· 7 years ago
6492338
Roll chromium_revision d23ce0ebaa..40e1a3f583 (488066:488128)
by buildbot
· 7 years ago
817ebf1
Roll chromium_revision 281eabf35e..d23ce0ebaa (488004:488066)
by buildbot
· 7 years ago
1777c5f
Move temporal-layer properties to FrameConfig.
by pbos
· 7 years ago
c40e1d3
Roll chromium_revision c86185cb26..281eabf35e (487874:488004)
by buildbot
· 7 years ago
398c3fd
Introduce RtpTransportInternal and SrtpTransport.
by zstein
· 7 years ago
f6a861a
Remove remains of webrtc/base
by ehmaldonado
· 7 years ago
3c3b110
Roll chromium_revision 8779e5365c..c86185cb26 (487834:487874)
by buildbot
· 7 years ago
e76f55e
Disable flaky NoBandwidthDropAfterDtx test.
by tschumim
· 7 years ago
a4c2117
Roll chromium_revision 6ada1228ca..8779e5365c (487809:487834)
by buildbot
· 7 years ago
2a06617
Roll chromium_revision fee38fe5d6..6ada1228ca (487773:487809)
by buildbot
· 7 years ago
9c0e0fa
Fix fromAndroidGraphicsMatrix to use column-major order for output.
by sakal
· 7 years ago
b4aa4eb
Replace WEBRTC_TRACE logging in modules/audio_device/.. mac/ win/
by saza
· 7 years ago
c58f8c0
Adds a histogram metric tracking for how long audio RTP packets are sent
by saza
· 7 years ago
e1d4dca
Roll chromium_revision b845da41ed..fee38fe5d6 (487740:487773)
by buildbot
· 7 years ago
dc7feb0
Roll chromium_revision 9b8266f849..b845da41ed (487695:487740)
by buildbot
· 7 years ago
4076576
Roll chromium_revision 0c6d79c9a3..9b8266f849 (487542:487695)
by buildbot
· 7 years ago
0cf9a4a
Add texture support to HardwareVideoEncoder.
by Bjorn Mellem
· 7 years ago
f80f344
Roll chromium_revision 712d1cf93f..0c6d79c9a3 (487487:487542)
by buildbot
· 7 years ago
8fb2361
Add texture support to HardwareVideoDecoder.
by Bjorn Mellem
· 7 years ago
153e204
[Webrtc] Reenable libc++ on ubsan
by thomasanderson
· 7 years ago
3e45cb5
Mapping screen id from DirectX capturer to GDI capturer
by Zijie He
· 7 years ago
8d2c235
Roll chromium_revision c09d9b5e5a..712d1cf93f (487450:487487)
by buildbot
· 7 years ago
f549dff
Fix gtest-parallel-wrapper
by ehmaldonado
· 7 years ago
80c829f
Enable tracing on rtcstats_integrationtest.cc
by ehmaldonado
· 7 years ago
b878575
Roll chromium_revision 03613afa1f..c09d9b5e5a (487428:487450)
by buildbot
· 7 years ago
b5c1607
UBSan fuzzer bug in LowCutFilter::BiqueadFilter::Process
by Alex Loiko
· 7 years ago
fcf97c3
Fix fullscreen scaling in AppRTCMobile.
by sakal
· 7 years ago
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