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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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ac9f876bc06b8152ecc3a328147cdab19136361b
ac9f876
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
by kwiberg
· 8 years ago
fe69a74
Making ContinueSSL synchronously set the state to "open".
by Taylor Brandstetter
· 8 years ago
dd7fb43
Emit SignalReadyToSend even for "presumed writable" connections.
by deadbeef
· 8 years ago
89824f6
Relanding: Allow the DTLS fingerprint verification to occur after the handshake.
by deadbeef
· 8 years ago
3cdfcd8
Revert of Use sps and pps to determine decodability of H.264 frames. (patchset #4 id:60001 of https://codereview.webrtc.org/2341713002/ )
by stefan
· 8 years ago
61050f6
Fixig issues in BWE dynamics plot scripts.
by gaetano.carlucci
· 8 years ago
17b0263
Use sps and pps to determine decodability of H.264 frames.
by Stefan Holmer
· 8 years ago
55d932b
Add logging statements to places where the frame might be dropped in WebRTC pipeline.
by sakal
· 8 years ago
115bd15
New helper function test::ReadI420Buffer, refactor FrameReader to use it.
by nisse
· 8 years ago
6f112cc
Delete unused support for vp8 partitions.
by nisse
· 8 years ago
3cc47eb
Add sanity check for decreasing RTP timestamp in RtpToNtpMs.
by asapersson
· 8 years ago
f5297a0
Reland of Delete VideoFrameFactory, CapturedFrame, and related code. (patchset #1 id:1 of https://codereview.webrtc.org/2357113002/ )
by nisse
· 8 years ago
280de9e
Reland: Fix race / crash in OnNetworkRouteChanged().
by Stefan Holmer
· 8 years ago
20a52e1
Reland of Unify the macOS and iOS capturer implementations (patchset #1 id:1 of https://codereview.webrtc.org/2381853002/ )
by kthelgason
· 8 years ago
edbae5e
Remove Crit::Scope lock by using atomic bool property.
by denicija
· 8 years ago
eb5040a
Disable TCPChannelClientTest.testConnectIPv6
by ehmaldonado
· 8 years ago
15e4ec3
Remove compat for iOS 7/8
by Kári Tristan Helgason
· 8 years ago
3b703ed
Revert of Let ViEEncoder handle resolution changes. (patchset #17 id:340001 of https://codereview.webrtc.org/2351633002/ )
by perkj
· 8 years ago
b73d269
Replace RelayPort with TurnPort in p2ptransportchannel tests.
by Honghai Zhang
· 8 years ago
26105b4
Let ViEEncoder handle resolution changes.
by perkj
· 8 years ago
8b8459e
suppress memcheck test errors in the Opus decoder and encoder.
by honghaiz
· 8 years ago
75f6626
Revert of Replace RelayPort with TurnPort in p2ptransportchannel tests. (patchset #2 id:40001 of https://codereview.webrtc.org/2380923002/ )
by honghaiz
· 8 years ago
7851bda
Move RTCPHelp::RTCPReceiveInformation inside RTCPReceiver
by danilchap
· 8 years ago
c8d2171
Replace RelayPort with TurnPort in p2ptransportchannel tests.
by Honghai Zhang
· 8 years ago
7502401
Do not spam "Connect failed with 101/65" in logs.
by skvlad
· 8 years ago
591c709
Suppress a memcheck error in Opus decoder
by henrik.lundin
· 8 years ago
590cf28
Add autothread to pseudo-tcp fuzzer.
by phoglund
· 8 years ago
70736e4
Remove old presumably unused directory.
by sakal
· 8 years ago
8e6a761
ProbeController: Limit max probing bitrate
by isheriff
· 8 years ago
6060186
Add presubmit format requirement for webrtc/api/android
by magjed
· 8 years ago
5614566
Fix faulty include paths that break the build
by Henrik Lundin
· 8 years ago
5ec85fb
Revert of Fix race / crash in OnNetworkRouteChanged(). (patchset #5 id:80001 of https://codereview.webrtc.org/2366333003/ )
by stefan
· 8 years ago
b6760f9
Format all Java in WebRTC.
by sakal
· 8 years ago
a48ddb7
Add VideoSendStream::Stats::prefered_media_bitrate_bps
by Per
· 8 years ago
fd0d426
Fix race / crash in OnNetworkRouteChanged().
by stefan
· 8 years ago
eddb757
Revert of Unify the macOS and iOS capturer implementations (patchset #4 id:60001 of https://codereview.webrtc.org/2309253005/ )
by kthelgason
· 8 years ago
ff9793c
Android: Remove onOutputFormatRequest from the VideoCapturer interface
by magjed
· 8 years ago
90ce01d
The current default schedule delay of 30 ms prohibits
by isheriff
· 8 years ago
0fd22ef
Rename P2PTransportChannel worker_thread_ to network_thread_.
by johan
· 8 years ago
77eab70
Enable the -Wundef warning for clang
by kwiberg
· 8 years ago
51f2919
Update WebRTC to build against libsrtp 2.0
by mattdr
· 8 years ago
24c7c12
Move FunctionView from AudioCodingModule to the rtc namespace
by kwiberg
· 8 years ago
35d43b9
Roll chromium_revision bdaa23ddfe..316b880c55 (421490:421519)
by ehmaldonado
· 8 years ago
7e146cb
Fixing heap read overflow when "sctp-port" is in a video description.
by deadbeef
· 8 years ago
478681e
Move the QP scaling thresholds to the relevant encoders.
by kthelgason
· 8 years ago
e75f204
Expose Ivf logging through the native API
by palmkvist
· 8 years ago
242d8bd
Unify the macOS and iOS capturer implementations
by kthelgason
· 8 years ago
f5e3bbe
Roll chromium_revision 386676ff4e..bdaa23ddfe (421470:421490)
by buildbot
· 8 years ago
e5684c5
Delete method webrtc::VideoFrame::allocated_size and enum PlaneType.
by nisse
· 8 years ago
798896a
Replace RtcpReceiveTimeInfo with rtcp::ReceiveTimeInfo
by danilchap
· 8 years ago
9a8abcb
Roll chromium_revision dd442d4812..386676ff4e (421425:421470)
by ehmaldonado
· 8 years ago
8fea199
[GN] Add missing framework headers
by kthelgason
· 8 years ago
e0b2f15
Frame continuity is now tested as soon as a frame is inserted into the FrameBuffer.
by philipel
· 8 years ago
89a3a1a
Moved Gn target rtc_event_log to one directory above.
by charujain
· 8 years ago
b7446d7
GN: Fix incorrect include_dir for libjingle_peerconnection_jni target
by charujain
· 8 years ago
f363d14
Roll chromium_revision f86fb54ec3..dd442d4812 (420104:421425)
by Henrik Kjellander
· 8 years ago
0c9e567
Landmine to clobber on Android and Windows.
by kjellander
· 8 years ago
5e3b5d1
CQ: Remove GYP Release trybots since we now only run GYP.
by kjellander
· 8 years ago
e5e632f
Hooking up target audio bitrate to audio network adaptor.
by minyue
· 8 years ago
72bebf1
Roll chromium_revision cede888c27..f86fb54ec3 (419407:420104)
by buildbot
· 8 years ago
c3f549b
Update expected Xcode version to 8.0.
by kjellander
· 8 years ago
ee99696
Make 'webrtc' a static library.
by kjellander
· 8 years ago
822a16f
Reland of Unify rtcp packet setters (patchset #1 id:1 of https://codereview.webrtc.org/2372713005/ )
by danilchap
· 8 years ago
4151471
Add usage description strings to Info.plist
by Kári Tristan Helgason
· 8 years ago
efc6e41
Revert of Unify rtcp packet setters (patchset #8 id:130001 of https://codereview.webrtc.org/2348623003/ )
by kjellander
· 8 years ago
9532124
RTCPReceiver store cname as std::string. simplifying cname management.
by danilchap
· 8 years ago
f1363fd
Adds support for AVAudioSessionSilenceSecondaryAudioHintNotification on iOS
by henrika
· 8 years ago
46a8d18
ACM: Removed the code for InitialDelayManager
by ossu
· 8 years ago
29a44e3
This is a resubmission of https://codereview.webrtc.org/2047513002/
by kthelgason
· 8 years ago
5f8ebae
Add limitations of number of frames that can be created in I420BufferPool::CreateBuffer.
by perkj
· 8 years ago
c8299f9
Posting Opus's set-force-channels functionality to WebRTC.
by minyue
· 8 years ago
20e77c7
Unify rtcp packet setters
by danilchap
· 8 years ago
4ecd970
GN: Fix incorrect include_dir for video_coding on iOS
by kjellander
· 8 years ago
c1815cf
Reland of name AppRTCDemo on Android and iOS to AppRTCMobile (patchset #1 id:1 of https://codereview.webrtc.org/2358133003/ )
by Magnus Jedvert
· 8 years ago
0a52c70
THis CL enables possibility to select full-duplex OpenSL ES audio in AppRTCDemo, i.e., it adds support for OpenSL ES for input as well. The user must explicitly select this new mode in the debug UI hence it is not the default selection. There is no separate UI for input and output; instead both are enabled/disabled by the same switch.
by henrika
· 8 years ago
64ec8f8
Reland of Move MutableDataY{,U,V} methods to I420Buffer only. (patchset #1 id:1 of https://codereview.webrtc.org/2354223002/ )
by nisse
· 8 years ago
c637389
Delete unused file mock_audio_vector.h.
by nisse
· 8 years ago
de2920c
Delete unused file sessionid.h.
by nisse
· 8 years ago
89175a6
Trust that calls to RemoteEstimatorProxy::Process are done at the right frequency.
by stefan
· 8 years ago
25337bb
Android: Update clang-format to follow Google style guide
by magjed
· 8 years ago
fd8e33d
Removing a useless ctor in AudioNetworkAdaptorImpl.
by minyue
· 8 years ago
8af4fd0
Disabled flaky VideoSendStreamTest.ChangingNetworkRoute
by hbos
· 8 years ago
98088dc
header_usage.sh script: Exclude matches in gyp and gn files.
by nisse
· 8 years ago
660312b
Enable //build/config/clang:extra_warnings for rtc_media
by kjellander
· 8 years ago
464382d
Remove duplicated entry for bwe_simulations.cc
by kjellander
· 8 years ago
3901128
Remove unnecessary jsoncpp includes.
by kjellander
· 8 years ago
2068411
r14326 added '-Wno-unused-result' to 'WARNING_CFLAGS!' which removes the
by jianjun.zhu
· 8 years ago
c59bf04
Remove differ from ScreenCapturer implementations
by zijiehe
· 8 years ago
d3d230f
- Make RtpSenderAudio not inherit from DtmfQueue.
by solenberg
· 8 years ago
92ea601
Move class RTCPHelp::RTCPPacketInformation into RTCPReceiver
by danilchap
· 8 years ago
dda3666
Fixes minor issue in AudioDeviceTest.RunPlayoutAndRecordingInFullDuplex for iOS.
by henrika
· 8 years ago
f8c5f2b
Fix vie_encoder_unittest.cc.
by Per
· 8 years ago
44428a8
iOS: Always build H264 HW encoder/decoder
by magjed
· 8 years ago
512ecb3
Let ViEEncoder tell VideoSendStream about reconfigurations.
by Per
· 8 years ago
2a27b0a
Delete unused class FakeScreenCapturerFactory.
by nisse
· 8 years ago
347ec5c
Change thread check to race check. Also, add comment to explain implementation of RaceChecker.
by solenberg
· 8 years ago
f1b08da
Stopped using the NetEqDecoder enum internally in NetEq.
by ossu
· 8 years ago
1490f7a
Add histogram for end-to-end delay: "WebRTC.Video.EndToEndDelayInMs"
by asapersson
· 8 years ago
6d4c8c3
Renaming a proto target in GYP for audio network adaptor.
by minyue
· 8 years ago
e87d673
Return texture frame when dropping frames in CameraCapturer.
by sakal
· 8 years ago
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