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talk
9e083d2
Reland of Delete empty API files and cleaned up includes. (patchset #1 id:1 of https://codereview.webrtc.org/1813083002/ )
by perkj
· 9 years ago
eec21bd
Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.
by jbauch
· 9 years ago
194e3bc
Revert of Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. (patchset #4 id:60001 of https://codereview.webrtc.org/1785713005/ )
by kjellander
· 9 years ago
944c390
Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.
by jbauch
· 9 years ago
246b527
Revert of Delete empty API files and cleaned up includes. (patchset #2 id:20001 of https://codereview.webrtc.org/1809053002/ )
by deadbeef
· 9 years ago
c9022f5
Delete empty API files and cleaned up includes.
by perkj
· 9 years ago
505945a
Delete unused VideoCapturer statistics.
by Niels Möller
· 9 years ago
94a23f0
Reland "Add check_deps rules in DEPS files."
by kjellander@webrtc.org
· 9 years ago
d6c3954
Refactor VideoTracks to forward all sinks to its source
by perkj
· 9 years ago
8ad582d
Remove DeviceManager and DeviceInfo.
by solenberg
· 9 years ago
56cf60e
Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
by kjellander
· 9 years ago
086f851
Add check_deps rules in DEPS files.
by kjellander@webrtc.org
· 9 years ago
a1cf366
Handle iOS devices with no rear-facing camera
by hjon
· 9 years ago
84cc918
Replace scoped_ptr with unique_ptr in talk/
by kwiberg
· 9 years ago
2db1dbb
Remove references to build_with_libjingle and libjingle_java GYP variables.
by kjellander@webrtc.org
· 9 years ago
aac3eb2
Minor ObjC API tweaks.
by tkchin
· 9 years ago
a3ede6c
Renamed VideoSourceInterface to VideoTrackSourceInterface.
by perkj
· 9 years ago
6140fcc
Move RTCFileLogger to webrtc/base/objc.
by Jon Hjelle
· 9 years ago
9bf5cde
Update build_ios_libs.sh script to build new Objective-C API and gather header files.
by hjon
· 9 years ago
9b8df25
Move talk/session/media -> webrtc/pc
by kjellander@webrtc.org
· 9 years ago
5ad1297
Rename webrtc/media/webrtc -> webrtc/media/engine
by kjellander@webrtc.org
· 9 years ago
8fb3557
rtc::Buffer: Replace an internal rtc::scoped_ptr with std::unique_ptr
by kwiberg
· 9 years ago
162c339
Revert of Make cricket::VideoCapturer implement VideoSourceInterface (patchset #14 id:300001 of https://codereview.webrtc.org/1655793003/ )
by perkj
· 9 years ago
4d19c5b
This cl introduce a VideoSourceInterface and let cricket::VideoCapturer implement it.
by Per
· 9 years ago
4b2a5a8
Revert of Make cricket::VideoCapturer implement VideoSourceInterface (patchset #12 id:260001 of https://codereview.webrtc.org/1655793003/ )
by perkj
· 9 years ago
2f21789
This cl introduce a VideoSourceInterface and let cricket::VideoCapturer implement it.
by perkj
· 9 years ago
e2812e7
Cleanup after talk/media move.
by kjellander@webrtc.org
· 9 years ago
15583c1
Move talk/app/webrtc to webrtc/api
by Henrik Kjellander
· 9 years ago
dfb769d
Remove deprecated PeerConnectionObserver::OnStateChange and OnIceComplete
by perkj
· 9 years ago
47b6263
Remove Java PC support. This cl removes none Android Java support.
by perkj
· 9 years ago
f6b5509
Fix GYP and GN references that are invalid in Chromium builds.
by kjellander
· 9 years ago
fd6706a
Log Android HW decoder delay time statistics.
by glaznev
· 9 years ago
8e8908a
Delete FrameInput method and FrameInputWrapper class.
by nisse
· 9 years ago
a96e2d7
Move talk/media to webrtc/media
by kjellander
· 9 years ago
ae95ff3
Add more logging and fix PTS overflow for HW decoder.
by glaznev
· 9 years ago
20834ca
Adds a nullptr check to prevent a rare crash when starting or stopping an RtcEventLog.
by ivoc
· 9 years ago
ba4c0e4
Add send-side BWE to WebRtcVoiceEngine under a finch experiment.
by stefan
· 9 years ago
08582ff
Replace uses of cricket::VideoRenderer by rtc::VideoSinkInterface.
by nisse
· 9 years ago
8cb910d
Delete backwards compatibility cruft from cricket::VideoFrame and VideoSourceInterface.
by nisse
· 9 years ago
9031d63
Remove the network with empty name or NONE connection type from the network list.
by honghaiz
· 9 years ago
14d024d
Do not notify networkconnect if the connection type is known.
by Honghai Zhang
· 9 years ago
45b683f
Call static method getConnectionType using the class name.
by Honghai Zhang
· 9 years ago
cedff02
Remove dead code from WebRtcVideoEngine2.
by Peter Boström
· 9 years ago
e03ac51
Implement NullVideoDecoder to avoid crash on unsupported decoders.
by jbauch
· 9 years ago
1088001
Support multiple rtx codecs.
by Stefan Holmer
· 9 years ago
abe095b
Roll chromium_revision c6076f2..609aa24 (372974:373145)
by kjellander
· 9 years ago
7f77749
Disable flaky test WebRtcSessionTest.TestRtxRemovedByCreateAnswer on Win and Mac.
by honghaiz
· 9 years ago
27a3485
Fixing a DCHECK failure on unknown connection type from OS.
by honghaiz
· 9 years ago
a7ad7c3
Get the adapter type information from Android OS.
by honghaiz
· 9 years ago
ed3277b
Deprecate VideoDecoder::Reset() and remove calls.
by Peter Boström
· 9 years ago
ce23bee
Remove SendStreamFormat and ViewRequests.
by Peter Boström
· 9 years ago
9429148
Extra logging for HW codec.
by glaznev
· 9 years ago
a6c39d9
Remove unimplemented VideoChannel code.
by Peter Boström
· 9 years ago
eee86a6
Add option to disable particular HW video codec from app.
by Alex Glaznev
· 9 years ago
b163c3f
Delete unused members from VideoOptions
by nisse
· 9 years ago
378dc77
Consolidate setters into SetRecvParameters.
by pbos
· 9 years ago
46eed76
Removing "candidates" attribute from TransportDescription.
by deadbeef
· 9 years ago
6043f2e
Revert of Adding "first packet received" notification to PeerConnectionObserver. (patchset #5 id:80001 of https://codereview.webrtc.org/1581693006/ )
by terelius
· 9 years ago
e73afba
New rtc::VideoSinkInterface.
by nisse
· 9 years ago
bec70ab
https://github.com/w3c/webrtc-stats/pull/10/files added mediaType to the tracks. The closest in the current stats is the ssrc type.
by fippo
· 9 years ago
6a062bd
Deleted method AudioTrackInterface::GetRenderer.
by nisse
· 9 years ago
ab8f82f
Make ECDSA default for RTCPeerConnection
by tkchin
· 9 years ago
d162a5e
Add shouldDisableBuffering to RTCFileLogger.
by tkchin
· 9 years ago
919ff75
Use high QP threshold for HW VP8 encoder frame downscaling.
by glaznev
· 9 years ago
08a6eab
Adding "first packet received" notification to PeerConnectionObserver.
by Taylor Brandstetter
· 9 years ago
7b3c72f
Revert of Adding "first packet received" notification to PeerConnectionObserver. (patchset #4 id:60001 of https://codereview.webrtc.org/1581693006/ )
by deadbeef
· 9 years ago
42265a8
Adding "first packet received" notification to PeerConnectionObserver.
by Taylor Brandstetter
· 9 years ago
3afc8c4
Consolidate SetSendParameters into one setter.
by Peter Boström
· 9 years ago
ec2922f
Change PeerConnectionFactory.setVideoHwAccelerationOptions to create shared Egl context for harware encoders and decoders.
by Per
· 9 years ago
2098fca
Revert of New rtc::VideoSinkInterface. (patchset #7 id:120001 of https://codereview.webrtc.org/1594973006/ )
by nisse
· 9 years ago
a862d45
New rtc::VideoSinkInterface.
by Niels Möller
· 9 years ago
b11e97a
Move talk/media/webrtc/OWNERS to talk/media.
by Peter Boström
· 9 years ago
bab934b
H.264 video codec support using OpenH264 (http://www.openh264.org/) for encoding and FFmpeg (https://www.ffmpeg.org/) for decoding.
by hbos
· 9 years ago
3ea1852
Sync build_ios_libs.sh script with http://webrtc.org/native-code/ios/
by hjon
· 9 years ago
4cb3e39
Fix compilation if HAVE_WEBRTC_VIDEO is not defined.
by jbauch
· 9 years ago
5ad935c
Remove mutable from rtc::CriticalSection members.
by pbos
· 9 years ago
9de632a
Deleted unused enums MediaChannelOptions and VoiceMediaChannelOptions,
by nisse
· 9 years ago
0a37497
Deleted unused method SetDumpPath and unneeded includes.
by nisse
· 9 years ago
c8930ba
Disable WebRtcSessionTest.TestStunError on Win.
by minyue
· 9 years ago
8947a01
Fixing an uninitialized variable in webrtcsession_unittest.
by deadbeef
· 9 years ago
3c16978
Remove cast to LocalAudioSource from AudioRtpSender.
by Tommi
· 9 years ago
0b98cf7
Delete CaptureRenderAdapter::VideoRenderInfo struct, it is unused since the recent deletion of SetSize.
by nisse
· 9 years ago
5082c83
Make type and constructors in EglBase14 public.
by noahric
· 9 years ago
d26fadb
Delete GetRenderer method, used only by the tests.
by nisse
· 9 years ago
057ecf0
Making WebRtcSession fire a destroyed signal.
by deadbeef
· 9 years ago
1d61a51
Send key frame if time difference between incoming frames exceeds a certain limit.
by asapersson
· 9 years ago
8a2c31d
Make it possible to run peerconnection_unittests on Android.
by perkj
· 9 years ago
c4c8485
Deleted renderer-related SetSize methods, and all uses.
by nisse
· 9 years ago
81354f5
Added mute logic to VideoTrackRenderers.
by nisse
· 9 years ago
f5a3a93
Add 5-argument wrapper WebRtcVideoFrame::InitToBlack
by Niels Möller
· 9 years ago
8b1e431
Delete remnants of non-square pixel support from cricket::VideoFrame.
by nisse
· 9 years ago
cec0a08
Add a new interface for creating a udp socket in which it binds the socket to a network if the network handle is set.
by honghaiz
· 9 years ago
f4decb5
Add QP statistics logging to Android HW encoder.
by glaznev
· 9 years ago
884f585
Storing raw audio sink for default audio track.
by deadbeef
· 9 years ago
79a5a83
Adapt to boringssl's new defaults.
by torbjorng
· 9 years ago
d66b44d
Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
by ivoc
· 9 years ago
0f7d293
Revert changes to default option setting in https://codereview.webrtc.org/1500633002/
by solenberg
· 9 years ago
dc305db
Add ApplyPacketOptions()
by Sergey Ulanov
· 9 years ago
20ac434
Fix a test bot failure.
by Honghai Zhang
· 9 years ago
e1f9d83
Adding AddTrack/RemoveTrack to native PeerConnection API.
by deadbeef
· 9 years ago
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