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3a0a0f4b0dae11b20ebacdbd79d6c7f94edeac7b
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talk
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app
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webrtc
e9021a3
Propogate network-worker thread split to api
by danilchap
· 9 years ago
bfefb03
Replace scoped_ptr with unique_ptr everywhere
by kwiberg
· 9 years ago
0565451
Reland of Delete cricket::VideoFrame methods GetYPlane and GetYPitch. (patchset #1 id:1 of https://codereview.webrtc.org/1921493004/ )
by nisse
· 9 years ago
b99395a
Reland of Delete video_render module. (patchset #1 id:1 of https://codereview.webrtc.org/1923613003/ )
by nisse
· 9 years ago
0190367
Revert of Delete video_render module. (patchset #12 id:220001 of https://codereview.webrtc.org/1912143002/ )
by kjellander
· 9 years ago
97cfd1e
Delete video_render module.
by nisse
· 9 years ago
9eeb624
Build dynamic iOS SDK.
by tkchin
· 9 years ago
555604a
Replace scoped_ptr with unique_ptr in webrtc/base/
by jbauch
· 9 years ago
b05f994
Revert of Delete cricket::VideoFrame methods GetYPlane and GetYPitch. (patchset #5 id:80001 of https://codereview.webrtc.org/1901973002/ )
by terelius
· 9 years ago
1c27c6b
Delete cricket::VideoFrame methods GetYPlane and GetYPitch.
by nisse
· 9 years ago
001c20d
Move logic of gyp_webrtc into gyp_webrtc.py
by kjellander@webrtc.org
· 9 years ago
2ded9b1
Replace SetCapturer and SetCaptureDevice by SetSource. Drop return value.
by nisse
· 9 years ago
71a0c2f
Deprecate GetWidth() and GetHeight() methods. Replaced by width() and height().
by nisse
· 9 years ago
60083c8
Delete unused cricket::VideoFrame methods MakeExclusive and CopyToFrame.
by nisse
· 9 years ago
89717aa
Improve iOS frame capture threading.
by tkchin
· 9 years ago
d61bf80
Removed MediaStreamTrackInterface::set_state
by perkj
· 9 years ago
8f59762
Delete VideoRendererInterface.
by Niels Möller
· 9 years ago
c8f952d
Propagate MediaStreamSource state to video tracks the same way as audio.
by perkj
· 9 years ago
bc73fe1
Move build scripts to webrtc/build/ios
by hjon
· 9 years ago
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
a3ede6c
Renamed VideoSourceInterface to VideoTrackSourceInterface.
by perkj
· 9 years ago
6140fcc
Move RTCFileLogger to webrtc/base/objc.
by Jon Hjelle
· 9 years ago
9b8df25
Move talk/session/media -> webrtc/pc
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
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
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
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
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
9429148
Extra logging for HW codec.
by glaznev
· 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
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
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
5ad935c
Remove mutable from rtc::CriticalSection members.
by pbos
· 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
5082c83
Make type and constructors in EglBase14 public.
by noahric
· 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
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
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
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
67b1e1a
Put options as the argument of the java PeerConnectionFactory constructor.
by honghaiz
· 9 years ago
3542013
Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ )
by sprang
· 9 years ago
31c8d2e
Update with new default boringssl no-aes cipher suites. Re-enable tests.
by Torbjorn Granlund
· 9 years ago
268493a
Revert of Delete remnants of non-square pixel support from cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ )
by nisse
· 9 years ago
709513d
Delete remnants of non-square pixel support from cricket::VideoFrame.
by nisse
· 9 years ago
2d110be
Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
by deadbeef
· 9 years ago
e591f93
Storing raw audio sink for default audio track.
by deadbeef
· 9 years ago
6955870
Convert channel counts to size_t.
by Peter Kasting
· 9 years ago
3e1cfa7
Delete unused method webrtc::VideoRendererInterface::SetSize.
by nisse
· 9 years ago
127782b
Add default dummy implementation of cricket::VideoRenderer::SetSize, to easy later removal.
by nisse
· 9 years ago
a7446d2
Change DTLS default from 1.0 to 1.2 for webrtc.
by Guo-wei Shieh
· 9 years ago
f475d365
Properly handle different transports having different SSL roles.
by Taylor Brandstetter
· 9 years ago
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