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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
5263b3c1ddb10ecca58d9f08364aad2d6ba1d95d
/
talk
5263b3c
Add options for NetEq fast accelerate mode through libjingle
by Henrik Lundin
· 9 years ago
4765070
Rename I420VideoFrame to VideoFrame.
by Miguel Casas-Sanchez
· 9 years ago
c2cb266
Match video orientation with device orientation for portrait and portrait upside down
by Jon Hjelle
· 9 years ago
7be99bd
Revert "Match video orientation with device orientation for portrait and portrait upside down"
by Zeke Chin
· 9 years ago
14c2695
Match video orientation with device orientation for portrait and portrait upside down
by Jon Hjelle
· 9 years ago
bc7dd7e
Add RTCConfiguration constructor to RTCPeerConnection wrapper.
by Zeke Chin
· 9 years ago
d935f91
Don't try to parse empty Ice urls.
by Joachim Bauch
· 9 years ago
a8202aa
Roll chromium_revision 1b9c098..ccef3cb (330302:331232)
by Henrik Kjellander
· 9 years ago
5c6c6e0
Implements TODOs for webrtc::datachannel state management when the SCTP association is congested. Adds missing state variables for each step in the transitions between DataChannelInterface::DataStates (kConnecting, kOpen, etc.), and uses them.
by Lally Singh
· 9 years ago
c28a896
VoE: Initialize WebRtcVoiceMediaChannel with AudioOptions during creation
by Jelena Marusic
· 9 years ago
04e5b49
Make maximum SSL version configurable through PeerConnectionFactory::Options
by Joachim Bauch
· 9 years ago
e70028e
Protect access to shared list of SRTP sessions.
by Joachim Bauch
· 9 years ago
9e3cb33
AppRTCDemo: check for necessary permissions before starting the call.
by Alex Glaznev
· 9 years ago
5ee9f67
Remove webrtcvideoengine.cc.
by Peter Boström
· 9 years ago
7c4e745
Support multiple URLs in PeerConnectionInterface::IceServer
by Joachim Bauch
· 9 years ago
d4f769d
Stop video candidates getting down to audio.
by Donald Curtis
· 9 years ago
259bd20
Report ssrc_groups in GetStats().
by Peter Boström
· 9 years ago
3b187b9
Removed unnecessary includes of webrtcvideocapturer.h
by Henrik Boström
· 9 years ago
23c2e55
Remove remaining .mk files.
by Peter Boström
· 9 years ago
fec2c6d
Prevent potential double-free if srtp_create fails.
by Joachim Bauch
· 9 years ago
cbe408a
WebRtcVideoCapturer: Getting rid of the |critical_section_stopping_| lock and all of its critical sections.
by Henrik Boström
· 9 years ago
f09e09c
VoE: Remove unused interfaces
by Jelena Marusic
· 9 years ago
54be3e0
Remove some WebRtcVideoEngine2 unittest stubs.
by Peter Boström
· 9 years ago
0eefb4d
Detach base/logging.* from base/stream.*.
by Tommi
· 9 years ago
469c2c0
Make Config::default_value leak instead of having an exit-time destructor.
by Andrew MacDonald
· 9 years ago
4bf12ea
Revert "Fix sending wrong candidates down to transportchannel."
by Alejandro Luebs
· 9 years ago
f65de84
Fix sending wrong candidates down to transportchannel.
by Donald Curtis
· 9 years ago
3548dd2
Set local SSRCs on receivers added before senders.
by Peter Boström
· 9 years ago
915df4f
CaptureManager: Don't stop a capturer at UnregisterVideoCapturer if it did not start in the first place.
by Henrik Boström
· 9 years ago
9a416bd
Get rid of unnecessary Terminate() method and worker_thread_ from WebRtcVideoEngine2
by Fredrik Solenberg
· 9 years ago
98d8cf5
Hardware VP8 encoding: Use QP as metric for resize.
by jackychen
· 9 years ago
af55ccc
Add RtcpMuxPolicy support to PeerConnection.
by Peter Thatcher
· 9 years ago
76b62ff
Clean up now-unused code that was used for libpeerconnection.[so|dll].
by Tommi
· 9 years ago
fce3242
Remove linphonemediaengine.*
by Fredrik Solenberg
· 9 years ago
c3f4dbc
Remove rtp_rtcp/ dump functionality.
by Peter Boström
· 9 years ago
831c558
Allow setting maximum protocol version for SSL stream adapters.
by Joachim Bauch
· 9 years ago
4d71ede
Add HW fallback option to software encoding.
by Peter Boström
· 9 years ago
17b889b
Issue 4366: Adapted frames have wrong width and height and are cropped.
by Guo-wei Shieh
· 9 years ago
2f5be9a
Improve Android camera error handling.
by Alex Glaznev
· 9 years ago
ccb49e7
Remove Soundclip handling from libjingle.
by Fredrik Solenberg
· 9 years ago
b92be45
Support 720P in portait as maximum on iOS.
by Weiyong Yao
· 10 years ago
2e7a098
Ensure mediasession generated offers with RTX contain an RTX ssrc for each video ssrc.
by Noah Richards
· 10 years ago
7252a2b
Add HW fallback option to software decoding.
by Peter Boström
· 10 years ago
b261989
Adding support for OpenSL ES output in native WebRTC
by henrika
· 10 years ago
144d018
fix indent on tokenize_first function signatures
by Donald Curtis
· 10 years ago
0e07f92
Split fmtp on semicolons not spaces as per RFC6871
by Donald Curtis
· 10 years ago
4cd6940
Enable -Wformat-security warning and cleanup GYP.
by Henrik Kjellander
· 10 years ago
39f2b0c
Implemented video device info for iOS
by Yuriy Shevchuk
· 10 years ago
2013aec
Propagating RTT from send-only channel to receive-only channel.
by Minyue
· 10 years ago
300eeb6
Remove VideoEngine interfaces.
by Peter Boström
· 10 years ago
67c9df7
Base NACK on send codecs.
by Peter Boström
· 10 years ago
126c03e
Base decision to send REMB on send codecs.
by Peter Boström
· 10 years ago
64dad83
Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
by Henrik Lundin
· 10 years ago
1f62923
Revert r9164 "Adding a new constraint to set NetEq buffer capacity ..."
by Henrik Lundin
· 10 years ago
fd32f35
Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
by Henrik Lundin
· 10 years ago
4c277bb
Add basic SCTP packet logging.
by Lally Singh
· 10 years ago
cdb47a4
Revert r9159 "Adding a new constraint to set NetEq buffer capacity ..."
by Henrik Lundin
· 10 years ago
208a229
Adding a new constraint to set NetEq buffer capacity from peerconnection
by Henrik Lundin
· 10 years ago
d3ddc1b
Consistently use DCHECK, not ASSERT or assert in talk/media/webrtc/.
by Fredrik Solenberg
· 10 years ago
e444a3d
WebRtcVoiceEngine: Get rid of unnecessary template base class.
by Fredrik Solenberg
· 10 years ago
aaf8ff2
WebRtcVoiceEngine: virtual to override + git cl format.
by Fredrik Solenberg
· 10 years ago
6179b89
Remove unused API on WebRtcVoiceEngine.
by Fredrik Solenberg
· 10 years ago
4b60c73
Hook up libjingle WebRtcVoiceEngine to Call API for combined A/V BWE.
by Fredrik Solenberg
· 10 years ago
81ea54e
Remove WebRtcVideoEngine.
by Peter Boström
· 10 years ago
ccfc939
Reinterpret AudioOption delay_agnostic_aec to override HW-AEC
by Bjorn Volcker
· 10 years ago
57cc74e
iOS camera switching video capturer.
by Zeke Chin
· 10 years ago
c56ac1e
rtc::Buffer: Remove backwards compatibility band-aids
by Karl Wiberg
· 10 years ago
e433c0e
Restore back verbosity logging for camera captured frame.
by Alex Glaznev
· 10 years ago
cac1b38
Expose RTCConfiguration to java JNI and add an option to disable TCP
by Jiayang Liu
· 10 years ago
4eddf18
Don't crash if SetRemoteDescription is called first with BundlePolicy=max-bundle.
by Peter Thatcher
· 10 years ago
cbf0927
Revert "rtc::Buffer: Remove backwards compatibility band-aids"
by Karl Wiberg
· 10 years ago
9e1a6d7
rtc::Buffer: Remove backwards compatibility band-aids
by Karl Wiberg
· 10 years ago
f16fcbe
Remove ViECapture usage in VideoSendStream.
by Peter Boström
· 10 years ago
efbde37
Don't use CPU adaptation for screen content in the new API.
by Erik Språng
· 10 years ago
adf89b7
Added SetBitRate function to VoE API to allow changing the audio bitrate.
by Ivo Creusen
· 10 years ago
23fba1f
Add AudioReceiveStream to Call API.
by Fredrik Solenberg
· 10 years ago
10ba3ee
Roll chromium_revision a12e1e1..0cb2549 (326495:327252)
by Henrik Kjellander
· 10 years ago
94cc1fe
Remove ViEImageProcess usage in VideoSendStream.
by Peter Boström
· 10 years ago
1ba344a
Adds a MediaConstraint for the AudioOption aec_dump
by Bjorn Volcker
· 10 years ago
faa6d07
Remove a few verbose log messages from webrtcvideoengine2.
by Alex Glaznev
· 10 years ago
019087f
Add safeguards against signalling peer-reflexive candidates.
by Peter Thatcher
· 10 years ago
143cec1
Set correct encoder-specific settings for vpx in the new API.
by Erik Språng
· 10 years ago
e6cefb6
GYP variables for building expat, icu, libsrtp, usrsctp
by Henrik Kjellander
· 10 years ago
77d444a
Handle the case when hoststring is empty.
by Tommi
· 10 years ago
c4188fd
Use IncomingVideoStream in VideoReceiveStream.
by Peter Boström
· 10 years ago
24d4485
Enable -Wunused-private-field warning for talk/
by Henrik Kjellander
· 10 years ago
3525954
Use short include paths for icu headers.
by Henrik Kjellander
· 10 years ago
ee0b00e
Prevent recv-stream reconfig on identical codecs.
by Peter Boström
· 10 years ago
908e77b
Allow Java code to detect if VP8 and H.264 HW decoding is supported.
by Alex Glaznev
· 10 years ago
b672882
Move cricket::FakeCall and associates to a separate file.
by Fredrik Solenberg
· 10 years ago
7fb711f
Remove unused voice channel argument from cricket::VideoChannel ctor and corresponding field in class.
by Fredrik Solenberg
· 10 years ago
393347f
Report receive-side packet loss.
by Peter Boström
· 10 years ago
7c027b6
Enable more Clang warnings for talk/
by Henrik Kjellander
· 10 years ago
61b4d51
Dynamic resolution change for VP8 HW encode.
by jackychen
· 10 years ago
e62202f
Support handling multiple RTX but only generate SDP with RTX associated with VP8.
by Shao Changbin
· 10 years ago
c4905fb
Fix race condition in Android camera JNI code.
by Alex Glaznev
· 10 years ago
ac7d97f
Remove frame copy in RTCOpenGLVideoRenderer.
by Zeke Chin
· 10 years ago
8c05415
Add extra logging for Android camera JNI layer.
by Alex Glaznev
· 10 years ago
9478437
rtc::Buffer improvements
by Karl Wiberg
· 10 years ago
9154373
Do not define POSIX.
by Thiago Farina
· 10 years ago
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