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8e6fd46cc324f8946e68396edcb252ffaf2d4579
8e6fd46
Route time-stretching metrics through libjingle
by Henrik Lundin
· 10 years ago
76cda01
Document the time unit in EventWrapper.
by Wan-Teh Chang
· 10 years ago
907bfb2
Fix an apparent typo in a unittest that caused it to not actually check the new window list it fetched.
by Peter Kasting
· 10 years ago
a831dc3
Convert native handles to buffers before encoding.
by Peter Boström
· 10 years ago
9ba52f8
Remove intermediate RTCP CNAME buffers.
by Peter Boström
· 10 years ago
aff1c84
Roll chromium_revision ccef3cb..7779e7d (331232:332119)
by Henrik Kjellander
· 10 years ago
5263b3c
Add options for NetEq fast accelerate mode through libjingle
by Henrik Lundin
· 10 years ago
0908d0d
Fix issue with RTT computations in simulator.
by Stefan Holmer
· 10 years ago
9b07368
Revert "Roll chromium_revision ccef3cb..7779e7d (331232:332119)"
by Henrik Kjellander
· 10 years ago
a8d686d
Roll chromium_revision ccef3cb..7779e7d (331232:332119)
by Henrik Kjellander
· 10 years ago
f69f1fb
Testing and improving NADA algorithm.
by Cesar Magalhaes
· 10 years ago
4765070
Rename I420VideoFrame to VideoFrame.
by Miguel Casas-Sanchez
· 10 years ago
c2cb266
Match video orientation with device orientation for portrait and portrait upside down
by Jon Hjelle
· 10 years ago
7be99bd
Revert "Match video orientation with device orientation for portrait and portrait upside down"
by Zeke Chin
· 10 years ago
14c2695
Match video orientation with device orientation for portrait and portrait upside down
by Jon Hjelle
· 10 years ago
bc7dd7e
Add RTCConfiguration constructor to RTCPeerConnection wrapper.
by Zeke Chin
· 10 years ago
d935f91
Don't try to parse empty Ice urls.
by Joachim Bauch
· 10 years ago
a8202aa
Roll chromium_revision 1b9c098..ccef3cb (330302:331232)
by Henrik Kjellander
· 10 years ago
5a8bad6
Update a comment that mentions the nonexistent Reset() method.
by Wan-Teh Chang
· 10 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
· 10 years ago
c28a896
VoE: Initialize WebRtcVoiceMediaChannel with AudioOptions during creation
by Jelena Marusic
· 10 years ago
bf738d7
Temporarily disabling OpenSL ES for playout.
by henrika
· 10 years ago
04e5b49
Make maximum SSL version configurable through PeerConnectionFactory::Options
by Joachim Bauch
· 10 years ago
cc84649
Add LappedTransform accessors.
by Michael Graczyk
· 10 years ago
e70028e
Protect access to shared list of SRTP sessions.
by Joachim Bauch
· 10 years ago
9e3cb33
AppRTCDemo: check for necessary permissions before starting the call.
by Alex Glaznev
· 10 years ago
770cc38
Don't call CRYPTO_add in BoringSSL.
by Jiayang Liu
· 10 years ago
3544837
Disable reusing of ECDHE keys with NSS.
by Joachim Bauch
· 10 years ago
5ee9f67
Remove webrtcvideoengine.cc.
by Peter Boström
· 10 years ago
603175a
Improve comments.
by Wan-Teh Chang
· 10 years ago
7c4e745
Support multiple URLs in PeerConnectionInterface::IceServer
by Joachim Bauch
· 10 years ago
45b229c
Remove an unnecessary webrtc:: namespace prefix.
by Wan-Teh Chang
· 10 years ago
92d9489
Miscellaneous cleanups in VCMReceiver and its unit tests.
by Wan-Teh Chang
· 10 years ago
645299d
Add frequency smoothing to postfilter.
by Andrew MacDonald
· 10 years ago
d4f769d
Stop video candidates getting down to audio.
by Donald Curtis
· 10 years ago
a743794
audio_processing/aecm: Create() now returns a pointer to the object
by Bjorn Volcker
· 10 years ago
71861a0
Remove GetSendSideDelay from RtpRtcp.
by Peter Boström
· 10 years ago
7cd16b0
video_processing_unittest: Only create files for visual inspection if the boolean flag 'gen_files' is set.
by Henrik Boström
· 10 years ago
c3deaa3
common_audio/vad: Removes head allocation failure check
by Bjorn Volcker
· 10 years ago
796e172
Fixes crash in WebRtcAudioManager.setCommunicationMode
by henrika
· 10 years ago
c41fe5d
Force 8 kHz sampling rate on Android emulator.
by Patrik Höglund
· 10 years ago
2251d6e
Remove ViESender.
by Peter Boström
· 10 years ago
259bd20
Report ssrc_groups in GetStats().
by Peter Boström
· 10 years ago
8bb6ea3
Reset speech encoder before hooking it up to RED or CNG
by Karl Wiberg
· 10 years ago
8051832
Adding a new Matlab tool rtpAnalyze
by Henrik Lundin
· 10 years ago
3b187b9
Removed unnecessary includes of webrtcvideocapturer.h
by Henrik Boström
· 10 years ago
11beccd
Remove external report blocks from RtcpSender and rtp_rtcp interface.
by Erik Språng
· 10 years ago
23c2e55
Remove remaining .mk files.
by Peter Boström
· 10 years ago
b444b3f
Redirect logs to stderr in audioproc_f.
by Andrew MacDonald
· 10 years ago
9b720f7
Add GetChunkLength to LappedTransform.
by Michael Graczyk
· 10 years ago
fec2c6d
Prevent potential double-free if srtp_create fails.
by Joachim Bauch
· 10 years ago
1060260
Added buildbucket bucket definitions
by Henrik Kjellander
· 10 years ago
92fbbb2
Switch acm_receiver over to using base/logging.h
by Tommi
· 10 years ago
9303eaf
Don't unnecessarily set mode/category on AVAudioSession.
by Noah Richards
· 10 years ago
def39883
Configure default render delay as 10 ms.
by Peter Boström
· 10 years ago
cf808d2
Add new fast mode for NetEq's Accelerate operation
by Henrik Lundin
· 10 years ago
cbe408a
WebRtcVideoCapturer: Getting rid of the |critical_section_stopping_| lock and all of its critical sections.
by Henrik Boström
· 10 years ago
c065cc7
Clarify boolean flags in neteq_opus_quality_test.
by Minyue Li
· 10 years ago
c13cacb
Remove an unused method in NetEq::Expand
by Henrik Lundin
· 10 years ago
de4703c
Refactor common_audio/vad: Create now returns the handle directly instead of an error code
by Bjorn Volcker
· 10 years ago
afef4bf
Reland "Adding a test framework for conference mode application in VoE."
by Minyue
· 10 years ago
a4b7e5e
Revert "Adding a test framework for conference mode application in VoE."
by Minyue
· 10 years ago
6a1ba8c
Fix coding style nits.
by Wan-Teh Chang
· 10 years ago
e87d487
Fix ARM64 detection for VP8 and VP9 wrappers.
by Stefan Holmer
· 10 years ago
fc05205
Adding a test framework for conference mode application in VoE.
by Minyue
· 10 years ago
5d55c98
WebRTC 4521: Remove usage of deprecated timezone global variable
by Guo-wei Shieh
· 10 years ago
8d3ad82
Script for auto-rolling chromium_revision in DEPS.
by Henrik Kjellander
· 10 years ago
5a3ebd7
Revert "Remove default encoder/decoders."
by Peter Boström
· 10 years ago
e14e5f4
Solve TSan warning about unlocking an unlocked mutex.
by Brave Yao
· 10 years ago
f09e09c
VoE: Remove unused interfaces
by Jelena Marusic
· 10 years ago
32c2023
Attempt at fixing error on the Chrome Windows FYI bots.
by Tommi
· 10 years ago
905495c
Introduce NetEq::Config::ToString and use it in NetEq's constructor
by Henrik Lundin
· 10 years ago
e982a70
PRESUBMIT: Fix typo.
by Henrik Kjellander
· 10 years ago
54be3e0
Remove some WebRtcVideoEngine2 unittest stubs.
by Peter Boström
· 10 years ago
d8399e6
Also provide sample rate when registering decoders
by Karl Wiberg
· 10 years ago
323b132
Protect ACM decoder buffer in stereo.
by Minyue
· 10 years ago
57e5fd2
PRESUBMIT: Improve PyLint check and add GN format check.
by Henrik Kjellander
· 10 years ago
00aac5a
Some cleanup for base/logging and base/stream.h
by Tommi
· 10 years ago
23edcff
Move base/logging.* to rtc_base_approved.
by Tommi
· 10 years ago
ee369e4
Refactoring of AudioTrackJni and AudioRecordJni using new JVM/JNI classes
by henrika
· 10 years ago
a26c4e5
Script to generate CL descriptions when rolling chromium_revision.
by Henrik Kjellander
· 10 years ago
0eefb4d
Detach base/logging.* from base/stream.*.
by Tommi
· 10 years ago
469c2c0
Make Config::default_value leak instead of having an exit-time destructor.
by Andrew MacDonald
· 10 years ago
4bf12ea
Revert "Fix sending wrong candidates down to transportchannel."
by Alejandro Luebs
· 10 years ago
f65de84
Fix sending wrong candidates down to transportchannel.
by Donald Curtis
· 10 years ago
67b635a
Fix simulcast_encoder_adapter giving full target_bitrate to the 2nd layer of any simulcast setup during InitEncode.
by Noah Richards
· 10 years ago
e4cb4e9
Fix jitter buffer bug around out-of-order packets and non-RTX padding.
by Noah Richards
· 10 years ago
4774874
Enable AudioProcessing48kHzSupport by default
by Alejandro Luebs
· 10 years ago
3548dd21
Set local SSRCs on receivers added before senders.
by Peter Boström
· 10 years ago
367c868
AudioEncoderCng: Handle case where speech encoder is reset
by Henrik Lundin
· 10 years ago
f761d10
Update NetEq Quality Test.
by Minyue Li
· 10 years ago
915df4f
CaptureManager: Don't stop a capturer at UnregisterVideoCapturer if it did not start in the first place.
by Henrik Boström
· 10 years ago
9a416bd
Get rid of unnecessary Terminate() method and worker_thread_ from WebRtcVideoEngine2
by Fredrik Solenberg
· 10 years ago
5af6d47
Code style change for quality_scaler.
by jackychen
· 10 years ago
98d8cf5
Hardware VP8 encoding: Use QP as metric for resize.
by jackychen
· 10 years ago
5fdcdf6
Enable ciphers to get ECDHE with NSS.
by Joachim Bauch
· 10 years ago
6f2ef74
Keep track of DTLS packet sizes to prevent partial reads.
by Joachim Bauch
· 10 years ago
a3ba0c7
RTPFragmentationHeader::VerifyAndAllocateFragmentationHeader: Verify that size fits in 16 bits
by Magnus Jedvert
· 10 years ago
36a1438
Remove ViEFrameProviderBase.
by Peter Boström
· 10 years ago
af55ccc
Add RtcpMuxPolicy support to PeerConnection.
by Peter Thatcher
· 10 years ago
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