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6167b2621fcd1790e70c2306906661e520309dee
6167b26
Make RtpTransportControllerSend::send_side_cc_ a direct member.
by nisse
· 8 years ago
cde46b7
Resolve cyclic dependency between audio network adaptor and event log api
by michaelt
· 8 years ago
28dc285
Adding cbr support for Opus
by soren
· 8 years ago
388fe42
Make WARN_UNUSED_RESULT a no-op on gcc
by kwiberg
· 8 years ago
177b17e
Move AndroidVideoTrackSourceObserver from API to src
by magjed
· 8 years ago
639d46a
Delete system_wrappers logging facility.
by nisse
· 8 years ago
be77920
Revert of CQ: Remove Linux ARM64 Debug trybot from default set. (patchset #1 id:1 of https://codereview.webrtc.org/2790263003/ )
by kjellander
· 8 years ago
2418001
ACM: Change test output files from PCM to WAV
by henrik.lundin
· 8 years ago
4fcfdd8
Enable rtc_unittests on iOS simulator
by kjellander
· 8 years ago
a280f7c
Added integer parsing functions in base/string_to_number.h
by ossu
· 8 years ago
b1e3fc4
Enable tools_unittests and rtc_stats_unittests on iOS Simulator
by kjellander
· 8 years ago
e24991d
Adds AudioDeviceTest.RunPlayoutAndRecordingInFullDuplex unittest.
by henrika
· 8 years ago
978504e
Move rtp header extension length check from Packet::FindExtension to ExtensionT::Parse
by danilchap
· 8 years ago
ed6343d
Roll chromium_revision 875e8893e9..3014f8b41e (462360:462374)
by buildbot
· 8 years ago
251eb27
Roll chromium_revision 75820eb165..875e8893e9 (460410:462360)
by kjellander
· 8 years ago
f6a4f37
Reland of Fixed error for missing explict class initialization error on iOS WebRTC buildbots (patchset #1 id:1 of https://codereview.webrtc.org/2803933002/ )
by guidou
· 8 years ago
854e507
Revert of Fixed error for missing explict class initialization error on iOS WebRTC buildbots (patchset #1 id:1 of https://codereview.webrtc.org/2799813002/ )
by guidou
· 8 years ago
5ac18af
Fixed error for missing explicit class initialization error on iOS buildbots
by peah
· 8 years ago
7343c8e
DirectX capturer may crash after switching shared screen
by zijiehe
· 8 years ago
cf02cf1
Major AEC3 render pipeline changes
by peah
· 8 years ago
4aceaf2
Android: Move Histogram from api to src.
by sakal
· 8 years ago
c522e75
Use new RTCCameraVideoCapturer in AppRTCMobile.
by sakal
· 8 years ago
1ba21eb
Add [c]begin() and [c]end() member functions to rtc::Buffer
by kwiberg
· 8 years ago
dea682d
This CL fixes the following:
by alessiob
· 8 years ago
129fc9c
Enabling 'gn check' on //webrtc/tools.
by mbonadei
· 8 years ago
c88b5d5
Reland of PyLint fixes for tools-webrtc and webrtc/tools (patchset #1 id:1 of https://codereview.webrtc.org/2737233003/ )
by kjellander
· 8 years ago
adf0635
Make GetConfig() part of the AudioProcessing interface
by henrik.lundin
· 8 years ago
368f5cf
Replace use of system_wrappers/include/logging.h by base/logging.h.
by nisse
· 8 years ago
2299b0ab
Android: Remove VideoCapturerAndroid
by magjed
· 8 years ago
d60d06a
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
by ilnik
· 8 years ago
d8ce1e1
Move SelectMediaType from RampUpTester to BaseTest.
by nisse
· 8 years ago
6eca98b
Add histogram stats for number of cpu/quality adapt changes per minute for sent video streams:
by asapersson
· 8 years ago
d48dbda
Add a minimal RtpTransport class for use by BaseChannel.
by zstein
· 8 years ago
465faf0
[iOS] Changed ptr to const ref for RTCConfiguration initialization
by jtteh
· 8 years ago
ba13131
Trivial data() and mutable_data() implementations
by yujo
· 8 years ago
baf9b58
README.md with deps, build, usage, troubleshooting
by alessiob
· 8 years ago
c337258
Revert of Deliver video frames on Android, on the decode thread. (patchset #7 id:120001 of https://codereview.webrtc.org/2764573002/ )
by guidou
· 8 years ago
fbd4f85
Javascript audio player for the exported HTML file.
by alessiob
· 8 years ago
fab6707
Add number of quality adapt changes to VideoSendStream stats.
by asapersson
· 8 years ago
e3aa88b
Deliver video frames on Android, on the decode thread.
by tommi
· 8 years ago
aa7d935
Evaluation scores export library and CSS file for the exported HTML file
by alessiob
· 8 years ago
8edb839
Reland of Export library that generates an HTLM file with the scores organized in tables. (patchset #1 id:1 of https://codereview.webrtc.org/2791293002/ )
by alessiob
· 8 years ago
fab482b
Simplify RTPSender::RegisterRtpHeaderExtension
by danilchap
· 8 years ago
55a0135
Make sure we observe enough frames before scaling.
by kthelgason
· 8 years ago
751c9dc
Roll chromium_revision 581ff14023..75820eb165 (459789:460410)
by buildbot
· 8 years ago
5115645
CQ: Remove Linux ARM64 Debug trybot from default set.
by Henrik Kjellander
· 8 years ago
3153c6d
Fixing DTX unittest for AudioEncoderOpus.
by minyue
· 8 years ago
4271224
CQ: Remove linux32_rel from default trybots
by Henrik Kjellander
· 8 years ago
0786c04
Fix crash in XServerPixelBuffer.
by sergeyu
· 8 years ago
4eeb537
[iOS] Added an initialization method to RTCConfiguration that takes a
by jtteh
· 8 years ago
9ab17d3
Forward capturer_id to shared desktopframe
by zijiehe
· 8 years ago
c964d0b
Fixing some case-sensitive codec name comparisons.
by deadbeef
· 8 years ago
716d7ac
Revert of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #8 id:140001 of https://codereview.webrtc.org/2780943003/ )
by guidou
· 8 years ago
880c85b
Revert of Export library that generates an HTLM file with the scores organized in tables. (patchset #3 id:40001 of https://codereview.webrtc.org/2717973006/ )
by alessiob
· 8 years ago
29e3330
Export library that generates an HTLM file with the scores organized in tables.
by alessiob
· 8 years ago
c42f540
Move video_encoder.h and video_decoder.h to /api and create GN targets for them
by ilnik
· 8 years ago
eb4662a
Single simulation runner.
by alessiob
· 8 years ago
0deb594
POLQA evaluation score.
by alessiob
· 8 years ago
c533df2
Audio level evaluation score.
by alessiob
· 8 years ago
54ad3df
I added two factory classes to address an important comment I got in another CL from kjellander@webrtc.org.
by alessiob
· 8 years ago
8a1b3c9
Environmental noise generator implemented.
by alessiob
· 8 years ago
653063f
Add functions to get/set rtp header extension by id.
by danilchap
· 8 years ago
23425f9
Add methods to register congestion controller observer after construction.
by nisse
· 8 years ago
d197cd9
Revert of Add empty header to fix internal project. (patchset #1 id:1 of https://codereview.webrtc.org/2790493006/ )
by kthelgason
· 8 years ago
d1b0e0e
Merge UpdateBandwidthEstimate with Update in AimdRateControl.
by terelius
· 8 years ago
97a7fb0
Delete obsolete file mock_congestion_controller.h.
by nisse
· 8 years ago
52a0ce6
Delete unused method OnProbeBitrate.
by nisse
· 8 years ago
3a407ee
Making RtpSender tests cover BWE with overhead.
by minyue
· 8 years ago
c5d62e2
Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2783183003/ )
by sprang
· 8 years ago
76d9c9c
Reland of Enable trendline experiment and bayesian bitrate estimator experiment by default.
by stefan
· 8 years ago
bc436ed
Revert of Supporting 48kHz PCM file. (patchset #1 id:1 of https://codereview.webrtc.org/2790493004/ )
by lliuu
· 8 years ago
029f7cc
Revert of Enable trendline experiment and bayesian bitrate estimator experiment by default. (patchset #6 id:100001 of https://codereview.webrtc.org/2777333003/ )
by lliuu
· 8 years ago
8a8b238
ScreenCapturerWinDirectx should have two DxgiDuplicatorController::Context instances, each for one DesktopFrame. So both DesktopFrame instances can be correctly updated.
by zijiehe
· 8 years ago
5f93709
Supporting 48kHz PCM file.
by minyue
· 8 years ago
16d5bae
Add empty header to fix internal project.
by kthelgason
· 8 years ago
27925de
Enable trendline experiment and bayesian bitrate estimator experiment by default.
by stefan
· 8 years ago
9c79ed9
Add loss-based BWE experiment which allows us to try different parameters.
by Stefan Holmer
· 8 years ago
fdbfdc9
Let PacketRouter separate send and receive modules.
by nisse
· 8 years ago
ec6fbd2
Moves channel-dependent audio input processing to separate encoder task queue.
by henrika
· 8 years ago
36e9eb4
Do not report quality limited resolution stats when quality scaler is disabled.
by asapersson
· 8 years ago
3883ccb
New RTCCameraVideoCapturer.
by sakal
· 8 years ago
ee8b861
remove module-wide warning suppression.
by kthelgason
· 8 years ago
e6a8009
Remove voe_auto_test cases for VoEFile.
by solenberg
· 8 years ago
d00aad5
Revert of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #16 id:300001 of https://codereview.webrtc.org/2747863003/ )
by mbonadei
· 8 years ago
124a6fc
MacOS: Add metal renderer and view.
by denicija
· 8 years ago
515dff4
Revert of Adding PRESUBMIT check on google::protobuf (patchset #2 id:20001 of https://codereview.webrtc.org/2753823003/ )
by mbonadei
· 8 years ago
ff046c7
Remove ALL usage of CriticalSectionWrapper.
by kthelgason
· 8 years ago
5533bd3
Reland: Use native (optimized) functions for byte order conversion. (patchset #1 id:1 of https://codereview.webrtc.org/2755103002/ )
by jbauch
· 8 years ago
65a8308
Another landmine for low_bandwidth_audio_test
by Henrik Kjellander
· 8 years ago
f9ed235
Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ )
by lliuu
· 8 years ago
7a3615b
Revert of Enable the bayesian bitrate estimator by default. (patchset #5 id:80001 of https://codereview.webrtc.org/2749803002/ )
by lliuu
· 8 years ago
c53a17f
Enable the bayesian bitrate estimator by default.
by stefan
· 8 years ago
3ea3c77
Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2764133002/ )
by sprang
· 8 years ago
8a25652
Reduce flakiness in EndToEnd probing tests.
by philipel
· 8 years ago
b13237b
Fix deprecated methods in AppRTCMobile.
by kthelgason
· 8 years ago
18703f9
Disable flaky test EndToEndTest.TriggerMidCallProbing
by aleloi
· 8 years ago
6d305ba
Add Windows, Mac, Android support to low bandwidth audio test
by oprypin
· 8 years ago
30cbd0b
Landmine #2 for https://codereview.webrtc.org/2767383005
by oprypin
· 8 years ago
0318463
gtest-parallel: Concatenate the log files in the passed, failed and interrupted dirs.
by ehmaldonado
· 8 years ago
dd27055
Adding PRESUBMIT check on google::protobuf
by mbonadei
· 8 years ago
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