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gerrit-public.fairphone.software
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platform
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external
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webrtc
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64b626b03f578c936da1e48ccc172ee3b6487fba
64b626b
Use Abseil container algorithms in pc/
by Steve Anton
· 6 years ago
b7446ed
Removing receive RIDs and Simulcast Layers.
by Amit Hilbuch
· 6 years ago
9bcf80a
Roll chromium_revision fa9574f1d1..ed7fd9b77f (626644:626752)
by chromium-webrtc-autoroll
· 6 years ago
733e087
Ignore duplicated incoming RTCP packets in RTC event log parser.
by Bjorn Terelius
· 6 years ago
a75f618
Roll chromium_revision 0a788fbaed..fa9574f1d1 (626455:626644)
by chromium-webrtc-autoroll
· 6 years ago
bcd39d4
Creating Simulcast offer and answer in Peer Connection.
by Amit Hilbuch
· 6 years ago
e76ca61
Allow use of functions in absl/algorithms
by Steve Anton
· 6 years ago
48c5493
Add 'UpdateAllocationLimits' in media transport.
by Piotr (Peter) Slatala
· 6 years ago
435ea0a
Add is_fec property to RtpPacketToSend
by Niels Möller
· 6 years ago
a3ed451
Add static factory method from FrameGenerator for FrameGeneratorCapturer.
by Artem Titov
· 6 years ago
37ec55e
[clang-tidy] Apply performance-faster-string-find fixes.
by Mirko Bonadei
· 6 years ago
190713c
Remove +api from internal DEPS files.
by Mirko Bonadei
· 6 years ago
7d61352
Remove unused defines and methods in internal_defines.h
by Åsa Persson
· 6 years ago
739baf0
[clang-tidy] Apply performance-for-range-copy fixes.
by Mirko Bonadei
· 6 years ago
2d65fff
Roll chromium_revision 53292b65a5..0a788fbaed (626349:626455)
by chromium-webrtc-autoroll
· 6 years ago
8270904
Roll chromium_revision 334d413a77..53292b65a5 (626249:626349)
by chromium-webrtc-autoroll
· 6 years ago
f380284
(7) Rename files to snake_case: remove forwarding headers
by Steve Anton
· 6 years ago
55b91b9
Only create no-op DTLS if media transport is used for both media and data
by Piotr (Peter) Slatala
· 6 years ago
9058e07
Roll chromium_revision 3343618014..334d413a77 (626126:626249)
by chromium-webrtc-autoroll
· 6 years ago
d970807
Remove rtc_base/scoped_ref_ptr.h.
by Mirko Bonadei
· 6 years ago
9444f3a
Roll chromium_revision 6a5b2b19b1..3343618014 (626014:626126)
by chromium-webrtc-autoroll
· 6 years ago
d3a5aaa
Check "rtc_include_internal_audio_device" before creating unittest for audio_device_ios_objc
by Jiawei Ou
· 6 years ago
63a176b
Do not modify media transport config when falling back to RTP
by Piotr (Peter) Slatala
· 6 years ago
18f65dc
Don't attempt to unwrap RTP timestamps for RTX stream.
by Bjorn Terelius
· 6 years ago
44b31d6
Delete leftover method MaxConfiguredBitrateVideo and member remote_ssrc_
by Niels Möller
· 6 years ago
0ef117e
Improving robustness of stable bandwidth estimate test.
by Sebastian Jansson
· 6 years ago
bebca61
Delete unused method SetSelectiveRetransmissions
by Niels Möller
· 6 years ago
728b5a4
Fix initialization to prevent SIGSEGV
by Artem Titov
· 6 years ago
b2d7141
Revert "Always use real VideoStreamsFactory in full stack tests"
by Ilya Nikolaevskiy
· 6 years ago
da37473
Make webrtc::ParseCandidate() public.
by Guido Urdaneta
· 6 years ago
99ec6f3
AEC3: Remove unused kill-switches from AdjustConfig
by Gustaf Ullberg
· 6 years ago
a9316c9
frame_analyzer: exit with status 1 when video files fail to open
by Oleh Prypin
· 6 years ago
a8f9e25
Make sure lost packets are removed from FakeNetworkPipe.
by Johannes Kron
· 6 years ago
fe490d8
Roll chromium_revision b483b4fce1..6a5b2b19b1 (625914:626014)
by chromium-webrtc-autoroll
· 6 years ago
e47433f
AEC3: Remove legacy render buffering
by Gustaf Ullberg
· 6 years ago
8a40edd
Delete constant RTP_PAYLOAD_NAME_SIZE
by Niels Möller
· 6 years ago
76cf320
Roll chromium_revision eedb2069ef..b483b4fce1 (625788:625914)
by chromium-webrtc-autoroll
· 6 years ago
b8c81c3
Roll chromium_revision 3432970f4e..eedb2069ef (625619:625788)
by chromium-webrtc-autoroll
· 6 years ago
f50c6c2
Introduce VideoQualityAnalyzerInjectionHelper.
by Artem Titov
· 6 years ago
3ea55d5
Reland "Delete RtpUtility::Payload, and refactor RTPSender to not use it"
by Niels Möller
· 6 years ago
5affbf2
Turn off automatic quality scaling for simulcast in video_loopback.
by philipel
· 6 years ago
3770b99
Allow repeated feedback packets in log parser.
by Sebastian Jansson
· 6 years ago
84ca69a
Add RTC event logging of LossNotification RTCP messages
by Elad Alon
· 6 years ago
e2fffd7
Roll chromium_revision 1aa6cb924c..3432970f4e (625210:625619)
by chromium-webrtc-autoroll
· 6 years ago
68d5860
Override default manifest from Chromium in WebRTC.
by Sami Kalliomäki
· 6 years ago
a67a9d9
Handle zero number of spatial layers at calculation of VP9 SVC padding.
by Sergey Silkin
· 6 years ago
f8e7ccb
Create new RTCP feedback message - LossIndication
by Elad Alon
· 6 years ago
2d05050
Revert "Roll chromium_revision 1aa6cb924c..faaba5b0a8 (625210:625596)"
by Oleh Prypin
· 6 years ago
81d4bf7
Revert "Delete RtpUtility::Payload, and refactor RTPSender to not use it"
by Artem Titov
· 6 years ago
1e27fec
Negate flag name for prerender smoothing and update comments.
by Rasmus Brandt
· 6 years ago
2fd09a4
Remove deprecated code from audio device.
by Mirko Bonadei
· 6 years ago
88ca008
Roll chromium_revision 1aa6cb924c..faaba5b0a8 (625210:625596)
by chromium-webrtc-autoroll
· 6 years ago
fc2175d
Introduce QualityAnalyzingVideoEncoder and QualityAnalyzingVideoDecoder.
by Artem Titov
· 6 years ago
171df93
Delete RtpUtility::Payload, and refactor RTPSender to not use it
by Niels Möller
· 6 years ago
2820d17
Roll chromium_revision 1cac36a781..1aa6cb924c (624101:625210)
by chromium-webrtc-autoroll
· 6 years ago
dbb49df
Moving UniqueIdGenerator into rtc_base.
by Amit Hilbuch
· 6 years ago
6fde78c
Prevent mac_framework_bundle configs from getting reset
by Thomas Anderson
· 6 years ago
ce7032b
Fixing snake_case files that were renamed in PRESUBMIT.py
by Amit Hilbuch
· 6 years ago
6a32de4
Fix potential race in CallTest.
by Erik Språng
· 6 years ago
2c58ba1
Move simulcast hysteresis factor parsing to RateControlSettings
by Erik Språng
· 6 years ago
83d5e86
Use EncoderSimulcastProxy for all codecs
by Florent Castelli
· 6 years ago
b599787
Make UlpfecReceiverImpl use rtc::TimeMillis, not Clock::GetRealTimeClock
by Niels Möller
· 6 years ago
4b4266f
Move parsing of trusted rate controller to RateControlSettings
by Erik Språng
· 6 years ago
470a5ea
Introduces common AudioAllocationSettings class.
by Sebastian Jansson
· 6 years ago
33b716f
Publish task queue test suite.
by Danil Chapovalov
· 6 years ago
b0397d6
Always send abs-send-time when negotiated and do not filter it out.
by Konrad Hofbauer
· 6 years ago
ae6e0b2
[CodeHealth] Fix use after std::move instances.
by Yves Gerey
· 6 years ago
e54287a
Correctly specify Mac version as 10.13 for iOS simulator tests
by Oleh Prypin
· 6 years ago
df919fb
Don't pretend we've received an end-of-candidates indication.
by Jonas Olsson
· 6 years ago
28522dc
Rename new build targets to follow the recent large file rename
by Karl Wiberg
· 6 years ago
79f0d4d
Enables feature to account for unacknowledged data.
by Sebastian Jansson
· 6 years ago
7e4341d
Reland "Reland "Adds richer packet and ice processing to ParsedRtcEventLog.""
by Sebastian Jansson
· 6 years ago
41dd0bc
Fix typo in rtc_base/thread_checker.h.
by Mirko Bonadei
· 6 years ago
067dc86
Make SetFirstSubFrameInFrame and SetLastSubFrameInFrame protected
by Elad Alon
· 6 years ago
3fdf90d
PSFB without REMB magic word is not an error
by Elad Alon
· 6 years ago
18cf238
Always use real VideoStreamsFactory in full stack tests
by Ilya Nikolaevskiy
· 6 years ago
d47d3eb
Report rendered pixels statistic in full stack tests
by Ilya Nikolaevskiy
· 6 years ago
0500b52
Reduce webrtc_perf_tests duration on buildbots
by Ilya Nikolaevskiy
· 6 years ago
23213d9
Refactor FileRotatingStream to use FileWrapper rather than FileStream
by Niels Möller
· 6 years ago
efd7034
Include video_bitrate_allocator.h, now that's in api/
by Niels Möller
· 6 years ago
cd76eab
Parsing of pacing factor and alr probing in RateControlSettings
by Erik Språng
· 6 years ago
44f0f87
Remove NetworkManager::{set_ipv6_enabled,ipv6_enabled}.
by Mirko Bonadei
· 6 years ago
71b5a7d
Revert "Reland "Adds richer packet and ice processing to ParsedRtcEventLog.""
by Sebastian Jansson
· 6 years ago
6fc6a0c
Reland "Adds richer packet and ice processing to ParsedRtcEventLog."
by Sebastian Jansson
· 6 years ago
5be3bbd
Clarify and unify network delay plot annotations.
by Konrad Hofbauer
· 6 years ago
fe3dfee
Add Visual Studio Code project folder to gitignore file.
by Konrad Hofbauer
· 6 years ago
4007360
Remove dimensions subarray from internal iOS bots config
by Artem Titarenko
· 6 years ago
7121564
Move congestion window field trial parsing to new class.
by Erik Språng
· 6 years ago
b4c6d1e
Connect global task queue factory and rtc::TaskQueue
by Danil Chapovalov
· 6 years ago
443760d
Android: Add option to print native stack traces in PeerConnectionFactory API
by Magnus Jedvert
· 6 years ago
1b761ca
Remove simulcast constraints in SimulcastEncoderAdapter
by Florent Castelli
· 6 years ago
e6a4793
AEC3: avoiding a warning in the reverberation decay estimator.
by Jesús de Vicente Peña
· 6 years ago
7510e4a
Reland "Android: Add helper methods for printing native stack traces"
by Magnus Jedvert
· 6 years ago
dfc7d63
Deprecate FirstSubFrameInFrame() and LastSubFrameInFrame()
by Elad Alon
· 6 years ago
05acd2b
Removes clock from TransportFeedbackAdapter.
by Sebastian Jansson
· 6 years ago
805a27e
Reland "Refactor WebRtcVideoEngine tests to not use cricket::VideoCapturer, part 2."
by Niels Möller
· 6 years ago
5a6ae02
Reland "Trim down FileWrapper class to be merely a wrapper owning a FILE*"
by Niels Möller
· 6 years ago
700615f
Revert "Android: Add helper methods for printing native stack traces"
by Magnus Jedvert
· 6 years ago
d7bc097
Change webrtc-internal iOS pool for try and ci bots
by Artem Titarenko
· 6 years ago
36e0f57
Replace the usage of RTC_HISTOGRAM_COMMON_BLOCK with
by Qingsi Wang
· 6 years ago
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