Gitiles
Code Review
Sign In
gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
6bbeb080b89080f5ad0dfc7da19d1437b51394a3
6bbeb08
Extract rtc_base/base64.h and rtc_base/base64.cc into separate target.
by Artem Titov
· 6 years ago
6250fdd
Delete FakeWebRtcVcmFactory::OnDestroyed method.
by Niels Möller
· 6 years ago
9394f6f
Stop using the beamformer inside APM
by Sam Zackrisson
· 6 years ago
431abd9
Replace rtc::Optional with absl::optional in test and rtc_tools
by Danil Chapovalov
· 6 years ago
9bf3158
Pass buffer with size when writing rtp header extension
by Danil Chapovalov
· 6 years ago
0040b66
Replace rtc::Optional with absl::optional
by Danil Chapovalov
· 6 years ago
ae18886
Disable new RTC_CHECK unittest
by Jonas Olsson
· 6 years ago
00c7183
Replace rtc::Optional with absl::optional in media, ortc, p2p
by Danil Chapovalov
· 6 years ago
5a9ba68
Add base64 webrtc owned third_party dep
by Artem Titov
· 6 years ago
5adf07d
Make instructions for checkin_chrome_dep a bit clearer.
by Patrik Höglund
· 6 years ago
ce4829a
Adds trial to ignore video pacing for audio packets.
by Sebastian Jansson
· 6 years ago
f8e5c11
Refactor checks to use a copy of the new logging backend.
by Jonas Olsson
· 6 years ago
6a9bd74
Fix a downstream test failure.
by Ying Wang
· 6 years ago
c235a8d
Adds trial to always send padding packets when not sending video.
by Sebastian Jansson
· 6 years ago
fc50110
Remove stringstreams from modules/video_coding/
by Jonas Olsson
· 6 years ago
5c43150
Makes BBR congestion window more similar to QUIC.
by Sebastian Jansson
· 6 years ago
fb4d66b
Improves buffer time calculation in network control tester.
by Sebastian Jansson
· 6 years ago
b9b146c
Replace rtc::Optional with absl::optional in audio, call and video
by Danil Chapovalov
· 6 years ago
e61d72b
Disables congestion window in pacer when CongestionWindowPushback is enabled.
by Ying Wang
· 6 years ago
92b24f0
Delete an unneeded include of pathutils.h.
by Niels Möller
· 6 years ago
fc9dcb6
Remove wire-up for cancelled experement on VAAPI VP8 encoding
by Ilya Nikolaevskiy
· 6 years ago
d264df5
Replace rtc::Optional with absl::optional in modules/rtp_rtcp
by Danil Chapovalov
· 6 years ago
394b4eb
Delete unused methods on rtc::Pathname.
by Niels Möller
· 6 years ago
65c61dc
Android: Add helper class for generating OpenGL shaders
by Magnus Jedvert
· 6 years ago
8643b78
Moved NackModule and VCMPacket to their own targets
by Ilya Nikolaevskiy
· 6 years ago
88aee28
Remove support for old test modes in EncodeDecodeTest
by Karl Wiberg
· 6 years ago
d477129
Remove dead RED code in TestRedFec
by Karl Wiberg
· 6 years ago
8fbe4f1
Remove executable insert_packet_with_timing
by Karl Wiberg
· 6 years ago
0f173bd
Revert "Drop tools/gyp from dependencies."
by Artem Titov
· 6 years ago
0a5fdbb
Use RTC_HISTOGRAM_ENUMERATION to report SRTP/SRTCP unprotect error.
by Zhi Huang
· 6 years ago
9eb3886
Adds field trial parser.
by Sebastian Jansson
· 6 years ago
7c32c86
Metal view: Update drawable size when rotating.
by Peter Hanspers
· 6 years ago
724a97d
Drop tools/gyp from dependencies.
by Niels Möller
· 6 years ago
a6fc636
Add ivoc@ and saza@ to audio_processing OWNERS
by Sam Zackrisson
· 6 years ago
6507054
Android: Add tests for VideoFrame.Buffer.toI420() and cropAndScale()
by Magnus Jedvert
· 6 years ago
d1f970d
Change echo detector to scoped_refptr
by Ivo Creusen
· 6 years ago
4e952a3
Remove unused WavFile::FormatAsString method.
by Jonas Olsson
· 6 years ago
671cae2
Handle FileRotatingStreams with long file names
by Jonas Olsson
· 6 years ago
1b36894
Reland "Refactor the regathering of candidates in P2PTransportChannel."
by Qingsi Wang
· 6 years ago
5d16d7f
Add a DCHECK for null port in FakePortAllocator.
by Qingsi Wang
· 6 years ago
60b6c1d
[Unified Plan] Clear RtpSender "SSRC" when the SDP has no send streams
by Steve Anton
· 6 years ago
f7d7e90
Replace std:remove on vector::erase in streamparams_unittest.cc
by Artem Titov
· 6 years ago
aeb0a64
AEC3: Increase the range of reported echo path delay metrics
by Per Åhgren
· 6 years ago
b7700d3
Android: Fix VideoTrack behavior for adding/removing VideoSinks
by Magnus Jedvert
· 6 years ago
493c78a
Replace all use of rtc::Pathname in generator_unittest.cc.
by Niels Möller
· 6 years ago
fabb12e
Introduce list of fields to put into codec agnostic descriptor
by Danil Chapovalov
· 6 years ago
075cb2b
AEC3: Changes to how the reverberation decay is applied.
by Jesús de Vicente Peña
· 6 years ago
9633cff
Remove "webrtc_rtp" traces.
by Jonas Olsson
· 6 years ago
29c36b2
Add ow2_asm license
by Yura Yaroshevich
· 6 years ago
789221f
Adding WebRTC-Audio-ForceNoTWCC field trial
by Alex Narest
· 6 years ago
e3cf3d0
Use enum class for VideoCodecMode and VideoCodecComplexity.
by Niels Möller
· 6 years ago
037b37a
Add implementation of EncodedFrame::Timestamp.
by Niels Möller
· 6 years ago
e4a17c5
Moved timing related logic into its own function in webrtc::PayloadRouter.
by philipel
· 6 years ago
1f4d7a2
Revert "Refactor the regathering of candidates in P2PTransportChannel."
by Qingsi Wang
· 6 years ago
241d0c1
Remove ContinualGatheringPolicy::GATHER_CONTINUALLY_AND_RECOVER.
by Qingsi Wang
· 6 years ago
aed7164
Updated PeerConnection integration test to fix race condition.
by Seth Hampson
· 6 years ago
2cf61e3
Roll chromium_revision ef538e3112..9df92afb16 (566490:566630)
by Autoroller
· 6 years ago
14f8aba
Refactor the regathering of candidates in P2PTransportChannel.
by Qingsi Wang
· 6 years ago
b57e169
Add a flag to actively reset the SRTP parameters
by Zhi Huang
· 6 years ago
9dce71b
Reland "Use absl::optional instead or rtc::Optional"
by Danil Chapovalov
· 6 years ago
57dc9e3
Roll chromium_revision ffaf1e2ba6..ef538e3112 (565764:566490)
by Autoroller
· 6 years ago
abe301f
Add HeaderExtensions to RtpParameters
by Florent Castelli
· 6 years ago
867e510
Enable send side audio TWCC only if WebRTC-Audio-ForceNoTWCC is not enabled.
by Alex Narest
· 6 years ago
910540d
Explicitly setting use_lld=false on MSVC bots.
by Mirko Bonadei
· 6 years ago
7af087a
Metal renderer does not handle i420 frames correctly.
by Peter Hanspers
· 6 years ago
dd3e0ab
Make rtc_software_fallback_wrappers target visible.
by Anders Carlsson
· 6 years ago
cf15eb5
Release ADM after passing it to PCF in AppRTC
by Paulina Hensman
· 6 years ago
d294c85
LogMessage::UpdateMinLogSeverity: Don't ignore all but the last stream
by Karl Wiberg
· 6 years ago
798b282
Don't update internal state of the FrameBuffer2 when an undecodable frame is inserted.
by philipel
· 6 years ago
dadaaee
Remove stringstreams from p2p/
by Jonas Olsson
· 6 years ago
39da65b
remove unused UNSHIPPED trace macros
by Jonas Olsson
· 6 years ago
e1d617c
Delay the creation of the platform thread in TestAudioDeviceModule.
by Sami Kalliomäki
· 6 years ago
9614a31
Remove manual references to exe_and_shlib_deps
by Tom Anderson
· 6 years ago
7685e86
Pass the RtcEventLog instance to ICE via JsepTransportController.
by Qingsi Wang
· 6 years ago
6c789e0
Revert "Add a flag to actively reset the SRTP parameters"
by Zhi Huang
· 6 years ago
bae1031
Add a flag to actively reset the SRTP parameters
by Zhi Huang
· 6 years ago
a46bd4b
Reland "Move class VideoCodec from common_types.h to its own api header file."
by Niels Möller
· 6 years ago
c17ca53
Delete deprecated VideoTrackSource constructor.
by Niels Möller
· 6 years ago
2ac6446
Document that preferred VideoFrame constructor takes no RTP timestamp.
by Niels Möller
· 6 years ago
425f713
Delete unused methods in VCMFrameBuffer and VCMSessionInfo.
by Niels Möller
· 6 years ago
6d19180
Fix increase in send rate when not receiving feedback
by Anastasia Koloskova
· 6 years ago
6bac5c1
Remove TimingFrameFlags from its old place after it was moved
by Ilya Nikolaevskiy
· 6 years ago
80a0a4c
Remove software fallback in Android hardware codec factories.
by Anders Carlsson
· 6 years ago
24db1c9
Remove unused iostream import
by Jonas Olsson
· 6 years ago
fddaf75
AEC3: Increase the look window in the delay estimator.
by Per Åhgren
· 6 years ago
43568dd
Remove stringstreams from pc/
by Jonas Olsson
· 6 years ago
6cdd546
Remove default argument in GetSimulcastConfig.
by Rasmus Brandt
· 6 years ago
fe89271
Android: Use JavaToNativeString in PeerConnectionFactory_InitializeFieldTrials
by Magnus Jedvert
· 6 years ago
b7beba4
Roll chromium_revision e9fcce491e..ffaf1e2ba6 (565609:565764)
by Autoroller
· 6 years ago
1ef88ae
Add kNumValues to IceCandidate* enums and move kUnknown to the front.
by Qingsi Wang
· 6 years ago
b90e63c6
Fix: NetEq PacketBuffer logs discarded packet with wrong codec level when new packet replaces the lower level packet
by Peng Yu
· 6 years ago
ec9c745
Adds support for new Windows ADM with limited API support.
by henrika
· 6 years ago
488eb98
Setting resolution alignment to 4 on iOS.
by Peter Hanspers
· 6 years ago
443e71f
Revert "Disabling VeryLowBitrateVP9 to unblock roll."
by Patrik Höglund
· 6 years ago
2d61162
Remove stringstreams from media/sctp/
by Jonas Olsson
· 6 years ago
25b41f8
remove unused stringstream import
by Jonas Olsson
· 6 years ago
0428d37
Roll chromium_revision f9a24d5d6f..e9fcce491e (564886:565609)
by Autoroller
· 6 years ago
6aa415e
Add full stack test with link capacity of 150 kbit/s.
by Niels Möller
· 6 years ago
350531e
Revert "Move class VideoCodec from common_types.h to its own api header file."
by Danil Chapovalov
· 6 years ago
7252053
Add R8 to DEPS.
by Patrik Höglund
· 6 years ago
Next »