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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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bb56d4b0e20db884b8393d34e55bb3ca257b6970
bb56d4b
Revert "Refactors BitrateProber with unit types and absolute probe time."
by Erik Språng
· 5 years ago
ebf4552
Adds WebRTC-Audio-AgcMinMicLevelExperiment to AGC1
by henrika
· 5 years ago
2b9317a
Stop checking VP8BaseHeavyTl3RateAllocation field trial on every frame.
by Rasmus Brandt
· 5 years ago
a06048a
Return status instead of CHECKing in event log parser.
by Björn Terelius
· 5 years ago
cc9bf63
Revert "Correct AEC3 multichannel functionality activation"
by Per Åhgren
· 5 years ago
191e38f
Delete gturn support
by Niels Möller
· 5 years ago
9dda1b3
Correct AEC3 multichannel functionality activation
by Per Åhgren
· 5 years ago
8846c8a
RNN VAD: cast and scale quantized weights at init
by Alessio Bazzica
· 5 years ago
26452ff
Cleanup of TransportFeedbackAdapter.
by Sebastian Jansson
· 5 years ago
c3d1f9b
Enable injection of a custom NetEqFactory into PeerConnectionFactory.
by Ivo Creusen
· 5 years ago
2ebbff8
do not offer gcm as the preferred cipher suite
by Philipp Hancke
· 5 years ago
3ce44a3
Move NetEq headers to api/
by Ivo Creusen
· 5 years ago
739a5b3
Refactors BitrateProber with unit types and absolute probe time.
by Erik Språng
· 5 years ago
8d65e9a
Fixes pacing interval dependency and race in BandwidthEndToEndTest
by Erik Språng
· 5 years ago
caaa9e7
AEC3: Handle multichannel audio in single CNG instance
by Gustaf Ullberg
· 5 years ago
cd2a92f
Removes RPLR based FEC controller.
by Sebastian Jansson
· 5 years ago
d1ea4c9
Update comments on Audio Level RTP header extension.
by Minyue Li
· 5 years ago
577c580
Do not stop SingleThreadedTaskQueueForTestingTest near the end of the tests
by Danil Chapovalov
· 5 years ago
fb075d5
Removing unused Opus wrapper API: WebRTCOpus_DecodePlc.
by Minyue Li
· 5 years ago
0cbb58e
Reland "Refactoring of the noise suppressor and adding true multichannel support"
by Per Åhgren
· 5 years ago
159b417
Keep the video send stream alive if the encoder drop frames.
by Jakob Ivarsson
· 5 years ago
c35333d
Add RTC_EXPORT_TEMPLATE_{DECLARE,DEFINE} macros.
by Mirko Bonadei
· 5 years ago
de36595
Added new Apple devices.
by Yura Yaroshevich
· 5 years ago
32913c1
Removes the flakiness in PeerConnectionUsageHistogramTest.
by Qingsi Wang
· 5 years ago
5bd8cb7
Revert "Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5.""
by Henrik Boström
· 5 years ago
54d0278
Renaming opus_interface.c to opus_interface.cc.
by Minyue Li
· 5 years ago
09860e0
Split out counting unique rtp timestamps from packet_buffer
by Danil Chapovalov
· 5 years ago
a0adf3d
Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5."
by Trevor Hayes
· 5 years ago
9560d7d
Make update_rect optional in VideoFrame
by Ilya Nikolaevskiy
· 5 years ago
4e19670
[PeerConnection] Implement parameterless SetLocalDescription().
by Henrik Boström
· 5 years ago
9b66114
Disable rendering statistics while video is paused.
by Sami Kalliomäki
· 5 years ago
4778f6c
Revert "Refactoring of the noise suppressor and adding true multichannel support"
by Artem Titov
· 5 years ago
9c712bb
Fix invalid @Nullable handling in TextureBufferImpl.
by Sami Kalliomäki
· 5 years ago
f298855
Cleanup of feedback observer interface
by Sebastian Jansson
· 5 years ago
470b2d5
Stop relying on GN's sources_assignment_filter.
by Mirko Bonadei
· 5 years ago
ae40e19
AEC3: Adding a configurable render signal gain
by Per Åhgren
· 5 years ago
87a7b82
Refactoring of the noise suppressor and adding true multichannel support
by Per Åhgren
· 5 years ago
c6c3f86
Expose TLS version and SRTP cipher to API
by Harald Alvestrand
· 5 years ago
6981fb5
Add support to not use turn server as stun server.
by Honghai Zhang
· 5 years ago
74f96ec
Removes unused late feedback plot from analyzer.
by Sebastian Jansson
· 5 years ago
9cdc9cc
Cleanup of deprecated RTPSender code
by Erik Språng
· 5 years ago
cb30726
Remove deprecated Audio Processing APIs
by Gustaf Ullberg
· 5 years ago
6e4e688
Fixed MSAN issue with usrsctp reliability test.
by Yura Yaroshevich
· 5 years ago
fbec2ec
Detach H264 sps pps tracker from VCMPacket
by Danil Chapovalov
· 5 years ago
05c4792
Removes OnPacketAdded callback from feedback adapter.
by Sebastian Jansson
· 5 years ago
9c71e49
Remove redundant BitrateProber::OnIncomingPacket() call
by Erik Språng
· 5 years ago
01a21f7
Roll chromium_revision 9109135db0..7ce0264138 (709913:710014)
by chromium-webrtc-autoroll
· 5 years ago
77b7529
Reland "Use RtpSenderEgress directly instead of via RTPSender"
by Erik Språng
· 5 years ago
79e653c
Apply bitrate boosting depending on field-trial.
by Ivo Creusen
· 5 years ago
6f5b9e0
Roll chromium_revision d68d92fb45..9109135db0 (709806:709913)
by chromium-webrtc-autoroll
· 5 years ago
70770ac
Make AudioFrame member instead of raw pointer in APM test fixture
by Sam Zackrisson
· 5 years ago
cff20c2
Adds protected bitrate helper methods to RtpRtcpImpl
by Erik Språng
· 5 years ago
a3728d3
Reland "[PeerConnection] Use an OperationsChain in PeerConnection for async ops."
by Henrik Boström
· 5 years ago
f50d58b
Add .clangd to .gitignore
by Rasmus Brandt
· 5 years ago
5cb7807
Implement crypto stats on DTLS transport
by Harald Alvestrand
· 5 years ago
a81e2b4
Revert "Use RtpSenderEgress directly instead of via RTPSender"
by Erik Språng
· 5 years ago
b533010
Use RtpSenderEgress directly instead of via RTPSender
by Erik Språng
· 5 years ago
3eae7e4
Add exponential backoff of retransmissions for a given packet
by Erik Språng
· 5 years ago
632d57d
Ignore low probe results when using NetworkStateEstimator under field trial
by Per Kjellander
· 5 years ago
1230fb7
ICE : add field trial for initial select dampening
by Jonas Oreland
· 5 years ago
c189ace
Roll chromium_revision 04c3c4c8f1..d68d92fb45 (709704:709806)
by chromium-webrtc-autoroll
· 5 years ago
e38e119
Roll chromium_revision 98ef1d6866..04c3c4c8f1 (709549:709704)
by chromium-webrtc-autoroll
· 5 years ago
e95fc85
Roll chromium_revision 3c5165bebc..98ef1d6866 (709394:709549)
by chromium-webrtc-autoroll
· 5 years ago
91e3ebe
Revert "Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.""
by Mirko Bonadei
· 5 years ago
29db239
Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."
by Mirko Bonadei
· 5 years ago
e114fb6
Added usrsctp reliablitiy stress test.
by Yura Yaroshevich
· 5 years ago
67ac9e8
Prepares RTPSender for extracting RtpSenderEgress
by Erik Språng
· 5 years ago
492fdf4
Make rtc_json poisonous in WebRTC
by Sam Zackrisson
· 5 years ago
1a61739
Fix MemoryLogWriter so that it always writes the full data.
by Björn Terelius
· 5 years ago
53a31f7
Introduce injectable NetEqController interface.
by Ivo Creusen
· 5 years ago
16cec3b
Added allow_codec_switching parameter to RTCConfig.
by philipel
· 5 years ago
49c0880
Revert "[PeerConnection] Use an OperationsChain in PeerConnection for async ops."
by Henrik Boström
· 5 years ago
1dac707
Cleanup PacketBuffer tests to use immediate result
by Danil Chapovalov
· 5 years ago
4f2783b
Speculative Revert: "Use FakeRenderer when fuzzing"
by Patrik Höglund
· 5 years ago
b394a56
Cleanup of EchoControl interface after downstream fixes
by Gustaf Ullberg
· 5 years ago
e277bde
Add APM test of pre-amplifier gain
by Sam Zackrisson
· 5 years ago
e914c1e
Roll chromium_revision 64883b3ea2..3c5165bebc (709283:709394)
by chromium-webrtc-autoroll
· 5 years ago
1dddaa1
[PeerConnection] Use an OperationsChain in PeerConnection for async ops.
by Henrik Boström
· 5 years ago
ef0e4d0
Roll chromium_revision e1ab9e9b20..64883b3ea2 (709180:709283)
by chromium-webrtc-autoroll
· 5 years ago
c4af214
Roll chromium_revision d7338c33b2..e1ab9e9b20 (708965:709180)
by chromium-webrtc-autoroll
· 5 years ago
3cb6104
AEC3: Support negative delay with external delay estimator
by Gustaf Ullberg
· 5 years ago
a922904
Calls OnPacketsAcknowledged on RtpRtcp instead of RTPSender directly.
by Erik Språng
· 5 years ago
b2290f4
Revert "Reset end-of-frame flag in non-VCL packet."
by Sergey Silkin
· 5 years ago
fc78aac
Batches video frame packets when posting to pacer
by Erik Språng
· 5 years ago
2040dcf
Roll chromium_revision f656c810e4..d7338c33b2 (708845:708965)
by chromium-webrtc-autoroll
· 5 years ago
eec3919
Remove trial WebRTC-Bwe-ProbeRateFallback
by Per Kjellander
· 5 years ago
d113ee3
Removes deprecated WebRTC-Bwe-AimdRateControl-NetworkState trial.
by Sebastian Jansson
· 5 years ago
bd20077
Roll chromium_revision 510c0ca3d7..f656c810e4 (708742:708845)
by chromium-webrtc-autoroll
· 5 years ago
0e2b581
RTC_EXPORT webrtc::DesktopCapturerDifferWrapper.
by Mirko Bonadei
· 5 years ago
c1a8abc
Roll chromium_revision b5030084da..510c0ca3d7 (708640:708742)
by chromium-webrtc-autoroll
· 5 years ago
4ff1c87
Fix RTC_LOCKABLE RTC_EXPORT order for rtc::Thread.
by Mirko Bonadei
· 5 years ago
d7bf5c5
Revert "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."
by Mirko Bonadei
· 5 years ago
6e81567
Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."
by Mirko Bonadei
· 5 years ago
ce1ffcd
change PacketBuffer to return it's result rather that use callback
by Danil Chapovalov
· 5 years ago
2522b25
Roll chromium_revision 6dc3a51e22..b5030084da (708537:708640)
by chromium-webrtc-autoroll
· 5 years ago
6adb0a2
Do not compile webrtc_lib_link_test if is_asan=true.
by Mirko Bonadei
· 5 years ago
21bfa40
Update APM config on RuntimeSetting pre amplifier gain change
by Sam Zackrisson
· 5 years ago
4f178d0
Fix gtk color-space conversion in peerconnection_client
by Niels Möller
· 5 years ago
d81a04e
Roll chromium_revision c0cca6e419..6dc3a51e22 (708426:708537)
by chromium-webrtc-autoroll
· 5 years ago
0ff7c02
Add multipleTouchEnabled for subview of RTCMTLVideoView and RTCEAGLVideoView
by CZ Theng
· 5 years ago
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