Gitiles
Code Review
Sign In
gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
71061bcca8135375fd9c8af729facb1722103c86
71061bc
Replace calls to deprecated googletest APIs.
by Mirko Bonadei
· 5 years ago
7e8de0b
Revert "Implement true negotiation for DatagramTransport with fallback to RTP."
by Bjorn Mellem
· 5 years ago
b4f3f37
Revert "Remove an over-zealous DCHECK in jsep_transport.h"
by Bjorn Mellem
· 5 years ago
dea0a0c
Revert "Remove another DCHECK that fails during renegotiation."
by Bjorn Mellem
· 5 years ago
907b592
Remove another DCHECK that fails during renegotiation.
by Bjorn A Mellem
· 5 years ago
99daff4
Roll chromium_revision 9cc11952e4..add4dabf7f (666915:667020)
by chromium-webrtc-autoroll
· 5 years ago
ffa007a
Remove an over-zealous DCHECK in jsep_transport.h
by Bjorn A Mellem
· 5 years ago
71c6482
Implement true negotiation for DatagramTransport with fallback to RTP.
by Bjorn A Mellem
· 5 years ago
9e78458
Roll chromium_revision e9da1759f7..9cc11952e4 (666811:666915)
by chromium-webrtc-autoroll
· 5 years ago
fa6ce01
Roll chromium_revision 209e3dbd30..e9da1759f7 (666685:666811)
by chromium-webrtc-autoroll
· 5 years ago
da13ea2
Reland "Added OnIceCandidateError to API and implementation"
by Eldar Rello
· 5 years ago
334808d
Roll chromium_revision 3ab0cb191a..209e3dbd30 (666328:666685)
by chromium-webrtc-autoroll
· 5 years ago
3b8ed28
Revert "Added OnIceCandidateError to API and implementation"
by Yves Gerey
· 5 years ago
7b06b9b
Remove pthatcher@webrtc.org from OWNERS
by Steve Anton
· 5 years ago
b038947
Remove jiayl@webrtc.org from OWNERS
by Steve Anton
· 5 years ago
6de8b17
Roll chromium_revision 5a40f1184e..3ab0cb191a (666217:666328)
by chromium-webrtc-autoroll
· 5 years ago
9469c78
Added OnIceCandidateError to API and implementation
by Eldar Rello
· 5 years ago
ab62b2e
Don't copy video frame metadata in each encoder/decoder
by Ilya Nikolaevskiy
· 5 years ago
9930929
Adds srte@ as OWNER of units.
by Sebastian Jansson
· 5 years ago
4fc0855
Cleanup video frame metadata copying
by Ilya Nikolaevskiy
· 5 years ago
b64af4b
Add retransmission_allowed flag to encoder output
by Elad Alon
· 5 years ago
781653c
Added functions to control the VideoStreamDecoder playout delay.
by philipel
· 5 years ago
4d9e428
Remove some leftover TODOs for webrtc:10336
by Elad Alon
· 5 years ago
aa3e612
Roll chromium_revision fd17362e28..5a40f1184e (666098:666217)
by chromium-webrtc-autoroll
· 5 years ago
f91353e
FecControllerDefault nits (missing empty lines)
by Elad Alon
· 5 years ago
74e63b8
Add missing proxy function for overloaded StartRtcEventLog peer connection function.
by Tim Haloun
· 5 years ago
dd0094a
Deprecate RtpRtcp::SetKeyFrameRequestMethod
by Niels Möller
· 5 years ago
48edc92
Delete deprecated AudioDeviceWithDataObserver factory
by Danil Chapovalov
· 5 years ago
517d8a0
Delete unused enum ProtectionType
by Niels Möller
· 5 years ago
36690cd
Fix inverted RTC_DCHECK in RtpVideoStreamReceiver::RtcpFeedbackBuffer
by Elad Alon
· 5 years ago
ba96e2f
In FrameEncodeMetadataWriter don't clear known bitrate on Reset.
by Ilya Nikolaevskiy
· 5 years ago
015ff80
Roll chromium_revision abb1a36732..fd17362e28 (665960:666098)
by chromium-webrtc-autoroll
· 5 years ago
06a9926
Roll chromium_revision f8c14c5353..abb1a36732 (665857:665960)
by chromium-webrtc-autoroll
· 5 years ago
835baf7
Add amithi@ as pc OWNERS
by Steve Anton
· 5 years ago
e0f3704
Add cap to video jitter buffer size/latency in experiment branches only.
by “Michael
· 5 years ago
479a3c0
Add support for enabling and negotiating raw RTP packetization.
by Mirta Dvornicic
· 5 years ago
961407f
Delete unused method RtpRtcp::GetRtpPacketLossStats
by Niels Möller
· 5 years ago
65853dd
Roll chromium_revision 1070231d7d..f8c14c5353 (665750:665857)
by chromium-webrtc-autoroll
· 5 years ago
31f18e1
Android SurfaceTextureHelper: Avoid crashing if size hasn't been set
by Magnus Jedvert
· 5 years ago
f4c7ab1
in test/scenario pass TaskQueueFactory explicitly
by Danil Chapovalov
· 5 years ago
c8501f7
Fix bug in neteq_quality_test
by Pablo Barrera González
· 5 years ago
90bc1e1
Fix comment typo about degradation preference.
by Åsa Persson
· 5 years ago
bd00271
Roll chromium_revision 584b49b1a7..1070231d7d (665633:665750)
by chromium-webrtc-autoroll
· 5 years ago
292ce4e
Move datagram transport to JsepTransport
by Anton Sukhanov
· 5 years ago
9005e23
Roll chromium_revision a3e71ebfa3..584b49b1a7 (665525:665633)
by chromium-webrtc-autoroll
· 5 years ago
1716d39
Let SessionDescription take ownership of MediaDescription
by Harald Alvestrand
· 5 years ago
1fe119f
Change the gating of surfacing candidates on ICE transport type change
by Qingsi Wang
· 5 years ago
e86af2c
Allowing buffering a LNTF (loss notification) feedback message in RTCPSender
by Elad Alon
· 5 years ago
4e34c18
Check input file extension is not wav
by Pablo Barrera González
· 5 years ago
102b728
Prevent howling in RunPlayoutAndRecordingInFullDuplex
by Gustaf Ullberg
· 5 years ago
15f2200
Roll chromium_revision aaa0f87a5c..a3e71ebfa3 (665423:665525)
by chromium-webrtc-autoroll
· 5 years ago
d2a6686
Add RtpPacketInfo to hold information about a received RtpPacket.
by Chen Xing
· 5 years ago
1df841d
Target SDK level 29 in AppRTCMobile.
by Sami Kalliomäki
· 5 years ago
ef09c5b
Buffer RTCP feedback messages in RtpVideoStreamReceiver
by Elad Alon
· 5 years ago
4cd1c6a
Lockless SwapQueue
by Gustaf Ullberg
· 5 years ago
89bbf37
Allow neteq_quality_test to read a complete file
by Pablo Barrera González
· 5 years ago
7537838
Add fhernqvist to watchlist.
by Fredrik Hernqvist
· 5 years ago
62838fe
Expose audio decoder factory in neteq_quality_test
by Pablo Barrera González
· 5 years ago
695cf6a
Delete deprecated StartRtcEventLog override with PlatformFile
by Niels Möller
· 5 years ago
f330183
A threading explanation
by Harald Alvestrand
· 5 years ago
8db36de
Roll chromium_revision 97b44755d9..aaa0f87a5c (665322:665423)
by chromium-webrtc-autoroll
· 5 years ago
114e8bb
Roll chromium_revision ebd9263281..97b44755d9 (665197:665322)
by chromium-webrtc-autoroll
· 5 years ago
36e3147
Surface the standardized ICE connection state to mobile clients.
by Qingsi Wang
· 5 years ago
2dbc627
Check H264 packetization mode when using IsSameCodec
by Steve Anton
· 5 years ago
2229cf7
Roll chromium_revision c1296cf1c0..ebd9263281 (665078:665197)
by chromium-webrtc-autoroll
· 5 years ago
85b8ce2
In media engine replace forward declaration with proper includes
by Danil Chapovalov
· 5 years ago
d7e2fb3
mb: Implement 'quiet' flag in mb lookup
by Oleh Prypin
· 5 years ago
cecf87f
Reland "Change default secure SCTP protocol to UDP/DTLS/SCTP"
by Guido Urdaneta
· 5 years ago
4436887
Revert "Change default secure SCTP protocol to UDP/DTLS/SCTP"
by Guido Urdaneta
· 5 years ago
a937c6e
Remove Win32 ASan from mb config.
by Mirko Bonadei
· 5 years ago
ad4a3c8
Roll chromium_revision 81e506385d..c1296cf1c0 (664522:665078)
by chromium-webrtc-autoroll
· 5 years ago
e93d109
Add "Win asan 64-bit" in order to migrate away from the 2-bit version.
by Mirko Bonadei
· 5 years ago
eb22227
Add OnDatgramLost and default value for receive_timestamp.
by Bjorn A Mellem
· 5 years ago
d91969e
Explicitly close PeerConnections when using ScopedFieldTrials
by Steve Anton
· 5 years ago
220f4be
Remove some media/ --> pc/ test dependencies
by Steve Anton
· 5 years ago
57dc02a
Add receive_timestamp to DatagramAcks.
by Bjorn A Mellem
· 5 years ago
0c1c1b4
Move ownership of ICE from DtlsTransport to JsepTransport.
by Bjorn A Mellem
· 5 years ago
a913c12
Roll chromium_revision 2d1120f0c1..81e506385d (664417:664522)
by chromium-webrtc-autoroll
· 5 years ago
74bebc5
Add OnDatagramAcked interface
by Anton Sukhanov
· 5 years ago
72055b1
Roll chromium_revision 8891f34d24..2d1120f0c1 (664289:664417)
by chromium-webrtc-autoroll
· 5 years ago
740cc35
Roll chromium_revision d4906ebd49..8891f34d24 (664184:664289)
by chromium-webrtc-autoroll
· 5 years ago
6806550
Fix build with recent linux kernel.
by Emilio Cobos Álvarez
· 5 years ago
85a9d91
Add ability to set min/start/max bitrate on peer's PC in PC quality tests
by Artem Titov
· 5 years ago
845c6aa
Add support for early loss detection using transport feedback.
by Erik Språng
· 5 years ago
b3b3e3f
Add acked bandwidth estimator config for sample uncertainty in ALR.
by Christoffer Rodbro
· 5 years ago
7eb0a5e
AudioDecoderOpus: Add support for 16 kHz output sample rate
by Karl Wiberg
· 5 years ago
ed69d41
Remove deprecated RtcEventLog Create functions
by Danil Chapovalov
· 5 years ago
2f5554d
Make KeyFrameRequestSender injectable in RtpVideoStreamReceiver
by Niels Möller
· 5 years ago
e8e7d7b
Move Connection into it's own .h/.cc file.
by Jonas Oreland
· 5 years ago
28f0eb2
Move H.264 SPS VUI rewriting to FrameEncodeMetadataWriter.
by Mirta Dvornicic
· 5 years ago
a1d1a1e
WebRTC Opus C interface: Add support for non-48 kHz decode sample rate
by Karl Wiberg
· 5 years ago
232b6a1
Propagate screenshare info into video track and it's source.
by Artem Titov
· 5 years ago
98266a4
Roll chromium_revision 99181c0bec..d4906ebd49 (664078:664184)
by chromium-webrtc-autoroll
· 5 years ago
6737841
Add jitterBufferDelay and jitterBufferEmittedCount stats for video
by Guido Urdaneta
· 5 years ago
e4470cd
Roll chromium_revision 9b60f86c15..99181c0bec (663961:664078)
by chromium-webrtc-autoroll
· 5 years ago
686be20
Fix ICE connection in datagram_transport.
by Anton Sukhanov
· 5 years ago
44bd71c
Create a composite implementation of RtpTransportInternal.
by Bjorn A Mellem
· 5 years ago
64e97cf
Roll chromium_revision 09fae7ef1b..9b60f86c15 (663849:663961)
by chromium-webrtc-autoroll
· 5 years ago
f94e3d9
Roll chromium_revision 9809faf8ca..09fae7ef1b (663719:663849)
by chromium-webrtc-autoroll
· 5 years ago
ce33b6a
Implement QualityLimitationReasonTracker and expose "reason".
by Henrik Boström
· 5 years ago
Next »