Gitiles
Code Review
Sign In
gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
ace2a2bcce455c56dfcf12c1b788473f172245ae
ace2a2b
Roll chromium_revision 1fa0f66d36..fbc0a229ff (699974:700084)
by chromium-webrtc-autoroll
· 5 years ago
d6eab9e
Roll chromium_revision 489dde9b43..1fa0f66d36 (699869:699974)
by chromium-webrtc-autoroll
· 5 years ago
c5bc9d6
Treat wlan as a WiFi adapter name on all platforms.
by Qingsi Wang
· 5 years ago
30c2b66
Roll chromium_revision edf70056b5..489dde9b43 (699748:699869)
by chromium-webrtc-autoroll
· 5 years ago
71037a8
Implement TaskQueueBase interface by SingleThreadedTaskQueueForTesting
by Danil Chapovalov
· 5 years ago
ad10222
Cleanup of unused field trials and options in SendSideBandwidthEstimation
by Sebastian Jansson
· 5 years ago
d63f8f8
Roll chromium_revision b3fb292c9b..edf70056b5 (699622:699748)
by chromium-webrtc-autoroll
· 5 years ago
461ee85
Cleanup of target rates in GoogCC/SendSideBandwidthEstimation.
by Sebastian Jansson
· 5 years ago
7bdf073
First step of adding multi-channel support to the echo subtractor
by Per Åhgren
· 5 years ago
538ca57
Converts const methods in BitrateAllocator to non-member functions.
by Sebastian Jansson
· 5 years ago
e32ae4f
Invalidate encoder rates on VideoStreamEncoder::ReconfigureEncoder
by Evan Shrubsole
· 5 years ago
0e3b1ff
Moving e to comply to the rest of the stack/heap storage scheme
by Per Åhgren
· 5 years ago
90d6efb
Revert "VP9 encoder: handle disabled layers correctly"
by Ilya Nikolaevskiy
· 5 years ago
7911d37
AEC3: Simplify use of SignalTransition
by Gustaf Ullberg
· 5 years ago
01dd885
Moves contents of bitrate_controller to goog_cc
by Sebastian Jansson
· 5 years ago
88fe84b
VP9 encoder: handle disabled layers correctly
by Ilya Nikolaevskiy
· 5 years ago
b67c44c
Add unit tests for balanced degradation settings.
by Åsa Persson
· 5 years ago
70bc753
Add comments to MultiCodecReceiveTest.
by Åsa Persson
· 5 years ago
84004c4
Roll chromium_revision f24b5ede72..b3fb292c9b (699513:699622)
by chromium-webrtc-autoroll
· 5 years ago
fc604aa
Unset sinks when deleting CompositeDataChannelTransport.
by Bjorn A Mellem
· 5 years ago
9d91174
Roll chromium_revision ae812cd84f..f24b5ede72 (699366:699513)
by chromium-webrtc-autoroll
· 5 years ago
bc3eebc
Reland "Reland "Refactor SCTP data channels to use DataChannelTransportInterface.""
by Bjorn A Mellem
· 5 years ago
2225c06
Roll chromium_revision 9f21b695c1..ae812cd84f (699240:699366)
by chromium-webrtc-autoroll
· 5 years ago
f34116e
Replacing bandwidth adaptation trial with stable target in Opus encoder.
by Sebastian Jansson
· 5 years ago
c30bc16
Adds abs-send-time and size field outputs to event log parser.
by Sebastian Jansson
· 5 years ago
74344d2
Support 2 byte payload size DTX packets in NetEq simulation.
by Jakob Ivarsson
· 5 years ago
9d28102
Remove deprecated method
by Johannes Kron
· 5 years ago
af3fdc0
AEC3: Suppression filter handles multiple channels
by Gustaf Ullberg
· 5 years ago
67309ef
Add release callback and reference count to java EncodedImage class
by Niels Möller
· 5 years ago
1b57541
Always pass arguments to INSTANTIATE_TEST_SUITE_P.
by Mirko Bonadei
· 5 years ago
63df20a
Roll chromium_revision 1d4ed9e21d..9f21b695c1 (699120:699240)
by chromium-webrtc-autoroll
· 5 years ago
4c93aab
Handle macro _M_ARM64 for MSVC build
by Tom Tan
· 5 years ago
ef14f07
Delete AudioDecoder method IncomingPacket
by Niels Möller
· 5 years ago
82ce384
Add improvement directions to PC and Call framework metrics
by Artem Titov
· 5 years ago
834a554
Include module_common_types.h only where needed
by Niels Möller
· 5 years ago
bf5ee00
Disable prerender smoothing in MultiCodecReceiveTest.
by Åsa Persson
· 5 years ago
a8e6f34
Delete the BasicPortAllocator constructor that enables gturn
by Niels Möller
· 5 years ago
f2690a1
Delete unused method SendSideBandwidthEstimation::UpdateReceiverBlock
by Niels Möller
· 5 years ago
bc646ee
Roll chromium_revision 09b71d3027..1d4ed9e21d (698937:699120)
by chromium-webrtc-autoroll
· 5 years ago
cd40de9
Delete the deprecated GetTransportParametersOffer().
by Bjorn A Mellem
· 5 years ago
988e63e
Proxy OnRtcpPacketReceived to the worker thread in channel tests.
by Bjorn A Mellem
· 5 years ago
aab43db
Roll chromium_revision 82de2e611e..09b71d3027 (698813:698937)
by chromium-webrtc-autoroll
· 5 years ago
a99b89b
AEC3: Echo remover handles multiple capture signals.
by Gustaf Ullberg
· 5 years ago
3433d56
Reduce resolution and bitrates of smoke test
by Johannes Kron
· 5 years ago
f7457e5
Store PacketBuffer by value instead of as reference counted object
by Danil Chapovalov
· 5 years ago
3c5f91b
Roll chromium_revision e74d6b592b..82de2e611e (698711:698813)
by chromium-webrtc-autoroll
· 5 years ago
289f313
Roll chromium_revision 5cbf4ebd59..e74d6b592b (698593:698711)
by chromium-webrtc-autoroll
· 5 years ago
4854b9f
Roll chromium_revision 230cc8f7e4..5cbf4ebd59 (698466:698593)
by chromium-webrtc-autoroll
· 5 years ago
37ad5ab
Change DatagramTransportInterface methods to pure virtual.
by Bjorn A Mellem
· 5 years ago
88db835
Change DataChannelTransportInterface/Sink methods to pure virtual.
by Bjorn A Mellem
· 5 years ago
d702231
Cleanup deprecated monitoring of MediaTransport state.
by Bjorn A Mellem
· 5 years ago
5ac329c
Cap h264 fuzzer input to 200k.
by Patrik Höglund
· 5 years ago
03bbef5
Fix accidental change of transport time metric
by Johannes Kron
· 5 years ago
c2e9d84
Roll chromium_revision 303c57cf17..230cc8f7e4 (698351:698466)
by chromium-webrtc-autoroll
· 5 years ago
27b0e0d
Remove obsolete todo comment in simulcast.h
by Åsa Persson
· 5 years ago
544dfb5
Delete isac GetBandwidthInfo/SetBandwidthInfo
by Niels Möller
· 5 years ago
ef83cc5
Add fuzzer testing for Dependency Descriptor rtp header extension
by Danil Chapovalov
· 5 years ago
04fd215
Cleanup passing rtp packet to ulpfec receiver.
by Danil Chapovalov
· 5 years ago
0cff4fc
Removed unused frame_size param from RtpFrameObject ctor.
by philipel
· 5 years ago
48b32b7
Delete support for enabling adaptive isac mode
by Niels Möller
· 5 years ago
b5e4785
RtpFrameObject now takes an EncodedImageBuffer in its ctor.
by philipel
· 5 years ago
fb59a6a
Return `const char*` from ToString(RTCErrorType error).
by Mirko Bonadei
· 5 years ago
e0b3167
Delete dead code inside #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
by Niels Möller
· 5 years ago
feee1e4
Add flag to APM to force multichannel even with AEC3
by Sam Zackrisson
· 5 years ago
e24557f
Declare api:libjingle_peerconnection_api dependency on media:media_base
by Niels Möller
· 5 years ago
2051b8b
Roll chromium_revision a536fa4a4a..303c57cf17 (698214:698351)
by chromium-webrtc-autoroll
· 5 years ago
95c538f
Roll chromium_revision fc1e948f93..a536fa4a4a (698112:698214)
by chromium-webrtc-autoroll
· 5 years ago
f288c8e
Roll chromium_revision cf1a2beb4b..fc1e948f93 (697976:698112)
by chromium-webrtc-autoroll
· 5 years ago
c12db81
Add frame receive to frame rendered metric to video_quality_analyzer
by Johannes Kron
· 5 years ago
f0be5b5
Make GetBitstream non-virtual since it is no longer needed for testing.
by philipel
· 5 years ago
40de3cc
Propagating TargetRate struct to BitrateAllocator.
by Sebastian Jansson
· 5 years ago
ac315b2
Add support for max/min encode bitrate to peer connection quality test
by Johannes Kron
· 5 years ago
6a09263
Delete obsolete isac "assign" api
by Niels Möller
· 5 years ago
d8ffbb0
Roll chromium_revision afdb2e7a8b..cf1a2beb4b (697871:697976)
by chromium-webrtc-autoroll
· 5 years ago
76161f7
Move the call to GetBitstream out of the RtpFrameObject ctor.
by philipel
· 5 years ago
14137a1
Adds logging of audio sessions status on the recording side in ADM for Android.
by henrika
· 5 years ago
86873f0
Improve field trial error message.
by Björn Terelius
· 5 years ago
e942b14
New build target api:media_interface
by Niels Möller
· 5 years ago
0a5ed89
Adds remote estimates to rtc event log.
by Sebastian Jansson
· 5 years ago
6ed60e3
Implement Dependency Descriptor writer
by Danil Chapovalov
· 5 years ago
489843f
Improve trendline estimator logging.
by Björn Terelius
· 5 years ago
693bf1e
Delete modules/rtp_rtcp local DivideRoundToNearest in favor on one in rtc_base
by Danil Chapovalov
· 5 years ago
bd24260
Roll chromium_revision eae7ecf757..afdb2e7a8b (697744:697871)
by chromium-webrtc-autoroll
· 5 years ago
efa04ef
Roll chromium_revision 65274319fc..eae7ecf757 (697640:697744)
by chromium-webrtc-autoroll
· 5 years ago
93b1ea2
Using struct for bitrate allocation limits.
by Sebastian Jansson
· 5 years ago
1b83a9e
Only handle each RTCP once.
by Sebastian Jansson
· 5 years ago
4bad650
Roll chromium_revision 2bd75c72c1..65274319fc (697505:697640)
by chromium-webrtc-autoroll
· 5 years ago
7b04a91
Delete almost all default methods on PeerConnectionInterface
by Niels Möller
· 5 years ago
e607a06
Removed unused include from PacketBuffer.
by philipel
· 5 years ago
33b83fd
Introduce integer division helpers with non-default rounding
by Danil Chapovalov
· 5 years ago
b6a45dd
Revert "Fix minor regression caused by a8336d3"
by Evan Shrubsole
· 5 years ago
53227cc
Remove webrtc::MinPositive from api/.
by Mirko Bonadei
· 5 years ago
1162ba2
Add max/min encode bitrates to video config of peer connection tests
by Johannes Kron
· 5 years ago
7cfde54
Roll chromium_revision 51a0808947..2bd75c72c1 (697405:697505)
by chromium-webrtc-autoroll
· 5 years ago
738bfa7
Remove api/bitrate_constraints.h.
by Mirko Bonadei
· 5 years ago
c128df1
Update style guide for absl::make_unique.
by Mirko Bonadei
· 5 years ago
95c4b91
Roll chromium_revision 31d9542abc..51a0808947 (697288:697405)
by chromium-webrtc-autoroll
· 5 years ago
ee5ec9a
Replacing local closure classes with C++14 moving capture lambdas.
by Sebastian Jansson
· 5 years ago
4d461ba
Reusing MediaStreamAllocationConfig struct in ObserverConfig.
by Sebastian Jansson
· 5 years ago
86314cf
Cleaning up C++14 move into lambda TODOs.
by Sebastian Jansson
· 5 years ago
Next »