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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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b67c44c3f51cd1d9561fe0461880d4a6020a78f5
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
368d002
Roll chromium_revision dbd1569418..31d9542abc (697157:697288)
by chromium-webrtc-autoroll
· 5 years ago
9fa8ef1
absl::make_unique presubmit check.
by Mirko Bonadei
· 5 years ago
317a1f0
Use std::make_unique instead of absl::make_unique.
by Mirko Bonadei
· 5 years ago
809198e
Fix minor regression caused by a8336d3
by Evan Shrubsole
· 5 years ago
7d00342
Remove old packet socket factory header.
by Patrik Höglund
· 5 years ago
e1b7777
Removing deprecated min_pacing_rate alias in StreamsConfig.
by Sebastian Jansson
· 5 years ago
4a822f4
Roll chromium_revision 2e4ccff8a8..dbd1569418 (696956:697157)
by chromium-webrtc-autoroll
· 5 years ago
2c6ea52
In TaskQueueTest::PostDelayedAfterDesctruct increase timeout
by Danil Chapovalov
· 5 years ago
c1c6284
New (empty) build target api:media_stream_interface
by Niels Möller
· 5 years ago
1722182
Roll chromium_revision 3cf04dec00..2e4ccff8a8 (696812:696956)
by chromium-webrtc-autoroll
· 5 years ago
7262fc2
Refactor Rtp Receivers to accept SSRC 0.
by Saurav Das
· 5 years ago
3d16474
in RtcpTransciever use lambdas with move capture.
by Danil Chapovalov
· 5 years ago
3462793
Roll chromium_revision 1d12ff693d..3cf04dec00 (696696:696812)
by chromium-webrtc-autoroll
· 5 years ago
68ef259
Delete deprecated rtc_event.h file
by Danil Chapovalov
· 5 years ago
f5dec1c
Implement Dependency Descriptor reader
by Danil Chapovalov
· 5 years ago
d9cc8c0
Encoder switching based on network and/or resolution conditions.
by philipel
· 5 years ago
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