Log - 3f7e0ede1e71f37a5d093c5ee98f6a54b70d4e50 - platform/external/webrtc

d0704ce Remove RTCP tests from channel_unittest. by Bjorn A Mellem · 5 years ago
ee153c9 Send rtcp::RemoteEstimate and rtcp::TransportFeedback in one packet by Per Kjellander · 5 years ago
9e70f36 Roll chromium_revision 651f5a2987..a1c9c88904 (704530:704650) by chromium-webrtc-autoroll · 5 years ago
f17976d Use single thread vp9 decoder for fuzzing by Kuang-che Wu · 5 years ago
45eb135 Remove the unused `receive_timestamp` arg to NetEq::InsertPacket by Karl Wiberg · 5 years ago
c466f08 Cap vp9 fuzzer frame size to prevent OOM by Kuang-che Wu · 5 years ago
cd0eedb Don't allocate audio if we have no transport sequence number. by Sebastian Jansson · 5 years ago
9afdddf Enable capturing from camera in PC framework by Artem Titov · 5 years ago
1699981 Add void::RtcpFeedbackSenderInterface::SendCombinedRtcpPacket by Per Kjellander · 5 years ago
03f4b36 Roll chromium_revision d9b4f45e42..651f5a2987 (704251:704530) by chromium-webrtc-autoroll · 5 years ago
cbbfd08 Replace virtual RtcpPacket::SetSenderSsrc with base member by Danil Chapovalov · 5 years ago
907f154 Revert "Implement rollback for setRemoteDescription" by Alex Loiko · 5 years ago
28214cd Fix handling of large packets in RtxReceiveStream by Niels Möller · 5 years ago
8675eee Bypass unnecessary resampling. by Gustaf Ullberg · 5 years ago
ba700de Add missing dependencies to the static library. by Mirko Bonadei · 5 years ago
066c2ab Roll chromium_revision 8e1616e4fc..d9b4f45e42 (704145:704251) by chromium-webrtc-autoroll · 5 years ago
16d4c4d Implement rollback for setRemoteDescription by Eldar Rello · 5 years ago
5963c7c Count disabled due to low bw streams or layers as bw limited quality in GetStats by Ilya Nikolaevskiy · 5 years ago
955f8fd Add virtual method rtcp::RtcpPacket::SetSenderSsrc by Per Kjellander · 5 years ago
6f41f8e Roll chromium_revision b2d00427a6..8e1616e4fc (703937:704145) by chromium-webrtc-autoroll · 5 years ago
f3f03e2 Removing outdated tests. by Alex Loiko · 5 years ago
f980725 AEC3: Send the spectral power estimates for all channels to AecState by Per Åhgren · 5 years ago
d9755ee Delete large up-front allocation in LibvpxVp8Encoder::InitEncode by Niels Möller · 5 years ago
422b9e0 Run fullband processing at output rate on ARM by Gustaf Ullberg · 5 years ago
1d3008b AEC3: Remove redundant class by Per Åhgren · 5 years ago
9ddd729 Add Duration field to EventRateCounter by Evan Shrubsole · 5 years ago
0169a3e Delete AecState::EchoPathGain() by Sam Zackrisson · 5 years ago
e1092c0 Roll chromium_revision a78cc9b4cc..b2d00427a6 (703818:703937) by chromium-webrtc-autoroll · 5 years ago
6e9395c Roll chromium_revision baa7b58596..a78cc9b4cc (703669:703818) by chromium-webrtc-autoroll · 5 years ago
f77b939 Makes render time > decode time in VideoFrameMatcher. by Sebastian Jansson · 5 years ago
46b0140 Update filter analyzer for multi channel by Sam Zackrisson · 5 years ago
43bd760 Fix build errors of RTCAudioDeviceTests by Byoungchan Lee · 5 years ago
cfe5e2a Stop using goma for MSVC bots. by Mirko Bonadei · 5 years ago
fa77ba6 SetStreams API of RtpSender wrapped for iOS and Android by Cyril Lashkevich · 5 years ago
999afa9 Fix cropping in H264 decoder wrapper. by Sergey Silkin · 5 years ago
7f9a0f3 Roll chromium_revision 977e732442..baa7b58596 (703537:703669) by chromium-webrtc-autoroll · 5 years ago
d46d1e9 Add #COMPONENT to WebRTC. by Patrik Höglund · 5 years ago
e93b1fe Improve bitstream dumping logic to handle multiple SLs correctly by Ilya Nikolaevskiy · 5 years ago
b4161d3 AEC3: Add multichannel support to the residual echo estimator by Per Åhgren · 5 years ago
7e6abf0 Roll chromium_revision 5ac2340a23..977e732442 (703358:703537) by chromium-webrtc-autoroll · 5 years ago
ff27da5 Add/remove receive streams with SSRC 0 from media channels by Saurav Das · 5 years ago
a639f7a Roll chromium_revision 10156469d6..5ac2340a23 (703248:703358) by chromium-webrtc-autoroll · 5 years ago
7c06777 Cleanup includes in modules/include/module_common_types.h by Danil Chapovalov · 5 years ago
0824c6f Delete voice_detection() pointer to submodule by Sam Zackrisson · 5 years ago
24d251f Add 100 ms network delay to the SupportsFlexFEC* tests. by Björn Terelius · 5 years ago
0a6510d Removes rtp_transport checks in AudioSendStream by Sebastian Jansson · 5 years ago
99a2096 Added support for skipping get_audio events, adding dummy packets and setting a field trial string. by Ivo Creusen · 5 years ago
35cf9e7 Replaces static modifier functions in AudioSendStream. by Sebastian Jansson · 5 years ago
db0b3bc Roll chromium_revision 35431c5114..10156469d6 (703133:703248) by chromium-webrtc-autoroll · 5 years ago
b441acf AEC3: Add support in the echo subtractor for handling multiple channels by Per Åhgren · 5 years ago
d21db5d Roll chromium_revision e2b55cc552..35431c5114 (703005:703133) by chromium-webrtc-autoroll · 5 years ago
0e0a04c Roll chromium_revision b5ead1daa2..e2b55cc552 (702047:703005) by chromium-webrtc-autoroll · 5 years ago
2b84dad Fixed issue with H264 packet buffer where it was not detecting presence of sps/pps for idr frames by Shyam Sadhwani · 5 years ago
4f2e940 ACM: Adding support for more than 2 channels in the send pipeline by Per Åhgren · 5 years ago
dc34a25 Adds RTPSenderVideo::Config struct with red/ulpfec config by Erik Språng · 5 years ago
b9bfe65 Delete VCMEncodedFrame::VerifyAndAllocate by Niels Möller · 5 years ago
7536bc5 Account for IP and UDP headers in emulated network by Niels Möller · 5 years ago
ed8eadc Update RTC_LOGs in DtlsTransport to be able to distinguish errors. by Henrik Boström · 5 years ago
f83d0ef Revert "Remove an old hack from test_main_lib.cc." by Patrik Höglund · 5 years ago
82a5100 Replacing /target:target with /target in BUILD autofix. by Sebastian Jansson · 5 years ago
ea55b08 Adds support for passing a vector of packets to the paced sender. by Erik Språng · 5 years ago
79f3287 Cleanup of simple TODO(srte) comments. by Sebastian Jansson · 5 years ago
5114a92 Remove an old hack from test_main_lib.cc. by Patrik Höglund · 5 years ago
0429f78 Base overhead calculation for audio priority rate on available data. by Sebastian Jansson · 5 years ago
78c82a4 Adds trial to always start probes with a small padding packet. by Erik Språng · 5 years ago
608083b Reset QP sum when QP is not reported on decoded frame. by Mirta Dvornicic · 5 years ago
6cf554e Reduces locking in RtpSenderVideo. by Erik Språng · 5 years ago
f23131f Removing AudioAllocationSettings moving functionality to AudioSendStream. by Sebastian Jansson · 5 years ago
b96a311 Sum byte counts for all reports of type kStatsReportTypeSsrc by Niels Möller · 5 years ago
2077542 Roll chromium_revision 1fdb019b56..b5ead1daa2 (701929:702047) by chromium-webrtc-autoroll · 5 years ago
62aee93 Adds trial to calculate audio overhead based on available data. by Sebastian Jansson · 5 years ago
f1e97b9 Reland "Prepares RtpSenderVideo for batch forwarding of generated packets" by Erik Språng · 5 years ago
1413ede Roll chromium_revision 4ce9e096a5..1fdb019b56 (701829:701929) by chromium-webrtc-autoroll · 5 years ago
2e70719 Roll chromium_revision 443491f487..4ce9e096a5 (701518:701829) by chromium-webrtc-autoroll · 5 years ago
f4e0c29 SimulcastEncoderAdapter: support per layer fallback and single encoder proxying by Erik Språng · 5 years ago
fddbe6c Improve readability in GoogCcNetworkController::OnSentPacket by Elad Alon · 5 years ago
9d7eb28 Don't limit simulcast layers number for screenshare based on resolution by Ilya Nikolaevskiy · 5 years ago
64672dc Adds log output to peer connection level scenario framework. by Sebastian Jansson · 5 years ago
65235d3 Add GetStats at end of PeerConnection quality tests by Niels Möller · 5 years ago
7c2bed8 Avoid memcpy in JavaToNativeEncodedImage by Niels Möller · 5 years ago
55377fe Roll chromium_revision aa4c7d6aab..443491f487 (701411:701518) by chromium-webrtc-autoroll · 5 years ago
bfcec4c Delete old placeholders for moved api/ header files by Niels Möller · 5 years ago
7c079f6 Reland "Fix minor regression caused by a8336d3" by Evan Shrubsole · 5 years ago
8f736c0 AEC3: Analyze multi-channel SubtractorOutput in AecState by Sam Zackrisson · 5 years ago
b3bb204 Remove unused RtpFrameObject ctor. by philipel · 5 years ago
2f7d779 Use new RtpFrameObject ctor for fuzzing. by philipel · 5 years ago
2be50f5 Roll chromium_revision a935474316..aa4c7d6aab (701308:701411) by chromium-webrtc-autoroll · 5 years ago
8e1343a Add an alt-protocol to SDP to indicate which m= sections use a plugin transport. by Bjorn A Mellem · 5 years ago
894eb8b Roll chromium_revision 1d84c1e780..a935474316 (701137:701308) by chromium-webrtc-autoroll · 5 years ago
4aded80 Roll chromium_revision bd70e4cf18..1d84c1e780 (701000:701137) by chromium-webrtc-autoroll · 5 years ago
1e91551 Fix -Wtautological-constant-compare in test/fuzzers. by Mirko Bonadei · 5 years ago
ef3dbad New class ScopedJavaRefCounted by Niels Möller · 5 years ago
e00ea5e Refactoring CapBitrateToThresholds in SendSideBandwidthEstimation. by Sebastian Jansson · 5 years ago
002b6f4 Fixes for support of disabling lower spatial layers in VP9 by Ilya Nikolaevskiy · 5 years ago
32eae4c AEC3: use different seed for different channels in CNG by Sam Zackrisson · 5 years ago
09f1195 Always pass arguments to INSTANTIATE_TEST_SUITE_P. by Mirko Bonadei · 5 years ago
45b176f Downgrade fps in same step as resolution in initial drop due to size. by Åsa Persson · 5 years ago
08a9f98 Revert "Prepares RtpSenderVideo for batch forwarding of generated packets" by Erik Språng · 5 years ago
e7314cd In ulpfec receiver check for malformed packets to avoid DCHECKS tirggering by Ilya Nikolaevskiy · 5 years ago
2449d7a Refactor legacy FrameBuffer to use EncodedImageBuffer::Realloc by Niels Möller · 5 years ago