Log - 9f9562592f05c1f037c236f667e3d60065a67a13 - platform/external/webrtc

9f95625 When SDES is used, pass pre-shared key to media transport. by Piotr (Peter) Slatala · 7 years ago
7182286 Allow FakeNetworkPipe to wake up its processing thread by Sebastian Jansson · 7 years ago
693432d Add obj-c mapping from native configuration to RTCConfiguration by Piotr (Peter) Slatala · 7 years ago
e6caa9f export RTCRtpTransceiverInit by Piasy · 7 years ago
ed45c57 Corrects audio overhead correction in Scenario test. by Sebastian Jansson · 7 years ago
69807e8 Depend directly on destination targets. by Yves Gerey · 7 years ago
a8fa2d0 Move some methods from StreamInterface to FifoBuffer by Niels Möller · 7 years ago
21cddff Harmonize paths to dependent targets. by Yves Gerey · 7 years ago
b32bb95 Bugfix: FlexFEC causes retransmit bitrate increase. by Ying Wang · 7 years ago
8b7d206 AEC3: Decrease latency until the delay has been detected by Per Åhgren · 7 years ago
f577ab3 Roll chromium_revision 7e85c0922c..9996ac8918 (604065:604166) by chromium-webrtc-autoroll · 7 years ago
b00b28e Roll chromium_revision 0cb3899c4e..7e85c0922c (603959:604065) by chromium-webrtc-autoroll · 7 years ago
b3f887b Expose key derivation through a simple interface for use in WebRTC. by Benjamin Wright · 7 years ago
1a92cd7 Roll chromium_revision 34bb9a9162..0cb3899c4e (603839:603959) by chromium-webrtc-autoroll · 7 years ago
c78b0ea Create a MediaTransportState enum and add a state callback to MediaTransport. by Bjorn Mellem · 7 years ago
eaf337a Remove event wait logic from DesktopConfigurationMonitor by Emircan Uysaler · 7 years ago
746d46b AGC2: renaming GainCurveApplier to Limiter. by Alessio Bazzica · 7 years ago
fcc3981 Revert "Use only first payload timestamp for RTCP SR generation for audio" by Ilya Nikolaevskiy · 7 years ago
992a868 Fix for clock reset repair. by Christoffer Rodbro · 7 years ago
a2e133d Delete StreamInterface::ReadLine. by Niels Möller · 7 years ago
ed7b8b1 Update media transport settings struct by Piotr (Peter) Slatala · 7 years ago
3e67676 Add support for field trials in peerconnection_client|server by Bjorn Terelius · 7 years ago
9a0662a Use only first payload timestamp for RTCP SR generation for audio by Ilya Nikolaevskiy · 7 years ago
b26cf2f Add field trial to enable the new RTC event log format. by Bjorn Terelius · 7 years ago
97e35ce Revert "Disabled TestPacketBuffer.SeqNumWrapOneFrame test due to clang update" by Artem Titarenko · 7 years ago
0eb7d3ff Always call ConvertToI420 with positive crop_height by Robert Bares · 7 years ago
9862c2e Delete OptionsFile class. Refactored only user, TurnFileAuth. by Niels Möller · 7 years ago
3df6e715 Makes PacketResult::GetSentPacket const. by Sebastian Jansson · 7 years ago
b33168e Roll chromium_revision 89ed1da2c8..34bb9a9162 (603733:603839) by chromium-webrtc-autoroll · 7 years ago
946179c Delete unused function rtc::Flow. by Niels Möller · 7 years ago
42b4315 Add iOS SDK unit tests for nalu_rewriter by Artem Titarenko · 7 years ago
9190b82 Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap by Johannes Kron · 7 years ago
f43bcd4 Remove likely obsolete entries from WATCHLISTS by Oleh Prypin · 7 years ago
0ac98ab Roll chromium_revision 03b56190ff..89ed1da2c8 (603619:603733) by chromium-webrtc-autoroll · 7 years ago
770c32a Roll chromium_revision 55624cc6cd..03b56190ff (603513:603619) by chromium-webrtc-autoroll · 7 years ago
3b149e4 Added myself to the base watchlist to monitor ssl* changes. by Benjamin Wright · 7 years ago
5124a04 Roll chromium_revision 62e33bd2f0..55624cc6cd (603177:603513) by chromium-webrtc-autoroll · 7 years ago
6b9d823 Add TargetBitrate callback to MediaTransportInterface. by Piotr (Peter) Slatala · 7 years ago
c640a93 Fix import of chromium into webrtc. by Artem Titov · 7 years ago
a0677d1 Add MediaTransportSettings struct for configuring media transport. by Piotr (Peter) Slatala · 7 years ago
12048c7 Fix error handling in hex_decode. by Niels Möller · 7 years ago
ef45669 Adds GetSentPacket to PacketResult. by Sebastian Jansson · 7 years ago
449c1c0 Adds unit tests for safe reset trial. by Sebastian Jansson · 7 years ago
7286496 Download aap2 and bundletool as part of required dependencies. by Yves Gerey · 7 years ago
6fcf6ca Modified PressEnterToContinue() to actualy check if Enter is pressed by Danail Kirov · 7 years ago
2c16cc6 Replace some usage of EventWrapper with rtc::Event. by Niels Möller · 7 years ago
88d8d7d Add missing assignment in RTCConfiguration.mm by Piotr (Peter) Slatala · 7 years ago
f3ff14c Properly setup MockPeerConnectionObserver in tests. by Yves Gerey · 7 years ago
22a8f98 Formatted sslidenty.cc and moved non referenced functions into an by Benjamin Wright · 7 years ago
428320c Formatting OpenSSLCertificate and doing some minor code cleanup. by Benjamin Wright · 7 years ago
5d35554 Rename private member functions to use CamelCase. by Benjamin Wright · 7 years ago
61c5cc8 Makes OpenSSL concrete implementations final. by Benjamin Wright · 7 years ago
2616594 Refactor: Make SSLCertChain a final class. by Benjamin Wright · 7 years ago
150a907 FrameEncryption Video End To End Testcase. by Benjamin Wright · 7 years ago
c462a6e Prevent the frame decryptor being set if the channel is stopped. by Benjamin Wright · 7 years ago
625771d Roll chromium_revision a539a24569..62e33bd2f0 (603045:603177) by chromium-webrtc-autoroll · 7 years ago
59ebf23 Refactor structs in rtc_event_log_parser_new.h by Elad Alon · 7 years ago
ff43541 Delta compression efficiency improvement for non-existent base by Elad Alon · 7 years ago
436ebca Fix extra setdscp call introduced by bad merge. by Tim Haloun · 7 years ago
0f08d22 Add a function for enabling the congestion window and pushback controller in the webrtc::SendSideCongestionController. by erikvarga@webrtc.org · 7 years ago
99b71df Use function_video_(en|de)coder_factory from api by Danil Chapovalov · 7 years ago
88c2c50 Use monotonic clock to derive NTP timestamps in RTCP module by Ilya Nikolaevskiy · 7 years ago
fdee701 Add parser and unittests for new RTC event log format. by Bjorn Terelius · 7 years ago
916ae08 Makes critsect_.Leave() more visible in PacedSender. by Sebastian Jansson · 7 years ago
6dd7f91 Remove deprecated deregistration functions in RtcpTransceiver by Danil Chapovalov · 7 years ago
06aa209 Add support to adapt video without preserving aspect ratio by Magnus Jedvert · 7 years ago
9049037 Simplify api/DEPS presubmit check. by Mirko Bonadei · 7 years ago
7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 7 years ago
2769cd5 Roll chromium_revision f54583b6a0..a539a24569 (602763:603045) by chromium-webrtc-autoroll · 7 years ago
0d24772 Allocate CMBlockBuffers using a memory pool. by Kári Tristan Helgason · 7 years ago
c35096d Reland "Encode RTC event logs in new format." by Bjorn Terelius · 7 years ago
a5d543c Set minSdkVersion to 16 for AppRTCMobile_stubbed_video_io_test_apk. by Mirko Bonadei · 7 years ago
95dfa52 Clarify the desired semantics of AsyncResolverInterface::GetResolvedAddress. by Zach Stein · 7 years ago
327b753 Split out a separate target for VP8EncoderSimulcastProxy by Jonathan Yu · 7 years ago
5abd541 Stop exporting simulcast_encoder_adapter.h in :rtc_internal_video_codecs by Jonathan Yu · 7 years ago
b19b497 Refactor: Removing IgnoreBadCert from SSLStreamAdapter. Make test methods more explicit. by Benjamin Wright · 7 years ago
dcd40ca Roll chromium_revision d68fb50e14..f54583b6a0 (602627:602763) by chromium-webrtc-autoroll · 7 years ago
8c27cca Promotoing webrtc::CryptoOptions to RTCConfiguration. by Benjamin Wright · 7 years ago
78410ad Fixes use after free error when setting a new FrameEncryptor on ChannelSend. by Benjamin Wright · 7 years ago
f26e290 fuchsia: Stub out timing and memory functions by Scott Graham · 7 years ago
9c8ae4b Disable probe delay warning in release builds. by Jamie Walch · 7 years ago
6c6c9df Refactor: Renaming ssl_cert_chain to GetSSLCertificateChain() by Benjamin Wright · 7 years ago
359d60a Adds target rate to audio send stream stats. by Sebastian Jansson · 7 years ago
57ba7e1 Normalize baseline in network delay plot to RTT/2. by Bjorn Terelius · 7 years ago
039743e Reland "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase" by Niels Möller · 7 years ago
e2754c9 Fixes bug in AudioPriorityBitrateAllocationStrategy field trial. by Sebastian Jansson · 7 years ago
c0e4d45 Adds BitrateAllocation struct to OnBitrateUpdated. by Sebastian Jansson · 7 years ago
4ba6c26 Delete MessageData when a message is posted to a quitting MessageQueue by Niels Möller · 7 years ago
9516c38 [Fuzzer] Check FieldTrial bitmask size at compile time. by Yves Gerey · 7 years ago
1803bb2 Fix for clock read race in FakeNetworkPipe. by Christoffer Rodbro · 7 years ago
3284b61 Fix for packet loss tracking in network emulation. by Christoffer Rodbro · 7 years ago
2620470 Update fuzzer max input length handling by Sam Zackrisson · 7 years ago
ddc84e9 Publish function_video_(en|de)coder_factory into api by Danil Chapovalov · 7 years ago
23524ce Add HDR metadata struct by Johannes Kron · 7 years ago
977b46a Export symbols needed by the Chromium component build (part 7). by Mirko Bonadei · 7 years ago
3eb1c72 Removes deprecated BitrateAllocation alias. by Sebastian Jansson · 7 years ago
2506839 Add DCHECK for wrap around in RtpVideoSender::OnBitrateUpdated. by Bjorn Terelius · 7 years ago
370bae4 APM: Adding more explicit handling of failures in the json config data by Per Åhgren · 7 years ago
487e694 Use default value if field trial switch is set to an invalid number by Johannes Kron · 7 years ago
273c851 Remove obsolete android ndk copy from //third_party/android_tools/ndk by Yongje Lee · 7 years ago