Log - 8cc711a7e1fa2adb95cd3596d6a26e624017645d - platform/external/webrtc

8cc711a Update URI of TransportSequenceNumberV2 by Johannes Kron · 5 years ago
aba8dc2 Rename EndpointNode into EmulatedEndpoint by Artem Titov · 5 years ago
2057673 Roll chromium_revision 4029fd18b6..f2f5c1896e (639320:639430) by chromium-webrtc-autoroll · 5 years ago
abea6e5 Delete always-true member is_media_transport_factory_enabled_ by Niels Möller · 5 years ago
c0c3e96 Revert "DCHECK feedback_rtt is positive" by Yves Gerey · 5 years ago
3368721 Revert "Change clip_name -> clip_path in VideoQualityTestFixture::Params::Video." by Yves Gerey · 5 years ago
32f887b Roll chromium_revision e4e8741478..4029fd18b6 (639216:639320) by chromium-webrtc-autoroll · 5 years ago
fe07c42 Roll chromium_revision ecf963f5ce..e4e8741478 (639097:639216) by chromium-webrtc-autoroll · 5 years ago
e25f595 Guard preferred_dscp with the network interface lock by Steve Anton · 5 years ago
9a071d1 Roll chromium_revision e328c33c20..ecf963f5ce (638961:639097) by chromium-webrtc-autoroll · 5 years ago
42d8c93 Revert "Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current" by Yves Gerey · 5 years ago
cad95b8 Reland "Tune vp9 screenshare bitrate and framerate of spatial layers" by Ilya Nikolaevskiy · 5 years ago
62c7b39 Allow suppression of padding check in RtpHeaderParser. by Sebastian Jansson · 5 years ago
44dd9f2 Adds ChannelSend specific encoder task queue. by Sebastian Jansson · 5 years ago
e01857c Revert "Reland "Tune vp9 screenshare bitrate and framerate of spatial layers"" by Ilya Nikolaevskiy · 5 years ago
12abf67 Reland "Tune vp9 screenshare bitrate and framerate of spatial layers" by Ilya Nikolaevskiy · 5 years ago
304e9d2 Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current by Danil Chapovalov · 5 years ago
37d4f91 DCHECK feedback_rtt is positive by Evan Shrubsole · 5 years ago
184f6d5 Change clip_name -> clip_path in VideoQualityTestFixture::Params::Video. by Rasmus Brandt · 5 years ago
159e53a Fix LibvpxVp8Encoder::FrameDropThreshold by Elad Alon · 5 years ago
dac7aa0 Roll chromium_revision c71bb6f5f1..e328c33c20 (638607:638961) by chromium-webrtc-autoroll · 5 years ago
f0cbcd3 Use stdlib TaskQueue implementation in webrtc fuzzers by Danil Chapovalov · 5 years ago
3caf50d Make ChangeBitrateVP9 unittest a bit more lenient. by Yves Gerey · 5 years ago
7f1c589 Adding new top-level directory crypto/ by Benjamin Wright · 5 years ago
1109b59 Revert "Tune vp9 screenshare bitrate and framerate of spatial layers" by Jeroen de Borst · 5 years ago
2c7b982 Revert "Delete CodecSpecificInfo argument from VideoDecoder::Decode" by Jeroen de Borst · 5 years ago
7e70291 Fix unscoped variable in test/scenario/BUILD.gn. by Mirko Bonadei · 5 years ago
74350db Roll chromium_revision 4205483be6..c71bb6f5f1 (638505:638607) by chromium-webrtc-autoroll · 5 years ago
aaf3cb3 Tune vp9 screenshare bitrate and framerate of spatial layers by Ilya Nikolaevskiy · 5 years ago
39d3a7d Delete CodecSpecificInfo argument from VideoDecoder::Decode by Niels Möller · 5 years ago
1c90cab Fix UpdateRect handling for native buffers in VideoStreamEncoder by Ilya Nikolaevskiy · 5 years ago
6f0aafa Add PrintResults to VideoCodecTest. by Rasmus Brandt · 5 years ago
d5af402 Add overhead observers to MediaTransportInterface by Niels Möller · 5 years ago
06b77f9 Use min allocatable bitrate as lower bound for target bitrate. by Sebastian Jansson · 5 years ago
ffe9376 Bump iOS min supported version to 10.0 by Kári Tristan Helgason · 5 years ago
b859b32 Update more VideoEncoder implementations to drop CodecSpecificInfo input by Niels Möller · 5 years ago
6318f13 Stop using rtc::TaskQueue::Current in RtcpTransceiver by Danil Chapovalov · 5 years ago
dc62ae4 Cleanup of constraints configuration in GoogCcNetworkController. by Sebastian Jansson · 5 years ago
78b7d49 Roll chromium_revision 1af146a0f6..4205483be6 (638325:638505) by chromium-webrtc-autoroll · 5 years ago
8a1e35c Finally delete deprecated mac capturer. by Kári Tristan Helgason · 5 years ago
87e2d78 Prepare for splitting FrameType into AudioFrameType and VideoFrameType by Niels Möller · 5 years ago
0b69826 Don't inject worker queue into send streams. by Sebastian Jansson · 5 years ago
de3360e Create Vp8FrameBufferController by Elad Alon · 5 years ago
610c763 Add target bitrate headroom signal to VideoStreamEncoder. by Erik Språng · 5 years ago
e49d64e Roll chromium_revision 3eb6e6ce76..1af146a0f6 (638159:638325) by chromium-webrtc-autoroll · 5 years ago
7276b97 Disable DTLS 1.0, TLS 1.0 and TLS 1.1 downgrade in WebRTC. by Benjamin Wright · 5 years ago
4423c36 Migrate RepeatingTask to take raw pointer to TaskQueueBase instead of TaskQueue by Danil Chapovalov · 5 years ago
11e55ee Renaming min_pacing_rate to min_total_allocated_bitrate. by Sebastian Jansson · 5 years ago
7b41225 Throttle frame-rate In VP8 encoder in steady state for screenshare by Ilya Nikolaevskiy · 5 years ago
2ecc8c8 Roll chromium_revision 99baeeafe2..3eb6e6ce76 (638035:638159) by chromium-webrtc-autoroll · 5 years ago
8672cac Trigger audio bitrate allocation update on overhead change. by Sebastian Jansson · 5 years ago
ee5ccbc Move ownership of RTPSenderAudio to ChannelSend. by Niels Möller · 5 years ago
232b3fd Expose relative packet arrival delay metric in stats API. by Jakob Ivarsson · 5 years ago
67f862e Guard against calls to OnEncodedFrame after Release. by Sami Kalliomäki · 5 years ago
6117068 Throttle frame-rate In VP9 encoder in steady state for screenshare by Ilya Nikolaevskiy · 5 years ago
0cb858c New VCMPacket constructor without WebRtcRTPHeader argument by Niels Möller · 5 years ago
7bc331f Android: Use android_deps directly by Peter Wen · 5 years ago
c44f6cc Modernize RtpRtcp factory function: use unique_ptr as return type by Danil Chapovalov · 5 years ago
ede7cb2 Rewrite video_loopback to use new mac capturer. by Kári Tristan Helgason · 5 years ago
c8d2e73 Delete CodecSpecificInfo argument from VideoEncoder::Encode by Niels Möller · 5 years ago
1e42761 Removes verbose extension warning in Scenario tests. by Sebastian Jansson · 5 years ago
110c64b Delete unused key WebRTC-Audio-SendSideBwe-For-Video. by Christoffer Rodbro · 5 years ago
745cfb9 use link_deps in ana_debug_dump_proto by Takuto Ikuta · 5 years ago
d738071 Refactor FakeEncoder to avoid writing to a const EncodedImage by Niels Möller · 5 years ago
9df3353 Generic Frame Descriptor (GFD) VP8 templates. by philipel · 5 years ago
8fb1a6a Delete a few return values from audio streams and video send streams. by Niels Möller · 5 years ago
0da25a1 Update TransportSequenceNumberV2 extension to support fixed size by Johannes Kron · 5 years ago
f441ea9 Minor cleanup of probe_controller.cc by Jonas Olsson · 5 years ago
200feba Make AEC3 the default desktop AEC option in WebRTC by Per Åhgren · 5 years ago
359c9b9 Roll chromium_revision 49f30ad2d0..99baeeafe2 (637641:638035) by chromium-webrtc-autoroll · 5 years ago
be7af93 Add dsymutil as a mac cipd dependency. by Yves Gerey · 5 years ago
b443dfe Use metrics::Samples in a couple pc/ tests by Steve Anton · 5 years ago
e2a284d Adding metrics to measure usage of simulcast API. by Amit Hilbuch · 5 years ago
4eb5c14 Reland "Remove field trial include from decision logic." by Jakob Ivarsson · 5 years ago
07a4f2b Merge rtc_task_queue(_api|_impl)? build targets into one by Danil Chapovalov · 5 years ago
4450708 Add relative_packet_arrival_delay and jitter_buffer_packets_received statistics. by Jakob Ivarsson · 5 years ago
1aa7581 Replace all usage of rtc::NewClosure with webrtc::ToQueuedTask by Danil Chapovalov · 5 years ago
c1e6e86 Add metrics::Samples to facilitate easier testing by Steve Anton · 5 years ago
d36c086 Add support for simulcast streams in QualityAnalyzingVideoDecoder. by Artem Titov · 5 years ago
6ec2f54 Fix mis-spelled TODO items by Niels Möller · 5 years ago
7949f21 Revert "Removes lock from ChannelSend." by Sebastian Jansson · 5 years ago
9ef5e05 Fix target bitrate handling for a single layer VP9 screenshare by Ilya Nikolaevskiy · 5 years ago
977b335 Injecting Clock into audio streams. by Sebastian Jansson · 5 years ago
f23f161 Roll chromium_revision 5fe3fd14f6..49f30ad2d0 (637538:637641) by chromium-webrtc-autoroll · 5 years ago
423bae4 Revert "Remove field trial include from decision logic." by Jakob Ivarsson‎ · 5 years ago
bd7ed4b Include sign for infinity in ToString for data units. by Sebastian Jansson · 5 years ago
44ce4b4 Adding a placeholder audio_buffer build target by Per Åhgren · 5 years ago
15845af Reland "Another mock for GetSctpTransport" (and add test) by Harald Alvestrand · 5 years ago
9b93447 Removes lock from ChannelSend. by Sebastian Jansson · 5 years ago
c8eeb18 Fixes parsing of logs where receive time info is missing. by Sebastian Jansson · 5 years ago
07316a6 Roll chromium_revision 4229a4b64d..5fe3fd14f6 (637436:637538) by chromium-webrtc-autoroll · 5 years ago
1f8e445 Roll chromium_revision 5afa522447..4229a4b64d (637301:637436) by chromium-webrtc-autoroll · 5 years ago
572c60f Injecting Clock into video senders. by Sebastian Jansson · 5 years ago
8026d60 Injecting Clock in video receive. by Sebastian Jansson · 5 years ago
ef50b25 Remove lock in WebRtcVideoEngine by Steve Anton · 5 years ago
4cde9ad Fix some typos found in ivf_file_writer.cc by Elad Alon · 5 years ago
4e5f5ed Allow Clock injection in Call. by Sebastian Jansson · 5 years ago
5fe9510 Move ownership of RTPSenderVideo one more level up, to RtpVideoSender by Niels Möller · 5 years ago
ac6cf7f Roll chromium_revision e65d7afd91..5afa522447 (637200:637301) by chromium-webrtc-autoroll · 5 years ago
da6806c Injecting Clock into BitrateAllocator. by Sebastian Jansson · 5 years ago