Log - 1175ae0d800e10ebd021b9998d47c0f6e18ad899 - platform/external/webrtc

1175ae0 Add log based GoogCC simulation to visualizer. by Sebastian Jansson · 6 years ago
7ae8d64 Restore VideoCodecInitializer to use only the 1st stream maxFramerate by Ilya Nikolaevskiy · 6 years ago
77efcd8 Reland "Replacing rtc::Thread with task queue for TestAudioDeviceModule." by Sebastian Jansson · 6 years ago
793597b Removes TaskQueueBase::Current call in repeating task. by Sebastian Jansson · 6 years ago
cda86dd Removes usages of repeating task without task queue argument. by Sebastian Jansson · 6 years ago
e7a5f7b Modifying MediaChannel to accept CopyOnWriteBuffer by value. by Amit Hilbuch · 6 years ago
dfaea9d Fuzz rtc::StringToNumber. by Benjamin Wright · 6 years ago
6a5e976 Add generic depacketizer fuzzer to WebRTC. by Benjamin Wright · 6 years ago
ade5cb8 Field trial fuzzer. by Benjamin Wright · 6 years ago
fa852ef Revert "Replacing rtc::Thread with task queue for TestAudioDeviceModule." by Seth Hampson · 6 years ago
bd50a84 Revert "Reland "DCHECK feedback_rtt is positive"" by Seth Hampson · 6 years ago
ea07650 Roll chromium_revision 8158f07c24..48038209dc (639862:640007) by chromium-webrtc-autoroll · 6 years ago
1b871d0 Replacing rtc::Thread with task queue for TestAudioDeviceModule. by Sebastian Jansson · 6 years ago
ab0d03d Reland "DCHECK feedback_rtt is positive" by Evan Shrubsole · 6 years ago
ec65e1f Allow construction of TaskQueueForTest with TaskQueueBase by Sebastian Jansson · 6 years ago
1b4254a Check current buffer time span instead of number of samples in postpone decoding after expand. by Jakob Ivarsson · 6 years ago
075e7fd Delete VCMPacket constructor with WebRtcRTPHeader by Niels Möller · 6 years ago
c4b391a Revert "NetEQ RTP Play: Optionally write output audio file" by Ivo Creusen · 6 years ago
e096004 Enable configuring probes via field trial. by Jonas Olsson · 6 years ago
cb96809 Make FieldTrialOptionals operator bool() explicit by Jonas Olsson · 6 years ago
bf40c38 Pass flexfec_sender only to the protected media send stream. by Niels Möller · 6 years ago
fc6ab00 Introduce EmulatedRoute by Artem Titov · 6 years ago
a268b69 Rename EndpointConfig into EmulatedEndpointConfig by Artem Titov · 6 years ago
30e60d6 Remove dependency on DirectShow baseclasses (streams.h from the winsdk_samples directory). by Tommi · 6 years ago
2594f27 Change some RTC_DCHECKs to RTC_DCHECK_EQ by Ruslan Burakov · 6 years ago
4f779c6 Remove legacy empty task_queue BUILD targets and build arg by Danil Chapovalov · 6 years ago
70a8394 Delete use of WebRtcRTPHeader from FEC test code by Niels Möller · 6 years ago
2eb54a4 Roll chromium_revision ad6c7f0653..8158f07c24 (639761:639862) by chromium-webrtc-autoroll · 6 years ago
6330818 NetEQ RTP Play: Optionally write output audio file by Alessio Bazzica · 6 years ago
dba16fd Roll chromium_revision 02df973441..ad6c7f0653 (639636:639761) by chromium-webrtc-autoroll · 6 years ago
b935d48 Roll chromium_revision 20749e92ed..02df973441 (639533:639636) by chromium-webrtc-autoroll · 6 years ago
7b0b966 Roll chromium_revision f2f5c1896e..20749e92ed (639430:639533) by chromium-webrtc-autoroll · 6 years ago
471783f Remove rtc::QueuedTask alias, use webrtc::QueuedTask directly by Danil Chapovalov · 6 years ago
125b5d6 Refactor RtpVideoStreamReceiver::OnReceivedPayloadData without WebRtcRTPHeader by Niels Möller · 6 years ago
d155d68 Removes rtp level keep alive support. by Sebastian Jansson · 6 years ago
9ffb5df Removes unused mock_bitrate_controller. by Sebastian Jansson · 6 years ago
d71edac Add an input size limit to APM fuzzer by Sam Zackrisson · 6 years ago
a5c0ba1 Reland "Fix LibvpxVp8Encoder::FrameDropThreshold" by Elad Alon · 6 years ago
e448a3f Update DataChannel bufferedamount implementation. by Marina Ciocea · 6 years ago
ad89528 Reland "Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current" by Danil Chapovalov · 6 years ago
55500d2 Revert "Fix LibvpxVp8Encoder::FrameDropThreshold" by Yves Gerey · 6 years ago
8a0c1f5 Don't reset bitrate when allocatable minimum changes. by Sebastian Jansson · 6 years ago
effdfe2 Move dependency on chromium DefaultTaskQueueFactory by Danil Chapovalov · 6 years ago
d4a37a6 Support absl::string_view in RTC_CHECK. by Mirko Bonadei · 6 years ago
8cc711a Update URI of TransportSequenceNumberV2 by Johannes Kron · 6 years ago
aba8dc2 Rename EndpointNode into EmulatedEndpoint by Artem Titov · 6 years ago
2057673 Roll chromium_revision 4029fd18b6..f2f5c1896e (639320:639430) by chromium-webrtc-autoroll · 6 years ago
abea6e5 Delete always-true member is_media_transport_factory_enabled_ by Niels Möller · 6 years ago
c0c3e96 Revert "DCHECK feedback_rtt is positive" by Yves Gerey · 6 years ago
3368721 Revert "Change clip_name -> clip_path in VideoQualityTestFixture::Params::Video." by Yves Gerey · 6 years ago
32f887b Roll chromium_revision e4e8741478..4029fd18b6 (639216:639320) by chromium-webrtc-autoroll · 6 years ago
fe07c42 Roll chromium_revision ecf963f5ce..e4e8741478 (639097:639216) by chromium-webrtc-autoroll · 6 years ago
e25f595 Guard preferred_dscp with the network interface lock by Steve Anton · 6 years ago
9a071d1 Roll chromium_revision e328c33c20..ecf963f5ce (638961:639097) by chromium-webrtc-autoroll · 6 years ago
42d8c93 Revert "Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current" by Yves Gerey · 6 years ago
cad95b8 Reland "Tune vp9 screenshare bitrate and framerate of spatial layers" by Ilya Nikolaevskiy · 6 years ago
62c7b39 Allow suppression of padding check in RtpHeaderParser. by Sebastian Jansson · 6 years ago
44dd9f2 Adds ChannelSend specific encoder task queue. by Sebastian Jansson · 6 years ago
e01857c Revert "Reland "Tune vp9 screenshare bitrate and framerate of spatial layers"" by Ilya Nikolaevskiy · 6 years ago
12abf67 Reland "Tune vp9 screenshare bitrate and framerate of spatial layers" by Ilya Nikolaevskiy · 6 years ago
304e9d2 Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current by Danil Chapovalov · 6 years ago
37d4f91 DCHECK feedback_rtt is positive by Evan Shrubsole · 6 years ago
184f6d5 Change clip_name -> clip_path in VideoQualityTestFixture::Params::Video. by Rasmus Brandt · 6 years ago
159e53a Fix LibvpxVp8Encoder::FrameDropThreshold by Elad Alon · 6 years ago
dac7aa0 Roll chromium_revision c71bb6f5f1..e328c33c20 (638607:638961) by chromium-webrtc-autoroll · 6 years ago
f0cbcd3 Use stdlib TaskQueue implementation in webrtc fuzzers by Danil Chapovalov · 6 years ago
3caf50d Make ChangeBitrateVP9 unittest a bit more lenient. by Yves Gerey · 6 years ago
7f1c589 Adding new top-level directory crypto/ by Benjamin Wright · 6 years ago
1109b59 Revert "Tune vp9 screenshare bitrate and framerate of spatial layers" by Jeroen de Borst · 6 years ago
2c7b982 Revert "Delete CodecSpecificInfo argument from VideoDecoder::Decode" by Jeroen de Borst · 6 years ago
7e70291 Fix unscoped variable in test/scenario/BUILD.gn. by Mirko Bonadei · 6 years ago
74350db Roll chromium_revision 4205483be6..c71bb6f5f1 (638505:638607) by chromium-webrtc-autoroll · 6 years ago
aaf3cb3 Tune vp9 screenshare bitrate and framerate of spatial layers by Ilya Nikolaevskiy · 6 years ago
39d3a7d Delete CodecSpecificInfo argument from VideoDecoder::Decode by Niels Möller · 6 years ago
1c90cab Fix UpdateRect handling for native buffers in VideoStreamEncoder by Ilya Nikolaevskiy · 6 years ago
6f0aafa Add PrintResults to VideoCodecTest. by Rasmus Brandt · 6 years ago
d5af402 Add overhead observers to MediaTransportInterface by Niels Möller · 6 years ago
06b77f9 Use min allocatable bitrate as lower bound for target bitrate. by Sebastian Jansson · 6 years ago
ffe9376 Bump iOS min supported version to 10.0 by Kári Tristan Helgason · 6 years ago
b859b32 Update more VideoEncoder implementations to drop CodecSpecificInfo input by Niels Möller · 6 years ago
6318f13 Stop using rtc::TaskQueue::Current in RtcpTransceiver by Danil Chapovalov · 6 years ago
dc62ae4 Cleanup of constraints configuration in GoogCcNetworkController. by Sebastian Jansson · 6 years ago
78b7d49 Roll chromium_revision 1af146a0f6..4205483be6 (638325:638505) by chromium-webrtc-autoroll · 6 years ago
8a1e35c Finally delete deprecated mac capturer. by Kári Tristan Helgason · 6 years ago
87e2d78 Prepare for splitting FrameType into AudioFrameType and VideoFrameType by Niels Möller · 6 years ago
0b69826 Don't inject worker queue into send streams. by Sebastian Jansson · 6 years ago
de3360e Create Vp8FrameBufferController by Elad Alon · 6 years ago
610c763 Add target bitrate headroom signal to VideoStreamEncoder. by Erik Språng · 6 years ago
e49d64e Roll chromium_revision 3eb6e6ce76..1af146a0f6 (638159:638325) by chromium-webrtc-autoroll · 6 years ago
7276b97 Disable DTLS 1.0, TLS 1.0 and TLS 1.1 downgrade in WebRTC. by Benjamin Wright · 6 years ago
4423c36 Migrate RepeatingTask to take raw pointer to TaskQueueBase instead of TaskQueue by Danil Chapovalov · 6 years ago
11e55ee Renaming min_pacing_rate to min_total_allocated_bitrate. by Sebastian Jansson · 6 years ago
7b41225 Throttle frame-rate In VP8 encoder in steady state for screenshare by Ilya Nikolaevskiy · 6 years ago
2ecc8c8 Roll chromium_revision 99baeeafe2..3eb6e6ce76 (638035:638159) by chromium-webrtc-autoroll · 6 years ago
8672cac Trigger audio bitrate allocation update on overhead change. by Sebastian Jansson · 6 years ago
ee5ccbc Move ownership of RTPSenderAudio to ChannelSend. by Niels Möller · 6 years ago
232b3fd Expose relative packet arrival delay metric in stats API. by Jakob Ivarsson · 6 years ago
67f862e Guard against calls to OnEncodedFrame after Release. by Sami Kalliomäki · 6 years ago
6117068 Throttle frame-rate In VP9 encoder in steady state for screenshare by Ilya Nikolaevskiy · 6 years ago
0cb858c New VCMPacket constructor without WebRtcRTPHeader argument by Niels Möller · 6 years ago