Log - 7fa42778b4fc87d4319f418b2c7286e03c482ff7 - platform/external/webrtc

7fa4277 Fix for tsan failue in real time scenario tests. by Sebastian Jansson · 5 years ago
6dcd4dc New target for api/rtp_parameters.h and api/media_types.h. by Niels Möller · 5 years ago
7db900e Simplify pacer queue by Erik Språng · 5 years ago
228900f Add TURN_LOGGING_ID to android sdk by Jonas Oreland · 5 years ago
03307cb Roll chromium_revision acc73a9128..9dd4f35a9d (691348:691474) by chromium-webrtc-autoroll · 5 years ago
5245501 Roll chromium_revision 8538f0b743..acc73a9128 (691247:691348) by chromium-webrtc-autoroll · 5 years ago
703ea95 Only create a datagram RTP transport if the datagram transport should be used for RTP. by Bjorn A Mellem · 5 years ago
dbec6d3 Roll chromium_revision f706cf738b..8538f0b743 (690793:691247) by chromium-webrtc-autoroll · 5 years ago
fa046b3 Remove unused using statements in webrtc_sdp.cc by Elad Alon · 5 years ago
10b6361 Add license for android_ndk by Oleksandr Iakovenko · 5 years ago
d191533 Fix wrong-import-order pylint errors in quality_assessment.signal_processing module. by Oleksandr Iakovenko · 5 years ago
4b9701e Fix simulcast tests and PC framework for conference mode support by Artem Titov · 5 years ago
149dc72 Add support for RTCTransportStats.selectedCandidatePairChanges by Jonas Oreland · 5 years ago
3b69817 Revert "Reland "Preserve min and max playout delay from RTP header extension"" by Johannes Kron · 5 years ago
87bed47 Reland "Preserve min and max playout delay from RTP header extension" by Johannes Kron · 5 years ago
f31cc08 Revert "Preserve min and max playout delay from RTP header extension" by Johannes Kron · 5 years ago
fdd2340 Revert "Detect leaks of TextureBufferImpl objects." by Sami Kalliomäki · 5 years ago
050e38f Add --trace_event option to capture events in unit tests. by Yves Gerey · 5 years ago
7f65932 Fix for sanitizer bot failure in AudioUsesAbsSendTimeExtension by Sebastian Jansson · 5 years ago
85ba997 Preserve min and max playout delay from RTP header extension by Johannes Kron · 5 years ago
5e8ddc3 Reland "Delete mac_utils.h and mac_utils.cc" by Niels Möller · 5 years ago
fac7e31 Removes TransportSequenceNumberAllocator by Erik Språng · 5 years ago
a370556 Refactor to free up PacketBuffer as soon as possible by Johannes Kron · 5 years ago
caef51e Consolidate FEC book-keeping by Niels Möller · 5 years ago
2d5aec5 Roll chromium_revision abb1ee24a4..f706cf738b (690691:690793) by chromium-webrtc-autoroll · 5 years ago
70768f4 Remove usage of StorageType enum by Erik Språng · 5 years ago
44bd29a Detect leaks of TextureBufferImpl objects. by Sami Kalliomäki · 5 years ago
de21bf4 Roll chromium_revision 925c16d3e7..abb1ee24a4 (690586:690691) by chromium-webrtc-autoroll · 5 years ago
4271afb Fix the bug and reland "Make min video target bitrate configurable." by Ying Wang · 5 years ago
30ab015 BalancedDegradationSettings: add min bitrate configuration for resolution. by Åsa Persson · 5 years ago
31d1bce Fix deadlock in VideoSendStream tests, cause of flake on some bots. by Tommi · 5 years ago
0c141c5 Fix frames dropped statistics by Johannes Kron · 5 years ago
7e896d0 Revert "Make min video target bitrate configurable." by Mirko Bonadei · 5 years ago
3c02842 Add TURN_LOGGING_ID by Jonas Oreland · 5 years ago
0949c89 Roll chromium_revision c7011257bb..925c16d3e7 (690474:690586) by chromium-webrtc-autoroll · 5 years ago
f5e0e50 Roll chromium_revision 004b50827c..c7011257bb (690310:690474) by chromium-webrtc-autoroll · 5 years ago
1fbfecd Use a pass-through pacer instead of special-cased pacerless mode by Erik Språng · 5 years ago
c15f92a Cleanup, remove media_send_ssrc field by Erik Språng · 5 years ago
8a61d0f Remove deprecated RTPSender ctor variant by Erik Språng · 5 years ago
adfb4f7 Add ability to parse stable bwe experiment settings by Erik Språng · 5 years ago
a471e79 Make min video target bitrate configurable. by Ying Wang · 5 years ago
3b407ff Tune qp threshold for VP9 blocky video by Ilya Nikolaevskiy · 5 years ago
4869bd6 Add method CanAdaptUp based on bitrate to BalancedDegradationSettings. by Åsa Persson · 5 years ago
4208a13 Removes deprecated InsertPacket/TimeToSendPacket/TimeToSendPadding by Erik Språng · 5 years ago
6558fa5 Reintroduce command line controlled reference data updating for ApmTest.Process by Sam Zackrisson · 5 years ago
5cdd226 Roll chromium_revision 318f298cba..004b50827c (688507:690310) by chromium-webrtc-autoroll · 5 years ago
2ca0b36 Correct the handling of sample rates that don't scale well into even 10 ms chunks by Per Åhgren · 5 years ago
1fda027 [vp9] Array temporal_up_switch wasn't properly initialized. by Yves Gerey · 5 years ago
184b4af New empty build target api:rtp_parameters by Niels Möller · 5 years ago
0aefbf0 Use the AEC3 high-pass filter for the whole APM by Per Åhgren · 5 years ago
c8626b6 Reland "Reland Process 8 kHz audio as 16 kHz internally of the audio processing module" by Per Åhgren · 5 years ago
7c4b0c5 Revert "Reland Process 8 kHz audio as 16 kHz internally of the audio processing module" by Artem Titarenko · 5 years ago
6e706ed Add ObjC interface wrapping new GetImplementations method. by Kári Tristan Helgason · 5 years ago
b6b4dee Fix flake in SamplesStatsCounterTest.FullSimpleTest by Artem Titov · 5 years ago
bf45add Set required alignment to 2 for iOS. by Kári Tristan Helgason · 5 years ago
b7b8e30 Reland Process 8 kHz audio as 16 kHz internally of the audio processing module by Per Åhgren · 5 years ago
d77cc24 New const method StreamStatistician::GetStats by Niels Möller · 5 years ago
74154e6 Save delays in history for 2 seconds instead of fixed 100 packets. by Jakob Ivarsson · 5 years ago
4e615d5 Wire the stable target bitrate from GoogCC to the BitrateAllocator by Florent Castelli · 5 years ago
3dd1985 Delete unused function MediaTypeFromString by Niels Möller · 5 years ago
b88b44e Don't include duplicated and incomplete frames in stats. by Johannes Kron · 5 years ago
d47941e Reland "Simplification and refactoring of the AudioBuffer code" by Per Åhgren · 5 years ago
a2dae38 Revert "Reland "Delete mac_utils.h and mac_utils.cc"" by Niels Moller · 5 years ago
05f8f1d Add helper classes to send and receive abs-capture-time extensions. by Chen Xing · 5 years ago
9fd2908 Remove unused framerate parameter from simulcast bitrate allocator. by Jonas Olsson · 5 years ago
df57833 Reland "Delete mac_utils.h and mac_utils.cc" by Niels Möller · 5 years ago
224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 5 years ago
b689af4 Changes to enable use of DatagramTransport as a data channel transport. by Bjorn A Mellem · 5 years ago
f254e9e Revert "Simplification and refactoring of the AudioBuffer code" by Steve Anton · 5 years ago
f5815fa Remove WebRTC-Pacer-LegacyPacketReferencing flag and most usage by Erik Språng · 5 years ago
1c602e3 Process 8 kHz audio as 16 kHz internally of the audio processing module by Per Åhgren · 5 years ago
81c0cf2 Simplification and refactoring of the AudioBuffer code by Per Åhgren · 5 years ago
f69bd5f Delete AudioDeviceWindowsCore::WideToUTF8, replaced with rtc::ToUtf8 by Niels Möller · 5 years ago
70efdde Set local ssrc at construction of Rtp module by Erik Språng · 5 years ago
21e99da Add implemented-but-missing members to RTCMediaStreamTrackStats::Members by Henrik Boström · 5 years ago
1c2f637 Simplify the VideoFrameDumpingDecoder API. by Markus Handell · 5 years ago
54d5d2c Rename RtpRtcp::Configuration::media_send_ssrc to local_media_ssrc by Erik Språng · 5 years ago
e8ef87b Include menus & dialogs in frames captured by WindowCapturerWin by Bryan Ferguson · 5 years ago
364b267 Replace DatagramDtlsAdaptor with DatagramRtpTransport. by Bjorn A Mellem · 5 years ago
a310b38 Roll chromium_revision 5a34954f26..318f298cba (688384:688507) by chromium-webrtc-autoroll · 5 years ago
2dac4e4 Remove rtc_use_lto GN arg. by Mirko Bonadei · 5 years ago
5ceb4ac Delete some unused AudioCodingModule methods by Niels Möller · 5 years ago
728a0ee Reland "Introduce ability to test echo in PC level test framework" by Artem Titov · 5 years ago
a854921 Enable custom metrics gathering from stats API in PC framework. by Artem Titov · 5 years ago
e21f3f5 Revert "Delete mac_utils.h and mac_utils.cc" by Niels Moller · 5 years ago
6b117a5 Make the callbacks to PollStats for RampUp* tests more regular. by Tommi · 5 years ago
ada8e17 Delete mac_utils.h and mac_utils.cc by Niels Möller · 5 years ago
928146f Removing all external access to the integer sample data in AudioBuffer by Per Åhgren · 5 years ago
93d4c10 Declare references as constant in the metal renderers. by Kári Tristan Helgason · 5 years ago
2579f0c RTCError as return type for PeerConnectionInterface::SetConfiguration by Niels Möller · 5 years ago
7627fdd Sanitize the address field of peer-reflexive remote candidates. by Qingsi Wang · 5 years ago
7c78e42 Roll chromium_revision 21d23ea529..5a34954f26 (688221:688384) by chromium-webrtc-autoroll · 5 years ago
587991c Remove jeroendb@webrtc.org from OWNERS by Steve Anton · 5 years ago
a0b52b5 Remove zhihuang@webrtc.org from OWNERS by Steve Anton · 5 years ago
9bdb1b1 Roll chromium_revision afb0a631b9..21d23ea529 (688061:688221) by chromium-webrtc-autoroll · 5 years ago
1ba5dec Reland "Set the usage pattern bits for adding remote ICE candidates from SDP." by Qingsi Wang · 5 years ago
0d1996f Removes empty p2p/base/transport.h by Sebastian Jansson · 5 years ago
fdf3880 Make "WebRTC-BweAllocProbingOnlyInAlr/Enabled/" default and remove key. by Konrad Hofbauer · 5 years ago
e3a10e1 Remove usage of RtpRtcp::SetSSRC() in video/ by Erik Språng · 5 years ago
185243b Remove most of PacedSenderUnittest by Erik Språng · 5 years ago