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  • d5b0199959 Roll chromium_revision 4a5a62a362..bbca8ebcc5 (887362:887470) chromium-webrtc-autoroll 2021-05-27 23:01:17 -07:00
  • 3d46d0b200 Proxy: solve event tracing with compile time strings. Markus Handell 2021-05-27 21:42:57 +02:00
  • d325f322c7 Update WebRTC code version (2021-05-28T04:03:27). webrtc-version-updater 2021-05-27 21:03:29 -07:00
  • 319bac634b Roll chromium_revision f713d4fb04..4a5a62a362 (887229:887362) chromium-webrtc-autoroll 2021-05-27 15:02:34 -07:00
  • d595f6c02a Roll chromium_revision a0132a2044..f713d4fb04 (887053:887229) chromium-webrtc-autoroll 2021-05-27 11:02:28 -07:00
  • c39080c883 Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions Artem Titov 2021-05-27 17:08:49 +02:00
  • 236ac50628 dcsctp: Add public API for BufferedAmountLow Victor Boivie 2021-05-20 19:34:18 +02:00
  • bd9031bf22 dcsctp: Add OnTotalBufferedAmountLow in Send Queue Victor Boivie 2021-05-26 19:48:55 +02:00
  • 791adafa09 dcsctp: Add OnBufferedAmountLow in Send Queue Victor Boivie 2021-05-20 17:21:24 +02:00
  • a1b8201009 Move proxies into pc/. Markus Handell 2021-05-26 18:56:30 +02:00
  • 7d2e669a38 dcsctp: Allocate TSN for end of abandoned message Victor Boivie 2021-05-26 23:46:41 +02:00
  • 9700d88b1a dcsctp: Avoid recalculation of outstanding bytes Victor Boivie 2021-05-22 22:25:17 +02:00
  • 36ad60613d Update WebRTC code version (2021-05-27T04:02:45). webrtc-version-updater 2021-05-26 21:02:47 -07:00
  • e52cfab633 PipeWire capturer: request mouse cursor to be part of the stream Jan Grulich 2021-04-09 15:53:57 +02:00
  • 2182096e66 RtpFrameReferenceFinder return frames directly instead of via callback. philipel 2021-05-25 15:35:57 +02:00
  • 7f11067110 Clean up RtpSenderTest and remove RtpSenderEgress dependencies. Erik Språng 2021-05-26 16:41:20 +02:00
  • b4f3204a23 Remove Win UWP mb config Christoffer Jansson 2021-05-26 14:18:05 +02:00
  • 940108b04f Apply autoformat to the docs Artem Titov 2021-05-26 12:47:34 +02:00
  • 8f8bf252e6 Remove usage of InjectPacket and transport_ in rtp_sender_unittest Erik Språng 2021-05-26 12:29:22 +02:00
  • b412efdb78 payload type mapping: restrict lower range to <= 63 Philipp Hancke 2021-05-25 09:03:19 +02:00
  • a9af50f151 Introduce CreateDataChannelOrError Harald Alvestrand 2021-05-21 13:33:51 +00:00
  • 0d0ed76ac1 Fix RTP header extension encryption Lennart Grahl 2021-05-17 16:06:37 +02:00
  • 4a54be7118 doc: update dtls_transport.md to use new link style Philipp Hancke 2021-05-17 10:20:14 +02:00
  • fec79b74a8 add srtp docs Philipp Hancke 2021-05-17 10:06:58 +02:00
  • 770acabd5d Refactor mid/rid rtp tests to avoid using egress/transport logic. Erik Språng 2021-05-26 10:01:17 +02:00
  • a39d96666b Remove unused property isLocked from RTCAudioSession Byoungchan Lee 2021-04-15 07:54:38 +09:00
  • 8d9d575920 PipeWire capturer: fix stream width in PW 0.2 code Jan Grulich 2021-04-19 18:19:37 +02:00
  • 27df00746a Update WebRTC code version (2021-05-26T04:05:14). webrtc-version-updater 2021-05-25 21:05:16 -07:00
  • 048bf187d1 Roll chromium_revision 8a4c5eb899..19159a8788 (886374:886529) chromium-webrtc-autoroll 2021-05-25 17:02:32 -07:00
  • 0a52ede821 Support for map of string keys to uint64_t / double values in RTCStats Byoungchan Lee 2021-05-22 08:41:02 +09:00
  • cbeff55a6a Roll chromium_revision 1b27d646a6..8a4c5eb899 (886225:886374) chromium-webrtc-autoroll 2021-05-25 11:01:27 -07:00
  • 157371643d Enforce thread invoke policy for invokes to itself Artem Titov 2021-05-25 11:17:07 +02:00
  • 4fbc3fc59e Move SendPacketUpdates* tests to rtp_sender_egress_unittest. Erik Språng 2021-05-25 17:00:26 +02:00
  • fade919bb1 Partial revert: "Use unordered map in RtpDemuxer" Victor Boivie 2021-05-25 13:34:07 +02:00
  • 238da9a57e Remove obsolete SendPacketMatches* tests from rtp_sender_egress_unittest. Erik Språng 2021-05-25 12:58:43 +02:00
  • 552169c7db Refactor RtpPacketCounter tests and move to rtp_sender_egress_unittest. Erik Språng 2021-05-25 14:03:38 +02:00
  • af0dff0c7d dcsctp: start SCTP_DUMP on a new line Philipp Hancke 2021-05-25 13:41:28 +02:00
  • 2ab4764b9e Clean-up for calculation of upper bandwidth limit. Christoffer Rodbro 2021-05-25 11:16:42 +02:00
  • 81e13d3d6e Roll chromium_revision 2826799ea1..1b27d646a6 (885837:886225) chromium-webrtc-autoroll 2021-05-25 03:02:59 -07:00
  • 36005afeb4 Refactor and improve RtpSender packet history test. Erik Språng 2021-05-24 18:50:04 +02:00
  • 02c0295a98 Remove obsolete DCHECK in RtpPacket::CopyHeaderFrom Danil Chapovalov 2021-05-24 19:09:42 +02:00
  • 6396b48b18 Avoid modifying BWE internal state on reception of REMB feedback. Christoffer Rodbro 2021-05-24 21:20:23 +02:00
  • c09c58134a dcsctp: Limit the size of generated SACK chunks Victor Boivie 2021-05-19 14:03:22 +02:00
  • e4adedc0cd Update WebRTC code version (2021-05-25T04:03:57). webrtc-version-updater 2021-05-24 21:03:59 -07:00
  • 41a111d5b9 Switch to av_packet_alloc() Ted Meyer 2021-05-21 12:20:49 -07:00
  • 0f506780aa Remove usage of TOOLKIT_GTK define. Byoungchan Lee 2021-04-16 07:03:36 +09:00
  • 816134a8aa Reland "Fix race between enabled() and set_enabled() in VideoTrack." Tommi 2021-05-24 16:54:41 +02:00
  • 2f3c5e6752 Skip WindowCapturerTest.Capture on macOS. Mirko Bonadei 2021-05-24 14:50:53 +02:00
  • ae0d117d51 Implement the mixer-to-client per CSRC audio level extension (RFC 6465). Doudou Kisabaka 2021-05-24 13:04:45 +02:00
  • 096ad02c02 Revert "Fix race between enabled() and set_enabled() in VideoTrack." Evan Shrubsole 2021-05-24 14:06:09 +00:00
  • 2fa4774067 Revert "Deprecate microsecond timestamps in RTC event log." Björn Terelius 2021-05-24 13:10:41 +00:00
  • cf497890f3 Refactor some retransmission tests. Erik Språng 2021-05-24 14:26:59 +02:00
  • 006815eed0 Add temp peer_connection_interface include rtc_base/event.h Evan Shrubsole 2021-05-24 12:59:56 +02:00
  • e6ee8fab7e Deprecate microsecond timestamps in RTC event log. Björn Terelius 2021-05-24 10:19:55 +02:00
  • 5ffefe9d2d Fix race between enabled() and set_enabled() in VideoTrack. Tommi 2021-05-22 17:46:53 +02:00
  • 13e585196e Update WebRTC code version (2021-05-24T04:02:02). webrtc-version-updater 2021-05-23 21:02:05 -07:00
  • f3d71c2ed7 Roll chromium_revision 70eb2d0977..2826799ea1 (885736:885837) chromium-webrtc-autoroll 2021-05-23 13:01:47 -07:00
  • 93faab1b51 dcsctp: Implement Round Robin scheduler Victor Boivie 2021-05-20 13:08:21 +02:00
  • 70cd086644 SEA: Only spawn multi-layered encoders if active layers > 1. Henrik Boström 2021-05-21 11:58:17 +02:00
  • bcadacdb0f Update WebRTC code version (2021-05-23T04:03:13). webrtc-version-updater 2021-05-22 21:03:15 -07:00
  • 2440d34075 dcsctp: Rename FCFSSendQueue to RRSendQueue Victor Boivie 2021-05-20 13:47:32 +02:00
  • 913c3af879 Update WebRTC code version (2021-05-22T04:04:00). webrtc-version-updater 2021-05-21 21:04:01 -07:00
  • 7ee9b6b940 Roll chromium_revision 9431bab2be..70eb2d0977 (885625:885736) chromium-webrtc-autoroll 2021-05-21 19:01:31 -07:00
  • 9f6808b13c Remove cricket::DtlsTransportState. Mirko Bonadei 2021-05-21 20:46:09 +02:00
  • e976f75c1d Roll chromium_revision e5dd2eb61e..9431bab2be (884954:885625) chromium-webrtc-autoroll 2021-05-21 13:02:06 -07:00
  • 20f9401578 Remove GTEST_ALLOW_UNINSTANTIATED in datachannel tests Harald Alvestrand 2021-05-21 07:04:27 +00:00
  • 32ee3b88ea dcsctp: Ensure RTO is always greater than RTT Victor Boivie 2021-05-19 22:22:42 +02:00
  • cab90db24a Delete NtpOffsetMs and TimeMicrosToNtp methods. Paul Hallak 2021-05-21 19:08:21 +02:00
  • 46fbefa302 Convert to NTP time using the real clock. Paul Hallak 2021-05-21 18:53:28 +02:00
  • a6b0d53dc2 Delete the old flavor of RtcpTransceiverImpl::ReceivePacket Paul Hallak 2021-05-21 18:26:07 +02:00
  • fe3dd51f32 Use the injected clock in rtcp_transciever. Paul Hallak 2021-05-21 18:08:04 +02:00
  • 61a287a3cb Add accessor for UTC start time in event log Björn Terelius 2021-05-21 15:02:07 +02:00
  • 00f6e75671 Use webrtc::Clock to query for the NTP time and to convert timestamps to NTP. Paul Hallak 2021-05-21 14:00:28 +02:00
  • 1cb796f03a Add performance tracing for API calls (inside api proxies). Tommi 2021-05-21 08:50:09 +02:00
  • 47ed99872d Use the clock to convert absolute capture timestamps to NTP times. Paul Hallak 2021-05-21 10:34:34 +02:00
  • edc347c4e5 Introduce (Un)SubscribeDtlsTransportState methods. Mirko Bonadei 2021-05-13 16:50:45 +02:00
  • 95f1e5192c Do not attempt setting the absolute capture time extension if we don't get a timestamp. Paul Hallak 2021-05-21 08:53:46 +02:00
  • 2491dbdf0b Make Clock::ConvertTimestampToNtpTime pure virtual Paul Hallak 2021-05-21 07:31:52 +02:00
  • e93fe6c532 Enable Chromium to stop including api/proxy.h indirectly. Markus Handell 2021-05-20 22:46:31 +02:00
  • b59e9045bf Add the ability to convert a timestamp to NTP time. Paul Hallak 2021-05-20 17:21:49 +02:00
  • 0cff39137b Start with a BeginLog event in event log encoder unittest Björn Terelius 2021-05-19 01:10:09 +02:00
  • b8dc7fa5a6 Make AgcManagerDirect clipping parameters configurable Hanna Silen 2021-05-20 17:37:56 +02:00
  • e2b9fc6909 Move FecOverheadRate, BitrateCallbacks to rtp_sender_egress_unittest. Erik Språng 2021-05-20 15:26:56 +02:00
  • 63b3095d2b Make local to capturer clock offset a separate entry in PacketInfo. Minyue Li 2021-05-19 14:38:25 +02:00
  • cbde0cf994 Roll chromium_revision 6291fe6f0e..e5dd2eb61e (884821:884954) chromium-webrtc-autoroll 2021-05-20 03:01:30 -07:00
  • 0de1ed0244 Have only two pure virtual methods for webrtc::Clock, CurrentTime and CurrentNtpTime. Make all other methods non-virtual. Paul Hallak 2021-05-19 14:36:50 +02:00
  • 8ed1e9336e Switch from check_targets to no_check_targets in .gn Byoungchan Lee 2021-05-20 18:30:10 +09:00
  • 193f4bf6c1 Replace legacy getStats with standard getStats in the iOS example Jaehyun Ko 2021-05-20 15:04:02 +09:00
  • bd346d7439 Update WebRTC code version (2021-05-20T04:01:58). webrtc-version-updater 2021-05-19 21:02:00 -07:00
  • 069ed356be Roll chromium_revision 0321a6153c..6291fe6f0e (884706:884821) chromium-webrtc-autoroll 2021-05-19 19:01:49 -07:00
  • 4fb5684493 Roll chromium_revision b8d2317c2e..0321a6153c (884575:884706) chromium-webrtc-autoroll 2021-05-19 15:01:35 -07:00
  • 6c96611069 Roll chromium_revision 3bdbd47d23..b8d2317c2e (884422:884575) chromium-webrtc-autoroll 2021-05-19 11:16:15 -07:00
  • 86bd92fca9 Add test for many non-dropped packets in unreliable mode. Harald Alvestrand 2021-05-19 16:17:04 +00:00
  • e7481a4199 Add an UlpFec test to RtpRtcp unit tests. Erik Språng 2021-05-19 16:46:33 +02:00
  • 398def6828 Improvements to AEC3 logging to simplify debugging Gustaf Ullberg 2021-05-19 15:25:18 +02:00
  • aeb8ce882f AEC3: Change adaptation speed of the matched filter after a delay is found Gustaf Ullberg 2021-05-19 14:26:31 +02:00
  • 402ceffff1 sctp: Reduce logging level for common calls Victor Boivie 2021-05-19 14:26:22 +02:00
  • 3a45d32d4e dcsctp: Report duplicate TSNs Victor Boivie 2021-05-19 13:40:55 +02:00
  • 91fef0250b Roll chromium_revision 108d11241e..3bdbd47d23 (884294:884422) chromium-webrtc-autoroll 2021-05-19 05:01:48 -07:00
  • f6be1b22d6 Simplify RtpSenderTest.SendFlexfecPackets and move to RtpRtcp-level. Erik Språng 2021-05-19 11:12:39 +02:00