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  • bd1e5d5aa5 Reland "Ensure RTCRtpSenders are always created with one encoding" Florent Castelli 2023-01-17 18:08:26 +01:00
  • 828de8036d Populate RTCInboundRtpStreamStats::playoutId when appropriate Fredrik Hernqvist 2023-01-19 15:04:54 +01:00
  • 4abca6699d Ensure FakeNetwork propages arrival_time Per K 2023-01-19 15:11:07 +01:00
  • 9f9671fe7f Revert "Reland "Ensure RTCRtpSenders are always created with one encoding"" Evan Shrubsole 2023-01-19 13:56:21 +00:00
  • 3e61f881cd Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" Per Kjellander 2023-01-19 10:08:35 +00:00
  • fc5d627cef Reland "Ensure RTCRtpSenders are always created with one encoding" Florent Castelli 2023-01-17 18:08:26 +01:00
  • 949e356456 Update WebRTC code version (2023-01-19T04:04:31). webrtc-version-updater 2023-01-18 20:04:33 -08:00
  • abb64161e4 mouse_cursor_monitor: Annotate a method with RTC_EXPORT Salman 2023-01-18 22:04:09 +00:00
  • a0bc404607 Remove WebRTC-Dav1dDecoder kill switch. philipel 2023-01-18 15:27:47 +01:00
  • 299c02ede2 Roll chromium_revision 5c3b57c4c8..e551fb7716 (1093721:1093833) chromium-webrtc-autoroll 2023-01-18 06:02:26 -08:00
  • 73918995a4 Add Fuchsia perf output and fix upload Christoffer Jansson 2023-01-18 14:46:34 +01:00
  • 9ece54fa73 Delete unnecssary AudioReceiveStreamInterface::GetRtpExtensions Per K 2023-01-18 14:22:38 +01:00
  • 444741e78d replace use of iterators with for loops or auto Philipp Hancke 2023-01-18 10:27:57 +01:00
  • 3b96f2c770 Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp Per K 2023-01-18 14:06:19 +01:00
  • bdf30c871f Ensure VideoRtpReplayer use new PacketReceiver::DeliverRtp packet. Per K 2023-01-18 09:49:28 +01:00
  • e6b3f48a06 Reland "Move leb128 helper functions into own build target" Danil Chapovalov 2023-01-13 12:33:41 +01:00
  • 4885de46ef Remove test workaround to catch scenario when packet is resent before sent Danil Chapovalov 2022-12-23 17:06:27 +00:00
  • 4ccb616fa2 Old iOS sim bots clean up. Jeremy Leconte 2023-01-18 11:11:07 +01:00
  • 44e5d5a9d1 Revert "Ensure RTCRtpSenders are always created with one encoding" Evan Shrubsole 2023-01-18 10:28:12 +00:00
  • 2c0376e20c Run iOS sim bots on versions 14, 15 and 16. Jeremy Leconte 2023-01-18 11:02:08 +01:00
  • e351560d5f Disable RTCCameraVideoCapturerTestsWithMockedCaptureSession. Jeremy Leconte 2023-01-18 09:28:29 +01:00
  • 25ffa806e8 Roll chromium_revision 8496a82e1d..5c3b57c4c8 (1093621:1093721) chromium-webrtc-autoroll 2023-01-17 20:02:16 -08:00
  • d05daeb16d Roll chromium_revision 2762031951..8496a82e1d (1093486:1093621) chromium-webrtc-autoroll 2023-01-17 16:01:51 -08:00
  • 897f15d4e1 Roll chromium_revision f1d4d8b74c..2762031951 (1093304:1093486) chromium-webrtc-autoroll 2023-01-17 12:02:22 -08:00
  • b8023690d9 Ensure RTCRtpSenders are always created with one encoding Florent Castelli 2023-01-17 18:08:26 +01:00
  • 478f3b786e Avoid waking up encoder thread when audio send stream is stopped. Jakob Ivarsson 2023-01-17 15:40:36 +01:00
  • 9a8aa209ab Roll chromium_revision b4664ef98d..f1d4d8b74c (1093178:1093304) chromium-webrtc-autoroll 2023-01-17 06:06:09 -08:00
  • 1389c4b594 Add 444 10 bits support for H264 and VP9 Sergio Garcia Murillo 2023-01-09 18:29:34 +01:00
  • 94b05599ec Only fill send/recv stats if there are send/receive streams Philipp Hancke 2023-01-17 10:38:56 +01:00
  • 9b239001bc Update WebRTC code version (2023-01-17T04:11:09). webrtc-version-updater 2023-01-16 20:11:10 -08:00
  • 6d19e6fefc Roll chromium_revision de6dd5a1b5..b4664ef98d (1093073:1093178) chromium-webrtc-autoroll 2023-01-16 18:01:59 -08:00
  • f1fa6ac7bf Implement time functions for Fuchsia. Sarah Pham 2023-01-16 10:43:34 +11:00
  • 40b5bd72d0 APM: fix TS initialization bugs with WebRTC-Audio-GainController2 Alessio Bazzica 2023-01-16 20:19:48 +01:00
  • f7a46e55cb Roll chromium_revision 09a5d11a47..de6dd5a1b5 (1092898:1093073) chromium-webrtc-autoroll 2023-01-16 08:02:13 -08:00
  • a6b9924988 Remove all usage of //rtc_base target Florent Castelli 2023-01-16 10:53:26 +00:00
  • efbe753617 Add RTCAudioPlayoutStats to GetStats(). Fredrik Hernqvist 2023-01-13 16:42:36 +01:00
  • f7e40717ab Only generate codec stats for the video send/recv codec in use Philipp Hancke 2023-01-16 09:57:41 +01:00
  • 3dd73ae6f4 Surface the SetMetadata() method so that Chromium can use it. Henrik Boström 2023-01-16 10:41:28 +01:00
  • 6afa92ab20 Tooling to process RtcEventNetEqSetMinimumDelay Lionel Koenig 2023-01-16 11:23:36 +01:00
  • 55ac75f177 Make terelius owner of rtc_event to text Lionel Koenig 2023-01-16 10:41:17 +01:00
  • 5cca08648c Update WebRTC doc related to webrtc.org accounts. Jeremy Leconte 2023-01-16 10:05:25 +01:00
  • 2d14479605 Update WebRTC code version (2023-01-16T04:07:02). webrtc-version-updater 2023-01-15 20:07:03 -08:00
  • 332bc4bfdd Roll chromium_revision ad7c2cc677..09a5d11a47 (1092797:1092898) chromium-webrtc-autoroll 2023-01-15 16:01:13 -08:00
  • c01410ea1e Roll chromium_revision 2650ba360c..ad7c2cc677 (1092690:1092797) chromium-webrtc-autoroll 2023-01-14 10:01:20 -08:00
  • a1eb4fe664 Roll chromium_revision 05284a0a51..2650ba360c (1092572:1092690) chromium-webrtc-autoroll 2023-01-13 16:17:17 -08:00
  • 37d4d840b0 Roll chromium_revision 2b544cbc99..05284a0a51 (1092425:1092572) chromium-webrtc-autoroll 2023-01-13 12:02:06 -08:00
  • 612872b29d Add RtcEvent to store when MinimumSetDelay is set on NetEq Lionel Koenig 2022-12-13 08:58:13 +01:00
  • 8da0f3aecc Roll chromium_revision f6c2626500..2b544cbc99 (1092306:1092425) chromium-webrtc-autoroll 2023-01-13 08:03:05 -08:00
  • 6cf46b9497 Add RTPVideoHeader::SetFromMetadata() and FromMetadata(). Henrik Boström 2023-01-11 15:55:53 +01:00
  • dc39aebd08 Add GetRTPVideoHeaderCodecSpecifics() to metadata. Henrik Boström 2023-01-11 12:16:36 +01:00
  • bb25641dd9 [PCLF] Add an API to add extra audio/video RTP header extensions Artem Titov 2023-01-13 11:54:29 +01:00
  • 6a9af57a24 Revert "Move leb128 helper functions into own build target" Danil Chapovalov 2023-01-13 08:13:08 +00:00
  • 7aa145d73e Remove 'iOS Debug (simulator)' from LKGR. Jeremy Leconte 2023-01-13 11:49:26 +01:00
  • 3fab086614 Use RtcEventLog::EncodingType::NewFormat in VideoQualityTest. philipel 2022-12-30 13:44:52 +01:00
  • 04e9318e5c Roll chromium_revision 76506aa56a..f6c2626500 (1092201:1092306) chromium-webrtc-autoroll 2023-01-13 02:01:45 -08:00
  • 079e93de17 Add callback for raw frames for video capture Michael Olbrich 2022-12-06 16:27:44 +01:00
  • 9795589f50 Add "iOS Debug (simulator)" CI bot. Jeremy Leconte 2023-01-13 08:53:50 +01:00
  • e7e53feace Roll chromium_revision aff3b5b1f9..76506aa56a (1092063:1092201) chromium-webrtc-autoroll 2023-01-12 18:02:34 -08:00
  • f04a345762 Roll chromium_revision 8ff1e4f84e..aff3b5b1f9 (1091923:1092063) chromium-webrtc-autoroll 2023-01-12 14:02:12 -08:00
  • c19cc29e25 Roll chromium_revision 00abcc8f11..8ff1e4f84e (1091815:1091923) chromium-webrtc-autoroll 2023-01-12 10:01:57 -08:00
  • b3046c25aa Use PacketReceiver::DeliverRtpPaket in scenario tests Per K 2023-01-12 17:00:22 +01:00
  • 5b7896be29 Roll chromium_revision b924865c52..00abcc8f11 (1091696:1091815) chromium-webrtc-autoroll 2023-01-12 06:03:38 -08:00
  • b613d62285 [Unwrap] Delete webrtc::Unwrapper Evan Shrubsole 2023-01-12 12:28:01 +00:00
  • d8b6b06e70 [Unwrap] Delete rtc::TimestampWrapAroundHandler Evan Shrubsole 2023-01-12 12:28:10 +00:00
  • 83fd843535 Add a ios_x64_dbg_simulator try bot. Jeremy Leconte 2023-01-12 14:27:13 +01:00
  • 222c052864 [Unwrap] Delete SequenceNumbersConformanceTest Evan Shrubsole 2023-01-12 12:27:48 +00:00
  • 4387ad6cdc [Unwrap] Migrate dcsctp sequence numbers to SeqNumUnwrapper Evan Shrubsole 2023-01-09 13:28:53 +00:00
  • 17f783eee8 Skip trimming packet arrival history at the beginning Danil Chapovalov 2023-01-04 09:56:47 +01:00
  • 778742963a In remb parser discard bitrate larger than max int64_t Danil Chapovalov 2023-01-12 09:54:42 +01:00
  • 43d4eee8ce [Unwrap] Migrate rtp_rtcp_tests to RtpSequenceNumberUnwrapper Evan Shrubsole 2023-01-12 10:35:04 +00:00
  • 9337ac8650 [Unwrap] Migrate RtcEventLog parser to use RtpSequenceNumberUnwrapper Evan Shrubsole 2023-01-09 13:40:58 +00:00
  • d22dc86211 Roll chromium_revision 40afafa78c..b924865c52 (1091586:1091696) chromium-webrtc-autoroll 2023-01-11 22:02:15 -08:00
  • 0bef97c5ae Update WebRTC code version (2023-01-12T04:02:41). webrtc-version-updater 2023-01-11 20:02:43 -08:00
  • 402a440577 Roll chromium_revision 709ec8ac30..40afafa78c (1091458:1091586) chromium-webrtc-autoroll 2023-01-11 16:14:50 -08:00
  • b40657773f Roll chromium_revision 0c76207a3f..709ec8ac30 (1090534:1091458) chromium-webrtc-autoroll 2023-01-11 12:02:17 -08:00
  • e4c49e379a [Unwrap] Migrate RtpToNtpEstimator to use RtpTimestampUnwrapper Evan Shrubsole 2023-01-11 14:14:51 +00:00
  • e6b4cbe606 Add SVC fallback. Åsa Persson 2023-01-11 15:50:58 +01:00
  • d100a589c8 Add dimensions to video settings in objc sdk camera backend. Andreas Pehrson 2022-12-19 17:23:51 +01:00
  • b081042eec Remove dimension check in SimulcastUtility::ValidSimulcastParameters anurag 2023-01-10 15:15:10 -08:00
  • 8c347eb5ea [Unwrap] Migrate TransportFeedbackDemuxer to use RtpSequenceNumberUnwrapper Evan Shrubsole 2023-01-09 13:42:24 +00:00
  • 57e5562c3f [Unwrap] Use RtpTimestampUnwrapper in audio/channel_receive Evan Shrubsole 2023-01-09 13:41:24 +00:00
  • fa962ffc69 Move leb128 helper functions into own build target Danil Chapovalov 2023-01-10 19:14:37 +01:00
  • a5ba58662f Update visibility on rtc_base:log_sinks target Florent Castelli 2023-01-11 11:01:15 +01:00
  • 7b4c8adb75 Reland "[Unwrap] Migrate ReceiveStatisticsImpl to use RtpSequenceNumberUnwrapper" Evan Shrubsole 2023-01-09 14:34:34 +00:00
  • e137c4592e stats: deprecate timestamp_us constructor and method Philipp Hancke 2023-01-10 13:06:02 +01:00
  • 128afb1a7e Only build fuchsia_perf_tests on fuchsia os. Jeremy Leconte 2023-01-11 10:55:35 +01:00
  • 47d4be732f [Unwrap] Migrate TransportFeedbackAdapter to use RtpSequenceNumberUnwrapper Evan Shrubsole 2023-01-09 13:42:15 +00:00
  • 7b96ebbc56 Run perf tests assertion only in the "quick" perf test mode. Jeremy Leconte 2023-01-11 08:37:34 +01:00
  • 69ea6afa35 Refresh WebRTC mixins.pyl file. Jeremy Leconte 2023-01-11 10:13:32 +01:00
  • 154cbea357 Add RTC_EXPORT to symbols imported by CRD Salman 2023-01-10 18:07:20 +00:00
  • bfc26c65e6 Use libyuv rotate methods Sergio Garcia Murillo 2023-01-09 16:08:20 +01:00
  • 143c3b2b4c Update visibility on rtc_base:net_helper target Florent Castelli 2023-01-10 15:02:51 +01:00
  • 89ca299161 Use parsed packet from RtpTransport::DemuxPacket in engine and call Per K 2023-01-10 14:28:25 +01:00
  • 7ef0c1aff5 Implement RTCNonStandardStatsMember using StatExposureCriteria Evan Shrubsole 2022-10-31 11:05:33 +00:00
  • d7956891d0 [DVQA] Remove default value for report_infra_metrics in VideoQualityAnalyzerInjectionHelper Artem Titov 2023-01-09 15:39:15 +01:00
  • 854ca9a0a6 Delete stale TODO about GFD fuzzing Danil Chapovalov 2023-01-10 12:49:50 +01:00
  • c4991048b2 Revert "[Unwrap] Migrate ReceiveStatisticsImpl to use RtpSequenceNumberUnwrapper" Jeremy Leconte 2023-01-10 11:14:03 +00:00
  • 885ededbb8 Add move constructor and assign operator to RtpPacket Danil Chapovalov 2023-01-10 10:56:20 +01:00
  • c3891e3a4e [Unwrap] Migrate NetEqDelayAnalyzer to use RtpTimestampUnwrapper Evan Shrubsole 2023-01-09 13:30:53 +00:00
  • 4df20baff1 Implement GetParameters/GetSources support for unsignaled SSRCs. Henrik Boström 2023-01-09 13:57:15 +01:00