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  • fad9a6dae7 Delete deprecated Create method and config from AudioCodingModule Henrik Lundin 2023-02-02 14:12:54 +00:00
  • 101c6aab1b Remove leftover function signatures. Fredrik Solenberg 2023-02-01 10:30:05 +01:00
  • 6c60f72a6b Refactor video codec testing stats Sergey Silkin 2023-02-02 11:49:13 +01:00
  • 97d1c34769 Enable rotation tests marked as expected failures Andreas Pehrson 2023-01-13 14:23:27 +01:00
  • 65ab5fd728 Cleanup RemoteEstimatorProxy::IncomingPacket Danil Chapovalov 2023-02-01 18:42:07 +01:00
  • ba846ccf24 Add a test that shows when channel_receive fires RR Harald Alvestrand 2023-02-01 14:55:13 +00:00
  • 84f75699c6 Break apart AudioCodingModule and AcmReceiver Henrik Lundin 2023-02-01 12:07:10 +00:00
  • c5455e7b53 Allow RTX ssrc to be updated on receive streams Per K 2023-02-01 10:48:09 +01:00
  • be03c09718 Only serialize non-stopped RTP header extensions Philipp Hancke 2023-02-01 10:10:06 +01:00
  • 1f206b841e Use ArrayView in the IncomingRtcpPacket function. Harald Alvestrand 2023-02-01 11:12:46 +00:00
  • 16a8792e0a Propagate received video csrcs to encodedframe metadata Tony Herre 2023-02-01 10:26:01 +01:00
  • 0507fbdd05 [Desktop Capture] Remove disabled test. mark a. foltz 2023-01-31 15:41:50 -08:00
  • cb4b0a6ad2 Check FMA3 support before use it in SincResampler Henrik Lundin 2023-01-31 15:49:55 +00:00
  • 0b7184ce06 Add possibility to set MetricsSet metadata. Mirko Bonadei 2023-01-31 13:02:45 +01:00
  • 217b384c1b Remove rtp header extension from config of Call audio and video receivers Per K 2023-01-30 16:04:01 +01:00
  • 3541732527 Add a Config struct to AcmReceiver, and a ctor using it Henrik Lundin 2023-01-31 08:40:56 +00:00
  • 327405165b Add method to get FD for physical socket Artem Titov 2023-01-26 12:54:49 +01:00
  • d78f8e7fce Fix doc path Artem Titov 2023-01-31 10:41:45 +01:00
  • e5922834f8 Add 'metadata' field to MetricsSet proto. Mirko Bonadei 2023-01-31 09:48:59 +01:00
  • a617867a45 Reland "Migrate WebRTC documentation to new renderer" Artem Titov 2023-01-30 10:51:01 +00:00
  • 23671039b0 Add android-tiramisuprivacysandbox to DEPS. Mirko Bonadei 2023-01-31 09:46:19 +01:00
  • 3963a95a62 Enforce policy that SDP munging requires special approval Harald Alvestrand 2023-01-25 17:47:07 +00:00
  • 48220310ae shared_screencast_stream: Set WL capturer id Salman 2023-01-30 15:53:06 +00:00
  • 9214718cc5 Roll chromium_revision 2fd12db504..e182675fbb (1098462:1098562) chromium-webrtc-autoroll 2023-01-30 06:01:17 -08:00
  • dad91a69bf Send periodic TransportFeedback based on extension version Per K 2023-01-30 12:26:17 +01:00
  • 2ded55e0df Cleanup Thread::BlockingCall Danil Chapovalov 2023-01-27 15:06:37 +00:00
  • 815522782a sdp: add rtcp-fb:* lines for common feedback Philipp Hancke 2023-01-17 15:13:59 +01:00
  • 68564bb59c [infra] Clean up mb_config.pyl after reclient migration Junji Watanabe 2023-01-30 16:38:06 +09:00
  • 942abaace2 Roll chromium_revision 7faf2e057e..2fd12db504 (1098356:1098462) chromium-webrtc-autoroll 2023-01-29 22:02:04 -08:00
  • f6a1f7e010 Roll chromium_revision ff5a43c15d..7faf2e057e (1098253:1098356) chromium-webrtc-autoroll 2023-01-29 00:01:53 -08:00
  • 863a07fe6f Update WebRTC code version (2023-01-29T04:02:18). webrtc-version-updater 2023-01-28 20:02:20 -08:00
  • dc7a14bb66 Roll chromium_revision 5195bd9e41..ff5a43c15d (1098136:1098253) chromium-webrtc-autoroll 2023-01-27 20:01:38 -08:00
  • 6ff7713a70 base_capturer_pipewire: Send frames via callback Salman 2023-01-27 22:58:35 +00:00
  • bf216a7d3e Roll chromium_revision d9876846b5..5195bd9e41 (1097987:1098136) chromium-webrtc-autoroll 2023-01-27 14:02:25 -08:00
  • 26340b07a4 desktop_capturer: Support frame rate negotiation via pipewire Salman 2023-01-27 18:26:05 +00:00
  • 4c4566a8e6 Roll chromium_revision 6da6f5ebed..d9876846b5 (1097837:1097987) chromium-webrtc-autoroll 2023-01-27 10:01:22 -08:00
  • 64ce699f4b Propagate Video CSRCs modified by an insertable streams frame transform Tony Herre 2023-01-27 13:25:32 +01:00
  • db208317eb Update RTP timestamp based on capture timestamp when audio send stream is resumed. Jakob Ivarsson 2023-01-27 15:13:22 +01:00
  • 9d21eb3e3e Roll chromium_revision 1f33300013..6da6f5ebed (1097480:1097837) chromium-webrtc-autoroll 2023-01-27 02:17:07 -08:00
  • a2653bc236 Export some more symbols for use in chromium tests. Sameer Vijaykar 2023-01-26 16:34:09 +01:00
  • ca1cfd45de Add missing src/third_party/jdk11 dependency and roll chromium into webrtc landrey 2023-01-26 19:18:17 +00:00
  • 0f2ce5cc1c Revert "Migrate WebRTC documentation to new renderer" Artem Titov 2023-01-26 20:18:46 +00:00
  • b0d8a3728c Ensure CallTest derived tests per default set min/max audio bitrate. Per K 2023-01-26 16:11:31 +01:00
  • 6c032cb835 in rtcp::TransportFeedback do not memorise all described packet Danil Chapovalov 2023-01-26 11:19:09 +01:00
  • dcb09ff218 Reset encoder when audio send stream is stopped. Jakob Ivarsson 2023-01-25 20:03:56 +01:00
  • 3eceaf4669 Migrate WebRTC documentation to new renderer Artem Titov 2023-01-16 13:23:50 +01:00
  • 94d5f6af62 Add missing include Bjorn Terelius 2023-01-20 13:37:01 +01:00
  • 66efab2dd2 Measure RTCPMuxPolicy at time of connect Philipp Hancke 2023-01-25 11:37:45 +01:00
  • 73e0cc8969 Delete unused Audio Bwe integration test. Per K 2023-01-25 13:43:01 +01:00
  • cfbb247f6d Update WebRTC code version (2023-01-26T04:01:54). webrtc-version-updater 2023-01-25 20:01:56 -08:00
  • e15b9ff408 Add a basic unittest for webrtc::voe::ChannelReceive Harald Alvestrand 2023-01-25 16:32:54 +00:00
  • 664cf14f9f Reland "Delete PacketReceiver::DeliverPacket from all implementations" Per K 2023-01-25 15:19:11 +01:00
  • 7a67dce582 prefer absl::optional for rtx-time Philipp Hancke 2023-01-25 13:52:24 +01:00
  • 7c43d24000 Roll chromium_revision a484be4b74..e5191e93ab (1096680:1096792) chromium-webrtc-autoroll 2023-01-25 08:01:58 -08:00
  • 5adc2b6969 Correct RTCAudioPlayoutStats type and add kind field. Fredrik Hernqvist 2023-01-25 13:49:16 +01:00
  • 0c1c9720d8 Fix gtest-output and resultdb for fuchsia Christoffer Jansson 2023-01-25 14:49:50 +01:00
  • 5683a12728 Increase expiration timeout for Perf bots landrey 2023-01-25 13:07:53 +00:00
  • 5671c64103 Stop overriding extensions in rampup tests Instead, ensure extensions are registered so that both transport and send streams are aware. Per K 2023-01-25 12:58:50 +01:00
  • 87733493da Update win10 mixins. Jeremy Leconte 2023-01-25 11:32:59 +01:00
  • 92dcc2d1f4 Roll chromium_revision 98774e2693..a484be4b74 (1096567:1096680) chromium-webrtc-autoroll 2023-01-25 02:15:49 -08:00
  • 10f1bf3923 Remove unused enum FrameCombiner::LimiterType Alessio Bazzica 2023-01-24 17:30:13 +01:00
  • f2a083f262 Revert "Delete PacketReceiver::DeliverPacket from all implementations" Andrey Logvin 2023-01-25 08:57:56 +00:00
  • 07577b5434 Update WebRTC code version (2023-01-25T04:11:56). webrtc-version-updater 2023-01-24 20:11:57 -08:00
  • 508979bb72 Roll chromium_revision 87e5077aae..98774e2693 (1096404:1096567) chromium-webrtc-autoroll 2023-01-24 18:04:50 -08:00
  • cc1c932f10 Roll chromium_revision 78e2f876c5..87e5077aae (1096262:1096404) chromium-webrtc-autoroll 2023-01-24 14:02:58 -08:00
  • ea36cc273f Roll chromium_revision 632253d282..78e2f876c5 (1096118:1096262) chromium-webrtc-autoroll 2023-01-24 10:02:00 -08:00
  • 897ea04db5 Delete PacketReceiver::DeliverPacket from all implementations Per K 2023-01-20 08:39:26 +01:00
  • 0540627386 SVC: Add test for SVC fallback Florent Castelli 2022-11-29 15:27:37 +01:00
  • 1e43ce6f3f Roll chromium_revision f8b9751f30..632253d282 (1095656:1096118) chromium-webrtc-autoroll 2023-01-24 04:02:40 -08:00
  • e2c29c520a Use PacketReceiver::DeliverRtpPacket in RtpReplayer Per K 2023-01-24 10:59:04 +01:00
  • 0793ee7552 Remove FakePortAllocator's dependency on ScopedKeyValueConfig. Sameer Vijaykar 2023-01-23 16:31:29 +01:00
  • ace52a821c [infra] Remove CQ shadow builders with reclient Junji Watanabe 2023-01-24 10:37:33 +09:00
  • 2810c14ade [infra] Add todo for reclient migration cleanup Junji Watanabe 2023-01-24 10:55:39 +09:00
  • 22821deb38 Make capture timestamp optional in ADM. Jakob Ivarsson 2023-01-20 22:09:29 +01:00
  • 6e627290bf Roll chromium_revision cdf104c2c9..f8b9751f30 (1095545:1095656) chromium-webrtc-autoroll 2023-01-23 08:02:27 -08:00
  • 5b55b2750a Version 3: Various changes on the pre-echo delay estimator: - Lowering the energy threshold for updating the accumulated error. - Not using the pre-echo estimate in the initial frames when the matched filters have been recently initialized. - Slight speed up for the increases in the accumulated error. - Not periodically resetting the accumulated error. Jesús de Vicente Peña 2023-01-23 11:35:50 +01:00
  • 438b5b4ca5 WebRtcVideoChannel creates default stream with dummy SSRC on received RTX packet. Per K 2023-01-23 14:46:03 +01:00
  • 05ce032486 Remove the //rtc_base target Florent Castelli 2023-01-20 16:39:40 +01:00
  • 9ad10bc492 Only generate codec stats for the voice send and receive codec Philipp Hancke 2023-01-18 13:38:19 +01:00
  • 41748c9188 Roll chromium_revision a3aaddaf0c..cdf104c2c9 (1095442:1095545) chromium-webrtc-autoroll 2023-01-23 02:08:24 -08:00
  • d3b9e71b0e Migrate linux_libfuzzer_rel to use reclient Junji Watanabe 2023-01-23 12:16:09 +09:00
  • d506651607 Update WebRTC code version (2023-01-23T04:11:41). webrtc-version-updater 2023-01-22 20:11:42 -08:00
  • 5e7ae14c11 Roll chromium_revision 0c3ca8fbc7..a3aaddaf0c (1095332:1095442) chromium-webrtc-autoroll 2023-01-22 02:16:34 -08:00
  • c8270b1dc8 Roll chromium_revision e0c02c1406..0c3ca8fbc7 (1095203:1095332) chromium-webrtc-autoroll 2023-01-20 20:02:42 -08:00
  • d2c15aa550 Roll chromium_revision 92aea4500e..e0c02c1406 (1095006:1095203) chromium-webrtc-autoroll 2023-01-20 14:02:21 -08:00
  • 62ba379caa Add some RTC_EXPORT needed by Chromium. Mirko Bonadei 2023-01-20 17:32:07 +00:00
  • c4ea5aeca9 Avoid log spamming when the dependency descriptor fail to parse. philipel 2023-01-20 13:13:01 +01:00
  • 4893638e72 mark RTPHeader struct RTC_EXPORT Philipp Hancke 2023-01-20 15:23:50 +01:00
  • 7d79740bc1 Roll chromium_revision 2fb5e2fb16..92aea4500e (1094884:1095006) chromium-webrtc-autoroll 2023-01-20 06:01:30 -08:00
  • eeb223557f Retune AGC2 input volume controller speech ratio threshold Alessio Bazzica 2023-01-20 10:54:54 +01:00
  • 46053e4aae Handle the case of missing certificates. Henrik Boström 2023-01-20 13:18:53 +01:00
  • 124d7c3fe5 [Stats] Handle the case of missing certificates. Henrik Boström 2023-01-20 10:48:31 +01:00
  • bf27f35a04 Roll chromium_revision c97ba0b8b1..2fb5e2fb16 (1094776:1094884) chromium-webrtc-autoroll 2023-01-19 22:01:14 -08:00
  • 89870ffa95 Reland "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" Per Kjellander 2023-01-19 15:45:58 +00:00
  • 8e1d61338a Update WebRTC code version (2023-01-20T04:01:58). webrtc-version-updater 2023-01-19 20:02:00 -08:00
  • 5f59dedd23 Roll chromium_revision 09356cf622..c97ba0b8b1 (1094626:1094776) chromium-webrtc-autoroll 2023-01-19 16:05:31 -08:00
  • 74f108a671 Add Fuchsia support to memory_usage. Sarah Pham 2023-01-20 10:18:19 +11:00
  • 541024a588 Roll chromium_revision e551fb7716..09356cf622 (1093833:1094626) chromium-webrtc-autoroll 2023-01-19 12:01:44 -08:00
  • 41010f95bb Log resolved & unresolved turn server address as sensitive string. Yury Yarashevich 2023-01-19 17:56:03 +01:00
  • 4cb2ac0e30 User Timestamp and TimeDelta instead of raw ints in RtpSenderEgress Danil Chapovalov 2023-01-19 12:59:55 +01:00