admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Commit Graph

  • 5eda59c96f Replace legacy RtpRtcp::GetRemoteStat function with GetLatestReportBlockData Danil Chapovalov 2021-02-18 20:53:28 +01:00
  • fd87944042 Removed WebRTC-NetworkCondition-EncoderSwitch field trial. philipel 2021-03-12 15:00:51 +01:00
  • 7c7885c016 Remove NTP timestamp from PacketBuffer::Packet. philipel 2021-03-12 14:33:46 +01:00
  • 662b306bae Replace blocking invokes with PostTask in AndroidNetworkMonitor Niels Möller 2021-03-12 14:08:23 +01:00
  • 048e9c2f45 In full svc controller reuse unused frame configuration Danil Chapovalov 2021-03-12 13:57:54 +01:00
  • 8da67f6165 In ksvc controller reuse unused frame configuration Danil Chapovalov 2021-03-12 12:56:55 +01:00
  • 8647340436 Introduce WebRTC documentation structure and how-to Artem Titov 2021-03-12 12:42:24 +01:00
  • cf70793c5f Update WebRTC code version (2021-03-12T04:03:49). webrtc-version-updater 2021-03-11 20:03:51 -08:00
  • 8973655f42 measure ice candidate poolsize setting for different bundle policys Philipp Hancke 2021-03-05 12:07:05 +01:00
  • 727d2afc43 Revert "Parse encoded frame QP if not provided by encoder" Sergey Silkin 2021-03-11 17:05:44 +00:00
  • 2d9f53ca58 Expose addIceCandidate with completion handler. Yura Yaroshevich 2021-03-10 13:03:00 +03:00
  • 31c5c9da35 Revert "Reland "Enable quality scaling when allowed"" Ilya Nikolaevskiy 2021-03-11 15:13:57 +00:00
  • 0021fe7793 Reland "Enable quality scaling when allowed" Sergey Silkin 2021-03-11 13:28:23 +01:00
  • 7bf29bc3ed Roll chromium_revision fc9c86fd36..c3fb27225e (861807:861941) chromium-webrtc-autoroll 2021-03-11 04:06:28 -08:00
  • b7227a5a10 Fix handling of partial match for GetVpnUnderlyingAdapterType Jonas Oreland 2021-03-11 12:52:52 +01:00
  • fd1e9d1af4 [Stats] Add minimum RTCReceivedRtpStreamStats with jitter and packetsLost Di Wu 2021-03-09 09:25:28 -08:00
  • 8639673f0c Parse encoded frame QP if not provided by encoder Sergey Silkin 2021-03-11 08:07:37 +01:00
  • 8097935df3 Revert "Reduce complexity in the APM pipeline when the output is not used" Ilya Nikolaevskiy 2021-03-11 11:40:46 +00:00
  • be140b4187 Change ObjCNetworkMonitor::OnPathUpdate to use PostTask Niels Möller 2021-03-11 10:24:06 +01:00
  • aa6adffba3 Reduce complexity in the APM pipeline when the output is not used Per Åhgren 2021-03-11 08:57:07 +00:00
  • 54dbc3be3f Revert "[Battery]: Delay start of TaskQueuePacedSender." Artem Titov 2021-03-11 09:00:15 +00:00
  • 3135772326 Changed setActive of RTCAudio Session, and it's working Abby Yeh 2021-03-09 01:55:31 +08:00
  • fbb2dcfe38 Roll chromium_revision 43d5378f7f..fc9c86fd36 (861531:861807) chromium-webrtc-autoroll 2021-03-10 18:12:48 -08:00
  • b2e71b8b35 Reland "Fix race between destroying SctpTransport and receiving notification on timer thread." Taylor Brandstetter 2021-03-08 15:36:53 -08:00
  • 34fdc92119 Add audioproc_f support for testing the runtime settings of whether the output is used Per Åhgren 2021-03-10 09:51:52 +00:00
  • 3d792e990a Add controls for MTU size of virtual socket server Harald Alvestrand 2021-03-10 07:29:28 +00:00
  • bccfd26322 Allow webrtc mac cross compile Andrey Logvin 2021-03-08 19:12:20 +00:00
  • 048adc7136 Add missing remote-outbound stats to RTCPReceiver::NTP Alessio Bazzica 2021-03-10 15:05:55 +01:00
  • da2fd2a2b2 Fix problem with ipv4 over ipv6 on Android Jonas Oreland 2021-03-10 15:33:31 +01:00
  • 89cb65ed66 [Battery]: Delay start of TaskQueuePacedSender. Etienne Pierre-Doray 2021-02-23 15:31:19 -05:00
  • 79011ef4a7 Remove ModuleRtpRtcpImpl2::LastReceivedNTP Alessio Bazzica 2021-03-10 14:52:35 +01:00
  • ee8cd20ec5 Add a mutex free implementation of webrtc::ReceiveStatistics Per Kjellander 2021-03-10 12:31:38 +01:00
  • bb22066e60 Roll chromium_revision e1b9354ff4..43d5378f7f (854007:861531) chromium-webrtc-autoroll 2021-03-10 04:11:41 -08:00
  • 213dc2cfc5 Temporarily disable Opus decode test. Jakob Ivarsson 2021-03-10 12:38:34 +01:00
  • 8bfa2756a5 Fix nullability of completion handlers in iOS SDK. Yura Yaroshevich 2021-03-10 13:07:27 +03:00
  • 89127190ce Revert "Roll chromium_revision e1b9354ff4..74090df66c (854007:861144)" Ilya Nikolaevskiy 2021-03-10 10:02:53 +00:00
  • 6adb8d9282 Revert "Roll chromium_revision 74090df66c..f86a579769 (861144:861254)" Ilya Nikolaevskiy 2021-03-10 10:08:24 +00:00
  • 734ae5296b Revert "Roll chromium_revision f86a579769..dc3e6d8b69 (861254:861387)" Ilya Nikolaevskiy 2021-03-10 10:08:37 +00:00
  • 6a55e7307b Stop inheriting from has_slots in DtlsTransport. Mirko Bonadei 2021-02-27 09:19:32 +01:00
  • 68ef4e5b00 Update WebRTC code version (2021-03-10T04:02:37). webrtc-version-updater 2021-03-09 20:02:40 -08:00
  • 8114016f59 Roll chromium_revision f86a579769..dc3e6d8b69 (861254:861387) chromium-webrtc-autoroll 2021-03-09 18:02:13 -08:00
  • 14a626af56 Roll chromium_revision 74090df66c..f86a579769 (861144:861254) chromium-webrtc-autoroll 2021-03-09 12:03:07 -08:00
  • 5265b9367a Add build-id to libjingle_peerconnection_so.so Raman Budny 2021-03-09 18:59:35 +03:00
  • cb7ff13432 Roll chromium_revision e1b9354ff4..74090df66c (854007:861144) Artem Titov 2021-03-09 15:43:46 +01:00
  • dc08aea610 Fix chromium roll into WebRTC Artem Titov 2021-03-09 14:17:31 +01:00
  • c84a887b8a Allow port 80 for TURN servers Harald Alvestrand 2021-03-09 08:14:02 +00:00
  • 668dbf66ce [Stats] Populate "frames" stats for video source. Di Wu 2021-02-27 00:29:15 -08:00
  • 5ab9b32ee6 Update WebRTC code version (2021-03-09T04:03:43). webrtc-version-updater 2021-03-08 20:03:55 -08:00
  • 92d12707e0 Expose PeerConnection.restartIce in iOS SDK. Yura Yaroshevich 2021-03-06 23:55:17 +03:00
  • d67253532f Expose parameterless setLocalDescription() in iOS SDK. Yura Yaroshevich 2021-03-05 18:21:16 +03:00
  • f00cd533ae Do more actions on SDP fuzzing. Harald Alvestrand 2021-03-08 09:46:28 +00:00
  • a113d231a6 srtp: use srtp_crypto_policy_set_from_profile_for_* from libsrtp Philipp Hancke 2021-02-22 15:01:42 +01:00
  • c81665cf9d Change AndroidNetworkMonitor::NotifyConnectionTypeChanged to use Invoke Niels Möller 2021-03-02 11:22:59 +01:00
  • 82a94125a4 Reland "Add a fuzzer test that tries to connect a PeerConnection." Harald Alvestrand 2021-03-08 07:01:48 +00:00
  • be66d95ab7 srtp: document rationale for srtp overhead calculation Philipp Hancke 2021-03-01 14:56:22 +01:00
  • 456a2642d3 Update WebRTC code version (2021-03-08T04:01:44). webrtc-version-updater 2021-03-07 20:01:46 -08:00
  • aedb0f0c66 Update WebRTC code version (2021-03-07T04:02:41). webrtc-version-updater 2021-03-06 20:02:44 -08:00
  • eb449a979b Revert "Reland "Enable quality scaling when allowed"" Guido Urdaneta 2021-03-06 04:20:14 +00:00
  • 77f97ec066 Update WebRTC code version (2021-03-06T04:02:56). webrtc-version-updater 2021-03-05 20:02:59 -08:00
  • 964a88fce1 Prevent possible out-of-bounds access Sergey Silkin 2021-03-05 15:17:28 +01:00
  • 4a4273bf05 VP9 ResolutionBitrateLimits: If bitrates are configured, use intersection. Åsa Persson 2021-03-05 16:03:37 +01:00
  • a86b29be01 Add VP9-specific default resolution bitrate limits Sergey Silkin 2021-03-05 13:29:19 +01:00
  • 1a89bc8b62 build: improve rtc_include_tests documentation Philipp Hancke 2021-03-02 21:23:24 +01:00
  • 1413e2db4c Update WebRTC code version (2021-03-05T04:03:29). webrtc-version-updater 2021-03-04 20:03:31 -08:00
  • 8a38b1cf68 Revert "Fix race between destroying SctpTransport and receiving notification on timer thread." Taylor 2021-03-05 01:05:38 +00:00
  • 83be84bb74 Reland "Enable quality scaling when allowed" Sergey Silkin 2021-03-04 11:42:48 +01:00
  • d140c8f43b Added missing nullable annotations to iOS SDK. Yura Yaroshevich 2021-03-02 23:25:10 +03:00
  • 854d59f750 Temporarily disable remaining Opus bit exactness tests. Jakob Ivarsson 2021-03-04 13:05:19 +01:00
  • 3208bf102b Allow uninstantiated parametrized tests in data_channel_integration_tests.cc Bjorn Terelius 2021-03-04 10:53:08 +01:00
  • 609b524dd3 Revert "Enable quality scaling when allowed" Björn Terelius 2021-03-04 09:46:53 +00:00
  • ae44fde188 Revert "Add a fuzzer test that tries to connect a PeerConnection." Evan Shrubsole 2021-03-04 08:50:47 +00:00
  • d744c5a9e4 Update WebRTC code version (2021-03-04T04:02:19). webrtc-version-updater 2021-03-03 20:02:22 -08:00
  • 8cfb287735 Add AV1 encoder&decoder wrappers for iOS SDK. Yura Yaroshevich 2021-03-03 12:29:42 +03:00
  • 662d4c328f AV1 test: change ssim threshold Jerome Jiang 2021-03-03 09:52:43 -08:00
  • 83e6eceac7 Comment out uninstantiated parametrized PC full stack test Bjorn Terelius 2021-03-03 17:53:45 +01:00
  • e7a55813f9 Temporarily disable some Opus bit exactness tests. Jakob Ivarsson 2021-03-03 14:18:15 +01:00
  • b6b782da68 Fix potential unsafe access to VCMTimestampMap::data Johannes Kron 2021-03-03 14:39:44 +01:00
  • 752cbaba90 Enable quality scaling when allowed Sergey Silkin 2021-03-03 08:24:02 +01:00
  • db0b4a8935 Do not crash if codec is not available Sergey Silkin 2021-03-03 13:34:49 +01:00
  • 652ada5029 Enabling a safe fall-back functionality for overruns in the runtime settings Per Åhgren 2021-03-03 10:52:44 +00:00
  • 99bcf60a41 Check MID for illegal token characters. Harald Alvestrand 2021-03-03 07:44:39 +00:00
  • c67b77eee4 Add a fuzzer test that tries to connect a PeerConnection. Harald Alvestrand 2021-03-02 15:32:38 +00:00
  • 94f77ebbc5 Allow using test code in fuzzers Harald Alvestrand 2021-03-02 15:15:08 +00:00
  • 09932cded8 Manually update jdk DEPS. Bjorn Terelius 2021-03-02 12:59:43 +01:00
  • 07d83c8a9a Modified STUN verification functions Harald Alvestrand 2021-03-02 08:09:53 +00:00
  • d99da804d7 Parent CL. Lahiru Ginnaliya Gamathige 2021-03-02 01:03:51 -08:00
  • a88fe7be14 Fix race between destroying SctpTransport and receiving notification on timer thread. Taylor Brandstetter 2021-02-26 12:56:36 -08:00
  • 88a51b2902 Populate "total_round_trip_time" and "round_trip_time_measurements" for remote inbound RTP streams Di Wu 2021-03-01 11:22:06 -08:00
  • 625e1d9b16 VP8 ref finder unittest cleanup philipel 2021-03-01 17:15:03 +01:00
  • 86f04ad135 Populate “fractionLost” stats for remote inbound rtp streams Di Wu 2021-02-28 23:36:03 -08:00
  • 6512af04ba Add root doc site definition for WebRTC documentation Artem Titov 2021-03-01 17:30:29 +01:00
  • f57f2cd3d2 doc: add M88/M89 release notes Philipp Hancke 2021-03-01 14:49:54 +01:00
  • 64f7da0b16 Use input_state to get pixels for single active stream. Åsa Persson 2021-03-01 09:38:57 +01:00
  • 31e06cb63d addIceCandidate: prefer ice candidate sdpMid over sdpMLineIndex Philipp Hancke 2021-02-26 09:23:53 +01:00
  • df6d4ca3f5 Prevent TURN from connecting to ports < 1024 (except 443) Harald Alvestrand 2021-02-25 11:18:19 +00:00
  • bfc3ef0fec Mention ninja all in webrtc development documantation. Andrey Logvin 2021-03-01 12:01:11 +00:00
  • 4785402475 Replace RecursiveCriticalSection with Mutex in ProcessThreadImpl Niels Möller 2021-03-01 11:31:33 +01:00
  • 1e2da374b3 Update WebRTC code version (2021-02-28T04:02:30). webrtc-version-updater 2021-02-27 20:02:33 -08:00
  • ac615b6766 Update WebRTC code version (2021-02-27T04:03:14). webrtc-version-updater 2021-02-26 20:03:16 -08:00
  • c96aa30c13 Minor refactoring of RtpVideoSender Erik Språng 2021-02-25 17:05:21 +01:00