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  • 14cad9fa35 Fix clang-tidy: performance-inefficient-vector-operation. Mirko Bonadei 2021-02-03 09:48:01 +01:00
  • d5827024a0 Add hta@ to WebRTC's root OWNERS. Mirko Bonadei 2021-02-02 10:24:07 +01:00
  • b5823055be In VP9 encoder avoid crashing when encoder produce an unexpected frame Danil Chapovalov 2021-02-03 12:33:17 +01:00
  • 53610223a8 Delete unused function webrtc::AudioProcessing::MutateConfig Sam Zackrisson 2021-02-03 11:04:55 +01:00
  • 6f75f6b3bd APM: add AGC2 SIMD kill switches in AudioProcessing::Config::ToString() Alessio Bazzica 2021-02-02 16:52:39 +01:00
  • 47ec157fbf Revert "Prepare to avoid hops to worker for network events." Mirko Bonadei 2021-02-03 12:08:07 +00:00
  • 9111bd18b1 LibvpxVp8Encoder: add option to configure resolution_bitrate_limits. Åsa Persson 2021-02-03 10:16:30 +01:00
  • 22e37d8857 Don't log a message that a field is missing if the field trial key starts with "_" Ying Wang 2021-02-02 15:36:16 +01:00
  • 76a1041f0f Revert "Fix data race for config_ in AudioSendStream" Henrik Boström 2021-02-02 15:02:25 +00:00
  • 0e3cb9fb20 Create and initialize encoders only for active streams Sergey Silkin 2021-02-02 18:23:40 +01:00
  • 312ea0e144 Roll chromium_revision a0e7a1a1f9..415eaa7c56 (849895:850009) chromium-webrtc-autoroll 2021-02-03 00:01:54 -08:00
  • 6862818c56 Roll chromium_revision e753f3f38e..a0e7a1a1f9 (849529:849895) chromium-webrtc-autoroll 2021-02-02 18:03:28 -08:00
  • d48a2b14e7 Prepare to avoid hops to worker for network events. Tomas Gunnarsson 2021-02-02 17:57:36 +01:00
  • 16ab60c4c6 Use CallbackList in DtlsHandshakeError in dtls_transport. Lahiru Ginnaliya Gamathige 2021-02-02 07:27:09 -08:00
  • c8421c4c3e Replace rtc::ThreadChecker with webrtc::SequenceChecker Artem Titov 2021-02-02 10:57:19 +01:00
  • 7358b40f44 Remove usage of AsyncInvoker in test class FakeNetworkSocket Danil Chapovalov 2021-02-01 20:43:19 +01:00
  • eee0e9e9d4 Remove passing rtp packet metadata through webrtc as array of bytes Danil Chapovalov 2021-01-27 18:05:04 +01:00
  • a1ca64c5ed Roll chromium_revision 24c8af75a9..e753f3f38e (849426:849529) chromium-webrtc-autoroll 2021-02-02 02:02:57 -08:00
  • 3f412945f0 Update WebRTC code version (2021-02-02T04:02:53). webrtc-version-updater 2021-02-01 20:02:55 -08:00
  • 84635021c6 Roll chromium_revision 23e1598eba..24c8af75a9 (849278:849426) chromium-webrtc-autoroll 2021-02-01 20:02:04 -08:00
  • 2ab9b28c52 Get rid of unnecessary network thread Invoke in BaseChannel. Taylor Brandstetter 2021-02-01 14:39:07 -08:00
  • d1d2dc7c05 Roll chromium_revision 6aa02eda73..23e1598eba (849160:849278) chromium-webrtc-autoroll 2021-02-01 14:03:08 -08:00
  • 0f009dc13d Roll chromium_revision 402f104a74..6aa02eda73 (849004:849160) chromium-webrtc-autoroll 2021-02-01 10:02:32 -08:00
  • e7ded686d5 Fix integer overflow. Jakob Ivarsson 2021-02-01 17:09:33 +01:00
  • 78f87ab106 Delete use of RecursiveCriticalSection in JsepTransport Niels Möller 2021-02-01 12:19:19 +01:00
  • 51e5c4b0f4 Fix data race for config_ in AudioSendStream Artem Titov 2021-01-29 16:00:05 +01:00
  • e7c79fd3d6 Remove from chromium build targets that are not compatible with it. Andrey Logvin 2021-02-01 09:56:37 +00:00
  • d6604df27f Revert "Enable Video-QualityScaling experiment by default" Ilya Nikolaevskiy 2021-02-01 12:01:06 +00:00
  • 3faea70d1a allow empty scalability mode in AV1 encoder Sergio Garcia Murillo 2021-01-29 10:26:28 +01:00
  • c91c4233e3 LibvpxVp9Encoder: add option to configure resolution_bitrate_limits. Åsa Persson 2021-02-01 09:20:05 +01:00
  • 989e6e7d22 Switch WebRTC's MB to RBE-CAS. Mirko Bonadei 2021-01-29 14:34:52 +01:00
  • b853d72250 Update Apple device list Dave Cowart 2021-01-19 15:29:11 -06:00
  • 1f1e190acc Roll chromium_revision 878a605f67..402f104a74 (848896:849004) chromium-webrtc-autoroll 2021-01-31 22:02:04 -08:00
  • 22b5efa7d6 Update WebRTC code version (2021-02-01T04:03:13). webrtc-version-updater 2021-01-31 20:03:15 -08:00
  • 41bfcf4a63 Inject network thread to Call. Tomas Gunnarsson 2021-01-30 16:15:21 +01:00
  • cedc3c7200 Update WebRTC code version (2021-01-31T04:03:17). webrtc-version-updater 2021-01-30 20:03:29 -08:00
  • 5ac4212bec Roll chromium_revision 2e446035f5..878a605f67 (848796:848896) chromium-webrtc-autoroll 2021-01-30 12:02:22 -08:00
  • 692f56591f Update WebRTC code version (2021-01-30T04:04:03). webrtc-version-updater 2021-01-29 20:04:05 -08:00
  • b28d6ca755 Roll chromium_revision a635fd2809..2e446035f5 (848661:848796) chromium-webrtc-autoroll 2021-01-29 16:01:47 -08:00
  • d0acbd8645 Revert "Do all BaseChannel operations within a single Thread::Invoke." Taylor Brandstetter 2021-01-25 13:44:55 -08:00
  • 271adffe63 Roll chromium_revision 3353629fad..a635fd2809 (848531:848661) chromium-webrtc-autoroll 2021-01-29 12:02:28 -08:00
  • f9a6148e04 Roll chromium_revision bbd3f0121d..3353629fad (848401:848531) chromium-webrtc-autoroll 2021-01-29 08:02:29 -08:00
  • 7864600a6e Add absl_deps field for rtc_test and rtc_executable Andrey Logvin 2021-01-29 10:50:19 +00:00
  • b79acd8ce9 Format webrtc/modules/audio_processing/transient/BUILD.gn file Andrey Logvin 2021-01-29 10:36:11 +00:00
  • ee8c275fd9 Make DVQA CPU usage tests more stable Andrey Logvin 2021-01-29 15:27:10 +00:00
  • 5761e7b3ff Running apply-iwyu on ~all files in pc/ Harald Alvestrand 2021-01-29 14:45:08 +00:00
  • 5e227abfe9 Move under enable_google_benchmarks targets that rely on the benchmarks Andrey Logvin 2021-01-29 10:34:55 +00:00
  • 133c0524f3 Make the config_ member of JsepTransportController const Harald Alvestrand 2021-01-29 14:27:12 +00:00
  • 9673ca42ea Add field trial for bitrate limit interpolation for simulcast resolutions <180p. Rasmus Brandt 2021-01-29 13:21:16 +01:00
  • c5bdac64cf Fix call_tests target dependencies Andrey Logvin 2021-01-29 10:33:53 +00:00
  • cd467b51ab sdp: check that sctp is on a application content type Philipp Hancke 2021-01-29 09:29:47 +01:00
  • 1a29a5da84 Delete rtc::Bind Niels Möller 2021-01-18 11:35:23 +01:00
  • 4ea26e5663 Update WebRTC code version (2021-01-29T04:02:57). webrtc-version-updater 2021-01-28 20:02:59 -08:00
  • 5e32fb84a1 Roll chromium_revision 0584f34f9c..bbd3f0121d (848277:848401) chromium-webrtc-autoroll 2021-01-28 20:02:23 -08:00
  • 9e9bf75dc1 Add comment about setting transport_name field for RemoveIceCandidates. Taylor Brandstetter 2021-01-26 14:55:38 -08:00
  • 8bd0f97fe6 Address CL comments from 200161. Austin Orion 2021-01-28 12:18:26 -08:00
  • 79f645237d Roll chromium_revision 3042ccda4e..0584f34f9c (848168:848277) chromium-webrtc-autoroll 2021-01-28 14:02:48 -08:00
  • 76bbc98d72 Adding MockVoipEngine for downstream project's tests Tim Na 2021-01-28 10:52:38 -08:00
  • 4f5322cced Roll chromium_revision ba13ceb157..3042ccda4e (848044:848168) chromium-webrtc-autoroll 2021-01-28 10:01:37 -08:00
  • 066b5b6ed7 Enable Video-QualityScaling experiment by default Ilya Nikolaevskiy 2021-01-28 14:58:24 +01:00
  • 54b925cfc2 add metrics for bundle usage Philipp Hancke 2021-01-28 09:56:39 +01:00
  • 1aa1d6463a Ensure VideoLayersAllocation.frame_rate_fps can not overflow Per Kjellander 2021-01-28 08:56:43 +01:00
  • 37dfddd5e8 Avoid treating VP8 key frame in simulcast as delta frame Danil Chapovalov 2021-01-27 19:24:23 +01:00
  • 075fd4b7f9 Roll chromium_revision 84c1288826..ba13ceb157 (847739:848044) chromium-webrtc-autoroll 2021-01-28 02:06:34 -08:00
  • 3b68aa346a Move some RTC_LOG to RTC_DLOG. Mirko Bonadei 2021-01-28 09:21:06 +01:00
  • 70f9e249d5 Remove DtlsHandShakeError and replace it with a Function Pointer. Lahiru Ginnaliya Gamathige 2021-01-27 23:32:46 -08:00
  • b70c9531ee sdp: cross-check media type and protocol earlier Philipp Hancke 2021-01-13 10:40:06 +01:00
  • 2aad81259e Refactor and implement WgcCapturerWin, a source agnostic capturer. Austin Orion 2021-01-07 14:34:31 -08:00
  • ca5d4a4152 Roll chromium_revision 61b8ff5c89..84c1288826 (847529:847739) chromium-webrtc-autoroll 2021-01-27 12:04:46 -08:00
  • d19d0cf8ee Reland: Add ability to load CreateDirect3DDeviceFromDXGIDevice from d3d11.dll Austin Orion 2021-01-27 09:40:21 -08:00
  • 49dbad021e Fixing audio timestamp stall during inactivation (under a kill switch) Minyue Li 2021-01-26 12:50:06 +01:00
  • 14b0e73bd8 Roll chromium_revision e3ed290da5..61b8ff5c89 (846763:847529) Artem Titov 2021-01-27 09:16:07 +01:00
  • 934b5e2ac6 Update WebRTC code version (2021-01-27T04:02:39). webrtc-version-updater 2021-01-26 20:02:54 -08:00
  • fae4fb1345 video_replay: add support for IVF file output Philipp Hancke 2021-01-26 18:05:56 +01:00
  • 1d77c3ef17 Fix roll chromium_revision 18311e2720..e3ed290da5 (844473:846763) Mirko Bonadei 2021-01-26 17:10:04 +01:00
  • 103876f379 av1: turn off a few tools that are not used for rtc Jerome Jiang 2021-01-25 16:42:51 -08:00
  • cc8a1f8450 Add API to get current time mode from NetworkEmulationManager Artem Titov 2021-01-26 13:58:23 +01:00
  • 08f46909a8 Protect DefaultVideoQualityAnalyzer::peers_ with lock Artem Titov 2021-01-25 23:52:57 +01:00
  • c57089a97a Add new peer to injector when adding it to analyzer. Removed unused injector Artem Titov 2021-01-25 23:05:19 +01:00
  • 4f3a2eba6b Destroy previous offer instead of leaking it in PeerConnectionInterfaceTest.ExtmapAllowMixedIsConfigurable Artem Titov 2021-01-26 09:58:21 +01:00
  • 0a03ed8492 Update WebRTC code version (2021-01-26T04:02:59). webrtc-version-updater 2021-01-25 20:03:01 -08:00
  • 5312a8f532 Add option to attach custom object to an rtp packet Danil Chapovalov 2021-01-25 17:02:57 +01:00
  • ded6636cf4 Cleanup RtcpSender from legacy functionality Danil Chapovalov 2021-01-25 12:40:01 +01:00
  • 54fb32a15e IvfFileReader: Fix SpatialIndex values Florent Castelli 2021-01-25 12:53:30 +01:00
  • 437843f9a5 Update WebRTC code version (2021-01-25T04:04:10). webrtc-version-updater 2021-01-24 20:04:13 -08:00
  • b1b79f7012 Update WebRTC code version (2021-01-24T04:03:52). webrtc-version-updater 2021-01-23 20:03:55 -08:00
  • 90776cb0ef Enable RTC_NO_UNIQUE_ADDRESS on MSan builds. Mirko Bonadei 2021-01-23 10:17:02 +01:00
  • ef53a7fc0b Reset IO thread checker when iOS audio unit stops Brian Dai 2021-01-20 12:20:12 -08:00
  • bf95da8e73 Update WebRTC code version (2021-01-23T04:04:31). webrtc-version-updater 2021-01-22 20:04:34 -08:00
  • 8c007fffea Restrict usage of resolution bitrate limits to singlecast Sergey Silkin 2021-01-22 18:59:59 +01:00
  • 461b1d903f Restart CPU overuse detection when encoder settings has changed. Jakob Ivarsson 2021-01-22 16:27:43 +01:00
  • 2803a2def1 Make audio device mocks publicly visible Steve Anton 2021-01-21 18:56:01 -08:00
  • 8df643b387 Introduce FinalRefCountedObject template class Danil Chapovalov 2021-01-22 16:11:10 +01:00
  • cbacec52bc Monitor the "concealed samples" stat for the audio during negotiation. Harald Alvestrand 2021-01-14 13:13:42 +00:00
  • 11215febb9 Require scalability mode to initialize av1 encoder. Danil Chapovalov 2021-01-21 15:56:05 +01:00
  • 2ed56fefef Update WebRTC code version (2021-01-22T04:03:26). webrtc-version-updater 2021-01-21 20:03:29 -08:00
  • d2dd732d83 Introduce network emulated endpoint optional name for better logging Artem Titov 2021-01-21 17:28:17 +01:00
  • e4fd1ba319 Delete mutable rtc::CopyOnWriteBuffer::data Danil Chapovalov 2021-01-21 17:00:16 +01:00
  • 6031b74664 Implement a Neon optimized function to find the argmax element in an array. Ivo Creusen 2021-01-20 16:26:43 +01:00
  • 03eed7c8d0 Fixes issue triggered by WebRTC-VP9-PerformanceFlags trial. Erik Språng 2021-01-20 22:12:33 +01:00