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  • a7e34d33fe Add resolution_bitrate_limits to EncoderInfo field trial. Åsa Persson 2021-01-20 15:36:13 +01:00
  • 026ad9ac22 Update WebRTC code version (2021-01-21T04:03:14). webrtc-version-updater 2021-01-20 20:03:17 -08:00
  • 49b20f919a Fix race with SctpTransport destruction and usrsctp timer thread. Taylor Brandstetter 2021-01-19 19:05:17 -08:00
  • c2ae4c8a37 Allow separate dump sets for the data dumper in APM Per Åhgren 2021-01-20 15:42:13 +01:00
  • 0be1846477 Fix enabling DependencyDescriptor for VP9 with spatial layers Danil Chapovalov 2021-01-20 12:11:29 +01:00
  • 1657baf67b Add .cache to .gitignore. Rasmus Brandt 2021-01-20 15:06:45 +01:00
  • 4f281f142a Cleanup FakeRtcEventLog from thread awareness Danil Chapovalov 2021-01-18 13:29:00 +01:00
  • 812c73cdc2 Another ilbc cross correlation fix Ivo Creusen 2021-01-18 16:25:22 +01:00
  • 5eb43b4777 Prefix HAVE_SCTP macro with WEBRTC_. Mirko Bonadei 2021-01-18 13:24:40 +01:00
  • 6dcbcead3d Update WebRTC code version (2021-01-20T04:04:26). webrtc-version-updater 2021-01-19 20:04:35 -08:00
  • 33c0ab4948 Call MediaChannel::OnPacketReceived on the network thread. Tomas Gunnarsson 2021-01-18 10:49:05 +01:00
  • 1cbf21e157 ChannelStatistics RTT test case around remote SSRC change. Tim Na 2021-01-15 15:54:07 -08:00
  • a24d35eb09 AlignmentAdjuster: take reduced layers into account for default downscaling. Åsa Persson 2021-01-19 10:25:10 +01:00
  • 5c3ff6b60f Switch to enable the HMM transparent mode classifier Gustaf Ullberg 2021-01-19 15:33:50 +01:00
  • 801c9995c8 Signal extmap-allow-mixed by default on session level Emil Lundmark 2021-01-19 13:06:32 +01:00
  • 1e75df26e3 Remove lock from UlpfecReceiverImpl and replace with a sequence checker. Tomas Gunnarsson 2021-01-19 13:30:23 +01:00
  • 5eb527cf7f Replace sigslot usages with callback list library. Lahiru Ginnaliya Gamathige 2021-01-18 23:32:22 -08:00
  • a722d2ac1e Add DeliverPacketAsync method to PacketReceiver. Tomas Gunnarsson 2021-01-19 09:00:18 +01:00
  • 6cdb67fb33 Document expected thread safety of the TaskQueue interface Danil Chapovalov 2021-01-18 17:02:55 +01:00
  • 676d61fb3b Update WebRTC code version (2021-01-19T04:01:32). webrtc-version-updater 2021-01-18 20:01:34 -08:00
  • 29bd8638ad Add field trial for allowing cropped resolution when limiting max layers. Åsa Persson 2021-01-18 09:15:07 +01:00
  • 5cf0ef0022 Stricter compile-time thread annotations in JsepTransportController Niels Möller 2021-01-18 14:00:36 +01:00
  • 25b8235f03 Remove unused function VideoDecoder::PrefersLateDecoding. philipel 2021-01-14 21:49:06 +01:00
  • 42eef86c4f Remove unused code in APM Alessio Bazzica 2021-01-15 16:41:48 +01:00
  • 111a3712e7 Delete unused.h include from api as unused Danil Chapovalov 2021-01-18 11:13:30 +01:00
  • 3e9cb2cbf2 Move deprecated code to their own build targets. Niels Möller 2021-01-14 12:00:57 +01:00
  • 844c759766 fix variable naming in ReportSdpFormatReceived Philipp Hancke 2021-01-18 11:25:31 +01:00
  • ee95c1c8c7 Roll chromium_revision da24822732..18311e2720 (844357:844473) chromium-webrtc-autoroll 2021-01-18 02:08:14 -08:00
  • 77ceff9276 payload type mapper: use media constants Philipp Hancke 2021-01-15 08:55:24 +01:00
  • 4bab23f550 Update pc/ to use C++ lambdas instead of rtc::Bind Niels Möller 2021-01-18 09:24:33 +01:00
  • d51000f4a8 Delete RTC_WARN_UNUSED_RESULT as no longer used Danil Chapovalov 2021-01-15 19:47:23 +01:00
  • cc6ae44ae6 Reland "Improve structuring of test for audio glitches." Harald Alvestrand 2021-01-18 08:06:23 +00:00
  • c20e3332ed Update SCTP test to use C++ lambdas instead of rtc::Bind Niels Möller 2021-01-14 10:48:27 +01:00
  • 588526c278 Remove deprecated //rtc_base:async_resolver. Mirko Bonadei 2021-01-18 08:52:33 +01:00
  • e091fd21d6 Remove lock from RtpStreamReceiverController. Tomas Gunnarsson 2021-01-17 16:14:52 +01:00
  • 8467cf27ac Reduce redundant flags for audio stream playout state. Tomas Gunnarsson 2021-01-17 14:36:44 +01:00
  • 12971a22d2 Update WebRTC code version (2021-01-18T04:03:42). webrtc-version-updater 2021-01-17 20:03:45 -08:00
  • 09729d2ecf Update WebRTC code version (2021-01-17T04:01:53). webrtc-version-updater 2021-01-16 20:01:58 -08:00
  • 9c8dd87636 Fixing WebRTC/Chromium FYI build. Mirko Bonadei 2021-01-16 18:52:24 +01:00
  • be93b788c6 Move iOS bundle data for tests inside rtc_include_test (take 2). Mirko Bonadei 2021-01-16 13:51:25 +01:00
  • 2397b6e75d SimulcastEncoderAdapter: Add field trial for EncoderInfo settings. Åsa Persson 2021-01-15 15:03:31 +01:00
  • fd9500e3b5 In criket::BaseChannel replace AsyncInvoker with task queue functions Danil Chapovalov 2021-01-15 17:29:53 +01:00
  • 8ed61858c3 Move iOS bundle data for tests inside rtc_include_test. Mirko Bonadei 2021-01-16 10:39:20 +01:00
  • d705bbed2c Roll chromium_revision 3c2d1e3ba1..da24822732 (844134:844357) chromium-webrtc-autoroll 2021-01-16 00:08:31 -08:00
  • f4d1ecf85f Update WebRTC code version (2021-01-16T04:03:13). webrtc-version-updater 2021-01-15 20:03:15 -08:00
  • 61ede7ab71 Roll chromium_revision 72fe6d7aab..3c2d1e3ba1 (844008:844134) chromium-webrtc-autoroll 2021-01-15 10:03:22 -08:00
  • d723da1943 Reland "Default enable delay adaptation during DTX." Jakob Ivarsson 2021-01-15 17:44:56 +01:00
  • 098da17f35 Reland "Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in c++ code" Danil Chapovalov 2021-01-14 16:15:31 +01:00
  • ba91dbcb3e In SVC controllers add support for frames dropped by encoder Danil Chapovalov 2021-01-15 16:46:11 +01:00
  • e5f4c6b8d2 Reland "Refactor rtc_base build targets." Mirko Bonadei 2021-01-15 10:41:01 +01:00
  • 79d9c373c5 Revert "Default enable delay adaptation during DTX." Jakob Ivarsson 2021-01-15 15:57:58 +00:00
  • 59bdcbe3c9 Default enable delay adaptation during DTX. Jakob Ivarsson 2021-01-15 14:39:26 +01:00
  • 5ab6a8cea9 Refactors SimulcastEncoder Adapter. Erik Språng 2021-01-14 16:37:26 +01:00
  • 1528e2b3a7 Set AV1E_SET_ERROR_RESILIENT_MODE on T1 and T2 enhanced layers Sergio Garcia Murillo 2021-01-14 12:55:32 +01:00
  • 809a261631 Roll chromium_revision 42ab9dc8c8..72fe6d7aab (843550:844008) chromium-webrtc-autoroll 2021-01-15 02:02:19 -08:00
  • a86cef7e2c Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in audio_coding Danil Chapovalov 2021-01-14 18:35:41 +01:00
  • 87e9f6e666 Update p2p/ to use C++ lambdas instead of rtc::Bind Niels Möller 2021-01-14 12:20:27 +01:00
  • 3ae09f5419 Revert "Improve structuring of test for audio glitches." Alex Loiko 2021-01-15 09:58:50 +00:00
  • 98db5d1a11 Revert "Add ability to load CreateDirect3DDeviceFromDXGIDevice from d3d11.dll" Alex Loiko 2021-01-15 09:41:56 +00:00
  • b45d3aa30e Update android jni code to use C++ lambdas instead of rtc::Bind Niels Möller 2021-01-14 16:46:25 +01:00
  • d76dcbd963 Simplify FakeRtcEventLog, delete rtc::Bind usage Niels Möller 2021-01-14 09:50:26 +01:00
  • f9ee0e08ec Add cross trafic emulation api Andrey Logvin 2021-01-14 09:50:32 +00:00
  • a7ccacc87f Update WebRTC code version (2021-01-15T04:04:08). webrtc-version-updater 2021-01-14 20:04:10 -08:00
  • 6c8738ce6e mb: Fully remove references to 'masters' in favor of 'builder_groups'. Mirko Bonadei 2021-01-14 09:21:32 +01:00
  • 7acc2d9fe3 Revert "Refactor rtc_base build targets." Mirko Bonadei 2021-01-14 21:03:55 +00:00
  • 884118dad1 Delete unused functions in ModuleRtpRtcpImpl Danil Chapovalov 2021-01-12 21:18:26 +01:00
  • 23f60ebe01 Add ability to load CreateDirect3DDeviceFromDXGIDevice from d3d11.dll Austin Orion 2021-01-13 12:01:58 -08:00
  • c17bca7410 SetOfferedRtpHeaderExtensions: fix error code. Markus Handell 2021-01-14 17:08:01 +01:00
  • 1669103833 Roll chromium_revision 189823ba75..42ab9dc8c8 (843026:843550) chromium-webrtc-autoroll 2021-01-14 08:02:45 -08:00
  • f77aa81e54 Update AudioDeviceBuffer to use C++ lambdas instead of rtc::Bind Niels Möller 2021-01-14 11:50:19 +01:00
  • 42c0d707dd Include packetization in video codec string Emil Lundmark 2021-01-13 16:50:18 +01:00
  • 4319b1695e Revert "Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in c++ code" Danil Chapovalov 2021-01-14 15:02:22 +00:00
  • 8c2250eddc Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in c++ code Danil Chapovalov 2021-01-14 13:31:46 +01:00
  • 1921708141 SetNegotiatedHeaderExtensions_w: Set list synchronously. Markus Handell 2021-01-14 14:00:40 +01:00
  • c12f625938 Adds VideoDecoder::GetDecoderInfo() Erik Språng 2021-01-13 21:49:59 +01:00
  • a0bb2ef2dc Delete unused VideoType enum values Niels Möller 2020-12-17 13:22:20 +01:00
  • e61a40e4ef Fix typo in audio processing header. Hua, Chunbo 2021-01-08 16:34:49 +08:00
  • fdbaeda003 Improve structuring of test for audio glitches. Harald Alvestrand 2021-01-14 09:26:54 +00:00
  • dbcaff0925 Fix AudioProcessing::Config::ToString() implementation. Hua, Chunbo 2021-01-08 13:36:20 +08:00
  • d73426d660 Add new empty build targets rtp_rtcp_legacy and video_legacy. Niels Möller 2020-12-16 15:57:49 +01:00
  • b24e720907 Fix inconsistencies in network BUILD.gn file Andrey Logvin 2021-01-14 08:29:36 +00:00
  • db7920424c Change PeerConnectionE2EQualityTest to use lambdas instead of rtc::Bind Niels Möller 2021-01-11 15:18:44 +01:00
  • 76714a6cc8 AGC2 minor code clean up Alessio Bazzica 2021-01-13 18:08:40 +01:00
  • ece6712d6e Add av1 to lower range IDs. Jerome Jiang 2021-01-13 10:12:34 -08:00
  • 507eacfd35 Reland "ChannelStatistics used for RTP stats in VoipStatistics." Tim Na 2021-01-06 12:09:26 -08:00
  • 2297272aa5 Roll chromium_revision c0feffab8f..189823ba75 (842900:843026) chromium-webrtc-autoroll 2021-01-13 08:02:21 -08:00
  • 8606b9c218 Replace all uses of the word 'master' with 'builder_group' in //tools/mb Mirko Bonadei 2021-01-12 14:29:40 +01:00
  • 6e509f9167 Handle case createShader throws an exception. Sami Kalliomäki 2021-01-13 15:12:34 +01:00
  • 94324f2774 Add a test to detect excessive audio delay during renegotiation. Harald Alvestrand 2021-01-13 12:31:53 +00:00
  • 0e8dd039be Fix cpu usage dvqa test on windows Andrey Logvin 2021-01-13 11:03:34 +00:00
  • 0c2808e576 Remove CheckCompile method from MetaBuildWrapper. Mirko Bonadei 2021-01-12 14:34:25 +01:00
  • 086ed6e02e Roll chromium_revision b24f1becea..c0feffab8f (842774:842900) chromium-webrtc-autoroll 2021-01-12 22:03:50 -08:00
  • 81b58d14d7 Update WebRTC code version (2021-01-13T04:03:21). webrtc-version-updater 2021-01-12 20:03:29 -08:00
  • 46f94acf2e Roll chromium_revision e35e978ee1..b24f1becea (842656:842774) chromium-webrtc-autoroll 2021-01-12 16:02:16 -08:00
  • 37827c9058 Revert "ChannelStatistics used for RTP stats in VoipStatistics." Alex Loiko 2021-01-12 21:34:31 +00:00
  • cf4e534997 Roll chromium_revision 0026868b2d..e35e978ee1 (842481:842656) chromium-webrtc-autoroll 2021-01-12 12:03:37 -08:00
  • 444e04be69 ChannelStatistics used for RTP stats in VoipStatistics. Tim Na 2021-01-06 12:09:26 -08:00
  • 2accc7d6e0 Revert "Add task queue to RtpRtcpInterface::Configuration." Niels Moller 2021-01-12 15:54:16 +00:00
  • b942d45110 Fill fps allocation by LibaomAv1Encoder::GetEncoderInfo Danil Chapovalov 2021-01-12 14:45:07 +01:00
  • 6f597bd2ab Move logged types for RTC event log into event headers. Björn Terelius 2021-01-12 16:05:06 +01:00