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• 5a6f022c93 Update WebRTC code version (2023-11-14T04:05:37). webrtc-version-updater 2023-11-13 20:05:38 -08:00
• e76f837c8d Roll chromium_revision 8147f91b53..9fa5156191 (1223892:1224038) chromium-webrtc-autoroll 2023-11-13 18:16:09 -08:00
• 8a99fdaed5 Roll chromium_revision f5ebc0d33e..8147f91b53 (1223721:1223892) chromium-webrtc-autoroll 2023-11-13 14:01:18 -08:00
• 2a33e0ae84 Roll chromium_revision 5126f37e0f..f5ebc0d33e (1223587:1223721) chromium-webrtc-autoroll 2023-11-13 10:03:45 -08:00
• 9305b108bd Fix integer overflow. Jakob Ivarsson 2023-11-13 14:39:57 +01:00
• e75cd0c704 Remove DTLS 1.0 legacy code Philipp Hancke 2023-11-13 12:25:25 +01:00
• d431156c0e Move codecs handling from test to tester Sergey Silkin 2023-11-13 15:48:25 +01:00
• 9c91e48599 Make RtpSenderInternal::CheckCodecParameters pure virtual Harald Alvestrand 2023-11-13 13:37:50 +00:00
• 393cc11ca1 Roll chromium_revision 59a1669cee..5126f37e0f (1223462:1223587) chromium-webrtc-autoroll 2023-11-13 06:02:09 -08:00
• a6544377bc Remove not-needed webrtc:: prefixes in pc/ Harald Alvestrand 2023-11-13 09:33:56 +00:00
• 9f0d10b1f4 Update WebRTC code version (2023-11-13T04:05:34). webrtc-version-updater 2023-11-12 20:05:36 -08:00
• dfe62c8ad4 Roll chromium_revision 3db03206f6..59a1669cee (1223359:1223462) chromium-webrtc-autoroll 2023-11-12 20:02:26 -08:00
• 82cbbcc179 dcsctp: Convert use of TimeMs to webrtc::Timestamp Victor Boivie 2023-11-08 16:31:44 +01:00
• 5820a7f6f6 Roll chromium_revision 21005282cd..3db03206f6 (1223257:1223359) chromium-webrtc-autoroll 2023-11-11 20:02:14 -08:00
• d28b6fdf49 Update WebRTC code version (2023-11-12T04:10:51). webrtc-version-updater 2023-11-11 20:10:52 -08:00
• 9c69c4625b Revert "Measure wall clock time of capture and encode processing." Markus Handell 2023-11-10 18:18:19 +00:00
• 0cb9b28e5b Remove untested and unused SetSrtpSend/ReceiveKey functions Harald Alvestrand 2023-11-11 08:25:36 +00:00
• 369111bf30 Revert "VideoStreamEncoder: exclude screencast from capture time measurement." Markus Handell 2023-11-11 09:08:05 +00:00
• 93ec8048b2 Update WebRTC code version (2023-11-11T04:05:39). webrtc-version-updater 2023-11-10 20:05:41 -08:00
• f52faf2114 Roll chromium_revision 9913d5966b..21005282cd (1223151:1223257) chromium-webrtc-autoroll 2023-11-10 18:02:04 -08:00
• dd45ab650e Revert "Add RTC_EXPORT for Voucher::ScopedSetter and Voucher::Attachment." Markus Handell 2023-11-10 18:16:45 +00:00
• 1b4a91d8bb Roll chromium_revision f712caa32d..9913d5966b (1223010:1223151) chromium-webrtc-autoroll 2023-11-10 14:04:01 -08:00
• b237f1be56 Roll chromium_revision 20dfb16950..f712caa32d (1222900:1223010) chromium-webrtc-autoroll 2023-11-10 10:01:47 -08:00
• 2d43ab8508 dcsctp: Add Now() callback Victor Boivie 2023-11-08 15:37:33 +01:00
• 2baa7ae9e0 Roll chromium_revision 1eeab19714..20dfb16950 (1222790:1222900) chromium-webrtc-autoroll 2023-11-10 06:03:53 -08:00
• 7d62fe5702 Default enable NetEq experiments. Jakob Ivarsson 2023-11-09 16:23:07 +01:00
• 176eda9019 Roll chromium_revision fd386a7bdd..1eeab19714 (1222639:1222790) chromium-webrtc-autoroll 2023-11-10 00:04:23 -08:00
• 32052ed002 Update WebRTC code version (2023-11-10T04:01:48). webrtc-version-updater 2023-11-09 20:01:49 -08:00
• cc7861afb3 Roll chromium_revision 26a8c41e4b..fd386a7bdd (1222481:1222639) chromium-webrtc-autoroll 2023-11-09 16:04:35 -08:00
• 3e3881ae3c Reland "Make frame transformer MimeType pure virtual again" Philipp Hancke 2023-11-06 11:28:30 +01:00
• 283a5fd7ec Roll chromium_revision fb4e97f84b..26a8c41e4b (1222310:1222481) chromium-webrtc-autoroll 2023-11-09 12:01:57 -08:00
• 6324919690 Roll chromium_revision 2b21cf7f67..fb4e97f84b (1222194:1222310) chromium-webrtc-autoroll 2023-11-09 08:01:52 -08:00
• e567d8a112 Remove unused AudioFrameProcessor* parameter from WebRtcVoiceEngine constructor Danil Chapovalov 2023-11-09 11:48:22 +01:00
• fa4d7c92b7 libstdc++: replace deprecated std::is_pod<T> Stephan Hartmann 2023-10-31 15:43:33 +01:00
• be23ea4bb9 Revert "Make frame transformer MimeType pure virtual again" Mirko Bonadei 2023-11-09 13:27:24 +00:00
• 2d43014acb Roll chromium_revision 0f76018f3e..2b21cf7f67 (1222068:1222194) chromium-webrtc-autoroll 2023-11-09 04:01:44 -08:00
• 3ea9fc4cd8 Make frame transformer MimeType pure virtual again Philipp Hancke 2023-11-06 11:28:30 +01:00
• 36ed560339 Fall back to software encoding on wrong HW video encoder configuration Raman Budny 2023-11-06 15:01:31 +00:00
• 698e0bec17 Add Connection::OnReadPacket(const rtc::PacketReceived&) Per K 2023-11-09 08:49:06 +01:00
• 0ece6706c8 Update WebRTC code version (2023-11-09T04:08:00). webrtc-version-updater 2023-11-08 20:08:02 -08:00
• 534374ad57 Roll chromium_revision a4610eda72..0f76018f3e (1221833:1222068) chromium-webrtc-autoroll 2023-11-08 20:01:28 -08:00
• d3414d9688 Roll chromium_revision 7ae4a9b4a4..a4610eda72 (1221714:1221833) chromium-webrtc-autoroll 2023-11-08 12:17:39 -08:00
• ce627d4f44 Roll chromium_revision 157ae1ccb9..7ae4a9b4a4 (1221460:1221714) chromium-webrtc-autoroll 2023-11-08 10:01:19 -08:00
• bd396fdffa Add rtp packetizer for H265 qwu16 2023-10-26 16:52:22 +08:00
• 916ec25eed Roll chromium_revision 337b838893..157ae1ccb9 (1221347:1221460) chromium-webrtc-autoroll 2023-11-08 02:01:25 -08:00
• 977b56c9e9 Remove SSRCs from libSRTP when removing them from the rtp_demuxer Philipp Hancke 2023-11-08 10:20:51 +01:00
• 80056062f5 Update WebRTC code version (2023-11-08T04:03:41). webrtc-version-updater 2023-11-07 20:03:42 -08:00
• 3e436d4ee4 Roll chromium_revision bad490e614..337b838893 (1221203:1221347) chromium-webrtc-autoroll 2023-11-07 18:01:50 -08:00
• 8ed4b4d314 Roll chromium_revision 3df2d0c8e6..bad490e614 (1221033:1221203) chromium-webrtc-autoroll 2023-11-07 14:02:27 -08:00
• 4397482d71 dcsctp: Convert timers to rtc::TimeDelta Victor Boivie 2023-10-26 16:24:30 +02:00
• 644025c51f FrameCadenceAdapter: keeps max_fps in sync with constraints during 0Hz henrika 2023-11-07 18:22:02 +01:00
• ad9c4773f4 Roll chromium_revision fb6b63f0cb..3df2d0c8e6 (1220864:1221033) chromium-webrtc-autoroll 2023-11-07 10:18:28 -08:00
• 795a2ca30f Roll chromium_revision 15aec93e66..fb6b63f0cb (1220619:1220864) Mirko Bonadei 2023-11-07 14:16:52 +01:00
• 03bc3a0fa6 VideoStreamEncoder: exclude screencast from capture time measurement. Markus Handell 2023-11-07 13:20:50 +01:00
• 03457f6891 Remove unused Connection::SignalReadPacket Per K 2023-11-07 09:41:01 +01:00
• e925db88c1 Make stats member of packet buffer. Jakob Ivarsson 2023-11-07 10:10:47 +01:00
• bd523afd3a Add unit test to RenderSynchronizer Linus Nilsson 2023-10-27 15:12:48 +02:00
• 7946be7429 Refactor audio/video offer/answer creation helpers Philipp Hancke 2023-11-06 09:52:46 +01:00
• 5dbfe9621a Update WebRTC code version (2023-11-07T04:11:54). webrtc-version-updater 2023-11-06 20:11:55 -08:00
• eba274a63b Roll chromium_revision d51ff9503d..15aec93e66 (1220376:1220619) chromium-webrtc-autoroll 2023-11-06 16:03:48 -08:00
• be04c98d64 dcsctp: Migrate non-Timer related to rtc::TimeDelta Victor Boivie 2023-10-26 14:22:39 +02:00
• 0a33589db1 Roll chromium_revision 1204aeca4d..d51ff9503d (1220240:1220376) chromium-webrtc-autoroll 2023-11-06 10:01:25 -08:00
• 2b58ec2938 Deprecate call_factory and media_engine in PeerConnectionFactoryDependencies Danil Chapovalov 2023-11-06 15:32:56 +01:00
• 0873faae00 Remove smart flushing experiment. Jakob Ivarsson 2023-11-02 13:45:35 +01:00
• 9272771418 Roll chromium_revision c89d7a6d7f..1204aeca4d (1216881:1220240) Mirko Bonadei 2023-11-06 14:17:46 +00:00
• 24f1ff77e5 Add third_party/re2/src to DEPS. Mirko Bonadei 2023-11-06 13:42:41 +01:00
• 2bb48727a0 Add third_party/fuzztest/src to DEPS. Mirko Bonadei 2023-11-06 11:32:55 +01:00
• 99ea7c3eaa Use pass-by-value in delayed scheduling of OnSentPacket Harald Alvestrand 2023-11-05 07:51:31 +00:00
• 7fbcc8cef7 Rename PlanB helper function in tests Philipp Hancke 2023-11-02 10:53:21 +01:00
• 386873b7cc Update WebRTC code version (2023-11-05T04:02:21). webrtc-version-updater 2023-11-04 21:02:22 -07:00
• 37df1c07b2 Update WebRTC code version (2023-11-04T04:12:04). webrtc-version-updater 2023-11-03 21:12:06 -07:00
• 4be5927dc7 Reduces rate at which TryGetNextFrame returns NULL for WGC henrika 2023-11-02 10:16:00 +01:00
• 2931ddd2e9 Fire SentPacket in a PostTask when recursive Harald Alvestrand 2023-11-03 15:30:17 +00:00
• 97f3fb08a0 Assert some pointers that might be flaky. Harald Alvestrand 2023-11-03 13:09:55 +00:00
• 4fd1cc70da Add EnableMediaWithDefaultsAndTimeController Danil Chapovalov 2023-11-03 12:51:31 +01:00
• c63120a092 Migrate PeerConnection tests to EnableMedia api Danil Chapovalov 2023-11-03 11:32:24 +01:00
• 37f5172f6e dcsctp: Use TimeDelta in RetransmissionTimeout Victor Boivie 2023-10-20 13:33:32 +02:00
• d2098933e1 Expose audio mimeType for insertable streams Philipp Hancke 2023-11-01 19:18:32 +01:00
• 23cecc1d43 Move scoped_refptr from rtc:: to webrtc:: Harald Alvestrand 2023-11-02 14:09:30 +00:00
• 414ffac4b7 Add RTC_EXPORT for Voucher::ScopedSetter and Voucher::Attachment. Markus Handell 2023-11-03 00:14:48 +01:00
• c68da75d04 Update WebRTC code version (2023-11-03T04:11:45). webrtc-version-updater 2023-11-02 21:11:46 -07:00
• 779c9dede9 Migrate CreatePeerConnectionFactory implementation to EnableMedia api Danil Chapovalov 2023-11-02 17:29:21 +01:00
• b29ff000da Migrate webrtc_link_test to include EnableMedia api Danil Chapovalov 2023-11-02 19:18:04 +01:00
• 78f905e5cc Move some users to use webrtc::RefCountInterface Harald Alvestrand 2023-11-02 14:09:26 +00:00
• 9aa115358e Delete CreatePeerConnectionFactory variant that accepts unowned audio frame processor Danil Chapovalov 2023-10-30 13:22:20 +01:00
• 7e21b0ca9d Migrage objc sdk and examples to EnableMedia api Danil Chapovalov 2023-11-02 13:11:21 +01:00
• 166111da62 Migrate PeerConnectionIntegrationWrapper to EnableMedia api Danil Chapovalov 2023-11-01 14:43:07 +01:00
• a6ce338a2c Update WebRTC code version (2023-11-02T04:03:05). webrtc-version-updater 2023-11-01 21:03:06 -07:00
• b78e6a9305 dcsctp: Use TimeDelta in TX path Victor Boivie 2023-10-20 13:16:35 +02:00
• 8039cdbe48 Measure wall clock time of capture and encode processing. Markus Handell 2023-11-01 14:24:25 +01:00
• 4f4ae8a8f9 Remove templated fmtp SDP helper Philipp Hancke 2023-10-27 16:05:57 +02:00
• 554f7db01c Add EnableMediaWithDefaults to replace SetMediaEngineDefaults Danil Chapovalov 2023-11-01 11:08:20 +01:00
• efeeba0864 Try removing RTC_PUSH_IGNORING_WUNDEF() around proto includes Björn Terelius 2023-10-31 15:44:50 +01:00
• 766e658665 Update WebRTC code version (2023-11-01T04:11:40). webrtc-version-updater 2023-10-31 21:11:42 -07:00
• 7d1693f1c5 Do not allow estimate to increase above the estimate when HOLD started. Diep Bui 2023-11-01 00:09:35 +00:00
• 93214073f1 Mark EnableMedia with RTC_EXPORT Danil Chapovalov 2023-10-31 19:30:06 +01:00
• d7fb7e4a5f Enable H265 on Android bots. Mirko Bonadei 2023-10-31 15:31:51 +01:00
• 3ee8117856 H265 build fix for Android. philipel 2023-10-31 15:38:40 +01:00
• cf2fe18daa Use acked bitrate as a candidate if padding is sent. Diep Bui 2023-10-31 13:44:13 +00:00
• 86f09ae3f6 Fixes the OnFrameToEncode probe henrika 2023-10-31 14:49:40 +01:00