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

Commit Graph

  • 70b65bc375 Remove deprecated ReceiveSideCongestionController ctor Per K 2025-02-05 08:40:49 +00:00
  • 3164c2a4eb Restructure PeerConnection tests not to create PortAllocator directly Danil Chapovalov 2025-02-04 22:21:23 +01:00
  • e46d8e4ec4 Update WebRTC code version (2025-02-05T04:02:29). webrtc-version-updater 2025-02-04 20:02:31 -08:00
  • b1ec813339 Expose direct access to PeerConnection in PeerConnectionWrapper helper Danil Chapovalov 2025-02-04 17:50:44 +01:00
  • e828c6dba3 red: remove hardcoded parameters in favor of taking them from the codec Philipp Hancke 2025-01-28 13:30:38 -08:00
  • 9c56cb3eda Add include for <optional> Takuto Ikuta 2025-01-30 20:44:30 -08:00
  • 0533b5eafe Add set_timestamp() method to RTCStats. Henrik Boström 2025-02-03 14:24:10 +01:00
  • 9cc5bc8499 Remove rust dependency on android. Jeremy Leconte 2025-02-03 14:51:07 +01:00
  • a6f35491d6 [cpp23] Remove use of std::aligned_storage in webrtc Victor Hugo Vianna Silva 2025-02-03 10:43:14 +00:00
  • c262375415 Add test for preferring RTX payload to be "primary codec + 1". Henrik Boström 2025-02-03 14:18:23 +01:00
  • ae24807590 [ObjC] Avoid usage of variable after move in RTCNetworkMonitor. Yury Yarashevich 2025-02-03 12:01:36 +01:00
  • f80562d22a [ObjC] Validate and store strong ref to peer_connection before use. Yury Yarashevich 2025-01-30 13:18:48 +01:00
  • b60a5ab91c Choose RTX codec PT in lower range if codec is in lower range Henrik Boström 2025-02-03 12:25:50 +01:00
  • 1181edda58 [ObjC] Fix strong reference check in RTCNetworkMonitor. Yury Yarashevich 2025-01-30 16:57:45 +01:00
  • 6f17d09dd1 [ObjC] Init NSMutableDictionary with capacity. Yury Yarashevich 2025-02-03 11:26:21 +01:00
  • 34c15bc511 Restructure PeerConnectionBundleTest helper not to create PortAllocator Danil Chapovalov 2025-01-31 16:12:12 +01:00
  • 9830de94e2 Update WebRTC code version (2025-02-01T04:05:45). webrtc-version-updater 2025-01-31 20:06:08 -08:00
  • f68df0b95c Restore primary/rtx payload type assignment logic Philipp Hancke 2025-01-30 10:46:09 -08:00
  • c58a767a23 Reland "Get DeviceScaleFactor for the captured monitor/screen" Palak Agarwal 2025-01-31 12:56:30 +00:00
  • 18b94b517d [rtc_base] Replace manual element initialization and movement with C++17 standard functions Ho Cheung 2025-01-25 20:59:55 +08:00
  • de1735058b Revert "Reland "Allow sending to separate payload types for each simulcast index."" Jonas Oreland 2025-01-30 23:55:05 -08:00
  • 9a407346fd Revert "Get DeviceScaleFactor for the captured monitor/screen" Mirko Bonadei 2025-01-31 01:19:20 -08:00
  • 45ebd339bd Update WebRTC code version (2025-01-31T04:06:50). webrtc-version-updater 2025-01-30 20:06:51 -08:00
  • e20fbb00d0 Get DeviceScaleFactor for the captured monitor/screen Palak Agarwal 2025-01-29 11:55:20 +00:00
  • d643be9fdc Add a render error callback from AudioDeviceIOS to AudioDeviceModuleIOS. Peter Hanspers 2025-01-30 10:38:57 +01:00
  • 4b39cb3eb6 Reland "Move piggybacking controller from P2PTC to DTLS transport" Jonas Oreland 2025-01-30 10:33:31 +01:00
  • 4de5839c11 Revert "Move piggybacking controller from P2PTC to DTLS transport" Jonas Oreland 2025-01-30 00:22:37 -08:00
  • 29e639e0a4 Move piggybacking controller from P2PTC to DTLS transport Philipp Hancke 2025-01-29 11:05:50 -08:00
  • feabcdb76b Reduce redundant memory allocation when capturing a single monitor. fizzfang 2025-01-24 16:47:11 +08:00
  • eb688d6e80 Remove dependency to NetworkStateEstimator from TransportSequenceNumberFeedbackGenerator Per K 2025-01-29 08:32:42 +00:00
  • 3155346f66 Reland "Remove rtc_p2p" Jonas Oreland 2025-01-29 08:34:50 +01:00
  • c6278957d1 Revert "Remove rtc_p2p" Jonas Oreland 2025-01-28 22:46:46 -08:00
  • a347fdf3c3 Update WebRTC code version (2025-01-29T04:03:12). webrtc-version-updater 2025-01-28 20:03:41 -08:00
  • a05ad63aa3 Remove rtc_p2p Jonas Oreland 2025-01-28 20:16:07 +01:00
  • 406d195d16 Move the rtc_p2p file last in its BUILD file Harald Alvestrand 2025-01-28 15:51:24 +00:00
  • 5342220a1b Make corruption_detection_message publicly visible Fanny Linderborg 2025-01-28 09:30:09 +01:00
  • 87b7c1aa6e Reduce warning logging when minimum playout delay exceed maximum Danil Chapovalov 2025-01-28 10:51:08 +01:00
  • 4a210486d3 DTLS 1.3 - patch 5 Jonas Oreland 2025-01-28 11:00:15 +01:00
  • d8fea51d65 Revert "Cleanup usage of the global field trials in the PeerConnectionE2EQualityTest helper" Danil Chapovalov 2025-01-28 02:16:28 -08:00
  • 1a72c0ccb9 Move a test from media_session_unittest to codec_vendor_unittest Harald Alvestrand 2025-01-28 07:48:04 +00:00
  • 2bebeaffe5 Remove unused create_call.cc Evan Shrubsole 2025-01-27 17:56:01 +00:00
  • fb31c99e03 Update WebRTC code version (2025-01-28T04:03:17). webrtc-version-updater 2025-01-27 20:03:18 -08:00
  • bea44590f6 desktop_capture: Fix Xrandr / Xrender order Florent Castelli 2025-01-27 16:26:03 +01:00
  • 26617bef59 Make AV1 even payload size default-on when packetizer is used directly Danil Chapovalov 2025-01-23 18:24:56 +01:00
  • 13170bd177 Refactor media_session to move codec handling to new class Harald Alvestrand 2025-01-27 14:51:55 +00:00
  • 6b7099527a VideoEncoder: rtc::StringBuilder instead of rtc::SimpleStringBuilder. Henrik Boström 2025-01-27 13:49:51 +01:00
  • 39da6f3a75 Move corruption_detection_message from common_video to api/transport/rtp Fanny Linderborg 2025-01-24 10:50:45 +01:00
  • a97304ca03 Cleanup usage of the global field trials in the PeerConnectionE2EQualityTest helper Danil Chapovalov 2025-01-23 11:29:14 +01:00
  • 9ff254eaf2 srtp: stop using private libsrtp function to determine packet index Philipp Hancke 2024-12-18 10:01:44 -08:00
  • a9b40ae0a4 Update WebRTC code version (2025-01-26T04:03:33). webrtc-version-updater 2025-01-25 20:04:02 -08:00
  • 7b8b0f665e Update WebRTC code version (2025-01-25T04:02:12). webrtc-version-updater 2025-01-24 20:02:33 -08:00
  • 5090eaf363 Reland "srtp: spanify Protect + Unprotect" Philipp Hancke 2025-01-22 15:27:59 -08:00
  • 4e8c984d15 Obfuscate private keys in unit tests to avoid false lint errors Philipp Hancke 2025-01-23 14:04:38 -08:00
  • ede69fd577 Make IsSameRtpCodecIgnoringLevel work for any codec. Henrik Boström 2025-01-24 11:20:54 +01:00
  • f2ecdd7ea3 Use ElementsAreArray in corruption detection unittests Fanny Linderborg 2025-01-24 12:40:28 +01:00
  • 1cc5e54368 Add missing newline Fanny Linderborg 2025-01-24 10:25:56 +01:00
  • 589acd56d0 dtls-stun piggybacking: make it compatible with DTLS 1.3 Philipp Hancke 2025-01-23 09:02:38 -08:00
  • eafee5e3d6 fix: h26x packet buffer video artifacts k-wasniowski 2025-01-22 18:45:34 +01:00
  • 63d3cf0d46 Update WebRTC code version (2025-01-24T04:10:20). webrtc-version-updater 2025-01-23 20:10:50 -08:00
  • a70cc7886c Make mid_ a private member variable Tommi 2025-01-21 22:34:35 +01:00
  • 7fe307df59 Update WebRTC code version (2025-01-23T04:07:51). webrtc-version-updater 2025-01-22 20:08:07 -08:00
  • a2a528c20b Add PortAllocator min/max ports to JAVA API Youjie Zhou 2025-01-19 08:48:04 -08:00
  • 79e5e721b5 Add unidirectional codec support ("offer to send" use case). Henrik Boström 2025-01-22 15:13:37 +01:00
  • 49ac6b758c Reland "Allow sending to separate payload types for each simulcast index." Henrik Boström 2025-01-22 14:22:45 +01:00
  • 1f9e6046dd Start deprecation process for non-Optional datachannel parameters Harald Alvestrand 2025-01-21 15:11:57 +00:00
  • 0bebca526a Remove gunit.h EXPECT/ASSERT..WAIT macros Evan Shrubsole 2025-01-21 10:04:37 +00:00
  • f62fbe9eaf Update WebRTC code version (2025-01-22T04:05:41). webrtc-version-updater 2025-01-21 20:06:03 -08:00
  • 046c979cb5 Delete reference to "no_build_hooks" GN variable (part 2) Andrew Grieve 2025-01-20 11:29:38 -05:00
  • 76c8f303a8 Replace use of .name in test code with .mid() Tommi 2025-01-17 16:48:23 +01:00
  • 7a0bdb602c Update PeerConnectionSdpMethods::AddRemoteCandidate Tommi 2025-01-17 16:00:53 +01:00
  • 3fef8b27db Adding an error callback to AudioDeviceModuleIOS. Peter Hanspers 2025-01-21 14:54:51 +01:00
  • 32f3c6cef1 Add AbslStringify for RtcErrorType and RtcErrorDetail Harald Alvestrand 2025-01-21 12:29:00 +00:00
  • a6bccab358 [DVQA] Dont try to render a 'superfluous' frame. Jeremy Leconte 2025-01-20 11:08:39 +01:00
  • 283a84d92a Add matchers for RTCError, rename old matcher for RTCErrorOr. Henrik Boström 2025-01-21 10:39:45 +01:00
  • 860a13c6fd Misc improvements to RtpTransceiver unit tests and test utils. Henrik Boström 2025-01-20 14:31:12 +01:00
  • ee7371f1f8 Update WebRTC code version (2025-01-21T04:06:38). webrtc-version-updater 2025-01-20 20:06:59 -08:00
  • b4127b5597 Roll chromium_revision dcda5ff9c0..3462a5bab8 (1408528:1408687) chromium-webrtc-autoroll 2025-01-20 08:04:15 -08:00
  • d621b419a3 Make WebRTC-Video-AV1EvenPayloadSizes default-on. Erik Språng 2025-01-20 15:01:38 +01:00
  • d48113627a Fix hw decoder rendering delay after frame resize Anna Lemehova 2025-01-17 19:28:53 +02:00
  • fa73a2ed79 Convert timeouts in integration_test_helpers to TimeDelta Evan Shrubsole 2025-01-20 10:18:53 +00:00
  • f1b3e3e115 Replace gunit.h macros with WaitUntil in modules/ Evan Shrubsole 2025-01-13 09:04:29 +00:00
  • 2a858e21f6 Migrate last uses of gunit.h macros Evan Shrubsole 2025-01-20 09:18:30 +00:00
  • 4f56e15075 Roll chromium_revision 48223dfc0a..dcda5ff9c0 (1408397:1408528) chromium-webrtc-autoroll 2025-01-20 00:03:58 -08:00
  • 9165a9b436 Disable OpenSSL tests needing a fake clock when boringssl is not used Philipp Hancke 2025-01-16 17:05:59 -08:00
  • e6890adc3a Update WebRTC code version (2025-01-20T04:06:42). webrtc-version-updater 2025-01-19 20:07:06 -08:00
  • 0908c9b90a Update WebRTC code version (2025-01-19T04:03:25). webrtc-version-updater 2025-01-18 20:03:54 -08:00
  • 9aeeb6123d Roll chromium_revision bb864a5b8d..48223dfc0a (1408296:1408397) chromium-webrtc-autoroll 2025-01-18 20:06:35 -08:00
  • a85040ffd9 Revert "Reland "Use Payload Type suggester for all codec merging"" Fabian Reddig 2025-01-17 14:45:55 -08:00
  • 48ca8b3d13 Roll chromium_revision 62907d98e8..bb864a5b8d (1408002:1408296) chromium-webrtc-autoroll 2025-01-17 22:03:57 -08:00
  • 1305eb9083 Update WebRTC code version (2025-01-18T04:08:19). webrtc-version-updater 2025-01-17 20:08:19 -08:00
  • 5f27925a5b Roll chromium_revision 8bf5d05e2b..62907d98e8 (1407755:1408002) chromium-webrtc-autoroll 2025-01-17 10:06:58 -08:00
  • 9f68535e68 Fix setParameters() throwing when level-id does not match. Henrik Boström 2025-01-17 12:06:53 +01:00
  • b0038dd14a Replace gunit.h macros with WaitUntil in P2P Evan Shrubsole 2025-01-17 13:22:54 +00:00
  • d9593037dd Replace gunit.h macros with WaitUntil in rtc_base/ Evan Shrubsole 2025-01-17 13:19:45 +00:00
  • 762753d0a2 Slight restriction of access to ContentInfo and prefer mid to name. Tommi 2025-01-17 12:31:07 +01:00
  • 88833e6d22 Update video stats documentation. Åsa Persson 2025-01-14 14:41:11 +00:00
  • ceb5a3b11e Roll chromium_revision 839b9b8bb4..8bf5d05e2b (1406733:1407755) chromium-webrtc-autoroll 2025-01-17 00:04:13 -08:00
  • d23d04163d Fix to allow small negative jumps due to out of order packets in packet buffer thebongy 2025-01-16 19:41:42 +05:30
  • 3cc17eed68 dtls-stun-piggybacking: add missing configuration guard Philipp Hancke 2025-01-16 14:31:32 -08:00
  • a65c453f9e Reduce default max QP for AV1 from 56 to 52 Sergey Silkin 2025-01-10 11:26:41 +01:00