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  • 657b65fa18 Remove special-casing of TWCCv2 negotiation Philipp Hancke 2024-01-29 09:50:01 +01:00
  • 27a452da21 Rewrite webrtc_sdp unittest to use DTLS not SDES Harald Alvestrand 2024-01-31 11:54:24 +00:00
  • 4860148c51 Add WebRTC-LibaomAv1Encoder-MaxConsecFrameDrop parameter to explicitly limit the maximum consecutive frame drop Dan Tan 2024-01-31 09:11:18 -08:00
  • 1d3e286c7f Fix a fuzzer-found issue in G.722 decoder Henrik Lundin 2024-01-31 15:22:13 +00:00
  • 26ad5b82ce Fix a fuzzer-found issue in PCM/G.711 decoder Henrik Lundin 2024-01-31 15:02:55 +00:00
  • 9b7f3649af Fix a fuzzer-found issue in PCM16 decoder Henrik Lundin 2024-01-31 13:15:00 +00:00
  • 14b016fbf9 In RtcEventLogEncoderNewFormat use propagated instead of global field trials Danil Chapovalov 2024-01-05 13:51:00 +01:00
  • c0741e9f12 Revert "Take out Fuchsia-only SDES-enabling parameters" Olga Sharonova 2024-01-31 12:49:22 +00:00
  • 7cb4ce0079 Remove IceCandidateType::kNumValues Tommi 2024-01-29 15:14:33 +01:00
  • 3cbe63eac1 Do not register receiver for REMB until it starts receiving Philipp Hancke 2024-01-19 09:53:12 +01:00
  • c4dd03dfcb Remove kUnknown as a possible value for IceCandidateType. Tommi 2024-01-31 10:44:27 +01:00
  • 958c9ac546 Allow VideoCaptureModulePipeWire to be shared with more consumers Jan Grulich 2024-01-29 15:12:47 +01:00
  • 365cf14407 Revert "Test new tree." Mirko Bonadei 2024-01-31 08:35:44 +00:00
  • c6675ed967 Test new tree. Mirko Bonadei 2024-01-31 08:52:13 +01:00
  • cc83e32cdb Fix H.265 bitstream parser incorrect PPS reference. Qiu Jianlin 2024-01-31 11:13:07 +08:00
  • 056782c4b5 Implement Socket::RecvFrom(ReceiveBuffer& buffer) in PhysicalSocketServer Per K 2024-01-30 12:32:05 +01:00
  • 59f3b35013 Take out Fuchsia-only SDES-enabling parameters Harald Alvestrand 2024-01-30 07:39:42 +00:00
  • 765024e67b test: fix fuzzers line-endings Alfred E. Heggestad 2024-01-18 17:28:59 +01:00
  • 05a6f3b425 Update WebRTC code version (2024-01-30T04:07:38). webrtc-version-updater 2024-01-29 20:07:39 -08:00
  • 68e85b8d0d Adds WebRTC-Audio-PriorityBitrate for controlling audio/video rate allocation Dan Tan 2024-01-29 15:02:59 -08:00
  • 62cee88e4b Propagate Environment through QualityAnalyzingVideoDecoderFactory Danil Chapovalov 2024-01-29 18:38:21 +01:00
  • 7aa797244d Propagate sequence number to cloned encoded audio frames Tony Herre 2024-01-29 08:31:14 +01:00
  • f43e8ebab9 Add RTP depacketizer for H265 qwu16 2024-01-19 11:32:17 +08:00
  • 6adf2243b5 Compute scaling factors for not-explicitly configured layers in VP9 encoder Ilya Nikolaevskiy 2024-01-26 15:01:51 +01:00
  • 98db63cfb6 Introduce RtpTransportConfig:allow_bandwidht_estimation_probe_without_media Per K 2024-01-26 12:57:01 +01:00
  • 89db1c5827 Update WebRTC code version (2024-01-29T04:16:27). webrtc-version-updater 2024-01-28 20:16:28 -08:00
  • 0c4165e667 Update WebRTC code version (2024-01-27T04:11:16). webrtc-version-updater 2024-01-26 20:11:17 -08:00
  • 698b4e7087 Update more Candidate type checkers to use Candidate::is_* Tommi 2024-01-24 08:36:45 +01:00
  • 9c6874607a Consolidate encoded transform mocks into api/test/ Tony Herre 2024-01-26 11:16:45 +01:00
  • 1dccfeb395 Set InterLayerPredMode based on scalability mode for VP9. Åsa Persson 2024-01-26 09:50:23 +01:00
  • 979b6d62a8 Refactor RtpVideoSender::SetActiveModules. Per K 2024-01-26 11:18:26 +01:00
  • 9c166e064f Remove VideoSendStream::StartPerRtpStream Per K 2024-01-26 08:34:34 +01:00
  • 9a953b28f9 Detangle p2p/connection.cc and port.cc Harald Alvestrand 2024-01-26 07:11:50 +00:00
  • d213dd5517 Pass Environment to VideoDecoders through VideoCodecTester Danil Chapovalov 2024-01-25 15:48:24 +01:00
  • 523eff622e [Stats] Delete unused RTCStatsMember type alias. Henrik Boström 2024-01-22 13:52:38 +01:00
  • 6a3bbefd58 Reland "Enable DD and VLA header extensions by default for Simulcast/SVC" Philipp Hancke 2023-12-12 14:45:31 +01:00
  • 7f8470aeee Update WebRTC code version (2024-01-26T04:12:47). webrtc-version-updater 2024-01-25 20:12:48 -08:00
  • 0817380a56 Pass Environment when creating VideoDecoder in VideoReceiveStream2 Danil Chapovalov 2024-01-23 13:02:46 +01:00
  • ac58a334f7 [Stats] Migrate from the RTCStatsMember type alias to absl::optional. Henrik Boström 2024-01-22 13:51:36 +01:00
  • a310d78662 Refactor a lot of the p2p:rtc_p2p target Harald Alvestrand 2024-01-25 12:27:18 +00:00
  • e3a4bdb46f Roll chromium_revision 712952759e..844caa73fd (1251936:1252127) chromium-webrtc-autoroll 2024-01-25 08:17:44 -08:00
  • cc70a6d174 Guard GenerateUniqueId aginst concurrent access. Mirko Bonadei 2024-01-25 08:34:26 +00:00
  • c3624d02d0 Add field trial that enables Opus PLC. Jakob Ivarsson 2024-01-24 09:48:01 +01:00
  • de3c726121 Update to vpython 3.11 and remove .vpython (v2.x) Christoffer Dewerin 2024-01-22 12:01:44 +01:00
  • c13a7f9a00 Change string constant to constexpr char[] to unblock roll. Henrik Boström 2024-01-25 09:50:50 +01:00
  • b7e8a10fe4 Roll chromium_revision cf886b3ada..712952759e (1250272:1251936) chromium-webrtc-autoroll 2024-01-24 23:52:18 -08:00
  • a8375bb973 iOS: Fix building tests on real devices Florent Castelli 2024-01-24 15:41:06 +01:00
  • 5372bcec52 Reland "[Stats] Make RTCStatsMember<T> a type alias for absl::optional<T>." Henrik Boström 2024-01-22 09:56:09 +01:00
  • 44e4453067 Fix Port test and supply a legal value for the port type. Tommi 2024-01-23 22:40:29 +01:00
  • 1b61c7161e Expose setCodecPreferences/getCapabilities for iOS Karim H 2024-01-04 09:28:49 +04:00
  • c708c00f95 Add VideoDecoderFactory function to pass Environment for VideoDecoder construction Danil Chapovalov 2024-01-22 13:17:05 +01:00
  • 340d6c0236 Remove packet overhead lock and cached bitrate constraints. Jakob Ivarsson 2024-01-22 14:49:15 +01:00
  • eb76f193f3 Implement Newline Check in the Presubmit Byoungchan Lee 2024-01-23 01:55:14 +09:00
  • 25b29829dd Roll chromium_revision e1fb84c37d..cf886b3ada (1250109:1250272) chromium-webrtc-autoroll 2024-01-22 10:01:44 -08:00
  • be2786cd23 Move candidate type preference defaults to the Candidate class Tommi 2024-01-19 17:33:00 +01:00
  • fd54a619a5 Revert "[Stats] Make RTCStatsMember<T> a type alias for absl::optional<T>." Henrik Boström 2024-01-22 13:32:07 +00:00
  • 6fa743fbab Roll chromium_revision 336689d906..e1fb84c37d (1249985:1250109) chromium-webrtc-autoroll 2024-01-22 04:13:08 -08:00
  • 79ac694d9b [Stats] Make RTCStatsMember<T> a type alias for absl::optional<T>. Henrik Boström 2024-01-22 09:56:09 +01:00
  • f1fc6ab3ba Remove usage of the rtc::TaskQueue in video/ Danil Chapovalov 2024-01-22 10:51:53 +01:00
  • 348438154a [Stats] Delete ValueToString/ToJson, ToString does the job. Henrik Boström 2024-01-19 15:52:43 +01:00
  • 9074f0b7d7 Roll chromium_revision eb15a443b5..336689d906 (1249859:1249985) chromium-webrtc-autoroll 2024-01-21 20:01:21 -08:00
  • 1768705d99 Revert^4 "Delete pc/peerconnection build target" Harald Alvestrand 2024-01-21 21:09:13 +00:00
  • 11f87b2b29 Update WebRTC code version (2024-01-21T04:12:31). webrtc-version-updater 2024-01-20 20:12:33 -08:00
  • 7a1f85fb85 Roll chromium_revision fc694f0482..eb15a443b5 (1249756:1249859) chromium-webrtc-autoroll 2024-01-20 16:04:03 -08:00
  • 798e451878 Adds WebRTC-AV1-OverridePriorityBitrate to change bit rate allocation between audio and video Dan Tan 2024-01-18 11:13:32 -08:00
  • 5dc6c14747 Update WebRTC code version (2024-01-20T04:12:27). webrtc-version-updater 2024-01-19 20:12:29 -08:00
  • 1c1c2602cb Roll chromium_revision fbebadaea2..fc694f0482 (1249624:1249756) chromium-webrtc-autoroll 2024-01-19 18:01:24 -08:00
  • 7ee67e1ee9 Roll chromium_revision 4b479b78e7..fbebadaea2 (1247241:1249624) chromium-webrtc-autoroll 2024-01-19 14:57:38 -08:00
  • 25be2f802a Remove legacy, unused, preference() property from Candidate Tommi 2024-01-19 17:32:52 +01:00
  • 67ea392f27 Set visibility for RTC event log events BUILD targets Björn Terelius 2024-01-19 17:43:34 +01:00
  • 6a992129fb Tighten som DCHECKs to CHECKs in VP9 packetization. Erik Språng 2024-01-19 16:59:01 +01:00
  • a9ef127f6f Fix fuzzing issue for h265 bitstream parser qwu16 2024-01-17 09:34:52 +08:00
  • 0a7fc84887 Use is_* getters when checking the Candidate type Tommi 2024-01-19 13:11:37 +01:00
  • 0206a971f5 Uninherit TaskQueueForTest from legacy rtc::TaskQueue Danil Chapovalov 2024-01-19 14:08:53 +01:00
  • 37e9b378fd Use default H264 SDP parameters Sergey Silkin 2024-01-19 14:07:20 +01:00
  • 7209548090 Reland "[Stats] Attribute::ToString(), to replace member ValueToString/ToJson." Henrik Boström 2024-01-19 13:53:01 +01:00
  • 4c335b70e8 Record audio timestamps from iOS. Olov Brändström 2024-01-18 15:05:08 +01:00
  • ed1d084d0a [Stats] Replace all uses of is_defined() with has_value(). Henrik Boström 2024-01-19 11:27:31 +01:00
  • d7478a8453 Reland "Remove FLOW trace events as these are unused and don't work" Evan Shrubsole 2024-01-18 10:30:22 +00:00
  • 3e623ef57d Respect decoder implementation Sergey Silkin 2024-01-19 09:45:30 +00:00
  • 434f4cb44f Cleanup usage of rtc::TaskQueue in TestAudioDevice Danil Chapovalov 2024-01-18 18:42:43 +01:00
  • f49d96d6e4 Remove usage of the rtc::TaskQueue in test/ Danil Chapovalov 2024-01-18 18:30:42 +01:00
  • 3b2b2afdaa Move candidate types from port to candidate.h Tommi 2024-01-17 14:19:41 +01:00
  • 4d706a9fd1 Revert "Reland "[Stats] Attribute::ToString(), to replace member ValueToString/ToJson."" Mirko Bonadei 2024-01-19 07:48:13 +00:00
  • 55cdc29b9d Reland "[Stats] Attribute::ToString(), to replace member ValueToString/ToJson." Henrik Boström 2024-01-17 10:32:23 +01:00
  • 3abf8be180 Update WebRTC code version (2024-01-19T04:01:59). webrtc-version-updater 2024-01-18 20:02:02 -08:00
  • df0b363cf0 Reland "[Stats] Add value_or() and migrate from ValueOrDefault()." Henrik Boström 2024-01-17 09:54:07 +01:00
  • fb99c6ebb5 Disable the use of CGDisplayStreamCreate() for desktop capture on Sonoma Johannes Kron 2024-01-18 14:00:19 +01:00
  • 7d637a9788 Cleanup include and dependencies in api/metronome Danil Chapovalov 2024-01-17 17:49:37 +01:00
  • 0b6899272c Combine video_send_stream_impl.cc and video_send_stream.cc Per K 2024-01-18 09:42:58 +01:00
  • 68b580f116 Remove usage of rtc::TaskQueue in video replay tool Danil Chapovalov 2024-01-18 13:29:42 +01:00
  • b1799b0814 Cleanup usage of the rtc::TaskQueue in audio/ Danil Chapovalov 2024-01-18 11:57:48 +01:00
  • 192c0628cb Revert "Remove FLOW trace events as these are unused and don't work" Henrik Boström 2024-01-18 11:52:58 +00:00
  • 680025a612 Remove FLOW trace events as these are unused and don't work Evan Shrubsole 2024-01-18 10:30:22 +00:00
  • c0ac4df7a5 Reland "[Stats] Move metric names to Attribute, constructed via AttributeInit." Henrik Boström 2024-01-16 11:03:19 +01:00
  • 0f1b9a9589 Replace rtc::TaskQueue* with TaskQueueBase* in audio channel send frame transformer Danil Chapovalov 2024-01-17 19:32:15 +01:00
  • 7aff4d1a40 Stash and retry packets that are waiting for the dependency descriptor template structure. philipel 2024-01-17 19:26:58 +01:00
  • 787c8f8845 Update WebRTC code version (2024-01-18T04:03:20). webrtc-version-updater 2024-01-17 20:03:21 -08:00
  • 7f457533a2 Revert^3 "Delete pc/peerconnection build target" Mirko Bonadei 2024-01-17 19:03:20 +00:00
  • 02d9eceb3c Remove dependency on rtc::TaskQueue in AudioProcessing module Danil Chapovalov 2024-01-15 15:51:12 +01:00