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• a08189b948 Revert "Add InsertPacket method that takes RtpPacketInfo." Jakob Ivarsson‎ 2024-11-25 09:00:51 +00:00
• b40c559858 Set voice RTCP mode based on the RemoteContent and not based on the LocalContent. Tomas Lundqvist 2024-11-22 13:45:27 +00:00
• 06723eaab8 Default max limit probe to 2x current bwe Per Kjellander 2024-11-20 13:25:20 +00:00
• 924dc088dc Use 16bit unsigned for channel id for TURN Tommi 2024-11-22 15:08:09 +01:00
• 89432bc225 Update WebRTC code version (2024-11-24T04:08:23). webrtc-version-updater 2024-11-23 20:08:25 -08:00
• 38ddea5ee3 Add InsertPacket method that takes RtpPacketInfo. Jakob Ivarsson 2024-11-22 14:54:40 +00:00
• c63e43f27d Deprecate PeerConnectionFactoryDependencies::audio_processing Danil Chapovalov 2024-11-18 12:57:26 +01:00
• e5f6f1fab4 Add optional corruption filter settings to EncodedImage. Erik Språng 2024-11-22 11:33:04 +01:00
• 24992e9518 Report all usage patterns to UKM Harald Alvestrand 2024-11-22 10:04:05 +00:00
• 394da76a9c Propagate ECN information through Network Emulation Per K 2024-11-22 06:59:54 +00:00
• cd013b1d59 Opus decoder: stereo decoding by default (behind field trial) Alessio Bazzica 2024-11-21 14:59:24 +01:00
• 1b0371a54e Reduce the moving median window size in Remote ntp time estimator. Olov Brändström 2024-11-19 15:22:15 +01:00
• 319892c4d9 Update WebRTC code version (2024-11-22T04:05:20). webrtc-version-updater 2024-11-21 20:05:44 -08:00
• 253b8464ff Fix AV1 encoder do't set end_of_picture when the top layer is dropped 林恩 2024-10-18 11:43:18 +00:00
• 8da15c43dd Avoid depending on codec info for audio jitter stat. Jakob Ivarsson 2024-11-21 15:35:46 +00:00
• 2e7e049bb4 Don't use transport-cc if RFC8888 feedback is negotiated. Harald Alvestrand 2024-11-21 12:21:15 +00:00
• 5ad1daeed9 setParameters should not throw when only level mismatch. Qiu Jianlin 2024-11-20 23:01:36 +08:00
• 00e86b3cb0 Update WebRTC code version (2024-11-21T04:06:03). webrtc-version-updater 2024-11-20 20:06:22 -08:00
• 9aeed0c5f4 Avoid potential deadlock due to queue in corruption detection. Erik Språng 2024-11-20 14:01:33 +01:00
• c181432772 Add debug logging in WavWriterTest.LargeFile Björn Terelius 2024-11-20 14:30:17 +01:00
• 2a69ddbe9e Remove an unused conversion function. Harald Alvestrand 2024-11-19 13:14:58 +00:00
• da7b7ca1c1 Comment unused variables in implemented functions 15\n Dor Hen 2024-11-20 12:15:38 +02:00
• 4c5e72e3e0 Update WebRTC code version (2024-11-20T04:03:13). webrtc-version-updater 2024-11-19 20:03:32 -08:00
• d63aacb460 Use 'checkout_linux' instead of 'checkout_fuzzer' to checkout libFuzzer. Jeremy Leconte 2024-11-19 15:56:01 +01:00
• fb62f90706 Verify that transport-cc is used when RFC8888 field trial is off. Harald Alvestrand 2024-11-19 12:11:47 +00:00
• 2d47c9395b Correct H.265 level-id in fmtp line for offer/answer. Qiu Jianlin 2024-11-18 21:35:59 +08:00
• 5f163fcaa0 Align Int16FrameData test class with AudioFrame Tommi 2024-11-19 11:53:51 +01:00
• 8337c966d4 Use default probe duration if target higher than networkstate estimate Per K 2024-11-19 08:52:53 +00:00
• 999f02bd5f Implement playout stats for ios AudioDeviceModule Lionel Koenig Gélas 2024-11-14 14:37:28 +01:00
• aae790e3fe Update WebRTC code version (2024-11-19T04:07:36). webrtc-version-updater 2024-11-18 20:08:01 -08:00
• 4c9dbd508d Remove/update TODOs assigned to alessiob Alessio Bazzica 2024-11-18 16:52:05 +01:00
• 56085ea0d1 AGC2 test: add missing include Alessio Bazzica 2024-10-14 15:34:56 +02:00
• 331ca30635 Remove py_quality_assessment and old TODOs in conversational_speech Alessio Bazzica 2024-11-18 14:50:32 +01:00
• 7f775bc94c Ensure accurate FPS calculation for low frame rates Johannes Kron 2024-11-12 13:31:06 +01:00
• 17554c1c4c Add graph for ecn packet count in incoming/outgoing CCFB Per Kjellander 2024-11-18 12:40:41 +00:00
• 3ffe94314a Fix lint warnings in TaskQueueStdlib Björn Terelius 2024-11-18 11:35:47 +01:00
• 69cc695699 Comment unused variables in implemented functions 14\n Dor Hen 2024-11-11 15:51:41 +02:00
• bda11ca6da Add histogram WebRTC.Video.EstimatedClockDrift_ppm Johannes Kron 2024-11-15 15:54:45 +01:00
• c79be57b47 Reland "Set default scalability mode for H.265 to L1T1." Qiu Jianlin 2024-11-14 20:51:07 +08:00
• 752235261e Remove all references to codec-level transport-cc functions and flags. Harald Alvestrand 2024-11-18 07:18:35 +00:00
• 79c380c5b7 Always compile rtc_base/trace_categories.{h,cc}. Mirko Bonadei 2024-11-15 19:18:11 +01:00
• ab0b1888a5 Update WebRTC code version (2024-11-18T04:04:38). webrtc-version-updater 2024-11-17 20:04:39 -08:00
• 1dcf202ffe Fixes a linear interpolation bad access Guillaume Petit 2024-11-14 15:28:03 +01:00
• 9ab2ac126d Update WebRTC code version (2024-11-16T04:04:32). webrtc-version-updater 2024-11-15 20:04:56 -08:00
• dd8d2ab890 Allow union initiliazation for webrtc::webrtc_pc_e2e::AudioConfig. Jeremy Leconte 2024-11-15 12:48:39 +01:00
• b2fc13d094 fix stun prober return fail in windows Johnny 2024-11-15 09:59:42 +08:00
• f54707cd71 Reuse VP9 simulcast stream limits for H.265. Qiu Jianlin 2024-11-14 19:36:37 +08:00
• 7ef1360485 Fix issue that all macros not defined in rtc_pc_unittests Qiu Jianlin 2024-11-15 10:13:55 +08:00
• 0b333f2a40 Update WebRTC code version (2024-11-15T04:04:53). webrtc-version-updater 2024-11-14 20:04:53 -08:00
• 8d8fc3222a Cleanup WebRTC-LegacyTlsProtocols field trial from field trial list Philipp Hancke 2024-11-14 07:53:39 -08:00
• 3b2402bd23 Fix rust DEPS. Jeremy Leconte 2024-11-14 14:53:11 +01:00
• 2578802038 Add third_party/zstd to the DEPS file. Jeremy Leconte 2024-11-11 08:39:04 +01:00
• 0c6d31919e Enable RFC 8888 feedback if negotiated Harald Alvestrand 2024-11-13 11:54:41 +00:00
• d4a3002b9b srtp: remove deprecated non-span versions of key setters Elad Alon 2024-09-26 14:31:35 -07:00
• 54ed3ad524 Revert "Set default scalability mode for H.265 to L1T1." Ilya Nikolaevskiy 2024-11-13 15:56:37 +00:00
• 90da0650b5 Allow to specify a 'fps_hint' when creating a IvfVideoFrameGenerator. Jeremy Leconte 2024-11-13 15:39:42 +01:00
• 4c171e84c3 Prevent upscaling when calculating sample values. Emil Vardar 2024-11-13 09:25:18 +00:00
• ebb11c4c87 With stereo decoding and mono packets produce mono after CN/PLC Alessio Bazzica 2024-11-13 13:49:30 +01:00
• b7f5e7fb29 Fix video renderer slowdown by wrong RenderTime Shunbo Li 2024-11-12 10:37:46 +08:00
• 019bca9590 Remove deprecated CreateFromIvfFileFrameGenerator. Jeremy Leconte 2024-11-13 13:44:40 +01:00
• c3021aece3 Update WebRTC code version (2024-11-13T04:06:47). webrtc-version-updater 2024-11-12 20:06:48 -08:00
• 775639e930 Set default scalability mode for H.265 to L1T1. Qiu Jianlin 2024-11-11 23:21:25 +08:00
• 2031ab5c78 Update fuzzer documentation. Jeremy Leconte 2024-11-08 20:30:17 +01:00
• a5d71009ac PipeWire camera: use better unique device name for camera devices Jan Grulich 2024-11-07 15:39:25 +01:00
• 9fb71e3b01 Add RTC_DCHECK_IS_ON to unittest, as death is expected with DCHECK Liad Rubin 2024-10-27 12:16:50 +02:00
• 71ba9cbb80 Add corruption detection header default as stopped. Emil Vardar 2024-11-08 15:27:34 +00:00
• 849549d403 Enable rust toolchain for bots that depend on chromium base/. Jeremy Leconte 2024-11-08 21:02:24 +01:00
• 7a79d68645 Remove WebRTC-SetCodecPreferences-ReceiveOnlyFilterInsteadOfThrow killswitch Philipp Hancke 2024-11-06 14:09:41 -08:00
• 3c18829356 Update WebRTC code version (2024-11-09T04:02:55). webrtc-version-updater 2024-11-08 20:03:18 -08:00
• 8bc85f99c2 Checkout libFuzzer only when 'checkout_fuzzer' is True. Jeremy Leconte 2024-11-08 18:05:04 +01:00
• badfd6347e Refactor RTC event log analyzer to work as const Björn Terelius 2024-11-08 14:40:06 +01:00
• 416cb498cc Rename corruption related metrics according to WebRTC's Statistics API. Emil Vardar 2024-11-08 09:56:44 +00:00
• 6c8b8e0a2b Remove unused vars (Java) Björn Terelius 2024-11-08 11:13:06 +01:00
• 6308db99af Add missing licenses for the third_party target 'llvm-libc'. Jeremy Leconte 2024-11-08 10:34:19 +01:00
• c772fc227b Deprecate AudioProcessingBuilder in favor of the BuiltinAudioProcessingBuilder Danil Chapovalov 2024-11-05 10:48:41 +01:00
• 279858e9c8 Update WebRTC code version (2024-11-08T04:06:14). webrtc-version-updater 2024-11-07 20:06:50 -08:00
• ff9e7cb182 Include H.265 support in RTP video frame assembler. Qiu Jianlin 2024-11-07 23:41:08 +08:00
• f4abc03ca2 Reduce DTLS RSA certificate tests Philipp Hancke 2024-11-04 14:12:59 -08:00
• 05e5c32f98 Replace usage of AudioProcessingBuilder in EnableMediaWithDefaults Danil Chapovalov 2024-11-01 12:40:37 +01:00
• c7824dba06 With stereo decoding and mono packets produce mono DTX/concealment Alessio Bazzica 2024-11-07 15:41:29 +01:00
• 04d97b6c52 Fix "arithmetic on a pointer to void" warning. Jeremy Leconte 2024-11-07 11:34:56 +01:00
• 4405d06b97 Add H.265 to codecs that supports temporal scalability. Qiu Jianlin 2024-11-07 12:33:46 +08:00
• 7589689774 Replace cricket::LeastCommonMultiple and cricket::GreatestCommonDivisor with std::lcm and std::gcd. Evan Shrubsole 2024-11-06 15:14:40 +00:00
• e6f0c2fd23 SEA discards inactive encoders in implementation name Evan Shrubsole 2024-11-06 13:33:21 +00:00
• a5676c17b3 Change scale factor to 0.5 Emil Vardar 2024-11-06 09:56:18 +00:00
• 69af473a8a Remove unused android //base dependency Björn Terelius 2024-11-07 10:14:19 +01:00
• 757d6c3992 Update WebRTC code version (2024-11-07T04:07:11). webrtc-version-updater 2024-11-06 20:07:12 -08:00
• a287ffa681 Add unit tests for AudioDecoderOpusImpl for stereo Alessio Bazzica 2024-11-06 10:03:29 +01:00
• b444820216 Update freshness of adaptation documentation Evan Shrubsole 2024-11-06 14:17:19 +00:00
• 8702008d4f Add llvm-libc dependency, now required for libc++. Jeremy Leconte 2024-11-06 12:55:39 +01:00
• 2cc99f1544 Disable use_fuzztest_wrapper for bots not using libfuzzer. Jeremy Leconte 2024-11-06 10:26:33 +01:00
• faef5de87c Cleanup H.265 TODOs. Qiu Jianlin 2024-11-01 20:18:15 +08:00
• 1a93080b8d Extend ios RTCPeerConnectionFactoryBuilder to have custom audio processing builder and field trials Danil Chapovalov 2024-11-04 15:45:01 +01:00
• 3bd3815c4d Update WebRTC code version (2024-11-05T04:09:36). webrtc-version-updater 2024-11-04 20:10:08 -08:00
• 72b827cfe1 Delete android helper to create default AudioProcessing Danil Chapovalov 2024-11-01 14:18:38 +01:00
• 170a7b52fe Delete deprecated overloads of the AudioprocFloat test helper Danil Chapovalov 2024-11-04 13:52:16 +01:00
• 13a020c1bb Disable libevent task queue completely. Markus Handell 2024-11-04 12:41:26 +01:00
• 7855622b28 Update WebRTC code version (2024-11-04T04:02:56). webrtc-version-updater 2024-11-03 20:03:21 -08:00
• 1d3f51650c Send CCFB at least every 250ms even if BWE is zero. This aligns with current transport sequence number feedback Per Kjellander 2024-11-03 11:48:02 +00:00
• 7c7290ba48 Update WebRTC code version (2024-11-03T04:04:34). webrtc-version-updater 2024-11-02 21:04:57 -07:00