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  • 682755e49e Do not support frame tracking id extension in production Philipp Hancke 2023-05-22 11:57:14 +02:00
  • 5bcea256c8 Use version-less CIPD path for android_toolchain Prashanth Swaminathan 2023-06-08 18:00:53 -07:00
  • f781ff76fa Update WebRTC code version (2023-06-09T04:02:47). webrtc-version-updater 2023-06-08 21:02:49 -07:00
  • cde5354729 Implement DelayVariationCalculator for events analysis. Rasmus Brandt 2023-06-08 10:41:25 +02:00
  • f99e0f4aa0 Remove stale Android NDK [2/2] Prashanth Swaminathan 2023-06-02 15:53:50 -07:00
  • 40ad4ebfb5 Roll chromium_revision a8db252505..a3756bb36c (1153825:1154916) Prashanth Swaminathan 2023-06-08 08:26:18 -07:00
  • 4ee5e5f294 Disable VideoCaptureTest due to flakyness Björn Terelius 2023-06-08 11:22:16 +02:00
  • 37fb647e5e Disable the roll of 'android_ndk' Prashanth Swaminathan 2023-06-07 11:33:39 -07:00
  • 36c945b46e Update WebRTC code version (2023-06-08T04:11:54). webrtc-version-updater 2023-06-07 21:11:55 -07:00
  • 9d9c3f4544 [Analysis] Remove old threshold fields Beining Chen 2023-06-07 02:30:41 +00:00
  • 89f64b994f Make packet info optional and only set for primary packets in NetEq. Jakob Ivarsson 2023-06-07 19:18:18 +02:00
  • 9e639fa248 Migrate Android NDK to CIPD [1/2] Prashanth Swaminathan 2023-06-02 15:52:10 -07:00
  • fc260a1878 Add method SetTimestamp in TransformableAudioFrameInterface Palak Agarwal 2023-06-07 16:17:49 +02:00
  • 3403acb3c6 av1: 8 threads for >720p and tiles config Jerome Jiang 2023-06-07 10:17:41 -04:00
  • d615704551 Enable frame dropping in libaom AV1 encoder Sergey Silkin 2023-06-07 13:45:14 +02:00
  • a458fe5497 Update WebRTC code version (2023-06-07T04:12:21). webrtc-version-updater 2023-06-06 21:12:22 -07:00
  • 09e0086d26 Remove ImplForTesting function from MediaChannel Harald Alvestrand 2023-06-06 12:12:10 +00:00
  • bd66cfe5de Roll chromium_revision a5cd053713..a8db252505 (1153688:1153825) chromium-webrtc-autoroll 2023-06-06 07:10:30 -07:00
  • 847208e9d6 Remove transitional shim classes Harald Alvestrand 2023-06-06 09:45:45 +00:00
  • ade07ca45e Rename current flexfec implementation flexfec_03 Yosef Twaik 2023-06-04 16:16:01 +03:00
  • 43df03db38 Fix spelling mistake ReplaceRemoteDescriptionAndCheckE*r*or Philipp Hancke 2023-05-02 11:28:10 +02:00
  • 6d25e96115 Roll chromium_revision 404afa6a86..a5cd053713 (1153573:1153688) chromium-webrtc-autoroll 2023-06-05 23:03:07 -07:00
  • d3b71c778f Update WebRTC code version (2023-06-06T04:12:09). webrtc-version-updater 2023-06-05 21:12:10 -07:00
  • e00a12fb30 Roll chromium_revision 96ad22527d..404afa6a86 (1153423:1153573) chromium-webrtc-autoroll 2023-06-05 17:01:30 -07:00
  • 8c4b9ea535 Remove references to AudioCodec and VideoCodec constructors Florent Castelli 2023-06-02 16:06:57 +00:00
  • fd096da738 Roll chromium_revision 8f3397a259..96ad22527d (1153256:1153423) chromium-webrtc-autoroll 2023-06-05 13:02:55 -07:00
  • 77c6230ef5 Add create functions for voice media send and receive channels. Harald Alvestrand 2023-06-05 14:19:40 +00:00
  • be316dab88 Ensure that RTCErrorOr<T, E> doesn't require T to be default constructible Florent Castelli 2023-06-02 15:57:43 +00:00
  • 0740048170 Roll chromium_revision f28b824184..8f3397a259 (1152496:1153256) chromium-webrtc-autoroll 2023-06-05 09:01:49 -07:00
  • b0ef5e4bcd Declare factory functions for video sender and receiver Harald Alvestrand 2023-06-05 10:29:32 +00:00
  • 2f0c0787b9 Split WebRtcVoiceChannel into Send and Receive classes Harald Alvestrand 2023-06-05 10:34:57 +00:00
  • 1e04d61f21 Update WebRTC code version (2023-06-05T04:02:35). webrtc-version-updater 2023-06-04 21:02:36 -07:00
  • 816f5b1a39 Create VP9Encoder with a VP9 codec object Florent Castelli 2023-05-31 12:39:03 +00:00
  • 968e3c09db rtp_sender: fix typo with spatial_bitmask Alfred E. Heggestad 2023-06-01 12:07:28 +02:00
  • 079ce251de Update WebRTC code version (2023-06-04T04:02:33). webrtc-version-updater 2023-06-03 21:02:35 -07:00
  • e10f025c68 Update WebRTC code version (2023-06-03T04:02:02). webrtc-version-updater 2023-06-02 21:02:03 -07:00
  • 5278b39fab Add H264Encoder::Create() Florent Castelli 2023-05-31 12:44:32 +00:00
  • 811e24a117 Move functionality from AudioCodec and VideoCodec into cricket::Codec Florent Castelli 2023-05-31 12:35:47 +00:00
  • b8651debd4 Roll chromium_revision d48b2929db..f28b824184 (1152392:1152496) chromium-webrtc-autoroll 2023-06-02 07:02:11 -07:00
  • 54e95bc562 Propagate time of the last received packet with Timestamp type Danil Chapovalov 2023-06-02 14:54:45 +02:00
  • 9a34d80fc4 Apply the "shim" pattern for WebRtcVoiceEngine Harald Alvestrand 2023-06-02 11:57:12 +00:00
  • b15a9f01fc Fix perf tests. Jeremy Leconte 2023-06-02 14:14:17 +02:00
  • 3488726163 sdp: reject spec simulcast answers without the rid extension Philipp Hancke 2023-06-01 19:07:50 +02:00
  • f785bd46e8 Split WebRtcVideoMediaChannel into Send and Receive Harald Alvestrand 2023-05-31 18:12:24 +00:00
  • 4ad141e69b Add callback for send codec in audio too Harald Alvestrand 2023-06-02 09:59:45 +00:00
  • 371b7af673 Roll chromium_revision 2478b63fb4..d48b2929db (1151892:1152392) chromium-webrtc-autoroll 2023-06-02 01:02:14 -07:00
  • b29ee5bc8f Run the same perf tests on all platforms. Jeremy Leconte 2023-05-30 13:18:52 +02:00
  • 267040e29a Make native VideoTrack pointer public Jonas Oreland 2023-06-02 07:10:47 +02:00
  • cfc1a3a1a9 Update vpython3 requests Brian Sheedy 2023-06-01 12:22:51 -07:00
  • eeacddbd99 Disable flaky PictureIdTests. Jeremy Leconte 2023-06-02 08:45:02 +02:00
  • d454815b19 Use //third_party/cpu_features directly Prashanth Swaminathan 2023-06-01 08:00:10 -07:00
  • dab505be2c Update WebRTC code version (2023-06-02T04:02:59). webrtc-version-updater 2023-06-01 21:03:00 -07:00
  • 063b45bed7 Roll chromium_revision faf350b988..2478b63fb4 (1151758:1151892) chromium-webrtc-autoroll 2023-06-01 09:02:01 -07:00
  • dba22d3190 Move transceiver iteration loop over to the signaling thread. Tommi 2023-06-01 16:08:52 +02:00
  • 513ab0cb2c Add a -d option to apply-iwyu Harald Alvestrand 2023-05-23 10:18:36 +00:00
  • e24b34c178 Roll chromium_revision e26eb46a54..faf350b988 (1150524:1151758) chromium-webrtc-autoroll 2023-06-01 03:04:41 -07:00
  • b93f69a51a In VideoCaptureV4L2 create the capture thread last in StartCapture Andreas Pehrson 2023-05-17 13:52:47 +02:00
  • e44a155967 Add third_party/cpu_features license path. Jeremy Leconte 2023-06-01 11:07:22 +02:00
  • 2d598535aa Add SetRetransmissionMode() to FecController, this will be used to control RTX settings in FecController. Ying Wang 2023-06-01 13:55:36 +08:00
  • 3d6e88e6ac Remove low_bandwidth_audio_test. Jeremy Leconte 2023-05-31 10:10:44 +02:00
  • 6110fd933a Update WebRTC code version (2023-06-01T04:12:34). webrtc-version-updater 2023-05-31 21:12:35 -07:00
  • cb85143f93 Fix duplicate 'unix' OS and latest-revision deps Prashanth Swaminathan 2023-05-31 07:23:48 -07:00
  • 2197300977 Update ReceiveStatistics to use Timestamp/TimeDelta to represent time Danil Chapovalov 2023-05-31 13:21:01 +02:00
  • a9bba047b7 Updating AsyncAudioProcessing API, part 1. Peter Hanspers 2023-05-30 13:43:41 +02:00
  • 56d69e2c67 Add //third_party/cpu_features to DEPS Prashanth Swaminathan 2023-05-30 21:49:40 -07:00
  • c18f083900 Split MediaChannel concrete functions to MediaChannelUtil Harald Alvestrand 2023-05-31 06:55:00 +00:00
  • 94a9d55671 Update WebRTC code version (2023-05-31T04:11:01). webrtc-version-updater 2023-05-30 21:11:02 -07:00
  • b84fae66db Use sinf instead of std::sinf to improve libstdc++ compatibility Li-Yu Yu 2023-05-26 16:00:07 +08:00
  • 9fa5057eb9 Roll chromium_revision da88253915..e26eb46a54 (1150417:1150524) chromium-webrtc-autoroll 2023-05-30 07:02:04 -07:00
  • 6acfbb066e Replace std::optional with absl::optional in RtpPacketHistory Per K 2023-05-30 13:26:11 +02:00
  • d8098fb5fd Delete struct RTCPReportBlock as no longer used Danil Chapovalov 2023-05-30 10:58:00 +02:00
  • d8b88d8b94 Use the VideoMediaChannelShim for all cases Harald Alvestrand 2023-05-30 10:13:17 +00:00
  • 428836d1ea tools: fix small typo in python script Alfred E. Heggestad 2023-05-22 15:06:20 +02:00
  • 4bf52387e7 sdp: reject BUNDLE with RTP header extension id collisions Philipp Hancke 2023-05-25 16:11:36 +02:00
  • b18463409b Run webrtc_perf_tests on Fuchsia os. Jeremy Leconte 2023-05-30 09:49:41 +02:00
  • c73ea4fc57 More systematic null checks before calling native methods Xavier Lepaul 2023-05-30 09:39:55 +02:00
  • a3e9c0ae5a Roll chromium_revision c90a8a46d7..da88253915 (1150306:1150417) chromium-webrtc-autoroll 2023-05-30 01:03:06 -07:00
  • 97c9623839 Make a shim object implementing the VideoMediaChannel interface Harald Alvestrand 2023-05-30 04:04:07 +00:00
  • 4c1e9598a3 Change flexfec header reader to parse according to updated RFC. Yosef Twaik 2023-05-15 09:56:11 -07:00
  • e4a9a6dd6b Update WebRTC code version (2023-05-30T04:02:06). webrtc-version-updater 2023-05-29 21:02:07 -07:00
  • c5e4bcc70e Roll chromium_revision 599c746c73..c90a8a46d7 (1150194:1150306) chromium-webrtc-autoroll 2023-05-29 11:15:01 -07:00
  • 4b14cb763b Roll chromium_revision fa2e063162..599c746c73 (1150086:1150194) chromium-webrtc-autoroll 2023-05-28 23:02:34 -07:00
  • 4aaacb4382 Update WebRTC code version (2023-05-29T04:03:50). webrtc-version-updater 2023-05-28 21:03:52 -07:00
  • e641a970ef In RtcpReceiver remove redundand way to represent RTCP report blocks Danil Chapovalov 2023-05-23 10:39:09 +02:00
  • b9de471b34 Update WebRTC code version (2023-05-28T04:11:22). webrtc-version-updater 2023-05-27 21:11:23 -07:00
  • 98185b9a37 Roll chromium_revision 99b12997bf..fa2e063162 (1150050:1150086) chromium-webrtc-autoroll 2023-05-27 13:02:13 -07:00
  • a294353ccb Use type raw for video_codec_perf_tests. Mirko Bonadei 2023-05-27 15:07:46 +02:00
  • 01c2efc310 Roll chromium_revision bddf6cbe18..99b12997bf (1149812:1150050) chromium-webrtc-autoroll 2023-05-27 01:02:17 -07:00
  • 9bc8d055f0 Update WebRTC code version (2023-05-27T04:12:09). webrtc-version-updater 2023-05-26 21:12:10 -07:00
  • 9ac543c35a Roll chromium_revision 1fc947a5da..bddf6cbe18 (1149703:1149812) chromium-webrtc-autoroll 2023-05-26 11:06:27 -07:00
  • 87e74f9fb7 Remove unused combined_audio_video_bwe. Yury Yarashevich 2023-05-25 11:27:34 +02:00
  • 2bb686dbc0 Stop running low_bandwith_audio_tests. Jeremy Leconte 2023-05-26 16:08:32 +02:00
  • 6490999660 Roll chromium_revision aae661725b..1fc947a5da (1148994:1149703) chromium-webrtc-autoroll 2023-05-26 07:03:10 -07:00
  • f0820ffd88 Implement video versions of RTCInboundRtpStreamStats.jitterBuffer{Target,Minimum}Delay Rasmus Brandt 2023-05-25 09:37:16 +02:00
  • 9caef2a8b8 Use a constant for invalid PipeWire file descriptor Jan Grulich 2023-05-25 19:16:32 +02:00
  • 0f1a2c5d97 Change StreamDataCounters to use Timestamp instead of int64_t Danil Chapovalov 2023-05-26 11:30:26 +02:00
  • 5f32fa47a7 Delete MediaBaseChannel class Harald Alvestrand 2023-05-26 08:26:13 +00:00
  • 4f1dcbb1ac doc: Follow up link rename in I2dbe1ef0c74a0de8c5619b522fab39527e797d9c Li-Yu Yu 2023-05-26 15:50:54 +08:00
  • f53b3436e4 Cleanup RtcpTransceiver dependency on webrtc::Transport Danil Chapovalov 2023-05-25 12:52:21 +02:00
  • 5f38949c54 Allow single-mline offers without BUNDLE group when using max-bundle Philipp Hancke 2023-05-17 12:23:53 +02:00