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  • fd8d2654d7 Revert of Improve and re-enable FEC end-to-end tests. (patchset #3 id:40001 of https://codereview.webrtc.org/2675573004/ ) brandtr 2017-02-06 06:19:51 -08:00
  • d40b0f39e0 Improve and re-enable FEC end-to-end tests. brandtr 2017-02-06 05:54:43 -08:00
  • cb789bb510 Remove NewApi lint suppression. sakal 2017-02-06 05:34:26 -08:00
  • 93e1e23537 Use RateAccCounter for sent bitrate stats. Reports average of periodically computed stats over a call. asapersson 2017-02-06 05:18:35 -08:00
  • 447dba9586 Add debuggable=true to AppRTCMobile manifest. henrika 2017-02-06 04:58:01 -08:00
  • b114e9c159 Camera2Session: Add return statements after reportError where needed. sakal 2017-02-06 04:55:21 -08:00
  • 873fcb958f Drop the check for stray mobileprovision (no longer needed) oprypin 2017-02-06 03:43:58 -08:00
  • 61202ac2ea Ensure that AEC3 is not run in tandem with AEC2 peah 2017-02-06 03:39:42 -08:00
  • 237e1bbf76 Fix potential use after free in H264 packetizer. sprang 2017-02-06 03:02:15 -08:00
  • 60f7c633f9 Remove temporary AddRtxInfo member function. brandtr 2017-02-06 02:34:25 -08:00
  • d44ce0563f Reland of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #1 id:1 of https://codereview.webrtc.org/2668973003/ ) nisse 2017-02-06 02:23:00 -08:00
  • 656610fbe7 Move frame_generator_capture.{cc, h} and video_capturer.h to video_test_common. ehmaldonado 2017-02-06 02:21:11 -08:00
  • a7111eb45a Fixing an error in ANA FrameLengthController unittest. minyue 2017-02-06 02:20:00 -08:00
  • e702b30fec Adding C++ versions of currently spec'd "RtpParameters" structs. deadbeef 2017-02-04 12:09:01 -08:00
  • d1f5fdac5c Allow changing the minimal ICE ping timeout with PeerConnection.SetConfiguration. skvlad 2017-02-03 16:54:05 -08:00
  • 98c437458a Allow passing network config constraint as base64 encoded string to preserve values of serialized protos. The values are a serialized byte stream packed into a std::string. To be represented as a NSString they must be base64 encoded or bytes outside of the ASCII range will be encoded into multi byte UTF8 sequences by default. haysc 2017-02-03 13:03:39 -08:00
  • 390e64d7eb Add VP9 full stack tests: - ConferenceMotionHd2000kbps100msLimitedQueueVP9 jianj 2017-02-03 09:51:23 -08:00
  • 53b6cc3832 Reland of Enable audio streams to send padding. (patchset #4 id:60001 of https://codereview.webrtc.org/2652893004/ ) stefan 2017-02-03 08:13:57 -08:00
  • b11fb25c12 Protect APM in webkit builds. agouaillard 2017-02-03 06:37:05 -08:00
  • 9d58d94585 Fix and improve FlexFEC configuration for RTP/RTCP. brandtr 2017-02-03 04:43:41 -08:00
  • 4cb1b7546b Extends timer from 10 to 30 seconds for output volume check on Android. henrika 2017-02-03 03:39:47 -08:00
  • 77ce9a5541 Avoid calling PostTask in audio callbacks. henrika 2017-02-03 02:19:17 -08:00
  • 5f47126865 Added VP8 simulcast tests. Fixed analyzer to correctly infer timestamps. ilnik 2017-02-03 02:02:17 -08:00
  • 4b512d7e0c Fix Chromium FYI bot skvlad 2017-02-02 18:22:46 -08:00
  • d030912de4 Pick the DTLS handshake timeout based on the ICE RTT estimate skvlad 2017-02-02 17:18:37 -08:00
  • a24a9e2fa6 Get rid of unqualified std:: types. deadbeef 2017-02-02 16:04:36 -08:00
  • 6741516e18 Implement new PeerConnection certificate policy API in ObjC API hnsl 2017-02-02 13:04:27 -08:00
  • a5d94fff99 Objective-C API to set the ICE check rate through RTCConfiguration. skvlad 2017-02-02 13:02:30 -08:00
  • b55bd5fef0 Don't capture variables explicitly in lambda expression. ehmaldonado 2017-02-02 11:51:21 -08:00
  • 5107246d4b Allow applications to limit the ICE check rate through RTCConfiguration skvlad 2017-02-02 11:50:14 -08:00
  • e5bd70223d Reland of Make the new jitter buffer the default jitter buffer. (patchset #2 id:260001 of https://codereview.chromium.org/2656983002/ ) philipel 2017-02-02 09:53:00 -08:00
  • 8c61924b56 video_coding::PacketBuffer now group all H264 packets with the same timestamp into the same frame. philipel 2017-02-02 08:51:29 -08:00
  • 1dffc62843 Remove all occurrences of "using std::string". ehmaldonado 2017-02-02 08:10:00 -08:00
  • e372d3c519 Add event log visualization of rtp timestamps over time. stefan 2017-02-02 08:04:18 -08:00
  • a55f021d43 Add 120ms frame ability to ANA michaelt 2017-02-02 07:47:19 -08:00
  • ed01647ea9 Remove bad DCHECK added as part of https://codereview.webrtc.org/2452163004/ solenberg 2017-02-02 04:23:24 -08:00
  • b33eed2e42 Fix perf issue when timinig out receiver infos in RTCP. stefan 2017-02-02 03:57:02 -08:00
  • cc99bc25d8 Change StunMessage::AddAttribute return type from bool to void. nisse 2017-02-02 01:31:30 -08:00
  • f7826d668a Remove InlinedApi lint ignore. sakal 2017-02-02 00:53:33 -08:00
  • a29d5ec613 Make 'webrtc' target a complete static library on Linux, Android and Windows kjellander 2017-02-01 23:51:13 -08:00
  • 24af66397e Adding Java wrapper for DtmfSender. deadbeef 2017-02-01 21:53:09 -08:00
  • 20cb0c1c85 Move DTMF sender to RtpSender (as opposed to WebRtcSession). deadbeef 2017-02-01 20:27:00 -08:00
  • 2e03c66119 Adding build switch for Opus that supports 120ms ptime. minyue 2017-02-01 17:31:11 -08:00
  • d3d3ba5159 Revert of Enable audio streams to send padding. (patchset #4 id:60001 of https://codereview.webrtc.org/2652893004/ ) deadbeef 2017-02-01 15:45:53 -08:00
  • 1cbf518f01 Roll chromium_revision 6b2002254c..496a750d38 (447561:447619) buildbot 2017-02-01 14:55:04 -08:00
  • 353e7e1d8a Roll chromium_revision 9f2c537112..6b2002254c (447517:447561) buildbot 2017-02-01 11:51:51 -08:00
  • e35f89a484 Enable audio streams to send padding. stefan 2017-02-01 09:06:25 -08:00
  • 46fbb7d9d5 Roll chromium_revision ccc17b815a..9f2c537112 (447493:447517) buildbot 2017-02-01 08:52:09 -08:00
  • b1ca073db4 Rename adaptation api methods, extended vie_encoder unit test. sprang 2017-02-01 08:38:12 -08:00
  • d83b9670a6 Replace consecutive-losses count by a calculation of first-order-FEC recoverability. Note: * PLR is calculated over all of the known packets. * RPLR is calculated over all of the known packet *pairs*. That is, only over sets of subsequent packets where the reception status is known for both. elad.alon 2017-02-01 08:36:09 -08:00
  • 14245cc939 Revert of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #9 id:160001 of https://codereview.webrtc.org/2659563002/ ) nisse 2017-02-01 08:10:36 -08:00
  • 77f0580f83 Add new step graph type to event log visualization tool. Currently used for bitrate estimate and accumulated packet count, but could in general be used for any metric that is piecewise constant. terelius 2017-02-01 06:34:53 -08:00
  • a565f92e87 Roll chromium_revision e87481817b..ccc17b815a (447482:447493) buildbot 2017-02-01 05:44:39 -08:00
  • 099110cf4b Don't send audio packets if the network is down. stefan 2017-02-01 03:57:42 -08:00
  • 4637b6afca Consistent 30% improvement in audio mixer running time. aleloi 2017-02-01 03:43:31 -08:00
  • 35fc2aa82f Revert of Drop frames until specified bitrate is achieved. (patchset #12 id:240001 of https://codereview.webrtc.org/2630333002/ ) minyue 2017-02-01 03:14:00 -08:00
  • 2ad42ca0a0 Roll chromium_revision 8346af5a71..e87481817b (447464:447482) buildbot 2017-02-01 03:12:51 -08:00
  • 6d4dd593a8 Always call RemoteBitrateEstimator::IncomingPacket from Call. nisse 2017-02-01 03:06:58 -08:00
  • 803dc29bb6 Enable cpplint and fix cpplint errors in webrtc/api oprypin 2017-02-01 01:55:59 -08:00
  • 83399caec5 Drop frames until specified bitrate is achieved. kthelgason 2017-02-01 01:31:52 -08:00
  • fdd9b85652 Roll chromium_revision e4d460e023..8346af5a71 (447441:447464) buildbot 2017-01-31 23:42:16 -08:00
  • a1cf88db7c Roll chromium_revision 9d90548426..e4d460e023 (447390:447441) buildbot 2017-01-31 20:56:48 -08:00
  • 3f6d817b43 Roll chromium_revision 2ed48364ed..9d90548426 (447343:447390) buildbot 2017-01-31 18:07:27 -08:00
  • dc20e26594 Use correct calling convention for CreateThread callback on Windows. deadbeef 2017-01-31 15:10:44 -08:00
  • 3e4ebc7d89 Roll chromium_revision 0851a43de7..2ed48364ed (447237:447343) buildbot 2017-01-31 15:01:59 -08:00
  • ac61b745df Refactor FakeAudioDevice to have separate methods for starting recording and playout. Also, change FakeAudioDevice to generate a sine tone instead of using a file. perkj 2017-01-31 13:32:49 -08:00
  • 1c056254a6 Fix race condition in FrameBuffer2 philipel 2017-01-31 09:53:12 -08:00
  • 54340d8e75 Change opus min bitrate. michaelt 2017-01-31 09:06:53 -08:00
  • cf34fdea19 Roll chromium_revision 721746ebca..0851a43de7 (447221:447237) buildbot 2017-01-31 08:49:57 -08:00
  • 7f08e82251 Fix per regression in probing. stefan 2017-01-31 07:49:28 -08:00
  • 6fb4f5696d Reland of move usage of deprecated g_type_init API (patchset #1 id:1 of https://codereview.webrtc.org/2666103002/ ) oprypin 2017-01-31 06:50:14 -08:00
  • d1685ab547 Revert of Remove usage of deprecated g_type_init API (patchset #1 id:1 of https://codereview.webrtc.org/2660823003/ ) oprypin 2017-01-31 06:09:30 -08:00
  • 0fe1216c9c Move more calls to webrtc::field_trial::FindFullName into ctor, thereby minimizing the non-trivial cost of repeated string comparisons. elad.alon 2017-01-31 05:48:37 -08:00
  • 89f281c51a Roll chromium_revision f74de5a3c9..721746ebca (447212:447221) buildbot 2017-01-31 05:33:55 -08:00
  • b2caab7f60 Remove usage of deprecated g_type_init API oprypin 2017-01-31 05:16:13 -08:00
  • 3ebbcb528b Stop using VoEVideoSync in Call/VideoReceiveStream. solenberg 2017-01-31 03:58:40 -08:00
  • 63b14b7d15 Add override declarations to PeerConnectionObserver subclasses, and delete obsolete methods. nisse 2017-01-31 03:34:01 -08:00
  • 1783f169d9 Roll chromium_revision a2c4dd1ab5..f74de5a3c9 (447201:447212) buildbot 2017-01-31 03:00:05 -08:00
  • a7ee14ebf5 Suppress Memcheck:Uninitialized error when printing rtc::optional. philipel 2017-01-31 02:45:54 -08:00
  • 1e4e8cb43d Add CreatePeerConnectionFactory overloads that take audio codec factory args kwiberg 2017-01-31 01:48:08 -08:00
  • 7ce109acd3 Replace the easy cases of VERIFY usage. nisse 2017-01-31 00:57:56 -08:00
  • 96a9fa0291 Removing webrtc/build folder mbonadei 2017-01-31 00:53:25 -08:00
  • 9f9a1c7b6b Roll chromium_revision e7b7b06987..a2c4dd1ab5 (447179:447201) buildbot 2017-01-31 00:03:43 -08:00
  • a4def99f31 Roll chromium_revision 3eee970eb6..e7b7b06987 (447079:447179) buildbot 2017-01-30 20:35:38 -08:00
  • 02839ae98a Roll chromium_revision 5555191b32..3eee970eb6 (447020:447079) buildbot 2017-01-30 14:46:38 -08:00
  • c14b7ed479 iSAC float decoder: Don't read past end of initialized part of buffer kwiberg 2017-01-30 12:17:05 -08:00
  • ba3f4111ff Roll chromium_revision 716b1b3275..5555191b32 (446992:447020) buildbot 2017-01-30 11:46:58 -08:00
  • a6a6d65278 Instantly pass network changes to controllers in audio network adaptor. minyue 2017-01-30 10:50:00 -08:00
  • 7b8cddd34c Roll chromium_revision d7ee7cd5fa..716b1b3275 (446974:446992) buildbot 2017-01-30 09:21:04 -08:00
  • 4c9b4af53a Compute packet loss for event log visualization similar to how it is defined in RFC 3550. terelius 2017-01-30 08:44:51 -08:00
  • aa4b0775aa Simplify IsFmtpParam according to RFC 4855. ossu 2017-01-30 07:41:18 -08:00
  • 55d6539b86 Roll chromium_revision 4f0acca4ba..d7ee7cd5fa (446964:446974) buildbot 2017-01-30 06:03:30 -08:00
  • a6ca518dd5 iSAC: Untangle some cyclic dependencies kwiberg 2017-01-30 05:28:54 -08:00
  • 4fb97462a8 Add presubmit check to prevent package boundary violations. ehmaldonado 2017-01-30 05:27:22 -08:00
  • 9cbb0a16e5 Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) ehmaldonado 2017-01-30 03:07:03 -08:00
  • 26d79eead0 Roll chromium_revision d2bee43df5..4f0acca4ba (446960:446964) buildbot 2017-01-30 02:46:19 -08:00
  • 1c0dea8675 Delete VideoFrame::set_render_time_ms. nisse 2017-01-30 02:43:18 -08:00
  • a26330ac6f Only define NO_RETURN if undefined agouaillard 2017-01-30 00:54:19 -08:00
  • 2e6048461e Roll chromium_revision 169ed39de4..d2bee43df5 (446956:446960) buildbot 2017-01-30 00:07:17 -08:00
  • 6f873be073 Roll chromium_revision b84d9d8be2..169ed39de4 (446949:446956) buildbot 2017-01-29 20:30:57 -08:00