Commit Graph

  • eb538fdcc6 Pass arrival time as an int64_t rather than a double to the MedianSlopeEstimator to avoid precision loss. terelius 2016-12-16 02:37:10 -08:00
  • beafee3009 Move ios_helpers to sdk folder kthelgason 2016-12-16 02:12:42 -08:00
  • 8d66245ccc Adding Åsa and Erik as video owners. mflodman 2016-12-16 11:12:31 +01:00
  • 528ec3db0b Don't report packets with id -1 to the transport feedback adapter as they provide no value. stefan 2016-12-16 02:11:26 -08:00
  • 05d6e260f5 Initialize packetization mode in VideoToolbox kthelgason 2016-12-16 02:10:20 -08:00
  • 9a272059c2 whitespace by manual git cl land. Andrii Shyshkalov 2016-12-16 11:00:45 +01:00
  • 2d897bd31b CQ whitespace. tandrii 2016-12-16 01:59:35 -08:00
  • 4cd6221127 iOS: Add trendline filter to field trials. tkchin 2016-12-15 13:17:54 -08:00
  • 5bc3945f8f Fix integer overflow in ProbeController. sergeyu 2016-12-15 10:42:09 -08:00
  • b3564adc91 Avoid precision loss in TrendlineEstimator by passing the arrival time as an int64_t instead of a double. terelius 2016-12-15 08:20:25 -08:00
  • a97c5d233d Add ossu@ to OWNERS of audio/ and modules/audio_coding/ ossu 2016-12-15 07:52:08 -08:00
  • d79f97b542 Fixing loopback video test by reconfiguring the encoder to correct size. mflodman 2016-12-15 07:24:33 -08:00
  • 721d402d71 Create VideoReceiver with external VCMTiming object. philipel 2016-12-15 07:10:57 -08:00
  • ac8d5164f0 Improves release of allocated audio resources on Android. henrika 2016-12-15 15:42:56 +01:00
  • 43c382111d Revert of Avoid precision loss in TrendlineEstimator from int64_t -> double conversion (patchset #7 id:120001 of https://codereview.webrtc.org/2577463002/ ) terelius 2016-12-15 06:42:44 -08:00
  • 0bac07a89b Revert of Avoid precision loss in MedianSlopeEstimator from int64_t -> double conversion (patchset #3 id:40001 of https://codereview.webrtc.org/2578543002/ ) terelius 2016-12-15 06:41:39 -08:00
  • ebcbcc3b24 Pass arrival time as an int64_t rather than a double to the MedianSlopeEstimator to avoid precision loss. terelius 2016-12-15 06:31:19 -08:00
  • df2ceb88a8 Reland of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #1 id:1 of https://codereview.webrtc.org/2574123002/ ) nisse 2016-12-15 06:29:53 -08:00
  • bf65be5435 Wire-up audio BWE with overhead. michaelt 2016-12-15 06:24:49 -08:00
  • c12cbaf9dd Avoid precision loss in TrendlineEstimator by passing the arrival time as an int64_t instead of a double. terelius 2016-12-15 06:19:59 -08:00
  • 3168c7a04b Rename RTCOutboundRTPStreamStats *_rtt members to *_round_trip_time. hbos 2016-12-15 06:17:08 -08:00
  • 6a58f33450 Revert of Delete rtc::linked_ptr. Only use, in statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2581663002/ ) nisse 2016-12-15 03:54:47 -08:00
  • 7bf5369763 RTCStatsIntegrationTest: TestMemberIsIDReference on all defined IDs. hbos 2016-12-15 03:33:35 -08:00
  • 06035cf53a Reland of Delete rtc::linked_ptr. Only use, in statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2576673002/ ) nisse 2016-12-15 03:12:14 -08:00
  • 0b571a11a4 MB: Enable memcheck for the linux_memcheck trybot. Henrik Kjellander 2016-12-15 11:08:59 +01:00
  • e10e6d1f47 RTCOutboundRTPStreamStats.roundTripTime: Only report non-negative values. hbos 2016-12-15 01:54:29 -08:00
  • 24db179bbf Move tools/autoroller to tools-webrtc/ + rename script Henrik Kjellander 2016-12-15 10:20:10 +01:00
  • 41286496cb Move all codec specific definitions from modules_include hta 2016-12-15 00:54:05 -08:00
  • b5ffc149a9 Create top-level dir tools-webrtc and start moving things into it. Henrik Kjellander 2016-12-15 09:48:52 +01:00
  • ef753e2e02 Remove unnecessary third_party links: nss, llvm-build, syzygy kjellander 2016-12-14 23:48:17 -08:00
  • 3bc031beb9 Remove unused items in tools/ Henrik Kjellander 2016-12-15 08:45:49 +01:00
  • c3765f941d Roll chromium_revision b935b59277..5e0dca78b3 (438725:438769) buildbot 2016-12-14 23:07:25 -08:00
  • f2832a0177 Roll chromium_revision 9807edde11..b935b59277 (438688:438725) buildbot 2016-12-14 20:08:59 -08:00
  • 2c8d94be4b Roll chromium_revision 699f628e13..9807edde11 (438637:438688) buildbot 2016-12-14 17:09:39 -08:00
  • 2769ec62c0 Add WriteIsolatedOutput() functions WriteIsolatedOutput() functions write large content into swarming isolated output folder, which are useful to log large test data for debugging purpose. zijiehe 2016-12-14 15:03:03 -08:00
  • fba7900f76 Roll chromium_revision eae1bf6b1e..699f628e13 (438554:438637) buildbot 2016-12-14 14:30:11 -08:00
  • 88cf05cf73 Guard against uninitialized packetization modes. hta 2016-12-14 12:48:28 -08:00
  • d30bd18048 Roll chromium_revision a20ca9fe57..eae1bf6b1e (438523:438554) buildbot 2016-12-14 11:09:13 -08:00
  • 9ce8cbfe06 Roll chromium_revision 135e29eed5..a20ca9fe57 (438491:438523) buildbot 2016-12-14 08:15:03 -08:00
  • 9a394f0649 Skip RTCMediaStreamTrackStats.echoReturnLoss[Enhancement] default value. hbos 2016-12-14 07:58:22 -08:00
  • c3c2f31852 Adds basic Bluetooth support to AppRTCMobile henrika 2016-12-14 07:36:56 -08:00
  • db8af2a953 Run 'git cl format --full' on Base64. johan 2016-12-14 07:15:12 -08:00
  • 9006987243 Remove deprecated RTPSender::SendPadData danilchap 2016-12-14 06:16:33 -08:00
  • e2ec7c270b Remove static cast from H264SpropParameterSets. johan 2016-12-14 06:08:33 -08:00
  • 930959d261 Improvements to the reliability of the echo detector perf test. ivoc 2016-12-14 05:48:16 -08:00
  • a701469a93 Roll chromium_revision a8e17a3031..135e29eed5 (438476:438491) buildbot 2016-12-14 05:06:52 -08:00
  • 8fc0c4c32f Add vector<uint8_t> to Base64 decoded data types. johan 2016-12-14 04:12:57 -08:00
  • 0878f94650 Delete accidental drmemory symlink kjellander@webrtc.org 2016-12-14 13:05:06 +01:00
  • 665da28857 Autoroller: Add --ignore-unclean-workdir flag kjellander 2016-12-14 03:57:08 -08:00
  • a5bb562bfe Delete webrtc/transport.h. aleloi 2016-12-14 03:48:47 -08:00
  • 9e1e6c599d Corrected access of null pointer in audioproc_f: The previous CL that added the ability to add and artificial nearend signal had an issue with null pointer access. peah 2016-12-14 03:12:20 -08:00
  • 63e6a38745 Removes verification of audio parameters on Android henrika 2016-12-14 02:53:26 -08:00
  • fded4ccec4 Roll chromium_revision 5fb8c41aea..a8e17a3031 (438448:438476) buildbot 2016-12-14 02:24:40 -08:00
  • 0989fbcad2 Revert of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #5 id:80001 of https://codereview.webrtc.org/2541863002/ ) nisse 2016-12-14 02:06:44 -08:00
  • 7b2516620c Fix for left shift of negative value in NetEq. ivoc 2016-12-14 01:59:59 -08:00
  • bd6c6fa309 Delete method Pathname::url and base/urlencode* nisse 2016-12-14 01:42:53 -08:00
  • bb66ec3573 Disable flaky test VideoProcessorIntegrationTest.ProcessNoLossChangeFrameRateFrameDropVP9 skvlad 2016-12-14 01:17:28 -08:00
  • e0eae3cec6 This CL adds the basic framework for AEC3 in the audio processing module. It will be followed by a number of other CLs that extends this framework. peah 2016-12-14 01:16:23 -08:00
  • db397429d4 Delete unused class rtc::RegKey. nisse 2016-12-14 00:30:27 -08:00
  • e5dc62aeb8 PRESUBMIT: Add authorized-authors check + AUTHORS e-mails. kjellander 2016-12-14 00:16:21 -08:00
  • 43c5a974b4 Delete stl_util.h. Unused since cl https://codereview.webrtc.org/2447103002 nisse 2016-12-14 00:07:36 -08:00
  • 8afbc8cba6 Revert of Delete rtc::linked_ptr. Only use, in statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2567143003/ ) nisse 2016-12-14 00:06:31 -08:00
  • 36f74e5579 Delete rtc::linked_ptr. Only use, in statstypes.h, replaced with std::unique_ptr. nisse 2016-12-13 23:36:18 -08:00
  • dd3c811ef8 Roll chromium_revision cfd026f99e..5fb8c41aea (438418:438448) buildbot 2016-12-13 23:15:37 -08:00
  • a5073c0c97 Disable AudioDeviceTest.StartPlayoutOnTwoInstances on iOS Henrik Kjellander 2016-12-14 07:25:05 +01:00
  • 80df795e3c Roll chromium_revision e234d53ddf..cfd026f99e (438369:438418) buildbot 2016-12-13 20:07:04 -08:00
  • e26b89c2dc Roll chromium_revision b571577c64..e234d53ddf (438292:438369) buildbot 2016-12-13 17:21:41 -08:00
  • 62802a1b0e Fixing possible crash due to RefCountedChannel assignment operator. deadbeef 2016-12-13 16:38:36 -08:00
  • b236257763 Fixing integer overflow when parsing bandwidth attribute. deadbeef 2016-12-13 16:37:06 -08:00
  • 9396a08077 Roll chromium_revision 79b1930444..b571577c64 (438242:438292) buildbot 2016-12-13 14:50:04 -08:00
  • 95aa96465d Support external audio mixer in webrtc 2. gyzhou 2016-12-13 14:06:26 -08:00
  • 7af91ddd6b Removing "crypto_required" from MediaContentDescription. deadbeef 2016-12-13 11:29:11 -08:00
  • 00fd520c8c Roll chromium_revision 047b36f906..79b1930444 (438176:438242) buildbot 2016-12-13 11:21:26 -08:00
  • b68cc75f19 ParseCandidate(): Refactor to fix memcheck false positive. Also make supported protocols explicit in check. hnsl 2016-12-13 10:33:41 -08:00
  • f8b262e6e6 Roll chromium_revision e882052d97..047b36f906 (438143:438176) buildbot 2016-12-13 08:08:58 -08:00
  • 301fc4a712 Update common_audio/smoothing_filter. minyue 2016-12-13 06:52:56 -08:00
  • bfcf561923 Delete VideoFrame default constructor, and IsZeroSize method. nisse 2016-12-13 06:08:32 -08:00
  • 46711db355 Disable flaky QualityScaler tests for now. kthelgason 2016-12-13 05:31:13 -08:00
  • 277b250936 Refactor "secure bool" into explicit PROTO_TLS. hnsl 2016-12-13 05:17:23 -08:00
  • 1c4b5bcd6f Roll chromium_revision 632410c83c..e882052d97 (438112:438143) buildbot 2016-12-13 05:11:41 -08:00
  • 38b6dbc793 Autoroller: Support for rolling individual DEPS entries. kjellander 2016-12-13 03:35:21 -08:00
  • ef16e9960f Add a gtk3 port of peerconnection_client on Linux thomasanderson 2016-12-13 02:57:43 -08:00
  • 349092befe Logging basic bad call detection palmkvist 2016-12-13 02:45:57 -08:00
  • e381015ca0 Revert of New PeerConnectionInterface::GetStats: No bogus default implementation. (patchset #1 id:1 of https://codereview.webrtc.org/2566143002/ ) hbos 2016-12-13 02:35:19 -08:00
  • 414598973d Roll chromium_revision 2d6dcff9ac..632410c83c (438085:438112) buildbot 2016-12-13 02:28:34 -08:00
  • 07e276c8d5 Rename RtpStreamReceiver::SetCodec() to ::AddCodec(). johan 2016-12-13 02:23:03 -08:00
  • 4b9ff416b4 setup_links: Remove mojo and WebKit links. kjellander 2016-12-13 01:52:14 -08:00
  • 8f2309478d New PeerConnectionInterface::GetStats: No bogus default implementation. hbos 2016-12-13 01:45:09 -08:00
  • 03392d0047 Fix for negative shift value in NetEq. ivoc 2016-12-13 01:05:27 -08:00
  • 921019c48b Delete unused class AsyncFile. nisse 2016-12-12 23:55:56 -08:00
  • 1b7230015c Roll chromium_revision e5fe50e808..2d6dcff9ac (437879:438085) buildbot 2016-12-12 23:13:06 -08:00
  • 6de92f9255 Don't allow changing ICE pool size after SetLocalDescription. deadbeef 2016-12-12 18:49:32 -08:00
  • 25ed435afe Implement parsing/serialization of a=bundle-only. deadbeef 2016-12-12 18:37:36 -08:00
  • 39ce11f7f6 Revert of Support external audio mixer. (patchset #5 id:140001 of https://codereview.webrtc.org/2539213003/ ) gyzhou 2016-12-12 17:06:49 -08:00
  • f6bcac59e8 Support external audio mixer in webrtc. gyzhou 2016-12-12 16:25:06 -08:00
  • 1354901663 Making audio network adaptor config proto a JAVA package. minyue 2016-12-12 14:06:19 -08:00
  • 580df537ce Fix header guard in thread_annotations.h. noahric 2016-12-12 13:36:46 -08:00
  • e5ba75a658 Destroy encoders that fail to InitEncode. noahric 2016-12-12 13:08:27 -08:00
  • cb44343006 Add SSRC to RtpEncodingParameters for audio. deadbeef 2016-12-12 11:12:36 -08:00
  • ccecdd4560 Roll chromium_revision 45a928c03f..e5fe50e808 (437857:437879) buildbot 2016-12-12 08:10:10 -08:00