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  • 757146baf1 Remove PC factory options param from LocalAudioSource::Create. deadbeef 2017-02-10 21:26:48 -08:00
  • 3f35e484f8 Roll chromium_revision 0dce9fc553..799869657a (449806:449840) buildbot 2017-02-10 21:08:47 -08:00
  • 112b2e99d8 Switching some interfaces to use std::unique_ptr<>. deadbeef 2017-02-10 20:13:37 -08:00
  • 2c87d9991b Roll chromium_revision 037b19bb21..0dce9fc553 (449745:449806) buildbot 2017-02-10 18:18:12 -08:00
  • 81baed36bf Add ability to return moved value from FunctorMessageHandler, Optional. deadbeef 2017-02-10 18:11:11 -08:00
  • a4549d6588 Fix SDP parsing crash due to missing track ID in "a=msid". deadbeef 2017-02-10 17:26:22 -08:00
  • ef35b17c76 Roll chromium_revision cc96c42adf..037b19bb21 (449672:449745) buildbot 2017-02-10 15:07:54 -08:00
  • abdc857967 Update list of supported Android codecs based on field trial dynamically. glaznev 2017-02-10 14:40:57 -08:00
  • 9238245d9b Fix nr of bytes sent to Opus decoder in DTX mode flim 2017-02-10 13:50:38 -08:00
  • 90f1e1e0d7 Fixing SDP parsing crash due to invalid port numbers. deadbeef 2017-02-10 12:35:05 -08:00
  • 612a497d81 Roll chromium_revision 3f4b691682..cc96c42adf (449626:449672) buildbot 2017-02-10 12:06:22 -08:00
  • 5bd5ca344e Rename "PacketTransportInterface" to "PacketTransportInternal". deadbeef 2017-02-10 11:31:50 -08:00
  • 640a200e3d Roll chromium_revision 182bfc3b6c..3f4b691682 (449577:449626) buildbot 2017-02-10 09:39:11 -08:00
  • 88df0bc591 Make functions in fileutils.h use "const std::string&". ehmaldonado 2017-02-10 09:27:14 -08:00
  • 46a0021e4e Retransmitted packets are now counted in receive time ilnik 2017-02-10 09:16:05 -08:00
  • adb374b4ed Remove henrik.lundin from webrtc/common_video/OWNERS henrik.lundin 2017-02-10 08:44:31 -08:00
  • 087bd34d23 Move AudioDecoder and related stuff to the api/ directory kwiberg 2017-02-10 08:15:44 -08:00
  • 84a3759825 Change rtc::VideoSinkWants to have target and a max pixel count sprang 2017-02-10 07:04:27 -08:00
  • e9ad271db4 Increase the send-time history to 60 seconds. stefan 2017-02-10 06:09:28 -08:00
  • 4ca18696c5 Allow residual echo detector to be enabled/disabled using AudioOptions, and no longer disable it on mobile platforms. ivoc 2017-02-10 05:11:09 -08:00
  • bd344e5c14 Roll chromium_revision 7c7d51c594..182bfc3b6c (449558:449577) buildbot 2017-02-10 03:47:55 -08:00
  • 0d729b3039 Check for use_x11 before runnig desktop_capture_modules_tests on linux. ehmaldonado 2017-02-10 01:38:23 -08:00
  • 38e9324e4e Add script for plotting statistics from webrtc integration test logs. asapersson 2017-02-10 01:37:17 -08:00
  • 55c5be0a03 Remove unused methods in WebRtcVoiceEngine and VoiceMediaChannel. solenberg 2017-02-10 01:20:25 -08:00
  • 654d54c073 Use std::unique_ptr in VideoProcessor. asapersson 2017-02-10 00:16:07 -08:00
  • c29988cb9a Roll chromium_revision 033d9e89ed..7c7d51c594 (449543:449558) buildbot 2017-02-09 23:54:47 -08:00
  • 8a4976347c Roll chromium_revision 18995b57d5..033d9e89ed (449497:449543) buildbot 2017-02-09 21:01:31 -08:00
  • a03d438cb4 Roll chromium_revision 5cd331dec1..18995b57d5 (449426:449497) buildbot 2017-02-09 17:56:07 -08:00
  • faedf7f942 Getting rid of "benign blocking error" log spam. deadbeef 2017-02-09 15:09:22 -08:00
  • 22e39708c0 Roll chromium_revision ba3e8dd8fa..5cd331dec1 (449304:449426) buildbot 2017-02-09 14:37:37 -08:00
  • 5d52bf7049 Roll chromium_revision baaeb3f30e..ba3e8dd8fa (449272:449304) kjellander 2017-02-09 11:19:31 -08:00
  • 3795376ba1 replace NtpTime->Clock with Clock->NtpTime dependency danilchap 2017-02-09 11:15:25 -08:00
  • 1e1c84db10 Fixing typo ilnik 2017-02-09 08:32:53 -08:00
  • 85d5ac744b Fix bug in recv-bwe tests introduced when switching to send-side bwe by default in tests. Stefan Holmer 2017-02-09 16:25:16 +01:00
  • 86a6617ef4 Roll chromium_revision cbcac91f7f..baaeb3f30e (449250:449272) buildbot 2017-02-09 06:13:15 -08:00
  • 8443238e26 Remove rtcp_utility as mostly unused. Since the only used class is RTCPUtilitiy::NackStats, rename it to RtcpNackStats and move it into dedicated file. danilchap 2017-02-09 05:21:42 -08:00
  • 9def800b70 Added a flag to AudioCodecSpec to indicate adaptive bitrate support. ossu 2017-02-09 05:14:32 -08:00
  • 0289364abc Remove unused voe_stress_test.cc solenberg 2017-02-09 05:03:25 -08:00
  • 3dd5ad9d50 Reland of Added VP8 simulcast tests. Fixed analyzer to correctly infer timestamps. (patchset #2 id:150001 of https://codereview.webrtc.org/2687073002/ ) ilnik 2017-02-09 04:58:53 -08:00
  • cc452e1179 Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ ) sakal 2017-02-09 04:53:45 -08:00
  • e67c59e7d2 Revert of Added VP8 simulcast tests. Fixed analyzer to correctly infer timestamps. (patchset #5 id:80001 of https://codereview.webrtc.org/2668763004/ ) ilnik 2017-02-09 04:08:56 -08:00
  • ae8121701d Roll chromium_revision 2019b9e075..cbcac91f7f (449230:449250) buildbot 2017-02-09 03:30:33 -08:00
  • 1752a10791 Remove unused voe_cpu_test.cc. solenberg 2017-02-09 03:25:56 -08:00
  • a48e1b6bc3 Fix for left shift of potentially negative value. ivoc 2017-02-09 03:05:59 -08:00
  • 2324b35890 Remove unused voe_output_test.cc. solenberg 2017-02-09 01:23:25 -08:00
  • 3029210c35 Move Android video quality loopback script. kjellander 2017-02-09 01:05:54 -08:00
  • 234accd8e1 Roll chromium_revision cf2dce6a6d..2019b9e075 (448969:449230) kjellander 2017-02-08 23:56:43 -08:00
  • 94a2f21c05 Increase STUN RTOs to work better on poor networks, such as 2G networks. pthatcher 2017-02-08 14:42:22 -08:00
  • 1749bc372e Use fake clock in some more networks tests. pthatcher 2017-02-08 13:18:00 -08:00
  • 4da058c0dd Create an Obj-C wrapper of the RtpReceiverObserverInterface. zhihuang 2017-02-08 10:41:20 -08:00
  • bb46b95dbe Add option to print information about configured SSRCs from RTC event logs. terelius 2017-02-08 09:37:30 -08:00
  • ed1850a71b Log information (at level LS_INFO) about which overuse estimator is used. terelius 2017-02-08 08:45:20 -08:00
  • 273f31b85c Fix for flaky RemoveOverheadFromBandwidth test. michaelt 2017-02-08 08:21:52 -08:00
  • 87d11cdbca Reland of Avoid calling PostTask in audio callbacks (patchset #1 id:1 of https://codereview.webrtc.org/2684913003/ ) henrika 2017-02-08 07:16:56 -08:00
  • 5d83780c42 Fix flaky test introduced by r16478 stefan 2017-02-08 07:09:05 -08:00
  • 0e3213a18c Fix bug in BitrateProber where an old probe added at a high bitrate will stay active indefinitely if the bandwidth estimate becomes too low to probe at that bitrate. Stefan Holmer 2017-02-08 15:19:05 +01:00
  • 488c5dcd8b Add new target direct_transport and remove fake_network and direct_transport from test_common. The purpose is to be able to use this feature without building and using all of test_common. However, test_common still builds call_test.cc that is depending on direct_transport. perkj 2017-02-08 05:55:51 -08:00
  • 91873b74c5 Roll chromium_revision 70957b2671..cf2dce6a6d (448581:448969) buildbot 2017-02-08 05:49:35 -08:00
  • e525d6aba6 Revert Make the new jitter buffer the default jitter buffer. stefan 2017-02-08 05:25:42 -08:00
  • 498ee8e816 Remove repeat flag from SendRTCP danilchap 2017-02-08 05:24:31 -08:00
  • fd8f102a84 Revert of Avoid calling PostTask in audio callbacks (patchset #6 id:100001 of https://codereview.webrtc.org/2663383004/ ) henrika 2017-02-08 05:23:15 -08:00
  • 219208991b Adding full initial version of delay estimation functionality in echo canceller 3 peah 2017-02-08 05:08:56 -08:00
  • d4ed7f59e4 New tool for printing basic packet information from an RTC event log to stdout. terelius 2017-02-08 04:22:53 -08:00
  • abcef5d32e Replace std::tr1::tuple by ::testing::tuple. ehmaldonado 2017-02-08 04:07:11 -08:00
  • 5e5a072f2c iOS: Fix breakage caused by buildbot recipe update Henrik Kjellander 2017-02-08 12:04:20 +01:00
  • b10f32f9b2 Adding more comments to every header file in api/ subdirectory. deadbeef 2017-02-08 01:38:21 -08:00
  • 76e02cd4d8 Reland of ll chromium_revision 496a750d38..70957b2671 (447619:448581) (patchset #1 id:1 of https://codereview.webrtc.org/2680743003/ ) kjellander 2017-02-08 01:27:33 -08:00
  • 54b6e982df Added gn target for rtc_event_log2rtp_dump. ivoc 2017-02-08 00:28:09 -08:00
  • 7798501d7a Fix the Chrome crash caused by RtcEventLog zhihuang 2017-02-07 15:45:16 -08:00
  • 9dd77baca4 Clarifying error messages in ParseIceServerUrl for invalid transport parameters. zstein 2017-02-07 15:09:50 -08:00
  • 69fb2cca4d Revert of Add QP sum stats for received streams. (patchset #10 id:180001 of https://codereview.webrtc.org/2649133005/ ) skvlad 2017-02-07 10:59:25 -08:00
  • ed02c6d68f Revert of RTCInboundRTPStreamStats.qpSum collected. (patchset #4 id:80001 of https://codereview.webrtc.org/2675943002/ ) skvlad 2017-02-07 10:45:31 -08:00
  • 76bc8e858f Delete VideoReceiveStream::Config::pre_render_callback. nisse 2017-02-07 09:37:41 -08:00
  • cd195bea5e RTCInboundRTPStreamStats.qpSum collected. hbos 2017-02-07 08:31:27 -08:00
  • c16fa5ea69 Replace all use of the VERIFY macro. nisse 2017-02-07 07:18:43 -08:00
  • ff0e72fd16 Add QP sum stats for received streams. sakal 2017-02-07 07:15:17 -08:00
  • 7de8d64f89 Wire up audio packet loss to BWE. stefan 2017-02-07 07:14:08 -08:00
  • 2bc6864278 Reland of Drop frames until specified bitrate is achieved. (patchset #1 id:1 of https://codereview.webrtc.org/2666303002/ ) kthelgason 2017-02-07 07:02:22 -08:00
  • 338f78ac95 RTCIceCandidatePairStats.available[Outgoing/Incoming]Bitrate collected. hbos 2017-02-07 06:41:21 -08:00
  • 3443bb75a0 RTCRTPStreamStats.ssrc changed type to uint32_t. hbos 2017-02-07 06:28:11 -08:00
  • 87b8e9f3a2 Add missing dependency to audio_decoder_unittests. ehmaldonado 2017-02-07 06:26:29 -08:00
  • a53d4e7b4f Reduce parallel jobs in build_aar.py to 200 when building with goma. sakal 2017-02-07 06:19:20 -08:00
  • f81be0a83b Revert of Roll chromium_revision 496a750d38..70957b2671 (447619:448581) (patchset #1 id:1 of https://codereview.webrtc.org/2683593002/ ) kjellander 2017-02-07 05:48:12 -08:00
  • 585a9b191c Refactor and clean-up relating to RTCCodecStats. hbos 2017-02-07 04:59:16 -08:00
  • 040f5cc5d7 Roll chromium_revision 496a750d38..70957b2671 (447619:448581) buildbot 2017-02-07 04:54:19 -08:00
  • b99b5960e1 Add chromium-junit4 tag to instrumentation test AndroidManifests. sakal 2017-02-07 04:12:30 -08:00
  • e0ac5a6c15 Use std::unique_ptr in VideoProcessorIntegrationTest. asapersson 2017-02-07 03:54:04 -08:00
  • 1b21b9bbf8 Replace occurences of string by std::string. ehmaldonado 2017-02-07 03:40:28 -08:00
  • 1634e16042 Remove use of selectors matching Apple private API names. kthelgason 2017-02-07 02:48:55 -08:00
  • 4a9a595ab4 Make rtcp packets copyable That would simplify their usage in tests where perfomance is not critical. danilchap 2017-02-07 01:53:04 -08:00
  • 1959b63b61 Remove Assert lint suppression. sakal 2017-02-07 01:41:47 -08:00
  • 4709e8971b Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call. nisse 2017-02-07 01:18:43 -08:00
  • 6b3fcfd823 Add support for extra GN args to Android build script. kjellander 2017-02-07 01:11:06 -08:00
  • 6b34124a6d Remove unnecessary RTPHeaderParser, following https://codereview.webrtc.org/2659563002/ solenberg 2017-02-06 13:39:38 -08:00
  • f748ca47b0 Change order of tear down/create of default audio stream, to avoid starting/stopping audio card playout unnecessarily. solenberg 2017-02-06 13:03:19 -08:00
  • bd9a77f4e5 Remove most of the remaining calls to VoE APIs from Audio[Send|Receive]Stream. solenberg 2017-02-06 12:53:57 -08:00
  • f9b6e5e4b5 Fix KeepsHighBitrateWhenReconfiguringSender to avoid flakiness if probing succeeds in between encoder reconfigurations. Stefan Holmer 2017-02-06 17:17:57 +01:00
  • 7a2d8ca9bc Rewrite iOS FAT libraries build script in Python oprypin 2017-02-06 07:53:41 -08:00
  • 1134b7b918 Reland of Improve and re-enable FEC end-to-end tests. (patchset #1 id:1 of https://codereview.webrtc.org/2672373002/ ) brandtr 2017-02-06 06:35:47 -08:00
  • b77c716d8a Enable send-side BWE by default for video in call tests. stefan 2017-02-06 06:29:38 -08:00