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  • 11820502b8 Revert "Make new pacer padding more like old one" Mirko Bonadei 2019-07-12 08:33:22 +00:00
  • 32b183432d Use default task queue factory in fuzzers Danil Chapovalov 2019-07-11 13:58:14 +02:00
  • 44cec0b5bd Handle non-integer frame rates in video codec tests. Sergey Silkin 2019-07-11 14:20:38 +02:00
  • 3ae59d33a3 Use the dummy address 0.0.0.0:9 in the c= and the m= lines if the default connection address is a hostname candidate. Qingsi Wang 2019-07-11 21:53:45 -07:00
  • 58388aa07c Roll chromium_revision 3bcb109517..ade23986de (676409:676659) chromium-webrtc-autoroll 2019-07-11 17:04:10 -07:00
  • 6ff9ebd070 Revert "Refactor FEC code to use COW buffers" Qingsi Wang 2019-07-11 19:50:15 +00:00
  • 0f0668e328 Revert "Cleanup FEC code after refactoring" Qingsi Wang 2019-07-11 19:51:12 +00:00
  • 66b3860fc9 Remove WebRTC-SimulcastScreenshare and enable it by default Florent Castelli 2019-07-10 16:57:57 +02:00
  • ac6c09634f Integrate datagram feedback loop Anton Sukhanov 2019-07-10 15:44:56 -07:00
  • 1e19db1d7e Let git-hyper-blame ignore new format cleanup. Yves Gerey 2019-07-11 16:01:33 +02:00
  • 4e5a41a086 Cleanup FEC code after refactoring Ilya Nikolaevskiy 2019-07-10 11:28:51 +02:00
  • 7325bc3917 Refactor FEC code to use COW buffers Ilya Nikolaevskiy 2019-07-10 11:28:04 +02:00
  • 1300ce12e8 Roll chromium_revision 8bdf10dfd2..3bcb109517 (676126:676409) chromium-webrtc-autoroll 2019-07-11 07:01:52 -07:00
  • bb7727211c Make new pacer padding more like old one Erik Språng 2019-07-10 10:10:37 +02:00
  • 741b96b175 Pass RtpRtcp::Configuration to RtcpReceiver ctor and initialize ssrcs Erik Språng 2019-07-11 13:08:34 +02:00
  • 6f129b3b76 Optimize PacketRouter/RTPSender interactions. Erik Språng 2019-07-11 13:01:58 +02:00
  • 08511d20d0 Roll chromium_revision fff68b4152..8bdf10dfd2 (675999:676126) chromium-webrtc-autoroll 2019-07-10 15:03:07 -07:00
  • ee303fae48 Move datagram_dtls_adaptor from p2p/base/ to pc/ Anton Sukhanov 2019-07-10 11:07:40 -07:00
  • 642a49d1eb Add wrapper to normalize flags. Mirko Bonadei 2019-07-10 19:21:20 +02:00
  • 52e240e2c1 Use 16000Hz audio in PC test when specified Artem Titov 2019-07-10 19:13:19 +02:00
  • ff25b873bf Implements method on RtpPacket to extract extension. Anton Sukhanov 2019-07-09 13:04:07 -07:00
  • ca5f21e293 Make force_fieldtrials persistent string during entire program live. Ruslan Burakov 2019-07-10 16:08:48 +02:00
  • a1471b7a55 Roll chromium_revision 34d38f69ba..fff68b4152 (675873:675999) chromium-webrtc-autoroll 2019-07-10 07:03:11 -07:00
  • b1f2d60456 Reland "Fix collection of audio metrics from PC test framework for audio test" Artem Titov 2019-07-10 14:40:58 +02:00
  • ca160215c9 Update rtc_tools/rtp_generator to compile Ilya Nikolaevskiy 2019-07-10 15:34:45 +02:00
  • 16850598db Add support of quick test mode into PC framework Artem Titov 2019-07-10 14:34:02 +02:00
  • 41300af876 Poison default task queue factory Danil Chapovalov 2019-07-10 12:44:43 +02:00
  • 80cb3f6db6 Remove the injectable bitrate allocation strategy API. Jonas Olsson 2019-07-09 12:31:20 +02:00
  • 0d90a1d039 Do not use hungarian notation for DwmGetWindowAttribute's params Julien Isorce 2019-07-09 09:21:44 -07:00
  • d6f4f74dba Roll chromium_revision 58be81bf4b..34d38f69ba (675720:675873) chromium-webrtc-autoroll 2019-07-09 17:05:55 -07:00
  • 6f420e4248 Reland "Add ability to set RTCP sender ssrc at construction time" Erik Språng 2019-07-09 20:06:30 +02:00
  • 4876cb21c8 Revert "Fix collection of audio metrics from PC test framework for audio test" Mirko Bonadei 2019-07-09 20:08:55 +00:00
  • 49fa4ea0e7 Roll chromium_revision 2bc3837c3d..58be81bf4b (675350:675720) chromium-webrtc-autoroll 2019-07-09 11:01:26 -07:00
  • d0679bd7e2 Enables usage of ChannelMixer in WebRTC's output mixer. henrika 2019-07-09 15:37:45 +02:00
  • b249c54209 Delete GlobalTaskQueueFactory as now unused Danil Chapovalov 2019-07-08 09:28:10 +02:00
  • 6456e352ac Use max bitrate limit recommended by encoder. Sergey Silkin 2019-07-08 17:56:40 +02:00
  • 2d0880b569 Fix collection of audio metrics from PC test framework for audio test Artem Titov 2019-07-09 13:50:18 +02:00
  • bc558cebdc Add support of specifying audio sample rate for PC test framework Artem Titov 2019-07-08 19:13:21 +02:00
  • 495a1aee05 Remove cricket::WebRtcMediaEngineFactory as now unused Danil Chapovalov 2019-07-08 18:19:58 +02:00
  • cb60a8b64c Roll chromium_revision b624ecb939..2bc3837c3d (675206:675350) chromium-webrtc-autoroll 2019-07-08 13:03:05 -07:00
  • 5e25facefd CroppingWindowCapturerWin: filter out cloaked window. Julien Isorce 2019-07-08 11:35:18 -07:00
  • a0eefc17f7 Rename USE_NATIVE_MUTEX_ON_MAC to RTC_USE_NATIVE_MUTEX_ON_MAC. Mirko Bonadei 2019-07-08 14:11:02 +02:00
  • 34462f5dc3 Revert "Add ability to set RTCP sender ssrc at construction time" Ilya Nikolaevskiy 2019-07-08 16:21:41 +00:00
  • 4a126e45c3 Rename tests to prevent clashing with old audio test Artem Titov 2019-07-08 15:28:26 +02:00
  • 94c58fd815 Add ability to set RTCP sender ssrc at construction time Erik Språng 2019-07-06 12:42:27 +02:00
  • a4d873786f Format almost everything. Jonas Olsson 2019-07-05 19:08:33 +02:00
  • c93bfcfd2f Roll chromium_revision 619b1dc518..b624ecb939 (675093:675206) chromium-webrtc-autoroll 2019-07-08 05:03:17 -07:00
  • 48284b86d4 BalancedDegradationSettings: Add option to configure fps based on codec type. Åsa Persson 2019-07-08 10:01:12 +02:00
  • 79e4c92d70 Remove bwe_rtp_play and add rtp_to_text to the build. Mirko Bonadei 2019-07-08 11:52:24 +02:00
  • ab0b9d4236 Switch rtc_event_log2rtp_dump to ABSL_FLAG. Mirko Bonadei 2019-07-08 12:31:59 +02:00
  • eec86cdd96 Fix platform-specific header dependencies to be more precise Oleh Prypin 2019-07-08 09:47:41 +02:00
  • 86f8b3bd5d Roll chromium_revision 48f88caf2e..619b1dc518 (674992:675093) chromium-webrtc-autoroll 2019-07-05 15:05:49 -07:00
  • 9455c305e6 Remove deprecation notice for VideoFrame::ntp_time_ms Ilya Nikolaevskiy 2019-07-03 16:18:13 +02:00
  • f6468d2569 Wire up new PacedSender code path. Erik Språng 2019-07-05 16:53:43 +02:00
  • 668ce0c7fa Remove trial WebRTC-SimulcastMaxLayers and make its behavior default Florent Castelli 2019-07-04 17:06:04 +02:00
  • ecae9cd1a7 Android: Add error callback for GL_OUT_OF_MEMORY in EglRenderer Magnus Jedvert 2019-07-05 14:33:12 +02:00
  • 48b1b18065 Add ability to create EmulatedNetworkNode from BuiltInNetworkBehaviorConfig Artem Titov 2019-07-05 13:09:48 +02:00
  • c8263e0638 Introduce PC level audio quality test. Artem Titov 2019-07-05 12:34:36 +02:00
  • 386802ef7c Move network emulation framework under test/network Artem Titov 2019-07-05 10:48:17 +02:00
  • 13eb602e01 Roll chromium_revision 50acc956cd..48f88caf2e (674882:674992) chromium-webrtc-autoroll 2019-07-05 03:07:17 -07:00
  • fdf74bdb5f Remove non implemented function from WebRtcVideoChannel. philipel 2019-07-03 13:10:45 +02:00
  • 656590dabc Roll chromium_revision 778353d874..50acc956cd (674780:674882) chromium-webrtc-autoroll 2019-07-04 11:01:45 -07:00
  • 1efb4a283b Add field trial for forcing partition resilience mode in libvpx. Rasmus Brandt 2019-07-04 17:31:58 +02:00
  • 3fbf1e2e4a Reduce kMaxSimulcastStreams to 3 Florent Castelli 2019-07-04 17:09:05 +02:00
  • 9d96209240 Switch unpack_aecdump to ABSL_FLAG. Mirko Bonadei 2019-07-04 10:19:10 +02:00
  • 4580ca2e99 Reland: Add ability to set ssrcs of RtpSender at construction time Erik Språng 2019-07-04 10:38:43 +02:00
  • 8e3cb53892 Remove activity_metric tool. Mirko Bonadei 2019-07-04 11:40:44 +02:00
  • 543179e527 Roll chromium_revision 79b588ee95..778353d874 (674667:674780) chromium-webrtc-autoroll 2019-07-04 03:03:39 -07:00
  • 2250b05778 Adding support for channel mixing between different channel layouts. henrika 2019-07-04 11:27:52 +02:00
  • 3f2eeb8136 Adding test on GetSpanSamples() for NetEq PacketBuffer. Minyue Li 2019-07-03 16:43:29 +02:00
  • d2c336f892 [getStats] Implement "media-source" audio levels, fixing Chrome bug. Henrik Boström 2019-07-03 17:11:10 +02:00
  • e8fbc5d702 Refactor WebRtcOpus_PacketHasFec. Minyue Li 2019-07-03 10:14:18 +02:00
  • 67008dfb36 Revert "Replace the implementation of GetContributingSources() on the audio side." Artem Titov 2019-07-04 07:14:11 +00:00
  • c6c730b7fd Roll chromium_revision 54ec0ffb89..79b588ee95 (674538:674667) chromium-webrtc-autoroll 2019-07-03 17:03:02 -07:00
  • 697f8618db Roll chromium_revision 13b7238371..54ec0ffb89 (674397:674538) chromium-webrtc-autoroll 2019-07-03 11:02:39 -07:00
  • 2e60217390 Add speculative checks to RtpPacketHistory Erik Språng 2019-07-03 16:30:55 +02:00
  • 46dda83bcb Improve buffer level estimation with DTX and add CNG time stretching. Jakob Ivarsson 2019-07-03 16:00:30 +02:00
  • 3d642f8442 Rename ..BitrateThresholds to ..BitrateLimits. Sergey Silkin 2019-07-03 15:09:33 +02:00
  • cecee9903f Disable VP9 2nd profile test for ios arm64 Artem Titov 2019-07-03 11:55:28 +02:00
  • 51f599bdaa Make rtc_base/base_java public. Sami Kalliomäki 2019-07-03 14:13:08 +02:00
  • 81e1bf0396 Remove using DegradationPreference from scenario_config.h Artem Titov 2019-07-03 13:46:55 +02:00
  • 6542826868 Add new tests with lossy networks on PC test framework Artem Titov 2019-07-02 16:48:02 +02:00
  • cd8a6e2f38 Add writing and parsing of the abs-capture-time RTP header extension. Chen Xing 2019-07-01 10:56:51 +02:00
  • 53d45baa50 Make TaskQueueFactory required construction parameter for Call Danil Chapovalov 2019-07-03 14:56:33 +02:00
  • 84ce3c08a5 Macro rename s/CS_DEBUG_CHECKS/RTC_CS_DEBUG_CHECKS. Mirko Bonadei 2019-07-03 14:45:54 +02:00
  • a2b30d8940 Add functions to read from/write to bitstream values with known max value Danil Chapovalov 2019-07-03 14:41:50 +02:00
  • 9eee121a8f Switch py_quality_assessment to ABSL_FLAG. Mirko Bonadei 2019-07-03 13:48:25 +02:00
  • b60141b14f Save and serialize the receive RIDs in MediaContentDescription Florent Castelli 2019-07-03 12:47:54 +02:00
  • e8ed83003d WebRtcVideoChannel encoder fallback. philipel 2019-07-03 11:53:48 +02:00
  • e420c6aa01 Add missing include for memcpy/memcmp Artem Titov 2019-07-03 13:43:37 +02:00
  • 6a2c1bae24 Roll chromium_revision ba17fd6b36..13b7238371 (674288:674397) chromium-webrtc-autoroll 2019-07-03 03:03:27 -07:00
  • 8fa7151e4b Replace the implementation of GetContributingSources() on the audio side. Chen Xing 2019-07-02 15:11:38 +02:00
  • 16661ebfd9 Fix: report video_bwe_stats as bytes per second, as specified in the unit Artem Titov 2019-07-03 11:25:20 +02:00
  • 443b7ee635 Destroy existing encoder instance before creating a new one. Sergey Silkin 2019-06-28 12:53:07 +02:00
  • 2c5af4f6dc Add * and / operator into SamplesStatsCounter. Artem Titov 2019-07-03 10:40:16 +02:00
  • 1d46f9c599 Add RtpPacket::IsExtensionReserved(). Erik Språng 2019-07-02 21:24:47 +02:00
  • 6038926379 Roll chromium_revision 35be8751a4..ba17fd6b36 (674036:674288) chromium-webrtc-autoroll 2019-07-02 17:03:13 -07:00
  • 02d7d353a9 Revert "Add ability to set ssrcs of RtpSender at construction time" Amit Hilbuch 2019-07-02 21:04:57 +00:00
  • c4421979fe Check the rid direction matches the direction in simulcast description Florent Castelli 2019-07-02 20:27:42 +02:00
  • 238aab9948 Fix bug in use_datagram_transport configuration. Bjorn A Mellem 2019-07-02 11:06:00 -07:00