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  • 78460a712a Roll chromium_revision 8e79102feb..b98c4727fc (485221:485244) buildbot 2017-07-10 06:31:29 -07:00
  • 370dd47973 Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ ) ehmaldonado 2017-07-10 05:58:42 -07:00
  • 9483b49baf Remove remains of webrtc/base ehmaldonado 2017-07-10 04:50:54 -07:00
  • ccd7328c3c Roll chromium_revision a28aeb7be0..8e79102feb (485062:485221) buildbot 2017-07-10 03:38:51 -07:00
  • 6c9556e7f1 Prevent warnings in PacketRouterTest and PacketRouterRembTest eladalon 2017-07-10 03:33:00 -07:00
  • 6cc25614a9 Remove webrtc::VideoEncoderFactory magjed 2017-07-10 03:26:36 -07:00
  • c453b08f1e Adding log in OrtcFactoryIntegrationTest and fix a bug. minyue-webrtc 2017-07-10 11:56:05 +02:00
  • 0b249c24bf Refactor gunit for synergy to gtest. minyue-webrtc 2017-07-10 11:52:21 +02:00
  • 539d104e3e Revert of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt (patchset #2 id:20001 of https://codereview.webrtc.org/2964863002/ ) mbonadei 2017-07-10 02:40:49 -07:00
  • bffe597e69 Convert occurrences of deprecated WEBRTC_TRACE logging to LOG style logging in webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc. saza 2017-07-10 01:05:45 -07:00
  • 5de068082b Add a check in the BlockBuffer of AEC2 to guard for buffer overflows. saza 2017-07-10 01:01:09 -07:00
  • 3ffa72d0f0 Add AudioFrame::ResetWithoutMuting() to address performance regression. Jonathan Yu 2017-07-07 00:05:10 -07:00
  • f1377f7222 Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on the worker thread. perkj 2017-07-07 16:38:28 -07:00
  • cf39dd5d82 Add RTC_FROM_HERE location information to two DCHECKs in ProcessThread. tommi 2017-07-07 16:24:34 -07:00
  • f88f72dd16 Roll chromium_revision 85f2caa791..a28aeb7be0 (484906:485062) buildbot 2017-07-07 15:34:54 -07:00
  • 4dde3df3b5 Move SrtpSession and tests to their own files. zstein 2017-07-07 14:26:25 -07:00
  • 7d0a77eef4 Handle case where UDP packet contains multiple DTLS records. jbauch 2017-07-07 13:44:07 -07:00
  • 7480da4118 Trace loggging: Check for g_event_logger is not null before calling it. jtteh 2017-07-07 11:02:15 -07:00
  • fb660ae633 Decreased the adaptation rate for the adaptive filter in the echo canceller 3 peah 2017-07-07 07:59:24 -07:00
  • 991521e2f9 Roll chromium_revision 6ddb8d2861..85f2caa791 (484869:484906) buildbot 2017-07-07 06:49:42 -07:00
  • de5ff8e2c8 Fix a variable naming typo henrik.lundin 2017-07-07 05:29:47 -07:00
  • e67bedbac3 External APM usage downstream dependency support cleanup peah 2017-07-07 04:25:11 -07:00
  • a0349c138d Injectable Obj-C video codecs Anders Carlsson 2017-07-07 13:19:09 +02:00
  • c1a58bae41 Roll chromium_revision a5ed187842..6ddb8d2861 (484843:484869) buildbot 2017-07-07 03:45:17 -07:00
  • eaaae9e91b base->rtc_base: Update .c, .mm and .java files. ehmaldonado 2017-07-07 03:09:51 -07:00
  • 0e9bd8e15b Roll chromium_revision 4b357464fd..a5ed187842 (484824:484843) buildbot 2017-07-07 01:32:07 -07:00
  • f04afde85a Report interframe delay sum in old GetStats ilnik 2017-07-07 01:26:24 -07:00
  • 5b361730d0 Support re-entrant calls to MessageQueueManager::Clear. jbauch 2017-07-06 23:51:37 -07:00
  • 876088ac77 Roll chromium_revision baaa9eae93..4b357464fd (484696:484824) buildbot 2017-07-06 21:16:17 -07:00
  • 4a494ffd12 desktop_capture: crop border in window_capture on Win8/10 braveyao 2017-07-06 20:20:27 -07:00
  • f07e6b4c00 Roll chromium_revision 2e0945b687..baaa9eae93 (484611:484696) buildbot 2017-07-06 12:35:48 -07:00
  • c20978e581 Rename webrtc/base -> webrtc/rtc_base Edward Lemur 2017-07-06 19:44:34 +02:00
  • ea39dfa770 Roll chromium_revision c33c6bfd24..2e0945b687 (484321:484611) buildbot 2017-07-06 09:48:49 -07:00
  • 9e3f1e4ca2 Fixed a miscalculation of sent bitrate caused by mixup of time units Sebastian Jansson 2017-07-06 16:52:09 +02:00
  • d66072b71b Moving asm code out of common_audio_c sources list mbonadei 2017-07-06 07:44:14 -07:00
  • 3b03476233 Remove MAIN_NIB_FILE from Info.plist because the substitution is broken oprypin 2017-07-06 07:09:57 -07:00
  • a44910787b Let NetEq reset the AudioFrame during muted state henrik.lundin 2017-07-06 05:23:53 -07:00
  • 02569adfd4 Update screen simulcast config sprang 2017-07-06 05:05:50 -07:00
  • 168794c43c Implement RTP keepalive in native stack. sprang 2017-07-06 04:38:06 -07:00
  • 5c0d703382 Moving asm code out of isac_fix_c sources list mbonadei 2017-07-06 03:48:55 -07:00
  • 05db21d5b3 Reland of move webrtc/tools (patchset #1 id:1 of https://codereview.webrtc.org/2973493002/ ) ehmaldonado 2017-07-06 03:34:35 -07:00
  • 2edc6845ac Report timing frames info in GetStats. ilnik 2017-07-06 03:06:50 -07:00
  • 5b7fc8ce42 A few simplifications to CodecDatabase and VCMGenericDecoder. tommi 2017-07-05 16:45:57 -07:00
  • 7025244bc0 Roll chromium_revision f45f1f992e..c33c6bfd24 (484285:484321) buildbot 2017-07-05 12:15:57 -07:00
  • 6aa95117d8 Fix null ref in NetworkMonitorAutoDetect if Connectivity Manager service is unavailable bdodson 2017-07-05 09:55:09 -07:00
  • e4f63a1929 Roll chromium_revision c01b31617b..f45f1f992e (484252:484285) buildbot 2017-07-05 09:47:30 -07:00
  • b16a01f14f Revert "Reland "Adding ANA config event to debug dump."" minyue-webrtc 2017-07-05 16:04:13 +02:00
  • 63d146b743 NetEq: Rectify the implementation of PacketBuffer::DiscardOldPackets henrik.lundin 2017-07-05 07:03:34 -07:00
  • 440ea8cdff Roll chromium_revision 3fe2409358..c01b31617b (484231:484252) buildbot 2017-07-05 06:43:22 -07:00
  • 191113a46b Added implementation of four functions in the BBR congestion controller: gnish 2017-07-05 05:00:46 -07:00
  • bc0c4f581f Roll chromium_revision 6da2ebcead..3fe2409358 (484119:484231) buildbot 2017-07-05 03:51:03 -07:00
  • fae474c9cd Implement packet discard rate in NetEq. minyue-webrtc 2017-07-05 11:17:40 +02:00
  • 889d9654f7 Fix issue with zero rtt reports when using FlexFEC and add perf test. stefan 2017-07-05 03:03:02 -07:00
  • 070efc088e Improves WebRTC.Audio.AveragePlayoutCallbacksBetweenGlitches UMA stat henrika 2017-07-05 02:34:31 -07:00
  • abee2d89cd Roll chromium_revision 2fe6dc66f8..6da2ebcead (484092:484119) buildbot 2017-07-04 09:33:14 -07:00
  • 0fc6d871bd Roll chromium_revision cf58257d56..2fe6dc66f8 (483646:484092) buildbot 2017-07-04 08:00:54 -07:00
  • f720704493 Added philipel@webrtc.org to webrtc/modules/remote_bitrate_estimator/OWNERS. philipel 2017-07-04 07:57:46 -07:00
  • cb576c50ee Fixes build issue based on usage of Android O specific API henrika 2017-07-04 07:02:35 -07:00
  • c43f68e52c Fix do not unregister bluetooth receiver if it was not registered Gustavo Garcia 2017-06-29 11:25:16 +03:00
  • cc8856c9c2 Remove unused static VideoEncoder functions magjed 2017-07-04 06:03:41 -07:00
  • f612998fa0 Override bots to use libstdc++ on Linux oprypin 2017-07-04 05:41:47 -07:00
  • 8eadead3f4 Adds support for USB audio devices in AppRTCMobile on Android. henrika 2017-07-04 05:10:48 -07:00
  • a9521e248e Reduce send rate to 50% if overusing before we have an acknowledged bitrate. terelius 2017-07-04 04:52:58 -07:00
  • 2c3161c86e Changed default value for the duration of the echo in echocanceller 3 peah 2017-07-04 04:33:11 -07:00
  • 0d7f04daa0 Reland of Add received audio/video call duration metrics based on packets. saza 2017-07-04 04:05:06 -07:00
  • 38fecafa48 Revert of Remove webrtc/tools (patchset #1 id:1 of https://codereview.webrtc.org/2970743003/ ) ehmaldonado 2017-07-04 02:02:49 -07:00
  • d3588cfb31 Improved low-level echo handling in echo canceller 3 peah 2017-07-04 01:54:37 -07:00
  • ed56680adb Remove webrtc/tools ehmaldonado 2017-07-04 01:34:47 -07:00
  • 382f21cd9c Revert of Add received audio and video call duration metrics based on packets. (patchset #4 id:140001 of https://codereview.webrtc.org/2957073002/ ) saza 2017-07-04 01:11:49 -07:00
  • 746749237a Add received audio/video call duration metrics based on packets. saza 2017-07-04 00:19:22 -07:00
  • 2a2b297aa6 Add underscore at end of Call members' names eladalon 2017-07-03 09:25:27 -07:00
  • 4235d78b57 Disabling flaky complexity tests for the audio processing module. peah 2017-07-03 09:11:22 -07:00
  • 7ab7fd66c4 Fix gmock warnings emanating from FlexfecReceiveStreamTest eladalon 2017-07-03 06:57:13 -07:00
  • 7c7796b8ec Register FlexFEC SSRC to receive RTCP on sending side. brandtr 2017-07-03 06:02:53 -07:00
  • 48587f91f8 Changing AudioConferenceMixer logging to base/logging.h Alex Loiko 2017-05-30 16:59:40 +02:00
  • 4257ab2e02 Add received interframe delay UMA metrics ilnik 2017-07-03 01:15:58 -07:00
  • ad515c459b fix comment length jansson 2017-07-03 00:47:44 -07:00
  • a80c16a67c Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" Henrik Kjellander 2017-07-01 16:48:15 +02:00
  • dca1e09db7 Revert "Update includes for webrtc/{base => rtc_base} rename (1/3)" Henrik Kjellander 2017-07-01 16:42:22 +02:00
  • c8fa692ec4 Update includes for webrtc/{base => rtc_base} rename (1/3) kjellander 2017-06-30 14:02:00 -07:00
  • c3771cc4d3 Update includes for webrtc/{base => rtc_base} rename (2/3) kjellander 2017-06-30 13:42:44 -07:00
  • 89c4a7e57d Wire up experiment for improved screenshare bwe. sprang 2017-06-30 13:27:40 -07:00
  • e96c45b662 Reland "Update includes for webrtc/{base => rtc_base} rename (3/3)" kjellander 2017-06-30 10:45:21 -07:00
  • 224e65939a Reland of "VideoFrameBuffer: Remove deprecated functions" Magnus Jedvert 2017-06-30 12:14:42 +00:00
  • e75d96b5bd Revert of Test and fix for huge bwe drop after alr state. (patchset #13 id:320001 of https://codereview.webrtc.org/2931873002/ ) terelius 2017-06-30 08:11:44 -07:00
  • 2d54784d89 Reland "Adding ANA config event to debug dump." minyue-webrtc 2017-06-30 15:25:38 +02:00
  • 17675ceb13 Enable the injection of an APM into a peerconnection peah 2017-06-30 07:24:04 -07:00
  • fa8567868e Fix FecTest.FlexfecTest flakiness caused by seq. num. wraparound. brandtr 2017-06-30 07:22:15 -07:00
  • 697a590314 Added the ability to adjust the AEC3 performance for large rooms peah 2017-06-30 07:06:10 -07:00
  • 4583db4967 Enable -Wunused-function warning everywhere. Henrik Kjellander 2017-06-22 14:27:27 +02:00
  • 1009cfca47 More gracefully handle rtp timestamp jumps in the rtp to ntp estimator. stefan 2017-06-30 06:28:13 -07:00
  • 9addbebf42 Remove RtpDemuxer tweak for preventing multiple RSID inspections eladalon 2017-06-30 06:26:54 -07:00
  • 49085ef280 Improves audio-routing in combination with BT in AppRTCMobile on Android. henrika 2017-06-30 06:25:25 -07:00
  • 0072511073 Revert "Update includes for webrtc/{base => rtc_base} rename (3/3)" Henrik Kjellander 2017-06-30 15:14:45 +02:00
  • f1c5ebf829 Update includes for webrtc/{base => rtc_base} rename (3/3) kjellander 2017-06-30 05:27:14 -07:00
  • 96d74bb933 Opus implementation of the AudioDecoderFactoryTemplate API kwiberg 2017-06-30 05:24:56 -07:00
  • 3aba2d1af9 Fix android video_quality_loopback_test ehmaldonado 2017-06-30 05:12:09 -07:00
  • d76b75370c Disable AudioDeviceTest.StartStopRecording on iOS henrika 2017-06-30 05:08:40 -07:00
  • 96da0115d7 Opus implementation of the AudioEncoderFactoryTemplate API kwiberg 2017-06-30 04:23:22 -07:00
  • 41bafb2f2d Update PRESUBMIT.py for webrtc/{tools => rtc_tools} rename. kjellander 2017-06-30 04:14:54 -07:00