- ac8d516 Improves release of allocated audio resources on Android. by henrika · 8 years ago
- 43c3821 Revert of Avoid precision loss in TrendlineEstimator from int64_t -> double conversion (patchset #7 id:120001 of https://codereview.webrtc.org/2577463002/ ) by terelius · 8 years ago
- 0bac07a Revert of Avoid precision loss in MedianSlopeEstimator from int64_t -> double conversion (patchset #3 id:40001 of https://codereview.webrtc.org/2578543002/ ) by terelius · 8 years ago
- ebcbcc3 Pass arrival time as an int64_t rather than a double to the MedianSlopeEstimator to avoid precision loss. by terelius · 8 years ago
- df2ceb8 Reland of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #1 id:1 of https://codereview.webrtc.org/2574123002/ ) by nisse · 8 years ago
- bf65be5 Wire-up audio BWE with overhead. by michaelt · 8 years ago
- c12cbaf Avoid precision loss in TrendlineEstimator by passing the arrival time as an int64_t instead of a double. by terelius · 8 years ago
- 3168c7a Rename RTCOutboundRTPStreamStats *_rtt members to *_round_trip_time. by hbos · 8 years ago
- 6a58f33 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/ ) by nisse · 8 years ago
- 7bf5369 RTCStatsIntegrationTest: TestMemberIsIDReference on all defined IDs. by hbos · 8 years ago
- 06035cf 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/ ) by nisse · 8 years ago
- 0b571a1 MB: Enable memcheck for the linux_memcheck trybot. by Henrik Kjellander · 8 years ago
- e10e6d1 RTCOutboundRTPStreamStats.roundTripTime: Only report non-negative values. by hbos · 8 years ago
- 24db179 Move tools/autoroller to tools-webrtc/ + rename script by Henrik Kjellander · 8 years ago
- 4128649 Move all codec specific definitions from modules_include by hta · 8 years ago
- b5ffc14 Create top-level dir tools-webrtc and start moving things into it. by Henrik Kjellander · 8 years ago
- ef753e2 Remove unnecessary third_party links: nss, llvm-build, syzygy by kjellander · 8 years ago
- 3bc031b Remove unused items in tools/ by Henrik Kjellander · 8 years ago
- c3765f9 Roll chromium_revision b935b59277..5e0dca78b3 (438725:438769) by buildbot · 8 years ago
- f2832a0 Roll chromium_revision 9807edde11..b935b59277 (438688:438725) by buildbot · 8 years ago
- 2c8d94b Roll chromium_revision 699f628e13..9807edde11 (438637:438688) by buildbot · 8 years ago
- 2769ec6 Add WriteIsolatedOutput() functions by zijiehe · 8 years ago
- fba7900 Roll chromium_revision eae1bf6b1e..699f628e13 (438554:438637) by buildbot · 8 years ago
- 88cf05c Guard against uninitialized packetization modes. by hta · 8 years ago
- d30bd18 Roll chromium_revision a20ca9fe57..eae1bf6b1e (438523:438554) by buildbot · 8 years ago
- 9ce8cbf Roll chromium_revision 135e29eed5..a20ca9fe57 (438491:438523) by buildbot · 8 years ago
- 9a394f0 Skip RTCMediaStreamTrackStats.echoReturnLoss[Enhancement] default value. by hbos · 8 years ago
- c3c2f31 Adds basic Bluetooth support to AppRTCMobile by henrika · 8 years ago
- db8af2a Run 'git cl format --full' on Base64. by johan · 8 years ago
- 9006987 Remove deprecated RTPSender::SendPadData by danilchap · 8 years ago
- e2ec7c2 Remove static cast from H264SpropParameterSets. by johan · 8 years ago
- 930959d Improvements to the reliability of the echo detector perf test. by ivoc · 8 years ago
- a701469 Roll chromium_revision a8e17a3031..135e29eed5 (438476:438491) by buildbot · 8 years ago
- 8fc0c4c Add vector<uint8_t> to Base64 decoded data types. by johan · 8 years ago
- 0878f94 Delete accidental drmemory symlink by kjellander@webrtc.org · 8 years ago
- 665da28 Autoroller: Add --ignore-unclean-workdir flag by kjellander · 8 years ago
- a5bb562 Delete webrtc/transport.h. by aleloi · 8 years ago
- 9e1e6c5 Corrected access of null pointer in audioproc_f: by peah · 8 years ago
- 63e6a38 Removes verification of audio parameters on Android by henrika · 8 years ago
- fded4cc Roll chromium_revision 5fb8c41aea..a8e17a3031 (438448:438476) by buildbot · 8 years ago
- 0989fbc Revert of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #5 id:80001 of https://codereview.webrtc.org/2541863002/ ) by nisse · 8 years ago
- 7b25166 Fix for left shift of negative value in NetEq. by ivoc · 8 years ago
- bd6c6fa Delete method Pathname::url and base/urlencode* by nisse · 8 years ago
- bb66ec3 Disable flaky test VideoProcessorIntegrationTest.ProcessNoLossChangeFrameRateFrameDropVP9 by skvlad · 8 years ago
- e0eae3c This CL adds the basic framework for AEC3 in the audio processing module. by peah · 8 years ago
- db39742 Delete unused class rtc::RegKey. by nisse · 8 years ago
- e5dc62a PRESUBMIT: Add authorized-authors check + AUTHORS e-mails. by kjellander · 8 years ago
- 43c5a97 Delete stl_util.h. Unused since cl https://codereview.webrtc.org/2447103002 by nisse · 8 years ago
- 8afbc8c 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/ ) by nisse · 8 years ago
- 36f74e5 Delete rtc::linked_ptr. Only use, in statstypes.h, replaced with std::unique_ptr. by nisse · 8 years ago
- dd3c811 Roll chromium_revision cfd026f99e..5fb8c41aea (438418:438448) by buildbot · 8 years ago
- a5073c0 Disable AudioDeviceTest.StartPlayoutOnTwoInstances on iOS by Henrik Kjellander · 8 years ago
- 80df795 Roll chromium_revision e234d53ddf..cfd026f99e (438369:438418) by buildbot · 8 years ago
- e26b89c Roll chromium_revision b571577c64..e234d53ddf (438292:438369) by buildbot · 8 years ago
- 62802a1 Fixing possible crash due to RefCountedChannel assignment operator. by deadbeef · 8 years ago
- b236257 Fixing integer overflow when parsing bandwidth attribute. by deadbeef · 8 years ago
- 9396a08 Roll chromium_revision 79b1930444..b571577c64 (438242:438292) by buildbot · 8 years ago
- 95aa964 Support external audio mixer in webrtc 2. by gyzhou · 8 years ago
- 7af91dd Removing "crypto_required" from MediaContentDescription. by deadbeef · 8 years ago
- 00fd520 Roll chromium_revision 047b36f906..79b1930444 (438176:438242) by buildbot · 8 years ago
- b68cc75 ParseCandidate(): Refactor to fix memcheck false positive. by hnsl · 8 years ago
- f8b262e Roll chromium_revision e882052d97..047b36f906 (438143:438176) by buildbot · 8 years ago
- 301fc4a Update common_audio/smoothing_filter. by minyue · 8 years ago
- bfcf561 Delete VideoFrame default constructor, and IsZeroSize method. by nisse · 8 years ago
- 46711db Disable flaky QualityScaler tests for now. by kthelgason · 8 years ago
- 277b250 Refactor "secure bool" into explicit PROTO_TLS. by hnsl · 8 years ago
- 1c4b5bc Roll chromium_revision 632410c83c..e882052d97 (438112:438143) by buildbot · 8 years ago
- 38b6dbc Autoroller: Support for rolling individual DEPS entries. by kjellander · 8 years ago
- ef16e99 Add a gtk3 port of peerconnection_client on Linux by thomasanderson · 8 years ago
- 349092b Logging basic bad call detection by palmkvist · 8 years ago
- e381015 Revert of New PeerConnectionInterface::GetStats: No bogus default implementation. (patchset #1 id:1 of https://codereview.webrtc.org/2566143002/ ) by hbos · 8 years ago
- 4145989 Roll chromium_revision 2d6dcff9ac..632410c83c (438085:438112) by buildbot · 8 years ago
- 07e276c Rename RtpStreamReceiver::SetCodec() to ::AddCodec(). by johan · 8 years ago
- 4b9ff41 setup_links: Remove mojo and WebKit links. by kjellander · 8 years ago
- 8f23094 New PeerConnectionInterface::GetStats: No bogus default implementation. by hbos · 8 years ago
- 03392d0 Fix for negative shift value in NetEq. by ivoc · 8 years ago
- 921019c Delete unused class AsyncFile. by nisse · 8 years ago
- 1b72300 Roll chromium_revision e5fe50e808..2d6dcff9ac (437879:438085) by buildbot · 8 years ago
- 6de92f9 Don't allow changing ICE pool size after SetLocalDescription. by deadbeef · 8 years ago
- 25ed435 Implement parsing/serialization of a=bundle-only. by deadbeef · 8 years ago
- 39ce11f Revert of Support external audio mixer. (patchset #5 id:140001 of https://codereview.webrtc.org/2539213003/ ) by gyzhou · 8 years ago
- f6bcac5 Support external audio mixer in webrtc. by gyzhou · 8 years ago
- 1354901 Making audio network adaptor config proto a JAVA package. by minyue · 8 years ago
- 580df53 Fix header guard in thread_annotations.h. by noahric · 8 years ago
- e5ba75a Destroy encoders that fail to InitEncode. by noahric · 8 years ago
- cb44343 Add SSRC to RtpEncodingParameters for audio. by deadbeef · 8 years ago
- ccecdd4 Roll chromium_revision 45a928c03f..e5fe50e808 (437857:437879) by buildbot · 8 years ago
- 4f19b2f Add OWNERS to BWE modules. by stefan · 8 years ago
- fe793eb Remove sequenced task checker from FlexfecSender. by brandtr · 8 years ago
- e54b0c5 Roll chromium_revision 88e7649411..45a928c03f (437837:437857) by buildbot · 8 years ago
- a9a6d4b Delete voice_engine_configurations.h by henrik.lundin · 8 years ago
- ba7e71b remove googViewLimitedResolution stat by philipp.hancke · 8 years ago
- d2ce622 Disabling the potentially flaky test by peah · 8 years ago
- bd44bb0 Fix out of bound reads in ParseIceServerUrl() for various input. by hnsl · 8 years ago
- b010b8f Roll chromium_revision 4537fa801e..88e7649411 (437826:437837) by buildbot · 8 years ago
- 65a1e77 Try to deflake VideoSendStream tests with ULPFEC. by brandtr · 8 years ago
- e448dd5 RTCIceCandidatePairStats.consentRequestsSent set by RTCStatsCollector by hbos · 8 years ago
- b29b9c8 Replace VideoCaptureDataCallback by VideoSinkInterface. by nisse · 8 years ago
- 99f7bfd Change MANUAL -> DISABLED for ScreenCapturerIntegrationTest tests by Henrik Kjellander · 8 years ago
- 9e3e0da Roll chromium_revision d33aa11bc5..4537fa801e (437814:437826) by buildbot · 8 years ago