- 192164e Preparational work before introducing the locks in order to harmonize the code: by peah · 10 years ago
- 4d291f7 Applied the render queueing to the agc. by peah · 10 years ago
- 03179cd Roll chromium_revision 6fd4bdd..a2e8a40 (359891:359987) by kjellander · 10 years ago
- 740c4f1 Remove packet initializer in RtpRtcpRtxNackTest. by pbos · 10 years ago
- 854e84c Use webrtc/base/logging.h for video coding/processing. by pbos · 10 years ago
- c91d173 Revert of Several Tick counter improvements. (patchset #8 id:140001 of https://codereview.webrtc.org/1415923010/ ) by thaloun · 10 years ago
- fa6228e Introduced the render sample queue for the aec and aecm. by peah · 10 years ago
- 4c27e4b Several Tick counter improvements. by Tim Haloun · 10 years ago
- eb8b388 Fix VP9 support in AppRTCDemo. by Alex Glaznev · 10 years ago
- 6f8ce06 common_video: rename interface -> include by kjellander · 10 years ago
- 591cb1f Roll chromium_revision c958aa7..6fd4bdd (359816:359891) by kjellander · 10 years ago
- b27f590 Create rtc::AtomicInt POD struct. by pbos · 10 years ago
- 3528a27 Flesh out webrtc/.gitignore by brucedawson · 10 years ago
- 482b12e Remove BundleFilter filtering of RTCP. by pbos · 10 years ago
- 8b85de2 Converted a bunch of error checking in Audio[Receive|Send]Stream to RTC_CHECKs instead. They should never fail. by solenberg · 10 years ago
- 9a7c838 Adding stddef.h to opus_inst.h. by minyue · 10 years ago
- 3a94154 Move some send stream configuration into webrtc::AudioSendStream. by solenberg · 10 years ago
- 633a3aa ThreadUtils: Add joinUninterruptibly() with timeout by magjed · 10 years ago
- e155ae6 Move CNG and RED management into the Rent-A-Codec by kwiberg · 10 years ago
- 54e9232 Revert of Do not delete the turn port entry right away when the respective connection is deleted. (patchset #5 id:260001 of https://codereview.webrtc.org/1426673007/ ) by tommi · 10 years ago
- 2a654fa Roll chromium_revision cad2987..c958aa7 (359796:359816) by kjellander · 10 years ago
- 0b9e29c Remove include dirs from modules/{media_file,pacing} by Henrik Kjellander · 10 years ago
- 3e0f602 Android EglBase: Add support for creating EGLSurface from Surface, not SurfaceHolder by magjed · 10 years ago
- d9b75be Fix a data race in the thread unit tests. by nisse · 10 years ago
- 6f14be8 Add limit for minimum number of required samples before recording input and sent framerate stats. by asapersson · 10 years ago
- 3c735f4 Roll chromium_revision b77e5bb..cad2987 (359767:359796) by kjellander · 10 years ago
- 8c64860 Roll chromium_revision 3b7968d..b77e5bb (359482:359767) by kjellander · 10 years ago
- e58fe8e Do not delete the turn port entry right away when the respective connection is deleted. by honghaiz · 10 years ago
- fa5d0db cleanup: get rid of basicdefs.h include by tfarina · 10 years ago
- a4845ef Fix flaky tests by honghaiz · 10 years ago
- 4a41361 Android SurfaceViewRenderer: Never hold a pending frame indefinitely by magjed · 10 years ago
- c01c254 Revert of Android MediaCodecVideoDecoder: Manage lifetime of texture frames (patchset #12 id:320001 of https://codereview.webrtc.org/1422963003/ ) by Per · 10 years ago
- f8506cb rtcp::Ij renamed to rtcp::ExtendedJitterReport by danilchap · 10 years ago
- cbe9f51 Revert of Remove global list of SRTP sessions. (patchset #4 id:60001 of https://codereview.webrtc.org/1416093010/ ) by phoglund · 10 years ago
- 0fa9b22 Remove scoped_ptrs for VCM sender_ and receiver_. by pbos · 10 years ago
- df948f0 rtcp::ReportBlock refactored to contain parsing by danilchap · 10 years ago
- 0a41893 Remove BitrateController dependency fromVideoReceiveStream. by mflodman · 10 years ago
- 464c087 Rename screenshare test. by philipel · 10 years ago
- 0e7e259 Move BitrateAllocator from BitrateController logic to Call. by mflodman · 10 years ago
- 69191ed Roll chromium_revision 4771dd5..3b7968d (359351:359482) by kjellander · 10 years ago
- faac497 Fix for scenario where m-line is revived after being set to port 0. by deadbeef · 10 years ago
- 69d0d46 Roll chromium_revision e658ee0..4771dd5 (359300:359351) by kjellander · 10 years ago
- 2cd7afe Do not delete a connection until it has not received anything for 30 seconds. by Honghai Zhang · 10 years ago
- 8597543 Schedule a CreatePermissionRequest after the success of a previous request by Honghai Zhang · 10 years ago
- 68876f9 Introduces Android API level linting, fixes all current API lint errors. by Patrik Höglund · 10 years ago
- 56a34df Re-add a thread check in Call::Call that was removed by mistake in a rebase. by solenberg · 10 years ago
- 9576e54 Reland "Prepare MediaCodecVideoEncoder for surface textures."" by perkj · 10 years ago
- 8093d54 Change default SSRC for RTCP receiver reports to not collide with video. by solenberg · 10 years ago
- dfe434e Roll chromium_revision b0415d9..e658ee0 (359214:359300) by kjellander · 10 years ago
- 5dda80a Remove webrtc/modules/video_{capture,render}/include by Henrik Kjellander · 10 years ago
- e71b24e OpenSL ES stability improvements. by henrika · 10 years ago
- fc6affc Android SurfaceViewRenderer: Call glClear() for every frame to avoid bad GL state by magjed · 10 years ago
- 9683964 Trivial initialization fix in AudioDeviceIOS by henrika · 10 years ago
- 31c8167 Roll chromium_revision 7e059f9..b0415d9 (359143:359214) by kjellander · 10 years ago
- a8e9f5e A little cleanup in p2ptransportchannel and transportchannel. by honghaiz · 10 years ago
- 066ded9 Relax the stun ping check on valid result. by guoweis · 10 years ago
- 33daa7e Roll chromium_revision 4a38519..7e059f9 (359080:359143) by kjellander · 10 years ago
- 6b14f93 Adjust parameter for VP9 resize unittest. by Marco · 10 years ago
- 9b5ee9c Send back ping response if the ping comes from an unknown address. by honghaiz · 10 years ago
- 653b8e0 Reland of Adding the ability to change ICE servers through SetConfiguration. (patchset #1 id:1 of https://codereview.webrtc.org/1424803004/ ) by deadbeef · 10 years ago
- 9b72af9 Remove webrtc/modules/audio_processing/{aec,aecm,ns}/include by Henrik Kjellander · 10 years ago
- e03cab9 When running this code in chromium on a machine with IPv6 disabled, the RTC_DCHECK fails and in release build, it could leak to further crash in chromium's rtc_peer_connection_hanlder.cc. by Guo-wei Shieh · 10 years ago
- ee2bac2 AcmReceiver::InsertPacket and NetEq::InsertPacket: Take ArrayView arguments by kwiberg · 10 years ago
- 91d9260 Add receive bitrate UMA stats. by stefan · 10 years ago
- 4dc9411 CodecManager::RegisterEncoder: Call SetFec on new encoder, not old by kwiberg · 10 years ago
- 718b6c7 Add waiting to SetSendSsrc tests. by Peter Boström · 10 years ago
- 4b56904 Fix race in VideoSendStreamTest.RtcpSenderReportContainsMediaBytesSent. by stefan · 10 years ago
- 00ac85e Update temporal up switch field for non-flexible mode according to updates in the RTP payload profile. by asapersson · 10 years ago
- f616a35 Roll chromium_revision 5a2ae99..4a38519 (359027:359080) by kjellander · 10 years ago
- fa566d6 Remove webrtc/examples/android/media_demo. by Peter Boström · 10 years ago
- cbfabbf Fix potential tearing issue in VideoRendererGui. by perkj · 10 years ago
- 9cb8982 Patchset 1 is a pure by perkj · 10 years ago
- b2d1c50 SurfaceViewRenderer: Add resource name to log outputs and exceptions by magjed · 10 years ago
- 1323fc3 Remove webrtc/test/channel_transport/include by Henrik Kjellander · 10 years ago
- 5237aaf Convert usage of ARRAY_SIZE to arraysize. by tfarina · 10 years ago
- e134a53 Roll chromium_revision 6f156f7..5a2ae99 (358880:359027) by kjellander · 10 years ago
- ad13d2f Round Rate computations from RateTracker. by Tim Psiaki · 10 years ago
- 9cafd97 Remove global list of SRTP sessions. by jbauch · 10 years ago
- 9af97f8 WebRTC should generate default private address even when adapter enumeration is disabled. by Guo-wei Shieh · 10 years ago
- 542059e Roll chromium_revision bff0bbb..6f156f7 (358822:358880) by kjellander · 10 years ago
- be57983 Rename Maybe to Optional by Karl Wiberg · 10 years ago
- 5376100 Add icu as a dependency on Android. by kjellander · 10 years ago
- 69a7fd5 Support VP9 HW video decoding on Android. by Alex Glaznev · 10 years ago
- ed8275a CodecManager: Eliminate the stereo_send_ member by kwiberg · 10 years ago
- a34bb2a Remove icu as a dependency by Henrik Kjellander · 10 years ago
- c94bd9b If a desktop captured window switches on/off it full screen mode, the capture may be unexpectedly terminated. During the transition of full screen mode on/off, the window can be temporarily invisible. by gyzhou · 10 years ago
- d153a37 Remove contention between RTCP packets and encoding. by Peter Boström · 10 years ago
- cfc319b Reland of Work on flexible mode and screen sharing. (patchset #1 id:1 of https://codereview.webrtc.org/1438543002/ ) by philipel · 10 years ago
- c95c366 Move the Rent-A-Codec™ from CodecOwner to CodecManager by kwiberg · 10 years ago
- cf3e13d Roll chromium_revision 95473df..bff0bbb (358772:358822) by kjellander · 10 years ago
- 0be8f1d Revert of Work on flexible mode and screen sharing. (patchset #28 id:520001 of https://codereview.webrtc.org/1328113004/ ) by terelius · 10 years ago
- 3ed3487 Remove field trial check for VP9. by asapersson · 10 years ago
- 327d8ba Add DecodedImageCallback::Decoded() function with custom decode time value. by Per · 10 years ago
- 805fc71 Let Rent-A-Codec™ create and own speech encoders by kwiberg · 10 years ago
- 3cea256 Reland "Prevent Opus DTX from generating intermittent noise during silence" by minyue · 10 years ago
- 626252f Adding minyue@ to some watch lists. by minyue · 10 years ago
- 77ccfb4 Work on flexible mode and screen sharing. by philipel · 10 years ago
- ce83ae1 Improve informative message in codereview.settings. by Henrik Kjellander · 10 years ago
- c12be39 -Removed the indirect error message reporting in aec and aecm. by peah · 10 years ago
- 952892a Fix a 64-bit pointer truncation bug found by VC++ 2015 by brucedawson · 10 years ago