- f97e5d7 Disable VideoCaptureExternalTest.FrameRate on Mac by henrik.lundin · 9 years ago
- e477a02 GYP: Add webrtc/pc/pc.gyp:* to 'All' target. by Henrik Kjellander · 9 years ago
- c70e4ec Packet buffer for the new jitter buffer. by philipel · 9 years ago
- 66a510a Get VideoCapturer stats via VideoTrackSourceInterface in StatsCollector, by nisse · 9 years ago
- a19e26e Remove deprecated RtpReceiver::CreateAudioReceiver() function. by solenberg · 9 years ago
- 2d13bc6 Give a more specific URL for creating WebRTC checkout by brucedawson · 9 years ago
- 65fad3e Delete unused cricket::VideoFrame methods MakeExclusive and CopyToFrame. by nisse · 9 years ago
- b600d48 Only split into bands when the reverse stream is analyzed in the APM by Alejandro Luebs · 9 years ago
- 91d31fe Improve iOS frame capture threading. by tkchin · 9 years ago
- d8df5e5 Use mobile platform settings for VP8 and VP9 decoders on all Android builds. by Alex Glaznev · 9 years ago
- 56d9dfc Add the last_sent_packet_id to the candidate pair change signal by Honghai Zhang · 9 years ago
- e47efce Minor ObjC header updates. by tkchin · 9 years ago
- 6bf669c Remove webrtc::ScopedVector by kwiberg · 9 years ago
- 3c4cb59 Change VideoSourceInterface::needs_denoising() to return rtc::Optional<bool> by Per · 9 years ago
- e1e16f0 Reland of move {media,p2p,pc,xmllite,xmpp}_tests.gypi files. (patchset #1 id:1 of https://codereview.webrtc.org/1846693002/ ) by kjellander · 9 years ago
- 83147bf This CL addresses late feedback on https://codereview.webrtc.org/1683193003/ by Torbjorn Granlund · 9 years ago
- 6e044aa Android EGL: Synchronize calls to eglSwapBuffers and eglMakeCurrent by Magnus Jedvert · 9 years ago
- 7c6a74f Add RTCConfiguration getter and setter methods. The immediate plan is to move some flags into an embedded MediaConfig (https://codereview.webrtc.org/1818033002/), which will be possible after Chrome is updated to use these new setter methods. by Niels Möller · 9 years ago
- 892e6df Android SurfaceTextureHelper: Distinguish thread names for decoder and camera by magjed · 9 years ago
- a7f1aa5 Limit max spatial layers to be configured through field trial (3->2) to match current limit in VP9EncoderImpl::InitEncode. by asapersson · 9 years ago
- 018a259 Add histogram stats for average QP per frame for VP8 (for sent video streams): by asapersson · 9 years ago
- d25cdbb - Add temporary VoEBase::audio_device_module() method. by solenberg · 9 years ago
- 9b791e4 MB: Remove unnecessary configs. by kjellander · 9 years ago
- 78a25a2 Changed the names of some of the bitexactness unittests to by peah · 9 years ago
- 5fb249f Changed tests to be DISABLED on non-supported platforms rather than not to build at all. by peah · 9 years ago
- 10e2675 [rtcp] Sdes::Parse updated not to use RTCPUtility by danilchap · 9 years ago
- c6ff47e Signal ready-to-send when switching to a writable connection. by Honghai Zhang · 9 years ago
- 1f0f771 Revert of Remove {media,p2p,pc,xmllite,xmpp}_tests.gypi files. (patchset #1 id:1 of https://codereview.webrtc.org/1839763004/ ) by kjellander · 9 years ago
- 463cc09 Remove {media,p2p,pc,xmllite,xmpp}_tests.gypi files. by kjellander · 9 years ago
- da65a68 Make rtcp sender use max transfer unit. by danilchap · 9 years ago
- 88ef094 [rtcp] ReceiverReport::Parse updated not to use RTCPUtility by danilchap · 9 years ago
- acf1581 Fix division by zero in NonlinearBeamformer by aluebs · 9 years ago
- 5ea80b1 Renamed the test::BitExactFrame method to test::VectorDifferenceBounded. by peah · 9 years ago
- d46864d Shorten single-stream VP8 HW implementation names. by Peter Boström · 9 years ago
- ca09a4b Split ByteBuffer into writer/reader objects. by jbauch · 9 years ago
- d0e6010 Rent-A-Codec: Reference count the shared iSAC bandwidth estimation state by kwiberg · 9 years ago
- 80eacc6 AudioCodingModule: Add methods for injecting external encoder stacks by kwiberg · 9 years ago
- 84bb523 Reland https://codereview.webrtc.org/1802993002/ by solenberg · 9 years ago
- 5ed7972 New method I420Buffer::Copy. by nisse · 9 years ago
- 82688e9 Reset the BWE when the network changes. by Honghai Zhang · 9 years ago
- 603a219 Adding JNI binding for 'active' field in RTP encodings. by Taylor Brandstetter · 9 years ago
- 67d4abe Remove unused stuff from AudioFrame: by solenberg · 9 years ago
- fda92c0 Added missing TODOs in the beamformer unit test code. by peah · 9 years ago
- 14d1e8c Reland: Add IntelligibilityEnhancer support to audioproc_float by Alejandro Luebs · 9 years ago
- 88a9699 Making new unit test assertions use the standard timeout. by Taylor Brandstetter · 9 years ago
- 8263d42 Revert "Add IntelligibilityEnhancer support to audioproc_float" by Alejandro Luebs · 9 years ago
- 374ac25 Add IntelligibilityEnhancer support to audioproc_float by Alejandro Luebs · 9 years ago
- 8e9df80 Update QuicTransportChannel to latest version of libquic by mikescarlett · 9 years ago
- 14d9fb1 Don't call operator== with scoped_ptr<T> and T* by kwiberg · 9 years ago
- 920b86a Remove calls to rtc::scoped_ptr::accept by kwiberg · 9 years ago
- ce55976 Reland of Added a bitexactness test for the gain controller in the audio processing module. by peah · 9 years ago
- b9c001f Remove accidentally readded webrtc/base/sslstreamadapterhelper.cc by Henrik Kjellander · 9 years ago
- 12cfa58 Remove all uses of the HAVE_CONFIG_H define. by Henrik Kjellander · 9 years ago
- 9d541b1 [rtcp] SenderReport::Parse updated not to use RTCPUtility by danilchap · 9 years ago
- 893fb1f Move to x509 v3 as required by the WebRTC draft. by torbjorng · 9 years ago
- 8d360b6 Remove orphaned files. by torbjorng · 9 years ago
- 91b510a Added a bitexactness test for the beamformer in the audio processing module by peah · 9 years ago
- e884401 Android HW decoder: Add support for textures when using EGL 1.0 by magjed · 9 years ago
- 908eeb9 Change include in metrics.h (change to use systems_wrappers/include/logging.h, base logging breaks chromium.fyi). by asapersson · 9 years ago
- 7274c0a Fix typo in FakeAdmTest.TestProcess name. by Peter Boström · 9 years ago
- 63b8d3f Add macros for ability to log samples that are added to histograms (RTC_LOGGED_*). by asapersson · 9 years ago
- d27854b Allow passing in strings of length zero to FileWrapper::Write without closing the file. by terelius · 9 years ago
- 7a29e63 Re-reland of Added a bitexactness test for the echo canceller in the audio processing module. by peah · 9 years ago
- 019caf7 Fix PeerConnectionInterfaceTest.CloseAndTestStreamsAndStates by Per · 9 years ago
- 7766e51 Added a bitexactness test for the intelligibility enhancer in the audio processing module by peah · 9 years ago
- 770073f Build dynamic framework with podspec for Objective-C API. by Jon Hjelle · 9 years ago
- 08ef89b Revert of Added a bitexactness test for the gain controller in the audio processing module. (patchset #3 id:60001 of https://codereview.webrtc.org/1812433002/ ) by peah · 9 years ago
- d79962e Added the JNI interface to get and set RtpParameters and the maximum bitrate limits. by skvlad · 9 years ago
- 5c481f7 Added a bitexactness test for the gain controller in the audio processing module. by peah · 9 years ago
- d4e1bb9 Adding BlockMeanCalculator for AEC. by minyue · 9 years ago
- c92ec0a Fixed a potential deadlock problem in the AGC by peah · 9 years ago
- b120b5b Avoid clicks when muting/unmuting a voe::Channel. by solenberg · 9 years ago
- 57abdbf Don't override curve preferences in BoringSSL. by David Benjamin · 9 years ago
- df760cd Revert of Added a bitexactness test for the echo canceller in the audio processing module. (patchset #2 id:40001 of https://codereview.webrtc.org/1827833006/ ) by guidou · 9 years ago
- f2defc4 ReAdd dummy MediaStreamTrack::set_state to make Chrome build happy. by perkj · 9 years ago
- 2bef95e Revert of Remove code interfacing legacy openssl. (patchset #3 id:40001 of https://codereview.webrtc.org/1808763002/ ) by Torbjorn Granlund · 9 years ago
- 6338d74 More cleanup of cricket::VideoCapturer by perkj · 9 years ago
- 3c013b3 Removed MediaStreamTrackInterface::set_state by perkj · 9 years ago
- 3716131 Reland of Added a bitexactness test for the echo canceller in the audio processing module. by peah · 9 years ago
- a91cd72 Move swap_queue.h to base/ by terelius · 9 years ago
- 5d41412 Delete default_send_ssrc_. by nisse · 9 years ago
- ddd067c Add 16-bit network id to the candidate signaling. by honghaiz · 9 years ago
- 299d6e3 Switch to using EGL 1.0 for rendering and HW codec. by Alex Glaznev · 9 years ago
- 18de68d Add visibility flag to GYP. by tkchin · 9 years ago
- 6a45d45 Remove WEBRTC_IOS from RTCPeerConnectionFactory public header. by tkchin · 9 years ago
- a600add Adding comments about threading around CreatePeerConnectionFactory. by Taylor Brandstetter · 9 years ago
- f2d8df8 [rtcp] Pli::Parse updated not to use RTCPUtility by danilchap · 9 years ago
- a2761a5 Delete class webrtc::VideoRenderer and its header file. by nisse · 9 years ago
- dc4cf77 Delete cricket::VideoRenderer. by nisse · 9 years ago
- bc105ab Add Mic Toggle button to AppRTCDemo (Android). by solenberg · 9 years ago
- 576b207 Delete VideoRendererInterface. by Niels Möller · 9 years ago
- 8dde01d Propagate MediaStreamSource state to video tracks the same way as audio. by perkj · 9 years ago
- 9a604fe Remove redefined macros from BitrateAdjuster by emircan · 9 years ago
- 432a768 Adding support for RTCRtpEncodingParameters.active flag. by deadbeef · 9 years ago
- ee1e4ed Make the audio channel communicate network state changes to the call. by skvlad · 9 years ago
- fa97919 Make Android min-resolution rotation-agnostic. by Peter Boström · 9 years ago
- 43ea052 cleanup RTCPSender by danilchap · 9 years ago
- a1399c7 Reland of Added a bitexactness test for the echo control mobile in the audio processing module by peah · 9 years ago
- 3241363 Refactor some ObjC API init methods. by Tze Kwang Chin · 9 years ago
- ecf7587 Use rtc::scoped_refptr for WebRtcVideoCapturer. by Peter Boström · 9 years ago
