- d6024e3 Roll chromium_revision 310ea93..8cf53d6 (349094:351112) by kjellander · 9 years ago
- 70ab1a1 Exposing RtpSenders and RtpReceivers from PeerConnection. by deadbeef · 9 years ago
- 8e9cb09 Android: Add unittests for SurfaceTextureHelper by Magnus Jedvert · 9 years ago
- 4fa648b Adding 20-second timeout to Java and Objective-C tests. by deadbeef · 9 years ago
- 8108764 Analyze support in gyp_webrtc by Henrik Kjellander · 9 years ago
- 2d56668 Unify Transport and newapi::Transport interfaces. by pbos · 9 years ago
- 8387c5f Remove AMR format parameter from AudioCoder in utility by henrik.lundin · 9 years ago
- 1968d3f Simplify VCMTimestampMap. by pbos · 9 years ago
- 8c404fa When doing DisableEquivalentPhases, exclude those AllocationSequences by honghaiz · 9 years ago
- 1f429e3 Passing the new policy from PeerConnection RTCConfiguration to by honghaiz · 9 years ago
- cb3649b Android VideoCapturer: Send ByteBuffer instead of byte[] by Magnus Jedvert · 9 years ago
- 4b808ee ACM: Remove unused and deprecated types by Henrik Lundin · 9 years ago
- 1bd0e03 ACM: Removing runtime APIs related to playout mode by henrik.lundin · 9 years ago
- d417523 Minor fix for debug logging on Android by henrika · 9 years ago
- 4fbd145 Fix suspend below min bitrate in new API by making it possible to set min bitrate at the receive-side. by stefan · 9 years ago
- d2413e5 Fix the C++ SurfaceTextureHolder by Per · 9 years ago
- 1ab271c Android SurfaceTextureHelper: Don't wait for pending frames in disconnect() by Magnus Jedvert · 9 years ago
- 3e9eb4b Add C++ SurfaceTextureHandler by Per · 9 years ago
- 82d6f2a ACM: Remove ACMVQMonCallback object by Henrik Lundin · 9 years ago
- 69984f0 Fixes logging levels in WebRtcAudioXXX.java classes by henrika · 9 years ago
- d6d27e7 Update isolate.gypi to support Swarming + move .isolate files by Henrik Kjellander · 9 years ago
- c97be6a Disable TestUdpReadyToSendIPv4 under MSan. by deadbeef · 9 years ago
- 4d47aa3 Fallback to system log when webrtc tracing not enabled. by Jiayang Liu · 9 years ago
- 1741770 Implement a high-QP threshold for Android H.264. by Peter Boström · 9 years ago
- a323fd6 Removes Nexus 6 from OpenSL ES blacklist. by henrika · 9 years ago
- 88799d9 RTCEAGLVideoView: Fix GL_FRAMEBUFFER_INCOMPLETE_ATTACHMENT error. by christoffer · 9 years ago
- 94c0105 Added peah@ to the watch lists by peah · 9 years ago
- 702f397 GN: Do not use forward_dependent_configs_from variable. by tfarina · 9 years ago
- 5c389d3 Split webrtc/video into webrtc/{audio,call,video}. by Peter Boström · 9 years ago
- 82e2055 Modifies invalid DCHECK in AudioRecordJni::OnCacheDirectBufferAddress() by henrika · 9 years ago
- 44d5d7b Autoroll: Update checkout before reading chromium_revision. by Henrik Kjellander · 9 years ago
- 495d2fd Reland of "Android GlRectDrawer: Add test for RGB rendering" by Magnus Jedvert · 9 years ago
- 3fd7be4 Revert of Don't link with audio codecs that we don't use (patchset #4 id:60001 of https://codereview.webrtc.org/1349393003/ ) by solenberg · 9 years ago
- a53e383 Revert of CodecOwner: Don't look at definitions for classes we don't link with (patchset #1 id:1 of https://codereview.webrtc.org/1364233002/ ) by solenberg · 9 years ago
- 67e0cf1 Android AppRTCDemo: Add slider for changing camera capture quality during call by Magnus Jedvert · 9 years ago
- 574d5da CodecOwner::SetEncoders: Return error code when given bad arguments by kwiberg · 9 years ago
- 6979b02 Adding stub files for RtpSender/RtpReceiver. by deadbeef · 9 years ago
- 4ba059d Remove custom handler since the logger already logs to console by default. by Jiayang Liu · 9 years ago
- 8937437 Do not prune if the current best connection is weak. by honghaiz · 9 years ago
- ea70d77 VideoCapturerAndroid: Add test for making calls on stopped camera by Magnus Jedvert · 9 years ago
- 59e72ab Enable logging for Mac by default on debug builds. by deadbeef · 9 years ago
- f4d38ea CodecOwner: Don't look at definitions for classes we don't link with by kwiberg · 9 years ago
- 34fbfff Remove VideoMediaChannel::SetRender(). by Peter Boström · 9 years ago
- 5e9a1bc Revert of Android GlRectDrawer: Add test for RGB rendering (patchset #3 id:40001 of https://codereview.webrtc.org/1367923002/ ) by magjed · 9 years ago
- a58ea78 1. Add receiving state as part of the connection sorting criteria. So if a connection's receiving state changes, it will re-select a better connection if there is any. by honghaiz · 9 years ago
- 8a88dd2 Stability improvement for audio recording on Android by henrika · 9 years ago
- 2bc68c7 Wire up QualityScaler for H.264 on Android. by Peter Boström · 9 years ago
- 7076729 Enable SurfaceViewRenderer for AppRTCDemo by Magnus Jedvert · 9 years ago
- 9236bb1 Minor fix for improving logging of supported platform effects by henrika · 9 years ago
- 6b8d355 Reland "Wire up send-side bandwidth estimation." by Erik Språng · 9 years ago
- 8c266e6 H264 bitstream parser. by Peter Boström · 9 years ago
- 6b20ad9 Android GlRectDrawer: Add test for RGB rendering by Magnus Jedvert · 9 years ago
- 2efe58b VideoCapturerAndroidTest: Dispose PeerConnectionFactory with pending frames by Magnus Jedvert · 9 years ago
- ec249d4 ACMCodecDB: Remove unused stuff, and move private stuff to anonymous namespace by kwiberg · 9 years ago
- 4a3ccad Remove SetAudioDelayOffset() and friends. by solenberg · 9 years ago
- f66a925 Don't link with audio codecs that we don't use by kwiberg · 9 years ago
- 61e933e Remove ChannelManager::GetCapabilities() by solenberg · 9 years ago
- c675ddd video_capture: Better support for UYVY by will.newton · 9 years ago
- 74d85e1 Reduce locking in overuse frame detector now that (as of r9508) the observer_ and options_ can only be set at construction time. E.g. no lock is any longer held while doing the callback. by asapersson · 9 years ago
- facbbec Remove use of DeviceManager from ChannelManager. by solenberg · 9 years ago
- 7603c76 Revert of Adding PeerConnectionInterface::SetConfiguration method. (patchset #4 id:60001 of https://codereview.webrtc.org/1317353005/ ) by deadbeef · 9 years ago
- 70702af Adding PeerConnectionInterface::SetConfiguration method. by deadbeef · 9 years ago
- 53eee43 Address the comment from 1367553002. by Guo-wei Shieh · 9 years ago
- 2e4b620 TcpPort doesn't connect when calling gmail with non-proxied UDP disabled. by Guo-wei Shieh · 9 years ago
- cdfe20b Fix the maximum native sample rate in AudioProcessing by Alejandro Luebs · 9 years ago
- cbecd35 Reland of TransportController refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/1358413003/ ) by deadbeef · 9 years ago
- d0b5b09 Add myself as OWNER of webrtc/voice_engine and talk/media/webrtc. by Fredrik Solenberg · 9 years ago
- 7cf0445 Remove ViEChannel::StartSend deadlock suppression. by Peter Boström · 9 years ago
- b7af7b0 Add myself to watchlist for a few subtrees of the repo. by Fredrik Solenberg · 9 years ago
- 8bffba7 Fix BWE bug where audio has timestamps in us. by Stefan Holmer · 9 years ago
- 6d92bf5 Returning correct duration estimate on Opus DTX packets. by minyuel · 9 years ago
- c14f5ff Improving support for Android Audio Effects in WebRTC. by henrika · 9 years ago
- c9bbeb0 Revert of Wire up send-side bandwidth estimation. (patchset #8 id:140001 of https://codereview.webrtc.org/1338203003/ ) by Erik Språng · 9 years ago
- d5c75b1 Reduce LS_INFO spam from voice_engine/. by Peter Boström · 9 years ago
- 7d17336 Remove the [Un]RegisterVoiceProcessor() API. by Fredrik Solenberg · 9 years ago
- 0967734 Remove VoEFile from VoeWrapper and the remaining places in libjingle where it was being used. by Fredrik Solenberg · 9 years ago
- f706c8a VideoCapturerAndroid: Fix threading issues by Magnus Jedvert · 9 years ago
- a81a42f Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ ) by torbjorng · 9 years ago
- 2d4e6c5 Fixing camera capture for video_loopback by ivica · 9 years ago
- 47ee2f3 TransportController refactoring. by deadbeef · 9 years ago
- 8967183 Simple cleanups of AudioDecoder and AudioEncoder classes by kwiberg · 9 years ago
- c1a1b35 Remove the SetLocalMonitor() API. by solenberg · 9 years ago
- 07d0936 Purge nss files and dependencies. by torbjorng · 9 years ago
- 7404368 Move AudioDecoderIsac* to its own files by Karl Wiberg · 9 years ago
- 7083e11 Remove callback_cs_ in ViEEncoder. by Peter Boström · 9 years ago
- 8212265 Android: Add class ThreadUtils with helper function joinUninterruptibly() by Magnus Jedvert · 9 years ago
- 6faf5be Move AudioDecoderPcm* next to AudioEncoderPcm* by kwiberg · 9 years ago
- d4818e7 Using static frame generator when no scrolling by ivica · 9 years ago
- 9b5476d sslidentity.cc/IntKeyTypeFamilyToKeyType function added, converting from int to KeyType. by Henrik Boström · 9 years ago
- ef165ee Wire up send-side bandwidth estimation. by sprang · 9 years ago
- 22011c1 Remove Channel::SetRingbackTone() and Channel::PlayRingbackTone(), and the code beneath it (within libjingle). by solenberg · 9 years ago
- 7317248 Rename CaptureThread to EncodingThread. by Peter Boström · 9 years ago
- ef5d5e4 Add field trial for automic resize in MediaCodecVideoEncoder. by asapersson · 9 years ago
- 1356ba5 Fixing target_bitrate_bps for a FullStackTest by ivica · 9 years ago
- 8f4f00f CQ: Update trybots by Henrik Kjellander · 9 years ago
- e4ba6ce Log the tag in native log stream. by Jiayang Liu · 9 years ago
- ebbf8a8 Make sure rtp_rtcp module doesn't directly reference anything in the pacer module, and remove build dependencies on it. by sprang · 9 years ago
- 04ac81f Replace readable with receiving where receiving means receiving anything (stun ping, response or data packet). by Peter Thatcher · 9 years ago
- 5bfc6cb Revert "Android: Enable C99 mode instead of C89 (default)." by Henrik Kjellander · 9 years ago
- 275a2f1 Revert of Replace readable with receiving where receiving means receiving anything (stun ping, response or da… (patchset #7 id:340001 of https://codereview.webrtc.org/1351673003/ ) by tommi · 9 years ago