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  • 6031001565 If the configured start bitrate is higher than the configures max bitrate, cap the star rate accordingly. mflodman@webrtc.org 2013-12-20 15:07:12 +00:00
  • 8dbca8d665 Race condition in ViECapturer::RegisterObserver sprang@webrtc.org 2013-12-20 11:36:03 +00:00
  • a463d73b99 Update WebRTC to version 3.48 tnakamura@webrtc.org 2013-12-19 22:38:38 +00:00
  • 54ae4ffb9e Add callbacks for receive channel RTCP statistics. sprang@webrtc.org 2013-12-19 13:26:02 +00:00
  • e682aa5077 Refactoring MediaOptimization so it can easily be turned into a thread-safe class. andresp@webrtc.org 2013-12-19 10:59:48 +00:00
  • faada6e604 Integrate fake_network_pipe into direct_transport. stefan@webrtc.org 2013-12-18 20:28:25 +00:00
  • 8f99a18119 Port scale and compare functions to pepper_33 and mips. BUG=none TEST=validator passes with new toolchain. R=andrew@webrtc.org, pbos@webrtc.org fbarchard@google.com 2013-12-18 19:51:37 +00:00
  • 5fe2d65c43 Remove metrics_unittests kjellander@webrtc.org 2013-12-18 13:27:37 +00:00
  • 8a54417968 Remove media_file from VideoEngine dependencies. pbos@webrtc.org 2013-12-18 10:00:29 +00:00
  • b429e516a9 cpplint cleaning new API and its implementation files. mflodman@webrtc.org 2013-12-18 09:46:22 +00:00
  • bcd124cdba Reduced execution time for CallTest::ReceivesPliAndRecovers, by dropping only one packet and made it predictable by removing rand(). mflodman@webrtc.org 2013-12-18 09:45:45 +00:00
  • 1fa41be66a Speeding up CallTest.ReceivesAndRetransmitsNack and removed the random packet loss. mflodman@webrtc.org 2013-12-18 09:44:53 +00:00
  • 8ae72560dd Make MouseCursor mutable sergeyu@chromium.org 2013-12-18 02:18:01 +00:00
  • f8be8df33a audio_processing_unittest: unbreak clang compilation. fischman@webrtc.org 2013-12-17 23:46:39 +00:00
  • 179908c81c JNI Audio: remove dead members. fischman@webrtc.org 2013-12-17 23:46:14 +00:00
  • e4c927208b Revert "Make MouseCursor mutable" sergeyu@chromium.org 2013-12-17 22:48:50 +00:00
  • 8fd1d26536 Make MouseCursor mutable sergeyu@chromium.org 2013-12-17 22:19:12 +00:00
  • af320fd2f7 The designated initializer method declaration in the Objective-C headers for RTCICEServer does't match its implementation. fischman@webrtc.org 2013-12-17 21:33:27 +00:00
  • 50f7b2da5d roll libyuv to r915 for webview jpeg build fix and NaCL pepper_33 initial support. BUG=none TEST=try bots R=andrew@webrtc.org, pbos@webrtc.org fbarchard@google.com 2013-12-17 18:18:17 +00:00
  • 052fa6243a Stop transport in test SuspendBelowMinBitrate. pbos@webrtc.org 2013-12-17 11:19:58 +00:00
  • e6b871bb29 Added method for getting default module state and protect agains a read/write race for child_modules_. mflodman@webrtc.org 2013-12-17 08:30:40 +00:00
  • 9df6674b26 Scale down by 4x with box filter. Fix for 1 pixel wide bilinear filter. Fix for I420ToARGB overread on V plane that causes valgrind fail. BUG=none TESTED=gcl try libyuv_r911 --bot=linux_valgrind R=andrew@webrtc.org fbarchard@google.com 2013-12-17 08:25:31 +00:00
  • eb7b7bce3d Modify video_render/ to allow a single old frame. pbos@webrtc.org 2013-12-16 18:24:37 +00:00
  • 5b3c67ef25 objc/README: Remove outdated advice about target_os. fischman@webrtc.org 2013-12-16 17:15:19 +00:00
  • 919f87fb36 Delete capturers after destroying streams in test. pbos@webrtc.org 2013-12-16 14:55:54 +00:00
  • e7b1e11283 Revert 5285 "Revert 5228 "Use the RTT from RtcpRttStats class if..." asapersson@webrtc.org 2013-12-16 14:40:36 +00:00
  • 1e7d61270c Simplification of histogram normalization in delay estimator. bjornv@webrtc.org 2013-12-16 13:37:28 +00:00
  • 5ab756703e Revert r5294 to re-roll r5293. pbos@webrtc.org 2013-12-16 12:24:44 +00:00
  • 5c64508b03 Adds robust validation functionality to the delay estimator bjornv@webrtc.org 2013-12-16 10:57:53 +00:00
  • 87ad57bc75 Incorrect iterator++ in ModuleRtpRtcpImpl::RegisterVideoBitrateObserver sprang@webrtc.org 2013-12-16 07:43:51 +00:00
  • 41e2615e02 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." turaj@webrtc.org 2013-12-15 18:42:32 +00:00
  • 341e91441a Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. solenberg@webrtc.org 2013-12-13 23:57:54 +00:00
  • e1bc6c8d8b Disabled tests on Android. The issue 2723 is filed to investigate the reason for tests failing. turaj@webrtc.org 2013-12-13 22:04:18 +00:00
  • dd393e7b9d Measure pacer queue size based on when packets are inserted rather than captured. stefan@webrtc.org 2013-12-13 22:03:27 +00:00
  • 167b6dfc73 Fix jitter buffer delay estimate. turaj@webrtc.org 2013-12-13 21:05:07 +00:00
  • 24301a67c6 Update talk to 58174641 together with http://review.webrtc.org/4319005/. wu@webrtc.org 2013-12-13 19:17:43 +00:00
  • 92c2793154 Adding REMB to receive stream configuration, the send side will always react to incoming REMB for now. mflodman@webrtc.org 2013-12-13 16:36:28 +00:00
  • 86bb56a7f5 Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..." asapersson@webrtc.org 2013-12-13 16:16:45 +00:00
  • 0a222eba69 Merge metrics_unittests into video_engine_tests. pbos@webrtc.org 2013-12-13 14:31:47 +00:00
  • 1d096901ac Move realtime tests to webrtc_perf_tests. pbos@webrtc.org 2013-12-13 12:48:05 +00:00
  • 62451dcba0 Update talk to 58157731. mallinath@webrtc.org 2013-12-13 12:29:34 +00:00
  • 6811b6e308 Callback for send bitrate estimates - new roll sprang@webrtc.org 2013-12-13 09:46:59 +00:00
  • f3973e81d5 Make sure channels in the same call are in the same channel group. mflodman@webrtc.org 2013-12-13 09:40:45 +00:00
  • e9abd591d7 Making RemoteRateControl::min_configured_bit_rate_ configurable henrik.lundin@webrtc.org 2013-12-13 08:42:42 +00:00
  • a9890800e0 Update talk to 58127566 together with https://webrtc-codereview.appspot.com/5309005/. wu@webrtc.org 2013-12-13 00:21:03 +00:00
  • a92baead39 ACM 2 compatibility with ACM 1. turaj@webrtc.org 2013-12-13 00:10:44 +00:00
  • 2018269dc3 Revert 5274 "Update talk to 58113193 together with https://webrt..." wu@webrtc.org 2013-12-12 22:54:25 +00:00
  • a129b6cd13 Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. wu@webrtc.org 2013-12-12 22:40:39 +00:00
  • 451745ec05 Complete rewrite of demo application. henrike@webrtc.org 2013-12-12 16:55:37 +00:00
  • 88ac63abc6 Remove overloaded CpuOveruseMeasure function. asapersson@webrtc.org 2013-12-12 14:37:33 +00:00
  • df7b1d6e39 AppRTCDemo(android): make ant be quiet on success and not overly noisy on failure. Also silence a 'cd' that would otherwise emit the path/to/talk. fischman@webrtc.org 2013-12-11 22:36:22 +00:00
  • 9ee75e9c77 Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency). henrike@webrtc.org 2013-12-11 21:42:44 +00:00
  • f41f06b916 PeerConnection(java): rationalize pointer-to-jlong conversion. fischman@webrtc.org 2013-12-11 21:07:18 +00:00
  • 9caf2765b2 Update talk to 58037405. wu@webrtc.org 2013-12-11 18:25:07 +00:00
  • 391b4db7de Fix common_video_unittests in apk_tests.gyp. pbos@webrtc.org 2013-12-11 16:48:53 +00:00
  • 724947b8ef Add SwapFrame() to VideoSendStreamInput. pbos@webrtc.org 2013-12-11 16:26:16 +00:00
  • 4c3faa9d73 Disable a libjingle unittest which is failing after a chromium roll out. turaj@webrtc.org 2013-12-11 16:12:31 +00:00
  • df02283279 Adds audio volume demo to the index page. hta@webrtc.org 2013-12-11 14:44:10 +00:00
  • 59d5705385 Fix memory tools error introduced in roll @ r5260 kjellander@webrtc.org 2013-12-11 14:16:53 +00:00
  • 096e8d9f94 Revert 5259 "Callback for send bitrate estimates" sprang@webrtc.org 2013-12-11 14:07:33 +00:00
  • f9bdbe3619 Roll chromium_revision 232627:238260 kjellander@webrtc.org 2013-12-11 13:37:12 +00:00
  • 2656cf9f4c Callback for send bitrate estimates sprang@webrtc.org 2013-12-11 12:53:03 +00:00
  • 26c40ba166 Removed audio element from volume measuring demo. hta@webrtc.org 2013-12-11 11:12:39 +00:00
  • 1133ffda4b Merged OWNERS of JS demo directories hta@webrtc.org 2013-12-11 08:51:56 +00:00
  • c4038d795d Rewriting the SoundMeter class to be RMS and be encapsulated differently hta@webrtc.org 2013-12-11 08:36:16 +00:00
  • 77507eff4f Correctly define OVERRIDE when building with g++ 4.7 and C++11 support andrew@webrtc.org 2013-12-11 00:07:11 +00:00
  • 7ae8495779 Removed unnecessary Pulse init from VoE startup. Saves 10% (~260ms) of the total PeerConnectionTest wallclock time. fischman@webrtc.org 2013-12-10 21:01:34 +00:00
  • 762fcdcca9 Correctly define OVERRIDE when building with g++ 4.7 and C++11 support andrew@webrtc.org 2013-12-10 19:20:46 +00:00
  • 8b8819262f Improve VideoSendStreamTest::MaxPacketSize sprang@webrtc.org 2013-12-10 10:05:17 +00:00
  • 917306d3fd Change uses of the obsolete armv7 setting to arm_version==7. kjellander@webrtc.org 2013-12-10 09:26:07 +00:00
  • eb7def234e Fix compilation errors on Fedora 20. fischman@webrtc.org 2013-12-09 21:34:30 +00:00
  • c329529047 Apply transaction to setting connected to Room entities, to resolve a possible race condition at two clients connecting simultaneously. braveyao@webrtc.org 2013-12-09 19:37:45 +00:00
  • 70ddf9355f libyuv r905 with yuv off by 1 fix for valgrind overread BUG=none TEST=valgrind build bots R=andrew@webrtc.org, mflodman@webrtc.org fbarchard@google.com 2013-12-09 18:17:42 +00:00
  • de7c9e8884 Ensure WEBRTC_MODULE_UTILITY_VIDEO is undefined for enable_video==0. andrew@webrtc.org 2013-12-09 16:23:00 +00:00
  • 5e13ac967b Add shape in DesktopFrame. sergeyu@chromium.org 2013-12-07 01:03:28 +00:00
  • 4acf4507b8 libyuv roll to r888 with valgrind overread fixes. BUG=none TEST=try bots R=andrew@webrtc.org, mflodman@webrtc.org fbarchard@google.com 2013-12-06 18:14:11 +00:00
  • 8d0ca7f5d2 Add new method to MockAudioProcessing. andrew@webrtc.org 2013-12-06 17:52:27 +00:00
  • 797522f9f2 Revert 5229 "Make VideoSendStreamTest::MaxPacketSize test a whol..." andrew@webrtc.org 2013-12-06 17:42:32 +00:00
  • 863b536100 Allow opening an AEC dump from an existing file handle. henrikg@webrtc.org 2013-12-06 16:05:17 +00:00
  • 0f3d0bb601 Stop video capturers in multi-stream test. pbos@webrtc.org 2013-12-06 15:48:17 +00:00
  • 758db4baea Demo showing how to measure volume using WebAudio hta@webrtc.org 2013-12-06 14:47:34 +00:00
  • 88615f0659 Fix use of uninitialized memory in RtpSenderTest::StreamDataCountersCallbacks sprang@webrtc.org 2013-12-06 13:16:44 +00:00
  • 7f73280ded Fraction lost statistics not being reported sprang@webrtc.org 2013-12-06 11:56:55 +00:00
  • 32f485b16a Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc. sergeyu@chromium.org 2013-12-05 22:36:21 +00:00
  • 57a5f64264 revert r5230 sergeyu@chromium.org 2013-12-05 22:14:46 +00:00
  • a1b21cd777 Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc. sergeyu@chromium.org 2013-12-05 21:28:34 +00:00
  • 7104fc1906 Make VideoSendStreamTest::MaxPacketSize test a whole range of frame sizes, to make sure all corner cases are covered. sprang@webrtc.org 2013-12-05 16:15:11 +00:00
  • 96a9b2dcdc Use the RTT from RtcpRttStats class if provided when sending/receiving NACK. asapersson@webrtc.org 2013-12-05 15:06:56 +00:00
  • ebad765ee0 Add callbacks for send channel rtp statistics sprang@webrtc.org 2013-12-05 14:29:02 +00:00
  • 5cea89f3e1 Remove CallTest dependency on voice_engine/test/. pbos@webrtc.org 2013-12-05 14:24:17 +00:00
  • 0a3c1471b8 Add API to query video engine for the send-side delay. stefan@webrtc.org 2013-12-05 14:05:07 +00:00
  • 07fcc4f2fa Fixing the android build henrik.lundin@webrtc.org 2013-12-05 13:24:25 +00:00
  • c49d5b7df8 Move implementation files out of the webrtc/ root. pbos@webrtc.org 2013-12-05 12:11:47 +00:00
  • 245037df09 Remove default implementations for SuspendBelowMinBitrate henrik.lundin@webrtc.org 2013-12-05 12:01:45 +00:00
  • b88fc18aba Fix bug where fraction_lost is always set to 0 when getting received RTCP statistics. stefan@webrtc.org 2013-12-05 11:36:46 +00:00
  • a6ad6e5b58 Add callbacks for send channel rtcp statistics sprang@webrtc.org 2013-12-05 09:48:44 +00:00
  • c4726d06fa Make RTPSender::SendPadData public. stefan@webrtc.org 2013-12-05 09:16:33 +00:00
  • 5bc25c41fc Update libjingle to 57692857 sergeyu@chromium.org 2013-12-05 00:24:06 +00:00
  • 3d9981d58a Remove unused ThreadData struct. andrew@webrtc.org 2013-12-04 17:13:47 +00:00
  • 3054ba6bb2 Remove the long disabled WEBRTC_SVNREVISION define. andrew@webrtc.org 2013-12-04 17:00:44 +00:00