Log - 1bd3140abdb08564cbe9bf55046aac430d93a9c8 - fp2-dev/platform/external/chromium_org/third_party/webrtc

1bd3140 Possible divide by 0 in ACM. by tina.legrand@webrtc.org · 11 years ago
cbb535a Error in update of read index in ACM by tina.legrand@webrtc.org · 11 years ago
98ac1e8 Rename unit_test.{cc,h} under module_unittest. by pbos@webrtc.org · 11 years ago
438be80 Remove log of undefined input values in GetCodec. by pbos@webrtc.org · 11 years ago
5cf0fd1 Diff NTP and internal once in VideoCaptureImpl. by pbos@webrtc.org · 11 years ago
50c1aef Build all java files into jar for each module on Android by fischman@webrtc.org · 11 years ago
5e742a8 WebRTCViEDemo: Use global reference when passing variables across different threads by yujie.mao@webrtc.org · 11 years ago
e63c003 Android opengles renderer: add thread sync to swap frame and draw native. by braveyao@webrtc.org · 11 years ago
af60a80 Suppress excessive logging in video_coding by hclam@chromium.org · 11 years ago
8a5cb95 Moves tools/update.py to trunk/webrtc/tools and updates it so that it no longer pulls any information from the DEPS file. by henrike@webrtc.org · 11 years ago
c1624d5 Removes unused main function that is poluting the build. by henrike@webrtc.org · 11 years ago
c7eab28 Re-add WebRTCDemo dependencies as dependencies (not just inputs) because they also need to be built for this target! by fischman@webrtc.org · 11 years ago
d305e11 Move TickTime::QueryOsForTicks out-of-line by fischman@webrtc.org · 11 years ago
8148118 Removes kStateFree and kStateDecoding, added a free_frames_ list which simplifies finding a free frame. by stefan@webrtc.org · 11 years ago
c10fc53 Fixed bad parameter passing in compare_videos.py by phoglund@webrtc.org · 11 years ago
3656192 Fix unnamed-type-template-args warnings on clang. by pbos@webrtc.org · 11 years ago
555f1cd Correctly rebuild WebRTCDemo-debug.apk after modules/ source file changes. by fischman@webrtc.org · 11 years ago
f8f91d6 Adding a first simple version of overuse detection, but not hooked up. by mflodman@webrtc.org · 11 years ago
0291c80 Removed ViE file API. by mflodman@webrtc.org · 11 years ago
0be9202 Do basic parsing of RTCP headers in PcapFileReader to enable log filtering. by solenberg@webrtc.org · 11 years ago
20cfda6 Remove unused multi stream bandwidth estimator. by solenberg@webrtc.org · 11 years ago
8ee45da Make sure padding packets are sent. by stefan@webrtc.org · 11 years ago
c8c333d mac: Mark kCGLPFAFullScreen as allowed in a 10.6-only function. by sergeyu@chromium.org · 11 years ago
0d35c78 Fix memory bot failure by hclam@chromium.org · 11 years ago
0f6f7cb Enqueue packet in pacer if sending fails by hclam@chromium.org · 11 years ago
44050b2 VCM: removing max jitter estimate by mikhal@webrtc.org · 11 years ago
f47d0f8 Remove unneeded *_NOT_SUPPORTED from VoEAudioProcessing. by andrew@webrtc.org · 11 years ago
7533659 Fixes some pacer/padding issues found while testing. by stefan@webrtc.org · 11 years ago
c9cb798 Use 3 threads for higher than 720p resolutions by fbarchard@google.com · 11 years ago
5b9adb0 Add a log message to see video delay break down by hclam@chromium.org · 11 years ago
b82ee51 Make ScreenCapturerMac work in versions of OSX before Lion. by sergeyu@chromium.org · 11 years ago
9ca71b1 Enable ScreenCapturer unittests by sergeyu@chromium.org · 11 years ago
ba458e2 Use intptr_t to represent window IDs on all platforms. by sergeyu@chromium.org · 11 years ago
69f7605 Wire up pacer-based padding. by stefan@webrtc.org · 11 years ago
67ca2b4 Revert r4145 "Revert 4127 "Switch frame list implementation to std::map."" by stefan@webrtc.org · 11 years ago
376ae3e Revert r4146 "Revert 4104 "Refactor jitter buffer to use separate lists for de..."" by stefan@webrtc.org · 11 years ago
b06dd93 Fix AV sync issue by hclam@chromium.org · 11 years ago
9540e2a Log current and target AV delay in ViESyncModule by hclam@chromium.org · 11 years ago
8dbc8ab Merge more tests into modules_{unit,integration}tests. by kjellander@webrtc.org · 11 years ago
6828566 WebRTCDemo: ensures that using front and back camera work as expected. by henrike@webrtc.org · 11 years ago
19b339a Fixes linker issue with no op trace. by henrike@webrtc.org · 11 years ago
7b2c430 Risk of division by zero. by turaj@webrtc.org · 11 years ago
cefb004 Revert 4211 "Build all java files into jar for each module on An..." by fischman@webrtc.org · 11 years ago
8ed5369 Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
83163e0 Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
789e98b Fix breakage due to test_fec conversion to gtest. by kjellander@webrtc.org · 11 years ago
f04f54a Convert test_fec to gtest by kjellander@webrtc.org · 11 years ago
f09f7b2 Disable ViEStandardIntegrationTest.RunsRtpRtcpTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
6ab2b1f G722_1/G722_1C codecs won't instantiate by tina.legrand@webrtc.org · 11 years ago
46cec2a Reorganize test targets in WebRTC by kjellander@webrtc.org · 11 years ago
0604490 Build all java files into jar for each module on Android by fischman@webrtc.org · 11 years ago
e8c9ecd Allow the screen capturer to capture oversized cursors and cursors without alpha channel (Windows). by alexeypa@chromium.org · 11 years ago
a463a4b Landing binary cursor image files to be used in a follow up CL. by alexeypa@chromium.org · 11 years ago
f1bcae0 Updated WebRTC version to 3.33 by elham@webrtc.org · 11 years ago
b4c89a4 Making no NACK mode work again in VideoEngine. by mflodman@webrtc.org · 11 years ago
63988b2 RW lock access to ssrc maps in VideoCall. by pbos@webrtc.org · 11 years ago
203d656 Add back the WEBRTC_DIRECT_TRACE flag. by solenberg@webrtc.org · 11 years ago
0a86a9f AudioDeviceAndroidOpenSLES: NULL variables might be referenced in StopPlayout() by braveyao@webrtc.org · 11 years ago
1125d89 Revert some variables to uint32_t to fix compile errors on Mac gcc. by andrew@webrtc.org · 11 years ago
64e2651 Allow audio devices with up to 64 channels on Mac. by andrew@webrtc.org · 11 years ago
0e7cd85 Fixed Rtp/Rtcp tests by pwestin@webrtc.org · 11 years ago
cfe2a74 Fix relative path to .gitignore and other minor changes. by andrew@webrtc.org · 11 years ago
a18c6e5 Removing functionality for inserting pre-encoded frames instead of raw by mflodman@webrtc.org · 11 years ago
029c3f4 Add script for appending entries to .gitignore. by andrew@webrtc.org · 11 years ago
9aeef32 Fix size_t to int conversion error on Win64. by andrew@webrtc.org · 11 years ago
a653113 Remove fake screen capturer because it's not used anywhere. by sergeyu@chromium.org · 11 years ago
5f545ff Fix for STL vector function data not available. by pwestin@webrtc.org · 11 years ago
4aa9f1a Connect ACM with RTP module for audio NACK. by pwestin@webrtc.org · 11 years ago
d0631e3 Nack for audio. by turaj@webrtc.org · 11 years ago
31fad0b Fix leaks in DesktopRegion by sergeyu@chromium.org · 11 years ago
69bab25 Implement DetectNumberOfCores on Android and make it consistent on Linux and Android by fischman@webrtc.org · 11 years ago
b8171ff Wire up Nack for Voe by pwestin@webrtc.org · 11 years ago
6fb2ca3 Fix init list for VideoSendStream::Config::Rtp. by pbos@webrtc.org · 11 years ago
6f1c3ef Stats+Config moved into VideoSend/ReceiveStreams. by pbos@webrtc.org · 11 years ago
53304e8 Merge webrtc_utility_unittests into modules_unittests. by kjellander@webrtc.org · 11 years ago
266fc69 Restore relative include paths to libyuv. by andrew@webrtc.org · 11 years ago
9b82368 Issue 1847, memcopy is wrong and unnecessary, it is sufficient to store the pointer before clearing the instance, and write back the pointer. by turaj@webrtc.org · 11 years ago
ae05178 resolve b9050210. Avoid pushing sync packet before any packet received. Do not turn on AV-sync if initial delay is zero. by turaj@webrtc.org · 11 years ago
6c82a7e Move screen capturers from chromium to webrtc. by sergeyu@chromium.org · 11 years ago
12bce3b Refactor padding and rtp header functionality. by stefan@webrtc.org · 11 years ago
d8ecee5 Update the remote bitrate estimator before passing the packet to the RTP module. by stefan@webrtc.org · 11 years ago
e54928f Remove XvRenderer. by pbos@webrtc.org · 11 years ago
751253d Fix build error introduced with r4168. by stefan@webrtc.org · 11 years ago
695ff2a Add support for padding in pacer. by stefan@webrtc.org · 11 years ago
026d1ce Include files from webrtc/.. paths in common_video/ by pbos@webrtc.org · 11 years ago
cff5c03 Include files from webrtc/.. paths in tools/ by pbos@webrtc.org · 11 years ago
1cc4ed7 Disable neteq_unittests on Win x64 in code. by kjellander@webrtc.org · 11 years ago
ec5caf3 Disable audio_decoder_unittests on Win x64 in code. by kjellander@webrtc.org · 11 years ago
f4fc8ba Disable audio_coding_unittests on Win x64 in code. by kjellander@webrtc.org · 11 years ago
3b35ec6 Do not hold a lock when calling VCMReceiveCallback::FrameToRender. by fischman@webrtc.org · 11 years ago
aec1bc8 Optimized DesktopRegion implementation. by sergeyu@chromium.org · 11 years ago
ad9ee0d Removed unused class members to enable clang=1 android build. by fischman@webrtc.org · 11 years ago
0016110 Setting SSRC in vie_loopback_test by mikhal@webrtc.org · 11 years ago
915ca75 Fix error in mixing test for supported sample rates. by andrew@webrtc.org · 11 years ago
a80d94b Change SetRTPAudioLevelIndicationStatus to ignore the id in the case of disabling. by wu@webrtc.org · 11 years ago
92bfbbd Replace the old resampler with SincResampler in the voice engine signal path. by andrew@webrtc.org · 11 years ago
379dce7 Remove ancient and unused CNG test. by andrew@webrtc.org · 11 years ago
2753b76 Add dummy audio NACK APIs by niklas.enbom@webrtc.org · 11 years ago
40954f0 Prevent excessive logging in jitter buffer by hclam@chromium.org · 11 years ago
8bf7456 Revert 4104 "Refactor jitter buffer to use separate lists for de..." by tnakamura@webrtc.org · 11 years ago