Log - 5ab7b9395381147c7e022b7c5881b6de34ee1bf5 - fp2-dev/platform/external/chromium_org/third_party/webrtc

5ab7b93 Proper spacing for end-of-namespace comments. by pbos@webrtc.org · 11 years ago
a539d8e In call to Opus decoder: frame length too large by tina.legrand@webrtc.org · 11 years ago
817d63c Possible divide by 0 in ACM. by tina.legrand@webrtc.org · 11 years ago
3b7be22 Error in update of read index in ACM by tina.legrand@webrtc.org · 11 years ago
eaf7428 Rename unit_test.{cc,h} under module_unittest. by pbos@webrtc.org · 11 years ago
363852e Remove log of undefined input values in GetCodec. by pbos@webrtc.org · 11 years ago
787640d Diff NTP and internal once in VideoCaptureImpl. by pbos@webrtc.org · 11 years ago
f808b77 Build all java files into jar for each module on Android by fischman@webrtc.org · 11 years ago
5934a6a WebRTCViEDemo: Use global reference when passing variables across different threads by yujie.mao@webrtc.org · 11 years ago
cdfff5b Android opengles renderer: add thread sync to swap frame and draw native. by braveyao@webrtc.org · 11 years ago
c9ba795 Suppress excessive logging in video_coding by hclam@chromium.org · 11 years ago
0c9b40d 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
399baf7 Removes unused main function that is poluting the build. by henrike@webrtc.org · 11 years ago
040e75f 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
1d9d1ea Move TickTime::QueryOsForTicks out-of-line by fischman@webrtc.org · 11 years ago
f4a9648 Removes kStateFree and kStateDecoding, added a free_frames_ list which simplifies finding a free frame. by stefan@webrtc.org · 11 years ago
9f56b60 Fixed bad parameter passing in compare_videos.py by phoglund@webrtc.org · 11 years ago
4171693 Fix unnamed-type-template-args warnings on clang. by pbos@webrtc.org · 11 years ago
d32fe69 Correctly rebuild WebRTCDemo-debug.apk after modules/ source file changes. by fischman@webrtc.org · 11 years ago
9165c4d Adding a first simple version of overuse detection, but not hooked up. by mflodman@webrtc.org · 11 years ago
7401259 Removed ViE file API. by mflodman@webrtc.org · 11 years ago
f4ac411 Do basic parsing of RTCP headers in PcapFileReader to enable log filtering. by solenberg@webrtc.org · 11 years ago
281399a Remove unused multi stream bandwidth estimator. by solenberg@webrtc.org · 11 years ago
a1e06c2 Make sure padding packets are sent. by stefan@webrtc.org · 11 years ago
6cf9867 mac: Mark kCGLPFAFullScreen as allowed in a 10.6-only function. by sergeyu@chromium.org · 11 years ago
381c0a0 Fix memory bot failure by hclam@chromium.org · 11 years ago
19f7ac1 Enqueue packet in pacer if sending fails by hclam@chromium.org · 11 years ago
b211732 VCM: removing max jitter estimate by mikhal@webrtc.org · 11 years ago
f4e4324 Remove unneeded *_NOT_SUPPORTED from VoEAudioProcessing. by andrew@webrtc.org · 11 years ago
0d2e502 Fixes some pacer/padding issues found while testing. by stefan@webrtc.org · 11 years ago
350c135 Use 3 threads for higher than 720p resolutions by fbarchard@google.com · 11 years ago
6b7e468 Add a log message to see video delay break down by hclam@chromium.org · 11 years ago
037b232 Make ScreenCapturerMac work in versions of OSX before Lion. by sergeyu@chromium.org · 11 years ago
b06153b Enable ScreenCapturer unittests by sergeyu@chromium.org · 11 years ago
d85ab45 Use intptr_t to represent window IDs on all platforms. by sergeyu@chromium.org · 11 years ago
39f2547 Wire up pacer-based padding. by stefan@webrtc.org · 11 years ago
3777209 Revert r4145 "Revert 4127 "Switch frame list implementation to std::map."" by stefan@webrtc.org · 11 years ago
81751f0 Revert r4146 "Revert 4104 "Refactor jitter buffer to use separate lists for de..."" by stefan@webrtc.org · 11 years ago
2eb0c7a Fix AV sync issue by hclam@chromium.org · 11 years ago
eaeb84f Log current and target AV delay in ViESyncModule by hclam@chromium.org · 11 years ago
e797d9e Merge more tests into modules_{unit,integration}tests. by kjellander@webrtc.org · 11 years ago
d6f0906 WebRTCDemo: ensures that using front and back camera work as expected. by henrike@webrtc.org · 11 years ago
798d5c1 Fixes linker issue with no op trace. by henrike@webrtc.org · 11 years ago
92717b6 Risk of division by zero. by turaj@webrtc.org · 11 years ago
9fefc91 Revert 4211 "Build all java files into jar for each module on An..." by fischman@webrtc.org · 11 years ago
3b37c8a Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
936844c Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
74819dd Fix breakage due to test_fec conversion to gtest. by kjellander@webrtc.org · 11 years ago
86fb841 Convert test_fec to gtest by kjellander@webrtc.org · 11 years ago
af38b53 Disable ViEStandardIntegrationTest.RunsRtpRtcpTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
d6ac5c3 G722_1/G722_1C codecs won't instantiate by tina.legrand@webrtc.org · 11 years ago
1df6cc7 Reorganize test targets in WebRTC by kjellander@webrtc.org · 11 years ago
f92d9ad Build all java files into jar for each module on Android by fischman@webrtc.org · 11 years ago
3da595e Allow the screen capturer to capture oversized cursors and cursors without alpha channel (Windows). by alexeypa@chromium.org · 11 years ago
a83e538 Landing binary cursor image files to be used in a follow up CL. by alexeypa@chromium.org · 11 years ago
ccc21d2 Updated WebRTC version to 3.33 by elham@webrtc.org · 11 years ago
091c4f8 Making no NACK mode work again in VideoEngine. by mflodman@webrtc.org · 11 years ago
7556bbe RW lock access to ssrc maps in VideoCall. by pbos@webrtc.org · 11 years ago
a6e8ec3 Add back the WEBRTC_DIRECT_TRACE flag. by solenberg@webrtc.org · 11 years ago
ffce2b1 AudioDeviceAndroidOpenSLES: NULL variables might be referenced in StopPlayout() by braveyao@webrtc.org · 11 years ago
6e5b871 Revert some variables to uint32_t to fix compile errors on Mac gcc. by andrew@webrtc.org · 11 years ago
4477bd5 Allow audio devices with up to 64 channels on Mac. by andrew@webrtc.org · 11 years ago
b42bf4b Fixed Rtp/Rtcp tests by pwestin@webrtc.org · 11 years ago
9c49814 Fix relative path to .gitignore and other minor changes. by andrew@webrtc.org · 11 years ago
a052973 Removing functionality for inserting pre-encoded frames instead of raw by mflodman@webrtc.org · 11 years ago
306e331 Add script for appending entries to .gitignore. by andrew@webrtc.org · 11 years ago
4dca856 Fix size_t to int conversion error on Win64. by andrew@webrtc.org · 11 years ago
1632b97 Remove fake screen capturer because it's not used anywhere. by sergeyu@chromium.org · 11 years ago
f22cc80 Fix for STL vector function data not available. by pwestin@webrtc.org · 11 years ago
f94ddea Connect ACM with RTP module for audio NACK. by pwestin@webrtc.org · 11 years ago
a629133 Nack for audio. by turaj@webrtc.org · 11 years ago
a1e84f1 Fix leaks in DesktopRegion by sergeyu@chromium.org · 11 years ago
018870d Implement DetectNumberOfCores on Android and make it consistent on Linux and Android by fischman@webrtc.org · 11 years ago
4673a99 Wire up Nack for Voe by pwestin@webrtc.org · 11 years ago
767ca95 Fix init list for VideoSendStream::Config::Rtp. by pbos@webrtc.org · 11 years ago
2c343fc Stats+Config moved into VideoSend/ReceiveStreams. by pbos@webrtc.org · 11 years ago
561fe8b Merge webrtc_utility_unittests into modules_unittests. by kjellander@webrtc.org · 11 years ago
14035f1 Restore relative include paths to libyuv. by andrew@webrtc.org · 11 years ago
fd5d808 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
07e10ab 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
910a3c6 Move screen capturers from chromium to webrtc. by sergeyu@chromium.org · 11 years ago
71fe9ac Refactor padding and rtp header functionality. by stefan@webrtc.org · 11 years ago
4dc727b Update the remote bitrate estimator before passing the packet to the RTP module. by stefan@webrtc.org · 11 years ago
84119ff Remove XvRenderer. by pbos@webrtc.org · 11 years ago
5eb0408 Fix build error introduced with r4168. by stefan@webrtc.org · 11 years ago
39278fb Add support for padding in pacer. by stefan@webrtc.org · 11 years ago
1ad8620 Include files from webrtc/.. paths in common_video/ by pbos@webrtc.org · 11 years ago
7ddad3e Include files from webrtc/.. paths in tools/ by pbos@webrtc.org · 11 years ago
76070bf Disable neteq_unittests on Win x64 in code. by kjellander@webrtc.org · 11 years ago
aa28a43 Disable audio_decoder_unittests on Win x64 in code. by kjellander@webrtc.org · 11 years ago
9d86b92 Disable audio_coding_unittests on Win x64 in code. by kjellander@webrtc.org · 11 years ago
9d59eaf Do not hold a lock when calling VCMReceiveCallback::FrameToRender. by fischman@webrtc.org · 11 years ago
1bcdae5 Optimized DesktopRegion implementation. by sergeyu@chromium.org · 11 years ago
26c4698 Removed unused class members to enable clang=1 android build. by fischman@webrtc.org · 11 years ago
52f0bce Setting SSRC in vie_loopback_test by mikhal@webrtc.org · 11 years ago
dff82e7 Fix error in mixing test for supported sample rates. by andrew@webrtc.org · 11 years ago
5276226 Change SetRTPAudioLevelIndicationStatus to ignore the id in the case of disabling. by wu@webrtc.org · 11 years ago
8f515b1 Replace the old resampler with SincResampler in the voice engine signal path. by andrew@webrtc.org · 11 years ago
a654283 Remove ancient and unused CNG test. by andrew@webrtc.org · 11 years ago
ad09c1a Add dummy audio NACK APIs by niklas.enbom@webrtc.org · 11 years ago