Log - 6241beefa225f16b7a3f5ec52089d307f8631636 - platform/external/webrtc

6241bee Refer to Chrome's DEPS to make rolling easier. by andrew@webrtc.org · 13 years ago
946601e Change default packetization mode to an equal size mode. by marpan@webrtc.org · 13 years ago
70efc32 Factory method for the ADM in the interface file. by henrike@webrtc.org · 13 years ago
dc3179d Implemented quick builds (e.g. only do full clean if the previous build failed). by phoglund@webrtc.org · 13 years ago
6eb0ca2 Two problems are fixed: by xians@webrtc.org · 13 years ago
a556b0d Reverting r1749. by mflodman@webrtc.org · 13 years ago
cb57f9b Updated libyuv revision to include mjpg and added mjpg to type conversion. by mflodman@webrtc.org · 13 years ago
4f9e44f Prepared for MJPG capturing on Linux. MJPG is conversion is not available in libyuv yet, so this CL is only made as preparation. by mflodman@webrtc.org · 13 years ago
682cd4e Add android target Review URL: https://webrtc-codereview.appspot.com/396016 by leozwang@webrtc.org · 13 years ago
4ad4c24 Add android to audio device module by leozwang@webrtc.org · 13 years ago
c2e9cd3 Renaming windows slaves to follow naming convention better. by kjellander@webrtc.org · 13 years ago
f25ab5d Enabling metrics_unittests that was created in http://webrtc-codereview.appspot.com/333025/ by kjellander@webrtc.org · 13 years ago
0fe2171 Relax libyuv test threshold and upgrade to libyuv r182. by stefan@webrtc.org · 13 years ago
539ef94 Remove the deprecated kTraceModuleCall trace from audio coding module. by xians@webrtc.org · 13 years ago
20e9cf2 Add android to video capture module by leozwang@webrtc.org · 13 years ago
1181b31 Pull chromium version of libjingle and webrtc and build peerconnection sample server and client. by wu@webrtc.org · 13 years ago
29fafef Fix building errors Review URL: https://webrtc-codereview.appspot.com/399012 by leozwang@webrtc.org · 13 years ago
51198f1 More PRESUBMIT checks. by kjellander@webrtc.org · 13 years ago
0d757b8 Fixing coverity issues in capture module. by mallinath@webrtc.org · 13 years ago
b9432ce Added simple wrapping for waterfall display. by kjellander@webrtc.org · 13 years ago
7cb0c24 Trying to free up hellner from review work, since he mainly works in libJingle. by niklas.enbom@webrtc.org · 13 years ago
8435e8e Remove the deprecated kTraceModuleCall trace from audio processing module. by xians@webrtc.org · 13 years ago
4e20a09 Making buildbot output more compact. by kjellander@webrtc.org · 13 years ago
3f6bf49 Fixed flunk settings: the builds show now halt only when compile and sync-kind operations fail. by phoglund@webrtc.org · 13 years ago
a475556 Assume 200 ms RTT if we're only receiving. by stefan@webrtc.org · 13 years ago
20aabbb Remove the deprecated kTraceModuleCall trace from audio device module. by xians@webrtc.org · 13 years ago
9a798d3 Remove the deprecated kTraceModuleCall trace from video processing module. by xians@webrtc.org · 13 years ago
ca98118 Upgrade libvpx to Duclair. by stefan@webrtc.org · 13 years ago
b45ceed Rewrote the call report test. by phoglund@webrtc.org · 13 years ago
843c8c7 Remove the deprecated kTraceModuleCall trace from video modules. by xians@webrtc.org · 13 years ago
6bde7a8 Remove the deprecated kTraceModuleCall trace from utility module. by xians@webrtc.org · 13 years ago
57fb09a Remove the deprecated kTraceModuleCall trace from udp transport module. by xians@webrtc.org · 13 years ago
03039d5 Remove the deprecated kTraceModuleCall trace from media file module. by xians@webrtc.org · 13 years ago
56cfe80 Remove the deprecated kTraceModuleCall trace from conference mixer. by xians@webrtc.org · 13 years ago
145f04f Changing Celt to use stereo as default. by tina.legrand@webrtc.org · 13 years ago
bd5648f Reverting 1718: failed linux video test. by marpan@webrtc.org · 13 years ago
883e716 Removed unused motion vector metrics from VideoContentMetrics; by marpan@webrtc.org · 13 years ago
f3760dc Fixes coverity warning that I missed in system wrappers. by henrike@webrtc.org · 13 years ago
b317286 Added a retry mechanism to vie_auto_test's verifying tests to make them less flaky. by phoglund@webrtc.org · 13 years ago
4cb0601 Disabled RTPModule VP8 packetizer assert. by mflodman@webrtc.org · 13 years ago
8bfee84 Initial revision of a ViE fuzz test. The idea is to inject randomized RTP packets and see what the video engine does. by phoglund@webrtc.org · 13 years ago
a52838b Update Android.mk and add test app by leozwang@webrtc.org · 13 years ago
79e29e5 Adding option to change bitrate for Celt. by tina.legrand@webrtc.org · 13 years ago
133d1a1 Add a new folder so that we can pull webrtc and libjingle together and build peerconnection sample client and server. by wu@webrtc.org · 13 years ago
ee62835 Updating the object-c++ file after change in the API by mallinath@webrtc.org · 13 years ago
8b4a98d Change in the interface file for GetBestMatchedCapability method. Updating mac files. by mallinath@webrtc.org · 13 years ago
69f8be3 Change the ExternalRenderer to provide both rtp timestamp and the render time. by wu@webrtc.org · 13 years ago
12984f0 Fixing Coverity issues by mallinath@webrtc.org · 13 years ago
3ab6dda Truncated the volume to 255 when the users set the volume above 100%. by xians@webrtc.org · 13 years ago
f7b6078 Allow multiple send channels for REMB. Current implementation splits the remote estimate evenly between all senders. by mflodman@webrtc.org · 13 years ago
439be29 Add APIs for getting receive-side estimated bandwidth and codec target rate. by stefan@webrtc.org · 13 years ago
6f5f9ff Moved coverage directory since we have to symlink the coverage folder from /var/www. by phoglund@webrtc.org · 13 years ago
590e5eb Convert audio layer to WAV on Vista RTM(without any Service Pack) by braveyao@webrtc.org · 13 years ago
d6d014f Fixes memory leaks introduced in 1698. by henrike@webrtc.org · 13 years ago
57193f0 Use http rather than https in DEPS. by andrew@webrtc.org · 13 years ago
cb33353 Remove common_settings.gypi. by andrew@webrtc.org · 13 years ago
f5da4da Removes a global non POD instance from the RTP_RTCP module that was introduced in https://code.google.com/p/webrtc/source/detail?r=1076. by henrike@webrtc.org · 13 years ago
0a272eb Disable SetAffinity on android by leozwang@webrtc.org · 13 years ago
05e0601 Fixes coverity warnings in the udp_transport module. by henrike@webrtc.org · 13 years ago
6b9253e Fixe issues reported by Coverity for modules/utility. by henrike@webrtc.org · 13 years ago
735478a Fixed bad parameter to android. by phoglund@webrtc.org · 13 years ago
f148b9e - Moved methods where they should be. by phoglund@webrtc.org · 13 years ago
cd46385 Fixing Android.mk for jpeg library by kjellander@webrtc.org · 13 years ago
0a57aae Converted old jpeg_test tool to gtest unit test. by kjellander@webrtc.org · 13 years ago
38a0d28 Enabling jpeg_unittests in buildbots. by kjellander@webrtc.org · 13 years ago
8bd6f19 Disable flaky CpuTest.Usage on Windows. by andrew@webrtc.org · 13 years ago
b38a66a Fixes a coverity warning in the mixer module. by henrike@webrtc.org · 13 years ago
1322614 Added a hack, python script for removing the logging of a trace depending on a keyword. Current implementation requires the keyword to be typed in the file; just change the 'trace_remove_key_word' from kTraceModuleCall to whatever. Hack should be fine since it works and it's a tool that has only limited utility. by henrike@webrtc.org · 13 years ago
79a99de Reverting 1680: valgrind memory leak reported. by marpan@webrtc.org · 13 years ago
738bcdc Fix to coverity issue 10339. by marpan@webrtc.org · 13 years ago
737c023 Properly disable sse2 source on non-x86. by andrew@webrtc.org · 13 years ago
59d6cec Fix the crash at playing 48kHz stereo wav file. by braveyao@webrtc.org · 13 years ago
4e34dcb Allow for spatial-downsampling without reinitializaing encoder. Change of frame size will automatically trigger new key frame in codec. This feature is set off in vie_encoder until we upgrade to the new libvpx. by marpan@webrtc.org · 13 years ago
124e563 Adding video_render_module_test to LinuxVideo slave. by kjellander@webrtc.org · 13 years ago
d7d4688 Update receive only channels with RTT. by mflodman@webrtc.org · 13 years ago
e8b1a0f Fixed incorrect default argument. by phoglund@webrtc.org · 13 years ago
6adfad1 Optimized coverage computations and fixed coverage on the integration bot. by phoglund@webrtc.org · 13 years ago
c76c096 Bugfix issue 273, workaround for compiler issue. by pwestin@webrtc.org · 13 years ago
52fd98d Removing encoder reset. Function did not make sence. by pwestin@webrtc.org · 13 years ago
567d507 Fixes a bug when number of media packets in a frame is larger than maximum allowed for the generateFEC. by marpan@webrtc.org · 13 years ago
292da24 New attempt. by phoglund@webrtc.org · 13 years ago
dbe1e13 Fixed compilation error on Windows. by phoglund@webrtc.org · 13 years ago
8224e19 Fixed incorrect packet loss reported to encoder. by mflodman@webrtc.org · 13 years ago
6b3bb89 Rewrote file test. by phoglund@webrtc.org · 13 years ago
5e95481 Clanup handling of key frame requests and FIR. by pwestin@webrtc.org · 13 years ago
486a3ba Enabling video_codecs_test_framework_integrationtests. by kjellander@webrtc.org · 13 years ago
a1e9e3f Enabling rtp_rtcp_unittests since issue 268 is now fixed and commited. by kjellander@webrtc.org · 13 years ago
caef503 Removing PeerConnection sample client and libjingle from webrtc. by wu@webrtc.org · 13 years ago
75f1948 Restore AECM Coverity fix. by andrew@webrtc.org · 13 years ago
aaa76f3 Rewrote network test. by phoglund@webrtc.org · 13 years ago
bf03338 Enable lcov on LinuxVideo and disable on others except Linux32DBG. by kjellander@webrtc.org · 13 years ago
4b37741 Fix release build errors. by stefan@webrtc.org · 13 years ago
3dbed85 This CL makes the playout delay value thread safe. by xians@webrtc.org · 13 years ago
9c84b0d Fix build errors with GCC. by stefan@webrtc.org · 13 years ago
7adab09 This removes the knowledge of frame completeness from the FEC decoder. by stefan@webrtc.org · 13 years ago
cde1c0b Fixed bad gtest filter. by phoglund@webrtc.org · 13 years ago
cf6a295 Making video codecs test framework integration test execute in a reproducable fashion. by kjellander@webrtc.org · 13 years ago
d5657c2 Refactored files according to google style since http://review.webrtc.org/314001/ is blocked on this and formatting changes should not be done with code changes. by henrike@webrtc.org · 13 years ago
68da6ad Remove WebRtc_ types. by andrew@webrtc.org · 13 years ago
454a27c The pthread_t is non-pointer type. by wu@webrtc.org · 13 years ago