Log - 00d566ec297d52c8e732d7299d69e4e580707d61 - fp2-dev/platform/external/chromium_org/third_party/webrtc

f13f1fc Updated the sync module with a slow moving filter by pwestin@webrtc.org · 11 years ago
c11933f Removed unused variable. by mflodman@webrtc.org · 11 years ago
bea854a Fixing Coverity issues. by mflodman@webrtc.org · 11 years ago
b6e175d Ensure build_demo.py run subprocesses with bash shell. by kjellander@webrtc.org · 11 years ago
06e8026 New ViE interface. by mflodman@webrtc.org · 11 years ago
237fe4f Remove vim/emacs modelines from .gypi files by pbos@webrtc.org · 11 years ago
7bc7e02 Adding a payload type for RTX. by mflodman@webrtc.org · 11 years ago
9b53152 Change capture interface to use NTP capture time. by stefan@webrtc.org · 11 years ago
f272497 Adding playout buffer status to the voe video sync by pwestin@webrtc.org · 11 years ago
de1c434 Updated WebRTC version number to 3.29 by elham@webrtc.org · 11 years ago
6e816cb WebRTCDemo: Enable making multiple calls. by fischman@webrtc.org · 11 years ago
74472fe More trace events by hclam@chromium.org · 11 years ago
73ebe67 Revert "With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps." by stefan@webrtc.org · 11 years ago
67879bc WebRtc_Word32 -> int32_t in video_engine/ by pbos@webrtc.org · 11 years ago
65deb26 With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps. by stefan@webrtc.org · 11 years ago
208a648 Always set render delay in ViEChannel::RegisterExternalDecoder. by pbos@webrtc.org · 11 years ago
8ec8955 Remove the old unused udp_transport by pwestin@webrtc.org · 11 years ago
a9f28d5 Clean packets on the network when closing + made loopback test actually run again. by mflodman@webrtc.org · 11 years ago
004f462 Add GYP target for WebRTC Video demo for Android. by kjellander@webrtc.org · 11 years ago
2ed1cd9 Permit arbitrary payload names for kVideoCodecGeneric. by pbos@webrtc.org · 11 years ago
ef91cbf Remove WEBRTC_*_ENGINE_NETWORK_API use by pwestin@webrtc.org · 11 years ago
dded206 Adds event traces and counters for WebRTC receive side. by edjee@google.com · 11 years ago
065b64d Remove UDP transport API from ViE by pwestin@webrtc.org · 11 years ago
f3ac3ba Updated Webrtc version to 3.28 by elham@webrtc.org · 11 years ago
fa2dd22 Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested. by solenberg@webrtc.org · 11 years ago
0c0795e Fix broken audio. by leozwang@webrtc.org · 11 years ago
88f12ab Fix flakiness in network up/down event tests when running under memcheck. by stefan@webrtc.org · 11 years ago
6fc5215 WebRTCDemo: remove unnecessary stop & start during orientation change which isn't necessary since API v14. by fischman@webrtc.org · 11 years ago
dca71b2 Add interface to signal a network down event. by stefan@webrtc.org · 11 years ago
f49577f Updated WebRTC version to 3.27 by elham@webrtc.org · 11 years ago
7fd368f Bugfix for extended RTP/RTCP test by pwestin@webrtc.org · 11 years ago
c075e25 Move the VIE tests to use external transport instead of the built in udp transport by pwestin@webrtc.org · 11 years ago
e1198e6 Add min and target bitrate to VideoCodec. by marpan@webrtc.org · 11 years ago
6313692 Follow-up fix for r3681. by stefan@webrtc.org · 11 years ago
ffe2ec6 Updated WebRTC version number to 3.26 by elham@webrtc.org · 11 years ago
72e204a Change VCM interface to take target bitrate in bits per second. by stefan@webrtc.org · 11 years ago
e3339fc Generic video-codec support. by pbos@webrtc.org · 11 years ago
946d240 Adding RTX on source by mikhal@webrtc.org · 11 years ago
912b7f7 Revert r3667 and r3665 Review URL: https://webrtc-codereview.appspot.com/1199004 by pwestin@webrtc.org · 11 years ago
40749c1 Added destructors for tests to control destruct order by pwestin@webrtc.org · 11 years ago
b793abe Increasing size of nack list in buffered mode. by mikhal@webrtc.org · 11 years ago
2daec4c Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work. by pwestin@webrtc.org · 11 years ago
8623e9b Lazy capture_device_info acquisition. by pbos@webrtc.org · 11 years ago
78e450f Enabling bufffering mode with no sync module or VoE by mikhal@webrtc.org · 11 years ago
640d1bb Updated version number to 3.25 by elham@webrtc.org · 11 years ago
03c41d7 Update integration tests for idempotent RTP header settings. by bemasc@google.com · 11 years ago
c6242c9 Destroy VCM and VPM instead of delete. by mflodman@webrtc.org · 12 years ago
1601d4a Handle multiple calls to set initial delay by mikhal@webrtc.org · 12 years ago
55e6f58 Stop and restart fix. by mflodman@webrtc.org · 12 years ago
c86dbab Fixed typo in vie_autotest_loopback.cc. by pbos@webrtc.org · 12 years ago
0329e59 Rename webrtc::StatsObserver to webrtc::CallStatsObserver by fischman@webrtc.org · 12 years ago
08998cd fixing nack list size calculation by mikhal@webrtc.org · 12 years ago
d7d9c5a Updated version number to 3.24 by elham@webrtc.org · 12 years ago
f4d3788 Refactoring temporal layers implementation and adding VideoCodecMode for easier control of codec settings. by stefan@webrtc.org · 12 years ago
0583ee6 Add VoE interface to VieRTP test by mikhal@webrtc.org · 12 years ago
9d6fcb3 Adding a receive side API for buffering mode. by mikhal@webrtc.org · 12 years ago
d380c03 Roll Chromium revision 176094:182149 by kjellander@webrtc.org · 12 years ago
5328d5e Remove MultiStreamMode from test. by stefan@webrtc.org · 12 years ago
7e63b04 Reset ssrc when calling SetSendCodec. by mflodman@webrtc.org · 12 years ago
2b9a2e0 Sync libvpx and its gyp wrapper from Chromium. by andrew@webrtc.org · 12 years ago
f093410 Increase maximum resolution to 4k x 3k. by fbarchard@google.com · 12 years ago
64e4c0e Android NDK build tools by kjellander@webrtc.org · 12 years ago
7189656 Set SingleStream BWE in unittests. by stefan@webrtc.org · 12 years ago
0c66de6 Updates to send side streaming mode: by mikhal@webrtc.org · 12 years ago
ce42660 Update version number to 3.23 by tnakamura@webrtc.org · 12 years ago
b2feb57 Made it possible to render custom call output to file. by phoglund@webrtc.org · 12 years ago
cd1ac8b Don't report an error for GetEstimatedReceiveBandwidth if there is no valid by mflodman@webrtc.org · 12 years ago
89b36e9 Enable indefinitely running vie_auto_test option by kjellander@webrtc.org · 12 years ago
8461b0a Updated version number to 3.22 by elham@webrtc.org · 12 years ago
1e7f77a Fixing/disabling Windows x64 warnings by kjellander@webrtc.org · 12 years ago
1619664 Adding a send side API for streaming by mikhal@webrtc.org · 12 years ago
7fff32c Fix mismatch between different NACK list lengths and packet buffers. by stefan@webrtc.org · 12 years ago
2a5dbce Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEstimator::Process trigger new REMB messages. by stefan@webrtc.org · 12 years ago
08f721b Set frame length for frame converting in external renderer by braveyao@webrtc.org · 12 years ago
ca0e88a VP8: Making key frame interval a tunnable parameter by mikhal@webrtc.org · 12 years ago
2be6c3e Fix webrtc compilation errors for Chrome Win64 by andrew@webrtc.org · 12 years ago
03db776 Moving ViE test files and deleting files no longer used. by mflodman@webrtc.org · 12 years ago
b8d25e1 Updated version number to 3.21 by elham@webrtc.org · 12 years ago
9669d71 Remove '<(library)' in gyp files. by wjia@webrtc.org · 12 years ago
04ceffe Convert psnr and ssim to strings before printing them. by stefan@webrtc.org · 12 years ago
05d046c Removing outdated comment by mikhal@webrtc.org · 12 years ago
7a7d234 Made ViEToFileRenderer use a separate thread for rendering frames to file. by stefan@webrtc.org · 12 years ago
e7a9bd4 logical 'and' of mutually exclusive tests is always false in ViECodecImpl::CodecValid() by braveyao@webrtc.org · 12 years ago
6bc469c Disable full stack PSNR/SSIM triggers on Mac and Win for now due to flakiness. Adding plots of PSNR and SSIM. by stefan@webrtc.org · 12 years ago
3212036 Disable PSNR/SSIM thresholds for the Gilber-Elliot test. by stefan@webrtc.org · 12 years ago
b790741 Generalized mechanism for excluding gtests on platforms, disabled broken tests on mac. by phoglund@webrtc.org · 12 years ago
0a11c53 Disabled GQoS since it breaks ViE auto test. by henrika@webrtc.org · 12 years ago
71f3f68 Enable external encoders with internal picture source. by stefan@webrtc.org · 12 years ago
026e6b6 Using Convert in lieu of ExtractBuffer: Less error prone (as we don't need to compute buffer sizes etc.). This cl is first in a series (doing all of WebRtc would make it quite a big cl). While at it, fixing a few headers. by mikhal@webrtc.org · 12 years ago
5750956 Updated version number to 3.20 by elham@webrtc.org · 12 years ago
6b04e54 Removed spaces from full stack test labels, consolidated graphs by phoglund@webrtc.org · 12 years ago
ac09423 Changed assert to log. by mflodman@webrtc.org · 12 years ago
a0f23f1 Make protection method, filename and resolution configurable for FullStackTest. by stefan@webrtc.org · 12 years ago
067911c vie auto test: Adding a constructor for NetworkParameters by mikhal@webrtc.org · 12 years ago
574c33c ViE autotest: Adding loss models to the external transport by mikhal@webrtc.org · 12 years ago
32c4e46 Updated version number to 3.19 by elham@webrtc.org · 12 years ago
f314c80 Added API to get receive side video delay. by mflodman@webrtc.org · 12 years ago
eaa05c8 Remove latency excl network and add render time diff stats. by stefan@webrtc.org · 12 years ago
afdaa2c Fix for buffer overflow, WebRTC issue 1196 by elham@webrtc.org · 12 years ago
4f92005 Added jitter to fake network pipe. by mflodman@webrtc.org · 12 years ago