Gitiles
Code Review
Sign In
gerrit-public.fairphone.software
/
fp2-dev
/
platform
/
external
/
chromium_org
/
third_party
/
webrtc
/
9d939ee2a490791105b3117b63dc11a8a3233671
9d939ee
Adds all unittests to android NDK-APK framework.
by henrike@webrtc.org
· 11 years ago
aa0dac5
Add some virtual and OVERRIDEs in webrtc/common_audio/
by pbos@webrtc.org
· 11 years ago
0bf6b98
Fix some chromium-style warnings in webrtc/modules/audio_processing/
by pbos@webrtc.org
· 11 years ago
f72eb49
Fix crash in DesktopRegion::Intersect().
by sergeyu@chromium.org
· 11 years ago
42ef0f5
Fix some chromium-style warnings in webrtc/system_wrappers/
by pbos@webrtc.org
· 11 years ago
28dda63
Removed lines preventing simultaneous kHardNack and decoding with errors. Also made changes recommended by gcl lint (with the exception of changing non-const references to pointers).
by agalusza@google.com
· 11 years ago
26a30e6
Unbreak clang/android build of webrtc.
by fischman@webrtc.org
· 11 years ago
53d1ade
Adding possibility to use encoding time when trigger underuse for frame based overuse detection.
by mflodman@webrtc.org
· 11 years ago
9b748e5
Merge r4374 from stable to trunk.
by xians@webrtc.org
· 11 years ago
2d4c1a1
Merge r4394 from stable to trunk.
by xians@webrtc.org
· 11 years ago
f686778
Merge r4326 from stable to trunk.
by xians@webrtc.org
· 11 years ago
dadf0f7
Handel zero correlation if at the same time distortion is also zero.
by turaj@webrtc.org
· 11 years ago
e2df770
Add some virtual and OVERRIDEs in webrtc/modules/audio_coding/
by pbos@webrtc.org
· 11 years ago
7df7f61
Fix some chromium-style warnings in webrtc/modules/desktop_capture/
by pbos@webrtc.org
· 11 years ago
463eb03
Fix some chromium-style warnings in webrtc/modules/pacing/
by pbos@webrtc.org
· 11 years ago
d0557b5
Fix some chromium-style warnings in webrtc/modules/rtp_rtcp/
by pbos@webrtc.org
· 11 years ago
ff3f7f6
Fix some chromium-style warnings in webrtc/modules/remote_bitrate_estimator/
by pbos@webrtc.org
· 11 years ago
ee34820
Fix some chromium-style warnings in webrtc/modules/bitrate_controller/
by pbos@webrtc.org
· 11 years ago
81e21c6
Added libjingle_peerconnection_java_unittest to buildbot_tests.py
by phoglund@webrtc.org
· 11 years ago
fe8ba4d
Move internal aec_core defines out of header.
by andrew@webrtc.org
· 11 years ago
3ea4830
Add svn:ignore properties for all spuriously-removed dirs on Linux64 Release (internal).
by fischman@webrtc.org
· 11 years ago
b36d452
Correcting Turaj's email.
by turaj@webrtc.org
· 11 years ago
80df10d
Fix some chromium-style warnings in webrtc/modules/video_coding/
by pbos@webrtc.org
· 11 years ago
ae6d494
Fix some chromium-style warnings in webrtc/test/
by pbos@webrtc.org
· 11 years ago
9bf2b46
Fix some chromium-style warnings in webrtc/tools/
by pbos@webrtc.org
· 11 years ago
e142b98
Fix some chromium-style warnings in webrtc/modules/audio_device/
by pbos@webrtc.org
· 11 years ago
46688dd
Sets up framework for decoding with errors: collects frame sizes (in number of packets) in JB and passes this information to VCMSessionInfo with rtt_ms as FrameData.
by agalusza@google.com
· 11 years ago
03bfae8
PeerConnectionTest.java: make the test work for the bots' v4l2loopback.
by fischman@webrtc.org
· 11 years ago
2a61170
Land http://webrtc-codereview.appspot.com/1632005/
by niklas.enbom@webrtc.org
· 11 years ago
0b8a595
Updated WebRTC version to 3.37 TBR=tnakamura@webrtc.org
by elham@webrtc.org
· 11 years ago
263411c
Improved error messages when binaries are missing. Also stderr = stdout now.
by phoglund@webrtc.org
· 11 years ago
6429cdb
To fix a bug in InverseFFTAndWindow() function in AECM.
by kma@webrtc.org
· 11 years ago
7b97b16
Updated WebRtcNsx_PrepareSpectrumNeon() in accordance with the new real FFT interface in APM. For reference, you can check https://webrtc-codereview.appspot.com/1830004/diff/92001/webrtc/modules/audio_processing/ns/nsx_core.c, line 594 "static void PrepareSpectrumC()".
by kma@webrtc.org
· 11 years ago
1095587
Access receiving_ under receive_cs critical section
by braveyao@webrtc.org
· 11 years ago
dbdcf16
Don't set clang_use_chrome_plugins in common.gypi
by sergeyu@chromium.org
· 11 years ago
0bd4d89
Fixes resources and data path in modules_unittests.isolate.
by henrike@webrtc.org
· 11 years ago
fd87865
Downstream latest Chromium SincResampler changes.
by andrew@webrtc.org
· 11 years ago
b31f64f
Update include paths in device_info_external.cc
by sergeyu@chromium.org
· 11 years ago
d13f24b
Add a Config class interface to AudioProcessing for passing options.
by andrew@webrtc.org
· 11 years ago
2cf4d85
Fix include path in video_capture_external.cc
by niklas.enbom@webrtc.org
· 11 years ago
a3c7fa2
Formalized Real 16-bit FFT for APM.
by kma@webrtc.org
· 11 years ago
51f7c7e
Fix ScreenCapturerLinux not to use XDamage when requested.
by sergeyu@chromium.org
· 11 years ago
530f40f
webrtc/common_types.h: Document bitrate fields' units.
by fischman@webrtc.org
· 11 years ago
7b87e6b
Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. Also disables 64 bit Mac builds for libjingle
by henrike@webrtc.org
· 11 years ago
ecbeb2b
Hooking up first simple CPU adaptation version.
by mflodman@webrtc.org
· 11 years ago
cb2fb3f
Revert 4382 "Makes webrtc and libjingle build from the same gyp-..."
by henrike@webrtc.org
· 11 years ago
4432261
Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. Also disables 64 bit Mac builds for libjingle
by henrike@webrtc.org
· 11 years ago
675eead
Modified the presubmit checks such that difference license templates are checked for in webrtc and talk folder.
by henrike@webrtc.org
· 11 years ago
95a4477
Correctly rebuild WebRTCDemo after jni/ source file changes
by yujie.mao@webrtc.org
· 11 years ago
331a9b2
Revert 4372 "Makes webrtc and libjingle build from the same gyp-..."
by henrike@webrtc.org
· 11 years ago
30a83c1
Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches.
by henrike@webrtc.org
· 11 years ago
7c4152b
AppRTCDemo: build fixes for iOS build in webrtc
by fischman@webrtc.org
· 11 years ago
382ef1e
Undo libvpx include changes in r4348 to fix build.
by tnakamura@webrtc.org
· 11 years ago
a491674
Default constructor for RtcpAppHandler.
by pbos@webrtc.org
· 11 years ago
62365d0
clean up incomplete revert in r4357 Also revert r4319, will follow up with pbos
by tnakamura@webrtc.org
· 11 years ago
1628267
Revert r4301
by tnakamura@webrtc.org
· 11 years ago
8eea45a
Fixes: Resolves conflict that will happen when merging libjingle's and WebRTC's supplemental.gyp. By separating build_with_chromium and build_with_libjingle one can now just define build_with_libjingle in libjingle's supplemental.gyp. Once that is done it will be possible to merge the two supplemental.gyp-files. I.e. in WebRTC the supplemental.gyp would only set build_with_chromium to 0 since there is no longer any reason to disable logging and tests as they will be accessible in the same repository as libjingle.
by henrike@webrtc.org
· 11 years ago
c64c74c
Include files from webrtc/.. paths in signal_processing/.
by pbos@webrtc.org
· 11 years ago
93c9ccb
Include files from webrtc/.. paths in media_file/.
by pbos@webrtc.org
· 11 years ago
d92e5df
Make sure first RTP packet counts as in-order.
by pbos@webrtc.org
· 11 years ago
3fa41a6
Include files from webrtc/.. paths in bitrate_controller/.
by pbos@webrtc.org
· 11 years ago
2a2e7ff
Include files from webrtc/.. paths in video_coding/.
by pbos@webrtc.org
· 11 years ago
1a9da45
Revert r4320 "Fix three uninitialized members in rtp_receiver_impl.cc"
by elham@webrtc.org
· 11 years ago
11c4464
Revert r4321 "Fix uninitialized value warning in rtp_payload_registry and make sure we return an error if the payload type isn't registered"
by elham@webrtc.org
· 11 years ago
35c7707
Revert r4322 "Support sending multiple report blocks and keeping track of statistics on
by elham@webrtc.org
· 11 years ago
9706ba7
Revert r4328
by elham@webrtc.org
· 11 years ago
71e2b87
Updated WebRTC version to 3.36 TBR=tnakamura@webrtc.org
by elham@webrtc.org
· 11 years ago
fa0f507
Remove dead video_capture for QuickTime.
by pbos@webrtc.org
· 11 years ago
adb1418
Include files from webrtc/.. paths in video_capture/.
by pbos@webrtc.org
· 11 years ago
fab7ea2
Include files from webrtc/.. paths in utility/.
by pbos@webrtc.org
· 11 years ago
3aefd72
Remove dead code testAPI.cc.
by pbos@webrtc.org
· 11 years ago
9c45631
Include files from webrtc/.. paths in video_render/.
by pbos@webrtc.org
· 11 years ago
f58779b
Fix some voe_auto_test uninitialised-value errors.
by pbos@webrtc.org
· 11 years ago
7c413cd
Include files from webrtc/.. paths in audio_device/.
by pbos@webrtc.org
· 11 years ago
784d202
Fix root-relative includes for pacing/.
by pbos@webrtc.org
· 11 years ago
5f999a3
Fixes a crash when sending SR reports from a sender only module.
by stefan@webrtc.org
· 11 years ago
c80c63d
ModuleRTPRTCP call rtcp_sender_.TMMBR() directly instead of calling its own API.
by braveyao@webrtc.org
· 11 years ago
633c018
Sorted headers under rtp_rtcp/.
by pbos@webrtc.org
· 11 years ago
3265bbd
Include files from webrtc/.. paths in video_engine/.
by pbos@webrtc.org
· 11 years ago
8d671f5
Direct3D renderer for new VideoEngine API tests.
by pbos@webrtc.org
· 11 years ago
45ab259
Support sending multiple report blocks and keeping track of statistics on several SSRCs.
by stefan@webrtc.org
· 11 years ago
2211018
Fix uninitialized value warning in rtp_payload_registry and make sure we return an error if the payload type isn't registered.
by stefan@webrtc.org
· 11 years ago
aa64b24
Fix three uninitialized members in rtp_receiver_impl.cc.
by stefan@webrtc.org
· 11 years ago
efa245a
Initialize payload-type frequency in channel.cc.
by pbos@webrtc.org
· 11 years ago
c2ae12e
Update version number to 3.35
by tnakamura@webrtc.org
· 11 years ago
faf2dba
Update version number to 3.34
by tnakamura@webrtc.org
· 11 years ago
0114e3d
Add root_path_android.cc to webrtc/test/Android.mk.
by pbos@webrtc.org
· 11 years ago
31c750c
Fixed implicit-int-conversion bugs.
by pbos@webrtc.org
· 11 years ago
7ca2327
Fix a circular dependency by removing an unnecessary dependency, add a missing include_tests check and missing lib references for android.
by stefan@webrtc.org
· 11 years ago
6dd15dc
Create gyp target for bwe components.
by stefan@webrtc.org
· 11 years ago
12d5ede
Initial port of FullStackTest to new VideoEngine API.
by pbos@webrtc.org
· 11 years ago
5ca7ffd
Arguments need to be separated when implementing gyp-actions.
by henrike@webrtc.org
· 11 years ago
7183bbb
Cleanup WebRTC tracing
by hclam@chromium.org
· 11 years ago
80fa00b
Added modules_unittests.isolate for ndk-apk builds.
by henrike@webrtc.org
· 11 years ago
6f44ab3
Disables unit tests that don't work on Android for Android.
by henrike@webrtc.org
· 11 years ago
5b871f8
Fixes build breakage when building WebRTC in Chromium and having include_tests=1.
by henrike@webrtc.org
· 11 years ago
13efe02
Fixes broken gyp-condition.
by henrike@webrtc.org
· 11 years ago
0642536
Unreverts revert: Makes it possible to find files used by some unit tests when running them as Chrome native tests.
by henrike@webrtc.org
· 11 years ago
b7f287d
Use scoped_ptr<> for loopback.cc
by pbos@webrtc.org
· 11 years ago
a430fef
Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the
by stefan@webrtc.org
· 11 years ago
Next »