Gitiles
Code Review
Sign In
gerrit-public.fairphone.software
/
fp2-dev
/
platform
/
external
/
chromium_org
/
third_party
/
webrtc
/
12e3ee7f2bb45b822f97f61c86a7a63e833a7ff2
12e3ee7
Isolate GYP target and .isolate files for tests
by kjellander@webrtc.org
· 11 years ago
95e1642
The video capture module for iOS.
by sjlee@webrtc.org
· 11 years ago
e238c24
Remove ViEBase::Init() call from VideoCall.
by pbos@webrtc.org
· 11 years ago
297e5ed
Remove VideoEngine class from new VideoEngine API.
by pbos@webrtc.org
· 11 years ago
87ae02e
Disable CanTransmitExtraRtpPacketsWithoutError on Windows.
by pbos@webrtc.org
· 11 years ago
f478f1e
Revert r4539 "Disable racy part of RunsRtpRtcpTestWithoutErrors".
by marpan@webrtc.org
· 11 years ago
c92c9ad
Disable racy part of RunsRtpRtcpTestWithoutErrors.
by pbos@webrtc.org
· 11 years ago
78dbe0b
Add native_handle.h to gyp.
by wuchengli@chromium.org
· 11 years ago
619cc69
To allow the propagation of under-run in NetEq.
by minyue@webrtc.org
· 11 years ago
80882f3
Replace MapWrapper with std::map<>.
by pbos@webrtc.org
· 11 years ago
a24f40d
Updated WebRTC version to 3.39
by elham@webrtc.org
· 11 years ago
f83e3a5
Signal when shutting down DirectTransport.
by pbos@webrtc.org
· 11 years ago
b0fc85b
Avoid acquiring VCM::_receiveCritSect during decode callback.
by wuchengli@chromium.org
· 11 years ago
48ac502
Run loopback tests with network thread.
by pbos@webrtc.org
· 11 years ago
edc86e5
Added Opus stereo support
by minyue@webrtc.org
· 11 years ago
6d94c78
Fix crash in screen capturer on Mac
by sergeyu@chromium.org
· 11 years ago
0bb1b31
Hand over loopback packets to a network thread.
by pbos@webrtc.org
· 11 years ago
16c8462
Don't pace out packets or generate padding when the pacer is disabled.
by stefan@webrtc.org
· 11 years ago
4ab008f
Remove include_dirs from test/test.gyp.
by pbos@webrtc.org
· 11 years ago
a0b4f27
Remove unused unreferenced code in webrtc/
by pbos@webrtc.org
· 11 years ago
8a11920
Revert "Avoid acquiring VCM::_receiveCritSect during decode callback."
by wuchengli@chromium.org
· 11 years ago
334bf81
Avoid acquiring VCM::_receiveCritSect during decode callback.
by wuchengli@chromium.org
· 11 years ago
165febc
Allowing decoding with errors, when disabling nack.
by mikhal@webrtc.org
· 11 years ago
a2505ea
Fix duplicate code
by niklas.enbom@webrtc.org
· 11 years ago
54e9955
Delete Channels without ChannelManager lock.
by pbos@webrtc.org
· 11 years ago
be78a05
Adding call to Opus PLC
by tina.legrand@webrtc.org
· 11 years ago
b263e41
Added logic for kSelectiveErrors to VCMJitterBuffer and corresponding unit tests.
by agalusza@google.com
· 11 years ago
9277c94
Ref-counted rewrite of ChannelManager.
by pbos@webrtc.org
· 11 years ago
5ce0e78
Code formatting on files touched in r4447.
by pbos@webrtc.org
· 11 years ago
7cbdbe6
Added configuration of max delay to ACM and NetEq
by pwestin@webrtc.org
· 11 years ago
547e0e3
Added Decoding with errors API to video_coding.h and removed unused DecodeError enum.
by agalusza@google.com
· 11 years ago
5c7fa98
Add turaj@webrtc.org to NetEq owners.
by turaj@webrtc.org
· 11 years ago
babb161
Disabled flaky HardwareTest.BuiltInWasapiAECWorksForAudioWindowsCoreAudioLayer.
by phoglund@webrtc.org
· 11 years ago
60ba778
Disabled SsrcPropagatesCorrectly on Linux.
by phoglund@webrtc.org
· 11 years ago
79df0bc
Better error treatment in NetEqImpl::InsertPacketInternal()
by minyue@webrtc.org
· 11 years ago
0c31023
removed NetEq::EnableDtmf()
by minyue@webrtc.org
· 11 years ago
df8d03f
* Update libjingle to 50389769.
by wu@webrtc.org
· 11 years ago
0f807b2
Invert dependency between webrtc_utility and media_file targets to reflect reality.
by fischman@webrtc.org
· 11 years ago
d5fb79c
Updated WebRTC version number to 3.38
by elham@webrtc.org
· 11 years ago
50ff6a5
Switch C++-style C headers with their C equivalents.
by pbos@webrtc.org
· 11 years ago
30c741a
Fix implicit int->bool conversion in VideoSendStream::DeliverRtcp.
by pbos@webrtc.org
· 11 years ago
2c00af7
Use RtpHeaderParser in VideoCall implementation.
by pbos@webrtc.org
· 11 years ago
bf9bc32
Glue code and tests for NACK in new VideoEngine API.
by pbos@webrtc.org
· 11 years ago
dac40f8
Fix send times in video_full_stack.
by pbos@webrtc.org
· 11 years ago
46d2ca1
Add back is.FrameProvider() call lost in r4194.
by pbos@webrtc.org
· 11 years ago
0b6e893
Remove redundant conditions key.
by andrew@webrtc.org
· 11 years ago
75370f1
Add one API for implementing Initial delay.
by turaj@webrtc.org
· 11 years ago
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
Next »