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andrew 88703d756a Disable base/logging.h stderr logs by default for webrtc/ tests. 
base/logging.h dumped to stderr by default in debug mode, but webrtc
"trace" (via system_wrappers/../logging.h) has that feature disabled by
default. This makes the two consistent.

Bonus: log the filename:line in base/logging.h, which exists in the
system_wrappers variant.

TEST=neteq_impl.cc logs (which use base/logging.h) no longer appear in
debug mode, unless --logs=true is passed. Filenames appear correctly.

Review URL: https://codereview.webrtc.org/1331503002

Cr-Commit-Position: refs/heads/master@{#9868}
2015-09-07 07:35:03 +00:00
..
base
Disable base/logging.h stderr logs by default for webrtc/ tests.
2015-09-07 07:35:03 +00:00
build
Revert of purge nss files and dependencies (patchset #1 id:1 of https://codereview.webrtc.org/1313233005/ )
2015-09-05 11:39:24 +00:00
common_audio
Convert some more things to size_t.
2015-08-29 00:31:15 +00:00
common_video
VideoFrameBuffer: Make non-const data access explicit
2015-08-26 14:06:30 +00:00
examples
Handle all RTCICEConnectionState values in ARDVideoCallViewController
2015-09-01 22:35:42 +00:00
libjingle
Remove webrtc/libjingle/{examples,session}.
2015-06-12 11:13:46 +00:00
modules
Fix ScreenCapturerWinGdi to handle DesktopFrameWin::Create() errors.
2015-09-05 01:38:15 +00:00
overrides
Attempt at fixing error on the Chrome Windows FYI bots.
2015-05-25 19:22:23 +00:00
p2p
Remove AsyncHttpRequest, AutoPortAllocator, ConnectivityChecker, and HttpPortAllocator.
2015-09-04 11:21:14 +00:00
sound
Remove henrike@ from OWNERS
2015-04-01 15:08:49 +00:00
system_wrappers
Disable base/logging.h stderr logs by default for webrtc/ tests.
2015-09-07 07:35:03 +00:00
test
Disable base/logging.h stderr logs by default for webrtc/ tests.
2015-09-07 07:35:03 +00:00
tools
Update a ton of audio code to use size_t more correctly and in general reduce
2015-08-24 21:52:45 +00:00
video
Changed LogRtpHeader to read the header length from the packet instead of requiring an extra argument.
2015-09-04 10:39:51 +00:00
video_engine
Update SendTimeHistory to store complete PacketInfo, not just send time
2015-09-04 11:43:23 +00:00
voice_engine
Stylizing AudioConferenceMixer.
2015-08-31 14:04:44 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
audio_receive_stream.h
Control combined_audio_video_bwe with config bool.
2015-07-24 14:10:25 +00:00
audio_send_stream.h
Add send transports to individual webrtc::Call streams.
2015-08-28 11:07:15 +00:00
BUILD.gn
GN: Fix compilation with NaCl toolchain
2015-09-02 21:22:48 +00:00
call.h
Add send transports to individual webrtc::Call streams.
2015-08-28 11:07:15 +00:00
codereview.settings
Update deeper codereview.settings files to match the root.
2015-06-24 19:40:39 +00:00
common_types.cc
Separating voice activity flag from audio level in RtpHeaderExtension.
2015-08-10 13:08:46 +00:00
common_types.h
Update a ton of audio code to use size_t more correctly and in general reduce
2015-08-24 21:52:45 +00:00
common.gyp
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
common.h
Make Config::default_value leak instead of having an exit-time destructor.
2015-05-23 00:50:33 +00:00
config.cc
Add support for transport wide sequence numbers
2015-08-03 11:38:48 +00:00
config.h
Add support for transport wide sequence numbers
2015-08-03 11:38:48 +00:00
engine_configurations.h
Remove VideoEngine interfaces.
2015-05-12 14:51:08 +00:00
frame_callback.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
libjingle_examples.gyp
Fix build when using Xcode 7 which contains .tbd files instead of .dylib
2015-09-05 03:01:08 +00:00
LICENSE
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
stream.h
Define Stream base classes
2015-07-16 07:30:20 +00:00
supplement.gypi
Move target_subarch from gyp_webrtc to supplement.gypi
2015-04-16 07:24:23 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Clear ARM NEON flag
2015-05-20 05:20:04 +00:00
video_decoder.h
iOS HW H264 support.
2015-06-29 21:35:08 +00:00
video_encoder.h
Allow for framerate reduction for HW encoder.
2015-07-13 23:26:40 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Re-land "Convert native handles to buffers before encoding."
2015-06-05 09:08:12 +00:00
video_receive_stream.h
Wire up currently-received video codec to stats.
2015-08-28 14:35:40 +00:00
video_renderer.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
video_send_stream.h
Pass the encoder's internal source property through to video_sender to request a keyframe from the external encoder
2015-09-04 01:24:53 +00:00
webrtc_examples.gyp
Adding support for OpenSL ES output in native WebRTC
2015-05-18 14:49:04 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Includes webrtc/build/protoc.gypi instead of build/protoc.gypi
2015-07-30 10:45:24 +00:00
webrtc.gyp
Includes webrtc/build/protoc.gypi instead of build/protoc.gypi
2015-07-30 10:45:24 +00:00

README.chromium 

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.