admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc
History
Bjorn Terelius c159b046d7 Renamed the ACMDump to RtcEventLog and moved it to webrtc/video, since it is not specific to the audio coding module. Updated .gyp and .gn files accordingly. 
Placed the protobuf structures in the namespace webrtc::rtclog. Removed the message field from the DebugEvent structure, since it was not used.

Added an interface to set config information for VideoReceiveStream and VideoSendStream in the event log.

Added function to log full RTCP packets and changed RTP-logging to only log headers.

Significantly extended the unit tests for RtcEventLog.

R=ivoc@webrtc.org, minyue@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1230973005 .

Cr-Commit-Position: refs/heads/master@{#9656}
2015-07-30 09:06:09 +00:00
..
base
Remove CircularFileStream / replace it with CallSessionFileRotatingStream.
2015-07-23 19:27:06 +00:00
build
Ignore genperf lib in merge_libs.py.
2015-07-21 19:54:44 +00:00
common_audio
Split MoveReadPosition into Forward and Backward versions.
2015-07-28 07:53:05 +00:00
common_video
Re-land "Convert native handles to buffers before encoding."
2015-06-05 09:08:12 +00:00
examples/android/media_demo
VoE: Remove unused interfaces
2015-05-26 08:25:00 +00:00
libjingle
Remove webrtc/libjingle/{examples,session}.
2015-06-12 11:13:46 +00:00
modules
Renamed the ACMDump to RtcEventLog and moved it to webrtc/video, since it is not specific to the audio coding module. Updated .gyp and .gn files accordingly.
2015-07-30 09:06:09 +00:00
overrides
Attempt at fixing error on the Chrome Windows FYI bots.
2015-05-25 19:22:23 +00:00
p2p
Fix an NPE when creating TurnPort with a NULL socket.
2015-07-17 23:22:02 +00:00
sound
Remove henrike@ from OWNERS
2015-04-01 15:08:49 +00:00
system_wrappers
When we trace to file, add eol of each trace message.
2015-07-28 02:46:01 +00:00
test
Add a frame generator that allows scrolling over a larger still image, for use with new screen sharing quality tests.
2015-07-29 14:58:17 +00:00
tools
Configure AudioProcessing directly in agc_harness.
2015-07-23 01:27:16 +00:00
video
Renamed the ACMDump to RtcEventLog and moved it to webrtc/video, since it is not specific to the audio coding module. Updated .gyp and .gn files accordingly.
2015-07-30 09:06:09 +00:00
video_engine
Remove webrtc::Config from ViEChannelGroup.
2015-07-27 15:37:14 +00:00
voice_engine
Removing AudioMixerStatusReceiver and ParticipantStatistics.
2015-07-28 09:07:35 +00:00
.gitignore
…
audio_receive_stream.h
Control combined_audio_video_bwe with config bool.
2015-07-24 14:10:25 +00:00
audio_send_stream.h
Define Stream base classes
2015-07-16 07:30:20 +00:00
BUILD.gn
Renamed the ACMDump to RtcEventLog and moved it to webrtc/video, since it is not specific to the audio coding module. Updated .gyp and .gn files accordingly.
2015-07-30 09:06:09 +00:00
call.h
Define Stream base classes
2015-07-16 07:30:20 +00:00
codereview.settings
Update deeper codereview.settings files to match the root.
2015-06-24 19:40:39 +00:00
common_types.cc
Parsing of transport wide sequence number rtp extension header.
2015-03-17 14:33:46 +00:00
common_types.h
Revert the process noise co-variance of the bitrate over-use estimator to its value prior to r9545.
2015-07-10 09:28:46 +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 AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
config.h
Add options for NetEq fast accelerate mode through libjingle
2015-06-01 08:29:55 +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
LICENSE
…
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
…
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
…
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
…
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
Define Stream base classes
2015-07-16 07:30:20 +00:00
video_renderer.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
video_send_stream.h
Define Stream base classes
2015-07-16 07:30:20 +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
Renamed the ACMDump to RtcEventLog and moved it to webrtc/video, since it is not specific to the audio coding module. Updated .gyp and .gn files accordingly.
2015-07-30 09:06:09 +00:00
webrtc.gyp
Renamed the ACMDump to RtcEventLog and moved it to webrtc/video, since it is not specific to the audio coding module. Updated .gyp and .gn files accordingly.
2015-07-30 09:06:09 +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.