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stefan 2328a94ec7 Add average rtt to CallStatsObserver and an average rtt histogram. 
TBR=mflodman@webrtc.org

BUG=webrtc:4711,webrtc:4548

Review URL: https://codereview.webrtc.org/1279543005

Cr-Commit-Position: refs/heads/master@{#9687}
2015-08-07 11:27:56 +00:00
..
base
Regenerate bind.h using pump.py
2015-08-06 10:41:04 +00:00
build
Suppress a race in libjingle_peerconnection_unittest
2015-08-06 11:34:39 +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
Move all the examples from the talk directory into the webrtc examples directory.
2015-08-05 22:48:29 +00:00
libjingle
Remove webrtc/libjingle/{examples,session}.
2015-06-12 11:13:46 +00:00
modules
Add average rtt to CallStatsObserver and an average rtt histogram.
2015-08-07 11:27:56 +00:00
overrides
Attempt at fixing error on the Chrome Windows FYI bots.
2015-05-25 19:22:23 +00:00
p2p
Fix Turn TCP port issue.
2015-08-03 17:23:40 +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 support for transport wide sequence numbers
2015-08-03 11:38:48 +00:00
tools
Configure AudioProcessing directly in agc_harness.
2015-07-23 01:27:16 +00:00
video
Selectable number of TL screenshare loopback test. Also contains some tweaks to make a single TL perform better.
2015-08-05 09:02:09 +00:00
video_engine
Add average rtt to CallStatsObserver and an average rtt histogram.
2015-08-07 11:27:56 +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
Includes webrtc/build/protoc.gypi instead of build/protoc.gypi
2015-07-30 10:45:24 +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
Integration of VP9 packetization.
2015-07-31 13:10:16 +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
Move all the examples from the talk directory into the webrtc examples directory.
2015-08-05 22:48:29 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
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
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.