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webrtc_m130/webrtc
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Henrik Lundin 1bb8cf846d NetEq/ACM: Refactor how packet waiting times are calculated 
With this change, the aggregates for packet waiting times are
calculated in NetEq's StatisticsCalculator insead of in
AcmReceiver. This simplifies things somewhat, and avoids having to
copy the raw data on polling.

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

Review URL: https://codereview.webrtc.org/1296633002 .

Cr-Commit-Position: refs/heads/master@{#9778}
2015-08-25 11:08:17 +00:00
..
base
RTCCertificates added to RTCConfiguration, used by WebRtcSession/-DescriptionFactory.
2015-08-25 07:53:36 +00:00
build
Re-enable LLVM LTO on Neon targets.
2015-08-18 04:05:59 +00:00
common_audio
Add a base class to Wav{Reader,Writer} to access shared parameters.
2015-08-25 00:29:34 +00:00
common_video
Re-land "Convert native handles to buffers before encoding."
2015-06-05 09:08:12 +00:00
examples
Android: Remove VideoRenderer.Callbacks.canApplyRotation()
2015-08-25 07:32:16 +00:00
libjingle
Remove webrtc/libjingle/{examples,session}.
2015-06-12 11:13:46 +00:00
modules
NetEq/ACM: Refactor how packet waiting times are calculated
2015-08-25 11:08:17 +00:00
overrides
Attempt at fixing error on the Chrome Windows FYI bots.
2015-05-25 19:22:23 +00:00
p2p
TcpPort Reconnect should inform upper layer to start sending again.
2015-08-24 18:58:07 +00:00
sound
Remove henrike@ from OWNERS
2015-04-01 15:08:49 +00:00
system_wrappers
Update a ton of audio code to use size_t more correctly and in general reduce
2015-08-24 21:52:45 +00:00
test
Update a ton of audio code to use size_t more correctly and in general reduce
2015-08-24 21:52:45 +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
Enabling screensharing perf test.
2015-08-11 17:43:09 +00:00
video_engine
Add average rtt to CallStatsObserver and an average rtt histogram.
2015-08-07 11:27:56 +00:00
voice_engine
Update a ton of audio code to use size_t more correctly and in general reduce
2015-08-24 21:52:45 +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
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
…
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
Add stats overlay to iOS AppRTCDemo.
2015-08-14 18:00:11 +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
…
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
…
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.