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sprang 233bd87d45 Add RemoteEstimatorProxy for capturing receive times 
For use when send-side bandwidth estimation is enabled.

Receive times need to be captured, buffered and then sent using
TransportFeedback RTCP messaged back to the send side.

BUG=webrtc:4173

Review URL: https://codereview.webrtc.org/1290813008

Cr-Commit-Position: refs/heads/master@{#9898}
2015-09-08 20:25:20 +00:00
..
base
Remove arraysize.h gcc hack and Chromium override.
2015-09-08 07:57:41 +00:00
build
GN: Fix invalid configuration for Android GCC build.
2015-09-07 10:57:57 +00:00
common_audio
Alphabetize common_audio/OWNERS.
2015-09-08 17:35:51 +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
Add RemoteEstimatorProxy for capturing receive times
2015-09-08 20:25:20 +00:00
overrides
Remove arraysize.h gcc hack and Chromium override.
2015-09-08 07:57:41 +00:00
p2p
Use of override keyword to fix chromium trybot
2015-09-08 12:10:15 +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
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +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
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
video_engine
Remove unused overuse detection metric (capture jitter).
2015-09-08 08:52:54 +00:00
voice_engine
Remove unnecessary fields from VoE SharedData.
2015-09-08 09:16:11 +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
Add send transports to individual webrtc::Call streams.
2015-08-28 11:07:15 +00:00
BUILD.gn
GN: Fix invalid configuration for Android GCC build.
2015-09-07 10:57:57 +00:00
call.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +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
…
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
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +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
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
Make LoadObserver settable per video send stream. Gives client flexibility and makes the implementation slightly simpler. See discussion in: https://codereview.webrtc.org/1269863005/
2015-09-08 12:13:25 +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
Full stack graphs
2015-09-07 13:04:23 +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.