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2224294c52fc0bb95eedb8187a62bedf1b5eb853
platform-external-webrtc/webrtc
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Karl Wiberg 2224294c52 iSAC: Functions for importing and exporting bandwidth est. info 
They make it possible to send bandwidth estimation info from decoder
to encoder even if they are separate objects (which we want them to be
because multithreading).

R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1208923002.

Cr-Commit-Position: refs/heads/master@{#9535}
2015-07-03 02:04:46 +00:00
..
base
Fix some clang warnings with -Wmissing-braces in WebRTC.
2015-07-02 00:52:18 +00:00
build
iOS HW H264 support.
2015-06-29 21:35:08 +00:00
common_audio
Remove a cast again, after it was shown to worsen Windows perf.
2015-06-29 22:16:48 +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
iSAC: Functions for importing and exporting bandwidth est. info
2015-07-03 02:04:46 +00:00
overrides
Attempt at fixing error on the Chrome Windows FYI bots.
2015-05-25 19:22:23 +00:00
p2p
Remove iostream which causes a new static initializer
2015-06-23 04:54:22 +00:00
sound
Remove henrike@ from OWNERS
2015-04-01 15:08:49 +00:00
system_wrappers
Fix some clang warnings with -Wmissing-braces in WebRTC.
2015-07-02 00:52:18 +00:00
test
Move frame input (ViECapturer) to webrtc/video/.
2015-06-26 04:58:23 +00:00
tools
Pull the Voice Activity Detector out from the AGC
2015-06-25 19:28:55 +00:00
video
Add test for dropping repeated NTP timestamps.
2015-07-02 11:14:50 +00:00
video_engine
Remove decoder-thread instantiation for senders.
2015-07-02 13:32:21 +00:00
voice_engine
Reformat existing code. There should be no functional effects.
2015-06-11 21:31:48 +00:00
.gitignore
…
audio_receive_stream.h
VoE2 API draft
2015-06-08 11:05:07 +00:00
audio_send_stream.h
VoE2 API draft
2015-06-08 11:05:07 +00:00
BUILD.gn
iOS HW H264 support.
2015-06-29 21:35:08 +00:00
call.h
C++11 in-class member initialization in Call configs.
2015-06-11 10:38:46 +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
Parsing of transport wide sequence number rtp extension header.
2015-03-17 14:33:46 +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
experiments.h
…
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
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +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
iOS HW H264 support.
2015-06-29 21:35:08 +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
C++11 in-class member initialization in Call configs.
2015-06-11 10:38:46 +00:00
video_renderer.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
video_send_stream.h
Move frame input (ViECapturer) to webrtc/video/.
2015-06-26 04:58:23 +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
Move frame input (ViECapturer) to webrtc/video/.
2015-06-26 04:58:23 +00:00
webrtc.gyp
Merge video_engine_core into webrtc target.
2015-06-22 05:57:26 +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.