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platform-external-webrtc/webrtc
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Sergey Ulanov 8eb76ff32a Make SHA1 computation thread-safe. 
Previously SHA1Transform() kept a static buffer. As result SHA1 was not
always computed correctly when running that code in parallel on multiple
threads. That was causing spurious messages about invalid Message
Integrity attribute when running some tests in chromoting.

R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/57379004

Cr-Commit-Position: refs/heads/master@{#9238}
2015-05-20 18:25:21 +00:00
..
base
Make SHA1 computation thread-safe.
2015-05-20 18:25:21 +00:00
build
Remove unnecessary dependencies for voe when building with include_internal_audio_device==0.
2015-05-20 07:44:23 +00:00
common_audio
Clear ARM NEON flag
2015-05-20 05:20:04 +00:00
common_video
Move IncomingVideoFrames to common_video/.
2015-05-13 11:28:04 +00:00
examples/android/media_demo
Remove rtp_rtcp/ dump functionality.
2015-05-20 12:10:56 +00:00
libjingle
GYP variables for building expat, icu, libsrtp, usrsctp
2015-04-27 12:38:37 +00:00
modules
Ensures that RECORD_AUDIO permission is required to start recording.
2015-05-20 14:08:44 +00:00
overrides
Further updates to fix libjingle logging.
2015-05-14 02:18:15 +00:00
p2p
Allow setting maximum protocol version for SSL stream adapters.
2015-05-20 10:48:24 +00:00
sound
Remove henrike@ from OWNERS
2015-04-01 15:08:49 +00:00
system_wrappers
Move SetCurrentThreadName to platform_thread.* in rtc_base_approved,
2015-05-18 11:50:31 +00:00
test
Remove VideoEngine interfaces.
2015-05-12 14:51:08 +00:00
tools
Disables mic bump-up level if not built with chromium
2015-04-22 04:39:47 +00:00
video
Add HW fallback option to software encoding.
2015-05-19 21:09:17 +00:00
video_engine
Remove rtp_rtcp/ dump functionality.
2015-05-20 12:10:56 +00:00
voice_engine
Remove rtp_rtcp/ dump functionality.
2015-05-20 12:10:56 +00:00
.gitignore
…
audio_receive_stream.h
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
BUILD.gn
Clear ARM NEON flag
2015-05-20 05:20:04 +00:00
call.h
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
codereview.settings
Add codereview.settings to the /webrtc subdirectory
2014-12-05 13:43:35 +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
…
config.cc
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
config.h
Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
2015-05-11 10:44:20 +00:00
engine_configurations.h
Remove VideoEngine interfaces.
2015-05-12 14:51:08 +00:00
experiments.h
Remove no longer used SkipEncodingUnusedStreams.
2014-07-22 07:17:17 +00:00
frame_callback.h
Changing include guard in frame_callback.h.
2015-02-03 14:51:39 +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
Add HW fallback option to software decoding.
2015-05-18 17:41:50 +00:00
video_encoder.h
Add HW fallback option to software encoding.
2015-05-19 21:09:17 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Reland "Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection.""
2015-04-09 11:44:19 +00:00
video_receive_stream.h
Use IncomingVideoStream in VideoReceiveStream.
2015-04-24 13:15:40 +00:00
video_renderer.h
Use VideoReceiveStream as an ExternalRenderer.
2015-02-09 15:15:24 +00:00
video_send_stream.h
Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.""
2015-03-18 09:51:44 +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
Add HW fallback option to software encoding.
2015-05-19 21:09:17 +00:00
webrtc.gyp
Rename GYP and GN targets for video capture+render.
2015-02-11 07:47:47 +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.