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be81fa538d915542c0be7a304398aa60b4e3de67
platform-external-webrtc/webrtc
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phoglund be81fa538d Rewrote perform_action_on_all_files to be parallell. 
This speeds up decode for a 720p workload from 25s to 5s on my machine.
I did some benchmarking and it appears we spend the vast majority of
the time on zxing decoding, so now we will bring all processors on the
the machine to bear.

BUG=webrtc:4986

Review URL: https://codereview.webrtc.org/1428433002

Cr-Commit-Position: refs/heads/master@{#10443}
2015-10-29 09:22:51 +00:00
..
audio
system_wrappers: rename interface -> include
2015-10-28 17:17:50 +00:00
base
system_wrappers: rename interface -> include
2015-10-28 17:17:50 +00:00
build
Landmine: clobber to remove out/{Debug,Release}/args.gn
2015-10-27 11:00:00 +00:00
call
system_wrappers: rename interface -> include
2015-10-28 17:17:50 +00:00
common_audio
system_wrappers: rename interface -> include
2015-10-28 17:17:50 +00:00
common_video
system_wrappers: rename interface -> include
2015-10-28 17:17:50 +00:00
examples
Android: Replace EGL14 with EGL10
2015-10-22 23:52:45 +00:00
libjingle
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
modules
Update reference indices according to updates in the RTP payload profile.
2015-10-29 08:45:47 +00:00
p2p
system_wrappers: rename interface -> include
2015-10-28 17:17:50 +00:00
sound
Revert "Add GN Build file for rtc_sound target."
2015-10-26 16:39:21 +00:00
system_wrappers
system_wrappers: rename interface -> include
2015-10-28 17:17:50 +00:00
test
system_wrappers: rename interface -> include
2015-10-28 17:17:50 +00:00
tools
Rewrote perform_action_on_all_files to be parallell.
2015-10-29 09:22:51 +00:00
video
system_wrappers: rename interface -> include
2015-10-28 17:17:50 +00:00
video_engine
system_wrappers: rename interface -> include
2015-10-28 17:17:50 +00:00
voice_engine
system_wrappers: rename interface -> include
2015-10-28 17:17:50 +00:00
.gitignore
…
audio_receive_stream.h
Re-Land: Implement AudioReceiveStream::GetStats().
2015-10-22 08:49:39 +00:00
audio_send_stream.h
Implement AudioSendStream::GetStats().
2015-10-27 10:35:30 +00:00
BUILD.gn
Revert "Add GN Build file for rtc_sound target."
2015-10-26 16:39:21 +00:00
call.h
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
codereview.settings
Update lower-level codereview.settings files.
2015-10-06 21:19:34 +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
Remove VideoFrameType aliases for FrameType.
2015-10-23 13:58:27 +00:00
common.gyp
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
common.h
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
config.cc
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
config.h
Revert of Adding support for simulcast and spatial layers into VideoQualityTest (patchset #10 id:180001 of https://codereview.webrtc.org/1353263005/ )
2015-10-12 13:33:24 +00:00
engine_configurations.h
Don't link with audio codecs that we don't use
2015-10-01 04:54:29 +00:00
frame_callback.h
Unify FrameType and VideoFrameType.
2015-10-19 09:39:15 +00:00
libjingle_examples.gyp
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +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
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
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
typedefs.h
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
video_decoder.h
iOS HW H264 support.
2015-06-29 21:35:08 +00:00
video_encoder.h
Allow encoders to fall back dynamically to software.
2015-10-22 06:54:57 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Remove VideoFrameType aliases for FrameType.
2015-10-23 13:58:27 +00:00
video_receive_stream.h
Fix RTPPayloadRegistry to correctly restore RTX, if a valid mapping exists.
2015-10-14 18:29:56 +00:00
video_renderer.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
video_send_stream.h
Reland of Collecting encode_time_ms for each frame (patchset #1 id:1 of https://codereview.webrtc.org/1383283005/ )
2015-10-07 09:43:25 +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
Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams.
2015-10-16 21:35:11 +00:00
webrtc.gyp
Remove references to libpeerconnection.
2015-10-08 12:41:05 +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.