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ivica c1cc854d54 Fixing perf regression caused by refactoring full stack tests 
Calling CreateCapturer after CreateStreams. The wrong order of calling those methods seems to have caused perf regressions.

Testing has been done here: https://codereview.webrtc.org/1371113004/

BUG=chromium:534220

Review URL: https://codereview.webrtc.org/1394463002

Cr-Commit-Position: refs/heads/master@{#10212}
2015-10-08 10:44:11 +00:00
..
audio
Split webrtc/video into webrtc/{audio,call,video}.
2015-09-25 11:58:39 +00:00
base
Revert of Provide RSA2048 as per RFC (patchset #9 id:200001 of https://codereview.webrtc.org/1329493005/ )
2015-10-08 09:30:21 +00:00
build
Don't link with audio codecs that we don't use
2015-10-01 04:54:29 +00:00
call
Add a PacketOptions struct to webrtc::Transport.
2015-10-02 10:39:40 +00:00
common_audio
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
common_video
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
examples
Use WebRTC logging in MediaCodec JNI code.
2015-10-07 23:51:20 +00:00
libjingle
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
modules
Disable pacer disabling.
2015-10-08 09:44:29 +00:00
p2p
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
sound
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
system_wrappers
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
test
Using different sequence numbers for different SSRCs
2015-10-08 06:44:35 +00:00
tools
Fix minor GYP error in webrtc/tools/internal_tools.gyp
2015-10-07 12:51:40 +00:00
video
Fixing perf regression caused by refactoring full stack tests
2015-10-08 10:44:11 +00:00
video_engine
Disable pacer disabling.
2015-10-08 09:44:29 +00:00
voice_engine
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
audio_receive_stream.h
Align new VoE API with design.
2015-10-01 15:13:46 +00:00
audio_send_stream.h
Align new VoE API with design.
2015-10-01 15:13:46 +00:00
BUILD.gn
Update build files to use webrtc_overrides in Chromium instead of overrides.
2015-09-30 10:49:11 +00:00
call.h
Align new VoE API with design.
2015-10-01 15:13:46 +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
Unify newapi::RtcpMode and RTCPMethod.
2015-10-02 09:37:18 +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
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
Don't link with audio codecs that we don't use
2015-10-01 04:54:29 +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
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
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
Add a PacketOptions struct to webrtc::Transport.
2015-10-02 10:39:40 +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 for framerate reduction for HW encoder.
2015-07-13 23:26:40 +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 kSkipFrame usage.
2015-10-06 12:05:03 +00:00
video_receive_stream.h
Unify newapi::RtcpMode and RTCPMethod.
2015-10-02 09:37:18 +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
Remove AgcManager.
2015-09-29 22:45:23 +00:00
webrtc.gyp
Split webrtc/video into webrtc/{audio,call,video}.
2015-09-25 11:58:39 +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.