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Bjorn Volcker 40a6d593d2 audio_processing/tests: Adds a flag to unpack input data to text file 
For quick and easy aecdump verifiation storing data as text speeds up the issue tracking process, since anyone can simply view values like mic volume.

BUG=4609
TESTED=verified unpacking an aecdump with flag --txt stores that data in text files
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50849004

Cr-Commit-Position: refs/heads/master@{#9142}
2015-05-06 08:51:47 +00:00
..
base
rtc::Buffer: Remove backwards compatibility band-aids
2015-05-04 12:54:56 +00:00
build
GYP variables for building expat, icu, libsrtp, usrsctp
2015-04-27 12:38:37 +00:00
common_audio
Add 3-band filter-bank implementation
2015-04-27 18:34:16 +00:00
common_video
Remove i420_video_frame.h from common_video.gyp
2015-04-10 12:40:21 +00:00
examples/android
Use short include paths for icu headers.
2015-04-23 06:58:02 +00:00
libjingle
GYP variables for building expat, icu, libsrtp, usrsctp
2015-04-27 12:38:37 +00:00
modules
audio_processing/tests: Adds a flag to unpack input data to text file
2015-05-06 08:51:47 +00:00
overrides
Cleanup: Remove DISALLOW_EVIL_CONSTRUCTORS macro.
2015-04-04 23:56:56 +00:00
p2p
Don't crash if SetRemoteDescription is called first with BundlePolicy=max-bundle.
2015-04-30 17:56:21 +00:00
sound
Remove henrike@ from OWNERS
2015-04-01 15:08:49 +00:00
system_wrappers
Move rtc::AtomicOps to webrtc/base/atomicops.h.
2015-04-30 12:16:14 +00:00
test
Use rtc::CriticalSection in webrtc/video/.
2015-05-01 14:25:53 +00:00
tools
Disables mic bump-up level if not built with chromium
2015-04-22 04:39:47 +00:00
video
Remove ViECodec usage in VideoSendStream.
2015-05-05 13:16:40 +00:00
video_engine
Remove ViECodec usage in VideoSendStream.
2015-05-05 13:16:40 +00:00
voice_engine
VoE: apply new style guide on VoE interfaces and their implementations
2015-05-04 12:15:41 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
audio_receive_stream.h
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
BUILD.gn
Let Chromium declare the mips_dsp_rev build variable.
2015-03-04 09:51:17 +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
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
config.cc
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
config.h
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
engine_configurations.h
Delete all codec-specific subclasses of ACMGenericCodec
2015-02-23 09:26:51 +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
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
supplement.gypi
Move target_subarch from gyp_webrtc to supplement.gypi
2015-04-16 07:24:23 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
2015-03-04 13:04:54 +00:00
video_decoder.h
Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*."
2015-03-16 13:48:18 +00:00
video_encoder.h
Dynamic resolution change for VP8 HW encode.
2015-04-21 22:29:53 +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
GYP variables for building expat, icu, libsrtp, usrsctp
2015-04-27 12:38:37 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Add a BWE fairness test.
2015-04-16 18:20:26 +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.