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bjornv@webrtc.org 63da1dd972 audio_processing: Now records mic volume level also when using new AGC 
Previously only mic level calculated by the legacy agc was logged in aecdebug dumps.
Now we log it for any agc.
In addition, it is now possible to turn on and off debug recording in the test tool voe_cmd_test.

BUG=4274
TESTED=verified using voe_cmd_test
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39839004

Cr-Commit-Position: refs/heads/master@{#8274}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8274 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-06 19:44:46 +00:00
..
base
Temporarily disable a couple of ThreadChecker tests on Mac.
2015-02-06 16:39:18 +00:00
build
Move use of DEPTH into build_with_chromium==1.
2015-01-30 14:55:20 +00:00
common_audio
Move channel_buffer.{h,cc} to common_audio.
2015-01-28 19:57:44 +00:00
common_video
Revert "Cleanup: unify rotation to be enum based instead of int for degree."
2015-02-06 12:58:46 +00:00
examples
Add arraysize() macro from Chromium, and make use of it in a few places.
2015-01-28 21:37:13 +00:00
libjingle
Enable Clang warning implicit-fallthrough and annotate the code.
2015-01-28 18:38:13 +00:00
modules
audio_processing: Now records mic volume level also when using new AGC
2015-02-06 19:44:46 +00:00
overrides
Remove COMPILE_ASSERT and use static_assert everywhere
2015-01-14 10:51:54 +00:00
p2p
VirtualSocketServer out-of-order issue with closing TCP sockets
2015-02-06 16:33:47 +00:00
sound
rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation.
2014-10-01 16:33:03 +00:00
system_wrappers
Temporarily change ThreadPosix to CHECK (crash) if we ever spend more than 30 seconds waiting for thread shutdown. There are cases on build bots where it looks like we're hitting this problem, but reproducing locally has been a struggle.
2015-02-06 19:27:12 +00:00
test
Fix the value of the first byte of nal unit
2015-02-05 17:56:44 +00:00
tools
Re-land "Remove <(webrtc_root) from source file entries."
2015-01-29 14:30:41 +00:00
video
Add tracing for slow paths in new video API.
2015-01-29 12:33:42 +00:00
video_engine
Implementing a packet router class, used to route RTP packets to the
2015-02-06 13:10:39 +00:00
voice_engine
audio_processing: Now records mic volume level also when using new AGC
2015-02-06 19:44:46 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
BUILD.gn
Make it easier to use external libyuv + cleanup GYP files.
2015-01-26 19:17:26 +00:00
call.h
Use int64_t more consistently for times, in particular for RTT values.
2015-01-12 21:51:21 +00:00
codereview.settings
Add codereview.settings to the /webrtc subdirectory
2014-12-05 13:43:35 +00:00
common_types.h
Add method for incrementing RtpPacketCounter. Removes duplicate code.
2015-02-04 08:35:21 +00:00
common.gyp
Make it easier to use external libyuv + cleanup GYP files.
2015-01-26 19:17:26 +00:00
common.h
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
config.cc
Revert 8028 "Support associated payload type when registering Rt..."
2015-01-09 20:22:46 +00:00
config.h
Revert 8028 "Support associated payload type when registering Rt..."
2015-01-09 20:22:46 +00:00
engine_configurations.h
Add VP9 codec to VCM and vie_auto_test.
2014-11-01 06:10:48 +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
Consolidate all third party licenses in LICENSE_THIRD_PARTY.
2013-05-05 18:54:10 +00:00
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
…
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
…
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
supplement.gypi
Roll chromium_revision 289723:291647
2014-08-25 14:16:32 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Enable Clang warning implicit-fallthrough and annotate the code.
2015-01-28 18:38:13 +00:00
video_decoder.h
Add support for VP9 in webrtc::Call and video_loopback.
2014-11-04 19:41:15 +00:00
video_encoder.h
Use int64_t more consistently for times, in particular for RTT values.
2015-01-12 21:51:21 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
video_receive_stream.h
Log configs when creating video streams in Call.
2015-01-15 10:09:39 +00:00
video_renderer.h
Separate Call API/build files from video_engine/.
2013-10-28 16:32:01 +00:00
video_send_stream.h
Log configs when creating video streams in Call.
2015-01-15 10:09:39 +00:00
webrtc_examples.gyp
Move internal capture+render to build_with_chromium==0 condition
2015-01-20 11:40:45 +00:00
webrtc_perf_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
webrtc_tests.gypi
Re-land "Remove <(webrtc_root) from source file entries."
2015-01-29 14:30:41 +00:00
webrtc.gyp
Make it easier to use external libyuv + cleanup GYP files.
2015-01-26 19:17:26 +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.