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aleloi 616df1e95c Added a level indicator to new mixer. 
Added a level indicator to the new mixer. The level indicator is
webrtc::voe::AudioLevel. It computes the current audio level, which is
used all the way up to peerconnection.

This is part of the project to rewrite the old conference mixer and
output mixer.

NOTRY=True

Review-Url: https://codereview.webrtc.org/2230823004
Cr-Commit-Position: refs/heads/master@{#13878}
2016-08-24 08:17:20 +00:00
..
androidjunit
Re-order and remove unused Java imports.
2016-08-02 13:44:19 +00:00
api
Fix AppRTC Android Demo GN build when is_component_build=true.
2016-08-24 07:48:30 +00:00
audio
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
2016-08-16 09:40:59 +00:00
base
Migrated GN target :g711_test
2016-08-23 15:08:30 +00:00
build
MB: Flip Android bots to GN by default.
2016-08-24 07:34:59 +00:00
call
Do not update stats for WebRTC.Call.EstimatedSendBitrateInKbps if we are not sending video.
2016-08-23 20:00:21 +00:00
common_audio
WebRtcSpl_SynthesisQMF: Fix UBSan fuzzer bug (left shift of negative value)
2016-08-18 10:17:10 +00:00
common_video
Revert of Add pps id and sps id parsing to the h.264 depacketizer. (patchset #5 id:80001 of https://codereview.webrtc.org/2238253002/ )
2016-08-22 12:34:43 +00:00
examples
Remove outdated symlink
2016-08-24 07:49:33 +00:00
libjingle
GN: Update xmpp and p2p to match Chromium build
2016-06-18 11:23:08 +00:00
media
Refactor WebRtcVideoCapturer.
2016-08-23 12:50:20 +00:00
modules
Added a level indicator to new mixer.
2016-08-24 08:17:20 +00:00
p2p
Add logs and small change in BasicPortAllocator.
2016-08-23 22:47:45 +00:00
pc
Some cleanup in BaseChannel RTCP mux code.
2016-08-22 23:00:37 +00:00
sdk
Add support for more resolutions on iOS/macOS
2016-08-19 08:24:49 +00:00
system_wrappers
GN: Fix test_support_unittests and MIPS compile issue.
2016-08-23 11:52:19 +00:00
test
GN: Fix test_support_unittests and MIPS compile issue.
2016-08-23 11:52:19 +00:00
tools
Define a protobuf format for representing plots. Add code to convert the C-representation generated by the RtcEventLog analysis tool, to the new protobuf format.
2016-08-24 01:15:31 +00:00
video
Fixing TSan data race warning in video end-to-end tests.
2016-08-23 01:14:21 +00:00
voice_engine
Added a level indicator to new mixer.
2016-08-24 08:17:20 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume().
2016-06-17 15:30:58 +00:00
audio_send_stream.h
Variable audio bitrate.
2016-07-26 11:44:12 +00:00
audio_sink.h
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
audio_state.h
Add comments about the Audio parts of the public Call API being WIP.
2015-12-03 12:06:31 +00:00
BUILD.gn
GN: Add "//build/config/sanitizers:deps" as a dependency to executable targets.
2016-08-19 09:11:15 +00:00
call.h
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
2016-08-16 09:40:59 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
FileWrapper[Impl] modifications and actually remove the "Impl" class.
2016-06-15 17:30:18 +00:00
common_types.h
Reland of actor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131913003/ )
2016-07-13 16:11:38 +00:00
common.gyp
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +00:00
common.h
New module for the adaptive level controlling functionality in the audio processing module
2016-06-29 22:26:19 +00:00
config.cc
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
config.h
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
2016-08-16 09:40:59 +00:00
DEPS
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
2016-06-13 14:35:01 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Add stefan@ to webrtc/OWNERS.
2016-08-10 15:49:59 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
rtc_unittests.isolate
…
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +00:00
transport.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Add EncodedImageCallback::OnEncodedImage().
2016-08-03 00:46:47 +00:00
video_engine_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
video_engine_tests.isolate
Add audio streams to CallTest and a first A/V call test.
2016-01-07 16:43:31 +00:00
video_frame.h
Add TODO comments on deprecated VideoFrame methods.
2016-06-22 15:47:52 +00:00
video_receive_stream.h
Add periodic logging of video stats.
2016-08-11 15:41:26 +00:00
video_send_stream.h
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
2016-08-16 09:40:59 +00:00
webrtc_examples.gyp
Add GN targets for AppRTC Demo on Android.
2016-06-29 12:55:12 +00:00
webrtc_nonparallel_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
Add an HD resolution perf test.
2016-08-11 14:01:03 +00:00
webrtc_tests.gypi
GN: Fix test_support_unittests and MIPS compile issue.
2016-08-23 11:52:19 +00:00
webrtc.gyp
Remove webrtc_all target
2016-06-14 09:09:24 +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.