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aleloi 30be5d7cf4 Updated mixer unittests and fixed a related bug in the new mixer. 
Changes to the mixer unittests:

Removed the tests related to the former 'OutputMixer', as it's going
to be removed. Removed incorrect comparison tests with the old mixer
because doing identical mixing decisions with the old mixer proved
unviable.

When the new mixer went from kMaximumAmountOfMixedAudioSources in the
last iteration to kMaximumAmountOfMixedAudioSources+1, it could hit an
RTC_NOTREACHED(); Added fix to mixer and test
AudioMixer.RampedOutSourcesShouldNotBeMarkedMixed that covers that
case.

NOTRY=True

Review-Url: https://codereview.webrtc.org/2253153004
Cr-Commit-Position: refs/heads/master@{#13880}
2016-08-24 08:38:50 +00:00
..
androidjunit
Re-order and remove unused Java imports.
2016-08-02 13:44:19 +00:00
api
RTCStats and RTCStatsReport added (webrtc/stats).
2016-08-24 08:33:19 +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Remove outdated symlink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
Updated mixer unittests and fixed a related bug in the new mixer.
2016-08-24 08:38:50 +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
stats
RTCStats and RTCStatsReport added (webrtc/stats).
2016-08-24 08:33:19 +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
RTCStats and RTCStatsReport added (webrtc/stats).
2016-08-24 08:33:19 +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
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
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
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
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.