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webrtc_m130/webrtc
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peah 48407f7122 Changed queue implementation to the proposed vector-based solution. 
Added unit tests.

BUG=webrtc:5099
TBR=hlundin-webrtc

Review URL: https://codereview.webrtc.org/1398473004

Cr-Commit-Position: refs/heads/master@{#10562}
2015-11-09 13:24:56 +00:00
..
audio
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
2015-11-06 23:34:58 +00:00
base
Pass audio to AudioEncoder::Encode() in an ArrayView
2015-11-06 09:21:39 +00:00
build
Remove ICU usage from jni_helpers.cc.
2015-11-06 21:56:11 +00:00
call
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
2015-11-06 23:34:58 +00:00
common_audio
Changed queue implementation to the proposed vector-based solution.
2015-11-09 13:24:56 +00:00
common_video
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
examples
Remove ICU usage from jni_helpers.cc.
2015-11-06 21:56:11 +00:00
libjingle
Switch usage of _DEBUG macro to NDEBUG.
2015-10-30 23:08:54 +00:00
modules
Added unittest of the locking functionality in the audio processing module
2015-11-09 11:13:25 +00:00
p2p
Stop a session when a new connection becomes writable.
2015-11-04 20:07:49 +00:00
sound
Fix chromium-style warnings in webrtc/sound/.
2015-10-29 11:50:15 +00:00
system_wrappers
Remove old system_wrappers event_tracer.h.
2015-11-06 10:16:56 +00:00
test
Change to use local Random object instead of global rand() in the RtcEventLog unit test.
2015-11-06 13:14:01 +00:00
tools
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
video
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
2015-11-06 23:34:58 +00:00
video_engine
Removed vie_defines.h
2015-11-06 12:33:56 +00:00
voice_engine
Removing trace checks in VoETestManager.
2015-11-04 15:11:52 +00:00
.gitignore
…
audio_receive_stream.h
…
audio_send_stream.h
…
audio_state.h
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
2015-11-06 23:34:58 +00:00
BUILD.gn
Remove unused crypto.gni import.
2015-11-04 22:44:04 +00:00
call.h
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
2015-11-06 23:34:58 +00:00
codereview.settings
…
common_types.cc
…
common_types.h
Adding support for simulcast and spatial layers into VideoQualityTest
2015-11-02 15:23:24 +00:00
common.gyp
…
common.h
…
config.cc
…
config.h
Adding support for simulcast and spatial layers into VideoQualityTest
2015-11-02 15:23:24 +00:00
engine_configurations.h
…
frame_callback.h
…
libjingle_examples.gyp
…
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
…
stream.h
…
supplement.gypi
…
transport.h
…
typedefs.h
…
video_decoder.h
…
video_encoder.h
Remove redudant encoder rate calls.
2015-10-29 15:30:29 +00:00
video_engine_tests.isolate
…
video_frame.h
…
video_receive_stream.h
…
video_renderer.h
…
video_send_stream.h
…
webrtc_examples.gyp
Remove ICU usage from jni_helpers.cc.
2015-11-06 21:56:11 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
2015-11-06 23:34:58 +00:00
webrtc.gyp
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
2015-11-06 23:34:58 +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.