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Karl Wiberg be57983f4b Rename Maybe to Optional 
And add examples of good and bad usage to the documentation.

R=aluebs@webrtc.org, henrik.lundin@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1432553007 .

Cr-Commit-Position: refs/heads/master@{#10588}
2015-11-10 21:34:32 +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
Rename Maybe to Optional
2015-11-10 21:34:32 +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
Rename Maybe to Optional
2015-11-10 21:34:32 +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
Remove contention between RTCP packets and encoding.
2015-11-10 15:27:21 +00:00
video_engine
Remove contention between RTCP packets and encoding.
2015-11-10 15:27:21 +00:00
voice_engine
Reland "Prevent Opus DTX from generating intermittent noise during silence"
2015-11-10 11:49:32 +00:00
.gitignore
…
audio_receive_stream.h
Re-Land: Implement AudioReceiveStream::GetStats().
2015-10-22 08:49:39 +00:00
audio_send_stream.h
Implement AudioSendStream::GetStats().
2015-10-27 10:35:30 +00:00
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
Improve informative message in codereview.settings.
2015-11-10 10:08:36 +00:00
common_types.cc
Separating voice activity flag from audio level in RtpHeaderExtension.
2015-08-10 13:08:46 +00:00
common_types.h
Adding support for simulcast and spatial layers into VideoQualityTest
2015-11-02 15:23:24 +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 RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
config.cc
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
config.h
Adding support for simulcast and spatial layers into VideoQualityTest
2015-11-02 15:23:24 +00:00
engine_configurations.h
Don't link with audio codecs that we don't use
2015-10-01 04:54:29 +00:00
frame_callback.h
Unify FrameType and VideoFrameType.
2015-10-19 09:39:15 +00:00
libjingle_examples.gyp
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
stream.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
supplement.gypi
Move target_subarch from gyp_webrtc to supplement.gypi
2015-04-16 07:24:23 +00:00
transport.h
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
typedefs.h
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
video_decoder.h
Add DecodedImageCallback::Decoded() function with custom decode time value.
2015-11-10 13:00:45 +00:00
video_encoder.h
Remove redudant encoder rate calls.
2015-10-29 15:30:29 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Remove VideoFrameType aliases for FrameType.
2015-10-23 13:58:27 +00:00
video_receive_stream.h
Fix RTPPayloadRegistry to correctly restore RTX, if a valid mapping exists.
2015-10-14 18:29:56 +00:00
video_renderer.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
video_send_stream.h
Reland of Collecting encode_time_ms for each frame (patchset #1 id:1 of https://codereview.webrtc.org/1383283005/ )
2015-10-07 09:43:25 +00:00
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
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
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.