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webrtc_m130/webrtc
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danilchap b1ac203480 Introduce helper class NtpTime 
Seconds and fractions parts of the ntp time presented with two values, but used as one.
This helper structure can make that use more clear.
(initially introduced into rtp_rtcp as https://codereview.webrtc.org/1435833003)

BUG=webrtc:5260

Review URL: https://codereview.webrtc.org/1482593002

Cr-Commit-Position: refs/heads/master@{#10814}
2015-11-26 17:01:18 +00:00
..
audio
Open backdoor in VoiceEngineImpl to get at the actual voe::Channel objects from an ID.
2015-11-25 16:16:57 +00:00
base
Clean up PlatformThread.
2015-11-26 16:45:57 +00:00
build
iOS: Set enable_protobuf=1 by default.
2015-11-20 08:37:50 +00:00
call
Rewrote pacer and bandwidth UMA stats.
2015-11-26 14:36:55 +00:00
common_audio
Add aecdump support to audioproc_f
2015-11-20 08:11:58 +00:00
common_video
Clean up PlatformThread.
2015-11-26 16:45:57 +00:00
examples
Add option to capture to texture in AppRTCDemo for Android.
2015-11-24 10:13:34 +00:00
libjingle
cleanup: get rid of basicdefs.h include
2015-11-13 22:37:43 +00:00
modules
Clean up PlatformThread.
2015-11-26 16:45:57 +00:00
p2p
Destroy a Connection if a CreatePermission request fails.
2015-11-25 17:00:12 +00:00
sound
Convert usage of ARRAY_SIZE to arraysize.
2015-11-11 07:44:39 +00:00
system_wrappers
Introduce helper class NtpTime
2015-11-26 17:01:18 +00:00
test
Clean up PlatformThread.
2015-11-26 16:45:57 +00:00
tools
Remove webrtc/test/channel_transport/include
2015-11-11 09:34:35 +00:00
video
Clean up PlatformThread.
2015-11-26 16:45:57 +00:00
video_engine
Clean up PlatformThread.
2015-11-26 16:45:57 +00:00
voice_engine
Clean up PlatformThread.
2015-11-26 16:45:57 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Re-Land: Implement AudioReceiveStream::GetStats().
2015-10-22 08:49:39 +00:00
audio_send_stream.h
Move some send stream configuration into webrtc::AudioSendStream.
2015-11-16 15:34:59 +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
Add fuzzing of VP8 QP parsing.
2015-11-26 13:52:28 +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
Make overuse estimator one dimensional.
2015-11-26 14:35:10 +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
Move some send stream configuration into webrtc::AudioSendStream.
2015-11-16 15:34:59 +00:00
engine_configurations.h
Remove VIDEOCODEC_* from engine_configurations.h.
2015-11-24 12:56:05 +00:00
frame_callback.h
Unify FrameType and VideoFrameType.
2015-10-19 09:39:15 +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
…
stream.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
supplement.gypi
Roll chromium_revision 68cf0b8..aa8e58a (361406:361601)
2015-11-25 21:20:11 +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
common_video: rename interface -> include
2015-11-16 21:52:31 +00:00
video_receive_stream.h
Require negotiation to send transport cc feedback over RTCP.
2015-11-21 02:05:53 +00:00
video_renderer.h
Remove frame time scheduing in IncomingVideoStream
2015-11-25 02:08:03 +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 webrtc/examples/android/media_demo.
2015-11-11 12:11:21 +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
Revert of Created a test that reports the statistics for the duration of APM stream processing API calls. (patchset #15 id:280001 of https://codereview.webrtc.org/1436553004/ )
2015-11-25 20:27:46 +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.