admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc
History
philipel 536378bf37 Allow FakeNetworkPipe to drop packets in bursts. 
The fake network pipe will still only drop packets at an average rate of
|loss_percent| but in bursts at an average length specified by
|avg_burst_loss_length|.

Also added the flag -avg_burst_loss_length to video loopback.

BUG=

Review-Url: https://codereview.webrtc.org/1995683003
Cr-Commit-Position: refs/heads/master@{#12969}
2016-05-31 10:20:28 +00:00
..
api
Revert of Android: Change camera fps range selection (patchset #4 id:100001 of https://codereview.webrtc.org/2013413002/ )
2016-05-31 08:44:34 +00:00
audio
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
base
Disabled ProfilerTest.TestFunction due to flakiness.
2016-05-30 14:36:27 +00:00
build
Reland "Set defines for Chromium"
2016-05-30 13:27:56 +00:00
call
Take ownership and close the platform file even if we fail to start logging.
2016-05-27 09:22:54 +00:00
common_audio
Reland of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2018553002/ )
2016-05-26 21:48:27 +00:00
common_video
Delete IsMutable and IsExclusive methods.
2016-05-26 13:50:00 +00:00
examples
Reland of Android: Add FramerateRange class (patchset #1 id:1 of https://codereview.webrtc.org/2024573002/ )
2016-05-30 11:00:27 +00:00
libjingle
Remove unused libjingle_xmpphelp target.
2016-05-24 18:33:39 +00:00
media
Moved CreateBuiltinDecoderFactory out to VoEBaseImpl.
2016-05-30 15:11:36 +00:00
modules
AudioDecoder: New method SampleRateHz, + implementations for our codecs
2016-05-31 09:46:28 +00:00
p2p
Handle any number of role conflicts; not just the first.
2016-05-27 20:34:45 +00:00
pc
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
sdk
Revert of Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface. (patchset #2 id:20001 of https://codereview.webrtc.org/2013523002/ )
2016-05-27 13:08:58 +00:00
system_wrappers
Fix iOS GN build and cleanup system_wrappers
2016-05-25 18:37:17 +00:00
test
Allow FakeNetworkPipe to drop packets in bursts.
2016-05-31 10:20:28 +00:00
tools
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +00:00
video
Allow FakeNetworkPipe to drop packets in bursts.
2016-05-31 10:20:28 +00:00
voice_engine
Moved CreateBuiltinDecoderFactory out to VoEBaseImpl.
2016-05-30 15:11:36 +00:00
.gitignore
…
audio_receive_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
audio_send_stream.h
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +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
…
BUILD.gn
Reland "Set defines for Chromium"
2016-05-30 13:27:56 +00:00
call.h
Update the call when the network route changes
2016-04-19 22:41:53 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
…
common_types.h
Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket.
2016-05-03 06:44:11 +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
#include "webrtc/base/constructormagic.h" where appropriate
2016-04-26 15:14:48 +00:00
config.cc
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
config.h
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
DEPS
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Add root owners to webrtc/OWNERS
2016-05-11 12:47:43 +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
Disable libyuv jpeg support on Android
2016-05-17 15:39:15 +00:00
transport.h
…
typedefs.h
Remove runtime NEON detection
2016-05-17 17:56:48 +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
Remove QualityScaler framerate reduction.
2016-04-26 11:37:20 +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
…
video_frame.h
Reland of Delete webrtc::VideoFrame methods buffer and stride. (patchset #1 id:1 of https://codereview.webrtc.org/1983583002/ )
2016-05-17 11:05:51 +00:00
video_receive_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
video_send_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
webrtc_examples.gyp
UI for AppRTC Android demo that doesn't require Design Support Library
2016-05-24 07:18:18 +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
…
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
Revert of Android: Change camera fps range selection (patchset #4 id:100001 of https://codereview.webrtc.org/2013413002/ )
2016-05-31 08:44:34 +00:00
webrtc.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

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.