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Danil Chapovalov 895a2d24e9 Reenable EndToEndTest.ReceivedFecPacketsNotNacked 
Test adjusted to be for VP8

BUG=webrtc:4328
R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/2017713002 .

Cr-Commit-Position: refs/heads/master@{#12946}
2016-05-27 12:12:19 +00:00
..
api
VideoCapturerAndroid: Replace static create() with ctor
2016-05-27 09:35:14 +00:00
audio
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
base
Lower event-tracing log-thread priority.
2016-05-27 09:11:52 +00:00
build
iOS: Update MB config for moved iOS GN bots.
2016-05-26 09:49:25 +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
Android: Add FramerateRange class
2016-05-27 08:36:01 +00:00
libjingle
Remove unused libjingle_xmpphelp target.
2016-05-24 18:33:39 +00:00
media
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
modules
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
p2p
Move the ICE state transition ASSERTS to a lower level.
2016-05-26 23:08:31 +00:00
pc
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
sdk
Fix iOS GN build and cleanup system_wrappers
2016-05-25 18:37:17 +00:00
system_wrappers
Fix iOS GN build and cleanup system_wrappers
2016-05-25 18:37:17 +00:00
test
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
tools
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +00:00
video
Reenable EndToEndTest.ReceivedFecPacketsNotNacked
2016-05-27 12:12:19 +00:00
voice_engine
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 20:03:13 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
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
Add comments about the Audio parts of the public Call API being WIP.
2015-12-03 12:06:31 +00:00
BUILD.gn
If building a fuzzer, define WEBRTC_UNSAFE_FUZZER_MODE.
2016-05-26 08:03:10 +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
Separating voice activity flag from audio level in RtpHeaderExtension.
2015-08-10 13:08:46 +00:00
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
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
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
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
supplement.gypi
Disable libyuv jpeg support on Android
2016-05-17 15:39:15 +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
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
Add audio streams to CallTest and a first A/V call test.
2016-01-07 16:43:31 +00:00
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
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
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Split TaskQueue out into a separate build target.
2016-05-24 18:40:44 +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.