admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc
History
nisse 744494f451 Make the FakeWebRtcVideoCaptureModule class initialize frame data. 
Needed to avoid DrMemory warnings, if the frame is passed to libyuv
AVX assembly functions.

BUG=libyuv:377

Review-Url: https://codereview.webrtc.org/1985693002
Cr-Commit-Position: refs/heads/master@{#12765}
2016-05-17 06:51:11 +00:00
..
api
Surface the IntelligibilityEnhancer on MediaConstraints
2016-05-16 22:32:45 +00:00
audio
Remove SendPacer from ViEEncoder and make sure SendPacer starts at a valid bitrate
2016-05-11 13:01:19 +00:00
base
Makes ECDSA the default certificate to use (generated if no other certificates
2016-05-16 09:45:45 +00:00
build
Reland of New task queueing primitive for async tasks: TaskQueue.
2016-05-14 18:31:49 +00:00
call
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +00:00
common_audio
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +00:00
common_video
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +00:00
examples
Fixes a bug where AppRTC Android Demo crashes with empty roomId.
2016-05-16 14:24:23 +00:00
libjingle
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +00:00
media
Make the FakeWebRtcVideoCaptureModule class initialize frame data.
2016-05-17 06:51:11 +00:00
modules
Surface the IntelligibilityEnhancer on MediaConstraints
2016-05-16 22:32:45 +00:00
p2p
Revert of Increase the stun ping interval. (patchset #5 id:80001 of https://codereview.webrtc.org/1944003002/ )
2016-05-16 21:12:22 +00:00
pc
Adding getParameters/setParameters APIs to RtpReceiver.
2016-05-16 18:40:38 +00:00
sdk
Adding getParameters/setParameters APIs to RtpReceiver.
2016-05-16 18:40:38 +00:00
system_wrappers
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +00:00
test
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +00:00
tools
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +00:00
video
Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #2 id:290001 of https://codereview.webrtc.org/1963413004/ )
2016-05-14 10:18:13 +00:00
voice_engine
Removing some old code which looked like it had to do with NACK handling but in reality did nothing.
2016-05-11 11:05:13 +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
Revert of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} (patchset #2 id:20001 of https://codereview.webrtc.org/1973313002/ )
2016-05-13 17:28:59 +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
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
common.h
#include "webrtc/base/constructormagic.h" where appropriate
2016-04-26 15:14:48 +00:00
config.cc
Change transport sequence number extension strings to specify what revision is implemented.
2016-03-01 15:27:27 +00:00
config.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +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
Remove use_openssl from webrtc
2016-04-18 18:13:17 +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
Update attributes to match gclibc's ansidecl.h
2016-01-08 12:51:47 +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
Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #2 id:290001 of https://codereview.webrtc.org/1963413004/ )
2016-05-14 10:18:13 +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
New UI for AppRTC Android Demo that is easier to use and better follows
2016-05-16 14:21:45 +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
New parser for event log. Manually parse the outermost EventStream to more easily deal with corrupt or partially written logs.
2016-05-13 07:43:04 +00:00
webrtc.gyp
New parser for event log. Manually parse the outermost EventStream to more easily deal with corrupt or partially written logs.
2016-05-13 07:43:04 +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.