zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
Files
20ef654174e245b3a06c9e9045bb97be9acd90cf
platform-external-webrtc/webrtc
History
hbos 20ef654174 RTCCertificate::Expires() and ::HasExpired() implemented using SSLCertificate::CertificateExpirationTime(). 
NOPRESUBMIT=true
BUG=chromium:544894

Review URL: https://codereview.webrtc.org/1494103003

Cr-Commit-Position: refs/heads/master@{#10930}
2015-12-08 09:42:46 +00:00
..
audio
Merge webrtc/video_engine/ into webrtc/video/
2015-12-08 08:09:07 +00:00
base
RTCCertificate::Expires() and ::HasExpired() implemented using SSLCertificate::CertificateExpirationTime().
2015-12-08 09:42:46 +00:00
build
Remove 'video_engine_core_unittests' binary.
2015-12-04 09:59:02 +00:00
call
Merge webrtc/video_engine/ into webrtc/video/
2015-12-08 08:09:07 +00:00
common_audio
Add aecdump support to audioproc_f
2015-11-20 08:11:58 +00:00
common_video
Nuke webrtc/common_video/plane.*.
2015-12-01 13:10:38 +00:00
examples
Implement standalone event tracing in AppRTCDemo.
2015-12-07 22:17:26 +00:00
libjingle
cleanup: get rid of basicdefs.h include
2015-11-13 22:37:43 +00:00
modules
There was an old scaling for CNG 48 kHz in the code, from the time where Audio Coding Module didn't have full 48 kHz support. This CL removes the scaling.
2015-12-08 09:13:08 +00:00
p2p
Add logs when stun or turn host lookup is completed.
2015-12-07 20:06:27 +00:00
sound
Convert usage of ARRAY_SIZE to arraysize.
2015-11-11 07:44:39 +00:00
system_wrappers
Revert of "Create rtc::AtomicInt POD struct." (patchset #3 id:40001 of https://codereview.webrtc.org/1498953002/ )
2015-12-07 22:29:21 +00:00
test
Merge webrtc/video_engine/ into webrtc/video/
2015-12-08 08:09:07 +00:00
tools
Use @webrtc.org addresses for OWNERS.
2015-12-07 18:53:25 +00:00
video
Add test which verifies that the RTP header extensions are set correctly for FEC packets.
2015-12-08 08:10:58 +00:00
voice_engine
Prepare the AudioSendStream to be hooked up to send-side BWE.
2015-12-07 09:26:32 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Add comments about the Audio parts of the public Call API being WIP.
2015-12-03 12:06:31 +00:00
audio_send_stream.h
Prepare the AudioSendStream to be hooked up to send-side BWE.
2015-12-07 09:26:32 +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
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
Revert of Make overuse estimator one dimensional. (patchset #5 id:80001 of https://codereview.webrtc.org/1376423002/ )
2015-11-27 09:02:35 +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
Prepare the AudioSendStream to be hooked up to send-side BWE.
2015-12-07 09:26:32 +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
Merge webrtc/video_engine/ into webrtc/video/
2015-12-08 08:09:07 +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
…
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
Merge webrtc/video_engine/ into webrtc/video/
2015-12-08 08:09:07 +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
Fix Mac ObjC PeerConnection API compilation.
2015-12-03 00:42:41 +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
Merge webrtc/video_engine/ into webrtc/video/
2015-12-08 08:09:07 +00:00
webrtc.gyp
Remove 'video_engine_core_unittests' binary.
2015-12-04 09:59:02 +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.