zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
Files
ab7073a1e84d59a8acde008a7767f5034050fac0
platform-external-webrtc/webrtc
History
glaznev@webrtc.org ab7073a1e8 Partial implementation of rtc::LogMessage in chromium overrides. 
rtc::LogMessage::LogToDebug used in peerconnection_jni.cc.

BUG=https://crbug.com/412276
R=glaznev@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7186 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-15 19:16:21 +00:00
..
base
Fix ThreadChecker unittests when DCHECK_ALWAYS_ON is defined
2014-09-15 11:19:35 +00:00
build
Set minimum SDK level to 10.7 for Mac and iOS.
2014-09-15 08:02:43 +00:00
common_audio
Add ctors to ChannelBuffer to enable copying on construction.
2014-09-08 20:27:04 +00:00
common_video
Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."
2014-09-11 09:48:30 +00:00
examples
Fix compile error on JDK 1.7.
2014-09-10 12:35:59 +00:00
libjingle/xmllite
Put base tests in webrtc_tests.gyp
2014-09-10 17:28:19 +00:00
modules
audio_processing: Correct sample rate in aec_debug_dump
2014-09-15 13:23:07 +00:00
overrides/webrtc/base
Partial implementation of rtc::LogMessage in chromium overrides.
2014-09-15 19:16:21 +00:00
sound
Put base tests in webrtc_tests.gyp
2014-09-10 17:28:19 +00:00
system_wrappers
Put base tests in webrtc_tests.gyp
2014-09-10 17:28:19 +00:00
test
Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."
2014-09-11 09:48:30 +00:00
tools
Fix printing of error stack in rtcbot when a test fails via test.fail().
2014-09-10 14:35:35 +00:00
video
Re-enable rampup_tests.cc for Android.
2014-09-15 12:27:35 +00:00
video_engine
Mark all virtual overrides in the hierarchy of RtpData and RtpReceiver as such.
2014-09-12 07:42:33 +00:00
voice_engine
Calculating round-trip-time in send-only channel in VoE.
2014-09-11 07:51:53 +00:00
.gitignore
…
BUILD.gn
Fix the RTC+Chromium GN build.
2014-09-09 19:15:33 +00:00
call.h
Network up/down signaling in Call.
2014-09-03 16:17:12 +00:00
common_types.h
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
2014-07-11 13:44:02 +00:00
common.gyp
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
common.h
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
config.cc
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
config.h
Check before send/receive rtp header extensions.
2014-07-20 15:27:35 +00:00
engine_configurations.h
Add boilerplate code for H.264.
2014-07-04 12:42:07 +00:00
experiments.h
Remove no longer used SkipEncodingUnusedStreams.
2014-07-22 07:17:17 +00:00
frame_callback.h
Wire up statistics in video receive stream of new API
2014-02-07 12:06:29 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Consolidate all third party licenses in LICENSE_THIRD_PARTY.
2013-05-05 18:54:10 +00:00
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
…
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
…
supplement.gypi
Roll chromium_revision 289723:291647
2014-08-25 14:16:32 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Add CHECK and friends from Chromium.
2014-08-28 16:28:26 +00:00
video_engine_tests.isolate
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
video_receive_stream.h
Change return value for number of discarded packets to be int.
2014-09-04 07:07:44 +00:00
video_renderer.h
Separate Call API/build files from video_engine/.
2013-10-28 16:32:01 +00:00
video_send_stream.h
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
2014-07-11 13:44:02 +00:00
webrtc_examples.gyp
Fix rm command for class cleanup in r7091
2014-09-06 22:11:28 +00:00
webrtc_perf_tests.isolate
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
webrtc_tests.gypi
Put base tests in webrtc_tests.gyp
2014-09-10 17:28:19 +00:00
webrtc.gyp
Put base tests in webrtc_tests.gyp
2014-09-10 17:28:19 +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.