generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
webrtc/webrtc
History
kwiberg@webrtc.org 00ba1a7dfd Move the AudioDecoder interface out of NetEq 
It belongs with the codecs, next to the AudioEncoder interface.

R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7798 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 14:23:23 +00:00
..
base
Use c++11 features in webrtc/base/network.cc as a test to see if we can use them.
2014-12-01 23:18:27 +00:00
build
Add Android test runner script for WebRTC.
2014-12-03 09:57:08 +00:00
common_audio
Remove -flax-vector-conversions flag for ARM NEON building.
2014-12-02 19:36:14 +00:00
common_video
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
examples
Update Android projects to API level 21.
2014-10-31 23:26:10 +00:00
libjingle
move xmpp and p2p to webrtc
2014-10-28 22:20:11 +00:00
modules
Move the AudioDecoder interface out of NetEq
2014-12-03 14:23:23 +00:00
overrides
webrtc/overrides: add OWNERS-file.
2014-09-17 08:04:28 +00:00
p2p
TurnPort should ignore STUN binding reponses when using shared socket.
2014-12-03 02:17:07 +00:00
sound
rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation.
2014-10-01 16:33:03 +00:00
system_wrappers
Add framerate for complete received frames to histogram stats:
2014-11-28 10:17:13 +00:00
test
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
tools
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
video
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
video_engine
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
voice_engine
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
.gitignore
…
BUILD.gn
Enabling building with NEON on ARM64
2014-11-26 17:01:40 +00:00
call.h
Implement settable min/start/max bitrates in Call.
2014-11-25 14:03:34 +00:00
common_types.h
Add receive bitrates to histogram stats:
2014-11-27 07:38:56 +00:00
common.gyp
…
common.h
…
config.cc
Log formatting fix for VideoEncoderConfig.
2014-11-06 09:35:08 +00:00
config.h
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
engine_configurations.h
Add VP9 codec to VCM and vie_auto_test.
2014-11-01 06:10:48 +00:00
experiments.h
…
frame_callback.h
…
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
supplement.gypi
…
transport.h
…
typedefs.h
Annotate COMPILE_ASSERT with __attribute__((unused)).
2014-11-17 13:47:38 +00:00
video_decoder.h
Add support for VP9 in webrtc::Call and video_loopback.
2014-11-04 19:41:15 +00:00
video_encoder.h
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
video_receive_stream.h
Wire up bandwidth stats to the new API and webrtcvideoengine2.
2014-11-05 14:05:29 +00:00
video_renderer.h
…
video_send_stream.h
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
webrtc_examples.gyp
Add macros and APIs for webrtc histograms.
2014-10-23 12:57:56 +00:00
webrtc_perf_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
webrtc_tests.gypi
Put send-side bwe probing under finch experiment.
2014-11-10 13:55:16 +00:00
webrtc.gyp
Add VP9 codec to VCM and vie_auto_test.
2014-11-01 06:10:48 +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.