admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc
History
Alex Glaznev 1a9e2108e3 Initial implementation of Android audio recording error handling. 
BUG=b/34128648
R=henrika@webrtc.org

Review-Url: https://codereview.webrtc.org/2620453004 .
Cr-Commit-Position: refs/heads/master@{#16017}
2017-01-11 21:00:01 +00:00
..
androidjunit
…
api
Reland of: Adding error output param to SetConfiguration, using new RTCError type.
2017-01-11 20:28:30 +00:00
audio
GN: Refactor webrtc_nonparallel_tests and audio_tests to avoid crossing package boundaries.
2017-01-05 14:03:24 +00:00
base
Replace ASSERT(false) by RTC_NOTREACHED().
2017-01-11 13:56:46 +00:00
build
Disable some filesystem tests that don't make sense on iOS.
2017-01-10 11:00:41 +00:00
call
Clean up ramp-up tests and make sure they all pass.
2017-01-09 12:19:24 +00:00
common_audio
Fix for left shift of potentially negative value.
2017-01-10 11:37:20 +00:00
common_video
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
examples
Initial implementation of Android audio recording error handling.
2017-01-11 21:00:01 +00:00
logging
Convert rtc_event_log from webrtc::Clock to rtc::TimeMicros.
2016-12-20 13:03:58 +00:00
media
Reland of: Adding error output param to SetConfiguration, using new RTCError type.
2017-01-11 20:28:30 +00:00
modules
Initial implementation of Android audio recording error handling.
2017-01-11 21:00:01 +00:00
p2p
Replace ASSERT(false) by RTC_NOTREACHED().
2017-01-11 13:56:46 +00:00
pc
Replace ASSERT(false) by RTC_NOTREACHED().
2017-01-11 13:56:46 +00:00
sdk
Update Android instrumentation tests to use JUnit4.
2017-01-11 14:22:56 +00:00
stats
RTCMediaStreamTrackStats.ssrcIds removed.
2017-01-11 12:47:23 +00:00
system_wrappers
Reland of Delete unused code from systeminfo. (patchset #1 id:1 of https://codereview.webrtc.org/2584563004/ )
2017-01-09 09:22:16 +00:00
test
Don't detect a new frame if a previous packet is used in a previous frame.
2017-01-11 10:01:56 +00:00
tools
Add method needed to extract frame capture and arrival timestamps from rtc event logs.
2016-12-20 16:51:52 +00:00
video
Make OverheadObserver::OnOverheadChanged count RTP headers only
2017-01-10 16:58:32 +00:00
voice_engine
Make OverheadObserver::OnOverheadChanged count RTP headers only
2017-01-10 16:58:32 +00:00
.gitignore
…
BUILD.gn
Reland of "Remove webrtc/libjingle/{xmllite,xmpp}"
2017-01-09 07:02:01 +00:00
call.h
…
codereview.settings
…
common_types.cc
…
common_types.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
config.cc
…
config.h
…
DEPS
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
…
video_decoder.h
…
video_encoder.h
…
video_frame.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
video_receive_stream.h
…
video_send_stream.h
…

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.