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sprang@webrtc.org a45cac0fb7 Avoid potential dead lock in StreamStatisticianImpl 
Extract callbacks for rtp/rtcp data, from StreamStatisticianImpl to
ReceiveStatisticsImpl, into separate methods with guards agains having
incorrect lock order.

BUG=2856
R=andresp@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5441 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27 16:22:08 +00:00
..
build
Add new API (webrtc.gyp:webrtc) to merge_libs.gyp.
2014-01-24 10:43:47 +00:00
common_audio
Remove the requirement to call set_sample_rate_hz and friends.
2014-01-07 17:45:09 +00:00
common_video
Add SwapFrame() to VideoSendStreamInput.
2013-12-11 16:26:16 +00:00
examples
Android, WebRTCDemo: fixes crash issue when pressing switch camera button on devices with only one camera.
2014-01-24 18:42:12 +00:00
modules
Avoid potential dead lock in StreamStatisticianImpl
2014-01-27 16:22:08 +00:00
system_wrappers
Removes the remaining uses of the list wrapper class and the list wrapper class.
2014-01-13 22:41:34 +00:00
test
Wire up RTX in VideoReceiveStream.
2014-01-24 09:30:53 +00:00
tools
Now printing less output from compare_videos.py.
2014-01-07 17:59:30 +00:00
video
Drop early packets when not sending in TransportAdapter.
2014-01-27 13:03:02 +00:00
video_engine
Fix deadlock on register/unregister observer while there is a an going callback.
2014-01-23 23:09:25 +00:00
voice_engine
Output logs to stderr from voe_cmd_test by default.
2014-01-21 22:17:43 +00:00
.gitignore
…
call.h
Remove outdated DestroyVideoSendStream comment.
2014-01-07 12:27:22 +00:00
common_types.h
Remove empty VideoCodecGeneric struct.
2014-01-14 12:55:59 +00:00
common.h
…
config.h
Wire up RTX in VideoReceiveStream.
2014-01-24 09:30:53 +00:00
engine_configurations.h
Adds a new voice engine warning for the typing noise off state.
2013-09-17 18:09:20 +00:00
experiments.h
Adds support for sending redundant payloads over RTX.
2013-12-04 10:24:26 +00:00
frame_callback.h
Implement and test EncodedImageCallback in new ViE API.
2013-11-26 11:41:59 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
supplement.gypi
Removes script for generating supplement.gypi also adds git ignore for tools/gn.
2014-01-21 15:54:56 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Correctly define OVERRIDE when building with g++ 4.7 and C++11 support
2013-12-10 19:20:46 +00:00
video_engine_tests.isolate
Merge metrics_unittests into video_engine_tests.
2013-12-13 14:31:47 +00:00
video_receive_stream.h
Wire up RTX in VideoReceiveStream.
2014-01-24 09:30:53 +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
Wire up RTX in VideoReceiveStream.
2014-01-24 09:30:53 +00:00
webrtc_examples.gyp
Android example apps: fixes issue where useful failure information was suppressed.
2014-01-21 19:03:51 +00:00
webrtc_perf_tests.isolate
Move realtime tests to webrtc_perf_tests.
2013-12-13 12:48:05 +00:00
webrtc_tests.gypi
Adding NetEq performance test to webrtc_perf_tests
2014-01-10 08:24:04 +00:00
webrtc.gyp
Integrate fake_network_pipe into direct_transport.
2013-12-18 20:28:25 +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.