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Niklas Enbom b4c5eaa0d6 Fix a time control bug, that the VCMReceiver::FrameForDecoding may over sleep. 
Remark: a unit test to verify VCMReiceiver::FrameForDecoding will be in a separate CL.

BUG=4726
R=stefan@webrtc.org, wtc@chromium.org

Review URL: https://webrtc-codereview.appspot.com/53619004

Cr-Commit-Position: refs/heads/master@{#9362}
2015-06-03 16:34:31 +00:00
..
base SSL_set_read_ahead no longer needed with BoringSSL. 2015-06-02 21:07:50 +00:00
build Roll chromium_revision ccef3cb..7779e7d (331232:332119) 2015-06-01 09:49:28 +00:00
common_audio Add LappedTransform accessors. 2015-05-29 01:01:43 +00:00
common_video Revert "Convert native handles to buffers before encoding." 2015-06-02 13:04:31 +00:00
examples/android/media_demo VoE: Remove unused interfaces 2015-05-26 08:25:00 +00:00
libjingle VoE: Initialize WebRtcVoiceMediaChannel with AudioOptions during creation 2015-05-29 13:05:52 +00:00
modules Fix a time control bug, that the VCMReceiver::FrameForDecoding may over sleep. 2015-06-03 16:34:31 +00:00
overrides Attempt at fixing error on the Chrome Windows FYI bots. 2015-05-25 19:22:23 +00:00
p2p Make maximum SSL version configurable through PeerConnectionFactory::Options 2015-05-29 07:40:51 +00:00
sound Remove henrike@ from OWNERS 2015-04-01 15:08:49 +00:00
system_wrappers Document the time unit in EventWrapper. 2015-06-02 00:33:51 +00:00
test Revert "Convert native handles to buffers before encoding." 2015-06-02 13:04:31 +00:00
tools PRESUBMIT: Improve PyLint check and add GN format check. 2015-05-25 10:55:50 +00:00
video Revert "Convert native handles to buffers before encoding." 2015-06-02 13:04:31 +00:00
video_engine Revert "Convert native handles to buffers before encoding." 2015-06-02 13:04:31 +00:00
voice_engine Add options for NetEq fast accelerate mode through libjingle 2015-06-01 08:29:55 +00:00
.gitignore .gitignore: Add *.mk, created as part of ChromiumOS build 2013-01-04 21:25:42 +00:00
audio_receive_stream.h Add AudioReceiveStream to Call API. 2015-04-29 13:24:10 +00:00
BUILD.gn PRESUBMIT: Improve PyLint check and add GN format check. 2015-05-25 10:55:50 +00:00
call.h Add AudioReceiveStream to Call API. 2015-04-29 13:24:10 +00:00
codereview.settings Add codereview.settings to the /webrtc subdirectory 2014-12-05 13:43:35 +00:00
common_types.cc Parsing of transport wide sequence number rtp extension header. 2015-03-17 14:33:46 +00:00
common_types.h Parsing of transport wide sequence number rtp extension header. 2015-03-17 14:33:46 +00:00
common.gyp Fix style violations in common_types.h and config.h 2015-02-26 14:01:28 +00:00
common.h Make Config::default_value leak instead of having an exit-time destructor. 2015-05-23 00:50:33 +00:00
config.cc Add AudioReceiveStream to Call API. 2015-04-29 13:24:10 +00:00
config.h Add options for NetEq fast accelerate mode through libjingle 2015-06-01 08:29:55 +00:00
engine_configurations.h Remove VideoEngine interfaces. 2015-05-12 14:51:08 +00:00
experiments.h Remove no longer used SkipEncodingUnusedStreams. 2014-07-22 07:17:17 +00:00
frame_callback.h Rename I420VideoFrame to VideoFrame. 2015-05-30 00:21:56 +00:00
LICENSE Move src/ -> webrtc/ 2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY Add the Ooura FFT to RealFourier. 2015-03-19 20:07:43 +00:00
OWNERS GN: Add BUILD.gn files + kjellander to OWNERS 2014-06-23 19:21:07 +00:00
PATENTS Move src/ -> webrtc/ 2012-10-22 18:19:23 +00:00
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 Move src/ -> webrtc/ 2012-10-22 18:19:23 +00:00
rtc_unittests.isolate Update all .isolate files for the new format. 2014-10-31 18:08:09 +00:00
supplement.gypi Move target_subarch from gyp_webrtc to supplement.gypi 2015-04-16 07:24:23 +00:00
transport.h Rename newapi::Transport::SendRTP()->SendRtp(). 2013-11-20 12:17:04 +00:00
typedefs.h Clear ARM NEON flag 2015-05-20 05:20:04 +00:00
video_decoder.h Rename I420VideoFrame to VideoFrame. 2015-05-30 00:21:56 +00:00
video_encoder.h Revert "Convert native handles to buffers before encoding." 2015-06-02 13:04:31 +00:00
video_engine_tests.isolate Update isolate files for Android APK tests. 2014-11-13 08:35:05 +00:00
video_frame.h Revert "Convert native handles to buffers before encoding." 2015-06-02 13:04:31 +00:00
video_receive_stream.h Configure default render delay as 10 ms. 2015-05-27 15:59:21 +00:00
video_renderer.h Rename I420VideoFrame to VideoFrame. 2015-05-30 00:21:56 +00:00
video_send_stream.h Rename I420VideoFrame to VideoFrame. 2015-05-30 00:21:56 +00:00
webrtc_examples.gyp Adding support for OpenSL ES output in native WebRTC 2015-05-18 14:49:04 +00:00
webrtc_perf_tests.isolate Enable GoogleWifiTrace3Mbps simulations. 2015-05-04 12:26:26 +00:00
webrtc_tests.gypi Add HW fallback option to software encoding. 2015-05-19 21:09:17 +00:00
webrtc.gyp Rename GYP and GN targets for video capture+render. 2015-02-11 07:47:47 +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.