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henrik.lundin@webrtc.org a32487f97b Disable AudioEncoderCopyRedDeathTest.NullSpeechEncoder 
Fails linux memcheck.

BUG=4108
TBR=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7920 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 21:04:55 +00:00
..
base Revert "Add adapter_type into Candidate object." 2014-12-16 05:28:10 +00:00
build Move isolate path into webrtc/build/android/test_runner.py 2014-12-11 11:59:46 +00:00
common_audio Add (safe) uint32_t cast to fix Win64 build. 2014-12-16 20:47:42 +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 Make one OWNERS files for all of webrtc/libjingle so we don't need approval from webrtc/OWNERS every time we want to add a directory. 2014-12-16 21:04:41 +00:00
modules Disable AudioEncoderCopyRedDeathTest.NullSpeechEncoder 2014-12-16 21:04:55 +00:00
overrides webrtc/overrides: add OWNERS-file. 2014-09-17 08:04:28 +00:00
p2p Revert "Add adapter_type into Candidate object." 2014-12-16 05:28:10 +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 video send bitrates to histogram stats: 2014-12-09 09:47:53 +00:00
test Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess. 2014-12-15 22:09:40 +00:00
tools Merge in AGC manager and AGC tools. 2014-12-15 16:33:16 +00:00
video Wire up RTT statistics to webrtc::Call. 2014-12-11 13:26:09 +00:00
video_engine Add field to counters for when first rtp/rtcp packet is sent/received. 2014-12-16 12:03:11 +00:00
voice_engine Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess. 2014-12-15 22:09:40 +00:00
.gitignore .gitignore: Add *.mk, created as part of ChromiumOS build 2013-01-04 21:25:42 +00:00
BUILD.gn Enabling building with NEON on ARM64 2014-11-26 17:01:40 +00:00
call.h Wire up RTT statistics to webrtc::Call. 2014-12-11 13:26:09 +00:00
codereview.settings Add codereview.settings to the /webrtc subdirectory 2014-12-05 13:43:35 +00:00
common_types.h Add field to counters for when first rtp/rtcp packet is sent/received. 2014-12-16 12:03:11 +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 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 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 Move src/ -> webrtc/ 2012-10-22 18:19:23 +00:00
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 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 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 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 Separate Call API/build files from video_engine/. 2013-10-28 16:32:01 +00:00
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 Add AGC manager tests. 2014-12-16 14:48:47 +00:00
webrtc.gyp Merge in AGC manager and AGC tools. 2014-12-15 16:33:16 +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.