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mflodman@webrtc.org a98e796615 Remove default RTP module functionality for setting CSRC. 
ViECapturer is always calling DeliverFrame with an empty CSRC vector, so
this is basically a dead path already. I added a DCHECK in ViEEncoder to
verify this is true.

BUG=769
TEST=Manually verified in Chromium.
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39059004

Cr-Commit-Position: refs/heads/master@{#8335}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8335 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-11 15:46:20 +00:00
..
base
Don't rely on webrtc/base/scoped_ptr.h to include stuff for you
2015-02-11 14:31:19 +00:00
build
Restructure GYP for vp9, opus and direct trace
2015-02-09 10:01:40 +00:00
common_audio
C++ readability review for ajm.
2015-02-11 01:10:17 +00:00
common_video
Cleanup: unify rotation to be enum based instead of int for degree.
2015-02-09 01:29:45 +00:00
examples
Refactoring WebRTC Java/JNI audio recording in C++ and Java.
2015-02-11 08:39:19 +00:00
libjingle
Enable Clang warning implicit-fallthrough and annotate the code.
2015-01-28 18:38:13 +00:00
modules
Remove default RTP module functionality for setting CSRC.
2015-02-11 15:46:20 +00:00
overrides
Clean kForever from basictypes and move it to the interfaces that actually have it.
2015-02-09 14:19:39 +00:00
p2p
Remove USE_WEBRTC_DEV_BRANCH.
2015-02-09 10:17:12 +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
Fix false positive DHECK in event_posix.cc
2015-02-11 15:19:22 +00:00
test
Rename GYP and GN targets for video capture+render.
2015-02-11 07:47:47 +00:00
tools
Re-land "Remove <(webrtc_root) from source file entries."
2015-01-29 14:30:41 +00:00
video
Remove VideoSendStreamTest.ProducesStats.
2015-02-11 14:25:23 +00:00
video_engine
Remove default RTP module functionality for setting CSRC.
2015-02-11 15:46:20 +00:00
voice_engine
audio_processing: Now records mic volume level also when using new AGC
2015-02-06 19:44:46 +00:00
.gitignore
…
BUILD.gn
Rename GYP and GN targets for video capture+render.
2015-02-11 07:47:47 +00:00
call.h
Use int64_t more consistently for times, in particular for RTT values.
2015-01-12 21:51:21 +00:00
codereview.settings
Add codereview.settings to the /webrtc subdirectory
2014-12-05 13:43:35 +00:00
common_types.h
Add method for incrementing RtpPacketCounter. Removes duplicate code.
2015-02-04 08:35:21 +00:00
common.gyp
Make it easier to use external libyuv + cleanup GYP files.
2015-01-26 19:17:26 +00:00
common.h
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
config.cc
Revert 8028 "Support associated payload type when registering Rt..."
2015-01-09 20:22:46 +00:00
config.h
Revert 8028 "Support associated payload type when registering Rt..."
2015-01-09 20:22:46 +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
Changing include guard in frame_callback.h.
2015-02-03 14:51:39 +00:00
LICENSE
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
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
Enable Clang warning implicit-fallthrough and annotate the code.
2015-01-28 18:38:13 +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
Remove default arguments in EncodedImageCallback.
2015-02-09 09:14:48 +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
Log configs when creating video streams in Call.
2015-01-15 10:09:39 +00:00
video_renderer.h
Use VideoReceiveStream as an ExternalRenderer.
2015-02-09 15:15:24 +00:00
video_send_stream.h
Log configs when creating video streams in Call.
2015-01-15 10:09:39 +00:00
webrtc_examples.gyp
Move internal capture+render to build_with_chromium==0 condition
2015-01-20 11:40:45 +00:00
webrtc_perf_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
webrtc_tests.gypi
Rename GYP and GN targets for video capture+render.
2015-02-11 07:47:47 +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.