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pbos@webrtc.org ebb467fdc8 Avoid NACK-list flush error on keyframe packets. 
Receiver code used to indicate a flush error even if the incoming packet
is a keyframe, forcing a request of a keyframe. Now it takes this
keyframe into account and doesn't error as the stream is decodable from
this point.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15549005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6188 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-19 15:28:02 +00:00
..
base
libvpx's UNUSED macro conflicts with webrtc/base's. Added missing include of assert.h. Globally defined function "Unused" in talk/base and its copy (webrtc/base) is causing a conflict.
2014-05-16 16:54:44 +00:00
build
removed webrtc_base_tests_utils from merge libs as it was breaking some builds.
2014-05-15 21:45:09 +00:00
common_audio
common_audio/signal_processing: Removes macro WEBRTC_SPL_UMUL_RSFT16
2014-05-16 12:11:20 +00:00
common_video
Added include of assert.h for files calling assert but missing the include.
2014-04-29 17:54:17 +00:00
examples
WebRTCDemo: couldn't run a second time. The reason is voe could register/unregister for each run, but vie would expect initialization only once per process.
2014-05-15 03:18:15 +00:00
modules
Avoid NACK-list flush error on keyframe packets.
2014-05-19 15:28:02 +00:00
overrides/webrtc/base
Fix Windows debug compile of overrides/ logging.
2014-05-15 13:56:56 +00:00
system_wrappers
Wire up --force_fieldtrials for vie_auto_test and for test targets linking with test/test.gyp:{test_main|test_support_main}
2014-05-16 09:39:51 +00:00
test
Wire up --force_fieldtrials for vie_auto_test and for test targets linking with test/test.gyp:{test_main|test_support_main}
2014-05-16 09:39:51 +00:00
tools
Add webrtc field trials API.
2014-05-14 12:24:04 +00:00
video
Fix Win VideoSendStream::...::ToString() compiles.
2014-05-15 10:03:24 +00:00
video_engine
Wire up --force_fieldtrials for vie_auto_test and for test targets linking with test/test.gyp:{test_main|test_support_main}
2014-05-16 09:39:51 +00:00
voice_engine
Fix flaky test SendRtpRtcpHeaderExtensionsTest.SentPackets*.
2014-05-16 11:27:09 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
call.h
Add DeliveryStatus enum to DeliverPacket().
2014-05-14 13:57:12 +00:00
common_types.h
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +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
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
config.h
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
engine_configurations.h
Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs.
2014-05-12 12:19:19 +00:00
experiments.h
Adding API for setting bandwidth estimation configurations.
2014-03-25 10:37:31 +00:00
frame_callback.h
Wire up statistics in video receive stream of new API
2014-02-07 12:06:29 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Consolidate all third party licenses in LICENSE_THIRD_PARTY.
2013-05-05 18:54:10 +00:00
OWNERS
Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
2014-04-14 20:08:03 +00:00
PATENTS
…
PRESUBMIT.py
Made the presubmit script accept license headers back to 2003
2014-05-15 18:21:17 +00:00
README.chromium
…
supplement.gypi
Roll chromium_revision 260462:266514
2014-04-29 09:36:40 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Remove ALLOW_UNUSED.
2014-05-05 18:18:02 +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
Rename Start/Stop in Video{Send,Receive}Streams.
2014-04-24 11:13:21 +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
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
webrtc_examples.gyp
Add webrtc field trials API.
2014-05-14 12:24:04 +00:00
webrtc_perf_tests.isolate
Move realtime tests to webrtc_perf_tests.
2013-12-13 12:48:05 +00:00
webrtc_tests.gypi
Add webrtc field trials API.
2014-05-14 12:24:04 +00:00
webrtc.gyp
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +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.