admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc
History
henrik.lundin fd2be2718d Fuzzer tests for AudioDecoder's DecodeRedundant and IncomingPacket 
This CL adds new fuzzer tests for the DecodeRedundant and
IncomingPacket methods of AudioDecoder. In practice, only Opus has
DecodeRedundant, and only iSAC has IncomingPacket. Did some minor work
to generalize the helper function reading values from the fuzzed
input.

BUG=webrtc:5306
R=pbos@webrtc.org
NOTRY=true

Review URL: https://codereview.webrtc.org/1607173003

Cr-Commit-Position: refs/heads/master@{#11533}
2016-02-09 08:00:32 +00:00
..
api
Fix GYP and GN references that are invalid in Chromium builds.
2016-02-08 07:04:33 +00:00
audio
Clean up of CongestionController.
2016-02-08 13:31:53 +00:00
base
Fix race between Thread ctor/dtor and MessageQueueManager registrations.
2016-02-05 08:25:04 +00:00
build
Revert of Analyze support in gyp_webrtc (patchset #1 id:1 of https://codereview.webrtc.org/1369683004/ )
2016-02-09 07:46:39 +00:00
call
Clean up of CongestionController.
2016-02-08 13:31:53 +00:00
common_audio
Move gtest_prod_util.h out of webrtc/test tree.
2016-02-05 08:23:57 +00:00
common_video
H264: Improve FFmpeg decoder performance by using I420BufferPool.
2016-02-05 16:08:39 +00:00
examples
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
libjingle
Remove DISABLED_ON_ macros.
2016-01-04 21:44:16 +00:00
media
Add perkj+magjed to webrtc/media/OWNERS
2016-02-08 14:46:04 +00:00
modules
Enable cpplint for webrtc/modules/pacing and fix all uncovered cpplint errors.
2016-02-09 07:18:30 +00:00
p2p
Create QuicSession
2016-02-09 01:35:55 +00:00
sound
OWNERS: Add * to .gyp{i,} everywhere.
2015-12-16 19:44:39 +00:00
system_wrappers
Integrate helper macros for calling histograms with different names (real-time vs screenshare and rampup metrics).
2016-02-04 08:33:29 +00:00
test
Fuzzer tests for AudioDecoder's DecodeRedundant and IncomingPacket
2016-02-09 08:00:32 +00:00
tools
Move gtest_prod_util.h out of webrtc/test tree.
2016-02-05 08:23:57 +00:00
video
Clean up of CongestionController.
2016-02-08 13:31:53 +00:00
voice_engine
Remove mutable from rtc::CriticalSections.
2016-02-01 17:00:59 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Add send-side BWE to WebRtcVoiceEngine under a finch experiment.
2016-02-04 12:12:31 +00:00
audio_send_stream.h
Prepare the AudioSendStream to be hooked up to send-side BWE.
2015-12-07 09:26:32 +00:00
audio_state.h
Add comments about the Audio parts of the public Call API being WIP.
2015-12-03 12:06:31 +00:00
BUILD.gn
Move gtest_prod_util.h out of webrtc/test tree.
2016-02-05 08:23:57 +00:00
call.h
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
2015-11-06 23:34:58 +00:00
codereview.settings
Improve informative message in codereview.settings.
2015-11-10 10:08:36 +00:00
common_types.cc
Separating voice activity flag from audio level in RtpHeaderExtension.
2015-08-10 13:08:46 +00:00
common_types.h
Remove extra_options from VideoCodec.
2016-01-19 15:26:24 +00:00
common.gyp
Move gtest_prod_util.h out of webrtc/test tree.
2016-02-05 08:23:57 +00:00
common.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
config.cc
Support multiple rtx codecs.
2016-02-03 12:30:10 +00:00
config.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
engine_configurations.h
Remove VIDEOCODEC_* from engine_configurations.h.
2015-11-24 12:56:05 +00:00
frame_callback.h
Measure encoding time on encode callbacks.
2016-02-05 10:13:41 +00:00
libjingle_media_unittest.isolate
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
…
stream.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
supplement.gypi
Roll chromium_revision 68cf0b8..aa8e58a (361406:361601)
2015-11-25 21:20:11 +00:00
transport.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
typedefs.h
Update attributes to match gclibc's ansidecl.h
2016-01-08 12:51:47 +00:00
video_decoder.h
Implement NullVideoDecoder to avoid crash on unsupported decoders.
2016-02-03 13:51:56 +00:00
video_encoder.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
video_engine_tests.isolate
Add audio streams to CallTest and a first A/V call test.
2016-01-07 16:43:31 +00:00
video_frame.h
Added EncodedImage::GetBufferPaddingBytes.
2016-01-21 13:43:18 +00:00
video_receive_stream.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
video_renderer.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
video_send_stream.h
Measure encoding time on encode callbacks.
2016-02-05 10:13:41 +00:00
webrtc_examples.gyp
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
webrtc.gyp
Move talk/media to webrtc/media
2016-02-05 07:52:35 +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.