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kjellander 86512b401e Rename iOS test specs to match buildbot names. 
I really prefer not using spaces in any filenames but
if we were to rename all the bots all existing URLs to builds
would stop working (or we'd loose the build history), so I'd
like to see if this works first.
The bots that hits the errors are the new ones I'm experimenting
with in client.webrtc.fyi. Example failing build:
https://build.chromium.org/p/client.webrtc.fyi/builders/iOS64%20Simulator%20Debug

BUG=chromium:498746
NOTRY=True
TBR=phoglund@webrtc.org

Review URL: https://codereview.webrtc.org/1665783002

Cr-Commit-Position: refs/heads/master@{#11473}
2016-02-03 12:41:10 +00:00
..
api
Make ECDSA default for RTCPeerConnection
2016-01-28 01:50:15 +00:00
audio
Use separate rtp module lists for send and receive in PacketRouter.
2016-02-01 12:40:04 +00:00
base
Add iOS tracing.
2016-02-03 09:51:22 +00:00
build
Rename iOS test specs to match buildbot names.
2016-02-03 12:41:10 +00:00
call
Use separate rtp module lists for send and receive in PacketRouter.
2016-02-01 12:40:04 +00:00
common_audio
Convert channel counts to size_t.
2016-01-13 00:26:55 +00:00
common_video
H.264 video codec support using OpenH264 (http://www.openh264.org/) for encoding and FFmpeg (https://www.ffmpeg.org/) for decoding.
2016-01-27 09:36:07 +00:00
examples
Add iOS tracing.
2016-02-03 09:51:22 +00:00
libjingle
Remove DISABLED_ON_ macros.
2016-01-04 21:44:16 +00:00
media
New rtc::VideoSinkInterface.
2016-01-28 12:47:13 +00:00
modules
Support multiple rtx codecs.
2016-02-03 12:30:10 +00:00
p2p
Reset TURN port NONCE when a new socket is created.
2016-02-01 23:19:24 +00:00
sound
OWNERS: Add * to .gyp{i,} everywhere.
2015-12-16 19:44:39 +00:00
system_wrappers
Add helper macros for calling a histogram with different names.
2016-02-02 15:13:16 +00:00
test
Remove implicit downcast in producer_fec_fuzzer.cc.
2016-02-02 21:31:11 +00:00
tools
Convert channel counts to size_t.
2016-01-13 00:26:55 +00:00
video
Support multiple rtx codecs.
2016-02-03 12:30:10 +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
Storing raw audio sink for default audio track.
2016-01-15 17:20:08 +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
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +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
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +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
Unify FrameType and VideoFrameType.
2015-10-19 09:39:15 +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
Deprecate VideoDecoder::Reset() and remove calls.
2016-02-02 14:40:13 +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
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
webrtc_examples.gyp
Disable warnings failing when using Clang on Windows.
2015-12-16 22:05:36 +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
Make ECDSA default for RTCPeerConnection
2016-01-28 01:50:15 +00:00
webrtc.gyp
Make ECDSA default for RTCPeerConnection
2016-01-28 01:50:15 +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.