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jbauch eaa9c1db73 Remove HAVE_SRTP define and unmaintained code. 
It was defined unconditionally and the code for non-HAVE_SRTP was unmaintained
and failed to compile.

BUG=webrtc:7294

Review-Url: https://codereview.webrtc.org/2729373002
Cr-Commit-Position: refs/heads/master@{#17074}
2017-03-06 19:32:22 +00:00
..
api
Reland "Enable GN check for webrtc/examples"
2017-03-06 08:29:21 +00:00
audio
Adding placeholder for low bandwidth audio test
2017-03-06 12:01:16 +00:00
base
Fixing race between CallbackCanceled and CancelCallback in AsyncInvoker.
2017-03-03 18:33:18 +00:00
call
Add support for Location (RTC_FROM_HERE) to ProcessThread::RegisterModule.
2017-03-03 11:20:24 +00:00
common_audio
Enable GN check for webrtc/common_audio
2017-03-01 15:07:10 +00:00
common_video
Ignore aud and sei NALus when parsing bitstream.
2017-03-06 10:49:36 +00:00
examples
Add kjellander to OWNERS for *.py in examples/android{app,tests}
2017-03-06 11:56:57 +00:00
logging
Remove unused include from rtc_event_log_parser.cc
2017-02-28 09:46:19 +00:00
media
Enable GN check for webrtc/media
2017-03-06 14:04:55 +00:00
modules
Fix segmentation fault in AudioEncoderOpusTest.EncodeAtMinBitrate.
2017-03-06 14:49:27 +00:00
ortc
Revert of Enable GN check for webrtc/{ortc,p2p} (patchset #4 id:60001 of https://codereview.webrtc.org/2714263004/ )
2017-03-04 23:08:44 +00:00
p2p
Revert of Enable GN check for webrtc/{ortc,p2p} (patchset #4 id:60001 of https://codereview.webrtc.org/2714263004/ )
2017-03-04 23:08:44 +00:00
pc
Remove HAVE_SRTP define and unmaintained code.
2017-03-06 19:32:22 +00:00
sdk
Android HW decoder: Don't check slice_height for texture output
2017-03-06 13:20:49 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Check __GLIBC_PREREQ availability before use.
2017-03-06 15:34:06 +00:00
test
Revert of Added large room scenario to full-stack tests. Added thumbnail streams functionality to call test/v… (patchset #8 id:140001 of https://codereview.webrtc.org/2730073002/ )
2017-03-06 15:35:13 +00:00
tools
Rename webrtc::PacketInfo to webrtc::PacketFeedback. This resolves ambiguity with a similarly named RTCPReceiver::PacketInformation and RtpPacketizerVp9::PacketInfo.
2017-03-06 13:32:21 +00:00
video
Revert of Added large room scenario to full-stack tests. Added thumbnail streams functionality to call test/v… (patchset #8 id:140001 of https://codereview.webrtc.org/2730073002/ )
2017-03-06 15:35:13 +00:00
voice_engine
Rename webrtc::PacketInfo to webrtc::PacketFeedback. This resolves ambiguity with a similarly named RTCPReceiver::PacketInformation and RtpPacketizerVp9::PacketInfo.
2017-03-06 13:32:21 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
Remove HAVE_SRTP define and unmaintained code.
2017-03-06 19:32:22 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Remove the Windows Wave audio device implementation.
2017-02-17 22:48:07 +00:00
config.cc
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
config.h
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
DEPS
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
no_op_function.cc
Moving no_op_function.cc out of webrtc/build
2017-01-24 13:25:12 +00:00
OWNERS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Only define NO_RETURN if undefined
2017-01-30 08:54:19 +00:00
video_decoder.h
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
video_receive_stream.h
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +00:00
video_send_stream.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
webrtc.gni
Reland of Setting is_component_build to false by default (patchset #1 id:1 of https://codereview.webrtc.org/2731703004/ )
2017-03-04 03:41:59 +00:00
whitespace.txt
Moving whitespace file up by one folder
2017-01-24 13:19:42 +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.