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webrtc_m130/webrtc
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mattdr 0d8ade543d Remove remnants of libsrtp1 
Now that Chromium has taken libsrtp2, remove any compatibility bridge code in WebRTC that was only needed for libsrtp1.

Remove SRTP_RELATIVE_PATH now that Google's internal copy of libsrtp and the Chromium copy have the same directory structure.

Fix some include orderings per the Chromium C++ style guide.

Remove the `extern "C"` blocks now that the libsrtp headers include them (https://github.com/cisco/libsrtp/pull/195).

BUG=webrtc:6376

Review-Url: https://codereview.webrtc.org/2447893002
Cr-Commit-Position: refs/heads/master@{#14776}
2016-10-25 16:47:31 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Revert of Clean up logging in AudioSendStream::SetupSendCodec(). (patchset #3 id:40001 of https://codereview.webrtc.org/2446963003/ )
2016-10-25 14:56:42 +00:00
audio
Revert of Clean up logging in AudioSendStream::SetupSendCodec(). (patchset #3 id:40001 of https://codereview.webrtc.org/2446963003/ )
2016-10-25 14:56:42 +00:00
base
Delete left-over file profiler_unittest.cc.
2016-10-25 07:33:44 +00:00
build
Remove remnants of libsrtp1
2016-10-25 16:47:31 +00:00
call
Fix BWE simulations so that it uses the delay based BWE.
2016-10-25 14:04:44 +00:00
common_audio
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
common_video
Reland of Delete webrtc::VideoFrame::CopyFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2397943003/ )
2016-10-24 07:44:17 +00:00
examples
Fix org.mockito.Matchers deprecation warnings in DirectRTCClientTest.
2016-10-20 12:49:12 +00:00
libjingle
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
logging
Revert of Clean up logging in AudioSendStream::SetupSendCodec(). (patchset #3 id:40001 of https://codereview.webrtc.org/2446963003/ )
2016-10-25 14:56:42 +00:00
media
Remove dead dependencies on xmllite and xmpp.
2016-10-25 08:27:17 +00:00
modules
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
p2p
Remove dead dependencies on xmllite and xmpp.
2016-10-25 08:27:17 +00:00
pc
Remove remnants of libsrtp1
2016-10-25 16:47:31 +00:00
sdk
iOS: Optimize video scaling and cropping
2016-10-20 10:34:32 +00:00
stats
RTCStats equality operator added.
2016-10-25 11:31:27 +00:00
system_wrappers
Delete DataLog abstraction, which was almost unused.
2016-10-21 08:53:01 +00:00
test
GN: move webrtc/video/ targets from webrtc/BUILD.gn into webrtc/video/BUILD.gn
2016-10-24 09:09:40 +00:00
tools
During a fix of an unrelated issue, a bug was introduced in the rtp analyzer tool: when the number of data points was divisible by RTPStatitstics.PLOT_RESOLUTION_MS (which is 50), pyplot.plot was called with arrays of different lengths. One of the arrays could be one element larger.
2016-10-18 08:44:50 +00:00
video
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
voice_engine
Stop using old AudioCodingModule::RegisterReceiveCodec overloads
2016-10-24 20:47:16 +00:00
.gitignore
…
BUILD.gn
Remove remnants of libsrtp1
2016-10-25 16:47:31 +00:00
call.h
Integrate FlexfecReceiveStream with Call.
2016-10-24 06:37:24 +00:00
codereview.settings
…
common_types.cc
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
common_types.h
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
common.gyp
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
config.cc
Revert of Clean up logging in AudioSendStream::SetupSendCodec(). (patchset #3 id:40001 of https://codereview.webrtc.org/2446963003/ )
2016-10-25 14:56:42 +00:00
config.h
Revert of Clean up logging in AudioSendStream::SetupSendCodec(). (patchset #3 id:40001 of https://codereview.webrtc.org/2446963003/ )
2016-10-25 14:56:42 +00:00
DEPS
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 10:07:13 +00:00
OWNERS
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +00:00
transport.h
…
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
video_encoder.h
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
video_frame.h
Reland of Delete webrtc::VideoFrame::CopyFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2397943003/ )
2016-10-24 07:44:17 +00:00
video_receive_stream.h
Rename FecConfig to UlpfecConfig in config.h.
2016-10-05 06:28:43 +00:00
video_send_stream.h
Implement framesEncoded stat in video send ssrc stats.
2016-10-24 09:09:39 +00:00
voice_engine_configurations.h
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
webrtc_examples.gyp
Reland of name AppRTCDemo on Android and iOS to AppRTCMobile (patchset #1 id:1 of https://codereview.webrtc.org/2358133003/ )
2016-09-27 08:10:52 +00:00
webrtc_tests.gypi
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +00:00
webrtc.gyp
Moved RtcEventLog files from call/ to logging/
2016-10-04 01:31:32 +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.