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deadbeef 7af91ddd6b Removing "crypto_required" from MediaContentDescription. 
"Crypto required" is a property of the PeerConnection of construction
time; it has nothing to do with SDP. So I'm moving it out of
MediaContentDescription and putting it in the BaseChannel constructor
instead. This is more intuitive, and provides the added assurance that
"secure_required_" can't be flipped from "true" to "false".

BUG=None

Review-Url: https://codereview.webrtc.org/2537343003
Cr-Commit-Position: refs/heads/master@{#15579}
2016-12-13 19:29:16 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Removing "crypto_required" from MediaContentDescription.
2016-12-13 19:29:16 +00:00
audio
Revert of Support external audio mixer. (patchset #5 id:140001 of https://codereview.webrtc.org/2539213003/ )
2016-12-13 01:07:00 +00:00
base
Delete unused class AsyncFile.
2016-12-13 07:56:01 +00:00
build
Move /webrtc/api/android files to /webrtc/sdk/android
2016-12-06 12:29:45 +00:00
call
Disable failing perf test on Android.
2016-12-09 14:15:22 +00:00
common_audio
Update common_audio/smoothing_filter.
2016-12-13 14:53:07 +00:00
common_video
Delete VideoFrame default constructor, and IsZeroSize method.
2016-12-13 14:08:39 +00:00
examples
Add a gtk3 port of peerconnection_client on Linux
2016-12-13 10:57:50 +00:00
libjingle
Remove extra uses of basictypes.h.
2016-12-09 14:54:08 +00:00
logging
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
media
Delete VideoFrame default constructor, and IsZeroSize method.
2016-12-13 14:08:39 +00:00
modules
Delete VideoFrame default constructor, and IsZeroSize method.
2016-12-13 14:08:39 +00:00
p2p
Removing "crypto_required" from MediaContentDescription.
2016-12-13 19:29:16 +00:00
pc
Removing "crypto_required" from MediaContentDescription.
2016-12-13 19:29:16 +00:00
sdk
Delete VideoFrame default constructor, and IsZeroSize method.
2016-12-13 14:08:39 +00:00
stats
Collecting RTCIceCandidatePairStats.transport_id and improved unittests.
2016-11-30 09:50:36 +00:00
system_wrappers
Add comment to metrics.h
2016-12-06 11:59:08 +00:00
test
Delete VideoFrame default constructor, and IsZeroSize method.
2016-12-13 14:08:39 +00:00
tools
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
video
Delete VideoFrame default constructor, and IsZeroSize method.
2016-12-13 14:08:39 +00:00
voice_engine
Delete voice_engine_configurations.h
2016-12-12 13:03:08 +00:00
.gitignore
…
BUILD.gn
Delete voice_engine_configurations.h
2016-12-12 13:03:08 +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
Revert of Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabl… (patchset #1 id:40001 of https://codereview.webrtc.org/2532053002/ )
2016-12-09 10:35:30 +00:00
config.cc
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
config.h
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
DEPS
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +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
…
transport.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
typedefs.h
Remove extra uses of basictypes.h.
2016-12-09 14:54:08 +00:00
video_decoder.h
Move VideoDecoder::Create() logic to separate internal video decoder factory
2016-12-01 08:27:35 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Delete VideoFrame default constructor, and IsZeroSize method.
2016-12-13 14:08:39 +00:00
video_receive_stream.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
video_send_stream.h
Implement periodic bandwidth probing in application-limited region.
2016-11-28 21:11:24 +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.