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webrtc_m130/webrtc
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stefan fca900aa37 Fix two invalid DCHECKs related to audio BWE. 
These are invalid since:
- An allocated bitrate of 0 means that the stream should be disabled. Changing the behavior to send audio at min bitrate even though the allocator asks for the stream to be disabled.
- It should be OK to set a min bitrate equal to the start bitrate.

BUG=webrtc:5079

Review-Url: https://codereview.webrtc.org/2806163003
Cr-Commit-Position: refs/heads/master@{#17613}
2017-04-10 10:53:00 +00:00
..
api
Added the GetSources() to the RtpReceiverInterface and implemented
2017-04-07 17:57:22 +00:00
audio
Fix two invalid DCHECKs related to audio BWE.
2017-04-10 10:53:00 +00:00
base
Revert of Delete class ScopedPtrCollection. Replaced with vector of unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2808463002/ )
2017-04-10 07:31:33 +00:00
call
Fix two invalid DCHECKs related to audio BWE.
2017-04-10 10:53:00 +00:00
common_audio
Ignore some UBSan errors
2017-04-10 09:29:33 +00:00
common_video
Delete unneeded includes of deprecated system_wrappers include files.
2017-03-30 07:31:15 +00:00
examples
Use new RTCCameraVideoCapturer in AppRTCMobile.
2017-04-05 19:17:48 +00:00
logging
Reland of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #1 id:1 of https://codereview.webrtc.org/2786363002/ )
2017-04-07 07:59:12 +00:00
media
Added the GetSources() to the RtpReceiverInterface and implemented
2017-04-07 17:57:22 +00:00
modules
Ignore some UBSan errors
2017-04-10 09:29:33 +00:00
ortc
Disable the ORTC integration tests on TSan.
2017-03-28 20:45:18 +00:00
p2p
Fix compilation issues of std::unique_ptr
2017-04-09 16:09:06 +00:00
pc
Fix compilation issues of std::unique_ptr
2017-04-09 16:09:06 +00:00
sdk
Move AndroidVideoTrackSourceObserver from API to src
2017-04-06 11:31:26 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Delete system_wrappers logging facility.
2017-04-06 10:53:06 +00:00
test
Added the GetSources() to the RtpReceiverInterface and implemented
2017-04-07 17:57:22 +00:00
tools
Resolve cyclic dependency between audio network adaptor and event log api
2017-04-06 12:59:10 +00:00
video
Add some unit tests to vie_encoder.
2017-04-10 08:12:52 +00:00
voice_engine
Added the GetSources() to the RtpReceiverInterface and implemented
2017-04-07 17:57:22 +00:00
.gitignore
…
BUILD.gn
Reland of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #1 id:1 of https://codereview.webrtc.org/2786363002/ )
2017-04-07 07:59:12 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
…
common_types.cc
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Revert of Supporting 48kHz PCM file. (patchset #1 id:1 of https://codereview.webrtc.org/2790493004/ )
2017-03-31 23:32:28 +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
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +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
…
typedefs.h
Make WARN_UNUSED_RESULT a no-op on gcc
2017-04-06 11:32:27 +00:00
video_decoder.h
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +00:00
video_encoder.h
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +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
Add |protected_by_flexfec| flag to VideoReceiveStream::Config.
2017-03-09 14:13:20 +00:00
video_send_stream.h
Add number of quality adapt changes to VideoSendStream stats.
2017-04-04 12:51:49 +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.