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webrtc_m130/webrtc
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andersc 151aa6b3f7 Adjust for initial cropping when adapting frame. 
When adapting a frame, any initial cropping coordinate was replaced
by the coordinate calculated for adapting. Offset initial coordinate
instead.

BUG=webrtc:7880

Review-Url: https://codereview.webrtc.org/2975613002
Cr-Commit-Position: refs/heads/master@{#19509}
2017-08-25 08:33:18 +00:00
..
api
Add stats totalSamplesReceived and concealedSamples
2017-08-25 00:50:42 +00:00
audio
Add stats totalSamplesReceived and concealedSamples
2017-08-25 00:50:42 +00:00
call
Add stats totalSamplesReceived and concealedSamples
2017-08-25 00:50:42 +00:00
common_audio
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
common_video
Change logging method from WEBRTC_TRACE macro to LOG macro.
2017-08-22 07:29:13 +00:00
examples
Rename all objc targets to be suffixed for consistency
2017-08-24 12:43:45 +00:00
logging
Move video send/receive stream headers to webrtc/call.
2017-08-22 12:43:23 +00:00
media
Add stats totalSamplesReceived and concealedSamples
2017-08-25 00:50:42 +00:00
modules
Utilizing the AEC3 config struct for constants.
2017-08-25 05:36:53 +00:00
ortc
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
p2p
Renamed dtlstransportchannel.h/.cc/_unittest.cc.
2017-08-24 01:05:50 +00:00
pc
Add stats totalSamplesReceived and concealedSamples
2017-08-25 00:50:42 +00:00
rtc_base
Fix rtc::CurrentThreadId() on Android
2017-08-24 13:41:05 +00:00
rtc_tools
Move PacedSender ownership to RtpTransportControllerSend.
2017-08-22 14:16:49 +00:00
sdk
Adjust for initial cropping when adapting frame.
2017-08-25 08:33:18 +00:00
stats
Add stats totalSamplesReceived and concealedSamples
2017-08-25 00:50:42 +00:00
system_wrappers
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
test
Remove deprecated DirectTransport constructors
2017-08-24 13:56:15 +00:00
video
Eliminate RtpVideoStreamReceiver::receive_cs_ in favor of using a SequencedTaskChecker
2017-08-25 07:49:08 +00:00
voice_engine
Delete unneeded includes of atomic32.h.
2017-08-24 10:30:05 +00:00
.gitignore
…
BUILD.gn
Rename all objc targets to be suffixed for consistency
2017-08-24 12:43:45 +00:00
codereview.settings
…
common_types.cc
Make CodecType conversion functions non-optional.
2017-08-24 10:52:48 +00:00
common_types.h
Add stats totalSamplesReceived and concealedSamples
2017-08-25 00:50:42 +00:00
config_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config.cc
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
config.h
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
DEPS
Move video send/receive stream headers to webrtc/call.
2017-08-22 12:43:23 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Don't add or rename files in webrtc/ and webrtc/api/ without a proper review
2017-05-08 13:01:54 +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_receive_stream.h
Move video send/receive stream headers to webrtc/call.
2017-08-22 12:43:23 +00:00
video_send_stream.h
Move video send/receive stream headers to webrtc/call.
2017-08-22 12:43:23 +00:00
webrtc.gni
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
whitespace.txt
…

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.