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nisse cbae0b475c Use I420Buffer rather than VideoFrameBuffer when writing pixels. 
Prepares for deleting VideoFrameBuffer::MutableDataY{,U,V}. Android
changes extracted from cl https://codereview.webrtc.org/2278883002/.

BUG=webrtc:5921

Review-Url: https://codereview.webrtc.org/2331383005
Cr-Commit-Position: refs/heads/master@{#14213}
2016-09-14 12:45:31 +00:00
..
androidjunit
Re-order and remove unused Java imports.
2016-08-02 13:44:19 +00:00
api
Use I420Buffer rather than VideoFrameBuffer when writing pixels.
2016-09-14 12:45:31 +00:00
audio
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
base
New method TimestampAligner::TranslateTimestamp
2016-09-14 07:37:03 +00:00
build
Enable BWE logging to command line when rtc_enable_bwe_test_logging is set to true
2016-09-14 12:04:43 +00:00
call
Remove RTC_LOGGED_* macro.
2016-09-10 05:40:34 +00:00
common_audio
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
common_video
GN: Declare resources for targets.
2016-09-14 12:10:06 +00:00
examples
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
libjingle
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
media
GN: Declare resources for targets.
2016-09-14 12:10:06 +00:00
modules
webrtc/modules/audio_processing: Use RTC_DCHECK() instead of assert()
2016-09-14 12:23:29 +00:00
p2p
Removing assert error when we fail to create a connection for a ping from an unknown address.
2016-09-10 03:48:08 +00:00
pc
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
sdk
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
stats
Delete Timing class, timing.h, and update all users.
2016-09-14 06:41:55 +00:00
system_wrappers
Remove RTC_LOGGED_* macro.
2016-09-10 05:40:34 +00:00
test
GN: Declare resources for targets.
2016-09-14 12:10:06 +00:00
tools
GN: Declare resources for targets.
2016-09-14 12:10:06 +00:00
video
Add periodic logging of number of captured and dropped frames in VideoCaptureInput. Logged every minute.
2016-09-12 07:10:53 +00:00
voice_engine
Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/
2016-09-13 14:29:19 +00:00
.gitignore
…
BUILD.gn
GN: Declare resources for targets.
2016-09-14 12:10:06 +00:00
call.h
This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads.
2016-09-01 08:17:43 +00:00
codereview.settings
…
common_types.cc
Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )
2016-09-08 10:25:05 +00:00
common_types.h
Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )
2016-09-08 10:25:05 +00:00
common.gyp
Moving/renaming webrtc/common.h.
2016-09-07 14:34:45 +00:00
config.cc
…
config.h
Moving/renaming webrtc/common.h.
2016-09-07 14:34:45 +00:00
DEPS
Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )
2016-09-08 10:25:05 +00:00
engine_configurations.h
Reland of move all reference to carbon api (patchset #1 id:1 of https://codereview.webrtc.org/2317343003/ )
2016-09-09 10:19:54 +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
…
rtc_unittests_apk.isolate
…
rtc_unittests.isolate
…
supplement.gypi
…
transport.h
…
typedefs.h
…
video_decoder.h
…
video_encoder.h
Add codec name to CodecSpecificInfo and get the codec name stats from there instead.
2016-09-01 07:21:18 +00:00
video_engine_tests_apk.isolate
…
video_engine_tests.isolate
…
video_frame.h
Introduce webrtc::VideoFrame::timestamp_us, and corresponding constructor.
2016-09-05 07:51:25 +00:00
video_receive_stream.h
Add periodic logging of video stats.
2016-08-11 15:41:26 +00:00
video_send_stream.h
This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads.
2016-09-01 08:17:43 +00:00
webrtc_examples.gyp
Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 10:07:13 +00:00
webrtc_nonparallel_tests_apk.isolate
…
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests_apk.isolate
…
webrtc_perf_tests.isolate
Add an HD resolution perf test.
2016-08-11 14:01:03 +00:00
webrtc_tests.gypi
Added build flag around the Intelligibility enhancer performance test code
2016-09-14 04:42:43 +00:00
webrtc.gyp
Move webrtc/audio_*.h to webrtc/api/call
2016-08-31 14:33:14 +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.