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nisse 76270de4bc Delete unused and almost unused frame-related methods. 
webrtc::VideoFrame::set_video_frame_buffer
webrtc::VideoFrame::ConvertNativeToI420Frame

cricket::WebRtcVideoFrame::InitToBlack

VideoFrameBuffer::data
VideoFrameBuffer::stride
VideoFrameBuffer::MutableData

TBR=tkchin@webrtc.org # Refactoring affecting RTCVideoFrame
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/2065733003
Cr-Commit-Position: refs/heads/master@{#13183}
2016-06-17 09:00:19 +00:00
..
androidjunit
Reland of Android: Change camera fps range selection (patchset #1 id:1 of https://codereview.webrtc.org/2021233002/ )
2016-06-01 07:44:07 +00:00
api
Delete unused and almost unused frame-related methods.
2016-06-17 09:00:19 +00:00
audio
Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream().
2016-06-16 17:53:28 +00:00
base
Android: Add Size class.
2016-06-17 08:02:10 +00:00
build
GN: Add modules_unittests
2016-06-13 07:19:53 +00:00
call
Changes synchronization offset perfomance tracking
2016-06-16 07:58:58 +00:00
common_audio
GN: Add common_audio_unittests and common_video_unittests
2016-06-08 08:28:15 +00:00
common_video
Delete unused and almost unused frame-related methods.
2016-06-17 09:00:19 +00:00
examples
Add RTCEventLog API to ObjC.
2016-06-14 22:03:19 +00:00
libjingle
Adding more detail to MessageQueue::Dispatch logging.
2016-06-10 21:17:33 +00:00
media
Delete unused and almost unused frame-related methods.
2016-06-17 09:00:19 +00:00
modules
Delete unused and almost unused frame-related methods.
2016-06-17 09:00:19 +00:00
p2p
Fix header size check in PseudoTcp::parse().
2016-06-17 01:08:19 +00:00
pc
Delete GetExecutablePath and related unused code.
2016-06-16 10:44:44 +00:00
sdk
Delete unused and almost unused frame-related methods.
2016-06-17 09:00:19 +00:00
system_wrappers
FileWrapper[Impl] modifications and actually remove the "Impl" class.
2016-06-15 17:30:18 +00:00
test
Roll chromium_revision 7fa6701bc5..1a73d11e65 (398458:399420)
2016-06-17 06:29:38 +00:00
tools
New misc scripts, header_usage.sh and author_line_count.sh.
2016-06-15 07:06:32 +00:00
video
Revert of Split IncomingVideoStream into two implementations, with smoothing and without. (patchset #4 id:290001 of https://codereview.webrtc.org/2071473002/ )
2016-06-16 22:44:11 +00:00
voice_engine
Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream().
2016-06-16 17:53:28 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Configure VoE NACK through AudioReceiveStream::Config, for receive streams. Also minor refactoring of WVoE unit test.
2016-06-14 19:13:07 +00:00
audio_send_stream.h
Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream().
2016-06-16 17:53:28 +00:00
audio_sink.h
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
audio_state.h
Add comments about the Audio parts of the public Call API being WIP.
2015-12-03 12:06:31 +00:00
BUILD.gn
GN: Add video_engine_tests
2016-06-14 10:53:09 +00:00
call.h
Movable support for VideoReceiveStream::Config and avoid copies.
2016-06-10 15:58:12 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
FileWrapper[Impl] modifications and actually remove the "Impl" class.
2016-06-15 17:30:18 +00:00
common_types.h
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
common.gyp
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +00:00
common.h
#include "webrtc/base/constructormagic.h" where appropriate
2016-04-26 15:14:48 +00:00
config.cc
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
config.h
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
DEPS
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
2016-06-13 14:35:01 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Add kwiberg@webrtc.org as root owner.
2016-06-14 09:39:40 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
rtc_unittests.isolate
…
supplement.gypi
Roll chromium_revision 086802955f..7fa6701bc5 (396351:398458)
2016-06-08 11:05:27 +00:00
transport.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
typedefs.h
Remove runtime NEON detection
2016-05-17 17:56:48 +00:00
video_decoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Remove QualityScaler framerate reduction.
2016-04-26 11:37:20 +00:00
video_engine_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
video_engine_tests.isolate
Add audio streams to CallTest and a first A/V call test.
2016-01-07 16:43:31 +00:00
video_frame.h
Delete unused and almost unused frame-related methods.
2016-06-17 09:00:19 +00:00
video_receive_stream.h
Movable support for VideoReceiveStream::Config and avoid copies.
2016-06-10 15:58:12 +00:00
video_send_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
webrtc_examples.gyp
Remove metrics_default from rtc_media dependencies.
2016-06-01 12:28:57 +00:00
webrtc_nonparallel_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
Fix missing resource file in webrtc_perf_tests.isolate
2016-06-17 07:39:49 +00:00
webrtc_tests.gypi
GN: Add SDK tests to rtc_unittests.
2016-06-08 08:14:22 +00:00
webrtc.gyp
Remove webrtc_all target
2016-06-14 09:09:24 +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.