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Bjorn Mellem 0cf9a4a482 Add texture support to HardwareVideoEncoder. 
HardwareVideoEncoderFactory can now take an EglBase.Context on creation.
When it does, it creates video encoders in texture mode.  It uses the
COLOR_FormatSurface colorFormat.  It passes the EglBase.Context to the
HardwareVideoEncoder.

The HardwareVideoEncoder sets up an input surface for its codec and handles
incoming frames by drawing them onto the input surface.

BUG=webrtc:7760
R=pthatcher@webrtc.org, sakal@webrtc.org

Review-Url: https://codereview.webrtc.org/2977153003 .
Cr-Commit-Position: refs/heads/master@{#19083}
2017-07-18 20:19:27 +00:00
..
api
Remove default implementation of PeerConnectionInterface::SetBitrate.
2017-07-17 22:09:30 +00:00
audio
Don't run NoBandwidthDropAfterDtx test on andriod because it's flaky.
2017-07-17 15:19:27 +00:00
base
Revert of Delete SignalThread class. (patchset #20 id:380001 of https://codereview.webrtc.org/2915253002/ )
2017-07-11 23:56:05 +00:00
call
Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy.
2017-07-14 19:17:49 +00:00
common_audio
Remove potential left shift of negative value in WebRtcSpl_AnalysisQMF
2017-07-18 10:02:28 +00:00
common_video
Use relative paths in GN files.
2017-07-11 13:20:45 +00:00
examples
Fix fullscreen scaling in AppRTCMobile.
2017-07-18 12:01:08 +00:00
logging
nit: Avoid pointer-to-unique_ptr in RtcEventLogImpl
2017-07-16 23:44:08 +00:00
media
Revert "Prefer external video codecs over internal in SDP"
2017-07-17 15:45:17 +00:00
modules
Mapping screen id from DirectX capturer to GDI capturer
2017-07-18 17:26:58 +00:00
ortc
Disabling test on iOS64 debug bot
2017-07-14 22:19:03 +00:00
p2p
Reinstate "Add additional check when setting RTCConfiguration"
2017-07-14 23:40:53 +00:00
pc
Enable tracing on rtcstats_integrationtest.cc
2017-07-18 14:35:19 +00:00
rtc_base
Reinstate "API for periodically regathering ICE candidates"
2017-07-14 22:26:05 +00:00
rtc_tools
Added -show_detector_state which show the detector state in the total bitrate graph.
2017-07-14 13:30:03 +00:00
sdk
Add texture support to HardwareVideoEncoder.
2017-07-18 20:19:27 +00:00
stats
Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy.
2017-07-14 19:17:49 +00:00
system_wrappers
Fix DetectNumberOfCores for 32-bit processes
2017-07-13 17:06:12 +00:00
test
Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy.
2017-07-14 19:17:49 +00:00
video
Add rtpdump and rtc log functionality to screenshare_loopback and video_loopback
2017-07-13 07:47:03 +00:00
voice_engine
Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy.
2017-07-14 19:17:49 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
Reimplemeted "Test and fix for huge bwe drop after alr state"
2017-07-17 08:41:41 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
common_types.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
DEPS
Move webrtc/{tools => rtc_tools}
2017-06-30 10:04:59 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
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
Report interframe delay sum in old GetStats
2017-07-07 08:26:24 +00:00
video_send_stream.h
Move RTP keep-alive config from VideoSendStream::Config to Call::Config
2017-07-11 10:44:17 +00:00
webrtc.gni
Adds detection of audio glitches for playout on iOS (reland)
2017-06-13 16:00:18 +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.