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andrew@webrtc.org 4165f7aa22 Add a variable for deciding when to use openmax_dl. 
Modifies the previous condition to additionally not use openmax_dl on
iOS. Remove the All target's direct dependency on it, as it is now
pulled in by the targets that need it.

Add gn support since an openmax_dl gn target is available.

BUG=chromium:415393, webrtc:3906
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7397 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-08 18:01:27 +00:00
..
base
CHECK/DCHECK: Explicitly state whether the condition can have side effects
2014-10-08 12:19:56 +00:00
build
Add a variable for deciding when to use openmax_dl.
2014-10-08 18:01:27 +00:00
common_audio
Add a variable for deciding when to use openmax_dl.
2014-10-08 18:01:27 +00:00
common_video
GN: Add common configs to all targets.
2014-09-28 17:37:22 +00:00
examples
Split video engine android initialization into each internal module initialization.
2014-09-17 11:44:51 +00:00
libjingle/xmllite
Mac: adds missing _DEBUG flag to mac debug builds.
2014-10-06 22:04:11 +00:00
modules
audio_coding: Replaced macro WEBRTC_SPL_RSHIFT_W16 with >>
2014-10-08 15:36:30 +00:00
overrides
webrtc/overrides: add OWNERS-file.
2014-09-17 08:04:28 +00:00
sound
rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation.
2014-10-01 16:33:03 +00:00
system_wrappers
Import LappedTransform and friends.
2014-10-01 17:42:18 +00:00
test
GN: Add common configs to tools and test.
2014-09-30 19:07:58 +00:00
tools
Check on the existence of report directory
2014-10-06 17:21:27 +00:00
video
Wire up CPU adaptation in WebRtcVideoEngine2.
2014-10-03 11:25:45 +00:00
video_engine
GN: Add common configs to all targets.
2014-09-28 17:37:22 +00:00
voice_engine
Reland "Remove DTMF status methods from Voice Engine" r7276
2014-10-01 08:23:21 +00:00
.gitignore
…
BUILD.gn
GN: Enable libvpx, add link target and convert some test targets
2014-09-30 18:05:02 +00:00
call.h
Wire up CPU adaptation in WebRtcVideoEngine2.
2014-10-03 11:25:45 +00:00
common_types.h
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
2014-07-11 13:44:02 +00:00
common.gyp
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
common.h
…
config.cc
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
config.h
Config struct for VideoEncoder.
2014-09-19 12:30:25 +00:00
engine_configurations.h
Add boilerplate code for H.264.
2014-07-04 12:42:07 +00:00
experiments.h
Remove no longer used SkipEncodingUnusedStreams.
2014-07-22 07:17:17 +00:00
frame_callback.h
Wire up statistics in video receive stream of new API
2014-02-07 12:06:29 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
…
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
…
rtc_unittests.isolate
Adds isolate for rtc_unittests and moves sound's unittests to rtc_unittest.
2014-09-30 14:21:10 +00:00
supplement.gypi
Roll chromium_revision 289723:291647
2014-08-25 14:16:32 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Add CHECK and friends from Chromium.
2014-08-28 16:28:26 +00:00
video_encoder.h
Expose VP8/H264 defaults through video_encoder.h.
2014-09-18 12:42:28 +00:00
video_engine_tests.isolate
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
video_frame.h
Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h.""
2014-09-17 09:02:25 +00:00
video_receive_stream.h
Change return value for number of discarded packets to be int.
2014-09-04 07:07:44 +00:00
video_renderer.h
Separate Call API/build files from video_engine/.
2013-10-28 16:32:01 +00:00
video_send_stream.h
Config struct for VideoEncoder.
2014-09-19 12:30:25 +00:00
webrtc_examples.gyp
Roll chromium_revision 6455c69..deaf2f7 (293954:295079)
2014-09-27 18:10:30 +00:00
webrtc_perf_tests.isolate
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
webrtc_tests.gypi
Moves xmllite's unittests to rtc_unittest.
2014-10-02 18:43:47 +00:00
webrtc.gyp
Moves xmllite's unittests to rtc_unittest.
2014-10-02 18:43:47 +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.