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platform-external-webrtc/webrtc
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Dan Willemsen aa415f46df Remove WEBRTC_BUILD_NEON_LIBS 
It doesn't work, and only complicates the Soong conversion (since it
can't even be tested). It shouldn't be that hard to reimplement in Soong
if we ever fix it.

Test: mmma external/webrtc
Change-Id: Ia5a23a6b7fb1ee17f7d9633016eb8ff73435b448
2017-10-25 15:19:20 -07:00
..
api
Leave only an empty top level OWNERS file.
2017-02-23 22:25:35 -08:00
audio
Leave only an empty top level OWNERS file.
2017-02-23 22:25:35 -08:00
base
Remove unused gnustl webrtc variants
2017-10-25 15:17:01 -07:00
build
Leave only an empty top level OWNERS file.
2017-02-23 22:25:35 -08:00
call
Leave only an empty top level OWNERS file.
2017-02-23 22:25:35 -08:00
common_audio
Remove unused gnustl webrtc variants
2017-10-25 15:17:01 -07:00
common_video
Leave only an empty top level OWNERS file.
2017-02-23 22:25:35 -08:00
examples
Leave only an empty top level OWNERS file.
2017-02-23 22:25:35 -08:00
libjingle
Leave only an empty top level OWNERS file.
2017-02-23 22:25:35 -08:00
modules
Remove WEBRTC_BUILD_NEON_LIBS
2017-10-25 15:19:20 -07:00
p2p
Leave only an empty top level OWNERS file.
2017-02-23 22:25:35 -08:00
sound
Leave only an empty top level OWNERS file.
2017-02-23 22:25:35 -08:00
system_wrappers
Remove unused gnustl webrtc variants
2017-10-25 15:17:01 -07:00
test
Merge "Leave only an empty top level OWNERS file." am: 82eac3519a am: 4cc6c9fa1f
2017-02-24 22:14:25 +00:00
tools
Leave only an empty top level OWNERS file.
2017-02-23 22:25:35 -08:00
video
Leave only an empty top level OWNERS file.
2017-02-23 22:25:35 -08:00
voice_engine
Leave only an empty top level OWNERS file.
2017-02-23 22:25:35 -08:00
.gitignore
Merge upstream SHA 04cb763
2016-01-15 11:28:47 -08:00
audio_receive_stream.h
Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
2016-01-13 20:00:29 +00:00
audio_send_stream.h
Prepare the AudioSendStream to be hooked up to send-side BWE.
2015-12-07 09:26:32 +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
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
call.h
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
2015-11-06 23:34:58 +00:00
codereview.settings
Improve informative message in codereview.settings.
2015-11-10 10:08:36 +00:00
common_types.cc
…
common_types.h
Convert channel counts to size_t.
2016-01-13 00:26:55 +00:00
common.gyp
…
common.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
config.cc
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
config.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
engine_configurations.h
Remove VIDEOCODEC_* from engine_configurations.h.
2015-11-24 12:56:05 +00:00
frame_callback.h
Unify FrameType and VideoFrameType.
2015-10-19 09:39:15 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
…
stream.h
…
supplement.gypi
Roll chromium_revision 68cf0b8..aa8e58a (361406:361601)
2015-11-25 21:20:11 +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
Update attributes to match gclibc's ansidecl.h
2016-01-08 12:51:47 +00:00
video_decoder.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
video_encoder.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +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
Add unit test for stand-alone denoiser and fixed some bugs.
2016-01-12 05:34:14 +00:00
video_receive_stream.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
video_renderer.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
video_send_stream.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
webrtc_examples.gyp
Disable warnings failing when using Clang on Windows.
2015-12-16 22:05:36 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
Step 1 to prepare call_test.* for combined audio/video tests.
2015-12-21 11:14:05 +00:00
webrtc.gyp
Remove 'video_engine_core_unittests' binary.
2015-12-04 09:59:02 +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.