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kjellander 499dcb1ac0 Remove references to .isolate files that are no longer needed. 
After the Android test framework in Chromium moved over to generated
scripts, these paths are no longer used, so let's clean them up.

BUG=webrtc:6323
NOTRY=True

Review-Url: https://codereview.webrtc.org/2327013002
Cr-Commit-Position: refs/heads/master@{#14151}
2016-09-09 08:39:00 +00:00
..
androidjunit
Re-order and remove unused Java imports.
2016-08-02 13:44:19 +00:00
api
Add sakal as an OWNER to some Android files.
2016-09-09 07:22:33 +00:00
audio
GN Templates: Move common_inherited_config to the template.
2016-09-05 13:10:23 +00:00
base
Revert of move all reference to carbon api (patchset #2 id:300001 of https://codereview.webrtc.org/2321493002/ )
2016-09-08 12:35:00 +00:00
build
Remove references to .isolate files that are no longer needed.
2016-09-09 08:39:00 +00:00
call
Update AvgCounter to have the ability to include last period metric for subsequent intervals without samples (e.g. for non-periodic updated stats).
2016-09-09 07:13:39 +00:00
common_audio
GN Templates: Move common_inherited_config to the template.
2016-09-05 13:10:23 +00:00
common_video
Add magjed@ as owner of webrtc/common_video
2016-09-08 21:15:34 +00:00
examples
Renamed RTCStatsReport to RTCLegacyStatsReport in objc files.
2016-09-09 08:36:32 +00:00
libjingle
GN Templates: Move common_inherited_config to the template.
2016-09-05 13:10:23 +00:00
media
Revert of move all reference to carbon api (patchset #2 id:300001 of https://codereview.webrtc.org/2321493002/ )
2016-09-08 12:35:00 +00:00
modules
This CL refactors the buffering of the incoming near-end signal inside
2016-09-08 16:49:50 +00:00
p2p
Fixed flaky StunRequestTests which depended on the wall clock
2016-09-07 00:18:58 +00:00
pc
GN Templates: Move common_inherited_config to the template.
2016-09-05 13:10:23 +00:00
sdk
Renamed RTCStatsReport to RTCLegacyStatsReport in objc files.
2016-09-09 08:36:32 +00:00
stats
GN Templates: Move common_inherited_config to the template.
2016-09-05 13:10:23 +00:00
system_wrappers
Make UMA stats creation available in the Java interface.
2016-09-09 07:11:54 +00:00
test
Change owner of webrtc/test/channel_transport to solenberg@.
2016-09-08 15:41:59 +00:00
tools
Revert of Don't use VoE legacy APIs in force_mic_volume_max tool. (patchset #5 id:80001 of https://codereview.webrtc.org/2268183007/ )
2016-09-08 14:31:02 +00:00
video
Update AvgCounter to have the ability to include last period metric for subsequent intervals without samples (e.g. for non-periodic updated stats).
2016-09-09 07:13:39 +00:00
voice_engine
FilePlayer: Remove backwards compatibility stuff that we no longer need
2016-09-08 11:32:40 +00:00
.gitignore
…
BUILD.gn
Revert of move all reference to carbon api (patchset #2 id:300001 of https://codereview.webrtc.org/2321493002/ )
2016-09-08 12:35:00 +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
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
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
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
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
Revert of move all reference to carbon api (patchset #2 id:300001 of https://codereview.webrtc.org/2321493002/ )
2016-09-08 12:35:00 +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
Add stefan@ to webrtc/OWNERS.
2016-08-10 15:49:59 +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
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +00:00
transport.h
…
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +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
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
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
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
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
Add an HD resolution perf test.
2016-08-11 14:01:03 +00:00
webrtc_tests.gypi
Revert of move all reference to carbon api (patchset #2 id:300001 of https://codereview.webrtc.org/2321493002/ )
2016-09-08 12:35:00 +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.