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platform-external-webrtc/webrtc
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hbos bd3dda6c86 Renamed RTCStatsReport to RTCLegacyStatsReport in objc files. 
This is to avoid a naming conflict with webrtc::RTCStatsReport that is
surfaced if you try to include it in peerconnectioninterface.h.

Background: The current stats is very much non-spec-compliant. A new
stats collection API is underway that is meant to be spec-compliant.
Some classes in Chromium and webrtc/sdk/objc have spec-compliant names
but non-spec-compliant behavior. These are being renamed to "Legacy" so
that new spec-compliant classes can be added with the correct names.

BUG=chromium:627816
TBR=tkchin@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2313943002
Cr-Commit-Position: refs/heads/master@{#14150}
2016-09-09 08:36:32 +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
Revert of Add a DEPS gclient hook to prune corrupt mockito remote. (patchset #1 id:1 of https://codereview.webrtc.org/2326523002/ )
2016-09-08 17:55:50 +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
…
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
…
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
…
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
…
video_engine_tests.isolate
…
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
…
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests_apk.isolate
…
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.