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kwiberg d313403564 rtc::Buffer: Let SetData and AppendData accept anything with .data() and .size() 
In addition to setting or appending from another Buffer, which was
already possible, this allows for e.g. std::vector and rtc::ArrayView
arguments.

Review-Url: https://codereview.webrtc.org/2293983002
Cr-Commit-Position: refs/heads/master@{#14073}
2016-09-05 14:46:24 +00:00
..
androidjunit
Re-order and remove unused Java imports.
2016-08-02 13:44:19 +00:00
api
GN Templates: Move common_inherited_config to the template.
2016-09-05 13:10:23 +00:00
audio
GN Templates: Move common_inherited_config to the template.
2016-09-05 13:10:23 +00:00
base
rtc::Buffer: Let SetData and AppendData accept anything with .data() and .size()
2016-09-05 14:46:24 +00:00
build
MB: Add WebRTC FYI bots to mb_config.pyl.
2016-09-05 13:44:02 +00:00
call
GN Templates: Move common_inherited_config to the template.
2016-09-05 13:10:23 +00:00
common_audio
GN Templates: Move common_inherited_config to the template.
2016-09-05 13:10:23 +00:00
common_video
GN Templates: Move common_inherited_config to the template.
2016-09-05 13:10:23 +00:00
examples
GN Templates: Move common_inherited_config to the template.
2016-09-05 13:10:23 +00:00
libjingle
GN Templates: Move common_inherited_config to the template.
2016-09-05 13:10:23 +00:00
media
GN Templates: Move common_inherited_config to the template.
2016-09-05 13:10:23 +00:00
modules
Use RtpPacketToSend in RtpSenderAudio.
2016-09-05 14:28:05 +00:00
p2p
GN Templates: Move common_inherited_config to the template.
2016-09-05 13:10:23 +00:00
pc
GN Templates: Move common_inherited_config to the template.
2016-09-05 13:10:23 +00:00
sdk
GN Templates: Move common_inherited_config to the template.
2016-09-05 13:10:23 +00:00
stats
GN Templates: Move common_inherited_config to the template.
2016-09-05 13:10:23 +00:00
system_wrappers
GN Templates: Move common_inherited_config to the template.
2016-09-05 13:10:23 +00:00
test
GN Templates: Move common_inherited_config to the template.
2016-09-05 13:10:23 +00:00
tools
GN Templates: Move common_inherited_config to the template.
2016-09-05 13:10:23 +00:00
video
GN Templates: Move common_inherited_config to the template.
2016-09-05 13:10:23 +00:00
voice_engine
GN Templates: Move common_inherited_config to the template.
2016-09-05 13:10:23 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
GN Templates: Move common_inherited_config to the template.
2016-09-05 13:10:23 +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
Revert of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #3 id:80001 of https://codereview.webrtc.org/2280703002/ )
2016-09-02 08:03:28 +00:00
common_types.h
Revert of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #3 id:80001 of https://codereview.webrtc.org/2280703002/ )
2016-09-02 08:03:28 +00:00
common.cc
Fix Chromium clang plugin warnings
2016-08-29 17:05:34 +00:00
common.gyp
Move webrtc/audio_*.h to webrtc/api/call
2016-08-31 14:33:14 +00:00
common.h
Fix Chromium clang plugin warnings
2016-08-29 17:05:34 +00:00
config.cc
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
config.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
DEPS
Revert of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #3 id:80001 of https://codereview.webrtc.org/2280703002/ )
2016-09-02 08:03:28 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +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
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
[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
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
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
Reland of Initial version of file wrapper
2016-09-02 11:38:43 +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.