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kjellander 4d02a358b4 GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} 
Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in
preparation for removing src/third_party/libjingle in Chromium.

BUG=webrtc:4256
NOTRY=True

Review-Url: https://codereview.webrtc.org/1929633002
Cr-Commit-Position: refs/heads/master@{#12724}
2016-05-13 12:52:20 +00:00
..
api
GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc}
2016-05-13 12:52:20 +00:00
audio
Remove SendPacer from ViEEncoder and make sure SendPacer starts at a valid bitrate
2016-05-11 13:01:19 +00:00
base
JNI+mm: Generate certificate if non-default key type is specified.
2016-05-13 11:50:50 +00:00
build
GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc}
2016-05-13 12:52:20 +00:00
call
Refactor before implementing per stream suspension.
2016-05-13 08:43:56 +00:00
common_audio
Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ )
2016-05-11 12:07:33 +00:00
common_video
Reland of Delete webrtc::VideoFrame methods buffer and stride. (patchset #1 id:1 of https://codereview.webrtc.org/1935443002/ )
2016-05-13 11:12:48 +00:00
examples
Add magjed@ and perkj@ as webrtc/examples/ owners
2016-05-12 10:48:26 +00:00
libjingle
GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc}
2016-05-13 12:52:20 +00:00
media
GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc}
2016-05-13 12:52:20 +00:00
modules
Reland of Delete webrtc::VideoFrame methods buffer and stride. (patchset #1 id:1 of https://codereview.webrtc.org/1935443002/ )
2016-05-13 11:12:48 +00:00
p2p
GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc}
2016-05-13 12:52:20 +00:00
pc
GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc}
2016-05-13 12:52:20 +00:00
sdk
JNI+mm: Generate certificate if non-default key type is specified.
2016-05-13 11:50:50 +00:00
system_wrappers
Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ )
2016-05-11 12:07:33 +00:00
test
Reland of Delete webrtc::VideoFrame methods buffer and stride. (patchset #1 id:1 of https://codereview.webrtc.org/1935443002/ )
2016-05-13 11:12:48 +00:00
tools
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +00:00
video
Reland of Delete webrtc::VideoFrame methods buffer and stride. (patchset #1 id:1 of https://codereview.webrtc.org/1935443002/ )
2016-05-13 11:12:48 +00:00
voice_engine
Removing some old code which looked like it had to do with NACK handling but in reality did nothing.
2016-05-11 11:05:13 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
audio_send_stream.h
Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ )
2016-05-11 12:07:33 +00:00
audio_sink.h
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +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
GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc}
2016-05-13 12:52:20 +00:00
call.h
Update the call when the network route changes
2016-04-19 22:41:53 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Separating voice activity flag from audio level in RtpHeaderExtension.
2015-08-10 13:08:46 +00:00
common_types.h
Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket.
2016-05-03 06:44:11 +00:00
common.gyp
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
common.h
#include "webrtc/base/constructormagic.h" where appropriate
2016-04-26 15:14:48 +00:00
config.cc
Change transport sequence number extension strings to specify what revision is implemented.
2016-03-01 15:27:27 +00:00
config.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
DEPS
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
Add root owners to webrtc/OWNERS
2016-05-11 12:47:43 +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_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
supplement.gypi
Remove use_openssl from webrtc
2016-04-18 18:13:17 +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
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Remove QualityScaler framerate reduction.
2016-04-26 11:37:20 +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
Reland of Delete webrtc::VideoFrame methods buffer and stride. (patchset #1 id:1 of https://codereview.webrtc.org/1935443002/ )
2016-05-13 11:12:48 +00:00
video_receive_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
video_send_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
webrtc_examples.gyp
Build dynamic iOS SDK.
2016-04-27 08:54:27 +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
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
New parser for event log. Manually parse the outermost EventStream to more easily deal with corrupt or partially written logs.
2016-05-13 07:43:04 +00:00
webrtc.gyp
New parser for event log. Manually parse the outermost EventStream to more easily deal with corrupt or partially written logs.
2016-05-13 07:43:04 +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.