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platform-external-webrtc/webrtc
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kjellander 94cee3111c GN: Enable api,media,pc and p2p for the 'webrtc' target. 
These parts were commented out to avoid breaking the Chromium
WebRTC FYI bots. Include them in the WebRTC build to make our bots
build as many as possible of our GN targets.

BUG=webrtc:5949
NOTRY=True
TBR=phoglund@webrtc.org

Review-Url: https://codereview.webrtc.org/2054903002
Cr-Commit-Position: refs/heads/master@{#13097}
2016-06-10 08:57:05 +00:00
..
androidjunit
Reland of Android: Change camera fps range selection (patchset #1 id:1 of https://codereview.webrtc.org/2021233002/ )
2016-06-01 07:44:07 +00:00
api
RTCPeerConnectionInterface.mm createNativeConfiguration and other clean-up.
2016-06-09 10:18:35 +00:00
audio
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
base
Moves macros ACCESS_ON/RUN_ON from thread_annotations to thread_checker
2016-06-09 14:55:32 +00:00
build
iOS: Disable Goma for iOS GN bots.
2016-06-06 16:09:55 +00:00
call
Implementing auto pausing of video streams.
2016-06-09 15:21:30 +00:00
common_audio
GN: Add common_audio_unittests and common_video_unittests
2016-06-08 08:28:15 +00:00
common_video
Reland of New method I420Buffer::SetToBlack. (patchset #1 id:1 of https://codereview.webrtc.org/2049023002/ )
2016-06-09 07:31:46 +00:00
examples
Use scoped_refptr for On(Add|Remove)Stream and OnDataChannel.
2016-05-31 20:02:30 +00:00
libjingle
Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc}
2016-06-03 10:09:40 +00:00
media
Reland of New method I420Buffer::SetToBlack. (patchset #1 id:1 of https://codereview.webrtc.org/2049023002/ )
2016-06-09 07:31:46 +00:00
modules
GN: Enable api,media,pc and p2p for the 'webrtc' target.
2016-06-10 08:57:05 +00:00
p2p
Before validating a STUN packet, check it's big enough for a header.
2016-06-09 15:45:52 +00:00
pc
Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc}
2016-06-03 10:09:40 +00:00
sdk
RTCPeerConnectionInterface.mm createNativeConfiguration and other clean-up.
2016-06-09 10:18:35 +00:00
system_wrappers
GN: Fix errors in rtc_include_tests conditions
2016-06-06 21:04:16 +00:00
test
GN: Enable api,media,pc and p2p for the 'webrtc' target.
2016-06-10 08:57:05 +00:00
tools
Change name of files and class in agc/histogram* in order to avoid issue file-name clash in build files
2016-06-08 13:42:08 +00:00
video
Implementing auto pausing of video streams.
2016-06-09 15:21:30 +00:00
voice_engine
AudioDecoderIsacT: Require caller to always specify sample rate
2016-06-02 09:59:02 +00:00
.gitignore
…
audio_receive_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
audio_send_stream.h
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +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
…
BUILD.gn
GN: Enable api,media,pc and p2p for the 'webrtc' target.
2016-06-10 08:57:05 +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
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
common_types.h
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
common.gyp
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +00:00
common.h
#include "webrtc/base/constructormagic.h" where appropriate
2016-04-26 15:14:48 +00:00
config.cc
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
config.h
Add sender controlled playout delay limits
2016-06-08 07:24:30 +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
…
OWNERS
Add root owners to webrtc/OWNERS
2016-05-11 12:47:43 +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
Roll chromium_revision 086802955f..7fa6701bc5 (396351:398458)
2016-06-08 11:05:27 +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
Remove runtime NEON detection
2016-05-17 17:56:48 +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
Add sender controlled playout delay limits
2016-06-08 07:24:30 +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
Remove metrics_default from rtc_media dependencies.
2016-06-01 12:28:57 +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
…
webrtc_tests.gypi
GN: Add SDK tests to rtc_unittests.
2016-06-08 08:14:22 +00:00
webrtc.gyp
GN: Add SDK tests to rtc_unittests.
2016-06-08 08:14:22 +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.