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webrtc_m130/webrtc
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kjellander a811968eec GN: Exclude frame_analyzer and rgba_to_i420_converter from Chromium builds. 
Chromium's ASan bot fails to link rgba_to_i420_converter so let's exclude it until resolved.

NOTRY=True
BUG=webrtc:5970
TBR=jochen@chromium.org

Review-Url: https://codereview.webrtc.org/2039173002
Cr-Commit-Position: refs/heads/master@{#13047}
2016-06-06 09:00:25 +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
Revert of Improving the fake clock and using it to fix a flaky STUN timeout test. (patchset #10 id:180001 of https://codereview.webrtc.org/2024813004/ )
2016-06-03 23:05:30 +00:00
audio
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
base
Revert of Improving the fake clock and using it to fix a flaky STUN timeout test. (patchset #10 id:180001 of https://codereview.webrtc.org/2024813004/ )
2016-06-03 23:05:30 +00:00
build
Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc}
2016-06-03 10:09:40 +00:00
call
Add missing dependencies on audio, video and call to the new GN files.
2016-06-03 20:14:37 +00:00
common_audio
Add clz functions (Count number of Leading Zero bits), 32-and 64-bit variants
2016-06-02 11:02:22 +00:00
common_video
Add enum value ScaleMethod::kFilterLinear.
2016-06-03 13:45:10 +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 GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc}
2016-06-03 10:09:40 +00:00
modules
Make WebRTCAudioTrack class public.
2016-06-03 22:33:39 +00:00
p2p
Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc}
2016-06-03 10:09:40 +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
Add flag in ios to support disabling high-cost networks.
2016-06-03 23:31:45 +00:00
system_wrappers
GN: Exclude tools and tests in Chromium builds.
2016-06-02 11:29:13 +00:00
test
Collapse most stdout spammy output of webrtc_perf_tests with PrintResultList
2016-06-03 16:27:45 +00:00
tools
GN: Exclude frame_analyzer and rgba_to_i420_converter from Chromium builds.
2016-06-06 09:00:25 +00:00
video
Always send RED headers if configured.
2016-06-03 07:16:54 +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
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 rtc_unittests
2016-06-03 13:57:06 +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
…
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
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
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
config.h
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +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
Disable libyuv jpeg support on Android
2016-05-17 15:39:15 +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
Reland of Delete webrtc::VideoFrame methods buffer and stride. (patchset #1 id:1 of https://codereview.webrtc.org/1983583002/ )
2016-05-17 11:05:51 +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
Revert of Improving the fake clock and using it to fix a flaky STUN timeout test. (patchset #10 id:180001 of https://codereview.webrtc.org/2024813004/ )
2016-06-03 23:05:30 +00:00
webrtc.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

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.