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webrtc_m130/webrtc
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perkj 5f8ebaeffd Add limitations of number of frames that can be created in I420BufferPool::CreateBuffer. 
If more than 60 frames are created and not returned, the implementation will crash.

I420BufferPool are currently used by the VP8 decoder, Quality scaler and VideoFrameFactory.

BUG=b/31390397
NOTRY=true  // Because of failing gclient runhooks on some bots

Review-Url: https://codereview.webrtc.org/2370653003
Cr-Commit-Position: refs/heads/master@{#14395}
2016-09-27 10:47:48 +00:00
..
androidjunit
Make android_junit_tests use LocalRobolectricTestRunner
2016-09-18 19:28:15 +00:00
api
Return texture frame when dropping frames in CameraCapturer.
2016-09-23 08:27:38 +00:00
audio
GN: Change rtc_source_set targets --> rtc_static_library
2016-09-23 07:38:58 +00:00
base
Remove unnecessary jsoncpp includes.
2016-09-26 09:52:40 +00:00
build
iOS: Always build H264 HW encoder/decoder
2016-09-23 14:01:44 +00:00
call
Unify rtcp packet setters
2016-09-27 08:37:51 +00:00
common_audio
GN: Change rtc_source_set targets --> rtc_static_library
2016-09-23 07:38:58 +00:00
common_video
Add limitations of number of frames that can be created in I420BufferPool::CreateBuffer.
2016-09-27 10:47:48 +00:00
examples
Reland of name AppRTCDemo on Android and iOS to AppRTCMobile (patchset #1 id:1 of https://codereview.webrtc.org/2358133003/ )
2016-09-27 08:10:52 +00:00
libjingle
GN: Change rtc_source_set targets --> rtc_static_library
2016-09-23 07:38:58 +00:00
media
Reland of Move MutableDataY{,U,V} methods to I420Buffer only. (patchset #1 id:1 of https://codereview.webrtc.org/2354223002/ )
2016-09-27 07:17:40 +00:00
modules
Posting Opus's set-force-channels functionality to WebRTC.
2016-09-27 09:08:54 +00:00
p2p
Delete unused file sessionid.h.
2016-09-27 06:28:51 +00:00
pc
GN: Change rtc_source_set targets --> rtc_static_library
2016-09-23 07:38:58 +00:00
sdk
GN: Change rtc_source_set targets --> rtc_static_library
2016-09-23 07:38:58 +00:00
stats
GN: Change rtc_source_set targets --> rtc_static_library
2016-09-23 07:38:58 +00:00
system_wrappers
GN: Change rtc_source_set targets --> rtc_static_library
2016-09-23 07:38:58 +00:00
test
GN: Change rtc_source_set targets --> rtc_static_library
2016-09-23 07:38:58 +00:00
tools
header_usage.sh script: Exclude matches in gyp and gn files.
2016-09-26 18:44:36 +00:00
video
Unify rtcp packet setters
2016-09-27 08:37:51 +00:00
voice_engine
GN: Change rtc_source_set targets --> rtc_static_library
2016-09-23 07:38:58 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
Reland of name AppRTCDemo on Android and iOS to AppRTCMobile (patchset #1 id:1 of https://codereview.webrtc.org/2358133003/ )
2016-09-27 08:10:52 +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
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
Add new decoding statistics for muted output
2016-09-20 08:47:19 +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
Reland of move all reference to carbon api (patchset #1 id:1 of https://codereview.webrtc.org/2317343003/ )
2016-09-09 10:19:54 +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
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
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_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
Replace VideoCapturerInput with VideoSinkInterface.
2016-09-16 14:53:48 +00:00
webrtc_examples.gyp
Reland of name AppRTCDemo on Android and iOS to AppRTCMobile (patchset #1 id:1 of https://codereview.webrtc.org/2358133003/ )
2016-09-27 08:10:52 +00:00
webrtc_tests.gypi
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +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.