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webrtc_m130/webrtc
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Niels Möller 47fe34c2bd Introduce an IsMutable method on VideoFrameBuffer. 
Unlike HasOneRef, it can be overridden to always return false in
immutable subclasses.

I'm also investigating overiding it in PooledI420Buffer, to directly
inherit I420Buffer but ignore the reference from the pool. Still
unclear if that will work out.

BUG=webrtc:5682

Committed: https://crrev.com/6bd10f2c1ac912cbe5addd880e559d59274c60e6
Cr-Commit-Position: refs/heads/master@{#12365}

R=magjed@webrtc.org, pbos@webrtc.org, perkj@webrtc.org

Review URL: https://codereview.webrtc.org/1881933004 .

Cr-Commit-Position: refs/heads/master@{#12404}
2016-04-18 11:03:11 +00:00
..
api
Tune QP-based quality thresholds.
2016-04-18 10:58:17 +00:00
audio
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
base
RTCCertificateGenerator added.
2016-04-15 15:55:30 +00:00
build
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
call
Suppress a flaky test: SwitchesToASTThenBackToTOFForVideo.
2016-04-15 13:34:54 +00:00
common_audio
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
common_video
Introduce an IsMutable method on VideoFrameBuffer.
2016-04-18 11:03:11 +00:00
examples
Use the new appr.tc URL
2016-04-15 14:16:19 +00:00
libjingle
Reland of move {media,p2p,pc,xmllite,xmpp}_tests.gypi files. (patchset #1 id:1 of https://codereview.webrtc.org/1846693002/ )
2016-03-31 14:23:52 +00:00
media
Introduce an IsMutable method on VideoFrameBuffer.
2016-04-18 11:03:11 +00:00
modules
Tune QP-based quality thresholds.
2016-04-18 10:58:17 +00:00
p2p
Make ReliableQuicStream::Write use base::StringPiece, not std::string
2016-04-14 18:12:51 +00:00
pc
Revert of Accept all the media profiles required by JSEP. (patchset #5 id:80001 of https://codereview.webrtc.org/1880913002/ )
2016-04-13 17:48:32 +00:00
system_wrappers
Add histogram stats for average QP per frame for VP9 (for sent video streams):
2016-04-18 09:58:52 +00:00
test
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
tools
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
video
Add histogram stats for average QP per frame for VP9 (for sent video streams):
2016-04-18 09:58:52 +00:00
voice_engine
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Replace scoped_ptr with unique_ptr in webrtc/audio/
2016-02-23 18:46:39 +00:00
audio_send_stream.h
Relanding https://codereview.webrtc.org/1715883002/ in pieces.
2016-03-11 11:06:48 +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: Update audio_sink.h location
2016-02-29 05:49:54 +00:00
call.h
Make the audio channel communicate network state changes to the call.
2016-03-22 22:32:31 +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
Tune BWE to be a bit less sensitive to spurious delay events.
2016-03-10 13:13:26 +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
Added support in the AEC for refined filter adaptation.
2016-04-15 18:23:36 +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
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
frame_callback.h
Measure encoding time on encode callbacks.
2016-02-05 10:13:41 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
Make pbos owner of additional video files.
2016-02-26 12:29:17 +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
stream.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
supplement.gypi
Roll chromium_revision 7542f07..38664e7 (377632:377790) + set SDK 10.11 on Mac
2016-02-26 11:21:18 +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
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +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
Delete method webrtc::VideoFrame::native_handle.
2016-04-15 10:43:45 +00:00
video_receive_stream.h
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +00:00
video_send_stream.h
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +00:00
webrtc_examples.gyp
Add test runner scripts for instrumentation tests
2016-04-08 13:54:32 +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
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc.gyp
GYP: Add webrtc/pc/pc.gyp:* to 'All' target.
2016-04-01 09:15:05 +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.