Seth Hampson 43745937a8 Adding shampson (me) as an owner to pc/ & api/. ...

With deadbeef removed from these OWNERS files, Steve is the only OWNER
on our team. I'm adding myself, because I have worked in these
directories and it makes sense to be able to distribute the code
reviews.

NOTRY=True

Bug: None
Change-Id: I48e88a07ee42254d937391f500f273656853d98b
Reviewed-on: https://webrtc-review.googlesource.com/86980
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23826}

2018-07-03 20:39:17 +00:00

..

audio

AEC3: Reverb modeling: Including the freq shape of the tails when modeling the reverberation

2018-06-26 16:17:45 +00:00

audio_codecs

Replace rtc::Optional with absl::optional in api

2018-06-21 12:50:03 +00:00

call

Pass packet retransmission information in PacketOptions.

2018-05-29 10:12:04 +00:00

ortc

Replace rtc::Optional with absl::optional in api

2018-06-21 12:50:03 +00:00

stats

Adding "is_standardized" flag to RTCStatsMember.

2018-06-28 00:43:46 +00:00

test

Generalize SimulcastEncoderAdapter, use for H264 & VP8.

2018-06-21 15:57:43 +00:00

transport

Replace rtc::Optional with absl::optional in api

2018-06-21 12:50:03 +00:00

units

Remove incorrect test from api/units/

2018-06-28 11:37:20 +00:00

video

Add bit depth information to PlanarYuvBuffer

2018-06-26 20:23:37 +00:00

video_codecs

Add functionality to set min bitrate for single stream through RtpEncodingParameters.

2018-06-25 10:10:48 +00:00

array_view_unittest.cc

ArrayView, adding ctor for fixed-size views of const(expr) std::array.

2018-05-15 13:49:02 +00:00

array_view.h

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

audio_options.cc

Remove chromium clang style errors affecting sdk/android/media_jni

2018-04-09 13:55:49 +00:00

audio_options.h

Replace rtc::Optional with absl::optional in api

2018-06-21 12:50:03 +00:00

bitrate_constraints.h

Convert video quality test from a TEST_F to a TEST fixture.

2018-06-21 15:49:43 +00:00

BUILD.gn

Convert video quality test from a TEST_F to a TEST fixture.

2018-06-21 15:49:43 +00:00

candidate.cc

This CL removes all usages of our custom ostream << overloads.

2018-04-03 12:51:00 +00:00

candidate.h

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

cryptoparams.h

Fix ortc_api circular deps.

2017-11-15 13:31:51 +00:00

datachannelinterface.h

Replace rtc::Optional with absl::optional in api

2018-06-21 12:50:03 +00:00

DEPS

Make rtc_software_fallback_wrappers target visible.

2018-06-12 12:51:34 +00:00

dtmfsenderinterface.h

Remove unused/deprecated DTMF methods

2018-06-20 21:00:10 +00:00

fakemetricsobserver.cc

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

fakemetricsobserver.h

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

fec_controller.h

Revert "Revert "Enables PeerConnectionFactory using external fec controller""

2018-02-20 12:41:55 +00:00

jsep.cc

Fix clang style errors in api/jsep.h

2017-12-06 18:12:06 +00:00

jsep.h

Replace rtc::Optional with absl::optional in api

2018-06-21 12:50:03 +00:00

jsepicecandidate.h

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

jsepsessiondescription.h

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

mediaconstraintsinterface.cc

Replace rtc::Optional with absl::optional in api

2018-06-21 12:50:03 +00:00

mediaconstraintsinterface.h

Replace rtc::Optional with absl::optional in api

2018-06-21 12:50:03 +00:00

mediastreaminterface.cc

Final name changing of MediaStreamInterface.label() to id().

2018-03-14 20:30:52 +00:00

mediastreaminterface.h

Replace rtc::Optional with absl::optional in api

2018-06-21 12:50:03 +00:00

mediastreamproxy.h

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

mediastreamtrackproxy.h

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

mediatypes.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

mediatypes.h

Convert video quality test from a TEST_F to a TEST fixture.

2018-06-21 15:49:43 +00:00

notifier.h

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

optional_unittest.cc

Reland "Use absl::optional instead or rtc::Optional"

2018-06-12 19:13:21 +00:00

optional.h

Reland "Use absl::optional instead or rtc::Optional"

2018-06-12 19:13:21 +00:00

OWNERS

Adding shampson (me) as an owner to pc/ & api/.

2018-07-03 20:39:17 +00:00

peerconnectionfactoryproxy.h

Add PeerConnection GetRtpSender/ReceiverCapabilities

2018-06-28 12:40:07 +00:00

peerconnectioninterface.h

Add PeerConnection GetRtpSender/ReceiverCapabilities

2018-06-28 12:40:07 +00:00

peerconnectionproxy.h

Remove unused/deprecated DTMF methods

2018-06-20 21:00:10 +00:00

proxy.cc

Enable the clang style plugin in primary p2p/ target

2017-11-01 00:19:05 +00:00

proxy.h

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

refcountedbase.h

New classes RefCounter and RefCountedBase.

2017-10-23 11:46:47 +00:00

rtcerror_unittest.cc

This CL removes all usages of our custom ostream << overloads.

2018-04-03 12:51:00 +00:00

rtcerror.cc

Reland "Reland "Remove our stream << overloads from non-test build targets.""

2018-05-03 10:41:41 +00:00

rtcerror.h

Reland "Reland "Remove our stream << overloads from non-test build targets.""

2018-05-03 10:41:41 +00:00

rtceventlogoutput.h

Move RtcEventLogOutput to api/

2017-10-06 13:58:14 +00:00

rtp_headers.cc

Fix circular dependencies in webrtc_common.

2017-12-15 14:33:26 +00:00

rtp_headers.h

Replace rtc::Optional with absl::optional in api

2018-06-21 12:50:03 +00:00

rtpparameters_unittest.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtpparameters.cc

Add Rtcp parameters for PeerConnection senders

2018-05-28 09:28:59 +00:00

rtpparameters.h

Add functionality to set min bitrate for single stream through RtpEncodingParameters.

2018-06-25 10:10:48 +00:00

rtpreceiverinterface.cc

Reland "Update RTCStatsCollector to work with RtpTransceivers"

2018-02-17 00:01:39 +00:00

rtpreceiverinterface.h

Replace rtc::Optional with absl::optional in api

2018-06-21 12:50:03 +00:00

rtpsenderinterface.h

Replace rtc::Optional with absl::optional in api

2018-06-21 12:50:03 +00:00

rtptransceiverinterface.h

Replace rtc::Optional with absl::optional in api

2018-06-21 12:50:03 +00:00

setremotedescriptionobserverinterface.h

Reland "SetRemoteDescriptionObserverInterface added."

2017-11-23 19:59:48 +00:00

statstypes.cc

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

statstypes.h

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

turncustomizer.h

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

umametrics.cc

Enable the clang style plugin in primary p2p/ target

2017-11-01 00:19:05 +00:00

umametrics.h

Add collection of usage signatures on PeerConnections

2018-05-31 13:07:09 +00:00

videosourceinterface.h

Move VideoStreamEncoderInterface to api/.

2018-05-21 19:50:37 +00:00

videosourceproxy.h

Replace rtc::Optional with absl::optional in api

2018-06-21 12:50:03 +00:00