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webrtc_m130/api
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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
…
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
…
candidate.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
cryptoparams.h
…
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
…
jsep.cc
…
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
…
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
…
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
…
proxy.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
refcountedbase.h
…
rtcerror_unittest.cc
…
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
…
rtp_headers.cc
…
rtp_headers.h
Replace rtc::Optional with absl::optional in api
2018-06-21 12:50:03 +00:00
rtpparameters_unittest.cc
…
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
…
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
…
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
…
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