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webrtc_m130/webrtc/api
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Steve Anton 300bf8e14b Reinstate "API for periodically regathering ICE candidates" 
Use rtc::SystemTimeNanos() instead of std::random_device() for PRNG seed
to avoid crashing when /dev/urandom is unavailable.

This reverts commit 3beb20720db349f651c2c04970c45b1b171c025c.

Bug: webrtc:7969
Change-Id: I5ed58a789939ee4caa99ac3abf9cab18e3e19c69
Reviewed-on: https://chromium-review.googlesource.com/572070
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19033}
2017-07-14 22:26:05 +00:00
..
audio
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
audio_codecs
Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )
2017-07-10 12:58:42 +00:00
call
Reland of Make AudioSinkInterface::Data hold a const pointer to the audio buffer. (patchset #1 id:1 of https://codereview.webrtc.org/2877013002/ )
2017-05-12 05:07:37 +00:00
ortc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
stats
Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy.
2017-07-14 19:17:49 +00:00
test
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
video
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
video_codecs
Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )
2017-07-10 12:58:42 +00:00
BUILD.gn
Use relative paths in GN files.
2017-07-11 13:20:45 +00:00
datachannel.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
datachannelinterface.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
DEPS
Opus implementation of the AudioEncoderFactoryTemplate API
2017-06-30 11:23:22 +00:00
dtmfsenderinterface.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
fakemetricsobserver.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
fakemetricsobserver.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
jsep.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
jsepicecandidate.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
jsepsessiondescription.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
mediaconstraintsinterface.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
mediaconstraintsinterface.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
mediastream.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediastreaminterface.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediastreaminterface.h
Revert of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt (patchset #2 id:20001 of https://codereview.webrtc.org/2964863002/ )
2017-07-10 09:40:49 +00:00
mediastreamproxy.h
Adding more comments to every header file in api/ subdirectory.
2017-02-08 09:38:21 +00:00
mediastreamtrack.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediastreamtrackproxy.h
Adding more comments to every header file in api/ subdirectory.
2017-02-08 09:38:21 +00:00
mediatypes.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
mediatypes.h
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
notifier.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
OWNERS
Don't add or rename files in webrtc/ and webrtc/api/ without a proper review
2017-05-08 13:01:54 +00:00
peerconnectionfactoryproxy.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
peerconnectioninterface.h
Reinstate "API for periodically regathering ICE candidates"
2017-07-14 22:26:05 +00:00
peerconnectionproxy.h
Relanding: Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP.
2017-06-02 21:37:37 +00:00
proxy.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtcerror_unittest.cc
Move rtc_api_unittests into rtc_unittests.
2017-02-24 09:17:43 +00:00
rtcerror.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtcerror.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtpparameters.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtpreceiverinterface.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtpsender.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
rtpsenderinterface.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
statstypes.cc
Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy.
2017-07-14 19:17:49 +00:00
statstypes.h
Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy.
2017-07-14 19:17:49 +00:00
streamcollection.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
umametrics.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
videosourceproxy.h
Adding more comments to every header file in api/ subdirectory.
2017-02-08 09:38:21 +00:00
videotracksource.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
webrtcsdp.h
Added api/webrtcsdp.h forwarding header to work around upstream projects.
2017-01-24 15:13:59 +00:00