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02455b27a278370368ea44154dba5c1504d4f1d5
platform-external-webrtc/webrtc/pc
History
jbauch d48f488bed Support GCM ciphers even if ENABLE_EXTERNAL_AUTH is defined. 
With ENABLE_EXTERNAL_AUTH, external auth will only be used depending
on the selected cipher (allowed for non-GCM, not allowed for GCM).

BUG=webrtc:5222, chromium:628400

Review-Url: https://codereview.webrtc.org/2720663003
Cr-Commit-Position: refs/heads/master@{#16955}
2017-03-01 23:34:36 +00:00
..
test
Switching some interfaces to use std::unique_ptr<>.
2017-02-11 04:13:37 +00:00
audiomonitor.cc
Adding more detail to MessageQueue::Dispatch logging.
2016-06-10 21:17:33 +00:00
audiomonitor.h
Simple lint fixes
2016-04-26 12:28:18 +00:00
audiotrack.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
audiotrack.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
BUILD.gn
Support GCM ciphers even if ENABLE_EXTERNAL_AUTH is defined.
2017-03-01 23:34:36 +00:00
bundlefilter_unittest.cc
Fix license headers in webrtc/pc
2016-02-12 08:05:07 +00:00
bundlefilter.cc
Fix license headers in webrtc/pc
2016-02-12 08:05:07 +00:00
bundlefilter.h
Simple lint fixes
2016-04-26 12:28:18 +00:00
channel_unittest.cc
Roll chromium_revision 33a7a547b9..0e44c5e141 (452838:453130)
2017-02-27 03:53:40 +00:00
channel.cc
Support GCM ciphers even if ENABLE_EXTERNAL_AUTH is defined.
2017-03-01 23:34:36 +00:00
channel.h
Rename "PacketTransportInterface" to "PacketTransportInternal".
2017-02-10 19:31:50 +00:00
channelmanager_unittest.cc
Removing unnecessary parameters from CreateXChannel methods.
2017-02-11 07:44:49 +00:00
channelmanager.cc
Support GCM ciphers even if ENABLE_EXTERNAL_AUTH is defined.
2017-03-01 23:34:36 +00:00
channelmanager.h
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
currentspeakermonitor_unittest.cc
Disable flaky tests on iOS
2017-02-27 06:10:14 +00:00
currentspeakermonitor.cc
Change default timestamp to 64 bits in all webrtc directories.
2016-05-06 18:29:27 +00:00
currentspeakermonitor.h
Reland of place basictypes.h with stdint.h for int_t types. (patchset #1 id:1 of https://codereview.webrtc.org/2603203003/ )
2017-01-02 16:42:32 +00:00
datachannel_unittest.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
datachannel.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
datachannel.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
DEPS
Add CreatePeerConnectionFactory overloads that take audio codec factory args
2017-01-31 09:48:08 +00:00
dtmfsender_unittest.cc
Move DTMF sender to RtpSender (as opposed to WebRtcSession).
2017-02-02 04:27:00 +00:00
dtmfsender.cc
Move DTMF sender to RtpSender (as opposed to WebRtcSession).
2017-02-02 04:27:00 +00:00
dtmfsender.h
Move DTMF sender to RtpSender (as opposed to WebRtcSession).
2017-02-02 04:27:00 +00:00
externalhmac.cc
Remove remnants of libsrtp1
2016-10-25 16:47:31 +00:00
externalhmac.h
Reland of place basictypes.h with stdint.h for int_t types. (patchset #1 id:1 of https://codereview.webrtc.org/2603203003/ )
2017-01-02 16:42:32 +00:00
fakemediacontroller.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
jsepicecandidate.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
jsepsessiondescription_unittest.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
jsepsessiondescription.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
localaudiosource_unittest.cc
Remove PC factory options param from LocalAudioSource::Create.
2017-02-11 05:26:48 +00:00
localaudiosource.cc
Remove PC factory options param from LocalAudioSource::Create.
2017-02-11 05:26:48 +00:00
localaudiosource.h
Remove PC factory options param from LocalAudioSource::Create.
2017-02-11 05:26:48 +00:00
mediaconstraintsinterface_unittest.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediacontroller.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediacontroller.h
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
mediamonitor.cc
Delete unneeded includes of base/common.h.
2017-01-25 09:47:24 +00:00
mediamonitor.h
Simple lint fixes
2016-04-26 12:28:18 +00:00
mediasession_unittest.cc
Use the same draft version in SDP data channel answers as used in the offer.
2017-02-18 03:48:38 +00:00
mediasession.cc
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
mediasession.h
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
mediastream_unittest.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediastream.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediastream.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediastreamobserver.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediastreamobserver.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediastreamtrack.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
OWNERS
Make hbos and hta rtcstats* OWNERS of webrtc/pc, not webrtc/api.
2017-02-28 14:46:44 +00:00
peerconnection_unittest.cc
Accept SDP with TRANSPORT attributes missing from bundled m= sections.
2017-02-23 03:35:18 +00:00
peerconnection.cc
Test field trial group with startswith rather than equals.
2017-02-28 16:50:47 +00:00
peerconnection.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
peerconnectionendtoend_unittest.cc
Relanding: Removing #defines previously used for building without BoringSSL/OpenSSL.
2017-01-24 03:39:57 +00:00
peerconnectionfactory_unittest.cc
Switching some interfaces to use std::unique_ptr<>.
2017-02-11 04:13:37 +00:00
peerconnectionfactory.cc
Changing some PeerConnection-related comments.
2017-02-13 17:49:58 +00:00
peerconnectionfactory.h
Switching some interfaces to use std::unique_ptr<>.
2017-02-11 04:13:37 +00:00
peerconnectioninterface_unittest.cc
Adding RTCErrorOr class to be used by ORTC APIs.
2017-02-17 07:31:33 +00:00
proxy_unittest.cc
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
quicdatachannel_unittest.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
quicdatachannel.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
quicdatachannel.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
quicdatatransport_unittest.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
quicdatatransport.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
quicdatatransport.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
remoteaudiosource.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
remoteaudiosource.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
rtcpmuxfilter_unittest.cc
Some cleanup in BaseChannel RTCP mux code.
2016-08-22 23:00:37 +00:00
rtcpmuxfilter.cc
Some cleanup in BaseChannel RTCP mux code.
2016-08-22 23:00:37 +00:00
rtcpmuxfilter.h
Remove extra uses of basictypes.h.
2016-12-09 14:54:08 +00:00
rtcstats_integrationtest.cc
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
rtcstatscollector_unittest.cc
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
rtcstatscollector.cc
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
rtcstatscollector.h
Move trackmediainfomap files from api/ to pc/.
2017-02-12 03:07:31 +00:00
rtpreceiver.cc
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
rtpreceiver.h
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
rtpsender.cc
Protect APM in webkit builds.
2017-02-03 14:37:05 +00:00
rtpsender.h
Move DTMF sender to RtpSender (as opposed to WebRtcSession).
2017-02-02 04:27:00 +00:00
rtpsenderreceiver_unittest.cc
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
sctputils_unittest.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
sctputils.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
sctputils.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
srtpfilter_unittest.cc
Support GCM ciphers even if ENABLE_EXTERNAL_AUTH is defined.
2017-03-01 23:34:36 +00:00
srtpfilter.cc
Support GCM ciphers even if ENABLE_EXTERNAL_AUTH is defined.
2017-03-01 23:34:36 +00:00
srtpfilter.h
Support GCM ciphers even if ENABLE_EXTERNAL_AUTH is defined.
2017-03-01 23:34:36 +00:00
statscollector_unittest.cc
Switching some interfaces to use std::unique_ptr<>.
2017-02-11 04:13:37 +00:00
statscollector.cc
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +00:00
statscollector.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
streamcollection.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
trackmediainfomap_unittest.cc
TrackMediaInfoMap: Allow same SSRC for send and receive side.
2017-02-20 10:05:13 +00:00
trackmediainfomap.cc
TrackMediaInfoMap: Allow same SSRC for send and receive side.
2017-02-20 10:05:13 +00:00
trackmediainfomap.h
Move trackmediainfomap files from api/ to pc/.
2017-02-12 03:07:31 +00:00
videocapturertracksource_unittest.cc
Switching some interfaces to use std::unique_ptr<>.
2017-02-11 04:13:37 +00:00
videocapturertracksource.cc
Fix occasional race in VideoCapturerTrackSource seen by memcheck bot.
2017-02-25 21:37:59 +00:00
videocapturertracksource.h
Fix occasional race in VideoCapturerTrackSource seen by memcheck bot.
2017-02-25 21:37:59 +00:00
videotrack_unittest.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
videotrack.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
videotrack.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
videotracksource.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
videotracksource.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
voicechannel.h
Simple lint fixes
2016-04-26 12:28:18 +00:00
webrtcsdp_unittest.cc
Use the same draft version in SDP data channel answers as used in the offer.
2017-02-18 03:48:38 +00:00
webrtcsdp.cc
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
webrtcsdp.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
webrtcsession_unittest.cc
Switching some interfaces to use std::unique_ptr<>.
2017-02-11 04:13:37 +00:00
webrtcsession.cc
Accept SDP with TRANSPORT attributes missing from bundled m= sections.
2017-02-23 03:35:18 +00:00
webrtcsession.h
Move DTMF sender to RtpSender (as opposed to WebRtcSession).
2017-02-02 04:27:00 +00:00
webrtcsessiondescriptionfactory.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
webrtcsessiondescriptionfactory.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00