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webrtc_m130/webrtc/pc
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Magnus Jedvert 26d5e2e280 Revert "Add additional check when setting RTCConfiguration" 
This reverts commit 8110beda7f98623e4510f99ed51a05d126437642.

Reason for revert:
Blocks reverting https://chromium-review.googlesource.com/c/562505


Original change's description:
> Add additional check when setting RTCConfiguration
> 
> Check that ice_regather_interval_range is set only when continual
> regathering is also set.
> 
> Bug: webrtc:7969
> Change-Id: Ifcfeee744d817cf00914418d7e682f11528faf05
> Reviewed-on: https://chromium-review.googlesource.com/569358
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#19009}

TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,steveanton@webrtc.org

Change-Id: I95955bb6ab0c5d0625e55a136e3773e9b90d74e2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7969
Reviewed-on: https://chromium-review.googlesource.com/571009
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19022}
2017-07-14 14:26:45 +00:00
..
test
Revert of Make the default ctor of rtc::Thread, protected (patchset #3 id:40001 of https://codereview.webrtc.org/2981623002/ )
2017-07-13 14:06:39 +00:00
audiomonitor.cc
Adding more detail to MessageQueue::Dispatch logging.
2016-06-10 21:17:33 +00:00
audiomonitor.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
audiotrack.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
audiotrack.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
BUILD.gn
Use relative paths in GN files.
2017-07-11 13:20:45 +00:00
bundlefilter_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
bundlefilter.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
bundlefilter.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
channel_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
channel.cc
Make BaseChannel::rtp_transport_ a unique_ptr.
2017-07-12 18:48:11 +00:00
channel.h
Make BaseChannel::rtp_transport_ a unique_ptr.
2017-07-12 18:48:11 +00:00
channelmanager_unittest.cc
Revert of Make the default ctor of rtc::Thread, protected (patchset #3 id:40001 of https://codereview.webrtc.org/2981623002/ )
2017-07-13 14:06:39 +00:00
channelmanager.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
channelmanager.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
createpeerconnectionfactory.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
currentspeakermonitor_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
currentspeakermonitor.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
currentspeakermonitor.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
datachannel_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
datachannel.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
datachannel.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
DEPS
Move SrtpSession and tests to their own files.
2017-07-07 21:26:25 +00:00
dtmfsender_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
dtmfsender.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
dtmfsender.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
externalhmac.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
externalhmac.h
Remove HAVE_SRTP define and unmaintained code.
2017-03-06 19:32:22 +00:00
iceserverparsing_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
iceserverparsing.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
iceserverparsing.h
Rewrite PeerConnection integration tests using better testing practices.
2017-03-30 04:08:16 +00:00
jsepicecandidate.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
jsepsessiondescription_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
jsepsessiondescription.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
localaudiosource_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +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
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
mediamonitor.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
mediamonitor.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
mediasession_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
mediasession.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
mediasession.h
Support encrypted RTP extensions (RFC 6904)
2017-06-29 19:31:36 +00:00
mediastream_unittest.cc
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
mediastream.cc
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
mediastreamobserver.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediastreamobserver.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +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_integrationtest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
peerconnection.cc
Revert "Add additional check when setting RTCConfiguration"
2017-07-14 14:26:45 +00:00
peerconnection.h
Revert "Add additional check when setting RTCConfiguration"
2017-07-14 14:26:45 +00:00
peerconnectionendtoend_unittest.cc
Revert of Make the default ctor of rtc::Thread, protected (patchset #3 id:40001 of https://codereview.webrtc.org/2981623002/ )
2017-07-13 14:06:39 +00:00
peerconnectionfactory_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
peerconnectionfactory.cc
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
peerconnectionfactory.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
peerconnectioninterface_unittest.cc
Revert "Add additional check when setting RTCConfiguration"
2017-07-14 14:26:45 +00:00
proxy_unittest.cc
Revert of Make the default ctor of rtc::Thread, protected (patchset #3 id:40001 of https://codereview.webrtc.org/2981623002/ )
2017-07-13 14:06:39 +00:00
quicdatachannel_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
quicdatachannel.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
quicdatachannel.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
quicdatatransport_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
quicdatatransport.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
quicdatatransport.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
remoteaudiosource.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
remoteaudiosource.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtcpmuxfilter_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtcpmuxfilter.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtcpmuxfilter.h
Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport.
2017-06-01 20:22:42 +00:00
rtcstats_integrationtest.cc
Revert of Make the default ctor of rtc::Thread, protected (patchset #3 id:40001 of https://codereview.webrtc.org/2981623002/ )
2017-07-13 14:06:39 +00:00
rtcstatscollector_unittest.cc
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
rtcstatscollector.cc
RTCStatsCollector: Get track IDs from senders/receivers instead of
2017-07-12 23:22:34 +00:00
rtcstatscollector.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtpreceiver.cc
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
rtpreceiver.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtpsender.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtpsender.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtpsenderreceiver_unittest.cc
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
rtptransport_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtptransport.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtptransport.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
sctputils_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
sctputils.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
sctputils.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
srtpfilter_unittest.cc
Move SrtpSession and tests to their own files.
2017-07-07 21:26:25 +00:00
srtpfilter.cc
Move SrtpSession and tests to their own files.
2017-07-07 21:26:25 +00:00
srtpfilter.h
Move SrtpSession and tests to their own files.
2017-07-07 21:26:25 +00:00
srtpsession_unittest.cc
Move SrtpSession and tests to their own files.
2017-07-07 21:26:25 +00:00
srtpsession.cc
Move SrtpSession and tests to their own files.
2017-07-07 21:26:25 +00:00
srtpsession.h
Move SrtpSession and tests to their own files.
2017-07-07 21:26:25 +00:00
srtptestutil.h
Move SrtpSession and tests to their own files.
2017-07-07 21:26:25 +00:00
statscollector_unittest.cc
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
statscollector.cc
Report interframe delay sum in old GetStats
2017-07-07 08:26:24 +00:00
statscollector.h
Revert "Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ )"
2017-06-02 13:44:03 +00:00
streamcollection.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
trackmediainfomap_unittest.cc
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
trackmediainfomap.cc
Add media related stats (audio level etc.) to unsignaled streams.
2017-05-19 20:09:47 +00:00
trackmediainfomap.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
videocapturertracksource_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
videocapturertracksource.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
videocapturertracksource.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
videotrack_unittest.cc
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
videotrack.cc
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
videotrack.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
videotracksource.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
videotracksource.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
voicechannel.h
Simple lint fixes
2016-04-26 12:28:18 +00:00
webrtcsdp_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
webrtcsdp.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
webrtcsdp.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
webrtcsession_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
webrtcsession.cc
API for periodically regathering ICE candidates
2017-07-11 21:49:38 +00:00
webrtcsession.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
webrtcsessiondescriptionfactory.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
webrtcsessiondescriptionfactory.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00