These metrics by themselves won't be as useful, unless they can be correlated to the use of the
feature 'WebRtcHideLocalIpsWithMdns'. This can be done by running a finch experiment where we turn
the feature on for a % of users, we can then compare these metrics for users with and without
the feature turned on.
A complementary change is required in Chrome:
tools/metrics/histograms/enums.xml
Bug: webrtc:9605 webrtc:10091 chromium:914452
Change-Id: Ibc6d16dec95a8e3943ce40063c02903769fe1cb4
Reviewed-on: https://webrtc-review.googlesource.com/c/113321
Commit-Queue: Jeroen de Borst <jeroendb@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26003}
The address and the related address of local candidates are sanitized
accordingly when the mDNS concealment of local IPs is enabled. Also,
remote hostname candidates created from signaling are sanitized in stats
as well. A couple of unit tests are revised to reflect the desired
behavior of AsyncResolverInterface so that when a hostname candidate is
resolved, the hostname is kept in the candidate address.
Bug: webrtc:9605, chromium:914452
Change-Id: Iad9ad04ce4e50304e44cf04b15b97a7ae2dec960
Reviewed-on: https://webrtc-review.googlesource.com/c/113643
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25996}
Using declarations should use fully qualified names (with leading `::`)
unless they are referring to a name inside the current namespace.
Source: https://abseil.io/tips/119.
This CL removes a lot of "using webrtc::*" adding a namespace to the
tests. It also removes some unneeded "using" declarations.
Bug: webrtc:9855
Change-Id: Id6eb843e9dcee2e458b1ffd0c499df390fa9c45d
Reviewed-on: https://webrtc-review.googlesource.com/c/114001
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25987}
These changes simplify the code, and also fix the issue where the peerconnectionstate would sometimes return to "new" during connection setup.
Bug: webrtc:9308
Change-Id: I895cd2f94a2b9688c821cca64d1a077317b99d44
Reviewed-on: https://webrtc-review.googlesource.com/c/111964
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25942}
Prior to this CL, if the "a=msid" attribute was missing it was treated
the same as if "no streams" were explicitly signaled (a=msid:-); the
receivers would not be associated with any streams.
In order to support legacy endpoints that don't recognize "a=msid" that
assume the Plan B behavior of a stream being created anyway, this CL
creates a stream with a random ID in such cases. For background, see
https://github.com/web-platform-tests/wpt/pull/14054.
Bug: chromium:907508
Change-Id: I9d9dd0e4ba8f9941f8652f4d7873adc560777cd9
Reviewed-on: https://webrtc-review.googlesource.com/c/112900
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25901}
The method for looking up track ID by SSRC was never updated for
Unified Plan so it only looked at the first audio section and the
first video section.
This CL changes the method to look through all audio and video
media sections rather than just the first.
Bug: chromium:906988
Change-Id: Ie79e6162b2bd24b8ac9e983b5fa7360c96f030da
Reviewed-on: https://webrtc-review.googlesource.com/c/112223
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25833}
This reverts commit 1e87b4f32b73526f9caaae2a7bccfbd0cd84dcb9.
Reason for revert: Breaks internal project
Original change's description:
> Replace the IceConnectionState implementation.
>
> PeerConnection::ice_connection_state() used to return a value based on both DTLS and ICE transports.
> Now that we have PeerConnection::peer_connection_state() to fill that role we can change the implementation of ice_connection_state over to match the spec.
>
> Bug: webrtc:6145
> Change-Id: Ia4f348f728f24faf4b976c63dea2187bb1f01ef0
> Reviewed-on: https://webrtc-review.googlesource.com/c/108780
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25773}
TBR=kwiberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org,jonasolsson@webrtc.org
Change-Id: Icc4368d120a4167286fa6ba2e884a3650b453eff
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6145
Reviewed-on: https://webrtc-review.googlesource.com/c/111925
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25775}
PeerConnection::ice_connection_state() used to return a value based on both DTLS and ICE transports.
Now that we have PeerConnection::peer_connection_state() to fill that role we can change the implementation of ice_connection_state over to match the spec.
Bug: webrtc:6145
Change-Id: Ia4f348f728f24faf4b976c63dea2187bb1f01ef0
Reviewed-on: https://webrtc-review.googlesource.com/c/108780
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25773}
This changes PeerConnection to allow sending and receiving data channel
messages over the media transport. If |use_media_transport_for_data_channels|
is set, PeerConnection will use a DCT_MEDIA_TRANSPORT mode for data
channels.
DCT_MEDIA_TRANSPORT acts exactly like DCT_SCTP within the data channel
and peer connection layers. On the transport layer, it uses the media
transport instead of SCTP. It appears as an RTP data channel in SDP
(just as media over media-transport appears as RTP in SDP).
Bug: webrtc:9719
Change-Id: I6a90142bd3f43668479c825ed02689dcd0d58b78
Reviewed-on: https://webrtc-review.googlesource.com/c/109740
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25575}
Underscore methods in the middle of classes is against the chromium style guide
this change is part of a long series of changes to refactor crypto code in
WebRTC to conform to the chromium standard better.
1. ssl_cert() -> GetSSLCertificate()
2. ssl_cert_chain() -> GetSSLCertificateChain()
3. Small tidying up in rtccertificategenerator.cc
Bug: webrtc:9860
Change-Id: I670f76e31d6d4f873034edb72d958b3c227379cb
Reviewed-on: https://webrtc-review.googlesource.com/c/107802
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25371}
Compute these states in jseptransportController and store them. Eventually they should be passed on to the peer connection observer and exposed in the blink layer.
Bug: webrtc:9308
Change-Id: Ifdec39c24a607fcb8211c4acf6b9704eaff371b1
Reviewed-on: https://webrtc-review.googlesource.com/c/103506
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25288}
Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class
that only handles SRTP configuration to a more generic structure that can be
used and extended for all per peer connection CryptoOptions that can be on a
given PeerConnection.
Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be
accessed as crypto_options.srtp.whatever_option_name. This is more inline with
other structures we have in WebRTC such as VideoConfig. As additional features
are added over time this will allow the structure to remain compartmentalized
and concerned components can only request a subset of the overall configuration
structure e.g:
void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config);
In addition to this it made little sense for sslstreamadapter.h to hold all
Srtp related configuration options. The header has become loo large and takes on
too many responsibilities and spilting this up will lead to more maintainable
code going forward.
This will be used in a future CL to enable configuration options for the newly
supported Frame Crypto.
Reland Fix:
- cryptooptions.h - now has enable_aes128_sha1_32_crypto_cipher as an optional
root level configuration.
- peerconnectionfactory - If this optional is set will now overwrite the
underyling value.
This along with the other field will be deprecated once dependent projects
are updated.
TBR=sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org
Bug: webrtc:9681
Change-Id: Iaa6b741baafb85d352e42f54226119f19d97151d
Reviewed-on: https://webrtc-review.googlesource.com/c/105560
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25135}
This reverts commit ac2f3d14e45398930bc35ff05ed7a3b9b617d328.
Reason for revert: Breaks downstream project
Original change's description:
> Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h
>
> Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class
> that only handles SRTP configuration to a more generic structure that can be
> used and extended for all per peer connection CryptoOptions that can be on a
> given PeerConnection.
>
> Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be
> accessed as crypto_options.srtp.whatever_option_name. This is more inline with
> other structures we have in WebRTC such as VideoConfig. As additional features
> are added over time this will allow the structure to remain compartmentalized
> and concerned components can only request a subset of the overall configuration
> structure e.g:
>
> void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config);
>
> In addition to this it made little sense for sslstreamadapter.h to hold all
> Srtp related configuration options. The header has become loo large and takes on
> too many responsibilities and spilting this up will lead to more maintainable
> code going forward.
>
> This will be used in a future CL to enable configuration options for the newly
> supported Frame Crypto.
>
> Change-Id: I99d1be36740c59548c8e62db52d68d738649707f
> Bug: webrtc:9681
> Reviewed-on: https://webrtc-review.googlesource.com/c/105180
> Reviewed-by: Emad Omara <emadomara@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25130}
TBR=steveanton@webrtc.org,sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org,benwright@webrtc.org
Bug: webrtc:9681
Change-Id: Ib0075c477c951b540d4deecb3b0cf8cf86ba0fff
Reviewed-on: https://webrtc-review.googlesource.com/c/105541
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25133}
Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class
that only handles SRTP configuration to a more generic structure that can be
used and extended for all per peer connection CryptoOptions that can be on a
given PeerConnection.
Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be
accessed as crypto_options.srtp.whatever_option_name. This is more inline with
other structures we have in WebRTC such as VideoConfig. As additional features
are added over time this will allow the structure to remain compartmentalized
and concerned components can only request a subset of the overall configuration
structure e.g:
void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config);
In addition to this it made little sense for sslstreamadapter.h to hold all
Srtp related configuration options. The header has become loo large and takes on
too many responsibilities and spilting this up will lead to more maintainable
code going forward.
This will be used in a future CL to enable configuration options for the newly
supported Frame Crypto.
Change-Id: I99d1be36740c59548c8e62db52d68d738649707f
Bug: webrtc:9681
Reviewed-on: https://webrtc-review.googlesource.com/c/105180
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25130}
This CL removes some deprecated build targets (and their headers)
from system_wrappers:
- field_trial_api
- field_trial_default
- metrics_api
- metrics_default
It also refreshes all the dependencies on field_trial.h and metrics.h.
A nice side effect is that it is finally possible to remove 'nogncheck'
from the following files (when it was used with field_trial_default
and metrics_default):
- sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h
- sdk/android/src/jni/pc/peerconnectionfactory.cc
- sdk/objc/api/peerconnection/RTCFieldTrials.mm
Bug: webrtc:9631
Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/100524
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24878}
RtpPacket::UpdateDelayStatistics was previously optimized with several
sanity checks added. These sanity checks caused many of the unit tests
in peerconnection_integration_unittests to fail and the CL was therefore
reverted. This CL contains the sanity checks along with patches so that
the unit tests pass.
Bug: webrtc:9439
Change-Id: Ia5f5e8b125e5f3f4b79d433e2282901143530a25
Reviewed-on: https://webrtc-review.googlesource.com/99802
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24813}
This is a reland of 1a2cc0acba6a66f89249455d8e5775849b56f755
Original change's description:
> Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*."
>
> This is a reland of 870bca1f418a1abf445169a638a61f9a649d557f
>
> Original change's description:
> > Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.
> >
> > We now use RTC_HISTOGRAM_* macros in system_wrappers/include/metrics.h
> > to report the metrics in pc/ and p2p/ that are currently been reported
> > using MetricsObserverInterface.
> >
> > TBR=tommi@webrtc.org
> >
> > Bug: webrtc:9409
> > Change-Id: I47c9975402293c72250203fa1ec19eb1668766f6
> > Reviewed-on: https://webrtc-review.googlesource.com/83782
> > Commit-Queue: Qingsi Wang <qingsi@google.com>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Taylor (left Google) <deadbeef@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#23914}
>
> TBR=steveanton@webrtc.org,hta@webrtc.org,tommi@webrtc.org
>
> Bug: webrtc:9409
> Change-Id: I37fc95ced60dea25aa9b4f5ad44bdf7174c8bd5c
> Reviewed-on: https://webrtc-review.googlesource.com/88060
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Cr-Commit-Position: refs/heads/master@{#23919}
TBR=steveanton@webrtc.org,tommi@webrtc.org
Bug: webrtc:9409
Change-Id: Ib55f0b6c9bcb9d9585924a4dfac5cf643ff4d76b
Reviewed-on: https://webrtc-review.googlesource.com/88343
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23957}
This reverts commit 1a2cc0acba6a66f89249455d8e5775849b56f755.
Reason for revert: It breaks internal Android debug build. Need further investigation.
Original change's description:
> Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*."
>
> This is a reland of 870bca1f418a1abf445169a638a61f9a649d557f
>
> Original change's description:
> > Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.
> >
> > We now use RTC_HISTOGRAM_* macros in system_wrappers/include/metrics.h
> > to report the metrics in pc/ and p2p/ that are currently been reported
> > using MetricsObserverInterface.
> >
> > TBR=tommi@webrtc.org
> >
> > Bug: webrtc:9409
> > Change-Id: I47c9975402293c72250203fa1ec19eb1668766f6
> > Reviewed-on: https://webrtc-review.googlesource.com/83782
> > Commit-Queue: Qingsi Wang <qingsi@google.com>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Taylor (left Google) <deadbeef@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#23914}
>
> TBR=steveanton@webrtc.org,hta@webrtc.org,tommi@webrtc.org
>
> Bug: webrtc:9409
> Change-Id: I37fc95ced60dea25aa9b4f5ad44bdf7174c8bd5c
> Reviewed-on: https://webrtc-review.googlesource.com/88060
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Cr-Commit-Position: refs/heads/master@{#23919}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org,tommi@webrtc.org,hta@webrtc.org,qingsi@google.com,qingsi@webrtc.org
Change-Id: I4a75fc7f52bfd0780526537a5a9a016fb9c20d6a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9409
Reviewed-on: https://webrtc-review.googlesource.com/88320
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#23938}
This is a reland of 870bca1f418a1abf445169a638a61f9a649d557f
Original change's description:
> Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.
>
> We now use RTC_HISTOGRAM_* macros in system_wrappers/include/metrics.h
> to report the metrics in pc/ and p2p/ that are currently been reported
> using MetricsObserverInterface.
>
> TBR=tommi@webrtc.org
>
> Bug: webrtc:9409
> Change-Id: I47c9975402293c72250203fa1ec19eb1668766f6
> Reviewed-on: https://webrtc-review.googlesource.com/83782
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Taylor (left Google) <deadbeef@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23914}
TBR=steveanton@webrtc.org,hta@webrtc.org,tommi@webrtc.org
Bug: webrtc:9409
Change-Id: I37fc95ced60dea25aa9b4f5ad44bdf7174c8bd5c
Reviewed-on: https://webrtc-review.googlesource.com/88060
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#23919}
This reverts commit 870bca1f418a1abf445169a638a61f9a649d557f.
Reason for revert: it breaks internal tests and builds
Original change's description:
> Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.
>
> We now use RTC_HISTOGRAM_* macros in system_wrappers/include/metrics.h
> to report the metrics in pc/ and p2p/ that are currently been reported
> using MetricsObserverInterface.
>
> TBR=tommi@webrtc.org
>
> Bug: webrtc:9409
> Change-Id: I47c9975402293c72250203fa1ec19eb1668766f6
> Reviewed-on: https://webrtc-review.googlesource.com/83782
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Taylor (left Google) <deadbeef@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23914}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hta@webrtc.org,tommi@webrtc.org
Change-Id: I1afd92d44f3b8cf3ae9aa6e6daa9a3a272e8097f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9409
Reviewed-on: https://webrtc-review.googlesource.com/88040
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#23916}
We now use RTC_HISTOGRAM_* macros in system_wrappers/include/metrics.h
to report the metrics in pc/ and p2p/ that are currently been reported
using MetricsObserverInterface.
TBR=tommi@webrtc.org
Bug: webrtc:9409
Change-Id: I47c9975402293c72250203fa1ec19eb1668766f6
Reviewed-on: https://webrtc-review.googlesource.com/83782
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor (left Google) <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23914}
Both incoming and outgoing datachannels should cause
the DATA_ADDED flag to be set.
This CL also moves all tests into their own file, and
improves scaffolding.
Bug: chromium:718508
Change-Id: I5c4c257ccb6f26799f7593bce8b27ebf59015b1e
Reviewed-on: https://webrtc-review.googlesource.com/85348
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23766}
EndToEndConnectionTimeWithTurnTurnPair was failing intermittently due to
a DCHECK being hit in ports.cc. This was caused by the ScopedFakeClock
being destroyed before the ports. The ports miscalculated a large
negative number for the rtt of a STUN request/response due to the global
clock changing. This fixes the problem by closing the PeerConnections
before the ScopedFakeClock goes out of scope.
Bug: webrtc:9422
Change-Id: Ia4aa3f638dff5da4317a35cf1514ec61472d0d74
Reviewed-on: https://webrtc-review.googlesource.com/84241
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23670}
This is the simplest possible fix, returning SSRC stats with a missing
track ID instead of returning no SSRC stats at all.
This means calling GetStats with the track selector argument will still not
work in this case.
Bug: webrtc:3342
Change-Id: I6b58fd5ac15b49274d3f1655e78ae36c4575e5fd
Reviewed-on: https://webrtc-review.googlesource.com/82260
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23667}
The PeerConnection integration test was creating TurnServers on the
stack on the signaling thread. This could cause a race condition problem
when the test was being taken down. Since the turn server was destructed
on the signaling thread, a socket might still try and send to it after
it was destroyed causing a seg fault. This change creates/destroys the
TestTurnServers on the network thread to fix this issue.
Bug: None
Change-Id: I080098502b737f0972ce2fa5357920de057a3312
Reviewed-on: https://webrtc-review.googlesource.com/81301
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23590}
This CL fixes a bug that the RtcEventLog owned by PeerConnection was not
passed to P2PTransportChannel after JsepTransportController was
introduced to deprecate the legacy TransportController.
Bug: webrtc:9337
Change-Id: I406cd9c0761dfe67f969aa99c6141e1ab38249d5
Reviewed-on: https://webrtc-review.googlesource.com/79964
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23572}
This generates a number that represent a set of bits that
indicates how a PeerConnection has been used over time.
Bug: chromium:718508
Change-Id: I6df177684c50bc825bc41ea97996574292084d41
Reviewed-on: https://webrtc-review.googlesource.com/79823
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23471}
End to end test for media sent over a TCP TURN server with both clients in relay
This test validates that media can be sent between two clients who are set up
to relay information with the server configured to use TCP instead of UDP.
Bug: webrtc:7668
Change-Id: I3efd04048589c144494f90f2cdf3df5f9f80300e
Reviewed-on: https://webrtc-review.googlesource.com/76507
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23354}
Previous code had a FakePeriodicVideoSource and a
VideoTrackSource, where the latter is reference counted and
outlives the former. That results in potential races when
RemoveSink is called on the VideoTrackSource after the
FakePeriodicVideoSource is destroyed, with a complicated sequence
to do correct shutdown.
The new class, FakePeriodicVideoTrackSource, owns a
FakePeriodicVideoSource, and they get the same lifetime.
Bug: webrtc:6353
Change-Id: Ic33b393e00a31fa28893dce2018948d3f90e0a9e
Reviewed-on: https://webrtc-review.googlesource.com/76961
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23320}
Previously, constructing a PeerConnection or WebRtcVideoEngine with
fake encoder/decoder factories would result in the real, built-in factories
also being used. In https://webrtc-review.googlesource.com/c/src/+/71162, this
changed, so to temporarily allow tests to continue working exactly the same as
before, the fake factories started encapsulating the real factories. This CL
removes that behavior and updates the tests accordingly.
Bug: webrtc:9228
Change-Id: Ida14a1e3f5f5a0e2f03100b7895b3b1bdf0a0a42
Reviewed-on: https://webrtc-review.googlesource.com/75260
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23209}
