Preparation for replacing use of TestVideoCapturer as an interface,
instead using VideoSourceInterface.
Methods kept as non-virtual on the subclass FrameGeneratorCapturer,
but it's changed to be started on creation.
Bug: webrtc:6353
Change-Id: Iae1c9a0ee55d730d4992204f62227ef2f057d58e
Reviewed-on: https://webrtc-review.googlesource.com/c/114425
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26037}
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}
Adding RidDescription to StreamParams that will contain the list of rids
for the track.
Adding receive_stream to MediaContentDescription to allow identifying
the stream that originates from the answerer (but is referenced by the
sender). For example, to signal that it will be received in Simulcast.
Bug: webrtc:10073.
Change-Id: Icd9a6b0a69d42bef51f525e673ce447255584334
Reviewed-on: https://webrtc-review.googlesource.com/c/113794
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25978}
Bug: None
Change-Id: I0fd0616132705c6d15a77fc442be47080f1b81b1
Reviewed-on: https://webrtc-review.googlesource.com/c/112721
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25975}
This changes MediaSessionFactory to take the unified plan
configuration option as an explicit setter rathen than a
MediaSessionOptions flag. This is fine since a PeerConnection will
always be in unified plan mode or not, and we know this at
construction.
Bug: None
Change-Id: Ifca45d1d7c9d62b2b41bb879f8665fb39b4cdcd0
Reviewed-on: https://webrtc-review.googlesource.com/c/113824
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25960}
Previously, the PeerConnection would look at the pending local
and remote descriptions also to determine if an m= section is
recycled. That is not quite spec compliant and breaks down under
some edge cases. This changes the PeerConnection to look only at
the *current* local or remote description (i.e., the descriptions
from the last time the PeerConnection was in a stable signaling
state) to determine if an m= section is recycled.
Additionally, the MediaSessionFactory only looked at the local
description to determine if an m= section is recycled. The full
criteria requires looking at the current local and current remote
m= sections. This change adds a state enum to the
MediaDescriptionOptions so that the MediaSessionFactory knows if
a media section is being recycled without duplicating the logic
in PeerConnection.
Tests are added to cover additional edge cases.
Bug: chromium:899680
Change-Id: I5bcf0f88957a61653269ed8bb50b2018500bc1d5
Reviewed-on: https://webrtc-review.googlesource.com/c/111293
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25959}
This change introduces a new measurement into the VideoReceiveStream::Stats
structure to measure the latency between the first frame being received and
the first frame being decoded in WebRTC. The goal here is to measure the latency
difference when a FrameEncryptor is attached and not attached.
Change-Id: I0f0178aff73b66f25dbc6617098033e226da2958
Bug: webrtc:10105
Reviewed-on: https://webrtc-review.googlesource.com/c/113328
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25956}
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}
Since a lot of native users have taken dependencies on our old, non-standard behaviour
we'll have to have two ice connection states living side by side until we can get rid
of the old one.
Bug: webrtc:6145
Change-Id: I9b673bffeb1dfcf410f7c56d4def5912121e644c
Reviewed-on: https://webrtc-review.googlesource.com/c/113421
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25929}
This was regression due to
https://webrtc-review.googlesource.com/c/src/+/111920
which broke downstream projects. The break was not caught in
unit tests, because unit tests use loopback and fake media
transports that do not use ICE.
Bug: None
Change-Id: If95935afed430d62d5ff9a2ee01d8eaccadc198d
Reviewed-on: https://webrtc-review.googlesource.com/c/113440
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25926}
This avoids use-after-free problems that occur when references
to webrtc::DtlsTransport objects are held outside of the PC.
Bug: chromium:907849
Change-Id: Id428c8e616482eff0f4327d2eac17e29bb3f6484
Reviewed-on: https://webrtc-review.googlesource.com/c/113303
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25915}
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}
Since not all fields are compared on NetworkRoute structs, the ==
operator overload doesn't really make the code easier to read. In fact
the feature that it only compares a subset of the fields is only used
once, at the other places, all fields are compared.
Removing the overload makes it more clear what is compared at each call
site.
Bug: webrtc:9883
Change-Id: I74f7eb32b602aa33fd282a815b71a172ae3f6a8b
Reviewed-on: https://webrtc-review.googlesource.com/c/113001
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25891}
Parsing simulcast according to:
https://tools.ietf.org/html/draft-ietf-mmusic-sdp-simulcast-13#section-5.1
Created SdpSerializer for making serialized components more testable.
Simulcast functionality is still not accessible to users.
Bug: webrtc:10055
Change-Id: Ia6e4cef756cb954521dd19e22911f8eb6498880e
Reviewed-on: https://webrtc-review.googlesource.com/c/112160
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25883}
These two configs are in both RTConfiguration and IceConfig,
but ParseIceConfig() function does not move them.
Bug: webrtc:10079
Change-Id: I11cbedfeabaf77228a253c7bc5e2781b28b08642
Reviewed-on: https://webrtc-review.googlesource.com/c/112546
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25860}
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 includes a refactoring of jseptransport to store a refcounted
object instead of a std::unique_ptr to the cricket::DtlsTransport.
Bug: chromium:907849
Change-Id: Ib557ce72c2e6ce8af297c2b8deb7ec3a103d6d31
Reviewed-on: https://webrtc-review.googlesource.com/c/111920
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25831}
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/106280.
This time the whole code base is covered.
Some files may have not been fixed though, whenever the IWYU tool
was breaking the build.
Bug: webrtc:8311
Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef
Reviewed-on: https://webrtc-review.googlesource.com/c/111965
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25830}
Note that this value will override the minimum delay that is used for audio/video sync.
Bug: webrtc:10053
Change-Id: Ia129f6c9ee9da5d00a3d955afaaa6e8f0c2bee33
Reviewed-on: https://webrtc-review.googlesource.com/c/112121
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25805}
The new wire format doesn't have much effect on compression unless
the log is encoded in reasonably large batches.
PeerConnection has two functions to start logging; one which takes
an output period (or batch size) in milliseconds and one which uses
a default period instead. This CL changes the default batch size to
5 seconds if the the new format is enabled as a field trial.
Bug: webrtc:8111
Change-Id: I638f6114325251b6a9acf4f863afe2688a3b0522
Reviewed-on: https://webrtc-review.googlesource.com/c/112130
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25803}
The stat will be exposed through origin trial described in:
https://docs.google.com/document/d/1stYIZhEmDZ7NJF9gjjsM66eLFJUdc-14a3QutrFbIwI
Change-Id: Ib191a11c6bd9e617abbe9dd82239b0c5b4e6b4e0
Bug: webrtc:10043
Reviewed-on: https://webrtc-review.googlesource.com/c/111922
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25802}
Auxiliary threads (worker, network) are still active
while PeerConnection is destructed, leading to race condition
in tests such as:
* RTCStatsIntegrationTest.GetStatsFromCaller
* RTCStatsIntegrationTest.GetsStatsWhileDestroyingPeerConnection
This CL prevents the conflict to happen by explicitly
closing the PeerConnection.
Bug: webrtc:9847
Change-Id: I40880bb9b193201711031b8c4563c6bbd4983c71
Reviewed-on: https://webrtc-review.googlesource.com/c/104820
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25801}
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 CL decouples //rtc_base:rtc_base_tests_utils from gunit by
moving gunit helpers (rtc_base/gunit.h) and rtc_base/testclient.h
(which depends on gunit helpers) to their own build target.
It also removes some unused dependencies in the WebRTC build graph.
Bug: None
Change-Id: Ia9820e84ff697da39b351eef73c45f6e4bdf2623
Reviewed-on: https://webrtc-review.googlesource.com/c/111861
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25769}
This seems to be the last piece of the puzzle.
Deprecation PSA:
https://groups.google.com/forum/#!msg/discuss-webrtc/NgqEPvkNuDE/7HtwnMmADgAJ
Bug: webrtc:8572
Change-Id: Ib04b843fe50b8f07742c85827af6881dcfdc2991
Reviewed-on: https://webrtc-review.googlesource.com/c/109005
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25765}
This code is never built by GN, and the header is never included.
Bug: webrtc:9855
Change-Id: I16e6a54cc95629917d454f91d9bcc99fc55d8a00
Reviewed-on: https://webrtc-review.googlesource.com/c/111754
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25753}
This CL moves webrtc::CreatePeerConnectionFactory definitions out of
pc:create_pc_factory and merges it with its declaration in the api/
directory.
In order to avoid circular dependencies a new build target is created:
* api:create_peerconnection_factory
Bug: webrtc:9862
Change-Id: Ie215c94460cba026f5bf7d11c9a5aa03792064af
Reviewed-on: https://webrtc-review.googlesource.com/c/111186
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25744}
Bug: None
Change-Id: I36931774438b80ce391e656b8db2f2bb6ed25d8b
Reviewed-on: https://webrtc-review.googlesource.com/c/110961
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25733}
JsepTransportController got a bit ugly with one super long method.
Splitting it to two, so that MediaTransport creation is separated.
Bug: webrtc:9719
Change-Id: I0b5aead2f96d79d6fc369a16810be58c8a661e71
Reviewed-on: https://webrtc-review.googlesource.com/c/111288
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25732}
So far ANA was not available for media transport interface. With recent changes to media transport, we can now account for packet overhead, network route (ip/tcp/udp/turn overheads) and we can also use bandwidth estimate from the media transport.
Bug: webrtc:9719
Change-Id: I98c9a09dd418b763c339ee2ee05592e164cf9199
Reviewed-on: https://webrtc-review.googlesource.com/c/110367
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25677}
The current implementation triggers vptr race condition due to the
test setup itself (see bug for the glorious details).
Disabling the test reduces TSAN noise and will help to detect more
critical defects.
Bug: webrtc:9847
Change-Id: I4912b00f1faad5f41ccaa4b55bc21b5215b816c9
Reviewed-on: https://webrtc-review.googlesource.com/c/110907
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25674}
The audio and video engine is exposed directly rather via redundant
wrapping functions. This reduces the amount of boiler plate code.
Bug: webrtc:9883
Change-Id: I203a945ee6079397e24a378966a569cd5626ac4a
Reviewed-on: https://webrtc-review.googlesource.com/c/106683
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25673}
