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}
This extends the API surface so that
custom certificates can be provided by an API user in both the standalone and
factory creation paths for the OpenSSLAdapter. Prior to this change the SSL
roots were hardcoded in a header file and directly included into
openssladapter.cc. This forces the 100 kilobytes of certificates to always be
compiled into the library. This is undesirable in certain linking cases where
these certificates can be shared from another binary that already has an
equivalent set of trusted roots hard coded into the binary.
Support for removing the hard coded SSL roots has also been added through a new
build flag. By default the hard coded SSL roots will be included and will be
used if no other trusted root certificates are provided.
The main goal of this CL is to reduce total binary size requirements of WebRTC
by about 100kb in certain applications where adding these certificates is
redundant.
Change-Id: Ifd36d92b5cb32d1b3098a61ddfc244d76df8f30f
Bug: chromium:526260
Change-Id: Ifd36d92b5cb32d1b3098a61ddfc244d76df8f30f
Reviewed-on: https://webrtc-review.googlesource.com/64841
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23180}
This removes the redundant type and replaces all usages. A slight change
in behavior is that we no longer get nanosecond resolution. This should
not matter since no current code requires nanosecond resolution.
Bug: webrtc:9155
Change-Id: I04334e08c686d95731621a6c8a7e40400d0ae3b2
Reviewed-on: https://webrtc-review.googlesource.com/71163
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23174}
To prepare for making the software codecs optional and injectable, these
codec factories provide a way to pass in identical factories as were the
default old behaviour.
Bug: webrtc:7925
Change-Id: I0c70fa3c56c999e9d1af6e172eff2fbba849e921
Reviewed-on: https://webrtc-review.googlesource.com/71162
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23096}
This is intended to exercise end-to-end sending with the MID RTP
header extension and demuxing by MID.
Bug: webrtc:4050
Change-Id: I81edb3687c65f5efce9591fa34cb03522ad675e5
Reviewed-on: https://webrtc-review.googlesource.com/71601
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23062}
This reverts commit edbd389ecc7388973b55e6e3787ed6a95254dc99.
Reason for revert: Breaking integration on Linux.
Original change's description:
> TCP TURN Integration Test
>
> This changeset adds a new integration test to do basic validation that TCP
> TURN functionality works in WebRTC. It simply sets up a TestTurnServer
> configured to relay over TCP and then allows the clients to connect to this
> server over TCP.
>
> Bug: webrtc:7668
> Change-Id: Id9f3b4e22f40ace7c7eeddf103b5d954a0872777
> Reviewed-on: https://webrtc-review.googlesource.com/70568
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23044}
TBR=deadbeef@webrtc.org,stefan@webrtc.org,pthatcher@webrtc.org,benwright@webrtc.org
Change-Id: Icdf8747d7a1a7bd2a1a29f1536821a0eacb7764e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7668
Reviewed-on: https://webrtc-review.googlesource.com/72961
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23045}
This changeset adds a new integration test to do basic validation that TCP
TURN functionality works in WebRTC. It simply sets up a TestTurnServer
configured to relay over TCP and then allows the clients to connect to this
server over TCP.
Bug: webrtc:7668
Change-Id: Id9f3b4e22f40ace7c7eeddf103b5d954a0872777
Reviewed-on: https://webrtc-review.googlesource.com/70568
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23044}
This is a reland of fc43d11717e16dd427ac84fee614e5511e43cefd
Original change's description:
> Add thread checker to PortAllocator and its subclasses and fix a bug
> causing memory contention by threads.
>
> PortAllocator and its subclasses assume all of their methods except the
> constructor must be called on the same thread (the network thread in
> practice). This CL adds a thread checker to PortAllocator and its
> subclasses for thread safety, and fixes bugs of invoking some of their
> methods in PeerConnection on the signaling thread.
>
> Bug: webrtc:9112
> Change-Id: I33ba9bae72ec09a45ec70435962f3f25cd31583c
> Reviewed-on: https://webrtc-review.googlesource.com/66945
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22814}
Bug: webrtc:9112
Change-Id: I5c7377f05c0daccbe469e2fdbdfacabc5c222f4c
Reviewed-on: https://webrtc-review.googlesource.com/69422
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22889}
BundleFilter is replaced by RtpDemuxer in RtpTransport for payload
type-based demuxing. RtpTransport will support MID-based demuxing later.
Each BaseChannel has its own RTP demuxing criteria and when connecting
to the RtpTransport, BaseChannel will register itself as a demuxer sink.
The inheritance model is changed. New inheritance chain:
DtlsSrtpTransport->SrtpTransport->RtpTranpsort
The JsepTransport2 is renamed to JsepTransport.
NOTE:
When RTCP packets are received, Call::DeliverRtcp will be called for
multiple times (webrtc:9035) which is an existing issue. With this CL,
it will become more of a problem and should be fixed.
Bug: webrtc:8587
Change-Id: Ibd880e7b744bd912336a691309950bc18e42cf62
Reviewed-on: https://webrtc-review.googlesource.com/65786
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22867}
This adds confusion to the native API and is only needed for
Chromium UMA metrics, so the appropriate metrics have been moved
upstream and kDefault option removed.
Bug: chromium:811683
Change-Id: I666d7f7793765b8d6edcd99416c8b6c957766f00
Reviewed-on: https://webrtc-review.googlesource.com/59261
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22864}
This was previously not working because the answerer wasn't generating
ICE credentials when it should have been.
This was fixed inadvertently by:
https://webrtc-review.googlesource.com/c/src/+/46380
But we should really also have a PeerConnection-level regression test
for this.
Bug: webrtc:6023
Change-Id: I3da900edcc8db8034ed61a7bb981d9c0e616254e
Reviewed-on: https://webrtc-review.googlesource.com/69403
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22832}
This changes the PeerConnection when in Unified Plan mode to reject
SDP applied with SetLocalDescription or SetRemoteDescription if the
SDP has multiple "Plan B tracks" (a=ssrc lines) in a media section.
The error is to inform developers that the given SDP will not be
interpreted as they might expect.
Bug: None
Change-Id: I7a0e11282fbf63dac06038cd22a66683517a87d0
Reviewed-on: https://webrtc-review.googlesource.com/68764
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22829}
This reverts commit fc43d11717e16dd427ac84fee614e5511e43cefd.
Reason for revert: Crashes downstream tests
Original change's description:
> Add thread checker to PortAllocator and its subclasses and fix a bug
> causing memory contention by threads.
>
> PortAllocator and its subclasses assume all of their methods except the
> constructor must be called on the same thread (the network thread in
> practice). This CL adds a thread checker to PortAllocator and its
> subclasses for thread safety, and fixes bugs of invoking some of their
> methods in PeerConnection on the signaling thread.
>
> Bug: webrtc:9112
> Change-Id: I33ba9bae72ec09a45ec70435962f3f25cd31583c
> Reviewed-on: https://webrtc-review.googlesource.com/66945
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22814}
TBR=deadbeef@webrtc.org,pthatcher@google.com,pthatcher@webrtc.org,qingsi@google.com,honghaiz@webrtc.org
Change-Id: I2db6561d5d6366d38caa58c3e719d0d48eda70c2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9112
Reviewed-on: https://webrtc-review.googlesource.com/69200
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22818}
causing memory contention by threads.
PortAllocator and its subclasses assume all of their methods except the
constructor must be called on the same thread (the network thread in
practice). This CL adds a thread checker to PortAllocator and its
subclasses for thread safety, and fixes bugs of invoking some of their
methods in PeerConnection on the signaling thread.
Bug: webrtc:9112
Change-Id: I33ba9bae72ec09a45ec70435962f3f25cd31583c
Reviewed-on: https://webrtc-review.googlesource.com/66945
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22814}
This profile will now not be used unless the application explicitly
sets the flag in CryptoOptions to true. As a result, an 80-bit
authentication tag will be used instead of a 32-bit one. See bug for
more details.
Bug: webrtc:7670
Change-Id: I7c0a118fd7b1e7aac23b9eb8717099f055de0441
Reviewed-on: https://webrtc-review.googlesource.com/66600
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22757}
Currently in the SDP we require an a=ssrc line in the m= section in
order for a StreamParams object to be created with that
MediaContentDescription. This change creates a StreamParams object
without ssrcs in the case that a=msid lines are signaled, but ssrcs
are not. When the remote description is set, this allows us to store
the "unsignaled" StreamParams object in the media channel to later
be used when the first packet is received and we create the
receive stream.
Bug: webrtc:7932, webrtc:7933
Change-Id: Ib6734abeee62b8ed688a8208722c402134c074ef
Reviewed-on: https://webrtc-review.googlesource.com/63141
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22712}
The RTCStatsCollector was only iterating through RtpTransceivers
in order to find the active transports for which to generate stats.
But for data channel only connections, there were no
RtpTransceivers so no transports were being identified.
This CL changes the stats collector to include the transport names
of the SCTP and RTP data channel if active.
Bug: chromium:826972
Change-Id: I762b253b3bbf0f0d7861bc281b8908decbb9b0d9
Reviewed-on: https://webrtc-review.googlesource.com/65788
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22697}
The TransportController will be replaced by the JsepTransportController
and JsepTransport will be replace be JsepTransport2.
The JsepTransportController will take the entire SessionDescription
and handle the RtcpMux, Sdes and BUNDLE internally.
The ownership model is also changed. The P2P layer transports are not
ref-counted and will be owned by the JsepTransport2.
In ORTC aspect, RtpTransportAdapter is now a wrapper over RtpTransport
or SrtpTransport and it implements the public and internal interface
by calling the transport underneath.
Bug: webrtc:8587
Change-Id: Ia7fa61288a566f211f8560072ea0eecaf19e48df
Reviewed-on: https://webrtc-review.googlesource.com/59586
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22693}
This flag (added to CryptoOptions) will allow applications to opt-in to
use of this suite, before it's disabled by default later. See bug for
more details.
TBR=magjed@webrtc.org
Bug: webrtc:7670
Change-Id: I800bedd4b26d807b6b7ac66b505d419c3323e454
Reviewed-on: https://webrtc-review.googlesource.com/64390
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22586}
Makes it easier to follow threads during debugging.
Bug: None
Change-Id: I88e68521e354224052500bc47f2300253b95a892
Reviewed-on: https://webrtc-review.googlesource.com/61429
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22405}
They were disabled since GetRemoteAudioSSLCertificate was written
in terms of voice/video channel, which were not methods supported
with Unified Plan. Now GetRemoteAudioSSLCertificate has been
rewritten to work with RtpTransceivers, so the test can be enabled.
Bug: webrtc:8764
Change-Id: I08b5fbcc0d69f36113a281c902db6508fa48ebdd
Reviewed-on: https://webrtc-review.googlesource.com/55923
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22115}
This reverts commit 6780c51b23516803dc27173d10ba98d018780447.
Reason for revert:
More details in crbug.com/810292
Original change's description:
> Use SRTP_AES128_CM_SHA1_80 by default instead of SRTP_AES128_CM_SHA1_32.
>
> A field has been added to "CryptoOptions" to enable SRTP_AES128_CM_SHA1_32
> from native apps if really necessary.
>
> R=deadbeef@webrtc.org
>
> Bug: webrtc:7670
> Change-Id: I36b6ab3e302fbf3cda2611ff196757e43a56e704
> Reviewed-on: https://webrtc-review.googlesource.com/41420
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Joachim Bauch <jbauch@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21952}
TBR=deadbeef@webrtc.org,magjed@webrtc.org,jbauch@webrtc.org
Change-Id: I643dbe023eca526f2cda4d97df045f2533741dd4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7670
Reviewed-on: https://webrtc-review.googlesource.com/49880
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21961}
A field has been added to "CryptoOptions" to enable SRTP_AES128_CM_SHA1_32
from native apps if really necessary.
R=deadbeef@webrtc.org
Bug: webrtc:7670
Change-Id: I36b6ab3e302fbf3cda2611ff196757e43a56e704
Reviewed-on: https://webrtc-review.googlesource.com/41420
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Joachim Bauch <jbauch@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21952}
This method returns the DTLS SSL certificate chain associated with the
audio transport on the remote side. This will become populated once the
DTLS connection with the peer has been completed.
TBR=deadbeef@webrtc.org
Bug: webrtc:8800
Change-Id: Ib90ccb3463415e798c17c187c5bdbfc4da26f11f
Reviewed-on: https://webrtc-review.googlesource.com/44140
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21785}
These tests verify the behavior between Plan B and
Unified Plan PeerConnections.
Bug: webrtc:7600
Change-Id: Ic41a0e692d32cde6fe7719ada2dbffd4281c008c
Reviewed-on: https://webrtc-review.googlesource.com/43244
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21782}
This is required to figure out when we can deprecate and remove
SDES.
Bug: chromium:804275
Change-Id: Ie234e6b3c8f5de8e78dda8d755d955caa61b7aa7
Reviewed-on: https://webrtc-review.googlesource.com/43340
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21746}
This also changes RtpReceiver and RemoteAudioSource to have two-step
initialization, since in Unified Plan RtpReceivers are created much
earlier than in Plan B.
Bug: webrtc:7600
Change-Id: Ia135d25eb8bcab22969007b3a825a5a43ce62bf4
Reviewed-on: https://webrtc-review.googlesource.com/39382
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21681}
This was causing ICE pings to continue going out on PeerConnections
that use DataChannels, even after closing the PeerConnection.
This CL adds a two-line fix, and an integration test that will catch
this and similar issues.
Bug: webrtc:7655
Change-Id: I589a2a1aaf6433c1d65be69a1267e1b52a33534b
Reviewed-on: https://webrtc-review.googlesource.com/37145
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21488}
This changes all internal code to use the media_description() helper
for ContentInfo along with the as_audio, as_video, and as_data casting
methods introduced in a previous CL. Reduces the total number of
pointer static_casts in pc/ from 351 to 122.
Bug: webrtc:8620
Change-Id: I996f49b55f1501c758a9e5223e30539a9f8d4eac
Reviewed-on: https://webrtc-review.googlesource.com/35921
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21419}
SDP is a detail of PeerConnection and is not used by anything in p2p, so
it belongs in the pc/ directory. This also allows
MediaContentDescription to be inlined in the future.
Bug: webrtc:8620
Change-Id: I38b65ede9942e29eb15035ab29f2be988da1e5ce
Reviewed-on: https://webrtc-review.googlesource.com/33781
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21376}
This moves all WebRTC internal code from using
SessionDescriptionInterface::type() which returns a string and
from using CreateSessionDescription with a string type parameter.
Bug: webrtc:8613
Change-Id: I1cdd93dc4b26dec157e22476fdac569d5da2810a
Reviewed-on: https://webrtc-review.googlesource.com/29500
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21147}
This reverts commit 33c5c7f5e4f018a5103770021328fc530d451d75.
Reason for revert: Fix broken API change.
TBR=sprang@webrtc.org,solenberg@webrtc.org
TBRing because only pc/ and api/ have changed since last LGTMed version.
Original change's description:
> Revert "Encode log events periodically instead of for every event."
>
> This reverts commit b154c27e72fddb6c0d7cac69a9c68fff22154519.
>
> Reason for revert: Broke the internal project.
>
> Original change's description:
> > Encode log events periodically instead of for every event.
> >
> > Updated unit test to take output_period and random seed as parameters.
> > Updated the peerconnection interface to allow passing in an output_period.
> >
> > This is in preparation of some upcoming CLs that will change the format
> > to store batches of delta-encoded values.
> >
> >
> > Bug: webrtc:8111
> > Change-Id: Id5d9844dfad8d8edad346cd7cbebc7eadaaa5416
> > Reviewed-on: https://webrtc-review.googlesource.com/22600
> > Commit-Queue: Björn Terelius <terelius@webrtc.org>
> > Reviewed-by: Elad Alon <eladalon@webrtc.org>
> > Reviewed-by: Tommi <tommi@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20736}
>
> Change-Id: I2257c46c014adb8c7c4fb28538635cabed1f2229
> Bug: webrtc:8111
> Reviewed-on: https://webrtc-review.googlesource.com/24160
> Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
> Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20738}
Bug: webrtc:8111
Change-Id: Ie69862cd52d11c1e15adeb6e2caacafe16863c80
Reviewed-on: https://webrtc-review.googlesource.com/24620
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20811}
PeerConnectionIntegrationTest.AddMediaToConnectedBundleDoesNotRestartIce
--> Fixed by an earlier CL (https://webrtc-review.googlesource.com/c/src/+/16261)
so re-enabled.
PeerConnectionIntegrationTest/PeerConnectionIntegrationIceStatesTest.VerifyIceStates
--> An existing bug causes this to by flaky when using a fake clock.
Fake clock removed with a TODO to change it back when the bug is fixed.
PeerConnectionIntegrationTest.TrackStatsUpdatedCorrectlyWhenUnsignaledSsrcChanges
--> The heuristic that >25% concealed audio samples is abnormal is
unfortunately not reliable enough on certain slow trybots. Bump the
threshold to 50% in hopes that is enough.
Bug: webrtc:8496
Change-Id: I17cfdf956a8a72ac399212c3c7efcdd2236be00d
Reviewed-on: https://webrtc-review.googlesource.com/20963
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20638}
EXPECT_EQ(0u, caller()->ice_connection_state_history().size());
Was failing since it cleared the connection state history before
the caller had finished transitioning to Completed, so the initial
transitions would be mistaken for later transitions.
Bug: None
Change-Id: I7043638f077ac5dcaeeca0d3ea6accc93c920364
Reviewed-on: https://webrtc-review.googlesource.com/16261
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20570}
Follow-up on https://webrtc-review.googlesource.com/c/src/+/17784.
Adds a new PC integration test using the newly added StartRecording API.
Bug: webrtc:7313
Change-Id: Ibd59910ca5d8f8ac96cfb891f41039759a18b6f6
Reviewed-on: https://webrtc-review.googlesource.com/17940
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20549}
Second attempt to land https://webrtc-review.googlesource.com/c/src/+/16180
Now removes voice_engine dependency from peerconnection and fixes a minor
const issue in NullAudioPoller.
TBR=solenberg
Bug: webrtc:7313
Change-Id: Ibfddbdc76118581e4a4dc64575203f84c1659e5c
Reviewed-on: https://webrtc-review.googlesource.com/17784
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20526}
This reverts commit 90bace095806a635411edd40fb8490a144e59e63.
Reason for revert: The original problem of this CL has been fixed in https://webrtc-review.googlesource.com/17540 but sounds like it is also adding voice_engine as a dependency of pc:peerconnection. We should investigate this because probably we can avoid it.
Original change's description:
> Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API
>
> (this CL is based on the work by Taylor and Steve in https://webrtc-review.googlesource.com/c/src/+/10201)
>
> This SetAudioPlayout method lets applications disable audio playout while
> still processing incoming audio data and generating statistics on the
> received audio.
>
> This may be useful if the application wants to set up media flows as
> soon as possible, but isn't ready to play audio yet. Currently, native
> applications don't have any API point to control this, unless they
> completely implement their own AudioDeviceModule.
>
> The SetAudioRecording works in a similar fashion but for the recorded
> audio. One difference is that calling SetAudioRecording(false) does not
> keep any audio processing alive.
>
> TBR=solenberg
>
> Bug: webrtc:7313
> Change-Id: I0aa075f6bfef9818f1080f85a8ff7842fb0750aa
> Reviewed-on: https://webrtc-review.googlesource.com/16180
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20499}
TBR=solenberg@webrtc.org,henrika@webrtc.org,kwiberg@webrtc.org
Change-Id: I8431227e21dbffcfed3dd0e6bd7ce26c4ce09394
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7313
Reviewed-on: https://webrtc-review.googlesource.com/17701
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20512}
(this CL is based on the work by Taylor and Steve in https://webrtc-review.googlesource.com/c/src/+/10201)
This SetAudioPlayout method lets applications disable audio playout while
still processing incoming audio data and generating statistics on the
received audio.
This may be useful if the application wants to set up media flows as
soon as possible, but isn't ready to play audio yet. Currently, native
applications don't have any API point to control this, unless they
completely implement their own AudioDeviceModule.
The SetAudioRecording works in a similar fashion but for the recorded
audio. One difference is that calling SetAudioRecording(false) does not
keep any audio processing alive.
TBR=solenberg
Bug: webrtc:7313
Change-Id: I0aa075f6bfef9818f1080f85a8ff7842fb0750aa
Reviewed-on: https://webrtc-review.googlesource.com/16180
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20499}
The DataChannel OPEN message is sent in-band in this test, and waiting
for the signaling state to change is not sufficient to guarantee that the
callee received this message
Bug: webrtc:8443
Change-Id: I76fa6348b6f8e1e70fb41a4e644aee805b2ef4de
Reviewed-on: https://webrtc-review.googlesource.com/15060
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20415}
