To allow end-to-end QuicDataChannel usage with a
PeerConnection, RTCConfiguration has been modified to
include a boolean for whether to do QUIC, since negotiation of
QUIC is not implemented. If one peer does QUIC, then it will be
assumed that the other peer must do QUIC or the connection
will fail.
PeerConnection has been modified to create data channels of type
QuicDataChannel when the peer wants to do QUIC.
WebRtcSession has ben modified to use a QuicDataTransport
instead of a DtlsTransportChannelWrapper/DataChannel
when QUIC should be used
QuicDataTransport implements the generic functions of
BaseChannel to manage the QuicTransportChannel.
Committed: https://crrev.com/34b54c36a533dadb6ceb70795119194e6f530ef5
Review-Url: https://codereview.webrtc.org/2166873002
Cr-Original-Commit-Position: refs/heads/master@{#13645}
Cr-Commit-Position: refs/heads/master@{#13657}
Reason for revert:
Reverting because it broke an RTP data channel test on the FYI bots.
Original issue's description:
> Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage.
>
> To allow end-to-end QuicDataChannel usage with a
> PeerConnection, RTCConfiguration has been modified to
> include a boolean for whether to do QUIC, since negotiation of
> QUIC is not implemented. If one peer does QUIC, then it will be
> assumed that the other peer must do QUIC or the connection
> will fail.
>
> PeerConnection has been modified to create data channels of type
> QuicDataChannel when the peer wants to do QUIC.
>
> WebRtcSession has ben modified to use a QuicDataTransport
> instead of a DtlsTransportChannelWrapper/DataChannel
> when QUIC should be used
>
> QuicDataTransport implements the generic functions of
> BaseChannel to manage the QuicTransportChannel.
>
> Committed: https://crrev.com/34b54c36a533dadb6ceb70795119194e6f530ef5
> Cr-Commit-Position: refs/heads/master@{#13645}
TBR=pthatcher@webrtc.org,zhihuang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2206793007
Cr-Commit-Position: refs/heads/master@{#13647}
To allow end-to-end QuicDataChannel usage with a
PeerConnection, RTCConfiguration has been modified to
include a boolean for whether to do QUIC, since negotiation of
QUIC is not implemented. If one peer does QUIC, then it will be
assumed that the other peer must do QUIC or the connection
will fail.
PeerConnection has been modified to create data channels of type
QuicDataChannel when the peer wants to do QUIC.
WebRtcSession has ben modified to use a QuicDataTransport
instead of a DtlsTransportChannelWrapper/DataChannel
when QUIC should be used
QuicDataTransport implements the generic functions of
BaseChannel to manage the QuicTransportChannel.
Review-Url: https://codereview.webrtc.org/2166873002
Cr-Commit-Position: refs/heads/master@{#13645}
GCM cipher suites are optional (disabled by default) and can be enabled
through "PeerConnectionFactoryInterface::Options".
If compiled with Chromium (i.e. "ENABLE_EXTERNAL_AUTH" is defined), no
GCM ciphers can be used yet (see https://crbug.com/628400).
BUG=webrtc:5222, 628400
Review-Url: https://codereview.webrtc.org/1528843005
Cr-Commit-Position: refs/heads/master@{#13635}
The RtpReceiverObserverInterface is created.
The SignalFirstPacketReceived will be forwarded from BaseChannel to WebRtcSession.
WebRtcSession will forward SignalFirstAudioPacketReceived and SignalFirstVideoPacketReceived to the RtpReceiverInterface.
The application can listen to the Signal by implementing and registering a RtpReceiverObserver.
Review-Url: https://codereview.webrtc.org/1999853002
Cr-Commit-Position: refs/heads/master@{#13139}
This is one less DtlsIdentityStoreInterface implementation, and one step closer
to removing this interface in favor of RTCCertificateGeneratorInterface.
This also removes PeerConnectionInterface::CreatePeerConnectionWithStore which
is no longer needed.
BUG=webrtc:5707, webrtc:5708
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2020623002 .
Cr-Commit-Position: refs/heads/master@{#12990}
The store was used in WebRtcSessionDescriptionFactory to generate certificates,
now a generator is used instead (new API). PeerConnection[Factory][Interface],
and WebRtcSession are updated to pass generators all the way down to the
WebRtcSessionDescriptionFactory instead of stores.
The webrtc implementation of a generator, RTCCertificateGenerator, is used as
the default generator (peerconnectionfactory.cc:189) instead of the webrtc
implementation of a store, DtlsIdentityStoreImpl.
The generator is fully parameterized and does not generate RSA-1024 unless you
ask for it (which makes sense not to do beforehand since ECDSA is now default).
The store was not fully parameterized (known filed bug).
The "top" layer, PeerConnectionFactoryInterface::CreatePeerConneciton, is
updated to take a generator instead of a store.
Many unittests still use a store, to allow them to continue to do so the
factory gets CreatePeerConnectionWithStore which uses the old function
signature (and invokes the new signature by wrapping the store in an
RTCCertificateGeneratorStoreWrapper). As soon as the FakeDtlsIdentityStore is
turned into a certificate generator instead of a store, the unittests will be
updated and we can remove CreatePeerConnectionWithStore.
This is a reupload of https://codereview.webrtc.org/2013523002/ with minor
changes.
BUG=webrtc:5707, webrtc:5708
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2017943002 .
Cr-Commit-Position: refs/heads/master@{#12984}
This will make it much less likely for application developers to not
realize the object is reference counted.
It also fixes a bug in the Java PeerConnection binding, by allowing a
reference to be transferred in the OnRemoveStream call via std::move.
BUG=webrtc:5128
R=pthatcher@webrtc.org, tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1972793003 .
Cr-Commit-Position: refs/heads/master@{#12976}
When this test was updated to use separate worker/signaling threads,
the virtual socket server wasn't moved over to the worker thread. So it
was set on the signaling thread and wasn't being used.
Review-Url: https://codereview.webrtc.org/2015763002
Cr-Commit-Position: refs/heads/master@{#12951}
Depends on this CL in order to work in Chromium:
https://codereview.chromium.org/1976673002/
It works by creating pooled PortAllocatorSessions which can be picked up
by a P2PTransportChannel when needed (after a local description is set).
This can optimize candidate gathering time when there is some time between
creating a PeerConnection and setting a local description.
R=pthatcher@webrtc.org
Committed: 48e9d05f51
Review URL: https://codereview.webrtc.org/1956453003 .
Cr-Commit-Position: refs/heads/master@{#12729}
Reason for revert:
Breaks remoting_unittests. They defined their own operator== which conflicts with this one.
I'll remove the operator== in a roll CL. But until it's approved, I'm reverting this so the FYI bots will pass.
Original issue's description:
> Implement RTCConfiguration.iceCandidatePoolSize.
>
> It works by creating pooled PortAllocatorSessions which can be picked up
> by a P2PTransportChannel when needed (after a local description is set).
>
> This can optimize candidate gathering time when there is some time between
> creating a PeerConnection and setting a local description.
>
> R=pthatcher@webrtc.org
>
> Committed: 48e9d05f51TBR=pthatcher@webrtc.org,honghaiz@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/1972043004
Cr-Commit-Position: refs/heads/master@{#12709}
It works by creating pooled PortAllocatorSessions which can be picked up
by a P2PTransportChannel when needed (after a local description is set).
This can optimize candidate gathering time when there is some time between
creating a PeerConnection and setting a local description.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1956453003 .
Cr-Commit-Position: refs/heads/master@{#12708}
But keep #including scoped_ptr.h in .h files, so as not to break
WebRTC users who expect those .h files to give them rtc::scoped_ptr.
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1930463002
Cr-Commit-Position: refs/heads/master@{#12530}
Split WebRtcVideoSendStream::OnFrame, to invoke encoder on the worker thread.
BUG=webrtc:5546
Review URL: https://codereview.webrtc.org/1875713002
Cr-Commit-Position: refs/heads/master@{#12471}
If a STUN ping arrives before the remote description does, a prflx
candidate will be created with an unknown generation.
Once the remote description does arrive, the candidate's generation
should be set so it can be sorted properly, and replaced by a non-prflx
candidate once the candidate is signaled.
BUG=webrtc:5752
R=honghaiz@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1868353004 .
Cr-Commit-Position: refs/heads/master@{#12433}
Reason for revert:
Causes P2PTestConductor.LocalP2PTestDtlsTransferCaller to fail on Win dbg.
https://build.chromium.org/p/client.webrtc/builders/Win32%20Debug/builds/7469/steps/peerconnection_unittests/logs/stdio
e:\b\build\slave\win\build\src\webrtc\api\peerconnection_unittest.cc(1221): error: Value of: initiating_client_->ice_connection_state()
Actual: 2
Expected: webrtc::PeerConnectionInterface::kIceConnectionCompleted
Which is: 3
Original issue's description:
> Changed P2PTestConductor to use a separate WorkerThread.
>
> P2PTestConductor currently use the current thread both as a signaling thread and a worker thread. Although convenient while debugging, it can also hide real bugs. An example is https://codereview.webrtc.org/1766653002/#ps420001 where the worker thread is deadlocked in the track proxy due to that the worker thread waits for the signaling thread but the proxy in turns invokes the worker thread..... That bug was only discovered on Android. I suggest we let the P2PTestConductor use a separate thread as a worker thread to better cover how PeerConnections are used in reality.
>
> BUG=webrtc:5426
>
> Committed: https://crrev.com/6172401972c54813698d73580779d675d99178b4
> Cr-Commit-Position: refs/heads/master@{#12252}
TBR=nisse@webrtc.org,pthatcher@webrtc.org,deadbeef@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1866503003
Cr-Commit-Position: refs/heads/master@{#12255}
P2PTestConductor currently use the current thread both as a signaling thread and a worker thread. Although convenient while debugging, it can also hide real bugs. An example is https://codereview.webrtc.org/1766653002/#ps420001 where the worker thread is deadlocked in the track proxy due to that the worker thread waits for the signaling thread but the proxy in turns invokes the worker thread..... That bug was only discovered on Android. I suggest we let the P2PTestConductor use a separate thread as a worker thread to better cover how PeerConnections are used in reality.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1859933002
Cr-Commit-Position: refs/heads/master@{#12252}
A handful of helpers were using SessionDescriptionInterface** output
arguments to return ownership. Chenge them to either use a
rtc::scoped_ptr<SessionDescriptionInterface>* output parameter, or to
simply return a rtc::scoped_ptr<SessionDescriptionInterface>. Not
using raw pointers for things you own is good in general; it will also
be very convenient when scoped_ptr is gone, since unique_ptr doesn't
have .accept() or .use() methods.
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1798173002
Cr-Commit-Position: refs/heads/master@{#12021}
The old code insists on exact cipher suite matches with hardwired expectations. It does this matching parameterized with key type (RSA vs ECDSA) and TLS version (DTLS vs TLS and version 1.0 vs 1.2).
This CL changes things to check against a white-list of cipher suites, with the check parameterized with key type (again RSA vs ECDSA). Then separately checks TLS version since the old implicit check of TLS version by means of resulting cipher suite was too blunt.
Using a white list for cipher suites isn't perfect, but it is safe and requires minimal maintenance. It allows compatibility with not just one exact version of underlying crypto lib, but any version with reasonable defaults.
The CL also re-enables critical tests which had to be disabled recently to allow a boringssl roll.
BUG=webrtc:5634
Review URL: https://codereview.webrtc.org/1774583002
Cr-Commit-Position: refs/heads/master@{#11951}
This CL also adds control flag in webrtcsession_unittests
that says whether to prefer constraints APIs or non-constraints APIs, and uses it in the test that was needed
to uncover the bug.
BUG=webrtc:4906
Review URL: https://codereview.webrtc.org/1775033002
Cr-Commit-Position: refs/heads/master@{#11947}
This involves changing a few verification functions for frames
received so that they always accept the result if there's no stream.
BUG=
Review URL: https://codereview.webrtc.org/1772353002
Cr-Commit-Position: refs/heads/master@{#11937}
Moved VideoSourceInterface to MediaStreamInterface.h
Renamed VideoSourceTest to VideoCapturerTrackSourceTest
Renamed VideoSource to VideoCaptureTrackSource and cl lint and cl format.
BUG=webrtc:5426
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1770003002 .
Cr-Commit-Position: refs/heads/master@{#11893}
Multiple sources with the same names forces ugly GYP hacks in
Chromium's libjingle.gyp. Rename the sources in WebRTC to
enable cleaning this up in Chromium.
To summarize:
webrtc/media/base/constants.{cc,h} -> mediaconstants.{cc,h}
webrtc/p2p/base/constants.{cc,h} -> p2pconstants.{cc,h}
This CL will require coordinating landing a roll in Chromium.
BUG=webrtc:4256
NOTRY=True
Review URL: https://codereview.webrtc.org/1750593002
Cr-Commit-Position: refs/heads/master@{#11842}
In addition to the code moved from talk/app/webrtc
there were some files in webrtc/api/objctests that still
had the libjingle license header.
BUG=webrtc:5418
TBR=tkchin@webrtc.org
NOTRY=True
Review URL: https://codereview.webrtc.org/1680293005
Cr-Commit-Position: refs/heads/master@{#11552}
The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.
License headers will be updated in a follow-up CL.
Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
except for these files:
talk/app/webrtc/peerconnectionendtoend_unittest.cc
talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
webrtc/media/devices/win32devicemanager.cc
The HAVE_SCTP define was added for the peerconnection_unittests target
in api_tests.gyp.
I also checked that none of
SRTP_RELATIVE_PATH
HAVE_SRTP
HAVE_WEBRTC_VIDEO
HAVE_WEBRTC_VOICE
were used by the talk/app/webrtc code.
For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle:
https://codereview.chromium.org/1615433002
BUG=webrtc:5418
NOPRESUBMIT=True
R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1610243002 .
Cr-Commit-Position: refs/heads/master@{#11545}
