DtlsTransportChannelWrapper is renamed to be DtlsTransport which inherits from
DtlsTransportInternal. There will be no concept of "channel" in p2p level.
Both P2PTransportChannel and DtlsTransport don't depend on TransportChannel
and TransportChannelImpl any more and they are removed in this CL.
BUG=none
Review-Url: https://codereview.webrtc.org/2606123002
Cr-Commit-Position: refs/heads/master@{#16173}
Reason for revert:
Failed the memory check.
May need to fix the memory leak.
Original issue's description:
> make the DtlsTransportWrapper inherit form DtlsTransportInternal
>
> BUG=none
>
> Review-Url: https://codereview.webrtc.org/2606123002
> Cr-Commit-Position: refs/heads/master@{#16160}
> Committed: 5aed06c8d3TBR=deadbeef@webrtc.org,pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=none
Review-Url: https://codereview.webrtc.org/2639203004
Cr-Commit-Position: refs/heads/master@{#16162}
Previously, BaseChannel supported a "no RTCP" mode, which wasn't
being used any more and is being deleted.
Also, "RTCP mux required" previously worked by calling "ActivateRtcpMux"
after construction. Now it works by explicitly passing a
"require_rtcp_mux" parameter into the constructor.
BUG=None
Review-Url: https://codereview.webrtc.org/2622613004
Cr-Commit-Position: refs/heads/master@{#16045}
The BaseChannel can set the transport directly without depending on
TransportController.
When initializing the network of the BaseChannel, the ChannelManager will
create TransportChannels with the TransportController.
When enabling bundling, WebRtcSession will get or create TransportChannels
with the TransportController.
When a TransportChannel of the BaseChannel needs to be destroyed, it will
fire a signal to notify the WebRtcSession.
BUG=none.
Review-Url: https://codereview.webrtc.org/2614263002
Cr-Commit-Position: refs/heads/master@{#16043}
Bulk of the changes were produced using
git grep -l ' ASSERT(' | grep -v test | grep -v 'common\.h' |\
xargs -n1 sed -i 's/ ASSERT(/ RTC_DCHECK(/'
followed by additional includes of base/checks.h in affected files,
and git cl format.
Also had to do some tweaks to #if !defined(NDEBUG) logic in the
taskrunner code (webrtc/base/task.cc, webrtc/base/taskparent.cc,
webrtc/base/taskparent.h, webrtc/base/taskrunner.cc), replaced to
consistently use RTC_DCHECK_IS_ON, and some of the checks needed
additional #if protection.
Test code was excluded, because it should probably use RTC_CHECK
rather than RTC_DCHECK.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2620303003
Cr-Commit-Position: refs/heads/master@{#16030}
Bulk of changes done using
git grep -l 'RTC_DCHECK(false)' | \
xargs sed -i 's/RTC_DCHECK(false)/RTC_NOTREACHED()/'
peerconnection.cc also used RTC_DCHECK(false && "msg") in two places,
which were updated manually.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2623313004
Cr-Commit-Position: refs/heads/master@{#16026}
This cl was produced by
git grep -l 'ASSERT(false)' |\
xargs -n1 sed -i 's/ASSERT(false)/RTC_NOTREACHED()/'
followed by additional includes of base/checks.h in affected files,
git cl format to adjust spacing in webrtc/base/transformadapter.cc.
Finally, to make presubmit happy, one unnamed TODO marker was deleted
in that file.
This is a step towards deletion of base/common.h.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2625003003
Cr-Commit-Position: refs/heads/master@{#16009}
The BaseChannel code is geared around RTP; the presence of media engines,
send and receive streams, SRTP, SDP directional attribute negotiation, etc.
It doesn't make sense to use it for SCTP as well. This separation should make
future work both on BaseChannel and the SCTP code paths easier.
SctpDataEngine now becomes SctpTransport, and is used by WebRtcSession
directly. cricket::DataChannel is also renamed, to RtpDataChannel, so it
doesn't get confused with webrtc::DataChannel any more.
Beyond just moving code around, some consequences of this CL:
- We'll now stop using the worker thread for SCTP. Packets will be
processed right on the network thread instead.
- The SDP directional attribute is ignored, as it's supposed to be.
BUG=None
Review-Url: https://codereview.webrtc.org/2564333002
Cr-Original-Commit-Position: refs/heads/master@{#15906}
Committed: 67b3bbe639
Review-Url: https://codereview.webrtc.org/2564333002
Cr-Commit-Position: refs/heads/master@{#15973}
Reason for revert:
Hitting DCHECK in chromium's WebrtcTransportTest.TerminateDataChannel and WebrtcTransportTest.DataStreamLate. Will investigate and reland.
Original issue's description:
> Separating SCTP code from BaseChannel/MediaChannel.
>
> The BaseChannel code is geared around RTP; the presence of media engines,
> send and receive streams, SRTP, SDP directional attribute negotiation, etc.
> It doesn't make sense to use it for SCTP as well. This separation should make
> future work both on BaseChannel and the SCTP code paths easier.
>
> SctpDataEngine now becomes SctpTransport, and is used by WebRtcSession
> directly. cricket::DataChannel is also renamed, to RtpDataChannel, so it
> doesn't get confused with webrtc::DataChannel any more.
>
> Beyond just moving code around, some consequences of this CL:
> - We'll now stop using the worker thread for SCTP. Packets will be
> processed right on the network thread instead.
> - The SDP directional attribute is ignored, as it's supposed to be.
>
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2564333002
> Cr-Commit-Position: refs/heads/master@{#15906}
> Committed: 67b3bbe639TBR=pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None
Review-Url: https://codereview.webrtc.org/2614813003
Cr-Commit-Position: refs/heads/master@{#15908}
The BaseChannel code is geared around RTP; the presence of media engines,
send and receive streams, SRTP, SDP directional attribute negotiation, etc.
It doesn't make sense to use it for SCTP as well. This separation should make
future work both on BaseChannel and the SCTP code paths easier.
SctpDataEngine now becomes SctpTransport, and is used by WebRtcSession
directly. cricket::DataChannel is also renamed, to RtpDataChannel, so it
doesn't get confused with webrtc::DataChannel any more.
Beyond just moving code around, some consequences of this CL:
- We'll now stop using the worker thread for SCTP. Packets will be
processed right on the network thread instead.
- The SDP directional attribute is ignored, as it's supposed to be.
BUG=None
Review-Url: https://codereview.webrtc.org/2564333002
Cr-Commit-Position: refs/heads/master@{#15906}
(This is a re-upload of https://codereview.webrtc.org/2567243003/, the
CQ stopped working there.)
The previously used WebRtcSession::GetTransportStats did a synchronous
invoke per channel (voice, video, data) on the signaling thread to the
network thread - e.g. 3 blocking invokes.
It is replaced by WebRtcSession::GetStats[_s] which can be invoked on
the signaling thread or on any thread if a ChannelNamePairs argument is
present (provided by WebRtcSession::GetChannelNamePairs on the signaling
thread).
With these changes, and changes allowing the getting of certificates
from any thread, the RTCStatsCollector can turn the 3 blocking thread
invokes into 1 non-blocking invoke.
BUG=webrtc:6875, chromium:627816
Review-Url: https://codereview.webrtc.org/2583883002
Cr-Commit-Position: refs/heads/master@{#15672}
"Crypto required" is a property of the PeerConnection of construction
time; it has nothing to do with SDP. So I'm moving it out of
MediaContentDescription and putting it in the BaseChannel constructor
instead. This is more intuitive, and provides the added assurance that
"secure_required_" can't be flipped from "true" to "false".
BUG=None
Review-Url: https://codereview.webrtc.org/2537343003
Cr-Commit-Position: refs/heads/master@{#15579}
Changing the configuration will cause subsequently generated offers to change
the ufrag/pwd as necessary, so that a new round of gathering is started that
uses the new configuration.
This CL also makes some minor unrelated changes: changing the reference SDP in
the PC tests to more match what we generate, and relaxing the network thread
requirement for JsepTransport (since there's no reason the "needs-ice-restart"
flag can't be accessed from the signaling thread).
BUG=webrtc:6714
Review-Url: https://codereview.webrtc.org/2563153002
Cr-Commit-Position: refs/heads/master@{#15527}
This error occurred if you created a data channel before there's a data
m= section. But this is expected to happen, since creating a data
channel is how you get a data m= section in the first place.
BUG=chromium:579430
NOTRY=true
TBR=pthatcher@webrtc.org
Review-Url: https://codereview.webrtc.org/2396013002
Cr-Commit-Position: refs/heads/master@{#14537}
Log the DTLS handshake error code in OpenSSLStreamAdapter.
Forward the error code to WebRTCSession with the Signals.
This part is only for the WebRTC native code.
To make it work, need another CL for Chromium.
BUG=webrtc:5959
Review-Url: https://codereview.webrtc.org/2167363002
Cr-Commit-Position: refs/heads/master@{#13940}
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}
Add IsClosed check when excuting some functions so that they can return early if the PeerConnection is closed.
The observer will not be called after the PeerConnection is closed.
BUG=webrtc:5861
Review-Url: https://codereview.webrtc.org/1975453002
Cr-Commit-Position: refs/heads/master@{#13544}
we will periodically check if any network does not have any connection on it and if yes, attempt to re-gather on those networks.
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2025573002 .
Cr-Commit-Position: refs/heads/master@{#13367}
Relanding again after fixing issue with RTC_DCHECKs.
This CL eliminates the need for the extra layer of indirection provided by
mediastreamprovider.h. It will thus make it easier to implement new
functionality in RtpSender/RtpReceiver.
It also brings us one step closer to the end goal of combining "senders"
and "send streams". Currently the sender still needs to go through the
BaseChannel and MediaChannel, using an SSRC as a key.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2046173002 .
Cr-Commit-Position: refs/heads/master@{#13305}
Reason for revert:
Broke video sending for iOS AppRTCDemo. To repro, run iOS AppRTCDemo in Release in loopback mode. The revision prior to this change worked.
Original issue's description:
> Reland of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
>
> This eliminates the need for the extra layer of indirection provided by
> mediastreamprovider.h. It will thus make it easier to implement new
> functionality in RtpSender/RtpReceiver.
>
> It also brings us one step closer to the end goal of combining "senders"
> and "send streams". Currently the sender still needs to go through the
> BaseChannel and MediaChannel, using an SSRC as a key.
>
> R=pthatcher@webrtc.org
>
> Committed: 2d5491783aTBR=pthatcher@webrtc.org,deadbeef@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/2092273003
Cr-Commit-Position: refs/heads/master@{#13289}
This eliminates the need for the extra layer of indirection provided by
mediastreamprovider.h. It will thus make it easier to implement new
functionality in RtpSender/RtpReceiver.
It also brings us one step closer to the end goal of combining "senders"
and "send streams". Currently the sender still needs to go through the
BaseChannel and MediaChannel, using an SSRC as a key.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2046173002 .
Cr-Commit-Position: refs/heads/master@{#13287}
Reason for revert:
Broke peerconnection_unittest somehow, due to introduction of thread check. Will fix and reland.
Original issue's description:
> Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
>
> This eliminates the need for the extra layer of indirection provided by
> mediastreamprovider.h. It will thus make it easier to implement new
> functionality in RtpSender/RtpReceiver.
>
> It also brings us one step closer to the end goal of combining "senders"
> and "send streams". Currently the sender still needs to go through the
> BaseChannel and MediaChannel, using an SSRC as a key.
>
> R=pthatcher@webrtc.org
>
> Committed: https://crrev.com/bc5831999d3354509d75357b659b4bb8e39f8a6c
> Cr-Commit-Position: refs/heads/master@{#13285}
TBR=pthatcher@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/2099843003
Cr-Commit-Position: refs/heads/master@{#13286}
This eliminates the need for the extra layer of indirection provided by
mediastreamprovider.h. It will thus make it easier to implement new
functionality in RtpSender/RtpReceiver.
It also brings us one step closer to the end goal of combining "senders"
and "send streams". Currently the sender still needs to go through the
BaseChannel and MediaChannel, using an SSRC as a key.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2046173002 .
Cr-Commit-Position: refs/heads/master@{#13285}
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 means there's only one thread hop to the worker thread.
At the video engine level, SetOptions and SetSource
are combined into one method (all within the same critical section)
which ensures that no frame will be encoded while SetVideoSend
is only partially finished.
BUG=webrtc:5691
Review-Url: https://codereview.webrtc.org/1838413002
Cr-Commit-Position: refs/heads/master@{#13022}
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}
Reason for revert:
There are more CreatePeerConnection calls than I anticipated/had found in Chromium, like remoting/protocol/webrtc_transport.cc. Reverting due to broken Chromium FYI bots.
Original issue's description:
> Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface.
>
> 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::CreatePeerConnection, is
> updated to take a generator instead of a store. But as to not break Chromium,
> the old function signature taking a store is kept. It is implemented to invoke
> the generator version by wrapping the store in an
> RTCCertificateGeneratorStoreWrapper. As soon as Chromium is updated to use the
> new function signature we can remove the old CreatePeerConnection.
> Due to having multiple CreatePeerConnection signatures, some calling places
> are updated to resolve the ambiguity introduced.
>
> BUG=webrtc:5707, webrtc:5708
> R=phoglund@webrtc.org, tommi@webrtc.org
> TBR=tkchin@webrc.org
>
> Committed: 400781a209TBR=tkchin@webrtc.org,tommi@webrtc.org,phoglund@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5707, webrtc:5708
Review-Url: https://codereview.webrtc.org/2020633002
Cr-Commit-Position: refs/heads/master@{#12948}
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::CreatePeerConnection, is
updated to take a generator instead of a store. But as to not break Chromium,
the old function signature taking a store is kept. It is implemented to invoke
the generator version by wrapping the store in an
RTCCertificateGeneratorStoreWrapper. As soon as Chromium is updated to use the
new function signature we can remove the old CreatePeerConnection.
Due to having multiple CreatePeerConnection signatures, some calling places
are updated to resolve the ambiguity introduced.
BUG=webrtc:5707, webrtc:5708
R=phoglund@webrtc.org, tommi@webrtc.orgTBR=tkchin@webrc.org
Review URL: https://codereview.webrtc.org/2013523002 .
Cr-Commit-Position: refs/heads/master@{#12947}
They don't really belong in PeerConnection because the state at that
level can change when a transport channel is removed. That makes almost
any state transition possible.
The asserts are now in P2PTransportChannel (the equivalent to
IceTransport in the spec). Currently it has a reduced set of states,
that don't even take into account writability, but I plan to change
that soon.
BUG=webrtc:4757
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2005573002 .
Cr-Commit-Position: refs/heads/master@{#12937}
With this change, when max-bundle and rtcp-mux are both enabled, we no
longer create and destroy a temporary transport channel when a media
channel gets added. Instead, the media channel uses the correct bundled
transport channel from the start.
This fixes a bug where adding a media type would cause the ICE state to
briefly become Disconnected and then immediately recover. The temporary
channel was created in a non-writable state, which caused the
TransportController to declare the ICE state to be Disconnected (as not
all transport channels were writable). Right after creation, the
temporary channel was then destroyed and the ICE state went back to the
correct one.
BUG=webrtc:5856
Review-Url: https://codereview.webrtc.org/1972493002
Cr-Commit-Position: refs/heads/master@{#12781}
This is similar to how a "receive" method is used to apply
RtpParameters to an RtpReceiver in ORTC. Currently, SetParameters
doesn't allow changing the parameters, so the main use of the API is
to retrieve the set of configured codecs. But other uses will likely
be made possible in the future.
R=glaznev@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1917193008 .
Cr-Commit-Position: refs/heads/master@{#12761}
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}
BaseChannel do calls to transport_channel on network_thread,
while keep calls to media_engine on worker_thread.
It still works when network_thread == worker_thread.
BUG=webrtc:5645
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1903393004 .
Cr-Commit-Position: refs/heads/master@{#12690}
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}
They're just no-ops now, and will soon go away.
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1914153002
Cr-Commit-Position: refs/heads/master@{#12510}
This propagated into various other places. Also had to #include headers that
were implicitly pulled by "scoped_ptr.h".
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1920043002
Cr-Commit-Position: refs/heads/master@{#12501}
