Also did some slight refactoring of the code that turns constraints
into rtc::Optionals. Used a template method to avoid code duplication,
and used the same pattern for "CopyConstraintsIntoAudioOptions" as was
being used for "CopyConstraintsIntoRtcConfiguration".
BUG=webrtc:6752
Review-Url: https://codereview.webrtc.org/2628523003
Cr-Commit-Position: refs/heads/master@{#16063}
This change moves the definition of the UMA MetricsObserverInterface from api/peerconnectioninterface.h into api/umametrics.h. This allows us to remove the unwanted dependency on peerconnectioninterface.h from files in webrtc/p2p.
This is a simple refactoring with no functional changes.
BUG=None
Review-Url: https://codereview.webrtc.org/2627093005
Cr-Commit-Position: refs/heads/master@{#16020}
Most notably, will return "INVALID_MODIFICATION" if a field in the
configuration was modified and modification of that field isn't supported.
Also changing RTCError to a class that wraps an enum type, because it will
eventually need to hold other information (like SDP line number), to match
the RTCError that was recently added to the spec:
https://github.com/w3c/webrtc-pc/pull/850
BUG=webrtc:6916
Review-Url: https://codereview.webrtc.org/2587133004
Cr-Original-Commit-Position: refs/heads/master@{#15777}
Committed: 7a5fa6cd61
Review-Url: https://codereview.webrtc.org/2587133004
Cr-Commit-Position: refs/heads/master@{#16016}
Refactor "OPT_SSLTCP" renaming it to "OPT_TLS_FAKE", making it clear
that it's not actually some kind of SSL over TCP. Also making it clear
that it's mutually exclusive with OPT_TLS. Maintaining deprecated
backwards compatible support for "OPT_SSLTCP".
Add "OPT_TLS_INSECURE" that implements the new certificate-check
disabled TLS mode, which is also mutually exclusive with the other
TLS options.
PortAllocator: Add a new TLS policy enum TlsCertPolicy which defines
the new insecure mode and added it as a RelayCredentials member.
TurnPort: Add new TLS policy member with appropriate getter and setter
to avoid constructor bloat. Initialize it from the RelayCredentials
after the TurnPort is created.
Expose the new feature in the PeerConnection API via
IceServer.tls_certificate_policy as well as via the Android JNI
PeerConnection API.
For security reasons we ensure that:
1) The policy is always explicitly initialized to secure.
2) API users have to explicitly integrate with the feature to
use it, and will otherwise get no change in behavior.
3) The feature is not immediately exposed in non-native
contexts. For example, disabling of certificate validation
is not implemented via URI parsing since this would
immediately allow it to be used from a web page.
This is a second attempt of https://codereview.webrtc.org/2557803002/
which was rolled back in https://codereview.webrtc.org/2590153002/
BUG=webrtc:6840
Review-Url: https://codereview.webrtc.org/2594623002
Cr-Commit-Position: refs/heads/master@{#15967}
Reason for revert:
Broke chromium FYI bot because the chromium mock PC overrides the method whose signature is changing.
Also broke a downstream internal test, which I need to investigate further.
Original issue's description:
> Adding error output param to SetConfiguration, using new RTCError type.
>
> Most notably, will return "INVALID_MODIFICATION" if a field in the
> configuration was modified and modification of that field isn't supported.
>
> Also changing RTCError to a class that wraps an enum type, because it will
> eventually need to hold other information (like SDP line number), to match
> the RTCError that was recently added to the spec:
> https://github.com/w3c/webrtc-pc/pull/850
>
> BUG=webrtc:6916
>
> Review-Url: https://codereview.webrtc.org/2587133004
> Cr-Commit-Position: refs/heads/master@{#15777}
> Committed: 7a5fa6cd61TBR=pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6916
Review-Url: https://codereview.webrtc.org/2600813002
Cr-Commit-Position: refs/heads/master@{#15778}
Most notably, will return "INVALID_MODIFICATION" if a field in the
configuration was modified and modification of that field isn't supported.
Also changing RTCError to a class that wraps an enum type, because it will
eventually need to hold other information (like SDP line number), to match
the RTCError that was recently added to the spec:
https://github.com/w3c/webrtc-pc/pull/850
BUG=webrtc:6916
Review-Url: https://codereview.webrtc.org/2587133004
Cr-Commit-Position: refs/heads/master@{#15777}
The new name, OnCompleteReports rather than OnComplete, is needed
because in C++ method lookup, overriding a method hides all otherwise
inherited methods with the same name, even if they have a different
signature. And here, the intention is that each subclass should
override one or the other of the two methods, and inherit the method it
doesn't override.
This cl is a prerequisite for
https://codereview.webrtc.org/2567143003/, because the Chrome glue
code needs to retain the stats report after the OnComplete method has
returned.
Currently, Chrome makes a copy of the stats mapping (which breaks when
changing ValuePtr from an rtc::linked_ptr to an std::unique_ptr). After
this cl, Chrome can be fixed to take ownership and no longer needs to
copy anything, unblocking cl 2567143003.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2584553002
Cr-Commit-Position: refs/heads/master@{#15708}
Reason for revert:
This CL broke all Chromium WebRTC FYI bots. A roll+fix was attempted here: https://codereview.chromium.org/2590783003/, but failed to land. I'm reverting this CL now to make the tree green again. Make the API change gradual when you reland so that we can update Chromium between.
Original issue's description:
> Add disabled certificate check support to IceServer PeerConnection API.
>
> Refactor "OPT_SSLTCP" renaming it to "OPT_TLS_FAKE", making it clear
> that it's not actually some kind of SSL over TCP. Also making it clear
> that it's mutually exclusive with OPT_TLS.
>
> Add "OPT_TLS_INSECURE" that implements the new certificate-check
> disabled TLS mode, which is also mutually exclusive with the other
> TLS options.
>
> PortAllocator: Add a new TLS policy enum TlsCertPolicy which defines
> the new insecure mode and added it as a RelayCredentials member.
>
> TurnPort: Add new TLS policy member with appropriate getter and setter
> to avoid constructor bloat. Initialize it from the RelayCredentials
> after the TurnPort is created.
>
> Expose the new feature in the PeerConnection API via
> IceServer.tls_certificate_policy as well as via the Android JNI
> PeerConnection API.
>
> For security reasons we ensure that:
>
> 1) The policy is always explicitly initialized to secure.
> 2) API users have to explicitly integrate with the feature to
> use it, and will otherwise get no change in behavior.
> 3) The feature is not immediately exposed in non-native
> contexts. For example, disabling of certificate validation
> is not implemented via URI parsing since this would
> immediately allow it to be used from a web page.
>
> BUG=webrtc:6840
>
> Review-Url: https://codereview.webrtc.org/2557803002
> Cr-Commit-Position: refs/heads/master@{#15670}
> Committed: b0f04fdb9eTBR=pthatcher@webrtc.org,deadbeef@webrtc.org,hnsl@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6840
Review-Url: https://codereview.webrtc.org/2590153002
Cr-Commit-Position: refs/heads/master@{#15703}
Refactor "OPT_SSLTCP" renaming it to "OPT_TLS_FAKE", making it clear
that it's not actually some kind of SSL over TCP. Also making it clear
that it's mutually exclusive with OPT_TLS.
Add "OPT_TLS_INSECURE" that implements the new certificate-check
disabled TLS mode, which is also mutually exclusive with the other
TLS options.
PortAllocator: Add a new TLS policy enum TlsCertPolicy which defines
the new insecure mode and added it as a RelayCredentials member.
TurnPort: Add new TLS policy member with appropriate getter and setter
to avoid constructor bloat. Initialize it from the RelayCredentials
after the TurnPort is created.
Expose the new feature in the PeerConnection API via
IceServer.tls_certificate_policy as well as via the Android JNI
PeerConnection API.
For security reasons we ensure that:
1) The policy is always explicitly initialized to secure.
2) API users have to explicitly integrate with the feature to
use it, and will otherwise get no change in behavior.
3) The feature is not immediately exposed in non-native
contexts. For example, disabling of certificate validation
is not implemented via URI parsing since this would
immediately allow it to be used from a web page.
BUG=webrtc:6840
Review-Url: https://codereview.webrtc.org/2557803002
Cr-Commit-Position: refs/heads/master@{#15670}
An external audio mixer will be passed from PeerConnectionFactory to
AudioTransportProxy.
This CL has rewritten based on reverted CL
https://codereview.chromium.org/2539213003/
The only difference is that
static MediaEngineInterface* Create(
webrtc::AudioDeviceModule* adm,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
audio_decoder_factory,
WebRtcVideoEncoderFactory* video_encoder_factory,
WebRtcVideoDecoderFactory* video_decoder_factory);
in media/engine/webrtcmediaengine.h is kept in this CL instead of
replaced for backward compatibility.
BUG=webrtc:6457
Review-Url: https://codereview.webrtc.org/2570993002
Cr-Commit-Position: refs/heads/master@{#15580}
The Chromium mock implementation implements the new GetStats API, so we
can remove this default implementation.
BUG=chromium:627816
Review-Url: https://codereview.webrtc.org/2566143002
Cr-Commit-Position: refs/heads/master@{#15563}
Reason for revert:
A interface change broke some downstream code in google3.
Original issue's description:
> Support external audio mixer in webrtc.
>
> An external audio mixer will be passed from PeerConnectionFactory to
> AudioTransportProxy.
>
> BUG=webrtc:6457
>
> Committed: https://crrev.com/f6bcac59e88c3be5ffd73bbb1098f2d82ee316a1
> Cr-Commit-Position: refs/heads/master@{#15556}
TBR=solenberg@webrtc.org,aleloi@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:6457
Review-Url: https://codereview.webrtc.org/2562333003
Cr-Commit-Position: refs/heads/master@{#15557}
An external audio mixer will be passed from PeerConnectionFactory to
AudioTransportProxy.
BUG=webrtc:6457
Review-Url: https://codereview.webrtc.org/2539213003
Cr-Commit-Position: refs/heads/master@{#15556}
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}
The enum is at about the same level of detail as DOMExceptions, and I
looked through the spec making sure that chromium will be able to perform
the DOMException mapping for each one.
The new enum is called RtcError and is outside the PeerConnectionInterface
scope, because we may want to use this for things not associated with a
PeerConnection in the future.
This CL doesn't yet use the error enum anywhere; that will probably happen
in follow-up CLs for the individual methods.
BUG=webrtc:6855
Review-Url: https://codereview.webrtc.org/2564683002
Cr-Commit-Position: refs/heads/master@{#15526}
Change the second parameter type to a const reference of vector so that
the vector will not be copied.
BUG=none
Review-Url: https://codereview.webrtc.org/2551603003
Cr-Commit-Position: refs/heads/master@{#15396}
Change the default value of rtcp-mux policy in RTCConfiguration.
Refactor the peerconnectioninterface and webrtcsession unit tests.
BUG=webrtc:6030
Review-Url: https://codereview.webrtc.org/2043193003
Cr-Commit-Position: refs/heads/master@{#15217}
Added the callback in native c++ API.
The callback function is called when a track is added and a new RtpReceiver is created.
The application can tell when tracks are added to the streams by listening to this callback.
BUG=webrtc:6112
Review-Url: https://codereview.webrtc.org/2505173002
Cr-Commit-Position: refs/heads/master@{#15142}
Just returns the configuration the PC was constructed with, or the last
one passed into SetConfiguration.
BUG=chromium:587453
Review-Url: https://codereview.webrtc.org/2504103002
Cr-Commit-Position: refs/heads/master@{#15111}
New file structure and targets:
rtc_stats_api
webrtc/api/stats/rtcstats.h
webrtc/api/stats/rtcstats_objects.h
webrtc/api/stats/rtcstatsreport.h
rtc_stats (dep on rtc_stats_api)
webrtc/stats/rtcstats.cc
webrtc/stats/rtcstats_objects.cc
webrtc/stats/rtcstatsreport.cc
libjingle_peerconnection (dep on rtc_stats)
webrtc/api/rtcstatscollector.cc
webrtc/api/rtcstatscollector.h
Placing rtc_stats_api headers in this separate target instead of
libjingle_peerconnection avoids a circular dependency
libjingle_peerconnection -> rtc_stats -> libjingle_peerconnection
Code changes:
PeerConnectionInterface::GetStats(RTCStatsCollectorCallback*) added for
the new stats collection API. Implemented by PeerConnection.
BUG=chromium:627816
Review-Url: https://codereview.webrtc.org/2331373004
Cr-Commit-Position: refs/heads/master@{#14246}
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}
When the flag prune_turn_ports is set, When a high-priority turn port becomes available, it will prune low-priority ones. The pruned port will not be used for creating connections locally and its candidates will not be sent over to the remove side (unless they have been sent before being pruned).
This effectively reduces the number of TURN candidates and connections created by TURN ports.
BUG=
R=deadbeef@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2093623004 .
Committed: https://crrev.com/17aac053f585e892114974d2eb248e05ad37f973
Cr-Original-Commit-Position: refs/heads/master@{#13335}
Cr-Commit-Position: refs/heads/master@{#13354}
Reason for revert:
Breaks Win32/Win64 Debug bots in client.webrtc waterfall
Original issue's description:
> Add config to prune low-priority TURN ports for creating connections
> When the flag prune_turn_ports is set, When a high-priority turn port becomes available, it will prune low-priority ones. The pruned port will not be used for creating connections locally and its candidates will not be sent over to the remove side (unless they have been sent before being pruned).
>
> This effectively reduces the number of TURN candidates and connections created by TURN ports.
>
> BUG=
> R=deadbeef@webrtc.org, pthatcher@webrtc.org
>
> Committed: https://crrev.com/17aac053f585e892114974d2eb248e05ad37f973
> Cr-Commit-Position: refs/heads/master@{#13335}
TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,honghaiz@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review-Url: https://codereview.webrtc.org/2111663003
Cr-Commit-Position: refs/heads/master@{#13342}
Reason for revert:
Reverting all CLs related to moving the eventlog, as they break Chromium tests.
Original issue's description:
> Move RtcEventLog object from inside VoiceEngine to Call.
>
> In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
> The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.
>
> BUG=webrtc:4741,webrtc:5603,chromium:609749
> R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org
>
> Committed: https://crrev.com/1895526c6130e3d0e9b154f95079b8eda7567016
> Cr-Commit-Position: refs/heads/master@{#13321}
TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741,webrtc:5603,chromium:609749
Review-Url: https://codereview.webrtc.org/2111813002
Cr-Commit-Position: refs/heads/master@{#13340}
Reason for revert:
Reverting all CLs related to moving the eventlog, as they break Chromium tests.
Original issue's description:
> Fix to make the start/stop functions for the Rtc Eventlog non-virtual.
>
> This is needed to prevent the Chromium import bot from breaking.
>
> BUG=
> R=tommi@webrtc.org
>
> Committed: https://crrev.com/df6ecea8ac7c4c3bddeda089d5fb9eccdf38a0a6
> Cr-Commit-Position: refs/heads/master@{#13324}
TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review-Url: https://codereview.webrtc.org/2111803002
Cr-Commit-Position: refs/heads/master@{#13339}
When the flag prune_turn_ports is set, When a high-priority turn port becomes available, it will prune low-priority ones. The pruned port will not be used for creating connections locally and its candidates will not be sent over to the remove side (unless they have been sent before being pruned).
This effectively reduces the number of TURN candidates and connections created by TURN ports.
BUG=
R=deadbeef@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2093623004 .
Cr-Commit-Position: refs/heads/master@{#13335}
In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.
BUG=webrtc:4741,webrtc:5603,chromium:609749
R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1748403002 .
Cr-Commit-Position: refs/heads/master@{#13321}
This CL turns nativeConfiguration into createNativeConfiguration returning a
pointer or nil on failure. This method's certificate generation is updated to
use the new API and reports failure (nil) if unsuccessful instead of relying on
the default certificate. We also remove the implicit assumption (now incorrect)
that RSA is the default. This is the same type of changes as was done in
https://codereview.webrtc.org/1965313002 but this file
(RTCPeerConnectionInterface.mm) was forgotten.
With no more usages of kIdentityName it and dtlsidentitystore.cc is removed.
Also removes unnecessary #include in peerconnectioninterface.h that was still
remnant due to an indirect include of kIdentityName.
RTCConfiguration+Private.h now lists method nativeEncryptionKeyTypeForKeyType
which was added in the above mentioned prior CL.
BUG=webrtc:5707, webrtc:5708
Review-Url: https://codereview.webrtc.org/2035473004
Cr-Commit-Position: refs/heads/master@{#13089}
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 allows webrtc to not gather on cellular networks if wifi or
other low cost networks are present.
BUG=
Review-Url: https://codereview.webrtc.org/1987833002
Cr-Commit-Position: refs/heads/master@{#12979}
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}
