15 Commits

Author SHA1 Message Date
ivoc
14d5dbe5b3 Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface"
The breaking tests in Chromium have been temporarily disabled, they will be fixed and reenabled soon.

Original CLs: https://codereview.webrtc.org/1748403002/, https://codereview.webrtc.org/2107253002/ and https://codereview.webrtc.org/2106103003/.

TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org,tkchin@webrtc.org
BUG=webrtc:4741, webrtc:5603, chromium:609749

Review-Url: https://codereview.webrtc.org/2110113003
Cr-Commit-Position: refs/heads/master@{#13379}
2016-07-04 14:07:03 +00:00
ivoc
9e03c3b372 Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ )
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}
2016-06-30 07:59:49 +00:00
Ivo Creusen
1895526c61 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

Review URL: https://codereview.webrtc.org/1748403002 .

Cr-Commit-Position: refs/heads/master@{#13321}
2016-06-29 11:57:01 +00:00
Taylor Brandstetter
5d97a9a05b Adding more detail to MessageQueue::Dispatch logging.
Every message will now be traced with the location from which it was
posted, including function name, file and line number.

This CL also writes a normal LOG message when the dispatch took more
than a certain amount of time (currently 50ms).

This logging should help us identify messages that are taking
longer than expected to be dispatched.

R=pthatcher@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/2019423006 .

Cr-Commit-Position: refs/heads/master@{#13104}
2016-06-10 21:17:33 +00:00
Henrik Boström
d03c23b216 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::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}
2016-06-01 09:44:29 +00:00
hbos
d7973ccdb5 Revert of Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface. (patchset #2 id:20001 of https://codereview.webrtc.org/2013523002/ )
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: 400781a209

TBR=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}
2016-05-27 13:08:58 +00:00
Henrik Boström
400781a209 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

Review URL: https://codereview.webrtc.org/2013523002 .

Cr-Commit-Position: refs/heads/master@{#12947}
2016-05-27 12:52:06 +00:00
ivoc
c1513ee1a3 Add a parameter to set a maximum file size when starting an RTC event log on the PeerConnectionFactory API.
The caller can set a negative or zero file size to avoid using a limit.
BUG=

Review-Url: https://codereview.webrtc.org/1974453002
Cr-Commit-Position: refs/heads/master@{#12730}
2016-05-13 15:30:44 +00:00
kwiberg
d1fe281e12 Replace scoped_ptr with unique_ptr in webrtc/api/
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}
2016-04-27 13:47:40 +00:00
nisse
72c8d2b708 Rename BEGIN_PROXY_MAP --> BEGIN_SIGNALLING_PROXY_MAP.
And BEGIN_WORKER_PROXY_MAP --> BEGIN_PROXY_MAP.

This rename was suggested by Tommi, with the idea that a proxy
invoking methods on the worker thread should be the common case.
It's a followup to https://codereview.webrtc.org/1861633002/

This cl also adds unittests for proxy calls to the
worker thread.

BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1871833002

Cr-Commit-Position: refs/heads/master@{#12374}
2016-04-15 10:49:13 +00:00
nisse
5b68ab50bb Extended proxy abstraction, to call certain methods to the worker thread.
Extracted from cl https://codereview.webrtc.org/1766653002/, where
AddOrUpdateSink results in a deadlock.

BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1861633002

Cr-Commit-Position: refs/heads/master@{#12281}
2016-04-07 14:46:00 +00:00
perkj
a3ede6c510 Renamed VideoSourceInterface to VideoTrackSourceInterface.
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}
2016-03-08 00:28:03 +00:00
hta
a2a49d9d9c This CL provides interfaces that do not use constraints for
all interfaces that formerly took constraints parameters
in name=value form.

This is in preparation for making Chrome only use these
explicit interfaces.

BUG=webrtc:4906

Review URL: https://codereview.webrtc.org/1717583002

Cr-Commit-Position: refs/heads/master@{#11870}
2016-03-04 10:51:44 +00:00
kjellander
b24317bfda Fix license headers in webrtc/api.
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}
2016-02-10 15:54:53 +00:00
Henrik Kjellander
15583c19d7 Move talk/app/webrtc to webrtc/api
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}
2016-02-10 09:53:26 +00:00