13 Commits

Author SHA1 Message Date
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
c36b31b78e Embed a cricket::MediaConfig in RTCConfiguration.
This eliminates some instances rtc:Optional and makes the code
simpler. No changes in defaults or other behaviour are intended.

BUG=webrtc:4906

Review URL: https://codereview.webrtc.org/1818033002

Cr-Commit-Position: refs/heads/master@{#12326}
2016-04-12 06:25:34 +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
Taylor Brandstetter
a8415fe9ea Adding comments about threading around CreatePeerConnectionFactory.
This has confused a lot of developers (understandably).

R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1828463002 .

Cr-Commit-Position: refs/heads/master@{#12105}
2016-03-23 17:38:16 +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
hbos
5291393510 DtlsIdentityStoreInterface::RequestIdentity: const& parameters
Changing from:

virtual void RequestIdentity(
    rtc::KeyParams key_params,
    rtc::Optional<uint64_t> expires,
    const rtc::scoped_refptr<DtlsIdentityRequestObserver>& observer);

to:

virtual void RequestIdentity(
    const rtc::KeyParams& key_params,
    const rtc::Optional<uint64_t>& expires_ms,
    const rtc::scoped_refptr<DtlsIdentityRequestObserver>& observer);

Making FakeDtlsIdentityStore DCHECK that |expires_ms| is not set, since it does not support that parameterization.

In a follow-up chromium CL the new signature will be used.

BUG=webrtc:5092, chromium:544902

Review URL: https://codereview.webrtc.org/1766673002

Cr-Commit-Position: refs/heads/master@{#11892}
2016-03-07 23:14:48 +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
Per
0f13ec1265 Removed VideoSource dependency to ChannelManager.
Instead VideoSource directly access the cricket::VideoCapturer via the worker_thread.

Document: https://docs.google.com/a/google.com/document/d/1mEIw_0uDzyHjL3l8a82WKp6AvgR8Tlwn9JGvhbUjVpY/edit?usp=sharing

BUG=webrtc:5426
R=nisse@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1759473003 .

Cr-Commit-Position: refs/heads/master@{#11852}
2016-03-03 08:22:41 +00:00
hbos
25359e0cc2 DtlsIdentityStoreInterface::RequestIdentity gets optional expires param.
This is a preparation CL. The expires param will be used in
a follow-up CL. Initially it will only be used by the
chromium implementation. Then we will either update the
webrtc implementation (DtlsIdentityStoreImpl) to use it or
we will remove that store completely as part of clean-up
work.

There are currently two versions of RequestIdentity, one
that takes KeyType and one that takes KeyParams.

The KeyType version is removed in favor of the new
KeyParams + expires version. The KeyParams version without
expires is kept as to not break chromium which currently
implements that. This is the version that can be removed in
a follow-up CL.

BUG=webrtc:5092, chromium:544902

Review URL: https://codereview.webrtc.org/1749193002

Cr-Commit-Position: refs/heads/master@{#11846}
2016-03-02 15:55:56 +00:00
nisse
51542be8ce Introduce struct MediaConfig, with construction-time settings.
Pass it to MediaController constructor and down to WebRtcVideoEngine2
and WebRtcVoiceEngine.

Follows discussion on https://codereview.webrtc.org/1646253004/

TBR=pthatcher@webrtc.org
BUG=webrtc:5438

Review URL: https://codereview.webrtc.org/1670153003

Cr-Commit-Position: refs/heads/master@{#11595}
2016-02-12 10:27:12 +00:00
kjellander@webrtc.org
5ad129741c Rename webrtc/media/webrtc -> webrtc/media/engine
BUG=webrtc:5420
NOTRY=True
R=pthatcher@google.com, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1684163002 .

Cr-Commit-Position: refs/heads/master@{#11591}
2016-02-12 05:39:50 +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