24 Commits

Author SHA1 Message Date
hbos
f9da44dbcf RTCPeerConnectionInterface.mm createNativeConfiguration and other clean-up.
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}
2016-06-09 10:18:35 +00:00
danilchap
e9021a3601 Propogate network-worker thread split to api
BUG=webrtc:5645

Review-Url: https://codereview.webrtc.org/1968393002
Cr-Commit-Position: refs/heads/master@{#12767}
2016-05-17 08:52:06 +00:00
perkj
9e083d2ac5 Reland of Delete empty API files and cleaned up includes. (patchset #1 id:1 of https://codereview.webrtc.org/1813083002/ )
Reason for revert:
New attempt. Cl for removing videosourceinterface.h dep in chrome is landed here: https://codereview.chromium.org/1810273003/

Original issue's description:
> Revert of Delete empty API files and cleaned up includes. (patchset #2 id:20001 of https://codereview.webrtc.org/1809053002/ )
>
> Reason for revert:
> Breaks Chromium build. Need to remove the references to the obsolete header files from Chromium and reland.
>
> Original issue's description:
> > Delete empty API files and cleaned up includes.
> >
> > TBR=glaznev@webrtc.org
> >
> > BUG=webrtc:5426
> >
> > Committed: https://crrev.com/c9022f508644dc33c01b05cb22ebfc2be145d6b2
> > Cr-Commit-Position: refs/heads/master@{#12039}
>
> TBR=nisse@webrtc.org,glaznev@webrtc.org,perkj@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5426
>
> Committed: https://crrev.com/246b5273986d5a5b140b3d1a656baa8d40c36276
> Cr-Commit-Position: refs/heads/master@{#12042}

TBR=nisse@webrtc.org,glaznev@webrtc.org,deadbeef@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1819733002

Cr-Commit-Position: refs/heads/master@{#12065}
2016-03-20 16:38:44 +00:00
deadbeef
246b527398 Revert of Delete empty API files and cleaned up includes. (patchset #2 id:20001 of https://codereview.webrtc.org/1809053002/ )
Reason for revert:
Breaks Chromium build. Need to remove the references to the obsolete header files from Chromium and reland.

Original issue's description:
> Delete empty API files and cleaned up includes.
>
> TBR=glaznev@webrtc.org
>
> BUG=webrtc:5426
>
> Committed: https://crrev.com/c9022f508644dc33c01b05cb22ebfc2be145d6b2
> Cr-Commit-Position: refs/heads/master@{#12039}

TBR=nisse@webrtc.org,glaznev@webrtc.org,perkj@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/1813083002

Cr-Commit-Position: refs/heads/master@{#12042}
2016-03-17 22:03:46 +00:00
perkj
c9022f5086 Delete empty API files and cleaned up includes.
TBR=glaznev@webrtc.org

BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1809053002

Cr-Commit-Position: refs/heads/master@{#12039}
2016-03-17 16:57:30 +00:00
kwiberg
84cc9189fb Replace scoped_ptr with unique_ptr in talk/
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1785353003

Cr-Commit-Position: refs/heads/master@{#11968}
2016-03-12 06:13:01 +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
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
Zeke Chin
71f6f4405c iOS HW H264 support.
First step towards supporting H264 on iOS. More tuning/experimentation
required in future CLs. Tested using AppRTCDemo on iPhone6 + iPad Mini.
Future work to get it working on OS/X, simulator (renders black screen
currently) and with the Android AppRTCDemo. Currently protected with a
compile time guard.

BUG=4081
R=andrew@webrtc.org, haysc@webrtc.org, holmer@google.com, jiayl@webrtc.org, kjellander@webrtc.org, pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1187573004.

Cr-Commit-Position: refs/heads/master@{#9515}
2015-06-29 21:35:08 +00:00
Zeke Chin
bc7dd7e023 Add RTCConfiguration constructor to RTCPeerConnection wrapper.
BUG=4658
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/56419004

Cr-Commit-Position: refs/heads/master@{#9335}
2015-05-29 21:59:23 +00:00
Zeke Chin
57cc74e32c iOS camera switching video capturer.
Introduces a new capture class derived from cricket::VideoCapturer that
provides the ability to switch cameras and updates AppRTCDemo to use it.
Some future work pending to clean up AppRTCDemo UI.

BUG=4070
R=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48279005

Cr-Commit-Position: refs/heads/master@{#9137}
2015-05-05 14:52:45 +00:00
jlmiller@webrtc.org
5f93d0a140 Update libjingle license statements at top of talk files for consistency
BUG=2133
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8105 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-20 21:36:13 +00:00
tkchin@webrtc.org
ef2a5dd398 Update AppRTCDemo UI.
- Removed log box. Debug logs still available through lldb.
- Remote video displayed in aspect fill format.
- Provide a hangup button.
- Added Default-568.png so we display properly on iPhone5+.

BUG=
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8081 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 22:38:21 +00:00
tkchin@webrtc.org
4e5115ae73 RTCPeerConnectionFactory: Explicitly create new worker and signaling threads.
There should be no change in behavior, since this is what the default
constructor does.

BUG=
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8007 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-07 06:35:18 +00:00
buildbot@webrtc.org
d4e598d57a (Auto)update libjingle 72097588-> 72159069
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-29 17:36:52 +00:00
tkchin@webrtc.org
acca675bcf Implement mac version of AppRTCDemo.
- Refactored and moved AppRTCDemo to support sharing AppRTC connection code between iOS and mac counterparts.
- Refactored OpenGL rendering code to be shared between iOS and mac counterparts.
- iOS AppRTCDemo now respects video aspect ratio.

BUG=2168
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6291 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-30 22:26:06 +00:00
mallinath@webrtc.org
a0d3067575 Use CreatePeerConnection method which accepts port_allocator.
Other method will be removed, in a different CL.

R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20369006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5987 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-26 00:00:15 +00:00
fischman@webrtc.org
7fa1fcb72c AppRTCDemo(ios): style/cleanup fixes following cr/62871616-p10
BUG=2168
R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/9709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5768 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-25 00:11:56 +00:00
fischman@webrtc.org
c693a2a624 PeerConnection(iOS): fix case in #import statements.
We've been skating by on OS/X's default case-insensitive filesystem, but this
is a bit silly.

This change brought to you by:
sed -i '' 's/\+internal\.h/+Internal.h/g' $(git grep -l '+internal.h')

BUG=3088
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5764 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-24 18:56:37 +00:00
henrike@webrtc.org
d3d6bce9ed (Auto)update libjingle 62865357-> 62871616
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5674 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-10 20:41:22 +00:00
fischman@webrtc.org
9ca93a8b8e Explicitly @synthesize ObjC @properties
This is required after https://code.google.com/p/gyp/source/detail?r=1768
turned on -Wobjc-missing-property-synthesis for ninja builds (until then it
was only enabled for xcode builds) to allow chromium_deps to roll in
webrtc/DEPS.

BUG=2560
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5047 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-29 00:14:15 +00:00
fischman@webrtc.org
1bc1954174 AppRTCDemo: builds using ninja on iOS for simulator and device!
Things included in this CL:
- updated READMEs to provide an exact/reproable set of steps for getting the app
  running.
- gyp changes to build the iOS AppRTCDemo sample app using gyp+ninja instead of
  the hand-crafted Xcode project (which has never worked in its checked-in
  form), including a gyp action to sign the sample app for deployment to an iOS
  device (the app can also be used in the simulator)
- deleted the busted hand-crafted Xcode project for the sample app
- updated the sample app to match the PeerConnection API that ended up landing
  (in a surprising twist of fate, the API landed quite a bit later than the
  sample app and this is the first time the CR-time changes in the API are
  reflected in the sample app)
- updated the sample app to reflect apprtc.appspot.com HTML/JS changes (equiv to
  the AppRTCClient.java changes in http://s10/47299162)
- picked up the iossim DEPS to enable launching the sample app in the simulator
  from the command-line.
- renamed some files to match capitalization of the classes they contain (Ice ->
  ICE) per ObjC naming guidelines.
- ran the files involved in this CL through clang-format to deal with xcode
  formatting craxy.

BUG=2106
RISK=P2
TESTED=unittest builds with ninja and passes on OS=mac; sample app builds with ninja and runs on simulator and device, though no audio flows from simulator/device (will fix in a follow-up CL)
R=andrew@webrtc.org, justincohen@google.com, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1874005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4466 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-01 18:29:45 +00:00
henrike@webrtc.org
723d683ecb Update talk folder to revision=49260075. Same as 369 in libjingle's google code repository.
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1797004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4338 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-12 16:04:50 +00:00
henrike@webrtc.org
28e2075280 Adds trunk/talk folder of revision 359 from libjingles google code to
trunk/talk


git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 00:45:36 +00:00