Commit Graph

47 Commits

Author SHA1 Message Date
kwiberg@webrtc.org
eebcab5ce9 rtc::Buffer: Rename length to size, for conformance with the STL
And add a constructor for creating an uninitialized Buffer of a
specified size.

(I intend to follow up with more Buffer changes, but since it's rather
widely used, the rename is quite noisy and works better as a separate
CL.)

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48579004

Cr-Commit-Position: refs/heads/master@{#8841}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8841 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-24 09:20:19 +00:00
guoweis@webrtc.org
00c509ad1c Add concept of whether video renderer supports rotation.
Rotation is best done when rendered in GPU, added the shader code which rotates the frame. For renderers which don't support rotation, the rotation will be done before sending down the frame to render. By default, assume renderer can't do rotation.

Tested with peerconnection_client on windows, AppRTCDemo on Mac.

BUG=4145
R=glaznev@webrtc.org, pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8660

Committed: https://code.google.com/p/webrtc/source/detail?r=8661

Review URL: https://webrtc-codereview.appspot.com/43569004

Cr-Commit-Position: refs/heads/master@{#8705}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8705 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-12 21:38:19 +00:00
tommi@webrtc.org
d3900296ae Use a variant for storing stats values in StatsCollector code.
This cuts down on the amount of string copying we currently do and paves the way for separating the code that fetches the stats from the code that populates the stats reports.  As is, that code is intertwined, so we populate the stats on both signaling and worker thread.

I'm also adding some documentation and TODOs for further improvements.

BUG=2822
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47459004

Cr-Commit-Position: refs/heads/master@{#8700}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8700 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-12 16:36:15 +00:00
magjed@webrtc.org
c2008a0e8c RTCOpenGLVideoRenderer: Add support for padded frames
This CL allows RTCOpenGLVideoRenderer to handle frames with pitch > width by making an intermediate frame copy.

BUG=4381,1128
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46509004

Cr-Commit-Position: refs/heads/master@{#8637}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8637 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-06 16:59:23 +00:00
tommi@webrtc.org
92f4018d80 Start using std::map for Values in the statscollector. This is in preparaton for more work which will cut down on the string copying work we do.
Rename "AddValue" methods to AddXxx where Xxx is the type being added. Moving forward, we'll support those types natively without conversion to string.

Normalizing the extraction code to have fewer places that add the same stats and data driven additions to reports instead of multiple call sites.

BUG=2822
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47369004

Cr-Commit-Position: refs/heads/master@{#8597}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8597 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 15:25:44 +00:00
kjellander@webrtc.org
14665ff7d4 Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
Clang version changed 223108:230914
Details: e144d30..6fdb142/tools/clang/scripts/update.sh

Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h

The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h"  -o -name "*.cc*" -o -name "*.mm*"`

which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override

Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h

Remaining uses of OVERRIDE was fixed by search+replace.

Manual edits were done to fix virtual destructors that were
overriding inherited ones.

Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc

This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.

BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41069004

Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 13:04:54 +00:00
magjed@webrtc.org
f09e7b8a4f WebRtcVideoFrame: DCHECK exclusive ownership for non-const pixel access
Add some const safety by DCHECK(HasOneRef()) in non-const GetYPlane. This CL also replaces all incorrect non-const calls with const calls for pixel data access in cricket::VideoFrame. It's easy to call the non-const version of e.g. GetYPlane by mistake, even if only const-access is needed. For example:
const scoped_ptr<cricket::VideoFrame> foo;
const uint8_t* y = foo->GetYPlane();
will actually call the non-const version of GetYPlane.

R=mflodman@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39079004

Cr-Commit-Position: refs/heads/master@{#8507}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8507 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-25 14:50:19 +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
tommi@webrtc.org
8e327c45d0 Update StatsCollector's interface in preparation of more changes.
This CL is the first of three and this one contains interface additions (not deletion for backwards compatibility) as well as a few necessary updates to internal code.

The next CL will be in Chromium to consume the new new methods and remove dependency on the old ones.

The third CL will then contain the bulk of the updates and improvements and be compatible with this interface.

BUG=2822
R=perkj@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8095

Review URL: https://webrtc-codereview.appspot.com/36829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8097 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-19 20:41:26 +00:00
tommi@webrtc.org
43e54e36bf Revert 8095 "Update StatsCollector's interface in preparation of..."
> Update StatsCollector's interface in preparation of more changes.
> 
> This CL is the first of three and this one contains interface additions (not deletion for backwards compatibility) as well as a few necessary updates to internal code.
> 
> The next CL will be in Chromium to consume the new new methods and remove dependency on the old ones.
> 
> The third CL will then contain the bulk of the updates and improvements and be compatible with this interface.
> 
> BUG=2822
> R=perkj@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/36829004

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8096 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-19 17:34:23 +00:00
tommi@webrtc.org
5b76fd79df Update StatsCollector's interface in preparation of more changes.
This CL is the first of three and this one contains interface additions (not deletion for backwards compatibility) as well as a few necessary updates to internal code.

The next CL will be in Chromium to consume the new new methods and remove dependency on the old ones.

The third CL will then contain the bulk of the updates and improvements and be compatible with this interface.

BUG=2822
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8095 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-19 16:49:33 +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
tkchin@webrtc.org
7ce4a584aa Add initWithCoder to RTCEAGLVideoView.
Allows for proper OpenGL initialization if view is created from
storyboard.

BUG=3896
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7970 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-19 20:47:35 +00:00
tommi@webrtc.org
e2e199b894 Clean up StatsObserver's OnComplete methods (address TODOs).
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7898 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 13:22:54 +00:00
tkchin@webrtc.org
ee9d61ce45 This fixes a small memory leak (found using Xcode/Instruments on iOS) in
the ObjC bindings of PeerConnection. The generated session description has
to be released by the recipient

BUG=3985
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28959004

Patch from Matthias Liebig <matthias.gcode@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7636 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 22:01:53 +00:00
tkchin@webrtc.org
8125744a5f Cleanup RTCVideoRenderer interface.
RTCVideoRenderer should be a protocol not a class. This change includes
an adapter for use with the C++ apis. The video views have been refactored
to implement that protocol.

BUG=3795
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7626 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 23:06:15 +00:00
perkj@webrtc.org
c2dd5ee2c0 Prepare for removal of PeerConnectionObserver::OnError.
Prepare for removal of constraints to PeerConnection::AddStream.

OnError has never been implemented and has been removed from the spec.
Also, constraints to PeerConnection::AddStream has also been removed from the spec and have never been implemented.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7605 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 11:31:29 +00:00
tkchin@webrtc.org
90750482fa Remove deprecated RTCVideoRenderer constructor.
Removes -[RTCVideoRenderer initWithView]. Also, fix potential issue where we hold on to a video frame longer than the lifetime of its associated track.

BUG=3341
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7032 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02 20:50:00 +00:00
buildbot@webrtc.org
a09a99950e (Auto)update libjingle 73222930-> 73226398
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6891 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-13 17:26:08 +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
tommi@webrtc.org
242068d58c A step towards changing StatsReport::Value::name to an enum.
The stats reporting code does a lot of unnecessary string copying.
This is a step in the direction of removing that and forcing use of only known constants.

This is a reland of an already reviewed cl that got reverted by mistake.

TBR=xians@google.com

Review URL: https://webrtc-codereview.appspot.com/12989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6678 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-14 20:19:56 +00:00
tkchin@webrtc.org
738df8913d Fix retain cycle in RTCEAGLVideoView.
CADisplayLink increases its target's refcount. In order to break retain cycle, we wrap CADisplayLink in a new RTCDisplayLinkTimer class and use that instead.

R=fischman@webrtc.org, noahric@chromium.org
BUG=3391

Review URL: https://webrtc-codereview.appspot.com/16599006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6331 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-04 20:19:39 +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
tkchin@webrtc.org
1732a591e7 Add a UIView for rendering a video track.
RTCEAGLVideoView provides functionality to render a supplied RTCVideoTrack using OpenGLES2.

R=fischman@webrtc.org
BUG=3188

Review URL: https://webrtc-codereview.appspot.com/12489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6192 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-19 23:26:01 +00:00
tkchin@webrtc.org
ff2733204d Implement ObjC DataChannel wrapper
R=fischman@webrtc.org
BUG=3112

Review URL: https://webrtc-codereview.appspot.com/16369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6031 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-30 18:32:33 +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
tkchin@webrtc.org
19b1be159e Provide GetStats method in RTCPeerConnection
BUG=3144
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12069006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5960 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 21:05:38 +00:00
tkchin@webrtc.org
ec3d8ecdcc Fix typo by renaming RTCSessionDescriptonDelegate -> RTCSessionsDescriptionDelegate
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5946 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-21 18:47:24 +00:00
fischman@webrtc.org
49c5ba32bb AppRTCDemo(iOS): now works in the iOS Simulator!
...which has no camera device emulation or pass-through, so no local video
view.

R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/10919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5815 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-31 20:22:19 +00:00
fischman@webrtc.org
76d4f389bb AppRTCDemo(iOS): allow rooms with no incoming audio.
Also fix a compile-time warning for a leftover unimplemented method
(RTCVideoRenderer:setTransform).

R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/10629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5780 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-25 17:40:38 +00:00
fischman@webrtc.org
385a722646 PeerConnection(iOS): make ARC-clean talk/.../objc* and talk/examples/ios
- Removes a strong-reference cycle between RTCPeerConnection and
  RTCPeerConnectionObserver
- Gives RTCPeerConnectionObserver a virtual dtor
- Ensures RTCPeerConnectionTest tears down correctly
- Ensures AppRTCDemo tears down correctly

This is the talk/ half; the webrtc/ half is in https://webrtc-codereview.appspot.com/10539005

BUG=3054,3055,3100
R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/10499005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5771 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-25 05:16:29 +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
mallinath@webrtc.org
0d92ef67c4 Libjingle source code has some spelling mistakes and one of them is "renegotation", which should be "renegotiation".
This CL is attempting to correct those.

BUG=2810
TBR=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5411 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-22 02:21:22 +00:00
fischman@webrtc.org
af320fd2f7 The designated initializer method declaration in the Objective-C headers for RTCICEServer does't match its implementation.
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6019004

Patch from Rafael Lopez Diez <rafalopezdiez@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5309 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-17 21:33:27 +00:00
fischman@webrtc.org
5b3c67ef25 objc/README: Remove outdated advice about target_os.
BUG=chromium:248168
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5979005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5302 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 17:15:19 +00:00
fischman@webrtc.org
e0034557a7 RTCPeerConnection(objc): avoid leaking ICE candidate on addition.
BUG=2670
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5199 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-02 18:49:54 +00:00
fischman@webrtc.org
7b273a545d PeerConnection iOS: update README instructions
This is needed to account for https://codereview.chromium.org/25535004/

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5079 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-04 18:48:12 +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
c31d4d0324 AppRTCDemo(iOS): prefer ISAC as audio codec
This makes audio flow well bidirectionally to an iPod Touch (5th gen).
Also:
- Update to new turnserver JSON style:
  - separate username field
  - multiple URLs for the same server (e.g. both UDP & TCP)
- Added more explicit logging for ICE Connected since it's useful for debugging
- Give focus to the input field on app launch since that's the only useful
  thing to have focus on, anyway.
- Fix minor typos
- Cleaned up trailing whitespace and hard tabs

BUG=2191
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2127004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4687 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 21:49:58 +00:00
fischman@webrtc.org
825e9b0a9b talk/objc/README: s/libjingle/webrtc/ in repository path.
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1985004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4501 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-07 16:52:03 +00:00
fischman@webrtc.org
86d7a198ec ObjC PeerConnection README: note workaround needed for crbug.com/248168
BUG=2106
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1940004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4467 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-01 19:27:54 +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