Commit Graph

15 Commits

Author SHA1 Message Date
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
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
61e78fca6c AppRTCDemo(iOS): remote-video reliability fixes
Previously GAE Channel callbacks would be handled by JS string-encoding the
payload into a URL.  Unfortunately this is limited to the (undocumented,
silently problematic) maximum URL length UIWebView supports.  Replaced this
scheme by a notification from JS to ObjC and a getter from ObjC to JS (which
happens out-of-line to avoid worrying about UIWebView's re-entrancy, or lack
thereof).  Part of this change also moved from a combination of: JSON,
URL-escaping, and ad-hoc :-separated values to simply JSON.

Also incidentally:
- Removed outdated TODO about onRenegotiationNeeded, which is unneeded
- Move handling of PeerConnection callbacks to the main queue to avoid having
  to think about concurrency too hard.
- Replaced a bunch of NSOrderedSame with isEqualToString for clearer code and
  not having to worry about the fact that [nil compare:@"foo"]==NSOrderedSame
  is always true (yay ObjC!).
- Auto-scroll messages view.

BUG=3117
R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/10899006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5814 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-31 20:16:49 +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
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
82387e4608 Add ability to receive calls for iOS
BUG=2701
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7989005

Patch from Sajid Hussain <shussain@temasys.com.sg>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5518 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 18:47:11 +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
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
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