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AppRTCDemo(iOS): allow rooms with no incoming audio.

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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
This commit is contained in:
fischman@webrtc.org
2014-03-25 17:40:38 +00:00
parent 6e3dbc2a77
commit 76d4f389bb
2 changed files with 2 additions and 5 deletions
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3
talk/app/webrtc/objc/public/RTCVideoRenderer.h
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@ -43,9 +43,6 @@ struct CGRect;
// of frames. // of frames.
- (id)initWithDelegate:(id<RTCVideoRendererDelegate>)delegate; - (id)initWithDelegate:(id<RTCVideoRendererDelegate>)delegate;
// Set an affine transform on relevant UIViews.
- (void)setTransform:(CGAffineTransform)transform;
// Starts rendering. // Starts rendering.
- (void)start; - (void)start;
// Stops rendering. It can be restarted again using the 'start' method above. // Stops rendering. It can be restarted again using the 'start' method above.

4
talk/examples/ios/AppRTCDemo/APPRTCAppDelegate.m
View File

@ -77,8 +77,8 @@
addedStream:(RTCMediaStream*)stream { addedStream:(RTCMediaStream*)stream {
NSLog(@"PCO onAddStream."); NSLog(@"PCO onAddStream.");
dispatch_async(dispatch_get_main_queue(), ^(void) { dispatch_async(dispatch_get_main_queue(), ^(void) {
NSAssert([stream.audioTracks count] >= 1, NSAssert([stream.audioTracks count] <= 1,
@"Expected at least 1 audio stream"); @"Expected at most 1 audio stream");
NSAssert([stream.videoTracks count] <= 1, NSAssert([stream.videoTracks count] <= 1,
@"Expected at most 1 video stream"); @"Expected at most 1 video stream");
if ([stream.videoTracks count] != 0) { if ([stream.videoTracks count] != 0) {