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
diff --git a/talk/app/webrtc/objc/public/RTCVideoRenderer.h b/talk/app/webrtc/objc/public/RTCVideoRenderer.h
index 8a0b74c..d6b6a65 100644
--- a/talk/app/webrtc/objc/public/RTCVideoRenderer.h
+++ b/talk/app/webrtc/objc/public/RTCVideoRenderer.h
@@ -43,9 +43,6 @@
// of frames.
- (id)initWithDelegate:(id<RTCVideoRendererDelegate>)delegate;
-// Set an affine transform on relevant UIViews.
-- (void)setTransform:(CGAffineTransform)transform;
-
// Starts rendering.
- (void)start;
// Stops rendering. It can be restarted again using the 'start' method above.
diff --git a/talk/examples/ios/AppRTCDemo/APPRTCAppDelegate.m b/talk/examples/ios/AppRTCDemo/APPRTCAppDelegate.m
index 8610477..8400778 100644
--- a/talk/examples/ios/AppRTCDemo/APPRTCAppDelegate.m
+++ b/talk/examples/ios/AppRTCDemo/APPRTCAppDelegate.m
@@ -77,8 +77,8 @@
addedStream:(RTCMediaStream*)stream {
NSLog(@"PCO onAddStream.");
dispatch_async(dispatch_get_main_queue(), ^(void) {
- NSAssert([stream.audioTracks count] >= 1,
- @"Expected at least 1 audio stream");
+ NSAssert([stream.audioTracks count] <= 1,
+ @"Expected at most 1 audio stream");
NSAssert([stream.videoTracks count] <= 1,
@"Expected at most 1 video stream");
if ([stream.videoTracks count] != 0) {
