commit 76d4f389bb09609bf0b52323ebe71e6a8653f341
author fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Tue Mar 25 17:40:38 2014 +0000
committer fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Tue Mar 25 17:40:38 2014 +0000
tree 32ee9a000a5a7f3e1888f03d210e270681085be8
parent 6e3dbc2a77eb96b050c4909c4206348f1b15550c

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

talk/examples/ios/AppRTCDemo/APPRTCAppDelegate.m

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) {