Implement VideoSendStream::SetCodec().
Removing assertion that SSRC count should be the same as the number of
streams in the codec. It makes sense that you don't always use the same
number of streams under one call. Dropping resolution due to CPU overuse
for instance can require less streams, but the SSRCs should stay
allocated so that operations can resume when not overusing any more.
This change also means we can get rid of the ugly SendStreamState whose
content wasn't defined. Instead we use SetCodec to change resolution
etc. on the fly. Should something else have to be replaced on the fly
then that functionality simply has to be implemented.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3499005
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5123 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/call.h b/call.h
index 5239f5a..5f4c7b6 100644
--- a/call.h
+++ b/call.h
@@ -66,7 +66,7 @@
// Returns the internal state of the send stream, for resume sending with a
// new stream with different settings.
// Note: Only the last returned send-stream state is valid.
- virtual SendStreamState* DestroySendStream(VideoSendStream* send_stream) = 0;
+ virtual void DestroySendStream(VideoSendStream* send_stream) = 0;
virtual VideoReceiveStream::Config GetDefaultReceiveConfig() = 0;