[asterisk-commits] phsultan: branch phsultan/rtmp-support r280056 - /team/phsultan/rtmp-support/...

[SVN commits to the Asterisk project]

Author: phsultan
Date: Wed Jul 28 04:24:26 2010
New Revision: 280056

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=280056
Log:
Set the cutoff frequency to half the output sampling rate.

Let's try this way to prevent audio aliasing, and get a better audio output
when downsampling.

Modified:
    team/phsultan/rtmp-support/apps/app_ffplayback.c

Modified: team/phsultan/rtmp-support/apps/app_ffplayback.c
URL: http://svnview.digium.com/svn/asterisk/team/phsultan/rtmp-support/apps/app_ffplayback.c?view=diff&rev=280056&r1=280055&r2=280056
==============================================================================
--- team/phsultan/rtmp-support/apps/app_ffplayback.c (original)
+++ team/phsultan/rtmp-support/apps/app_ffplayback.c Wed Jul 28 04:24:26 2010
@@ -438,7 +438,7 @@
 			1, in_acctx->channels,
 			16000, in_acctx->sample_rate,
 			SAMPLE_FMT_S16, in_acctx->sample_fmt,
-			16, 10, 1, 0.8); 
+			16, 10, 1, 1.0); 
 
 	/* Init audio frame */
 	trans_pvt = ast_translator_build_path(chan->writeformat, AST_FORMAT_SLINEAR16);