From: Dan Dennedy <dan@dennedy.org>
Date: Wed, 9 Dec 2009 03:55:15 +0000 (-0800)
Subject: Fix underlinking libm by removing math function.
X-Git-Url: https://git.sesse.net/?a=commitdiff_plain;h=f1f090f6cb54d86aee2f951601758f426c956573;p=mlt

Fix underlinking libm by removing math function.
---

diff --git a/configure b/configure
index ff9d2e2d..d957a8db 100755
--- a/configure
+++ b/configure
@@ -1,6 +1,6 @@
 #!/bin/sh
 
-export version=0.4.8
+export version=0.4.9
 export soversion=2
 
 show_help()
diff --git a/src/framework/mlt.h b/src/framework/mlt.h
index 3711d2ed..0b211d73 100644
--- a/src/framework/mlt.h
+++ b/src/framework/mlt.h
@@ -23,8 +23,8 @@
 #ifndef _MLT_H_
 #define _MLT_H_
 
-#define LIBMLT_VERSION_INT ((0<<16)+(4<<8)+8)
-#define LIBMLT_VERSION     0.4.8
+#define LIBMLT_VERSION_INT ((0<<16)+(4<<8)+9)
+#define LIBMLT_VERSION     0.4.9
 
 #ifdef __cplusplus
 extern "C"
diff --git a/src/modules/avformat/producer_avformat.c b/src/modules/avformat/producer_avformat.c
index 2d0be647..78b39060 100644
--- a/src/modules/avformat/producer_avformat.c
+++ b/src/modules/avformat/producer_avformat.c
@@ -1422,7 +1422,7 @@ static int decode_audio( producer_avformat this, int *ignore, AVPacket *pkt, int
 		{
 			// Figure out how many samples will be needed after resampling
 			int convert_samples = data_size / codec_context->channels / ( av_get_bits_per_sample_format( codec_context->sample_fmt ) / 8 );
-			int samples_needed = lrint( this->resample_factor * convert_samples );
+			int samples_needed = this->resample_factor * convert_samples + 1;
 			
 			// Resize audio buffer to prevent overflow
 			if ( audio_used * channels + samples_needed > this->audio_buffer_size[ index ] )