From f1f090f6cb54d86aee2f951601758f426c956573 Mon Sep 17 00:00:00 2001 From: Dan Dennedy Date: Tue, 8 Dec 2009 19:55:15 -0800 Subject: [PATCH] Fix underlinking libm by removing math function. --- configure | 2 +- src/framework/mlt.h | 4 ++-- src/modules/avformat/producer_avformat.c | 2 +- 3 files changed, 4 insertions(+), 4 deletions(-) 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 ] ) -- 2.39.2