From 3021e900b5c60c059218e8e0400a77641f98929c Mon Sep 17 00:00:00 2001 From: Brian Matherly Date: Wed, 14 Mar 2018 21:30:10 -0500 Subject: [PATCH] Fix for loop initial declarations. Conform to C89 mode loop declarations. --- src/modules/avformat/filter_swresample.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/src/modules/avformat/filter_swresample.c b/src/modules/avformat/filter_swresample.c index d479857d1..177b38569 100644 --- a/src/modules/avformat/filter_swresample.c +++ b/src/modules/avformat/filter_swresample.c @@ -76,7 +76,8 @@ static void audio_format_planes( mlt_audio_format format, int samples, int chann { int plane_count = audio_plane_count( format, channels ); size_t plane_size = audio_plane_size( format, samples, channels ); - for( int p = 0; p < plane_count; p++ ) + int p = 0; + for( p = 0; p < plane_count; p++ ) { planes[p] = buffer + ( p * plane_size ); } @@ -89,7 +90,8 @@ static void collapse_channels( mlt_audio_format format, int channels, int alloca { size_t src_plane_size = audio_plane_size( format, allocated_samples, channels ); size_t dst_plane_size = audio_plane_size( format, used_samples, channels ); - for( int p = 0; p < plane_count; p++ ) + int p = 0; + for( p = 0; p < plane_count; p++ ) { uint8_t* src = buffer + ( p * src_plane_size ); uint8_t* dst = buffer + ( p * dst_plane_size ); @@ -138,12 +140,13 @@ static int configure_swr_context( mlt_filter filter ) int64_t custom_out_layout = 0; double* matrix = av_mallocz_array( pdata->in_channels * pdata->out_channels, sizeof(double) ); int stride = pdata->in_channels; + int i = 0; - for( int i = 0; i < pdata->in_channels; i++ ) + for( i = 0; i < pdata->in_channels; i++ ) { custom_in_layout = (custom_in_layout << 1) | 0x01; } - for( int i = 0; i < pdata->out_channels; i++ ) + for( i = 0; i < pdata->out_channels; i++ ) { custom_out_layout = (custom_out_layout << 1) | 0x01; if( i <= pdata->in_channels ) @@ -241,6 +244,7 @@ static int filter_get_audio( mlt_frame frame, void **buffer, mlt_audio_format *f if( !error ) { int in_samples = *samples; + int out_samples = 0; int alloc_samples = in_samples; if( in_frequency != out_frequency ) { @@ -255,7 +259,7 @@ static int filter_get_audio( mlt_frame frame, void **buffer, mlt_audio_format *f audio_format_planes( in_format, in_samples, in_channels, *buffer, pdata->in_buffers ); audio_format_planes( out_format, alloc_samples, out_channels, out_buffer, pdata->out_buffers ); - int out_samples = swr_convert( pdata->ctx, pdata->out_buffers, alloc_samples, (const uint8_t**)pdata->in_buffers, in_samples ); + out_samples = swr_convert( pdata->ctx, pdata->out_buffers, alloc_samples, (const uint8_t**)pdata->in_buffers, in_samples ); if( out_samples > 0 ) { collapse_channels( out_format, out_channels, alloc_samples, out_samples, out_buffer );