Commit 18ea923e authored by Rémi Denis-Courmont's avatar Rémi Denis-Courmont

a52: cleanup and fix buffer size

This closes a heap overflow on corrupt files.
Pointed-out-by: default avatarClément Lecigne <clemun@gmail.com>
(cherry picked from commit a6e127f134e42f5d1d5ea9d9b8d055ce42e8caf8)
parent 5e0c3e5a
......@@ -271,10 +271,10 @@ static void Exchange( sample_t * p_out, const sample_t * p_in )
}
/*****************************************************************************
* DoWork: decode an ATSC A/52 frame.
* Convert: decode an ATSC A/52 frame.
*****************************************************************************/
static void DoWork( filter_t * p_filter,
aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
static block_t *Convert( filter_t *p_filter, block_t *p_in_buf )
{
filter_sys_t *p_sys = p_filter->p_sys;
#ifdef LIBA52_FIXED
......@@ -283,9 +283,11 @@ static void DoWork( filter_t * p_filter,
sample_t i_sample_level = 1;
#endif
int i_flags = p_sys->i_flags;
int i_bytes_per_block = 256 * p_sys->i_nb_channels
* sizeof(sample_t);
int i;
size_t i_bytes_per_block = 256 * p_sys->i_nb_channels * sizeof(sample_t);
block_t *p_out_buf = filter_NewAudioBuffer( p_filter, 6 * i_bytes_per_block );
if( unlikely(p_out_buf == NULL) )
goto out;
/* Do the actual decoding now. */
a52_frame( p_sys->p_liba52, p_in_buf->p_buffer,
......@@ -307,7 +309,7 @@ static void DoWork( filter_t * p_filter,
a52_dynrng( p_sys->p_liba52, NULL, NULL );
}
for ( i = 0; i < 6; i++ )
for( unsigned i = 0; i < 6; i++ )
{
sample_t * p_samples;
......@@ -342,7 +344,12 @@ static void DoWork( filter_t * p_filter,
}
p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
p_out_buf->i_buffer = i_bytes_per_block * 6;
p_out_buf->i_dts = p_in_buf->i_dts;
p_out_buf->i_pts = p_in_buf->i_pts;
p_out_buf->i_length = p_in_buf->i_length;
out:
block_Release( p_in_buf );
return p_out_buf;
}
/*****************************************************************************
......@@ -388,36 +395,3 @@ static void CloseFilter( vlc_object_t *p_this )
a52_free( p_sys->p_liba52 );
free( p_sys );
}
static block_t *Convert( filter_t *p_filter, block_t *p_block )
{
if( !p_block || !p_block->i_nb_samples )
{
if( p_block )
block_Release( p_block );
return NULL;
}
size_t i_out_size = p_block->i_nb_samples *
p_filter->fmt_out.audio.i_bitspersample *
p_filter->fmt_out.audio.i_channels / 8;
block_t *p_out = filter_NewAudioBuffer( p_filter, i_out_size );
if( !p_out )
{
msg_Warn( p_filter, "can't get output buffer" );
block_Release( p_block );
return NULL;
}
p_out->i_nb_samples = p_block->i_nb_samples;
p_out->i_dts = p_block->i_dts;
p_out->i_pts = p_block->i_pts;
p_out->i_length = p_block->i_length;
DoWork( p_filter, p_block, p_out );
block_Release( p_block );
return p_out;
}
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