[vlc-commits] commit: Fixed bandlimited audio filter with low output samplerate. ( Laurent Aimar )
git at videolan.org
git at videolan.org
Sun May 2 02:26:14 CEST 2010
vlc | branch: master | Laurent Aimar <fenrir at videolan.org> | Sun May 2 02:13:03 2010 +0200| [f9f6f88c53d72b8b1cf2b05a37e5ced786a2fbbf] | committer: Laurent Aimar
Fixed bandlimited audio filter with low output samplerate.
It closes #3543.
> http://git.videolan.org/gitweb.cgi/vlc.git/?a=commit;h=f9f6f88c53d72b8b1cf2b05a37e5ced786a2fbbf
---
modules/audio_filter/resampler/bandlimited.c | 70 ++++++++++++++++++--------
1 files changed, 48 insertions(+), 22 deletions(-)
diff --git a/modules/audio_filter/resampler/bandlimited.c b/modules/audio_filter/resampler/bandlimited.c
index 80ef03b..91a2085 100644
--- a/modules/audio_filter/resampler/bandlimited.c
+++ b/modules/audio_filter/resampler/bandlimited.c
@@ -57,7 +57,7 @@ static void CloseFilter( vlc_object_t * );
static block_t *Resample( filter_t *, block_t * );
static void ResampleFloat( filter_t *p_filter,
- block_t *p_out_buf, size_t *pi_out,
+ block_t **pp_out_buf, size_t *pi_out,
float **pp_in,
int i_in, int i_in_end,
double d_factor, bool b_factor_old,
@@ -147,7 +147,6 @@ static block_t *Resample( filter_t * p_filter, block_t * p_in_buf )
block_Release( p_in_buf );
return NULL;
}
- float *p_out = (float *)p_out_buf->p_buffer;
if( (p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY) || p_sys->b_first )
{
@@ -194,7 +193,7 @@ static block_t *Resample( filter_t * p_filter, block_t * p_in_buf )
const float *p_in_orig = p_in;
/* Make sure the output buffer is reset */
- memset( p_out, 0, p_out_buf->i_buffer );
+ memset( p_out_buf->p_buffer, 0, p_out_buf->i_buffer );
/* Calculate the new length of the filter wing */
d_factor = (double)i_out_rate / p_filter->fmt_in.audio.i_rate;
@@ -214,7 +213,7 @@ static block_t *Resample( filter_t * p_filter, block_t * p_in_buf )
i_old_in_end = __MIN( i_filter_wing, i_in_nb - p_sys->i_old_wing );
ResampleFloat( p_filter,
- p_out_buf, &i_out, &p_in,
+ &p_out_buf, &i_out, &p_in,
i_in, i_old_in_end,
p_sys->d_old_factor, true,
i_nb_channels, i_bytes_per_frame );
@@ -226,20 +225,23 @@ static block_t *Resample( filter_t * p_filter, block_t * p_in_buf )
p_sys->d_old_factor = d_factor;
p_sys->i_old_wing = i_filter_wing;
}
- ResampleFloat( p_filter,
- p_out_buf, &i_out, &p_in,
- i_in, i_in_nb - i_filter_wing,
- d_factor, false,
- i_nb_channels, i_bytes_per_frame );
-
- /* Finalize aout buffer */
- p_out_buf->i_nb_samples = i_out;
- p_out_buf->i_pts = date_Get( &p_sys->end_date );
- p_out_buf->i_length = date_Increment( &p_sys->end_date,
- p_out_buf->i_nb_samples ) - p_out_buf->i_pts;
-
- p_out_buf->i_buffer = p_out_buf->i_nb_samples *
- i_nb_channels * sizeof(int32_t);
+ if( p_out_buf )
+ {
+ ResampleFloat( p_filter,
+ &p_out_buf, &i_out, &p_in,
+ i_in, i_in_nb - i_filter_wing,
+ d_factor, false,
+ i_nb_channels, i_bytes_per_frame );
+
+ /* Finalize aout buffer */
+ p_out_buf->i_nb_samples = i_out;
+ p_out_buf->i_pts = date_Get( &p_sys->end_date );
+ p_out_buf->i_length = date_Increment( &p_sys->end_date,
+ p_out_buf->i_nb_samples ) - p_out_buf->i_pts;
+
+ p_out_buf->i_buffer = p_out_buf->i_nb_samples *
+ i_nb_channels * sizeof(int32_t);
+ }
/* Buffer i_filter_wing * 2 samples for next time */
if( p_sys->i_old_wing )
@@ -435,8 +437,28 @@ static void FilterFloatUD( const float Imp[], const float ImpD[], uint16_t Nwing
}
}
+static int ReallocBuffer( block_t **pp_out_buf,
+ float **pp_out, size_t i_out,
+ int i_nb_channels, int i_bytes_per_frame )
+{
+ if( i_out < (*pp_out_buf)->i_buffer/i_bytes_per_frame )
+ return VLC_SUCCESS;
+
+ /* It may happen when the wing size changes */
+ const unsigned i_extra_frame = 256;
+ *pp_out_buf = block_Realloc( *pp_out_buf, 0,
+ (*pp_out_buf)->i_buffer +
+ i_extra_frame * i_bytes_per_frame );
+ if( !*pp_out_buf )
+ return VLC_EGENERIC;
+
+ *pp_out = (float*)(*pp_out_buf)->p_buffer + i_out * i_nb_channels;
+ memset( *pp_out, 0, i_extra_frame * i_bytes_per_frame );
+ return VLC_SUCCESS;
+}
+
static void ResampleFloat( filter_t *p_filter,
- block_t *p_out_buf, size_t *pi_out,
+ block_t **pp_out_buf, size_t *pi_out,
float **pp_in,
int i_in, int i_in_end,
double d_factor, bool b_factor_old,
@@ -446,12 +468,15 @@ static void ResampleFloat( filter_t *p_filter,
float *p_in = *pp_in;
size_t i_out = *pi_out;
- float *p_out = (float*)p_out_buf->p_buffer + i_out * i_nb_channels;
+ float *p_out = (float*)(*pp_out_buf)->p_buffer + i_out * i_nb_channels;
for( ; i_in < i_in_end; i_in++ )
{
if( b_factor_old && d_factor == 1 )
{
+ if( ReallocBuffer( pp_out_buf, &p_out,
+ i_out, i_nb_channels, i_bytes_per_frame ) )
+ return;
/* Just copy the samples */
memcpy( p_out, p_in, i_bytes_per_frame );
p_in += i_nb_channels;
@@ -462,8 +487,9 @@ static void ResampleFloat( filter_t *p_filter,
while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
{
- if( p_out_buf->i_buffer/i_bytes_per_frame <= i_out )
- break;
+ if( ReallocBuffer( pp_out_buf, &p_out,
+ i_out, i_nb_channels, i_bytes_per_frame ) )
+ return;
if( d_factor >= 1 )
{
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