#include <vlc_filter.h>
#include <vlc_block.h>
+#include <assert.h>
+
#include "bandlimited.h"
/*****************************************************************************
* Local prototypes
*****************************************************************************/
-/* audio filter2 */
+/* audio filter */
static int OpenFilter ( vlc_object_t * );
static void CloseFilter( vlc_object_t * );
static block_t *Resample( filter_t *, block_t * );
-
-static void FilterFloatUP( const float Imp[], const float ImpD[], uint16_t Nwing,
- float *f_in, float *f_out, uint32_t ui_remainder,
- uint32_t ui_output_rate, int16_t Inc,
- int i_nb_channels );
-
-static void FilterFloatUD( const float Imp[], const float ImpD[], uint16_t Nwing,
- float *f_in, float *f_out, uint32_t ui_remainder,
- uint32_t ui_output_rate, uint32_t ui_input_rate,
- int16_t Inc, int i_nb_channels );
+static void ResampleFloat( filter_t *p_filter,
+ 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,
+ int i_nb_channels, int i_bytes_per_frame );
/*****************************************************************************
* Local structures
struct filter_sys_t
{
int32_t *p_buf; /* this filter introduces a delay */
- int i_buf_size;
+ size_t i_buf_size;
double d_old_factor;
int i_old_wing;
set_category( CAT_AUDIO )
set_subcategory( SUBCAT_AUDIO_MISC )
set_description( N_("Audio filter for band-limited interpolation resampling") )
- set_capability( "audio filter2", 20 )
+ set_capability( "audio filter", 20 )
set_callbacks( OpenFilter, CloseFilter )
vlc_module_end ()
date_Increment( &p_sys->end_date,
p_in_buf->i_nb_samples ) - p_in_buf->i_pts;
}
- p_in_buf->i_flags |= BLOCK_FLAG_DISCONTINUITY;
p_sys->i_old_wing = 0;
+ p_sys->b_first = true;
return p_in_buf;
}
+ p_filter->p_sys->i_buf_size;
block_t *p_out_buf = filter_NewAudioBuffer( p_filter, i_out_size );
if( !p_out_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 )
{
p_sys->b_first = false;
}
- int i_in_nb = p_in_buf->i_nb_samples;
- int i_in, i_out = 0;
+ size_t i_in_nb = p_in_buf->i_nb_samples;
+ size_t i_in, i_out = 0;
double d_factor, d_scale_factor, d_old_scale_factor;
- int i_filter_wing;
+ size_t i_filter_wing;
#if 0
msg_Err( p_filter, "old rate: %i, old factor: %f, old wing: %i, i_in: %i",
p_sys->i_old_wing, i_in_nb );
#endif
- /* Prepare the source buffer */
- i_in_nb += (p_sys->i_old_wing * 2);
-
- float p_in_orig[i_in_nb * p_filter->fmt_in.audio.i_bytes_per_frame / 4],
- *p_in = p_in_orig;
+ /* Same format in and out... */
+ assert( p_filter->fmt_in.audio.i_bytes_per_frame == i_bytes_per_frame );
- /* Copy all our samples in p_in */
+ /* Prepare the source buffer */
if( p_sys->i_old_wing )
- {
- vlc_memcpy( p_in, p_sys->p_buf,
- p_sys->i_old_wing * 2 *
- p_filter->fmt_in.audio.i_bytes_per_frame );
+ { /* Copy all our samples in p_in_buf */
+ /* Normally, there should be enough room for the old wing in the
+ * buffer head room. Otherwise, we need to copy memory anyway. */
+ p_in_buf = block_Realloc( p_in_buf,
+ p_sys->i_old_wing * 2 * i_bytes_per_frame,
+ p_in_buf->i_buffer );
+ if( unlikely(p_in_buf == NULL) )
+ return NULL;
+ memcpy( p_in_buf->p_buffer, p_sys->p_buf,
+ p_sys->i_old_wing * 2 * i_bytes_per_frame );
}
- /* XXX: why i_nb_channels instead of i_bytes_per_frame??? */
- vlc_memcpy( p_in + p_sys->i_old_wing * 2 * i_nb_channels,
- p_in_buf->p_buffer,
- p_in_buf->i_nb_samples * p_filter->fmt_in.audio.i_bytes_per_frame );
- block_Release( p_in_buf );
+ i_in_nb += (p_sys->i_old_wing * 2);
+ float *p_in = (float *)p_in_buf->p_buffer;
+ 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;
/* Apply the old rate until we have enough samples for the new one */
i_in = p_sys->i_old_wing;
p_in += p_sys->i_old_wing * i_nb_channels;
- for( ; i_in < i_filter_wing &&
- (i_in + p_sys->i_old_wing) < i_in_nb; i_in++ )
- {
- if( p_sys->d_old_factor == 1 )
- {
- /* Just copy the samples */
- memcpy( p_out, p_in,
- p_filter->fmt_in.audio.i_bytes_per_frame );
- p_in += i_nb_channels;
- p_out += i_nb_channels;
- i_out++;
- continue;
- }
-
- while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
- {
-
- if( p_sys->d_old_factor >= 1 )
- {
- /* FilterFloatUP() is faster if we can use it */
-
- /* Perform left-wing inner product */
- FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
- SMALL_FILTER_NWING, p_in, p_out,
- p_sys->i_remainder,
- p_filter->fmt_out.audio.i_rate,
- -1, i_nb_channels );
- /* Perform right-wing inner product */
- FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
- SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
- p_filter->fmt_out.audio.i_rate -
- p_sys->i_remainder,
- p_filter->fmt_out.audio.i_rate,
- 1, i_nb_channels );
-
-#if 0
- /* Normalize for unity filter gain */
- for( i = 0; i < i_nb_channels; i++ )
- {
- *(p_out+i) *= d_old_scale_factor;
- }
-#endif
-
- /* Sanity check */
- if( p_out_buf->i_buffer/p_filter->fmt_in.audio.i_bytes_per_frame
- <= (unsigned int)i_out+1 )
- {
- p_out += i_nb_channels;
- i_out++;
- p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
- break;
- }
- }
- else
- {
- /* Perform left-wing inner product */
- FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
- SMALL_FILTER_NWING, p_in, p_out,
- p_sys->i_remainder,
- p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
- -1, i_nb_channels );
- /* Perform right-wing inner product */
- FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
- SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
- p_filter->fmt_out.audio.i_rate -
- p_sys->i_remainder,
- p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
- 1, i_nb_channels );
- }
- p_out += i_nb_channels;
- i_out++;
+ size_t i_old_in_end = 0;
+ if( p_sys->i_old_wing <= i_in_nb )
+ i_old_in_end = __MIN( i_filter_wing, i_in_nb - p_sys->i_old_wing );
- p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
- }
-
- p_in += i_nb_channels;
- p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
- }
+ ResampleFloat( p_filter,
+ &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 );
+ i_in = __MAX( i_in, i_old_in_end );
/* Apply the new rate for the rest of the samples */
if( i_in < i_in_nb - i_filter_wing )
p_sys->d_old_factor = d_factor;
p_sys->i_old_wing = i_filter_wing;
}
- for( ; i_in < i_in_nb - i_filter_wing; i_in++ )
+ if( p_out_buf )
{
- while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
- {
-
- if( d_factor >= 1 )
- {
- /* FilterFloatUP() is faster if we can use it */
-
- /* Perform left-wing inner product */
- FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
- SMALL_FILTER_NWING, p_in, p_out,
- p_sys->i_remainder,
- p_filter->fmt_out.audio.i_rate,
- -1, i_nb_channels );
-
- /* Perform right-wing inner product */
- FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
- SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
- p_filter->fmt_out.audio.i_rate -
- p_sys->i_remainder,
- p_filter->fmt_out.audio.i_rate,
- 1, i_nb_channels );
-
-#if 0
- /* Normalize for unity filter gain */
- for( int i = 0; i < i_nb_channels; i++ )
- {
- *(p_out+i) *= d_old_scale_factor;
- }
-#endif
- /* Sanity check */
- if( p_out_buf->i_buffer/p_filter->fmt_in.audio.i_bytes_per_frame
- <= (unsigned int)i_out+1 )
- {
- p_out += i_nb_channels;
- i_out++;
- p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
- break;
- }
- }
- else
- {
- /* Perform left-wing inner product */
- FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
- SMALL_FILTER_NWING, p_in, p_out,
- p_sys->i_remainder,
- p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
- -1, i_nb_channels );
- /* Perform right-wing inner product */
- FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
- SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
- p_filter->fmt_out.audio.i_rate -
- p_sys->i_remainder,
- p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
- 1, i_nb_channels );
- }
-
- p_out += i_nb_channels;
- i_out++;
-
- p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
- }
-
- p_in += i_nb_channels;
- p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
+ 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_dts =
+ 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 )
{
+ size_t newsize = p_sys->i_old_wing * 2 * i_bytes_per_frame;
+ if( newsize > p_sys->i_buf_size )
+ {
+ free( p_sys->p_buf );
+ p_sys->p_buf = malloc( newsize );
+ if( p_sys->p_buf != NULL )
+ p_sys->i_buf_size = newsize;
+ else
+ {
+ p_sys->i_buf_size = p_sys->i_old_wing = 0; /* oops! */
+ block_Release( p_in_buf );
+ return p_out_buf;
+ }
+ }
memcpy( p_sys->p_buf,
p_in_orig + (i_in_nb - 2 * p_sys->i_old_wing) *
i_nb_channels, (2 * p_sys->i_old_wing) *
i_out * p_filter->fmt_in.audio.i_bytes_per_frame );
#endif
- /* 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);
+ block_Release( p_in_buf );
return p_out_buf;
}
filter_t *p_filter = (filter_t *)p_this;
filter_sys_t *p_sys;
unsigned int i_out_rate = p_filter->fmt_out.audio.i_rate;
- double d_factor;
- int i_filter_wing;
- if( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate ||
- p_filter->fmt_in.i_codec != VLC_CODEC_FL32 )
+ if ( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate
+ || p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format
+ || p_filter->fmt_in.audio.i_physical_channels
+ != p_filter->fmt_out.audio.i_physical_channels
+ || p_filter->fmt_in.audio.i_original_channels
+ != p_filter->fmt_out.audio.i_original_channels
+ || p_filter->fmt_in.audio.i_format != VLC_CODEC_FL32 )
{
return VLC_EGENERIC;
}
#if !defined( SYS_DARWIN )
- if( !config_GetInt( p_this, "hq-resampling" ) )
+ if( !var_InheritBool( p_this, "hq-resampling" ) )
{
return VLC_EGENERIC;
}
if( p_sys == NULL )
return VLC_ENOMEM;
- /* Calculate worst case for the length of the filter wing */
- d_factor = (double)i_out_rate / p_filter->fmt_in.audio.i_rate;
- i_filter_wing = ((SMALL_FILTER_NMULT + 1)/2.0)
- * __MAX(1.0, 1.0/d_factor) + 10;
- p_sys->i_buf_size = p_filter->fmt_in.audio.i_channels *
- sizeof(int32_t) * 2 * i_filter_wing;
-
- /* Allocate enough memory to buffer previous samples */
- p_sys->p_buf = malloc( p_sys->i_buf_size );
- if( p_sys->p_buf == NULL )
- {
- free( p_sys );
- return VLC_ENOMEM;
- }
+ p_sys->p_buf = NULL;
+ p_sys->i_buf_size = 0;
p_sys->i_old_wing = 0;
p_sys->b_first = true;
free( p_filter->p_sys );
}
-void FilterFloatUP( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
- float *p_out, uint32_t ui_remainder,
- uint32_t ui_output_rate, int16_t Inc, int i_nb_channels )
+static void FilterFloatUP( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
+ float *p_out, uint32_t ui_remainder,
+ uint32_t ui_output_rate, int16_t Inc, int i_nb_channels )
{
const float *Hp, *Hdp, *End;
float t, temp;
}
}
-void FilterFloatUD( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
- float *p_out, uint32_t ui_remainder,
- uint32_t ui_output_rate, uint32_t ui_input_rate,
- int16_t Inc, int i_nb_channels )
+static void FilterFloatUD( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
+ float *p_out, uint32_t ui_remainder,
+ uint32_t ui_output_rate, uint32_t ui_input_rate,
+ int16_t Inc, int i_nb_channels )
{
const float *Hp, *Hdp, *End;
float t, temp;
p_in += (Inc * i_nb_channels); /* Input signal step */
}
}
+
+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 **pp_out_buf, size_t *pi_out,
+ float **pp_in,
+ int i_in, int i_in_end,
+ double d_factor, bool b_factor_old,
+ int i_nb_channels, int i_bytes_per_frame )
+{
+ filter_sys_t *p_sys = p_filter->p_sys;
+
+ float *p_in = *pp_in;
+ size_t i_out = *pi_out;
+ 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;
+ p_out += i_nb_channels;
+ i_out++;
+ continue;
+ }
+
+ while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
+ {
+ if( ReallocBuffer( pp_out_buf, &p_out,
+ i_out, i_nb_channels, i_bytes_per_frame ) )
+ return;
+
+ if( d_factor >= 1 )
+ {
+ /* FilterFloatUP() is faster if we can use it */
+
+ /* Perform left-wing inner product */
+ FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
+ SMALL_FILTER_NWING, p_in, p_out,
+ p_sys->i_remainder,
+ p_filter->fmt_out.audio.i_rate,
+ -1, i_nb_channels );
+ /* Perform right-wing inner product */
+ FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
+ SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
+ p_filter->fmt_out.audio.i_rate -
+ p_sys->i_remainder,
+ p_filter->fmt_out.audio.i_rate,
+ 1, i_nb_channels );
+
+#if 0
+ /* Normalize for unity filter gain */
+ for( i = 0; i < i_nb_channels; i++ )
+ {
+ *(p_out+i) *= d_old_scale_factor;
+ }
+#endif
+ }
+ else
+ {
+ /* Perform left-wing inner product */
+ FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
+ SMALL_FILTER_NWING, p_in, p_out,
+ p_sys->i_remainder,
+ p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
+ -1, i_nb_channels );
+ /* Perform right-wing inner product */
+ FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
+ SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
+ p_filter->fmt_out.audio.i_rate -
+ p_sys->i_remainder,
+ p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
+ 1, i_nb_channels );
+ }
+
+ p_out += i_nb_channels;
+ i_out++;
+
+ p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
+ }
+
+ p_in += i_nb_channels;
+ p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
+ }
+
+ *pp_in = p_in;
+ *pi_out = i_out;
+}
+
+
projects / vlc / blobdiff