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 *p_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
/* 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, 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_out_buf->i_buffer/i_bytes_per_frame <= i_out )
- break;
+ 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 );
- 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
- }
- 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_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++ )
- {
- 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;
-
- assert( i_out < p_out_buf->i_buffer/i_bytes_per_frame );
- 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
- }
- 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;
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 void ResampleFloat( filter_t *p_filter,
+ block_t *p_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*)p_out_buf->p_buffer + i_out * i_nb_channels;
+
+ for( ; i_in < i_in_end; i_in++ )
+ {
+ if( b_factor_old && d_factor == 1 )
+ {
+ /* 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( p_out_buf->i_buffer/i_bytes_per_frame <= i_out )
+ break;
+
+ 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;
+}
+
+