#include <vlc_common.h>
#include <vlc_plugin.h>
#include <vlc_aout.h>
+#include <vlc_filter.h>
+#include <vlc_block.h>
#include "bandlimited.h"
static void DoWork ( aout_instance_t *, aout_filter_t *, aout_buffer_t *,
aout_buffer_t * );
+/* audio filter2 */
+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,
/*****************************************************************************
* Local structures
*****************************************************************************/
-struct aout_filter_sys_t
+struct filter_sys_t
{
int32_t *p_buf; /* this filter introduces a delay */
int i_buf_size;
unsigned int i_remainder; /* remainder of previous sample */
audio_date_t end_date;
+
+ bool b_first;
+ bool b_filter2;
};
/*****************************************************************************
set_description( N_("Audio filter for band-limited interpolation resampling") );
set_capability( "audio filter", 20 );
set_callbacks( Create, Close );
+
+ add_submodule();
+ set_description( _("Audio filter for band-limited interpolation resampling") );
+ set_capability( "audio filter2", 20 );
+ set_callbacks( OpenFilter, CloseFilter );
vlc_module_end();
/*****************************************************************************
static int Create( vlc_object_t *p_this )
{
aout_filter_t * p_filter = (aout_filter_t *)p_this;
+ struct filter_sys_t * p_sys;
double d_factor;
int i_filter_wing;
#endif
/* Allocate the memory needed to store the module's structure */
- p_filter->p_sys = malloc( sizeof(struct aout_filter_sys_t) );
- if( p_filter->p_sys == NULL )
+ p_sys = malloc( sizeof(filter_sys_t) );
+ p_filter->p_sys = (struct aout_filter_sys_t *)p_sys;
+ if( p_sys == NULL )
return VLC_ENOMEM;
/* Calculate worst case for the length of the filter wing */
/ p_filter->input.i_rate / AOUT_MAX_INPUT_RATE;
i_filter_wing = ((SMALL_FILTER_NMULT + 1)/2.0)
* __MAX(1.0, 1.0/d_factor) + 10;
- p_filter->p_sys->i_buf_size = aout_FormatNbChannels( &p_filter->input ) *
+ p_sys->i_buf_size = aout_FormatNbChannels( &p_filter->input ) *
sizeof(int32_t) * 2 * i_filter_wing;
/* Allocate enough memory to buffer previous samples */
- p_filter->p_sys->p_buf = malloc( p_filter->p_sys->i_buf_size );
- if( p_filter->p_sys->p_buf == NULL )
+ p_sys->p_buf = malloc( p_sys->i_buf_size );
+ if( p_sys->p_buf == NULL )
+ {
+ free( p_sys );
return VLC_ENOMEM;
+ }
- p_filter->p_sys->i_old_wing = 0;
+ p_sys->i_old_wing = 0;
p_filter->pf_do_work = DoWork;
/* We don't want a new buffer to be created because we're not sure we'll
static void Close( vlc_object_t * p_this )
{
aout_filter_t * p_filter = (aout_filter_t *)p_this;
- free( p_filter->p_sys->p_buf );
- free( p_filter->p_sys );
+ filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
+ free( p_sys->p_buf );
+ free( p_sys );
}
/*****************************************************************************
static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
{
+ filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
float *p_in, *p_in_orig, *p_out = (float *)p_out_buf->p_buffer;
int i_nb_channels = aout_FormatNbChannels( &p_filter->input );
int i_in_nb = p_in_buf->i_nb_samples;
int i_in, i_out = 0;
+ unsigned int i_out_rate;
double d_factor, d_scale_factor, d_old_scale_factor;
int i_filter_wing;
-#if 0
- int i;
-#endif
+
+ if( p_sys->b_filter2 )
+ i_out_rate = p_filter->output.i_rate;
+ else
+ i_out_rate = p_aout->mixer.mixer.i_rate;
/* Check if we really need to run the resampler */
- if( p_aout->mixer.mixer.i_rate == p_filter->input.i_rate )
+ if( i_out_rate == p_filter->input.i_rate )
{
if( /*p_filter->b_continuity && /--* What difference does it make ? :) */
- p_filter->p_sys->i_old_wing &&
+ p_sys->i_old_wing &&
p_in_buf->i_size >=
- p_in_buf->i_nb_bytes + p_filter->p_sys->i_old_wing *
+ p_in_buf->i_nb_bytes + p_sys->i_old_wing *
p_filter->input.i_bytes_per_frame )
{
/* output the whole thing with the samples from last time */
memmove( ((float *)(p_in_buf->p_buffer)) +
- i_nb_channels * p_filter->p_sys->i_old_wing,
+ i_nb_channels * p_sys->i_old_wing,
p_in_buf->p_buffer, p_in_buf->i_nb_bytes );
- memcpy( p_in_buf->p_buffer, p_filter->p_sys->p_buf +
- i_nb_channels * p_filter->p_sys->i_old_wing,
- p_filter->p_sys->i_old_wing *
+ memcpy( p_in_buf->p_buffer, p_sys->p_buf +
+ i_nb_channels * p_sys->i_old_wing,
+ p_sys->i_old_wing *
p_filter->input.i_bytes_per_frame );
p_out_buf->i_nb_samples = p_in_buf->i_nb_samples +
- p_filter->p_sys->i_old_wing;
+ p_sys->i_old_wing;
- p_out_buf->start_date = aout_DateGet( &p_filter->p_sys->end_date );
+ p_out_buf->start_date = aout_DateGet( &p_sys->end_date );
p_out_buf->end_date =
- aout_DateIncrement( &p_filter->p_sys->end_date,
+ aout_DateIncrement( &p_sys->end_date,
p_out_buf->i_nb_samples );
p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples *
p_filter->input.i_bytes_per_frame;
}
p_filter->b_continuity = false;
- p_filter->p_sys->i_old_wing = 0;
+ p_sys->i_old_wing = 0;
return;
}
/* Continuity in sound samples has been broken, we'd better reset
* everything. */
p_filter->b_continuity = true;
- p_filter->p_sys->i_remainder = 0;
- aout_DateInit( &p_filter->p_sys->end_date, p_filter->output.i_rate );
- aout_DateSet( &p_filter->p_sys->end_date, p_in_buf->start_date );
- p_filter->p_sys->i_old_rate = p_filter->input.i_rate;
- p_filter->p_sys->d_old_factor = 1;
- p_filter->p_sys->i_old_wing = 0;
+ p_sys->i_remainder = 0;
+ aout_DateInit( &p_sys->end_date, i_out_rate );
+ aout_DateSet( &p_sys->end_date, p_in_buf->start_date );
+ p_sys->i_old_rate = p_filter->input.i_rate;
+ p_sys->d_old_factor = 1;
+ p_sys->i_old_wing = 0;
}
#if 0
msg_Err( p_filter, "old rate: %i, old factor: %f, old wing: %i, i_in: %i",
- p_filter->p_sys->i_old_rate, p_filter->p_sys->d_old_factor,
- p_filter->p_sys->i_old_wing, i_in_nb );
+ p_sys->i_old_rate, p_sys->d_old_factor,
+ p_sys->i_old_wing, i_in_nb );
#endif
/* Prepare the source buffer */
- i_in_nb += (p_filter->p_sys->i_old_wing * 2);
+ i_in_nb += (p_sys->i_old_wing * 2);
#ifdef HAVE_ALLOCA
p_in = p_in_orig = (float *)alloca( i_in_nb *
p_filter->input.i_bytes_per_frame );
}
/* Copy all our samples in p_in */
- if( p_filter->p_sys->i_old_wing )
+ if( p_sys->i_old_wing )
{
- vlc_memcpy( p_in, p_filter->p_sys->p_buf,
- p_filter->p_sys->i_old_wing * 2 *
+ vlc_memcpy( p_in, p_sys->p_buf,
+ p_sys->i_old_wing * 2 *
p_filter->input.i_bytes_per_frame );
}
- vlc_memcpy( p_in + p_filter->p_sys->i_old_wing * 2 * i_nb_channels,
+ 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->input.i_bytes_per_frame );
memset( p_out, 0, p_out_buf->i_size );
/* Calculate the new length of the filter wing */
- d_factor = (double)p_aout->mixer.mixer.i_rate / p_filter->input.i_rate;
+ d_factor = (double)i_out_rate / p_filter->input.i_rate;
i_filter_wing = ((SMALL_FILTER_NMULT+1)/2.0) * __MAX(1.0,1.0/d_factor) + 1;
/* Account for increased filter gain when using factors less than 1 */
d_old_scale_factor = SMALL_FILTER_SCALE *
- p_filter->p_sys->d_old_factor + 0.5;
+ p_sys->d_old_factor + 0.5;
d_scale_factor = SMALL_FILTER_SCALE * d_factor + 0.5;
/* Apply the old rate until we have enough samples for the new one */
- i_in = p_filter->p_sys->i_old_wing;
- p_in += p_filter->p_sys->i_old_wing * i_nb_channels;
+ 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_filter->p_sys->i_old_wing) < i_in_nb; i_in++ )
+ (i_in + p_sys->i_old_wing) < i_in_nb; i_in++ )
{
- if( p_filter->p_sys->d_old_factor == 1 )
+ if( p_sys->d_old_factor == 1 )
{
/* Just copy the samples */
memcpy( p_out, p_in,
continue;
}
- while( p_filter->p_sys->i_remainder < p_filter->output.i_rate )
+ while( p_sys->i_remainder < p_filter->output.i_rate )
{
- if( p_filter->p_sys->d_old_factor >= 1 )
+ 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_filter->p_sys->i_remainder,
+ p_sys->i_remainder,
p_filter->output.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->output.i_rate -
- p_filter->p_sys->i_remainder,
+ p_sys->i_remainder,
p_filter->output.i_rate,
1, i_nb_channels );
{
p_out += i_nb_channels;
i_out++;
- p_filter->p_sys->i_remainder += p_filter->input.i_rate;
+ p_sys->i_remainder += p_filter->input.i_rate;
break;
}
}
/* Perform left-wing inner product */
FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
SMALL_FILTER_NWING, p_in, p_out,
- p_filter->p_sys->i_remainder,
+ p_sys->i_remainder,
p_filter->output.i_rate, p_filter->input.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->output.i_rate -
- p_filter->p_sys->i_remainder,
+ p_sys->i_remainder,
p_filter->output.i_rate, p_filter->input.i_rate,
1, i_nb_channels );
}
p_out += i_nb_channels;
i_out++;
- p_filter->p_sys->i_remainder += p_filter->input.i_rate;
+ p_sys->i_remainder += p_filter->input.i_rate;
}
p_in += i_nb_channels;
- p_filter->p_sys->i_remainder -= p_filter->output.i_rate;
+ p_sys->i_remainder -= p_filter->output.i_rate;
}
/* Apply the new rate for the rest of the samples */
if( i_in < i_in_nb - i_filter_wing )
{
- p_filter->p_sys->i_old_rate = p_filter->input.i_rate;
- p_filter->p_sys->d_old_factor = d_factor;
- p_filter->p_sys->i_old_wing = i_filter_wing;
+ p_sys->i_old_rate = p_filter->input.i_rate;
+ 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_filter->p_sys->i_remainder < p_filter->output.i_rate )
+ while( p_sys->i_remainder < p_filter->output.i_rate )
{
if( d_factor >= 1 )
/* Perform left-wing inner product */
FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
SMALL_FILTER_NWING, p_in, p_out,
- p_filter->p_sys->i_remainder,
+ p_sys->i_remainder,
p_filter->output.i_rate,
-1, i_nb_channels );
FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
p_filter->output.i_rate -
- p_filter->p_sys->i_remainder,
+ p_sys->i_remainder,
p_filter->output.i_rate,
1, i_nb_channels );
#if 0
/* Normalize for unity filter gain */
- for( i = 0; i < i_nb_channels; i++ )
+ for( int i = 0; i < i_nb_channels; i++ )
{
*(p_out+i) *= d_old_scale_factor;
}
{
p_out += i_nb_channels;
i_out++;
- p_filter->p_sys->i_remainder += p_filter->input.i_rate;
+ p_sys->i_remainder += p_filter->input.i_rate;
break;
}
}
/* Perform left-wing inner product */
FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
SMALL_FILTER_NWING, p_in, p_out,
- p_filter->p_sys->i_remainder,
+ p_sys->i_remainder,
p_filter->output.i_rate, p_filter->input.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->output.i_rate -
- p_filter->p_sys->i_remainder,
+ p_sys->i_remainder,
p_filter->output.i_rate, p_filter->input.i_rate,
1, i_nb_channels );
}
p_out += i_nb_channels;
i_out++;
- p_filter->p_sys->i_remainder += p_filter->input.i_rate;
+ p_sys->i_remainder += p_filter->input.i_rate;
}
p_in += i_nb_channels;
- p_filter->p_sys->i_remainder -= p_filter->output.i_rate;
+ p_sys->i_remainder -= p_filter->output.i_rate;
}
/* Buffer i_filter_wing * 2 samples for next time */
- if( p_filter->p_sys->i_old_wing )
+ if( p_sys->i_old_wing )
{
- memcpy( p_filter->p_sys->p_buf,
- p_in_orig + (i_in_nb - 2 * p_filter->p_sys->i_old_wing) *
- i_nb_channels, (2 * p_filter->p_sys->i_old_wing) *
+ 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) *
p_filter->input.i_bytes_per_frame );
}
/* Finalize aout buffer */
p_out_buf->i_nb_samples = i_out;
- p_out_buf->start_date = aout_DateGet( &p_filter->p_sys->end_date );
- p_out_buf->end_date = aout_DateIncrement( &p_filter->p_sys->end_date,
+ p_out_buf->start_date = aout_DateGet( &p_sys->end_date );
+ p_out_buf->end_date = aout_DateIncrement( &p_sys->end_date,
p_out_buf->i_nb_samples );
p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples *
}
+/*****************************************************************************
+ * OpenFilter:
+ *****************************************************************************/
+static int OpenFilter( vlc_object_t *p_this )
+{
+ 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_FOURCC('f','l','3','2') )
+ {
+ return VLC_EGENERIC;
+ }
+
+#if !defined( SYS_DARWIN )
+ if( !config_GetInt( p_this, "hq-resampling" ) )
+ {
+ return VLC_EGENERIC;
+ }
+#endif
+
+ /* Allocate the memory needed to store the module's structure */
+ p_filter->p_sys = p_sys = malloc( sizeof(struct filter_sys_t) );
+ 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_filter->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_filter->p_sys->p_buf = malloc( p_filter->p_sys->i_buf_size );
+ if( p_filter->p_sys->p_buf == NULL )
+ {
+ free( p_sys );
+ return VLC_ENOMEM;
+ }
+
+ p_filter->p_sys->i_old_wing = 0;
+ p_sys->b_first = true;
+ p_sys->b_filter2 = true;
+ p_filter->pf_audio_filter = Resample;
+
+ msg_Dbg( p_this, "%4.4s/%iKHz/%i->%4.4s/%iKHz/%i",
+ (char *)&p_filter->fmt_in.i_codec,
+ p_filter->fmt_in.audio.i_rate,
+ p_filter->fmt_in.audio.i_channels,
+ (char *)&p_filter->fmt_out.i_codec,
+ p_filter->fmt_out.audio.i_rate,
+ p_filter->fmt_out.audio.i_channels);
+
+ p_filter->fmt_out = p_filter->fmt_in;
+ p_filter->fmt_out.audio.i_rate = i_out_rate;
+
+ return 0;
+}
+
+/*****************************************************************************
+ * CloseFilter : deallocate data structures
+ *****************************************************************************/
+static void CloseFilter( vlc_object_t *p_this )
+{
+ filter_t *p_filter = (filter_t *)p_this;
+ free( p_filter->p_sys->p_buf );
+ free( p_filter->p_sys );
+}
+
+/*****************************************************************************
+ * Resample
+ *****************************************************************************/
+static block_t *Resample( filter_t *p_filter, block_t *p_block )
+{
+ aout_filter_t aout_filter;
+ aout_buffer_t in_buf, out_buf;
+ block_t *p_out;
+ int i_out_size;
+ int i_bytes_per_frame;
+
+ if( !p_block || !p_block->i_samples )
+ {
+ if( p_block ) p_block->pf_release( p_block );
+ return NULL;
+ }
+
+ i_bytes_per_frame = p_filter->fmt_out.audio.i_channels *
+ p_filter->fmt_out.audio.i_bitspersample / 8;
+
+ i_out_size = i_bytes_per_frame * ( 1 + (p_block->i_samples *
+ p_filter->fmt_out.audio.i_rate / p_filter->fmt_in.audio.i_rate));
+
+ p_out = p_filter->pf_audio_buffer_new( p_filter, i_out_size );
+ if( !p_out )
+ {
+ msg_Warn( p_filter, "can't get output buffer" );
+ p_block->pf_release( p_block );
+ return NULL;
+ }
+
+ p_out->i_samples = i_out_size / i_bytes_per_frame;
+ p_out->i_dts = p_block->i_dts;
+ p_out->i_pts = p_block->i_pts;
+ p_out->i_length = p_block->i_length;
+
+ aout_filter.p_sys = (struct aout_filter_sys_t *)p_filter->p_sys;
+ aout_filter.input = p_filter->fmt_in.audio;
+ aout_filter.input.i_bytes_per_frame = p_filter->fmt_in.audio.i_channels *
+ p_filter->fmt_in.audio.i_bitspersample / 8;
+ aout_filter.output = p_filter->fmt_out.audio;
+ aout_filter.output.i_bytes_per_frame = p_filter->fmt_out.audio.i_channels *
+ p_filter->fmt_out.audio.i_bitspersample / 8;
+ aout_filter.b_continuity = !p_filter->p_sys->b_first;
+ p_filter->p_sys->b_first = false;
+
+ in_buf.p_buffer = p_block->p_buffer;
+ in_buf.i_nb_bytes = in_buf.i_size = p_block->i_buffer;
+ in_buf.i_nb_samples = p_block->i_samples;
+ out_buf.p_buffer = p_out->p_buffer;
+ out_buf.i_nb_bytes = out_buf.i_size = p_out->i_buffer;
+ out_buf.i_nb_samples = p_out->i_samples;
+
+ DoWork( (aout_instance_t *)p_filter, &aout_filter, &in_buf, &out_buf );
+
+ p_block->pf_release( p_block );
+
+ p_out->i_buffer = out_buf.i_nb_bytes;
+ p_out->i_samples = out_buf.i_nb_samples;
+
+ return p_out;
+}
+
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 )