/*****************************************************************************
* DoWork: convert a buffer
*****************************************************************************/
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_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( p_sys->b_filter2 )
i_out_rate = p_filter->output.i_rate;
else
i_out_rate = p_aout->mixer_format.i_rate;
/* Check if we really need to run the resampler */
if( i_out_rate == p_filter->input.i_rate )
{
if( /*p_filter->b_continuity && /--* What difference does it make ? :) */
p_sys->i_old_wing &&
p_in_buf->i_size >=
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_sys->i_old_wing,
p_in_buf->p_buffer, p_in_buf->i_nb_bytes );
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_sys->i_old_wing;
p_out_buf->start_date = date_Get( &p_sys->end_date );
p_out_buf->end_date =
date_Increment( &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_sys->i_old_wing = 0;
return;
}
if( !p_filter->b_continuity )
{
/* Continuity in sound samples has been broken, we'd better reset
* everything. */
p_filter->b_continuity = true;
p_sys->i_remainder = 0;
date_Init( &p_sys->end_date, i_out_rate, 1 );
date_Set( &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_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_sys->i_old_wing * 2);
float p_in_orig[i_in_nb * p_filter->input.i_bytes_per_frame / 4],
*p_in = p_in_orig;
/* Copy all our samples in p_in */
if( p_sys->i_old_wing )
{
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_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 );
/* Make sure the output buffer is reset */
memset( p_out, 0, p_out_buf->i_size );
/* Calculate the new length of the filter wing */
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_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_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->input.i_bytes_per_frame );
p_in += i_nb_channels;
p_out += i_nb_channels;
i_out++;
continue;
}
while( p_sys->i_remainder < p_filter->output.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->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_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++ )
{
*(p_out+i) *= d_old_scale_factor;
}
#endif
/* Sanity check */
if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame
<= (unsigned int)i_out+1 )
{
p_out += i_nb_channels;
i_out++;
p_sys->i_remainder += p_filter->input.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->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_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_sys->i_remainder += p_filter->input.i_rate;
}
p_in += i_nb_channels;
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_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_sys->i_remainder < p_filter->output.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->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_sys->i_remainder,
p_filter->output.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_size/p_filter->input.i_bytes_per_frame
<= (unsigned int)i_out+1 )
{
p_out += i_nb_channels;
i_out++;
p_sys->i_remainder += p_filter->input.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->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_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_sys->i_remainder += p_filter->input.i_rate;
}
p_in += i_nb_channels;
p_sys->i_remainder -= p_filter->output.i_rate;
}
/* Buffer i_filter_wing * 2 samples for next time */
if( 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 );
}
#if 0
msg_Err( p_filter, "p_out size: %i, nb bytes out: %i", p_out_buf->i_size,
i_out * p_filter->input.i_bytes_per_frame );
#endif
/* Finalize aout buffer */
p_out_buf->i_nb_samples = i_out;
p_out_buf->start_date = date_Get( &p_sys->end_date );
p_out_buf->end_date = date_Increment( &p_sys->end_date,
p_out_buf->i_nb_samples );
p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples *
i_nb_channels * sizeof(int32_t);
}
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;
}
/*****************************************************************************
* Resample: convert a buffer
*****************************************************************************/
static block_t *Resample( filter_t * p_filter, block_t * p_in_buf )
{
if( !p_in_buf || !p_in_buf->i_nb_samples )
{
if( p_in_buf )
block_Release( p_in_buf );
return NULL;
}
filter_sys_t *p_sys = p_filter->p_sys;
unsigned int i_out_rate = p_filter->fmt_out.audio.i_rate;
int i_nb_channels = aout_FormatNbChannels( &p_filter->fmt_in.audio );
/* Check if we really need to run the resampler */
if( i_out_rate == p_filter->fmt_in.audio.i_rate )
{
if( !(p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY) &&
p_sys->i_old_wing )
{
/* output the whole thing with the samples from last time */
p_in_buf = block_Realloc( p_in_buf,
p_sys->i_old_wing * p_filter->fmt_in.audio.i_bytes_per_frame,
p_in_buf->i_buffer );
if( !p_in_buf )
return NULL;
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->fmt_in.audio.i_bytes_per_frame );
p_in_buf->i_nb_samples += p_sys->i_old_wing;
p_in_buf->i_pts = date_Get( &p_sys->end_date );
p_in_buf->i_length =
date_Increment( &p_sys->end_date,
p_in_buf->i_nb_samples ) - p_in_buf->i_pts;
}
p_sys->i_old_wing = 0;
p_sys->b_first = true;
return p_in_buf;
}
unsigned i_bytes_per_frame = p_filter->fmt_out.audio.i_channels *
p_filter->fmt_out.audio.i_bitspersample / 8;
size_t i_out_size = i_bytes_per_frame * ( 1 + ( p_in_buf->i_nb_samples *
p_filter->fmt_out.audio.i_rate / p_filter->fmt_in.audio.i_rate) )
+ p_filter->p_sys->i_buf_size;
block_t *p_out_buf = filter_NewAudioBuffer( p_filter, i_out_size );
if( !p_out_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 )
{
/* Continuity in sound samples has been broken, we'd better reset
* everything. */
p_out_buf->i_flags |= BLOCK_FLAG_DISCONTINUITY;
p_sys->i_remainder = 0;
date_Init( &p_sys->end_date, i_out_rate, 1 );
date_Set( &p_sys->end_date, p_in_buf->i_pts );
p_sys->d_old_factor = 1;
p_sys->i_old_wing = 0;
p_sys->b_first = false;
}
int i_in_nb = p_in_buf->i_nb_samples;
int i_in, i_out = 0;
double d_factor, d_scale_factor, d_old_scale_factor;
int 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_rate, p_sys->d_old_factor,
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;
/* Copy all our samples in p_in */
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 );
}
/* 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 );
/* Make sure the output buffer is reset */
memset( p_out, 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;
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_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_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++;
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;
}
/* 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( 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;
}
/* 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 )
{
size_t newsize = p_sys->i_old_wing * 2
* p_filter->fmt_in.audio.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! */
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) *
p_filter->fmt_in.audio.i_bytes_per_frame );
}
#if 0
msg_Err( p_filter, "p_out size: %i, nb bytes out: %i", p_out_buf->i_buffer,
i_out * p_filter->fmt_in.audio.i_bytes_per_frame );
#endif
return p_out_buf;
}