/*****************************************************************************
* Local prototypes
*****************************************************************************/
-static int Create ( vlc_object_t * );
-static void Close ( vlc_object_t * );
-static void DoWork ( aout_instance_t *, aout_filter_t *, aout_buffer_t *,
- aout_buffer_t * );
/* audio filter2 */
static int OpenFilter ( vlc_object_t * );
struct filter_sys_t
{
int32_t *p_buf; /* this filter introduces a delay */
- int i_buf_size;
+ size_t i_buf_size;
- int i_old_rate;
double d_old_factor;
int i_old_wing;
unsigned int i_remainder; /* remainder of previous sample */
-
- audio_date_t end_date;
-
bool b_first;
- bool b_filter2;
+
+ date_t end_date;
};
/*****************************************************************************
set_category( CAT_AUDIO )
set_subcategory( SUBCAT_AUDIO_MISC )
set_description( N_("Audio filter for band-limited interpolation resampling") )
- set_capability( "audio filter", 20 )
- set_callbacks( Create, Close )
-
- add_submodule ()
- set_description( N_("Audio filter for band-limited interpolation resampling") )
set_capability( "audio filter2", 20 )
set_callbacks( OpenFilter, CloseFilter )
vlc_module_end ()
/*****************************************************************************
- * Create: allocate linear resampler
+ * Resample: convert a buffer
*****************************************************************************/
-static int Create( vlc_object_t *p_this )
+static block_t *Resample( filter_t * p_filter, block_t * p_in_buf )
{
- aout_filter_t * p_filter = (aout_filter_t *)p_this;
- struct filter_sys_t * p_sys;
- double d_factor;
- int i_filter_wing;
-
- if ( p_filter->input.i_rate == p_filter->output.i_rate
- || p_filter->input.i_format != p_filter->output.i_format
- || p_filter->input.i_physical_channels
- != p_filter->output.i_physical_channels
- || p_filter->input.i_original_channels
- != p_filter->output.i_original_channels
- || p_filter->input.i_format != VLC_CODEC_FL32 )
- {
- return VLC_EGENERIC;
- }
-
-#if !defined( __APPLE__ )
- if( !config_GetInt( p_this, "hq-resampling" ) )
+ if( !p_in_buf || !p_in_buf->i_nb_samples )
{
- return VLC_EGENERIC;
- }
-#endif
-
- /* Allocate the memory needed to store the module's structure */
- p_sys = malloc( sizeof(filter_sys_t) );
- if( p_sys == NULL )
- return VLC_ENOMEM;
- p_filter->p_sys = (struct aout_filter_sys_t *)p_sys;
-
- /* Calculate worst case for the length of the filter wing */
- d_factor = (double)p_filter->output.i_rate
- / 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_sys->i_buf_size = aout_FormatNbChannels( &p_filter->input ) *
- 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;
+ if( p_in_buf )
+ block_Release( p_in_buf );
+ return NULL;
}
- p_sys->i_old_wing = 0;
- p_sys->b_filter2 = false; /* It seams to be a good valuefor this module */
- p_filter->pf_do_work = DoWork;
-
- /* We don't want a new buffer to be created because we're not sure we'll
- * actually need to resample anything. */
- p_filter->b_in_place = true;
-
- return VLC_SUCCESS;
-}
-
-/*****************************************************************************
- * Close: free our resources
- *****************************************************************************/
-static void Close( vlc_object_t * p_this )
-{
- aout_filter_t * p_filter = (aout_filter_t *)p_this;
- filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
- free( p_sys->p_buf );
- free( p_sys );
-}
-
-/*****************************************************************************
- * 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.mixer.i_rate;
+ 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->input.i_rate )
+ if( i_out_rate == p_filter->fmt_in.audio.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 )
+ if( !(p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY) &&
+ p_sys->i_old_wing )
{
/* 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 );
+ 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->input.i_bytes_per_frame );
+ p_filter->fmt_in.audio.i_bytes_per_frame );
- p_out_buf->i_nb_samples = p_in_buf->i_nb_samples +
- p_sys->i_old_wing;
+ p_in_buf->i_nb_samples += p_sys->i_old_wing;
- 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 *
- p_filter->input.i_bytes_per_frame;
+ 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_filter->b_continuity = false;
p_sys->i_old_wing = 0;
- return;
+ p_sys->b_first = true;
+ return p_in_buf;
}
- if( !p_filter->b_continuity )
+ 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_filter->b_continuity = true;
+ p_out_buf->i_flags |= BLOCK_FLAG_DISCONTINUITY;
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;
+ 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,
/* 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],
+ 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 */
{
vlc_memcpy( p_in, p_sys->p_buf,
p_sys->i_old_wing * 2 *
- p_filter->input.i_bytes_per_frame );
+ 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->input.i_bytes_per_frame );
+ 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_size );
+ 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->input.i_rate;
+ 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 */
{
/* Just copy the samples */
memcpy( p_out, p_in,
- p_filter->input.i_bytes_per_frame );
+ 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->output.i_rate )
+ while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
{
if( p_sys->d_old_factor >= 1 )
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,
+ 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->output.i_rate -
+ p_filter->fmt_out.audio.i_rate -
p_sys->i_remainder,
- p_filter->output.i_rate,
+ p_filter->fmt_out.audio.i_rate,
1, i_nb_channels );
#if 0
#endif
/* Sanity check */
- if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame
+ 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->input.i_rate;
+ p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
break;
}
}
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,
+ 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->output.i_rate -
+ p_filter->fmt_out.audio.i_rate -
p_sys->i_remainder,
- p_filter->output.i_rate, p_filter->input.i_rate,
+ 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->input.i_rate;
+ p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
}
p_in += i_nb_channels;
- p_sys->i_remainder -= p_filter->output.i_rate;
+ 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->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 )
+ while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
{
if( d_factor >= 1 )
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,
+ 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->output.i_rate -
+ p_filter->fmt_out.audio.i_rate -
p_sys->i_remainder,
- p_filter->output.i_rate,
+ p_filter->fmt_out.audio.i_rate,
1, i_nb_channels );
#if 0
}
#endif
/* Sanity check */
- if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame
+ 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->input.i_rate;
+ p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
break;
}
}
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,
+ 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->output.i_rate -
+ p_filter->fmt_out.audio.i_rate -
p_sys->i_remainder,
- p_filter->output.i_rate, p_filter->input.i_rate,
+ 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->input.i_rate;
+ p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
}
p_in += i_nb_channels;
- p_sys->i_remainder -= p_filter->output.i_rate;
+ 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->input.i_bytes_per_frame );
+ 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_size,
- i_out * p_filter->input.i_bytes_per_frame );
+ 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
- /* Finalize aout buffer */
- p_out_buf->i_nb_samples = i_out;
- 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 *
- i_nb_channels * sizeof(int32_t);
-
+ 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_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_sys->p_buf = NULL;
+ p_sys->i_buf_size = 0;
- p_filter->p_sys->i_old_wing = 0;
+ 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",
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 )
- block_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_filter->p_sys->i_buf_size;
-
- 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" );
- block_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 );
-
- block_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 )
projects / vlc / blobdiff