/*****************************************************************************
* 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 * );
int i_old_wing;
unsigned int i_remainder; /* remainder of previous sample */
+ bool b_first;
date_t end_date;
-
- bool b_filter2;
};
/*****************************************************************************
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->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( __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->fmt_out.audio.i_rate
- / p_filter->fmt_in.audio.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->fmt_in.audio ) *
- 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;
-
+ 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 );
- 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->fmt_out.audio.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->fmt_in.audio.i_rate )
{
-#if 0 /* FIXME: needs audio filter2 to use block_Realloc */
if( /*p_filter->b_continuity && /--* What difference does it make ? :) */
p_sys->i_old_wing )
{
p_sys->i_old_wing * p_filter->fmt_in.audio.i_bytes_per_frame,
p_in_buf->i_buffer );
if( !p_in_buf )
- abort();
+ 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_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->i_pts = date_Get( &p_sys->end_date );
- p_out_buf->i_length =
+ p_in_buf->i_pts = date_Get( &p_sys->end_date );
+ p_in_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 *
- p_filter->fmt_in.audio.i_bytes_per_frame;
+ p_in_buf->i_nb_samples ) - p_in_buf->i_pts;
}
-#endif
- p_out_buf->i_flags |= BLOCK_FLAG_DISCONTINUITY;
+ p_in_buf->i_flags |= BLOCK_FLAG_DISCONTINUITY;
p_sys->i_old_wing = 0;
- return;
+ return p_in_buf;
}
- if( p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY )
+ 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. */
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 * 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 );
p_out_buf->i_buffer = p_out_buf->i_nb_samples *
i_nb_channels * sizeof(int32_t);
-
+ return p_out_buf;
}
/*****************************************************************************
}
p_sys->i_old_wing = 0;
- p_sys->b_filter2 = true;
+ p_sys->b_first = 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_nb_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_nb_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_nb_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.fmt_in.audio = p_filter->fmt_in.audio;
- aout_filter.fmt_in.audio.i_bytes_per_frame = p_filter->fmt_in.audio.i_channels *
- p_filter->fmt_in.audio.i_bitspersample / 8;
- aout_filter.fmt_out.audio = p_filter->fmt_out.audio;
- aout_filter.fmt_out.audio.i_bytes_per_frame = p_filter->fmt_out.audio.i_channels *
- p_filter->fmt_out.audio.i_bitspersample / 8;
-
- in_buf.p_buffer = p_block->p_buffer;
- in_buf.i_buffer = p_block->i_buffer;
- in_buf.i_nb_samples = p_block->i_nb_samples;
- out_buf.p_buffer = p_out->p_buffer;
- out_buf.i_buffer = p_out->i_buffer;
- out_buf.i_nb_samples = p_out->i_nb_samples;
-
- DoWork( (aout_instance_t *)p_filter, &aout_filter, &in_buf, &out_buf );
-
- block_Release( p_block );
-
- p_out->i_buffer = out_buf.i_buffer;
- p_out->i_nb_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 )
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