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Commit f4d3c850 authored by Rémi Denis-Courmont's avatar Rémi Denis-Courmont
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band-limited resampler: switch to audio filter2

parent dbd392ea
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Showing with 37 additions and 176 deletions
modules/audio_filter/resampler/bandlimited.c
View file @ f4d3c850
......@@ -48,10 +48,6 @@
/*****************************************************************************
* 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 * );
......@@ -81,10 +77,9 @@ struct filter_sys_t
int i_old_wing;
unsigned int i_remainder; /* remainder of previous sample */
bool b_first;
date_t end_date;
bool b_filter2;
};
/*****************************************************************************
......@@ -94,112 +89,29 @@ vlc_module_begin ()
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 )
{
......@@ -208,30 +120,35 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
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. */
......@@ -241,8 +158,14 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
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,
......@@ -262,9 +185,11 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
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 );
......@@ -457,7 +382,7 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
p_out_buf->i_buffer = p_out_buf->i_nb_samples *
i_nb_channels * sizeof(int32_t);
return p_out_buf;
}
/*****************************************************************************
......@@ -505,7 +430,7 @@ static int OpenFilter( vlc_object_t *p_this )
}
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",
......@@ -532,70 +457,6 @@ static void CloseFilter( vlc_object_t *p_this )
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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