X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=modules%2Faudio_filter%2Fresampler%2Fbandlimited.c;h=0fba703f447eaeed348dae654a1472c6550b575d;hb=d8e4bc56fd3fe95a18abccddc7f3943b00575c5e;hp=a2ad7bc68e4a218adb3b722521b051d6ea581597;hpb=faceb389ec12c13cc41901bc62265045f6f1ff39;p=vlc diff --git a/modules/audio_filter/resampler/bandlimited.c b/modules/audio_filter/resampler/bandlimited.c index a2ad7bc68e..0fba703f44 100644 --- a/modules/audio_filter/resampler/bandlimited.c +++ b/modules/audio_filter/resampler/bandlimited.c @@ -1,8 +1,8 @@ /***************************************************************************** - * bandlimited.c : bandlimited interpolation resampler + * bandlimited.c : band-limited interpolation resampler ***************************************************************************** - * Copyright (C) 2002 VideoLAN - * $Id: bandlimited.c,v 1.4 2003/03/05 19:31:32 gbazin Exp $ + * Copyright (C) 2002, 2006 the VideoLAN team + * $Id$ * * Authors: Gildas Bazin * @@ -10,7 +10,7 @@ * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. - * + * * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the @@ -18,13 +18,13 @@ * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111, USA. + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA. *****************************************************************************/ /***************************************************************************** * Preamble: * - * This implementation of the bandlimited interpolationis based on the + * This implementation of the band-limited interpolationis based on the * following paper: * http://ccrma-www.stanford.edu/~jos/resample/resample.html * @@ -32,28 +32,35 @@ * filter is 13 samples. * *****************************************************************************/ -#include /* malloc(), free() */ -#include -#include -#include "audio_output.h" -#include "aout_internal.h" +#ifdef HAVE_CONFIG_H +# include "config.h" +#endif + +#include +#include +#include +#include +#include + #include "bandlimited.h" /***************************************************************************** * 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 * ); -static void FilterFloatUP( float Imp[], float ImpD[], uint16_t Nwing, +/* 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, int i_nb_channels ); -static void FilterFloatUD( float Imp[], float ImpD[], uint16_t Nwing, +static void FilterFloatUD( const float Imp[], const float ImpD[], uint16_t Nwing, float *f_in, float *f_out, uint32_t ui_remainder, uint32_t ui_output_rate, uint32_t ui_input_rate, int16_t Inc, int i_nb_channels ); @@ -61,235 +68,176 @@ static void FilterFloatUD( float Imp[], float ImpD[], uint16_t Nwing, /***************************************************************************** * Local structures *****************************************************************************/ -struct aout_filter_sys_t +struct filter_sys_t { int32_t *p_buf; /* this filter introduces a delay */ int i_buf_size; - int i_old_rate; double d_old_factor; int i_old_wing; unsigned int i_remainder; /* remainder of previous sample */ + bool b_first; - audio_date_t end_date; + date_t end_date; }; /***************************************************************************** * Module descriptor *****************************************************************************/ -vlc_module_begin(); - set_description( _("audio filter for bandlimited interpolation resampling") ); - set_capability( "audio filter", 20 ); - set_callbacks( Create, Close ); -vlc_module_end(); +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 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; - 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_FOURCC('f','l','3','2') ) - { - return VLC_EGENERIC; - } - - /* 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 ) + if( !p_in_buf || !p_in_buf->i_nb_samples ) { - msg_Err( p_filter, "out of memory" ); - return VLC_ENOMEM; + if( p_in_buf ) + block_Release( p_in_buf ); + return NULL; } - /* Calculate worst case for the length of the filter wing */ - d_factor = (double)p_filter->output.i_rate - / p_filter->input.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 = 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 ) - { - msg_Err( p_filter, "out of memory" ); - return VLC_ENOMEM; - } - - 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 = VLC_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; - free( p_filter->p_sys->p_buf ); - free( p_filter->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 ) -{ - 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; - double d_factor, d_scale_factor, d_old_scale_factor; - int i_filter_wing; -#if 0 - int i; -#endif + 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( p_aout->mixer.mixer.i_rate == p_filter->input.i_rate ) + if( i_out_rate == p_filter->fmt_in.audio.i_rate ) { - if( p_filter->b_continuity && - p_in_buf->i_size >= - p_in_buf->i_nb_bytes + sizeof(float) * i_nb_channels ) + if( !(p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY) && + p_sys->i_old_wing ) { - if( p_filter->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_filter->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 * - 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_out_buf->end_date = - aout_DateIncrement( &p_filter->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; - } + /* 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_filter->b_continuity = VLC_FALSE; - return; + p_sys->i_old_wing = 0; + 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 = VLC_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_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_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); -#ifdef HAVE_ALLOCA - p_in = p_in_orig = (float *)alloca( i_in_nb * - p_filter->input.i_bytes_per_frame ); -#else - p_in = p_in_orig = (float *)malloc( i_in_nb * - p_filter->input.i_bytes_per_frame ); -#endif - if( p_in == NULL ) - { - return; - } + 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_filter->p_sys->i_old_wing ) + if( p_sys->i_old_wing ) { - p_aout->p_vlc->pf_memcpy( p_in, p_filter->p_sys->p_buf, - p_filter->p_sys->i_old_wing * 2 * - p_filter->input.i_bytes_per_frame ); + vlc_memcpy( p_in, p_sys->p_buf, + p_sys->i_old_wing * 2 * + p_filter->fmt_in.audio.i_bytes_per_frame ); } - p_aout->p_vlc->pf_memcpy( p_in + p_filter->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 ); + /* 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_size ); + memset( p_out, 0, p_out_buf->i_buffer ); /* 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->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_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_buf->p_buffer, p_in, - p_filter->input.i_bytes_per_frame ); + 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_filter->p_sys->i_remainder < p_filter->output.i_rate ) + while( p_sys->i_remainder < p_filter->fmt_out.audio.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_filter->output.i_rate, + 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->output.i_rate - - p_filter->p_sys->i_remainder, - p_filter->output.i_rate, + p_filter->fmt_out.audio.i_rate - + p_sys->i_remainder, + p_filter->fmt_out.audio.i_rate, 1, i_nb_channels ); #if 0 @@ -301,12 +249,12 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter, #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_filter->p_sys->i_remainder += p_filter->input.i_rate; + p_sys->i_remainder += p_filter->fmt_in.audio.i_rate; break; } } @@ -315,38 +263,37 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter, /* 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_filter->output.i_rate, p_filter->input.i_rate, + 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->output.i_rate - - p_filter->p_sys->i_remainder, - p_filter->output.i_rate, p_filter->input.i_rate, + 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_filter->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_filter->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_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->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->fmt_out.audio.i_rate ) { if( d_factor >= 1 ) @@ -356,32 +303,32 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter, /* 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_filter->output.i_rate, + 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->output.i_rate - - p_filter->p_sys->i_remainder, - p_filter->output.i_rate, + 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++ ) + 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 + 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_filter->p_sys->i_remainder += p_filter->input.i_rate; + p_sys->i_remainder += p_filter->fmt_in.audio.i_rate; break; } } @@ -390,64 +337,130 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter, /* 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_filter->output.i_rate, p_filter->input.i_rate, + 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->output.i_rate - - p_filter->p_sys->i_remainder, - p_filter->output.i_rate, p_filter->input.i_rate, + 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_filter->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_filter->p_sys->i_remainder -= p_filter->output.i_rate; + p_sys->i_remainder -= p_filter->fmt_out.audio.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) * - p_filter->input.i_bytes_per_frame ); + 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_size, - i_out * p_filter->input.i_bytes_per_frame ); -#endif - - /* Free the temp buffer */ -#ifndef HAVE_ALLOCA - free( p_in_orig ); + 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 = p_in_buf->start_date; + 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->end_date = aout_DateIncrement( &p_filter->p_sys->end_date, - p_out_buf->i_nb_samples ); - - p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples * + p_out_buf->i_buffer = p_out_buf->i_nb_samples * i_nb_channels * sizeof(int32_t); + return p_out_buf; +} + +/***************************************************************************** + * 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_CODEC_FL32 ) + { + 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_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_sys->p_buf = malloc( p_sys->i_buf_size ); + if( p_sys->p_buf == NULL ) + { + free( p_sys ); + return VLC_ENOMEM; + } + + p_sys->i_old_wing = 0; + p_sys->b_first = 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 ); } -void FilterFloatUP( float Imp[], float ImpD[], uint16_t Nwing, float *p_in, +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 ) { - float *Hp, *Hdp, *End; + const float *Hp, *Hdp, *End; float t, temp; uint32_t ui_linear_remainder; int i; @@ -486,12 +499,12 @@ void FilterFloatUP( float Imp[], float ImpD[], uint16_t Nwing, float *p_in, } } -void FilterFloatUD( float Imp[], float ImpD[], uint16_t Nwing, float *p_in, +void FilterFloatUD( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in, float *p_out, uint32_t ui_remainder, uint32_t ui_output_rate, uint32_t ui_input_rate, int16_t Inc, int i_nb_channels ) { - float *Hp, *Hdp, *End; + const float *Hp, *Hdp, *End; float t, temp; uint32_t ui_linear_remainder; int i, ui_counter = 0;