X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=modules%2Faudio_filter%2Fresampler%2Fbandlimited.c;h=8167271fa1801ae2e4540794f8767afd9aa7fb23;hb=b3b215771577a03077dbed7e43b9f1e89515ed77;hp=2cbe087c665430078710d2797d630ae128c21679;hpb=2cb472dba008f7d877ffe6bae9c5575253365282;p=vlc diff --git a/modules/audio_filter/resampler/bandlimited.c b/modules/audio_filter/resampler/bandlimited.c index 2cbe087c66..8167271fa1 100644 --- a/modules/audio_filter/resampler/bandlimited.c +++ b/modules/audio_filter/resampler/bandlimited.c @@ -1,7 +1,7 @@ /***************************************************************************** * bandlimited.c : band-limited interpolation resampler ***************************************************************************** - * Copyright (C) 2002 the VideoLAN team + * 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 @@ -32,12 +32,17 @@ * 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" /***************************************************************************** @@ -48,12 +53,18 @@ 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 ); @@ -61,7 +72,7 @@ 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; @@ -72,19 +83,27 @@ struct aout_filter_sys_t unsigned int i_remainder; /* remainder of previous sample */ - audio_date_t end_date; + date_t end_date; + + bool b_first; + bool b_filter2; }; /***************************************************************************** * Module descriptor *****************************************************************************/ -vlc_module_begin(); - set_category( CAT_AUDIO ); - set_subcategory( SUBCAT_AUDIO_MISC ); - set_description( _("audio filter for band-limited 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 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 @@ -92,6 +111,7 @@ vlc_module_end(); static int Create( vlc_object_t *p_this ) { aout_filter_t * p_filter = (aout_filter_t *)p_this; + struct filter_sys_t * p_sys; double d_factor; int i_filter_wing; @@ -101,12 +121,12 @@ static int Create( vlc_object_t *p_this ) != 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') ) + || p_filter->input.i_format != VLC_CODEC_FL32 ) { return VLC_EGENERIC; } -#if !defined( SYS_DARWIN ) +#if !defined( __APPLE__ ) if( !config_GetInt( p_this, "hq-resampling" ) ) { return VLC_EGENERIC; @@ -114,35 +134,34 @@ static int Create( vlc_object_t *p_this ) #endif /* 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 ) - { - msg_Err( p_filter, "out of memory" ); + 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; + / 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_filter->p_sys->i_buf_size = aout_FormatNbChannels( &p_filter->input ) * + 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 ) + p_sys->p_buf = malloc( p_sys->i_buf_size ); + if( p_sys->p_buf == NULL ) { - msg_Err( p_filter, "out of memory" ); + free( p_sys ); return VLC_ENOMEM; } - p_filter->p_sys->i_old_wing = 0; + 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 = VLC_TRUE; + p_filter->b_in_place = true; return VLC_SUCCESS; } @@ -153,8 +172,9 @@ static int Create( vlc_object_t *p_this ) 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 ); + filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys; + free( p_sys->p_buf ); + free( p_sys ); } /***************************************************************************** @@ -163,48 +183,53 @@ static void Close( vlc_object_t * p_this ) 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; + 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 0 - int i; -#endif + + 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( p_aout->mixer.mixer.i_rate == p_filter->input.i_rate ) + if( i_out_rate == p_filter->input.i_rate ) { - if( //p_filter->b_continuity && /* What difference does it make ? :) */ - p_filter->p_sys->i_old_wing && - p_in_buf->i_size >= - p_in_buf->i_nb_bytes + p_filter->p_sys->i_old_wing * - p_filter->input.i_bytes_per_frame ) +#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 ) { /* 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_in_buf = block_Realloc( p_in_buf, + p_sys->i_old_wing * p_filter->input.i_bytes_per_frame, + p_in_buf->i_buffer ); + if( !p_in_buf ) + abort(); + 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_filter->p_sys->i_old_wing; + p_sys->i_old_wing; - p_out_buf->start_date = aout_DateGet( &p_filter->p_sys->end_date ); - p_out_buf->end_date = - aout_DateIncrement( &p_filter->p_sys->end_date, - p_out_buf->i_nb_samples ); + 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_nb_bytes = p_out_buf->i_nb_samples * + p_out_buf->i_buffer = p_out_buf->i_nb_samples * p_filter->input.i_bytes_per_frame; } - p_filter->b_continuity = VLC_FALSE; - p_filter->p_sys->i_old_wing = 0; +#endif + p_filter->b_continuity = false; + p_sys->i_old_wing = 0; return; } @@ -212,94 +237,85 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter, { /* 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_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->i_pts ); + 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_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->input.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->input.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 ); + 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 ); + 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->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_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, p_in, - p_filter->input.i_bytes_per_frame ); + 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_filter->p_sys->i_remainder < p_filter->output.i_rate ) + while( p_sys->i_remainder < p_filter->output.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_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_filter->p_sys->i_remainder, + p_sys->i_remainder, p_filter->output.i_rate, 1, i_nb_channels ); @@ -312,12 +328,12 @@ 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->input.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->input.i_rate; break; } } @@ -326,14 +342,14 @@ 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_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_filter->p_sys->i_remainder, + p_sys->i_remainder, p_filter->output.i_rate, p_filter->input.i_rate, 1, i_nb_channels ); } @@ -341,23 +357,23 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter, p_out += i_nb_channels; i_out++; - p_filter->p_sys->i_remainder += p_filter->input.i_rate; + p_sys->i_remainder += p_filter->input.i_rate; } p_in += i_nb_channels; - p_filter->p_sys->i_remainder -= p_filter->output.i_rate; + 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_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->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_filter->p_sys->i_remainder < p_filter->output.i_rate ) + while( p_sys->i_remainder < p_filter->output.i_rate ) { if( d_factor >= 1 ) @@ -367,7 +383,7 @@ 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_sys->i_remainder, p_filter->output.i_rate, -1, i_nb_channels ); @@ -375,24 +391,24 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter, 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_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++ ) + 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->input.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->input.i_rate; break; } } @@ -401,14 +417,14 @@ 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_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_filter->p_sys->i_remainder, + p_sys->i_remainder, p_filter->output.i_rate, p_filter->input.i_rate, 1, i_nb_channels ); } @@ -416,48 +432,182 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter, p_out += i_nb_channels; i_out++; - p_filter->p_sys->i_remainder += p_filter->input.i_rate; + p_sys->i_remainder += p_filter->input.i_rate; } p_in += i_nb_channels; - p_filter->p_sys->i_remainder -= p_filter->output.i_rate; + p_sys->i_remainder -= p_filter->output.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) * + 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, + msg_Err( p_filter, "p_out size: %i, nb bytes out: %i", p_out_buf->i_buffer, i_out * p_filter->input.i_bytes_per_frame ); #endif - /* Free the temp buffer */ -#ifndef HAVE_ALLOCA - free( p_in_orig ); -#endif - /* Finalize aout buffer */ p_out_buf->i_nb_samples = i_out; - p_out_buf->start_date = aout_DateGet( &p_filter->p_sys->end_date ); - p_out_buf->end_date = aout_DateIncrement( &p_filter->p_sys->end_date, - p_out_buf->i_nb_samples ); + 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_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); } -void FilterFloatUP( float Imp[], float ImpD[], uint16_t Nwing, float *p_in, +/***************************************************************************** + * 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_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_filter->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", + (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 ); +} + +/***************************************************************************** + * 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.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_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 ) { - float *Hp, *Hdp, *End; + const float *Hp, *Hdp, *End; float t, temp; uint32_t ui_linear_remainder; int i; @@ -496,12 +646,12 @@ 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;