* modules/audio_filter/resampler/bandlimited.[ch]: new high-quality
   bandlimited interpolation resampler.
   It is not yet enabled by default because it requires a bit more works
   (not fully stable yet because of buffer overflows under certain conditions).

* src/audio_output/input.c: the resamplers are now always active (expcept
   in spdif mode).

* modules/audio_filter/resampler/*: modified the resampler to return the
   input buffer when no resampling is needed.

include/aout_internal.h

@@ -2,7 +2,7 @@
  * aout_internal.h : internal defines for audio output
  *****************************************************************************
  * Copyright (C) 2002 VideoLAN
- * $Id: aout_internal.h,v 1.38 2003/02/09 01:13:43 massiot Exp $
+ * $Id: aout_internal.h,v 1.39 2003/03/04 03:27:40 gbazin Exp $
  *
  * Authors: Christophe Massiot <massiot@via.ecp.fr>
  *
@@ -112,7 +112,7 @@ typedef struct aout_filter_t
                                          struct aout_buffer_t *,
                                          struct aout_buffer_t * );
     vlc_bool_t              b_in_place;
-    vlc_bool_t              b_reinit;
+    vlc_bool_t              b_continuity;
 } aout_filter_t;
 
 /*****************************************************************************

modules/audio_filter/resampler/Modules.am

 SOURCES_trivial_resampler = modules/audio_filter/resampler/trivial.c
 SOURCES_ugly_resampler = modules/audio_filter/resampler/ugly.c
 SOURCES_linear_resampler = modules/audio_filter/resampler/linear.c
-#SOURCES_fast = modules/audio_filter/resampler/fast.c
+SOURCES_bandlimited_resampler = modules/audio_filter/resampler/bandlimited.c \
+	modules/audio_filter/resampler/bandlimited.h

modules/audio_filter/resampler/bandlimited.c

+/*****************************************************************************
+ * bandlimited.c : bandlimited interpolation resampler
+ *****************************************************************************
+ * Copyright (C) 2002 VideoLAN
+ * $Id: bandlimited.c,v 1.1 2003/03/04 03:27:40 gbazin Exp $
+ *
+ * Authors: Gildas Bazin <gbazin@netcourrier.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * 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
+ * GNU General Public License for more details.
+ *
+ * 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.
+ *****************************************************************************/
+
+/*****************************************************************************
+ * Preamble:
+ *
+ * This implementation of the bandlimited interpolationis based on the
+ * following paper:
+ * http://ccrma-www.stanford.edu/~jos/resample/resample.html
+ *
+ * It uses a Kaiser-windowed sinc-function low-pass filter and the width of the
+ * filter is 13 samples.
+ *
+ *****************************************************************************/
+#include <stdlib.h>                                      /* malloc(), free() */
+#include <string.h>
+
+#include <vlc/vlc.h>
+#include "audio_output.h"
+#include "aout_internal.h"
+#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,
+                           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,
+                           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 );
+
+/*****************************************************************************
+ * Local structures
+ *****************************************************************************/
+struct aout_filter_sys_t
+{
+    int32_t *p_buf;                        /* this filter introduces a delay */
+    int i_buf_size;
+
+    int i_old_rate;
+    int d_old_factor;
+    int i_old_wing;
+
+    unsigned int i_remainder;                /* remainder of previous sample */
+
+    audio_date_t end_date;
+};
+
+/*****************************************************************************
+ * Module descriptor
+ *****************************************************************************/
+vlc_module_begin();
+    set_description( _("audio filter for bandlimited interpolation resampling") );
+    set_capability( "audio filter", 4 );
+    set_callbacks( Create, Close );
+vlc_module_end();
+
+/*****************************************************************************
+ * Create: allocate linear resampler
+ *****************************************************************************/
+static int Create( vlc_object_t *p_this )
+{
+    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 )
+    {
+        msg_Err( p_filter, "out of memory" );
+        return VLC_ENOMEM;
+    }
+
+    /* 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 = (double)p_aout->mixer.mixer.i_rate
+                        / p_filter->input.i_rate;
+    int i_filter_wing, i_left_over;
+
+    /* Check if we really need to run the resampler */
+    if( p_aout->mixer.mixer.i_rate == p_filter->input.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_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,
+                        i_nb_channels * 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;
+
+                aout_DateSet( &p_filter->p_sys->end_date,
+                              p_in_buf->start_date );
+
+                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 *
+                    i_nb_channels * sizeof(int32_t);
+            }
+        }
+        p_filter->b_continuity = VLC_FALSE;
+        return;
+    }
+
+    if( !p_filter->b_continuity )
+    {
+        /* 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 );
+
+        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;
+    }
+
+#if 0
+    msg_Err( p_filter, "old rate: %i, old factor: %i, 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 );
+#endif
+
+    /* Calculate the length of the filter wing */
+    d_factor = (double)p_aout->mixer.mixer.i_rate / p_filter->input.i_rate;
+    i_filter_wing = ((SMALL_FILTER_NMULT+1)/2.0) * __MAX(1.0,1.0/d_factor) + 1;
+
+    /* Check if we have enough buffered data to start with the new rate. */
+    i_left_over = i_filter_wing - p_filter->p_sys->i_old_wing;
+
+    /* 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;
+    }
+
+    /* Copy all our samples in p_in */
+    if( p_filter->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 );
+    }
+    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 );
+
+    /* Make sure the output buffer is reset */
+    memset( p_out, 0, p_out_buf->i_size );
+
+#if 0
+    /* Account for increased filter gain when using factors less than 1 */
+    if( d_factor < 1 )
+    {
+        LpScl = SMALL_FILTER_SCALE * d_factor + 0.5;
+    }
+#endif
+
+    /* Apply the old rate until we have enough samples for the new one */
+    for( i_in = p_filter->p_sys->i_old_wing; i_in < i_left_over; i_in++ )
+    {
+        if( p_filter->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 );          
+            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 )
+        {
+
+            if( p_filter->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,
+                               -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,
+                               1, i_nb_channels );
+
+            }
+            else
+            {
+                /* 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,
+                               -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,
+                               1, i_nb_channels );
+            }
+
+#if 0
+            v *= LpScl;             /* Normalize for unity filter gain */
+#endif
+
+            p_out += i_nb_channels;
+            i_out++;
+
+            p_filter->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;
+    }
+
+    /* 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;
+    }
+    for( ; i_in < i_in_nb - i_filter_wing; i_in++ )
+    {
+        while( p_filter->p_sys->i_remainder < p_filter->output.i_rate )
+        {
+
+            if( d_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,
+                               -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,
+                               1, i_nb_channels );
+            }
+            else
+            {
+                /* 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,
+                               -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,
+                               1, i_nb_channels );
+            }
+
+#if 0
+            v *= LpScl;             /* Normalize for unity filter gain */
+#endif
+
+            p_out += i_nb_channels;
+            i_out++;
+
+            p_filter->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;
+    }
+
+    /* Buffer i_filter_wing * 2 samples for next time */
+    if( p_filter->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 );
+    }
+
+#if 0
+    msg_Err( p_filter, "pout size: %i, nb_samples 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 );
+#endif
+
+    /* Finalize aout buffer */
+    p_out_buf->i_nb_samples = i_out;
+    p_out_buf->start_date = p_in_buf->start_date;
+
+    if( p_in_buf->start_date !=
+        aout_DateGet( &p_filter->p_sys->end_date ) )
+    {
+        aout_DateSet( &p_filter->p_sys->end_date, p_in_buf->start_date );
+    }
+
+    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 *
+        i_nb_channels * sizeof(int32_t);
+
+}
+
+void FilterFloatUP( float Imp[], 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;
+    float t, temp;
+    uint32_t ui_linear_remainder;
+    int i;
+
+    Hp = &Imp[(ui_remainder<<Nhc)/ui_output_rate];
+    Hdp = &ImpD[(ui_remainder<<Nhc)/ui_output_rate];
+
+    End = &Imp[Nwing];
+
+    ui_linear_remainder = (ui_remainder<<Nhc) -
+                            (ui_remainder<<Nhc)/ui_output_rate*ui_output_rate;
+
+    if (Inc == 1)               /* If doing right wing...              */
+    {                           /* ...drop extra coeff, so when Ph is  */
+        End--;                  /*    0.5, we don't do too many mult's */
+        if (ui_remainder == 0)  /* If the phase is zero...           */
+        {                       /* ...then we've already skipped the */
+            Hp += Npc;          /*    first sample, so we must also  */
+            Hdp += Npc;         /*    skip ahead in Imp[] and ImpD[] */
+        }
+    }
+
+    while (Hp < End) {
+        t = *Hp;                /* Get filter coeff */
+                                /* t is now interp'd filter coeff */
+        t += *Hdp * ui_linear_remainder / ui_output_rate / Npc;
+        for( i = 0; i < i_nb_channels; i++ )
+        {
+            temp = t;
+            temp *= *(p_in+i);  /* Mult coeff by input sample */
+            *(p_out+i) += temp; /* The filter output */
+        }
+        Hdp += Npc;             /* Filter coeff differences step */
+        Hp += Npc;              /* Filter coeff step */
+        p_in += (Inc * i_nb_channels); /* Input signal step */
+    }
+}
+
+void FilterFloatUD( float Imp[], 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;
+    float t, temp;
+    uint32_t ui_linear_remainder;
+    int i, ui_counter = 0;
+
+    Hp = Imp + (ui_remainder<<Nhc) / ui_input_rate;
+    Hdp = ImpD  + (ui_remainder<<Nhc) / ui_input_rate;
+
+    End = &Imp[Nwing];
+
+    if (Inc == 1)               /* If doing right wing...              */
+    {                           /* ...drop extra coeff, so when Ph is  */
+        End--;                  /*    0.5, we don't do too many mult's */
+        if (ui_remainder == 0)  /* If the phase is zero...           */
+        {                       /* ...then we've already skipped the */
+            Hp = Imp +          /* first sample, so we must also  */
+                  (ui_output_rate << Nhc) / ui_input_rate;
+            Hdp = ImpD +        /* skip ahead in Imp[] and ImpD[] */
+                  (ui_output_rate << Nhc) / ui_input_rate;
+            ui_counter++;
+        }
+    }
+
+    while (Hp < End) {
+        t = *Hp;                /* Get filter coeff */
+                                /* t is now interp'd filter coeff */
+        ui_linear_remainder =
+          ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) -
+          ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) /
+          ui_input_rate * ui_input_rate;
+          //(ui_remainder<<Nhc)* ui_output_rate/ui_input_rate -
+          //(ui_remainder<<Nhc) / ui_input_rate * ui_output_rate;
+        t += *Hdp * ui_linear_remainder / ui_input_rate / Npc;
+        for( i = 0; i < i_nb_channels; i++ )
+        {
+            temp = t;
+            temp *= *(p_in+i);  /* Mult coeff by input sample */
+            *(p_out+i) += temp; /* The filter output */
+        }
+
+        ui_counter++;
+
+        /* Filter coeff step */
+        Hp = Imp + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
+                    / ui_input_rate;
+        /* Filter coeff differences step */
+        Hdp = ImpD + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
+                     / ui_input_rate;
+
+        p_in += (Inc * i_nb_channels); /* Input signal step */
+    }
+}

modules/audio_filter/resampler/bandlimited.h

+/*****************************************************************************
+ * bandlimited.h : bandlimited interpolation resampler
+ *****************************************************************************
+ * Copyright (C) 2002 VideoLAN
+ * $Id: bandlimited.h,v 1.1 2003/03/04 03:27:40 gbazin Exp $
+ *
+ * Authors: Gildas Bazin <gbazin@netcourrier.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * 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
+ * GNU General Public License for more details.
+ *
+ * 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.
+ *****************************************************************************/
+
+/* Conversion constants */
+#define Nhc       8
+#define Npc      (1<<Nhc)
+
+/* Description of constants:
+ *
+ * Npc - is the number of look-up values available for the lowpass filter
+ *    between the beginning of its impulse response and the "cutoff time"
+ *    of the filter.  The cutoff time is defined as the reciprocal of the
+ *    lowpass-filter cut off frequence in Hz.  For example, if the
+ *    lowpass filter were a sinc function, Npc would be the index of the
+ *    impulse-response lookup-table corresponding to the first zero-
+ *    crossing of the sinc function.  (The inverse first zero-crossing
+ *    time of a sinc function equals its nominal cutoff frequency in Hz.)
+ *    Npc must be a power of 2 due to the details of the current
+ *    implementation. The default value of 512 is sufficiently high that
+ *    using linear interpolation to fill in between the table entries
+ *    gives approximately 16-bit accuracy in filter coefficients.
+ *
+ * Nhc - is log base 2 of Npc.
+ */
+
+#define SMALL_FILTER_NMULT ((int16_t)13)
+#define SMALL_FILTER_SCALE 13128 /* Unity-gain scale factor */
+#define SMALL_FILTER_NWING 1536 /* Filter table length */
+static float SMALL_FILTER_FLOAT_IMP[] /* Impulse response */ = {
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+0.000001,
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+0.000001,
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+0.000001,
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+0.000001,
+0.000001,
+0.000001,
+0.000001,
+0.000001,
+0.000001,
+0.000001
+};

modules/audio_filter/resampler/linear.c

@@ -2,7 +2,7 @@
  * linear.c : linear interpolation resampler
  *****************************************************************************
  * Copyright (C) 2002 VideoLAN
- * $Id: linear.c,v 1.9 2003/02/17 09:47:16 gbazin Exp $
+ * $Id: linear.c,v 1.10 2003/03/04 03:27:40 gbazin Exp $
  *
  * Authors: Gildas Bazin <gbazin@netcourrier.com>
  *          Sigmund Augdal <sigmunau@idi.ntnu.no>
@@ -94,7 +94,10 @@ static int Create( vlc_object_t *p_this )
     }
 
     p_filter->pf_do_work = DoWork;
-    p_filter->b_in_place = VLC_FALSE;
+
+    /* 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;
 }
@@ -115,18 +118,46 @@ 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 = (float*)p_in_buf->p_buffer;
-    float* p_out = (float*)p_out_buf->p_buffer;
-    float* p_prev_sample = (float*)p_filter->p_sys->p_prev_sample;
+    float *p_in, *p_out = (float *)p_out_buf->p_buffer;
+    float *p_prev_sample = (float *)p_filter->p_sys->p_prev_sample;
 
     int i_nb_channels = aout_FormatNbChannels( &p_filter->input );
     int i_in_nb = p_in_buf->i_nb_samples;
     int i_chan, i_in, i_out = 0;
 
+    /* Check if we really need to run the resampler */
+    if( p_aout->mixer.mixer.i_rate == p_filter->input.i_rate )
+    {
+        if( p_filter->b_continuity &&
+	    p_in_buf->i_size >=
+	      p_in_buf->i_nb_bytes + sizeof(float) * i_nb_channels )
+	{
+	    /* output the whole thing with the last sample from last time */
+	    memmove( ((float *)(p_in_buf->p_buffer)) + i_nb_channels,
+		     p_in_buf->p_buffer, p_in_buf->i_nb_bytes );
+	    memcpy( p_in_buf->p_buffer, p_prev_sample,
+		    i_nb_channels * sizeof(float) );
+	}
+        p_filter->b_continuity = VLC_FALSE;
+        return;
+    }
+
+#ifdef HAVE_ALLOCA
+    p_in = (float *)alloca( p_in_buf->i_nb_bytes );
+#else
+    p_in = (float *)malloc( p_in_buf->i_nb_bytes );
+#endif
+    if( p_in == NULL )
+    {
+        return;
+    }
+
+    p_aout->p_vlc->pf_memcpy( p_in, p_in_buf->p_buffer, p_in_buf->i_nb_bytes );
+
     /* Take care of the previous input sample (if any) */
-    if( p_filter->b_reinit )
+    if( !p_filter->b_continuity )
     {
-        p_filter->b_reinit = VLC_FALSE;
+        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 );
     }

modules/audio_filter/resampler/trivial.c

@@ -2,7 +2,7 @@
  * trivial.c : trivial resampler (skips samples or pads with zeroes)
  *****************************************************************************
  * Copyright (C) 2002 VideoLAN
- * $Id: trivial.c,v 1.10 2002/11/20 16:43:32 sam Exp $
+ * $Id: trivial.c,v 1.11 2003/03/04 03:27:40 gbazin Exp $
  *
  * Authors: Christophe Massiot <massiot@via.ecp.fr>
  *
@@ -68,7 +68,7 @@ static int Create( vlc_object_t *p_this )
     }
 
     p_filter->pf_do_work = DoWork;
-    p_filter->b_in_place = 1;
+    p_filter->b_in_place = VLC_TRUE;
 
     return 0;
 }
@@ -85,6 +85,12 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
     int i_sample_bytes = aout_FormatNbChannels( &p_filter->input )
                           * sizeof(int32_t);
 
+    /* Check if we really need to run the resampler */
+    if( p_aout->mixer.mixer.i_rate == p_filter->input.i_rate )
+    {
+        return;
+    }
+
     if ( p_out_buf != p_in_buf )
     {
         /* For whatever reason the buffer allocator decided to allocate

modules/audio_filter/resampler/ugly.c

@@ -2,7 +2,7 @@
  * ugly.c : ugly resampler (changes pitch)
  *****************************************************************************
  * Copyright (C) 2002 VideoLAN
- * $Id: ugly.c,v 1.8 2002/12/06 16:34:04 sam Exp $
+ * $Id: ugly.c,v 1.9 2003/03/04 03:27:40 gbazin Exp $
  *
  * Authors: Samuel Hocevar <sam@zoy.org>
  *
@@ -68,7 +68,10 @@ static int Create( vlc_object_t *p_this )
     }
 
     p_filter->pf_do_work = DoWork;
-    p_filter->b_in_place = VLC_FALSE;
+
+    /* 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;
 }
@@ -79,8 +82,7 @@ static int Create( 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 )
 {
-    int32_t* p_in = (int32_t*)p_in_buf->p_buffer;
-    int32_t* p_out = (int32_t*)p_out_buf->p_buffer;
+    int32_t *p_in, *p_out = (int32_t*)p_out_buf->p_buffer;
 
     unsigned int i_nb_channels = aout_FormatNbChannels( &p_filter->input );
     unsigned int i_in_nb = p_in_buf->i_nb_samples;
@@ -89,6 +91,24 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
     unsigned int i_sample_bytes = i_nb_channels * sizeof(int32_t);
     unsigned int i_out, i_chan, i_remainder = 0;
 
+    /* Check if we really need to run the resampler */
+    if( p_aout->mixer.mixer.i_rate == p_filter->input.i_rate )
+    {
+        return;
+    }
+
+#ifdef HAVE_ALLOCA
+    p_in = (int32_t *)alloca( p_in_buf->i_nb_bytes );
+#else
+    p_in = (int32_t *)malloc( p_in_buf->i_nb_bytes );
+#endif
+    if( p_in == NULL )
+    {
+        return;
+    }
+
+    p_aout->p_vlc->pf_memcpy( p_in, p_in_buf->p_buffer, p_in_buf->i_nb_bytes );
+
     for( i_out = i_out_nb ; i_out-- ; )
     {
         for( i_chan = i_nb_channels ; i_chan ; )

src/audio_output/filters.c

@@ -2,7 +2,7 @@
  * filters.c : audio output filters management
  *****************************************************************************
  * Copyright (C) 2002 VideoLAN
- * $Id: filters.c,v 1.17 2003/01/21 10:29:12 massiot Exp $
+ * $Id: filters.c,v 1.18 2003/03/04 03:27:40 gbazin Exp $
  *
  * Authors: Christophe Massiot <massiot@via.ecp.fr>
  *
@@ -60,7 +60,7 @@ static aout_filter_t * FindFilter( aout_instance_t * p_aout,
         return NULL;
     }
 
-    p_filter->b_reinit = VLC_TRUE;
+    p_filter->b_continuity = VLC_FALSE;
 
     return p_filter;
 }

src/audio_output/input.c

@@ -2,7 +2,7 @@
  * input.c : internal management of input streams for the audio output
  *****************************************************************************
  * Copyright (C) 2002 VideoLAN
- * $Id: input.c,v 1.32 2003/02/21 22:59:38 gbazin Exp $
+ * $Id: input.c,v 1.33 2003/03/04 03:27:40 gbazin Exp $
  *
  * Authors: Christophe Massiot <massiot@via.ecp.fr>
  *
@@ -130,7 +130,7 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input )
         }
 
         /* success */
-        p_headphone_filter->b_reinit = VLC_TRUE;
+        p_headphone_filter->b_continuity = VLC_FALSE;
         p_input->pp_filters[p_input->i_nb_filters++] = p_headphone_filter;
     }
 
@@ -248,7 +248,7 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
         if ( p_input->i_nb_resamplers != 0 )
         {
             p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate;
-            p_input->pp_resamplers[0]->b_reinit = VLC_TRUE;
+            p_input->pp_resamplers[0]->b_continuity = VLC_FALSE;
         }
         start_date = 0;
     }
@@ -260,7 +260,12 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
         msg_Warn( p_aout, "PTS is out of range ("I64Fd"), dropping buffer",
                   mdate() - p_buffer->start_date );
         aout_BufferFree( p_buffer );
-
+        p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
+        if ( p_input->i_nb_resamplers != 0 )
+        {
+            p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate;
+            p_input->pp_resamplers[0]->b_continuity = VLC_FALSE;
+        }
         return 0;
     }
 
@@ -353,8 +358,7 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
     p_buffer->start_date = start_date;
 
     /* Actually run the resampler now. */
-    if ( p_input->i_nb_resamplers > 0 && p_aout->mixer.mixer.i_rate !=
-         p_input->pp_resamplers[0]->input.i_rate )
+    if ( p_input->i_nb_resamplers > 0 )
     {
         aout_FiltersPlay( p_aout, p_input->pp_resamplers,
                           p_input->i_nb_resamplers,