Added fixed32tos16 filter based upon libmad downscale functionality.

modules/audio_filter/converter/fixed32tos16.c

+/*****************************************************************************
+ * fixed32tos16.c : converter from fised32 to signed 16 bits integer
+ *****************************************************************************
+ * Copyright (C) 2002 VideoLAN
+ * $Id: fixed32tos16.c,v 1.1 2002/08/12 20:34:56 jpsaman Exp $
+ *
+ * Authors: Jean-Paul Saman <jpsaman@wxs.nl>
+ *
+ * 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
+ *****************************************************************************/
+#include <errno.h>
+#include <stdlib.h>                                      /* malloc(), free() */
+#include <string.h>
+
+#include <vlc/vlc.h>
+#include "audio_output.h"
+#include "aout_internal.h"
+
+/*****************************************************************************
+ * Local prototypes
+ *****************************************************************************/
+static int  Create    ( vlc_object_t * );
+
+static void DoWork    ( aout_instance_t *, aout_filter_t *, aout_buffer_t *,
+                        aout_buffer_t * );
+
+/*****************************************************************************
+ * Module descriptor
+ *****************************************************************************/
+vlc_module_begin();
+    set_description( _("aout filter for fixed32->s16 conversion") );
+    set_capability( "audio filter", 10 );
+    set_callbacks( Create, NULL );
+vlc_module_end();
+
+/*****************************************************************************
+ * Create: allocate trivial mixer
+ *****************************************************************************
+ * This function allocates and initializes a Crop vout method.
+ *****************************************************************************/
+static int Create( vlc_object_t *p_this )
+{
+    aout_filter_t * p_filter = (aout_filter_t *)p_this;
+
+    if ( p_filter->input.i_format != AOUT_FMT_FIXED32
+          && p_filter->output.i_format != AOUT_FMT_S16_NE )
+    {
+        return -1;
+    }
+
+    if ( p_filter->input.i_rate != p_filter->output.i_rate
+          || p_filter->input.i_channels != p_filter->output.i_channels )
+    {
+        return -1;
+    }
+
+    p_filter->pf_do_work = DoWork;
+    p_filter->b_in_place = 1;
+
+    printf( "fixed32tos16 spawned.\n" );
+    return 0;
+}
+
+/*****************************************************************************
+ * support routines borrowed from mpg321 (file: mad.c), which is distributed
+ * under GPL license
+ *
+ * mpg321 was written by Joe Drew <drew@debian.org>, and based upon 'plaympeg'
+ * from the smpeg sources, which was written by various people from Loki Software
+ * (http://www.lokigames.com).
+ *
+ * It also incorporates some source from mad, written by Robert Leslie
+ *****************************************************************************/
+
+/* The following two routines and data structure are from the ever-brilliant
+     Rob Leslie.
+*/
+
+#define VLC_F_FRACBITS  28
+
+# if VLC_F_FRACBITS == 28
+#  define VLC_F(x)		((vlc_fixed_t) (x##L))
+# endif
+
+# define VLC_F_ONE		VLC_F(0x10000000)
+
+struct audio_dither {
+    vlc_fixed_t error[3];
+    vlc_fixed_t random;
+};
+
+/********************************************************************
+ * NAME:                prng()
+ * DESCRIPTION: 32-bit pseudo-random number generator
+ ********************************************************************/
+static inline unsigned long prng(unsigned long state)
+{
+    return (state * 0x0019660dL + 0x3c6ef35fL) & 0xffffffffL;
+}
+
+/********************************************************************
+ * NAME:        mpg321_s24_to_s16_pcm()
+ * DESCRIPTION: generic linear sample quantize and dither routine
+ ********************************************************************/
+static inline s16 mpg321_s24_to_s16_pcm(unsigned int bits, vlc_fixed_t sample,
+                                    struct audio_dither *dither)
+{
+    unsigned int scalebits;
+    vlc_fixed_t output, mask, random;
+
+    enum {
+        MIN = -VLC_F_ONE,
+        MAX = VLC_F_ONE - 1
+    };
+
+    /* noise shape */
+    sample += dither->error[0] - dither->error[1] + dither->error[2];
+
+    dither->error[2] = dither->error[1];
+    dither->error[1] = dither->error[0] / 2;
+
+    /* bias */
+    output = sample + (1L << (VLC_F_FRACBITS + 1 - bits - 1));
+
+    scalebits = VLC_F_FRACBITS + 1 - bits;
+    mask = (1L << scalebits) - 1;
+
+    /* dither */
+    random    = prng(dither->random);
+    output += (random & mask) - (dither->random & mask);
+
+    dither->random = random;
+
+    /* clip */
+    if (output > MAX) {
+        output = MAX;
+
+        if (sample > MAX)
+            sample = MAX;
+    }
+    else if (output < MIN) {
+        output = MIN;
+
+        if (sample < MIN)
+            sample = MIN;
+    }
+
+    /* quantize */
+    output &= ~mask;
+
+    /* error feedback */
+    dither->error[0] = sample - output;
+
+    /* scale */
+    return output >> scalebits;
+}
+
+/*****************************************************************************
+ * s24_to_s16_pcm: Scale a 24 bit pcm sample to a 16 bit pcm sample.
+ *****************************************************************************/
+static inline s16 s24_to_s16_pcm(vlc_fixed_t sample)
+{
+  /* round */
+  sample += (1L << (VLC_F_FRACBITS - 16));
+
+  /* clip */
+  if (sample >= VLC_F_ONE)
+    sample = VLC_F_ONE - 1;
+  else if (sample < -VLC_F_ONE)
+    sample = -VLC_F_ONE;
+
+  /* quantize */
+  return (sample >> (VLC_F_FRACBITS + 1 - 16));
+}
+
+/*****************************************************************************
+ * 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 )
+{
+    int i;
+    vlc_fixed_t * p_in = (vlc_fixed_t *)p_in_buf->p_buffer;
+    s16 * p_out = (s16 *)p_out_buf->p_buffer;
+    s16 sample;
+//    static struct audio_dither dither;
+
+    for ( i = 0; i < p_in_buf->i_nb_samples * p_filter->input.i_channels; i++ )
+    {
+        /* Accurate scaling */
+//        p_out = mpg321_s24_to_s16_pcm(16, *p_in++, &dither);
+        /* Fast Scaling */
+        sample = s24_to_s16_pcm(*p_in++);
+
+#ifndef WORDS_BIGENDIAN
+        *p_out++ = (s16) (sample >> 0);
+        *p_out++ = (s16) (sample >> 8);
+#else
+        *p_out++ = (s16) (sample >> 8);
+        *p_out++ = (s16) (sample >> 0);
+#endif
+//        p_in++; p_out++;
+    }
+    p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
+}