git.sesse.net Git - vlc/blob - modules/audio_filter/channel_mixer/trivial.c

   1 /*****************************************************************************
   2  * trivial.c : trivial channel mixer plug-in (drops unwanted channels)
   3  *****************************************************************************
   4  * Copyright (C) 2002, 2006 VLC authors and VideoLAN
   5  * $Id$
   6  *
   7  * Authors: Christophe Massiot <massiot@via.ecp.fr>
   8  *
   9  * This program is free software; you can redistribute it and/or modify it
  10  * under the terms of the GNU Lesser General Public License as published by
  11  * the Free Software Foundation; either version 2.1 of the License, or
  12  * (at your option) any later version.
  13  *
  14  * This program is distributed in the hope that it will be useful,
  15  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  16  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
  17  * GNU Lesser General Public License for more details.
  18  *
  19  * You should have received a copy of the GNU Lesser General Public License
  20  * along with this program; if not, write to the Free Software Foundation,
  21  * Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
  22  *****************************************************************************/
  23
  24 /*****************************************************************************
  25  * Preamble
  26  *****************************************************************************/
  27
  28 #ifdef HAVE_CONFIG_H
  29 # include "config.h"
  30 #endif
  31
  32 #include <vlc_common.h>
  33 #include <vlc_plugin.h>
  34 #include <vlc_aout.h>
  35 #include <vlc_filter.h>
  36
  37 /*****************************************************************************
  38  * Local prototypes
  39  *****************************************************************************/
  40 static int  Create    ( vlc_object_t * );
  41
  42 static block_t *DoWork( filter_t *, block_t * );
  43
  44 /*****************************************************************************
  45  * Module descriptor
  46  *****************************************************************************/
  47 vlc_module_begin ()
  48     set_description( N_("Audio filter for trivial channel mixing") )
  49     set_capability( "audio converter", 1 )
  50     set_category( CAT_AUDIO )
  51     set_subcategory( SUBCAT_AUDIO_MISC )
  52     set_callbacks( Create, NULL )
  53 vlc_module_end ()
  54
  55 /*****************************************************************************
  56  * Create: allocate trivial channel mixer
  57  *****************************************************************************/
  58 static int Create( vlc_object_t *p_this )
  59 {
  60     filter_t * p_filter = (filter_t *)p_this;
  61
  62     if ( (p_filter->fmt_in.audio.i_physical_channels
  63            == p_filter->fmt_out.audio.i_physical_channels
  64            && p_filter->fmt_in.audio.i_original_channels
  65                == p_filter->fmt_out.audio.i_original_channels)
  66           || p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format
  67           || p_filter->fmt_in.audio.i_rate != p_filter->fmt_out.audio.i_rate
  68           || p_filter->fmt_in.audio.i_format != VLC_CODEC_FL32 )
  69     {
  70         return VLC_EGENERIC;
  71     }
  72
  73     p_filter->pf_audio_filter = DoWork;
  74     return VLC_SUCCESS;
  75 }
  76
  77 /*****************************************************************************
  78  * SparseCopy: trivially downmix or upmix a buffer
  79  *****************************************************************************/
  80 static void SparseCopy( float * p_dest, const float * p_src, size_t i_len,
  81                         int i_output_stride, int i_input_stride )
  82 {
  83     int i;
  84     for ( i = i_len; i--; )
  85     {
  86         int j;
  87         for ( j = 0; j < i_output_stride; j++ )
  88         {
  89             p_dest[j] = p_src[j % i_input_stride];
  90         }
  91         p_src += i_input_stride;
  92         p_dest += i_output_stride;
  93     }
  94 }
  95
  96 /*****************************************************************************
  97  * DoWork: convert a buffer
  98  *****************************************************************************/
  99 static block_t *DoWork( filter_t * p_filter, block_t * p_in_buf )
 100 {
 101     int i_input_nb = aout_FormatNbChannels( &p_filter->fmt_in.audio );
 102     int i_output_nb = aout_FormatNbChannels( &p_filter->fmt_out.audio );
 103
 104     block_t *p_out_buf;
 105     if( i_input_nb >= i_output_nb )
 106     {
 107         p_out_buf = p_in_buf; /* mix in place */
 108         p_out_buf->i_buffer = p_in_buf->i_buffer / i_input_nb * i_output_nb;
 109     }
 110     else
 111     {
 112         p_out_buf = block_Alloc(
 113                               p_in_buf->i_buffer / i_input_nb * i_output_nb );
 114         if( !p_out_buf )
 115             goto out;
 116         p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
 117         p_out_buf->i_dts        = p_in_buf->i_dts;
 118         p_out_buf->i_pts        = p_in_buf->i_pts;
 119         p_out_buf->i_length     = p_in_buf->i_length;
 120     }
 121
 122     float * p_dest = (float *)p_out_buf->p_buffer;
 123     const float * p_src = (float *)p_in_buf->p_buffer;
 124
 125     if ( (p_filter->fmt_out.audio.i_original_channels & AOUT_CHAN_PHYSMASK)
 126                 != (p_filter->fmt_in.audio.i_original_channels & AOUT_CHAN_PHYSMASK)
 127            && (p_filter->fmt_in.audio.i_original_channels & AOUT_CHAN_PHYSMASK)
 128                 == (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT) )
 129     {
 130         int i;
 131         /* This is a bit special. */
 132         if ( !(p_filter->fmt_out.audio.i_original_channels & AOUT_CHAN_LEFT) )
 133         {
 134             p_src++;
 135         }
 136         if ( p_filter->fmt_out.audio.i_physical_channels == AOUT_CHAN_CENTER )
 137         {
 138             /* Mono mode */
 139             for ( i = p_in_buf->i_nb_samples; i--; )
 140             {
 141                 *p_dest = *p_src;
 142                 p_dest++;
 143                 p_src += 2;
 144             }
 145         }
 146         else
 147         {
 148             /* Fake-stereo mode */
 149             for ( i = p_in_buf->i_nb_samples; i--; )
 150             {
 151                 *p_dest = *p_src;
 152                 p_dest++;
 153                 *p_dest = *p_src;
 154                 p_dest++;
 155                 p_src += 2;
 156             }
 157         }
 158     }
 159     else if ( p_filter->fmt_out.audio.i_original_channels
 160                                     & AOUT_CHAN_REVERSESTEREO )
 161     {
 162         /* Reverse-stereo mode */
 163         int i;
 164         for ( i = p_in_buf->i_nb_samples; i--; )
 165         {
 166             float i_tmp = p_src[0];
 167             p_dest[0] = p_src[1];
 168             p_dest[1] = i_tmp;
 169
 170             p_dest += 2;
 171             p_src += 2;
 172         }
 173     }
 174     else
 175     {
 176         SparseCopy( p_dest, p_src, p_in_buf->i_nb_samples, i_output_nb,
 177                     i_input_nb );
 178     }
 179 out:
 180     if( p_in_buf != p_out_buf )
 181         block_Release( p_in_buf );
 182     return p_out_buf;
 183 }
 184