]> git.sesse.net Git - vlc/blobdiff - src/audio_output/filters.c
* modules/control/http.c: portability fix.
[vlc] / src / audio_output / filters.c
index be5f7aefda42a6f915566b19beb66258de0dd77c..93d12408546ed9725222388ce97fda37ea865231 100644 (file)
@@ -1,8 +1,8 @@
 /*****************************************************************************
  * filters.c : audio output filters management
  *****************************************************************************
- * Copyright (C) 2002 VideoLAN
- * $Id: filters.c,v 1.13 2002/11/11 22:27:00 gbazin Exp $
+ * Copyright (C) 2002-2004 VideoLAN
+ * $Id: filters.c,v 1.20 2004/03/03 20:39:52 gbazin Exp $
  *
  * Authors: Christophe Massiot <massiot@via.ecp.fr>
  *
@@ -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
@@ -30,7 +30,7 @@
 #include <vlc/vlc.h>
 
 #ifdef HAVE_ALLOCA_H
-#   include <alloca.h> 
+#   include <alloca.h>
 #endif
 
 #include "audio_output.h"
@@ -52,7 +52,7 @@ static aout_filter_t * FindFilter( aout_instance_t * p_aout,
     memcpy( &p_filter->input, p_input_format, sizeof(audio_sample_format_t) );
     memcpy( &p_filter->output, p_output_format,
             sizeof(audio_sample_format_t) );
-    p_filter->p_module = module_Need( p_filter, "audio filter", NULL );
+    p_filter->p_module = module_Need( p_filter, "audio filter", NULL, 0 );
     if ( p_filter->p_module == NULL )
     {
         vlc_object_detach( p_filter );
@@ -60,13 +60,13 @@ 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;
 }
 
 /*****************************************************************************
- * SplitConversion: split a conversion in two parts 
+ * SplitConversion: split a conversion in two parts
  *****************************************************************************
  * Returns the number of conversions required by the first part - 0 if only
  * one conversion was asked.
@@ -74,16 +74,18 @@ static aout_filter_t * FindFilter( aout_instance_t * p_aout,
  * developer passed SplitConversion( toto, titi, titi, ... ). That is legal.
  * SplitConversion( toto, titi, toto, ... ) isn't.
  *****************************************************************************/
-static int SplitConversion( aout_instance_t * p_aout,
-                             const audio_sample_format_t * p_input_format,
-                             const audio_sample_format_t * p_output_format,
-                             audio_sample_format_t * p_middle_format )
+static int SplitConversion( const audio_sample_format_t * p_input_format,
+                            const audio_sample_format_t * p_output_format,
+                            audio_sample_format_t * p_middle_format )
 {
     vlc_bool_t b_format =
              (p_input_format->i_format != p_output_format->i_format);
     vlc_bool_t b_rate = (p_input_format->i_rate != p_output_format->i_rate);
     vlc_bool_t b_channels =
-             (p_input_format->i_channels != p_output_format->i_channels);
+        (p_input_format->i_physical_channels
+          != p_output_format->i_physical_channels)
+     || (p_input_format->i_original_channels
+          != p_output_format->i_original_channels);
     int i_nb_conversions = b_format + b_rate + b_channels;
 
     if ( i_nb_conversions <= 1 ) return 0;
@@ -95,16 +97,22 @@ static int SplitConversion( aout_instance_t * p_aout,
         if ( !b_format || !b_channels )
         {
             p_middle_format->i_rate = p_input_format->i_rate;
+            aout_FormatPrepare( p_middle_format );
             return 1;
         }
 
         /* !b_rate */
-        p_middle_format->i_channels = p_input_format->i_channels;
+        p_middle_format->i_physical_channels
+             = p_input_format->i_physical_channels;
+        p_middle_format->i_original_channels
+             = p_input_format->i_original_channels;
+        aout_FormatPrepare( p_middle_format );
         return 1;
     }
 
     /* i_nb_conversion == 3 */
     p_middle_format->i_rate = p_input_format->i_rate;
+    aout_FormatPrepare( p_middle_format );
     return 2;
 }
 
@@ -144,7 +152,7 @@ int aout_FiltersCreatePipeline( aout_instance_t * p_aout,
     /* We'll have to split the conversion. We always do the downmixing
      * before the resampling, because the audio decoder can probably do it
      * for us. */
-    i_nb_conversions = SplitConversion( p_aout, p_input_format,
+    i_nb_conversions = SplitConversion( p_input_format,
                                         p_output_format, &temp_format );
     if ( !i_nb_conversions )
     {
@@ -157,10 +165,8 @@ int aout_FiltersCreatePipeline( aout_instance_t * p_aout,
     if ( pp_filters[0] == NULL && i_nb_conversions == 2 )
     {
         /* Try with only one conversion. */
-        SplitConversion( p_aout, p_input_format, &temp_format,
-                         &temp_format );
-        pp_filters[0] = FindFilter( p_aout, p_input_format,
-                                    &temp_format );
+        SplitConversion( p_input_format, &temp_format, &temp_format );
+        pp_filters[0] = FindFilter( p_aout, p_input_format, &temp_format );
     }
     if ( pp_filters[0] == NULL )
     {
@@ -176,9 +182,8 @@ int aout_FiltersCreatePipeline( aout_instance_t * p_aout,
     if ( pp_filters[1] == NULL )
     {
         /* Try to split the conversion. */
-        i_nb_conversions = SplitConversion( p_aout,
-                                    &pp_filters[0]->output,
-                                    p_output_format, &temp_format );
+        i_nb_conversions = SplitConversion( &pp_filters[0]->output,
+                                           p_output_format, &temp_format );
         if ( !i_nb_conversions )
         {
             vlc_object_detach( pp_filters[0] );
@@ -248,6 +253,8 @@ void aout_FiltersHintBuffers( aout_instance_t * p_aout,
 {
     int i;
 
+    (void)p_aout; /* unused */
+
     for ( i = i_nb_filters - 1; i >= 0; i-- )
     {
         aout_filter_t * p_filter = pp_filters[i];
@@ -294,13 +301,11 @@ void aout_FiltersPlay( aout_instance_t * p_aout,
         aout_filter_t * p_filter = pp_filters[i];
         aout_buffer_t * p_output_buffer;
 
-        /* We need this because resamplers can produce more samples than
-           (i_in_nb * p_filter->output.i_rate / p_filter->input.i_rate) */
-        int i_compensate_rounding = 2 * p_filter->input.i_rate
-            / p_filter->output.i_rate;
-
+        /* Resamplers can produce slightly more samples than (i_in_nb *
+         * p_filter->output.i_rate / p_filter->input.i_rate) so we need
+         * slightly bigger buffers. */
         aout_BufferAlloc( &p_filter->output_alloc,
-            ((mtime_t)(*pp_input_buffer)->i_nb_samples + i_compensate_rounding)
+            ((mtime_t)(*pp_input_buffer)->i_nb_samples + 2)
             * 1000000 / p_filter->input.i_rate,
             *pp_input_buffer, p_output_buffer );
         if ( p_output_buffer == NULL )