Changeset 1206810b7e59fcec539825b4fb9e8794152222a2

Timestamp:

04/03/03 20:28:39 (6 years ago)

Author:

Gildas Bazin <gbazin@videolan.org>

git-committer:

Gildas Bazin <gbazin@videolan.org> 1046806119 +0000

git-parent:

[eb1ba666bb488d9ed579789c3f4c9df802774c54]

git-author:

Gildas Bazin <gbazin@videolan.org> 1046806119 +0000

Message:

* modules/audio_filter/resampler/bandlimited.c: fix bug that was affecting

quality badly + some clean-up.
Changed the module priority so it is now the default resampler.

Files:

• modules/audio_filter/resampler/bandlimited.c (modified) (13 diffs)

modules/audio_filter/resampler/bandlimited.c

@@ -3 +3 @@
- *****************************************************************************
- * Copyright (C) 2002 VideoLAN
-1 2003/03/04 03:27:40
+2 2003/03/04 19:28:39
- *
- * Authors: Gildas Bazin <gbazin@netcourrier.com>
@@ -68 +68 @@
-
-    int i_old_rate;
-int
+double
-    int i_old_wing;
-
@@ -81 +81 @@
-vlc_module_begin();
-    set_description( _("audio filter for bandlimited interpolation resampling") );
-4
+20
-    set_callbacks( Create, Close );
-vlc_module_end();
@@ -159 +159 @@
-    int i_in_nb = p_in_buf->i_nb_samples;
-    int i_in, i_out = 0;
-
-
-
-
+
+
+
+
+
-
-    /* Check if we really need to run the resampler */
@@ -214 +215 @@
-
-#if 0
-i
+f
-             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 */
@@ -255 +249 @@
-    memset( p_out, 0, p_out_buf->i_size );
-
-
+
+
+
+
-    /* Account for increased filter gain when using factors less than 1 */
-
-
-
-
-
+
+
+
-
-    /* Apply the old rate until we have enough samples for the new one */
-
+
+
+
+
-    {
-        if( p_filter->p_sys->d_old_factor == 1 )
@@ -298 +296 @@
-                               1, i_nb_channels );
-
+#if 0
+                /* Normalize for unity filter gain */
+                for( i = 0; i < i_nb_channels; i++ )
+                {
+                    *(p_out+i) *= d_old_scale_factor;
+                }
+#endif
-            }
-            else
@@ -316 +321 @@
-            }
-
-#if 0
-            v *= LpScl;             /* Normalize for unity filter gain */
-#endif
-
-            p_out += i_nb_channels;
-            i_out++;
@@ -331 +332 @@
-
-    /* Apply the new rate for the rest of the samples */
+    /* TODO: Check we have enough future samples for the new rate */
-    if( i_in < i_in_nb - i_filter_wing )
-    {
@@ -360 +362 @@
-                               p_filter->output.i_rate,
-                               1, i_nb_channels );
+
+#if 0
+                /* Normalize for unity filter gain */
+                for( i = 0; i < i_nb_channels; i++ )
+                {
+                    *(p_out+i) *= d_old_scale_factor;
+                }
+#endif
-            }
-            else
@@ -378 +388 @@
-            }
-
-#if 0
-            v *= LpScl;             /* Normalize for unity filter gain */
-#endif
-
-            p_out += i_nb_channels;
-            i_out++;
@@ -402 +408 @@
-
-#if 0
-out size: %i, nb_sampl
+_out size: %i, nb byt
-             i_out * p_filter->input.i_bytes_per_frame );
-#endif
@@ -507 +513 @@
-          ((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++ )