Changeset a8af64458655e9f803cb2f754cfa04affaf944b8

Timestamp:

06/28/08 11:37:27 (2 months ago)

Author:

Rémi Denis-Courmont <rdenis@simphalempin.com>

git-committer:

Rémi Denis-Courmont <rdenis@simphalempin.com> 1214645847 +0300

git-parent:

[3c5f3a24ced2212136529fb8637072070d5ca58f]

git-author:

Rov Juvano <rovjuvano@users.sourceforge.net> 1214150967 -0400

Message:

Use the Scaletempo audio filter

Scaletempo maintains the audio pitch when playback rate != 1.0 (i.e.
no chipmunk effect). This fixes the pitch scaling caused by using the
resampler to handle playback rate.

Signed-off-by: Rémi Denis-Courmont <rdenis@simphalempin.com>

Files:

• include/vlc_aout.h (modified) (1 diff)
• src/audio_output/input.c (modified) (7 diffs)

include/vlc_aout.h

@@ -255 +255 @@
-    aout_filter_t *         pp_filters[AOUT_MAX_FILTERS];
-    int                     i_nb_filters;
+
+    aout_filter_t *         p_playback_rate_filter;
-
-    /* resamplers */

src/audio_output/input.c

@@ -402 +402 @@
-    p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
-
+    p_input->p_playback_rate_filter = NULL;
+    for( int i = 0; i < p_input->i_nb_filters; i++ )
+    {
+        aout_filter_t *p_filter = p_input->pp_filters[i];
+        if( strcmp( "scaletempo", p_filter->psz_object_name ) == 0 )
+        {
+          p_input->p_playback_rate_filter = p_filter;
+          break;
+        }
+    }
+    if( ! p_input->p_playback_rate_filter && p_input->i_nb_resamplers > 0 )
+    {
+        p_input->p_playback_rate_filter = p_input->pp_resamplers[0];
+    }
+
-    aout_FiltersHintBuffers( p_aout, p_input->pp_filters,
-                             p_input->i_nb_filters,
@@ -476 +491 @@
-    }
-
-i_nb_resamplers <= 0
+p_playback_rate_filter == NULL
-    {
-        inputDrop( p_aout, p_input, p_buffer );
@@ -503 +518 @@
-#endif
-
- by modifying resampler input rate
+, but keep drift correction
-    if( i_input_rate != p_input->i_last_input_rate )
-    {
-p_resamplers[0]
+_playback_rate_filter
-#define F(r,ir) ( INPUT_RATE_DEFAULT * (r) / (ir) )
-        const int i_delta = *pi_rate - F(p_input->input.i_rate,p_input->i_last_input_rate);
@@ -552 +567 @@
-    /* If the audio drift is too big then it's not worth trying to resample
-     * the audio. */
+    mtime_t i_pts_tolerance = 3 * AOUT_PTS_TOLERANCE * i_input_rate / INPUT_RATE_DEFAULT;
-    if ( start_date != 0 &&
-3 * AOUT_PTS_TOLERANCE
+i_pts_tolerance
-    {
-        msg_Warn( p_aout, "audio drift is too big (%"PRId64"), clearing out",
@@ -567 +583 @@
-    }
-    else if ( start_date != 0 &&
-3 * AOUT_PTS_TOLERANCE 
+i_pts_tolerance
-    {
-        msg_Warn( p_aout, "audio drift is too big (%"PRId64"), dropping buffer",
@@ -630 +646 @@
-        /* Check if everything is back to normal, in which case we can stop the
-         * resampling */
-
+
+
+
+
+
-        {
-            p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
@@ -734 +754 @@
-    if( p_input->i_nb_resamplers != 0 )
-    {
-
-
+
+
+
+
-        p_input->pp_resamplers[0]->b_continuity = false;
-    }