* Makefile.old: Fixed BeOS compilation.

* ALL: Fewer overhead when resampling.

diff --git a/Makefile.old b/Makefile.old
index 436b1fe0cf61e2f140c8fbbd7b1a88ec59078765..93cef4cd0e704f2f56d227eb282f9c0ff2a4747c 100644 (file)
--- a/Makefile.old
+++ b/Makefile.old
@@ -61,9 +61,9 @@ LIBVLC_OBJ += src/misc/win32_specific.o
 RESOURCE_OBJ :=        share/vlc_win32_rc.o
 endif
 
-LIBVLC_OBJ += $(CPP_OBJ) $(M_OBJ) $(BUILTIN_OBJ)
+C_OBJ := $(VLC_OBJ) $(LIBVLC_OBJ) $(BUILTIN_OBJ)
 
-C_OBJ := $(VLC_OBJ) $(LIBVLC_OBJ)
+LIBVLC_OBJ += $(CPP_OBJ) $(M_OBJ) $(BUILTIN_OBJ)
 
 VLC_OBJ += $(RESOURCE_OBJ)
 

diff --git a/include/aout_internal.h b/include/aout_internal.h
index a4e9cb61fdf274bf3a20f1e526b4b08e52c71e39..a1210fbb1a23c01663236c63bc53b95f85ebc286 100644 (file)
--- a/include/aout_internal.h
+++ b/include/aout_internal.h
@@ -2,7 +2,7 @@
  * aout_internal.h : internal defines for audio output
  *****************************************************************************
  * Copyright (C) 2002 VideoLAN
- * $Id: aout_internal.h,v 1.21 2002/09/26 22:40:18 massiot Exp $
+ * $Id: aout_internal.h,v 1.22 2002/09/27 23:38:04 massiot Exp $
  *
  * Authors: Christophe Massiot <massiot@via.ecp.fr>
  *
@@ -144,6 +144,10 @@ struct aout_input_t
     aout_filter_t *         pp_filters[AOUT_MAX_FILTERS];
     int                     i_nb_filters;
 
+    /* resamplers */
+    aout_filter_t *         pp_resamplers[AOUT_MAX_FILTERS];
+    int                     i_nb_resamplers;
+
     aout_fifo_t             fifo;
 
     /* Mixer information */

diff --git a/include/vlc_config.h b/include/vlc_config.h
index e04f4c2e0aa9bc4362a1f8cb06c50577a98eca18..d38c27fa6e3cd3a4544973e3cb6719a9c2ab7298 100644 (file)
--- a/include/vlc_config.h
+++ b/include/vlc_config.h
@@ -147,6 +147,9 @@
  * time, without resampling */
 #define AOUT_PTS_TOLERANCE              (mtime_t)(.02*CLOCK_FREQ)
 
+/* Max acceptable resampling (in %) */
+#define AOUT_MAX_RESAMPLING             10
+
 /*****************************************************************************
  * Video configuration
  *****************************************************************************/

diff --git a/modules/audio_filter/converter/s16tofloat32swab.c b/modules/audio_filter/converter/s16tofloat32swab.c
index b5ac29d08f3d43a1d3e1345882473de41c6dcdaa..cc25b540f8ffd48270a0808cef97c46d03b32e3c 100644 (file)
--- a/modules/audio_filter/converter/s16tofloat32swab.c
+++ b/modules/audio_filter/converter/s16tofloat32swab.c
@@ -3,7 +3,7 @@
  *                      with endianness change
  *****************************************************************************
  * Copyright (C) 2002 VideoLAN
- * $Id: s16tofloat32swab.c,v 1.4 2002/09/26 22:40:19 massiot Exp $
+ * $Id: s16tofloat32swab.c,v 1.5 2002/09/27 23:38:04 massiot Exp $
  *
  * Authors: Samuel Hocevar <sam@zoy.org>
  *          Henri Fallon <henri@videolan.org>
@@ -103,7 +103,7 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
 
 #ifdef HAVE_SWAB
     s16 * p_swabbed = alloca( i * sizeof(s16) );
-    swab( p_in_buf->p_buffer, p_swabbed, i * sizeof(s16) );
+    swab( p_in_buf->p_buffer, (void *)p_swabbed, i * sizeof(s16) );
     p_in = p_swabbed + i - 1;
 #else
     byte_t p_tmp[2];

diff --git a/src/audio_output/input.c b/src/audio_output/input.c
index 18b42ef15fcc235c02bbcc5e10ea6d30b5b25628..b40367b0eaaa7238210c4f8cae51d4da960d3158 100644 (file)
--- a/src/audio_output/input.c
+++ b/src/audio_output/input.c
@@ -2,7 +2,7 @@
  * input.c : internal management of input streams for the audio output
  *****************************************************************************
  * Copyright (C) 2002 VideoLAN
- * $Id: input.c,v 1.13 2002/09/26 22:40:25 massiot Exp $
+ * $Id: input.c,v 1.14 2002/09/27 23:38:04 massiot Exp $
  *
  * Authors: Christophe Massiot <massiot@via.ecp.fr>
  *
@@ -41,14 +41,19 @@
  *****************************************************************************/
 int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input )
 {
+    audio_sample_format_t intermediate_format;
+
     /* Prepare FIFO. */
     aout_FifoInit( p_aout, &p_input->fifo, p_aout->mixer.mixer.i_rate );
     p_input->p_first_byte_to_mix = NULL;
 
     /* Create filters. */
+    memcpy( &intermediate_format, &p_aout->mixer.mixer,
+            sizeof(audio_sample_format_t) );
+    intermediate_format.i_rate = p_input->input.i_rate;
     if ( aout_FiltersCreatePipeline( p_aout, p_input->pp_filters,
                                      &p_input->i_nb_filters, &p_input->input,
-                                     &p_aout->mixer.mixer ) < 0 )
+                                     &intermediate_format ) < 0 )
     {
         msg_Err( p_aout, "couldn't set an input pipeline" );
 
@@ -58,10 +63,37 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input )
         return -1;
     }
 
+    /* Create resamplers. */
+    intermediate_format.i_rate = (p_input->input.i_rate
+                             * (100 + AOUT_MAX_RESAMPLING)) / 100;
+    if ( intermediate_format.i_rate == p_aout->mixer.mixer.i_rate )
+    {
+        /* Just in case... */
+        intermediate_format.i_rate++;
+    }
+    if ( aout_FiltersCreatePipeline( p_aout, p_input->pp_resamplers,
+                                     &p_input->i_nb_resamplers,
+                                     &intermediate_format,
+                                     &p_aout->mixer.mixer ) < 0 )
+    {
+        msg_Err( p_aout, "couldn't set a resampler pipeline" );
+
+        aout_FiltersDestroyPipeline( p_aout, p_input->pp_filters,
+                                     p_input->i_nb_filters );
+        aout_FifoDestroy( p_aout, &p_input->fifo );
+        p_input->b_error = 1;
+
+        return -1;
+    }
+
     /* Prepare hints for the buffer allocator. */
     p_input->input_alloc.i_alloc_type = AOUT_ALLOC_HEAP;
     p_input->input_alloc.i_bytes_per_sec = -1;
 
+    aout_FiltersHintBuffers( p_aout, p_input->pp_resamplers,
+                             p_input->i_nb_resamplers,
+                             &p_input->input_alloc );
+
     aout_FiltersHintBuffers( p_aout, p_input->pp_filters,
                              p_input->i_nb_filters,
                              &p_input->input_alloc );
@@ -91,6 +123,8 @@ int aout_InputDelete( aout_instance_t * p_aout, aout_input_t * p_input )
 
     aout_FiltersDestroyPipeline( p_aout, p_input->pp_filters,
                                  p_input->i_nb_filters );
+    aout_FiltersDestroyPipeline( p_aout, p_input->pp_resamplers,
+                                 p_input->i_nb_resamplers );
     aout_FifoDestroy( p_aout, &p_input->fifo );
 
     return 0;
@@ -140,6 +174,10 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
 
     if ( start_date == 0 ) start_date = p_buffer->start_date;
 
+    /* Run pre-filters. */
+    aout_FiltersPlay( p_aout, p_input->pp_filters, p_input->i_nb_filters,
+                      &p_buffer );
+
     if ( start_date < p_buffer->start_date - AOUT_PTS_TOLERANCE
           || start_date > p_buffer->start_date + AOUT_PTS_TOLERANCE )
     {
@@ -150,81 +188,45 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
          *    synchronization
          * Solution : resample the buffer to avoid a scratch.
          */
-        audio_sample_format_t new_input;
-        int i_ratio, i_nb_filters;
+        int i_ratio;
         mtime_t old_duration;
-        aout_filter_t * pp_filters[AOUT_MAX_FILTERS];
-        aout_buffer_t * p_new_buffer;
-        aout_alloc_t dummy_alloc;
         mtime_t drift = p_buffer->start_date - start_date;
 
         msg_Warn( p_aout, "buffer is %lld %s, resampling",
                          drift > 0 ? drift : -drift,
                          drift > 0 ? "in advance" : "late" );
-        memcpy( &new_input, &p_input->input,
-                sizeof(audio_sample_format_t) );
         old_duration = p_buffer->end_date - p_buffer->start_date;
         duration = p_buffer->end_date - start_date;
-        i_ratio = duration * 100 / old_duration;
+        i_ratio = (duration * 100) / old_duration;
         /* If the ratio is too != 100, the sound quality will be awful. */
-        if ( i_ratio < 66 /* % */ )
+        if ( i_ratio < 100 - AOUT_MAX_RESAMPLING /* % */ )
         {
-            duration = old_duration * 66 / 100;
+            duration = (old_duration * (100 - AOUT_MAX_RESAMPLING)) / 100;
         }
-        if ( i_ratio > 150 /* % */ )
+        if ( i_ratio > 100 + AOUT_MAX_RESAMPLING /* % */ )
         {
-            duration = old_duration * 150 / 100;
+            duration = (old_duration * (100 + AOUT_MAX_RESAMPLING)) / 100;
         }
-        new_input.i_rate = new_input.i_rate * old_duration / duration;
-        aout_FormatPrepare( &new_input );
-
-        if ( aout_FiltersCreatePipeline( p_aout, pp_filters,
-                                         &i_nb_filters, &new_input,
-                                         &p_aout->mixer.mixer ) < 0 )
-        {
-            msg_Err( p_aout, "couldn't set an input pipeline for resampling" );
-            vlc_mutex_lock( &p_aout->mixer_lock );
-            aout_FifoSet( p_aout, &p_input->fifo, 0 );
-            vlc_mutex_unlock( &p_aout->mixer_lock );
-            aout_BufferFree( p_buffer );
-
-            vlc_mutex_unlock( &p_input->lock );
-            return 0;
-        }
-
-        dummy_alloc.i_alloc_type = AOUT_ALLOC_HEAP;
-        dummy_alloc.i_bytes_per_sec = -1;
-        aout_FiltersHintBuffers( p_aout, pp_filters, i_nb_filters,
-                                 &dummy_alloc );
-        dummy_alloc.i_bytes_per_sec = __MAX(
-                                    dummy_alloc.i_bytes_per_sec,
-                                    new_input.i_bytes_per_frame
-                                     * new_input.i_rate
-                                     / new_input.i_frame_length );
-        dummy_alloc.i_alloc_type = AOUT_ALLOC_HEAP;
-
-        aout_BufferAlloc( &dummy_alloc, duration, NULL, p_new_buffer );
-        memcpy( p_new_buffer->p_buffer, p_buffer->p_buffer,
-                p_buffer->i_nb_bytes );
-        p_new_buffer->i_nb_samples = p_buffer->i_nb_samples;
-        p_new_buffer->i_nb_bytes = p_buffer->i_nb_bytes;
-
-        aout_BufferFree( p_buffer );
-        p_buffer = p_new_buffer;
+        p_input->pp_resamplers[0]->input.i_rate 
+            = (p_input->input.i_rate * old_duration) / duration;
 
-        aout_FiltersPlay( p_aout, pp_filters, i_nb_filters,
+        aout_FiltersPlay( p_aout, p_input->pp_resamplers,
+                          p_input->i_nb_resamplers,
                           &p_buffer );
-
-        aout_FiltersDestroyPipeline( p_aout, pp_filters,
-                                     i_nb_filters );
     }
     else
     {
-        /* No resampling needed (except maybe the one imposed by the
-         * output). */
         duration = p_buffer->end_date - p_buffer->start_date;
-        aout_FiltersPlay( p_aout, p_input->pp_filters, p_input->i_nb_filters,
-                          &p_buffer );
+
+        if ( p_input->input.i_rate != p_aout->mixer.mixer.i_rate )
+        {
+            /* Standard resampling is needed ! */
+            p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate;
+
+            aout_FiltersPlay( p_aout, p_input->pp_resamplers,
+                              p_input->i_nb_resamplers,
+                              &p_buffer );
+        }
     }
 
     /* Adding the start date will be managed by aout_FifoPush(). */