band-limited resampler: switch to audio filter2

modules/audio_filter/resampler/bandlimited.c

@@ -48,10 +48,6 @@
 /*****************************************************************************
  * Local prototypes
  *****************************************************************************/
-static int  Create    ( vlc_object_t * );
-static void Close     ( vlc_object_t * );
-static void DoWork    ( aout_instance_t *, aout_filter_t *, aout_buffer_t *,
-                        aout_buffer_t * );
 
 /* audio filter2 */
 static int  OpenFilter ( vlc_object_t * );
@@ -81,10 +77,9 @@ struct filter_sys_t
     int i_old_wing;
 
     unsigned int i_remainder;                /* remainder of previous sample */
+    bool b_first;
 
     date_t end_date;
-
-    bool b_filter2;
 };
 
 /*****************************************************************************
@@ -94,112 +89,29 @@ vlc_module_begin ()
     set_category( CAT_AUDIO )
     set_subcategory( SUBCAT_AUDIO_MISC )
     set_description( N_("Audio filter for band-limited interpolation resampling") )
-    set_capability( "audio filter", 20 )
-    set_callbacks( Create, Close )
-
-    add_submodule ()
-    set_description( N_("Audio filter for band-limited interpolation resampling") )
     set_capability( "audio filter2", 20 )
     set_callbacks( OpenFilter, CloseFilter )
 vlc_module_end ()
 
 /*****************************************************************************
- * Create: allocate linear resampler
+ * Resample: convert a buffer
  *****************************************************************************/
-static int Create( vlc_object_t *p_this )
+static block_t *Resample( filter_t * p_filter, block_t * p_in_buf )
 {
-    aout_filter_t * p_filter = (aout_filter_t *)p_this;
-    struct filter_sys_t * p_sys;
-    double d_factor;
-    int i_filter_wing;
-
-    if ( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate
-          || p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format
-          || p_filter->fmt_in.audio.i_physical_channels
-              != p_filter->fmt_out.audio.i_physical_channels
-          || p_filter->fmt_in.audio.i_original_channels
-              != p_filter->fmt_out.audio.i_original_channels
-          || p_filter->fmt_in.audio.i_format != VLC_CODEC_FL32 )
-    {
-        return VLC_EGENERIC;
-    }
-
-#if !defined( __APPLE__ )
-    if( !config_GetInt( p_this, "hq-resampling" ) )
+    if( !p_in_buf || !p_in_buf->i_nb_samples )
     {
-        return VLC_EGENERIC;
-    }
-#endif
-
-    /* Allocate the memory needed to store the module's structure */
-    p_sys = malloc( sizeof(filter_sys_t) );
-    if( p_sys == NULL )
-        return VLC_ENOMEM;
-    p_filter->p_sys = (struct aout_filter_sys_t *)p_sys;
-
-    /* Calculate worst case for the length of the filter wing */
-    d_factor = (double)p_filter->fmt_out.audio.i_rate
-                        / p_filter->fmt_in.audio.i_rate / AOUT_MAX_INPUT_RATE;
-    i_filter_wing = ((SMALL_FILTER_NMULT + 1)/2.0)
-                      * __MAX(1.0, 1.0/d_factor) + 10;
-    p_sys->i_buf_size = aout_FormatNbChannels( &p_filter->fmt_in.audio ) *
-        sizeof(int32_t) * 2 * i_filter_wing;
-
-    /* Allocate enough memory to buffer previous samples */
-    p_sys->p_buf = malloc( p_sys->i_buf_size );
-    if( p_sys->p_buf == NULL )
-    {
-        free( p_sys );
-        return VLC_ENOMEM;
+        if( p_in_buf )
+            block_Release( p_in_buf );
+        return NULL;
     }
 
-    p_sys->i_old_wing = 0;
-    p_sys->b_filter2 = false;           /* It seams to be a good valuefor this module */
-    p_filter->pf_do_work = DoWork;
-
-    /* We don't want a new buffer to be created because we're not sure we'll
-     * actually need to resample anything. */
-    p_filter->b_in_place = true;
-
-    return VLC_SUCCESS;
-}
-
-/*****************************************************************************
- * Close: free our resources
- *****************************************************************************/
-static void Close( vlc_object_t * p_this )
-{
-    aout_filter_t * p_filter = (aout_filter_t *)p_this;
-    filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
-    free( p_sys->p_buf );
-    free( p_sys );
-}
-
-/*****************************************************************************
- * DoWork: convert a buffer
- *****************************************************************************/
-static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
-                    aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
-{
-    filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
-    float *p_out = (float *)p_out_buf->p_buffer;
-
+    filter_sys_t *p_sys = p_filter->p_sys;
+    unsigned int i_out_rate = p_filter->fmt_out.audio.i_rate;
     int i_nb_channels = aout_FormatNbChannels( &p_filter->fmt_in.audio );
-    int i_in_nb = p_in_buf->i_nb_samples;
-    int i_in, i_out = 0;
-    unsigned int i_out_rate;
-    double d_factor, d_scale_factor, d_old_scale_factor;
-    int i_filter_wing;
-
-    if( p_sys->b_filter2 )
-        i_out_rate = p_filter->fmt_out.audio.i_rate;
-    else
-        i_out_rate = p_aout->mixer_format.i_rate;
 
     /* Check if we really need to run the resampler */
     if( i_out_rate == p_filter->fmt_in.audio.i_rate )
     {
-#if 0   /* FIXME: needs audio filter2 to use block_Realloc */
         if( /*p_filter->b_continuity && /--* What difference does it make ? :) */
             p_sys->i_old_wing )
         {
@@ -208,30 +120,35 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
                 p_sys->i_old_wing * p_filter->fmt_in.audio.i_bytes_per_frame,
                 p_in_buf->i_buffer );
             if( !p_in_buf )
-                abort();
+                return NULL;
             memcpy( p_in_buf->p_buffer, p_sys->p_buf +
                     i_nb_channels * p_sys->i_old_wing,
                     p_sys->i_old_wing *
                     p_filter->fmt_in.audio.i_bytes_per_frame );
 
-            p_out_buf->i_nb_samples = p_in_buf->i_nb_samples +
-                p_sys->i_old_wing;
+            p_in_buf->i_nb_samples += p_sys->i_old_wing;
 
-            p_out_buf->i_pts = date_Get( &p_sys->end_date );
-            p_out_buf->i_length =
+            p_in_buf->i_pts = date_Get( &p_sys->end_date );
+            p_in_buf->i_length =
                 date_Increment( &p_sys->end_date,
-                                p_out_buf->i_nb_samples ) - p_out_buf->i_pts;
-
-            p_out_buf->i_buffer = p_out_buf->i_nb_samples *
-                p_filter->fmt_in.audio.i_bytes_per_frame;
+                                p_in_buf->i_nb_samples ) - p_in_buf->i_pts;
         }
-#endif
-        p_out_buf->i_flags |= BLOCK_FLAG_DISCONTINUITY;
+        p_in_buf->i_flags |= BLOCK_FLAG_DISCONTINUITY;
         p_sys->i_old_wing = 0;
-        return;
+        return p_in_buf;
     }
 
-    if( p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY )
+    unsigned i_bytes_per_frame = p_filter->fmt_out.audio.i_channels *
+                                 p_filter->fmt_out.audio.i_bitspersample / 8;
+    size_t i_out_size = i_bytes_per_frame * ( 1 + ( p_in_buf->i_nb_samples *
+              p_filter->fmt_out.audio.i_rate / p_filter->fmt_in.audio.i_rate) )
+            + p_filter->p_sys->i_buf_size;
+    block_t *p_out_buf = filter_NewAudioBuffer( p_filter, i_out_size );
+    if( !p_out_buf )
+        return NULL;
+    float *p_out = (float *)p_out_buf->p_buffer;
+
+    if( (p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY) || p_sys->b_first )
     {
         /* Continuity in sound samples has been broken, we'd better reset
          * everything. */
@@ -241,8 +158,14 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
         date_Set( &p_sys->end_date, p_in_buf->i_pts );
         p_sys->d_old_factor = 1;
         p_sys->i_old_wing   = 0;
+        p_sys->b_first = false;
     }
 
+    int i_in_nb = p_in_buf->i_nb_samples;
+    int i_in, i_out = 0;
+    double d_factor, d_scale_factor, d_old_scale_factor;
+    int i_filter_wing;
+
 #if 0
     msg_Err( p_filter, "old rate: %i, old factor: %f, old wing: %i, i_in: %i",
              p_sys->i_old_rate, p_sys->d_old_factor,
@@ -262,9 +185,11 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
                     p_sys->i_old_wing * 2 *
                       p_filter->fmt_in.audio.i_bytes_per_frame );
     }
+    /* XXX: why i_nb_channels instead of i_bytes_per_frame??? */
     vlc_memcpy( p_in + p_sys->i_old_wing * 2 * i_nb_channels,
                 p_in_buf->p_buffer,
                 p_in_buf->i_nb_samples * p_filter->fmt_in.audio.i_bytes_per_frame );
+    block_Release( p_in_buf );
 
     /* Make sure the output buffer is reset */
     memset( p_out, 0, p_out_buf->i_buffer );
@@ -457,7 +382,7 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
 
     p_out_buf->i_buffer = p_out_buf->i_nb_samples *
         i_nb_channels * sizeof(int32_t);
-
+    return p_out_buf;
 }
 
 /*****************************************************************************
@@ -505,7 +430,7 @@ static int OpenFilter( vlc_object_t *p_this )
     }
 
     p_sys->i_old_wing = 0;
-    p_sys->b_filter2 = true;
+    p_sys->b_first = true;
     p_filter->pf_audio_filter = Resample;
 
     msg_Dbg( p_this, "%4.4s/%iKHz/%i->%4.4s/%iKHz/%i",
@@ -532,70 +457,6 @@ static void CloseFilter( vlc_object_t *p_this )
     free( p_filter->p_sys );
 }
 
-/*****************************************************************************
- * Resample
- *****************************************************************************/
-static block_t *Resample( filter_t *p_filter, block_t *p_block )
-{
-    aout_filter_t aout_filter;
-    aout_buffer_t in_buf, out_buf;
-    block_t *p_out;
-    int i_out_size;
-    int i_bytes_per_frame;
-
-    if( !p_block || !p_block->i_nb_samples )
-    {
-        if( p_block )
-            block_Release( p_block );
-        return NULL;
-    }
-
-    i_bytes_per_frame = p_filter->fmt_out.audio.i_channels *
-                  p_filter->fmt_out.audio.i_bitspersample / 8;
-
-    i_out_size = i_bytes_per_frame * ( 1 + ( p_block->i_nb_samples *
-                                             p_filter->fmt_out.audio.i_rate /
-                                             p_filter->fmt_in.audio.i_rate) ) +
-                 p_filter->p_sys->i_buf_size;
-
-    p_out = p_filter->pf_audio_buffer_new( p_filter, i_out_size );
-    if( !p_out )
-    {
-        msg_Warn( p_filter, "can't get output buffer" );
-        block_Release( p_block );
-        return NULL;
-    }
-
-    p_out->i_nb_samples = i_out_size / i_bytes_per_frame;
-    p_out->i_dts = p_block->i_dts;
-    p_out->i_pts = p_block->i_pts;
-    p_out->i_length = p_block->i_length;
-
-    aout_filter.p_sys = (struct aout_filter_sys_t *)p_filter->p_sys;
-    aout_filter.fmt_in.audio = p_filter->fmt_in.audio;
-    aout_filter.fmt_in.audio.i_bytes_per_frame = p_filter->fmt_in.audio.i_channels *
-                  p_filter->fmt_in.audio.i_bitspersample / 8;
-    aout_filter.fmt_out.audio = p_filter->fmt_out.audio;
-    aout_filter.fmt_out.audio.i_bytes_per_frame = p_filter->fmt_out.audio.i_channels *
-                  p_filter->fmt_out.audio.i_bitspersample / 8;
-
-    in_buf.p_buffer = p_block->p_buffer;
-    in_buf.i_buffer = p_block->i_buffer;
-    in_buf.i_nb_samples = p_block->i_nb_samples;
-    out_buf.p_buffer = p_out->p_buffer;
-    out_buf.i_buffer = p_out->i_buffer;
-    out_buf.i_nb_samples = p_out->i_nb_samples;
-
-    DoWork( (aout_instance_t *)p_filter, &aout_filter, &in_buf, &out_buf );
-
-    block_Release( p_block );
-
-    p_out->i_buffer = out_buf.i_buffer;
-    p_out->i_nb_samples = out_buf.i_nb_samples;
-
-    return p_out;
-}
-
 void FilterFloatUP( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
                     float *p_out, uint32_t ui_remainder,
                     uint32_t ui_output_rate, int16_t Inc, int i_nb_channels )