]> git.sesse.net Git - vlc/blobdiff - modules/audio_filter/resampler/bandlimited.c
Use var_Inherit* instead of var_CreateGet*.
[vlc] / modules / audio_filter / resampler / bandlimited.c
index 112f6242c3c1786045b96a9247e846aafe0af8c5..d0f447d79b78f64b01ca7a53f035728e0c5054e2 100644 (file)
@@ -56,16 +56,12 @@ static int  OpenFilter ( vlc_object_t * );
 static void CloseFilter( vlc_object_t * );
 static block_t *Resample( filter_t *, block_t * );
 
-
-static void FilterFloatUP( const float Imp[], const float ImpD[], uint16_t Nwing,
-                           float *f_in, float *f_out, uint32_t ui_remainder,
-                           uint32_t ui_output_rate, int16_t Inc,
-                           int i_nb_channels );
-
-static void FilterFloatUD( const float Imp[], const float ImpD[], uint16_t Nwing,
-                           float *f_in, float *f_out, uint32_t ui_remainder,
-                           uint32_t ui_output_rate, uint32_t ui_input_rate,
-                           int16_t Inc, int i_nb_channels );
+static void ResampleFloat( filter_t *p_filter,
+                           block_t **pp_out_buf,  size_t *pi_out,
+                           float **pp_in,
+                           int i_in, int i_in_end,
+                           double d_factor, bool b_factor_old,
+                           int i_nb_channels, int i_bytes_per_frame );
 
 /*****************************************************************************
  * Local structures
@@ -147,8 +143,10 @@ static block_t *Resample( filter_t * p_filter, block_t * p_in_buf )
             + p_filter->p_sys->i_buf_size;
     block_t *p_out_buf = filter_NewAudioBuffer( p_filter, i_out_size );
     if( !p_out_buf )
+    {
+        block_Release( p_in_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 )
     {
@@ -195,7 +193,7 @@ static block_t *Resample( filter_t * p_filter, block_t * p_in_buf )
     const float *p_in_orig = p_in;
 
     /* Make sure the output buffer is reset */
-    memset( p_out, 0, p_out_buf->i_buffer );
+    memset( p_out_buf->p_buffer, 0, p_out_buf->i_buffer );
 
     /* Calculate the new length of the filter wing */
     d_factor = (double)i_out_rate / p_filter->fmt_in.audio.i_rate;
@@ -209,83 +207,17 @@ static block_t *Resample( filter_t * p_filter, block_t * p_in_buf )
     /* Apply the old rate until we have enough samples for the new one */
     i_in = p_sys->i_old_wing;
     p_in += p_sys->i_old_wing * i_nb_channels;
-    for( ; i_in < i_filter_wing &&
-           (i_in + p_sys->i_old_wing) < i_in_nb; i_in++ )
-    {
-        if( p_sys->d_old_factor == 1 )
-        {
-            /* Just copy the samples */
-            memcpy( p_out, p_in, i_bytes_per_frame );
-            p_in += i_nb_channels;
-            p_out += i_nb_channels;
-            i_out++;
-            continue;
-        }
-
-        while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
-        {
-
-            if( p_sys->d_old_factor >= 1 )
-            {
-                /* FilterFloatUP() is faster if we can use it */
-
-                /* Perform left-wing inner product */
-                FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
-                               SMALL_FILTER_NWING, p_in, p_out,
-                               p_sys->i_remainder,
-                               p_filter->fmt_out.audio.i_rate,
-                               -1, i_nb_channels );
-                /* Perform right-wing inner product */
-                FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
-                               SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
-                               p_filter->fmt_out.audio.i_rate -
-                               p_sys->i_remainder,
-                               p_filter->fmt_out.audio.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
-
-                /* Sanity check */
-                if( p_out_buf->i_buffer/i_bytes_per_frame <= i_out+1 )
-                {
-                    p_out += i_nb_channels;
-                    i_out++;
-                    p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
-                    break;
-                }
-            }
-            else
-            {
-                /* Perform left-wing inner product */
-                FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
-                               SMALL_FILTER_NWING, p_in, p_out,
-                               p_sys->i_remainder,
-                               p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
-                               -1, i_nb_channels );
-                /* Perform right-wing inner product */
-                FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
-                               SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
-                               p_filter->fmt_out.audio.i_rate -
-                               p_sys->i_remainder,
-                               p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
-                               1, i_nb_channels );
-            }
-
-            p_out += i_nb_channels;
-            i_out++;
 
-            p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
-        }
+    size_t i_old_in_end = 0;
+    if( p_sys->i_old_wing <= i_in_nb )
+        i_old_in_end = __MIN( i_filter_wing, i_in_nb - p_sys->i_old_wing );
 
-        p_in += i_nb_channels;
-        p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
-    }
+    ResampleFloat( p_filter,
+                   &p_out_buf, &i_out, &p_in,
+                   i_in, i_old_in_end,
+                   p_sys->d_old_factor, true,
+                   i_nb_channels, i_bytes_per_frame );
+    i_in = __MAX( i_in, i_old_in_end );
 
     /* Apply the new rate for the rest of the samples */
     if( i_in < i_in_nb - i_filter_wing )
@@ -293,82 +225,25 @@ static block_t *Resample( filter_t * p_filter, block_t * p_in_buf )
         p_sys->d_old_factor = d_factor;
         p_sys->i_old_wing   = i_filter_wing;
     }
-    for( ; i_in < i_in_nb - i_filter_wing; i_in++ )
+    if( p_out_buf )
     {
-        while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
-        {
-
-            if( d_factor >= 1 )
-            {
-                /* FilterFloatUP() is faster if we can use it */
-
-                /* Perform left-wing inner product */
-                FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
-                               SMALL_FILTER_NWING, p_in, p_out,
-                               p_sys->i_remainder,
-                               p_filter->fmt_out.audio.i_rate,
-                               -1, i_nb_channels );
-
-                /* Perform right-wing inner product */
-                FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
-                               SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
-                               p_filter->fmt_out.audio.i_rate -
-                               p_sys->i_remainder,
-                               p_filter->fmt_out.audio.i_rate,
-                               1, i_nb_channels );
-
-#if 0
-                /* Normalize for unity filter gain */
-                for( int i = 0; i < i_nb_channels; i++ )
-                {
-                    *(p_out+i) *= d_old_scale_factor;
-                }
-#endif
-                /* Sanity check */
-                if( p_out_buf->i_buffer/i_bytes_per_frame <= i_out+1 )
-                {
-                    p_out += i_nb_channels;
-                    i_out++;
-                    p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
-                    break;
-                }
-            }
-            else
-            {
-                /* Perform left-wing inner product */
-                FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
-                               SMALL_FILTER_NWING, p_in, p_out,
-                               p_sys->i_remainder,
-                               p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
-                               -1, i_nb_channels );
-                /* Perform right-wing inner product */
-                FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
-                               SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
-                               p_filter->fmt_out.audio.i_rate -
-                               p_sys->i_remainder,
-                               p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
-                               1, i_nb_channels );
-            }
-
-            p_out += i_nb_channels;
-            i_out++;
-
-            p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
-        }
-
-        p_in += i_nb_channels;
-        p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
+        ResampleFloat( p_filter,
+                       &p_out_buf, &i_out, &p_in,
+                       i_in, i_in_nb - i_filter_wing,
+                       d_factor, false,
+                       i_nb_channels, i_bytes_per_frame );
+
+        /* Finalize aout buffer */
+        p_out_buf->i_nb_samples = i_out;
+        p_out_buf->i_dts =
+        p_out_buf->i_pts = date_Get( &p_sys->end_date );
+        p_out_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 *
+            i_nb_channels * sizeof(int32_t);
     }
 
-    /* Finalize aout buffer */
-    p_out_buf->i_nb_samples = i_out;
-    p_out_buf->i_pts = date_Get( &p_sys->end_date );
-    p_out_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 *
-        i_nb_channels * sizeof(int32_t);
-
     /* Buffer i_filter_wing * 2 samples for next time */
     if( p_sys->i_old_wing )
     {
@@ -382,6 +257,7 @@ static block_t *Resample( filter_t * p_filter, block_t * p_in_buf )
             else
             {
                 p_sys->i_buf_size = p_sys->i_old_wing = 0; /* oops! */
+                block_Release( p_in_buf );
                 return p_out_buf;
             }
         }
@@ -396,6 +272,7 @@ static block_t *Resample( filter_t * p_filter, block_t * p_in_buf )
              i_out * p_filter->fmt_in.audio.i_bytes_per_frame );
 #endif
 
+    block_Release( p_in_buf );
     return p_out_buf;
 }
 
@@ -420,7 +297,7 @@ static int OpenFilter( vlc_object_t *p_this )
     }
 
 #if !defined( SYS_DARWIN )
-    if( !var_InheritInteger( p_this, "hq-resampling" ) )
+    if( !var_InheritBool( p_this, "hq-resampling" ) )
     {
         return VLC_EGENERIC;
     }
@@ -462,9 +339,9 @@ static void CloseFilter( vlc_object_t *p_this )
     free( p_filter->p_sys );
 }
 
-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 )
+static 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 )
 {
     const float *Hp, *Hdp, *End;
     float t, temp;
@@ -505,10 +382,10 @@ void FilterFloatUP( const float Imp[], const float ImpD[], uint16_t Nwing, float
     }
 }
 
-void FilterFloatUD( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
-                    float *p_out, uint32_t ui_remainder,
-                    uint32_t ui_output_rate, uint32_t ui_input_rate,
-                    int16_t Inc, int i_nb_channels )
+static void FilterFloatUD( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
+                           float *p_out, uint32_t ui_remainder,
+                           uint32_t ui_output_rate, uint32_t ui_input_rate,
+                           int16_t Inc, int i_nb_channels )
 {
     const float *Hp, *Hdp, *End;
     float t, temp;
@@ -560,3 +437,116 @@ void FilterFloatUD( const float Imp[], const float ImpD[], uint16_t Nwing, float
         p_in += (Inc * i_nb_channels); /* Input signal step */
     }
 }
+
+static int ReallocBuffer( block_t **pp_out_buf,
+                          float **pp_out, size_t i_out,
+                          int i_nb_channels, int i_bytes_per_frame )
+{
+    if( i_out < (*pp_out_buf)->i_buffer/i_bytes_per_frame )
+        return VLC_SUCCESS;
+
+    /* It may happen when the wing size changes */
+    const unsigned i_extra_frame = 256;
+    *pp_out_buf = block_Realloc( *pp_out_buf, 0,
+                                 (*pp_out_buf)->i_buffer +
+                                    i_extra_frame * i_bytes_per_frame );
+    if( !*pp_out_buf )
+        return VLC_EGENERIC;
+
+    *pp_out = (float*)(*pp_out_buf)->p_buffer + i_out * i_nb_channels;
+    memset( *pp_out, 0, i_extra_frame * i_bytes_per_frame );
+    return VLC_SUCCESS;
+}
+
+static void ResampleFloat( filter_t *p_filter,
+                           block_t **pp_out_buf,  size_t *pi_out,
+                           float **pp_in,
+                           int i_in, int i_in_end,
+                           double d_factor, bool b_factor_old,
+                           int i_nb_channels, int i_bytes_per_frame )
+{
+    filter_sys_t *p_sys = p_filter->p_sys;
+
+    float *p_in = *pp_in;
+    size_t i_out = *pi_out;
+    float *p_out = (float*)(*pp_out_buf)->p_buffer + i_out * i_nb_channels;
+
+    for( ; i_in < i_in_end; i_in++ )
+    {
+        if( b_factor_old && d_factor == 1 )
+        {
+            if( ReallocBuffer( pp_out_buf, &p_out,
+                               i_out, i_nb_channels, i_bytes_per_frame ) )
+                return;
+            /* Just copy the samples */
+            memcpy( p_out, p_in, i_bytes_per_frame );
+            p_in += i_nb_channels;
+            p_out += i_nb_channels;
+            i_out++;
+            continue;
+        }
+
+        while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
+        {
+            if( ReallocBuffer( pp_out_buf, &p_out,
+                               i_out, i_nb_channels, i_bytes_per_frame ) )
+                return;
+
+            if( d_factor >= 1 )
+            {
+                /* FilterFloatUP() is faster if we can use it */
+
+                /* Perform left-wing inner product */
+                FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
+                               SMALL_FILTER_NWING, p_in, p_out,
+                               p_sys->i_remainder,
+                               p_filter->fmt_out.audio.i_rate,
+                               -1, i_nb_channels );
+                /* Perform right-wing inner product */
+                FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
+                               SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
+                               p_filter->fmt_out.audio.i_rate -
+                               p_sys->i_remainder,
+                               p_filter->fmt_out.audio.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
+            {
+                /* Perform left-wing inner product */
+                FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
+                               SMALL_FILTER_NWING, p_in, p_out,
+                               p_sys->i_remainder,
+                               p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
+                               -1, i_nb_channels );
+                /* Perform right-wing inner product */
+                FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
+                               SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
+                               p_filter->fmt_out.audio.i_rate -
+                               p_sys->i_remainder,
+                               p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
+                               1, i_nb_channels );
+            }
+
+            p_out += i_nb_channels;
+            i_out++;
+
+            p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
+        }
+
+        p_in += i_nb_channels;
+        p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
+    }
+
+    *pp_in  = p_in;
+    *pi_out = i_out;
+}
+
+