]> git.sesse.net Git - vlc/blobdiff - src/audio_output/filters.c
aout_InputPlay: compute drift once (rather than 2-4 times)
[vlc] / src / audio_output / filters.c
index b5ceb86b96b23a9be9c26466a2524308e9f1d31a..c3a8e9b635c54efe00548fb8c8a20a47bf057dd5 100644 (file)
@@ -59,7 +59,6 @@ static filter_t * FindFilter( aout_instance_t * p_aout,
                                   VLC_OBJECT_GENERIC, typename );
 
     if ( p_filter == NULL ) return NULL;
-    vlc_object_attach( p_filter, p_aout );
 
     memcpy( &p_filter->fmt_in.audio, p_input_format,
             sizeof(audio_sample_format_t) );
@@ -68,6 +67,7 @@ static filter_t * FindFilter( aout_instance_t * p_aout,
             sizeof(audio_sample_format_t) );
     p_filter->fmt_out.i_codec = p_output_format->i_format;
     p_filter->pf_audio_buffer_new = aout_FilterBufferNew;
+    p_filter->p_owner = NULL;
 
     p_filter->p_module = module_need( p_filter, "audio filter", NULL, false );
     if ( p_filter->p_module == NULL )
@@ -80,212 +80,112 @@ static filter_t * FindFilter( aout_instance_t * p_aout,
     return p_filter;
 }
 
-/*****************************************************************************
- * SplitConversion: split a conversion in two parts
- *****************************************************************************
- * Returns the number of conversions required by the first part - 0 if only
- * one conversion was asked.
- * Beware : p_output_format can be modified during this function if the
- * developer passed SplitConversion( toto, titi, titi, ... ). That is legal.
- * SplitConversion( toto, titi, toto, ... ) isn't.
- *****************************************************************************/
-static int SplitConversion( const audio_sample_format_t * p_input_format,
-                            const audio_sample_format_t * p_output_format,
-                            audio_sample_format_t * p_middle_format )
+/**
+ * Splits audio format conversion in two simpler conversions
+ * @return 0 on successful split, -1 if the input and output formats are too
+ * similar to split the conversion.
+ */
+static int SplitConversion( const audio_sample_format_t *restrict infmt,
+                            const audio_sample_format_t *restrict outfmt,
+                            audio_sample_format_t *midfmt )
 {
-    bool b_format =
-             (p_input_format->i_format != p_output_format->i_format);
-    bool b_rate = (p_input_format->i_rate != p_output_format->i_rate);
-    bool b_channels =
-        (p_input_format->i_physical_channels
-          != p_output_format->i_physical_channels)
-     || (p_input_format->i_original_channels
-          != p_output_format->i_original_channels);
-    int i_nb_conversions = b_format + b_rate + b_channels;
-
-    if ( i_nb_conversions <= 1 ) return 0;
-
-    memcpy( p_middle_format, p_output_format, sizeof(audio_sample_format_t) );
+    *midfmt = *outfmt;
 
-    if ( i_nb_conversions == 2 )
+    if( infmt->i_rate != outfmt->i_rate )
+        midfmt->i_rate = infmt->i_rate;
+    else
+    if( infmt->i_physical_channels != outfmt->i_physical_channels
+     || infmt->i_original_channels != outfmt->i_original_channels )
     {
-        if ( !b_format || !b_channels )
-        {
-            p_middle_format->i_rate = p_input_format->i_rate;
-            aout_FormatPrepare( p_middle_format );
-            return 1;
-        }
-
-        /* !b_rate */
-        p_middle_format->i_physical_channels
-             = p_input_format->i_physical_channels;
-        p_middle_format->i_original_channels
-             = p_input_format->i_original_channels;
-        aout_FormatPrepare( p_middle_format );
-        return 1;
+        midfmt->i_physical_channels = infmt->i_physical_channels;
+        midfmt->i_original_channels = infmt->i_original_channels;
     }
+    else
+        return -1;
 
-    /* i_nb_conversion == 3 */
-    p_middle_format->i_rate = p_input_format->i_rate;
-    aout_FormatPrepare( p_middle_format );
-    return 2;
+    aout_FormatPrepare( midfmt );
+    return AOUT_FMTS_IDENTICAL( infmt, midfmt ) ? -1 : 0;
 }
 
-static void ReleaseFilter( filter_t * p_filter )
+/**
+ * Allocates audio format conversion filters
+ * @param obj parent VLC object for new filters
+ * @param filters table of filters [IN/OUT]
+ * @param nb_filters pointer to the number of filters in the table [IN/OUT]
+ * @param infmt input audio format
+ * @param outfmt output audio format
+ * @return 0 on success, -1 on failure
+ */
+int aout_FiltersCreatePipeline( aout_instance_t *obj,
+                                filter_t **filters,
+                                int *nb_filters,
+                                const audio_sample_format_t *restrict infmt,
+                                const audio_sample_format_t *restrict outfmt )
 {
-    module_unneed( p_filter, p_filter->p_module );
-    vlc_object_release( p_filter );
-}
+    audio_sample_format_t curfmt = *outfmt;
+    unsigned i = 0, max = *nb_filters - AOUT_MAX_FILTERS;
 
-/*****************************************************************************
- * aout_FiltersCreatePipeline: create a filters pipeline to transform a sample
- *                             format to another
- *****************************************************************************
- * pi_nb_filters must be initialized before calling this function
- *****************************************************************************/
-int aout_FiltersCreatePipeline( aout_instance_t * p_aout,
-                                filter_t ** pp_filters_start,
-                                int * pi_nb_filters,
-                                const audio_sample_format_t * p_input_format,
-                                const audio_sample_format_t * p_output_format )
-{
-    filter_t** pp_filters = pp_filters_start + *pi_nb_filters;
-    audio_sample_format_t temp_format;
-    int i_nb_conversions;
+    filters += *nb_filters;
+    aout_FormatsPrint( obj, "filter(s)", infmt, outfmt );
 
-    if ( AOUT_FMTS_IDENTICAL( p_input_format, p_output_format ) )
+    while( !AOUT_FMTS_IDENTICAL( infmt, &curfmt ) )
     {
-        msg_Dbg( p_aout, "no need for any filter" );
-        return 0;
-    }
-
-    aout_FormatsPrint( p_aout, "filter(s)", p_input_format, p_output_format );
-
-    if( *pi_nb_filters + 1 > AOUT_MAX_FILTERS )
-    {
-        msg_Err( p_aout, "max filter reached (%d)", AOUT_MAX_FILTERS );
-        dialog_Fatal( p_aout, _("Audio filtering failed"),
-                      _("The maximum number of filters (%d) was reached."),
-                      AOUT_MAX_FILTERS );
-        return -1;
-    }
-
-    /* Try to find a filter to do the whole conversion. */
-    pp_filters[0] = FindFilter( p_aout, p_input_format, p_output_format );
-    if ( pp_filters[0] != NULL )
-    {
-        msg_Dbg( p_aout, "found a filter for the whole conversion" );
-        ++*pi_nb_filters;
-        return 0;
-    }
+        if( i >= max )
+        {
+            msg_Err( obj, "max (%u) filters reached", AOUT_MAX_FILTERS );
+            dialog_Fatal( obj, _("Audio filtering failed"),
+                          _("The maximum number of filters (%u) was reached."),
+                          AOUT_MAX_FILTERS );
+            goto rollback;
+        }
 
-    /* We'll have to split the conversion. We always do the downmixing
-     * before the resampling, because the audio decoder can probably do it
-     * for us. */
-    i_nb_conversions = SplitConversion( p_input_format,
-                                        p_output_format, &temp_format );
-    if ( !i_nb_conversions )
-    {
-        /* There was only one conversion to do, and we already failed. */
-        msg_Err( p_aout, "couldn't find a filter for the conversion "
-                "%4.4s -> %4.4s",
-                &p_input_format->i_format, &p_output_format->i_format );
-        return -1;
-    }
+        /* Make room and prepend a filter */
+        memmove( filters + 1, filters, i * sizeof( *filters ) );
 
-    pp_filters[0] = FindFilter( p_aout, p_input_format, &temp_format );
-    if ( pp_filters[0] == NULL && i_nb_conversions == 2 )
-    {
-        /* Try with only one conversion. */
-        SplitConversion( p_input_format, &temp_format, &temp_format );
-        pp_filters[0] = FindFilter( p_aout, p_input_format, &temp_format );
-    }
-    if ( pp_filters[0] == NULL )
-    {
-        msg_Err( p_aout,
-              "couldn't find a filter for the first part of the conversion" );
-        return -1;
-    }
-
-    /* We have the first stage of the conversion. Find a filter for
-     * the rest. */
-    if( *pi_nb_filters + 2 > AOUT_MAX_FILTERS )
-    {
-        ReleaseFilter( pp_filters[0] );
-        msg_Err( p_aout, "max filter reached (%d)", AOUT_MAX_FILTERS );
-        dialog_Fatal( p_aout, _("Audio filtering failed"),
-                      _("The maximum number of filters (%d) was reached."),
-                      AOUT_MAX_FILTERS );
-        return -1;
-    }
-    pp_filters[1] = FindFilter( p_aout, &pp_filters[0]->fmt_out.audio,
-                                p_output_format );
-    if ( pp_filters[1] == NULL )
-    {
-        /* Try to split the conversion. */
-        i_nb_conversions = SplitConversion( &pp_filters[0]->fmt_out.audio,
-                                           p_output_format, &temp_format );
-        if ( !i_nb_conversions )
+        *filters = FindFilter( obj, infmt, &curfmt );
+        if( *filters != NULL )
         {
-            ReleaseFilter( pp_filters[0] );
-            msg_Err( p_aout,
-              "couldn't find a filter for the second part of the conversion" );
-            return -1;
+            i++;
+            break; /* done! */
         }
-        if( *pi_nb_filters + 3 > AOUT_MAX_FILTERS )
+
+        audio_sample_format_t midfmt;
+        /* Split the conversion */
+        if( SplitConversion( infmt, &curfmt, &midfmt ) )
         {
-            ReleaseFilter( pp_filters[0] );
-            msg_Err( p_aout, "max filter reached (%d)", AOUT_MAX_FILTERS );
-            dialog_Fatal( p_aout, _("Audio filtering failed"),
-                          _("The maximum number of filters (%d) was reached."),
-                          AOUT_MAX_FILTERS );
-            return -1;
+            msg_Err( obj, "conversion pipeline failed: %4.4s -> %4.4s",
+                     (const char *)&infmt->i_format,
+                     (const char *)&outfmt->i_format );
+            goto rollback;
         }
-        pp_filters[1] = FindFilter( p_aout, &pp_filters[0]->fmt_out.audio,
-                                    &temp_format );
-        pp_filters[2] = FindFilter( p_aout, &temp_format,
-                                    p_output_format );
 
-        if ( pp_filters[1] == NULL || pp_filters[2] == NULL )
+        *filters = FindFilter( obj, &midfmt, &curfmt );
+        if( *filters == NULL )
         {
-            ReleaseFilter( pp_filters[0] );
-            if ( pp_filters[1] != NULL )
-            {
-                ReleaseFilter( pp_filters[1] );
-            }
-            if ( pp_filters[2] != NULL )
-            {
-                ReleaseFilter( pp_filters[2] );
-            }
-            msg_Err( p_aout,
-               "couldn't find filters for the second part of the conversion" );
-            return -1;
+            msg_Err( obj, "cannot find filter for simple conversion" );
+            goto rollback;
         }
-        *pi_nb_filters += 3;
-        msg_Dbg( p_aout, "found 3 filters for the whole conversion" );
-    }
-    else
-    {
-        *pi_nb_filters += 2;
-        msg_Dbg( p_aout, "found 2 filters for the whole conversion" );
+        curfmt = midfmt;
+        i++;
     }
 
+    msg_Dbg( obj, "conversion pipeline completed" );
+    *nb_filters += i;
     return 0;
+
+rollback:
+    aout_FiltersDestroyPipeline( filters, i );
+    return -1;
 }
 
-/*****************************************************************************
- * aout_FiltersDestroyPipeline: deallocate a filters pipeline
- *****************************************************************************/
-void aout_FiltersDestroyPipeline( aout_instance_t * p_aout,
-                                  filter_t ** pp_filters,
-                                  int i_nb_filters )
+/**
+ * Destroys a chain of audio filters.
+ */
+void aout_FiltersDestroyPipeline( filter_t *const *filters, unsigned n )
 {
-    int i;
-    (void)p_aout;
-
-    for ( i = 0; i < i_nb_filters; i++ )
+    for( unsigned i = 0; i < n; i++ )
     {
-        filter_t *p_filter = pp_filters[i];
+        filter_t *p_filter = filters[i];
 
         module_unneed( p_filter, p_filter->p_module );
         free( p_filter->p_owner );
@@ -293,40 +193,10 @@ void aout_FiltersDestroyPipeline( aout_instance_t * p_aout,
     }
 }
 
-/*****************************************************************************
- * aout_FiltersHintBuffers: fill in aout_alloc_t structures to optimize
- *                          buffer allocations
- *****************************************************************************/
-void aout_FiltersHintBuffers( aout_instance_t * p_aout,
-                              filter_t ** pp_filters,
-                              int i_nb_filters, aout_alloc_t * p_first_alloc )
-{
-    int i;
-
-    (void)p_aout; /* unused */
-
-    for ( i = i_nb_filters - 1; i >= 0; i-- )
-    {
-        filter_t * p_filter = pp_filters[i];
-
-        int i_output_size = p_filter->fmt_out.audio.i_bytes_per_frame
-                         * p_filter->fmt_out.audio.i_rate * AOUT_MAX_INPUT_RATE
-                         / p_filter->fmt_out.audio.i_frame_length;
-        int i_input_size = p_filter->fmt_in.audio.i_bytes_per_frame
-                         * p_filter->fmt_in.audio.i_rate * AOUT_MAX_INPUT_RATE
-                         / p_filter->fmt_in.audio.i_frame_length;
-
-        if( i_output_size > p_first_alloc->i_bytes_per_sec )
-            p_first_alloc->i_bytes_per_sec = i_output_size;
-        if( i_input_size > p_first_alloc->i_bytes_per_sec )
-            p_first_alloc->i_bytes_per_sec = i_input_size;
-    }
-}
-
-/*****************************************************************************
- * aout_FiltersPlay: play a buffer
- *****************************************************************************/
-void aout_FiltersPlay( filter_t ** pp_filters,
+/**
+ * Filters an audio buffer through a chain of filters.
+ */
+void aout_FiltersPlay( filter_t *const *pp_filters,
                        unsigned i_nb_filters, block_t ** pp_block )
 {
     block_t *p_block = *pp_block;
@@ -342,4 +212,3 @@ void aout_FiltersPlay( filter_t ** pp_filters,
     }
     *pp_block = p_block;
 }
-