/*****************************************************************************
* filters.c : audio output filters management
*****************************************************************************
- * Copyright (C) 2002 VideoLAN
- * $Id: filters.c,v 1.10 2002/09/22 14:53:52 massiot Exp $
+ * Copyright (C) 2002-2007 the VideoLAN team
+ * $Id$
*
* Authors: Christophe Massiot <massiot@via.ecp.fr>
*
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
- *
+ *
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111, USA.
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
*****************************************************************************/
/*****************************************************************************
* Preamble
*****************************************************************************/
-#include <stdlib.h> /* calloc(), malloc(), free() */
-#include <string.h>
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
-#include <vlc/vlc.h>
+#include <vlc_common.h>
+#include <vlc_interface.h>
#ifdef HAVE_ALLOCA_H
-# include <alloca.h>
+# include <alloca.h>
#endif
-#include "audio_output.h"
+#include <vlc_aout.h>
#include "aout_internal.h"
+#include <libvlc.h>
/*****************************************************************************
* FindFilter: find an audio filter for a specific transformation
const audio_sample_format_t * p_input_format,
const audio_sample_format_t * p_output_format )
{
- aout_filter_t * p_filter = vlc_object_create( p_aout,
- sizeof(aout_filter_t) );
+ static const char typename[] = "audio output";
+ aout_filter_t * p_filter;
+
+ p_filter = vlc_custom_create( p_aout, sizeof(*p_filter),
+ VLC_OBJECT_GENERIC, typename );
if ( p_filter == NULL ) return NULL;
vlc_object_attach( p_filter, p_aout );
memcpy( &p_filter->input, p_input_format, sizeof(audio_sample_format_t) );
memcpy( &p_filter->output, p_output_format,
sizeof(audio_sample_format_t) );
- p_filter->p_module = module_Need( p_filter, "audio filter", NULL );
+ p_filter->p_module = module_Need( p_filter, "audio filter", NULL, 0 );
if ( p_filter->p_module == NULL )
{
vlc_object_detach( p_filter );
- vlc_object_destroy( p_filter );
+ vlc_object_release( p_filter );
return NULL;
}
+ p_filter->b_continuity = false;
+
return p_filter;
}
/*****************************************************************************
- * SplitConversion: split a conversion in two parts
+ * SplitConversion: split a conversion in two parts
*****************************************************************************
* Returns the number of conversions required by the first part - 0 if only
* one conversion was asked.
* developer passed SplitConversion( toto, titi, titi, ... ). That is legal.
* SplitConversion( toto, titi, toto, ... ) isn't.
*****************************************************************************/
-static int SplitConversion( aout_instance_t * p_aout,
- const audio_sample_format_t * p_input_format,
- const audio_sample_format_t * p_output_format,
- audio_sample_format_t * p_middle_format )
+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 )
{
- vlc_bool_t b_format =
+ bool b_format =
(p_input_format->i_format != p_output_format->i_format);
- vlc_bool_t b_rate = (p_input_format->i_rate != p_output_format->i_rate);
- vlc_bool_t b_channels =
- (p_input_format->i_channels != p_output_format->i_channels);
+ 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;
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_channels = p_input_format->i_channels;
+ 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;
}
/* i_nb_conversion == 3 */
p_middle_format->i_rate = p_input_format->i_rate;
+ aout_FormatPrepare( p_middle_format );
return 2;
}
+static void ReleaseFilter( aout_filter_t * p_filter )
+{
+ module_Unneed( p_filter, p_filter->p_module );
+ vlc_object_detach( p_filter );
+ vlc_object_release( p_filter );
+}
+
/*****************************************************************************
* aout_FiltersCreatePipeline: create a filters pipeline to transform a sample
* format to another
*****************************************************************************
- * TODO : allow the user to add/remove specific filters
+ * pi_nb_filters must be initialized before calling this function
*****************************************************************************/
int aout_FiltersCreatePipeline( aout_instance_t * p_aout,
- aout_filter_t ** pp_filters,
+ 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 )
{
+ aout_filter_t** pp_filters = pp_filters_start + *pi_nb_filters;
audio_sample_format_t temp_format;
int i_nb_conversions;
if ( AOUT_FMTS_IDENTICAL( p_input_format, p_output_format ) )
{
msg_Dbg( p_aout, "no need for any filter" );
- *pi_nb_filters = 0;
return 0;
}
- msg_Dbg( p_aout, "filter(s) format=%d->%d rate=%d->%d channels=%d->%d",
- p_input_format->i_format, p_output_format->i_format,
- p_input_format->i_rate, p_output_format->i_rate,
- p_input_format->i_channels, p_output_format->i_channels );
+ 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 );
+ intf_UserFatal( p_aout, false, _("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 = 1;
+ ++*pi_nb_filters;
return 0;
}
/* 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_aout, p_input_format,
+ i_nb_conversions = SplitConversion( p_input_format,
p_output_format, &temp_format );
if ( !i_nb_conversions )
{
if ( pp_filters[0] == NULL && i_nb_conversions == 2 )
{
/* Try with only one conversion. */
- SplitConversion( p_aout, p_input_format, &temp_format,
- &temp_format );
- pp_filters[0] = FindFilter( p_aout, p_input_format,
- &temp_format );
+ SplitConversion( p_input_format, &temp_format, &temp_format );
+ pp_filters[0] = FindFilter( p_aout, p_input_format, &temp_format );
}
if ( pp_filters[0] == NULL )
{
/* 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 );
+ intf_UserFatal( p_aout, false, _("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]->output,
p_output_format );
if ( pp_filters[1] == NULL )
{
/* Try to split the conversion. */
- i_nb_conversions = SplitConversion( p_aout,
- &pp_filters[0]->output,
- p_output_format, &temp_format );
+ i_nb_conversions = SplitConversion( &pp_filters[0]->output,
+ p_output_format, &temp_format );
if ( !i_nb_conversions )
{
- vlc_object_detach( pp_filters[0] );
- vlc_object_destroy( pp_filters[0] );
+ ReleaseFilter( pp_filters[0] );
msg_Err( p_aout,
"couldn't find a filter for the second part of the conversion" );
+ return -1;
+ }
+ if( *pi_nb_filters + 3 > AOUT_MAX_FILTERS )
+ {
+ ReleaseFilter( pp_filters[0] );
+ msg_Err( p_aout, "max filter reached (%d)", AOUT_MAX_FILTERS );
+ intf_UserFatal( p_aout, false, _("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]->output,
&temp_format );
if ( pp_filters[1] == NULL || pp_filters[2] == NULL )
{
- vlc_object_detach( pp_filters[0] );
- vlc_object_destroy( pp_filters[0] );
+ ReleaseFilter( pp_filters[0] );
if ( pp_filters[1] != NULL )
{
- vlc_object_detach( pp_filters[1] );
- vlc_object_destroy( pp_filters[1] );
+ ReleaseFilter( pp_filters[1] );
}
if ( pp_filters[2] != NULL )
{
- vlc_object_detach( pp_filters[2] );
- vlc_object_destroy( pp_filters[2] );
+ ReleaseFilter( pp_filters[2] );
}
msg_Err( p_aout,
"couldn't find filters for the second part of the conversion" );
+ return -1;
}
- *pi_nb_filters = 3;
+ *pi_nb_filters += 3;
+ msg_Dbg( p_aout, "found 3 filters for the whole conversion" );
}
else
{
- *pi_nb_filters = 2;
+ *pi_nb_filters += 2;
+ msg_Dbg( p_aout, "found 2 filters for the whole conversion" );
}
- /* We have enough filters. */
- msg_Dbg( p_aout, "found %d filters for the whole conversion",
- *pi_nb_filters );
return 0;
}
int i_nb_filters )
{
int i;
+ (void)p_aout;
for ( i = 0; i < i_nb_filters; i++ )
{
module_Unneed( pp_filters[i], pp_filters[i]->p_module );
vlc_object_detach( pp_filters[i] );
- vlc_object_destroy( pp_filters[i] );
+ vlc_object_release( pp_filters[i] );
}
}
{
int i;
+ (void)p_aout; /* unused */
+
for ( i = i_nb_filters - 1; i >= 0; i-- )
{
aout_filter_t * p_filter = pp_filters[i];
int i_output_size = p_filter->output.i_bytes_per_frame
- * p_filter->output.i_rate
+ * p_filter->output.i_rate * AOUT_MAX_INPUT_RATE
/ p_filter->output.i_frame_length;
int i_input_size = p_filter->input.i_bytes_per_frame
- * p_filter->input.i_rate
+ * p_filter->input.i_rate * AOUT_MAX_INPUT_RATE
/ p_filter->input.i_frame_length;
p_first_alloc->i_bytes_per_sec = __MAX( p_first_alloc->i_bytes_per_sec,
aout_filter_t * p_filter = pp_filters[i];
aout_buffer_t * p_output_buffer;
+ /* Resamplers can produce slightly more samples than (i_in_nb *
+ * p_filter->output.i_rate / p_filter->input.i_rate) so we need
+ * slightly bigger buffers. */
aout_BufferAlloc( &p_filter->output_alloc,
- (mtime_t)(*pp_input_buffer)->i_nb_samples * 1000000
- / p_filter->input.i_rate, *pp_input_buffer,
- p_output_buffer );
+ ((mtime_t)(*pp_input_buffer)->i_nb_samples + 2)
+ * 1000000 / p_filter->input.i_rate,
+ *pp_input_buffer, p_output_buffer );
if ( p_output_buffer == NULL )
- {
- msg_Err( p_aout, "out of memory" );
return;
- }
/* Please note that p_output_buffer->i_nb_samples & i_nb_bytes
* shall be set by the filter plug-in. */
aout_BufferFree( *pp_input_buffer );
*pp_input_buffer = p_output_buffer;
}
+
+ if( p_output_buffer->i_nb_samples <= 0 )
+ break;
}
}