/*****************************************************************************
* input.c : internal management of input streams for the audio output
*****************************************************************************
- * Copyright (C) 2002 VideoLAN
- * $Id: input.c,v 1.16 2002/10/09 22:54:22 massiot Exp $
+ * Copyright (C) 2002-2004 VideoLAN
+ * $Id$
*
* 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
#include <string.h>
#include <vlc/vlc.h>
+#include <vlc/input.h> /* for input_thread_t and i_pts_delay */
#ifdef HAVE_ALLOCA_H
# include <alloca.h>
#include "audio_output.h"
#include "aout_internal.h"
+static int VisualizationCallback( vlc_object_t *, char const *,
+ vlc_value_t, vlc_value_t, void * );
+static int EqualizerCallback( vlc_object_t *, char const *,
+ vlc_value_t, vlc_value_t, void * );
+static aout_filter_t * allocateUserChannelMixer( aout_instance_t *,
+ audio_sample_format_t *,
+ audio_sample_format_t * );
+
/*****************************************************************************
* aout_InputNew : allocate a new input and rework the filter pipeline
*****************************************************************************/
int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input )
{
- audio_sample_format_t intermediate_format;
+ audio_sample_format_t user_filter_format;
+ audio_sample_format_t intermediate_format;/* input of resampler */
+ vlc_value_t val, text;
+ char * psz_filters;
+ aout_filter_t * p_user_channel_mixer;
+
+ aout_FormatPrint( p_aout, "input", &p_input->input );
/* 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. */
+ /* Prepare format structure */
memcpy( &intermediate_format, &p_aout->mixer.mixer,
sizeof(audio_sample_format_t) );
intermediate_format.i_rate = p_input->input.i_rate;
+
+ /* Try to use the channel mixer chosen by the user */
+ memcpy ( &user_filter_format, &intermediate_format,
+ sizeof(audio_sample_format_t) );
+ user_filter_format.i_physical_channels = p_input->input.i_physical_channels;
+ user_filter_format.i_original_channels = p_input->input.i_original_channels;
+ user_filter_format.i_bytes_per_frame = user_filter_format.i_bytes_per_frame
+ * aout_FormatNbChannels( &user_filter_format )
+ / aout_FormatNbChannels( &intermediate_format );
+ p_user_channel_mixer = allocateUserChannelMixer( p_aout, &user_filter_format,
+ &intermediate_format );
+ /* If it failed, let the main pipeline do channel mixing */
+ if ( ! p_user_channel_mixer )
+ {
+ memcpy ( &user_filter_format, &intermediate_format,
+ sizeof(audio_sample_format_t) );
+ }
+
+ /* Create filters. */
if ( aout_FiltersCreatePipeline( p_aout, p_input->pp_filters,
- &p_input->i_nb_filters, &p_input->input,
- &intermediate_format ) < 0 )
+ &p_input->i_nb_filters,
+ &p_input->input,
+ &user_filter_format
+ ) < 0 )
{
msg_Err( p_aout, "couldn't set an input pipeline" );
aout_FifoDestroy( p_aout, &p_input->fifo );
p_input->b_error = 1;
-
return -1;
}
+ /* Now add user filters */
+ if( var_Type( p_aout, "visual" ) == 0 )
+ {
+ module_t *p_module;
+ var_Create( p_aout, "visual", VLC_VAR_STRING | VLC_VAR_HASCHOICE );
+ text.psz_string = _("Visualizations");
+ var_Change( p_aout, "visual", VLC_VAR_SETTEXT, &text, NULL );
+ val.psz_string = ""; text.psz_string = _("Disable");
+ var_Change( p_aout, "visual", VLC_VAR_ADDCHOICE, &val, &text );
+ val.psz_string = "random"; text.psz_string = _("Random");
+ var_Change( p_aout, "visual", VLC_VAR_ADDCHOICE, &val, &text );
+ val.psz_string = "scope"; text.psz_string = _("Scope");
+ var_Change( p_aout, "visual", VLC_VAR_ADDCHOICE, &val, &text );
+ val.psz_string = "spectrum"; text.psz_string = _("Spectrum");
+ var_Change( p_aout, "visual", VLC_VAR_ADDCHOICE, &val, &text );
+
+ /* Look for goom plugin */
+ p_module = config_FindModule( VLC_OBJECT(p_aout), "goom" );
+ if( p_module )
+ {
+ val.psz_string = "goom"; text.psz_string = _("Goom");
+ var_Change( p_aout, "visual", VLC_VAR_ADDCHOICE, &val, &text );
+ }
+
+ if( var_Get( p_aout, "effect-list", &val ) == VLC_SUCCESS )
+ {
+ var_Set( p_aout, "visual", val );
+ if( val.psz_string ) free( val.psz_string );
+ }
+ var_AddCallback( p_aout, "visual", VisualizationCallback, NULL );
+ }
+
+ if( var_Type( p_aout, "equalizer" ) == 0 )
+ {
+ module_config_t *p_config;
+ int i;
+
+ p_config = config_FindConfig( VLC_OBJECT(p_aout), "equalizer-preset" );
+ if( p_config && p_config->i_list )
+ {
+ var_Create( p_aout, "equalizer",
+ VLC_VAR_STRING | VLC_VAR_HASCHOICE );
+ text.psz_string = _("Equalizer");
+ var_Change( p_aout, "equalizer", VLC_VAR_SETTEXT, &text, NULL );
+
+ val.psz_string = ""; text.psz_string = _("Disable");
+ var_Change( p_aout, "equalizer", VLC_VAR_ADDCHOICE, &val, &text );
+
+ for( i = 0; i < p_config->i_list; i++ )
+ {
+ val.psz_string = p_config->ppsz_list[i];
+ text.psz_string = p_config->ppsz_list_text[i];
+ var_Change( p_aout, "equalizer", VLC_VAR_ADDCHOICE,
+ &val, &text );
+ }
+
+ var_AddCallback( p_aout, "equalizer", EqualizerCallback, NULL );
+ }
+ }
+
+ if( var_Type( p_aout, "audio-filter" ) == 0 )
+ {
+ var_Create( p_aout, "audio-filter",
+ VLC_VAR_STRING | VLC_VAR_DOINHERIT );
+ text.psz_string = _("Audio filters");
+ var_Change( p_aout, "audio-filter", VLC_VAR_SETTEXT, &text, NULL );
+ }
+
+ var_Get( p_aout, "audio-filter", &val );
+ psz_filters = val.psz_string;
+ if( psz_filters && *psz_filters )
+ {
+ char *psz_parser = psz_filters;
+ char *psz_next;
+
+ while( psz_parser && *psz_parser )
+ {
+ aout_filter_t * p_filter;
+
+ if( p_input->i_nb_filters >= AOUT_MAX_FILTERS )
+ {
+ msg_Dbg( p_aout, "max filter reached (%d)", AOUT_MAX_FILTERS );
+ break;
+ }
+
+ while( *psz_parser == ' ' && *psz_parser == ',' )
+ {
+ psz_parser++;
+ }
+ if( ( psz_next = strchr( psz_parser , ',' ) ) )
+ {
+ *psz_next++ = '\0';
+ }
+ if( *psz_parser =='\0' )
+ {
+ break;
+ }
+
+ msg_Dbg( p_aout, "user filter \"%s\"", psz_parser );
+
+ /* Create a VLC object */
+ p_filter = vlc_object_create( p_aout, sizeof(aout_filter_t) );
+ if( p_filter == NULL )
+ {
+ msg_Err( p_aout, "cannot add user filter %s (skipped)",
+ psz_parser );
+ psz_parser = psz_next;
+ continue;
+ }
+
+ vlc_object_attach( p_filter , p_aout );
+ memcpy( &p_filter->input, &user_filter_format,
+ sizeof(audio_sample_format_t) );
+ memcpy( &p_filter->output, &user_filter_format,
+ sizeof(audio_sample_format_t) );
+
+ p_filter->p_module =
+ module_Need( p_filter,"audio filter", psz_parser, VLC_TRUE );
+
+ if( p_filter->p_module== NULL )
+ {
+ msg_Err( p_aout, "cannot add user filter %s (skipped)",
+ psz_parser );
+
+ vlc_object_detach( p_filter );
+ vlc_object_destroy( p_filter );
+ psz_parser = psz_next;
+ continue;
+
+ }
+ p_filter->b_continuity = VLC_FALSE;
+
+ p_input->pp_filters[p_input->i_nb_filters++] = p_filter;
+
+ /* next filter if any */
+ psz_parser = psz_next;
+ }
+ }
+ if( psz_filters ) free( psz_filters );
+
+ /* Attach the user channel mixer */
+ if ( p_user_channel_mixer )
+ {
+ p_input->pp_filters[p_input->i_nb_filters++] = p_user_channel_mixer;
+ }
+
/* 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;
else
{
/* Create resamplers. */
- intermediate_format.i_rate = (p_input->input.i_rate
+ intermediate_format.i_rate = (__MAX(p_input->input.i_rate,
+ p_aout->mixer.mixer.i_rate)
* (100 + AOUT_MAX_RESAMPLING)) / 100;
if ( intermediate_format.i_rate == p_aout->mixer.mixer.i_rate )
{
aout_FiltersDestroyPipeline( p_aout, p_input->pp_filters,
p_input->i_nb_filters );
aout_FifoDestroy( p_aout, &p_input->fifo );
+ var_Destroy( p_aout, "visual" );
p_input->b_error = 1;
return -1;
aout_FiltersHintBuffers( p_aout, p_input->pp_resamplers,
p_input->i_nb_resamplers,
&p_input->input_alloc );
+
+ /* Setup the initial rate of the resampler */
+ p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate;
}
+ p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
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_filters,
p_input->i_nb_filters,
&p_input->input_alloc );
/* i_bytes_per_sec is still == -1 if no filters */
p_input->input_alloc.i_bytes_per_sec = __MAX(
p_input->input_alloc.i_bytes_per_sec,
- p_input->input.i_bytes_per_frame
+ (int)(p_input->input.i_bytes_per_frame
* p_input->input.i_rate
- / p_input->input.i_frame_length );
+ / p_input->input.i_frame_length) );
/* Allocate in the heap, it is more convenient for the decoder. */
p_input->input_alloc.i_alloc_type = AOUT_ALLOC_HEAP;
- p_input->b_error = 0;
+ p_input->b_error = VLC_FALSE;
+ p_input->b_restart = VLC_FALSE;
return 0;
}
int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
aout_buffer_t * p_buffer )
{
- mtime_t start_date, duration;
+ mtime_t start_date;
+
+ if( p_input->b_restart )
+ {
+ aout_fifo_t fifo, dummy_fifo;
+ byte_t *p_first_byte_to_mix;
+
+ vlc_mutex_lock( &p_aout->mixer_lock );
+
+ /* A little trick to avoid loosing our input fifo */
+ aout_FifoInit( p_aout, &dummy_fifo, p_aout->mixer.mixer.i_rate );
+ p_first_byte_to_mix = p_input->p_first_byte_to_mix;
+ fifo = p_input->fifo;
+ p_input->fifo = dummy_fifo;
+ aout_InputDelete( p_aout, p_input );
+ aout_InputNew( p_aout, p_input );
+ p_input->p_first_byte_to_mix = p_first_byte_to_mix;
+ p_input->fifo = fifo;
+
+ vlc_mutex_unlock( &p_aout->mixer_lock );
+ }
/* We don't care if someone changes the start date behind our back after
* this. We'll deal with that when pushing the buffer, and compensate
/* The decoder is _very_ late. This can only happen if the user
* pauses the stream (or if the decoder is buggy, which cannot
* happen :). */
- msg_Warn( p_aout, "computed PTS is out of range (%lld), clearing out",
- start_date );
+ msg_Warn( p_aout, "computed PTS is out of range ("I64Fd"), "
+ "clearing out", mdate() - start_date );
vlc_mutex_lock( &p_aout->input_fifos_lock );
aout_FifoSet( p_aout, &p_input->fifo, 0 );
+ p_input->p_first_byte_to_mix = NULL;
vlc_mutex_unlock( &p_aout->input_fifos_lock );
+ if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE )
+ msg_Warn( p_aout, "timing screwed, stopping resampling" );
+ p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
+ if ( p_input->i_nb_resamplers != 0 )
+ {
+ p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate;
+ p_input->pp_resamplers[0]->b_continuity = VLC_FALSE;
+ }
start_date = 0;
- }
+ }
if ( p_buffer->start_date < mdate() + AOUT_MIN_PREPARE_TIME )
{
/* The decoder gives us f*cked up PTS. It's its business, but we
* can't present it anyway, so drop the buffer. */
- msg_Warn( p_aout, "PTS is out of range (%lld), dropping buffer",
+ msg_Warn( p_aout, "PTS is out of range ("I64Fd"), dropping buffer",
mdate() - p_buffer->start_date );
aout_BufferFree( p_buffer );
-
+ p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
+ if ( p_input->i_nb_resamplers != 0 )
+ {
+ p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate;
+ p_input->pp_resamplers[0]->b_continuity = VLC_FALSE;
+ }
return 0;
}
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 )
+ /* Run the resampler if needed.
+ * We first need to calculate the output rate of this resampler. */
+ if ( ( p_input->i_resampling_type == AOUT_RESAMPLING_NONE ) &&
+ ( start_date < p_buffer->start_date - AOUT_PTS_TOLERANCE
+ || start_date > p_buffer->start_date + AOUT_PTS_TOLERANCE ) &&
+ p_input->i_nb_resamplers > 0 )
{
/* Can happen in several circumstances :
* 1. A problem at the input (clock drift)
* synchronization
* Solution : resample the buffer to avoid a scratch.
*/
- int i_ratio;
- mtime_t old_duration;
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" );
- old_duration = p_buffer->end_date - p_buffer->start_date;
- duration = p_buffer->end_date - start_date;
- i_ratio = (duration * 100) / old_duration;
- /* If the ratio is too != 100, the sound quality will be awful. */
- if ( i_ratio < 100 - AOUT_MAX_RESAMPLING /* % */ )
+ p_input->i_resamp_start_date = mdate();
+ p_input->i_resamp_start_drift = (int)drift;
+
+ if ( drift > 0 )
+ p_input->i_resampling_type = AOUT_RESAMPLING_DOWN;
+ else
+ p_input->i_resampling_type = AOUT_RESAMPLING_UP;
+
+ msg_Warn( p_aout, "buffer is "I64Fd" %s, triggering %ssampling",
+ drift > 0 ? drift : -drift,
+ drift > 0 ? "in advance" : "late",
+ drift > 0 ? "down" : "up");
+ }
+
+ if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE )
+ {
+ /* Resampling has been triggered previously (because of dates
+ * mismatch). We want the resampling to happen progressively so
+ * it isn't too audible to the listener. */
+
+ if( p_input->i_resampling_type == AOUT_RESAMPLING_UP )
{
- duration = (old_duration * (100 - AOUT_MAX_RESAMPLING)) / 100;
+ p_input->pp_resamplers[0]->input.i_rate += 10; /* Hz */
}
- if ( i_ratio > 100 + AOUT_MAX_RESAMPLING /* % */ )
+ else
{
- duration = (old_duration * (100 + AOUT_MAX_RESAMPLING)) / 100;
+ p_input->pp_resamplers[0]->input.i_rate -= 10; /* Hz */
}
- p_input->pp_resamplers[0]->input.i_rate
- = (p_input->input.i_rate * old_duration) / duration;
+ /* Check if everything is back to normal, in which case we can stop the
+ * resampling */
+ if( p_input->pp_resamplers[0]->input.i_rate ==
+ p_input->input.i_rate )
+ {
+ p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
+ msg_Warn( p_aout, "resampling stopped after "I64Fi" usec",
+ mdate() - p_input->i_resamp_start_date );
+ }
+ else if( abs( (int)(p_buffer->start_date - start_date) ) <
+ abs( p_input->i_resamp_start_drift ) / 2 )
+ {
+ /* if we reduced the drift from half, then it is time to switch
+ * back the resampling direction. */
+ if( p_input->i_resampling_type == AOUT_RESAMPLING_UP )
+ p_input->i_resampling_type = AOUT_RESAMPLING_DOWN;
+ else
+ p_input->i_resampling_type = AOUT_RESAMPLING_UP;
+ p_input->i_resamp_start_drift = 0;
+ }
+ else if( p_input->i_resamp_start_drift &&
+ ( abs( (int)(p_buffer->start_date - start_date) ) >
+ abs( p_input->i_resamp_start_drift ) * 3 / 2 ) )
+ {
+ /* If the drift is increasing and not decreasing, than something
+ * is bad. We'd better stop the resampling right now. */
+ msg_Warn( p_aout, "timing screwed, stopping resampling" );
+ p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
+ p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate;
+ }
+ }
+
+ /* Adding the start date will be managed by aout_FifoPush(). */
+ p_buffer->end_date = start_date +
+ (p_buffer->end_date - p_buffer->start_date);
+ p_buffer->start_date = start_date;
+
+ /* Actually run the resampler now. */
+ if ( p_input->i_nb_resamplers > 0 )
+ {
aout_FiltersPlay( p_aout, p_input->pp_resamplers,
p_input->i_nb_resamplers,
&p_buffer );
}
+
+ vlc_mutex_lock( &p_aout->input_fifos_lock );
+ aout_FifoPush( p_aout, &p_input->fifo, p_buffer );
+ vlc_mutex_unlock( &p_aout->input_fifos_lock );
+
+ return 0;
+}
+
+static int ChangeFiltersString( aout_instance_t * p_aout,
+ char *psz_name, vlc_bool_t b_add )
+{
+ vlc_value_t val;
+ char *psz_parser;
+
+ var_Get( p_aout, "audio-filter", &val );
+
+ if( !val.psz_string ) val.psz_string = strdup("");
+
+ psz_parser = strstr( val.psz_string, psz_name );
+
+ if( b_add )
+ {
+ if( !psz_parser )
+ {
+ psz_parser = val.psz_string;
+ asprintf( &val.psz_string, (*val.psz_string) ? "%s,%s" : "%s%s",
+ val.psz_string, psz_name );
+ free( psz_parser );
+ }
+ else
+ {
+ return 0;
+ }
+ }
else
{
- duration = p_buffer->end_date - p_buffer->start_date;
+ if( psz_parser )
+ {
+ memmove( psz_parser, psz_parser + strlen(psz_name) +
+ (*(psz_parser + strlen(psz_name)) == ',' ? 1 : 0 ),
+ strlen(psz_parser + strlen(psz_name)) + 1 );
+ }
+ else
+ {
+ free( val.psz_string );
+ return 0;
+ }
+ }
+
+ var_Set( p_aout, "audio-filter", val );
+ free( val.psz_string );
+ return 1;
+}
+
+static int VisualizationCallback( vlc_object_t *p_this, char const *psz_cmd,
+ vlc_value_t oldval, vlc_value_t newval, void *p_data )
+{
+ aout_instance_t *p_aout = (aout_instance_t *)p_this;
+ char *psz_mode = newval.psz_string;
+ vlc_value_t val;
+ int i;
- if ( p_input->input.i_rate != p_aout->mixer.mixer.i_rate )
+ if( !psz_mode || !*psz_mode )
+ {
+ ChangeFiltersString( p_aout, "goom", VLC_FALSE );
+ ChangeFiltersString( p_aout, "visual", VLC_FALSE );
+ }
+ else
+ {
+ if( !strcmp( "goom", psz_mode ) )
{
- /* Standard resampling is needed ! */
- p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate;
+ ChangeFiltersString( p_aout, "visual", VLC_FALSE );
+ ChangeFiltersString( p_aout, "goom", VLC_TRUE );
+ }
+ else
+ {
+ val.psz_string = psz_mode;
+ var_Create( p_aout, "effect-list", VLC_VAR_STRING );
+ var_Set( p_aout, "effect-list", val );
- aout_FiltersPlay( p_aout, p_input->pp_resamplers,
- p_input->i_nb_resamplers,
- &p_buffer );
+ ChangeFiltersString( p_aout, "goom", VLC_FALSE );
+ ChangeFiltersString( p_aout, "visual", VLC_TRUE );
}
}
- /* Adding the start date will be managed by aout_FifoPush(). */
- p_buffer->start_date = start_date;
- p_buffer->end_date = start_date + duration;
+ /* That sucks */
+ for( i = 0; i < p_aout->i_nb_inputs; i++ )
+ {
+ p_aout->pp_inputs[i]->b_restart = VLC_TRUE;
+ }
- vlc_mutex_lock( &p_aout->input_fifos_lock );
- aout_FifoPush( p_aout, &p_input->fifo, p_buffer );
- vlc_mutex_unlock( &p_aout->input_fifos_lock );
+ return VLC_SUCCESS;
+}
- return 0;
+static int EqualizerCallback( vlc_object_t *p_this, char const *psz_cmd,
+ vlc_value_t oldval, vlc_value_t newval, void *p_data )
+{
+ aout_instance_t *p_aout = (aout_instance_t *)p_this;
+ char *psz_mode = newval.psz_string;
+ vlc_value_t val;
+ int i;
+ int i_ret;
+
+ if( !psz_mode || !*psz_mode )
+ {
+ i_ret = ChangeFiltersString( p_aout, "equalizer", VLC_FALSE );
+ }
+ else
+ {
+ val.psz_string = psz_mode;
+ var_Create( p_aout, "equalizer-preset", VLC_VAR_STRING );
+ var_Set( p_aout, "equalizer-preset", val );
+ i_ret = ChangeFiltersString( p_aout, "equalizer", VLC_TRUE );
+
+ }
+
+ /* That sucks */
+ if( i_ret == 1 )
+ {
+ for( i = 0; i < p_aout->i_nb_inputs; i++ )
+ {
+ p_aout->pp_inputs[i]->b_restart = VLC_TRUE;
+ }
+ }
+
+ return VLC_SUCCESS;
+}
+
+static aout_filter_t * allocateUserChannelMixer( aout_instance_t * p_aout,
+ audio_sample_format_t * p_input_format,
+ audio_sample_format_t * p_output_format )
+{
+ aout_filter_t * p_channel_mixer;
+
+ /* Retreive user preferred channel mixer */
+ char * psz_name = config_GetPsz( p_aout, "audio-channel-mixer" );
+
+ /* Not specified => let the main pipeline do the mixing */
+ if ( ! psz_name ) return NULL;
+
+ /* Debug information */
+ aout_FormatsPrint( p_aout, "channel mixer", p_input_format,
+ p_output_format );
+
+ /* Create a VLC object */
+ p_channel_mixer = vlc_object_create( p_aout, sizeof(aout_filter_t) );
+ if( p_channel_mixer == NULL )
+ {
+ msg_Err( p_aout, "cannot add user channel mixer %s", psz_name );
+ return NULL;
+ }
+ vlc_object_attach( p_channel_mixer , p_aout );
+
+ /* Attach the suitable module */
+ memcpy( &p_channel_mixer->input, p_input_format,
+ sizeof(audio_sample_format_t) );
+ memcpy( &p_channel_mixer->output, p_output_format,
+ sizeof(audio_sample_format_t) );
+ p_channel_mixer->p_module =
+ module_Need( p_channel_mixer,"audio filter", psz_name, VLC_TRUE );
+ if( p_channel_mixer->p_module== NULL )
+ {
+ msg_Err( p_aout, "cannot add user channel mixer %s", psz_name );
+ vlc_object_detach( p_channel_mixer );
+ vlc_object_destroy( p_channel_mixer );
+ return NULL;
+ }
+ p_channel_mixer->b_continuity = VLC_FALSE;
+
+ /* Ok */
+ return p_channel_mixer;
}
projects / vlc / blobdiff