* Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
*/
-#include "filter_resample.h"
-
+#include <framework/mlt_filter.h>
#include <framework/mlt_frame.h>
+#include <framework/mlt_log.h>
#include <stdio.h>
#include <stdlib.h>
#include <samplerate.h>
-#define __USE_ISOC99 1
-#include <math.h>
+#include <string.h>
-#define BUFFER_LEN 20480
+// BUFFER_LEN is based on a maximum of 96KHz, 5 fps, 8 channels
+// TODO: dynamically allocate larger buffer size
+#define BUFFER_LEN ((96000/5) * 8 * sizeof(float))
#define RESAMPLE_TYPE SRC_SINC_FASTEST
/** Get the audio.
*/
-static int resample_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
+static int resample_get_audio( mlt_frame frame, void **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
{
- // Get the properties of the a frame
- mlt_properties properties = mlt_frame_properties( frame );
- int output_rate = mlt_properties_get_int( properties, "resample.frequency" );
- SRC_STATE *state = mlt_properties_get_data( properties, "resample.state", NULL );
- SRC_DATA data;
- float *input_buffer = mlt_properties_get_data( properties, "resample.input_buffer", NULL );
- float *output_buffer = mlt_properties_get_data( properties, "resample.output_buffer", NULL );
- int i;
+ // Get the filter service
+ mlt_filter filter = mlt_frame_pop_audio( frame );
+
+ // Get the filter properties
+ mlt_properties filter_properties = MLT_FILTER_PROPERTIES( filter );
+
+ // Get the resample information
+ int output_rate = mlt_properties_get_int( filter_properties, "frequency" );
+ // If no resample frequency is specified, default to requested value
if ( output_rate == 0 )
output_rate = *frequency;
- // Restore the original get_audio
- frame->get_audio = mlt_properties_get_data( properties, "resample.get_audio", NULL );
-
// Get the producer's audio
- mlt_frame_get_audio( frame, buffer, format, frequency, channels, samples );
-
- // Return now if now work to do
- if ( output_rate == *frequency )
- return 0;
-
- //fprintf( stderr, "resample_get_audio: input_rate %d output_rate %d\n", *frequency, output_rate );
-
- // Convert to floating point
- for ( i = 0; i < *samples * *channels; ++i )
- input_buffer[ i ] = ( float )( (*buffer)[ i ] ) / 32768;
-
- // Resample
- data.data_in = input_buffer;
- data.data_out = output_buffer;
- data.src_ratio = ( float ) output_rate / ( float ) *frequency;
- data.input_frames = *samples;
- data.output_frames = BUFFER_LEN / *channels;
- data.end_of_input = 0;
- i = src_process( state, &data );
- if ( i == 0 )
+ int error = mlt_frame_get_audio( frame, buffer, format, frequency, channels, samples );
+ if ( error ) return error;
+
+ // Return now if no work to do
+ if ( output_rate != *frequency && *frequency > 0 && *channels > 0 )
{
- if ( data.output_frames_gen > *samples )
+ mlt_log_debug( MLT_FILTER_SERVICE(filter), "channels %d samples %d frequency %d -> %d\n",
+ *channels, *samples, *frequency, output_rate );
+
+ // Do not convert to float unless we need to change the rate
+ if ( *format != mlt_audio_f32le )
+ frame->convert_audio( frame, buffer, format, mlt_audio_f32le );
+
+ mlt_service_lock( MLT_FILTER_SERVICE(filter) );
+
+ SRC_DATA data;
+ data.data_in = *buffer;
+ data.data_out = mlt_properties_get_data( filter_properties, "output_buffer", NULL );
+ data.src_ratio = ( float ) output_rate / ( float ) *frequency;
+ data.input_frames = *samples;
+ data.output_frames = BUFFER_LEN / *channels;
+ data.end_of_input = 0;
+
+ SRC_STATE *state = mlt_properties_get_data( filter_properties, "state", NULL );
+ if ( !state || mlt_properties_get_int( filter_properties, "channels" ) != *channels )
+ {
+ // Recreate the resampler if the number of channels changed
+ state = src_new( RESAMPLE_TYPE, *channels, &error );
+ mlt_properties_set_data( filter_properties, "state", state, 0, (mlt_destructor) src_delete, NULL );
+ mlt_properties_set_int( filter_properties, "channels", *channels );
+ }
+
+ // Resample the audio
+ error = src_process( state, &data );
+ if ( !error )
{
- *buffer = mlt_pool_alloc( data.output_frames_gen * *channels * sizeof( int16_t ) );
- mlt_properties_set_data( properties, "audio", *buffer, *channels * data.output_frames_gen * 2, mlt_pool_release, NULL );
+ // Update output variables
+ *samples = data.output_frames_gen;
+ *frequency = output_rate;
+ *buffer = data.data_out;
}
- *samples = data.output_frames_gen;
- *frequency = output_rate;
-
- // Convert from floating back to signed 16bit
- for ( i = 0; i < *samples * *channels; ++i )
+ else
{
- float sample = output_buffer[ i ];
- if ( sample > 1.0 )
- sample = 1.0;
- if ( sample < -1.0 )
- sample = -1.0;
- if ( sample >= 0 )
- (*buffer)[ i ] = lrint( 32767.0 * sample );
- else
- (*buffer)[ i ] = lrint( 32768.0 * sample );
+ mlt_log_error( MLT_FILTER_SERVICE( filter ), "%s %d,%d,%d\n", src_strerror( error ), *frequency, *samples, output_rate );
}
+ mlt_service_unlock( MLT_FILTER_SERVICE(filter) );
}
- else
- fprintf( stderr, "resample_get_audio: %s %d,%d,%d\n", src_strerror( i ), *frequency, *samples, output_rate );
-
- return 0;
+
+ return error;
}
/** Filter processing.
static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
{
- mlt_properties properties = mlt_filter_properties( this );
- mlt_properties frame_props = mlt_frame_properties( frame );
-
- // Propogate the frequency property if supplied
- if ( mlt_properties_get( properties, "frequency" ) != NULL )
- mlt_properties_set_int( frame_props, "resample.frequency", mlt_properties_get_int( properties, "frequency" ) );
-
- // Propogate the other properties
- mlt_properties_set_int( frame_props, "resample.channels", mlt_properties_get_int( properties, "channels" ) );
- mlt_properties_set_data( frame_props, "resample.state", mlt_properties_get_data( properties, "state", NULL ), 0, NULL, NULL );
- mlt_properties_set_data( frame_props, "resample.input_buffer", mlt_properties_get_data( properties, "input_buffer", NULL ), 0, NULL, NULL );
- mlt_properties_set_data( frame_props, "resample.output_buffer", mlt_properties_get_data( properties, "output_buffer", NULL ), 0, NULL, NULL );
-
- // Backup the original get_audio (it's still needed)
- mlt_properties_set_data( frame_props, "resample.get_audio", frame->get_audio, 0, NULL, NULL );
-
- // Override the get_audio method
- frame->get_audio = resample_get_audio;
+ if ( mlt_frame_is_test_audio( frame ) == 0 )
+ {
+ mlt_frame_push_audio( frame, this );
+ mlt_frame_push_audio( frame, resample_get_audio );
+ }
return frame;
}
/** Constructor for the filter.
*/
-mlt_filter filter_resample_init( char *arg )
+mlt_filter filter_resample_init( mlt_profile profile, mlt_service_type type, const char *id, char *arg )
{
mlt_filter this = mlt_filter_new( );
if ( this != NULL )
SRC_STATE *state = src_new( RESAMPLE_TYPE, 2 /* channels */, &error );
if ( error == 0 )
{
- void *input_buffer = mlt_pool_alloc( BUFFER_LEN );
void *output_buffer = mlt_pool_alloc( BUFFER_LEN );
this->process = filter_process;
if ( arg != NULL )
- mlt_properties_set_int( mlt_filter_properties( this ), "frequency", atoi( arg ) );
- mlt_properties_set_int( mlt_filter_properties( this ), "channels", 2 );
- mlt_properties_set_data( mlt_filter_properties( this ), "state", state, 0, (mlt_destructor)src_delete, NULL );
- mlt_properties_set_data( mlt_filter_properties( this ), "input_buffer", input_buffer, BUFFER_LEN, mlt_pool_release, NULL );
- mlt_properties_set_data( mlt_filter_properties( this ), "output_buffer", output_buffer, BUFFER_LEN, mlt_pool_release, NULL );
+ mlt_properties_set_int( MLT_FILTER_PROPERTIES( this ), "frequency", atoi( arg ) );
+ mlt_properties_set_int( MLT_FILTER_PROPERTIES( this ), "channels", 2 );
+ mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "state", state, 0, (mlt_destructor)src_delete, NULL );
+ mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "output_buffer", output_buffer, BUFFER_LEN, mlt_pool_release, NULL );
}
else
{
}
return this;
}
-
projects / mlt / blobdiff