int mlt_consumer_init( mlt_consumer self, void *child, mlt_profile profile )
{
int error = 0;
memset( self, 0, sizeof( struct mlt_consumer_s ) );
self->child = child;
error = mlt_service_init( &self->parent, self );
if ( error == 0 )
{
// Get the properties from the service
mlt_properties properties = MLT_SERVICE_PROPERTIES( &self->parent );
// Apply profile to properties
if ( profile == NULL )
{
// Normally the application creates the profile and controls its lifetime
// This is the fallback exception handling
profile = mlt_profile_init( NULL );
mlt_properties properties = MLT_CONSUMER_PROPERTIES( self );
mlt_properties_set_data( properties, "_profile", profile, 0, (mlt_destructor)mlt_profile_close, NULL );
}
apply_profile_properties( self, profile, properties );
// Default rescaler for all consumers
mlt_properties_set( properties, "rescale", "bilinear" );
// Default read ahead buffer size
mlt_properties_set_int( properties, "buffer", 25 );
mlt_properties_set_int( properties, "drop_max", 5 );
// Default audio frequency and channels
mlt_properties_set_int( properties, "frequency", 48000 );
mlt_properties_set_int( properties, "channels", 2 );
// Default of all consumers is real time
mlt_properties_set_int( properties, "real_time", 1 );
// Default to environment test card
mlt_properties_set( properties, "test_card", mlt_environment( "MLT_TEST_CARD" ) );
// Hmm - default all consumers to yuv422 :-/
self->format = mlt_image_yuv422;
mlt_properties_set( properties, "mlt_image_format", mlt_image_format_name( self->format ) );
mlt_properties_set( properties, "mlt_audio_format", mlt_audio_format_name( mlt_audio_s16 ) );
mlt_events_register( properties, "consumer-frame-show", ( mlt_transmitter )mlt_consumer_frame_show );
mlt_events_register( properties, "consumer-frame-render", ( mlt_transmitter )mlt_consumer_frame_render );
mlt_events_register( properties, "consumer-stopped", NULL );
mlt_events_listen( properties, self, "consumer-frame-show", ( mlt_listener )on_consumer_frame_show );
// Register a property-changed listener to handle the profile property -
// subsequent properties can override the profile
self->event_listener = mlt_events_listen( properties, self, "property-changed", ( mlt_listener )mlt_consumer_property_changed );
// Create the push mutex and condition
pthread_mutex_init( &self->put_mutex, NULL );
pthread_cond_init( &self->put_cond, NULL );
}
return error;
}
static int convert_audio( mlt_frame frame, void **audio, mlt_audio_format *format, mlt_audio_format requested_format )
{
int error = 1;
mlt_properties properties = MLT_FRAME_PROPERTIES( frame );
int channels = mlt_properties_get_int( properties, "audio_channels" );
int samples = mlt_properties_get_int( properties, "audio_samples" );
int size = mlt_audio_format_size( requested_format, samples, channels );
if ( *format != requested_format )
{
mlt_log_debug( NULL, "[filter audioconvert] %s -> %s %d channels %d samples\n",
mlt_audio_format_name( *format ), mlt_audio_format_name( requested_format ),
channels, samples );
switch ( *format )
{
case mlt_audio_s16:
switch ( requested_format )
{
case mlt_audio_s32:
{
int32_t *buffer = mlt_pool_alloc( size );
int32_t *p = buffer;
int c;
for ( c = 0; c < channels; c++ )
{
int16_t *q = (int16_t*) *audio + c;
int i = samples + 1;
while ( --i )
{
*p++ = (int32_t) *q << 16;
q += channels;
}
}
*audio = buffer;
error = 0;
break;
}
case mlt_audio_float:
{
float *buffer = mlt_pool_alloc( size );
float *p = buffer;
int c;
for ( c = 0; c < channels; c++ )
{
int16_t *q = (int16_t*) *audio + c;
int i = samples + 1;
while ( --i )
{
*p++ = (float)( *q ) / 32768.0;
q += channels;
}
}
*audio = buffer;
error = 0;
break;
}
case mlt_audio_s32le:
{
int32_t *buffer = mlt_pool_alloc( size );
int32_t *p = buffer;
int16_t *q = (int16_t*) *audio;
int i = samples * channels + 1;
while ( --i )
*p++ = (int32_t) *q++ << 16;
*audio = buffer;
error = 0;
break;
}
case mlt_audio_f32le:
{
float *buffer = mlt_pool_alloc( size );
float *p = buffer;
int16_t *q = (int16_t*) *audio;
int i = samples * channels + 1;
while ( --i )
{
float f = (float)( *q++ ) / 32768.0;
f = f > 1.0 ? 1.0 : f < -1.0 ? -1.0 : f;
*p++ = f;
}
*audio = buffer;
error = 0;
break;
}
case mlt_audio_u8:
{
uint8_t *buffer = mlt_pool_alloc( size );
uint8_t *p = buffer;
int16_t *q = (int16_t*) *audio;
int i = samples * channels + 1;
while ( --i )
*p++ = ( *q++ >> 8 ) + 128;
*audio = buffer;
error = 0;
break;
}
default:
break;
}
break;
case mlt_audio_s32:
switch ( requested_format )
{
case mlt_audio_s16:
{
int16_t *buffer = mlt_pool_alloc( size );
int16_t *p = buffer;
int32_t *q = (int32_t*) *audio;
int s, c;
for ( s = 0; s < samples; s++ )
for ( c = 0; c < channels; c++ )
*p++ = *( q + c * samples + s ) >> 16;
*audio = buffer;
error = 0;
break;
}
case mlt_audio_float:
{
float *buffer = mlt_pool_alloc( size );
float *p = buffer;
int32_t *q = (int32_t*) *audio;
int i = samples * channels + 1;
while ( --i )
*p++ = (float)( *q++ ) / 2147483648.0;
*audio = buffer;
error = 0;
break;
}
case mlt_audio_s32le:
{
int32_t *buffer = mlt_pool_alloc( size );
int32_t *p = buffer;
int32_t *q = (int32_t*) *audio;
int s, c;
for ( s = 0; s < samples; s++ )
for ( c = 0; c < channels; c++ )
*p++ = *( q + c * samples + s );
*audio = buffer;
error = 0;
break;
}
case mlt_audio_f32le:
{
float *buffer = mlt_pool_alloc( size );
float *p = buffer;
int32_t *q = (int32_t*) *audio;
int s, c;
for ( s = 0; s < samples; s++ )
for ( c = 0; c < channels; c++ )
{
float f = (float)( *( q + c * samples + s ) ) / 2147483648.0;
f = f > 1.0 ? 1.0 : f < -1.0 ? -1.0 : f;
*p++ = f;
}
*audio = buffer;
error = 0;
break;
}
case mlt_audio_u8:
{
uint8_t *buffer = mlt_pool_alloc( size );
uint8_t *p = buffer;
int32_t *q = (int32_t*) *audio;
int s, c;
for ( s = 0; s < samples; s++ )
for ( c = 0; c < channels; c++ )
*p++ = ( q[c * samples + s] >> 24 ) + 128;
*audio = buffer;
error = 0;
break;
}
default:
break;
}
break;
case mlt_audio_float:
switch ( requested_format )
{
case mlt_audio_s16:
{
int16_t *buffer = mlt_pool_alloc( size );
int16_t *p = buffer;
float *q = (float*) *audio;
int s, c;
for ( s = 0; s < samples; s++ )
for ( c = 0; c < channels; c++ )
{
float f = *( q + c * samples + s );
f = f > 1.0 ? 1.0 : f < -1.0 ? -1.0 : f;
*p++ = 32767 * f;
}
*audio = buffer;
error = 0;
break;
}
case mlt_audio_s32:
{
int32_t *buffer = mlt_pool_alloc( size );
int32_t *p = buffer;
float *q = (float*) *audio;
int i = samples * channels + 1;
while ( --i )
{
float f = *q++;
f = f > 1.0 ? 1.0 : f < -1.0 ? -1.0 : f;
*p++ = ( f > 0 ? 2147483647LL : 2147483648LL ) * f;
}
*audio = buffer;
error = 0;
break;
}
case mlt_audio_s32le:
{
int32_t *buffer = mlt_pool_alloc( size );
int32_t *p = buffer;
float *q = (float*) *audio;
int s, c;
for ( s = 0; s < samples; s++ )
for ( c = 0; c < channels; c++ )
{
float f = *( q + c * samples + s );
f = f > 1.0 ? 1.0 : f < -1.0 ? -1.0 : f;
*p++ = ( f > 0 ? 2147483647LL : 2147483648LL ) * f;
}
*audio = buffer;
error = 0;
break;
}
case mlt_audio_f32le:
{
float *buffer = mlt_pool_alloc( size );
float *p = buffer;
float *q = (float*) *audio;
int s, c;
for ( s = 0; s < samples; s++ )
for ( c = 0; c < channels; c++ )
*p++ = *( q + c * samples + s );
*audio = buffer;
error = 0;
break;
}
case mlt_audio_u8:
{
uint8_t *buffer = mlt_pool_alloc( size );
uint8_t *p = buffer;
float *q = (float*) *audio;
int s, c;
for ( s = 0; s < samples; s++ )
for ( c = 0; c < channels; c++ )
{
float f = *( q + c * samples + s );
f = f > 1.0 ? 1.0 : f < -1.0 ? -1.0 : f;
*p++ = ( 127 * f ) + 128;
}
*audio = buffer;
error = 0;
break;
}
default:
break;
}
break;
case mlt_audio_s32le:
switch ( requested_format )
{
case mlt_audio_s16:
{
int16_t *buffer = mlt_pool_alloc( size );
int16_t *p = buffer;
int32_t *q = (int32_t*) *audio;
int i = samples * channels + 1;
while ( --i )
*p++ = *q++ >> 16;
*audio = buffer;
error = 0;
break;
}
case mlt_audio_s32:
{
int32_t *buffer = mlt_pool_alloc( size );
int32_t *p = buffer;
int c;
for ( c = 0; c < channels; c++ )
{
int32_t *q = (int32_t*) *audio + c;
int i = samples + 1;
while ( --i )
{
*p++ = *q;
q += channels;
}
}
*audio = buffer;
error = 0;
break;
}
case mlt_audio_float:
{
float *buffer = mlt_pool_alloc( size );
float *p = buffer;
int c;
for ( c = 0; c < channels; c++ )
{
int32_t *q = (int32_t*) *audio + c;
int i = samples + 1;
while ( --i )
{
*p++ = (float)( *q ) / 2147483648.0;
q += channels;
}
}
*audio = buffer;
error = 0;
break;
}
case mlt_audio_f32le:
{
float *buffer = mlt_pool_alloc( size );
float *p = buffer;
int32_t *q = (int32_t*) *audio;
int i = samples * channels + 1;
while ( --i )
*p++ = (float)( *q++ ) / 2147483648.0;
*audio = buffer;
error = 0;
break;
}
case mlt_audio_u8:
{
uint8_t *buffer = mlt_pool_alloc( size );
uint8_t *p = buffer;
int32_t *q = (int32_t*) *audio;
int i = samples * channels + 1;
while ( --i )
*p++ = ( *q++ >> 24 ) + 128;
*audio = buffer;
error = 0;
break;
}
default:
break;
}
break;
case mlt_audio_f32le:
switch ( requested_format )
{
case mlt_audio_s16:
{
int16_t *buffer = mlt_pool_alloc( size );
int16_t *p = buffer;
float *q = (float*) *audio;
int i = samples * channels + 1;
while ( --i )
{
float f = *q++;
f = f > 1.0 ? 1.0 : f < -1.0 ? -1.0 : f;
*p++ = 32767 * f;
}
*audio = buffer;
error = 0;
break;
}
case mlt_audio_float:
{
float *buffer = mlt_pool_alloc( size );
float *p = buffer;
int c;
for ( c = 0; c < channels; c++ )
{
float *q = (float*) *audio + c;
int i = samples + 1;
while ( --i )
{
*p++ = *q;
q += channels;
}
}
*audio = buffer;
error = 0;
break;
}
case mlt_audio_s32:
{
int32_t *buffer = mlt_pool_alloc( size );
int32_t *p = buffer;
int c;
for ( c = 0; c < channels; c++ )
{
float *q = (float*) *audio + c;
int i = samples + 1;
while ( --i )
{
float f = *q;
f = f > 1.0 ? 1.0 : f < -1.0 ? -1.0 : f;
*p++ = ( f > 0 ? 2147483647LL : 2147483648LL ) * f;
q += channels;
}
}
*audio = buffer;
error = 0;
break;
}
case mlt_audio_s32le:
{
int32_t *buffer = mlt_pool_alloc( size );
int32_t *p = buffer;
float *q = (float*) *audio;
int i = samples * channels + 1;
while ( --i )
{
float f = *q++;
f = f > 1.0 ? 1.0 : f < -1.0 ? -1.0 : f;
*p++ = ( f > 0 ? 2147483647LL : 2147483648LL ) * f;
}
*audio = buffer;
error = 0;
break;
}
case mlt_audio_u8:
{
uint8_t *buffer = mlt_pool_alloc( size );
uint8_t *p = buffer;
float *q = (float*) *audio;
int i = samples * channels + 1;
while ( --i )
{
float f = *q++;
f = f > 1.0 ? 1.0 : f < -1.0 ? -1.0 : f;
*p++ = ( 127 * f ) + 128;
}
*audio = buffer;
error = 0;
break;
}
default:
break;
}
break;
case mlt_audio_u8:
switch ( requested_format )
{
case mlt_audio_s32:
{
int32_t *buffer = mlt_pool_alloc( size );
int32_t *p = buffer;
int c;
for ( c = 0; c < channels; c++ )
{
uint8_t *q = (uint8_t*) *audio + c;
int i = samples + 1;
while ( --i )
{
*p++ = ( (int32_t) *q - 128 ) << 24;
q += channels;
}
}
*audio = buffer;
error = 0;
break;
}
case mlt_audio_float:
{
float *buffer = mlt_pool_alloc( size );
float *p = buffer;
int c;
for ( c = 0; c < channels; c++ )
{
uint8_t *q = (uint8_t*) *audio + c;
int i = samples + 1;
while ( --i )
{
*p++ = ( (float) *q - 128 ) / 256.0f;
q += channels;
}
}
*audio = buffer;
error = 0;
break;
}
case mlt_audio_s16:
{
int16_t *buffer = mlt_pool_alloc( size );
int16_t *p = buffer;
uint8_t *q = (uint8_t*) *audio;
int i = samples * channels + 1;
while ( --i )
*p++ = ( (int16_t) *q++ - 128 ) << 8;
*audio = buffer;
error = 0;
break;
}
case mlt_audio_s32le:
{
int32_t *buffer = mlt_pool_alloc( size );
int32_t *p = buffer;
uint8_t *q = (uint8_t*) *audio;
int i = samples * channels + 1;
while ( --i )
*p++ = ( (int32_t) *q++ - 128 ) << 24;
*audio = buffer;
error = 0;
break;
}
case mlt_audio_f32le:
{
float *buffer = mlt_pool_alloc( size );
float *p = buffer;
uint8_t *q = (uint8_t*) *audio;
int i = samples * channels + 1;
while ( --i )
{
float f = ( (float) *q++ - 128 ) / 256.0f;
f = f > 1.0 ? 1.0 : f < -1.0 ? -1.0 : f;
*p++ = f;
}
*audio = buffer;
error = 0;
break;
}
default:
break;
}
break;
default:
break;
}
}
