/*
* filter_audioconvert.c -- convert from one audio format to another
- * Copyright (C) 2009 Ushodaya Enterprises Limited
+ * Copyright (C) 2009-2012 Ushodaya Enterprises Limited
* Author: Dan Dennedy <dan@dennedy.org>
*
* This library is free software; you can redistribute it and/or
#include <stdio.h>
#include <stdlib.h>
+#ifndef CLAMP
+#define CLAMP( x, min, max ) (x) < (min) ? (min) : (x) > (max) ? (max) : (x)
+#endif
+
static int convert_audio( mlt_frame frame, void **audio, mlt_audio_format *format, mlt_audio_format requested_format )
{
- int error = 0;
+ 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 = 0;
+ int size = mlt_audio_format_size( requested_format, samples, channels );
if ( *format != requested_format )
{
{
case mlt_audio_s32:
{
- size = channels * samples * sizeof(int32_t);
int32_t *buffer = mlt_pool_alloc( size );
int32_t *p = buffer;
int c;
}
}
*audio = buffer;
+ error = 0;
break;
}
case mlt_audio_float:
{
- size = channels * samples * sizeof(float);
float *buffer = mlt_pool_alloc( size );
float *p = buffer;
int c;
}
}
*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;
+ *p++ = CLAMP( f, -1.0f, 1.0f );
+ }
+ *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:
- error = 1;
+ break;
}
break;
case mlt_audio_s32:
{
case mlt_audio_s16:
{
- size = channels * samples * sizeof(int16_t);
int16_t *buffer = mlt_pool_alloc( size );
int16_t *p = buffer;
int32_t *q = (int32_t*) *audio;
for ( c = 0; c < channels; c++ )
*p++ = *( q + c * samples + s ) >> 16;
*audio = buffer;
+ error = 0;
break;
}
case mlt_audio_float:
{
- size = channels * samples * sizeof(float);
float *buffer = mlt_pool_alloc( size );
float *p = buffer;
int32_t *q = (int32_t*) *audio;
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;
+ *p++ = CLAMP( f, -1.0f, 1.0f );
+ }
+ *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:
- error = 1;
+ break;
}
break;
case mlt_audio_float:
{
case mlt_audio_s16:
{
- size = channels * samples * sizeof(int16_t);
int16_t *buffer = mlt_pool_alloc( size );
int16_t *p = buffer;
float *q = (float*) *audio;
for ( c = 0; c < channels; c++ )
{
float f = *( q + c * samples + s );
- f = f > 1.0 ? 1.0 : f < -1.0 ? -1.0 : f;
+ f = CLAMP( f, -1.0f, 1.0f );
*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 = CLAMP( f, -1.0f, 1.0f );
+ int64_t pcm = ( f > 0.0f ? 2147483647LL : 2147483648LL ) * f;
+ *p++ = CLAMP( pcm, -2147483648LL, 2147483647LL );
+ }
+ *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 = CLAMP( f, -1.0f, 1.0f );
+ int64_t pcm = ( f > 0.0f ? 2147483647LL : 2147483648LL ) * f;
+ *p++ = CLAMP( pcm, -2147483648LL, 2147483647LL );
+ }
+ *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 = CLAMP( f, -1.0f, 1.0f );
+ *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 = CLAMP( f, -1.0f , 1.0f );
+ *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:
{
- size = channels * samples * sizeof(int32_t);
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 = CLAMP( f, -1.0f , 1.0f );
+ int64_t pcm = ( f > 0.0f ? 2147483647LL : 2147483648LL ) * f;
+ *p++ = CLAMP( pcm, -2147483648LL, 2147483647LL );
+ 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 = CLAMP( f, -1.0f , 1.0f );
+ int64_t pcm = ( f > 0.0f ? 2147483647LL : 2147483648LL ) * f;
+ *p++ = CLAMP( pcm, -2147483648LL, 2147483647LL );
+ }
+ *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++ = ( f > 0 ? 2147483647LL : 2147483648LL ) * f;
+ f = CLAMP( f, -1.0f , 1.0f );
+ *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;
+ *p++ = CLAMP( f, -1.0f, 1.0f );
+ }
+ *audio = buffer;
+ error = 0;
break;
}
default:
- error = 1;
+ break;
}
break;
default:
- error = 1;
+ break;
}
}
- if ( size )
+ if ( !error )
{
mlt_frame_set_audio( frame, *audio, requested_format, size, mlt_pool_release );
*format = requested_format;
mlt_filter filter_audioconvert_init( mlt_profile profile, mlt_service_type type, const char *id, char *arg )
{
- mlt_filter this = calloc( sizeof( struct mlt_filter_s ), 1 );
+ mlt_filter this = calloc( 1, sizeof( struct mlt_filter_s ) );
if ( mlt_filter_init( this, this ) == 0 )
this->process = filter_process;
return this;
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