X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;ds=inline;f=src%2Fmodules%2Fcore%2Ffilter_audioconvert.c;h=a38219286f7c10968b2589296f240b3e0d91d398;hb=31e70f01d93d5465d96b2842780fccd1d9bc6ade;hp=eeff0792b91075256f245f150fe4b8e3de5c8dce;hpb=835fd43a4200772b82f52e9714cd2f02c0f23db0;p=mlt

diff --git a/src/modules/core/filter_audioconvert.c b/src/modules/core/filter_audioconvert.c
index eeff0792..a3821928 100644
--- a/src/modules/core/filter_audioconvert.c
+++ b/src/modules/core/filter_audioconvert.c
@@ -1,6 +1,6 @@
 /*
  * 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
@@ -27,7 +27,7 @@
 
 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" );
@@ -59,6 +59,7 @@ static int convert_audio( mlt_frame frame, void **audio, mlt_audio_format *forma
 					}
 				}
 				*audio = buffer;
+				error = 0;
 				break;
 			}
 			case mlt_audio_float:
@@ -77,10 +78,51 @@ static int convert_audio( mlt_frame frame, void **audio, mlt_audio_format *forma
 					}
 				}
 				*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:
-				error = 1;
+				break;
 			}
 			break;
 		case mlt_audio_s32:
@@ -96,6 +138,7 @@ static int convert_audio( mlt_frame frame, void **audio, mlt_audio_format *forma
 					for ( c = 0; c < channels; c++ )
 						*p++ = *( q + c * samples + s ) >> 16;
 				*audio = buffer;
+				error = 0;
 				break;
 			}
 			case mlt_audio_float:
@@ -107,10 +150,54 @@ static int convert_audio( mlt_frame frame, void **audio, mlt_audio_format *forma
 				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:
-				error = 1;
+				break;
 			}
 			break;
 		case mlt_audio_float:
@@ -130,6 +217,7 @@ static int convert_audio( mlt_frame frame, void **audio, mlt_audio_format *forma
 						*p++ = 32767 * f;
 					}
 				*audio = buffer;
+				error = 0;
 				break;
 			}
 			case mlt_audio_s32:
@@ -145,14 +233,323 @@ static int convert_audio( mlt_frame frame, void **audio, mlt_audio_format *forma
 					*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:
-				error = 1;
+				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:
-			error = 1;
+			break;
 		}
 	}
 	if ( !error )
@@ -177,7 +574,7 @@ static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
 
 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;