X-Git-Url: http://research.m1stereo.tv/gitweb?a=blobdiff_plain;f=src%2Fframework%2Fmlt_frame.c;h=880cdef4395fa3b61bb9f52daea727ecb209320f;hb=0bd5d91026b8bd143f957f119d61d5fedd45cf70;hp=f8f07870035051b0f7a03b056cdfefc0e6d504d6;hpb=c28f352daacca421f977bca75eefd36a73189dac;p=melted

diff --git a/src/framework/mlt_frame.c b/src/framework/mlt_frame.c
index f8f0787..880cdef 100644
--- a/src/framework/mlt_frame.c
+++ b/src/framework/mlt_frame.c
@@ -263,16 +263,7 @@ int mlt_frame_get_audio( mlt_frame this, int16_t **buffer, mlt_audio_format *for
 
 void mlt_frame_close( mlt_frame this )
 {
-	mlt_frame frame = mlt_frame_pop_frame( this );
-	
-	while ( frame != NULL )
-	{
-		mlt_frame_close( frame);
-		frame = mlt_frame_pop_frame( this );
-	}
-	
 	mlt_properties_close( &this->parent );
-
 	free( this );
 }
 
@@ -465,22 +456,32 @@ int mlt_frame_composite_yuv( mlt_frame this, mlt_frame that, int x, int y, float
 	if ( p_alpha )
 		p_alpha += x_src + y_src * stride_src / 2;
 
+	uint8_t *p = p_src;
+	uint8_t *q = p_dest;
+	uint8_t *o = p_dest;
+	uint8_t *z = p_alpha;
+
+	uint8_t Y;
+	uint8_t UV;
+	uint8_t a;
+	float value;
+
 	// now do the compositing only to cropped extents
 	for ( i = 0; i < height_src; i++ )
 	{
-		uint8_t *p = p_src;
-		uint8_t *q = p_dest;
-		uint8_t *o = p_dest;
-		uint8_t *z = p_alpha;
+		p = p_src;
+		q = p_dest;
+		o = p_dest;
+		z = p_alpha;
 
 		for ( j = 0; j < width_src; j ++ )
 		{
-				uint8_t y = *p ++;
-				uint8_t uv = *p ++;
-				uint8_t a = ( z == NULL ) ? 255 : *z ++;
-				float value = ( weight * ( float ) a / 255.0 );
-				*o ++ = (uint8_t)( y * value + *q++ * ( 1 - value ) );
-				*o ++ = (uint8_t)( uv * value + *q++ * ( 1 - value ) );
+			Y = *p ++;
+			UV = *p ++;
+			a = ( z == NULL ) ? 255 : *z ++;
+			value = ( weight * ( float ) a / 255.0 );
+			*o ++ = (uint8_t)( Y * value + *q++ * ( 1 - value ) );
+			*o ++ = (uint8_t)( UV * value + *q++ * ( 1 - value ) );
 		}
 
 		p_src += stride_src;
@@ -661,14 +662,14 @@ uint8_t *mlt_frame_rescale_yuv422( mlt_frame this, int owidth, int oheight )
 		uint8_t *in_ptr;
 
 		// Generate the affine transform scaling values
-		float scale_width = ( float )iwidth / ( float )owidth;
-		float scale_height = ( float )iheight / ( float )oheight;
+		int scale_width = ( iwidth << 16 ) / owidth;
+		int scale_height = ( iheight << 16 ) / oheight;
 
     	// Loop for the entirety of our output height.
     	for ( y = - out_y_range; y < out_y_range ; y ++ )
     	{
 			// Calculate the derived y value
-			dy = scale_height * y;
+			dy = ( scale_height * y ) >> 16;
 
         	// Start at the beginning of the line
         	out_ptr = out_line;
@@ -680,22 +681,13 @@ uint8_t *mlt_frame_rescale_yuv422( mlt_frame this, int owidth, int oheight )
         	for ( x = - out_x_range; x < out_x_range; x += 1 )
         	{
 				// Calculated the derived x
-				dx = scale_width * x;
-
-            	// Check if x and y are in the valid input range.
-            	if ( abs( dx ) < in_x_range && abs( dy ) < in_y_range  )
-            	{
-                	// We're in the input range for this row.
-					in_ptr = in_line + ( dx >> 1 ) * 4 + 2 * ( x & 1 );
-                	*out_ptr ++ = *in_ptr ++;
-                	*out_ptr ++ = *in_ptr ++;
-            	}
-            	else
-            	{
-                	// We're not in the input range for this row.
-                	*out_ptr ++ = 16;
-                	*out_ptr ++ = 128;
-            	}
+				dx = ( scale_width * x ) >> 16;
+
+               	// We're in the input range for this row.
+				in_ptr = in_line + ( dx << 1 );
+               	*out_ptr ++ = *in_ptr ++;
+				in_ptr = in_line + ( ( dx >> 1 ) << 2 ) + ( ( x & 1 ) << 1 ) + 1;
+               	*out_ptr ++ = *in_ptr;
         	}
 
         	// Move to next output line
@@ -722,15 +714,18 @@ int mlt_frame_mix_audio( mlt_frame this, mlt_frame that, float weight, int16_t *
 	int16_t *src, *dest;
 	//static int16_t *extra_src = NULL, *extra_dest = NULL;
 	static int extra_src_samples = 0, extra_dest_samples = 0;
-	int frequency_src = 0, frequency_dest = 0;
-	int channels_src = 0, channels_dest = 0;
-	int samples_src = 0, samples_dest = 0;
+	int frequency_src = *channels, frequency_dest = *channels;
+	int channels_src = *channels, channels_dest = *channels;
+	int samples_src = *samples, samples_dest = *samples;
 	int i, j;
+	double d = 0, s = 0;
 
 	mlt_frame_get_audio( this, &p_dest, format, &frequency_dest, &channels_dest, &samples_dest );
-	//fprintf( stderr, "frame dest samples %d channels %d position %f\n", samples_dest, channels_dest, mlt_properties_get_position( mlt_frame_properties( this ), "position" ) );
+	//fprintf( stderr, "frame dest samples %d channels %d position %lld\n", samples_dest, channels_dest, mlt_properties_get_position( mlt_frame_properties( this ), "position" ) );
 	mlt_frame_get_audio( that, &p_src, format, &frequency_src, &channels_src, &samples_src );
 	//fprintf( stderr, "frame src  samples %d channels %d\n", samples_src, channels_src );
+	src = p_src;
+	dest = p_dest;
 	if ( channels_src > 6 )
 		channels_src = 0;
 	if ( channels_dest > 6 )
@@ -758,9 +753,6 @@ int mlt_frame_mix_audio( mlt_frame this, mlt_frame that, float weight, int16_t *
 	}
 	else
 		src = p_src;
-#else
-	src = p_src;
-	dest = p_dest;
 #endif
 
 	// determine number of samples to process	
@@ -777,8 +769,10 @@ int mlt_frame_mix_audio( mlt_frame this, mlt_frame that, float weight, int16_t *
 	{
 		for ( j = 0; j < *channels; j++ )
 		{
-			double d = (double) dest[ i * channels_dest + j ];
-			double s = (double) src[ i * channels_src + j ];
+			if ( j < channels_dest )
+				d = (double) dest[ i * channels_dest + j ];
+			if ( j < channels_src )
+				s = (double) src[ i * channels_src + j ];
 			dest[ i * channels_dest + j ] = s * weight + d * ( 1.0 - weight );
 		}
 	}