new files, based on SoX (http://sox.sf.net): better resampling code. Note that my mixer.cpp changes are on purpose not yet in CVS since they are not complete. Only reasons I checkin these files is that it's much more comfortable to have CVS, since I need to rewrite parts of resample.cpp now (I already have lots of modifications in). Also expect more OO in the future

svn-id: r9176

1 files changed, 164 insertions, 0 deletions

diff --git a/sound/rate.cpp b/sound/rate.cpp
new file mode 100644
index 0000000000..bde477647a
--- /dev/null
+++ b/sound/rate.cpp
@@ -0,0 +1,164 @@
+/*
+* August 21, 1998
+* Copyright 1998 Fabrice Bellard.
+*
+* [Rewrote completly the code of Lance Norskog And Sundry
+* Contributors with a more efficient algorithm.]
+*
+* This source code is freely redistributable and may be used for
+* any purpose.	 This copyright notice must be maintained. 
+* Lance Norskog And Sundry Contributors are not responsible for 
+* the consequences of using this software.  
+*/
+
+/*
+ * Sound Tools rate change effect file.
+ */
+
+#include "rate.h"
+
+#include <math.h>
+/*
+ * Linear Interpolation.
+ *
+ * The use of fractional increment allows us to use no buffer. It
+ * avoid the problems at the end of the buffer we had with the old
+ * method which stored a possibly big buffer of size
+ * lcm(in_rate,out_rate).
+ *
+ * Limited to 16 bit samples and sampling frequency <= 65535 Hz. If
+ * the input & output frequencies are equal, a delay of one sample is
+ * introduced.	Limited to processing 32-bit count worth of samples.
+ *
+ * 1 << FRAC_BITS evaluating to zero in several places.	 Changed with
+ * an (unsigned long) cast to make it safe.  MarkMLl 2/1/99
+ *
+ * Replaced all uses of floating point arithmetic by fixed point
+ * calculations (Max Horn 2003-07-18).
+ */
+
+#define FRAC_BITS 16
+
+/* Private data */
+
+typedef struct ratestuff
+{
+	unsigned long opos_frac;  /* fractional position of the output stream in input stream unit */
+	unsigned long opos;
+
+	unsigned long opos_inc_frac;  /* fractional position increment in the output stream */
+	unsigned long opos_inc;
+
+	unsigned long ipos;	 /* position in the input stream (integer) */
+
+	st_sample_t ilast; /* last sample in the input stream */
+} *rate_t;
+
+/*
+ * Process options
+ */
+int st_rate_getopts(eff_t effp, int n, char **argv)
+{
+	if (n) {
+		st_fail("Rate effect takes no options.");
+		return (ST_EOF);
+	}
+
+	return (ST_SUCCESS);
+}
+
+/*
+ * Prepare processing.
+ */
+int st_rate_start(eff_t effp, st_rate_t inrate, st_rate_t outrate)
+{
+	rate_t rate = (rate_t) effp->priv;
+	unsigned long incr;
+
+	if (inrate == outrate) {
+		st_fail("Input and Output rates must be different to use rate effect");
+		return (ST_EOF);
+	}
+
+	if (inrate >= 65536 || outrate >= 65536) {
+		st_fail("rate effect can only handle rates < 65536");
+		return (ST_EOF);
+	}
+
+	rate->opos_frac = 0;
+	rate->opos = 0;
+
+	/* increment */
+	incr = (inrate << FRAC_BITS) / outrate;
+
+	rate->opos_inc_frac = incr & ((1UL << FRAC_BITS) - 1);
+	rate->opos_inc = incr >> FRAC_BITS;
+
+	rate->ipos = 0;
+
+	rate->ilast = 0;
+	return (ST_SUCCESS);
+}
+
+/*
+ * Processed signed long samples from ibuf to obuf.
+ * Return number of samples processed.
+ */
+int st_rate_flow(eff_t effp, InputStream &input, st_sample_t *obuf, st_size_t *osamp)
+{
+	rate_t rate = (rate_t) effp->priv;
+	st_sample_t *ostart, *oend;
+	st_sample_t ilast, icur, out;
+	unsigned long tmp;
+
+	ilast = rate->ilast;
+
+	ostart = obuf;
+	oend = obuf + *osamp;
+
+	while (obuf < oend && !input.eof()) {
+
+		/* read as many input samples so that ipos > opos */
+		while (rate->ipos <= rate->opos) {
+			ilast = input.read();
+			rate->ipos++;
+			/* See if we finished the input buffer yet */
+
+			if (input.eof())
+				goto the_end;
+		}
+
+		icur = input.peek();
+
+		/* interpolate */
+		out = ilast + (((icur - ilast) * rate->opos_frac) >> FRAC_BITS);
+
+		/* output sample & increment position */
+
+		clampedAdd(*obuf++, out);
+		#if 1	// FIXME: Hack to generate stereo output
+
+		clampedAdd(*obuf++, out);
+		#endif
+
+		tmp = rate->opos_frac + rate->opos_inc_frac;
+		rate->opos = rate->opos + rate->opos_inc + (tmp >> FRAC_BITS);
+		rate->opos_frac = tmp & ((1UL << FRAC_BITS) - 1);
+	}
+
+the_end:
+	*osamp = obuf - ostart;
+	rate->ilast = ilast;
+	return (ST_SUCCESS);
+}
+
+/*
+ * Do anything required when you stop reading samples.	
+ * Don't close input file! 
+ */
+int st_rate_stop(eff_t effp)
+{
+	/* nothing to do */
+	return (ST_SUCCESS);
+}
+