--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/spandsp-0.0.6pre17/src/spandsp/time_scale.h	Fri Jun 25 15:50:58 2010 +0200
@@ -0,0 +1,118 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * time_scale.h - Time scaling for linear speech data
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2004 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU Lesser General Public License version 2.1,
+ * as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id: time_scale.h,v 1.20 2009/02/10 13:06:47 steveu Exp $
+ */
+
+#if !defined(_SPANDSP_TIME_SCALE_H_)
+#define _SPANDSP_TIME_SCALE_H_
+
+/*! \page time_scale_page Time scaling speech
+\section time_scale_page_sec_1 What does it do?
+The time scaling module allows speech files to be played back at a
+different speed from the speed at which they were recorded. If this
+were done by simply speeding up or slowing down replay, the pitch of
+the voice would change, and sound very odd. This module keeps the pitch
+of the voice at its original level.
+
+The speed of the voice may be altered over a wide range. However, the practical
+useful rates are between about half normal speed and twice normal speed.
+
+\section time_scale_page_sec_2 How does it work?
+The time scaling module is based on the Pointer Interval Controlled
+OverLap and Add (PICOLA) method, developed by Morita Naotaka.
+Mikio Ikeda has an excellent web page on this subject at
+http://keizai.yokkaichi-u.ac.jp/~ikeda/research/picola.html
+There is also working code there. This implementation uses
+exactly the same algorithms, but the code is a complete rewrite.
+Mikio's code batch processes files. This version works incrementally
+on streams, and allows multiple streams to be processed concurrently.
+
+\section time_scale_page_sec_3 How do I used it?
+The output buffer must be big enough to hold the maximum number of samples which
+could result from the data in the input buffer, which is:
+
+    input_len*playout_rate + sample_rate/TIME_SCALE_MIN_PITCH + 1
+*/
+
+/*! Audio time scaling descriptor. */
+typedef struct time_scale_state_s time_scale_state_t;
+
+#if defined(__cplusplus)
+extern "C"
+{
+#endif
+
+/*! Initialise a time scale context. This must be called before the first
+    use of the context, to initialise its contents.
+    \brief Initialise a time scale context.
+    \param s The time scale context.
+    \param sample_rate The sample rate of the signal.
+    \param playout_rate The ratio between the output speed and the input speed.
+    \return A pointer to the context, or NULL if there was a problem. */
+SPAN_DECLARE(time_scale_state_t *) time_scale_init(time_scale_state_t *s, int sample_rate, float playout_rate);
+
+/*! \brief Release a time scale context.
+    \param s The time scale context.
+    \return 0 for OK, else -1. */
+SPAN_DECLARE(int) time_scale_release(time_scale_state_t *s);
+
+/*! \brief Free a time scale context.
+    \param s The time scale context.
+    \return 0 for OK, else -1. */
+SPAN_DECLARE(int) time_scale_free(time_scale_state_t *s);
+
+/*! Change the time scale rate.
+    \brief Change the time scale rate.
+    \param s The time scale context.
+    \param playout_rate The ratio between the output speed and the input speed.
+    \return 0 if changed OK, else -1. */
+SPAN_DECLARE(int) time_scale_rate(time_scale_state_t *s, float playout_rate);
+
+/*! Find the maximum possible samples which could result from scaling the specified
+    number of input samples, at the current playback rate.
+    \brief Find the maximum possible output samples.
+    \param s The time scale context.
+    \param input_len The number of input samples.
+    \return The maximum possible output samples. */
+SPAN_DECLARE(int) time_scale_max_output_len(time_scale_state_t *s, int input_len);
+
+/*! Time scale a chunk of audio samples.
+    \brief Time scale a chunk of audio samples.
+    \param s The time scale context.
+    \param out The output audio sample buffer. This must be large enough to accept
+           the longest possible result from processing the input data. See the
+           algorithm documentation for how the longest possible result may be calculated.
+    \param in The input audio sample buffer.
+    \param len The number of input samples.
+    \return The number of output samples.
+*/
+SPAN_DECLARE(int) time_scale(time_scale_state_t *s, int16_t out[], int16_t in[], int len);
+
+#if defined(__cplusplus)
+}
+#endif
+
+#endif
+/*- End of file ------------------------------------------------------------*/