--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/spandsp-0.0.6pre17/tests/sig_tone_tests.c	Fri Jun 25 15:50:58 2010 +0200
@@ -0,0 +1,261 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * sig_tone_tests.c
+ *
+ * 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 General Public License version 2, 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 General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id: sig_tone_tests.c,v 1.27 2009/09/23 16:02:59 steveu Exp $
+ */
+
+/*! \file */
+
+/*! \page sig_tone_tests_page The signaling tone processor tests
+\section sig_tone_tests_sec_1 What does it do?
+???.
+
+\section sig_tone_tests_sec_2 How does it work?
+???.
+*/
+
+#if defined(HAVE_CONFIG_H)
+#include "config.h"
+#endif
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <memory.h>
+#include <sndfile.h>
+
+//#if defined(WITH_SPANDSP_INTERNALS)
+#define SPANDSP_EXPOSE_INTERNAL_STRUCTURES
+//#endif
+
+#include "spandsp.h"
+#include "spandsp-sim.h"
+
+#define OUT_FILE_NAME               "sig_tone.wav"
+
+#define SAMPLES_PER_CHUNK           160
+
+static int sampleno = 0;
+static int tone_1_present = 0;
+static int tone_2_present = 0;
+static int tx_section = 0;
+static int dial_pulses = 0;
+
+static void tx_handler(void *user_data, int what, int level, int duration)
+{
+    sig_tone_tx_state_t *s;
+    
+    s = (sig_tone_tx_state_t *) user_data;
+    //printf("What - %d, duration - %d\n", what, duration);
+    if ((what & SIG_TONE_TX_UPDATE_REQUEST))
+    {
+        printf("Tx: update request\n");
+        /* The sig tone transmit side wants to know what to do next */
+        switch (tx_section)
+        {
+        case 0:
+            printf("33ms break - %d samples\n", ms_to_samples(33));
+            tx_section++;
+            sig_tone_tx_set_mode(s, SIG_TONE_1_PRESENT, ms_to_samples(33));
+            break;
+        case 1:
+            printf("67ms make - %d samples\n", ms_to_samples(67));
+            if (++dial_pulses == 9)
+                tx_section++;
+            else
+                tx_section--;
+            sig_tone_tx_set_mode(s, 0, ms_to_samples(67));
+            break;
+        case 2:
+            tx_section++;
+            sig_tone_tx_set_mode(s, SIG_TONE_1_PRESENT, ms_to_samples(600));
+            break;
+        case 3:
+            sig_tone_tx_set_mode(s, SIG_TONE_1_PRESENT | SIG_TONE_TX_PASSTHROUGH, 0);
+            break;
+        }
+        /*endswitch*/
+    }
+    /*endif*/
+}
+/*- End of function --------------------------------------------------------*/
+
+static void rx_handler(void *user_data, int what, int level, int duration)
+{
+    float ms;
+
+    ms = 1000.0f*(float) duration/(float) SAMPLE_RATE;
+    printf("What - %d, duration - %d\n", what, duration);
+    if ((what & SIG_TONE_1_CHANGE))
+    {
+        tone_1_present = what & SIG_TONE_1_PRESENT;
+        printf("Rx: tone 1 is %s after %d samples (%fms)\n", (tone_1_present)  ?  "on"  : "off", duration, ms);
+    }
+    /*endif*/
+    if ((what & SIG_TONE_2_CHANGE))
+    {
+        tone_2_present = what & SIG_TONE_2_PRESENT;
+        printf("Rx: tone 2 is %s after %d samples (%fms)\n", (tone_2_present)  ?  "on"  : "off", duration, ms);
+    }
+    /*endif*/
+}
+/*- End of function --------------------------------------------------------*/
+
+static void map_frequency_response(sig_tone_rx_state_t *s)
+{
+    int16_t buf[8192];
+    int i;
+    int len;
+    double sumin;
+    double sumout;
+    swept_tone_state_t *swept;
+    
+    /* Things like noise don't highlight the frequency response of the high Q notch
+       very well. We use a slowly swept frequency to check it. */
+    swept = swept_tone_init(NULL, 200.0f, 3900.0f, -10.0f, 120*SAMPLE_RATE, 0);
+    for (;;)
+    {
+        if ((len = swept_tone(swept, buf, SAMPLES_PER_CHUNK)) <= 0)
+            break;
+        sumin = 0.0;
+        for (i = 0;  i < len;  i++)
+            sumin += (double) buf[i]*(double) buf[i];
+        sig_tone_rx(s, buf, len);
+        sumout = 0.0;
+        for (i = 0;  i < len;  i++)
+            sumout += (double) buf[i]*(double) buf[i];
+        /*endfor*/
+        printf("%7.1f %f\n", swept_tone_current_frequency(swept), 10.0*log10(sumout/sumin));
+    }
+    /*endfor*/
+    swept_tone_free(swept);
+}
+/*- End of function --------------------------------------------------------*/
+
+int main(int argc, char *argv[])
+{
+    int16_t amp[SAMPLES_PER_CHUNK];
+    int16_t out_amp[2*SAMPLES_PER_CHUNK];
+    SNDFILE *outhandle;
+    int outframes;
+    int i;
+    int type;
+    int rx_samples;
+    int tx_samples;
+    sig_tone_tx_state_t tx_state;
+    sig_tone_rx_state_t rx_state;
+    awgn_state_t noise_source;
+    codec_munge_state_t *munge;
+
+    if ((outhandle = sf_open_telephony_write(OUT_FILE_NAME, 2)) == NULL)
+    {
+        fprintf(stderr, "    Cannot create audio file '%s'\n", OUT_FILE_NAME);
+        exit(2);
+    }
+    /*endif*/
+
+    awgn_init_dbm0(&noise_source, 1234567, -20.0f);
+
+    for (type = 1;  type <= 3;  type++)
+    {
+        sampleno = 0;
+        tone_1_present = 0;
+        tone_2_present = 0;
+        tx_section = 0;
+        munge = NULL;
+        switch (type)
+        {
+        case 1:
+            printf("2280Hz tests.\n");
+            munge = codec_munge_init(MUNGE_CODEC_ALAW, 0);
+            sig_tone_tx_init(&tx_state, SIG_TONE_2280HZ, tx_handler, &tx_state);
+            sig_tone_rx_init(&rx_state, SIG_TONE_2280HZ, rx_handler, &rx_state);
+            rx_state.current_rx_tone |= SIG_TONE_RX_PASSTHROUGH;
+            break;
+        case 2:
+            printf("2600Hz tests.\n");
+            munge = codec_munge_init(MUNGE_CODEC_ULAW, 0);
+            sig_tone_tx_init(&tx_state, SIG_TONE_2600HZ, tx_handler, &tx_state);
+            sig_tone_rx_init(&rx_state, SIG_TONE_2600HZ, rx_handler, &rx_state);
+            rx_state.current_rx_tone |= SIG_TONE_RX_PASSTHROUGH;
+            break;
+        case 3:
+            printf("2400Hz/2600Hz tests.\n");
+            munge = codec_munge_init(MUNGE_CODEC_ULAW, 0);
+            sig_tone_tx_init(&tx_state, SIG_TONE_2400HZ_2600HZ, tx_handler, &tx_state);
+            sig_tone_rx_init(&rx_state, SIG_TONE_2400HZ_2600HZ, rx_handler, &rx_state);
+            rx_state.current_rx_tone |= SIG_TONE_RX_PASSTHROUGH;
+            break;
+        }
+        /*endswitch*/
+        /* Set to the default of hook condition */
+        sig_tone_rx_set_mode(&rx_state, SIG_TONE_RX_PASSTHROUGH | SIG_TONE_RX_FILTER_TONE, 0);
+        sig_tone_tx_set_mode(&tx_state, SIG_TONE_1_PRESENT | SIG_TONE_2_PRESENT | SIG_TONE_TX_PASSTHROUGH, 0);
+
+        map_frequency_response(&rx_state);
+
+        sig_tone_rx_set_mode(&rx_state, SIG_TONE_RX_PASSTHROUGH, 0);
+        for (sampleno = 0;  sampleno < 30000;  sampleno += SAMPLES_PER_CHUNK)
+        {
+            if (sampleno == 8000)
+            {
+                /* 100ms seize */
+                printf("100ms seize - %d samples\n", ms_to_samples(100));
+                dial_pulses = 0;
+                sig_tone_tx_set_mode(&tx_state, 0, ms_to_samples(100));
+            }
+            for (i = 0;  i < SAMPLES_PER_CHUNK;  i++)
+                amp[i] = awgn(&noise_source);
+            /*endfor*/
+            tx_samples = sig_tone_tx(&tx_state, amp, SAMPLES_PER_CHUNK);
+            for (i = 0;  i < tx_samples;  i++)
+                out_amp[2*i] = amp[i];
+            /*endfor*/
+            codec_munge(munge, amp, tx_samples);
+            rx_samples = sig_tone_rx(&rx_state, amp, tx_samples);
+            for (i = 0;  i < rx_samples;  i++)
+                out_amp[2*i + 1] = amp[i];
+            /*endfor*/
+            outframes = sf_writef_short(outhandle, out_amp, rx_samples);
+            if (outframes != rx_samples)
+            {
+                fprintf(stderr, "    Error writing audio file\n");
+                exit(2);
+            }
+            /*endif*/
+        }
+        /*endfor*/
+    }
+    /*endfor*/
+    if (sf_close(outhandle) != 0)
+    {
+        fprintf(stderr, "    Cannot close audio file '%s'\n", OUT_FILE_NAME);
+        exit(2);
+    }
+    /*endif*/
+    
+    printf("Tests completed.\n");
+    return  0;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/