audiostuff: spandsp-0.0.3/spandsp-0.0.3/tests/modem_echo

comparison spandsp-0.0.3/spandsp-0.0.3/tests/modem_echo_tests.c @ 5:f762bf195c4b

import spandsp-0.0.3

author	Peter Meerwald <pmeerw@cosy.sbg.ac.at>
date	Fri, 25 Jun 2010 16:00:21 +0200
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-:26cd8f1ef0b1
+:f762bf195c4b
+/*
+* SpanDSP - a series of DSP components for telephony
+*
+* modem_echo_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: modem_echo_tests.c,v 1.22 2006/11/19 14:07:27 steveu Exp $
+*/
+/*! \page modem_echo_can_tests_page Line echo cancellation for modems tests
+\section modem_echo_can_tests_page_sec_1 What does it do?
+Currently the echo cancellation tests only provide simple exercising of the
+cancellor in the way it might be used for line echo cancellation. The test code
+is in echotests.c.
+The goal is to test the echo cancellor again the G.16X specs. Clearly, that also
+means the goal for the cancellor itself is to comply with those specs. Right
+now, the only aspect of these tests implemented is the line impulse response
+models in g168tests.c.
+\section modem_echo_can_tests_page_sec_2 How does it work?
+The current test consists of feeding a wave file of real speech to the echo
+cancellor as the transmit signal. A very simple model of a telephone line is
+used to simulate a simple echo from the transmit signal. A second wave file of
+real speech is also used to simulate a signal received form the far end of the
+line. This is gated so it is only placed for one second every 10 seconds,
+simulating the double talk condition. The resulting echo cancelled signal can
+either be store in a file for further analysis, or played back as the data is
+processed.
+A number of modified versions of this test have been performed. The signal level
+of the two speech sources has been varied. Several simple models of the
+telephone line have been used. Although the current cancellor design has known
+limitations, it seems stable for all these test conditions. No instability has
+been observed in the current version due to arithmetic overflow when the speech
+is very loud (with earlier versions, well, ....:) ). The lack of saturating
+arithmetic in general purpose CPUs is a huge disadvantage here, as software
+saturation logic would cause a major slow down. Floating point would be good,
+but is not usable in the Linux kernel. Anyway, the bottom line seems to be the
+current design is genuinely useful, if imperfect.
+\section modem_echo_can_tests_page_sec_2 How do I use it?
+Build the tests with the command "./build". Currently there is no proper make
+setup, or way to build individual tests. "./build" will built all the tests
+which currently exist for the DSP functions. The echo cancellation test assumes
+there are two wave files containing mono, 16 bit signed PCM speech data, sampled
+at 8kHz. These should be called local_sound.wav and far_sound.wav. A third wave
+file will be produced. This very crudely starts with the first 256 bytes from
+the local_sound.wav file, followed by the results of the echo cancellation. The
+resulting audio is also played to the /dev/dsp device. A printf near the end of
+echo_tests.c is commented out with a \#if. If this is enabled, detailed
+information about the results of the echo cancellation will be written to
+stdout. By saving this into a file, Grace (recommended), GnuPlot, or some other
+plotting package may be used to graphically display the functioning of the
+cancellor.
+*/
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+#if defined(HAVE_FL_FL_H)  &&  defined(HAVE_FL_FL_CARTESIAN_H)  &&  defined(HAVE_FL_FL_AUDIO_METER_H)
+#define ENABLE_GUI
+#endif
+#include <stdlib.h>
+#include <inttypes.h>
+#include <string.h>
+#include <time.h>
+#include <stdio.h>
+#include <fcntl.h>
+#include <audiofile.h>
+#include <tiffio.h>
+#if defined(HAVE_TGMATH_H)
+#include <tgmath.h>
+#endif
+#if defined(HAVE_MATH_H)
+#define GEN_CONST
+#include <math.h>
+#endif
+#include "spandsp.h"
+#include "spandsp/g168models.h"
+#if defined(ENABLE_GUI)
+#include "echo_monitor.h"
+#endif
+#if !defined(NULL)
+#define NULL (void *) 0
+#endif
+typedef struct
+{
+const char *name;
+int max;
+int cur;
+AFfilehandle handle;
+int16_t signal[8000];
+} signal_source_t;
+signal_source_t local_css;
+fir32_state_t line_model;
+AFfilehandle resulthandle;
+int16_t residue_sound[8000];
+int residue_cur = 0;
+int do_codec_munge = TRUE;
+int use_gui = FALSE;
+static const int16_t tone_1khz[] = {10362, 7327, 0, -7327, -10362, -7327, 0, 7327};
+static inline void put_residue(int16_t tx, int16_t residue)
+{
+int outframes;
+residue_sound[residue_cur++] = tx;
+residue_sound[residue_cur++] = residue;
+if (residue_cur >= 8000)
+{
+residue_cur >>= 1;
+outframes = afWriteFrames(resulthandle,
+AF_DEFAULT_TRACK,
+residue_sound,
+residue_cur);
+if (outframes != residue_cur)
+{
+fprintf(stderr, "    Error writing residue sound\n");
+exit(2);
+}
+residue_cur = 0;
+}
+}
+/*- End of function --------------------------------------------------------*/
+static void signal_load(signal_source_t *sig, const char *name)
+{
+float x;
+sig->name = name;
+if ((sig->handle = afOpenFile(sig->name, "r", 0)) == AF_NULL_FILEHANDLE)
+{
+fprintf(stderr, "    Cannot open sound file '%s'\n", sig->name);
+exit(2);
+}
+if ((x = afGetFrameSize(sig->handle, AF_DEFAULT_TRACK, 1)) != 2.0)
+{
+fprintf(stderr, "    Unexpected frame size in wave file '%s'\n", sig->name);
+exit(2);
+}
+if ((x = afGetRate(sig->handle, AF_DEFAULT_TRACK)) != (float) SAMPLE_RATE)
+{
+printf("    Unexpected sample rate in wave file '%s'\n", sig->name);
+exit(2);
+}
+if ((x = afGetChannels(sig->handle, AF_DEFAULT_TRACK)) != 1.0)
+{
+printf("    Unexpected number of channels in wave file '%s'\n", sig->name);
+exit(2);
+}
+sig->max = afReadFrames(sig->handle, AF_DEFAULT_TRACK, sig->signal, 8000);
+if (sig->max < 0)
+{
+fprintf(stderr, "    Error reading sound file '%s'\n", sig->name);
+exit(2);
+}
+}
+/*- End of function --------------------------------------------------------*/
+static void signal_free(signal_source_t *sig)
+{
+if (afCloseFile(sig->handle) != 0)
+{
+fprintf(stderr, "    Cannot close sound file '%s'\n", sig->name);
+exit(2);
+}
+}
+/*- End of function --------------------------------------------------------*/
+static void signal_restart(signal_source_t *sig)
+{
+sig->cur = 0;
+}
+/*- End of function --------------------------------------------------------*/
+static int16_t signal_amp(signal_source_t *sig)
+{
+int16_t tx;
+tx = sig->signal[sig->cur++];
+if (sig->cur >= sig->max)
+sig->cur = 0;
+return tx;
+}
+/*- End of function --------------------------------------------------------*/
+static inline int16_t codec_munge(int16_t amp)
+{
+if (do_codec_munge)
+return alaw_to_linear(linear_to_alaw(amp));
+return amp;
+}
+/*- End of function --------------------------------------------------------*/
+static void channel_model_create(int model)
+{
+static const int32_t *line_models[] =
+{
+line_model_d2_coeffs,
+line_model_d3_coeffs,
+line_model_d4_coeffs,
+line_model_d5_coeffs,
+line_model_d6_coeffs,
+line_model_d7_coeffs,
+line_model_d8_coeffs,
+line_model_d9_coeffs
+};
+static int line_model_sizes[] =
+{
+sizeof(line_model_d2_coeffs)/sizeof(int32_t),
+sizeof(line_model_d3_coeffs)/sizeof(int32_t),
+sizeof(line_model_d4_coeffs)/sizeof(int32_t),
+sizeof(line_model_d5_coeffs)/sizeof(int32_t),
+sizeof(line_model_d6_coeffs)/sizeof(int32_t),
+sizeof(line_model_d7_coeffs)/sizeof(int32_t),
+sizeof(line_model_d8_coeffs)/sizeof(int32_t),
+sizeof(line_model_d9_coeffs)/sizeof(int32_t)
+};
+fir32_create(&line_model, line_models[model], line_model_sizes[model]);
+}
+/*- End of function --------------------------------------------------------*/
+static int16_t channel_model(int16_t local, int16_t far)
+{
+int16_t echo;
+int16_t rx;
+/* Channel modelling is merely simulating the effects of A-law distortion
+and using one of the echo models from G.168 */
+/* The local tx signal will have gone through an A-law munging before
+it reached the line's analogue area where the echo occurs. */
+echo = fir32(&line_model, codec_munge(local/8));
+/* The far end signal will have been through an A-law munging, although
+this should not affect things. */
+rx = echo + codec_munge(far);
+/* This mixed echo and far end signal will have been through an A-law munging when it came back into
+the digital network. */
+rx = codec_munge(rx);
+return  rx;
+}
+/*- End of function --------------------------------------------------------*/
+int main(int argc, char *argv[])
+{
+modem_echo_can_state_t *ctx;
+//awgn_state_t local_noise_source;
+awgn_state_t far_noise_source;
+int i;
+int clean;
+int16_t rx;
+int16_t tx;
+int local_cur;
+int far_cur;
+int result_cur;
+AFfilesetup filesetup;
+int line_model_no;
+time_t now;
+power_meter_t power_before;
+power_meter_t power_after;
+float unadapted_output_power;
+float unadapted_echo_power;
+float adapted_output_power;
+float adapted_echo_power;
+#if defined(ENABLE_GUI)
+int16_t amp[2];
+#endif
+line_model_no = 0;
+use_gui = FALSE;
+for (i = 1;  i < argc;  i++)
+{
+if (strcmp(argv[i], "-g") == 0)
+{
+use_gui = TRUE;
+continue;
+}
+line_model_no = atoi(argv[1]);
+}
+time(&now);
+ctx = modem_echo_can_create(256);
+awgn_init_dbm0(&far_noise_source, 7162534, -50.0f);
+signal_load(&local_css, "sound_c1_8k.wav");
+filesetup = afNewFileSetup();
+if (filesetup == AF_NULL_FILESETUP)
+{
+fprintf(stderr, "    Failed to create file setup\n");
+exit(2);
+}
+afInitSampleFormat(filesetup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 16);
+afInitRate(filesetup, AF_DEFAULT_TRACK, (float) SAMPLE_RATE);
+afInitFileFormat(filesetup, AF_FILE_WAVE);
+afInitChannels(filesetup, AF_DEFAULT_TRACK, 2);
+resulthandle = afOpenFile("modem_echo.wav", "w", filesetup);
+if (resulthandle == AF_NULL_FILEHANDLE)
+{
+fprintf(stderr, "    Failed to open result file\n");
+exit(2);
+}
+#if defined(ENABLE_GUI)
+if (use_gui)
+start_echo_can_monitor(256);
+#endif
+local_cur = 0;
+far_cur = 0;
+result_cur = 0;
+channel_model_create(line_model_no);
+#if defined(ENABLE_GUI)
+if (use_gui)
+echo_can_monitor_line_model_update(line_model.coeffs, line_model.taps);
+#endif
+modem_echo_can_flush(ctx);
+power_meter_init(&power_before, 5);
+power_meter_init(&power_after, 5);
+/* Measure the echo power before adaption */
+modem_echo_can_adaption_mode(ctx, FALSE);
+for (i = 0;  i < 8000*5;  i++)
+{
+tx = tone_1khz[i & 7];
+rx = channel_model(tx, 0);
+clean = modem_echo_can_update(ctx, tx, rx);
+power_meter_update(&power_before, rx);
+power_meter_update(&power_after, clean);
+}
+unadapted_output_power = power_meter_dbm0(&power_before);
+unadapted_echo_power = power_meter_dbm0(&power_after);
+printf("Pre-adaption: output power %10.5fdBm0, echo power %10.5fdBm0\n", unadapted_output_power, unadapted_echo_power);
+/* Converge the canceller */
+signal_restart(&local_css);
+modem_echo_can_adaption_mode(ctx, TRUE);
+for (i = 0;  i < 800*2;  i++)
+{
+clean = modem_echo_can_update(ctx, 0, 0);
+put_residue(0, clean);
+}
+for (i = 0;  i < 8000*50;  i++)
+{
+tx = signal_amp(&local_css);
+rx = channel_model(tx, 0);
+clean = modem_echo_can_update(ctx, tx, rx);
+power_meter_update(&power_before, rx);
+power_meter_update(&power_after, clean);
+#if 0
+if (i%800 == 0)
+printf("Powers %10.5fdBm0 %10.5fdBm0\n", power_meter_dbm0(&power_before), power_meter_dbm0(&power_after));
+#endif
+put_residue(tx, clean);
+#if defined(ENABLE_GUI)
+if (use_gui)
+{
+echo_can_monitor_can_update(ctx->fir_taps16, 256);
+amp[0] = tx;
+echo_can_monitor_line_spectrum_update(amp, 1);
+}
+#endif
+}
+/* Now lets see how well adapted we are */
+modem_echo_can_adaption_mode(ctx, FALSE);
+for (i = 0;  i < 8000*5;  i++)
+{
+tx = tone_1khz[i & 7];
+rx = channel_model(tx, 0);
+clean = modem_echo_can_update(ctx, tx, rx);
+power_meter_update(&power_before, rx);
+power_meter_update(&power_after, clean);
+}
+adapted_output_power = power_meter_dbm0(&power_before);
+adapted_echo_power = power_meter_dbm0(&power_after);
+printf("Post-adaption: output power %10.5fdBm0, echo power %10.5fdBm0\n", adapted_output_power, adapted_echo_power);
+if (fabsf(adapted_output_power - unadapted_output_power) > 0.1f
+||
+adapted_echo_power > unadapted_echo_power - 30.0f)
+{
+printf("Tests failed.\n");
+exit(2);
+}
+modem_echo_can_free(ctx);
+signal_free(&local_css);
+if (afCloseFile(resulthandle) != 0)
+{
+fprintf(stderr, "    Cannot close speech file '%s'\n", "result_sound.wav");
+exit(2);
+}
+afFreeFileSetup(filesetup);
+#if defined(ENABLE_GUI)
+if (use_gui)
+echo_can_monitor_wait_to_end();
+#endif
+printf("Tests passed.\n");
+return  0;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

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comparison spandsp-0.0.3/spandsp-0.0.3/tests/modem_echo_tests.c @ 5:f762bf195c4b