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asterisk/funcs/func_pitchshift.c

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2010, Digium, Inc.
*
* David Vossel <dvossel@digium.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
*
* \brief Pitch Shift Audio Effect
*
* \author David Vossel <dvossel@digium.com>
*
* \ingroup functions
*/
/************************* SMB FUNCTION LICENSE *********************************
*
* SYNOPSIS: Routine for doing pitch shifting while maintaining
* duration using the Short Time Fourier Transform.
*
* DESCRIPTION: The routine takes a pitchShift factor value which is between 0.5
* (one octave down) and 2. (one octave up). A value of exactly 1 does not change
* the pitch. num_samps_to_process tells the routine how many samples in indata[0...
* num_samps_to_process-1] should be pitch shifted and moved to outdata[0 ...
* num_samps_to_process-1]. The two buffers can be identical (ie. it can process the
* data in-place). fft_frame_size defines the FFT frame size used for the
* processing. Typical values are 1024, 2048 and 4096. It may be any value <=
* MAX_FRAME_LENGTH but it MUST be a power of 2. osamp is the STFT
* oversampling factor which also determines the overlap between adjacent STFT
* frames. It should at least be 4 for moderate scaling ratios. A value of 32 is
* recommended for best quality. sampleRate takes the sample rate for the signal
* in unit Hz, ie. 44100 for 44.1 kHz audio. The data passed to the routine in
* indata[] should be in the range [-1.0, 1.0), which is also the output range
* for the data, make sure you scale the data accordingly (for 16bit signed integers
* you would have to divide (and multiply) by 32768).
*
* COPYRIGHT 1999-2009 Stephan M. Bernsee <smb [AT] dspdimension [DOT] com>
*
* The Wide Open License (WOL)
*
* Permission to use, copy, modify, distribute and sell this software and its
* documentation for any purpose is hereby granted without fee, provided that
* the above copyright notice and this license appear in all source copies.
* THIS SOFTWARE IS PROVIDED "AS IS" WITHOUT EXPRESS OR IMPLIED WARRANTY OF
* ANY KIND. See http://www.dspguru.com/wol.htm for more information.
*
*****************************************************************************/
/*** MODULEINFO
<support_level>extended</support_level>
***/
#include "asterisk.h"
#include "asterisk/module.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/utils.h"
#include "asterisk/audiohook.h"
#include <math.h>
/*** DOCUMENTATION
<function name="PITCH_SHIFT" language="en_US">
<synopsis>
Pitch shift both tx and rx audio streams on a channel.
</synopsis>
<syntax>
<parameter name="channel direction" required="true">
<para>Direction can be either <literal>rx</literal>, <literal>tx</literal>, or
<literal>both</literal>. The direction can either be set to a valid floating
point number between 0.1 and 4.0 or one of the enum values listed below. A value
of 1.0 has no effect. Greater than 1 raises the pitch. Lower than 1 lowers
the pitch.</para>
<para>The pitch amount can also be set by the following values</para>
<enumlist>
<enum name = "highest" />
<enum name = "higher" />
<enum name = "high" />
<enum name = "low" />
<enum name = "lower" />
<enum name = "lowest" />
</enumlist>
</parameter>
</syntax>
<description>
<para>Examples:</para>
<example title="Raises pitch an octave">
exten => 1,1,Set(PITCH_SHIFT(tx)=highest)
</example>
<example title="Raises pitch more">
exten => 1,1,Set(PITCH_SHIFT(rx)=higher)
</example>
<example title="Raises pitch">
exten => 1,1,Set(PITCH_SHIFT(both)=high)
</example>
<example title="Lowers pitch">
exten => 1,1,Set(PITCH_SHIFT(rx)=low)
</example>
<example title="Lowers pitch more">
exten => 1,1,Set(PITCH_SHIFT(tx)=lower)
</example>
<example title="Lowers pitch an octave">
exten => 1,1,Set(PITCH_SHIFT(both)=lowest)
</example>
<example title="Lowers pitch">
exten => 1,1,Set(PITCH_SHIFT(rx)=0.8)
</example>
<example title="Raises pitch">
exten => 1,1,Set(PITCH_SHIFT(tx)=1.5)
</example>
</description>
</function>
***/
#ifndef M_PI
#define M_PI 3.14159265358979323846
#endif
#define MAX_FRAME_LENGTH 256
#define HIGHEST 2
#define HIGHER 1.5
#define HIGH 1.25
#define LOW .85
#define LOWER .7
#define LOWEST .5
struct fft_data {
float in_fifo[MAX_FRAME_LENGTH];
float out_fifo[MAX_FRAME_LENGTH];
float fft_worksp[2*MAX_FRAME_LENGTH];
float last_phase[MAX_FRAME_LENGTH/2+1];
float sum_phase[MAX_FRAME_LENGTH/2+1];
float output_accum[2*MAX_FRAME_LENGTH];
float ana_freq[MAX_FRAME_LENGTH];
float ana_magn[MAX_FRAME_LENGTH];
float syn_freq[MAX_FRAME_LENGTH];
float sys_magn[MAX_FRAME_LENGTH];
long gRover;
float shift_amount;
};
struct pitchshift_data {
struct ast_audiohook audiohook;
struct fft_data rx;
struct fft_data tx;
};
static void smb_fft(float *fft_buffer, long fft_frame_size, long sign);
static void smb_pitch_shift(float pitchShift, long num_samps_to_process, long fft_frame_size, long osamp, float sample_rate, int16_t *indata, int16_t *outdata, struct fft_data *fft_data);
static int pitch_shift(struct ast_frame *f, float amount, struct fft_data *fft_data);
static void destroy_callback(void *data)
{
struct pitchshift_data *shift = data;
ast_audiohook_destroy(&shift->audiohook);
ast_free(shift);
};
static const struct ast_datastore_info pitchshift_datastore = {
.type = "pitchshift",
.destroy = destroy_callback
};
static int pitchshift_cb(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *f, enum ast_audiohook_direction direction)
{
struct ast_datastore *datastore = NULL;
struct pitchshift_data *shift = NULL;
if (!f) {
return 0;
}
if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE) {
return -1;
}
if (!(datastore = ast_channel_datastore_find(chan, &pitchshift_datastore, NULL))) {
return -1;
}
shift = datastore->data;
if (direction == AST_AUDIOHOOK_DIRECTION_WRITE) {
pitch_shift(f, shift->tx.shift_amount, &shift->tx);
} else {
pitch_shift(f, shift->rx.shift_amount, &shift->rx);
}
return 0;
}
static int pitchshift_helper(struct ast_channel *chan, const char *cmd, char *data, const char *value)
{
struct ast_datastore *datastore = NULL;
struct pitchshift_data *shift = NULL;
int new = 0;
float amount = 0;
if (!chan) {
ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
return -1;
}
ast_channel_lock(chan);
if (!(datastore = ast_channel_datastore_find(chan, &pitchshift_datastore, NULL))) {
ast_channel_unlock(chan);
if (!(datastore = ast_datastore_alloc(&pitchshift_datastore, NULL))) {
return 0;
}
if (!(shift = ast_calloc(1, sizeof(*shift)))) {
ast_datastore_free(datastore);
return 0;
}
ast_audiohook_init(&shift->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "pitch_shift", AST_AUDIOHOOK_MANIPULATE_ALL_RATES);
shift->audiohook.manipulate_callback = pitchshift_cb;
datastore->data = shift;
new = 1;
} else {
ast_channel_unlock(chan);
shift = datastore->data;
}
if (!strcasecmp(value, "highest")) {
amount = HIGHEST;
} else if (!strcasecmp(value, "higher")) {
amount = HIGHER;
} else if (!strcasecmp(value, "high")) {
amount = HIGH;
} else if (!strcasecmp(value, "lowest")) {
amount = LOWEST;
} else if (!strcasecmp(value, "lower")) {
amount = LOWER;
} else if (!strcasecmp(value, "low")) {
amount = LOW;
} else {
if (!sscanf(value, "%30f", &amount) || (amount <= 0) || (amount > 4)) {
goto cleanup_error;
}
}
if (!strcasecmp(data, "rx")) {
shift->rx.shift_amount = amount;
} else if (!strcasecmp(data, "tx")) {
shift->tx.shift_amount = amount;
} else if (!strcasecmp(data, "both")) {
shift->rx.shift_amount = amount;
shift->tx.shift_amount = amount;
} else {
goto cleanup_error;
}
if (new) {
ast_channel_lock(chan);
ast_channel_datastore_add(chan, datastore);
ast_channel_unlock(chan);
ast_audiohook_attach(chan, &shift->audiohook);
}
return 0;
cleanup_error:
ast_log(LOG_ERROR, "Invalid argument provided to the %s function\n", cmd);
if (new) {
ast_datastore_free(datastore);
}
return -1;
}
static void smb_fft(float *fft_buffer, long fft_frame_size, long sign)
{
float wr, wi, arg, *p1, *p2, temp;
float tr, ti, ur, ui, *p1r, *p1i, *p2r, *p2i;
long i, bitm, j, le, le2, k;
for (i = 2; i < 2 * fft_frame_size - 2; i += 2) {
for (bitm = 2, j = 0; bitm < 2 * fft_frame_size; bitm <<= 1) {
if (i & bitm) {
j++;
}
j <<= 1;
}
if (i < j) {
p1 = fft_buffer + i; p2 = fft_buffer + j;
temp = *p1; *(p1++) = *p2;
*(p2++) = temp; temp = *p1;
*p1 = *p2; *p2 = temp;
}
}
for (k = 0, le = 2; k < (long) (log(fft_frame_size) / log(2.) + .5); k++) {
le <<= 1;
le2 = le>>1;
ur = 1.0;
ui = 0.0;
arg = M_PI / (le2>>1);
wr = cos(arg);
wi = sign * sin(arg);
for (j = 0; j < le2; j += 2) {
p1r = fft_buffer+j; p1i = p1r + 1;
p2r = p1r + le2; p2i = p2r + 1;
for (i = j; i < 2 * fft_frame_size; i += le) {
tr = *p2r * ur - *p2i * ui;
ti = *p2r * ui + *p2i * ur;
*p2r = *p1r - tr; *p2i = *p1i - ti;
*p1r += tr; *p1i += ti;
p1r += le; p1i += le;
p2r += le; p2i += le;
}
tr = ur * wr - ui * wi;
ui = ur * wi + ui * wr;
ur = tr;
}
}
}
static void smb_pitch_shift(float pitchShift, long num_samps_to_process, long fft_frame_size, long osamp, float sample_rate, int16_t *indata, int16_t *outdata, struct fft_data *fft_data)
{
float *in_fifo = fft_data->in_fifo;
float *out_fifo = fft_data->out_fifo;
float *fft_worksp = fft_data->fft_worksp;
float *last_phase = fft_data->last_phase;
float *sum_phase = fft_data->sum_phase;
float *output_accum = fft_data->output_accum;
float *ana_freq = fft_data->ana_freq;
float *ana_magn = fft_data->ana_magn;
float *syn_freq = fft_data->syn_freq;
float *sys_magn = fft_data->sys_magn;
double magn, phase, tmp, window, real, imag;
double freq_per_bin, expect;
long i,k, qpd, index, in_fifo_latency, step_size, fft_frame_size2;
/* set up some handy variables */
fft_frame_size2 = fft_frame_size / 2;
step_size = fft_frame_size / osamp;
freq_per_bin = sample_rate / (double) fft_frame_size;
expect = 2. * M_PI * (double) step_size / (double) fft_frame_size;
in_fifo_latency = fft_frame_size-step_size;
if (fft_data->gRover == 0) {
fft_data->gRover = in_fifo_latency;
}
/* main processing loop */
for (i = 0; i < num_samps_to_process; i++){
/* As long as we have not yet collected enough data just read in */
in_fifo[fft_data->gRover] = indata[i];
outdata[i] = out_fifo[fft_data->gRover - in_fifo_latency];
fft_data->gRover++;
/* now we have enough data for processing */
if (fft_data->gRover >= fft_frame_size) {
fft_data->gRover = in_fifo_latency;
/* do windowing and re,im interleave */
for (k = 0; k < fft_frame_size;k++) {
window = -.5 * cos(2. * M_PI * (double) k / (double) fft_frame_size) + .5;
fft_worksp[2*k] = in_fifo[k] * window;
fft_worksp[2*k+1] = 0.;
}
/* ***************** ANALYSIS ******************* */
/* do transform */
smb_fft(fft_worksp, fft_frame_size, -1);
/* this is the analysis step */
for (k = 0; k <= fft_frame_size2; k++) {
/* de-interlace FFT buffer */
real = fft_worksp[2*k];
imag = fft_worksp[2*k+1];
/* compute magnitude and phase */
magn = 2. * sqrt(real * real + imag * imag);
phase = atan2(imag, real);
/* compute phase difference */
tmp = phase - last_phase[k];
last_phase[k] = phase;
/* subtract expected phase difference */
tmp -= (double) k * expect;
/* map delta phase into +/- Pi interval */
qpd = tmp / M_PI;
if (qpd >= 0) {
qpd += qpd & 1;
} else {
qpd -= qpd & 1;
}
tmp -= M_PI * (double) qpd;
/* get deviation from bin frequency from the +/- Pi interval */
tmp = osamp * tmp / (2. * M_PI);
/* compute the k-th partials' true frequency */
tmp = (double) k * freq_per_bin + tmp * freq_per_bin;
/* store magnitude and true frequency in analysis arrays */
ana_magn[k] = magn;
ana_freq[k] = tmp;
}
/* ***************** PROCESSING ******************* */
/* this does the actual pitch shifting */
memset(sys_magn, 0, fft_frame_size * sizeof(float));
memset(syn_freq, 0, fft_frame_size * sizeof(float));
for (k = 0; k <= fft_frame_size2; k++) {
index = k * pitchShift;
if (index <= fft_frame_size2) {
sys_magn[index] += ana_magn[k];
syn_freq[index] = ana_freq[k] * pitchShift;
}
}
/* ***************** SYNTHESIS ******************* */
/* this is the synthesis step */
for (k = 0; k <= fft_frame_size2; k++) {
/* get magnitude and true frequency from synthesis arrays */
magn = sys_magn[k];
tmp = syn_freq[k];
/* subtract bin mid frequency */
tmp -= (double) k * freq_per_bin;
/* get bin deviation from freq deviation */
tmp /= freq_per_bin;
/* take osamp into account */
tmp = 2. * M_PI * tmp / osamp;
/* add the overlap phase advance back in */
tmp += (double) k * expect;
/* accumulate delta phase to get bin phase */
sum_phase[k] += tmp;
phase = sum_phase[k];
/* get real and imag part and re-interleave */
fft_worksp[2*k] = magn * cos(phase);
fft_worksp[2*k+1] = magn * sin(phase);
}
/* zero negative frequencies */
for (k = fft_frame_size + 2; k < 2 * fft_frame_size; k++) {
fft_worksp[k] = 0.;
}
/* do inverse transform */
smb_fft(fft_worksp, fft_frame_size, 1);
/* do windowing and add to output accumulator */
for (k = 0; k < fft_frame_size; k++) {
window = -.5 * cos(2. * M_PI * (double) k / (double) fft_frame_size) + .5;
output_accum[k] += 2. * window * fft_worksp[2*k] / (fft_frame_size2 * osamp);
}
for (k = 0; k < step_size; k++) {
out_fifo[k] = output_accum[k];
}
/* shift accumulator */
memmove(output_accum, output_accum+step_size, fft_frame_size * sizeof(float));
/* move input FIFO */
for (k = 0; k < in_fifo_latency; k++) {
in_fifo[k] = in_fifo[k+step_size];
}
}
}
}
static int pitch_shift(struct ast_frame *f, float amount, struct fft_data *fft)
{
int16_t *fun = (int16_t *) f->data.ptr;
int samples;
/* an amount of 1 has no effect */
if (!amount || amount == 1 || !fun || (f->samples % 32)) {
return 0;
}
for (samples = 0; samples < f->samples; samples += 32) {
smb_pitch_shift(amount, 32, MAX_FRAME_LENGTH, 32, ast_format_get_sample_rate(f->subclass.format), fun+samples, fun+samples, fft);
}
return 0;
}
static struct ast_custom_function pitch_shift_function = {
.name = "PITCH_SHIFT",
.write = pitchshift_helper,
};
static int unload_module(void)
{
return ast_custom_function_unregister(&pitch_shift_function);
}
static int load_module(void)
{
int res = ast_custom_function_register(&pitch_shift_function);
return res ? AST_MODULE_LOAD_DECLINE : AST_MODULE_LOAD_SUCCESS;
}
AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "Audio Effects Dialplan Functions");
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