func_pitchshift.c

Go to the documentation of this file.

/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 2010, Digium, Inc.
 *
 * David Vossel <[email protected]>
 *
 * 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 <[email protected]>
 *
 * \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>
 <para>exten => 1,1,Set(PITCH_SHIFT(tx)=highest); raises pitch an octave </para>
 <para>exten => 1,1,Set(PITCH_SHIFT(rx)=higher) ; raises pitch more </para>
 <para>exten => 1,1,Set(PITCH_SHIFT(both)=high) ; raises pitch </para>
 <para>exten => 1,1,Set(PITCH_SHIFT(rx)=low) ; lowers pitch </para>
 <para>exten => 1,1,Set(PITCH_SHIFT(tx)=lower) ; lowers pitch more </para>
 <para>exten => 1,1,Set(PITCH_SHIFT(both)=lowest) ; lowers pitch an octave </para>

 <para>exten => 1,1,Set(PITCH_SHIFT(rx)=0.8) ; lowers pitch </para>
 <para>exten => 1,1,Set(PITCH_SHIFT(tx)=1.5) ; raises pitch </para>
 </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, expct;
 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;
 expct = 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 * expct;

 /* 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 * expct;

 /* 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");