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">
        <since>
            <version>1.8.0</version>
        </since>
        <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");