--- /dev/null
+/*
+ * 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.
+*
+*****************************************************************************/
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#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" />
+ </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>
+ ***/
+
+#define M_PI 3.14159265358979323846
+#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) ||
+ (f->frametype != AST_FRAME_VOICE) ||
+ ((f->subclass.codec != AST_FORMAT_SLINEAR) &&
+ (f->subclass.codec != AST_FORMAT_SLINEAR16))) {
+ 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;
+
+ 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");
+ 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_rate(f->subclass.codec), 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(ASTERISK_GPL_KEY, "Audio Effects Dialplan Functions");
projects / thirdparty / asterisk.git / commitdiff
