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PITCH_SHIFT dialplan function
The PITCH_SHIFT function can be used on a channel to independently modify the pitch of both rx and tx audio streams. Now you can improve your conference calls by assigning a random pitch effect to everyone entering a meetme room, or just make your day more interesting by making your co-workers sound funny. These are just some of the numerious practical uses for this function. Enjoy! https://reviewboard.asterisk.org/r/526/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251038 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
David Vossel
2
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2
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@@ -143,6 +143,8 @@ Applications
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Dialplan Functions
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Dialplan Functions
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------------------
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------------------
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* PITCH_SHIFT dialplan function added. This function can be used to modify the
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pitch of a channel's tx and rx audio streams.
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* Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
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* Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
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setting various connected line and redirecting party information.
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setting various connected line and redirecting party information.
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* CALLERID and CONNECTEDLINE dialplan functions have been extended to
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* CALLERID and CONNECTEDLINE dialplan functions have been extended to
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503
funcs/func_pitchshift.c
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503
funcs/func_pitchshift.c
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@@ -0,0 +1,503 @@
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/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2010, Digium, Inc.
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*
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* David Vossel <dvossel@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*! \file
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*
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* \brief Pitch Shift Audio Effect
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*
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* \author David Vossel <dvossel@digium.com>
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*
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* \ingroup functions
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*/
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/************************* SMB FUNCTION LICENSE *********************************
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*
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* SYNOPSIS: Routine for doing pitch shifting while maintaining
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* duration using the Short Time Fourier Transform.
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*
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* DESCRIPTION: The routine takes a pitchShift factor value which is between 0.5
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* (one octave down) and 2. (one octave up). A value of exactly 1 does not change
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* the pitch. num_samps_to_process tells the routine how many samples in indata[0...
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* num_samps_to_process-1] should be pitch shifted and moved to outdata[0 ...
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* num_samps_to_process-1]. The two buffers can be identical (ie. it can process the
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* data in-place). fft_frame_size defines the FFT frame size used for the
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* processing. Typical values are 1024, 2048 and 4096. It may be any value <=
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* MAX_FRAME_LENGTH but it MUST be a power of 2. osamp is the STFT
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* oversampling factor which also determines the overlap between adjacent STFT
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* frames. It should at least be 4 for moderate scaling ratios. A value of 32 is
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* recommended for best quality. sampleRate takes the sample rate for the signal
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* in unit Hz, ie. 44100 for 44.1 kHz audio. The data passed to the routine in
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* indata[] should be in the range [-1.0, 1.0), which is also the output range
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* for the data, make sure you scale the data accordingly (for 16bit signed integers
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* you would have to divide (and multiply) by 32768).
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*
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* COPYRIGHT 1999-2009 Stephan M. Bernsee <smb [AT] dspdimension [DOT] com>
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*
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* The Wide Open License (WOL)
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*
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* Permission to use, copy, modify, distribute and sell this software and its
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* documentation for any purpose is hereby granted without fee, provided that
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* the above copyright notice and this license appear in all source copies.
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* THIS SOFTWARE IS PROVIDED "AS IS" WITHOUT EXPRESS OR IMPLIED WARRANTY OF
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* ANY KIND. See http://www.dspguru.com/wol.htm for more information.
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*
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*****************************************************************************/
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include "asterisk/module.h"
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#include "asterisk/channel.h"
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#include "asterisk/pbx.h"
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#include "asterisk/utils.h"
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#include "asterisk/audiohook.h"
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#include <math.h>
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/*** DOCUMENTATION
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<function name="PITCH_SHIFT" language="en_US">
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<synopsis>
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Pitch shift both tx and rx audio streams on a channel.
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</synopsis>
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<syntax>
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<parameter name="channel direction" required="true">
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<para>Direction can be either <literal>rx</literal>, <literal>tx</literal>, or
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<literal>both</literal>. The direction can either be set to a valid floating
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point number between 0.1 and 4.0 or one of the enum values listed below. A value
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of 1.0 has no effect. Greater than 1 raises the pitch. Lower than 1 lowers
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the pitch.</para>
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<para>The pitch amount can also be set by the following values</para>
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<enumlist>
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<enum name = "highest" />
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<enum name = "higher" />
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<enum name = "high" />
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<enum name = "low" />
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<enum name = "lower" />
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<enum name = "lowest" />
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</parameter>
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</syntax>
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<description>
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<para>Examples:</para>
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<para>exten => 1,1,Set(PITCH_SHIFT(tx)=highest); raises pitch an octave </para>
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<para>exten => 1,1,Set(PITCH_SHIFT(rx)=higher) ; raises pitch more </para>
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<para>exten => 1,1,Set(PITCH_SHIFT(both)=high) ; raises pitch </para>
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<para>exten => 1,1,Set(PITCH_SHIFT(rx)=low) ; lowers pitch </para>
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<para>exten => 1,1,Set(PITCH_SHIFT(tx)=lower) ; lowers pitch more </para>
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<para>exten => 1,1,Set(PITCH_SHIFT(both)=lowest) ; lowers pitch an octave </para>
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<para>exten => 1,1,Set(PITCH_SHIFT(rx)=0.8) ; lowers pitch </para>
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<para>exten => 1,1,Set(PITCH_SHIFT(tx)=1.5) ; raises pitch </para>
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</description>
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</function>
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***/
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#define M_PI 3.14159265358979323846
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#define MAX_FRAME_LENGTH 256
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#define HIGHEST 2
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#define HIGHER 1.5
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#define HIGH 1.25
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#define LOW .85
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#define LOWER .7
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#define LOWEST .5
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struct fft_data {
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float in_fifo[MAX_FRAME_LENGTH];
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float out_fifo[MAX_FRAME_LENGTH];
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float fft_worksp[2*MAX_FRAME_LENGTH];
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float last_phase[MAX_FRAME_LENGTH/2+1];
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float sum_phase[MAX_FRAME_LENGTH/2+1];
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float output_accum[2*MAX_FRAME_LENGTH];
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float ana_freq[MAX_FRAME_LENGTH];
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float ana_magn[MAX_FRAME_LENGTH];
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float syn_freq[MAX_FRAME_LENGTH];
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float sys_magn[MAX_FRAME_LENGTH];
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long gRover;
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float shift_amount;
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};
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struct pitchshift_data {
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struct ast_audiohook audiohook;
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struct fft_data rx;
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struct fft_data tx;
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};
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static void smb_fft(float *fft_buffer, long fft_frame_size, long sign);
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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);
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static int pitch_shift(struct ast_frame *f, float amount, struct fft_data *fft_data);
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static void destroy_callback(void *data)
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{
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struct pitchshift_data *shift = data;
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ast_audiohook_destroy(&shift->audiohook);
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ast_free(shift);
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};
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static const struct ast_datastore_info pitchshift_datastore = {
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.type = "pitchshift",
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.destroy = destroy_callback
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};
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static int pitchshift_cb(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *f, enum ast_audiohook_direction direction)
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{
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struct ast_datastore *datastore = NULL;
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struct pitchshift_data *shift = NULL;
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if (!f) {
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return 0;
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}
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if ((audiohook->status == AST_AUDIOHOOK_STATUS_DONE) ||
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(f->frametype != AST_FRAME_VOICE) ||
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((f->subclass.codec != AST_FORMAT_SLINEAR) &&
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(f->subclass.codec != AST_FORMAT_SLINEAR16))) {
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return -1;
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}
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if (!(datastore = ast_channel_datastore_find(chan, &pitchshift_datastore, NULL))) {
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return -1;
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}
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shift = datastore->data;
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if (direction == AST_AUDIOHOOK_DIRECTION_WRITE) {
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pitch_shift(f, shift->tx.shift_amount, &shift->tx);
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} else {
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pitch_shift(f, shift->rx.shift_amount, &shift->rx);
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}
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return 0;
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}
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static int pitchshift_helper(struct ast_channel *chan, const char *cmd, char *data, const char *value)
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{
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struct ast_datastore *datastore = NULL;
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struct pitchshift_data *shift = NULL;
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int new = 0;
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float amount = 0;
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ast_channel_lock(chan);
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if (!(datastore = ast_channel_datastore_find(chan, &pitchshift_datastore, NULL))) {
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ast_channel_unlock(chan);
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if (!(datastore = ast_datastore_alloc(&pitchshift_datastore, NULL))) {
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return 0;
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}
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if (!(shift = ast_calloc(1, sizeof(*shift)))) {
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ast_datastore_free(datastore);
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return 0;
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}
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ast_audiohook_init(&shift->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "pitch_shift");
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shift->audiohook.manipulate_callback = pitchshift_cb;
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datastore->data = shift;
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new = 1;
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} else {
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ast_channel_unlock(chan);
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shift = datastore->data;
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}
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if (!strcasecmp(value, "highest")) {
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amount = HIGHEST;
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} else if (!strcasecmp(value, "higher")) {
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amount = HIGHER;
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} else if (!strcasecmp(value, "high")) {
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amount = HIGH;
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} else if (!strcasecmp(value, "lowest")) {
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amount = LOWEST;
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} else if (!strcasecmp(value, "lower")) {
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amount = LOWER;
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} else if (!strcasecmp(value, "low")) {
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amount = LOW;
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} else {
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if (!sscanf(value, "%30f", &amount) || (amount <= 0) || (amount > 4)) {
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goto cleanup_error;
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}
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}
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if (!strcasecmp(data, "rx")) {
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shift->rx.shift_amount = amount;
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} else if (!strcasecmp(data, "tx")) {
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shift->tx.shift_amount = amount;
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} else if (!strcasecmp(data, "both")) {
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shift->rx.shift_amount = amount;
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shift->tx.shift_amount = amount;
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} else {
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goto cleanup_error;
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}
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if (new) {
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ast_channel_lock(chan);
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ast_channel_datastore_add(chan, datastore);
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ast_channel_unlock(chan);
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ast_audiohook_attach(chan, &shift->audiohook);
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}
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return 0;
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cleanup_error:
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ast_log(LOG_ERROR, "Invalid argument provided to the %s function\n", cmd);
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if (new) {
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ast_datastore_free(datastore);
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}
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return -1;
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}
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static void smb_fft(float *fft_buffer, long fft_frame_size, long sign)
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{
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float wr, wi, arg, *p1, *p2, temp;
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float tr, ti, ur, ui, *p1r, *p1i, *p2r, *p2i;
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long i, bitm, j, le, le2, k;
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for (i = 2; i < 2 * fft_frame_size - 2; i += 2) {
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for (bitm = 2, j = 0; bitm < 2 * fft_frame_size; bitm <<= 1) {
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if (i & bitm) {
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j++;
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}
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j <<= 1;
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|
}
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|
if (i < j) {
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p1 = fft_buffer + i; p2 = fft_buffer + j;
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|
temp = *p1; *(p1++) = *p2;
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|
*(p2++) = temp; temp = *p1;
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*p1 = *p2; *p2 = temp;
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|
}
|
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|
}
|
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|
for (k = 0, le = 2; k < (long) (log(fft_frame_size) / log(2.) + .5); k++) {
|
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|
le <<= 1;
|
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|
le2 = le>>1;
|
||||||
|
ur = 1.0;
|
||||||
|
ui = 0.0;
|
||||||
|
arg = M_PI / (le2>>1);
|
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|
wr = cos(arg);
|
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|
wi = sign * sin(arg);
|
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|
for (j = 0; j < le2; j += 2) {
|
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|
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");
|
||||||
Reference in New Issue
Block a user