/***************************************************************************
 *   Copyright (C) 2005 to 2013 by Jonathan Duddington                     *
 *   email: jonsd@users.sourceforge.net                                    *
 *                                                                         *
 *   This program is free software; you can redistribute it and/or modify  *
 *   it under the terms of the GNU General Public License as published by  *
 *   the Free Software Foundation; either version 3 of the License, or     *
 *   (at your option) any later version.                                   *
 *                                                                         *
 *   This program is distributed in the hope that it will be useful,       *
 *   but WITHOUT ANY WARRANTY; without even the implied warranty of        *
 *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the         *
 *   GNU General Public License for more details.                          *
 *                                                                         *
 *   You should have received a copy of the GNU General Public License     *
 *   along with this program; if not, see:                                 *
 *               <http://www.gnu.org/licenses/>.                           *
 ***************************************************************************/

#include "StdAfx.h"

// this version keeps wavemult window as a constant fraction
// of the cycle length - but that spreads out the HF peaks too much

#include <stdio.h>
#include <string.h>
#include <stdlib.h>
#include <math.h>



#include "speak_lib.h"
#include "speech.h"
#include "phoneme.h"
#include "synthesize.h"
#include "voice.h"

#ifdef INCLUDE_SONIC
#include "sonic.h"
#endif

#ifdef USE_PORTAUDIO
#include "portaudio.h"
#undef USE_PORTAUDIO
// determine portaudio version by looking for a #define which is not in V18
#ifdef paNeverDropInput
#define USE_PORTAUDIO   19
#else
#define USE_PORTAUDIO   18
#endif
#endif

#define N_SINTAB  2048
#include "sintab.h"


#define PI  3.1415927
#define PI2 6.283185307
#define N_WAV_BUF   10

voice_t *wvoice;

FILE *f_log = NULL;
int option_waveout = 0;
static int option_harmonic1 = 10;   // 10
int option_log_frames = 0;
static int flutter_amp = 64;

static int general_amplitude = 60;
static int consonant_amp = 26;   // 24

int embedded_value[N_EMBEDDED_VALUES];

static int PHASE_INC_FACTOR;
int samplerate = 0;       // this is set by Wavegeninit()
int samplerate_native=0;
extern int option_device_number;
extern int option_quiet;

static wavegen_peaks_t peaks[N_PEAKS];
static int peak_harmonic[N_PEAKS];
static int peak_height[N_PEAKS];

int echo_head;
int echo_tail;
int echo_amp = 0;
short echo_buf[N_ECHO_BUF];
static int echo_length = 0;   // period (in sample\) to ensure completion of echo at the end of speech, set in WavegenSetEcho()

static int voicing;
static RESONATOR rbreath[N_PEAKS];

static int harm_sqrt_n = 0;


#define N_LOWHARM  30
static int harm_inc[N_LOWHARM];    // only for these harmonics do we interpolate amplitude between steps
static int *harmspect;
static int hswitch=0;
static int hspect[2][MAX_HARMONIC];         // 2 copies, we interpolate between then
static int max_hval=0;

static int nsamples=0;       // number to do
static int modulation_type = 0;
static int glottal_flag = 0;
static int glottal_reduce = 0;


WGEN_DATA wdata;

static int amp_ix;
static int amp_inc;
static unsigned char *amplitude_env = NULL;

static int samplecount=0;    // number done
static int samplecount_start=0;  // count at start of this segment
static int end_wave=0;      // continue to end of wave cycle
static int wavephase;
static int phaseinc;
static int cycle_samples;         // number of samples in a cycle at current pitch
static int cbytes;
static int hf_factor;

static double minus_pi_t;
static double two_pi_t;


unsigned char *out_ptr;
unsigned char *out_start;
unsigned char *out_end;
int outbuf_size = 0;

// the queue of operations passed to wavegen from sythesize
long64 wcmdq[N_WCMDQ][4];
int wcmdq_head=0;
int wcmdq_tail=0;

// pitch,speed,
int embedded_default[N_EMBEDDED_VALUES]        = {0,    50,175,100,50, 0, 0, 0,175,0,0,0,0,0,0};
static int embedded_max[N_EMBEDDED_VALUES]     = {0,0x7fff,750,300,99,99,99, 0,750,0,0,0,0,4,0};

#define N_CALLBACK_IX N_WAV_BUF-2   // adjust this delay to match display with the currently spoken word
int current_source_index=0;

extern FILE *f_wave;

#if (USE_PORTAUDIO == 18)
static PortAudioStream *pa_stream=NULL;
#endif
#if (USE_PORTAUDIO == 19)
static PaStream *pa_stream=NULL;
#endif

#ifdef INCLUDE_SONIC
static sonicStream sonicSpeedupStream = NULL;
double sonicSpeed = 1.0;
#endif

// 1st index=roughness
// 2nd index=modulation_type
// value: bits 0-3  amplitude (16ths), bits 4-7 every n cycles
#define N_ROUGHNESS 8
static unsigned char modulation_tab[N_ROUGHNESS][8] = {
	{0, 0x00, 0x00, 0x00, 0, 0x46, 0xf2, 0x29},
	{0, 0x2f, 0x00, 0x2f, 0, 0x45, 0xf2, 0x29},
	{0, 0x2f, 0x00, 0x2e, 0, 0x45, 0xf2, 0x28},
	{0, 0x2e, 0x00, 0x2d, 0, 0x34, 0xf2, 0x28},
	{0, 0x2d, 0x2d, 0x2c, 0, 0x34, 0xf2, 0x28},
	{0, 0x2b, 0x2b, 0x2b, 0, 0x34, 0xf2, 0x28},
	{0, 0x2a, 0x2a, 0x2a, 0, 0x34, 0xf2, 0x28},
	{0, 0x29, 0x29, 0x29, 0, 0x34, 0xf2, 0x28},
};

// Flutter table, to add natural variations to the pitch
#define N_FLUTTER  0x170
static int Flutter_inc;
static const unsigned char Flutter_tab[N_FLUTTER] = {
   0x80, 0x9b, 0xb5, 0xcb, 0xdc, 0xe8, 0xed, 0xec,
   0xe6, 0xdc, 0xce, 0xbf, 0xb0, 0xa3, 0x98, 0x90,
   0x8c, 0x8b, 0x8c, 0x8f, 0x92, 0x94, 0x95, 0x92,
   0x8c, 0x83, 0x78, 0x69, 0x59, 0x49, 0x3c, 0x31,
   0x2a, 0x29, 0x2d, 0x36, 0x44, 0x56, 0x69, 0x7d,
   0x8f, 0x9f, 0xaa, 0xb1, 0xb2, 0xad, 0xa4, 0x96,
   0x87, 0x78, 0x69, 0x5c, 0x53, 0x4f, 0x4f, 0x55,
   0x5e, 0x6b, 0x7a, 0x88, 0x96, 0xa2, 0xab, 0xb0,

   0xb1, 0xae, 0xa8, 0xa0, 0x98, 0x91, 0x8b, 0x88,
   0x89, 0x8d, 0x94, 0x9d, 0xa8, 0xb2, 0xbb, 0xc0,
   0xc1, 0xbd, 0xb4, 0xa5, 0x92, 0x7c, 0x63, 0x4a,
   0x32, 0x1e, 0x0e, 0x05, 0x02, 0x05, 0x0f, 0x1e,
   0x30, 0x44, 0x59, 0x6d, 0x7f, 0x8c, 0x96, 0x9c,
   0x9f, 0x9f, 0x9d, 0x9b, 0x99, 0x99, 0x9c, 0xa1,
   0xa9, 0xb3, 0xbf, 0xca, 0xd5, 0xdc, 0xe0, 0xde,
   0xd8, 0xcc, 0xbb, 0xa6, 0x8f, 0x77, 0x60, 0x4b,

   0x3a, 0x2e, 0x28, 0x29, 0x2f, 0x3a, 0x48, 0x59,
   0x6a, 0x7a, 0x86, 0x90, 0x94, 0x95, 0x91, 0x89,
   0x80, 0x75, 0x6b, 0x62, 0x5c, 0x5a, 0x5c, 0x61,
   0x69, 0x74, 0x80, 0x8a, 0x94, 0x9a, 0x9e, 0x9d,
   0x98, 0x90, 0x86, 0x7c, 0x71, 0x68, 0x62, 0x60,
   0x63, 0x6b, 0x78, 0x88, 0x9b, 0xaf, 0xc2, 0xd2,
   0xdf, 0xe6, 0xe7, 0xe2, 0xd7, 0xc6, 0xb2, 0x9c,
   0x84, 0x6f, 0x5b, 0x4b, 0x40, 0x39, 0x37, 0x38,

   0x3d, 0x43, 0x4a, 0x50, 0x54, 0x56, 0x55, 0x52,
   0x4d, 0x48, 0x42, 0x3f, 0x3e, 0x41, 0x49, 0x56,
   0x67, 0x7c, 0x93, 0xab, 0xc3, 0xd9, 0xea, 0xf6,
   0xfc, 0xfb, 0xf4, 0xe7, 0xd5, 0xc0, 0xaa, 0x94,
   0x80, 0x71, 0x64, 0x5d, 0x5a, 0x5c, 0x61, 0x68,
   0x70, 0x77, 0x7d, 0x7f, 0x7f, 0x7b, 0x74, 0x6b,
   0x61, 0x57, 0x4e, 0x48, 0x46, 0x48, 0x4e, 0x59,
   0x66, 0x75, 0x84, 0x93, 0x9f, 0xa7, 0xab, 0xaa,

   0xa4, 0x99, 0x8b, 0x7b, 0x6a, 0x5b, 0x4e, 0x46,
   0x43, 0x45, 0x4d, 0x5a, 0x6b, 0x7f, 0x92, 0xa6,
   0xb8, 0xc5, 0xcf, 0xd3, 0xd2, 0xcd, 0xc4, 0xb9,
   0xad, 0xa1, 0x96, 0x8e, 0x89, 0x87, 0x87, 0x8a,
   0x8d, 0x91, 0x92, 0x91, 0x8c, 0x84, 0x78, 0x68,
   0x55, 0x41, 0x2e, 0x1c, 0x0e, 0x05, 0x01, 0x05,
   0x0f, 0x1f, 0x34, 0x4d, 0x68, 0x81, 0x9a, 0xb0,
   0xc1, 0xcd, 0xd3, 0xd3, 0xd0, 0xc8, 0xbf, 0xb5,

   0xab, 0xa4, 0x9f, 0x9c, 0x9d, 0xa0, 0xa5, 0xaa,
   0xae, 0xb1, 0xb0, 0xab, 0xa3, 0x96, 0x87, 0x76,
   0x63, 0x51, 0x42, 0x36, 0x2f, 0x2d, 0x31, 0x3a,
   0x48, 0x59, 0x6b, 0x7e, 0x8e, 0x9c, 0xa6, 0xaa,
   0xa9, 0xa3, 0x98, 0x8a, 0x7b, 0x6c, 0x5d, 0x52,
   0x4a, 0x48, 0x4a, 0x50, 0x5a, 0x67, 0x75, 0x82
};

// waveform shape table for HF peaks, formants 6,7,8
#define N_WAVEMULT 128
static int wavemult_offset=0;
static int wavemult_max=0;

// the presets are for 22050 Hz sample rate.
// A different rate will need to recalculate the presets in WavegenInit()
static unsigned char wavemult[N_WAVEMULT] = {
  0,  0,  0,  2,  3,  5,  8, 11, 14, 18, 22, 27, 32, 37, 43, 49,
    55, 62, 69, 76, 83, 90, 98,105,113,121,128,136,144,152,159,166,
   174,181,188,194,201,207,213,218,224,228,233,237,240,244,246,249,
   251,252,253,253,253,253,252,251,249,246,244,240,237,233,228,224,
   218,213,207,201,194,188,181,174,166,159,152,144,136,128,121,113,
   105, 98, 90, 83, 76, 69, 62, 55, 49, 43, 37, 32, 27, 22, 18, 14,
    11,  8,  5,  3,  2,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,
     0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0 };


// set from y = pow(2,x) * 128,  x=-1 to 1
unsigned char pitch_adjust_tab[MAX_PITCH_VALUE+1] = {
    64, 65, 66, 67, 68, 69, 70, 71,
    72, 73, 74, 75, 76, 77, 78, 79,
    80, 81, 82, 83, 84, 86, 87, 88,
    89, 91, 92, 93, 94, 96, 97, 98,
   100,101,103,104,105,107,108,110,
   111,113,115,116,118,119,121,123,
   124,126,128,130,132,133,135,137,
   139,141,143,145,147,149,151,153,
   155,158,160,162,164,167,169,171,
   174,176,179,181,184,186,189,191,
   194,197,199,202,205,208,211,214,
   217,220,223,226,229,232,236,239,
   242,246,249,252, 254,255 };


#ifdef LOG_FRAMES
static void LogMarker(int type, int value, int value2)
{//===================================================
	char buf[20];
	int *p;

	if(option_log_frames == 0)
		return;

	if((type == espeakEVENT_PHONEME) || (type == espeakEVENT_SENTENCE))
	{
		f_log=fopen("log-espeakedit","a");
		if(f_log)
		{
			if(type == espeakEVENT_PHONEME)
			{
				p = (int *)buf;
				p[0] = value;
				p[1] = value2;
				buf[8] = 0;
				fprintf(f_log,"Phoneme [%s]\n", buf);
			}
			else
				fprintf(f_log,"\n");
			fclose(f_log);
			f_log = NULL;
		}
	}
}
#endif

void WcmdqStop()
{//=============
	wcmdq_head = 0;
	wcmdq_tail = 0;

#ifdef INCLUDE_SONIC
	if(sonicSpeedupStream != NULL)
	{
		sonicDestroyStream(sonicSpeedupStream);
		sonicSpeedupStream = NULL;
	}
#endif

#ifdef USE_PORTAUDIO
	Pa_AbortStream(pa_stream);
#endif
	if(mbrola_name[0] != 0)
		MbrolaReset();
}


int WcmdqFree()
{//============
	int i;
	i = wcmdq_head - wcmdq_tail;
	if(i <= 0) i += N_WCMDQ;
	return(i);
}

int WcmdqUsed()
{//============
   return(N_WCMDQ - WcmdqFree());
}


void WcmdqInc()
{//============
	wcmdq_tail++;
	if(wcmdq_tail >= N_WCMDQ) wcmdq_tail=0;
}

static void WcmdqIncHead()
{//=======================
	wcmdq_head++;
	if(wcmdq_head >= N_WCMDQ) wcmdq_head=0;
}



// data points from which to make the presets for pk_shape1 and pk_shape2
#define PEAKSHAPEW 256
static const float pk_shape_x[2][8] = {
	{0,-0.6f, 0.0f, 0.6f, 1.4f, 2.5f, 4.5f, 5.5f},
	{0,-0.6f, 0.0f, 0.6f, 1.4f, 2.0f, 4.5f, 5.5f }};
static const float pk_shape_y[2][8] = {
	{0,  67,  81,  67,  31,  14,   0,  -6} ,
	{0,  77,  81,  77,  31,   7,   0,  -6 }};

unsigned char pk_shape1[PEAKSHAPEW+1] = {
   255,254,254,254,254,254,253,253,252,251,251,250,249,248,247,246,
   245,244,242,241,239,238,236,234,233,231,229,227,225,223,220,218,
   216,213,211,209,207,205,203,201,199,197,195,193,191,189,187,185,
   183,180,178,176,173,171,169,166,164,161,159,156,154,151,148,146,
   143,140,138,135,132,129,126,123,120,118,115,112,108,105,102, 99,
    96, 95, 93, 91, 90, 88, 86, 85, 83, 82, 80, 79, 77, 76, 74, 73,
    72, 70, 69, 68, 67, 66, 64, 63, 62, 61, 60, 59, 58, 57, 56, 55,
    55, 54, 53, 52, 52, 51, 50, 50, 49, 48, 48, 47, 47, 46, 46, 46,
    45, 45, 45, 44, 44, 44, 44, 44, 44, 44, 43, 43, 43, 43, 44, 43,
    42, 42, 41, 40, 40, 39, 38, 38, 37, 36, 36, 35, 35, 34, 33, 33,
    32, 32, 31, 30, 30, 29, 29, 28, 28, 27, 26, 26, 25, 25, 24, 24,
    23, 23, 22, 22, 21, 21, 20, 20, 19, 19, 18, 18, 18, 17, 17, 16,
    16, 15, 15, 15, 14, 14, 13, 13, 13, 12, 12, 11, 11, 11, 10, 10,
    10,  9,  9,  9,  8,  8,  8,  7,  7,  7,  7,  6,  6,  6,  5,  5,
     5,  5,  4,  4,  4,  4,  4,  3,  3,  3,  3,  2,  2,  2,  2,  2,
     2,  1,  1,  1,  1,  1,  1,  0,  0,  0,  0,  0,  0,  0,  0,  0,
     0 };

static unsigned char pk_shape2[PEAKSHAPEW+1] = {
   255,254,254,254,254,254,254,254,254,254,253,253,253,253,252,252,
   252,251,251,251,250,250,249,249,248,248,247,247,246,245,245,244,
   243,243,242,241,239,237,235,233,231,229,227,225,223,221,218,216,
   213,211,208,205,203,200,197,194,191,187,184,181,178,174,171,167,
   163,160,156,152,148,144,140,136,132,127,123,119,114,110,105,100,
    96, 94, 91, 88, 86, 83, 81, 78, 76, 74, 71, 69, 66, 64, 62, 60,
    57, 55, 53, 51, 49, 47, 44, 42, 40, 38, 36, 34, 32, 30, 29, 27,
    25, 23, 21, 19, 18, 16, 14, 12, 11,  9,  7,  6,  4,  3,  1,  0,
     0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,
     0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,
     0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,
     0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,
     0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,
     0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,
     0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,
     0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,
     0 };

static unsigned char *pk_shape;


static void WavegenInitPkData(int which)
{//=====================================
// this is only needed to set up the presets for pk_shape1 and pk_shape2
// These have already been pre-calculated and preset
#ifdef deleted
	int ix;
	int p;
	float x;
	float y[PEAKSHAPEW];
	float maxy=0;

	if(which==0)
		pk_shape = pk_shape1;
	else
		pk_shape = pk_shape2;

	p = 0;
	for(ix=0;ix<PEAKSHAPEW;ix++)
	{
		x = (4.5*ix)/PEAKSHAPEW;
		if(x >= pk_shape_x[which][p+3]) p++;
		y[ix] = polint(&pk_shape_x[which][p],&pk_shape_y[which][p],3,x);
		if(y[ix] > maxy) maxy = y[ix];
	}
	for(ix=0;ix<PEAKSHAPEW;ix++)
	{
		p = (int)(y[ix]*255/maxy);
      pk_shape[ix] = (p >= 0) ? p : 0;
	}
	pk_shape[PEAKSHAPEW]=0;
#endif
}  //  end of WavegenInitPkData



#ifdef USE_PORTAUDIO
// PortAudio interface

static int userdata[4];
static PaError pa_init_err=0;
static int out_channels=1;

unsigned char *outbuffer = NULL;
int outbuffer_size = 0;


#if USE_PORTAUDIO == 18
static int WaveCallback(void *inputBuffer, void *outputBuffer,
		unsigned long framesPerBuffer, PaTimestamp outTime, void *userData )
#else
static int WaveCallback(const void *inputBuffer, void *outputBuffer,
		long unsigned int framesPerBuffer, const PaStreamCallbackTimeInfo *outTime,
		PaStreamCallbackFlags flags, void *userData )
#endif
{
	int ix;
	int result;
	unsigned char *p;
	unsigned char *out_buf;
	unsigned char *out_end2;
	int pa_size;

	pa_size = framesPerBuffer*2;

	// make a buffer 3x size of the portaudio output
	ix = pa_size*3;
	if(ix > outbuffer_size)
	{
		outbuffer = (unsigned char *)realloc(outbuffer, ix);
		if(outbuffer == NULL)
		{
			fprintf(stderr, "espeak: out of memory\n");
		}
		outbuffer_size = ix;
		out_ptr = NULL;
	}
	if(out_ptr == NULL)
	{
		out_ptr = out_start = outbuffer;
		out_end = out_start + outbuffer_size;
	}
	out_end2 = &outbuffer[pa_size];  // top of data needed for the portaudio buffer

#ifdef LIBRARY
	event_list_ix = 0;
#endif

	result = WavegenFill(1);

	// copy from the outbut buffer into the portaudio buffer
	if(result && (out_ptr > out_end2))
	{
		result = 0;   // don't end yet, there is more data in the buffer than can fit in portaudio
	}

	while(out_ptr < out_end2)
		*out_ptr++ = 0;  // fill with zeros up to the size of the portaudio buffer

	memcpy(outputBuffer, outbuffer, pa_size);

	// move the remaining contents of the start of the output buffer
	for(p = out_end2; p < out_end; p++)
	{
		p[-pa_size] = p[0];
	}
	out_ptr -= pa_size;

#ifdef LIBRARY
	count_samples += framesPerBuffer;
	if(synth_callback)
	{
		// synchronous-playback mode, allow the calling process to abort the speech
		event_list[event_list_ix].type = espeakEVENT_LIST_TERMINATED; // indicates end of event list
		event_list[event_list_ix].user_data = 0;

		if(synth_callback(NULL,0,event_list) == 1)
		{
			SpeakNextClause(NULL,NULL,2);  // stop speaking
			result = 1;
		}
	}
#endif

#ifdef ARCH_BIG
	{
		// swap the order of bytes in each sound sample in the portaudio buffer
		int c;
		unsigned char *buf_end;
		out_buf = (unsigned char *)outputBuffer;
		buf_end = out_buf + framesPerBuffer*2;
		while(out_buf < buf_end)
		{
			c = out_buf[0];
			out_buf[0] = out_buf[1];
			out_buf[1] = c;
			out_buf += 2;
		}
	}
#endif

	if(out_channels == 2)
	{
		// sound output can only do stereo, not mono.  Duplicate each sound sample to
		// produce 2 channels.
		out_buf = (unsigned char *)outputBuffer;
		for(ix=framesPerBuffer-1; ix>=0; ix--)
		{
			p = &out_buf[ix*4];
			p[3] = p[1] = out_buf[ix*2 + 1];
			p[2] = p[0] = out_buf[ix*2];
		}
	}

#if USE_PORTAUDIO == 18
#ifdef PLATFORM_WINDOWS
	return(result);
#endif
	if(result != 0)
	{
		static int end_timer = 0;
		if(end_timer == 0)
			end_timer = 4;
		if(end_timer > 0)
		{
			end_timer--;
			if(end_timer == 0)
				return(1);
		}
	}
	return(0);
#else
	return(result);
#endif

}  //  end of WaveCallBack


#if USE_PORTAUDIO == 19
/* This is a fixed version of Pa_OpenDefaultStream() for use if the version in portaudio V19
   is broken */

static PaError Pa_OpenDefaultStream2( PaStream** stream,
                              int inputChannelCount,
                              int outputChannelCount,
                              PaSampleFormat sampleFormat,
                              double sampleRate,
                              unsigned long framesPerBuffer,
                              PaStreamCallback *streamCallback,
                              void *userData )
{
	PaError result;
	PaStreamParameters hostApiOutputParameters;

	if(option_device_number >= 0)
		hostApiOutputParameters.device = option_device_number;
	else
		hostApiOutputParameters.device = Pa_GetDefaultOutputDevice();

	if( hostApiOutputParameters.device == paNoDevice )
		return paDeviceUnavailable;

	hostApiOutputParameters.channelCount = outputChannelCount;
	hostApiOutputParameters.sampleFormat = sampleFormat;
	/* defaultHighOutputLatency is used below instead of
	   defaultLowOutputLatency because it is more important for the default
	   stream to work reliably than it is for it to work with the lowest
	   latency.
	*/
	hostApiOutputParameters.suggestedLatency =
	      Pa_GetDeviceInfo( hostApiOutputParameters.device )->defaultHighOutputLatency;
	hostApiOutputParameters.hostApiSpecificStreamInfo = NULL;

	result = Pa_OpenStream(
		stream, NULL, &hostApiOutputParameters, sampleRate, framesPerBuffer, paNoFlag, streamCallback, userData );

	return(result);
}
#endif


int WavegenOpenSound()
{//===================
	PaError err, err2;
	PaError active;

	if(option_waveout || option_quiet)
	{
		// writing to WAV file, not to portaudio
		return(0);
	}

#if USE_PORTAUDIO == 18
	active = Pa_StreamActive(pa_stream);
#else
	active = Pa_IsStreamActive(pa_stream);
#endif

	if(active == 1)
		return(0);
	if(active < 0)
	{
		out_channels = 1;

#if USE_PORTAUDIO == 18
		err2 = Pa_OpenDefaultStream(&pa_stream,0,1,paInt16,samplerate,512,N_WAV_BUF,WaveCallback,(void *)userdata);

		if(err2 == paInvalidChannelCount)
		{
			// failed to open with mono, try stereo
			out_channels=2;
			err2 = Pa_OpenDefaultStream(&pa_stream,0,2,paInt16,samplerate,512,N_WAV_BUF,WaveCallback,(void *)userdata);
		}
#else
		err2 = Pa_OpenDefaultStream2(&pa_stream,0,1,paInt16,(double)samplerate,512,WaveCallback,(void *)userdata);

		if(err2 == paInvalidChannelCount)
		{
			// failed to open with mono, try stereo
			out_channels=2;
			err2 = Pa_OpenDefaultStream(&pa_stream,0,2,paInt16,(double)samplerate,512,WaveCallback,(void *)userdata);
		}
#endif
	}
	err = Pa_StartStream(pa_stream);

#if USE_PORTAUDIO == 19
	if(err == paStreamIsNotStopped)
	{
		// not sure why we need this, but PA v19 seems to need it
		err = Pa_StopStream(pa_stream);
		err = Pa_StartStream(pa_stream);
	}
#endif

	if(err != paNoError)
	{
		// exit speak if we can't open the sound device - this is OK if speak is being run for each utterance
		exit(2);
	}

	return(0);
}



int WavegenCloseSound()
{//====================
	PaError active;

	// check whether speaking has finished, and close the stream
	if(pa_stream != NULL)
	{
#if USE_PORTAUDIO == 18
		active = Pa_StreamActive(pa_stream);
#else
		active = Pa_IsStreamActive(pa_stream);
#endif
		if(WcmdqUsed() == 0)   // also check that the queue is empty
		{
			if(active == 0)
			{
				Pa_CloseStream(pa_stream);
				pa_stream = NULL;
				return(1);
			}
		}
		else
		{
			WavegenOpenSound();  // still items in the queue, shouldn't be closed
		}
	}
	return(0);
}


int WavegenInitSound()
{//===================
	PaError err;

	if(option_quiet)
		return(0);

	// PortAudio sound output library
	err = Pa_Initialize();
	pa_init_err = err;
	if(err != paNoError)
	{
		fprintf(stderr,"Failed to initialise the PortAudio sound\n");
		return(1);
	}
	return(0);
}
#else
int WavegenOpenSound()
{//===================
	return(0);
}
int WavegenCloseSound()
{//====================
	return(0);
}
int WavegenInitSound()
{//===================
	return(0);
}
#endif


void WavegenInit(int rate, int wavemult_fact)
{//==========================================
	int  ix;
	double x;

	if(wavemult_fact == 0)
		wavemult_fact=60;  // default

	wvoice = NULL;
	samplerate = samplerate_native = rate;
	PHASE_INC_FACTOR = 0x8000000 / samplerate;   // assumes pitch is Hz*32
	Flutter_inc = (64 * samplerate)/rate;
	samplecount = 0;
	nsamples = 0;
	wavephase = 0x7fffffff;
	max_hval = 0;

	wdata.amplitude = 32;
	wdata.amplitude_fmt = 100;

	for(ix=0; ix<N_EMBEDDED_VALUES; ix++)
		embedded_value[ix] = embedded_default[ix];


	// set up window to generate a spread of harmonics from a
	// single peak for HF peaks
	wavemult_max = (samplerate * wavemult_fact)/(256 * 50);
	if(wavemult_max > N_WAVEMULT) wavemult_max = N_WAVEMULT;

	wavemult_offset = wavemult_max/2;

	if(samplerate != 22050)
	{
		// wavemult table has preset values for 22050 Hz, we only need to
		// recalculate them if we have a different sample rate
		for(ix=0; ix<wavemult_max; ix++)
		{
			x = 127*(1.0 - cos(PI2*ix/wavemult_max));
			wavemult[ix] = (int)x;
		}
	}

	WavegenInitPkData(1);
	WavegenInitPkData(0);
	pk_shape = pk_shape2;         // pk_shape2

#ifdef INCLUDE_KLATT
	KlattInit();
#endif

#ifdef LOG_FRAMES
remove("log-espeakedit");
remove("log-klatt");
#endif
}  // end of WavegenInit


int GetAmplitude(void)
{//===================
	int amp;

	// normal, none, reduced, moderate, strong
	static const unsigned char amp_emphasis[5] = {16, 16, 10, 16, 22};

	amp = (embedded_value[EMBED_A])*55/100;
	general_amplitude = amp * amp_emphasis[embedded_value[EMBED_F]] / 16;
	return(general_amplitude);
}


static void WavegenSetEcho(void)
{//=============================
	int delay;
	int amp;

	voicing = wvoice->voicing;
	delay = wvoice->echo_delay;
	amp = wvoice->echo_amp;

	if(delay >= N_ECHO_BUF)
		delay = N_ECHO_BUF-1;
	if(amp > 100)
		amp = 100;

	memset(echo_buf,0,sizeof(echo_buf));
	echo_tail = 0;

	if(embedded_value[EMBED_H] > 0)
	{
		// set echo from an embedded command in the text
		amp = embedded_value[EMBED_H];
		delay = 130;
	}

	if(delay == 0)
		amp = 0;

	echo_head = (delay * samplerate)/1000;
	echo_length = echo_head;       // ensure completion of echo at the end of speech. Use 1 delay period?
	if(amp == 0)
		echo_length = 0;
	if(amp > 20)
		echo_length = echo_head * 2;    // perhaps allow 2 echo periods if the echo is loud.

	// echo_amp units are 1/256ths of the amplitude of the original sound.
	echo_amp = amp;
	// compensate (partially) for increase in amplitude due to echo
	general_amplitude = GetAmplitude();
	general_amplitude = ((general_amplitude * (500-amp))/500);
}  // end of WavegenSetEcho



int PeaksToHarmspect(wavegen_peaks_t *peaks, int pitch, int *htab, int control)
{//============================================================================
// Calculate the amplitude of each  harmonics from the formants
// Only for formants 0 to 5

// control 0=initial call, 1=every 64 cycles

   // pitch and freqs are Hz<<16

	int f;
	wavegen_peaks_t *p;
	int fp;   // centre freq of peak
	int fhi;  // high freq of peak
	int h;    // harmonic number
	int pk;
	int hmax;
	int hmax_samplerate;      // highest harmonic allowed for the samplerate
	int x;
	int ix;
	int h1;

#ifdef SPECT_EDITOR
	if(harm_sqrt_n > 0)
		return(HarmToHarmspect(pitch,htab));
#endif

	// initialise as much of *out as we will need
	if(wvoice == NULL)
		return(1);
	hmax = (peaks[wvoice->n_harmonic_peaks].freq + peaks[wvoice->n_harmonic_peaks].right)/pitch;
	if(hmax >= MAX_HARMONIC)
		hmax = MAX_HARMONIC-1;

	// restrict highest harmonic to half the samplerate
	hmax_samplerate = (((samplerate * 19)/40) << 16)/pitch;   // only 95% of Nyquist freq
//	hmax_samplerate = (samplerate << 16)/(pitch*2);

	if(hmax > hmax_samplerate)
		hmax = hmax_samplerate;

	for(h=0;h<=hmax;h++)
		htab[h]=0;

	h=0;
	for(pk=0; pk<=wvoice->n_harmonic_peaks; pk++)
	{
		p = &peaks[pk];
		if((p->height == 0) || (fp = p->freq)==0)
			continue;

		fhi = p->freq + p->right;
		h = ((p->freq - p->left) / pitch) + 1;
		if(h <= 0) h = 1;

		for(f=pitch*h; f < fp; f+=pitch)
		{
			htab[h++] += pk_shape[(fp-f)/(p->left>>8)] * p->height;
		}
		for(; f < fhi; f+=pitch)
		{
			htab[h++] += pk_shape[(f-fp)/(p->right>>8)] * p->height;
		}
	}

{
int y;
int h2;
	// increase bass
	y = peaks[1].height * 10;   // addition as a multiple of 1/256s
	h2 = (1000<<16)/pitch;       // decrease until 1000Hz
	if(h2 > 0)
	{
		x = y/h2;
		h = 1;
		while(y > 0)
		{
			htab[h++] += y;
			y -= x;
		}
	}
}

	// find the nearest harmonic for HF peaks where we don't use shape
	for(; pk<N_PEAKS; pk++)
	{
		x = peaks[pk].height >> 14;
		peak_height[pk] = (x * x * 5)/2;

		// find the nearest harmonic for HF peaks where we don't use shape
		if(control == 0)
		{
			// set this initially, but make changes only at the quiet point
			peak_harmonic[pk] = peaks[pk].freq / pitch;
		}
		// only use harmonics up to half the samplerate
		if(peak_harmonic[pk] >= hmax_samplerate)
			peak_height[pk] = 0;
	}

	// convert from the square-rooted values
	f = 0;
	for(h=0; h<=hmax; h++, f+=pitch)
	{
		x = htab[h] >> 15;
		htab[h] = (x * x) >> 8;

		if((ix = (f >> 19)) < N_TONE_ADJUST)
		{
			htab[h] = (htab[h] * wvoice->tone_adjust[ix]) >> 13;  // index tone_adjust with Hz/8
		}
	}

	// adjust the amplitude of the first harmonic, affects tonal quality
	h1 = htab[1] * option_harmonic1;
	htab[1] = h1/8;


	// calc intermediate increments of LF harmonics
	if(control & 1)
	{
		for(h=1; h<N_LOWHARM; h++)
		{
			harm_inc[h] = (htab[h] - harmspect[h]) >> 3;
		}
	}

	return(hmax);  // highest harmonic number
}  // end of PeaksToHarmspect



static void AdvanceParameters()
{//============================
// Called every 64 samples to increment the formant freq, height, and widths

	int x;
	int ix;
	static int Flutter_ix = 0;

	// advance the pitch
	wdata.pitch_ix += wdata.pitch_inc;
	if((ix = wdata.pitch_ix>>8) > 127) ix = 127;
	x = wdata.pitch_env[ix] * wdata.pitch_range;
	wdata.pitch = (x>>8) + wdata.pitch_base;

	amp_ix += amp_inc;

	/* add pitch flutter */
	if(Flutter_ix >= (N_FLUTTER*64))
		Flutter_ix = 0;
	x = ((int)(Flutter_tab[Flutter_ix >> 6])-0x80) * flutter_amp;
	Flutter_ix += Flutter_inc;
	wdata.pitch += x;
	if(wdata.pitch < 102400)
		wdata.pitch = 102400;   // min pitch, 25 Hz  (25 << 12)

	if(samplecount == samplecount_start)
		return;

	for(ix=0; ix <= wvoice->n_harmonic_peaks; ix++)
	{
		peaks[ix].freq1 += peaks[ix].freq_inc;
		peaks[ix].freq = (int)peaks[ix].freq1;
		peaks[ix].height1 += peaks[ix].height_inc;
		if((peaks[ix].height = (int)peaks[ix].height1) < 0)
			peaks[ix].height = 0;
		peaks[ix].left1 += peaks[ix].left_inc;
		peaks[ix].left = (int)peaks[ix].left1;
		if(ix < 3)
		{
			peaks[ix].right1 += peaks[ix].right_inc;
			peaks[ix].right = (int)peaks[ix].right1;
		}
		else
		{
			peaks[ix].right = peaks[ix].left;
		}
	}
	for(;ix < 8; ix++)
	{
		// formants 6,7,8 don't have a width parameter
		if(ix < 7)
		{
			peaks[ix].freq1 += peaks[ix].freq_inc;
			peaks[ix].freq = (int)peaks[ix].freq1;
		}
		peaks[ix].height1 += peaks[ix].height_inc;
		if((peaks[ix].height = (int)peaks[ix].height1) < 0)
			peaks[ix].height = 0;
	}

#ifdef SPECT_EDITOR
	if(harm_sqrt_n != 0)
	{
		// We are generating from a harmonic spectrum at a given pitch, not from formant peaks
		for(ix=0; ix<harm_sqrt_n; ix++)
			harm_sqrt[ix] += harm_sqrt_inc[ix];
	}
#endif
}  //  end of AdvanceParameters


#ifndef PLATFORM_RISCOS
static double resonator(RESONATOR *r, double input)
{//================================================
	double x;

	x = r->a * input + r->b * r->x1 + r->c * r->x2;
	r->x2 = r->x1;
	r->x1 = x;

 return x;
}



static void setresonator(RESONATOR *rp, int freq, int bwidth, int init)
{//====================================================================
// freq    Frequency of resonator in Hz
// bwidth  Bandwidth of resonator in Hz
// init    Initialize internal data

	double x;
	double arg;

	if(init)
	{
		rp->x1 = 0;
		rp->x2 = 0;
	}

   // x  =  exp(-pi * bwidth * t)
	arg = minus_pi_t * bwidth;
	x = exp(arg);

	// c  =  -(x*x)
	rp->c = -(x * x);

	// b = x * 2*cos(2 pi * freq * t)

	arg = two_pi_t * freq;
	rp->b = x * cos(arg) * 2.0;

	// a = 1.0 - b - c
	rp->a = 1.0 - rp->b - rp->c;
}  // end if setresonator
#endif


void InitBreath(void)
{//==================
#ifndef PLATFORM_RISCOS
	int ix;

	minus_pi_t = -PI / samplerate;
	two_pi_t = -2.0 * minus_pi_t;

	for(ix=0; ix<N_PEAKS; ix++)
	{
		setresonator(&rbreath[ix],2000,200,1);
	}
#endif
}  // end of InitBreath



static void SetBreath()
{//====================
#ifndef PLATFORM_RISCOS
	int pk;

	if(wvoice->breath[0] == 0)
		return;

	for(pk=1; pk<N_PEAKS; pk++)
	{
		if(wvoice->breath[pk] != 0)
		{
			// breath[0] indicates that some breath formants are needed
			// set the freq from the current ynthesis formant and the width from the voice data
			setresonator(&rbreath[pk], peaks[pk].freq >> 16, wvoice->breathw[pk],0);
		}
	}
#endif
}  // end of SetBreath


static int ApplyBreath(void)
{//=========================
	int value = 0;
#ifndef PLATFORM_RISCOS
	int noise;
	int ix;
	int amp;

	// use two random numbers, for alternate formants
	noise = (rand() & 0x3fff) - 0x2000;

	for(ix=1; ix < N_PEAKS; ix++)
	{
		if((amp = wvoice->breath[ix]) != 0)
		{
			amp *= (peaks[ix].height >> 14);
			value += (int)resonator(&rbreath[ix],noise) * amp;
		}
	}
#endif
	return (value);
}



int Wavegen()
{//==========
	unsigned short waveph;
	unsigned short theta;
	int total;
	int h;
	int ix;
	int z, z1, z2;
	int echo;
	int ov;
	static int maxh, maxh2;
	int pk;
	signed char c;
	int sample;
	int amp;
	int modn_amp, modn_period;
	static int agc = 256;
	static int h_switch_sign = 0;
	static int cycle_count = 0;
	static int amplitude2 = 0;   // adjusted for pitch

	// continue until the output buffer is full, or
	// the required number of samples have been produced

	for(;;)
	{
		if((end_wave==0) && (samplecount==nsamples))
			return(0);

		if((samplecount & 0x3f) == 0)
		{
			// every 64 samples, adjust the parameters
			if(samplecount == 0)
			{
				hswitch = 0;
				harmspect = hspect[0];
				maxh2 = PeaksToHarmspect(peaks, wdata.pitch<<4, hspect[0], 0);

				// adjust amplitude to compensate for fewer harmonics at higher pitch
//				amplitude2 = (wdata.amplitude * wdata.pitch)/(100 << 11);
				amplitude2 = (wdata.amplitude * (wdata.pitch >> 8) * wdata.amplitude_fmt)/(10000 << 3);

            // switch sign of harmonics above about 900Hz, to reduce max peak amplitude
				h_switch_sign = 890 / (wdata.pitch >> 12);
			}
			else
				AdvanceParameters();

			// pitch is Hz<<12
			phaseinc = (wdata.pitch>>7) * PHASE_INC_FACTOR;
			cycle_samples = samplerate/(wdata.pitch >> 12);  // sr/(pitch*2)
			hf_factor = wdata.pitch >> 11;

			maxh = maxh2;
			harmspect = hspect[hswitch];
			hswitch ^= 1;
			maxh2 = PeaksToHarmspect(peaks, wdata.pitch<<4, hspect[hswitch], 1);

			SetBreath();
		}
		else
		if((samplecount & 0x07) == 0)
		{
			for(h=1; h<N_LOWHARM && h<=maxh2 && h<=maxh; h++)
			{
				harmspect[h] += harm_inc[h];
			}

			// bring automctic gain control back towards unity
			if(agc < 256) agc++;
		}

		samplecount++;

		if(wavephase > 0)
		{
			wavephase += phaseinc;
			if(wavephase < 0)
			{
				// sign has changed, reached a quiet point in the waveform
				cbytes = wavemult_offset - (cycle_samples)/2;
				if(samplecount > nsamples)
					return(0);

				cycle_count++;

				for(pk=wvoice->n_harmonic_peaks+1; pk<N_PEAKS; pk++)
				{
					// find the nearest harmonic for HF peaks where we don't use shape
					peak_harmonic[pk] = ((peaks[pk].freq / (wdata.pitch*8)) + 1) / 2;
				}

				// adjust amplitude to compensate for fewer harmonics at higher pitch
//				amplitude2 = (wdata.amplitude * wdata.pitch)/(100 << 11);
				amplitude2 = (wdata.amplitude * (wdata.pitch >> 8) * wdata.amplitude_fmt)/(10000 << 3);

				if(glottal_flag > 0)
				{
					if(glottal_flag == 3)
					{
						if((nsamples-samplecount) < (cycle_samples*2))
						{
							// Vowel before glottal-stop.
							// This is the start of the penultimate cycle, reduce its amplitude
							glottal_flag = 2;
							amplitude2 = (amplitude2 *  glottal_reduce)/256;
						}
					}
					else
					if(glottal_flag == 4)
					{
						// Vowel following a glottal-stop.
						// This is the start of the second cycle, reduce its amplitude
						glottal_flag = 2;
						amplitude2 = (amplitude2 * glottal_reduce)/256;
					}
					else
					{
						glottal_flag--;
					}
				}

				if(amplitude_env != NULL)
				{
					// amplitude envelope is only used for creaky voice effect on certain vowels/tones
					if((ix = amp_ix>>8) > 127) ix = 127;
					amp = amplitude_env[ix];
					amplitude2 = (amplitude2 * amp)/128;
//					if(amp < 255)
//						modulation_type = 7;
				}

				// introduce roughness into the sound by reducing the amplitude of
				modn_period = 0;
				if(voice->roughness < N_ROUGHNESS)
				{
					modn_period = modulation_tab[voice->roughness][modulation_type];
					modn_amp = modn_period & 0xf;
					modn_period = modn_period >> 4;
				}

				if(modn_period != 0)
				{
					if(modn_period==0xf)
					{
						// just once */
						amplitude2 = (amplitude2 * modn_amp)/16;
						modulation_type = 0;
					}
					else
					{
						// reduce amplitude every [modn_period} cycles
						if((cycle_count % modn_period)==0)
							amplitude2 = (amplitude2 * modn_amp)/16;
					}
				}
			}
		}
		else
		{
			wavephase += phaseinc;
		}
		waveph = (unsigned short)(wavephase >> 16);
		total = 0;

		// apply HF peaks, formants 6,7,8
		// add a single harmonic and then spread this my multiplying by a
		// window.  This is to reduce the processing power needed to add the
		// higher frequence harmonics.
		cbytes++;
		if(cbytes >=0 && cbytes<wavemult_max)
		{
			for(pk=wvoice->n_harmonic_peaks+1; pk<N_PEAKS; pk++)
			{
				theta = peak_harmonic[pk] * waveph;
				total += (long)sin_tab[theta >> 5] * peak_height[pk];
			}

			// spread the peaks by multiplying by a window
			total = (long)(total / hf_factor) * wavemult[cbytes];
		}

		// apply main peaks, formants 0 to 5
#ifdef USE_ASSEMBLER_1
		// use an optimised routine for this loop, if available
		total += AddSineWaves(waveph, h_switch_sign, maxh, harmspect);  // call an assembler code routine
#else
		theta = waveph;

		for(h=1; h<=h_switch_sign; h++)
		{
			total += ((int)sin_tab[theta >> 5] * harmspect[h]);
			theta += waveph;
		}
		while(h<=maxh)
		{
			total -= ((int)sin_tab[theta >> 5] * harmspect[h]);
			theta += waveph;
			h++;
		}
#endif

		if(voicing != 64)
		{
			total = (total >> 6) * voicing;
		}

#ifndef PLATFORM_RISCOS
		if(wvoice->breath[0])
		{
			total +=  ApplyBreath();
		}
#endif

		// mix with sampled wave if required
		z2 = 0;
		if(wdata.mix_wavefile_ix < wdata.n_mix_wavefile)
		{
			if(wdata.mix_wave_scale == 0)
			{
				// a 16 bit sample
				c = wdata.mix_wavefile[wdata.mix_wavefile_ix+wdata.mix_wavefile_offset+1];
				sample = wdata.mix_wavefile[wdata.mix_wavefile_ix+wdata.mix_wavefile_offset] + (c * 256);
				wdata.mix_wavefile_ix += 2;
			}
			else
			{
				// a 8 bit sample, scaled
				sample = (signed char)wdata.mix_wavefile[wdata.mix_wavefile_offset+wdata.mix_wavefile_ix++] * wdata.mix_wave_scale;
			}
			z2 = (sample * wdata.amplitude_v) >> 10;
			z2 = (z2 * wdata.mix_wave_amp)/32;

			if((wdata.mix_wavefile_ix + wdata.mix_wavefile_offset) >= wdata.mix_wavefile_max)  // reached the end of available WAV data
				wdata.mix_wavefile_offset -= (wdata.mix_wavefile_max*3)/4;
		}

		z1 = z2 + (((total>>8) * amplitude2) >> 13);

		echo = (echo_buf[echo_tail++] * echo_amp);
		z1 += echo >> 8;
		if(echo_tail >= N_ECHO_BUF)
			echo_tail=0;

		z = (z1 * agc) >> 8;

		// check for overflow, 16bit signed samples
		if(z >= 32768)
		{
			ov = 8388608/z1 - 1;      // 8388608 is 2^23, i.e. max value * 256
			if(ov < agc) agc = ov;    // set agc to number of 1/256ths to multiply the sample by
			z = (z1 * agc) >> 8;      // reduce sample by agc value to prevent overflow
		}
		else
		if(z <= -32768)
		{
			ov = -8388608/z1 - 1;
			if(ov < agc) agc = ov;
			z = (z1 * agc) >> 8;
		}
		*out_ptr++ = z;
		*out_ptr++ = z >> 8;

		echo_buf[echo_head++] = z;
		if(echo_head >= N_ECHO_BUF)
			echo_head = 0;

		if(out_ptr >= out_end)
			return(1);
	}
	return(0);
}  //  end of Wavegen


static int PlaySilence(int length, int resume)
{//===========================================
	static int n_samples;
	int value=0;

	nsamples = 0;
	samplecount = 0;
	wavephase = 0x7fffffff;

	if(length == 0)
		return(0);

	if(resume==0)
		n_samples = length;

	while(n_samples-- > 0)
	{
		value = (echo_buf[echo_tail++] * echo_amp) >> 8;

		if(echo_tail >= N_ECHO_BUF)
			echo_tail = 0;

		*out_ptr++ = value;
		*out_ptr++ = value >> 8;

		echo_buf[echo_head++] = value;
		if(echo_head >= N_ECHO_BUF)
			echo_head = 0;

		if(out_ptr >= out_end)
			return(1);
	}
	return(0);
}  // end of PlaySilence



static int PlayWave(int length, int resume, unsigned char *data, int scale, int amp)
{//=================================================================================
	static int n_samples;
	static int ix=0;
	int value;
	signed char c;

	if(resume==0)
	{
		n_samples = length;
		ix = 0;
	}

	nsamples = 0;
	samplecount = 0;

	while(n_samples-- > 0)
	{
		if(scale == 0)
		{
			// 16 bits data
			c = data[ix+1];
			value = data[ix] + (c * 256);
			ix+=2;
		}
		else
		{
			// 8 bit data, shift by the specified scale factor
			value = (signed char)data[ix++] * scale;
		}
		value *= (consonant_amp * general_amplitude);   // reduce strength of consonant
		value = value >> 10;
		value = (value * amp)/32;

		value += ((echo_buf[echo_tail++] * echo_amp) >> 8);

		if(value > 32767)
			value = 32768;
		else
		if(value < -32768)
			value = -32768;

		if(echo_tail >= N_ECHO_BUF)
			echo_tail = 0;

		out_ptr[0] = value;
		out_ptr[1] = value >> 8;
		out_ptr+=2;

		echo_buf[echo_head++] = (value*3)/4;
		if(echo_head >= N_ECHO_BUF)
			echo_head = 0;

		if(out_ptr >= out_end)
			return(1);
	}
	return(0);
}


static int SetWithRange0(int value, int max)
{//=========================================
	if(value < 0)
		return(0);
	if(value > max)
		return(max);
	return(value);
}


static void SetPitchFormants()
{//===========================
	int ix;
	int factor = 256;
	int pitch_value;

	// adjust formants to give better results for a different voice pitch
	if((pitch_value = embedded_value[EMBED_P]) > MAX_PITCH_VALUE)
		pitch_value = MAX_PITCH_VALUE;

	if(pitch_value > 50)
	{
		// only adjust if the pitch is higher than normal
		factor = 256 + (25 * (pitch_value - 50))/50;
	}

	for(ix=0; ix<=5; ix++)
	{
		wvoice->freq[ix] = (wvoice->freq2[ix] * factor)/256;
	}

	factor = embedded_value[EMBED_T]*3;
	wvoice->height[0] = (wvoice->height2[0] * (256 - factor*2))/256;
	wvoice->height[1] = (wvoice->height2[1] * (256 - factor))/256;
}


void SetEmbedded(int control, int value)
{//=====================================
	// there was an embedded command in the text at this point
	int sign=0;
	int command;

	command = control & 0x1f;
	if((control & 0x60) == 0x60)
		sign = -1;
	else
	if((control & 0x60) == 0x40)
		sign = 1;

	if(command < N_EMBEDDED_VALUES)
	{
		if(sign == 0)
			embedded_value[command] = value;
		else
			embedded_value[command] += (value * sign);
		embedded_value[command] = SetWithRange0(embedded_value[command],embedded_max[command]);
	}

	switch(command)
	{
	case EMBED_T:
		WavegenSetEcho();   // and drop through to case P
	case EMBED_P:
		SetPitchFormants();
		break;

	case EMBED_A:  // amplitude
		general_amplitude = GetAmplitude();
		break;

	case EMBED_F:   // emphasis
		general_amplitude = GetAmplitude();
		break;

	case EMBED_H:
		WavegenSetEcho();
		break;
	}
}


void WavegenSetVoice(voice_t *v)
{//=============================
	static voice_t v2;

	memcpy(&v2,v,sizeof(v2));
	wvoice = &v2;

	if(v->peak_shape==0)
		pk_shape = pk_shape1;
	else
		pk_shape = pk_shape2;

	consonant_amp = (v->consonant_amp * 26) /100;
	if(samplerate <= 11000)
	{
		consonant_amp = consonant_amp*2;  // emphasize consonants at low sample rates
		option_harmonic1 = 6;
	}
	WavegenSetEcho();
	SetPitchFormants();
	MarkerEvent(espeakEVENT_SAMPLERATE, 0, wvoice->samplerate, 0, out_ptr);
//	WVoiceChanged(wvoice);
}


static void SetAmplitude(int length, unsigned char *amp_env, int value)
{//====================================================================
	amp_ix = 0;
	if(length==0)
		amp_inc = 0;
	else
		amp_inc = (256 * ENV_LEN * STEPSIZE)/length;

	wdata.amplitude = (value * general_amplitude)/16;
	wdata.amplitude_v = (wdata.amplitude * wvoice->consonant_ampv * 15)/100;           // for wave mixed with voiced sounds

	amplitude_env = amp_env;
}


void SetPitch2(voice_t *voice, int pitch1, int pitch2, int *pitch_base, int *pitch_range)
{//======================================================================================
	int x;
	int base;
	int range;
	int pitch_value;

	if(pitch1 > pitch2)
	{
		x = pitch1;   // swap values
		pitch1 = pitch2;
		pitch2 = x;
	}

	if((pitch_value = embedded_value[EMBED_P]) > MAX_PITCH_VALUE)
		pitch_value = MAX_PITCH_VALUE;
	pitch_value -= embedded_value[EMBED_T];   // adjust tone for announcing punctuation
	if(pitch_value < 0)
		pitch_value = 0;

	base = (voice->pitch_base * pitch_adjust_tab[pitch_value])/128;
	range =  (voice->pitch_range * embedded_value[EMBED_R])/50;

	// compensate for change in pitch when the range is narrowed or widened
	base -= (range - voice->pitch_range)*18;

	*pitch_base = base + (pitch1 * range)/2;
	*pitch_range = base + (pitch2 * range)/2 - *pitch_base;
}


void SetPitch(int length, unsigned char *env, int pitch1, int pitch2)
{//==================================================================
// length in samples

#ifdef LOG_FRAMES
if(option_log_frames)
{
	f_log=fopen("log-espeakedit","a");
	if(f_log != NULL)
	{
		fprintf(f_log,"	  pitch %3d %3d  %3dmS\n",pitch1,pitch2,(length*1000)/samplerate);
		fclose(f_log);
		f_log=NULL;
	}
}
#endif
	if((wdata.pitch_env = env)==NULL)
		wdata.pitch_env = env_fall;  // default

	wdata.pitch_ix = 0;
	if(length==0)
		wdata.pitch_inc = 0;
	else
		wdata.pitch_inc = (256 * ENV_LEN * STEPSIZE)/length;

	SetPitch2(wvoice, pitch1, pitch2, &wdata.pitch_base, &wdata.pitch_range);
	// set initial pitch
	wdata.pitch = ((wdata.pitch_env[0] * wdata.pitch_range) >>8) + wdata.pitch_base;   // Hz << 12

	flutter_amp = wvoice->flutter;

}  // end of SetPitch





void SetSynth(int length, int modn, frame_t *fr1, frame_t *fr2, voice_t *v)
{//========================================================================
	int ix;
	DOUBLEX next;
	int length2;
	int length4;
	int qix;
	int cmd;
	static int glottal_reduce_tab1[4] = {0x30, 0x30, 0x40, 0x50};  // vowel before [?], amp * 1/256
//	static int glottal_reduce_tab1[4] = {0x30, 0x40, 0x50, 0x60};  // vowel before [?], amp * 1/256
	static int glottal_reduce_tab2[4] = {0x90, 0xa0, 0xb0, 0xc0};  // vowel after [?], amp * 1/256

#ifdef LOG_FRAMES
if(option_log_frames)
{
	f_log=fopen("log-espeakedit","a");
	if(f_log != NULL)
	{
		fprintf(f_log,"%3dmS  %3d %3d %4d %4d (%3d %3d %3d %3d)  to  %3d %3d %4d %4d (%3d %3d %3d %3d)\n",length*1000/samplerate,
			fr1->ffreq[0],fr1->ffreq[1],fr1->ffreq[2],fr1->ffreq[3], fr1->fheight[0],fr1->fheight[1],fr1->fheight[2],fr1->fheight[3],
			fr2->ffreq[0],fr2->ffreq[1],fr2->ffreq[2],fr2->ffreq[3], fr2->fheight[0],fr2->fheight[1],fr2->fheight[2],fr2->fheight[3] );

	fclose(f_log);
	f_log=NULL;
	}
}
#endif

	harm_sqrt_n = 0;
	end_wave = 1;

	// any additional information in the param1 ?
	modulation_type = modn & 0xff;

	glottal_flag = 0;
	if(modn & 0x400)
	{
		glottal_flag = 3;  // before a glottal stop
		glottal_reduce = glottal_reduce_tab1[(modn >> 8) & 3];
	}
	if(modn & 0x800)
	{
		glottal_flag = 4;  // after a glottal stop
		glottal_reduce = glottal_reduce_tab2[(modn >> 8) & 3];
	}

	for(qix=wcmdq_head+1;;qix++)
	{
		if(qix >= N_WCMDQ) qix = 0;
		if(qix == wcmdq_tail) break;

		cmd = wcmdq[qix][0];
		if(cmd==WCMD_SPECT)
		{
			end_wave = 0;  // next wave generation is from another spectrum
			break;
		}
		if((cmd==WCMD_WAVE) || (cmd==WCMD_PAUSE))
			break;   // next is not from spectrum, so continue until end of wave cycle
	}

	// round the length to a multiple of the stepsize
	length2 = (length + STEPSIZE/2) & ~0x3f;
	if(length2 == 0)
		length2 = STEPSIZE;

	// add this length to any left over from the previous synth
	samplecount_start = samplecount;
	nsamples += length2;

	length4 = length2/4;

	peaks[7].freq = (7800  * v->freq[7] + v->freqadd[7]*256) << 8;
	peaks[8].freq = (9000  * v->freq[8] + v->freqadd[8]*256) << 8;

	for(ix=0; ix < 8; ix++)
	{
		if(ix < 7)
		{
			peaks[ix].freq1 = (fr1->ffreq[ix] * v->freq[ix] + v->freqadd[ix]*256) << 8;
			peaks[ix].freq = (int)peaks[ix].freq1;
			next = (fr2->ffreq[ix] * v->freq[ix] + v->freqadd[ix]*256) << 8;
			peaks[ix].freq_inc =  ((next - peaks[ix].freq1) * (STEPSIZE/4)) / length4;  // lower headroom for fixed point math
		}

		peaks[ix].height1 = (fr1->fheight[ix] * v->height[ix]) << 6;
		peaks[ix].height = (int)peaks[ix].height1;
		next = (fr2->fheight[ix] * v->height[ix]) << 6;
		peaks[ix].height_inc =  ((next - peaks[ix].height1) * STEPSIZE) / length2;

		if((ix <= 5) && (ix <= wvoice->n_harmonic_peaks))
		{
			peaks[ix].left1 = (fr1->fwidth[ix] * v->width[ix]) << 10;
			peaks[ix].left = (int)peaks[ix].left1;
			next = (fr2->fwidth[ix] * v->width[ix]) << 10;
			peaks[ix].left_inc =  ((next - peaks[ix].left1) * STEPSIZE) / length2;

			if(ix < 3)
			{
				peaks[ix].right1 = (fr1->fright[ix] * v->width[ix]) << 10;
				peaks[ix].right = (int)peaks[ix].right1;
				next = (fr2->fright[ix] * v->width[ix]) << 10;
				peaks[ix].right_inc = ((next - peaks[ix].right1) * STEPSIZE) / length2;
			}
			else
			{
				peaks[ix].right = peaks[ix].left;
			}
		}
	}
}  // end of SetSynth


static int Wavegen2(int length, int modulation, int resume, frame_t *fr1, frame_t *fr2)
{//====================================================================================
	if(resume==0)
		SetSynth(length, modulation, fr1, fr2, wvoice);

	return(Wavegen());
}

void Write4Bytes(FILE *f, int value)
{//=================================
// Write 4 bytes to a file, least significant first
	int ix;

	for(ix=0; ix<4; ix++)
	{
		fputc(value & 0xff,f);
		value = value >> 8;
	}
}



int WavegenFill2(int fill_zeros)
{//============================
// Pick up next wavegen commands from the queue
// return: 0  output buffer has been filled
// return: 1  input command queue is now empty

	long64 *q;
	int length;
	int result;
	int marker_type;
	static int resume=0;
	static int echo_complete=0;

	while(out_ptr < out_end)
	{
		if(WcmdqUsed() <= 0)
		{
			if(echo_complete > 0)
			{
				// continue to play silence until echo is completed
				resume = PlaySilence(echo_complete,resume);
				if(resume == 1)
					return(0);  // not yet finished
			}

			if(fill_zeros)
			{
				while(out_ptr < out_end)
					*out_ptr++ = 0;
			}
			return(1);              // queue empty, close sound channel
		}

		result = 0;
		q = wcmdq[wcmdq_head];
		length = q[1];

		switch(q[0] & 0xff)
		{
		case WCMD_PITCH:
			SetPitch(length,(unsigned char *)q[2],q[3] >> 16,q[3] & 0xffff);
			break;

		case WCMD_PAUSE:
			if(resume==0)
			{
				echo_complete -= length;
			}
			wdata.n_mix_wavefile = 0;
			wdata.amplitude_fmt = 100;
#ifdef INCLUDE_KLATT
			KlattReset(1);
#endif
			result = PlaySilence(length,resume);
			break;

		case WCMD_WAVE:
			echo_complete = echo_length;
			wdata.n_mix_wavefile = 0;
#ifdef INCLUDE_KLATT
			KlattReset(1);
#endif
			result = PlayWave(length,resume,(unsigned char*)q[2], q[3] & 0xff, q[3] >> 8);
			break;

		case WCMD_WAVE2:
			// wave file to be played at the same time as synthesis
			wdata.mix_wave_amp = q[3] >> 8;
			wdata.mix_wave_scale = q[3] & 0xff;
			wdata.n_mix_wavefile = (length & 0xffff);
			wdata.mix_wavefile_max = (length >> 16) & 0xffff;
			if(wdata.mix_wave_scale == 0)
			{
				wdata.n_mix_wavefile *= 2;
				wdata.mix_wavefile_max *= 2;
			}
			wdata.mix_wavefile_ix = 0;
			wdata.mix_wavefile_offset = 0;
			wdata.mix_wavefile = (unsigned char *)q[2];
			break;

		case WCMD_SPECT2:   // as WCMD_SPECT but stop any concurrent wave file
			wdata.n_mix_wavefile = 0;   // ... and drop through to WCMD_SPECT case
		case WCMD_SPECT:
			echo_complete = echo_length;
			result = Wavegen2(length & 0xffff,q[1] >> 16,resume,(frame_t *)q[2],(frame_t *)q[3]);
			break;

#ifdef INCLUDE_KLATT
		case WCMD_KLATT2:   // as WCMD_SPECT but stop any concurrent wave file
			wdata.n_mix_wavefile = 0;   // ... and drop through to WCMD_SPECT case
		case WCMD_KLATT:
			echo_complete = echo_length;
			result = Wavegen_Klatt2(length & 0xffff,q[1] >> 16,resume,(frame_t *)q[2],(frame_t *)q[3]);
			break;
#endif

		case WCMD_MARKER:
			marker_type = q[0] >> 8;
			MarkerEvent(marker_type, q[1],q[2],q[3],out_ptr);
#ifdef LOG_FRAMES
			LogMarker(marker_type, q[2], q[3]);
#endif
			if(marker_type == 1)  // word marker
			{
				current_source_index = q[1] & 0xffffff;
			}
			break;

		case WCMD_AMPLITUDE:
			SetAmplitude(length,(unsigned char *)q[2],q[3]);
			break;

		case WCMD_VOICE:
			WavegenSetVoice((voice_t *)q[2]);
			free((voice_t *)q[2]);
			break;

		case WCMD_EMBEDDED:
			SetEmbedded(q[1],q[2]);
			break;

		case WCMD_MBROLA_DATA:
			result = MbrolaFill(length, resume, (general_amplitude * wvoice->voicing)/64);
			break;

		case WCMD_FMT_AMPLITUDE:
			if((wdata.amplitude_fmt = q[1]) == 0)
				wdata.amplitude_fmt = 100;  // percentage, but value=0 means 100%
			break;

#ifdef INCLUDE_SONIC
		case WCMD_SONIC_SPEED:
			sonicSpeed = (double)q[1] / 1024;
			break;
#endif
		}

		if(result==0)
		{
			WcmdqIncHead();
			resume=0;
		}
		else
		{
			resume=1;
		}
	}

	return(0);
}  // end of WavegenFill2


#ifdef INCLUDE_SONIC
/* Speed up the audio samples with libsonic. */
static int SpeedUp(short *outbuf, int length_in, int length_out, int end_of_text)
{//==============================================================================
	if(length_in >0)
	{
		if(sonicSpeedupStream == NULL)
		{
			sonicSpeedupStream = sonicCreateStream(22050, 1);
		}
		if(sonicGetSpeed(sonicSpeedupStream) != sonicSpeed)
		{
		        sonicSetSpeed(sonicSpeedupStream, sonicSpeed);
		}

		sonicWriteShortToStream(sonicSpeedupStream, outbuf, length_in);
	}

	if(sonicSpeedupStream == NULL)
		return(0);

	if(end_of_text)
	{
		sonicFlushStream(sonicSpeedupStream);
	}
	return sonicReadShortFromStream(sonicSpeedupStream, outbuf, length_out);
}  // end of SpeedUp
#endif


/* Call WavegenFill2, and then speed up the output samples. */
int WavegenFill(int fill_zeros)
{//============================
	int finished;
	unsigned char *p_start;

	p_start = out_ptr;

	// fill_zeros is ignored. It is now done in the portaudio callback
	finished = WavegenFill2(0);

#ifdef INCLUDE_SONIC
	if(sonicSpeed > 1.0)
	{
		int length;
		int max_length;

		max_length = (out_end - p_start);
		length =  2*SpeedUp((short *)p_start, (out_ptr-p_start)/2, max_length/2, finished);
		out_ptr = p_start + length;

		if(length >= max_length)
			finished = 0;   // there may be more data to flush
	}
#endif
	return finished;
}  // end of WavegenFill