/*
    MACLIB:  A companion library to SDL for working with Macintosh (tm) data
    Copyright (C) 1997-2021 Sam Lantinga <slouken@libsdl.org>

    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 2 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, write to the Free Software
    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
*/

/* Microsoft WAVE file loading routines */

#include <stdlib.h>
#include <string.h>
#include "SDL_endian.h"
#include "SDL_rwops.h"
#include "Mac_Wave.h"

#if SDL_BYTEORDER == SDL_BIG_ENDIAN
#define wavesex32(x)	\
        ((x << 24) | ((x & 0xff00) << 8) | ((x >> 8) & 0xff00) | (x >> 24))
#define wavesex16(x)	((x << 8) | (x >> 8))
#define snd_sex32(x)	(x)
#define snd_sex16(x)	(x)
#else
#define wavesex32(x)	(x)
#define wavesex16(x)	(x)
#define snd_sex32(x)	\
        ((x << 24) | ((x & 0xff00) << 8) | ((x >> 8) & 0xff00) | (x >> 24))
#define snd_sex16(x)	((x << 8) | (x >> 8))
#endif /* Endianness */

/*******************************************/
/* Define values for Macintosh SND format */
/*******************************************/

/* Different sound header formats */
#define FORMAT_1	0x0001
#define FORMAT_2	0x0002

/* The different types of sound data */
#define SAMPLED_SND	0x0005

/* Initialization commands */
#define MONO_SOUND	0x00000080
#define STEREO_SOUND	0x000000A0

/* The different sound commands; we only support BUFFER_CMD */
#define SOUND_CMD	0x8050		/* Different from BUFFER_CMD? */
#define BUFFER_CMD	0x8051

/* Different original sampling rates -- rate = (#define)>>16 */
#define rate44khz	0xAC440000		/* 44100.0 */
#define rate22khz	0x56EE8BA3		/* 22254.5 */
#define rate11khz	0x2B7745D0		/* 11127.3 */
#define rate11khz2	0x2B7745D1		/* 11127.3 (?) */

#define stdSH		0x00
#define extSH		0xFF
#define cmpSH		0xFE

/*******************************************/

#define snd_copy16(V, D)						\
{									\
	V = *((Uint16 *)D);						\
	D += 2;								\
	V = snd_sex16(V);						\
}
#define snd_copy32(V, D)						\
{									\
	memcpy(&V, D, sizeof(Uint32));					\
	D += 4;								\
	V = snd_sex32(V);						\
}

/*******************************************/

void
Wave:: Init(void) {
	sound_data = NULL;
	errstr = NULL;
}
void
Wave:: Free(void) {
	if ( sound_data ) {
		delete[] sound_data;
		sound_data = NULL;
	}
}

int
Wave:: Load(const char *wavefile, Uint16 desired_rate)
{
	Uint8 *samples;

	/* Free any existing WAVE data */
	Free();
	Init();

	/* Load the WAVE file */
	if ( SDL_LoadWAV(wavefile, &spec, &samples, &sound_datalen) == NULL ) {
		error("%s", SDL_GetError());
		return(-1);
	}
	/* Copy malloc()'d data to new'd data */
	sound_data = new Uint8[sound_datalen];
	memcpy(sound_data, samples, sound_datalen);
	SDL_FreeWAV(samples);

	/* Set the desired sample frequency */
	Frequency(desired_rate);

	/* Rewind and go! */
	Rewind();
	return(0);
}

Uint32
Wave:: Frequency(Uint16 desired_rate)
{
	if ( (desired_rate > 0) && (desired_rate != spec.freq) ) {
		Uint8 *samples;
		Uint32 datalen, samplesize;

		samples = sound_data;
		samplesize = SampleSize();
		datalen = ConvertRate(spec.freq, desired_rate,
				&samples, sound_datalen/samplesize, samplesize);
		if ( samples != sound_data ) {
			/* Create new sound data */
			delete[] sound_data; 
			sound_data = samples;
			sound_datalen = datalen*samplesize;

			/* Adjust the format */
			spec.freq = desired_rate;
		}
	}
	return(spec.freq);
}

/* Most of this information came from the "Inside Macintosh" book series */
int
Wave:: Load(Mac_ResData *snd, Uint16 desired_rate)
{
	Uint8 *data;
	Uint8 *samples;
	Uint16 snd_version;
	int snd_channels;

	/* Free any existing WAVE data */
	Free();
	Init();

	/* Start loading the WAVE from the SND */
	data = snd->data;
	snd_copy16(snd_version, data);

	snd_channels = 1;			/* Is this always true? */
	if ( snd_version == FORMAT_1 ) {
		Uint16 n_types;		/* Number of sound data types */
		Uint16 f_type;		/* First sound data type */
		Uint32 init_op;		/* Initialization option (unused) */

		snd_copy16(n_types, data);
		if ( n_types != 1 ) {
			error("Multi-type sound not supported");
			return(-1);
		}
		snd_copy16(f_type, data);
		if ( f_type != SAMPLED_SND ) {
			error("Not a sampled sound resource");
			return(-1);
		}
		snd_copy32(init_op, data);
	} else if ( snd_version == FORMAT_2 ) {
		Uint16 ref_cnt;		/* (unused) */
	
		snd_copy16(ref_cnt, data);
	} else {
		error("Unknown sound format: 0x%X", snd_version);
		return(-1);
	}

	/* Next is the Sound commands section */
	{
		Uint16 num_cmds;	/* Number of sound commands */
		Uint16 command;		/* The first sound command */
		Uint16 param1;		/* BUFFER_CMD parameter 1 */
		Uint32 param2;		/* Offset to sampled data */

		snd_copy16(num_cmds, data);
		if ( num_cmds != 1 ) {
			error("Multi-command sound not supported");
			return(-1);
		}
		snd_copy16(command, data);
		if ( (command != BUFFER_CMD) && (command != SOUND_CMD) ) {
			error("Unknown sound command: 0x%X\n", command);
			return(-1);
		}
		snd_copy16(param1, data);
		/* Param1 is ignored (should be 0x0000) */

		snd_copy32(param2, data);
		/* Set 'data' to the offset of the sampled data */
		if ( param2 > snd->length ) {
			error("Offset too large -- corrupt sound?");
			return(-1);
		}
		data = snd->data+param2;
	}

	/* Next is the sampled sound header */
	{
		Uint32 sample_offset;
		Uint32 num_samples;
		Uint32 sample_rate;
		Uint32 loop_start, loop_end;
		Uint8  encoding;
		Uint8  freq_base;

		snd_copy32(sample_offset, data);
		/* FIXME: What's the interpretation of this offset? */
		if ( sample_offset != 0 ) {
			error("Sound samples don't immediately follow header");
			return(-1);
		}
		snd_copy32(num_samples, data);
		snd_copy32(sample_rate, data);
		/* Sound loops are ignored for now */
		snd_copy32(loop_start, data);
		snd_copy32(loop_end, data);
		encoding = *data++;
		if ( encoding != stdSH ) {
			error("Non-standard sound encoding: 0x%X", encoding);
			return(-1);
		}
		/* Frequency base might be used later */
		freq_base = *data++;
		
		/* Now allocate room for the sound */
		if ( num_samples > snd->length-(data-snd->data) ) {
			error("truncated sound resource");
			return(-1);
		}

		/* Convert the audio data to desired sample rates */
		samples = data;
		switch ( sample_rate ) {
			case rate11khz:
			case rate11khz2:
				/* Assuming 8-bit mono samples */
				if ( desired_rate == 0 )
					desired_rate = 11025;
				num_samples =
				  ConvertRate(sample_rate>>16, desired_rate,
						&samples, num_samples, 1);
				break;
			case rate22khz:
				/* Assuming 8-bit mono samples */
				if ( desired_rate == 0 )
					desired_rate = 22050;
				num_samples =
				  ConvertRate(sample_rate>>16, desired_rate,
						&samples, num_samples, 1);
				break;
			case rate44khz:
			default:
				if ( desired_rate == 0 ) {
					desired_rate = (sample_rate>>16);
					break;
				}
				num_samples =
				  ConvertRate(sample_rate>>16, desired_rate,
						&samples, num_samples, 1);
				break;
		}
		sample_rate = desired_rate;

		/* Fill in the audio spec */
		spec.freq = sample_rate;
		spec.format = AUDIO_U8;		/* The only format? */
		spec.channels = snd_channels;
		spec.samples = 4096;
		spec.callback = NULL;
		spec.userdata = NULL;

		/* Save the audio data */
		sound_datalen = num_samples*snd_channels;
		if ( samples == data ) {
			sound_data = new Uint8[sound_datalen];
			memcpy(sound_data, samples, sound_datalen);
		} else {
			sound_data = samples;
		}
	}
	Rewind();
	return(0);
}

#define SLOW_CONVERT
#ifdef SLOW_CONVERT
/* This is relatively accurate, but not very fast */
Uint32
Wave:: ConvertRate(Uint16 rate_in, Uint16 rate_out, 
			Uint8 **samples, Uint32 n_samples, Uint8 s_size)
{
	double ipos, i_size;
	Uint32 opos;
	Uint32 n_in, n_out;
	Uint8 *input, *output;

	n_in = n_samples*s_size;
	input = *samples;
	n_out = (Uint32)(((double)rate_out/rate_in)*n_samples)+1;
	output = new Uint8[n_out*s_size];
	i_size = ((double)rate_in/rate_out)*s_size;
#ifdef CONVERTRATE_DEBUG
printf("%g seconds of input\n", (double)n_samples/rate_in);
printf("Input rate: %hu, Output rate: %hu, Input increment: %g\n", rate_in, rate_out, i_size/s_size);
printf("%g seconds of output\n", (double)n_out/rate_out);
#endif
	for ( ipos = 0, opos = 0; (Uint32)ipos < n_in; ) {
#ifdef CONVERTRATE_DEBUG
if ( opos >= n_out*s_size ) printf("Warning: buffer output overflow!\n");
#endif
		memcpy(&output[opos], &input[(Uint32)ipos], s_size);
		ipos += i_size;
		opos += s_size;
	}
	*samples = output;
	return(opos/s_size);
}
#else
#define CHECK_ERROR
/* This assumes that the rate in and the rate out are close to eachother,
   i.e. only one sample needs to be inserted or deleted for a certain
   number of skips.  It's fast, but if inaccurate it can lead to output
   distortion and buffer overrun.
*/
Uint32
Wave:: ConvertRate(Uint16 rate_in, Uint16 rate_out, 
			Uint8 **samples, Uint32 n_samples, Uint8 s_size)
{
	Uint8 *s_in, *s_out;
	Uint32 n_in,  n_out;		/* number of bytes in sample */
	Uint32 ideal_out;
	Uint32 p_in,  p_out;		/* byte position in sample */
	Uint16 run, runs, i;
	Uint8 free_sin;
	Uint8 srate_in, srate_out;
	Uint8 skip_in, stuff_out;
	int cumulative_error;

#ifdef CONVERTRATE_DEBUG
printf("%g seconds of input\n", (double)n_samples/rate_in);
#endif
	/* Initialize variables for input samples */
	srate_in = rate_in>>8;
	srate_out = rate_out>>8;
	free_sin = 0;
	s_in = *samples;
	n_in = n_samples*s_size;
	skip_in = srate_in/srate_out;
	stuff_out = srate_out/srate_in;

#ifdef CONVERTRATE_DEBUG
printf("Rate in: 0x%X, Rate out: 0x%X, Rate in/out = %d, Rate out/in = %d\n",
				rate_in>>8, rate_out>>8, skip_in, stuff_out);
#endif
	/* Use simple integer rate conversion to get sample rates close */
	if ( skip_in > 0 ) {
		/* rate_in is approximately some multiple of rate_out */
		n_out = n_in/skip_in;
		s_out = new Uint8[n_out];
		p_in  = 0;
		p_out = 0;
		while ( p_in < n_in ) {
			memcpy(&s_out[p_out], &s_in[p_in], s_size);
			p_in  += s_size*skip_in;
			p_out += s_size*1; 
		}
		free_sin = 1;
		s_in = s_out;
		n_in = n_out;
		rate_in /= skip_in;
	} else
	if ( stuff_out > 0 ) {
		/* rate_out is approximately some multiple of rate_in */
		n_out = n_in*stuff_out;
		s_out = new Uint8[n_out];
		p_in  = 0;
		p_out = 0;
		while ( p_in < n_in ) {
			for ( run = 0; run < stuff_out; ++run ) {
				memcpy(&s_out[p_out], &s_in[p_in], s_size);
				p_out += s_size;
			}
			p_in  += s_size;
		}
		free_sin = 1;
		s_in = s_out;
		n_in = n_out;
		rate_in *= stuff_out;
	}

	/* The algorithm: 
		Since rate_in is close to rate_out, we can go 'run' samples 
		before having to insert or delete a sample.  We always insert
		or delete a full sample, doing no floating point carry over
		for the next run.  If rate_in is relatively close to rate_out,
		the error is negligible.  The above rate change code works
		to shift standard Macintosh frequencies close to standard PC
		frequencies for no error with relatively short samples.
	*/
	if ( rate_in > rate_out ) {
		run = (rate_out/(rate_in-rate_out))*s_size;
		ideal_out = (Uint32)(((double)rate_out/rate_in)*n_in)*s_size;
		runs = ideal_out/run;
		n_out = (Uint32)run*runs+(n_in-(Uint32)(run+s_size)*runs);
		s_out = new Uint8[n_out];
		p_in = 0;
		p_out = 0;
#ifdef CHECK_ERROR
		printf("Ideal output: %lu bytes\n", ideal_out);
		cumulative_error = n_out - ideal_out;
		if ( cumulative_error != 0 ) {
			printf(
"WARNING: error in converting samples from rate %lu to %lu: %d bytes\n", 
					rate_in, rate_out, cumulative_error);
		}
#endif /* CHECK_ERROR */
		for ( i=0; i<runs; ++i ) {
			memcpy(&s_out[p_out], &s_in[p_in], run);
			p_in += run;
			p_out += run;
			p_in += s_size;
		}
		run = n_in-p_in;
		memcpy(&s_out[p_out], &s_in[p_in], run);
		p_out += run;
#ifdef CONVERTRATE_DEBUG
printf("rate_in = %hu, rate_out = %hu, n_in = %lu, n_out = %lu, p_out = %lu\n",
					rate_in, rate_out, n_in, n_out, p_out);
#endif
		if ( free_sin )
			delete[] s_in;
		*samples = s_out;
	} else
	if ( rate_out > rate_in ) {
		run = (rate_in/(rate_out-rate_in))*s_size;
		ideal_out = (Uint32)(((double)rate_in/rate_out)*n_in)*s_size;
		runs = ideal_out/run;
		n_out = (Uint32)runs*(run+s_size)+(n_in-(Uint32)runs*run);
		s_out = new Uint8[n_out];
		p_in = 0;
		p_out = 0;
#ifdef CHECK_ERROR
		printf("Ideal output: %lu bytes\n", ideal_out);
		cumulative_error = n_out - ideal_out;
		if ( cumulative_error != 0 ) {
			printf(
"WARNING: error in converting samples from rate %lu to %lu: %d bytes\n", 
					rate_in, rate_out, cumulative_error);
		}
#endif /* CHECK_ERROR */
		for ( i=0; i<runs; ++i ) {
			memcpy(&s_out[p_out], &s_in[p_in], run);
			p_in += run;
			p_out += run;
			memcpy(&s_out[p_out], &s_in[p_in], s_size);
			p_out += s_size;
		}
		run = n_in-p_in;
		memcpy(&s_out[p_out], &s_in[p_in], run);
		p_out += run;
#ifdef CONVERTRATE_DEBUG
printf("rate_in = %hu, rate_out = %hu, n_in = %lu, n_out = %lu, p_out = %lu\n",
					rate_in, rate_out, n_in, n_out, p_out);
#endif
		if ( free_sin )
			delete[] s_in;
		*samples = s_out;
	} else {
		p_out = n_in;
		*samples = s_in;
	}
#ifdef CONVERTRATE_DEBUG
printf("%g seconds of output\n", (double)(p_out/s_size)/rate_out);
#endif
	return(p_out/s_size);
}
#endif

int
Wave:: Save(const char *wavefile)
{
	/* Normally, these chunks come consecutively in a WAVE file */
	SDL_RWops *dst;
	Uint32 wavelen;
	struct WaveFMT {
		Uint32	FMTchunk;
		Uint32	fmtlen;
		Uint16	encoding;	
		Uint16	channels;	/* 1 = mono, 2 = stereo */
		Uint32	frequency;	/* One of 11025, 22050, or 44100 Hz */
		Uint32	byterate;	/* Average bytes per second */
		Uint16	samplesize;	/* Bytes per sample block */
		Uint16	bitspersample;	/* One of 8, 12, 16 */
	} format;

	/* Open the WAV file for writing */
	dst = SDL_RWFromFile(wavefile, "wb");
	if ( dst == NULL ) {
		error("Couldn't open %s for writing", wavefile);
		return(-1);
	}

	/* Convert all the information about the sound chunk to WAVE format */
	memcpy(&format.FMTchunk, "fmt ", 4);
	format.fmtlen = sizeof(format)-2*sizeof(Uint32);
	format.encoding = 1;
	format.channels = spec.channels;
	format.frequency = spec.freq;
	format.byterate = format.frequency*format.channels;
	format.samplesize = ((spec.format&0xFF)/8)*spec.channels;
	format.bitspersample = (spec.format&0xFF);

	/* Swap the WAVE format information to little-endian format */
	format.fmtlen		= SDL_SwapLE32(format.fmtlen);
	format.encoding		= SDL_SwapLE16(format.encoding);
	format.channels		= SDL_SwapLE16(format.channels);
	format.frequency	= SDL_SwapLE32(format.frequency);
	format.byterate		= SDL_SwapLE32(format.byterate);
	format.samplesize	= SDL_SwapLE16(format.samplesize);
	format.bitspersample	= SDL_SwapLE16(format.bitspersample);

	/* Figure out how big the RIFF chunk will be */
	wavelen = sizeof(Uint32)+sizeof(format)+2*sizeof(Uint32)+sound_datalen;

	/* Save the WAVE */
	if ( ! SDL_RWwrite(dst, "RIFF", sizeof(Uint32), 1) ||
	     ! SDL_WriteLE32(dst, wavelen) ||
	     ! SDL_RWwrite(dst, "WAVE", sizeof(Uint32), 1) ||
	     ! SDL_RWwrite(dst, &format, sizeof(format), 1) ||
	     ! SDL_RWwrite(dst, "data", sizeof(Uint32), 1) ||
	     ! SDL_WriteLE32(dst, sound_datalen) ||
	     ! SDL_RWwrite(dst, sound_data, sound_datalen, 1) ||
	     					(SDL_RWclose(dst) != 0) ) {
		error("Couldn't write to %s", wavefile);
		SDL_RWclose(dst);
		return(-1);
	}
	return(0);
}