/*

    SDL_mixer:  An audio mixer library based on the SDL library

    Copyright (C) 1997-2009 Sam Lantinga

    This library is free software; you can redistribute it and/or

    modify it under the terms of the GNU Library General Public

    License as published by the Free Software Foundation; either

    version 2 of the License, or (at your option) any later version.

    This library 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

    Library General Public License for more details.

    You should have received a copy of the GNU Library General Public

    License along with this library; if not, write to the Free

    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

    Sam Lantinga

    slouken@libsdl.org

*/

/* $Id$ */

/* This file supports streaming WAV files, without volume adjustment */

#include <stdlib.h>

#include <string.h>

#include "SDL_audio.h"

#include "SDL_mutex.h"

#include "SDL_rwops.h"

#include "SDL_endian.h"

#include "SDL_mixer.h"

#include "wavestream.h"

/*

    Taken with permission from SDL_wave.h, part of the SDL library,

    available at: http://www.libsdl.org/

    and placed under the same license as this mixer library.

*/

/* WAVE files are little-endian */

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

/* Define values for Microsoft WAVE format */

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

#define RIFF		0x46464952		/* "RIFF" */

#define WAVE		0x45564157		/* "WAVE" */

#define FACT		0x74636166		/* "fact" */

#define LIST		0x5453494c		/* "LIST" */

#define FMT		0x20746D66		/* "fmt " */

#define DATA		0x61746164		/* "data" */

#define PCM_CODE	1

#define ADPCM_CODE	2

#define WAVE_MONO	1

#define WAVE_STEREO	2

/* Normally, these three chunks come consecutively in a WAVE file */

typedef struct WaveFMT {

/* Not saved in the chunk we read:

	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	blockalign;		/* Bytes per sample block */

	Uint16	bitspersample;		/* One of 8, 12, 16, or 4 for ADPCM */

} WaveFMT;

/* The general chunk found in the WAVE file */

typedef struct Chunk {

	Uint32 magic;

	Uint32 length;

	Uint8 *data;			/* Data includes magic and length */

} Chunk;

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

/* Define values for AIFF (IFF audio) format */

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

#define FORM		0x4d524f46		/* "FORM" */

#define AIFF		0x46464941		/* "AIFF" */

#define SSND		0x444e5353		/* "SSND" */

#define COMM		0x4d4d4f43		/* "COMM" */

/* Currently we only support a single stream at a time */

static WAVStream *music = NULL;

/* This is the format of the audio mixer data */

static SDL_AudioSpec mixer;

static int wavestream_volume = MIX_MAX_VOLUME;

/* Function to load the WAV/AIFF stream */

static SDL_RWops *LoadWAVStream (SDL_RWops *rw, SDL_AudioSpec *spec,

					long *start, long *stop);

static SDL_RWops *LoadAIFFStream (SDL_RWops *rw, SDL_AudioSpec *spec,

					long *start, long *stop);

/* Initialize the WAVStream player, with the given mixer settings

   This function returns 0, or -1 if there was an error.

 */

int WAVStream_Init(SDL_AudioSpec *mixerfmt)

{

	mixer = *mixerfmt;

	return(0);

}

void WAVStream_SetVolume(int volume)

{

	wavestream_volume = volume;

}

WAVStream *WAVStream_LoadSong(const char *file, const char *magic)

{

	SDL_RWops *rw;

	WAVStream *wave;

	rw = SDL_RWFromFile(file, "rb");

	if ( rw == NULL ) {

		SDL_SetError("Couldn't open %s", file);

		return NULL;

	}

	wave = WAVStream_LoadSong_RW(rw, magic);

	if ( wave == NULL ) {

		SDL_FreeRW(rw);

		return NULL;

	}

	return wave;

}

/* Load a WAV stream from the given file */

WAVStream *WAVStream_LoadSong_RW(SDL_RWops *rw, const char *magic)

{

	WAVStream *wave;

	SDL_AudioSpec wavespec;

	if ( ! mixer.format ) {

		Mix_SetError("WAV music output not started");

		return(NULL);

	}

	wave = (WAVStream *)malloc(sizeof *wave);

	if ( wave ) {

		memset(wave, 0, (sizeof *wave));

		if ( strcmp(magic, "RIFF") == 0 ) {

			wave->rw = LoadWAVStream(rw, &wavespec,

					&wave->start, &wave->stop);

		} else

		if ( strcmp(magic, "FORM") == 0 ) {

			wave->rw = LoadAIFFStream(rw, &wavespec,

					&wave->start, &wave->stop);

		} else {

			Mix_SetError("Unknown WAVE format");

		}

		if ( wave->rw == NULL ) {

			free(wave);

			return(NULL);

		}

		SDL_BuildAudioCVT(&wave->cvt,

			wavespec.format, wavespec.channels, wavespec.freq,

			mixer.format, mixer.channels, mixer.freq);

	}

	return(wave);

}

/* Start playback of a given WAV stream */

void WAVStream_Start(WAVStream *wave)

{

	SDL_RWseek (wave->rw, wave->start, SEEK_SET);

	music = wave;

}

/* Play some of a stream previously started with WAVStream_Start() */

void WAVStream_PlaySome(Uint8 *stream, int len)

{

	long pos;

	if ( music && ((pos=SDL_RWtell(music->rw)) < music->stop) ) {

		if ( music->cvt.needed ) {

			int original_len;

			original_len=(int)((double)len/music->cvt.len_ratio);

			if ( music->cvt.len != original_len ) {

				int worksize;

				if ( music->cvt.buf != NULL ) {

					free(music->cvt.buf);

				}

				worksize = original_len*music->cvt.len_mult;

				music->cvt.buf=(Uint8 *)malloc(worksize);

				if ( music->cvt.buf == NULL ) {

					return;

				}

				music->cvt.len = original_len;

			}

			if ( (music->stop - pos) < original_len ) {

				original_len = (music->stop - pos);

			}

			original_len = SDL_RWread(music->rw, music->cvt.buf,1,original_len);

			/* At least at the time of writing, SDL_ConvertAudio()

			   does byte-order swapping starting at the end of the

			   buffer. Thus, if we are reading 16-bit samples, we

			   had better make damn sure that we get an even

			   number of bytes, or we'll get garbage.

			 */

			if ( (music->cvt.src_format & 0x0010) && (original_len & 1) ) {

				original_len--;

			}

			music->cvt.len = original_len;

			SDL_ConvertAudio(&music->cvt);

			SDL_MixAudio(stream, music->cvt.buf, music->cvt.len_cvt, wavestream_volume);

		} else {

			Uint8 *data;

			if ( (music->stop - pos) < len ) {

				len = (music->stop - pos);

			}

			data = SDL_stack_alloc(Uint8, len);

			if (data)

			{		

				SDL_RWread(music->rw, data, len, 1);

				SDL_MixAudio(stream, data, len, wavestream_volume);

				SDL_stack_free(data);

			}	

		}

	}

}

/* Stop playback of a stream previously started with WAVStream_Start() */

void WAVStream_Stop(void)

{

	music = NULL;

}

/* Close the given WAV stream */

void WAVStream_FreeSong(WAVStream *wave)

{

	if ( wave ) {

		/* Clean up associated data */

		if ( wave->freerw ) {

			SDL_FreeRW(wave->rw);

		}

		if ( wave->cvt.buf ) {

			free(wave->cvt.buf);

		}

		free(wave);

	}

}

/* Return non-zero if a stream is currently playing */

int WAVStream_Active(void)

{

	int active;

	active = 0;

	if ( music && (SDL_RWtell(music->rw) < music->stop) ) {

		active = 1;

	}

	return(active);

}

static int ReadChunk(SDL_RWops *src, Chunk *chunk, int read_data)

{

	chunk->magic	= SDL_ReadLE32(src);

	chunk->length	= SDL_ReadLE32(src);

	if ( read_data ) {

		chunk->data = (Uint8 *)malloc(chunk->length);

		if ( chunk->data == NULL ) {

			Mix_SetError("Out of memory");

			return(-1);

		}

		if ( SDL_RWread(src, chunk->data, chunk->length, 1) != 1 ) {

			Mix_SetError("Couldn't read chunk");

			free(chunk->data);

			return(-1);

		}

	} else {

		SDL_RWseek(src, chunk->length, SEEK_CUR);

	}

	return(chunk->length);

}

static SDL_RWops *LoadWAVStream (SDL_RWops *src, SDL_AudioSpec *spec,

					long *start, long *stop)

{

	int was_error;

	Chunk chunk;

	int lenread;

	/* WAV magic header */

	Uint32 RIFFchunk;

	Uint32 wavelen;

	Uint32 WAVEmagic;

	/* FMT chunk */

	WaveFMT *format = NULL;

	was_error = 0;

	/* Check the magic header */

	RIFFchunk	= SDL_ReadLE32(src);

	wavelen		= SDL_ReadLE32(src);

	WAVEmagic	= SDL_ReadLE32(src);

	if ( (RIFFchunk != RIFF) || (WAVEmagic != WAVE) ) {

		Mix_SetError("Unrecognized file type (not WAVE)");

		was_error = 1;

		goto done;

	}

	/* Read the audio data format chunk */

	chunk.data = NULL;

	do {

		/* FIXME! Add this logic to SDL_LoadWAV_RW() */

		if ( chunk.data ) {

			free(chunk.data);

		}

		lenread = ReadChunk(src, &chunk, 1);

		if ( lenread < 0 ) {

			was_error = 1;

			goto done;

		}

	} while ( (chunk.magic == FACT) || (chunk.magic == LIST) );

	/* Decode the audio data format */

	format = (WaveFMT *)chunk.data;

	if ( chunk.magic != FMT ) {

		free(chunk.data);

		Mix_SetError("Complex WAVE files not supported");

		was_error = 1;

		goto done;

	}

	switch (SDL_SwapLE16(format->encoding)) {

		case PCM_CODE:

			/* We can understand this */

			break;

		default:

			Mix_SetError("Unknown WAVE data format");

			was_error = 1;

			goto done;

	}

	memset(spec, 0, (sizeof *spec));

	spec->freq = SDL_SwapLE32(format->frequency);

	switch (SDL_SwapLE16(format->bitspersample)) {

		case 8:

			spec->format = AUDIO_U8;

			break;

		case 16:

			spec->format = AUDIO_S16;

			break;

		default:

			Mix_SetError("Unknown PCM data format");

			was_error = 1;

			goto done;

	}

	spec->channels = (Uint8) SDL_SwapLE16(format->channels);

	spec->samples = 4096;		/* Good default buffer size */

	/* Set the file offset to the DATA chunk data */

	chunk.data = NULL;

	do {

		*start = SDL_RWtell(src) + 2*sizeof(Uint32);

		lenread = ReadChunk(src, &chunk, 0);

		if ( lenread < 0 ) {

			was_error = 1;

			goto done;

		}

	} while ( chunk.magic != DATA );

	*stop = SDL_RWtell(src);

done:

	if ( format != NULL ) {

		free(format);

	}

	if ( was_error ) {

		return NULL;

	}

	return(src);

}

/* I couldn't get SANE_to_double() to work, so I stole this from libsndfile.

 * I don't pretend to fully understand it.

 */

static Uint32 SANE_to_Uint32 (Uint8 *sanebuf)

{

	/* Negative number? */

	if (sanebuf[0] & 0x80)

		return 0;

	/* Less than 1? */

	if (sanebuf[0] <= 0x3F)

		return 1;

	/* Way too big? */

	if (sanebuf[0] > 0x40)

		return 0x4000000;

	/* Still too big? */

	if (sanebuf[0] == 0x40 && sanebuf[1] > 0x1C)

		return 800000000;

	return ((sanebuf[2] << 23) | (sanebuf[3] << 15) | (sanebuf[4] << 7)

		| (sanebuf[5] >> 1)) >> (29 - sanebuf[1]);

}

static SDL_RWops *LoadAIFFStream (SDL_RWops *src, SDL_AudioSpec *spec,

					long *start, long *stop)

{

	int was_error;

	int found_SSND;

	int found_COMM;

	Uint32 chunk_type;

	Uint32 chunk_length;

	long next_chunk;

	/* AIFF magic header */

	Uint32 FORMchunk;

	Uint32 AIFFmagic;

	/* SSND chunk        */

	Uint32 offset;

	Uint32 blocksize;

	/* COMM format chunk */

	Uint16 channels = 0;

	Uint32 numsamples = 0;

	Uint16 samplesize = 0;

	Uint8 sane_freq[10];

	Uint32 frequency = 0;

	was_error = 0;

	/* Check the magic header */

	FORMchunk	= SDL_ReadLE32(src);

	chunk_length	= SDL_ReadBE32(src);

	AIFFmagic	= SDL_ReadLE32(src);

	if ( (FORMchunk != FORM) || (AIFFmagic != AIFF) ) {

		Mix_SetError("Unrecognized file type (not AIFF)");

		was_error = 1;

		goto done;

	}

	/* From what I understand of the specification, chunks may appear in

         * any order, and we should just ignore unknown ones.

	 *

	 * TODO: Better sanity-checking. E.g. what happens if the AIFF file

	 *       contains compressed sound data?

         */

	found_SSND = 0;

	found_COMM = 0;

	do {

	    chunk_type		= SDL_ReadLE32(src);

	    chunk_length	= SDL_ReadBE32(src);

	    next_chunk		= SDL_RWtell(src) + chunk_length;

	    /* Paranoia to avoid infinite loops */

	    if (chunk_length == 0)

		break;

            switch (chunk_type) {

		case SSND:

		    found_SSND		= 1;

		    offset		= SDL_ReadBE32(src);

		    blocksize		= SDL_ReadBE32(src);

		    *start		= SDL_RWtell(src) + offset;

		    break;

		case COMM:

		    found_COMM		= 1;

		    /* Read the audio data format chunk */

		    channels		= SDL_ReadBE16(src);

		    numsamples		= SDL_ReadBE32(src);

		    samplesize		= SDL_ReadBE16(src);

		    SDL_RWread(src, sane_freq, sizeof(sane_freq), 1);

		    frequency		= SANE_to_Uint32(sane_freq);

		    break;

		default:

		    break;

	    }

	} while ((!found_SSND || !found_COMM)

		 && SDL_RWseek(src, next_chunk, SEEK_SET) != -1);

	if (!found_SSND) {

	    Mix_SetError("Bad AIFF file (no SSND chunk)");

	    was_error = 1;

	    goto done;

	}

	if (!found_COMM) {

	    Mix_SetError("Bad AIFF file (no COMM chunk)");

	    was_error = 1;

	    goto done;

	}

	*stop = *start + channels * numsamples * (samplesize / 8);

	/* Decode the audio data format */

	memset(spec, 0, (sizeof *spec));

	spec->freq = frequency;

	switch (samplesize) {

		case 8:

			spec->format = AUDIO_S8;

			break;

		case 16:

			spec->format = AUDIO_S16MSB;

			break;

		default:

			Mix_SetError("Unknown samplesize in data format");

			was_error = 1;

			goto done;

	}

	spec->channels = (Uint8) channels;

	spec->samples = 4096;		/* Good default buffer size */

done:

	if ( was_error ) {

		return NULL;

	}

	return(src);

}