view decoders/aiff.c @ 474:c66080364dff

Most decoders now report total sample play time, now. Technically, this breaks binary compatibility with the 1.0 branch, since it extends the Sound_Sample struct, but most (all?) programs are just passing pointers allocated by SDL_sound around, and might be okay. Source-level compatibility is not broken...yet! :) --ryan. -------- Original Message -------- Subject: SDL_sound patch: Finding total length of time of sound file. Date: Sun, 26 Jan 2003 09:31:17 -0800 (PST) Hi Ryan, I am working with Eric Wing and helping him modify SDL_sound. AS part of our efforts in improving and enhancing SDL_sound, we like to submit this patch. We modified the codecs to find the total time of a sound file. Below is the explanation of the patch. The patch is appended as an attachment to this email. * MOTIVATION: We needed the ability to get the total play time of a sample (And we noticed that we're not the only ones). Since SDL_sound blocks direct access to the specific decoders, there is no way for a user to know this information short of decoding the whole thing. Because of this, we believe this will be a useful addition, even though the accuracy may not be perfect (subject to each decoder) or the information may not always be available. * CONTRIBUTORS: Wesley Leong (modified the majority of the codecs and verified the results) Eric Wing (showed everyone how to do modify codec, modified mikmod) Wang Lam (modified a handful of codecs, researched into specs and int overflow) Ahilan Anantha (modified a few codecs and helped with integer math) * GENERAL ISSUES: We chose the value to be milliseconds as an Sint32. Milliseconds because that's what Sound_Seek takes as a parameter and -1 to allow for instances/codecs where the value could not be determined. We are not sure if this is the final convention you want, so we are willing to work with you on this. We also expect the total_time field to be set on open and never again modified by SDL_sound. Users may access it directly much like the sample buffer and buffer_size. We thought about recomputing the time on DecodeAll, but since users may seek or decode small chunks first, not all the data may be there. So this is better done by the user. This may be good information to document. Currently, all the main codecs are implemented except for QuickTime.
author Ryan C. Gordon <icculus@icculus.org>
date Sat, 08 May 2004 08:19:50 +0000
parents 9d0b5ec9cc26
children 3e705c9180e5
line wrap: on
line source

/*
 * SDL_sound -- An abstract sound format decoding API.
 * Copyright (C) 2001  Ryan C. Gordon.
 *
 * This library is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 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
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with this library; if not, write to the Free Software
 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
 */

/*
 * AIFF decoder for SDL_sound
 *
 * [Insert something profound about the AIFF file format here.]
 *
 * This code was ripped from a decoder I had written for SDL_mixer, which was
 * based on SDL_mixer's old AIFF music loader. (This loader was unfortunately
 * completely broken, but it was still useful because all the pieces were
 * still there, so to speak.)
 *
 * When rewriting it for SDL_sound, I changed its structure to be more like
 * the WAV loader Ryan wrote. Had they not both been part of the same project
 * it would have been embarrassing how similar they are.
 *
 * It is not the most feature-complete AIFF loader the world has ever seen.
 * For instance, it only makes a token attempt at implementing the AIFF-C
 * standard; basically the parts of it that I can easily understand and test.
 * It's a start, though.
 *
 * Please see the file COPYING in the source's root directory.
 *
 *  This file was written by Torbjörn Andersson. (d91tan@Update.UU.SE)
 */

#if HAVE_CONFIG_H
#  include <config.h>
#endif

#ifdef SOUND_SUPPORTS_AIFF

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

#include "SDL_sound.h"

#define __SDL_SOUND_INTERNAL__
#include "SDL_sound_internal.h"

static Uint32 SANE_to_Uint32 (Uint8 *sanebuf);


static int AIFF_init(void);
static void AIFF_quit(void);
static int AIFF_open(Sound_Sample *sample, const char *ext);
static void AIFF_close(Sound_Sample *sample);
static Uint32 AIFF_read(Sound_Sample *sample);
static int AIFF_rewind(Sound_Sample *sample);
static int AIFF_seek(Sound_Sample *sample, Uint32 ms);

static const char *extensions_aiff[] = { "AIFF", NULL };
const Sound_DecoderFunctions __Sound_DecoderFunctions_AIFF =
{
    {
        extensions_aiff,
        "Audio Interchange File Format",
        "Torbjörn Andersson <d91tan@Update.UU.SE>",
        "http://www.icculus.org/SDL_sound/"
    },

    AIFF_init,      /*   init() method */
    AIFF_quit,      /*   quit() method */
    AIFF_open,      /*   open() method */
    AIFF_close,     /*  close() method */
    AIFF_read,      /*   read() method */
    AIFF_rewind,    /* rewind() method */
    AIFF_seek       /*   seek() method */
};


/*****************************************************************************
 * aiff_t is what we store in our internal->decoder_private field...         *
 *****************************************************************************/
typedef struct S_AIFF_FMT_T
{
    Uint32 type;

    Uint32 total_bytes;
    Uint32 data_starting_offset;

    void (*free)(struct S_AIFF_FMT_T *fmt);
    Uint32 (*read_sample)(Sound_Sample *sample);
    int (*rewind_sample)(Sound_Sample *sample);
    int (*seek_sample)(Sound_Sample *sample, Uint32 ms);


#if 0
/*
   this is ripped from wav.c as ann example of format-specific data.
   please replace with something more appropriate when the need arises.
*/
    union
    {
        struct
        {
            Uint16 cbSize;
            Uint16 wSamplesPerBlock;
            Uint16 wNumCoef;
            ADPCMCOEFSET *aCoeff;
        } adpcm;

        /* put other format-specific data here... */
    } fmt;
#endif
} fmt_t;


typedef struct
{
    fmt_t fmt;
    Sint32 bytesLeft;
} aiff_t;



    /* Chunk management code... */

#define formID 0x4D524F46  /* "FORM", in ascii. */
#define aiffID 0x46464941  /* "AIFF", in ascii. */
#define aifcID 0x43464941  /* "AIFC", in ascii. */
#define ssndID 0x444E5353  /* "SSND", in ascii. */


/*****************************************************************************
 * The COMM chunk...                                                         *
 *****************************************************************************/

#define commID 0x4D4D4F43  /* "COMM", in ascii. */

/* format/compression types... */
#define noneID 0x454E4F4E  /* "NONE", in ascii. */

typedef struct
{
    Uint32 ckID;
    Uint32 ckDataSize;
    Uint16 numChannels;
    Uint32 numSampleFrames;
    Uint16 sampleSize;
    Uint32 sampleRate;
        /*
         * We don't handle AIFF-C compressed audio yet, but for those
         * interested the allowed compression types are supposed to be
         *
         *   compressionType   compressionName   meaning
         *   ---------------------------------------------------------------
         *   'NONE'            "not compressed"  uncompressed, that is,
         *                                        straight digitized samples
         *   'ACE2'            "ACE 2-to-1"      2-to-1 IIGS ACE (Audio
         *                                        Compression / Expansion)
         *   'ACE8'            "ACE 8-to-3"      8-to-3 IIGS ACE (Audio
         *                                        Compression / Expansion)
         *   'MAC3'            "MACE 3-to-1"     3-to-1 Macintosh Audio
         *                                        Compression / Expansion
         *   'MAC6'            "MACE 6-to-1"     6-to-1 Macintosh Audio
         *                                        Compression / Expansion
         *
         * A pstring is a "Pascal-style string", that is, "one byte followed
         * by test bytes followed when needed by one pad byte. The total
         * number of bytes in a pstring must be even. The pad byte is
         * included when the number of text bytes is even, so the total of
         * text bytes + one count byte + one pad byte will be even. This pad
         * byte is not reflected in the count."
         *
         * As for how these compression algorithms work, your guess is as
         * good as mine.
         */
    Uint32 compressionType;
#if 0
    pstring compressionName;
#endif
} comm_t;


/*
 * Read in a comm_t from disk. This makes this process safe regardless of
 *  the processor's byte order or how the comm_t structure is packed.
 */

static int read_comm_chunk(SDL_RWops *rw, comm_t *comm)
{
    Uint8 sampleRate[10];

    /* skip reading the chunk ID, since it was already read at this point... */
    comm->ckID = commID;

    if (SDL_RWread(rw, &comm->ckDataSize, sizeof (comm->ckDataSize), 1) != 1)
        return(0);
    comm->ckDataSize = SDL_SwapBE32(comm->ckDataSize);

    if (SDL_RWread(rw, &comm->numChannels, sizeof (comm->numChannels), 1) != 1)
        return(0);
    comm->numChannels = SDL_SwapBE16(comm->numChannels);

    if (SDL_RWread(rw, &comm->numSampleFrames,
                   sizeof (comm->numSampleFrames), 1) != 1)
        return(0);
    comm->numSampleFrames = SDL_SwapBE32(comm->numSampleFrames);

    if (SDL_RWread(rw, &comm->sampleSize, sizeof (comm->sampleSize), 1) != 1)
        return(0);
    comm->sampleSize = SDL_SwapBE16(comm->sampleSize);

    if (SDL_RWread(rw, sampleRate, sizeof (sampleRate), 1) != 1)
        return(0);
    comm->sampleRate = SANE_to_Uint32(sampleRate);

    if (comm->ckDataSize > sizeof(comm->numChannels)
                         + sizeof(comm->numSampleFrames)
                         + sizeof(comm->sampleSize)
                         + sizeof(sampleRate))
    {
        if (SDL_RWread(rw, &comm->compressionType,
                       sizeof (comm->compressionType), 1) != 1)
            return(0);
        comm->compressionType = SDL_SwapBE32(comm->compressionType);
    } /* if */
    else
    {
        comm->compressionType = noneID;
    } /* else */

    return(1);
} /* read_comm_chunk */



/*****************************************************************************
 * The SSND chunk...                                                         *
 *****************************************************************************/

typedef struct
{
    Uint32 ckID;
    Uint32 ckDataSize;
    Uint32 offset;
    Uint32 blockSize;
    /*
     * Then, comm->numSampleFrames sample frames. (It's better to get the
     * length from numSampleFrames than from ckDataSize.)
     */
} ssnd_t;


static int read_ssnd_chunk(SDL_RWops *rw, ssnd_t *ssnd)
{
    /* skip reading the chunk ID, since it was already read at this point... */
    ssnd->ckID = ssndID;

    if (SDL_RWread(rw, &ssnd->ckDataSize, sizeof (ssnd->ckDataSize), 1) != 1)
        return(0);
    ssnd->ckDataSize = SDL_SwapBE32(ssnd->ckDataSize);

    if (SDL_RWread(rw, &ssnd->offset, sizeof (ssnd->offset), 1) != 1)
        return(0);
    ssnd->offset = SDL_SwapBE32(ssnd->offset);

    if (SDL_RWread(rw, &ssnd->blockSize, sizeof (ssnd->blockSize), 1) != 1)
        return(0);
    ssnd->blockSize = SDL_SwapBE32(ssnd->blockSize);

    /* Leave the SDL_RWops position indicator at the start of the samples */
    if (SDL_RWseek(rw, (int) ssnd->offset, SEEK_CUR) == -1)
        return(0);

    return(1);
} /* read_ssnd_chunk */



/*****************************************************************************
 * Normal, uncompressed aiff handler...                                      *
 *****************************************************************************/

static Uint32 read_sample_fmt_normal(Sound_Sample *sample)
{
    Uint32 retval;
    Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
    aiff_t *a = (aiff_t *) internal->decoder_private;
    Uint32 max = (internal->buffer_size < (Uint32) a->bytesLeft) ?
                    internal->buffer_size : (Uint32) a->bytesLeft;

    assert(max > 0);

        /*
         * We don't actually do any decoding, so we read the AIFF data
         *  directly into the internal buffer...
         */
    retval = SDL_RWread(internal->rw, internal->buffer, 1, max);

    a->bytesLeft -= retval;

        /* Make sure the read went smoothly... */
    if ((retval == 0) || (a->bytesLeft == 0))
        sample->flags |= SOUND_SAMPLEFLAG_EOF;

    else if (retval == -1)
        sample->flags |= SOUND_SAMPLEFLAG_ERROR;

        /* (next call this EAGAIN may turn into an EOF or error.) */
    else if (retval < internal->buffer_size)
        sample->flags |= SOUND_SAMPLEFLAG_EAGAIN;

    return(retval);
} /* read_sample_fmt_normal */


static int rewind_sample_fmt_normal(Sound_Sample *sample)
{
    /* no-op. */
    return(1);
} /* rewind_sample_fmt_normal */


static int seek_sample_fmt_normal(Sound_Sample *sample, Uint32 ms)
{
    Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
    aiff_t *a = (aiff_t *) internal->decoder_private;
    fmt_t *fmt = &a->fmt;
    int offset = __Sound_convertMsToBytePos(&sample->actual, ms);
    int pos = (int) (fmt->data_starting_offset + offset);
    int rc = SDL_RWseek(internal->rw, pos, SEEK_SET);
    BAIL_IF_MACRO(rc != pos, ERR_IO_ERROR, 0);
    a->bytesLeft = fmt->total_bytes - offset;
    return(1);  /* success. */
} /* seek_sample_fmt_normal */


static void free_fmt_normal(fmt_t *fmt)
{
    /* it's a no-op. */
} /* free_fmt_normal */


static int read_fmt_normal(SDL_RWops *rw, fmt_t *fmt)
{
    /* (don't need to read more from the RWops...) */
    fmt->free = free_fmt_normal;
    fmt->read_sample = read_sample_fmt_normal;
    fmt->rewind_sample = rewind_sample_fmt_normal;
    fmt->seek_sample = seek_sample_fmt_normal;
    return(1);
} /* read_fmt_normal */




/*****************************************************************************
 * Everything else...                                                        *
 *****************************************************************************/

static int AIFF_init(void)
{
    return(1);  /* always succeeds. */
} /* AIFF_init */


static void AIFF_quit(void)
{
    /* it's a no-op. */
} /* AIFF_quit */


/* 
 * Sample rate is encoded as an "80 bit IEEE Standard 754 floating point
 * number (Standard Apple Numeric Environment [SANE] data type Extended)".
 * Whose bright idea was that?
 *
 * This function was adapted from libsndfile, and while I do know a little
 * bit about the IEEE floating point standard I don't pretend to fully
 * understand this.
 */
static Uint32 SANE_to_Uint32 (Uint8 *sanebuf)
{
    /* Is the frequency outside of what we can represent with Uint32? */
    if ( (sanebuf[0] & 0x80)
      || (sanebuf[0] <= 0x3F)
      || (sanebuf[0] > 0x40)
      || (sanebuf[0] == 0x40 && sanebuf[1] > 0x1C) )
        return 0;

    return ((sanebuf[2] << 23) | (sanebuf[3] << 15) | (sanebuf[4] << 7)
        | (sanebuf[5] >> 1)) >> (29 - sanebuf[1]);
} /* SANE_to_Uint32 */


static int find_chunk(SDL_RWops *rw, Uint32 id)
{
    Sint32 siz = 0;
    Uint32 _id = 0;

    while (1)
    {
        BAIL_IF_MACRO(SDL_RWread(rw, &_id, sizeof (_id), 1) != 1, NULL, 0);
        if (SDL_SwapLE32(_id) == id)
            return(1);

        BAIL_IF_MACRO(SDL_RWread(rw, &siz, sizeof (siz), 1) != 1, NULL, 0);
        siz = SDL_SwapBE32(siz);
        assert(siz > 0);
        BAIL_IF_MACRO(SDL_RWseek(rw, siz, SEEK_CUR) == -1, NULL, 0);
    } /* while */

    return(0);  /* shouldn't hit this, but just in case... */
} /* find_chunk */


static int read_fmt(SDL_RWops *rw, comm_t *c, fmt_t *fmt)
{
    fmt->type = c->compressionType;

    /* if it's in this switch statement, we support the format. */
    switch (fmt->type)
    {
        case noneID:
            SNDDBG(("AIFF: Appears to be uncompressed audio.\n"));
            return(read_fmt_normal(rw, fmt));

        /* add other types here. */

        default:
            SNDDBG(("AIFF: Format %lu is unknown.\n",
                    (unsigned int) fmt->type));
            BAIL_MACRO("AIFF: Unsupported format", 0);
    } /* switch */

    assert(0);  /* shouldn't hit this point. */
    return(0);
} /* read_fmt */


static int AIFF_open(Sound_Sample *sample, const char *ext)
{
    Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
    SDL_RWops *rw = internal->rw;
    Uint32 chunk_id;
    int bytes_per_sample;
    long pos;
    comm_t c;
    ssnd_t s;
    aiff_t *a;

    BAIL_IF_MACRO(SDL_ReadLE32(rw) != formID, "AIFF: Not a FORM file.", 0);
        SDL_ReadBE32(rw);  /* throw the length away; we don't need it. */

    chunk_id = SDL_ReadLE32(rw);
    BAIL_IF_MACRO(chunk_id != aiffID && chunk_id != aifcID,
        "AIFF: Not an AIFF or AIFC file.", 0);

    /* Chunks may appear in any order, so we establish base camp here. */
    pos = SDL_RWtell(rw);

    BAIL_IF_MACRO(!find_chunk(rw, commID), "AIFF: No common chunk.", 0);
    BAIL_IF_MACRO(!read_comm_chunk(rw, &c),
                  "AIFF: Can't read common chunk.", 0);

    sample->actual.channels = (Uint8) c.numChannels;
    sample->actual.rate = c.sampleRate;

    /* Really, sample->total_time = (c.numSampleFrames*1000) c.sampleRate */
    sample->total_time = (c.numSampleFrames / c.sampleRate) * 1000;
    sample->total_time += (c.numSampleFrames % c.sampleRate) 
                       *  1000 / c.sampleRate;

    if (c.sampleSize <= 8)
    {
        sample->actual.format = AUDIO_S8;
        bytes_per_sample = c.numChannels;
    } /* if */
    else if (c.sampleSize <= 16)
    {
        sample->actual.format = AUDIO_S16MSB;
        bytes_per_sample = 2 * c.numChannels;
    } /* if */
    else
    {
        BAIL_MACRO("AIFF: Unsupported sample size.", 0);
    } /* else */

    BAIL_IF_MACRO(c.sampleRate == 0, "AIFF: Unsupported sample rate.", 0);

    a = (aiff_t *) malloc(sizeof(aiff_t));
    BAIL_IF_MACRO(a == NULL, ERR_OUT_OF_MEMORY, 0);

    if (!read_fmt(rw, &c, &(a->fmt)))
    {
        free(a);
        return(0);
    } /* if */

    SDL_RWseek(rw, pos, SEEK_SET);    /* if the seek fails, let it go... */

    if (!find_chunk(rw, ssndID))
    {
        free(a);
        BAIL_MACRO("AIFF: No sound data chunk.", 0);
    } /* if */

    if (!read_ssnd_chunk(rw, &s))
    {
        free(a);
        BAIL_MACRO("AIFF: Can't read sound data chunk.", 0);
    } /* if */

    a->fmt.total_bytes = a->bytesLeft = bytes_per_sample * c.numSampleFrames;
    a->fmt.data_starting_offset = SDL_RWtell(rw);
    internal->decoder_private = (void *) a;

    sample->flags = SOUND_SAMPLEFLAG_CANSEEK;

    SNDDBG(("AIFF: Accepting data stream.\n"));
    return(1); /* we'll handle this data. */
} /* AIFF_open */


static void AIFF_close(Sound_Sample *sample)
{
    Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
    aiff_t *a = (aiff_t *) internal->decoder_private;
    a->fmt.free(&(a->fmt));
    free(a);
} /* AIFF_close */


static Uint32 AIFF_read(Sound_Sample *sample)
{
    Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
    aiff_t *a = (aiff_t *) internal->decoder_private;
    return(a->fmt.read_sample(sample));
} /* AIFF_read */


static int AIFF_rewind(Sound_Sample *sample)
{
    Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
    aiff_t *a = (aiff_t *) internal->decoder_private;
    fmt_t *fmt = &a->fmt;
    int rc = SDL_RWseek(internal->rw, fmt->data_starting_offset, SEEK_SET);
    BAIL_IF_MACRO(rc != fmt->data_starting_offset, ERR_IO_ERROR, 0);
    a->bytesLeft = fmt->total_bytes;
    return(fmt->rewind_sample(sample));
} /* AIFF_rewind */


static int AIFF_seek(Sound_Sample *sample, Uint32 ms)
{
    Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
    aiff_t *a = (aiff_t *) internal->decoder_private;
    return(a->fmt.seek_sample(sample, ms));
} /* AIFF_seek */

#endif /* SOUND_SUPPORTS_AIFF */

/* end of aiff.c ... */