gwcmad.c

/* MP3 support for SoX
 *
 * Uses libmad for MP3 decoding
 * and libmp3lame for MP3 encoding
 *
 * Written by Fabrizio Gennari <fabrizio.ge@tiscali.it>
 *
 * The decoding part is based on the decoder-tutorial program madlld
 * written by Bertrand Petit <madlld@phoe.fmug.org>,
 */

#include <string.h>

#include <mad.h>

#ifdef HAVE_LAME_LAME_H
#include <lame/lame.h>
#endif

#if HAVE_ID3TAG && HAVE_UNISTD_H
  #include <id3tag.h>
  #include <unistd.h>
#else
  #define ID3_TAG_FLAG_FOOTERPRESENT 0x10
#endif

#define INPUT_BUFFER_SIZE       (sox_globals.bufsiz)

/* Private data */
typedef struct {
	int fd ;
        struct mad_stream       Stream;
        struct mad_frame        Frame;
        struct mad_synth        Synth;
        mad_timer_t             Timer;
        unsigned char           *InputBuffer;
        ptrdiff_t             cursamp;
        size_t              FrameCount;
#ifdef HAVE_LAME_LAME_H
        lame_global_flags       *gfp;
#endif /*HAVE_LAME_LAME_H*/
} priv_t;

static priv_t private_data ;

/* This function merges the functions tagtype() and id3_tag_query()
   from MAD's libid3tag, so we don't have to link to it
   Returns 0 if the frame is not an ID3 tag, tag length if it is */

static int tagtype(const unsigned char *data, size_t length)
{
    if (length >= 3 && data[0] == 'T' && data[1] == 'A' && data[2] == 'G')
    {
        return 128; /* ID3V1 */
    }

    if (length >= 10 &&
        (data[0] == 'I' && data[1] == 'D' && data[2] == '3') &&
        data[3] < 0xff && data[4] < 0xff &&
        data[6] < 0x80 && data[7] < 0x80 && data[8] < 0x80 && data[9] < 0x80)
    {     /* ID3V2 */
        unsigned char flags;
        unsigned int size;
        flags = data[5];
        size = 10 + (data[6]<<21) + (data[7]<<14) + (data[8]<<7) + data[9];
        if (flags & ID3_TAG_FLAG_FOOTERPRESENT)
            size += 10;
        for (; size < length && !data[size]; ++size);  /* Consume padding */
        return size;
    }

    return 0;
}

#include "mp3-duration.h"

/*
 * (Re)fill the stream buffer that is to be decoded.  If any data
 * still exists in the buffer then they are first shifted to be
 * front of the stream buffer.
 */
static int sox_mp3_input(sox_format_t * ft)
{
    priv_t *p = &private_data ;
    size_t bytes_read;
    size_t remaining;

    remaining = p->Stream.bufend - p->Stream.next_frame;

    /* libmad does not consume all the buffer it's given. Some
     * data, part of a truncated frame, is left unused at the
     * end of the buffer. That data must be put back at the
     * beginning of the buffer and taken in account for
     * refilling the buffer. This means that the input buffer
     * must be large enough to hold a complete frame at the
     * highest observable bit-rate (currently 448 kb/s).
     * TODO: Is 2016 bytes the size of the largest frame?
     * (448000*(1152/32000))/8
     */
    memmove(p->InputBuffer, p->Stream.next_frame, remaining);

    bytes_read = readbuf(p->fd, p->InputBuffer+remaining,
                            INPUT_BUFFER_SIZE-remaining);
    if (bytes_read == 0)
    {
        return SOX_EOF;
    }

    mad_stream_buffer(&p->Stream, p->InputBuffer, bytes_read+remaining);
    p->Stream.error = 0;

    return SOX_SUCCESS;
}

/* Attempts to read an ID3 tag at the current location in stream and
 * consume it all.  Returns SOX_EOF if no tag is found.  Its up to
 * caller to recover.
 * */
static int sox_mp3_inputtag(sox_format_t * ft)
{
    priv_t *p = &private_data ;
    int rc = SOX_EOF;
    size_t remaining;
    size_t tagsize;


    /* FIXME: This needs some more work if we are to ever
     * look at the ID3 frame.  This is because the Stream
     * may not be able to hold the complete ID3 frame.
     * We should consume the whole frame inside tagtype()
     * instead of outside of tagframe().  That would support
     * recovering when Stream contains less then 8-bytes (header)
     * and also when ID3v2 is bigger then Stream buffer size.
     * Need to pass in stream so that buffer can be
     * consumed as well as letting additional data to be
     * read in.
     */
    remaining = p->Stream.bufend - p->Stream.next_frame;
    if ((tagsize = tagtype(p->Stream.this_frame, remaining)))
    {
        mad_stream_skip(&p->Stream, tagsize);
        rc = SOX_SUCCESS;
    }

    /* We know that a valid frame hasn't been found yet
     * so help libmad out and go back into frame seek mode.
     * This is true whether an ID3 tag was found or not.
     */
    mad_stream_sync(&p->Stream);

    return rc;
}

static int startread(sox_format_t * ft)
{
    priv_t *p = &private_data ;
    size_t ReadSize;

    p->InputBuffer = NULL;

    p->InputBuffer=lsx_malloc(INPUT_BUFFER_SIZE);

    if (ft->seekable) {
#if HAVE_ID3TAG && HAVE_UNISTD_H
      read_comments(ft);
      rewind(ft->fp);
      if (!ft->signal.length)
#endif
        ft->signal.length = mp3_duration_ms(ft->fp, p->InputBuffer);
    }

    mad_stream_init(&p->Stream);
    mad_frame_init(&p->Frame);
    mad_synth_init(&p->Synth);
    mad_timer_reset(&p->Timer);

    ft->encoding.encoding = SOX_ENCODING_MP3;

    /* Decode at least one valid frame to find out the input
     * format.  The decoded frame will be saved off so that it
     * can be processed later.
     */
    ReadSize = lsx_readbuf(ft, p->InputBuffer, INPUT_BUFFER_SIZE);
    if (ReadSize != INPUT_BUFFER_SIZE && ferror(ft->fp))
      return SOX_EOF;

    mad_stream_buffer(&p->Stream, p->InputBuffer, ReadSize);

    /* Find a valid frame before starting up.  This makes sure
     * that we have a valid MP3 and also skips past ID3v2 tags
     * at the beginning of the audio file.
     */
    p->Stream.error = 0;
    while (mad_frame_decode(&p->Frame,&p->Stream))
    {
        /* check whether input buffer needs a refill */
        if (p->Stream.error == MAD_ERROR_BUFLEN)
        {
            if (sox_mp3_input(ft) == SOX_EOF)
                return SOX_EOF;

            continue;
        }

        /* Consume any ID3 tags */
        sox_mp3_inputtag(ft);

        /* FIXME: We should probably detect when we've read
         * a bunch of non-ID3 data and still haven't found a
         * frame.  In that case we can abort early without
         * scanning the whole file.
         */
        p->Stream.error = 0;
    }

    if (p->Stream.error)
    {
        lsx_fail_errno(ft,SOX_EOF,"No valid MP3 frame found");
        return SOX_EOF;
    }

    switch(p->Frame.header.mode)
    {
        case MAD_MODE_SINGLE_CHANNEL:
        case MAD_MODE_DUAL_CHANNEL:
        case MAD_MODE_JOINT_STEREO:
        case MAD_MODE_STEREO:
            ft->signal.channels = MAD_NCHANNELS(&p->Frame.header);
            break;
        default:
            lsx_fail_errno(ft, SOX_EFMT, "Cannot determine number of channels");
            return SOX_EOF;
    }

    p->FrameCount=1;

    mad_timer_add(&p->Timer,p->Frame.header.duration);
    mad_synth_frame(&p->Synth,&p->Frame);
    ft->signal.rate=p->Synth.pcm.samplerate;
    ft->signal.length = ft->signal.length * .001 * ft->signal.rate + .5;
    ft->signal.length *= ft->signal.channels;  /* Keep separate from line above! */

    p->cursamp = 0;

    return SOX_SUCCESS;
}

/*
 * Read up to len samples from p->Synth
 * If needed, read some more MP3 data, decode them and synth them
 * Place in buf[].
 * Return number of samples read.
 */
static size_t sox_mp3read(sox_format_t * ft, sox_sample_t *buf, size_t len)
{
    priv_t *p = &private_data ;
    size_t donow,i,done=0;
    mad_fixed_t sample;
    size_t chan;

    do {
        size_t x = (p->Synth.pcm.length - p->cursamp)*ft->signal.channels;
        donow=min(len, x);
        i=0;
        while(i<donow){
            for(chan=0;chan<ft->signal.channels;chan++){
                sample=p->Synth.pcm.samples[chan][p->cursamp];
                if (sample < -MAD_F_ONE)
                    sample=-MAD_F_ONE;
                else if (sample >= MAD_F_ONE)
                    sample=MAD_F_ONE-1;
                *buf++=(sox_sample_t)(sample<<(32-1-MAD_F_FRACBITS));
                i++;
            }
            p->cursamp++;
        };

        len-=donow;
        done+=donow;

        if (len==0) break;

        /* check whether input buffer needs a refill */
        if (p->Stream.error == MAD_ERROR_BUFLEN)
        {
            if (sox_mp3_input(ft) == SOX_EOF)
                return 0;
        }

        if (mad_frame_decode(&p->Frame,&p->Stream))
        {
            if(MAD_RECOVERABLE(p->Stream.error))
            {
                sox_mp3_inputtag(ft);
                continue;
            }
            else
            {
                if (p->Stream.error == MAD_ERROR_BUFLEN)
                    continue;
                else
                {
                    lsx_report("unrecoverable frame level error (%s).",
                              mad_stream_errorstr(&p->Stream));
                    return done;
                }
            }
        }
        p->FrameCount++;
        mad_timer_add(&p->Timer,p->Frame.header.duration);
        mad_synth_frame(&p->Synth,&p->Frame);
        p->cursamp=0;
    } while(1);

    return done;
}

static int stopread(sox_format_t * ft)
{
    priv_t *p = &private_data ;

  mad_synth_finish(&p->Synth);
  mad_frame_finish(&p->Frame);
  mad_stream_finish(&p->Stream);

  free(p->InputBuffer);

  return SOX_SUCCESS;
}

#ifdef HAVE_LAME_LAME_H
static void null_error_func(const char* string UNUSED, va_list va UNUSED)
{
  return;
}

static int startwrite(sox_format_t * ft)
{
    priv_t *p = &private_data ;

  if (ft->encoding.encoding != SOX_ENCODING_MP3) {
    if(ft->encoding.encoding != SOX_ENCODING_UNKNOWN)
      lsx_report("Encoding forced to MP3");
    ft->encoding.encoding = SOX_ENCODING_MP3;
  }

  p->gfp = lame_init();
  if (p->gfp == NULL){
    lsx_fail_errno(ft,SOX_EOF,"Initialization of LAME library failed");
    return(SOX_EOF);
  }

  if (ft->signal.channels != SOX_ENCODING_UNKNOWN) {
    if ( (lame_set_num_channels(p->gfp,(int)ft->signal.channels)) < 0) {
        lsx_fail_errno(ft,SOX_EOF,"Unsupported number of channels");
        return(SOX_EOF);
    }
  }
  else
    ft->signal.channels = lame_get_num_channels(p->gfp); /* LAME default */

  lame_set_in_samplerate(p->gfp,(int)ft->signal.rate);

  lame_set_bWriteVbrTag(p->gfp, 0); /* disable writing VBR tag */

  /* The bitrate, mode, quality and other settings are the default ones,
     since SoX's command line options do not allow to set them */

  /* FIXME: Someone who knows about lame could implement adjustable compression
     here.  E.g. by using the -C value as an index into a table of params or
     as a compressed bit-rate. */
  if (ft->encoding.compression != HUGE_VAL)
      lsx_warn("-C option not supported for mp3; using default compression rate");
  if (lame_init_params(p->gfp) < 0){
        lsx_fail_errno(ft,SOX_EOF,"LAME initialization failed");
        return(SOX_EOF);
  }
  lame_set_errorf(p->gfp,null_error_func);
  lame_set_debugf(p->gfp,null_error_func);
  lame_set_msgf  (p->gfp,null_error_func);

  return(SOX_SUCCESS);
}

static size_t sox_mp3write(sox_format_t * ft, const sox_sample_t *buf, size_t samp)
{
    priv_t *p = &private_data ;
    unsigned char *mp3buffer;
    size_t mp3buffer_size;
    short signed int *buffer_l, *buffer_r = NULL;
    int nsamples = samp/ft->signal.channels;
    int i,j;
    ptrdiff_t done = 0;
    size_t written;

    /* NOTE: This logic assumes that "short int" is 16-bits
     * on all platforms.  It happens to be for all that I know
     * about.
     *
     * Lame ultimately wants data scaled to 16-bit samples
     * and assumes for the majority of cases that your passing
     * in something scaled based on passed in datatype
     * (16, 32, 64, and float).
     *
     * If we used long buffers then this means it expects
     * different scalling between 32-bit and 64-bit CPU's.
     *
     * We might as well scale it ourselfs to 16-bit to allow
     * lsx_malloc()'ing a smaller buffer and call a consistent
     * interface.
     */
    buffer_l = lsx_malloc(nsamples * sizeof(short signed int));

    if (ft->signal.channels == 2)
    {
        /* lame doesn't support interleaved samples so we must break
         * them out into separate buffers.
         */
        buffer_r = lsx_malloc(nsamples* sizeof(short signed int));
        j=0;
        for (i=0; i<nsamples; i++)
        {
            buffer_l[i]=SOX_SAMPLE_TO_SIGNED_16BIT(buf[j++], ft->clips);
            buffer_r[i]=SOX_SAMPLE_TO_SIGNED_16BIT(buf[j++], ft->clips);
        }
    }
    else
    {
        j=0;
        for (i=0; i<nsamples; i++)
        {
            buffer_l[i]=SOX_SAMPLE_TO_SIGNED_16BIT(buf[j++], ft->clips);
        }
    }

    mp3buffer_size = 1.25 * nsamples + 7200;
    mp3buffer = lsx_malloc(mp3buffer_size);

    if ((written = lame_encode_buffer(p->gfp,buffer_l, buffer_r,
                                      nsamples, mp3buffer,
                                      (int)mp3buffer_size)) > mp3buffer_size){
        lsx_fail_errno(ft,SOX_EOF,"Encoding failed");
        goto end;
    }

    if (lsx_writebuf(ft, mp3buffer, written) < written)
    {
        lsx_fail_errno(ft,SOX_EOF,"File write failed");
        goto end;
    }

    done = nsamples*ft->signal.channels;

end:
    free(mp3buffer);
    if (ft->signal.channels == 2)
        free(buffer_r);
    free(buffer_l);

    return done;
}

static int stopwrite(sox_format_t * ft)
{
  priv_t *p = &private_data ;
  unsigned char mp3buffer[7200];
  int written;
  size_t written2;

  if ((written=lame_encode_flush(p->gfp, mp3buffer, 7200)) <0){
    lsx_fail_errno(ft,SOX_EOF,"Encoding failed");
  }
  else if (lsx_writebuf(ft, mp3buffer, written2 = written) < written2){
    lsx_fail_errno(ft,SOX_EOF,"File write failed");
  }

  lame_close(p->gfp);
  return SOX_SUCCESS;
}

#else /* HAVE_LAME_LAME_H */
static int startwrite(sox_format_t * ft UNUSED)
{
  lsx_fail_errno(ft,SOX_EOF,"SoX was compiled without MP3 encoding support");
  return SOX_EOF;
}
#define sox_mp3write NULL
#define stopwrite NULL
#endif /* HAVE_LAME_LAME_H */

SOX_FORMAT_HANDLER(mp3)
{
  static char const * const names[] = {"mp3", "mp2", "audio/mpeg", NULL};
  static unsigned const write_encodings[] = {
    SOX_ENCODING_GSM, 0, 0};
  static sox_format_handler_t const handler = {SOX_LIB_VERSION_CODE,
    "MPEG Layer 3 lossy audio compression", names, 0,
    startread, sox_mp3read, stopread,
    startwrite, sox_mp3write, stopwrite,
    NULL, write_encodings, NULL, sizeof(priv_t)
  };
  return &handler;
}