ref: af81c85a823c9ce0b2524e89faf6e42e3cbf4d0f
dir: /src/mp3.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 "st_i.h"
#include <string.h>
#if defined(HAVE_LIBMAD) || defined(HAVE_LAME)
#if defined(HAVE_LIBMAD)
#include <mad.h>
#endif
#if defined(HAVE_LAME)
#include <lame/lame.h>
#include <math.h>
#endif
#ifndef MIN
#define MIN(s1,s2) ((s1)<(s2)?(s1):(s2))
#endif
#define INPUT_BUFFER_SIZE (ST_BUFSIZ)
/* Private data */
struct mp3priv {
#if defined(HAVE_LIBMAD)
struct mad_stream *Stream;
struct mad_frame *Frame;
struct mad_synth *Synth;
mad_timer_t *Timer;
unsigned char *InputBuffer;
st_ssize_t cursamp;
st_size_t FrameCount;
#endif /*HAVE_LIBMAD*/
#if defined(HAVE_LAME)
lame_global_flags *gfp;
#endif /*HAVE_LAME*/
};
#if defined(HAVE_LIBMAD)
/* 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 */
#define ID3_TAG_FLAG_FOOTERPRESENT 0x10
static int tagtype(const unsigned char *data, int length)
{
/* TODO: It would be nice to look for Xing VBR headers
* or TLE fields in ID3 to detect length of file
* and set ft->length.
* For CBR, we should fstat the file and divided
* by bitrate to find 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 = (data[6]<<21) + (data[7]<<14) + (data[8]<<7) + data[9];
if (flags & ID3_TAG_FLAG_FOOTERPRESENT)
size += 10;
return 10 + size;
}
return 0;
}
/*
* (Re)fill the stream buffer whish 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 st_mp3_input(ft_t ft)
{
struct mp3priv *p = (struct mp3priv *) ft->priv;
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 = st_readbuf(ft, p->InputBuffer+remaining, 1,
INPUT_BUFFER_SIZE-remaining);
if (bytes_read == 0)
{
return ST_EOF;
}
mad_stream_buffer(p->Stream, p->InputBuffer, bytes_read+remaining);
p->Stream->error = 0;
return ST_SUCCESS;
}
/* Attempts to read an ID3 tag at the current location in stream and
* consume it all. Returns ST_EOF if no tag is found. Its up to
* caller to recover.
* */
static int st_mp3_inputtag(ft_t ft)
{
struct mp3priv *p = (struct mp3priv *) ft->priv;
int rc = ST_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 = ST_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;
}
int st_mp3startread(ft_t ft)
{
struct mp3priv *p = (struct mp3priv *) ft->priv;
size_t ReadSize;
p->Stream = NULL;
p->Frame = NULL;
p->Synth = NULL;
p->Timer = NULL;
p->InputBuffer = NULL;
p->Stream=(struct mad_stream *)malloc(sizeof(struct mad_stream));
p->Frame=(struct mad_frame *)malloc(sizeof(struct mad_frame));
p->Synth=(struct mad_synth *)malloc(sizeof(struct mad_synth));
p->Timer=(mad_timer_t *)malloc(sizeof(mad_timer_t));
p->InputBuffer=(unsigned char *)malloc(INPUT_BUFFER_SIZE);
if (!p->Stream || !p->Frame || !p->Synth ||
!p->Timer || !p->InputBuffer)
{
st_fail_errno(ft, ST_ENOMEM, "Could not allocate memory");
goto error;
}
mad_stream_init(p->Stream);
mad_frame_init(p->Frame);
mad_synth_init(p->Synth);
mad_timer_reset(p->Timer);
ft->info.encoding = ST_ENCODING_MP3;
ft->info.size = ST_SIZE_WORD;
/* 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=st_readbuf(ft, p->InputBuffer, 1, INPUT_BUFFER_SIZE);
if(ReadSize<=0)
{
if(st_error(ft))
st_fail_errno(ft,ST_EOF,"read error on bitstream");
if(st_eof(ft))
st_fail_errno(ft,ST_EOF,"end of input stream");
return(ST_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 (st_mp3_input(ft) == ST_EOF)
return ST_EOF;
continue;
}
/* Consume any ID3 tags */
st_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)
{
st_fail_errno(ft,ST_EOF,"No valid MP3 frame found");
return ST_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->info.channels = MAD_NCHANNELS(&p->Frame->header);
break;
default:
st_fail_errno(ft, ST_EFMT, "Cannot determine number of channels");
return ST_EOF;
}
p->FrameCount=1;
mad_timer_add(p->Timer,p->Frame->header.duration);
mad_synth_frame(p->Synth,p->Frame);
ft->info.rate=p->Synth->pcm.samplerate;
p->cursamp = 0;
return ST_SUCCESS;
error:
if (p->Stream) free(p->Stream);
if (p->Frame) free(p->Frame);
if (p->Synth) free(p->Synth);
if (p->Timer) free(p->Timer);
if (p->InputBuffer) free(p->InputBuffer);
return ST_EOF;
}
/*
* 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.
*/
st_ssize_t st_mp3read(ft_t ft, st_sample_t *buf, st_ssize_t len)
{
struct mp3priv *p = (struct mp3priv *) ft->priv;
st_ssize_t donow,i,done=0;
mad_fixed_t sample;
int chan;
do {
donow=MIN(len,(p->Synth->pcm.length - p->cursamp)*ft->info.channels);
i=0;
while(i<donow){
for(chan=0;chan<ft->info.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++=(st_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 (st_mp3_input(ft) == ST_EOF)
return ST_EOF;
}
if (mad_frame_decode(p->Frame,p->Stream))
{
if(MAD_RECOVERABLE(p->Stream->error))
{
st_mp3_inputtag(ft);
continue;
}
else
{
if (p->Stream->error == MAD_ERROR_BUFLEN)
continue;
else
{
st_report("unrecoverable frame level error (%s).\n",
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;
}
int st_mp3stopread(ft_t ft)
{
struct mp3priv *p=(struct mp3priv*) ft->priv;
mad_synth_finish(p->Synth);
mad_frame_finish(p->Frame);
mad_stream_finish(p->Stream);
free(p->Stream);
free(p->Frame);
free(p->Synth);
free(p->Timer);
free(p->InputBuffer);
return ST_SUCCESS;
}
#else /*HAVE_LIBMAD*/
int st_mp3startread(ft_t ft)
{
st_fail_errno(ft,ST_EOF,"SoX was compiled without MP3 decoding support");
return ST_EOF;
}
st_ssize_t st_mp3read(ft_t ft, st_sample_t *buf, st_ssize_t samp)
{
st_fail_errno(ft,ST_EOF,"SoX was compiled without MP3 decoding support");
return ST_EOF;
}
int st_mp3stopread(ft_t ft)
{
st_fail_errno(ft,ST_EOF,"SoX was compiled without MP3 decoding support");
return ST_EOF;
}
#endif /*HAVE_LIBMAD*/
#if defined (HAVE_LAME)
void null_error_func(const char* string, va_list va){
return;
}
int st_mp3startwrite(ft_t ft)
{
struct mp3priv *p = (struct mp3priv *) ft->priv;
if (ft->info.encoding != ST_ENCODING_MP3){
if(ft->info.encoding != -1)
st_report("Encoding forced to MP3");
ft->info.encoding = ST_ENCODING_MP3;
}
p->gfp = lame_init();
if (p->gfp == NULL){
st_fail_errno(ft,ST_EOF,"Initialization of LAME library failed");
return(ST_EOF);
}
if (ft->info.channels != -1){
if ( (lame_set_num_channels(p->gfp,ft->info.channels)) < 0) {
st_fail_errno(ft,ST_EOF,"Unsupported number of channels");
return(ST_EOF);
}
}
else
ft->info.channels = lame_get_num_channels(p->gfp); /* LAME default */
lame_set_in_samplerate(p->gfp,ft->info.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->info.compression != HUGE_VAL)
st_warn("-C option not supported for mp3; using default compression rate");
if (lame_init_params(p->gfp) < 0){
st_fail_errno(ft,ST_EOF,"LAME initialization failed");
return(ST_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(ST_SUCCESS);
}
st_ssize_t st_mp3write(ft_t ft, st_sample_t *buf, st_ssize_t samp)
{
struct mp3priv *p = (struct mp3priv *)ft->priv;
char *mp3buffer;
int mp3buffer_size;
short signed int *buffer_l, *buffer_r = NULL;
int nsamples = samp/ft->info.channels;
int i,j;
st_ssize_t done = 0;
int 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
* malloc()'ing a smaller buffer and call a consistent
* interface.
*/
if ((buffer_l =
(short signed int *)malloc(nsamples*
sizeof(short signed int))) == NULL)
{
st_fail_errno(ft, ST_ENOMEM, "Memory allocation failed");
goto end4;
}
if (ft->info.channels == 2)
{
/* lame doesn't support iterleaved samples so we must break
* them out into seperate buffers.
*/
if ((buffer_r =
(short signed int *)malloc(nsamples*
sizeof(short signed int))) == NULL)
{
st_fail_errno(ft,ST_ENOMEM,"Memory allocation failed");
goto end3;
}
j=0;
for (i=0; i<nsamples; i++)
{
buffer_l[i]=ST_SAMPLE_TO_SIGNED_WORD(buf[j++]);
buffer_r[i]=ST_SAMPLE_TO_SIGNED_WORD(buf[j++]);
}
}
else
{
j=0;
for (i=0; i<nsamples; i++)
{
buffer_l[i]=ST_SAMPLE_TO_SIGNED_WORD(buf[j++]);
}
}
mp3buffer_size = 1.25 * nsamples + 7200;
if ((mp3buffer=(char *)malloc(mp3buffer_size)) == NULL)
{
st_fail_errno(ft,ST_ENOMEM,"Memory allocation failed");
goto end2;
}
if ((written = lame_encode_buffer(p->gfp,buffer_l, buffer_r,
nsamples, (unsigned char *)mp3buffer,
mp3buffer_size)) < 0){
st_fail_errno(ft,ST_EOF,"Encoding failed");
goto end;
}
if (st_writebuf(ft, mp3buffer, 1, written) < written)
{
st_fail_errno(ft,ST_EOF,"File write failed");
goto end;
}
done = nsamples*ft->info.channels;
end:
free(mp3buffer);
end2:
if (ft->info.channels == 2)
free(buffer_r);
end3:
free(buffer_l);
end4:
return done;
}
int st_mp3stopwrite(ft_t ft)
{
struct mp3priv *p = (struct mp3priv *) ft->priv;
char mp3buffer[7200];
int written;
if ( (written=lame_encode_flush(p->gfp, (unsigned char *)mp3buffer, 7200)) <0){
st_fail_errno(ft,ST_EOF,"Encoding failed");
}
else if (st_writebuf(ft, mp3buffer, 1, written) < written){
st_fail_errno(ft,ST_EOF,"File write failed");
}
lame_close(p->gfp);
return ST_SUCCESS;
}
#else /* HAVE_LAME */
int st_mp3startwrite(ft_t ft)
{
st_fail_errno(ft,ST_EOF,"Sorry, no MP3 encoding support");
return ST_EOF;
}
st_ssize_t st_mp3write(ft_t ft, st_sample_t *buf, st_ssize_t samp)
{
st_fail_errno(ft,ST_EOF,"Sorry, no MP3 encoding support");
return ST_EOF;
}
int st_mp3stopwrite(ft_t ft)
{
st_fail_errno(ft,ST_EOF,"Sorry, no MP3 encoding support");
return ST_EOF;
}
#endif /* HAVE_LAME */
/* MP3 */
static char *mp3names[] = {
"mp3",
"mp2",
NULL,
};
static st_format_t st_mp3_format = {
mp3names,
NULL,
ST_FILE_STEREO,
st_mp3startread,
st_mp3read,
st_mp3stopread,
st_mp3startwrite,
st_mp3write,
st_mp3stopwrite,
st_format_nothing_seek
};
const st_format_t *st_mp3_format_fn(void)
{
return &st_mp3_format;
}
#endif