ref: 36cc2a77dc286bd596f59166f6737ffb4a5e2e85
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 "sox_i.h"
#include <string.h>
#ifdef HAVE_MAD_H
#include <mad.h>
#endif
#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 {
#ifdef HAVE_MAD_H
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;
#endif /*HAVE_MAD_H*/
#ifdef HAVE_LAME_LAME_H
lame_global_flags *gfp;
#endif /*HAVE_LAME_LAME_H*/
} priv_t;
#ifdef HAVE_MAD_H
/* 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 = (priv_t *) 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 = lsx_readbuf(ft, 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 = (priv_t *) ft->priv;
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 = (priv_t *) ft->priv;
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 = (priv_t *) ft->priv;
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=(priv_t*) ft->priv;
mad_synth_finish(&p->Synth);
mad_frame_finish(&p->Frame);
mad_stream_finish(&p->Stream);
free(p->InputBuffer);
return SOX_SUCCESS;
}
#else /*HAVE_MAD_H*/
static int startread(sox_format_t * ft)
{
lsx_fail_errno(ft,SOX_EOF,"SoX was compiled without MP3 decoding support");
return SOX_EOF;
}
#define sox_mp3read NULL
#define stopread NULL
#endif /*HAVE_MAD_H*/
#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 = (priv_t *) ft->priv;
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 = (priv_t *)ft->priv;
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;
SOX_SAMPLE_LOCALS;
/* 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 iterleaved samples so we must break
* them out into seperate 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 = (priv_t *) ft->priv;
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;
}