ref: dfb4b522830edab8f3856289d326d6cf5e930644
dir: /sys/src/cmd/audio/mp3enc/vorbis_interface.c/
/* $Id: vorbis_interface.c,v 1.16 2001/03/18 14:31:29 aleidinger Exp $ */
/* Compile lame with
#! /bin/bash
OGGVORBIS_ROOT=/home/cvs/vorbis
export CPPFLAGS="-I${OGGVORBIS_ROOT}/ogg/include -I${OGGVORBIS_ROOT}/vorbis/lib"
export CONFIG_DEFS="-DUSE_FFT3DN"
make clean
../configure
make
You can also do this with the "--with-vorbis" options in configure.
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
/* LAME interface to libvorbis */
#ifdef HAVE_VORBIS
#include <stdlib.h>
#include <limits.h>
#include <time.h>
#include "vorbis/codec.h"
#include "modes/modes.h"
#include "lame.h"
#include "util.h"
#ifdef WITH_DMALLOC
#include <dmalloc.h>
#endif
short int convbuffer [4096]; /* take 8 KByte out of the data segment, not the stack */
int convsize;
ogg_sync_state oy; // sync and verify incoming physical bitstream
ogg_stream_state os; // take physical pages, weld into a logical stream of packets
ogg_page og; // one Ogg bitstream page. Vorbis packets are inside
ogg_packet op; // one raw packet of data for decode
vorbis_info vi; // struct that stores all the static vorbis bitstream settings
vorbis_comment vc; // struct that stores all the bitstream user comments
vorbis_dsp_state vd; // central working state for the packet->PCM decoder
vorbis_block vb; // local working space for packet->PCM decode
int lame_decode_ogg_initfile( lame_global_flags* gfp,
FILE* fd,
mp3data_struct* mp3data )
{
lame_internal_flags *gfc = gfp->internal_flags;
char *buffer;
int bytes;
int i;
/********** Decode setup ************/
ogg_sync_init(&oy); /* Now we can read pages */
/* grab some data at the head of the stream. We want the first page
(which is guaranteed to be small and only contain the Vorbis
stream initial header) We need the first page to get the stream
serialno. */
/* submit a 4k block to libvorbis' Ogg layer */
buffer=ogg_sync_buffer(&oy,4096);
bytes=fread(buffer,1,4096,fd);
ogg_sync_wrote(&oy,bytes);
/* Get the first page. */
if(ogg_sync_pageout(&oy,&og)!=1){
/* error case. Must not be Vorbis data */
ERRORF( gfc, "Error initializing Ogg bitstream data.\n" );
return -1;
}
/* Get the serial number and set up the rest of decode. */
/* serialno first; use it to set up a logical stream */
ogg_stream_init(&os,ogg_page_serialno(&og));
/* extract the initial header from the first page and verify that the
Ogg bitstream is in fact Vorbis data */
/* I handle the initial header first instead of just having the code
read all three Vorbis headers at once because reading the initial
header is an easy way to identify a Vorbis bitstream and it's
useful to see that functionality seperated out. */
vorbis_info_init(&vi);
vorbis_comment_init(&vc);
if(ogg_stream_pagein(&os,&og)<0){
/* error; stream version mismatch perhaps */
ERRORF( gfc, "Error reading first page of Ogg bitstream data.\n" );
return -1;
}
if(ogg_stream_packetout(&os,&op)!=1){
/* no page? must not be vorbis */
ERRORF( gfc, "Error reading initial header packet.\n" );
return -1;
}
if(vorbis_synthesis_headerin(&vi,&vc,&op)<0){
/* error case; not a vorbis header */
ERRORF( gfc, "This Ogg bitstream does not contain Vorbis "
"audio data.\n");
return -1;
}
/* At this point, we're sure we're Vorbis. We've set up the logical
(Ogg) bitstream decoder. Get the comment and codebook headers and
set up the Vorbis decoder */
/* The next two packets in order are the comment and codebook headers.
They're likely large and may span multiple pages. Thus we reead
and submit data until we get our two pacakets, watching that no
pages are missing. If a page is missing, error out; losing a
header page is the only place where missing data is fatal. */
i=0;
while(i<2){
while(i<2){
int result=ogg_sync_pageout(&oy,&og);
if(result==0)break; /* Need more data */
/* Don't complain about missing or corrupt data yet. We'll
catch it at the packet output phase */
if(result==1){
ogg_stream_pagein(&os,&og); /* we can ignore any errors here
as they'll also become apparent
at packetout */
while(i<2){
result=ogg_stream_packetout(&os,&op);
if(result==0)break;
if(result==-1){
/* Uh oh; data at some point was corrupted or missing!
We can't tolerate that in a header. Die. */
ERRORF( gfc, "Corrupt secondary header. Exiting.\n" );
return -1;
}
vorbis_synthesis_headerin(&vi,&vc,&op);
i++;
}
}
}
/* no harm in not checking before adding more */
buffer=ogg_sync_buffer(&oy,4096);
bytes=fread(buffer,1,4096,fd);
if(bytes==0 && i<2){
ERRORF( gfc, "End of file before finding all Vorbis headers!\n" );
return -1;
}
ogg_sync_wrote(&oy,bytes);
}
/* Throw the comments plus a few lines about the bitstream we're
decoding */
{
/*
char **ptr=vc.user_comments;
while(*ptr){
MSGF( gfc, "%s\n", *ptr );
++ptr;
}
MSGF( gfc, "\nBitstream is %d channel, %ldHz\n", vi.channels, vi.rate );
MSGF( gfc, "Encoded by: %s\n\n", vc.vendor );
*/
}
/* OK, got and parsed all three headers. Initialize the Vorbis
packet->PCM decoder. */
vorbis_synthesis_init(&vd,&vi); /* central decode state */
vorbis_block_init(&vd,&vb); /* local state for most of the decode
so multiple block decodes can
proceed in parallel. We could init
multiple vorbis_block structures
for vd here */
mp3data->stereo = vi.channels;
mp3data->samplerate = vi.rate;
mp3data->bitrate = 0; //ov_bitrate_instant(&vf);
mp3data->nsamp=MAX_U_32_NUM;
return 0;
}
/*
For lame_decode_fromfile: return code
-1 error, or eof
0 ok, but need more data before outputing any samples
n number of samples output.
*/
int lame_decode_ogg_fromfile( lame_global_flags* gfp,
FILE* fd,
short int pcm_l[],
short int pcm_r[],
mp3data_struct* mp3data )
{
lame_internal_flags *gfc = gfp->internal_flags;
int samples,result,i,j,eof=0,eos=0,bout=0;
double **pcm;
while(1){
/*
**pcm is a multichannel double vector. In stereo, for
example, pcm[0] is left, and pcm[1] is right. samples is
the size of each channel. Convert the float values
(-1.<=range<=1.) to whatever PCM format and write it out */
/* unpack the buffer, if it has at least 1024 samples */
convsize=1024;
samples=vorbis_synthesis_pcmout(&vd,&pcm);
if (samples >= convsize || eos || eof) {
/* read 1024 samples, or if eos, read what ever is in buffer */
int clipflag=0;
bout=(samples<convsize?samples:convsize);
/* convert doubles to 16 bit signed ints (host order) and
interleave */
for(i=0;i<vi.channels;i++){
double *mono=pcm[i];
for(j=0;j<bout;j++){
int val=mono[j]*32767.;
/* might as well guard against clipping */
if(val>32767){
val=32767;
clipflag=1;
}
if(val<-32768){
val=-32768;
clipflag=1;
}
if (i==0) pcm_l[j]=val;
if (i==1) pcm_r[j]=val;
}
}
/*
if(clipflag)
MSGF( gfc, "Clipping in frame %ld\n", vd.sequence );
*/
/* tell libvorbis how many samples we actually consumed */
vorbis_synthesis_read(&vd,bout);
break;
}
result=ogg_sync_pageout(&oy,&og);
if(result==0) {
/* need more data */
}else if (result==-1){ /* missing or corrupt data at this page position */
ERRORF( gfc, "Corrupt or missing data in bitstream; "
"continuing...\n");
}else{
/* decode this page */
ogg_stream_pagein(&os,&og); /* can safely ignore errors at
this point */
do {
result=ogg_stream_packetout(&os,&op);
if(result==0) {
/* need more data */
} else if(result==-1){ /* missing or corrupt data at this page position */
/* no reason to complain; already complained above */
}else{
/* we have a packet. Decode it */
vorbis_synthesis(&vb,&op);
vorbis_synthesis_blockin(&vd,&vb);
}
} while (result!=0);
}
/* is this the last page? */
if(ogg_page_eos(&og))eos=1;
if(!eos){
char *buffer;
int bytes;
buffer=ogg_sync_buffer(&oy,4096);
bytes=fread(buffer,1,4096,fd);
ogg_sync_wrote(&oy,bytes);
if(bytes==0)eof=1;
}
}
mp3data->stereo = vi.channels;
mp3data->samplerate = vi.rate;
mp3data->bitrate = 0; //ov_bitrate_instant(&vf);
/* mp3data->nsamp=MAX_U_32_NUM;*/
if (bout==0) {
/* clean up this logical bitstream; before exit we see if we're
followed by another [chained] */
ogg_stream_clear(&os);
/* ogg_page and ogg_packet structs always point to storage in
libvorbis. They're never freed or manipulated directly */
vorbis_block_clear(&vb);
vorbis_dsp_clear(&vd);
vorbis_info_clear(&vi); /* must be called last */
/* OK, clean up the framer */
ogg_sync_clear(&oy);
return -1;
}
return bout;
}
ogg_stream_state os2; // take physical pages, weld into a logical stream of packets
ogg_page og2; // one Ogg bitstream page. Vorbis packets are inside
ogg_packet op2; // one raw packet of data for decode
vorbis_info vi2; // struct that stores all the static vorbis bitstream settings
vorbis_comment vc2; // struct that stores all the bitstream user comments
vorbis_dsp_state vd2; // central working state for the packet->PCM decoder
vorbis_block vb2; // local working space for packet->PCM decode
#define MAX_COMMENT_LENGTH 255
int lame_encode_ogg_init(lame_global_flags *gfp)
{
lame_internal_flags *gfc=gfp->internal_flags;
char comment[MAX_COMMENT_LENGTH+1];
/********** Encode setup ************/
/* choose an encoding mode */
/* (mode 0: 44kHz stereo uncoupled, roughly 128kbps VBR) */
if (gfp->compression_ratio < 5.01) {
memcpy(&vi2,&info_E,sizeof(vi2));
MSGF( gfc, "Encoding with Vorbis mode info_E \n" );
} else if (gfp->compression_ratio < 6) {
memcpy(&vi2,&info_D,sizeof(vi2));
MSGF( gfc, "Encoding with Vorbis mode info_D \n" );
} else if (gfp->compression_ratio < 8) {
memcpy(&vi2,&info_C,sizeof(vi2));
MSGF( gfc, "Encoding with Vorbis mode info_C \n" );
} else if (gfp->compression_ratio < 10) {
memcpy(&vi2,&info_B,sizeof(vi2));
MSGF( gfc, "Encoding with Vorbis mode info_B \n" );
} else if (gfp->compression_ratio < 12) {
memcpy(&vi2,&info_A,sizeof(vi2));
MSGF( gfc, "Encoding with Vorbis mode info_A \n" );
} else {
memcpy(&vi2,&info_A,sizeof(vi2));
MSGF( gfc, "Encoding with Vorbis mode info_A \n" );
}
vi2.channels = gfc->channels_out;
vi2.rate = gfp->out_samplerate;
/* add a comment */
vorbis_comment_init(&vc2);
vorbis_comment_add(&vc2,"Track encoded using L.A.M.E. libvorbis interface.");
/* Add ID3-style comments to the output using (for the time being) the
"private data members" in the "id3tag_spec" data structure. This was
from a patch by Ralph Giles <giles@a3a32260.sympatico.bconnected.net> */
#ifdef THIS_CODE_IS_NOT_BROKEN_ANYMORE
if(gfp->tag_spec.title) {
strcpy(comment,"TITLE=");
strncat(comment,gfp->tag_spec.title,MAX_COMMENT_LENGTH-strlen(comment));
vorbis_comment_add(&vc2,comment);
}
if(gfp->tag_spec.artist) {
strcpy(comment,"ARTIST=");
strncat(comment,gfp->tag_spec.artist,MAX_COMMENT_LENGTH-strlen(comment));
vorbis_comment_add(&vc2,comment);
}
if(gfp->tag_spec.album) {
strcpy(comment,"ALBUM=");
strncat(comment,gfp->tag_spec.album,MAX_COMMENT_LENGTH-strlen(comment));
vorbis_comment_add(&vc2,comment);
}
/* pretend that the ID3 fields are equivalent to the Vorbis fields */
if(gfp->tag_spec.year) {
sprintf(comment, "DATE=%d", gfp->tag_spec.year);
vorbis_comment_add(&vc2,comment);
}
if(gfp->tag_spec.comment) {
strcpy(comment,"DESCRIPTION=");
strncat(comment,gfp->tag_spec.comment,MAX_COMMENT_LENGTH-strlen(comment));
vorbis_comment_add(&vc2,comment);
}
/* TODO -- support for track and genre */
#endif
/* set up the analysis state and auxiliary encoding storage */
vorbis_analysis_init(&vd2,&vi2);
vorbis_block_init(&vd2,&vb2);
/* set up our packet->stream encoder */
/* pick a random serial number; that way we can more likely build
chained streams just by concatenation */
srand(time(NULL));
ogg_stream_init(&os2,rand());
/* Vorbis streams begin with three headers; the initial header (with
most of the codec setup parameters) which is mandated by the Ogg
bitstream spec. The second header holds any comment fields. The
third header holds the bitstream codebook. We merely need to
make the headers, then pass them to libvorbis one at a time;
libvorbis handles the additional Ogg bitstream constraints */
{
ogg_packet header;
ogg_packet header_comm;
ogg_packet header_code;
vorbis_analysis_headerout(&vd2,&vc2,&header,&header_comm,&header_code);
ogg_stream_packetin(&os2,&header); /* automatically placed in its own
page */
ogg_stream_packetin(&os2,&header_comm);
ogg_stream_packetin(&os2,&header_code);
/* no need to write out here. We'll get to that in the main loop */
}
return 0;
}
int lame_encode_ogg_finish(lame_global_flags *gfp,
char *mp3buf, int mp3buf_size)
{
int eos=0,bytes=0;
vorbis_analysis_wrote(&vd2,0);
while(vorbis_analysis_blockout(&vd2,&vb2)==1){
/* analysis */
vorbis_analysis(&vb2,&op2);
/* weld the packet into the bitstream */
ogg_stream_packetin(&os2,&op2);
/* write out pages (if any) */
while(!eos){
int result=ogg_stream_pageout(&os2,&og2);
if(result==0)break;
/* check if mp3buffer is big enough for the output */
bytes += og2.header_len + og2.body_len;
if (bytes > mp3buf_size && mp3buf_size>0)
return -5;
memcpy(mp3buf,og2.header,og2.header_len);
memcpy(mp3buf+og2.header_len,og2.body,og2.body_len);
/* this could be set above, but for illustrative purposes, I do
it here (to show that vorbis does know where the stream ends) */
if(ogg_page_eos(&og2))eos=1;
}
}
/* clean up and exit. vorbis_info_clear() must be called last */
ogg_stream_clear(&os2);
vorbis_block_clear(&vb2);
vorbis_dsp_clear(&vd2);
/* ogg_page and ogg_packet structs always point to storage in
libvorbis. They're never freed or manipulated directly */
return bytes;
}
int lame_encode_ogg_frame (
lame_global_flags* gfp,
const sample_t* inbuf_l,
const sample_t* inbuf_r,
unsigned char* mp3buf,
size_t mp3buf_size )
{
lame_internal_flags *gfc = gfp->internal_flags;
int i;
int eos = 0;
int bytes = 0;
/* expose the buffer to submit data */
double **buffer = vorbis_analysis_buffer(&vd2,gfp->framesize);
/* change level of input by -90 dB (no de-interleaving!) */
for ( i = 0; i < gfp->framesize; i++ )
buffer [0] [i] = (1/32768.) * inbuf_l [i];
if ( gfc->channels_out == 2 )
for ( i = 0; i < gfp->framesize; i++ )
buffer [1] [i] = (1/32768.) * inbuf_r [i];
/* tell the library how much we actually submitted */
vorbis_analysis_wrote(&vd2,i);
/* vorbis does some data preanalysis, then divvies up blocks for
more involved (potentially parallel) processing. Get a single
block for encoding now */
while(vorbis_analysis_blockout(&vd2,&vb2)==1){
int result;
/* analysis */
vorbis_analysis(&vb2,&op2);
/* weld the packet into the bitstream */
ogg_stream_packetin(&os2,&op2);
/* write out pages (if any) */
do {
result=ogg_stream_pageout(&os2,&og2);
if (result==0) break;
/* check if mp3buffer is big enough for the output */
bytes += og2.header_len + og2.body_len;
/*
DEBUGF("\n\n*********\ndecoded bytes=%i %i \n",bytes,mp3buf_size);
*/
if (bytes > mp3buf_size && mp3buf_size>0)
return -6;
memcpy(mp3buf,og2.header,og2.header_len);
memcpy(mp3buf+og2.header_len,og2.body,og2.body_len);
mp3buf += og2.header_len + og2.body_len;
if(ogg_page_eos(&og2))eos=1;
} while (1);
}
(gfp -> frameNum)++;
return bytes;
}
#endif
/* end of vorbis_interface.c */