ref: 0e1665104b189774fef617caa7355059a7306c67
dir: /src/opusdec.c/
/* Copyright (c) 2002-2007 Jean-Marc Valin
Copyright (c) 2008 CSIRO
Copyright (c) 2007-2012 Xiph.Org Foundation
File: opusdec.c
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <stdio.h>
#if !defined WIN32 && !defined _WIN32
#include <unistd.h>
#endif
#ifdef HAVE_GETOPT_H
#include <getopt.h>
#endif
/*#ifndef HAVE_GETOPT_LONG
#include "getopt_win.h"
#endif*/
#include <stdlib.h>
#include <limits.h>
#include <string.h>
#include <opus.h>
#include <opus_multistream.h>
#include <ogg/ogg.h>
#if defined WIN32 || defined _WIN32
#include "wave_out.h"
/* We need the following two to set stdout to binary */
#include <io.h>
#include <fcntl.h>
#endif
#include <math.h>
#ifdef HAVE_LRINTF
# define float2int(x) lrintf(x)
#else
# define float2int(flt) ((int)(floor(.5+flt)))
#endif
#ifdef _WIN32
#define I64FORMAT "I64d"
#else
#define I64FORMAT "lld"
#endif
#if defined HAVE_SYS_SOUNDCARD_H || defined HAVE_MACHINE_SOUNDCARD_H || HAVE_SOUNDCARD_H
#ifdef HAVE_SYS_SOUNDCARD_H
# include <sys/soundcard.h>
#elif HAVE_MACHINE_SOUNDCARD_H
# include <machine/soundcard.h>
#else
# include <soundcard.h>
#endif
#include <sys/types.h>
#include <sys/stat.h>
#include <fcntl.h>
#include <sys/ioctl.h>
#elif defined HAVE_SYS_AUDIOIO_H
#include <sys/types.h>
#include <fcntl.h>
#include <sys/ioctl.h>
#include <sys/audioio.h>
#ifndef AUDIO_ENCODING_SLINEAR
#define AUDIO_ENCODING_SLINEAR AUDIO_ENCODING_LINEAR /* Solaris */
#endif
#endif
#include <string.h>
#include "wav_io.h"
#include "opus_header.h"
#include "diag_range.h"
#include "speex_resampler.h"
#include "stack_alloc.h"
#define MINI(_a,_b) ((_a)<(_b)?(_a):(_b))
#define MAXI(_a,_b) ((_a)>(_b)?(_a):(_b))
#define CLAMPI(_a,_b,_c) (MAXI(_a,MINI(_b,_c)))
/* 120ms at 48000 */
#define MAX_FRAME_SIZE (960*6)
#define readint(buf, base) (((buf[base+3]<<24)&0xff000000)| \
((buf[base+2]<<16)&0xff0000)| \
((buf[base+1]<<8)&0xff00)| \
(buf[base]&0xff))
typedef struct shapestate shapestate;
struct shapestate {
float * b_buf;
float * a_buf;
int fs;
int mute;
};
static unsigned int rngseed = 22222;
static inline unsigned int fast_rand(void) {
rngseed = (rngseed * 96314165) + 907633515;
return rngseed;
}
#ifndef HAVE_FMINF
# define fminf(_x,_y) ((_x)<(_y)?(_x):(_y))
#endif
#ifndef HAVE_FMAXF
# define fmaxf(_x,_y) ((_x)>(_y)?(_x):(_y))
#endif
/* This implements a 16 bit quantization with full triangular dither
and IIR noise shaping. The noise shaping filters were designed by
Sebastian Gesemann based on the LAME ATH curves with flattening
to limit their peak gain to 20dB.
(Everyone elses' noise shaping filters are mildly crazy)
The 48kHz version of this filter is just a warped version of the
44.1kHz filter and probably could be improved by shifting the
HF shelf up in frequency a little bit since 48k has a bit more
room and being more conservative against bat-ears is probably
more important than more noise suppression.
This process can increase the peak level of the signal (in theory
by the peak error of 1.5 +20dB though this much is unobservable rare)
so to avoid clipping the signal is attenuated by a couple thousandths
of a dB. Initially the approach taken here was to only attenuate by
the 99.9th percentile, making clipping rare but not impossible (like
SoX) but the limited gain of the filter means that the worst case was
only two thousandths of a dB more, so this just uses the worst case.
The attenuation is probably also helpful to prevent clipping in the DAC
reconstruction filters or downstream resampling in any case.*/
static inline void shape_dither_toshort(shapestate *_ss, short *_o, float *_i, int _n, int _CC)
{
const float gains[3]={32768.f-15.f,32768.f-15.f,32768.f-3.f};
const float fcoef[3][8] =
{
{2.2374f, -.7339f, -.1251f, -.6033f, 0.9030f, .0116f, -.5853f, -.2571f}, /* 48.0kHz noise shaping filter sd=2.34*/
{2.2061f, -.4706f, -.2534f, -.6214f, 1.0587f, .0676f, -.6054f, -.2738f}, /* 44.1kHz noise shaping filter sd=2.51*/
{1.0000f, 0.0000f, 0.0000f, 0.0000f, 0.0000f,0.0000f, 0.0000f, 0.0000f}, /* lowpass noise shaping filter sd=0.65*/
};
int i;
int rate=_ss->fs==44100?1:(_ss->fs==48000?0:2);
float gain=gains[rate];
float *b_buf;
float *a_buf;
int mute=_ss->mute;
b_buf=_ss->b_buf;
a_buf=_ss->a_buf;
if(mute>64)
memset(a_buf,0,sizeof(float)*_CC*4);
for(i=0;i<_n;i++)
{
int c;
int pos = i*_CC;
int silent=1;
for(c=0;c<_CC;c++)
{
int j, si;
float r,s,err=0;
silent&=_i[pos+c]==0;
s=_i[pos+c]*gain;
for(j=0;j<4;j++)
err += fcoef[rate][j]*b_buf[c*4+j] - fcoef[rate][j+4]*a_buf[c*4+j];
memmove(&a_buf[c*4+1],&a_buf[c*4],sizeof(float)*3);
memmove(&b_buf[c*4+1],&b_buf[c*4],sizeof(float)*3);
a_buf[c*4]=err;
s = s - err;
r=(float)fast_rand()*(1/(float)UINT_MAX) - (float)fast_rand()*(1/(float)UINT_MAX);
if (mute>16)r=0;
/*Clamp in float out of paranoia that the input will be >96dBFS and wrap if the
integer is clamped.*/
_o[pos+c] = si = float2int(fmaxf(-32768,fminf(s + r,32767)));
/*Including clipping in the noise shaping is generally disastrous:
the futile effort to restore the clipped energy results in more clipping.
However, small amounts-- at the level which could normally be created by
dither and rounding-- are harmless and can even reduce clipping somewhat
due to the clipping sometimes reducing the dither+rounding error.*/
b_buf[c*4] = (mute>16)?0:fmaxf(-1.5f,fminf(si - s,1.5f));
}
mute++;
if(!silent)mute=0;
}
_ss->mute=MINI(mute,960);
}
static void print_comments(char *comments, int length)
{
char *c=comments;
int len, i, nb_fields, err=0;
if (length<(8+4+4))
{
fprintf (stderr, "Invalid/corrupted comments\n");
return;
}
if (strncmp(c, "OpusTags", 8) != 0)
{
fprintf (stderr, "Invalid/corrupted comments\n");
return;
}
c += 8;
fprintf(stderr, "Encoded with ");
len=readint(c, 0);
c+=4;
if (len < 0 || len>(length-16))
{
fprintf (stderr, "Invalid/corrupted comments\n");
return;
}
err&=fwrite(c, 1, len, stderr)!=(unsigned)len;
c+=len;
fprintf (stderr, "\n");
/*The -16 check above makes sure we can read this.*/
nb_fields=readint(c, 0);
c+=4;
length-=16+len;
if (nb_fields < 0 || nb_fields>(length>>2))
{
fprintf (stderr, "Invalid/corrupted comments\n");
return;
}
for (i=0;i<nb_fields;i++)
{
if (length<4)
{
fprintf (stderr, "Invalid/corrupted comments\n");
return;
}
len=readint(c, 0);
c+=4;
length-=4;
if (len < 0 || len>length)
{
fprintf (stderr, "Invalid/corrupted comments\n");
return;
}
err&=fwrite(c, 1, len, stderr)!=(unsigned)len;
c+=len;
length-=len;
fprintf (stderr, "\n");
}
}
FILE *out_file_open(char *outFile, int *wav_format, int rate, int mapping_family, int *channels)
{
FILE *fout=NULL;
/*Open output file*/
if (strlen(outFile)==0)
{
#if defined HAVE_SYS_SOUNDCARD_H
int audio_fd, format, stereo;
audio_fd=open("/dev/dsp", O_WRONLY);
if (audio_fd<0)
{
perror("Cannot open /dev/dsp");
exit(1);
}
format=AFMT_S16_NE;
if (ioctl(audio_fd, SNDCTL_DSP_SETFMT, &format)==-1)
{
perror("SNDCTL_DSP_SETFMT");
close(audio_fd);
exit(1);
}
stereo=0;
if (*channels==2)
stereo=1;
if (ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo)==-1)
{
perror("SNDCTL_DSP_STEREO");
close(audio_fd);
exit(1);
}
if (stereo!=0)
{
if (*channels==1)
fprintf (stderr, "Cannot set mono mode, will decode in stereo\n");
*channels=2;
}
if (ioctl(audio_fd, SNDCTL_DSP_SPEED, &rate)==-1)
{
perror("SNDCTL_DSP_SPEED");
close(audio_fd);
exit(1);
}
fout = fdopen(audio_fd, "w");
if(!fout)
{
perror("Cannot open output");
exit(1);
}
#elif defined HAVE_SYS_AUDIOIO_H
audio_info_t info;
int audio_fd;
audio_fd = open("/dev/audio", O_WRONLY);
if (audio_fd<0)
{
perror("Cannot open /dev/audio");
exit(1);
}
AUDIO_INITINFO(&info);
#ifdef AUMODE_PLAY /* NetBSD/OpenBSD */
info.mode = AUMODE_PLAY;
#endif
info.play.encoding = AUDIO_ENCODING_SLINEAR;
info.play.precision = 16;
info.play.input_sample_rate = rate;
info.play.channels = *channels;
if (ioctl(audio_fd, AUDIO_SETINFO, &info) < 0)
{
perror ("AUDIO_SETINFO");
exit(1);
}
fout = fdopen(audio_fd, "w");
if(!fout)
{
perror("Cannot open output");
exit(1);
}
#elif defined WIN32 || defined _WIN32
{
unsigned int opus_channels = *channels;
if (Set_WIN_Params (INVALID_FILEDESC, rate, SAMPLE_SIZE, opus_channels))
{
fprintf (stderr, "Can't access %s\n", "WAVE OUT");
exit(1);
}
}
#else
fprintf (stderr, "No soundcard support\n");
exit(1);
#endif
} else {
if (strcmp(outFile,"-")==0)
{
#if defined WIN32 || defined _WIN32
_setmode(_fileno(stdout), _O_BINARY);
#endif
fout=stdout;
}
else
{
fout = fopen(outFile, "wb");
if (!fout)
{
perror(outFile);
exit(1);
}
if (*wav_format)
{
*wav_format = write_wav_header(fout, rate, mapping_family, *channels);
if (*wav_format < 0)
{
fprintf (stderr, "Error writing WAV header.\n");
exit(1);
}
}
}
}
return fout;
}
void usage(void)
{
printf ("Usage: opusdec [options] input_file.opus [output_file]\n");
printf ("\n");
printf ("Decodes a Opus file and produce a WAV file or raw file\n");
printf ("\n");
printf ("input_file can be:\n");
printf (" filename.opus regular Opus file\n");
printf (" - stdin\n");
printf ("\n");
printf ("output_file can be:\n");
printf (" filename.wav Wav file\n");
printf (" filename.* Raw PCM file (any extension other than .wav)\n");
printf (" - stdout\n");
printf (" (nothing) Will be played to soundcard\n");
printf ("\n");
printf ("Options:\n");
printf (" --rate n Force decoding at sampling rate n Hz\n");
printf (" --no-dither Do not dither 16-bit output\n");
printf (" --packet-loss n Simulate n %% random packet loss\n");
printf (" --save-range file Saves check values for every frame to a file\n");
printf (" -h, --help This help\n");
printf (" -v, --version Version information\n");
printf (" --quiet Quiet mode\n");
printf ("\n");
}
void version(void)
{
printf("opusdec %s %s (using %s)\n",PACKAGE,VERSION,opus_get_version_string());
printf("Copyright (C) 2008-2012 Xiph.Org Foundation\n");
}
void version_short(void)
{
printf("opusdec %s %s (using %s)\n",PACKAGE,VERSION,opus_get_version_string());
printf("Copyright (C) 2008-2012 Xiph.Org Foundation\n");
}
static OpusMSDecoder *process_header(ogg_packet *op, opus_int32 *rate, int *mapping_family, int *channels, int *preskip, float *gain, int *streams, int wav_format, int quiet)
{
int err;
OpusMSDecoder *st;
OpusHeader header;
if (opus_header_parse(op->packet, op->bytes, &header)==0)
{
fprintf(stderr, "Cannot parse header\n");
return NULL;
}
*mapping_family = header.channel_mapping;
*channels = header.channels;
if(wav_format)adjust_wav_mapping(*mapping_family, *channels, header.stream_map);
if(!*rate)*rate=header.input_sample_rate;
/*If the rate is unspecified we decode to 48000*/
if(*rate==0)*rate=48000;
if(*rate<8000||*rate>192000){
fprintf(stderr,"Warning: Crazy input_rate %d, decoding to 48000 instead.\n",*rate);
*rate=48000;
}
*preskip = header.preskip;
st = opus_multistream_decoder_create(48000, header.channels, header.nb_streams, header.nb_coupled, header.stream_map, &err);
if(err != OPUS_OK){
fprintf(stderr, "Cannot create encoder: %s\n", opus_strerror(err));
return NULL;
}
if (!st)
{
fprintf (stderr, "Decoder initialization failed: %s\n", opus_strerror(err));
return NULL;
}
*streams=header.nb_streams;
if(header.gain!=0)
{
/*Gain API added in a newer libopus version..*/
#ifdef OPUS_SET_GAIN
err=opus_multistream_decoder_ctl(st,OPUS_SET_GAIN(header.gain));
if(err==OPUS_UNIMPLEMENTED)
{
#endif
*gain = pow(10., header.gain/5120.);
#ifdef OPUS_SET_GAIN
} else if (err!=OPUS_OK)
{
fprintf (stderr, "Error setting gain: %s\n", opus_strerror(err));
return NULL;
}
#endif
}
if (!quiet)
{
fprintf(stderr, "Decoding %d Hz %saudio", *rate, *rate!=(int)header.input_sample_rate?"(forced) ":"");
fprintf(stderr, " (%d channel%s)",*channels,*channels>1?"s":"");
if(header.version!=1)fprintf(stderr, ", Header v%d",header.version);
fprintf(stderr, "\n");
if (header.gain!=0)printf("Playback gain: %f (%f dB)\n", *gain, header.gain/256.);
}
return st;
}
opus_int64 audio_write(float *pcm, int channels, int frame_size, FILE *fout, SpeexResamplerState *resampler,
int *skip, shapestate *shapemem, int file, opus_int64 maxout)
{
opus_int64 sampout=0;
int i,ret,tmp_skip;
unsigned out_len;
short *out;
float *buf;
float *output;
out=alloca(sizeof(short)*MAX_FRAME_SIZE*channels);
buf=alloca(sizeof(float)*MAX_FRAME_SIZE*channels);
maxout=maxout<0?0:maxout;
do {
if (skip){
tmp_skip = (*skip>frame_size) ? (int)frame_size : *skip;
*skip -= tmp_skip;
} else {
tmp_skip = 0;
}
if (resampler){
unsigned in_len;
output=buf;
in_len = frame_size-tmp_skip;
out_len = 1024<maxout?1024:maxout;
speex_resampler_process_interleaved_float(resampler, pcm+channels*tmp_skip, &in_len, buf, &out_len);
pcm += channels*(in_len+tmp_skip);
frame_size -= in_len+tmp_skip;
} else {
output=pcm+channels*tmp_skip;
out_len=frame_size-tmp_skip;
frame_size=0;
}
/*Convert to short and save to output file*/
if (shapemem){
shape_dither_toshort(shapemem,out,output,out_len,channels);
}else{
for (i=0;i<(int)out_len*channels;i++)
out[i]=(short)float2int(fmaxf(-32768,fminf(output[i]*32768.f,32767)));
}
if ((le_short(1)!=1)&&file){
for (i=0;i<(int)out_len*channels;i++)
out[i]=le_short(out[i]);
}
if(maxout>0)
{
#if defined WIN32 || defined _WIN32
if(!file){
ret=WIN_Play_Samples (out, sizeof(short) * channels * (out_len<maxout?out_len:maxout));
if(ret>0)ret/=sizeof(short)*channels;
else fprintf(stderr, "Error playing audio.\n");
}else
#endif
ret=fwrite(out, 2*channels, out_len<maxout?out_len:maxout, fout);
sampout+=ret;
maxout-=ret;
}
} while (frame_size>0 && maxout>0);
return sampout;
}
int main(int argc, char **argv)
{
int c;
int option_index = 0;
char *inFile, *outFile;
FILE *fin, *fout=NULL, *frange=NULL;
float *output;
int frame_size=0;
OpusMSDecoder *st=NULL;
opus_int64 packet_count=0;
int total_links=0;
int stream_init = 0;
int quiet = 0;
ogg_int64_t page_granule=0;
ogg_int64_t link_out=0;
struct option long_options[] =
{
{"help", no_argument, NULL, 0},
{"quiet", no_argument, NULL, 0},
{"version", no_argument, NULL, 0},
{"version-short", no_argument, NULL, 0},
{"rate", required_argument, NULL, 0},
{"no-dither", no_argument, NULL, 0},
{"packet-loss", required_argument, NULL, 0},
{"save-range", required_argument, NULL, 0},
{0, 0, 0, 0}
};
ogg_sync_state oy;
ogg_page og;
ogg_packet op;
ogg_stream_state os;
int close_in=0;
int eos=0;
ogg_int64_t audio_size=0;
float loss_percent=-1;
int channels=-1;
int mapping_family;
int rate=0;
int wav_format=0;
int preskip=0;
int gran_offset=0;
int has_opus_stream=0;
ogg_int32_t opus_serialno;
int dither=1;
shapestate shapemem;
SpeexResamplerState *resampler=NULL;
float gain=1;
int streams=0;
output=0;
shapemem.a_buf=0;
shapemem.b_buf=0;
shapemem.mute=960;
shapemem.fs=0;
/*Process options*/
while(1)
{
c = getopt_long (argc, argv, "hvV",
long_options, &option_index);
if (c==-1)
break;
switch(c)
{
case 0:
if (strcmp(long_options[option_index].name,"help")==0)
{
usage();
exit(0);
} else if (strcmp(long_options[option_index].name,"quiet")==0)
{
quiet = 1;
} else if (strcmp(long_options[option_index].name,"version")==0)
{
version();
exit(0);
} else if (strcmp(long_options[option_index].name,"version-short")==0)
{
version_short();
exit(0);
} else if (strcmp(long_options[option_index].name,"no-dither")==0)
{
dither=0;
} else if (strcmp(long_options[option_index].name,"rate")==0)
{
rate=atoi (optarg);
}else if(strcmp(long_options[option_index].name,"save-range")==0){
frange=fopen(optarg,"w");
if(frange==NULL){
perror(optarg);
fprintf(stderr,"Could not open save-range file: %s\n",optarg);
fprintf(stderr,"Must provide a writable file name.\n");
exit(1);
}
} else if (strcmp(long_options[option_index].name,"packet-loss")==0)
{
loss_percent = atof(optarg);
}
break;
case 'h':
usage();
exit(0);
break;
case 'v':
version();
exit(0);
break;
case '?':
usage();
exit(1);
break;
}
}
if (argc-optind!=2 && argc-optind!=1)
{
usage();
exit(1);
}
inFile=argv[optind];
if (argc-optind==2)
outFile=argv[optind+1];
else
outFile = "";
wav_format = strlen(outFile)>=4 && (
strcmp(outFile+strlen(outFile)-4,".wav")==0
|| strcmp(outFile+strlen(outFile)-4,".WAV")==0);
/*Open input file*/
if (strcmp(inFile, "-")==0)
{
#if defined WIN32 || defined _WIN32
_setmode(_fileno(stdin), _O_BINARY);
#endif
fin=stdin;
}
else
{
fin = fopen(inFile, "rb");
if (!fin)
{
perror(inFile);
exit(1);
}
close_in=1;
}
/*Init Ogg data struct*/
ogg_sync_init(&oy);
/*Main decoding loop*/
while (1)
{
char *data;
int i, nb_read;
/*Get the ogg buffer for writing*/
data = ogg_sync_buffer(&oy, 200);
/*Read bitstream from input file*/
nb_read = fread(data, sizeof(char), 200, fin);
ogg_sync_wrote(&oy, nb_read);
/*Loop for all complete pages we got (most likely only one)*/
while (ogg_sync_pageout(&oy, &og)==1)
{
if (stream_init == 0) {
ogg_stream_init(&os, ogg_page_serialno(&og));
stream_init = 1;
}
if (ogg_page_serialno(&og) != os.serialno) {
/* so all streams are read. */
ogg_stream_reset_serialno(&os, ogg_page_serialno(&og));
}
/*Add page to the bitstream*/
ogg_stream_pagein(&os, &og);
page_granule = ogg_page_granulepos(&og);
/*Extract all available packets*/
while (ogg_stream_packetout(&os, &op) == 1)
{
if (op.b_o_s && op.bytes>=8 && !memcmp(op.packet, "OpusHead", 8)) {
if(!has_opus_stream)
{
opus_serialno = os.serialno;
has_opus_stream = 1;
link_out = 0;
packet_count = 0;
eos = 0;
total_links++;
} else {
fprintf(stderr,"Warning: ignoring opus stream %" I64FORMAT "\n",(long long)os.serialno);
}
}
if (!has_opus_stream || os.serialno != opus_serialno)
break;
/*If first packet, process as OPUS header*/
if (packet_count==0)
{
st = process_header(&op, &rate, &mapping_family, &channels, &preskip, &gain, &streams, wav_format, quiet);
if (!st)
exit(1);
gran_offset=preskip;
if(!shapemem.a_buf)
{
shapemem.a_buf=calloc(channels,sizeof(float)*4);
shapemem.b_buf=calloc(channels,sizeof(float)*4);
shapemem.fs=rate;
}
if(!output)output=malloc(sizeof(float)*MAX_FRAME_SIZE*channels);
if (rate != 48000)
{
int err;
resampler = speex_resampler_init(channels, 48000, rate, 5, &err);
if (err!=0)
fprintf(stderr, "resampler error: %s\n", speex_resampler_strerror(err));
speex_resampler_skip_zeros(resampler);
}
if(!fout)fout=out_file_open(outFile, &wav_format, rate, mapping_family, &channels);
} else if (packet_count==1)
{
if (!quiet)
print_comments((char*)op.packet, op.bytes);
} else {
int ret;
opus_int64 maxout;
opus_int64 outsamp;
int lost=0;
if (loss_percent>0 && 100*((float)rand())/RAND_MAX<loss_percent)
lost=1;
/*End of stream condition*/
if (op.e_o_s && os.serialno == opus_serialno)eos=1; /* don't care for anything except opus eos */
/*Decode frame*/
if (!lost)
ret = opus_multistream_decode_float(st, (unsigned char*)op.packet, op.bytes, output, MAX_FRAME_SIZE, 0);
else
ret = opus_multistream_decode_float(st, NULL, 0, output, MAX_FRAME_SIZE, 0);
if (ret<0)
{
fprintf (stderr, "Decoding error: %s\n", opus_strerror(ret));
break;
}
frame_size = ret;
if(frange!=NULL){
OpusDecoder *od;
opus_uint32 rngs[256];
for(i=0;i<streams;i++){
ret=opus_multistream_decoder_ctl(st,OPUS_MULTISTREAM_GET_DECODER_STATE(i,&od));
ret=opus_decoder_ctl(od,OPUS_GET_FINAL_RANGE(&rngs[i]));
}
save_range(frange,frame_size*(48000/48000/*decoding_rate*/),op.packet,op.bytes,
rngs,streams);
}
/* Apply header gain */
for (i=0;i<frame_size*channels;i++)
output[i] *= gain;
maxout=((page_granule-gran_offset)*rate/48000)-link_out;
outsamp=audio_write(output, channels, frame_size, fout, resampler, &preskip, dither?&shapemem:0, strlen(outFile)!=0,0>maxout?0:maxout);
link_out+=outsamp;
audio_size+=sizeof(short)*outsamp*channels;
}
packet_count++;
}
/* Drain the resampler */
if(eos && resampler)
{
float *zeros;
int drain;
zeros=(float *)calloc(100*channels,sizeof(float));
drain = speex_resampler_get_input_latency(resampler);
do {
opus_int64 outsamp;
int tmp = drain;
if (tmp > 100)
tmp = 100;
outsamp=audio_write(zeros, channels, tmp, fout, resampler, NULL, &shapemem, strlen(outFile)==0, ((page_granule-gran_offset)*rate/48000)-link_out);
link_out+=outsamp;
audio_size+=sizeof(short)*outsamp*channels;
drain -= tmp;
} while (drain>0);
free(zeros);
speex_resampler_destroy(resampler);
resampler=NULL;
}
if(eos)
{
has_opus_stream=0;
if(st)opus_multistream_decoder_destroy(st);
st=NULL;
}
}
if (feof(fin))
break;
}
if (fout && wav_format>=0 && audio_size<0x7FFFFFFF)
{
if (fseek(fout,4,SEEK_SET)==0)
{
int tmp;
tmp = le_int(audio_size+20+wav_format);
if(fwrite(&tmp,4,1,fout)!=1)fprintf(stderr,"Error writing end length.\n");
if (fseek(fout,16+wav_format,SEEK_CUR)==0)
{
tmp = le_int(audio_size);
if(fwrite(&tmp,4,1,fout)!=1)fprintf(stderr,"Error writing header length.\n");
} else
{
fprintf (stderr, "First seek worked, second didn't\n");
}
} else {
fprintf (stderr, "Cannot seek on wave file, size will be incorrect\n");
}
}
if(!total_links)fprintf (stderr, "This doesn't look like a Opus file\n");
if (stream_init)
ogg_stream_clear(&os);
ogg_sync_clear(&oy);
#if defined WIN32 || defined _WIN32
if (strlen(outFile)==0)
WIN_Audio_close ();
#endif
if(shapemem.a_buf)free(shapemem.a_buf);
if(shapemem.b_buf)free(shapemem.b_buf);
if(output)free(output);
if(frange)fclose(frange);
if (close_in)
fclose(fin);
if (fout != NULL)
fclose(fout);
return 0;
}