/*

 * MP3/MPlayer plugin to VDR (C++)

 *

 * (C) 2001-2007 Stefan Huelswitt <s.huelswitt@gmx.de>

 *

 * This code is free software; you can redistribute it and/or

 * modify it under the terms of the GNU General Public License

 * as published by the Free Software Foundation; either version 2

 * of the License, or (at your option) any later version.

 *

 * This code is distributed in the hope that it will be useful,

 * but WITHOUT ANY WARRANTY; without even the implied warranty of

 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the

 * GNU General Public License for more details.

 *

 * You should have received a copy of the GNU General Public License

 * along with this program; if not, write to the Free Software

 * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.

 * Or, point your browser to http://www.gnu.org/copyleft/gpl.html

 */

#include <stdlib.h>

#include <stdio.h>

#include <sys/ioctl.h>

#include <math.h>

#ifdef WITH_OSS

#include <sys/soundcard.h>

#endif

#include <mad.h>

#include <id3tag.h>

#include <vdr/player.h>

#include <vdr/ringbuffer.h>

#include <vdr/thread.h>

#include <vdr/tools.h>

#include "common.h"

#include "setup-mp3.h"

#include "player-mp3.h"

#include "data-mp3.h"

#include "decoder.h"

#include "decoder-core.h"

#ifndef NO_DEBUG

//#define DEBUG_MODE      // debug playmode changes

#define DEBUG_BGR       // debug backround scan thread

#define DEBUG_DELAY 300 // debug write/decode delays

//#define ACC_DUMP        // dump limiter lookup table to /tmp/limiter

#endif

#if !defined(NO_DEBUG) && defined(DEBUG_MODE)

#define dm(x) { (x); }

#else

#define dm(x) ; 

#endif

#if !defined(NO_DEBUG) && defined(DEBUG_BGR)

#define db(x) { (x); }

#else

#define db(x) ; 

#endif

// ----------------------------------------------------------------

#define MP3BUFSIZE (1024*1024)                               // output ringbuffer size

#define OUT_BITS 16                                          // output 16 bit samples to DVB driver

#define OUT_FACT (OUT_BITS/8*2)                              // output factor is 16 bit & 2 channels -> 4 bytes

// cResample

#define MAX_NSAMPLES (1152*7)                                // max. buffer for resampled frame

// cNormalize

#define MIN_GAIN   0.03                                      // min. gain required to launch the normalizer

#define MAX_GAIN   3.0                                       // max. allowed gain

#define USE_FAST_LIMITER

#define LIM_ACC    12                                        // bit, accuracy for lookup table

#define F_LIM_MAX  (mad_fixed_t)((1<<(MAD_F_FRACBITS+2))-1)  // max. value covered by lookup table

#define LIM_SHIFT  (MAD_F_FRACBITS-LIM_ACC)                  // shift value for table lookup

#define F_LIM_JMP  (mad_fixed_t)(1<<LIM_SHIFT)               // lookup table jump between values

// cLevel

#define POW_WIN 100                                          // window width for smoothing power values

#define EPSILON 0.00000000001                                // anything less than EPSILON is considered zero

// --- cResample ------------------------------------------------------------

// The resample code has been adapted from the madplay project

// (resample.c) found in the libmad distribution

class cResample {

private:

  mad_fixed_t ratio;

  mad_fixed_t step;

  mad_fixed_t last;

  mad_fixed_t resampled[MAX_NSAMPLES];

public:

  bool SetInputRate(unsigned int oldrate, unsigned int newrate);

  unsigned int ResampleBlock(unsigned int nsamples, const mad_fixed_t *old);

  const mad_fixed_t *Resampled(void) { return resampled; }

  };

bool cResample::SetInputRate(unsigned int oldrate, unsigned int newrate)

{

  if(oldrate<8000 || oldrate>newrate*6) { // out of range

    esyslog("WARNING: samplerate %d out of range 8000-%d\n",oldrate,newrate*6);

    return 0;

    }

  ratio=mad_f_tofixed((double)oldrate/(double)newrate);

  step=0; last=0;

#ifdef DEBUG

  static mad_fixed_t oldratio=0;

  if(oldratio!=ratio) {

    printf("mad: new resample ratio %f (from %d kHz to %d kHz)\n",mad_f_todouble(ratio),oldrate,newrate);

    oldratio=ratio;

    }

#endif

  return ratio!=MAD_F_ONE;

}

unsigned int cResample::ResampleBlock(unsigned int nsamples, const mad_fixed_t *old)

{

  // This resampling algorithm is based on a linear interpolation, which is

  // not at all the best sounding but is relatively fast and efficient.

  //

  // A better algorithm would be one that implements a bandlimited

  // interpolation.

  mad_fixed_t *nsam=resampled;

  const mad_fixed_t *end=old+nsamples;

  const mad_fixed_t *begin=nsam;

  if(step < 0) {

    step = mad_f_fracpart(-step);

    while (step < MAD_F_ONE) {

      *nsam++ = step ? last+mad_f_mul(*old-last,step) : last;

      step += ratio;

      if(((step + 0x00000080L) & 0x0fffff00L) == 0)

	step = (step + 0x00000080L) & ~0x0fffffffL;

      }

    step -= MAD_F_ONE;

    }

  while (end - old > 1 + mad_f_intpart(step)) {

    old += mad_f_intpart(step);

    step = mad_f_fracpart(step);

    *nsam++ = step ? *old + mad_f_mul(old[1] - old[0], step) : *old;

    step += ratio;

    if (((step + 0x00000080L) & 0x0fffff00L) == 0)

      step = (step + 0x00000080L) & ~0x0fffffffL;

    }

  if (end - old == 1 + mad_f_intpart(step)) {

    last = end[-1];

    step = -step;

    }

  else step -= mad_f_fromint(end - old);

  return nsam-begin;

}

// --- cLevel ----------------------------------------------------------------

// The normalize algorithm and parts of the code has been adapted from the

// Normalize 0.7 project. (C) 1999-2002, Chris Vaill <cvaill@cs.columbia.edu>

// A little background on how normalize computes the volume

// of a wav file, in case you want to know just how your

// files are being munged:

//

// The volumes calculated are RMS amplitudes, which corre­

// spond (roughly) to perceived volume. Taking the RMS ampli­

// tude of an entire file would not give us quite the measure

// we want, though, because a quiet song punctuated by short

// loud parts would average out to a quiet song, and the

// adjustment we would compute would make the loud parts

// excessively loud.

//

// What we want is to consider the maximum volume of the

// file, and normalize according to that. We break up the

// signal into 100 chunks per second, and get the signal

// power of each chunk, in order to get an estimation of

// "instantaneous power" over time. This "instantaneous

// power" signal varies too much to get a good measure of the

// original signal's maximum sustained power, so we run a

// smoothing algorithm over the power signal (specifically, a

// mean filter with a window width of 100 elements). The max­

// imum point of the smoothed power signal turns out to be a

// good measure of the maximum sustained power of the file.

// We can then take the square root of the power to get maxi­

// mum sustained RMS amplitude.

class cLevel {

private:

  double maxpow;

  mad_fixed_t peak;

  struct Power {

    // smooth

    int npow, wpow;

    double powsum, pows[POW_WIN];

    // sum

    unsigned int nsum;

    double sum;

    } power[2];

  //

  inline void AddPower(struct Power *p, double pow);

public:

  void Init(void);

  void GetPower(struct mad_pcm *pcm);

  double GetLevel(void);

  double GetPeak(void);

  };

void cLevel::Init(void)

{

  for(int l=0 ; l<2 ; l++) {

    struct Power *p=&power[l];

    p->sum=p->powsum=0.0; p->wpow=p->npow=p->nsum=0;

    for(int i=POW_WIN-1 ; i>=0 ; i--) p->pows[i]=0.0;

    }

  maxpow=0.0; peak=0;

}

void cLevel::GetPower(struct mad_pcm *pcm)

{

  for(int i=0 ; i<pcm->channels ; i++) {

    struct Power *p=&power[i];

    mad_fixed_t *data=pcm->samples[i];

    for(int n=pcm->length ; n>0 ; n--) {

      if(*data < -peak) peak = -*data;

      if(*data >  peak) peak =  *data;

      double s=mad_f_todouble(*data++);

      p->sum+=(s*s);

      if(++(p->nsum)>=pcm->samplerate/100) {

        AddPower(p,p->sum/(double)p->nsum);

        p->sum=0.0; p->nsum=0;

        }

      }

    }

}

void cLevel::AddPower(struct Power *p, double pow)

{

  p->powsum+=pow;

  if(p->npow>=POW_WIN) {

    if(p->powsum>maxpow) maxpow=p->powsum;

    p->powsum-=p->pows[p->wpow];

    }

  else p->npow++;

  p->pows[p->wpow]=pow;

  p->wpow=(p->wpow+1) % POW_WIN;

}

double cLevel::GetLevel(void)

{

  if(maxpow<EPSILON) {

    // Either this whole file has zero power, or was too short to ever

    // fill the smoothing buffer.  In the latter case, we need to just

    // get maxpow from whatever data we did collect.

    if(power[0].powsum>maxpow) maxpow=power[0].powsum;

    if(power[1].powsum>maxpow) maxpow=power[1].powsum;

    }

  double level=sqrt(maxpow/(double)POW_WIN);     // adjust for the smoothing window size and root

  d(printf("norm: new volumen level=%f peak=%f\n",level,mad_f_todouble(peak)))

  return level;

}

double cLevel::GetPeak(void)

{

  return mad_f_todouble(peak);

}

// --- cNormalize ------------------------------------------------------------

class cNormalize {

private:

  mad_fixed_t gain;

  double d_limlvl, one_limlvl;

  mad_fixed_t limlvl;

  bool dogain, dolimit;

#ifdef DEBUG

  // stats

  unsigned long limited, clipped, total;

  mad_fixed_t peak;

#endif

  // limiter

#ifdef USE_FAST_LIMITER

  mad_fixed_t *table, tablestart;

  int tablesize;

  inline mad_fixed_t FastLimiter(mad_fixed_t x);

#endif

  inline mad_fixed_t Limiter(mad_fixed_t x);

public:

  cNormalize(void);

  ~cNormalize();

  void Init(double Level, double Peak);

  void Stats(void);

  void AddGain(struct mad_pcm *pcm);

  };

cNormalize::cNormalize(void)

{

  d_limlvl=(double)MP3Setup.LimiterLevel/100.0;

  one_limlvl=1-d_limlvl;

  limlvl=mad_f_tofixed(d_limlvl);

  d(printf("norm: lim_lev=%f lim_acc=%d\n",d_limlvl,LIM_ACC))

#ifdef USE_FAST_LIMITER

  mad_fixed_t start=limlvl & ~(F_LIM_JMP-1);

  tablestart=start;

  tablesize=(unsigned int)(F_LIM_MAX-start)/F_LIM_JMP + 2;

  table=new mad_fixed_t[tablesize];

  if(table) {

    d(printf("norm: table size=%d start=%08x jump=%08x\n",tablesize,start,F_LIM_JMP))

    for(int i=0 ; i<tablesize ; i++) {

      table[i]=Limiter(start);

      start+=F_LIM_JMP;

      }

    tablesize--; // avoid a -1 in FastLimiter()

    // do a quick accuracy check, just to be sure that FastLimiter() is working

    // as expected :-)

#ifdef ACC_DUMP

    FILE *out=fopen("/tmp/limiter","w");

#endif

    mad_fixed_t maxdiff=0;

    for(mad_fixed_t x=F_LIM_MAX ; x>=limlvl ; x-=mad_f_tofixed(1e-4)) {

      mad_fixed_t diff=mad_f_abs(Limiter(x)-FastLimiter(x));

      if(diff>maxdiff) maxdiff=diff;

#ifdef ACC_DUMP

      fprintf(out,"%0.10f\t%0.10f\t%0.10f\t%0.10f\t%0.10f\n",

        mad_f_todouble(x),mad_f_todouble(Limiter(x)),mad_f_todouble(FastLimiter(x)),mad_f_todouble(diff),mad_f_todouble(maxdiff));

      if(ferror(out)) break;

#endif

      }

#ifdef ACC_DUMP

    fclose(out);

#endif

    d(printf("norm: accuracy %.12f\n",mad_f_todouble(maxdiff)))

    if(mad_f_todouble(maxdiff)>1e-6) {

      esyslog("ERROR: accuracy check failed, normalizer disabled");

      delete table; table=0;

      }

    }

  else esyslog("ERROR: no memory for lookup table, normalizer disabled");

#endif // USE_FAST_LIMITER

}

cNormalize::~cNormalize()

{

#ifdef USE_FAST_LIMITER

  delete[] table;

#endif

}

void cNormalize::Init(double Level, double Peak)

{

  double Target=(double)MP3Setup.TargetLevel/100.0;

  double dgain=Target/Level;

  if(dgain>MAX_GAIN) dgain=MAX_GAIN;

  gain=mad_f_tofixed(dgain);

  // Check if we actually need to apply a gain

  dogain=(Target>0.0 && fabs(1-dgain)>MIN_GAIN);

#ifdef USE_FAST_LIMITER

  if(!table) dogain=false;

#endif

  // Check if we actually need to do limiting:

  // we have to if limiter is enabled, if gain>1 and if the peaks will clip.

  dolimit=(d_limlvl<1.0 && dgain>1.0 && Peak*dgain>1.0);

#ifdef DEBUG

  printf("norm: gain=%f dogain=%d dolimit=%d (target=%f level=%f peak=%f)\n",dgain,dogain,dolimit,Target,Level,Peak);

  limited=clipped=total=0; peak=0;

#endif

}

void cNormalize::Stats(void)

{

#ifdef DEBUG

  if(total)

    printf("norm: stats tot=%ld lim=%ld/%.3f%% clip=%ld/%.3f%% peak=%.3f\n",

           total,limited,(double)limited/total*100.0,clipped,(double)clipped/total*100.0,mad_f_todouble(peak));

#endif

}

mad_fixed_t cNormalize::Limiter(mad_fixed_t x)

{

// Limiter function:

//

//        / x                                                (for x <= lev)

//   x' = |

//        \ tanh((x - lev) / (1-lev)) * (1-lev) + lev        (for x > lev)

//

// call only with x>=0. For negative samples, preserve sign outside this function

//

// With limiter level = 0, this is equivalent to a tanh() function;

// with limiter level = 1, this is equivalent to clipping.

  if(x>limlvl) {

#ifdef DEBUG

    if(x>MAD_F_ONE) clipped++;

    limited++;

#endif

    x=mad_f_tofixed(tanh((mad_f_todouble(x)-d_limlvl) / one_limlvl) * one_limlvl + d_limlvl);

    }

  return x;

}

#ifdef USE_FAST_LIMITER

mad_fixed_t cNormalize::FastLimiter(mad_fixed_t x)

{

// The fast algorithm is based on a linear interpolation between the

// the values in the lookup table. Relays heavly on libmads fixed point format.

  if(x>limlvl) {

    int i=(unsigned int)(x-tablestart)/F_LIM_JMP;

#ifdef DEBUG

    if(x>MAD_F_ONE) clipped++;

    limited++;

    if(i>=tablesize) printf("norm: overflow x=%f x-ts=%f i=%d tsize=%d\n",

                            mad_f_todouble(x),mad_f_todouble(x-tablestart),i,tablesize);

#endif

    mad_fixed_t r=x & (F_LIM_JMP-1);

    x=MAD_F_ONE;

    if(i<tablesize) {

      mad_fixed_t *ptr=&table[i];

      x=*ptr;

      mad_fixed_t d=*(ptr+1)-x;

      //x+=mad_f_mul(d,r)<<LIM_ACC;                // this is not accurate as mad_f_mul() does >>MAD_F_FRACBITS

                                                   // which is senseless in the case of following <<LIM_ACC.

      x+=((long long)d*(long long)r)>>LIM_SHIFT;   // better, don't know if works on all machines

      }

    }

  return x;

}

#endif

#ifdef USE_FAST_LIMITER

#define LIMITER_FUNC FastLimiter

#else

#define LIMITER_FUNC Limiter

#endif

void cNormalize::AddGain(struct mad_pcm *pcm)

{

  if(dogain) {

    for(int i=0 ; i<pcm->channels ; i++) {

      mad_fixed_t *data=pcm->samples[i];

#ifdef DEBUG

      total+=pcm->length;

#endif

      if(dolimit) {

        for(int n=pcm->length ; n>0 ; n--) {

          mad_fixed_t s=mad_f_mul(*data,gain);

          if(s<0) {

            s=-s;

#ifdef DEBUG

            if(s>peak) peak=s;

#endif

            s=LIMITER_FUNC(s);

            s=-s;

            }

          else {

#ifdef DEBUG

            if(s>peak) peak=s;

#endif

            s=LIMITER_FUNC(s);

            }

          *data++=s;

          }

        }

      else {

        for(int n=pcm->length ; n>0 ; n--) {

          mad_fixed_t s=mad_f_mul(*data,gain);

#ifdef DEBUG

          if(s>peak) peak=s;

          else if(-s>peak) peak=-s;

#endif

          if(s>MAD_F_ONE) s=MAD_F_ONE;   // do clipping

          if(s<-MAD_F_ONE) s=-MAD_F_ONE;

          *data++=s;

          }

        }

      }

    }

}

// --- cScale ----------------------------------------------------------------

// The dither code has been adapted from the madplay project

// (audio.c) found in the libmad distribution

enum eAudioMode { amRoundBE, amDitherBE, amRoundLE, amDitherLE };

class cScale {

private:

  enum { MIN=-MAD_F_ONE, MAX=MAD_F_ONE - 1 };

#ifdef DEBUG

  // audio stats

  unsigned long clipped_samples;

  mad_fixed_t peak_clipping;

  mad_fixed_t peak_sample;

#endif

  // dither

  struct dither {

    mad_fixed_t error[3];

    mad_fixed_t random;

    } leftD, rightD;

  //

  inline mad_fixed_t Clip(mad_fixed_t sample, bool stats=true);

  inline unsigned long Prng(unsigned long state);

  signed long LinearRound(mad_fixed_t sample);

  signed long LinearDither(mad_fixed_t sample, struct dither *dither);

public:

  void Init(void);

  void Stats(void);

  unsigned int ScaleBlock(unsigned char *data, unsigned int size, unsigned int &nsamples, const mad_fixed_t * &left, const mad_fixed_t * &right, eAudioMode mode);

  };

void cScale::Init(void)

{

#ifdef DEBUG

  clipped_samples=0; peak_clipping=peak_sample=0;

#endif

  memset(&leftD,0,sizeof(leftD));

  memset(&rightD,0,sizeof(rightD));

}

void cScale::Stats(void)

{

#ifdef DEBUG

  printf("mp3: scale stats clipped=%ld peak_clip=%f peak=%f\n",

         clipped_samples,mad_f_todouble(peak_clipping),mad_f_todouble(peak_sample));

#endif

}

// gather signal statistics while clipping

mad_fixed_t cScale::Clip(mad_fixed_t sample, bool stats)

{

#ifndef DEBUG

  if (sample > MAX) sample = MAX;

  if (sample < MIN) sample = MIN;

#else

  if(!stats) {

    if (sample > MAX) sample = MAX;

    if (sample < MIN) sample = MIN;

    }

  else {

    if (sample >= peak_sample) {

      if (sample > MAX) {

        ++clipped_samples;

        if (sample - MAX > peak_clipping)

	  peak_clipping = sample - MAX;

        sample = MAX;

        }

      peak_sample = sample;

      }

    else if (sample < -peak_sample) {

      if (sample < MIN) {

        ++clipped_samples;

        if (MIN - sample > peak_clipping)

	  peak_clipping = MIN - sample;

        sample = MIN;

        }

      peak_sample = -sample;

      }

    }

#endif

  return sample;

}

// generic linear sample quantize routine

signed long cScale::LinearRound(mad_fixed_t sample)

{

  // round

  sample += (1L << (MAD_F_FRACBITS - OUT_BITS));

  // clip

  sample=Clip(sample);

  // quantize and scale

  return sample >> (MAD_F_FRACBITS + 1 - OUT_BITS);

}

// 32-bit pseudo-random number generator

unsigned long cScale::Prng(unsigned long state)

{

  return (state * 0x0019660dL + 0x3c6ef35fL) & 0xffffffffL;

}

// generic linear sample quantize and dither routine

signed long cScale::LinearDither(mad_fixed_t sample, struct dither *dither)

{

  // noise shape

  sample += dither->error[0] - dither->error[1] + dither->error[2];

  dither->error[2] = dither->error[1];

  dither->error[1] = dither->error[0] / 2;

  // bias

  mad_fixed_t output = sample + (1L << (MAD_F_FRACBITS + 1 - OUT_BITS - 1));

  const int scalebits = MAD_F_FRACBITS + 1 - OUT_BITS;

  const mad_fixed_t mask = (1L << scalebits) - 1;

  // dither

  const mad_fixed_t random = Prng(dither->random);

  output += (random & mask) - (dither->random & mask);

  dither->random = random;

  // clip

  output=Clip(output);

  sample=Clip(sample,false);

  // quantize

  output &= ~mask;

  // error feedback

  dither->error[0] = sample - output;

  // scale

  return output >> scalebits;

}

#define PUT_BE(data,sample) { *data++=(sample)>>8; *data++=(sample)>>0; }

#define PUT_LE(data,sample) { *data++=(sample)>>0; *data++=(sample)>>8; }

// write a block of signed 16-bit PCM samples

unsigned int cScale::ScaleBlock(unsigned char *data, unsigned int size, unsigned int &nsamples, const mad_fixed_t * &left, const mad_fixed_t * &right, eAudioMode mode)

{

  unsigned int len=size/OUT_FACT;

  if(len>nsamples) { len=nsamples; size=len*OUT_FACT; }

  nsamples-=len;

  switch(mode) {

    case amRoundBE:

      while(len--) {

        signed int sample=LinearRound(*left++);

        PUT_BE(data,sample);

        if(right) sample=LinearRound(*right++);

        PUT_BE(data,sample);

        }

      break;

    case amDitherBE:

      while(len--) {

	signed int sample=LinearDither(*left++,&leftD);

        PUT_BE(data,sample);

	if(right) sample=LinearDither(*right++,&rightD);

        PUT_BE(data,sample);

        }

      break;

    case amRoundLE:

      while(len--) {

        signed int sample=LinearRound(*left++);

        PUT_LE(data,sample);

        if(right) sample=LinearRound(*right++);

        PUT_LE(data,sample);

        }

      break;

    case amDitherLE:

      while(len--) {

	signed int sample=LinearDither(*left++,&leftD);

        PUT_LE(data,sample);

	if(right) sample=LinearDither(*right++,&rightD);

        PUT_LE(data,sample);

        }

      break;

    }

 return size;

}

// --- cShuffle ----------------------------------------------------------------

class cShuffle {

private:

  int *shuffle, max;

  unsigned int seed;

  //

  int Index(int pos);

public:

  cShuffle(void);

  ~cShuffle();

  void Shuffle(int num, int curr);

  void Del(int pos);

  void Flush(void);

  int First(void);

  int Next(int curr);

  int Prev(int curr);

  int Goto(int pos, int curr);

  };

cShuffle::cShuffle(void)

{

  shuffle=0; max=0;

  seed=time(0);

}

cShuffle::~cShuffle(void)

{

  Flush();

}

void cShuffle::Flush(void)

{

  delete shuffle; shuffle=0;

  max=0;

}

int cShuffle::Index(int pos)

{

  if(pos>=0)

    for(int i=0; i<max; i++) if(shuffle[i]==pos) return i;

  return -1;

}

void cShuffle::Shuffle(int num, int curr)

{

  int oldmax=0;

  if(num!=max) {

    int *ns=new int[num];

    if(shuffle) {

      if(num>max) {

        memcpy(ns,shuffle,max*sizeof(int));

        oldmax=max;

        }

      delete shuffle;

      }

    shuffle=ns; max=num;

    }

  if(!oldmax) curr=-1;

  for(int i=oldmax ; i<max ; i++) shuffle[i]=i;

  int in=Index(curr)+1; if(in<0) in=0;

  if((max-in)>=2) {

    for(int i=in ; i<max ; i++) {

      int ran=(rand_r(&seed) % ((max-in)*4-4))/4; ran+=((ran+in) >= i);

      int t=shuffle[i];

      shuffle[i]=shuffle[ran+in];

      shuffle[ran+in]=t;

      }

    }

#ifdef DEBUG

  printf("shuffle: order (%d , %d -> %d) ",num,curr,in);

  for(int i=0 ; i<max ; i++) printf("%d ",shuffle[i]);

  printf("\n");

#endif

}

void cShuffle::Del(int pos)

{

  int i=Index(pos);

  if(i>=0) {

    if(i+1<max) memmove(&shuffle[i],&shuffle[i+1],(max-i-1)*sizeof(int));

    max--;

    }

}

int cShuffle::First(void)

{

  return shuffle[0];

}

int cShuffle::Next(int curr)

{

  int i=Index(curr);

  return (i>=0 && i+1<max) ? shuffle[i+1] : -1;

}

int cShuffle::Prev(int curr)

{

  int i=Index(curr);

  return (i>0) ? shuffle[i-1] : -1;

}

int cShuffle::Goto(int pos, int curr)

{

  int i=Index(curr);

  int g=Index(pos);

  if(g>=0) {

    if(g<i) {

      for(int l=g; l<i; l++) shuffle[l]=shuffle[l+1];

      shuffle[i]=pos;

      }

    else if(g>i) {

      for(int l=g; l>i+1; l--) shuffle[l]=shuffle[l-1];

      shuffle[i+1]=pos;

      }

#ifdef DEBUG

    printf("shuffle: goto order (%d -> %d , %d -> %d) ",pos,g,curr,i);

    for(int i=0 ; i<max ; i++) printf("%d ",shuffle[i]);

    printf("\n");

#endif

    return pos;

    }

  return -1;

}

// --- cPlayManager ------------------------------------------------------------

#define SCANNED_ID3 1

#define SCANNED_LVL 2

cPlayManager *mgr=0;

cPlayManager::cPlayManager(void)

{

  curr=0; currIndex=-1;

  scan=0; stopscan=throttle=pass2=release=false;

  play=0; playNew=eol=false;

  shuffle=new cShuffle;

  loopMode=(MP3Setup.InitLoopMode>0);

  shuffleMode=(MP3Setup.InitShuffleMode>0);

}

cPlayManager::~cPlayManager()

{

  Flush();

  Release();

  listMutex.Lock();

  stopscan=true; bgCond.Broadcast();

  listMutex.Unlock();

  Cancel(2);

  delete shuffle;

}

void cPlayManager::ThrottleWait(void)

{

  while(!stopscan && !release && throttle) {

    db(printf("mgr: background scan throttled\n"))

    bgCond.Wait(listMutex);

    db(printf("mgr: background scan throttle wakeup\n"))

    }

}

void cPlayManager::Action(void)

{

  db(printf("mgr: background scan thread started (pid=%d)\n", getpid()))

  nice(5);

  listMutex.Lock();

  while(!stopscan) {

    for(scan=list.First(); !stopscan && !release && scan; scan=list.Next(scan)) {

      ThrottleWait();

      listMutex.Unlock();

      if(!(scan->user & SCANNED_ID3)) {

        db(printf("mgr: scanning (id3) %s\n",scan->Name()))

        cSongInfo *si=scan->Info(true);

        if(si && si->Level>0.0) scan->user|=SCANNED_LVL;

        scan->user|=SCANNED_ID3;

        }

      listMutex.Lock();

      }

    if(MP3Setup.BgrScan>1) {

      pass2=true;

      for(scan=list.First(); !stopscan && !release && scan; scan=list.Next(scan)) {

        if(scan==curr) continue;

        ThrottleWait();

        listMutex.Unlock();

        if(!(scan->user & SCANNED_LVL)) {

          cDecoder *dec=scan->Decoder();

          if(dec) {

            cSongInfo *si=scan->Info(false);

            if(!dec->IsStream() && (!si || si->Level<=0.0) && dec->Start()) {

              db(printf("mgr: scanning (lvl) %s\n",scan->Name()))

              cLevel level;

              level.Init();

              bool go=true;

              while(go && !release) {

                if(throttle) {

                  listMutex.Lock(); ThrottleWait(); listMutex.Unlock();

                  continue;

                  }

                struct Decode *ds=dec->Decode();

                switch(ds->status) {

                  case dsPlay:

                    level.GetPower(ds->pcm);

                    break;

                  case dsSkip:

                  case dsSoftError:

                    break;

                  case dsEof:

                    {

                    double l=level.GetLevel();

                    if(l>0.0) {

                      cSongInfo *si=dec->SongInfo(false);

                      cFileInfo *fi=dec->FileInfo();

                      if(si && fi) {

                        si->Level=l;

                        si->Peak=level.GetPeak();

                        InfoCache.Cache(si,fi);

                        }

                      }

                    }

                    //fall through

                  case dsOK:

                  case dsError:

                    scan->user|=SCANNED_LVL;

                    go=false;

                    break;

                  }

                }

              }

            else scan->user|=SCANNED_LVL;

            dec->Stop();

            }

          }

        listMutex.Lock();

        }

      pass2=false;

      }

    do {

      scan=0; release=false; fgCond.Broadcast();

      db(printf("mgr: background scan idle\n"))

      bgCond.Wait(listMutex);

      db(printf("mgr: background scan idle wakeup\n"))

      } while(!stopscan && (release || throttle));

    }

  listMutex.Unlock();

  db(printf("mgr: background scan thread ended (pid=%d)\n", getpid()))

}

void cPlayManager::Throttle(bool thr)

{

  if(MP3Setup.BgrScan) {

    if(!thr && throttle) {

      db(printf("mgr: bgr-scan -> run (%d)\n",time_ms()))

      listMutex.Lock();

      throttle=false; bgCond.Broadcast();

      listMutex.Unlock();

      }

    if(thr && !throttle) {

      db(printf("mgr: bgr-scan -> throttle (%d)\n",time_ms()))

      throttle=true;

      }

    }

}

void cPlayManager::ToggleShuffle(void)

{

  shuffleMode=!shuffleMode;

  d(printf("mgr: shuffle mode toggled : %d\n",shuffleMode))

  if(shuffleMode && !eol) {

    curr=0; currIndex=-1;

    shuffle->Shuffle(maxIndex+1,-1);

    Next();

    }

}

void cPlayManager::ToggleLoop(void)

{

  loopMode=!loopMode;

  d(printf("mgr: loop mode toggled : %d\n",loopMode))

}

bool cPlayManager::Info(int num, cMP3PlayInfo *pi)

{

  cSong *s;

  int idx=num-1;

  if(idx<0) { idx=currIndex; s=curr; }

  else      { s=list.Get(idx); }

  memset(pi,0,sizeof(*pi));

  pi->Num=idx+1;

  pi->MaxNum=maxIndex+1;

  pi->Loop=loopMode;

  pi->Shuffle=shuffleMode;

  bool res=false;

  if(s) {

    strn0cpy(pi->Title,s->Name(),sizeof(pi->Title));

    strn0cpy(pi->Filename,s->FullPath(),sizeof(pi->Filename));

    cSongInfo *si=s->Info(false);

    if(si && si->HasInfo()) {

      static const char *modestr[] = { "Mono","Dual","Joint-Stereo","Stereo" };

      if(si->Title)  strn0cpy(pi->Title,si->Title,sizeof(pi->Title));

      if(si->Artist) strn0cpy(pi->Artist,si->Artist,sizeof(pi->Artist));

      if(si->Album)  strn0cpy(pi->Album,si->Album,sizeof(pi->Album));

      strn0cpy(pi->SMode,modestr[si->ChMode],sizeof(pi->SMode));

      pi->Year=si->Year;

      pi->SampleFreq=si->SampleFreq;

      pi->Bitrate=si->Bitrate;

      pi->MaxBitrate=si->MaxBitrate;

      res=true;

      }

    }

  pi->Hash=MakeHashBuff((char *)pi,(char *)&pi->Loop-(char *)pi);

  return res;

}

void cPlayManager::Add(cPlayList *pl)

{

  cMutexLock lock(&listMutex);

  bool real=false;

  for(cSong *song=pl->First(); song; song=pl->cList<cSong>::Next(song)) {

    cSong *ns=new cSong(song);

    list.Add(ns);

    real=true;

    }

  if(real) {

    if(MP3Setup.BgrScan) { stopscan=false; if(!Active()) Start(); }

    else stopscan=true;

    bgCond.Broadcast();

    maxIndex=list.Count()-1;

    if(shuffleMode) shuffle->Shuffle(maxIndex+1,currIndex);

    if(!curr) Next();

    }

}