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player-mp3.c
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/*
* MP3/MPlayer plugin to VDR (C++)
*
* (C) 2001-2005 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 <linux/dvb/video.h>
#include "common.h"
#include "setup-karaoke.h"
#include "player-mp3.h"
#include "data-mp3.h"
#include "decoder.h"
#include "decoder-core.h"
//#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
#ifdef DEBUG_MODE
#define dm(x) { (x); }
#else
#define dm(x) ;
#endif
#ifdef 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("[vdr-karaoke] 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("[karaoke] 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("[karaoke] 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)KaraokeSetup.LimiterLevel/100.0;
one_limlvl=1-d_limlvl;
limlvl=mad_f_tofixed(d_limlvl);
d(printf("[karaoke] 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("[karaoke] 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("[karaoke] norm: accuracy %.12f\n",mad_f_todouble(maxdiff)))
if(mad_f_todouble(maxdiff)>1e-6) {
esyslog("[vdr-karaoke] ERROR: accuracy check failed, normalizer disabled");
delete table; table=0;
}
}
else esyslog("[vdr-karaoke] 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)KaraokeSetup.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("[karaoke] 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("[karaoke] 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("[karaoke] 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("[karaoke] karaoke: 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("[karaoke] 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("[karaoke] 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;
char Songname[1024];
bool scrollmode;
bool CanLoadCover;
char coverpicture[1024];
cPlayManager::cPlayManager(void)
{
scrollmode=0;
curr=0; currIndex=-1;
scan=0; stopscan=throttle=pass2=release=false;
play=0; playNew=eol=false;
shuffle=new cShuffle;
loopMode=(KaraokeSetup.InitLoopMode>0);
shuffleMode=(KaraokeSetup.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("[karaoke] mgr: background scan throttled\n"))
bgCond.Wait(listMutex);
db(printf("[karaoke] mgr: background scan throttle wakeup\n"))
}
}
void cPlayManager::Action(void)
{
db(printf("[karaoke] 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("[karaoke] 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(KaraokeSetup.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("[karaoke] 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("[karaoke] mgr: background scan idle\n"))
bgCond.Wait(listMutex);
db(printf("[karaoke] mgr: background scan idle wakeup\n"))
} while(!stopscan && (release || throttle));
}
listMutex.Unlock();
db(printf("[karaoke] mgr: background scan thread ended (pid=%d)\n", getpid()))
}
void cPlayManager::Throttle(bool thr)
{
if(KaraokeSetup.BgrScan) {
if(!thr && throttle) {
db(printf("[karaoke] mgr: bgr-scan -> run (%d)\n",time_ms()))
listMutex.Lock();
throttle=false; bgCond.Broadcast();
listMutex.Unlock();
}
if(thr && !throttle) {
db(printf("[karaoke] mgr: bgr-scan -> throttle (%d)\n",time_ms()))
throttle=true;
}
}
}
void cPlayManager::ToggleShuffle(void)
{
shuffleMode=!shuffleMode;
d(printf("[karaoke] mgr: shuffle mode toggled : %d\n",shuffleMode))
}
void cPlayManager::ToggleLoop(void)
{
loopMode=!loopMode;
d(printf("[karaoke] 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(KaraokeSetup.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();
}
}
void cPlayManager::Flush(void)
{
cMutexLock lock(&listMutex);
Halt();
list.Clear();
shuffle->Flush();