mp3dec: resample to target frequency, play: guess mp3 from filename
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cinap_lenrek
parent
173a01ef8b
commit
b70a7fc963
2 changed files with 74 additions and 85 deletions
11
rc/bin/play
11
rc/bin/play
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@ -32,13 +32,16 @@ fn play1 {
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case *plain*
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sed 's/
//g' | while(j=`{read}){
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echo $"j >[1=2]
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t=$typ
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if(~ $"j http:* https:* HTTP:* HTTPS:*){
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if(~ $#typ 0) typ=mp3
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hget -r 'Icy-MetaData: 0' $j | play1 $typ
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if(~ $#t 0) t=mp3
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hget -r 'Icy-MetaData: 0' $j | play1 $t
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}
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if not {
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if(test -r $"j)
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play1 <$"j
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if(test -r $"j){
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if(~ $#t 0 && ~ $"j *mp3*) t=mp3
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play1 $t <$"j
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}
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if not {
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echo $argv0: 'can''t open file:' $"j >[1=2]
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}
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@ -29,16 +29,12 @@ input(void *, struct mad_stream *stream)
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return MAD_FLOW_CONTINUE;
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}
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/*
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* Dither 28-bit down to 16-bit. From mpg321.
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* I'm skeptical, but it's supposed to make the
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* samples sound better than just truncation.
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*/
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typedef struct Dither Dither;
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struct Dither
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typedef struct Chan Chan;
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struct Chan
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{
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mad_fixed_t error[3];
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mad_fixed_t random;
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ulong phase;
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mad_fixed_t last;
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mad_fixed_t rand;
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};
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#define PRNG(x) (((x)*0x19660dL + 0x3c6ef35fL) & 0xffffffffL)
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@ -47,94 +43,84 @@ enum
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{
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FracBits = MAD_F_FRACBITS,
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OutBits = 16,
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Round = 1 << (FracBits+1-OutBits-1), // sic
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ScaleBits = FracBits + 1 - OutBits,
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LowMask = (1<<ScaleBits) - 1,
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Min = -MAD_F_ONE,
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Max = MAD_F_ONE - 1,
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};
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int
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audiodither(mad_fixed_t v, Dither *d)
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static uchar*
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resample(Chan *c, mad_fixed_t *src, uchar *dst, int mono, ulong delta, ulong count)
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{
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int out;
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mad_fixed_t random;
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mad_fixed_t last, val, out, rand;
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ulong phase, pos;
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vlong v;
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/* noise shape */
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v += d->error[0] - d->error[1] + d->error[2];
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d->error[2] = d->error[1];
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d->error[1] = d->error[0] / 2;
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/* bias */
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out = v + Round;
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/* dither */
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random = PRNG(d->random);
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out += (random & LowMask) - (d->random & LowMask);
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d->random = random;
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/* clip */
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if(out > Max){
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out = Max;
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if(v > Max)
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v = Max;
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}else if(out < Min){
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out = Min;
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if(v < Min)
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v = Min;
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rand = c->rand;
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last = c->last;
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phase = c->phase;
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pos = phase >> 16;
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while(pos < count){
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val = src[pos];
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if(pos)
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last = src[pos-1];
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/* interpolate */
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v = val;
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v -= last;
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v *= (phase & 0xFFFF);
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out = last + (v >> 16);
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/* dithering */
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out += (rand & LowMask) - LowMask/2;
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rand = PRNG(rand);
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/* cliping */
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if(out > Max)
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out = Max;
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else if(out < Min)
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out = Min;
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out >>= ScaleBits;
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*dst++ = out;
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*dst++ = out >> 8;
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if(mono){
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*dst++ = out;
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*dst++ = out >> 8;
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} else
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dst += 2;
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phase += delta;
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pos = phase >> 16;
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}
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/* quantize */
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out &= ~LowMask;
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/* error feedback */
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d->error[0] = v - out;
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/* scale */
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return out >> ScaleBits;
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c->rand = rand;
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c->last = val;
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if(delta < 0x10000)
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c->phase = phase & 0xFFFF;
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else
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c->phase = phase - (count << 16);
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return dst;
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}
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static enum mad_flow
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output(void *, struct mad_header const* header, struct mad_pcm *pcm)
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{
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int i, n, v;
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mad_fixed_t const *left, *right;
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static Dither d;
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static uchar buf[16384], *p;
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static uchar *buf;
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static int nbuf;
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static Chan c1, c0;
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ulong n, delta;
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uchar *p;
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if(pcm->samplerate != rate){
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rate = pcm->samplerate;
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fprint(2, "warning: audio sample rate is %d Hz\n", rate);
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delta = (pcm->samplerate << 16) / rate;
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n = 4 * (pcm->samplerate + pcm->length * rate) / pcm->samplerate;
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if(n > nbuf){
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nbuf = n;
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buf = realloc(buf, nbuf);
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}
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p = buf;
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memset(&d, 0, sizeof d);
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n = pcm->length;
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switch(pcm->channels){
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case 1:
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left = pcm->samples[0];
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for(i=0; i<n; i++){
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v = audiodither(*left++, &d);
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/* stereoize */
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*p++ = v;
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*p++ = v>>8;
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*p++ = v;
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*p++ = v>>8;
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}
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break;
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case 2:
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left = pcm->samples[0];
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right = pcm->samples[1];
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for(i=0; i<n; i++){
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v = audiodither(*left++, &d);
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*p++ = v;
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*p++ = v>>8;
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v = audiodither(*right++, &d);
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*p++ = v;
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*p++ = v>>8;
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}
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break;
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}
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assert(p<=buf+sizeof buf);
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if(pcm->channels == 2)
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resample(&c1, pcm->samples[1], buf+2, 0, delta, pcm->length);
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p = resample(&c0, pcm->samples[0], buf, pcm->channels == 1, delta, pcm->length);
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write(1, buf, p-buf);
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return MAD_FLOW_CONTINUE;
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}
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