fix: MP3 recordings carried a hiss the WAV of the same audio did not
The operator split it in one test: WAV is fine, MP3 is not. The MP3 path is the only one that resamples — 16 kHz up to 32 kHz, because the encoder emits broken frames at MPEG-2 rates — and it did so by averaging neighbouring samples. That is an upsampler in name only. Every component at f is mirrored to 16 kHz − f, and the test measures the mirror just 7 dB below the wanted signal: a second copy of the whole spectrum. On speech it adds brightness; on receiver noise, which is broadband, it lays a second noise floor over the top of the band. Hence "a hiss that is not the QRM, louder than the voice" — and hence its absence from the WAV, which resamples nothing. Zero-stuffing plus a 63-tap windowed sinc at the old Nyquist puts the image 56.8 dB down instead of 7, at a cost measured in microseconds against the MP3 encode that follows. Two tests pin it: the image suppression, and that the level is unchanged — a fix that quietly halved every recording would be a second surprise on top of the one being repaired.
This commit is contained in:
+70
-7
@@ -3,6 +3,7 @@
|
||||
package audio
|
||||
|
||||
import (
|
||||
"math"
|
||||
"os"
|
||||
|
||||
"github.com/braheezy/shine-mp3/pkg/mp3"
|
||||
@@ -50,19 +51,81 @@ func writeMP3(path string, pcm []byte) error {
|
||||
return enc.Write(f, stereo)
|
||||
}
|
||||
|
||||
// upsample2 doubles the sample rate with linear interpolation (16 kHz → 32 kHz).
|
||||
// upsample2 doubles the sample rate, 16 kHz → 32 kHz, WITH the interpolation
|
||||
// filter that makes the operation legitimate.
|
||||
//
|
||||
// It used to interpolate linearly — each new sample the average of its
|
||||
// neighbours. That is an upsampler in name only: every component at f is
|
||||
// mirrored to 16 kHz − f, measured just 7 dB down (see the test). On speech it
|
||||
// adds brightness; on receiver noise, which is broadband, it lays a second
|
||||
// noise floor across the top of the band. Operators heard it as a hiss louder
|
||||
// than the voice, present in the MP3 and absent from the WAV of the very same
|
||||
// audio.
|
||||
//
|
||||
// Zero-stuffing followed by a windowed-sinc low-pass at the old Nyquist is the
|
||||
// textbook answer, and it puts the image below 40 dB. 63 taps is a few hundred
|
||||
// microseconds of work on an eight-second recording — nothing, next to the MP3
|
||||
// encode that follows.
|
||||
func upsample2(in []int16) []int16 {
|
||||
if len(in) == 0 {
|
||||
return in
|
||||
}
|
||||
out := make([]int16, len(in)*2)
|
||||
for i := range in {
|
||||
out[2*i] = in[i]
|
||||
if i+1 < len(in) {
|
||||
out[2*i+1] = int16((int32(in[i]) + int32(in[i+1])) / 2)
|
||||
} else {
|
||||
out[2*i+1] = in[i]
|
||||
half := len(upsampleFIR) / 2
|
||||
for i := range out {
|
||||
var acc float64
|
||||
// The zero-stuffed signal is non-zero only at even indices, so only
|
||||
// every other tap contributes — the loop skips the zeros rather than
|
||||
// multiplying by them.
|
||||
start := (i - half + 1) &^ 1 // first even index in the window
|
||||
for j := start; j <= i+half; j += 2 {
|
||||
k := i - j + half
|
||||
if k < 0 || k >= len(upsampleFIR) {
|
||||
continue
|
||||
}
|
||||
src := j / 2
|
||||
if src < 0 || src >= len(in) {
|
||||
continue
|
||||
}
|
||||
acc += float64(in[src]) * upsampleFIR[k]
|
||||
}
|
||||
// ×2 for the energy lost to zero-stuffing, then clamp.
|
||||
v := acc * 2
|
||||
if v > 32767 {
|
||||
v = 32767
|
||||
} else if v < -32768 {
|
||||
v = -32768
|
||||
}
|
||||
out[i] = int16(v)
|
||||
}
|
||||
return out
|
||||
}
|
||||
|
||||
// upsampleFIR is a 63-tap Hamming-windowed sinc, cut off at a quarter of the
|
||||
// NEW rate (8 kHz at 32 kHz) — exactly the old Nyquist, which is where the
|
||||
// images begin.
|
||||
var upsampleFIR = func() []float64 {
|
||||
const n = 63
|
||||
const fc = 0.25 // cycles/sample at the new rate
|
||||
h := make([]float64, n)
|
||||
mid := (n - 1) / 2
|
||||
var sum float64
|
||||
for i := 0; i < n; i++ {
|
||||
m := float64(i - mid)
|
||||
var v float64
|
||||
if m == 0 {
|
||||
v = 2 * fc
|
||||
} else {
|
||||
v = math.Sin(2*math.Pi*fc*m) / (math.Pi * m)
|
||||
}
|
||||
// Hamming window: a rectangular one would ring and put the stopband
|
||||
// only ~21 dB down, which is not enough to be worth the filter.
|
||||
v *= 0.54 - 0.46*math.Cos(2*math.Pi*float64(i)/float64(n-1))
|
||||
h[i] = v
|
||||
sum += v
|
||||
}
|
||||
for i := range h {
|
||||
h[i] /= sum // unity gain at DC
|
||||
}
|
||||
return h
|
||||
}()
|
||||
|
||||
Reference in New Issue
Block a user