faiface/beep

What's the proper way to close underlying streamers from beep.Seq (or beep.Resample)

Koumbaya opened this issue · 1 comments

Hi, I'm confused about the proper method for closing underlying streamers in a beep.Seq if I can't defer the .Close where I instantiate the streamer ?
Most of the examples are withing a single scope so defer does the job, but for a more complex app, how do I access the .Close function ?
I've been using speaker.Clear() if I want to stop a sequence and play another, but doesn't that mean that the files in the sequence don't get closed properly ?

ewintr commented

I had the same question for beep.Resample and I was able to quickly myself by copy/pasting the code for the resampler and add the Close() function myself. Since it is all interfaces, it doesn't matter from which package the code is.

The following is the same Resampler as from beep, but with an added function so that it satisfies the StreamCloser interface. You can use it like resampled := yourpackage.Resample(4, format.SampleRate, sr, originalStreamer)

I could turn this into a pull request and add it to Beep. But the repo seems quite dead.

package yourpackage

import (
	"fmt"

	"github.com/faiface/beep"
)

// Resample takes a Streamer which is assumed to stream at the old sample rate and returns a
// Streamer, which streams the data from the original Streamer resampled to the new sample rate.
//
// This is, for example, useful when mixing multiple Streamer with different sample rates, either
// through a beep.Mixer, or through a speaker. Speaker has a constant sample rate. Thus, playing
// Streamer which stream at a different sample rate will lead to a changed speed and pitch of the
// playback.
//
//	sr := beep.SampleRate(48000)
//	speaker.Init(sr, sr.N(time.Second/2))
//	speaker.Play(beep.Resample(3, format.SampleRate, sr, s))
//
// In the example, the original sample rate of the source if format.SampleRate. We want to play it
// at the speaker's native sample rate and thus we need to resample.
//
// The quality argument specifies the quality of the resampling process. Higher quality implies
// worse performance. Values below 1 or above 64 are invalid and Resample will panic. Here's a table
// for deciding which quality to pick.
//
//	quality | use case
//	--------|---------
//	1       | very high performance, on-the-fly resampling, low quality
//	3-4     | good performance, on-the-fly resampling, good quality
//	6       | higher CPU usage, usually not suitable for on-the-fly resampling, very good quality
//	>6      | even higher CPU usage, for offline resampling, very good quality
//
// Sane quality values are usually below 16. Higher values will consume too much CPU, giving
// negligible quality improvements.
//
// Resample propagates errors from s.
func Resample(quality int, old, new beep.SampleRate, s beep.StreamCloser) *Resampler {
	return ResampleRatio(quality, float64(old)/float64(new), s)
}

// ResampleRatio is same as Resample, except it takes the ratio of the old and the new sample rate,
// specifically, the old sample rate divided by the new sample rate. Aside from correcting the
// sample rate, this can be used to change the speed of the audio. For example, resampling at the
// ratio of 2 and playing at the original sample rate will cause doubled speed in playback.
func ResampleRatio(quality int, ratio float64, s beep.StreamCloser) *Resampler {
	if quality < 1 || 64 < quality {
		panic(fmt.Errorf("resample: invalid quality: %d", quality))
	}
	return &Resampler{
		s:     s,
		ratio: ratio,
		first: true,
		buf1:  make([][2]float64, 512),
		buf2:  make([][2]float64, 512),
		pts:   make([]point, quality*2),
		off:   0,
		pos:   0,
	}
}

// Resampler is a Streamer created by Resample and ResampleRatio functions. It allows dynamic
// changing of the resampling ratio, which can be useful for dynamically changing the speed of
// streaming.
type Resampler struct {
	s          beep.StreamCloser // the orignal streamer
	ratio      float64           // old sample rate / new sample rate
	first      bool              // true when Stream was not called before
	buf1, buf2 [][2]float64      // buf1 contains previous buf2, new data goes into buf2, buf1 is because interpolation might require old samples
	pts        []point           // pts is for points used for interpolation
	off        int               // off is the position of the start of buf2 in the original data
	pos        int               // pos is the current position in the resampled data
}

// Stream streams the original audio resampled according to the current ratio.
func (r *Resampler) Stream(samples [][2]float64) (n int, ok bool) {
	// if it's the first time, we need to fill buf2 with initial data, buf1 remains zeroed
	if r.first {
		sn, _ := r.s.Stream(r.buf2)
		r.buf2 = r.buf2[:sn]
		r.first = false
	}
	// we start resampling, sample by sample
	for len(samples) > 0 {
	again:
		for c := range samples[0] {
			// calculate the current position in the original data
			j := float64(r.pos) * r.ratio

			// find quality*2 closest samples to j and translate them to points for interpolation
			for pi := range r.pts {
				// calculate the index of one of the closest samples
				k := int(j) + pi - len(r.pts)/2 + 1

				var y float64
				switch {
				// the sample is in buf1
				case k < r.off:
					y = r.buf1[len(r.buf1)+k-r.off][c]
				// the sample is in buf2
				case k < r.off+len(r.buf2):
					y = r.buf2[k-r.off][c]
				// the sample is beyond buf2, so we need to load new data
				case k >= r.off+len(r.buf2):
					// we load into buf1
					sn, _ := r.s.Stream(r.buf1)
					// this condition happens when the original Streamer got
					// drained and j is after the end of the
					// original data
					if int(j) >= r.off+len(r.buf2)+sn {
						return n, n > 0
					}
					// this condition happens when the original Streamer got
					// drained and this one of the closest samples is after the
					// end of the original data
					if k >= r.off+len(r.buf2)+sn {
						y = 0
						break
					}
					// otherwise everything is fine, we swap buffers and start
					// calculating the sample again
					r.off += len(r.buf2)
					r.buf1 = r.buf1[:sn]
					r.buf1, r.buf2 = r.buf2, r.buf1
					goto again
				}

				r.pts[pi] = point{float64(k), y}
			}

			// calculate the resampled sample using polynomial interpolation from the
			// quality*2 closest samples
			samples[0][c] = lagrange(r.pts, j)
		}
		samples = samples[1:]
		n++
		r.pos++
	}
	return n, true
}

// Err propagates the original Streamer's errors.
func (r *Resampler) Err() error {
	return r.s.Err()
}

// Close propagates the original Streamers Close method.
func (r *Resampler) Close() error {
	return r.s.Close()
}

// Ratio returns the current resampling ratio.
func (r *Resampler) Ratio() float64 {
	return r.ratio
}

// SetRatio sets the resampling ratio. This does not cause any glitches in the stream.
func (r *Resampler) SetRatio(ratio float64) {
	r.pos = int(float64(r.pos) * r.ratio / ratio)
	r.ratio = ratio
}

// lagrange calculates the value at x of a polynomial of order len(pts)+1 which goes through all
// points in pts
func lagrange(pts []point, x float64) (y float64) {
	y = 0.0
	for j := range pts {
		l := 1.0
		for m := range pts {
			if j == m {
				continue
			}
			l *= (x - pts[m].X) / (pts[j].X - pts[m].X)
		}
		y += pts[j].Y * l
	}
	return y
}

type point struct {
	X, Y float64
}