I Like English 3 Audio

[Beverly Zielonko]

Atthis time you can't add anything to itunes on ios. You can do a screen recording but then you can't convert the file to mp3. Foiled again by ios. Looks like you maybe able to export just the audio with Luma Fusion so I think I will try that. Still don't see how to get the audio into itunes on ios but maybe apple will fix that some day.

I recall using Koala to extract screen recorded audio into Koala and saving the result as a Koala Wave Sample file. Koala has a lot of cool use cases for processing audio in unusual ways. I used this workflow to make short samples from a Zappa YouTube Video and triggered them to make a new work in Koala:

Yes! I agree. I love the little add audio button in Anki. I use it to quickly add correct pronunciation for language learning. With pictures and audio, I can help my youngest son learn our tribal language even though he is too little to read much. It is nice that audio can be added to RemNote, but a little record button would be so much quicker. It would be especially handy for language teachers to reinforce correct pronunciation or just give students a sense of things like sentence cadence when sending notes to students.

Recently moved study sessions & note-taking over to Remote & abounded Anki. However, Anki has a feature allowing recorded audio to the back or front of a flashcard. I would like to see them implemented on remote if possible. For example, suppose I get a card wrong. In that case, I can play the audio that I recorded to explain the concept, thus improving my ability to make connections when studying actively.

If I may also suggest, I suggest using a site like vocaroo.com to record/upload audio you already have and share them as links. Red Gregory has also shared another way to get audio, if you need to do so here: =1141

No problem!

I agree with you as well, which is why I also voted for this feature request. It would be tedious to have to do it manually like that

I hope something like this becomes a feature in the future!

In a previous post we had an encounter with jitter in audio. The talk about jitter is hard to miss these days. There are plenty of products on the market claiming to have low jitter or to reduce jitter.

If you want to know how my simulation works, read on. Otherwise, you can skip this section where you can do some fun jittery listening. To simulate jitter, I used handy software called Matlab. Matlab is great for doing all sorts of mathematical things. I wrote a little script that does this:

But here, instead of jumping 100 samples, I jump 100 plus (or minus) a number (lets call that number x). Doing this simulates a situation where your clock is ticking a little earlier or later than it is supposed to.

Noise jitter is when the ticks of the clock are off by a random length of time. That means x is a random number. I used a normally distributed random number, which is what we are more likely to have in the real world.

Periodic jitter is when x oscillates in time. This jitter can be caused by the 60 Hz (or its harmonics) from your power supply leaking into your clock generation circuitry.

Every circuit is unique, so in the real world periodic jitter can come in different frequencies. However, for the sake of simplicity, in this experiment, I used 60Hz periodic jitter.

There are seven versions of each sample. Three have periodic jitter, three have random jitter, and one is the original - untouched. In the samples we have 2s (microseconds), 4s, 8s, 16s RMS jitter. It is unlikely in the real world for a digital-to-analog converter to have a jitter higher than 2s RMS. I threw in the 4s, 8s, and 16s samples to see what some extreme cases sound like. As you scroll down you can listen to them and make your best guess of which is which. Then you can click the sample's title (i.e. 1kHz v1) to see if your guesses are right.

Before you move on, here is something to keep in mind. The clock interval of a 44.1 kHz signal is 22.7 s, meaning, there is a sample every 22.7 s. Jitter is expressed in RMS microseconds. So when we say there is a 2s RMS jitter, it means the clock was, on average, off by 2s.

This wraps up our mini experiment. Of course, the system that you used to listen to the samples also adds its own jitter on top of what was added by me. However, these samples should still give you a good idea about the sound of jitter. I hope this post has shed some light on this illusive phenomenon. How did jitter affect your experience? Did it effect the music the way you imagined? Do you see a flaw in the experiment?

Just as the title says, I have a voice over audio file. When I play it in Itunes, Quicktime player, a random voice memo app and in preview it sounds perfectly fine. Its not until I bring it into premiere pro all of a sudden it sounds compressed, fuzzy and robotic. Its as if I have the mic inside my mouth. None of this distortian is on the original file and makes it unuseable.

convert the file to something uncompressed like 48k 16bit aiff... and if this file worked previously on this system, I'm guessing you did some sort of update... these things usually don't happen on there own (I did say usually).

The easiest way to do it I know is to open Resolve (the free version will do) place two mixes on a timeline one under the other (should be perfect in sync) and invert the audio phase of one of the tracks on Fairlight page

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ah yes same technique used by noise canceling headphones they just put the inverted outsode noise back onto your signal bascially. I used this a lot in audacity to find out if a stereo track was Dolby or not.

I did some of my own research and found out that SID-chips had only few hardware supported synthesizing features. Including three audio oscillators with four possible waveforms (sawtooth, triangle, pulse, noise), with ADSR envelopes and ring modulators. Accompanied with oscillator sync and ring modulators. Also read there was a way to play single PCM sound as well.

I used to write music on the C64 for games, demos and even services (I wrote the official QuantumLink theme, even). As for your question, the four different waveforms were typically overlaid with the sync and ring mods (less often ring, because it was unpredictable on different versions of the SID chip), and sometimes used cleanly.

For example, a typical 'snare' sound would be composed of a noise waveform with a very fast attack and sustain, and depending on whether you wanted a drumstick or brush sound, either a very fast decay and moderately short release, or a short decay and slower release.

Getting the right sound was typically trial and error, and the limitations were pretty heavy. You really never got to the point of piano or guitar sound due to the simple waveforms without overlayable harmonic waveforms, about the best you could get was things that sounded beepy, things that sounded marimba-y, and things that sounded like a snare drum.

One of the tricks used most often to extend sound was done with fast machine code playback routines that could change the played notes on voices so quickly as to give the impression of a fuller, harmonic tone. We just called it arpeggiation, although at 10 to 12 note changes a second it sounded more like a buzzy chord.

As for the sampled waveforms, they were only available as single bit and later 4 bit samples. These sounded terrible despite our best attempts, because basically the method of playback for a sample on the 64 was to play a white noise waveform and rapidly alter the volume on the SID chip to produce a rising and falling wave. Do it fast enough and it sort of sounds like the original sound, poorly tuned in on a staticky radio.

This type of music you are describing falls into the category of "chiptunes". I'd recommend checking out some modern trackers like MilkyTracker, which are used to create music in this style. There are libraries like libmodplug that allow you to play tracker in your software.

First off, you are right on the specs you gave. But the thing is: neither the waveform needs to be fixed for an instrument, nor the pitch, nor pulsewidth, nor filters (cutoff, type and resonance can be changed on the fly). So that's different to some retro-synth VST instruments you may know, where you have mostly fixed parameters.

A C64 tracker normally has tables for that. A wave-and-arpeggio-table is used to adjust waveform and pitch during playing a note. Also there are pulse and filter tables used. Depending on how much processing time can be spent, the SID registers are written to @25Hz or more (some tracks go up to socalled 8-speed, which is 400Hz), standard on PAL systems is 50Hz (screen refresh rate).So by changing waveform, pitch, pulse and filter at a rate just a little lower or even (for bass) close to the musical frequencies, you can produce really nifty sounds from just one voice. Then imagine: 3 channels makes 3 nifty sounds at the same time :)

As for Pianos and Guitars: I'd love to get a wavetable that really sounds like that. As others pointed out: SID is missing the harmonics - although you may use HardSync - which synchronizes two voices to add some kind of harmonics, but I have not managed real-sounding brass, piano or strings so far. On that point: SID sounds like SID.

Regarding sample playback - you may not believe it, but after decades some demo scene enthusiasts have managed to find a way to play back samples via PWM on one voice and still be able to play SID instruments and use the filters on the other 2 voices. Still those samples have a max of 12kHz sampling rate and afaik are only 8 bit (please correct me if I'm wrong), but this really sounds great. Example: =8_M260qMbWA

To sum it up: SID sounds are so unique, because the Parameters are adjusted at rates of 25-400Hz, typically 50Hz. And actually the features of that chip were way ahead of the industry in 1982 when it was released.

A few of our users have reported that participants in meetings tell them that they sound like a robot (choppy and odd audio artifacts). I have heard it when meeting with some of our Ops team members. The users have external bluetooth headphones.

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