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15

Your best bet is to see what formats youtube-dl returns by running at a command-line, youtube-dl -F URL which will produce output looking like this format code extension resolution note 249 webm audio only DASH audio 53k , opus @ 50k (48000Hz), 4.25MiB 250 webm audio only DASH audio 72k , opus @ 70k (48000Hz), 5.53MiB ...


13

This guide worked 100% for me - it takes you through each step. Afterwards, the m4a file plays! Note: this is completely free. http://sysfrontier.com/en/2014/12/31/hello-world/ Also pasting instructions in case the link goes down in future: How to fix corrupted voice memo (m4a) files. Do you have broken voice memo files? You can fix those files by ...


10

Whichever app you use, you'll want to line them up and then invert the phase on one of them. Depending on how identical, and how they were made, it might easy to line them up and get an accurate result, otherwise you'll have to spend some time trying to line it up as best you can by zooming in on the samples. The idea being that if you have two things that ...


7

The purpose of a direct box is simply to take a quarter inch guitar line signal and convert it to a balanced signal that won't degrade over the longer run to the sound system. A DI doesn't do anything that can't be done with a decent recording interface since you aren't going a long distance with the signal. You are best off to get an audio interface that ...


7

Audacity is a great tool for this kind of work. You might be able to make a slightly better sound using a sampler that can crossfade across different sounds for different pitches, but I'm not sure if the result would be worth the extra effort. It sounds to me like the problem with your "current best" is that you either aren't looking for zero crossings when ...


7

You can use MIDI CC 5 - Controls portamento rate to slide between 2 notes played subsequently. MIDI CC 65 - Portamento On/Off switch | 0 to 63 = Off, 64 to 127 = On MIDI CC 84 - Controls the amount of Portamento.


7

Once a signal is digitized it is treated as a number (as you quite rightly point out) and for 16bits the range of numbers are -32768 to +32767. The numbers are created by an analogue to digital converter. The analogue to digital converter (ADC) will have a maximum input range from -X volts to +X volts (i.e. real signals that you could measure inside your ...


7

There are three parameters of this filter that are described in the phrase "100 Hz 12 dB per octave low pass filter". I'll cover them in reverse order. Low pass filter - This means the filter does not change lower frequencies ("passes" those frequencies through) and blocks higher frequencies. Sometimes these filters are called "high cut filters", but that ...


6

SoundIt does just that. And it lets you stream the sound output of your PC to any web browser so you can stream to any phone or even another PC.


6

That's still an EQ, you just cut the bands to -infinity. More specifically, you would want to use a shelf EQ if you have it. Also sometimes called high pass and low pass filters.


6

The only "natural" (i.e. "using only the data in the sample") way to change the pitch of a recorded sample is to change the speed of its playback. So a sample that sounds an octave lower should take twice as long to play back. But you have a tool that somehow manages to play that sample back octave lower, while somehow keeping the playback time the same. ...


6

Audio CDs are encoded with 16-bit values. Higher bitrates are generally used for editing, not for playback. You can write 24-bit WAV files to an optical disc, of course, but it will not be a standards complying audio CD.


6

I strongly disagree with a lot of Dave's recommendations, based on my experience recording vehicle sounds for racing games, advertisement and sound engineering: an engine under load doesn't sound like an engine revving while standing still at all. "patching in after the fact" will be loads more work than recording audio and video at the same time. Expect ...


5

This is a matter of taste more than a technical issue, so I'll be using terms like "groove" and "feel" rather than more quantifiable criteria. It comes down to making it seem natural instead of jarring. Since you mention club DJs, I presume you mean "in front of dancers," and the human body is going to move differently at 90bpm than it is at 130. A simple ...


5

Go to Devices/Device Setup then under VST Audio System on the right side of the window Tick the option underneath ASIO Driver that says release driver when app is in background... ;-p that worked for me.


5

A good digital signal should either get there or not. Minor differences in cable quality are not going to have any impact at all other than latency. Latency is simply the delay in how long it takes to get to point a to point b and won't impact the sound quality significantly. USB is a digital signal, so a "premium" USB cable is a load of crap. There is ...


5

Your best bet will be to run your source material through a bitcrusher and a sample-rate reducer. However, that will just make it sound like the kind of low-quality samples that could be played back on old machines. Actual chiptune music tends to involve very careful use of low-resolution waveforms and FM synthesis.


5

I think the most straight forward solution would be using a DI unit. Once you connect it to your source(PC), you will have a balanced signal which you can safely carry on a balanced cable for 100ft (or even more).


5

Mixing audio in a computer is very analogous to how sound mixes in the real world. In the real world, sound is simply pressure waves that cause vibrations in our eardrum that get converted to what we hear by our brain. In the computer, that level of pressure of the wave is represented as a series of samples that describe how strong the pressure wave was. ...


5

The audio interface would be used instead of your sound card. The laptop itself shouldn't play a role in the quality of the sound. (Though it can happen... see @left's comment below) The sound quality will be affected by the signal chain. I.e., the original audio quality, the settings of the software you use to edit the audio, the interface you ...


5

When no specific feature is to be observed in sound, the best way to represent it is to display its spectrogram since it reflects what the ear does: measuring energy among frequencies as its varies with time. You can get that representation using the free and open source command-line tool SoX: sox sound-original.wav -n spectrogram -o sound-original.png sox ...


5

Nothing posted by Schizomorph is wrong - if it's busted, it's busted... However, one small ray of light is azimuth/stereo balance correction in software. Back in the old days I had a bit of freeware (and the source code) by Airwindows - one of the most amazing, if just a little esoteric, sound designers on the planet - that could do this task [non-realtime],...


5

I am an engineering student my self and I use Audacity to generate Sine, Sawtooth, Square waves. It's easy as pie. Download Audacity from here. Open Audacity and click on Generate from the Toolbar on top. Select the waveform you need, the frequency, the amplitude and duration you want it. Play the track.


5

What you are asking is impossible to determine from a spectrogram. Spectrograms indicate frequencies present, not the quality of the audio. There may be loss of fine detail in one that would barely show up on a spectrogram at all or there could be noise and artifacts introduced that would make the spectrogram look more full. Spectrograms do not ...


4

I had a quick look at your device, it says the USB input is 16 bit / 48 kHz and that the device uses asynchronous sample rate conversion and 24 bit DACs @ 192 kHz. S/PDIF and Toslink are identical at the bitstream level and both are consumer variants of AES3 which supports 16 or 20 bit at 44.1 or 48 kHz. the HiFace2 says it has very good clocks, which ...


4

There is no such tool that I'm aware of and I'm not even sure what exactly the output of such a tool would look like. What is available, however, is the ability to use the integration between Adobe applications to work on the audio track in Premiere directly in Audition so that you shouldn't have to manage it yourself like this. In fact, I believe it may ...


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