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
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
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 ...
There are two approaches in which you can learn it.
First one: use your ears! Experiment with what happens when you tweak the EQ settings for various kinds of audio, e.g. voice recordings, full mixed music tracks or white noise. You will soon get a feeling about what the highs/mids/lows are and which frequencies are prominent in which instruments. As a ...
The best course I can recommend is to use the best tools and resources you have available. Headphones are not ideal, but they're better than bad speakers, and probably better than even good speakers in a bad listening environment.
A lot depends on what you're mixing and what the target environment will be. Mixing music or voice that will mainly be consumed ...
Here is a legend with details:
If this is what you have, then the yellow jacket RCA is video, the red is audio right channel, while the white is audio left channel.
Yes, it is possible, but it isn't easy.
There are a number of free tools for editing audio in the frequency domain. I haven't had much luck isolating specific sounds with them, but I have been able to do some sonic manipulation that wasn't possible with traiditional audio editors.
Tapestrea, from the Princeton Sound Lab.
The latest ...
Your equipment is just fine. The differences between a lossless encoding or the original recording and a 320 kbps MP3 encoding are very subtle and most humans cannot distinguish between them. Lowering the bitrate should make the difference more obvious.
Try converting your FLAC file to MP3 (or any other lossy compression) at several different bitrates, ...
The basic idea
The characteristic "wobble" bass popular in mid-late 2000s dubstep is frequently (but not always) created by sweeping a lowpass filter over a harmonically-rich sound of some kind, usually some oscillators. The filter cuts out the higher harmonics, creating an effect similar to closing your lips while saying "ahhhhh. By moving the filter ...
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 ...
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 ...
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. ...
I would use the sound desk. It's always on & you won't need to swap it out every couple weeks. Hopefully... :)
Drawbacks? You may need to use a ground lift on your keyboard if there's hum. For a relatively stationary system like a church...no need to fiddle with batteries.
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 ...
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.
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.
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 ...
Does more impedance mean less volume ?
As to impedance, take a look at Bill Griddle's answer:
In essence: yes, higher impedance = lower volume. But higher volume is not necessarily better quality. Most professional headphones has a high impedance. Are ...
You didn't supply very much information, such as your OS and your desired output format, so I can not give you a detailed answer.
CAF is a container format that can support several audio formats, but Apple Lossless (ALAC) is probably the most common.
ffmpeg can decode ALAC in CAF:
ffmpeg -i input.caf output.wav
This can be turned into a "batch" command ...
how it generates its sounds
This depends on various factors. It can use samples, or a basic synthesizer algorithm, a table etc. Only the authors will know for sure. To me it sounded like low-resolution samples (or heavily compressed mp3), but my mini laptop does not work well for sound analysis.
How to technically generate the sounds depends on which ...
Like most things, it depends. There are several factors at play: the impedance of the source, the nature of the circuit (balanced vs unbalanced), the capacitance of the cable, etc.
Ideally you want a low impedance source driving a low-capacitance, balanced circuit.
The impedance of the source is a big factor. A low driving impedance will not be affected as ...