Hot answers tagged audio-quality
10
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 ...
8
As the mic is mounted on a stand a pop-filter would be a very easy solution.
If this is visually not acceptable (as it's a video project) you could think of adding a physical barrier between the person and the stand. For example a (DIY version of) this barrier.
6
When you import a lossy audio file into editing software, that file is decoded into an audio waveform. This is a "lossless" waveform in that it's not stored in a compressed format, but it's not the same as the original waveform that your lossy file was approximating.
So there are already three concepts involved:
The original waveform, which is to say, ...
6
I'd start with the hum, as this becomes more difficult after any other editing step. But right at the beginning, you should be able to do it very well with a multiple-notch filter.
Then I'd tackle clicks and pops. For this you get the best results by doing all of them manually, but it's of course very laborous. But fully automatic tools, in my experience, ...
6
Branding. Price. Policy on stifling competition.
In all reality, a good cable shouldn't do anything between Point A and Point B. Any cable claims to that is either lying or doing something you don't want it to.
I've found that the best cables I can get are the ones I make for myself at 1/10th the cost of the big brands. Even the lifetime warranty isn't ...
6
You'll actually get arguably better quality from your CD.
At best those MP3s will be encoded from the same masters as the CDs, except lossily compressed to MP3, while your CDs are uncompressed PCM audio. Any improvements to the production process - and AFAIK there haven't been any major improvements in quality at the professional level (take that with a ...
6
1/4" TS/TRS is a far superior interconnect than RCA/phono in every way, except size.
If you have to choose an output for an instrument, it should be 1/4" if at all possible.
1/4" TS cables are more robust, much more easily repairable, and are used throughout the professional audio industry.
RCA cables were invented (by RCA) for use in consumer ...
5
The warnings from WLPhoenix and Josh are absolutely correct: most cable connections do never have a significant influence on the sound, regardless of the cable brand.
There is one important exception: guitar cables. Because most electric guitars have only primitive passive high-impedance electronics (the same applies to passive electric basses), supplied ...
5
It is definitely possible that the quality is degraded. All parts of the recording chain have some upper limit on the frequency range they can handle properly. On high-end turntables, this may well exceed the audible range by quite a lot and therefore not matter, but if it was digitalised at 44.1 kHz (which it almost certainly was, if it came back as a ...
5
Does more impedance mean less volume ?
As to impedance, take a look at Bill Griddle's answer:
What is the effect of impedance on headphone sound quality?
In essence: yes, higher impedance = lower volume. But higher volume is not necessarily better quality. Most professional headphones has a high impedance. Are you A DJ then low impedance is more ...
4
If you mean bad in the sense in which a bad cable is bad – i.e., broken or fine – simply connect the microphone to a mixer with headphone output or an active speaker, tap on the microphone and say something into it. You'll notice if one doesn't work, there will be no sound! Or e.g. heavily distorted sound, or failing sound when you jiggle the plug etc.
...
4
Travis, you have, perhaps unwittingly, stumbled upon a question that has raged for quite some time. In short, there is no answer. Or more to the point, there is no agreed-upon answer. The Wikipedia entry on this is fairly concise for the tl;dr crowd...
There is debate among audiophiles surrounding the impact that high-end
cables have on audio ...
4
The problem is not really going to be the mic in your headset, although obviously a cheap mic is likely to be lower quality than a good one.
The problem is that recording in to a normal PC is at a very low voltage, so the difference between noise (mains hum, interference etc) and the signal you want is not that great, so when you amplify the signal you also ...
3
You have happened upon one of the edge cases where the sampling frequency above what humans can hear actually does matter. That edge case is FM synthesis.
There is plenty of anecdotal discussion on forums all over the internet that boils down to people arguing about the original Yamaha DX series of FM gear vs softsynths like FM8. These discussions typically ...
3
Note that the Zoom H4n can record 24 bit samples. If you set your input level so that the loudest sound peaks at half your available headroom, your recording will effectively use 23 bits. That's still plenty of bits left if you plan to downsample to 16bits for your finished product. Unless you are looking for the coloration of a specific limiter or preamp, ...
3
To quantify this problem, you want to know the difference between your recording level ("how high the meters are when you're making sound into your mic") and the noise level ("how loud things are when you aren't making noise"). In simple terms, the difference between these is called signal/noise ratio, or SNR. When you boost the signal, you are also boosting ...
3
While I think that the answer provided by leftaroundabout is totally spot on and correct, I would like to add the following to simplify:
Think of the equalizer as set of filters. What are you filtering? The musical or sonic spectra, typically in the 20 Hz to 20 K Hz range (human hearing). Each filter addresses a select band of frequencies from this spectra.
...
3
Practically any time you convert from one lossy format to another, you will lose quality. How much quality, and what precise effect it will have depends on your encoder, and its algorithm.
It also depends on what you mean by quality. Are you referring to bitrate? Or the encoder settings used to generate the file?
It may be possible to transcode from one ...
3
Technically, the PMD661 and that mic should be clean as a whistle. Maybe your gain is turned up too high? I don't know if changing the recording method would change that, but it could if you use a device with a better mic pre-amp, so the gain isn't so high and the hiss isn't introduced. Personally, I would try borrow another device and see if the hiss ...
3
Sometimes a pop filter is not enough.
If you have an audio engineer on site monitoring the signal while the singers are recording this live or in a studio have the engineer work out a hand signal that can be interpreted as "back off from the mic".
Perhaps a reverse high five or a simple hand retracting would work. The key here is to have both the audio ...
3
Something that helps in recording studios is to get people to sing over the mic rather than directly into it. You'll notice that they generally hang a condensor mic from the roof and position it at eye level, higher than the mouth. This gets rid of all sorts of pops and plosives, as the wind generated from speaking doesn't crash into the mic.
For a speaking ...
3
There's no magic bullet, but here's a few things you can try.
First you'll need a program to edit the audio. I'm on a mac and use Apple's Logic. Its a couple hundred bucks and very powerful for the cost.
To clean up the hiss noise that was undoubtedly recorded with the laptop mic, use a de-noiser plug-in. Plug ins for audio work similarly to layer ...
3
You are right!
The quality of the audio will be decreased if the volume lowered by software means. As you have correctly assumed, reducing volume in software is actually similar to reducing the bit depth.
Generally, every 6 dB of attenuation is equivalent to reducing the bit depth by one.
Brief explanation: Max volume level of your sound card output, ...
2
There isn't a single, simple answer for this question. This forum post explains:
You can estimate the approximate Audio information [1] content of FM by using the bandwidth and dynamic range. FM pretty much tops out at about 15K and the dynamic range is about 70db. Each bit represents approx 6db so we need just under 12 bits and a sampling rate of 30K.
...
2
"Drags" and "buzzes" sound to me like a buffer problem. While this isn't necessarily part of Kontakt per se, it is sensitive to your free computer resources and perhaps Kontakt (combined with whatever software you've got going in the background) has used up enough resources that it's becoming hard to keep processing the buffer. This problem isn't uncommon ...
2
Check this Wikipedia article
The article doesn't go into full detail, but let's address yo your problem:
YES Audio loses quality very easily specially when changing its sampling rate and in your particular case it probably has to do with aliasing, which is a consequence of re sampling audio.
Basically - as people tell me at least - you should only raise ...
2
Adobe Audition CS5.5 has a Match Volume feature for batch processing multiple files.
Open Audition
Effects > Match Volume
Drag the files you want to match into the window
Set the level you want to match all the clips to
Click Match Volume
Alternatively you can click Batch and you will get options to rename the input & output files and choose the ...
2
I'm afraid this might sound cliche or condescending, but try them out and trust your ears. If you knew what each mic was you could look up all their specs, but that information may not be all that useful to you. I could, and would be happy to answer questions about what those specs mean, but I think it's much more useful to you to just trust that the one ...
2
neroAacEnc is also available for Linux. If you find that is works for you in OS X then you can use ffmpeg to pipe to this encoder and then mux with ffmpeg:
ffmpeg -i input -f wav - | neroAacEnc -ignorelength -if - -of audio.mp4
ffmpeg -i video.mp4 -i audio.mp4 -c copy -map 0 output.mkv
However, FFmpeg now also supports the external encoding library from ...
2
The Zoom H4N has built in preamps and microphones.
Preamp is short for pre amplifier. It raises the signal from mic level (very low) to line level so it can be recorded or sent to an amplifier or a signal processing device. One reason you might want to use an external preamp with the Zoom is because it might have a limiter to prevent clipping, or a high ...
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