16

FLAC compression levels are (only) a trade of between encoding time and file size. The decoding time is pretty much independent of compression rate. In the following I will refer to the compression levels 0, ..., 8 as FLAC-0, ..., FLAC-8. In short: I recommend FLAC-4! The Easy Solutions Obviously: If I don't care about encoding time and since space is ...


8

It's important to remember that a compressor does not make a sound louder, it reduces dynamic range. By bringing down the peak values, you give yourself more headroom to bring up the overall signal level (so the quieter parts can be louder). It is that "make up gain" which makes a sound louder, not the act of compression. The choice between using ...


5

In theory there is no difference. In practice, using parallel compression gives you more control over how the sound is processed going in and coming out. For example,you could process just the low end of a drum loop to add more power to the the kick. Also, since it uses bussing you could send the output of multiple tracks to a single parallel compressor and ...


5

No. Once lossy formats are encoded, any data not saved within the file is lost. You could convert a lossy MP3 to a WAV or an M4A file but the quality of the WAV or M4A would be exactly the same as the original MP3.


5

Here, is there any common low-end level that every song should has? No or Does it just set by ear? Yes. In fact, it should be set by ear, not by eyes. Don't bother with what the spectrum analyzer tells you, it is dumb and doesn't know or say very much. You have to listen for the right sounds. or Does it depend on the music genres? Definitely. Hip-...


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

It kinda matters what order you put them in but everyone has a different order that they like and then sometimes you adjust for issues. Most Channel strips let you change the order of at least some of the modules. I tend to default to EQ/DYN/De-es/multiband but everyone is different. Sometimes you might need 2 of certain processes, 1 to fix an issue, ...


4

GIGO [garbage in, garbage out] does apply... ...however - a good mix, master or especially restoration engineer could pull apparent newness out of a low quality file - that's essentially the same task as recovering a track from an old 78 RPM record, or remastering the Beatles albums from the original multitracks. You work to eliminate the 'bad' & ...


3

I would apply them in order of de-esser, EQ and compressor. The first two could be done in either order, but the compressor should generally be last. You could EQ with or without the De-esser applied, but the De-esser will offer you less control over the sound than a good EQ. The compression should be last because it deals with overall signal power, which ...


3

First of all, there are no absolutes. Recording is an art form (albeit one with technical considerations), and there are no rules with art. Your only guidelines are "does this sound good" and "do I like it?". Aside from that, you're free to experiment. Experimenting, however, does have its pitfalls. By placing anything in the signal chain which will alter ...


3

i used no compression. just a slow attack on the ADSR :-)


3

The problem is that Google Play Music does not currently have support for Podcasts. Due to this, it is going to be next to impossible for you to get a file down to the necessary size without breaking it into sections and publishing them as an album. Even at the 320kbps quality that Google Play produces for end user consumption from the FLAC files, you ...


3

It can actually be a very good idea to put two or more compressors (with different parameter settings) in sequence. For instance, limiting1 can allow you to make drums louder, but it tends to make transients a bit murky. One way to counteract this is to put a slower compressor (possibly with sidechain low-cut) before the limiter, to “pump” these ...


3

As you specify using a 'stereo linked compressor', neither the timing relation or the level differences between both channels should be modified. Therefore, the stereo width should be unaffected. That's what 'stereo linked compressors' are made for. Nevertheless, there could be unexpected perceptive differences, especially if you're hitting hard into the ...


3

Multiband Compression is not something Standard.Most of the time it's a complex procedure but you have to keep in mind that you are doing this to a whole mix which means your dB reduction should be very careful or you might kill it. That's a general guideline though , there are techniques that use aggressive/heavy compression. I would recommend reading the ...


3

I think what you are looking for is an expander, here's a link that explains it Quoted here: Audio expansion means to expand the dynamic range of a signal. It is basically the opposite of audio compression. Like compressors and limiters, an audio expander has an adjustable threshold and ratio. Whereas compression and limiting take effect whenever the ...


3

Yes. Without decompression a flac file contains just pseudo-random data. A media player requires an ordered stream of PCM stereo audio samples in order for them to be converted to analogue audio via a DAC chip. Therefore, we implement 'codecs' (Code-Decode) to handle the decoding of the flac data to the ordered stream of PCM samples we require to be fed to ...


3

Briefly, threshold is the level at which the compressor is triggered, attack is how quickly the compressor reacts to the trigger, and release is how long the compressor stays activated before returning to the normal level. Essentially you set the threshold to the sound level that you want to start to compress, then adjust the attack and release to shape the ...


3

They give a small improvement to the compression at the cost of encoding time. As neither requires specifying --lax, they do not make the resulting files less compliant. -8e, -8p and -8ep could be thought of as compression levels better/slower than -8. They both independently brute-force a setting/value in order to find the best combination for every ...


2

Maybe you could try to reduce the sample rate. Does they say anything about a required sample rate ? I guess you're recording in 44Hz, you should try to reduce it to 22Hz. It will lower the quality, but it can be unnoticed, especially if it's mostly voice, and your file will be around half the initial size. I just tried it on a FLAC song and the result is ...


2

FLAC is a lossless audio encoder so the bitrate is nothing other than an indication of the compression ratio FLAC has been able to achieve. Since FLAC is not allowed to change the material at all, the better the material matches its predictions (and the more time FLAC may spend on compression), the higher the compression ratio will be. That means that ...


2

There are any number of possible reasons. It could be lower noise, it could have a softer feel to activation of the compressor. It may be able to activate the compression more quickly. It may simply be better marketing allowing for a higher price based on brand. You'd really have to compare the specific models to see the differences.


2

If you love the sound that you're getting, you're doing it right. What matters, is that you know your intended purpose for that vocal. Will you need the flexibility on the back end or do you already know the sound you're looking for? Compression and EQ before Tape was, and still is, a common signal chain in certain musical situations when you already ...


2

I'm honestly not sure what you are asking here. Compression does not increase the "volume" of a signal, it decreases it. Compression makes a quiet portion of the sounds louder relative to a louder portion by reducing the signal strength when the signal strength is high. Often a gain is applied after compression to keep the signal strength up, but this ...


2

Ogg Vorbis doesn't suffer from much of the horriffic issues that occur with MP3, and generally produces smaller files. Hi Quality Voice Recorder will do what you want. If you need smaller, obviously gsm which is the encoding used for mobile phone calls should be suffficient, but there is less software compatible with this format.


2

After a bit more looking into it and experimenting I've found that the middle equation should be replaced with xG - ((R - 1)/R)(xG - T + W/2)^2/(2W) And it works correctly


2

To my knowledge, this is not possible. To add and crossfade audio data requires to decode the mp3 binary stream to pcm, then re-encode to mp3 the resulting pcm stream if you want an mp3 output. As mp3 is not very good at cascading, you will (probably) have degradation, the overall degradation will depend on the input and output mp3 bitrate.


2

There are programs that can split and concatenate MP3s without re-encoding. So one thing you could do is to split a few seconds off every MP3 at the beginning and end, apply your cross-fades to them (with re-encoding), and then concatenate everything. Regarding "how much worse" with re-encoding: If you choose a high bitrate for re-encoding, it shouldn't be ...


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