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Sound Recording For Atrac? Better Quality?

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Guest tony wong

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Guest tony wong

what will happen to the quality if the music company themselves make the MD?

what I mean is not directly copying from the PCM/cd, but a real mastering/mixing for MD

is it gonna have better quality on sound?

hope people who don't understand this topic just don't reply

yes, u can ask

just don't leave ur non-sense reply

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I don't think so, because in that case they have two options:

1) Mix and master as they've always done (going for the best sound quality for their music) and then live with any artifacts from the ATRAC encoding

2) Make consessions to the quality by attenuating parts of the music/spectrum that will be a problem for the ATRAC encoder.

In case 2), it is no longer about making the music sound best, but it is about making the music sound best on a certain device/medium, which is not really the best philisophy behind mixing/mastering.

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Guest tony wong

I don't think so, because in that case they have two options:

1) Mix and master as they've always done (going for the best sound quality for their music) and then live with any artifacts from the ATRAC encoding

2) Make consessions to the quality by attenuating parts of the music/spectrum that will be a problem for the ATRAC encoder.

In case 2), it is no longer about making the music sound best, but it is about making the music sound best on a certain device/medium, which is not really the best philisophy behind mixing/mastering.

thx

what I mean is they mastered for MD 2nd time after they first made the mastering for CD

anyway, thx god for there's someone know what I am talking about laugh.gif

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thx

what I mean is they mastered for MD 2nd time after they first made the mastering for CD

anyway, thx god for there's someone know what I am talking about laugh.gif

ok

I don't really know what a mixing/master engineer should do to make a piece of music sound optimal on a MD. I do a lot of mixing (and mastering also) myself and if I'd would mix a piece of music for low-bandwidth purposes (for instance MP3@64), I would attenuate things like the hi-hat, the snare, and sounds with a sharp attack in general to avoid ringing and pre-echo. But, I would do that because I'd know that the resulting audio will sound bad anyway, and it would be just a matter of letting it sound as less bad as possible.

With MD, however, we're speaking of high bitrates (at least I assume that you're aiming at SP or higher) and in that case the music should sound like the artists/producers/engineers have meant it to sound. In my opinion, if a medium isn't capable of reproducing that sound, it's a problem of the medium and not of the content. Fortunately, to some extent SP is perfectly capable of reproducing PCM quality.

A final remark: I'm speaking about media that use compression here. In other fields, re-mastering is very common. For instance, I believe that audiotracks of movies get a re-mastering treatment for usage on a DVD, because in a typical home theater less dynamics are required than in a cinema.

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I believe you would get better quality. An MD is a recorder and makes the logic decisions about what data to record, what to throw away in real time. This happens even if you record a CD the recording is a known length (by you, not the MD). A record company could use a ATRAC algorithm that examined the entire track before encoding. Since you could have more powerful processors I would think it possible to have a more sophisticated algorithm also.

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Guest tony wong

I believe you would get better quality.  An MD is a recorder and makes the logic decisions about what data to record, what to throw away in real time.  This happens even if you record a CD the recording is a known length (by you, not the MD).  A record company could use a ATRAC algorithm that examined the entire track before encoding. Since you could have more powerful processors I would think it possible to have a more sophisticated algorithm also.

no, I mean a different(I haven't stated this before, sorry)

but urs would work as well

what I am thinking of is : fighting with ATRAC's weak point

u know in some certain of the sound will be weakened by ATRAC, just push it higher

but until this moment, I know I am stupid

we do make copy with cd, 16bit 44.1kHz sound

the source would be different for music company mastering

and the equipment is different

their equipment is possibly able to accept 30 bit 60kHz(take it for example, just example) and the unit possibly take longer time to process these source

probably 1hour for 5 sec

sorry for being stupid this time

conclusion : personal recording from consumer MD equipment(yes even the most expensive units) will be much worst than mastering MD material from music company

assumed that they do have some equipment like I stated above unsure.gif

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I don't think so, because in that case they have two options:

1) Mix and master as they've always done (going for the best sound quality for their music) and then live with any artifacts from the ATRAC encoding

2) Make consessions to the quality by attenuating parts of the music/spectrum that will be a problem for the ATRAC encoder.

In case 2), it is no longer about making the music sound best, but it is about making the music sound best on a certain device/medium, which is not really the best philisophy behind mixing/mastering.

I'd said you're somewhere on the fence in terms of right/wrong here, bug80.

Traditionally, the mastering process has had two 'layers': the first is to make the stereo master, the 2nd is to optimise a duplication master for each medium the recording will be distribuuted on.

These days this is generally for CD, and most mastering engineers end up "bit pushing" their recordings to make them sound as loud as possible on a 16-bit medium.

The end result is basically the same as back when they made separate distro masters for cassette and LP: each one is compressed slightly differently, and probably has a slightly different EQ curve as well.

The process of bitpushing can both add to and subtract from normal artifacting, for example. On one hand, it can add distortion and artifacting caused by clipped peaks that PCM can play out much as limited analogue signals do, but most forms of discrete transforms will make only into distortion. On another hand, compression and limiting [what bitpushing really is] can both exaggerate [under certain circumstances] and lessen the effects that extremely hard transients have on discrete transform encoding [such as ringing and pre-echo].

Basically, your option [2] there is pretty much what they've always done, and what they still do, though to my knowledge they don't master things to compensate for artifacting with compression, exactly. Most of the time now they appear to just try and squeeze tha maximum volume possible out of the medium, at the expense of dynamic range.

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what I am thinking of is : fighting with ATRAC's weak point u know in some certain of the sound will be weakened by ATRAC, just push it higher

Compensating levels for compression loss would have one effect and one effect only: increasing the audible artifacts.

Okay, two effects: it would also utterly destroy the original timbre [tonal balance] of the recording.

An example -

One of the most common masking principles used by lossy compression algorithms is that of louder adjacent tones masking quieter ones, i.e. a tone at 1,000Hz at -6dBfs will mask out one at 1,050Hz and -12dBfs [grossly simplified, yes].

What you're suggesting is that the quieter tone be boosted so that the encoder is forced to include its information, correct?

The result of this would be to:

1) destroy the timbre of the instrument that adjacent tone/harmonic comes from

2) overstress the encoder, which now has more data that it wants to encode, which then makes priority decisions almost impossible, and most likely ends up with far worse artifacting, since the encoder no whas too few bits to store the "desired" information.

There is only one true way, IMO, to compensate for a lossy encoder's artifacting, and that is to increase the bitrate until the artifacting is made impossible by the actual presence of the data whose loss causes the artifacts.

i.e. record uncompressed.

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I'd said you're somewhere on the fence in terms of right/wrong here, bug80.

Traditionally, the mastering process has had two 'layers': the first is to make the stereo master, the 2nd is to optimise a duplication master for each medium the recording will be distribuuted on.

These days this is generally for CD, and most mastering engineers end up "bit pushing" their recordings to make them sound as loud as possible on a 16-bit medium. 

The end result is basically the same as back when they made separate distro masters for cassette and LP: each one is compressed slightly differently, and probably has a slightly different EQ curve as well.

The process of bitpushing can both add to and subtract from normal artifacting, for example.  On one hand, it can add distortion and artifacting caused by clipped peaks that PCM can play out much as limited analogue signals do, but most forms of discrete transforms will make only into distortion.  On another hand, compression and limiting [what bitpushing really is] can both exaggerate [under certain circumstances] and lessen the effects that extremely hard transients have on discrete transform encoding [such as ringing and pre-echo]. 

Basically, your option [2] there is pretty much what they've always done, and what they still do, though to my knowledge they don't master things to compensate for artifacting with compression, exactly.  Most of the time now they appear to just try and squeeze tha maximum volume possible out of the medium, at the expense of dynamic range.

Yes you´re right. Nowadays it is no longer about the music but about making the record as loud as possible in the mastering process (grandpa speaks, haha). It´s a shame really.

By the way, I´ve heard that even different masters are made for different continents (for example: a master with a small low/high boost for the VS and a master with a small mid boost for Europe), because opinions about what sounds good differs between different countries.

If you look at it that way, then mastering for a certain medium isn´t such a bad idea (if we don´t take the question into account if there exists a method to overcome compression artifacts in a clever way).

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Different masters have traditionally been made for [this is not always the case any more]:

* radio

* cassette

* LP

* CD

&c.

As far as differences between countries, I have heard of this but never really noted it in action. It would make sense under circumstances, as the equipment [especially for broadcasting] in different regions use different loading/EQ curves [loading being more or less dynamic range compression].

My guess is that this would be more for standards compliance than to suit the tastes of listeners in a particular region, though I may likely be wrong on this.

Terms to look up if you're interested include AES/EBU and SMPTE.

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