I found an interesting article here.
1. Do you agree with him ?
2. If so, why don't implement thoose functions directly inside encoders ?
Tnx
Full Version: Web Mastering...
1. No. This guy does not know what he's talking about. He's way off. For one thing, he thinks you can "increase dynamic range" by compressing the music.
2. Any adjustments you make to the audio before compressing it to a low quality file "for analog modem delivery" must be done based on that particular piece of audio. No encoder can make those decisions. The closest you'll get is using presets for various types of program material, and that's not very close either.
-- Uosdwis
2. Any adjustments you make to the audio before compressing it to a low quality file "for analog modem delivery" must be done based on that particular piece of audio. No encoder can make those decisions. The closest you'll get is using presets for various types of program material, and that's not very close either.
-- Uosdwis
2. I asked for options, not for decisions...
Anyway, i have coded some *really different* material with OGGDropXPd @ 32 KHz, mono, q=-1, lowpass=19500 and sounds really good to me.
Any other XPerience ?
Anyway, i have coded some *really different* material with OGGDropXPd @ 32 KHz, mono, q=-1, lowpass=19500 and sounds really good to me.
Any other XPerience ?
Well, obviously, if you include an option to enchance audio for modem streaming, the encoder (in reality, the programmer) will make decisions whether or not to normalize, compress or equalize the audio it is fed. This is what you meant by "options", isn't it?
What I was saying is that it's not likely you can make options or presets like these that will work as intended on whatever people may feed the encoder, except marginally. This is also why mastering plugins and Finalizers does not replace mastering engineers.
-- Uosdwis
What I was saying is that it's not likely you can make options or presets like these that will work as intended on whatever people may feed the encoder, except marginally. This is also why mastering plugins and Finalizers does not replace mastering engineers.
-- Uosdwis
QUOTE
During the audio encoding process for analog modem delivery, your sound files go through drastic changes as bits of data are thrown out and compressed to reduce file sizes. This process eliminates certain frequencies and accentuates background noises that you may not have noticed in the original audio source. Generally, the two areas to focus on are the highest ¼ of your audio frequencies and the lowest ¼ of your audio frequencies. The higher frequencies (above 10,000Hz) tend to carry a lot of background noise, such as hiss and scratchiness in an audio file. Whereas the lower audio frequencies (below 75Hz) tend to carry a lot of the rumble that can occur from background noises, such as H/VAC systems or microphone rumble caused by wind.
To correct this problem, eliminate the low-end frequencies between 20Hz and 75Hz and the highest frequencies above 10,000Hz with the equalization faders in your sound editor. Then boost the low-end bass frequencies between 75Hz and 110Hz to compensate for the elimination of the bass sounds below 75Hz. It's also a good idea to boost the mid-range vocal frequencies between 2,000Hz and 7,000Hz to add brightness to a mix and make the vocals clearer after Web conversion, as shown in Figure 1.
To correct this problem, eliminate the low-end frequencies between 20Hz and 75Hz and the highest frequencies above 10,000Hz with the equalization faders in your sound editor. Then boost the low-end bass frequencies between 75Hz and 110Hz to compensate for the elimination of the bass sounds below 75Hz. It's also a good idea to boost the mid-range vocal frequencies between 2,000Hz and 7,000Hz to add brightness to a mix and make the vocals clearer after Web conversion, as shown in Figure 1.
It seems this tutorial is aimed at very low bitrate compression, for 56k modem streaming. Dismiss the We'll have you posting high-quality sound files in no time statement !
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