That is a DVD recorder, it has no CD recording capability. 48kHz@16bit is the base samplerate and bit depth for DVD audio streams no matter what format it is - MP2, AC3, DTS, or PCM (96kHz@24bit is also possible by spec, but I don't recall seeing it as an option outside of DVD-Audio discs). Some speaker setups allow outputting 96kHz audio, and the 96kHz functions of the player allow for output to those types of setups. DVD-Audio discs can support the same ranges, including the higher samplerates such as 96kHz, but they also have dedicated audio streams and use MLP (Meridian Lossless Packing) encoding usually. An extended variant of MLP is the Dolby TrueHD format used on HD-DVD and Blu-ray, both of which also support Enhanced AC3 (EAC3), a similarly extended version of AC3 that also is advertised as being more efficient at encoding the stream intrinsically and not simply supporting more channels and whatnot.post-it wrote:.. well if its any conciliation to your waiting, I'll have to back down from my claims. Even after buying a DCD/CD recorder, "pdf Manual is available on-line
http://www.funai-corp.com/6pdf/om/WV10D6.pdf " I could not find reference to the 44.1k question!
.. in the manual that came with this expensive liar at $49.95 it only mentions 96k and 48k sample rates .. pages 15 & 67 of the paperback!
.. I also asked the salesman why the CD's said Remastered, 32bit and AC_5 3D sound tracks if CD's only record at 44.1k sampling rate 16bit Stereo .. he and a few other salesmen broke-out laughing T_T ever feel like you've been set-up by someone?
.. then they laughed even harder once I said that THAT was False Advertising!
.. I have no proof of what it is that I have.
.. I have no way of proving that a CD Player can decode AC_5 because everything today is all DVD Recorders and for Computers.
.. somethings not right here; how can it be available when you walk into the stores but be un-explainable because of wording on-line !?!??
.. I'll just shut my mouth and enjoy "what the Internet says is not possible." Another-Words, you win! Model SV2000 ISBN: 0-53818-57028-9 CD Player.
48 kHz?
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Qyot27
- Surreptitious fluffy bunny
- Joined: Fri Aug 30, 2002 12:08 pm
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- Location: St. Pete, FL
- Contact:
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post-it
- Joined: Wed Jul 17, 2002 5:21 am
- Status: Hunting Tanks
- Location: Chilliwack - Fishing
.. ok, so they were DVD Audio "CD's" .. T_T .. I never heard of such non-sense; and, of course, a DVD Player could play them back >_<
.. now on to the real problem with 48k audio from a CD; in many of the recordings from CD's, when Cool Edit is used, the CD being recorded actually sounds better than the original CD itself. My question is, "where are these LC-AAC Encoders" that I've been reading about?
D-sound - sucks!
MP3sx - sucks!
Nero AAC - doesn't sound right!
.. also, I'd like to Encode the Audio Tracks that I've remixxed at 96k sampling rate; not 48k nor 44.1k .. is there an Encoder for That kind of MP3/MP4 Audio ??
.. now on to the real problem with 48k audio from a CD; in many of the recordings from CD's, when Cool Edit is used, the CD being recorded actually sounds better than the original CD itself. My question is, "where are these LC-AAC Encoders" that I've been reading about?
D-sound - sucks!
MP3sx - sucks!
Nero AAC - doesn't sound right!
.. also, I'd like to Encode the Audio Tracks that I've remixxed at 96k sampling rate; not 48k nor 44.1k .. is there an Encoder for That kind of MP3/MP4 Audio ??
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bobbias
- Joined: Fri Aug 01, 2003 10:15 pm
- Location: Midland, Ontario, Canada
Yes, there is such thing as a DVD Music Disk. They are different than conventional CDs. DVDs are actually very different from CDs in quite a few ways (Physically they are thicker (IIRC), have a different material that the information is recorded on, etc.)
It's no use trying to convert the music to 96khz. The only reason anyone uses 96khz is when they are directly recording sound. The reason they use higher sample rates is this:
The maximum frequency that can be recorded at a given sample rate is exactly half the sample rate. Therefore, at 44.1khz, the maximum frequency that could be recorded is 22.05khz. The human ear cannot hear more than about 20khz, so 2.05khz is easily enough, and anything above that is going into sounds we cannot hear.
Also, if you have a recording is 44.1khz, and convert it to 96khz or any other higher sample rate, there will be no difference in the sound. These frequencies above 22.05khz will NOT magically appear. You will simply make the file bigger, and waste space.
SO, that means that you should give up working in anything above 44.1khz, because really, there's no point to it. Would it make sense to take an old AMV in some crappy format that looks decent, and encode it to some lossless format like Lagarith just because Lagarith has the potential to look better? You'd just end up wasting space.
Encoding anything that was originally at 44.1khz into 96khz is pretty much the same as taking that old file and encoding it to Lagarith. They're both an exercise in futility and proof that whoever is doing that doesn't know enough about what they're working with. Next time, do some more research.
It's no use trying to convert the music to 96khz. The only reason anyone uses 96khz is when they are directly recording sound. The reason they use higher sample rates is this:
The maximum frequency that can be recorded at a given sample rate is exactly half the sample rate. Therefore, at 44.1khz, the maximum frequency that could be recorded is 22.05khz. The human ear cannot hear more than about 20khz, so 2.05khz is easily enough, and anything above that is going into sounds we cannot hear.
Also, if you have a recording is 44.1khz, and convert it to 96khz or any other higher sample rate, there will be no difference in the sound. These frequencies above 22.05khz will NOT magically appear. You will simply make the file bigger, and waste space.
SO, that means that you should give up working in anything above 44.1khz, because really, there's no point to it. Would it make sense to take an old AMV in some crappy format that looks decent, and encode it to some lossless format like Lagarith just because Lagarith has the potential to look better? You'd just end up wasting space.
Encoding anything that was originally at 44.1khz into 96khz is pretty much the same as taking that old file and encoding it to Lagarith. They're both an exercise in futility and proof that whoever is doing that doesn't know enough about what they're working with. Next time, do some more research.
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post-it
- Joined: Wed Jul 17, 2002 5:21 am
- Status: Hunting Tanks
- Location: Chilliwack - Fishing
.. converting something to 96k is not what I'm trying to seek here, my goal is Clarity in the higher frequency ranges of what I'm creating. If a tree of bells are shaken and twin-high-octave-tubular-bells are being played, 44.1k has a hard time distinguishing the difference. 48k is better, at least the sample rate makes it sound better. 88.2k is still experimental. I would prefer having the option of encoding things at 92k if only for experimental reasons.
.. My theory is that the higher the sample rate, the cleaner the encode and playback will be. Why bother with such things? The human voice and kettle drums are over samples so much that they can sound realistic at 44.1k however, high pitched bells and flutes sound washed-out from under-sampling.
.. At thirty to sixty inches per second, in a recording studio, not much is missed! I'd like to have that option available on my computer when I use it as a recoding studio. That sample rate would be 336k which is totally silly; so for now I'll settle for 96k sample rate at 640k bit rate joint stereo. My question is, "where is this magical encoder that can handle that sample rate?"
.. If the designed software is available I'd like to know where its at. If not, then the program needs to be made. 96K at +600k are the current standards for direct recordings .. or haven't you done your research Mr. bobbias. a 16k signal at 44.1k sample rate is only 2.5 samplings -- I want to do better than that!
.. My theory is that the higher the sample rate, the cleaner the encode and playback will be. Why bother with such things? The human voice and kettle drums are over samples so much that they can sound realistic at 44.1k however, high pitched bells and flutes sound washed-out from under-sampling.
.. At thirty to sixty inches per second, in a recording studio, not much is missed! I'd like to have that option available on my computer when I use it as a recoding studio. That sample rate would be 336k which is totally silly; so for now I'll settle for 96k sample rate at 640k bit rate joint stereo. My question is, "where is this magical encoder that can handle that sample rate?"
.. If the designed software is available I'd like to know where its at. If not, then the program needs to be made. 96K at +600k are the current standards for direct recordings .. or haven't you done your research Mr. bobbias. a 16k signal at 44.1k sample rate is only 2.5 samplings -- I want to do better than that!
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bobbias
- Joined: Fri Aug 01, 2003 10:15 pm
- Location: Midland, Ontario, Canada
Unless you are recording directly from a microphone, there is absolutely no point in dealing with anything above 44.1khz. I used 96khz as a reference, but the same sort of thing applies to anything over 44.1khz. If you are referring to a recording already on your computer, or a CD, then you are shit out of luck, because converting from 44.1khz to anything above it will NOT save the sounds you'd like to hear.
If you are recording these sounds on your computer, you need to know what sort of capabilities your sound card has, because not everything can record at anything higher than 44.1khz, which would mean you're shit out of luck for recording too. Nowadays most cards can record at that, but not everyone has new cards.
In any case, unless you are recording the tubular bells yourself, there's no way you're going to hear those extra sounds, which, by the way, are out of the human hearing range. Anything that 48k and above records that is beyond what 44.1k can handle CAN'T BE HEARD BY THE HUMAN EAR.
If you're concerned with how certain things sound in different codecs, like comparing AAC, MP3, and OGG Vorbis, I cound understand talking about certain frequencies being lost and distorted in things like tubular bells and such, because "Smearing" in the high end is a common issue that CODECS have. Unfortunately, an MP3 of tubular bells in 44.1k will sound the same as one in 48k if they are both at the same bitrate. If you have the CD and it sounds normal on the CD, but it sounds like crap in an MP3, at 44.1k, then it's not the sample rate or the bit rate. It's the fact that when MP3 compresses sound, it throws out certain things that our brain can't quite understand properly. Unfortunately, stuff like tubular bells get distorted fairly badly. Try saving it in FLAC or some other lossless codec.
"At thirty to sixty inches per second, in a recording studio, not much is missed!"
What the hell does that mean? I spent a semester in our highschool studio and I have absolutely no idea what you're talking about.
If you are recording these sounds on your computer, you need to know what sort of capabilities your sound card has, because not everything can record at anything higher than 44.1khz, which would mean you're shit out of luck for recording too. Nowadays most cards can record at that, but not everyone has new cards.
In any case, unless you are recording the tubular bells yourself, there's no way you're going to hear those extra sounds, which, by the way, are out of the human hearing range. Anything that 48k and above records that is beyond what 44.1k can handle CAN'T BE HEARD BY THE HUMAN EAR.
If you're concerned with how certain things sound in different codecs, like comparing AAC, MP3, and OGG Vorbis, I cound understand talking about certain frequencies being lost and distorted in things like tubular bells and such, because "Smearing" in the high end is a common issue that CODECS have. Unfortunately, an MP3 of tubular bells in 44.1k will sound the same as one in 48k if they are both at the same bitrate. If you have the CD and it sounds normal on the CD, but it sounds like crap in an MP3, at 44.1k, then it's not the sample rate or the bit rate. It's the fact that when MP3 compresses sound, it throws out certain things that our brain can't quite understand properly. Unfortunately, stuff like tubular bells get distorted fairly badly. Try saving it in FLAC or some other lossless codec.
"At thirty to sixty inches per second, in a recording studio, not much is missed!"
What the hell does that mean? I spent a semester in our highschool studio and I have absolutely no idea what you're talking about.
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post-it
- Joined: Wed Jul 17, 2002 5:21 am
- Status: Hunting Tanks
- Location: Chilliwack - Fishing
.. thought so; newbe!
.. Recording Studios use 2" perforated reel to reels which move at 30" to 60 " per second to kill background wow & flutter plus hiss is no longer an issue which was common knowledge to anyone in the Music Business up until a few years ago. The overall response was 2hz - 70khz at 32 channels with a depth of 160dbs. Larger studios ran at 3" which didn't really matter at that time because DBX was now helping to make background noise a thing of the past plus it helped improve the high end frequencies by encoding the bells shimmers - previously only heard inside the studio itself.
.. When CD's were introduced, the first thing everyone noticed was how FLAT and LIFELESS the recordings were; they were right - the shimmer and subtle vibrations were all but erased from the CD itself. CD's had a limitation of sixty DBs which was masked by the first fifty DBs being understood as a threshold .. artist's were not buying this theory back then .. we still don't buy that theory today. however, the tone-def number-crunchers kept people off balance long enough for CD's to become popular and make 44.1k a standard.
.. The advantage of Reel to Reels, Cassette and LP's were their ability to hide signals like "Quad Stereo" and "Stereo Quad" plus "FM carriers for CD-4" in those tapes and on those records. CD's came along and didn't have the bandwidth to hold those carriers. 48k was tryed and it failed. 88.2k would allow "SQ" to respond again and 92k allowed "QS" and "DBX" to be down-mixxed to 68K signals once more.
.. Digitally, 48k sampling rates just do not hold enough frequency response to be useful in the Music industry. Yamaha came out with a standard that the Digital world totally overlooked called VQF .. but you already know about its devide in threes' pattern don't you.
.. Once again I ask the question of, "Where Are These 96k Encoders" on the internet; if they make them then where are they, if not we'll have to program them.
.. I'm sure that LAME has a 96k sample rate joint stereo encoders somewhere; all I'm asking is "where are they hiding it at" ??
.. If such a beast has not been made; then make the bloody thing already - quit screwing around like BOB the BIAS here and get crack'n.
~fin~
.. Recording Studios use 2" perforated reel to reels which move at 30" to 60 " per second to kill background wow & flutter plus hiss is no longer an issue which was common knowledge to anyone in the Music Business up until a few years ago. The overall response was 2hz - 70khz at 32 channels with a depth of 160dbs. Larger studios ran at 3" which didn't really matter at that time because DBX was now helping to make background noise a thing of the past plus it helped improve the high end frequencies by encoding the bells shimmers - previously only heard inside the studio itself.
.. When CD's were introduced, the first thing everyone noticed was how FLAT and LIFELESS the recordings were; they were right - the shimmer and subtle vibrations were all but erased from the CD itself. CD's had a limitation of sixty DBs which was masked by the first fifty DBs being understood as a threshold .. artist's were not buying this theory back then .. we still don't buy that theory today. however, the tone-def number-crunchers kept people off balance long enough for CD's to become popular and make 44.1k a standard.
.. The advantage of Reel to Reels, Cassette and LP's were their ability to hide signals like "Quad Stereo" and "Stereo Quad" plus "FM carriers for CD-4" in those tapes and on those records. CD's came along and didn't have the bandwidth to hold those carriers. 48k was tryed and it failed. 88.2k would allow "SQ" to respond again and 92k allowed "QS" and "DBX" to be down-mixxed to 68K signals once more.
.. Digitally, 48k sampling rates just do not hold enough frequency response to be useful in the Music industry. Yamaha came out with a standard that the Digital world totally overlooked called VQF .. but you already know about its devide in threes' pattern don't you.
.. Once again I ask the question of, "Where Are These 96k Encoders" on the internet; if they make them then where are they, if not we'll have to program them.
.. I'm sure that LAME has a 96k sample rate joint stereo encoders somewhere; all I'm asking is "where are they hiding it at" ??
.. If such a beast has not been made; then make the bloody thing already - quit screwing around like BOB the BIAS here and get crack'n.
~fin~
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bobbias
- Joined: Fri Aug 01, 2003 10:15 pm
- Location: Midland, Ontario, Canada
I see what you're talking about here now (though your bad english makes it hard for me to read at times).
I will admit I don't know much about the older equipment used in recording. Nor do I know everything there is to know. I couldn't understand what you were asking, and assumed you were trying to upsample a 44.1khz CD signal to a higher samplerate in order to gain the frequency response which was lost when it recorded as 44.1khz.
I know CDs aren't the best, but they DO happen to be quite a bit more durable than tape.
Honestly, if you're going to be recording things in 96k, it's not worth it to use MP3. If you didn't already know MP3 is getting pretty old now, and the psychoacoustics systems that it uses to encode the data and drop certain frequencies is VERY damaging to anything over 20khz.
If you're so concerned about all of this, head over to the hydrogenaudio forums at hydrogenaudio.org. They know a boatload more than I do, and should be able to help you out better than I can.
I will admit I don't know much about the older equipment used in recording. Nor do I know everything there is to know. I couldn't understand what you were asking, and assumed you were trying to upsample a 44.1khz CD signal to a higher samplerate in order to gain the frequency response which was lost when it recorded as 44.1khz.
I know CDs aren't the best, but they DO happen to be quite a bit more durable than tape.
Honestly, if you're going to be recording things in 96k, it's not worth it to use MP3. If you didn't already know MP3 is getting pretty old now, and the psychoacoustics systems that it uses to encode the data and drop certain frequencies is VERY damaging to anything over 20khz.
If you're so concerned about all of this, head over to the hydrogenaudio forums at hydrogenaudio.org. They know a boatload more than I do, and should be able to help you out better than I can.
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Qyot27
- Surreptitious fluffy bunny
- Joined: Fri Aug 30, 2002 12:08 pm
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- Contact:
I actually don't even think 96kHz is in the MP3 specifications - I believe MP3's abilities cut off at 48kHz, 16-bit. AAC supports the higher frequency ranges and higher bit depths, but then again, AAC is also newer than MP3 is. Vorbis can do it as well, and it's listed in WMA encoding lists but I'd just as soon stay away from WMA period.bobbias wrote:Honestly, if you're going to be recording things in 96k, it's not worth it to use MP3.
I can see support for higher frequencies and bit depths coming in handy for converting between the MLP streams on DVD-Audio discs and the high-def disc formats without the need to downsample the frequency and drop the bit depth, as streams of 96kHz@24-bit calibre are available there. Regular CDs, however, don't support that, and neither do DVD-Video discs (as far as MLP goes anyway, as it eats up too much space to be feasible outside of dedicated DVD-Audio discs and/or DualDisc implementations).
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bobbias
- Joined: Fri Aug 01, 2003 10:15 pm
- Location: Midland, Ontario, Canada
Well, I don't know all the technical specs for MP3. I was just saying that in general, if your source is at 44.1khz, there's no point in upsampling it because all that will do is increase the file size without doing anything with the extra frequency response. Not to mention that MP3 would damage a good portion of the extra frequency response range anyway because of the psychoacoustics it employs. AAC and Vorbis are both considered better compression schemes than MP3 because they are considered transparent at lower bitrates, the sound overall better at low bitrates compared to MP3, and they tend to sound better at any given rate than MP3.
I don't know all the technical specs, but I have hung around hydrogenaudio long enough to find out that MP3 rarely EVER sounds as good as almost every single alternative there is. (I also know from personal experience.)
I don't know all the technical specs, but I have hung around hydrogenaudio long enough to find out that MP3 rarely EVER sounds as good as almost every single alternative there is. (I also know from personal experience.)
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Qyot27
- Surreptitious fluffy bunny
- Joined: Fri Aug 30, 2002 12:08 pm
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- Location: St. Pete, FL
- Contact:
True. The only examples nowadays where I upconvert from any frequency to 48 is in the case of encoding video for DVD (or in expectation for DVD), as 44.1kHz isn't supported. Whether one's standalone supports it is a different issue, but I still go for the standard itself because then I have a guarantee that it'll play regardless.bobbias wrote:Well, I don't know all the technical specs for MP3. I was just saying that in general, if your source is at 44.1khz, there's no point in upsampling it because all that will do is increase the file size without doing anything with the extra frequency response. Not to mention that MP3 would damage a good portion of the extra frequency response range anyway because of the psychoacoustics it employs. AAC and Vorbis are both considered better compression schemes than MP3 because they are considered transparent at lower bitrates, the sound overall better at low bitrates compared to MP3, and they tend to sound better at any given rate than MP3.
I don't know all the technical specs, but I have hung around hydrogenaudio long enough to find out that MP3 rarely EVER sounds as good as almost every single alternative there is. (I also know from personal experience.)