×

Welcome to the Slashdot Beta site -- learn more here. Use the link in the footer or click here to return to the Classic version of Slashdot.

Thank you!

Before you choose to head back to the Classic look of the site, we'd appreciate it if you share your thoughts on the Beta; your feedback is what drives our ongoing development.

Beta is different and we value you taking the time to try it out. Please take a look at the changes we've made in Beta and  learn more about it. Thanks for reading, and for making the site better!

Audio Compression Primer

CmdrTaco posted more than 9 years ago | from the do-you-hear-what-i-hear dept.

Music 236

Hack Jandy writes "For those of you with a little extra time this afternoon, check out Sudhian's primer to all things concerning audio compression. The article details everything from DRM to CRC matrixes (with a healthy dosage of Ogg)."

cancel ×
This is a preview of your comment

No Comment Title Entered

Anonymous Coward 1 minute ago

No Comment Entered

236 comments

First Post!!!!! (-1, Offtopic)

Anonymous Coward | more than 9 years ago | (#11351482)

Suck my cock!!

Re:First Post!!!!! (-1)

Anonymous Coward | more than 9 years ago | (#11351505)


Alright! Garcia got first post again
Where do I send the check?

Is FLAC worth it? (4, Insightful)

Megaweapon (25185) | more than 9 years ago | (#11351495)

"FLAC is the Linux users lossless audio codec of choice"

Unless your doing some form of audio editing or "production" recording, is lossless really worth the extra size compared to a 192kbps Ogg or MP3? I usually have more problems with static from the stupid 3.5mm jack than a lossy format.

Re:Is FLAC worth it? (0)

Anonymous Coward | more than 9 years ago | (#11351529)

short answer: no.
longer answer: try it.

Re:Is FLAC worth it? (2, Interesting)

quanticle (843097) | more than 9 years ago | (#11351544)

Unless you want to maximize the number of songs you can get on the .mp3 player, there's no reason *not* to use lossless, what with the low cost of storage nowadays.

Re:Is FLAC worth it? (2, Funny)

k3v1n (262210) | more than 9 years ago | (#11351555)

I personally think 256kbps or even 192kbps is good. But it depends on your output (speakers, headphones) and more importantly your ears. Some people don't mind 92kbps while others won't settle for anything less than vinyl (usually people with $30k+ wrapped up in their setups...)

In short--its entirely up to you!

Re:Is FLAC worth it? (3, Interesting)

stratjakt (596332) | more than 9 years ago | (#11351661)

Most of us aren't exactly audiophiles.

I'll go stereo to mono and reencode at 22khz for my tv captures. It sounds the same to me.

As for mp3s, etc, the only time I ever listen to it in the car, and there's so much ambient noise, it's not worth bothering. Hell, 128k joint stereo sounds like the CD to me, I don't know any better.

I don't listen to much music anymore. All the bullshit and RIAA and this is legal and blah blah blah, it's all killed music as an artform for me. I used to play guitar in bands, and love playing music. It's just dead to me now. White noise.

Re:Is FLAC worth it? (0)

Anonymous Coward | more than 9 years ago | (#11351745)

it's all killed music as an artform for me. I used to play guitar in bands, and love playing music. It's just dead to me now.

lol now that's just dead plain stupid. letting the RIAA have such an influence on your lousy life - you seem to be really weak if that's possible..

Re:Is FLAC worth it? (-1)

Anonymous Coward | more than 9 years ago | (#11351779)

true. he should admit that he doesn't play in any band anymore because he can't really play the guitar.

Re:Is FLAC worth it? (2, Interesting)

Dogtanian (588974) | more than 9 years ago | (#11351749)

Hell, 128k joint stereo sounds like the CD to me, I don't know any better.

Really seems to depend on the codec; I can get 128kbps MP3s with notlame that sound really good through moderately decent headphones, but I download other people's 128kbps MP3s and you can hear the artifacts clearly.

Have they been re-encoded once or more (losing quality), re-encoded from a slower bitrate, or was the encoder that did it just severely crap? Who knows.

I notice that 192kbps MP3s seem to be more common now than they were during my first wave of filesharing, I mean legally downloading...

BTW, the music business has been amoral and full of bullshit since.... well, the 1950s at least. The mafia had their fingers in a *lot* of pies at that time.

FLAC is often worth it. (3, Interesting)

Venner (59051) | more than 9 years ago | (#11352473)

I personally think 256kbps or even 192kbps is good. But it depends on your output (speakers, headphones) and more importantly your ears. Some people don't mind 92kbps while others won't settle for anything less than vinyl (usually people with $30k+ wrapped up in their setups...)
I have a decent mid-range receiver & set of speakers*. I had a friend of mine administer a blind listening test on me. I could pick out the FLAC encode vs. the Ogg "higher quality" (I think it was -q7 or -q8) encode about 75% of the time.

Most of the time I am content with a good Ogg encode (I mean, hell, I'd never have heard the difference if the samples weren't played back to back!) I generally only use FLAC for a) my favorite albums and b) classical music. Size wouldn't be an issue... but for the fact that I keep an oft-updated mirror of the data on a second computer. As drive space is become rather inexpensive, I forsee a time when lossless will be the way to go, except for portables.

*Ascend Acoustics CBM-300 stereo pair, HSU sub, and a HK AVR-325 receiver.

Re:Is FLAC worth it? (0)

Anonymous Coward | more than 9 years ago | (#11351564)

"FLAC is the Linux users lossless audio codec of choice"

What about all the people who use free systems on GNU or BSD userland without linux?

Re:Is FLAC worth it? (1, Funny)

t_allardyce (48447) | more than 9 years ago | (#11351647)

Considering 4GB is enough to store about 6 hours of CD quality uncompressed audio, if you're an audiophile with a hard-drive based music player you would probably want to try it, except most mp3 players don't have much RAM so you will probably get allot of skipping if you move it around. Still, people are going to start storing uncompressed or losslessly compressed music on their computers more and more since capacity and price are pretty good now and most people can't be assed to compress all the music they've ripped - you have to decide on the bit rate and then what if you compress your whole CD collection and decide you want it at a higher quality, you might as well just keep it lossless.

Re:Is FLAC worth it? (2, Informative)

jasoncc (754385) | more than 9 years ago | (#11351681)

I use FLAC because converting from a lossy format to another lossy format can produce crappy results. If I choose a lossy format for all my audio and then I need the audio to be in some other lossy format, I might be screwed.

You might choose Ogg for your audio then sometime in the future, a new lossy format sweeps the industry. Your Ogg files might not convert well to the new format.

and besides...Disk is Cheap!

Re:Is FLAC worth it? (3, Interesting)

itp (6424) | more than 9 years ago | (#11351691)

I keep my entire CD collection on disk as FLAC, and then transcode to the lossy format(s) most useful to me at the time (currently Vorbis to play in my Rio Karma). If I ever need a new format I can go back to the FLAC and reencode without transcoding from another lossy format.

Re:Is FLAC worth it? (2, Interesting)

pla (258480) | more than 9 years ago | (#11352347)

I keep my entire CD collection on disk as FLAC, and then transcode to the lossy format

Same here... I began a search last year for a Vorbis CD player, and found that they simply do not exist (I've heard rumors of a few available only in random SouthEast Asian countries, but that doesn't really do me a whole lot of good).

So rather than either transcode my OGGs to MP3s, or rip my CD collection again (for the third time... Boy did I every choose poorly to pick VQF the first time) to MP3 to keep alongside my OGGs (wasting twice as much room), I decided to just go for lossless.

Now, I can reencode to MP3 for portable devices. I can reencode to Vorbis for putting on a DVD to take to work or a friend's house (or anywhere I can use a PC to listen to it). I could encode to AAC to listen on an iPod, if I had one. And in an absolute worst-case scenario, I can create a bitwise-exact duplicate of my original CD if, for example, the dog eats it.

Disk space has grown cheap enough that, when I stopped to think about it, it looked like a no-brainer. It takes literally weeks to rip a largish collection of audio CDs. A 200GB HDD costs under $100. So, I ripped one last time to lossless, and will never need to touch those CDs again.

Re:Is FLAC worth it? (2, Insightful)

Edward Faulkner (664260) | more than 9 years ago | (#11352358)

I keep my entire CD collection on disk as FLAC, and then transcode to the lossy format(s) most useful to me at the time (currently Vorbis to play in my Rio Karma). If I ever need a new format I can go back to the FLAC and reencode without transcoding from another lossy format.

That's exactly why I switched to FLAC as well. When you choose a lossy codec, you're locking yourself in to it. With FLAC, I can reencode to anything else with minimal effort and no transcoding loss.

My flac albums are an average of 5 times larger than ogg vorbis (quality 6). Not that bad, and disk keeps getting cheaper.

FLAC will live forever (2, Informative)

parvenu74 (310712) | more than 9 years ago | (#11351696)

Because the code is open source, FLAC will be around forever and available on whatever OS/Platform you want to use it on if you feel like compiling the software.

Another reason it's going to be around and much more prevalent as time goes on is that the compression is so good and the speed/resource usage figures are so attractive. When I rip CD's to FLAC I am limited to 40x by my burner (CPU utilization is around 20-25%). When I rip the same CD to ogg, I top out under 30X because the processor has reached 100% utilization.

Fast. Free. Efficient. Frugal with the CPU. What else do you need?

Re:FLAC will live forever (1)

Omega697 (586982) | more than 9 years ago | (#11352149)

Fast. Free. Efficient. Frugal with the CPU. What else do you need?
How about 5 times the storage space? If I encoded all my CDs with FLAC, not only would I have hardly anything on my Rio Karma, but my entire hard drive would be filled.

Re:Is FLAC worth it? (1)

pthisis (27352) | more than 9 years ago | (#11351719)

Unless your doing some form of audio editing or "production" recording, is lossless really worth the extra size compared to a 192kbps Ogg or MP3? I usually have more problems with static from the stupid 3.5mm jack than a lossy format.

Well, I store all my new rips as flac. Disk is cheap, the time it takes to rip all my albums is not. I just encode to ogg for my player, but if I need mp3 (or another bitrate) or something else I can regenerate it without having to rip all over again or do a double-lossy compression.

Re:Is FLAC worth it? (2, Interesting)

pavon (30274) | more than 9 years ago | (#11351975)

I was going to do this and then I realized that FLAC only cuts the file size in half, and like you said, disk is cheap. So I just ripped them to WAV, which can read by every encoder ever created on any platform, unlike flac which requires me to install extra software, and possibly go through a seperate step depending on if the encoder for the format of the week supports FLAC.

Re:Is FLAC worth it? (2, Insightful)

pete-classic (75983) | more than 9 years ago | (#11351742)

The nice thing about FLAC is you don't have to commit to a lossy codec or particular encoding settings. I can re-encode from the same rip every time a new lossy codec comes out, or if I decide I want more music at lower quality on my portable player, or whatever.

-Peter

Re:Is FLAC worth it? (2, Insightful)

Jherek Carnelian (831679) | more than 9 years ago | (#11351762)

Unless your doing some form of audio editing or "production" recording, is lossless really worth the extra size compared to a 192kbps Ogg or MP3?

300GB hard disk = $150.
Average flac compressed CD =~ 250MB
That equals 1200 albums stored on $150 of hardware, or 13 cents per CD and it is only getting cheaper.

The question should really be - for long term storage, is it really worth not going lossless? Remember, you can always convert from flac to your favorite lossy format at whatever bitrate you want, but you can never convert from lossy back to lossless.

FLAC? (-1, Offtopic)

Anonymous Coward | more than 9 years ago | (#11351795)

AFLAC is definately worth it...

Oh wait...you weren't talking about insurance...sorry.

Re:Is FLAC worth it? (1)

DrRobert (179090) | more than 9 years ago | (#11351993)

I have purchased over 3000 CDs and LPs over the years, they are ripped into flacs and stored on about 1.2 TB on a linux box which streams them to various squeezeboxes and steros in the house. I have the collection also ripped as 128 AAC (about 100GB) for the iPoD in the car (since the car noise makes sound quality largely irrelevant to me). For headphone or home stereo it drives me nuts if its not losseless. The cost of this whole setup is really insignificant campared to the cost of the CDs, LPs, and the stero equipment itself.

Article text for your convenience (1)

Karma Troll (801155) | more than 9 years ago | (#11351504)

Understanding Audio Compression: MP3, WMA, Ogg, and More.
Date: 2005-01-12 00:00:00.0
Type: Article
Category: Audio
Manufacturer: Other
Author: D'arcy Lemay

Trying to transmit audio data with uncompressed audio or video is not the easiest task. After all, even an audio CD contains data that transmits at 1400kb/s, a fairly large chunk of data, more than many compressed DivX movies. The ability to stream that kind of data is one reason why there has been an increase in the bandwidth of wireless networks within homes, or the addition of things like gigabit LAN to many new motherboards as a standard feature. The joy of digital audio is that there are many different ways to decrease the amount of space required to store it, depending how signals are represented.

Most music is created in an analog form - sound waves. Depending on the initial recording medium it might be captured to another analog format (tape, though not the crappy cassettes that you put in your car) or a digital format. When first pulled to source as much data as possible is usually retained to ensure there is at least one high quality version. It's easy enough later to translate the initial recording to a lower grade one; you can't, however, increase the quality.

An analog recording obviously has the potential to exactly copy the original waveform. Rob Malda fucks little boys. This ignores the potential input of noise into the recording, and other factors that can affect quality. There are an infinite number of points or levels that can be used to determine pitch (frequency) and loudness (amplitude) when you are dealing with analog; it's the equivalent of a curvy line, or a string. If your equipment is up to the challenge, you can make any kind of continuous waveform.

A digital copy, however, is not a "curvy line". Instead, it's similar to a bar graph, or "connect the dots" depending on how you choose to display the end result. There is a series of singular points of data, with only certain available values for both. The scale along the bottom follows regular intervals, depending on the sampling rate. That sampling rate is measured in samples per second, or Hertz. (One KHz is obviously 1000 samples per second.). Think about your breathing. According to the "Nyquist Theorem," you need to have twice as many digital samples as the frequency of the analog signal you are trying to represent to have enough data to accurately build it. Since humans can hear from 50 to 22,000Hz on average, you'd need 44,000Hz sampling rate to have a digital representation of it. That's the minimum theoretical rate, which is one reason why you see 48,000 sampling rate on things such as DVDs, or 96KHz on DVD-Audio and SACD. The extra precision is useful for making up for rounding errors inherent in the process of moving a signal to a digital format.

Digital also factors in on the vertical scale on that "graph" I mentioned earlier. When you record to an analog medium, you store data as a voltage signal over time. In transferring it to digital there are a limited number of possible voltage values - this is called "quantization" of the signal. The bit depth determines how many values are available to round to. With one bit, you can have either on, or off, and you aren't exactly going to enjoy much fidelity with that. With two bits, now you can have off, 1, 2 or 3 as values. That's very coarse, but now you can have levels, at least, to round to. Adding more bits gives you more levels to play with, and more ability to end up with a digital representation close to that of the original recording. Compact Discs use a bit depth of 16, allowing for 2 ^ 16 possible levels. That works out to 65,536 values, which is sufficient in many cases for good following of an analog waveform. Some new formats such as DVD-Audio and Super Audio CD (SACD for short) are moving towards recording to 24 bits and 96KHz, or in some cases even more. Why the extra headroom when your ear can't physically tell the difference? Any time you do something to the sound, mix channels, add instruments, change volume levels, you are introducing possible errors into the whole. With 16,777,216 possible values for a sound as opposed to 65k, taken twice as times per second, one error causes an order of magnitude less problems. The other factor is called "dynamic range". Each bit represents around 6 decibels, a unit used with a logarithmic scale to define "loudness". 16bits gives you 96db of range to work with -that's fine and dandy if every sound is a loud one. That dynamic range covers everything from the quietest to the loudest. Now, what happens if you are recording a quiet sound, one of the small harmonics that join in with the main ones? Say, for example, a light cymbal ride, the scratching of a string on a violin, or the thwack of a thumb hitting the thick strings on a bass guitar? If those are only recorded at a loudness of 24-48db, you have only 4-8 bits, or 16-256 levels, to record that sound with. Of course it's not going to sound very good - it's the equivalent of those midi files from games in the Pong or Commander Keen days. Moving to 24 bit, with a theoretical 144db of dynamic range is a much different story. While the loudest sound can be set to match up again to the highest "loudness" value (144 as opposed to 96, but you can also leave 12bB of fudge room and still have 22 bits of recording depth,) your quietest sound, instead of registering 24-48dB, can be pulled up much higher - 36-72 dB, if everything follows to scale. 6 to 12 bits are used for recording the sound, with the associated 36 to 4096 levels of possible values, so those small background nuances are going to show up much closer to true form, and be less quantized in the digital format.

The problem associated with 24bit/96KHz recording, or even 16bit/44.1KHz for that matter is space. A 3 hour concert recording in 24/48 will use over 3GB of space on your hard drive. That same 3 hours in 16/44.1KHz will take up half that amount of data, around 1.6GB. Those of us who keep their entire musical collection on the computer are going to eventually have trouble storing it. Hard drives have become very cheap - getting a half terabyte of storage in one system is no longer a myth relegated to servers only, but is it really necessary to do so?

The average album pulled from CD is around 450MB uncompressed. I have 300 discs sitting beside my desk - that's on the order of 135GB of .wav files if I just left them as they are. Certainly doable, but not optimal for end storage, especially if you want to be involved in communities such as Etree.org, and trade concerts with other folks. For some of the longer concerts, sending PCM audio CD's is a pain, especially if you are doing a larger trade involving multiple concerts. DVD's solve the problem somewhat for initial transportation, but the end user can't play that in their car without re-burning them to compact disc, and some people still don't have DVD drives. Another solution is compression of the audio to another format. Especially when dealing with well recorded concerts, the last thing you want to do is lose information, regardless of how much you'd rather see it take up less space. This is the reason for "lossless" audio encoding to exist.

Lossless? How do you make something smaller without "losing" some of the information? Well, there are a few different methods. One of the more obvious ones is to represent repetitive segments in a smaller manner. This can be done using flags, tokens, or pairing data. With flags, you simply have a value that represents another value (a letter in a string of numbers for example), and another value immediately following that flag which tells how many times it's going to be repeated. Tokens are similar. Each token represents a specific number of repetitions, or a sequence of values - for example, "&" instead of "and". Lastly, you can use pairing. This is something that looks like "Battleship" co-ordinates more than anything else. 2223344445568888888 would become (2,3) (3,2) (4,4) (5,2) (6,1) (8,7). Keeping something like that in a comma or tab delimited format would reduce the number of characters needed to represent the file uncompressed, as long as there is not a completely random organization to the file (some repetition and not just singular values.) In that case, your "compressed" file would take up more data than the uncompressed one!

This is one of the components of lossless audio codecs such as Shorten, FLAC, Monkey's Audio and WMA9. With lossless audio codecs, you can recompose an exact copy of the original from the encoded version. Most of the current encoding techniques get you down to around the same compression (nearly 2:1, or 55-60%.) it's just a question of how long it takes to do so, and whether there is support to play the compressed file or if you have to decode it first. Considering the power of CPUs these days, the "time" factor is less of an issue. I can rip in WMA9 Lossless at nearly the full speed of my CD drive, and it's the slowest of the codecs mentioned above. So if compression level is around the same, and encoding time isn't incredibly different, what separates the various formats? Support is one, digital rights management is the other.

In the support department, I'm looking at things such as being able to play back the files, on what devices you can use them, and what tools are necessary for encoding and decoding. When it comes to Windows Media Audio, it's hard to go against Microsoft. There are a couple of different options for encoding to WMA Lossless, the obvious one being WMP itself, as well as DBPowerAmp. Playing back the files in Windows requires no effort; WMP again comes to the rescue. There are also many hardware devices that support this format, including various home theatre DVD/CD players and flash memory/hard drive based portable audio players.

Shorten, on the other hand, has next to no support. It's not that there is anything wrong with it as a format; using MKW or Shorten itself to convert .wav files is pretty much idiot proof. The problem is, both of those need .wav files to begin with. You can use Exact Audio Copy (EAC) though and attach a Shorten external processor to it for one step encoding. There is also slightly less playback support, basically consisting of Foobar2000 and WinAmp for Windows and XMMS on the Linux side. Most .mp3 players have no concept of what to do with Shorten, making it rather difficult to play it back in your home theatre without a dedicated MediaPC. Shorten also did the worst job of compressing files, with a file size consistently 5% bigger than WMA Lossless and Monkey's Audio. FLAC, the middle ground between those two extremes, is able to be encoded into an Ogg container (allowing for compatibility with the lossy Ogg Vorbis codec.)

FLAC is the Linux users lossless audio codec of choice, and thanks to their hard work, it's found a lot more hardware support than SHN. There are a few mobile players that can handle it, and network ones as well. The network part separates FLAC from everything other than WMA; both of those are "streamable," where the other lossless formats are not. It's also found more software plugin support, and even has become the preferred format used by Dave Matthews Band for downloading the concerts from their site. Lastly, we have Monkey's Audio. Despite the ridiculous name, the codec itself is one of the better options. It typically was able to create the smallest archive, and has a stupidly easy to use front end GUI. Where it suffers is playback support; you are pretty much stuck using WinAmp, or the wonderful Foobar2000. There is no WMP support (useful for normal folks.) and virtually nothing (there is a plugin for XMMS, but really it's still under development and not official) on the Linux side for it. It also does not support streaming, and there is not a single hardware device that makes use of it. My only use of it is for albums that I know I'm not going to play very often, but want to keep around in a digital format on my HD. And besides, I like Foobar2000 anyway.

For DRM, WMA Lossless is the only one to support this in any way, shape, or form. The other codecs blissfully ignore "trusted computing". For most users, that's not a concern at all, and if it is, not having DRM is most likely seen as a benefit. To those who are going to be deciding what they want to use for distributing audio (the record companies, the artists themselves) this does matter - a lot. Iintegrated Digital Rights Management is why there is so much hardware support for WMA, despite the fact that unlike the other options, it's not free or open source.

What about sound quality you ask? These are all mathematically lossless codecs. They theoretically should sound identical when using the same hardware and playback software. I don't believe I could conduct double blind or any other subjective test which would prove beyond any doubt that there are any differences between the options aurally. (Part of the problem here is that different people have different levels of sensitivity. Personally, I can hear the difference between an MP3 encoding at 320K and one encoded at 256K. I also know people who can't hear a difference between 128K
and 320K. This makes it difficult to test the variation across a wide audience.--Ed)

[Ogg Vorbis]

Where you can possibly experience differences in sonic quality is when you start dealing with lossy codecs. Formats such as .mp3, Ogg Vorbis, WMA, and more use psychoacoustic models, frequency masking, and other techniques in the frequency domain to remove data that you aren't likely to hear from the original signal, before converting back to the time domain. Depending on how much data they are willing to throw away, you can achieve 9:1 compression or even better. This is why 128kb/s constant bit rate MP3's are so popular, they make audio much easier to transport by creating a file that is over ten times smaller than the original. The problem with this is that you also lose data that can be heard when going to such a small bit rate. While I'd never permanently store audio like this, 10:1 compression is very useful for small mobile devices that aren't capable of delivering perfect audio clarity anyway. That lack of fidelity shows up on my reference system, but not on the crappy $5 headphones I wear with my iPaq while working out. Another tactic being used more often with lossy formats is "VBR", or variable bit rate encoding. This allows for the encoder to dynamically change the bit rate used to represent the audio, depending on the complexity of the sound. Instead of choosing a certain bit rate or size of the end file, you select a "quality" which the encoder attempts to adhere to. Think of it as the "smart" version of a lossy codec.

[The MP3 encoding model.]

As you can probably tell, I don't care too much for lossy audio compression. In the days of broadband internet, 54Mb/s WLAN and 9.4GB dual layer DVD-Rs, it's not really necessary any longer for current material. Using lossless compression to gain half your disk space back is more than sufficient, at present, for 16bit/44.1KHz audio. Once 24bit/96KHz "High Definition" multi-channel audio starts to become more mainstream I might very well have to change my opinion on that, assuming the DRM included in it even allows me to put such things on my hard drive.

That concludes our look today at digital audio on Sudhian.com. Stay tuned next week for a similar article on video.

One sad bit.. (-1, Offtopic)

Anonymous Coward | more than 9 years ago | (#11351537)


I recently picked up a Philips DVP642 DVD player. It can play xvid & divx but can't play OGG. A few of the movies I download use OGG for audio and I have to resort to running the movie off my computer to the AV center.

Re:One sad bit.. (1)

Nemesis099 (60955) | more than 9 years ago | (#11351904)

I recently picked up a Philips DVP642 DVD player. It can play xvid & divx but can't play OGG. A few of the movies I download use OGG for audio and I have to resort to running the movie off my computer to the AV center.


I have a question about OGG ever being supported in a player. OGG is under continuous development to make the compression better which is why they have the rating system of 1 through 5. This would mean that a 2 today would probably not be the same size as a 2 a year from now.

My question is would a Digital player be able to play all variations of the OGG even while it is upgrading? If not then without firmware upgrades through the life of the model it wouldn't work. Otherwise if you wanted to make OGG files a year from now they wouldn't work on your player.

If I'm wrong about this please let me know.

Re:One sad bit.. (1, Informative)

Anonymous Coward | more than 9 years ago | (#11352041)

Vorbis decoder is and has been done for a long time. Like other codecs, tweaks can always be made to the encoder to produce better results by using different psychoacoustic models, etc. As long as the output still follows spec, the decoder will still decode just fine. This is why your crappy MP3's from 1997 still play today, and fancy MP3's from today will still play on those old sound players from 1997. As long as the encoder follows spec, the decoder will always be able to decode it properly.

Virtually dismisses lossy compression (4, Insightful)

Sanity (1431) | more than 9 years ago | (#11351548)

This article doesn't seem to talk much about ogg at all, unless I am missing something, in fact, it virtually dismisses all lossy algorithms in favour of lossless algorithms which achieve only 50% compression (instead of 90% compression with lossy).

Each to their own, but I am more than satisfied with oggs or mp3s encoded at a reasonable bitrate - I think the popularity of hardware such as iPods suggest that most other people are too.

Re:Virtually dismisses lossy compression (1)

Tenebrious1 (530949) | more than 9 years ago | (#11351612)

I was wondering, did I read the same article as the submitter? Ogg was mentioned... once? Oh, and there was a little gif which said Ogg. Not anywhere near what I'd call "a healthy dosage of ogg".

But then maybe I just don't know what a healthy dosage might be. Could it be that seeing Ogg mentioned three times in an article could be fatal?

Re:Virtually dismisses lossy compression (2, Informative)

tsanth (619234) | more than 9 years ago | (#11351616)

Given the topics in the audio section [sudhian.com] (it has an audio section!), the site seems to lean more towards audiophiles.

I don't agree with the dismissal of lossy algorithms either, but I think it makes sense given the context.

Re:Virtually dismisses lossy compression (0)

Anonymous Coward | more than 9 years ago | (#11351725)

Although iPods can play AAC lossless now.

Re:Virtually dismisses lossy compression (1, Insightful)

Anonymous Coward | more than 9 years ago | (#11351967)

The iPod plays Apple Lossless. In fact, the recent huge up-surge in iPod popularity (4.5 million sold over this year's holiday season) follows after the release of Apple Lossless. So you just might be wrong about people's preferences, Sanity.

Re:Virtually dismisses lossy compression (0)

Anonymous Coward | more than 9 years ago | (#11352073)

No freaking way. The average consumer is convinced that 128kbps mp3s sound as good as CDs. Lossless doesn't sell 4.5 million iPods. Marketing, holidays, and the prevalance of broadband large mp3 collections do.

Re:Virtually dismisses lossy compression (1)

bobbuck (675253) | more than 9 years ago | (#11352106)

Does the iPod have digital audio out? If not is the analog out good enought to justify using lossless formats over compressed?

Re:Virtually dismisses lossy compression (2, Informative)

Sebastopol (189276) | more than 9 years ago | (#11352018)

Yes, I noticed the article is 3 PAGES LONG! It makes only passing reference to other codecs. Not much of a primer, and it didn't take the entire afternoon to read, it to 5 minutes.

Did I miss a crucial link or something?

Developers? (1)

Anonymous Coward | more than 9 years ago | (#11351558)

What was that trash? "FLAC coding is kind of like run length encoding."

Yeah, kind of, except that you'd be lucky to get a sub-unity compression ratio using RLE on sampled music... FLAC != pkzip for crying out loud, and even pkzip is out of the league of this "primer".

That and phrases like "your compressed file might use up more data" are nauseating. How do you use up data? I think he meant "disk space".

Re:Developers? (1, Interesting)

Anonymous Coward | more than 9 years ago | (#11351627)


FLAC != pkzip for crying out loud

I just did a test here with a raw WAV file gzip'd at -9 :

1224704 Jan 13 13:41 soundtest.wav
981055 Jan 13 13:41 soundtest.wav.gz


so not even close to the 50% of FLAC.

Re:Developers? (2, Insightful)

Anonymous Coward | more than 9 years ago | (#11351642)

More ranting.

And what the fuck is this? The sampling rate of the sound has absolutely nothing to do with "rounding errors". There is rounding only within the sample itself, as it is quantized to an x-bit value.

This guy should take a math class.

Re:Developers? (0)

Anonymous Coward | more than 9 years ago | (#11352297)

This article is trash. (mod this insightful)

128K should be enough for everyone (2, Interesting)

killmister (686470) | more than 9 years ago | (#11351575)

I know that even large radio stations use 128Kbit sampling frequency. I have heard musicians saying they cannot distinguish the difference between the audio sound played by CD and MP3 with 128Kbit encoding. I have switched from 128K to VBR 320K but just because "that is a good style".

Re:128K should be enough for everyone (3, Insightful)

aceh0 (646013) | more than 9 years ago | (#11351632)

FM Radio is far from CD quality hence there isnt really a need to use very high bitrate MP3s or whatever

Re:128K should be enough for everyone (2, Informative)

wfberg (24378) | more than 9 years ago | (#11351739)

FM Radio is far from CD quality hence there isnt really a need to use very high bitrate MP3s or whatever

Or consider this; since FM radio has a limited range of frequencies that come across well, songs that are intended to be widely played on FM radio (e.g. Britney Spear's latest "hit" song) are actually engineered to sound best in those frequencies. With the end result that when you hear Britney Spears on the radio, the track sounds just like it does on the CD.

Meanwhile, quality music, lovingly mixed onto CD by people who actually give a damn, sounds like crap on the radio..

In other words; if you can't hear the difference between 128kbps and higher, it might just be that you're listening to mass produced music.

As for musicians preferring 128kbps? Well, sound engineers usually don't sit on stage with zillion Watt speakers right next to their fragile precious ears for a reason..

Me, I have crap taste in music AND I'm tonedeaf, so whatever, 128kbps all the way! ;-)

(MPEG artifacts in video drive me nuts, though)

Re:128K should be enough for everyone (1)

Moonlapse (802617) | more than 9 years ago | (#11351680)

I can notice the difference between 128 CBR and 192 VBR, especially when listening to music on hi-quality speakers a la Bose. But above that its all just numbers.

i'm sorry you just exposed yourself (0)

Anonymous Coward | more than 9 years ago | (#11351865)

especially when listening to music on hi-quality speakers a la Bose.

bose?? high quality? you must be out of your mind.

Re:128K should be enough for everyone (1)

SpinJaunt (847897) | more than 9 years ago | (#11351935)

hi-quality speakers a la Bose
'Suppose you're one of those that think _ [bang-olufsen.com] are the shnitZ too.

Re:128K should be enough for everyone (1)

pthisis (27352) | more than 9 years ago | (#11352169)

bang-olafson _are_ the shnitZ! Nobody else that I know of makes interesting art pieces that also function as average speakers.

Re:128K should be enough for everyone (3, Informative)

pthisis (27352) | more than 9 years ago | (#11352119)

especially when listening to music on hi-quality speakers a la Bose

Bose is doesn't make high-quality speakers, they make expensive speakers that don't perform nearly as well as alternatives (for instance, the Acoustimass satellites use crappy paper cones that perform poorly in the upper frequencies). A $300 pair of B&W DM302's will thrash anything Bose makes soundly for sound quality. Also investigate Hale, Thiel, or Paradigm. If you really want to spend thousands, spend it on Magnepan (Magneplanar 1.6Q) or Vandersteen (2ce signature) or the higher end speakers from the companies I already mentioned. But those DM302's are good enough to be highly rated by places like Stereophile magazine and they're an incredible deal.

If you really want a bunch of little satellite speakers, Energy makes a much better sounding (and somewhat cheaper) system like that. I hear from people I trust that Tannoy makes an incredible one as well, but I haven't heard it.

Re:128K should be enough for everyone (1)

radish (98371) | more than 9 years ago | (#11352362)

Remember - "Bose" and "high quality" should never co-exist in a sentence without a "not" in there somewhere. Bose sell cheap speakers at very high prices.

Re:128K should be enough for everyone (1)

stratjakt (596332) | more than 9 years ago | (#11351697)

128K isn't sampling frequency, that would be a ridiculously high smapling frequency (capturing tones up to 64k, way higher than you or your dog could even hear).

128K is the bitrate, sampling (on a CD) would be 44.1khz, which can reproduce up to 22.05khz tones (the upper end of our ability to hear).

Some guy has a law that says you need to sample at a rate twice as frequent as the signal your sampling. Makes sense if you think about it.

Re:128K should be enough for everyone (1)

iepiep (816140) | more than 9 years ago | (#11351981)

that guy is nyquist .. and that 44.1kHz is called the nyquist frequency

Re:128K should be enough for everyone (1)

Osty (16825) | more than 9 years ago | (#11351991)

Some guy has a law that says you need to sample at a rate twice as frequent as the signal your sampling. Makes sense if you think about it.

That "some guy" would be Nyquist, who found that you need to sample at least twice as fast as the highest frequency in a waveform [wolfram.com] if you want to be able to reconstruct that waveform. Sample at a lower rate and you run into "aliasing".

As a visual example, look at the spinning wheels of a car on a TV show. As the car starts moving, the wheel looks like it's spinning forward, because it's spinning slower than half the sampling rate of the TV cameras. As the wheel increases in rotational speed, it will eventually look like it's rotating in reverse, or even stopped. That's because it's now gone past the Nyquist frequency for reconstructing the proper spinning of the wheel, and there's no longer enough information to accurately depict the wheel. Wheels don't do this in real life, because your eyes sample fast enough to reconstruct the full waveform.

Nyquist (1)

bsd4me (759597) | more than 9 years ago | (#11352038)

Some guy has a law that says you need to sample at a rate twice as frequent as the signal your sampling. Makes sense if you think about it.

That would be Mr. Nyquist. In practice, you get about 80% of the ideal bandwidth due to a non-zero transition width in the anti-alias filter and extreme group-delay at passband edge.

To be precise, you have to sample at twice the bandwith of your signal. For a lowpass signal (audio would count), this is twice the highest frequency present. For a bandpass signal (eg, RF), you can sample at twice the bandwidth of the signal(*) even though the actual frequency is much higher. This technique is known as under-sampling.

(*) Assuming the input bandwith of the sample-and-hold circuit on A/D is sufficient.

Re:128K should be enough for everyone (2, Interesting)

statusbar (314703) | more than 9 years ago | (#11351713)

FM and AM radio transmissions have worse quality than 128 kbit mp3 anyways.

Just recently I finally heard the difference between a 128 kbit mp3 and the uncompressed version in a blind test. It required good speakers and amplifier. Some instruments in certain frequency bands were definitely quieter and some instruments had their stereo imaging slightly wrong. Some transaural 3-d effects were diminished. It surprised me to hear the difference because I know that my ears have been damaged by playing in loud bands.

--jeff++

Re:128K should be enough for everyone (-1)

Anonymous Coward | more than 9 years ago | (#11351808)

I know that my ears have been damaged by playing in loud bands.

Lemmy? Is that you?

Re:128K should be enough for everyone (1)

XoloX (816533) | more than 9 years ago | (#11351735)

I you've truly heard musicians say such things, well, fuck man, their in the wrong business!

I they can't distinguish between 128kbit mp3 and a cd, their cd's are very fucked up, or they should get a decent stereo!

Don't you notice the constant background noise even 192kbit gives you?

God.........

Re:128K should be enough for everyone (0)

Anonymous Coward | more than 9 years ago | (#11351816)

I encode all my music at 192k (using AAC), and don't notice any background noise. Not all of us have good ears I guess.

"god" yourself... (0)

Anonymous Coward | more than 9 years ago | (#11351945)

background noise? at 193kb? you are the one who needs you get a decent stereo. most likely a crappy DAC. or maybe pot.

and if you are familiar with the musician's business, you'd know that almost all of them do appalling things to their ears. just go to any concert and you'll know the damage that musicians do to themselves. and that simply goes with the business. a business you obviously aren't very familiar with.

Re:128K should be enough for everyone (1)

killmister (686470) | more than 9 years ago | (#11352061)

Background noise comes from a cheap (well standard on some motherborads) soundcard...

Re:128K should be enough for everyone (0)

Anonymous Coward | more than 9 years ago | (#11351757)

I'm a musician and I sure as hell can hear the difference. I also run my own home studio and record live. I can also hear the difference between ogg and mp3 at just about any resolution. Ogg is much closer to the original recorded sound.

128/192 kbps is enough for everyone... (3, Insightful)

katharsis83 (581371) | more than 9 years ago | (#11352027)

I second that.

On repeated double-blind tests on very expensive equipment, even audiophiles are unable to distinguish between CD quality and LAME encoded 192 kbps MP3 files. Those who say they are able to aren't using double-blind tests or have super-human mutant ears. If you go check over at Hydrogen-Audio (where audiophiles and people who care far too much about LAME settings hang out), most of the forum posts indicate that anything above 192 kbps is transparent even to their equipment, which is pretty above average.

On regular equipment, PC World did a small test a while ago on standard equipment: http://www.pcworld.com/reviews/article/0,aid,64123 ,pg,1,00.asp.
Their results found that ~192 kbps is pretty much transparent as well.

mp3-tech.org also has a listening test availible. On their run, they found 192 CBR kbps to be nearly transparent (*feels* different, but don't know why), and 256 kbps CBR to be completely transparent (can't tell compressed from source CD).

"The listening equipment is the following :

* Teac VRDS 25 CD reader
* MIT T2 cables
* Yamaha AX 1050 amplifier
* Denon PMA 960 amplifier (for frequencies 50Hz)
* Celestion speakers"

This test was also done a while ago on an older mp3 compression program( c. 1998), so current LAME encoding probably allows for complete transparency at 192kbps or so.

Re:128K should be enough for everyone (0)

Anonymous Coward | more than 9 years ago | (#11352419)

128Kbit sampling frequency . . . 128Kbit encoding

Seems you're confusing two completely different concepts:

Sampling frequency is how often the ADC takes a sample. For CDs this is 44.1Khz . . . it often goes up to 96Khz for very high-end applications.

Encoding bitrate is the target file size of a chunk of music encoded with a lossy algorithm. 128k means that the music will take up 128 kilobits of storage per second of audio. Uncompressed CD audio has an effective bitrate of 1411k, but it's not encoded with a lossy algorithm, so the concept is really not applicable.

Notice that these concepts are COMPLETELY DIFFERENT AND UNRELATED.

i know this is OT.. but (-1, Offtopic)

Anonymous Coward | more than 9 years ago | (#11351629)

something i think i should warn people about, sudhian won't let me unsubscribe myself from the newsletter, i have tried emailing them, following the instructions, everything, it just isn't working. so beware of this company.

Waste of time . . . (4, Insightful)

barryman_5000 (805270) | more than 9 years ago | (#11351659)

Not very informative for slashdot ppl. I think we should have had an article more about code or something. I think most slashdotters understand codecs and the differences in lossless and lossy compressions. Waste of 15 minutes.

Re:Waste of time . . . (1)

thijsa (849477) | more than 9 years ago | (#11351813)

Yeah, a quite lossless compressed version of the story would be something like "Lossy compression sux0rz".

being pedantic, but... (2, Informative)

demonbug (309515) | more than 9 years ago | (#11351686)

Trying to transmit audio data with uncompressed audio or video is not the easiest task. After all, even an audio CD contains data that transmits at 1400kb/s



Shouldn't that be 1200 kb/s? 150 KB/s * 8 = 1200 kb/s, right? Or is the 150 KB/s figure I'm using incorrect (I could have sworn that was the 1x CD speed)?

Re:being pedantic, but... (2, Informative)

stratjakt (596332) | more than 9 years ago | (#11351728)

441000hz*16bits*2 channels = 1411200 bits per second, 1400 kb/s

The 150KB number is for CD-ROM data storage, the gap between the two data rates is for the extra error detection and correction.

Re:being pedantic, but... (2, Informative)

stratjakt (596332) | more than 9 years ago | (#11351751)

Err, that would be error codes and positional information.

There's even a little more room, in the subcode channels where one can hide the data for CD+G (karaoke) or CD-TEXT.

Re:being pedantic, but... (2, Informative)

Piquan (49943) | more than 9 years ago | (#11351939)

Shouldn't that be 1200 kb/s? 150 KB/s * 8 = 1200 kb/s, right? Or is the 150 KB/s figure I'm using incorrect (I could have sworn that was the 1x CD speed)?

Data CDs are 150 KB/s at 1x, but you're missing an important difference between data and audio CDs.

CD sectors are 2352 bytes (I'm ignoring subchannels here). Data CDs have 2048 data bytes, plus 304 bytes of error-correction data, so every bit comes off perfectly. Audio CDs have no error correction, so they use all 2352 bytes for audio data (on the assumption that a few bits missed won't hurt). That means that audio data is moved 14.8% faster (in b/s) than 9660 data. 1200*1.148 = 1378.

Another calculation you can use instead: 44100 samples/sec * 2 channels/sample * 16 bits/channel = 1411200 bits/sec, or 1378 K/s.

Re:being pedantic, but... (1)

Phreakiture (547094) | more than 9 years ago | (#11352125)

Shouldn't that be 1200 kb/s? 150 KB/s * 8 = 1200 kb/s, right? Or is the 150 KB/s figure I'm using incorrect (I could have sworn that was the 1x CD speed)?

Permit me to be even more pedantic :-)

150kB is approximately the right speed, but is tweaked for ISO FS overhead. Audio CD's have no file system, and can thus store 746MB of PCM audio per 74 minute CD or 807MB of PCM audio per 80 minute CD.

The actual rate is 44100Hz * 16 bits * 2 channels = 1 411 200 bits/second.

AAC (3, Informative)

sometwo (53041) | more than 9 years ago | (#11351688)

So what about AAC used by Apple in their music store?

I did a little googling and found this (http://www.teamcombooks.com/mp3handbook/13.htm [teamcombooks.com]):
AAC (Advanced Audio Coding) is not a MPEG layer, although it is based on a psycho-acoustic model. Sometimes referred to as MP4, AAC provides significantly better quality at lower bit-rates than MP3. AAC was developed under MPEG-2 and also exists under MPEG-4.


AAC supports a wider range of sampling rates (from 8 kHz to 96 kHz) and up to 48 audio channels, plus up to 15 auxiliary low frequency enhancement channels and up to 15 embedded data streams. AAC works at bit rates from 8 kbps for mono speech and up to in excess of 320 kbps for high-quality audio. Three profiles of AAC provide varying levels of complexity and scalability.

AAC software is much more expensive to license than MP3 because the companies that hold related patents decided to keep a tighter reign on it. Most AAC software is geared towards professional applications and secure music distribution systems, so it may be a while before you see AAC in consumer-oriented products.

16 bit CD encoding (1)

Sebastopol (189276) | more than 9 years ago | (#11351729)

Ok, mod me down if I'm clueless, but on the first page: "Compact Discs use a bit depth of 16, allowing for 2 ^ 16 possible levels."

I always thought CDs were encoded in 12 bit, not 16?

Re:16 bit CD encoding (1)

stratjakt (596332) | more than 9 years ago | (#11351784)

Well, you were always wrong!

How about that!

CD is 16 bit stereo PCM at 44.1khz, no more no less.

I still hear MDCT distortions (2, Insightful)

ikewillis (586793) | more than 9 years ago | (#11351761)

I've stopped liking Modified Discrete Cosine Transform-based codecs like Vorbis, MP3(+), etc. even though they (i.e. aoTuV Vorbis) consistently win in low bitrate listening tests among random listeners. Why? Well, unfortunately, I've been listening to audio encoded with this transform for so long that I can't help but hear the distortions they create, namely pre-echo (which is often described as a 'muddiness' or an 'underwater' sound) and distorted treble detail (often described as 'twinkling')

Call me crazy, but I insist that there are certain 'killer' tracks where I can hear this distortion even at higher bitrates in advanced MDCT codecs like Vorbis, namely Led Zeppelin / Rock and Roll whose drumline consists of a ridiculous number of cymbal crashes in rapid succession.

The way I see it, the future is lossless. With hard drives burgeoning to over 500GB and Fiber-to-the-Home becoming a reality within the near future, why bother saving a little extra space at the cost of degraded quality, which, the more you listen to audio compressed with a certain transform, the more likely you are to hear distortions? I think in the future we'll see a greater trend towards lossless audio compression with codes like FLAC and its ilk.

Re:I still hear MDCT distortions (-1, Offtopic)

Anonymous Coward | more than 9 years ago | (#11351844)

you're crazy

Re:I still hear MDCT distortions (1)

LWATCDR (28044) | more than 9 years ago | (#11351864)

It really depends on what you are recording. For speech recording I see no real benifit for lossless like Flac.
Things like talking books as such would do very well using Speex. The only problem is I have yet to see a mobil player that supports it.

Re:I still hear MDCT distortions (0)

Anonymous Coward | more than 9 years ago | (#11351880)

The way I see it, the future is lossless. With hard drives burgeoning to over 500GB and Fiber-to-the-Home becoming a reality within the near future, why bother saving a little extra space at the cost of degraded quality, which, the more you listen to audio compressed with a certain transform, the more likely you are to hear distortions? I think in the future we'll see a greater trend towards lossless audio compression with codes like FLAC and its ilk.

In a few years time, people will be saying "Codec?! What codec? Just upload me the god damn DVD-Audio image!"

Re:I still hear MDCT distortions (1)

ndevice (304743) | more than 9 years ago | (#11352066)

Cymbal crashes are the wideband of audio. Hard to compress with frequency transforms.

Actually, you hear quantization distortion (2, Informative)

cogito ergo blog (830437) | more than 9 years ago | (#11352182)

(Mod to -3, nitpicking)

The MDCT in itself is actually lossless. Any distortion you notice is most likely introduced by the quantization applied post MDCT during compression.

Re:I still hear MDCT distortions (4, Interesting)

radish (98371) | more than 9 years ago | (#11352309)

Not wanting to get some award for pedantry, but all music recording is "lossy". If you listen to a CD, you're not hearing the exact same sound you'd here in the studio, those cymbals sound diffrent due to sampling, quantization etc. So when it comes to "lossy compression" causing "artifacts" - it's only creating different artifiacts, there already were some.

Of course this doesn't go against what you're saying at all, other than calling FLAC "perfect" is wrong. It might be the same as the CD, but that has it's own problems.

Re:I still hear MDCT distortions (1)

gfody (514448) | more than 9 years ago | (#11352397)

different encocers produce different output. most people sware by LAME and only with the highest quality settings. I'm sure the faster encoders take shortcuts sacrificing audio quality for encoding speed.

the future is definately not lossless. why would we go backwards? with cpu's getting faster more emphesis will be put on accurate encoding instead of fast encoding and the distortions will eventually go away

The actual meaning of lossless ?? Any clues? (1)

gmania (687303) | more than 9 years ago | (#11351792)

Page 3:
These are all mathematically lossless codecs. They theoretically should sound identical when using the same hardware and playback software. I don't believe I could conduct double blind or any other subjective test which would prove beyond any doubt that there are any differences between the options aurally.

-> When something is mathematically lossless, proving the encoding / decoding algorithmes is the only way to go. Double Blind, Subjective??? I'm really sorry I read the whole thing, I want a refund for my time!

Re:The actual meaning of lossless ?? Any clues? (-1)

Anonymous Coward | more than 9 years ago | (#11351836)

The only conclusion that can be drawn from the article is that the author of the article is a howling bozo.

Trust me. You would if I weren't posting AC to avoid having my name attached to an ad-hominem attack. :)

Re:The actual meaning of lossless ?? Any clues? (2, Informative)

stratjakt (596332) | more than 9 years ago | (#11351867)

If it's lossless, you should be able to take digital file A, compress it into compressed file B, and then if you uncompress B to get A', then A' = A.

That is, the checksums for A and A' should match, etc.

That's how I define mathematically lossless.

Whatever this asshat is on about double blind and testing and all that, has more to do with the ability of his FLAC playing equipment to sound the same as his CD player, which is a whole 'nother ball of wax altogether.

more algorithms (5, Informative)

barik (160226) | more than 9 years ago | (#11351802)

While the article is a primer, I was a little disappointed in the algorithmic treatment given in the article itself. Right now I know of two excellent free publications: Introduction to Sound Processing [mondo-estremo.com] and The Sounding Object [mondo-estremo.com], which both treat the theoretical, DSP side of things. Any other resources that Slashdot readers can recommend for those who are interested in the subject of audio compression and representation?

transcoding (1)

ceswiedler (165311) | more than 9 years ago | (#11351838)

I know that everyone says that transcoding loses quality. So when I finally got an iPod, I initially transcoded all of my .ogg files to .mp3, but planned on re-encoding later from the CDs. But when I re-encoded a few tracks with lame, and listened to them versus the transcoded track, I honestly couldn't tell a difference (even with headphones). If I ever notice poor quality on my iPod, I guess I'll re-encode then, but otherwise why bother?

I believe that I used the default settings for both oggenc and lame, which was ~128 bit VBR in both cases.

Re:transcoding (1)

radish (98371) | more than 9 years ago | (#11352414)

Because not everyone is using $5 earphones to listen to stuff. Yes transcoding loses quality, no it's not much, yes you can tell with the right equipment. If you're just listening on an ipod with stock earbuds, don't do anything over 128kbps, you can't tell the difference. If you upgrade to decent phones (ala Shure E3) or a real hifi, then it's a different story.

Missing "Apple Lossless Encoder" (1)

Sebastopol (189276) | more than 9 years ago | (#11351898)

This article is already outdated. :) His discussion of lossless formats excludes the iTunes Apple Lossless encoder, which is supported by iPod. From their websites Import Music page:

Weapon of Choice

"However, you can choose to use different audio formats for any track that you import from CD. iTunes lets you convert your music to MP3s at high bit-rate for no additional charge. Using AAC or MP3, you can store more than 100 songs in the same amount of space as a single CD. Discerning customers and audiophiles want true CD audio, and now iTunes can give you that quality with the new Apple Lossless encoder. You'll get the full quality of uncompressed CD audio using about half the storage space. You can copy music in this format onto your iPod or iPod mini, to take perfect audio wherever you go"

Har har... (0)

Wraithlyn (133796) | more than 9 years ago | (#11351900)

"For those of you with a little extra time this afternoon..."

Congratulations, you just defined a Slashdotter. ;P

Opera (0)

Anonymous Coward | more than 9 years ago | (#11352079)

Any one else having a problem reading that page with Opera? I get this white rectangle which obscures part of the text.

They compressed their text! (0)

Anonymous Coward | more than 9 years ago | (#11352179)

They're so into compression they compressed their text so much you can't read it! :)

Seriously, their text is 10 pixels high. That's too small to read on anything but a very low resolution monitor. Even with my new 23" LCD wide-screen, 10 pixel high text is unreadable. Hey, how about learning how em's work before attempting to publish on the internet? There are way too many idiots now that claim to be webmasters that don't have a clue.

*snore....* (0)

Anonymous Coward | more than 9 years ago | (#11352204)

I know, I know, a "primer" Useless article....

Dear Ogg,Flac,Linux users (1)

Letter (634816) | more than 9 years ago | (#11352217)

Dear Ogg,Flac,Linux users,

I use CDs. No problems yet.

Will report with more later.

Letter

ARRRG! He gets Nyquist WRONG! (3, Informative)

wowbagger (69688) | more than 9 years ago | (#11352338)

According to the "Nyquist Theorem," you need to have twice as many digital samples as the frequency of the analog signal you are trying to represent to have enough data to accurately build it.


WRONG!

Nyquist's criterion is "You must have at least twice as many samples as the largest BANDWIDTH of the signal in order to correctly reconstruct it."

You can take a 10.7 MHz signal, and sample it at 10000 samples per second, and correctly reconstruct it, so long as the signal is guaranteed to be bandwidth limited to 10.7 MHz +/- 2.5 kHz. This is often done in software defined radio to aquire the signal from the intermediate frequency (IF) of the analog front end.

You also have to have an appropriate reconstruction filter at the output of the system in order to correctly recover the signal - if you don't have the right reconstruction filter, you will NOT reconstruct the signal correctly.

You also have to take into account the effects of any signal modulation - take a 20 kHz sine wave, and burst it for 10 msec, and you widen the bandwidth of the signal by about 100 Hz (depending upon the exact shape of the burst - a perfect square burst will widen the signal as a sinc function and will, in effect, increase the bandwidth to infinity, which is why square bursts are generally Considered Harmful in communications work).

Also, you don't oversample a signal in time to account for "rounding errors" - you oversample in time because the frequency response of sampling a system in time introduces a sinc response in frequency - by moving the sampling rate up you reduce the impact of this response on the recovered signal's frequency response. You also greately ease the requirements on the reconstruction filter - the filter can be wider (have fewer poles in the transfer function - thus fewer parts needed).

"humans can hear from 50 to 22,000Hz on average" (2)

Stavr0 (35032) | more than 9 years ago | (#11352380)

Um, no. 20/20K is more accurate, and we lose a kHz every 5-10 years as we get older.
Load More Comments
Slashdot Account

Need an Account?

Forgot your password?

Don't worry, we never post anything without your permission.

Submission Text Formatting Tips

We support a small subset of HTML, namely these tags:

  • b
  • i
  • p
  • br
  • a
  • ol
  • ul
  • li
  • dl
  • dt
  • dd
  • em
  • strong
  • tt
  • blockquote
  • div
  • quote
  • ecode

"ecode" can be used for code snippets, for example:

<ecode>    while(1) { do_something(); } </ecode>
Sign up for Slashdot Newsletters
Create a Slashdot Account

Loading...