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SanDisk, Music Publishers Push DRM-free SlotMusic Format
Strudelkugel writes "The LA Times and others are reporting the music industry is working with SanDisk to try unrestricted music files on microSD memory cards to improve sales of physical media: 'In addition to music, the slotMusic cards will come pre-loaded with other things, such as liner notes, album-cover artwork and sometimes video.' The important part: 'The music on slotMusic comes without copyright protection, so it can be used on almost all computers, mobile phones and music players — but it won't play on an iPod, which doesn't have a micro-SD memory slot. It has one gigabyte of memory, and the music tracks are played back at high quality.' Could it be the labels have finally recognized that providing features and convenience to customers is preferable to suing them?" Most computers also don't have microSD slots; according to EMI's press release, there will be a "tiny USB sleeve" packaged with each card, and the "high quality" format means up to 320kbps MP3. From the given description, it seems like it would be no harder to transfer the tracks to an iPod (via a computer) than to most other players.
Somehow I am a little doubtful, given that the article does not state which format the songs will be distributed in.
From the article:
Music, Retail and Tech Leaders to Offer "slotMusic(TM)": High Quality, DRM-Free MP3 Music on microSD(TM) Cards
My guess is, this is yet another "plays on most devices" that the record labels always cooks up
And your guess is wrong. This is genuinely good news, they're finally realizing that certain people will pirate regardless how inconvenient they make it.
Ms. Quinn, the author of the Los Angeles Times article, is not a very good technology writer. She not only quotes that it won't work with iPods (which is terribly misleading; the microSD card won't, but the contained DRM-free MP3s will be very easy to work with), but she also refers to this as a "new music format".
Medium, yes; format, no. Distributing on the microSD cards is new, but seems like something people may latch onto quickly. MP3 is old and a de facto universal format, which is what makes this even better.
"Give a man fire, and he'll be warm for a day; set a man on fire, and he'll be warm for the rest of his life
Stereo, for example, was invented to create more space for sounds in a recording.
No, it wasn't. Stereo is used to recreate the spatial component of music: when you record a number of instruments sitting at different positions in the studio, you should be able to hear where those instruments are. That has nothing to do with 'too many waveforms ...cramped together on the same output'.
In fact, in a stereo recording, most of the information will be played back by both speakers.
It is possible to make a recording where the left and right channels have nothing in common, but you'll find that those sound very unnatural, so these recordings are (thankfully) rare. It's like having half the musicians on the far left of the stage, and the other half on the far right, with nobody in the center.
you need different ranges assigned to different speakers that can give out that frequencies. but, there has to be more of the same speakers assigned to a particular frequency range - lets say, you got a certain size of tweeter. if there are 4 of this, and you divide a small incremental range of high frequency sound to four of these in small increments, you'll have, say, seperated two sopranos' (each soprano will have differences in their frequencies, even if minute and hardly identifiable by human ear) voices to two tweeters of the SAME kind, but while playing these two sopranos' voices, each of their voices will come from the different tweeters. this will increase the distinctiveness of each sound. here, the quality of the tweeters matter VERY much.
Nonsense. No audio system works like that.
1. you can't separate two voices or two instruments like this, because each voice produces a range of frequencies that mostly overlaps. They'll sound different because their harmonic spectrum (the relative volume of each harmonic) differs a bit, but there is no filter that can separate them.
2. A loudspeaker box usually contains a few drivers of different sizes, because the driver size needs to be matched more or less to the frequency. A 12" bass driver is too heavy to produce 10 kHz, conversely a 1" tweeter can't move enough air to produce convincing bass. The challenge is to use no more drivers than necessary, because dividing the frequency spectrum like this introduces all kinds of problems. The holy grail of loudspeaker design is the point source: a single point that can produce the entire spectrum.
The only reason loudspeaker arrays are used, is volume. Multiple parallel drivers can produce more volume than a single driver.
There are some interesting side effects to arrays. The dispersion pattern changes a bit, which can be beneficial if done right. But 'a sound stage that encompasses you'? No. That's due to the surfeit of power which sets up reverberations in the hall. You get the same effect cranking up your non-array home stereo.
"But pretty much anyone with decent equipment *can* hear the difference between 24bit and 16bit, or 48khz and 96khz."
Lots of people who pay large sums for audio equipment _claim_ they can hear such differences despite the fact that the original source signals from the best microphones in the world don't produce any useful information above 22KHz and have signal / noise ratios of 90db or less, so there won't be any extra musical information that requires the higher frequency response and dynamic range provided by more bits and higher sampling frequencies.
Studios use high sampling rates and word sizes (192 KHz 32-bit) because multiple tracks can act as input to other tracks, which means that noise accumulates, and positional differences of high frequency bits in lower sampling rates can combine to produce artefacts (both of these can and do also occur when mixing multiple tracks down). Neither of these is a factor in domestic listening however, because _any_ system below the native studio resolution of 192 KHz 32-bit will end up being dithered down using the same algorithms (often on the same hardware).
"That is a pretty well established fact"
Established by whom? Double-blind listening tests indicate that there's no objective difference between them on any level of equipment when they're only being used to play back pre-recorded sources, irrespective of the musical genre being used to evaluate them. There's plenty of psycho-acoustical information to indicate that rise-times in waveforms above the upper threshold of human hearing can have a notable effect on the way it's perceived, but the inability of microphones used in music recording applications to transduce those frequencies into useful signals means that it's of academic rather than practical interest (some microphones such as the ones used in bat detectors can respond to extremely high frequencies, but they have other characteristics that make them useless for recording music signals).
"Audio CDs are generally encoded as 48khz, 16bit, 1411kbps PCM audio"
The audio on digital video is recorded at 48KHz. CDs are 44.1 KHz.
"For comparison, get one of the few albums available in DVD Audio and compare them to the CD - especially at high volumes. "
You'll need one of the even fewer DVD Audio albums that isn't up-sampled and re-mixed from a 44.1 KHz 16 bit master, and therefore actually has some chance of containing real extra musical information that isn't on the CD version to make such a comparison valid, otherwise any perceivable differences will be nothing more than artefacts of the up-sampling and re-mastering process.
I'm not going to change your sheets again, Mr. Hastings.