Dial-Up Audio Public Listening Test Opened
CaptainCheese writes "Hydrogenaudio.org's Roberto Amorim just announced the opening of their 32kps multi-format listening test, intended to test the current 'dial-up' quality codecs.
From the Announcement: "The formats featured are Nero Digital Audio (HE-AAC+PS), Ogg Vorbis, WMA9 Std., MP3pro, Real Audio and QDesign Music Codec.
Lame MP3 is being used as low anchor, and a lowpass at 7kHz is being used as high anchor." These codec tests are unusual in that they adhere to ITU-R BS.1116-1. The test is open until July 11th and all are invited to participate. There's more info in the original test discussion, which indicates the originator is interested in 'testing formats working on dial-up streaming bitrates' - the test page notes: 'The real arena where codecs are competing, and most development is going, is at low bitrates.'"
...since I'm liable to vote for whichever one sounds most like the Centurions from Battlestar Galactica, or the voice communications from THX-1138. Not best quality, not most understandable, just coolest.
"A great democracy must be progressive or it will soon cease to be a great democracy." --Theodore Roosevelt
Now if only the companies who manufacture digital players would take a look and see that there is life beyond MP3. Nice that a few are starting to offer Ogg Vobis, but they are few and far between.
If it isn't, you'll only find out the most popular format, not the best.
I've seen the double-blind tests done at 128kbps and again fail to see the point.
What I really want to see is a rating of codecs that are able to achieve DBT-proven audible transparency and see them rated in terms of storage space (thus allowing the VBR schemes to finally compete).
Of course FLAC would come in last (considering WAV is the 'source'), but can my high quality VBR LAME MP3 pass for the original and take less space than MPC?
This comment does not necessarily represent the views and opinions of the author.
When the best analog quality is achieved by A->D, compression, modem error correction, and D->A, you know something is shit.
BUT...
Although it wouldn't help for internet music, better low-bitrate codecs could make internet talk radio more feasible. It lets companies save bandwidth on the server side and still maintain quality that at worst is a bit better than the phone connections of people calling in (VoIP notwithstanding).
The Cheese Stands Alone.
Wow! I thought people on this site would have been a little more understanding. Believe it or not there are other places in the world (such as Africa) where high-speed Internet is not the norm or even available. Plus if you stream audio, any attempt to lower bandwidth is a plus as it lowers your bills.
Get over yourselves please.
By the way, did you ever notice the lack of multimedia even on this site? Why might that be? Hmmm...
I remember back when I had dialup, it totally sucked, it took me absolutely forever to download ANYTHING, I would read a magazine while I used the computer because it would take so long for pages to load... when I got broadband at home it was a very happy day... but lots of people I know still don't have broadband in their area, which is why I think it is nice for people to do things like this, but also I think you could use this for voice over ip... just my two cents
No matter how optimized it is, won't it will still use too much bandwidth for dial-up users who actually want to do something else with their connection? All of the streams I ever tried to listen to, including the 8kbps ones, gladly used all of my available bandwidth. I don't know about anybody else, but I'm not interested in only getting a fraction of my 2 KB/sec max for browsing, using chats, or other tasks.
Audioscrobbler
Why don't you just USE THE PHONE?
Yes, I know there are applications for this, like doing some other thing while listening to audio, and the prohibitive internatioanl call rates, but still..
Make even shorter URLs - 8LN.org
I just took the test with sample 9, one of the speech ones, and it's amazing how much variability there is in the various codecs. One of them was so good I could only reliably hear the difference after a dozen repeated listenings, and another sounded like a cellphone in a tunnel. I'll be interested to see the results in a week or so.
Karma: Segmentation fault (tried to dereference a null post)
Something, though, tells me that this test isn't going to apply to this sort of dial-up audio.
Procrastination sucks.
doubtful...
"The object of war is not to die for your country, but to make the other bastard die for his." - Patton
Notice all the different non-standard switches I had to use, which together help noticably. That's the sort of stuff you need to do to LAME before it produces acceptable results at very low bitrates. It is optimized only for 44.1KHz, so we should keep that in mind when we see the results. Notice now that none of these switches are being used for this test, so I'm almost certain that LAME will come out looking much worse than it is.
I would love for there to be a LAME-based encoder that is optimized for speech, low bitrates and sample rates. If it is made, I am prepared to re-encode all the readings that are (and are about to be) posted on my site.
Hey, there is too much spacing in between the fonts. I am having difficulty reading your post.
As always, this is rigged !!
I just listened to this, and I can tell you that it sounds totally like crap. Here at home, I have a sound room with a bitchen 100% analog system... Vacuum tube amplifiers, gold wiring, the works. I play my records on this thing and they have that wonderful warm sound. But this 32 kps sound sounds like garbage.
Also don't forget that there are various setting for a codec that can affect the output. How to minamize all the variables say one, and have a meaningful comparison?
"By the way, did you ever notice the lack of multimedia even on this site? Why might that be? Hmmm..."
Because we don't need the whole holographic, 5.1 stereo, touch sensitive, Goatse.cx experience.
"When I first read this story's title on this, I immediately thought of They Might Be Giants' infamous Dial-A-Song program, which can incidentally be reached at (718) 387-6962."
Oh lovely. We Slashdotted the phone company. If you stand out on the balcony, you can see the telco going up in flames.
Maybe, say a MMORPG. Let's then say that I juin a team/guild/association. And let's say that, while we play and collarbate, we all want to chat. Well what if we were to use the phone system as you suggest? First we'd need to get a teleconferencing centre to conference the calls, which isn't free. Then we'd all have to call long distance, and pay per minute. Those in other countries would have to pay a LOT per minute. So for a game that we pay less than $20/month to play, we'd each be paying anywhere from $5-$100 per hour to talk to each other. Hmmm, am thinking NOT.
But what if we were to get a member, that has a big line, or has access to one, to setup a voice server, say TeamSpeak. We then all connect to that, and both voice and the game comes over the same line. Thus we pay nothing to chat. Seems to me like this is the answer people will opt for.
Indeed, in the MMORPGs I've played, it WAS the answer we opted for. We happily chatted away on TeamSpeak and played Galaxies, all over the same Internet connection. The wonders of TCP/IP, you can run more than one service on the same line at the same time. And yes, with a low bitrate speech codec you CAN play an MMORPG (they don't use much bandwith) and talk at the same time.
Apparently in the 128kbs the figures 50gig of data was leeched, and there were no extra submissions of test data. I wonder how it'll work out this time...
ROFL.
Listening at that low quality doesn't have as much commercial, and quite frankly, personal appeal as it did back in the 90's.
Not every location is set up for wireless broadband Internet access. Can you get affordable broadband on your mobile phone? GSM mobile phones receive and transmit voice at 13 kbps using the GSM RPE-LTP codec; one often has to pay extra just to get 32 kbps data. Also think about digital radio; lower bitrate for a given perceptual quality allows for more music choices in the same frequency, possibly reducing the number of actual stations that Clear Channel needs to own. Another application of low bitrate audio is in handheld video games; I've written a program to encode music at 30 kbps and then play it back on the Game Boy Advance, a machine thought not to be able to handle the mathematical complexity of MP3.
What I really want to see is a rating of codecs that are able to achieve DBT-proven audible transparency
If you're looking for transparency proven with ABX double-blind testing, you know where to find it.
DSL and cable work only as far as the cord can reach, and fixed wireless is just as fixed. Audio over a mobile last mile currently requires a low data rate codec, as bcombee pointed out. And zerblat wonders: can one get affordable broadband in developing countries?
but I could do without all the stupid bullshit. Maybe copeland secretly hates ruby and wants it to look more astroturfy than someone pitching C# and the MSDN.
THIS THING CAN TURN ON A DIME, MACROSSZERO STYLE ALSO FUCK BETA, ~NYORON
I have a nice 1.5Mbit connection.
Which can feed 7 listeners at 192 kbps or 46 listeners at 32 kbps, as you seem to recognize with talk radio. Wideband Speex, an audio codec designed for talk radio and telephony, sounds listenable even at 12 kbps (listen). However, more listeners for talk radio does mean a bigger audience for conservative spokesmen, whether you agree with them or not.
32kb/s music just doesn't cut it.
Have you actually tried listening to a recent codec at 32 kbps? Sure, it's not transparent, but often it'll do in a rather noisy environment such as while riding your bike or the city bus. If you want to try, grab a few 32 kbps Ogg Vorbis files from this page.
Audio on a phone is typically only up to 4kHz. Anything higher is cut out. NOTE HOW THERE IS NO BITS IN THAT NUMBER.
Many parts of the PSTN go over a digital connection similar to an ISDN channel. This runs at 64 kbps.
Compare a 56kbit wav to a 56kbit mp3 and you'll hear a huge difference...
To save Slashdot readers the trouble of going in and encoding it yourself, I've done it for you. Hear it here.
If 2 files of the same recording are available, and are the same size, which sounds better, an mp3 or a wav?
If you can't predict the answer yourself, then here are some test files.
A lot of people are complaining here that low-bitrate recording is useless / stupid / so 1988 etc. Fine for them :) I'm glad that people are fanatic about sound quality and that storage prices make it reasonable for many people to use nothing but lossless codecs etc., and to care about the difference between 192kbps and 256kbps MP3s. Lossless is certainly a good storage answer for the long term, as the file can be inflated and re-squashed with the latest n' greatest lossy codecs as appropriate.
However, there are reasons and times where the lossy stuff, even hugely lossy stuff like this listening test focuses on, makes a lot of sense and has no big downside. For me, squashing audiobooks is this way. I can fit about 30 hours of book into one CD-R size chunk of hard drive as a series of (extremely listenable) quality zero mono ogg vorbis files. Beats carrying 30 CDs around.
timothy
jrnl: http://tinyurl.com/c2l8yr / foes: http://tinyurl.com/ckjno5
In terms of voice I did some testing of my own a while back and actually (as much as I hate to admit it) realaudio came out ahead in most of my voice tests right down to below 24 kpbs testing. I think it really helps when you've specifically designed your codec for these ultra-low bitrates. IIRC, WMA didn't do that bad at these low bitrates either.
For 16 kbps and lower, it was pretty tough to find anything that sounded ok. This is where speex starts to look a bit better (although it didn't fare well in my testing).
Quite an experience to live in fear, isn't it? That's what it is to be a slave.