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Discussion in 'Pros' Racquets and Gear' started by vbranis, Mar 20, 2009.
From the IW stringing room.
can you measure the length? Is it extended?
I'm not sure about the length, I did not measure it.
Any string specs? It looks like he's hybriding with something, obviously one the types of string is Big Banger. Maybe it's just the lighting in the picture.
Yea, it is definitely extra long. i can tell even in this pic. looks like 27.5 or 28 inches long.
No it's full BB
GREAT JOB vbranis!
LM Instinct Tour XL, as many other posters have concluded before
how did u get the pics
Absolutely correct- you even see the LM ridges on the frame.
What color is that string considered to be? Natural/white?
mhm nice frame
I didn't see that at first thanks for pointing it out.
great French stringer for the FFT Jean Claude Poupon who does Mauresmo's racquets too
Correct, it's BB Original strung at 24/23 kilos.
What I want to know is if Gasquet uses the same rectangular handle shape that comes with head racquets, or does these guys change to a more rounded grip shape.
Is it just me, but my forehand is way better with a rounder grip shape.
If anyone has info or pics to show this, it would be great.
I prefer the rounded grip also.
That's probably one of the best pj's I've seen.
yea liquidmetal instinct XL
i think he should switch to the new one... thats wat i use
but then he would play like you..........
ha ha thats a good thing lol... but for his sake he should keep his own game
Is that the new model for 09?
no because the part that says "extreme pro" would be red
nice work man
Richard is still playing with the old graphics (and the same pj'd LM Istinct Tour XL of course )
Nice to know he uses full BB original, i might have to try that out.
looks like a prestige pro with yellow paint
Not sure if this has been asked before, but does anyone know what gripsize Gasqet uses???
too bad he wont be using it for awhile.
haha yea thats really too bad....
This is a VERY BAD joke, but it´s to bad to pass on...
When Gasquet makes a comeback the commentators really have to watch out when they say that Richard "goes for the line" or when he hits a "line-cleaner" ,,,, :twisted:
lol im looking forward to when someone uses hawkeye and gasquest just snorts the line right off the court.
I tried snorting Coke once, but the ice cubes got stuck in my nose.
Thats the worst joke out of all of them that people have posted!
It amazes me how they can paint the racket exactly like the original, even though it is a different shape and has the LM ridges. I take it its done by a machine?
that one was evil :lol:
I agree. Its crazy to think they can just paint any frame to look like any frame they want to look like.
Sweet pix!
How long is the Instinct XL ??
It's 27.5"
Is that rosin/coke I see on the handle...?
I'd like to know too... seeing how pros tend to use smaller grip.
why thank you THESEXPISTOL and mozzer i thoguht that it was so bad that it was almost, close to being funny.
i thought i saw gasquet getting in some training on a local beach
anyway iv heard his racket encorporates new secretive technology ; )
^^Cmon,, take it easy on Richard. All the pros does Coke. He just had the bad luck of being caught. that is all.
I think it's kinda lame how none of the other french players are standing up to him, but I guess they're kinda pissed with him for being such a flashy little pansy who bails on Davis Cup matches. :-\
I like gasquet, I really really hope he doesn't get suspended, that'd be really really lame.
Think you meant for him. Actually, this isn't really true. Maybe it's just the gaping hole in tennis reporting in the US press that leaves that impression.
In Lequipe, the French sports daily paper, Tsonga, Santoro, and Llodra (and maybe others that I didn't catch) have been on record supporting him, in the sense that the situation is lame. L, lame. I don't think it's any smart person's place to step up and say he didn't snort anything that night unless they were with him, and they haven't said that. But these are his closest friends on tour. What they've said is no way he's an addict, and this is all BS. This is pretty French imo - there's a clear (excuse the pun) line between performance enhancing and recreational drug use. At the same time, this is a nightmare for French tennis, and my take (based on the Roland Garros coverage here) is that it's the kind of situation that they're afraid to mention... like saying the word cancer. TV announcers were lowering they're voice mentioning Gasquet (the best grass courter) in reference to the grass season. Finally, there was an impassioned, articulate and common sense plea in Tennis magazine (French - not the bs US mag of the same name) for leniency on Gasquet, which I hope makes a difference. The Tour de France wouldn't ban a rider for 2 years for drinking too much wine, and Gasquet shouldn't suffer the same fate for partying too hard. Bad judgement, you win... if anything, he should be punished for partying and ruining his chances - for not taking his talent seriously, not for enhancing his chances.
Hi, is the gasquet racquet for sale?
This is a really old thread, plus I'm pretty sure the OP just strung the racquet for him.
How do you send messages to other people on this site? I'm new.:???:
Is it coincidence that a very similar racquet just came up on the big auction site?
Separate names with a comma.


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Josh Koscheck (15-5 MMA, 13-5 UFC) is the final obligation on the current contract of Matt Hughes (45-8 MMA, 18-6 UFC).
Consequently, the former UFC welterweight champ is playing his future by ear after he fights the onetime title challenger Saturday at UFC 135.
"The outcome of this fight won't really matter, to be honest," he said today. "After this fight, win or lose, doesn't matter, I'll talk to the UFC and we'll figure out what we want to do."
UFC 135 takes place Saturday at Denver's Pepsi Center. The event's main card airs on pay-per-view. Two preliminary bouts to air on Spike TV make up a lead-in to the main card, and the remainder of the preliminary card streams live on Facebook.
While co-headliner Hughes said he continues to get enjoyment out of his career as a fighter, there are some close to him that don't feel the same way.
"My wife tells me I'm done fighting," he admitted.
That declaration actually came 10 months prior when Hughes was knocked out by B.J. Penn in a brisk 21 seconds at UFC 121. The two had split a pair of earlier bouts.
Hughes isn't saying he's boarded that plane yet.
"We'll see what I feel like, and we'll see what the UFC wants to do," he said.
The loss to Penn snapped a three-fight win-streak for Hughes, who once reigned over the welterweight division with an iron fist. Victories over Matt Serra, Renzo Gracie and Ricardo Almeida boosted his stock. But dual losses to current champ Georges St-Pierre made a title shot a faraway prospect, and at 37, Hughes is no spring chicken.
But he's clearly not ready to give up his lifestyle, and that may keep him in the game.
"It's competition," Hughes said. "I think that's what drives your top athletes in the UFC is the fact that you love getting in there against one other person and mixing it up. I don't have to rely on four other basketball teammates to score a basket. I've just got to rely on myself. The fact that it's just me and another person competing in there, that's my big drive.
"I consider myself a lucky man. I love going to the gym every day, twice a day. I love traveling with my buddies. All in all, it's a great lifestyle."
For the latest on UFC 135, stay tuned to the UFC Rumors section of the site.
(Pictured: Matt Hughes)

subset: commoncrawl


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In the first video you see some of our favourite guys, the Norwegian director Petter Næss and Norwegian actor Stig Henrik Hoff arriving on the red carpet – you can see it here!
The second clip from the red carpet shows the two German actors David Kross and Florian Lukas with the Næss and Hoff – click here to see it!
Click here for the third clip mostly shows the British actor Lachlan Nieboer arriving at the red carpet event
The fourth clip is probably the one you've all been waiting for the most as it includes Rupert Grint and you can hear what he has to say to ICM as well as the surrounding journalists. Also, it shows all the actors on the red carpet posing together for pictures.
We also have some pictures from the red carpet event that was taken by the ICM girls – and you can see them all in our gallery by clicking here!
We will have a lot more stuff from yesterday's press day – but just to keep your interests peaked, you can see a few of our pictures in the gallery! There are pictures from the press conference, the photocall and the introduction leading up to the world premiere itself.
New articles from the Norwegian press has surfaced – you can read them here:
Dagavisen.no: Into the White is ready to premiere – SPOILER ALERT!
Filmweb.no: The Harry Potter Effect
NRK.no: Like being on the moon
Oa.no: Hoping for "Max Manus" potential
Side3.no: It helps with alcohol
Filmfest Oslo released their own video from the premiere on Facebook – and you can see it on their profile here!
3 Comments to "Rupert Grint: ICM Exclusive – Our Night on the Red Carpet!"
March 7th, 2012 at 5:18 AM
Thank you soooo much
March 8th, 2012 at 5:34 PM
So the Promotion is over for now 🙁 I hope there might be more in other Countries later
December 1st, 2012 at 3:21 AM
What a child!! causing nanny woerirs already picky not arriving by email as expected, parents not in to explain missing picky!!! Still, nannies are persistent after several answer machines and two mobile phone calls I tracked him/her down!!


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Rated 5 out of 5 by MinnieMae from Best looking acrylic glasses ever! Well, I must say, I looked everyplace I could think of to buy acrylic glasses to use by the pool. I always knew that DuraClear was the top of the line, but thought I'd find something comparable, and less expensive. The search was a waste of my time, except that when I was ready to buy the DuraClear, Wm/Sonoma offered them on sale, and free shipping, too. I couldn't resist. They arrived in about two/three days. I can't believe how lovely they are, and I examined every detail. You won't be disappointed with this purchase!Date published: 2014-06-10Rated 4 out of 5 by msmls from Pretty and practical I admit I was a little peeved that these took so long to arrive, especially since they were delayed for several weeks beyond the original backorder date and did not arrive in time for a party I was having (hence the 4 instead of 5 stars). However, once they did arrive, I was pleased with them. The colors are pretty, the material is sturdy (yes, they ARE plastic, as is clearly stated in the description -- several reviewers were upset that they aren't glass) and they are very practical for an outdoor party. Even though they are a heavy plastic, they're worth the money and will be used over and over again.Date published: 2013-08-20Rated 1 out of 5 by BrendaJC from Not Glass, Plastic Was expecting glass but they are a high quality acrylic I'm guessing? Some kind of plastic. Was giving as a gift but when I saw they were plastic I just couldn't so I'm keeping for myself. Just so you are not deceived.Date published: 2012-12-18Rated 5 out of 5 by WyoCowgirl from Perfect for the pool My sister had a set of these glasses and I always thought they were very classy. So I finally ordered my own set and am very pleased with the quality. Terrific value too. I will use them exclusively at the pool in the summertime.Date published: 2012-01-18


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Tarasco can relate as he tutors young Nats
First-base/outfield/baserunning coach brings lessons from playing days
By Bill Ladson / MLB.com
VIERA, Fla. -- After third-base coach Bo Porter announced that he was leaving the Nationals to become the manager of the Astros in September, Nationals skipper Davey Johnson quickly made up his mind that Tony Tarasco would replace Porter as the outfield and baserunning coach while serving as the new first-base coach.
Tarasco had been working in Washington's Minor League system since 2006 as a hitting coach and outfield coordinator. To Johnson, it was a no-brainer that Tarasco would join the Major League staff. It helped that Johnson managed Tarasco when both were with the Orioles in the late 1990s. Years later, as Washington's senior advisor, Johnson saw Tarasco worked well with outfielders in the system.
"I got to see Tony work with young players and how they responded to him," Johnson said. "The minute [Porter] left, I knew it was going to be Tony Tarasco. We took a long time announcing it. He has a lot of talent to work with."
How does Tarasco feel about working in the Major Leagues?
"I get to spend time with my former Major League manager," Tarasco said. "It's an opportunity to work under him and learn some things. I'm excited to be part of the staff and help these guys."
It was Tarasco who helped turn a below-average shortstop named Michael Taylor into an above-average outfielder. Bryce Harper never played the outfield until after he was the first overall pick in the 2010 First-Year Player Draft. But there was Tarasco teaching Harper the nuances of playing the outfield. Because of Tarasco's tutelage, Harper is more than an adequate outfielder.
"He was with me in the Minors the whole time," Harper said of Tarasco. "He really helped me out the whole time. He told me to be as smooth as I can, hit the cutoff guy, just do the right things in the outfield.
"He can make you go to the next level. Just be as soft as you can on your feet, try to catch everything you can. Be smart with your arm. He was my baserunning guy out there. He really taught me a lot."
Tarasco has the ability to connect with a lot of different personalities. When Harper was playing for Class A Hagerstown, for example, Tarasco wasn't afraid to tell Harper to not look ahead to the Major Leagues.
"Mentally, Tony would say, 'Stay in Hagerstown when you are in Hagerstown,'" Harper said. "'Stay in Harrisburg when you are there. Don't get too ahead of yourself. Make sure your mind is on [the Minor Leagues], not D.C. It's just a moment of time when you are up there. You need to keep focused on your [Minor League] team. Not the Nationals.'"
Tarasco said there are three reasons he works well with young players.
"I'm real. I come from the heart and it's about them," he said.
To Tarasco, it helps that he grew up in Los Angeles and hung out with a melting pot of people.
"I wasn't isolated to just one group or two groups," he said. "I grew up in a [multi-cultural environment]. That played a huge part in it. I listened to all kind of music growing up."
Tarasco also learned a lot while playing in the Braves' organization, the team that drafted him in the 1988 First-Year Player Draft. He learned to be on an even keel by being around coaches like Willie Stargell, who was the team's hitting coordinator.
"I took a lot from that guy. That guy represented a lot of things," Tarasco said. "He represented how to play the game -- mechanically, mentally. We used to call Willie even keel. Willie was the same guy. You didn't know he went 5-for-5 or 0-for-5. Me, Chipper Jones and Ryan Klesko spent a lot of time with Willie. I tip my hat to that organization for putting the people around me. I came up with some people who taught me how to play the game. I'm not reinventing anything. I'm giving the same things [the Braves] gave me."
While he plans to be a valuable teacher, Tarasco wants to be part of a ruthless mentality the Nationals have already established when it comes to running the bases.
"I would like to continue seeing them deliver the message to the opposing team that when we get off that bus, we are coming to play," Tarasco said. "I think with that attitude and that mentality, they have been able to turn some things around, and it has been a fun energy to watch."
Bill Ladson is a reporter for MLB.com and writes an MLBlog, All Nats All the time. He also could be found on Twitter @WashingNats. This story was not subject to the approval of Major League Baseball or its clubs.


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This item is currently not available.
This freestanding, walk-in 9' x 9' greenhouse sets up quickly to create a protected growing area for plants. Handy features make it an ideal place to start seeds, harden off transplants,
protect container plantings and extend the growing season. The clear glazing material allows maximum light and heat transmission. Open bottom lets you place over existing plants. Two large, zippered screen doors provide easy access. Screened vents provide good air circulation while keeping out bugs and birds. Includes access portholes for hoses and power cords. Sets up easily; no tools required. Folds to a compact size; includes bag for easy transport and storage.
(based on 3 reviews)
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(10 of 10 customers found this review helpful)
Not very sturdy
By Sara
from New Hampshire
Best Uses
Comments about : Clear Farmhouse Greenhouse, 9' x 9':
Within a couple days of putting this up there were small tears from the wind and then, even with trying to fix them, they eventually grew. The whole corner seam then ripped out within 10 days. Not worth your money, unless you have a pretty sheltered location in which to place it.
Bottom Line No, I would not recommend this to a friend
Merchant response: We're so sorry that this item hasn't worked out for you! We do offer a 100% satisfaction guarentee, so please call our customer service departement at 1-800-876-5520.
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Don't Buy
By Marge
from Wallingford, CT
The concept is a good one but the execution was poor. The instructions for set up are extremely poor and contrary to what the bos says, it is definitely not self erecting. It is cheaply made out of very poor quality materials. I hope it will get me through this one season. Save your money.
(6 of 7 customers found this review helpful)
So far so good.
By Flyboy Sam
from Seabeck, WA
The greenhouse arrives in double boxed cardboard and is enclosed in a bag the size of a truck tire. When I tried to right the bag, using the sewn in strap the bag ripped open due to the 70 lb weight. Assembly is NOT a one person job. It takes 2 people to get it done. The support tubing simply inserts together (not good enough) so I decided to reinforce the joins by wrapping all of them with foil tape. One X brace fit the corner pockets fine, but two of us could not get the second cross brace to form a straight "X". So I simply used more foil tape to bind them. The side windows unzip nicely but there is no way to secure the outer flap up. Velcro squares didn't hold. Because I intend to use the structure to also store firewood, I put a 10x10 tarp down on the bare ground. Pounding the ground stakes in was horrible, due to the local rocky soil. Even drilling into the ground didn't help.I had to buy stronger ones and quick-creted the critical ones into the ground. I consider these flaws to be relatively minor and overall am satisfied with my purchase. Now for the test to see if it will survive 8 months of winter. (Yes, I have a 20' brush to deal with snow loading).
Bottom Line Yes, I would recommend this to a friend
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If I downsize the embellishments & papers in a kit (for storage purposes) will it affect the outcome of the finished album?
No announcement yet.
new posts
From my personal point of view :-) When I design a kit, the very last stage is to "optimize" each file... and I always spend time on this task. So mine are altready optimized... so you wouldn't have to reduce their size.
I'm sure that most of the designers in the SBM store optimize them.
If you reduce the size, you lose quality... depending of the size of your pages.
I know some designers (no name :-)) who don't optimize them and the result is that for 100 embies you have about 180 Mb!!! if not more!
What you could do is use an external disk to store your kits...
Also note that if you have to reduce the size of each item of many kits it will be a VERY FASTIDIOUS task! and very long..
No seriously don't modify the size.
Hope that it helps.
You're welcome.
Sandrine, What is your interpretation of the word "optimize"?
I always try to flatten my layers on .jpg images and trim excess on .pngs and also reduce size if originally made large to view detail as they are being made. Will group layers or merge visible on .png files before saving too.
My Buddies - pkdoll, Marion and CraftyScraps
Optimize? means optimize. LOL. Seriously. I can't explain this *process* in just a few words. I'm going to give you some tips...
When you design, you should perform different steps... same thing when you develop app.
You should define what you will do and how... ie the theme, the palette, the list of embies and papers... if you need to create new shapes for your embies, new texture styles...
Once done, you start to design.
You have to create new stuff from scratch (shapes, styles...) and to "customize" existing stuff.
Test your styles to have an idea of the final weight of your embies once the style is applied... An embie is a series of layers with styles which determine the final weight, as well as the size of your original document, of course (always work in 300 dpi).
For new stuff you should "test" each new embie... papers... the best way is to create quickly a page in SBM and to export it as .jpeg - 300 dpi - 3600 x 3600.... and then you zoom +, +, +... to check any detail. You can also print.
Once everything is created, you should optimize all your files.
First, remove all the "blank" space around the embies and check your embies aren't cut...
If you use CU, PLEASE apply a COLOR AROUND the object (a different color of the object - 10 px)... Use the zoom... so if there's any LOST PIXEL around the object you will see them immediately.
You should check the size/weight of each of them: pixels - Kb or Mb and reduce it if necessary.
A flower... 1000 x 1000 px is fine... 300 x 300 px is TOO small.
The weight of your file is 2.5 Mb... it's TOO heavy... You should try to reduce the size and you should get a lower weight.
For the papers...
5 Mb is TOO heavy!!! You shouldn't have papers whose weight is higher than 3 Mb... and I would even say 2.5 Mb. Try to reduce the quality option... but don't select a value lower then 7. Check if you have unuseful layers... idf yes, remove them, and save again.
So that's some tips...
To conclude you should review all your files, and zoom at the embies/papers to detect any issue :-)
Hope that it helps.
I would say "THINK QUALITY" :-)
SandrineLast edited by KapiScrap; 03-09-2012, 01:06 PM.


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Investigation into age, identity delay Mejia's arrival
By Anthony DiComo / MLB.com
PORT ST. LUCIE, Fla. -- The United States government has delayed Jenrry Mejia's attempts to obtain a visa to enter the country, pending an investigation into his age and identity.
General manager Sandy Alderson said he believes the review of Mejia's documents is random, though such investigations are rare for international players who have already spent multiple summers in the States. Major League Baseball is conducting the identity review for the U.S. consulate, and the Mets informed the league that they consider it a priority.
Still, the club has no clear idea of how long it may take.
"It was unanticipated by the player, by us, by Major League Baseball for that matter," Alderson said. "It's very unusual for a Latin player who has been to the United States on multiple occasions and has been on a Major League roster for several years to be investigated. But as I said, we don't believe there's any specific reason for it. We think it was done on a random basis."
Mejia, who is officially listed as 22 years old, has played professionally in the United States since 2008. He is working out at the Mets' training facility in the Dominican Republic in an effort to remain on pace with his peers in Florida.
"I'm hoping that the work he's able to get there will not leave him too far behind when he actually gets here," Alderson said.
Though he will likely open this season at Triple-A Las Vegas, Mejia is a fringe candidate for the Mets' Opening Day bullpen. The length of his travel delay could affect that competition.
Alderson: Fifth year for Bourn was stumbling block
PORT ST. LUCIE, Fla. -- Even if the Mets had received a positive ruling on their Draft pick concerns in their pursuit of outfielder Michael Bourn, they may not have signed him to a free-agent contract. General manager Sandy Alderson said Wednesday that Bourn's desire for a fifth-year vesting option was critical to New York's decision to bow out of negotiations.
"There were two issues," Alderson said. "One was the fifth-year vest, which we viewed as a fifth year in the contract at the levels we were talking about. We weren't prepared to go to five years, and they had known that for some time. The vest was a big issue for us."
The Mets previously made their stance with regard to the 11th overall pick in the First-Year Player Draft well known. If the league ruled that the Mets would have to forfeit their first-round pick to sign Bourn, they would not have proceeded.
But it never reached that point because neither the Mets nor Bourn caved in negotiations. Bourn sought a fifth-year option, which he ultimately received in his four-year, $48 million deal with the Indians. Alderson indicated he was unwilling to add that clause, even if it meant losing out on a premium free agent.
"There's some disappointment, obviously," Alderson said. "We made a strong effort to try to sign him and weren't able to do it, which is unfortunate. But, nonetheless, we move on."
Per Mets' wishes, Santana doubtful for Classic
PORT ST. LUCIE, Fla. -- General manager Sandy Alderson again hinted that Johan Santana is unlikely to pitch in the World Baseball Classic, despite Santana's desire to represent his native Venezuela.
"It's a complicated situation, just because of his injury and the fact that he did not pitch at the end of last season," Alderson said. "Ultimately, he needs our permission to play, and what we're going to do is take a look at it over the next few days."
Because Santana finished last season on the disabled list, the World Baseball Classic would need to insure the $31 million remaining on his contract for him to gain eligibility. That in itself is unlikely, and even if it comes to pass, the Mets could still ask Santana not to play. The left-hander has said that he would respect those wishes.
"We're very concerned that he'd be ready to go and healthy for us," Alderson said. "There are reasons why players in his situation at the end of a previous season typically don't play in the WBC. So we'll have to take a hard look at it."
Worth noting
• The Mets have added two Citi Field Fireworks Nights to their promotional calendar on May 24 (following a 7:10 p.m. ET game against the Braves) and July 3 (7:10 p.m. against the D-backs), the team announced Wednesday.
• The club also announced that All-Star Six Packs are on sale now at mets.com. Each pack includes a guaranteed ticket to Opening Day or a Subway Series game, in addition to five other regular-season games and Major League Baseball's All-Star FanFest in July. The Mets plan to announce additional promotional dates in the weeks ahead.
Anthony DiComo is a reporter for MLB.com. Follow him on Twitter @AnthonyDicomo. This story was not subject to the approval of Major League Baseball or its clubs.


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I think the London Olympics from a spectator's POV were a spectacular success.
It was always in my mind that London, as one of the world's largest and most open cities might fall prey to a terrorist attack.
Thankfully that did not happen and I was glad to hear Sebastien Coe at the closing ceremonies thank "those who protected us."
Anyone who has read my posts about drugs and cheating knows where I stand on that issue.
If I have been tough on Bolt please take it as an overall and overwhelming feeling about what I see as incredible hypocrisy from the IOC and many of the governing bodies of various Olympic sports.
I am getting weary of hearing "I have never failed a drug test."
Several links I posted were made with the intent to show how easy it is to use banned substances and not get caught.
It can be easy enough to conclude the spirit of the Olympic movement exists today primarily as a way for certain bodies to make big money.
Like most sports...the Olympics are not the first to be changed by monetary greed.
But still.......I saw some amazing moments and I remain hopeful for the future.
For me the legacy of the London games will be the great spirit shown by the British people. Of course they cheered on their own athletes but having seen many events I think there was a wonderful and genuine show of good will and support at all of the venues for Olympians of all nationalities.
For now that is enough and I say "London, and Great Britain, well done.
My favorite moment of the games was the goal scored by Alex Morgan to beat Canada....but I am a football nut and a huge Alex Morgan fan.
Gabby the great was my favorite athlete at the games. And Mustafina captivated me as well.
My favorite moment of sportsmanship was by the Japanese Women's football team.
Despite a heartbreaking defeat they showed pure class. To hear the way they were cheered by the crowd as they approached the podium is what has always made the Olympics special to me.


text:
An infographic
highlighting the results of recent a survey of college students shows that while 86 percent of millennials visit Facebook every day, they do so to
browse other people's profiles, not learn about brands.
Sixty-nine percent of millennials rarely—or never—visit fan pages. Only 1 percent visit a brand page daily.
Millennials do like some organizations, but they are mostly organizations they're involved with offline or locally. Some examples are sororities and
fraternities, nonprofits and sports teams.
Some companies, though, do land a coveted spot on millennials' radar. Among those are Coca-Cola, Toms, Victoria's Secret, and Target.
How can you get millennials to like your brand on Facebook? Millennials say they are more likely to like an organization if it:
Home | Social Media | Media Relations | Crisis | Marketing | Writing & Editing | Events | Training | Awards | Videos
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text:
- Korean Grand Prix
'We just don't have the speed' - Raikkonen
- Race:
- Korean Grand Prix
- Drivers:
- Kimi Räikkönen
- Teams:
- Lotus F1
Kimi Raikkonen thinks Lotus is lacking the speed for him to mount a late comeback in the drivers' championship over the next four races.
Raikkonen finished fifth at the Korean Grand Prix but could not match the pace of the Red Bulls and Ferraris in the race. Lotus brought a new Coanda-effect exhaust system to its car this weekend, but Raikkonen said the E20 is still lacking the pace for him to close the 48-point gap in the remaining four races.
"We keep doing our best but we just don't have the speed," he said. "We'll keep trying and pushing but it doesn't look likely."
He said that fifth was the most Lotus could have hoped for in Korea.
"I think this was about where we should have been today," Raikkonen said. "The yellow flags at the start didn't help when we were fighting with Felipe [Massa] but even so we didn't quite have the speed to match the Red Bulls and Ferraris. We also lost quite a bit of time behind Lewis [Hamilton] - mainly because the new exhaust system means you lose a little bit of power so it was hard to get past - but with a few more practice sessions I'm sure we'll learn a lot more to be able to improve it. I was pretty much on my own at the end but we had to keep pushing to maintain position and it wasn't so easy."


text:
A man of integrity and honor, David Gage Smokler passed away April 6, 2012, with a life fully lived and a remarkable legacy at the age of 85. He was a loving son, husband, father, grandfather, father-in-law, uncle, cousin and friend.
A gentle, kind man, who never met a stranger, David found joy and beauty in all things simple. He was born in Dallas, Texas, December 17, 1926, to Louis Harold Smokler and Pansy Myra Gage. After graduating from Texas A&M as the University's youngest veterinarian at age 19 and returning from his service in the U.S. Army, David married Rosemary Self. Over their 62 years of marriage, he adored his three daughters: Patricia Lewis, Janet King, and Susan Graves. He was blessed with five grandchildren: Kara and Jared King, Alicia Lewis, and Heather and Dave Graves. David enjoyed the company and conversations he shared with "the chosen" sons-in-law: Ron Lewis, Chad King, and James Graves.
As a third generation dairyman with Holtex Farm, David (Doc) Smokler mentored many generations of dairymen and farmers. As a premier dairyman, he was a pioneer in artificial breeding starting in the 1940s. He was respected nationally and internationally in the Holstein community. He served as President of the Texas Holstein Association and the Holstein Association USA and led the industry in genetic breeding and dairy management.
When asked how he was--he often replied "nearly perfect." David always said few people had the fun of working their lifetime at something they truly enjoyed. He was fortunate to have found a career in which he was able to combine his passion for nature, the outdoors and animals.
A private, family celebration of his life was held. In lieu of flowers, the family prefers a memorial donation to the charity of your choice. Condolences may be offered at www.millerfuneral.com or 3740 Cartwright Rd., PMB 262, Missouri City, Texas 77459.


text:
SoXSection: User Commands (1)
Updated: December 11, 2001
Index Return to Main Contents
NAMEsox - Sound eXchange : universal sound sample translator
SYNOPSISsox infile1 [ infile2 ... ] outfile sox [ general options ] [ format options ] infile1
[ [ format options ] infile2 ... ] [ format options ] outfile
[ effect [ effect options ] ... ] soxmix infile1 infile2 [ infile3 ... ] outfile soxmix [ general options ] [ format options ] infile1
[ format options ] infile2
[ [ format options ] infile3 ... ]
[ format options ] outfile
[ effect [ effect options ] ... ]
General options:
[ -h ] [ -p ] [ -q ] [ -S ] [ -V ] Format options:
[ -t filetype ] [ -r rate ] [ -s/-u/-U/-A/-a/-i/-g/-f ]
[ -b/-w/-l/-d ] [ -v volume ]
[ -c channels ] [ -x ] [ -e ] Effects:
avg [ -l | -r | -f | -b | -1 | -2 | -3 | -4 | n,n,...,n ]
band [ -n ] center [ width ]
bandpass frequency bandwidth
bandreject frequency bandwidth
chorus gain-in gain out delay decay speed depth
-s | -t [ delay decay speed depth -s | -t ]
compand attack1,decay1[,attack2,decay2...]
[ gain [ initial-volume [ delay ] ] ]
dcshift shift [ limitergain ]
echo gain-in gain-out delay decay [ delay decay ... ]
echos gain-in gain-out delay decay [ delay decay ... ]
fade [ type ] fade-in-length
[ stop-time [ fade-out-length ] ]
filter [ low ]-[ high ] [ window-len [ beta ]]
flanger gain-in gain-out delay decay speed < -s | -t >
highp frequency
highpass frequency
lowp frequency
lowpass frequency
mcompand "attack1,decay1[,attack2,decay2...]
[ gain [ initial-volume [ delay ] ] ]" xover_freq
noiseprof [profile-file]
noisered profile-file [threshold]
pan direction
phaser gain-in gain-out delay decay speed < -s | -t >
pick [ -1 | -2 | -3 | -4 | -l | -r | -f | -b ]
pitch shift [ width interpole fade ]
polyphase [ -w < nut / ham > ]
[ -width < long / short / # > ]
[ -cutoff # ]
repeat count
resample [ -qs | -q | -ql ] [ rolloff [ beta ] ]
reverb gain-out reverb-time delay [ delay ... ]
silence above_periods [ duration threshold[ d | % ]
[ below_periods duration
threshold[ d | % ]]
speed [ -c ] factor
stat [ -s n ] [ -rms ] [ -v ] [ -d ]
stretch [ factor [ window fade shift fading ]
swap [ 1 2 | 1 2 3 4 ]
synth [ length ] type mix [ freq [ -freq2 ]
[ off ] [ ph ] [ p1 ] [ p2 ] [ p3 ]
trim start [ length ]
vibro speed [ depth ]
vol gain [ type [ limitergain ] ]
DESCRIPTIONSoX is a command line program that can convert most popular audio files to most other popular audio file formats. It can optionally change the audio sample data type and apply one or more sound effects to the file during this translation. If more than one input file is specified then they are concatenated into the output file. In this case, it has a restriction that all input files must be of the same data type and sample rates. soxmix is functionally the same as the command line program sox expect that it takes two or more files as input and mixes the audio together to produce a single file as output. It has a restriction that all input files must be of the same data type and sample rates. There are two types of audio file formats that SoX can work with. The first are self-describing file formats. These contain a header that completely describe the characteristics of the audio data that follows. The second type are header-less data, or sometimes called raw data. A user must pass enough information to SoX on the command line so that it knows what type of data it contains. Audio data can usually be totally described by four characteristics:
- rate
- The sample rate is in samples per second. For example, CD sample rates are at 44100.
- data size
- The precision the data is stored in. Most popular are 8-bit bytes or 16-bit words.
- data encoding
- What encoding the data type uses. Examples are u-law, ADPCM, or signed linear data.
- channels
- How many channels are contained in the audio data. Mono and Stereo are the two most common. Please refer to the soxexam(1) manual page for a long description with examples on how to use SoX with various types of file formats.
OPTIONSThe option syntax is a little grotty, but in essence:
sox file.au file.wav
translates a sound file in SUN Sparc .AU format into a Microsoft .WAV file, while
sox -v 0.5 file.au -r 12000 file.wav mask
does the same format translation but also lowers the amplitude by 1/2, changes the sampling rate to 12000 hertz, and applies the mask sound effect to the audio data. The following will mix two sound files together to to produce a single sound file.
soxmix music.wav voice.wav mixed.wav
SoX can be used as a part of pipe operations by using the special filenames of "-". If specified as an input name, it will read data from stdin. If specified as an output name, it will send data to stdout.
General options:
- -h
- Print version number and usage information.
- -p
- Run in preview mode and run fast. This will somewhat speed up SoX when the output format has a different number of channels and a different rate than the input file. Currently, this defaults to using the rate effect instead of the resample effect for sample rate changes.
- -q
- Run in quite mode when SoX wouldn't otherwise do that. Inverse of -S option.
- Print status while processing audio data. Tells how much of audio data has been processed in terms of audio running time instead of samples.
- Print a description of processing phases. Useful for figuring out exactly how SoX
is mangling your sound samples.
Format options:
Format options effect the input or output file that they immediately precede.
Self describing input files can obtain all the format information directly from the header and so don't generally need format options. Headerless input files lack this information and so format options must be used to inform SoX of the file's data type, sample rate, and number of channels.
By default, SoX attempts to write audio data using the same data type, sample rate, and channel count as the input data. If the user wants the output file to be of a different format then format options can be used to specify the differences.
If an output file format doesn't support the same data type, sample rate, or channel count as the input file format, then SoX will auto select the closest values it does support so that the user does not have to specify these format change options manually.
- -t filetype
- gives the type of the sound sample file. Useful when file extension is not standard or can not be determeind by looking at the header of the file.
- -r rate
Gives the sample rate in Hertz of the file. To cause the output file to have
a different sample rate than the input file, include this option as a part
of the output format options.
If the input and output files have different rates then a sample rate change effect must be ran. Since SoX has multiple rate changing effects, the user can specify which to use as an effect. If no rate change effect is specified then a default one will be chosen.
- -v volume
Change amplitude (floating point);
less than 1.0 decreases, greater than 1.0 increases. May use a negative
number to invert the phase of the audio data. It is interesting to note
that we perceive volume
logarithmically but this adjusts the amplitude linearly.
As with other format options, the volume option effects the file its specified with. This is useful whe processing mutiple input files as the volume adjustment can be specified for each input file or just once to adjust the output file. This can be compared to an audio mixer were you can control the volume of each input as well as a master volume (output side).
soxmix defaults the value of the -v option for each input file to 1/input_file_count. This means if your mixing two input files together then each input file's volume is adjusted by 0.5. This is done to prevent clipping of audio data during the mixing operation. Users will most likely not be happy with this large of a volume adjustment and can specify the -v option to override this default value.
Note: For the non-mixing case, see the stat effect for information on finding the maximum volume adjustment that can be done with this option without causing audio data to be clipped.
- -s/-u/-U/-A/-a/-i/-g/-f
The sample data encoding is signed linear (2's complement),
unsigned linear, u-law (logarithmic), A-law (logarithmic),
ADPCM, IMA_ADPCM, GSM, or Floating-point.
U-law (actually shorthand for mu-law) and A-law are the U.S. and international standards for logarithmic telephone sound compression. When uncompressed u-law has roughly the precision of 14-bit PCM audio and A-law has roughly the precision of 13-bit PCM audio.
A-law and u-law data is sometimes encoded using a reversed bit-ordering (ie. MSB becomes LSB). Internally, SoX understands how to work with this encoding but there is currently no command line option to specify it. If you need this support then you can use the psuedo file types of ".la" and ".lu" to inform sox of the encoding. See supported file types for more information.
ADPCM is a form of sound compression that has a good compromise between good sound quality and fast encoding/decoding time. It is used for telephone sound compression and places were full fidelity is not as important. When uncompressed it has roughly the precision of 16-bit PCM audio. Popular version of ADPCM include G.726, MS ADPCM, and IMA ADPCM. The -a flag has different meanings in different file handlers. In .wav files it represents MS ADPCM files, in all others it means G.726 ADPCM. IMA ADPCM is a specific form of ADPCM compression, slightly simpler and slightly lower fidelity than Microsoft's flavor of ADPCM. IMA ADPCM is also called DVI ADPCM.
GSM is a standard used for telephone sound compression in European countries and its gaining popularity because of its quality. It usually is CPU intensive to work with GSM audio data.
- -b/-w/-l/-d
- The sample data size is in bytes, 16-bit words, 32-bit long words, or 64-bit double long (long long) words.
- -x
- The sample data is in XINU format; that is, it comes from a machine with the opposite word order than yours and must be swapped according to the word-size given above. Only 16-bit and 32-bit integer data may be swapped. Machine-format floating-point data is not portable.
- -c channels
- The number of sound channels in the data file. This may be 1, 2, or 4; for mono, stereo, or quad sound data. To cause the output file to have a different number of channels than the input file, include this option with the output file options. If the input and output file have a different number of channels then the avg effect must be used. If the avg effect is not specified on the command line it will be invoked internally with default parameters.
- -e
- When specified after the last input filename (so that it applies to the output file) it allows you to avoid giving an output filename and will not produce an output file. It will apply any specified effects to the input file. This is mainly useful with the stat effect but can be used.
FILE TYPESSoX attempts to determine the file type of input files automatically by looking at the header of the audio file. When it is unable to detect the file type or if its an output file then it uses the file extension of the file to determine what type of file format handler to use. This can be overridden by specifying the "-t" option on the command line. The input and output files may be read from standard in and out. This is done by specifying '-' as the filename. File formats which have headers are checked, if that header doesn't seem right, the program exits with an appropriate message. The following file formats are supported:
- .8svx
- Amiga 8SVX musical instrument description format.
- .aiff
- AIFF files used on Apple IIc/IIgs and SGI. Note: the AIFF format supports only one SSND chunk. It does not support multiple sound chunks, or the 8SVX musical instrument description format. AIFF files are multimedia archives and can have multiple audio and picture chunks. You may need a separate archiver to work with them.
- .alsa
ALSA /dev/snd/pcmCxDxp device driver
This is a pseudo-file type and can be optionally compiled into SoX. Run sox -h to see if you have support for this file type. When this driver is used it allows you to open up the ALSA /dev/snd/pcmCxDxp file and configure it to use the same data format as passed in to SoX. It works for both playing and recording sound samples. When playing sound files it attempts to set up the ALSA driver to use the same format as the input file. It is suggested to always override the output values to use the highest quality samples your sound card can handle. Example: sox infile -t alsa -w -s /dev/snd/pcmC0D0p
- .au
- SUN Microsystems AU files. There are apparently many types of .au files; DEC has invented its own with a different magic number and word order. The .au handler can read these files but will not write them. Some .au files have valid AU headers and some do not. The latter are probably original SUN u-law 8000 hz samples. These can be dealt with using the .ul format (see below).
- .avr
Audio Visual Research
The AVR format is produced by a number of commercial packages on the Mac.
- .cdr
CD-R files are used in mastering music on Compact Disks. The audio data on a CD-R disk is a raw audio file with a format of stereo 16-bit signed samples at a 44khz sample rate. There is a special blocking/padding oddity at the end of the audio file and is why it needs its own handler.
- .cvs
Continuously Variable Slope Delta modulation
Used to compress speech audio for applications such as voice mail.
- .dat
Text Data files
These files contain a textual representation of the sample data. There is one line at the beginning that contains the sample rate. Subsequent lines contain two numeric data items: the time since the beginning of the first sample and the sample value. Values are normalized so that the maximum and minimum are 1.00 and -1.00. This file format can be used to create data files for external programs such as FFT analyzers or graph routines. SoX can also convert a file in this format back into one of the other file formats.
- .gsm
GSM 06.10 Lossy Speech Compression
A standard for compressing speech which is used in the Global Standard for Mobil telecommunications (GSM). Its good for its purpose, shrinking audio data size, but it will introduce lots of noise when a given sound sample is encoded and decoded multiple times. This format is used by some voice mail applications. It is rather CPU intensive.
GSM in SoX is optional and requires access to an external GSM library. To see if there is support for gsm run sox -h and look for it under the list of supported file formats.
- .hcom
- Macintosh HCOM files. These are (apparently) Mac FSSD files with some variant of Huffman compression. The Macintosh has wacky file formats and this format handler apparently doesn't handle all the ones it should. Mac users will need your usual arsenal of file converters to deal with an HCOM file under Unix or DOS.
- .maud
An Amiga format
An IFF-conform sound file type, registered by MS MacroSystem Computer GmbH, published along with the "Toccata" sound-card on the Amiga. Allows 8bit linear, 16bit linear, A-Law, u-law in mono and stereo.
- .mp3
MP3 Compressed Audio
MP3 audio files come from the MPEG standards for audio and video compression. They are a lossy compression format that achieves good compression rates with a minimum amount of quality loss. Also see Ogg Vorbis for a similar format. MP3 support in SoX is optional and requires access to either or both the external libmad and libmp3lame libraries. To see if there is support for Mp3 run sox -h and look for it under the list of supported file formats as "mp3".
- .nul
- Null file handler. This is a fake file hander that act as if its reading a stream of 0's from a while or fake writing output to a file. This is not a very useful file handler in most cases. It might be useful in some scripts were you do not want to read or write from a real file but would like to specify a filename for consistency.
- .ogg
Ogg Vorbis Compressed Audio.
Ogg Vorbis is a open, patent-free CODEC designed for compressing music and streaming audio. It is similar to MP3, VQF, AAC, and other lossy formats. SoX can decode all types of Ogg Vorbis files, but can only encode at 128 kbps. Decoding is somewhat CPU intensive and encoding is very CPU intensive.
Ogg Vorbis in SoX is optional and requires access to external Ogg Vorbis libraries. To see if there is support for Ogg Vorbis run sox -h and look for it under the list of supported file formats as "vorbis".
- ossdsp
OSS /dev/dsp device driver
This is a pseudo-file type and can be optionally compiled into SoX. Run sox -h to see if you have support for this file type. When this driver is used it allows you to open up the OSS /dev/dsp file and configure it to use the same data format as passed in to SoX. It works for both playing and recording sound samples. When playing sound files it attempts to set up the OSS driver to use the same format as the input file. It is suggested to always override the output values to use the highest quality samples your sound card can handle. Example: sox infile -t ossdsp -w -s /dev/dsp
- .prc
Psion record.app
Used in some Psion devices for System alarms. This format is newer then the .wve format that is used in some Psion devices.
- .sf
IRCAM Sound Files.
Sound Files are used by academic music software such as the CSound package, and the MixView sound sample editor.
- .sph
SPHERE (SPeech HEader Resources) is a file format defined by NIST (National Institute of Standards and Technology) and is used with speech audio. SoX can read these files when they contain u-law and PCM data. It will ignore any header information that says the data is compressed using shorten compression and will treat the data as either u-law or PCM. This will allow SoX and the command line shorten program to be ran together using pipes to uncompress the data and then pass the result to SoX for processing.
- .smp
Turtle Beach SampleVision files.
SMP files are for use with the PC-DOS package SampleVision by Turtle Beach Softworks. This package is for communication to several MIDI samplers. All sample rates are supported by the package, although not all are supported by the samplers themselves. Currently loop points are ignored.
- .snd
Under DOS this file format is the same as the .sndt format. Under all other platforms it is the same as the .au format.
- .sndt
SoundTool files.
This is an older DOS file format.
- sunau
Sun /dev/audio device driver
This is a pseudo-file type and can be optionally compiled into SoX. Run sox -h to see if you have support for this file type. When this driver is used it allows you to open up a Sun /dev/audio file and configure it to use the same data type as passed in to SoX. It works for both playing and recording sound samples. When playing sound files it attempts to set up the audio driver to use the same format as the input file. It is suggested to always override the output values to use the highest quality samples your hardware can handle. Example: sox infile -t sunau -w -s /dev/audio or sox infile -t sunau -U -c 1 /dev/audio for older sun equipment.
- .txw
Yamaha TX-16W sampler.
A file format from a Yamaha sampling keyboard which wrote IBM-PC format 3.5" floppies. Handles reading of files which do not have the sample rate field set to one of the expected by looking at some other bytes in the attack/loop length fields, and defaulting to 33kHz if the sample rate is still unknown.
- .vms
More info to come.
Used to compress speech audio for applications such as voice mail.
- .voc
Sound Blaster VOC files.
VOC files are multi-part and contain silence parts, looping, and different sample rates for different chunks. On input, the silence parts are filled out, loops are rejected, and sample data with a new sample rate is rejected. Silence with a different sample rate is generated appropriately. On output, silence is not detected, nor are impossible sample rates. Note, this version now supports playing VOC files with multiple blocks and supports playing files containing u-law and A-law samples.
- vorbis
- See .ogg format.
- .vox
- A headerless file of Dialogic/OKI ADPCM audio data commonly comes with the extension .vox. This ADPCM data has 12-bit precision packed into only 4-bits.
- .wav
Microsoft .WAV RIFF files.
These appear to be very similar to IFF files, but not the same. They are the native sound file format of Windows. (Obviously, Windows was of such incredible importance to the computer industry that it just had to have its own sound file format.)
Normally .wav files have all formatting information in their headers, and so do not need any format options specified for an input file. If any are, they will override the file header, and you will be warned to this effect. You had better know what you are doing! Output format options will cause a format conversion, and the .wav will written appropriately.
SoX currently can read PCM, ULAW, ALAW, MS ADPCM, and IMA (or DVI) ADPCM. It can write all of these formats including the ADPCM encoding. Big endian versions of RIFF files, called RIFX, can also be read and written. To write a RIFX file, use the -x option with the output file options.
- .wve
Psion 8-bit A-law
These are 8-bit A-law 8khz sound files used on the Psion palmtop portable computer.
- .raw
Raw files (no header).
The sample rate, size (byte, word, etc), and encoding (signed, unsigned, etc.) of the sample file must be given. The number of channels defaults to 1.
- .ub, .sb, .uw, .sw, .ul, .al, .lu, .la, .sl
- These are several suffices which serve as a shorthand for raw files with a given size and encoding. Thus, ub, sb, uw, sw, ul, al, lu, la and sl correspond to "unsigned byte", "signed byte", "unsigned word", "signed word", "u-law" (byte), "A-law" (byte), inverse bit order "u-law", inverse bit order "A-law", and "signed long". The sample rate defaults to 8000 hz if not explicitly set, and the number of channels defaults to 1. There are lots of Sparc samples floating around in u-law format with no header and fixed at a sample rate of 8000 hz. (Certain sound management software cheerfully ignores the headers.) Similarly, most Mac sound files are in unsigned byte format with a sample rate of 11025 or 22050 hz.
- .auto
- This is a ``meta-type'' and is the default file type if the user does not specify one. This file type attempts to guess the real type by looking for magic words in the header. If the type can't be guessed, the program exits with an error message. The input must be a plain file, not a pipe. This type can't be used for output files.
EFFECTSMultiple effects may be applied to the audio data by specifying them one after another at the end of the command line.
- avg [ -l | -r | -f | -b | -1 | -2 | -3 | -4 | n,n,...,n ]
Reduce the number of channels by averaging the samples,
or duplicate channels to increase the number of channels.
This effect is automatically used when the number of input
channels differ from the number of output channels. When reducing
the number of channels it is possible to manually specify the
avg effect and use the -l, -r, -f, -b,
-1, -2, -3, -4, options to select only
the left, right, front, back channel(s) or specific channel
for the output instead of averaging the channels.
The -l, and -r options will do averaging
in quad-channel files so select the exact channel to prevent this.
The avg effect can also be invoked with up to 16 double-precision numbers, seperated by commas, which specify the proportion (0.0 = 0% and 1.0 = 100%) of each input channel that is to be mixed into each output channel. In two-channel mode, 4 numbers are given: l->l, l->r, r->l, and r->r, respectively. In four-channel mode, the first 4 numbers give the proportions for the left-front output channel, as follows: lf->lf, rf->lf, lb->lf, and rb->rf. The next 4 give the right-front output in the same order, then left-back and right-back.
It is also possible to use the 16 numbers to expand or reduce the channel count; just specify 0 for unused channels.
Finally, certain reduced combination of numbers can be specified for certain input/output channel combinations.
In Ch Out Ch Num Mappings
_____ ______ ___ _____________________________
2 1 2 l->l, r->l
2 2 1 adjust balance
4 1 4 lf->l, rf->l, lb->l, rb-l
4 2 2 lf->l&rf->r, lb->l&rb->r
4 4 1 adjust balance
4 4 2 front balance, back balance
- band [ -n ] center [ width ]
- Apply a band-pass filter. The frequency response drops logarithmically around thecenter frequency. The width gives the slope of the drop. The frequencies at center + width and center - width will be half of their original amplitudes. Band defaults to a mode oriented to pitched signals, i.e. voice, singing, or instrumental music. The -n (for noise) option uses the alternate mode for un-pitched signals. Warning: -n introduces a power-gain of about 11dB in the filter, so beware of output clipping. Band introduces noise in the shape of the filter, i.e. peaking at the center frequency and settling around it. See filter for a bandpass effect with steeper shoulders.
- bandpass frequency bandwidth
- Butterworth bandpass filter. Description coming soon!
- bandrejectfrequency bandwidth
- Butterworth bandreject filter. Description coming soon!
- chorusgain-in gain-out delay decay speed depth
- | -t [ delay decay speed depth -s | -t ... ]
- Add a chorus to a sound sample. Each quadtuple delay/decay/speed/depth gives the delay in milliseconds and the decay (relative to gain-in) with a modulation speed in Hz using depth in milliseconds. The modulation is either sinusoidal (-s) or triangular (-t). Gain-out is the volume of the output.
- compand attack1,decay1[,attack2,decay2...]
[gain [initial-volume [delay ] ] ]
Compand (compress or expand) the dynamic range of a sample. The
attack and decay time specify the integration time over which the
absolute value of the input signal is integrated to determine its
volume; attacks refer to increases in volume and decays refer to
decreases. Where more than one pair of attack/decay parameters are
specified, each channel is treated separately and the number of pairs
must agree with the number of input channels. The second parameter is
a list of points on the compander's transfer function specified in dB
relative to the maximum possible signal amplitude. The input values
must be in a strictly increasing order but the transfer function does
not have to be monotonically rising. The special value -inf may
be used to indicate that the input volume should be associated output
volume. The points -inf,-inf and 0,0 are assumed; the
latter may be overridden, but the former may not.
The third (optional) parameter is a post-processing gain in dB which is applied after the compression has taken place; the fourth (optional) parameter is an initial volume to be assumed for each channel when the effect starts. This permits the user to supply a nominal level initially, so that, for example, a very large gain is not applied to initial signal levels before the companding action has begun to operate: it is quite probable that in such an event, the output would be severely clipped while the compander gain properly adjusts itself.
The fifth (optional) parameter is a delay in seconds. The input signal is analyzed immediately to control the compander, but it is delayed before being fed to the volume adjuster. Specifying a delay approximately equal to the attack/decay times allows the compander to effectively operate in a "predictive" rather than a reactive mode.
- copy
- Copy the input file to the output file. This is the default effect if both files have the same sampling rate.
- dcshift shift [ limitergain ]
DC Shift the audio data, with basic linear amplitude formula.
This is most useful if your audio data tends to not be centered around
a value of 0. Shifting it back will allow you to get the most volume
adjustments without clipping audio data.
The first option is the dcshift value. It is a floating point number that indicates the amount to shift.
An option limtergain value can be specified as well. It should have a value much less then 1.0 and is used only on peaks to prevent clipping.
- deemph
- Apply a treble attenuation shelving filter to samples in audio cd format. The frequency response of pre-emphasized recordings is rectified. The filtering is defined in the standard document ISO 908.
- earwax
Makes sound easier to listen to on headphones.
Adds audio-cues to samples in audio cd format so that
when listened to on headphones the stereo image is
moved from inside
your head (standard for headphones) to outside and in front of the
listener (standard for speakers). See
www.geocities.com/beinges for a full explanation.
- echo gain-in gain-out delay decay [ delay decay ... ]
- Add echoing to a sound sample. Each delay/decay part gives the delay in milliseconds and the decay (relative to gain-in) of that echo. Gain-out is the volume of the output.
- echos gain-in gain-out delay decay [ delay decay ... ]
- Add a sequence of echos to a sound sample. Each delay/decay part gives the delay in milliseconds and the decay (relative to gain-in) of that echo. Gain-out is the volume of the output.
- fade [ type ] fade-in-length
[ stop-time [ fade-out-length ] ]
Add a fade effect to the beginning, end, or both of the audio data.
For fade-ins, this starts from the first sample and ramps the volume of the audio from 0 to full volume over fade-in-length seconds. Specify 0 seconds if no fade-in is wanted.
For fade-outs, the audio data will be truncated at the stop-time and the volume will be ramped from full volume down to 0 starting at fade-out-length seconds before the stop-time. If fade-out-length is not specified, it defaults to the same value as fade-in-length. No fade-out is performed if the stop-time is not specified.
All times can be specified in either periods of time or sample counts. To specify time periods use the format hh:mm:ss.frac format. To specify using sample counts, specify the number of samples and append the letter 's' to the sample count (for example 8000s).
An optional type can be specified to change the type of envelope. Choices are q for quarter of a sinewave, h for half a sinewave, t for linear slope, l for logarithmic, and p for inverted parabola. The default is a linear slope.
- filter [ low ]-[ high ] [ window-len [ beta ] ]
Apply a Sinc-windowed lowpass, highpass, or bandpass filter of given
window length to the signal.
low refers to the frequency of the lower 6dB corner of the filter.
high refers to the frequency of the upper 6dB corner of the filter.
A lowpass filter is obtained by leaving low unspecified, or 0. A highpass filter is obtained by leaving high unspecified, or 0, or greater than or equal to the Nyquist frequency.
The window-len, if unspecified, defaults to 128. Longer windows give a sharper cutoff, smaller windows a more gradual cutoff.
The beta, if unspecified, defaults to 16. This selects a Kaiser window. You can select a Nuttall window by specifying anything <= 2.0 here. For more discussion of beta, look under the resample effect.
- flanger gain-in gain-out delay decay speed < -s | -t >
- Add a flanger to a sound sample. Each triple delay/decay/speed gives the delay in milliseconds and the decay (relative to gain-in) with a modulation speed in Hz. The modulation is either sinodial (-s) or triangular (-t). Gain-out is the volume of the output.
- highp frequency
- Apply a single pole recursive high-pass filter. The frequency response drops logarithmically with I frequency in the middle of the drop. The slope of the filter is quite gentle. See filter for a highpass effect with sharper cutoff.
- highpass frequency
- Butterworth highpass filter. Description coming soon!
- lowpfrequency
- Apply a single pole recursive low-pass filter. The frequency response drops logarithmically with frequency in the middle of the drop. The slope of the filter is quite gentle. See filter for a lowpass effect with sharper cutoff.
- lowpass frequency
- Butterworth lowpass filter. Description coming soon!
- mask
- Add "masking noise" to signal. This effect deliberately adds white noise to a sound in order to mask quantization effects, created by the process of playing a sound digitally. It tends to mask buzzing voices, for example. It adds 1/2 bit of noise to the sound file at the output bit depth.
- mcompand "attack1,decay1[,attack2,decay2...]
[gain [initial-volume [delay ] ] ]" xover_freq
Multi-band compander is similar to the single band compander but the audio file is first divided up into bands and then the compander is ran on each band. See the compand effect for definition of its options. Compand options are specified between double quotes and the crossover frequency for that band is specefied seperately with xover_fre. This can be repeated multiple times to create multiple bands.
- noiseprof [profile-file]
- noisered profile-file [threshold]
Noise reduction filter with profiling. This filter is moderately effective at
removing consistent background noise such as hiss or hum. To use it, first run
the noiseprof effect on a section of silence
(that is, a section which contains
nothing but noise). The noiseprof effect will print a noise profile
to profile-file, or to stdout if no profile-file is specified.
If there is sound output on stdout then the profile will instead be directed to
To actually remove the noise, run SoX again with the noisered filter. The filter needs one argument, profile-file, which contains the noise profile from noiseprof. thershold specifies how much noise should be removed, and may be between 0 and 1 with a default of 0.5. Higher values will remove more noise but present a greater possibility of distorting the desired audio signal. Experiment with different threshold values to find the optimal one for your sample.
- pan direction
- Pan the sound of an audio file from one channel to another. This is done by changing the volume of the input channels so that it fades out on one channel and fades-in on another. If the number of input channels is different then the number of output channels then this effect tries to intelligently handle this. For instance, if the input contains 1 channel and the output contains 2 channels, then it will create the missing channel itself. Thedirection is a value from -1.0 to 1.0. -1.0 represents far left and 1.0 represents far right. Numbers in between will start the pan effect without totally muting the opposite channel.
- phaser gain-in gain-out delay decay speed < -s | -t >
- Add a phaser to a sound sample. Each triple delay/decay/speed gives the delay in milliseconds and the decay (relative to gain-in) with a modulation speed in Hz. The modulation is either sinodial (-s) or triangular (-t). The decay should be less than 0.5 to avoid feedback. Gain-out is the volume of the output.
- pick [ -1 | -2 | -3 | -4 | -l | -r | -f | -b ]
- Pick a subset of channels to be copied into the output file. This effect is just an alias of the "avg" effect but is left here for historical reasons.
- pitch shift [ width interpole fade ]
- Change the pitch of file without affecting its duration by cross-fading shifted samples.shift is given in cents. Use a positive value to shift to treble, negative value to shift to bass. Default shift is 0. width of window is in ms. Default width is 20ms. Try 30ms to lower pitch, and 10ms to raise pitch. interpole option, can be "cubic" or "linear". Default is "cubic". The fade option, can be "cos", "hamming", "linear" or "trapezoid". Default is "cos".
- polyphase [ -w < nut / ham > ]
[ -width < long / short / # > ]
[ -cutoff # ]
Translate input sampling rate to output sampling rate via polyphase
interpolation, a DSP algorithm. This method is slow and uses lots
of RAM, but gives much better results than
-w < nut / ham > : select either a Nuttal (~90 dB stopband) or Hamming (~43 dB stopband) window. Default is nut.
-width long / short / # : specify the (approximate) width of the filter. long is 1024 samples; short is 128 samples. Alternatively, an exact number can be used. Default is long. The short option is not recommended, as it produces poor quality results.
-cutoff # : specify the filter cutoff frequency in terms of fraction of frequency bandwidth, also know as the Nyquist frequency. Please see the resample effect for further information on Nyquist frequency. If upsampling, then this is the fraction of the original signal that should go through. If downsampling, this is the fraction of the signal left after downsampling. Default is 0.95. Remember that this is a float.
- rate
Translate input sampling rate to output sampling rate
via linear interpolation to the Least Common Multiple
of the two sampling rates.
This is the default effect
if the two files have different sampling rates and the preview options
was specified.
This is fast but noisy:
the spectrum of the original sound will be shifted upwards
and duplicated faintly when up-translating by a multiple.
Lerp-ing is acceptable for cheap 8-bit sound hardware, but for CD-quality sound you should instead use either resample or polyphase. If you are wondering which rate changing effects to use, you will want to read a detailed analysis of all of them at http://leute.server.de/wilde/resample.html
- repeat count
- Repeats the audio data count times. Requires disk space to store the data to be repeated.
- resample [ -qs | -q | -ql ] [ rolloff [ beta ] ]
Translate input sampling rate to output sampling rate
via simulated analog filtration.
This method is slower than
but gives much better results.
By default, linear interpolation is used, with a window width about 45 samples at the lower of the two rate. This gives an accuracy of about 16 bits, but insufficient stopband rejection in the case that you want to have rolloff greater than about 0.80 of the Nyquist frequency.
The -q* options will change the default values for rolloff and beta as well as use quadratic interpolation of filter coefficients, resulting in about 24 bits precision. The -qs, -q, or -ql options specify increased accuracy at the cost of lower execution speed. It is optional to specify rolloff and beta parameters when using the -q* options.
Following is a table of the reasonable defaults which are built-in to SoX:
Option Window rolloff beta interpolation
------ ------ ------- ---- -------------
(none) 45 0.80 16 linear
-qs 45 0.80 16 quadratic
-q 75 0.875 16 quadratic
-ql 149 0.94 16 quadratic
------ ------ ------- ---- -------------
-qs, -q, or -ql use window lengths of 45, 75, or 149 samples, respectively, at the lower sample-rate of the two files. This means progressively sharper stop-band rejection, at proportionally slower execution times.
rolloff refers to the cut-off frequency of the low pass filter and is given in terms of the Nyquist frequency for the lower sample rate. rolloff therefore should be something between 0.0 and 1.0, in practice 0.8-0.95. The defaults are indicated above.
The Nyquist frequency is equal to (sample rate / 2). Logically, this is because the A/D converter needs at least 2 samples to detect 1 cycle at the Nyquist frequency. Frequencies higher then the Nyquist will actually appear as lower frequencies to the A/D converter and is called aliasing. Normally, A/D converts run the signal through a highpass filter first to avoid these problems.
Similar problems will happen in software when reducing the sample rate of an audio file (frequencies above the new Nyquist frequency can be aliased to lower frequencies). Therefore, a good resample effect will remove all frequency information above the new Nyquist frequency.
The rolloff refers to how close to the Nyquist frequency this cutoff is, with closer being better. When increasing the sample rate of an audio file you would not expect to have any frequencies exist that are past the original Nyquist frequency. Because of resampling properties, it is common to have aliasing data created that is above the old Nyquist frequency. In that case the rolloff refers to how close to the original Nyquist frequency to use a highpass filter to remove this false data, with closer also being better.
The beta parameter determines the type of filter window used. Any value greater than 2.0 is the beta for a Kaiser window. Beta <= 2.0 selects a Nuttall window. If unspecified, the default is a Kaiser window with beta 16.
In the case of Kaiser window (beta > 2.0), lower betas produce a somewhat faster transition from passband to stopband, at the cost of noticeable artifacts. A beta of 16 is the default, beta less than 10 is not recommended. If you want a sharper cutoff, don't use low beta's, use a longer sample window. A Nuttall window is selected by specifying any 'beta' <= 2, and the Nuttall window has somewhat steeper cutoff than the default Kaiser window. You will probably not need to use the beta parameter at all, unless you are just curious about comparing the effects of Nuttall vs. Kaiser windows.
This is the default effect if the two files have different sampling rates. Default parameters are, as indicated above, Kaiser window of length 45, rolloff 0.80, beta 16, linear interpolation.
NOTE: -qs is only slightly slower, but more accurate for 16-bit or higher precision.
NOTE: In many cases of up-sampling, no interpolation is needed, as exact filter coefficients can be computed in a reasonable amount of space. To be precise, this is done when
input_rate < output_rate
output_rate/gcd(input_rate,output_rate) <= 511
- reverb gain-out reverbe-time delay [ delay ... ]
- Add reverberation to a sound sample. Each delay is given in milliseconds and its feedback is depending on the reverb-time in milliseconds. Each delay should be in the range of half to quarter of reverb-time to get a realistic reverberation. Gain-out is the volume of the output.
- reverse
- Reverse the sound sample completely. Included for finding Satanic subliminals.
- silence above_periods [ duration threshold[ d | % ]
[ below_periods duration
- [ d | % ]]
Removes silence from the beginning, middle, or end of a sound file. Silence is anything below a specified threshold.
The above_periods value is used to indicate if sound should be trimmed at the beginning of the audio file. A value of zero indicates no silence should be trimmed from the beginning. When specifing an non-zero above_periods, it trims audio up until it finds non-silence. Normally, when trimming silence from beginning of audio the above_periods will be 1 but it can be increased to higher values to trim all data up to a specific count of non-silence periods. For example, if you had an audio file with two songs that each contained 2 seconds of silence before the song, you could specify an above_period of 2 to strip out both silence periods and the first song.
When above_periods is non-zero, you must also specify a duration and threshold. Duration indications the amount of time that non-silence must be detected before it stops trimming data. By increasing the duration, burst of noise can be treated as silence and trimmed off.
Threshold is used to indicate what sample value you should treat as silence. For digital audio, a value of 0 may be fine but for audio recorded from analog, you may wish to increase ths value to account for background noise.
When optionally trimming silence from the end of a sound file, you specify a below_periods count. In this case, below_period means to remove all audio data after silence is detected. Normally, this will be a value 1 of but it can be increased to skip over periods of silence that are wanted. For example, if you have a song with 2 seconds of silence in the middle and 2 second at the end, you could set below_period to a value of 2 to skip over the silence in the middle of the audio file.
For below_periods, duration specifies a period of silence that must exist before data is not copied any more. By specifying a higher duration, silence that is wanted can be left in the audio. For example, if you have a song with an expected 1 second of silence in the middle and 2 seconds of silence at the end, a duration of 2 seconds could be used to skip over the middle silence.
Unfortunetly, you must know the length of the silence at the end of your audio file to trim off silence reliably. A work around is to use the silence effect in combination with the reverse effect. By first reversing the audio, you can use the above_periods to reliably trim all audio from what looks like the front of the file. Then reverse the file again to get back to normal.
To remove silence from the middle of a file, specify a below_periods that is negative. This value is then treated as a positive value and is also used to indicate the effect should restart processing as specified by the above_periods, making it suitable for removing periods of silence in the middle of the sound file.
The period counts are in units of samples. Duration counts may be in the format of hh:mm:ss.frac, or the exact count of samples. Threshold numbers may be suffixed iwth d, or % to indicate the value is in decibels or a percentage of maximum value of the sample value (0% specifies pure digital silence).
- speed [ -c ] factor
- Speed up or down the sound, as a magnetic tape with a speed control. It affects both pitch and time. A factor of 1.0 means no change, and is the default. 2.0 doubles speed, thus time length is cut by a half and pitch is one octave higher. 0.5 halves speed thus time length doubles and pitch is one octave lower. If the optional -c parameter is used then the factor is specified in "cents".
- stat [-s n ] [-rms ] [ -v ] [ -d ]
- Do a statistical check on the input file, and print results on the standard error file. Audio data is passed unmodified from input to output file unless used along with the -eoption.
The "Volume Adjustment:" field in the statistics gives you the argument to the -v number which will make the sample as loud as possible without clipping.
The option -v will print out the "Volume Adjustment:" field's value only and return. This could be of use in scripts to auto convert the volume.
The -s n option is used to scale the input data by a given factor. The default value of n is the max value of a signed long variable (0x7fffffff). Internal effects always work with signed long PCM data and so the value should relate to this fact.
The -rms option will convert all output average values to root mean square format.
There is also an optional parameter -d that will print out a hex dump of the sound file from the internal buffer that is in 32-bit signed PCM data. This is mainly only of use in tracking down endian problems that creep in to SoX on cross-platform versions.
- stretch factor [window fade shift fading]
- Time stretch file by a given factor. Change duration without affecting the pitch.factor of stretching: >1.0 lengthen, <1.0 shorten duration. window size is in ms. Default is 20ms. The fade option, can be "lin". shift ratio, in [0.0 1.0]. Default depends on stretch factor. 1.0 to shorten, 0.8 to lengthen. The fading ratio, in [0.0 0.5]. The amount of a fade's default depends on factor and shift.
- swap [ 1 2 | 1 2 3 4 ]
- Swap channels in multi-channel sound files. Optionally, you may specify the channel order you would like the output in. This defaults to output channel 2 and then 1 for stereo and 2, 1, 4, 3 for quad-channels. An interesting feature is that you may duplicate a given channel by overwriting another. This is done by repeating an output channel on the command line. For example, swap 2 2 will overwrite channel 1 with channel 2's data; creating a stereo file with both channels containing the same audio data.
- synth [length ] type mix [ freq [ -freq2 ]
[ off ] [ ph ] [ p1 ] [ p2 ] [ p3 ]
The synth effect will generate various types of audio data. Although
this effect is used to generate audio data, an input file must be specified.
The length of the input audio file determines the length of the output
audio file.
<length> length in sec or hh:mm:ss.frac, 0=inputlength, default=0
<type> is sine, square, triangle, sawtooth, trapetz, exp, whitenoise, pinknoise, brownnoise, default=sine
<mix> is create, mix, amod, default=create
<freq> frequency at beginning in Hz, not used for noise..
<freq2> frequency at end in Hz, not used for noise.. <freq/2> can be given as %%n, where 'n' is the number of half notes in respect to A (440Hz)
<off> Bias (DC-offset) of signal in percent, default=0
<ph> phase shift 0..100 shift phase 0..2*Pi, not used for noise..
<p1> square: Ton/Toff, triangle+trapetz: rising slope time (0..100)
<p2> trapetz: ON time (0..100)
<p3> trapetz: falling slope position (0..100)
- trim start [ length ]
Trim can trim off unwanted audio data from the beginning and end of the
audio file. Audio samples are not sent to the output stream until
the start location is reached.
The optional length parameter tells the number of samples to output after the start sample and is used to trim off the back side of the audio data. Using a value of 0 for the start parameter will allow trimming off the back side only.
Both options can be specified using either an amount of time and an exact count of samples. The format for specifying lengths in time is hh:mm:ss.frac. A start value of 1:30.5 will not start until 1 minute, thirty and 1/2 seconds into the audio data. The format for specifying sample counts is the number of samples with the letter 's' appended to it. A value of 8000s will wait until 8000 samples are read before starting to process audio data.
- vibro speed [ depth ]
- Add the world-famous Fender Vibro-Champ sound effect to a sound sample by using a sine wave as the volume knob. Speedgives the Hertz value of the wave. This must be under 30. Depth gives the amount the volume is cut into by the sine wave, ranging 0.0 to 1.0 and defaulting to 0.5.
- vol gain [ type [ limitergain ] ]
The vol effect is much like the command line option -v. It allows you to
adjust the volume of an input file and allows you to specify the adjustment
in relation to amplitude, power, or dB. If type is not specified then
it defaults to amplitude.
When type is amplitude then a linear change of the amplitude is performed based on the gain. Therefore, a value of 1.0 will keep the volume the same, 0.0 to < 1.0 will cause the volume to decrease and values of > 1.0 will cause the volume to increase. Beware of clipping audio data when the gain is greater then 1.0. A negative value performs the same adjustment while also changing the phase.
When type is power then a value of 1.0 also means no change in volume.
When type is dB the amplitude is changed logarithmically. 0.0 is constant while +6 doubles the amplitude.
An optional limitergain value can be specified and should be a value much less then 1.0 (ie 0.05 or 0.02) and is used only on peaks to prevent clipping. Not specifying this parameter will cause no limiter to be used. In verbose mode, this effect will display the percentage of audio data that needed to be limited.
BUGSThe syntax is horrific. Thats the breaks when trying to handle all things from the command line. Please report any bugs found in this version of SoX to Chris Bagwell (firstname.lastname@example.com)
SEE ALSOplay(1), rec(1), soxexam(1)
NOTICESThe version of SoX that accompanies this manual page is support by Chris Bagwell (email@example.com). Please refer any questions regarding it to this address. You may obtain the latest version at the the web site http://sox.sourceforge.net/
AUTHORChris Bagwell (firstname.lastname@example.com). Updates by Anonymous
Random Man Pages:


text:
If you need a child object to create a parent, and can use a singleton, just lazily initialize it. The first pass will loop around a bit, but every pass after that, you're already done.
I don't know why you have the $b in there. Does every parent object wrap a child object?
The parent merely needs to use (and then discard) the child object. To make things a little more concrete, the Parent class (not it's real name) is there to provide both a reasonable object constructor for children and AUTOLOAD object properties accessor methods. For public methods, this class has a series of methods that return DBI handles, so that DB credentials are centralized in one place.
Child classes provide
might not return an object of type Child, then you're asking the wrong class. You just need a
method, which has a lazy initialization for creating a specific Parent handle to help decide which Child_nnn class to create.
There. That sorts it out. I'd be very confused if calling
returned a Child2 instead of a Child. It doesn't make sense from an OO perspective.


text:
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text:
Main drive for 2012 iMac?
I asked about another SSD raid setup but think this sounds even better. Could this be my main drive for iMac?
I was disappointed to read its noisy though I'm sure it's not that bad. Mac peeps prolly have higher standards. One possible use case for me if possible would be with windows and the new mech warrior game if that might work? The speed boost might really make my stock iMac a capable game machine for even new titles (it's easily handled the few older Mac titles I've experimented with.
But mainly my reason is speed without compromising my iMac's style. My two previous computers are a Linux box and MacBook Air both with 128GB SSDs and I find it hard to go back from near instant booting and app loading they give me to the retro waiting experienced with old timey spinning platters.
- Asked by Anthony T from San Francisco
- Asked about: LaCie SATA III SSD Thunderbolt Litt…
2 Answers from the Community
Alexander thanks for your reply.
Funny but I did decide on the Mercury Helios/Accelsior and it is very fast. It's not as fast as I was imagining (for $800) and I think that has to do with latency with Thunderbolt as I think you were suggesting- sustained reads and rights get a blazing 700 mbps so yes the target case is someone who has to do fast transfers and this is certainly blazing for that.
So that's 10 times faster than the stock HDD which gets in the 60s but Application loading and booting are definitely not 10 times faster but probably closer to 4 times faster (time to login screen went from one minute to fifteen seconds). I do agree now that the internal one would likely be faster but still it's a $1300 option.
The Mercury Helios is noisy...not super noisy by any means (I would have said really quiet before I owned my iMac) but when you're talking about an addition to a whisper quiet iMac it may be a concern. I got a 2 m thunderbolt cable so I may see about putting it a bit away off my desk. I'm also going to look into replacing the stock Helios fan with a super quiet fan.
- Answered by Anthony T from Cary
This can work as a boot drive on an iMac although I would not recommend it. It is very fast, particularly for larger files. However, application loading is sketchy when used as a boot drive. Sometimes fast, sometimes slow. Perhaps partially due to thunderbolt latency and partially due to this drive being designed for creative pros needing to stream large data files on the fly from something like the Blackmagic Cinema Camera. It is indeed a very loud drive. I am not picky at all but this thing, trust me, is very loud. For whatever reason, the LaCie LittleBig SSDs are noisy and continuously run(or at least sound like it). They generate a great deal of heat as well which is normal but scary if you're not prepared for it. I had to purchase a 10 foot Thunderbolt cable to hide this little guy behind my couch before replacing it with a 5Big. So it's unlikely that this will sit next to your iMac and add to its style. I returned the LittleBig and now a giant LaCie 5big 20TB sits next to my iMac. It is virtually silent which is odd considering it uses mechanical drives. LaCie makes a 10TB version as well which is only $400 dollars more than their 512GB SSD drive and quite a bit faster for video work. Overall, this an expensive drive that's really supposed to be a portable drive for creative professionals on a low-end Hollywood set or out in the field. For a main boot drive, I would ask "why not just purchase a 2012 iMac with a 768 GB internal SSD?" Although the specs for the 768GB flash drive aren't as fast on paper as the LaCie, the internal flash drive will be much faster overall as a main drive since it's specifically designed by Apple to work in this regard. The best external boot drive solution for an iMac that I've seen is to thunderbolt a Mercury Helios + Mercury Accelsior PCIe SSD. This will get you read/write speeds of 750MB+/s and is designed to be Mac Bootable
- Answered by Alexander H from San Jose