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eat slowly,
and to lie about your age.
~ Lucille Ball
"Oh, grow up!" he often says to me when I've said or done something particularly ridiculous, a smirk dancing upon his lips, a glint of humor in his eyes. "Would you really want me to?" is my usual rejoinder. We revel in our youthful silliness and utter disregard for the rules of behavior that most seem so urgently ready to apply to folks of our age. Another birthday has rolled around and I am now squarely centered in that "woman of a certain age" category. I look in the mirror and see the lines on my skin and the silver threaded through my dark hair, I feel the weight of the years upon my shoulders, pulling me down with unforgiving severity, gravity giving me a less-than-youthful appearance. These old bones creak and the back has a tendency to slouch, the elevator has taken precedence over the stairs and fabric seems to strain at snaps and buttons. But for all of the outward changes, that slow but inevitable metamorphoses that we each go through, the visible traces left by the advancing years, I somehow feel an inward subtle shift in the opposite direction.
So the "Oh, grow up!" followed immediately by the "Would you really want me to?" is a game we play, just more childish banter between two who simply do not feel that the years have made us grow old. We laugh in the face of Old Man Time and hold onto youth joyfully, in an ironclad grip.
What a crime to waste it on children.
~ George Bernard Shaw
The body is some strange foreign vessel, almost alien in its outlandishness. There is an odd disconnect throughout our youth and well into adulthood, this relationship we have with our outer shell, as if wearing someone else's ill-fitting clothing. As a child, we often have moments of not quite being able to control our movements nor do we quite understand the changes that happen seemingly overnight as we sleep; as a teen, there is discomfort and embarrassment in every lump and bump, every growth spurt and unruly, out-of-control development. There may be a brief moment when we achieve the perfect balance, when we reach some ideal age, that place in time where it all comes together effortlessly, without blemish, pure and sublime, our hair, our skin, our figure; we glance in the mirror and smile, content, self-confident, at ease and at peace with ourselves. "Ah, I have finally grown up and grown into the person I was meant to be all along!" we exclaim, nodding in approval as we turn to blow out the 30 or so candles. But the moment is fleeting; it rushes by, a whisper blown swiftly away on the wind. We wake up shortly after, minutes it seems, and the walls begin to crumble; the skin sags, ever so imperceptibly at first, but we notice it a bit more every day; the first gray hair sneaks in, almost as a fine joke; the knees creak and crack as we climb the stairs to the apartment and it seems just that much more difficult to push ourselves out of bed in the morning. We catch a glimpse of our face, our body as we walk in front of a mirror or plate glass window and are stunned, wondering when it was that we grew so old.
growing up is optional.
~ Chili Davis
Yet, although I reached my stride quite a number of years ago, my peak physical years have come and gone, and today, well, the lines are getting fuzzy, the streaks of sophisticated silver run their fingers brazenly through my unruly hair and keeping in shape takes more effort every day, my inner child is well and alive, thank you very much. Rebellious in nature, the youthful me bursts at the seams, a ball of energy, not willing to sit still and twiddle her thumbs allowing any old rather snooty Grande Dame to make the decisions. Some may say that there is something irreverent in the way I behave, that silliness does not become a woman of a certain age; others may shake their head in dismay at my adamant determination to simply not grow up, their eyes opened wide in disbelief at my jokes or antics. But although I have little control over the outer shell other than exercise, diet and a good haircut, a touch of makeup and the choice of what I wear, my spirit is my own to do with as I please.
If you don't mind, it doesn't matter.
~ Leroy "Satchel" Paige
Yes, another birthday has come and gone with much hurrah and I was spoiled and pampered by my men in their usual, quiet way. And as is the tradition every single year, I baked my own cake. French pastry shops are abundant in tarts of glistening fruit, creams of chocolate, raspberry or vanilla studded with poached pears or bright berries and crunchy with praline or biscuits, elegant verrines of layer upon delicate layer of mousses and bavaroises topped by froths of whipped Chantilly; one jaw-dropping gorgeous, ravishingly delicious delight after another, it is true, but a birthday is simply not a birthday without a layer cake. And there is no better way to have exactly what you love best than making it yourself. I toyed with the idea of repeating last year's wildly successful Espresso Chocolate Cake with Mocha Mascarpone Frosting, as it had indeed been one of the best cakes I have ever tasted. And although I had finally settled on the same flavor combination – a favorite – I turned instead to my favorite chocolate cake recipe, one that was handed down from my father, and my simple chocolate buttercream frosting. Yet I twisted and turned and added espresso to both the cake batter and the frosting, whisked in a container of fresh mascarpone to the buttercream for a richer, smoother, creamier frosting and voilà I created my perfect birthday cake!
A childhood delight to bring out the youthful frivolity, the joy and delight in each of us; dense, ultra moist, devilishly chocolaty layers with a diabolically inspired kiss of espresso, a cake at once flirtatious with its voluptuous swirls of mocha cream and serious in its sinful decadence. And what a cake! A flash to whip up and bring together, and oh so easy going down. Kid friendly indeed yet oh so incredibly adult.
And a perfect romantic dessert for St. Valentine's Day.
Makes a 8 ½ or 9-inch two layer cake or an 7-inch three layer cake.
1 ¾ cup flour
2 cups sugar
¾ cup unsweetened cocoa powder
1 ½ tsp baking powder
1 ½ tsp baking soda
1 tsp salt
2 large eggs
1 cup whole milk
½ cup vegetable oil
2 tsps vanilla
1 cup prepared coffee *
* If you prefer, the coffee can be replaced with water or a mixture of water and fruit juice.
Preheat oven to 350°F (180°C). Oil and flour two 8 ½ or 9-inch round cake pans or three 7-inch cake pans generously. (I oiled the pans, lined with parchment and then lightly oiled the paper and dusted with flour.)
Combine all of the dry ingredients in a large mixing bowl. Whisk or whiz them with the electric mixer on low speed for 30 seconds until everything is well combined. Add the eggs, milk, oil and vanilla. Beat on low until well blended then increase the mixer speed to medium and beat for about 2 minutes. Bring the 1 cup of coffee just to the boil and stir in carefully by hand until very well blended. Carefully divide the batter between the two prepared cake pans – it will be liquid. (If you want to make the smaller 3-layer cake and only have 2 cake pans: oil, line and flour the two pans and divide 2/3 of the batter between the two; the pans should be filled about 1/3 to ½ full. Bake the first two layers. When they are done, remove from the oven, allow to cool for several minutes, slide a sharp knife around the edges to loosen and invert (then upright) on cooling racks to completely cool. Clean, oil, line and dust with flour one of the pans and pour the remaining third of the batter into this pan and bake as directed.)
Bake in the preheated oven for 35 – 40 minutes or until the center is set (30 – 35 minutes for the smaller layers). Remove from oven and allow to cool for 10 – 15 minutes on cooling racks before turning them out onto the racks to cool completely.
11 - 12 oz (325 - 350 grams) powdered/confectioner's sugar
8 Tbs (120 grams) unsalted butter, softened to room temperature
1.8 oz (50 grams) unsweetened cocoa powder
4 Tbs very hot prepared coffee
3.5 – 5.3 oz (100 – 150 g) fresh mascarpone cheese
Using an electric hand mixer, cream the butter and the powdered sugar together. Add the cocoa powder and the hot coffee and beat, scraping down the sides as necessary, until well blended and fluffy. Beat in as much mascarpone as desired until smooth and whipped.
Chill in the refrigerator until firm enough so that, when spread and the layers are stacked, the frosting does not slide.
Frost the tops of the layers then stack, placing the bottom layer on a cake or serving plate. I slip strips of waxed paper or parchment under the edges of the cake before frosting the sides in order to keep the plate clean and frosting-free. Smooth the frosting on the sides of the cake. Pipe rosettes of frosting and decorate as desired. Gently slide the strips of parchment out from under the cake and retouch as needed. Chill in the refrigerator until the frosting has firmed. Because the frosting contains mascarpone, it is best to store uneaten cake in the refrigerator.


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Find hotels & motels in Chesapeake Bay using the list below. Find cheap and discounted hotel/motel rates in or nearby Chesapeake Bay, VA for your corporate or personal leisure travel. Discover the most favored Chesapeake Bay Virginia hotel rooms. Review the Chesapeake Bay hotel list below to find the perfect lodging. We also have a map of the Chesapeake Bay hotels.The Chesapeake Bay center is approximately 14 miles from Kilmarnock Museum and 19 miles from Thousand Trails Chesapeake Bay.
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subset: commoncrawl


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Explains how to build eye-catching banner ads in Flash that achieve design goals and satisfy the requirements of search engines and the sites where the ads will display.
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MILWAUKEE---The Chicago Wolves pulled within one halfway through the third period but could not find the equalizer and dropped a 4-2 decision to the Milwaukee Admirals in an Amtrak Rivalry matchup at the Bradley Center Friday night.
Forwards Darren Haydar and Jordan Schroeder each scored power-play goals for Chicago (0-2-0-1).
Milwaukee (2-0-0-0) opened scoring 2:54 into the game with its first of three power-play goals. Center Chris Mueller held the puck below the goal line and slid the puck to right wing Juuso Puustinen in the far circle who ripped a shot past Wolves goaltender Eddie Lack, top-shelf.
The Admirals doubled their lead to 2-0 when center Kyle Wilson's shot was stopped by Lack but the puck slipped through his pads, and right wing Ryan Thang pounced on the loose puck and pushed it into the empty net at 13:01 of the first period.
Haydar got Chicago on the board 1:56 into the second stanza when his pass attempt from the far circle was deflected but the puck bounced right back to his stick and he flicked a quick shot above the shoulder of Milwaukee netminder Jeremy Smith.
Milwaukee regained its two-goal cushion at 4:40 of the third period when a scramble ensued in front of Lack and Mueller poked the loose puck over the line.
Lack (0-1-1) recorded 31 saves on 34 shots.
Chicago cut its deficit to one with a power-play goal at 12:06 of the final frame. Schroeder skated into the Admirals zone and fired a shot from the far boards that beat Smith, five-hole.
Smith (2-0-0) kicked out 21 of 23 shots.
Forward Taylor Beck sealed the win for Milwaukee with an empty net marker with 54 seconds remaining in the game.
The Wolves host Milwaukee in the franchise's 18th home opener, presented by LifeSource, at the Allstate Arena Saturday at 7 p.m.
Chicago 0 1 1-- 2
Milwaukee 2 0 2-- 4
First Period---1, Milwaukee, Puustinen 2 (Mueller, Wilson), 2:54 pp; 2, Milwaukee, Thang 3 (Wilson, Mueller), 13:01 pp. Penalties---Rodin, Chicago (hooking), 2:06; Friesen, Chicago (tripping), 12:27.
Second Period---3, Chicago, Haydar 1 (Doell, Mancari), 1:56 pp. Penalties---Smith, Milwaukee (tripping), 1:34; Valentine, Milwaukee (checking to the head), 16:35.
Third Period---4, Milwaukee, Mueller 1 (Thang, Ellis), 4:40 pp; 5, Chicago, Schroeder 1 (Mancari), 12:06 pp; 6, Milwaukee, Beck 1 (Mueller, Ellis), 19:06 en. Penalties---Davies, Chicago (tripping), 3:44; Valentine, Milwaukee (boarding), 10:14.
Shots on goal---Chicago: 8-7-8--23. Milwaukee: 15-8-12--35. Power plays---Chicago: 2-3. Milwaukee: 3-3. Goalies--Chicago, Lack (31-34). Milwaukee, Smith (21-23). A---5,145. Referee---Francis Charron. Linesmen---Aaron Mills and Mathieu Chenier.


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TAMPA, Fla. --- University of Tennessee at Chattanooga freshman Maddison Melchionna (Afton, Tenn.) claimed the top spot in the USF Invitational women's 800 meter run in her first collegiate track meet while senior Nathan Wanuch (Mississauga, Ontario) finished second in the men's 3000 meter Steeplechase.
Melchionna finished the race with a time of 2:13.75 followed by classmate Jessica York (Normandy, Tenn.) who placed third with a time of 2:17.34 edging out sophomore Rebecca Greenwall (Bogart, Ga.) who was fourth with a time of 2:17.48.
"I expected Maddison to be good," UTC head coach Bill Gautier said. "She redshirted the indoor season, so I'm very happy for her. Jessica improved on her indoor time and Rebecca did a great job as always."
In the men's 3000 meter Steeplechase, Wanuch was second with a time of 9:30.89 in the season opener.
"Nathan ran really well," Gautier said. "The weather was just right for him and this was a good outdoor meet."
The weather, while good for running, was not just right in the men's triple jump. James Stovall scratched his two good jumps and finished sixth with a distance of 44-10.75
"When you're indoors, the conditions are perfect for the triple jump. But there was a bit of wind today that made it difficult," Gautier said. "He kept at it and did a really good job.
"The men's 1500 was a good race too. Mark [Cole] did a good job as always and Paul [Stuart] and John [Gilpin] ran a good race after sitting out the indoor season."
In the men's 1500 meter run, senior Mark Cole (McKenzie, Tenn.) finished second in his heat with a time of 4:06.14 to place sixth overall. Senior John Gilpin (Nashville, Tenn.) finished third in the same heat with a time of 4:08.02 for 11th overall. Sophomore Paul Stuart (Brentwood, Tenn.) ran the fast heat and finished ninth overall with a time of 4:07.51. Sophomore Marquis Carter (Memphis, Tenn.) who normally runs the shorter races, finished 18th overall in the 1500m with a time of 4:10.21.
Sophomore Iman Isang (Chattanooga, Tenn.) finished ninth in the men's 110 meter Hurdles with a time of 15.68 and ran a PR in the 100 meter Dash with a time of 11.40, besting his previous time of 11.53 set last season at the Western Carolina Catamount Classic.
"Today was a really good start to the season," Gautier said. "A lot of people got in some good races and hopefully we'll be able to build on this next week."
The Mocs are off to the Alabama Relays next weekend at Tuscaloosa, Ala.


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Wind: 0 mph
Former Harwood paraprofessional Denise Fuoco is appealing the school board's ruling to uphold the decision of Washington West Supervisory Union Superintendent Brigid Scheffert to terminate Fuoco after she failed to prove her "Highly Qualified Professional" status by passing a federally required exam.
The Harwood Union School Board's unanimous vote to uphold Fuoco's
dismissal followed a grievance hearing held December 9 that included
Vermont NEA representative Stuart Russo-Savage and Carla Bell from the
Harwood Educators Association.
Under the terms of her contract, Fuoco has the right to seek mediation if the grievance wasn't resolved with the initial hearing; from there, she may choose to submit the grievance to final and binding arbitration.
Fuoco said that at a meeting earlier this week, NEA and union officers met and agreed "to move forward with the next step." The NEA has 30 days to file for mediation from Tuesday, January 4.
The decision to uphold Fuoco's termination followed an executive session December 9 and public comment from several concerned parents of special needs students, Harwood faculty and staff who all spoke to Fuoco's skill and high level of professionalism.
Fuoco was fired after she failed to take a federally required standardized test to earn the "HQP" designation, a task made difficult in her case due to her dyslexia.
She has worked in the Washington West Supervisory Union (WWSU) for the past several years including seven years spent working at the Moretown Elementary School prior to joining Harwood.
While at Moretown, Fuoco was allowed the accommodation of submitting a portfolio of her work in lieu of taking the written test; she filed a grievance with the Harwood School Board after she was told that the accommodation would no longer be afforded to her and that she would need to take and pass the federal exam.
Under section 1119(c) through 1119(h) of the No Child Left Behind Act (NCLB) the federal requirements for Title I, part A, "instructional paraprofessionals" to achieve HQP status, paraeducators must have a high school diploma or its recognized equivalent.
In addition, they must have completed at least two years of study (defined as a minimum of 48 credit hours) at an institution of higher education or obtained an associate's degree (or higher) or meet the standard of quality and demonstrate it through a formal state or local assessment.
According to the Vermont Department of Education, "The latter option (II-C) may be fulfilled through standardized assessments such as Parapro or Paraeducator.net, or through a locally developed portfolio process LEAs may also choose to accept comparable HQP documentation from other school districts or states."
Sheffert said that the portfolio standard is significantly higher and would have to be implemented for all paraprofessionals in WWSU. In addition, she said upholding the state and federal regulations is essential to maintain funding directly tied to them.


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The road to recovery
Ronald Reagan once remarked, "To sit back hoping that someday, some way, someone will make things right is to go on feeding the crocodile, hoping he will eat you last – but eat you he will." There are many memorable quotes associated with business, or the economy, or the recession. And, even though at first […]
This land is our land
"And on the eighth day, God looked down on his planned paradise and said, "I need a caretaker." So God made a farmer. God said, "I need somebody willing to get up before dawn, milk cows, work all day in the field, milk cows again, eat supper, then go to town and stay past midnight […]
Watching wind grow in rural Illinois
Wind energy development in the United States is rushing past recent growth records. In 2011, 6,819 megawatts of generating capacity were installed; in 2012, that figure jumped to more than 13,000 MW, according to the American Wind Energy Association. In total, the U.S. boasts more than 60,000 MW of wind power capacity. Since 1 MW […]


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StartUpHire is the place to find Marketing jobs in and near Derry. Review the list of Marketing Derry career opportunities to the left. These Marketing jobs represent some of the most rewarding careers available.
You can refine your job search for Marketing career opportunities in Derry by adding a job title, company name, or keyword to the search box above … or click the "Advanced Search" link for more options.
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A selective two year Fellowship at a fast-paced, service-oriented start-up that provides promising seniors and recent grad with significant work experiences, increasing responsibilities, and ongoing training and professional development opportunities.7/29/2015 Allston, MA Save This Job
The Sales Executive follows a full sales life cycle to deliver BIA products and services to new and returning clients in all focus areas, including established markets in the institutional investor community and new verticals being pursued by the company.7/30/2015 Boston, MA Save This Job
Bit9 + Carbon Black is currently seeking a dynamic and experienced Product Marketing Manager to help drive market awareness and adoption of our leading endpoint threat prevention, detection and response solutions. In this pivotal role you will work with s7/31/2015 Waltham, MA Save This Job
Are you an "A+" player who likes a challenge Are you passionate about your work Do you like free espresso Will you fight others for the last slice of pizza Do you want to work at a high growth SaaS company in Cambridge, MA, the cradle of high tech Join th7/29/2015 Cambridge, MA Save This Job
Bit9 + Carbon Black is currently seeking an experienced Community Marketing Manager to grow and manage our community marketing program. The Bit9 + Carbon Community is growing quickly as curity engineers and enthusiasts, incident responders and IT professionals from around the7/1/2015 Waltham, MA Save This Job
Latin America Field Marketing Manager in Burlington Massachusetts United States Danaher Company Description Arbor Networks is a leading provider of network security and monitoring solutions for global networks. Its customers include 70 percent of the world?s ISPs and large6/8/2015 Burlington, MA Save This Job
Influitive is a rapidly-growing software startup in the advocate marketing space. We?ve secured funding from top-tier venture capitalists and our customer include more than 200 of the world?s best and fastest-growing technology companies. We?re passionate about creating a new6/6/2015 Boston, MA Save This Job
Department:Marketing Location:Waltham, MA Click Here to Complete Your Application Senior Marketing/Lead Generation Manager ZoomInfo is looking for a Lead Gen rock star to join the marketing team as its Senior Marketing/Lead Generation Manager. You will work closely with7/20/2015 Waltham, MA Save This Job
Reports to: VP, Marketing Location: Waltham, MA Position Summary CallMiner is seeking an amazing data-driven program marketing manager to help drive lead generation through content, web, and other forms of inbound marketing. Customer engagement and speech analytics or the7/20/2015 Waltham, MA Save This Job
Product Marketing Waltham, Massachusetts Description The Sr. Manager/Director of Partner Marketing will lead a team of Partner Marketers creating and implementing strategies to drive success of the Sitefinity Partner Program. He or she will work closely with the Partner Program7/14/2015 Waltham, MA Save This Job
Director of Digital Marketing in Burlington Massachusetts United States Danaher Company Description Arbor Networks helps secure the world?s largest enterprise and service provider networks from DDoS attacks and advanced threats. Arbor is the world?s leading provider of DDoS7/8/2015 Burlington, MA Save This Job
Boston, MA or Providence, RI Full Time Marketing Manager/Supervisor SHARE Help shape one of the New England's most innovative companies. Develop innovative, eye-catching product marketing content and campaigns to best position ShapeUp, increase awareness, and solidify7/2/2015 Boston, MA Save This Job
Wakefield, MAFull-time Company Description About Certica Solutions www.CerticaSolutions.com Certica Solutions provides data management and education content solutions which enable K-12 educators and administrators to assess, measure and improve instructional and organizational6/29/2015 Wakefield, MA Save This Job
Burlington, MA The Marketing Programs Manager will be instrumental in creating, executing and managing marketing programs to ensure Black Duck reaches its business goals. The candidate will be required to globally support integrated campaigns, events, Webinars, seminars,6/22/2015 Burlington, MA Save This Job
Job Description Job Description The PatientsLikeMe Marketing team is responsible for creating the strategy, positioning and execution plan for the marketing of PatientsLikeMe to both consumers and industry partners. As the Senior Marketing Manager, you will execute marketing6/15/2015 Cambridge, MA Save This Job
Location:Maynard, MA Job Code:TBD # of openings:1 DESCRIPTION Key Job Responsibilities Collaborate with sales stakeholders to build an integrated marketing and communications plan to support world-wide business to increase consumer engagement and ultimately drive sales Develop6/8/2015 Maynard, MA Save This Job
Core Security is seeking a motivated and energetic Marketing Intern to join our top-notch team. This paid internship is an ideal opportunity to gain valuable real-world experience at a successful, dynamic organization that is the independent leader in its field of security6/1/2015 Boston, MA Save This Job
CyberArk is seeking a senior business and marketing leader to lead the digital/web marketing team. This newly created position will be responsible for driving our digital strategy and web marketing efforts, including our corporate website, organic and paid search, display,5/31/2015 Newton, MA Save This Job
Job Function:Marketing Department:Marketing Location:Boston, MA Date Posted:6/23/2015 More Sharing ServicesShare Share on facebookShare on linkedinShare on googleShare on twitterEmail this job Marketing Specialist - Lux Research Inc. - Downtown Boston (Greater Boston Area) Lux7/14/2015 Boston, MA Save This Job
Job Function:Marketing Department:Marketing Location:Boston, MA Date Posted:6/23/2015 More Sharing ServicesShare Share on facebookShare on linkedinShare on googleShare on twitterEmail this job Marketing Specialist - Lux Research Inc. - Downtown Boston (Greater Boston Area) Lux7/14/2015 Boston, MA Save This Job


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Auditor David Irwin of the firm of Mauldin and Jenkins presented the audit report to board members at their March 21 meeting. He began by stating the county had a clean audit.
"That is what you want," he said.
Irwin added that the county's financial department is top-notch, and LaVita Crutchfield and staff are a joy to work with.
"They are receptive to change, and we don't get that a lot," he commented.
The auditor gave a quick overview of accounts and said the General Fund balance is strong. He said the purpose of audit recommendations to the county are to make sure everything is property reported.
Irwin noted one issue is segregation of duties for financial accounts. He explained the staff members reconciling the accounts should not be the same people who maintain them.
"You need different people reporting and balancing statements. This is a common occurrence in small government offices," he stated.
For the full story, pick up a copy of this week's newspaper or subscribe to our e-Edition at http://ee.jcnews.com.


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|Here is an email from a very satisfied customer who purchased the AMC-400 back in 2006. His name is Bob G and apparently his AMC-400 is holding up great! Here's what he wrote:Hi
My name is Bob G out here in Northern California, I bought my AMC-400 motorcycle carrier waaaaaay back in September 2006. I ride almost every weekend and I have hauled many dirt bikes on this thing and it still, to this day works just as good as the day I bought it. It's a awesome product and what you get for the price is un-touchable. I have sent the link to this site to many people asking about bike carriers on my favorite riding sites like Thumpertalk.com
Thank you for the awesome product
Bob G
Thanks for the great feedback Bob!
|Click here for information about the AMC-400 Hitch Mounted Aluminum Motorcycle Carrier.
|Click here for information about hitch mount motorcycle carriers.
|Thanks for reading! Jay Tamez – Discount Ramps LLC


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An encoder transforms at least a portion of a signal, counts the resulting transform coefficients having a zero value, and encodes the signal with the zero count. A decoder decodes the signal in order to recover the zero count. The decoder may also determine its own zero count of the signal as received and may compare the zero count that it determines to the recovered zero count. The decoder may be arranged to detect compression/decompression based upon results from the comparison, and/or the decoder may be arranged to prevent use of a device based upon results from the comparison.
1. A method of processing a signal comprising:
receiving the signal at a receiver to produce a received signal;
decoding the received signal so as to read a coefficient value count code from the received signal;
performing a transform of the received signal to produce transform coefficients;
counting those transform coefficients having a predetermined value;
comparing the coefficient value count contained in the coefficient value count code to the transform coefficient count; and
selectively further processing or discarding the received signal based upon the comparison of the coefficient value count contained in the coefficient value count code to the transform coefficient count.
2. The method of
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9. The method of
10. A decoder having an input and an output, wherein the input receives a signal, wherein the decoder:
decodes the input signal so as to read a coefficient value count code from the input signal;
transforms the input signal to produce transform coefficients;
counts those transform coefficients having a predetermined value;
compares the coefficient value count contained in the coefficient value count code to the transform coefficient count; and
selectively further processes or discards the input signal based upon the comparison of the coefficient value count contained in the coefficient value count code to the transform coefficient count.
11. The decoder of
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13. The decoder of
14. The decoder of
15. The encoder of
16. The decoder of
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24. The decoder of
This is a Divisional of U.S. application Ser. No. 09/616,116 filed Jul. 14, 2000 now U.S. Pat. No. 6,879,652. This application contains disclosure similar to the disclosures in U.S. patent application Ser. No. 09/116,397 filed Jul. 16, 1998, in U.S. patent application Ser. No. 09/427,970 filed Oct. 27, 1999, in U.S. patent application Ser. No. 09/428,425 filed Oct. 27, 1999, in U.S. patent application Ser. No. 09/543,480 filed Apr. 6, 2000, and in U.S. patent application Ser. No. 09/553,776 filed Apr. 21, 2000.
The present invention relates to the detection of signals, such as audio streams, which have been modified.
Video and/or audio received by video and/or audio receivers have been monitored for a variety of reasons. For example, the transmission of copyrighted video and/or audio is monitored in order to assess appropriate royalties. Other examples include monitoring to determine whether a receiver is authorized to receive the video and/or audio, and to determine the sources and/or identities of video and/or audio.
One approach to monitoring video and/or audio is to add ancillary codes to the video and/or audio at the time of transmission or recording and to detect and decode the ancillary codes at the time of receipt by a receiver or at the time of performance. There are many arrangements for adding an ancillary code to video and/or audio in such a way that the added ancillary code is not noticed when the video is viewed on a monitor and/or when the audio is reproduced by speakers. For example, it is well known in television broadcasting to hide ancillary codes in non-viewable portions of video by inserting them into either the video's vertical blanking interval or horizontal retrace interval. One such system is referred to as "AMOL" and is taught in U.S. Pat. No. 4,025,851.
Other known video encoding systems have sought to bury the ancillary code in a portion of a video signal's transmission bandwidth that otherwise carries little signal energy. An example of such a system is disclosed by Dougherty in U.S. Pat. No. 5,629,739.
An advantage of adding an ancillary code to audio is that the ancillary code can be detected in connection with radio transmissions and with pre-recorded music, as well as in connection with television transmissions. Moreover, ancillary codes, which are added to audio signals, are reproduced in the audio signal output of a speaker and, therefore, offer the possibility of non-intrusive interception such as by use of a microphone. Thus, the reception and/or performance of audio can be monitored by the use of portable metering equipment.
One known audio encoding system is disclosed by Crosby, in U.S. Pat. No. 3,845,391. In this system, an ancillary code is inserted in a narrow frequency "notch" from which the original audio signal is deleted. The notch is made at a fixed predetermined frequency (e.g., 40 Hz). This approach led to ancillary codes that were audible when the original audio signal containing the ancillary code was of low intensity.
A series of improvements followed the Crosby patent. Thus, Howard, in U.S. Pat. No. 4,703,476, teaches the use of two separate notch frequencies for the mark and the space portions of a code signal. Kramer, in U.S. Pat. Nos. 4,931,871 and in 4,945,412 teaches, inter alia, using a code signal having an amplitude that tracks the amplitude of the audio signal to which the ancillary code is added.
Microphone-equipped audio monitoring devices that can pick up and store inaudible ancillary codes transmitted in an audio signal are also known. For example, Aijalla et al., in WO 94/11989 and in U.S. Pat. No. 5,579,124, describe an arrangement in which spread spectrum techniques are used to add an ancillary code to an audio signal so that the ancillary code is either not perceptible, or can be heard only as low level "static" noise. Also, Jensen et al., in U.S. Pat. No. 5,450,490, teach an arrangement for adding an ancillary code at a fixed set of frequencies and using one of two masking signals, where the choice of masking signal is made on the basis of a frequency analysis of the audio signal to which the ancillary code is to be added.
Moreover, Preuss et al., in U.S. Pat. No. 5,319,735, teach a multi-band audio encoding arrangement in which a spread spectrum ancillary code is inserted in recorded music at a fixed ratio to the input signal intensity (code-to-music ratio) that is preferably 19 dB. Lee et al., in U.S. Pat. No. 5,687,191, teach an audio coding arrangement suitable for use with digitized audio signals in which the code intensity is made to match the input signal by calculating a signal-to-mask ratio in each of several frequency bands and by then inserting the code at an intensity that is a predetermined ratio of the audio input in that band. As reported in this patent, Lee et al. have also described a method of embedding digital information in a digital waveform in pending U.S. application Ser. No. 08/524,132.
It will be recognized that, because ancillary codes are preferably inserted at low intensities in order to prevent the ancillary code from distracting a listener of program audio, such ancillary codes may be vulnerable to various signal processing operations. For example, although Lee et al. discuss digitized audio signals, it may be noted that many of the earlier known approaches to encoding an audio signal are not compatible with current and proposed digital audio standards, particularly those employing signal compression methods that may reduce the signal's dynamic range (and thereby delete a low level ancillary code) or that otherwise may damage an ancillary code. In many applications, it is particularly important for an ancillary code to survive compression and subsequent de-compression by such algorithms as the AC-3 algorithm or the algorithms recommended in the ISO/IEC 11172 MPEG standard, which is expected to be widely used in future digital television transmission and reception systems.
It must also be recognized that the widespread availability of devices to store and transmit copyright protected digital music and images has forced owners of such copyrighted materials to seek methods to prevent unauthorized copying, transmission, and storage of their material. Unlike the analog domain, where repeated copying of music and video stored on media, such as tapes, results in a degradation of quality, digital representations can be copied without any loss of quality. The main constraints preventing illegal reproductions of copyrighted digital material is the large storage capacity and transmission bandwidth required for performing these operations. However, data compression algorithms have made the reproduction of digital material possible.
Data compression is typically achieved by means of "lossy compression" algorithms. In this approach, the inability of the human ear to detect the presence of a low power frequency f1 when there is a neighboring high power frequency f2 is exploited to modify the number of bits used to represent each spectral value. Thus, while a two-channel or stereo digital audio stream in its original form may carry data at a rate of 1.5 megabits/second, a compressed version of this stream may have a data rate of 96 kilobits/second.
A popular compression technology known as MP3 can compress original audio stored as digital files by a factor of ten. When decompressed, the resulting digital audio is virtually indistinguishable from the original. From a single compressed MP3 file, any number of identical digital audio files can be created. Currently, portable devices that can store audio in the form of MP3 files and play these files after decompression are available.
In order to protect copyrighted material, digital code insertion techniques have been developed where ancillary codes are inserted into audio as well as video digital data streams. The inserted ancillary codes are used as digital signatures to uniquely identify a piece of music or an image. As discussed above, many methods for embedding such imperceptible ancillary codes in both audio and video data are currently available. While such ancillary codes provide proof of ownership, there still exists a need for the prevention of distribution of illegally reproduced versions of digital music and video.
In an effort to satisfy this need, it has been proposed to use two separate ancillary codes that are periodically embedded in an audio stream. For example, it is suggested that the ancillary codes be embedded in the audio stream at least once every 15 seconds. The first ancillary code is a "robust" ancillary code that is present in the audio even after it has been subjected to fairly severe compression and decompression. The second ancillary code is a "fragile" ancillary code that is also embedded in the original audio and that is erased during the compression/decompression operation.
The robust ancillary code contains a specific bit that, if set, instructs the software in a compliant player not set, to allow the music to be played without such a search. If the compliant player is instructed to search for the presence of the fragile ancillary code, and if the fragile ancillary code cannot be detected by the compliant player, the compliant player will not play the music.
Additional bits in the robust ancillary code also determine whether copies of the music can be made. In all, twelve bits of data constitute an exemplary robust ancillary code and are arranged in a specified bit structure.
A problem with the "fragile" ancillary code is that it is fragile and may be difficult to receive even when there is no unauthorized compression/decompression. Accordingly, an embodiment of the present invention is directed to a pair of robust ancillary codes useful in detecting unauthorized compression. The first ancillary code consists of a number (such as twelve) of bits conforming to a specified bit structure such as that discussed above, and the second ancillary code consists of a number (such as eight) of bits forming a descriptor that characterizes a part of the audio signal in which the ancillary codes are embedded. In a player designed to detect compression, both of the ancillary codes are extracted irrespective of whether or not the audio material has been subjected to a compression/decompression operation. The detector in the player independently computes a descriptor for the received audio and compares this computed descriptor to the embedded descriptor. Any difference that exceeds a threshold indicates unauthorized compression.
According to one aspect of the present invention, an encoder has an input and an output. The input receives a signal. The encoder calculates a zero count of at least a portion of the signal and encodes the signal with the calculated zero count. The output carries the encoded signal.
According to another aspect of the present invention, a decoder has an input and an output. The input receives a signal. The decoder decodes the received signal so as to read a zero count code from the signal, and the output carries a signal based upon the decoded zero count code.
According to still another aspect of the present invention, a method of encoding a signal comprises a) performing a transform of the signal to produce coefficients, b) counting those coefficients having a predetermined value; and, c) encoding the signal with the count.
According to yet another aspect of the present invention, a method of decoding a received signal comprises a) decoding the received signal so as to read a coefficient value count code from the received signal; b) performing a transform of the received signal to produce transform coefficients; c) counting those transform coefficients having a predetermined value; and, d) comparing the coefficient value count contained in the coefficient value count code to the transform coefficient count.
According to a further aspect of the present invention, an electrical signal contains a count code related to a count of coefficients resulting from a transform of at least a portion of the electrical signal.
These and other features and advantages will become more apparent from a detailed consideration of the invention when taken in conjunction with the drawings in which:
Audio signals are usually digitized at sampling rates that range between thirty-two kHz and forty-eight kHz. For example, a sampling rate of 44.1 kHz is commonly used during the digital recording of music. However, digital television ("DTV") is likely to use a forty eight kHz sampling rate. Besides the sampling rate, another parameter of interest in digitizing an audio signal is the number of binary bits used to represent the audio signal at each of the instants when it is sampled. This number of binary bits can vary, for example, between sixteen and twenty four bits per sample. The amplitude dynamic range resulting from using sixteen bits per sample of the audio signal is ninety-six dB. This decibel measure is the ratio between the square of the highest audio amplitude (216=65536) and the lowest audio amplitude (12=1). The dynamic range resulting from using twenty-four bits per sample is 144 dB. Raw audio, which is sampled at the 44.1 kHz rate and which is converted to a sixteen-bit per sample representation, results in a data rate of 705.6 kbits/s.
As discussed above, compression of audio signals is performed in order to reduce this data rate to a level which makes it possible to transmit a stereo pair of such data on a channel with a throughput as low as 192 kbits/s. This compression typically is accomplished by transform coding. Most compression algorithms are based on the well-known Modified Discrete Cosine Transform(MDCT). This transform is an orthogonal lapped transform that has the property of Time Domain Aliasing Cancellation (TDAC) and was first described by Princen and Bradley in 1986. [Princen J, Bradley A, Analysis/Synthesis Filter Bank Design Based on Time Domain Aliasing Cancellation, IEEE Transactions ASSP-34, No. 5, October 1986, pp 1153-1161]. For example, this transform may be performed on a sampled block of audio containing N samples with amplitudes x(k), where k=0,1, . . . N−1, using the following equation:
An inverse transform to reconstruct the original audio from the spectral coefficients resulting from equation (1) is performed in order to decompress the compressed audio.
In order to compute the transform given by equation (1), an audio block is constructed by combining N/2 "old" samples with N/2 "new" samples of audio. In a subsequent audio block, the "new" samples would become "old" samples and so on. Because the blocks overlap, this type of block processing prevents errors that may occur at the boundary between one block and the previous or subsequent block. There are several well known algorithms available to compute the MDCT efficiently. Most of these use the Fast Fourier Transform. [Gluth R, regular FFT-Related Transform Kernels for DCT/DST-based polyphase filter banks, ICASSP 91, pp 2205-8, Vol. 3.]
As a specific example, N may equal 1024 samples per overlapped block, where each block includes 512 "old" samples (i.e., samples from a previous block ) and 512 "new" or current samples. The spectral representation of such a block is divided into critical bands where each band comprises a group of several neighboring frequencies. The power in each of these bands can be calculated by summing the squares of the amplitudes of the frequency components within the band.
Compression algorithms such as MPEG-II Layer 3 (popularly known as MP3) and Dolby's AC-3 reduce the number of bits required to represent each spectral coefficient based on the psycho-acoustic properties of the human auditory system. In fact, several of these coefficients which fall below a given threshold are set to zero. This threshold, which typically represents either (i) the acoustic energy required at the masked frequency in order to make it audible or (ii) an energy change in the existing spectral value that would be perceptible, is usually referred to as the masking threshold and can be dynamically computed for each band. The present invention recognizes that normal uncompressed audio contains far fewer zero coefficients than a corresponding compressed/decompressed version of the same audio.
In order for the encoder 12 to embed a "robust" digital ancillary code in an audio data stream in a manner compatible with compression technology, the encoder 12 should preferably use frequencies and critical bands that match those used in compression. The block length NC of the audio signal that is used for coding may be chosen such that, for example, jNC=Nd=1024, where j is an integer. A suitable value for NC may be, for example, 512. As depicted by a step 40 of the flow chart shown in
The frequencies resulting from the Fourier Transform are indexed in the range −256 to +255, where an index of 255 corresponds to exactly half the sampling frequency fS. Therefore, for a forty-eight kHz sampling frequency, the highest index would correspond to a frequency of twenty-four kHz. Accordingly, for purposes of this indexing, the index closest to a particular frequency component fj resulting from the Fourier Transform ℑ{v(t)} is given by the following equation:
The code frequencies fi used for coding a block may be chosen from the Fourier Transform ℑ{v(t)} at a step 46 in a particular frequency range, such as the range of 4.8 kHz to 6 kHz which may be chosen to exploit the higher auditory threshold in this band. Also, each successive bit of the code may use a different pair of code frequencies f1 and f0 denoted by corresponding code frequency indexes I1 and I0. There are two exemplary ways of selecting the code frequencies f1 and f0 at the step 46 so as to create an inaudible wide-band noise like code, although other ways of selecting the code frequencies f1 and f0 could be used.
One way of selecting the code frequencies f1 and f0 at the step 46 is to compute the code frequencies by use of a frequency hopping algorithm employing a hop sequence HS and a shift index Ishift. For example, if Ns bits are grouped together to form a pseudo-noise sequence, HS is an ordered sequence of Ns numbers representing the frequency deviation relative to a predetermined reference index I5k. For the case where Ns=7, a hop sequence HS={2, 5, 1, 4, 3, 2, 5} and a shift index Ishift=5, for example, could be used. In general, the indices for the Ns bits resulting from a hop sequence may be given by the following equations:
Another way of selecting the code frequencies at the step 46 is to determine a frequency index Imax at which the spectral power of the audio signal, as determined at the step 44, is a maximum in the low frequency band extending from zero Hz to two kHz. In other words, Imax is the index corresponding to the frequency having maximum power in the range of 0-2 kHz. It is useful to perform this calculation starting at index 1, because index 0 represents the "local" DC component and may be modified by high pass filters used in compression. The code frequency indices I1 and I0 are chosen relative to the frequency index Imax so that they lie in a higher frequency band at which the human ear is relatively less sensitive. Again, one possible choice for the reference frequency f5k is five kHz corresponding to a reference index I5k=53 such that I1 and I0 are given by the following equations:
Unlike many traditional coding methods, such as Frequency Shift Keying (FSK) or Phase Shift Keying (PSK), the present invention does not rely on a single fixed frequency. Accordingly, a "frequency-hopping" effect is created similar to that seen in spread spectrum modulation systems. However, unlike spread spectrum, the object of varying the coding frequencies of the present invention is to avoid the use of a constant code frequency which may render it audible.
For either of the two code frequencies selection approaches (a) and (b) described above, there are at least four modulation methods that can be implemented at a step 56 in order to encode a binary bit of data in an audio block, i.e., amplitude modulation, modulation by frequency swapping, phase modulation, and odd/even index modulation. These four methods of modulation are separately described below.
In order to code a binary '1' using amplitude modulation, the spectral power at I1 is increased to a level such that it constitutes a maximum in its corresponding neighborhood of frequencies. The neighborhood of indices corresponding to this neighborhood of frequencies is analyzed at a step 48 in order to determine how much the code frequencies f1 and f0 must be boosted and attenuated, respectively, so that they are detectable by the decoder 26. For index I1, the neighborhood may preferably extend from I1−2 to I1+2, and is constrained to cover a narrow enough range of frequencies that the neighborhood of I1 does not overlap the neighborhood of I0. Simultaneously, the spectral power at I0 is modified in order to make it a minimum in its neighborhood of indices ranging from I0−2 to I0+2. Conversely, in order to code a binary '0' using amplitude modulation, the power at I1 is attenuated and the power at I0 is increased in their corresponding neighborhoods.
As an example,
The spectral power modification process requires the computation of four values each in the neighborhood of I1 and I0. For the neighborhood of I1 these four values are as follows: (1) Imax1 which is the index of the frequency in the neighborhood of I1 having maximum power; (2) Pmax1 which is the spectral power at Imax1; (3) Imin1 which is the index of the frequency in the neighborhood of I1 having minimum power; and (4) Pmin1 which is the spectral power at Imin1. Corresponding values for the I0 neighborhood are Imax0, Pmax0, Imin0, and Pmin0.
If Imax1=I1, and if the binary value to be coded is a '1,' only a token increase in Pmax1 (i.e., the power at I1) is required at the step 56. Similarly, if Imin0=I0, then only a token decrease in Pmax0 (i.e., the power at I0) is required at the step 56. When Pmax1 is boosted, it is multiplied by a factor 1+A at the step 56, where A is in the range of about 1.5 to about 2.0. The choice of A is based on experimental audibility tests combined with compression survivability tests. The condition for imperceptibility requires a low value for A, whereas the condition for compression survivability requires a large value for A. A fixed value of A may not lend itself to only a token increase or decrease of power. Therefore, a more logical choice for A would be a value based on the local masking threshold. In this case, A is variable, and coding can be achieved with a minimal incremental power level change and yet survive compression.
In either case, the spectral power at I1 is given by the following equation:
The Fourier Transform of the block to be coded as determined at the step 44 also contains negative frequency components with indices ranging in index values from −256 to −1. Spectral amplitudes at frequency indices −I1 and −I0 must be set to values representing the complex conjugate of amplitudes at I1 and I0, respectively, according to the following equations:
Compression algorithms based on the effect of masking modify the amplitude of individual spectral components by means of a bit allocation algorithm. Frequency bands subjected to a high level of masking by the presence of high spectral energies in neighboring bands are assigned fewer bits, with the result that their amplitudes are coarsely quantized. However, the decompressed audio under most conditions tends to maintain relative amplitude levels at frequencies within a neighborhood. The selected frequencies in the encoded audio stream which have been amplified or attenuated at the step 56 will, therefore, maintain their relative positions even after a compression/decompression process.
It may happen that the Fourier Transform ℑ{v(t)} of a block may not result in a frequency component of sufficient amplitude at the frequencies f1 and f0 to permit encoding of a bit by boosting the power at the appropriate frequency. In this event, it is preferable not to encode this block and to instead encode a subsequent block where the power of the signal at the frequencies f1 and f0 is appropriate for encoding.
In this approach, which is a variation of the amplitude modulation approach described above in section (i), the spectral amplitudes at I1 and Imax1 are swapped when encoding a one bit while retaining the original phase angles at I1 and Imax1. A similar swap between the spectral amplitudes at I0 and Imax0 is also performed. When encoding a zero bit, the roles of I1 and I0 are reversed as in the case of amplitude modulation. As in the previous case, swapping is also applied to the corresponding negative frequency indices. This encoding approach results in a lower audibility level because the encoded signal undergoes only a minor frequency distortion. Both the unencoded and encoded signals have identical energy values.
The phase angle associated with a spectral component I0 is given by the following equation:
In order to accomplish this form of modulation, one of the spectral components may have to undergo a maximum phase change of 180�, which could make the code audible. In practice, however, it is not essential to perform phase modulation to this extent, as it is only necessary to ensure that the two components are either "close" to one another in phase or "far" apart. Therefore, at the step 48, a phase neighborhood extending over a range of �π/4 around φR, the reference component, and another neighborhood extending over a range of �π/4 around φR+π may be chosen. The modifiable spectral component has its phase angle φM modified at the step 56 so as to fall into one of these phase neighborhoods depending upon whether a binary '0' or a binary '1' is being encoded. If a modifiable spectral component is already in the appropriate phase neighborhood, no phase modification may be necessary. In typical audio streams, approximately 30% of the segments are "self-coded" in this manner and no modulation is required.
In this odd/even index modulation approach, a single code frequency index, I1, selected as in the case of the other modulation schemes, is used. A neighborhood defined by indexes I1, I1+1, I1+2, and I1+3, is analyzed to determine whether the index IM corresponding to the spectral component having the maximum power in this neighborhood is odd or even. If the bit to be encoded is a '1' and the index IM is odd, then the block being coded is assumed to be "auto-coded." Otherwise, an odd-indexed frequency in the neighborhood is selected for amplification in order to make it a maximum. A bit '0' is coded in a similar manner using an even index. In the neighborhood consisting of four indexes, the probability that the parity of the index of the frequency with maximum spectral power will match that required for coding the appropriate bit value is 0.25. Therefore, 25% of the blocks, on an average, would be auto-coded. This type of coding will significantly decrease code audibility.
It should be noted that these coding techniques preserve the power of the audio signal 14.
A practical problem associated with block coding by either amplitude or phase modulation of the type described above is that large discontinuities in the audio signal can arise at a boundary between successive blocks. These sharp transitions can render the code audible. In order to eliminate these sharp transitions, the time-domain signal v(t) can be multiplied by a smooth envelope or window function w(t) at the step 42 prior to performing the Fourier Transform at the step 44. No window function is required for the modulation by frequency swapping approach described herein. The frequency distortion is usually small enough to produce only minor edge discontinuities in the time domain between adjacent blocks.
The window function w(t) is depicted in
The modified frequency spectrum which now contains the binary code (either '0' or '1') is subjected to an inverse transform operation at a step 62 in order to obtain the encoded time domain signal, as will be discussed below. Following the step 62, the coded time domain signal is determined at a step 64 according to the following equation:
While individual bits of the "robust" ancillary code can be coded by the method described thus far, practical decoding of digital data also requires (i) synchronization, so as to locate the start of data, and (ii) built-in error correction, so as to provide for reliable data reception. The raw bit error rate resulting from coding by spectral modulation is high and can typically reach a value of 20%. In the presence of such error rates, both synchronization and error-correction may be achieved by using pseudo-noise (PN) sequences of ones and zeroes. A PN sequence can be generated, for example, by using an m-stage shift register 58 and an exclusive-OR gate 60 as shown in
An alternative method uses a plurality of PN15 sequences, each of which includes five bits of code data and 10 appended error correction bits. This representation provides a Hamming distance of 7 between any two 5-bit code data words. Up to three errors in a fifteen bit sequence can be detected and corrected. This PN15 sequence is ideally suited for a channel with a raw bit error rate of 20%.
If the first ancillary code contains the twelve bits as described above, and if eight bits are used to specify the number of zeros prior to compression and decompression as described below, the resulting twenty-bit data packet is converted into four groups each containing five bits of data. Ten bits are added to each five bit data group to form four unique 15-bit data PN sequences. A null block may also be added. A PN15 synchronization sequence and the four data sequences together, with each sequence also containing a null block, require 80 audio blocks with a total duration of 0.854 seconds. The structure of each data sequence may be given by the following: DDDDDEEEEEEEEEEN where "N" is a null block that represents no bit, "D" is a data bit, and "E" is an error correction bit. Other sequences may be used.
In terms of synchronization, a unique. synchronization sequence 66 (
As stated earlier, the code data to be transmitted is converted into four bit groups, each of which is represented by a PN15 sequence. As shown in
In the case of stereo signals, the left and right channels are encoded with identical digital data. In the case of mono signals, the left and right channels are combined to produce a single audio signal stream. Because the frequencies selected for modulation are identical in both channels, the resulting monophonic sound is also expected to have the desired spectral characteristics so that, when decoded, the same digital code is recovered.
As described above, the first ancillary code may contain twelve-bits conforming to a specified bit structure, and the second ancillary code may contain a number (such as eight) of bits forming a zero count descriptor that characterizes a part of the audio signal in which the ancillary codes are embedded. The above encoding techniques may be used to encode both the first and second ancillary codes. The zero count descriptor contained in the second ancillary code is generated as described below.
As noted above, each data sequence consists of fifteen data blocks and one null block of audio each of 10.66 millisecond duration. The synchronization sequence also contains sixteen blocks of audio with one of the blocks being a null block. The "zero count" may be computed, for example, on an audio segment containing the synchronization sequence as well as the first and second data sequences in accordance with
However, many implementations of popular decompression algorithms, such as Dolby's AC-3, make use of dithering when recreating an audio signal from a compressed digital audio bit stream. Dithering involves the replacement of the MDCT coefficients, which were set to zero during compression, by small random values prior to the inverse transformation that generates the decompressed time domain signal. The rationale for this dithering operation is that the original MDCT coefficients that were set-to zero had small non-zero values that contributed to the overall energy of the audio stream. Dithering is intended to compensate for this lost energy.
The small random values that are used in dithering are uniformly distributed around a zero mean. Therefore, a large number of zero coefficients are converted to non-zero values. As a result, dithering can result in a decrease in the zero count of the compressed signal, thereby making it more difficult to distinguish between original and compressed/decompressed audio. However, a large enough number of coefficients continue to retain a null value so that the zero count remains a useful tool in detecting compression/decompression.
Accordingly, prior to determining the zero count as described above, the encoder 12 computes a transform 85, such as an MDCT, of the original audio signal 14. The encoder 12 then modifies the transform of the original audio signal 14 by replacing at least some and preferably all of the coefficients whose values are zero with corresponding nominal randomly selected non-zero values 86. Following such modification, the encoder 12 reconstructs the audio by performing an inverse transform, such as an inverse MDCT, on the resulting transform coefficients 87. The resulting audio stream may be referred to as the zero suppressed main audio stream. This zero suppression processing does not perceptibly degrade the quality of the audio signal because the altered coefficients still have extremely low values.
This zero suppression process reduces the zero count significantly, typically by an order of magnitude. For example,
The zero suppressed main audio signal is then further processed as a zero suppressed auxiliary audio stream by non-compression type modifications (such as graphic equalization) that result in an increase of the zero count and that are typically found in receivers and/or players 88. As discussed above, and as shown in
The embedded ancillary code(s) are recovered by the decoder 26. The decoder 26, if necessary, converts the analog audio to a sampled digital output stream at a preferred sampling rate matching the sampling rate of the encoder 12. In decoding systems where there are limitations in terms of memory and computing power, a half-rate sampling could be used. In the case of half-rate sampling, each code block would consist of Nc/2=256 samples, and the resolution in the frequency domain (i.e., the frequency difference between successive spectral components) would remain the same as in the full sampling rate case. In the case where the receiver 20 provides digital outputs, the digital outputs are processed directly by the decoder 26 without sampling but at a data rate suitable for the decoder 26.
The task of decoding is primarily one of matching the decoded data bits with those of a PN15 sequence which could be either a synchronization sequence or a code data sequence representing one or more code data bits. The case of amplitude modulated audio blocks is considered here. However, decoding of phase modulated blocks is virtually identical, except for the spectral analysis, which would compare phase angles rather than amplitude distributions, and decoding of index modulated blocks would similarly analyze the parity of the frequency index with maximum power in the specified neighborhood. Audio blocks encoded by frequency swapping can also be decoded by the same process.
In a practical implementation of audio decoding, such as may be used in a home audience metering system, the ability to decode an audio stream in real-time is highly desirable. The decoder 26 may be arranged to run the decoding algorithm described below on Digital Signal Processing (DSP) based hardware typically used in such applications. As disclosed above, the incoming encoded audio signal may be made available to the decoder 26 from either the audio output 28 or from the microphone 30 placed in the vicinity of the speakers 24. In order to increase processing speed and reduce memory requirements, the decoder 26 may sample the incoming encoded audio signal at half (24 kHz) of the normal 48 kHz sampling rate.
Before recovering the actual data bits representing code information, it is necessary to locate the synchronization sequence. In order to search for the synchronization sequence within an incoming audio stream, blocks of 256 samples, each consisting of the most recently received sample and the 255 prior samples, could be analyzed. For real-time operation, this analysis, which includes computing the Fast Fourier Transform of the 256 sample block, has to be completed before the arrival of the next sample. Performing a 256-point Fast Fourier Transform on a 40 MHZ DSP processor takes about 600 microseconds. However, the time between samples is only 40 microseconds, making real time processing of the incoming coded audio signal as described above impractical with current hardware.
Therefore, instead of computing a normal Fast Fourier Transform on each 256 sample block, the decoder 26 may be arranged to achieve real-time decoding by implementing an incremental or sliding Fast Fourier Transform routine 100 (
Moreover, unlike a conventional transform which computes the complete spectrum consisting of 256 frequency "bins," the decoder 26 computes the spectral amplitude only at frequency indexes that belong to the neighborhoods of interest, i.e., the neighborhoods used by the encoder 12. In a typical example, frequency indexes ranging from 45 to 70 are adequate so that the corresponding frequency spectrum contains only twenty-six frequency bins. Any code that is recovered appears in one or more elements of the status information array SIS as soon as the end of a message block is encountered.
Additionally, it is noted that the frequency spectrum as analyzed by a Fast Fourier Transform typically changes very little over a small number of samples of an audio stream. Therefore, instead of processing each block of 256 samples consisting of one "new" sample and 255 "old" samples, 256 sample blocks may be processed such that, in each block of 256 samples to be processed, the last k samples are "new" and the remaining 256-k samples are from a previous analysis. In the case where k=4, processing speed may be increased by skipping through the audio stream in four sample increments, where a skip factor k is defined as k=4 to account for this operation.
Each element SIS[p] of the status information array SIS consists of five members: a previous condition status PCS, a next jump index JI, a group counter GC, a raw data array DA, and an output data array OP. The raw data array DA has the capacity to hold fifteen integers. The output data array OP stores ten integers, with each integer of the output data array OP corresponding to a five bit number extracted from a recovered PN15 sequence. This PN15 sequence, accordingly, has five actual data bits and ten other bits. These other bits may be used, for example, for error correction. It is assumed here that the useful data in a message block consists of 50 bits divided into 10 groups with each group containing 5 bits, although a message block of any size may be used.
The operation of the status information array SIS is explained in connection with
In order to first locate the synchronization sequence, the Fast Fourier Transform corresponding to the initial 256 sample block read at the processing stage 102 is tested at a processing stage 106 for a triple tone, which represents the first bit in the synchronization sequence. The presence of a triple tone may be determined by examining the initial 256 sample block for the indices I0, I1, and Imid used by the encoder 12 in generating the triple tone, as described above. The SIS[p] element of the SIS array that is associated with this initial block of 256 samples is SIS[0], where the status array index p is equal to 0.
If a triple tone is found at the processing stage 106, the values of certain members of the SIS[0] element of the status information array SIS are changed at a processing stage 108 as follows: the previous condition status PCS, which is initially set to 0, is changed to a 1 indicating that a triple tone was found in the sample block corresponding to SIS[0]; the value of the next jump index JI is incremented to 1; and, the first integer of the raw data member DA[0] in the raw data array DA is set to the value (0 or 1) of the triple tone. In this case, the first integer of the raw data member DA[0] in the raw data array DA is set to 1 because it is assumed in this analysis that the triple tone is the equivalent of a 1 bit. Also, the status array index p is incremented by one for the next sample block. If there is no triple tone, none of these changes in the SIS[0] element are made at the processing stage 108, but the status array index p is still incremented by one for the next sample block. Whether or not a triple tone is detected in this 256 sample block, the routine 100 enters an incremental FFT mode at a processing stage 110.
Accordingly, a new 256 sample block increment is read into the buffer at a processing stage 112 by adding four new samples to, and discarding the four oldest samples from, the initial 256 sample block processed at the processing stages 102-106. This new 256 sample block increment is analyzed at a processing stage 114 according to the following steps:
Because p is not yet equal to 64 as determined at a processing stage 118 and the group counter GC has not accumulated a count of 10 as determined at a processing stage 120, this analysis corresponding to the processing stages 112-120 proceeds in the manner described above in four sample increments where p is incremented for each four sample increment. When SIS[63] is reached where p=64, p is reset to 0 at the processing stage 118, and the 256 sample block increment now in the buffer is exactly 256 samples away from the location in the audio stream at which the SIS[0] element was last updated. Each time p reaches 64, the SIS array represented by the SIS[0]-SIS[63] elements is examined to determine whether the previous condition status PCS of any of these elements is one indicating a triple tone. If the previous condition status PCS of any of these elements corresponding to the current 64 sample block increments is not one, the processing stages 112-120 are repeated for the next 64 block increments. (Each block increment comprises 256 samples.)
Once the previous condition status PCS is equal to 1 for any of the SIS[0]-SIS[63] elements corresponding to any set of 64 sample block increments, and the corresponding raw data member DA[p] is set to the value of the triple tone bit, the next 64 block increments are analyzed at the processing stages 112-120 for the next bit in the synchronization sequence.
Each of the new block increments beginning where p was reset to 0 is analyzed for the next bit in the synchronization sequence. This analysis uses the second member of the hop sequence HS because the next jump index JI is equal to 1. From this hop sequence number and the shift index used in encoding, the I1 and I0 indexes can be determined, for example from equations (4) and (5). Then, the neighborhoods of the I1 and I0 indexes are analyzed to locate maximums and minimums in the case of amplitude modulation. If, for example, a power maximum at I1 and a power minimum at I0 are detected, the next bit in the synchronization sequence is taken to be 1. In order to allow for some variations in the signal that may arise due to compression or other forms of distortion, the index for either the maximum power or minimum power in a neighborhood is allowed to deviate by one from its expected value. For example, if a power maximum is found in the index I1, and if the power minimum in the index I0 neighborhood is found at I0−1, instead of I0, the next bit in the synchronization sequence is still taken to be 1. On the other hand, if a power minimum at I1 and a power maximum at I0 are detected using the same allowable variations discussed above, the next bit in the synchronization sequence is taken to be 0. However, if none of these conditions are satisfied, the output code is set to −1, indicating a sample block that cannot be decoded. Assuming that a 0 bit or a 1 bit is found, the second integer of the raw data member DA[1] in the raw data array DA is set to the appropriate value, and the next jump index JI of SIS[0] is incremented to 2, which corresponds to the third member of the hop sequence HS. From this hop sequence number and the shift index used in encoding, the I1 and I0 indexes can be determined. Then, the neighborhoods of the I1 and I0 indexes are analyzed to locate maximums and minimums in the case of amplitude modulation so that the value of the next bit can be decoded from the third set of 64 block increments, and so on for the remaining ones of the fifteen bits of the synchronization sequence. The fifteen bits stored in the raw data array DA may then be compared with a reference synchronization sequence to determine synchronization. If the number of errors between the fifteen bits stored in the raw data array DA and the reference synchronization sequence exceeds a previously set threshold, the extracted sequence is not acceptable as a synchronization, and the search for the synchronization sequence begins anew with a search for a triple tone.
If a valid synchronization sequence is thus detected, there is a valid synchronization, and the PN15 data sequences may then be extracted using the same analysis as is used for the synchronization sequence, except that detection of each PN15 data sequence is not conditioned upon detection of the triple tone which is reserved for the synchronization sequence. As each bit of a PN15 data sequence is found, it is inserted as a corresponding integer of the raw data array DA. When all integers of the raw data array DA are filled, (i) these integers are compared to each of the thirty-two possible PN15 sequences, (ii) the best matching sequence indicates which 5-bit number to select for writing into the appropriate array location of the output data array OP, and (iii) the group counter GC member is incremented to indicate that the first PN15 data sequence has been successfully extracted. If the group counter GC has not yet been incremented to 10 (this number depends on the number of groups of bits required to encode the first and second ancillary codes) as determined at the processing stage 120, program flow returns to the processing stage 112 in order to decode the next PN15 data sequence.
When the group counter GC has incremented to 10 (or other appropriate number such as four for the twelve-bit first ancillary code and the eight-bit second ancillary code described above) as determined at the processing stage 120, the output data array OP, which contains a full 50-bit message (or 20-bit message as appropriate), is read at a processing stage 122. It is possible that several adjacent elements of the status information array SIS, each representing a message block separated by four samples from its neighbor, may lead to the recovery of the same message because synchronization may occur at several locations in the audio stream which are close to one another. If all these messages are identical, there is a high probability that an error-free code has been received.
Once a message has been recovered and the message has been read at the processing stage 122, the previous condition status PCS of the corresponding SIS element is set to 0 at a processing stage 124 so that searching is resumed at a processing stage 126 for the triple tone of the synchronization sequence of the next message block.
The zero count ancillary code, which was encoded into the audio signal 14 by the encoder 12 either alone or with another ancillary code (such as the first ancillary code described above), is decoded by the decoder 26 using, for example, the decoding technique described above. For example, the decoded zero count may be used by the decoder 26 to determine if the audio signal 14 has undergone compression/decompression.
In order to detect compression/decompression, which increases the zero coefficient count of a transform of an audio signal, the decoder 26 decodes the zero count ancillary code. Also, the decoder 26, following non-compression type modifications (such as graphic equalization) which tend to increase the zero count of a transform of the signal, performs a transform (such as that exemplified by equation (1)) on the same portion of the audio signal 14 that was used by the encoder 12 to make the zero count calculation described above. The decoder 26 then counts the zero coefficients in the transform. For example, if the eight-bit zero count second ancillary code is appended to the twelve-bit first ancillary code as discussed above, the decoder 26 can make its zero count from the transformed portion of the received audio signal containing the synchronization sequence and the first two data sequences (containing the first ten bits of the twelve-bit first ancillary code).
Thereafter, the decoder 26 compares the zero count that it calculates to the zero count contained in the zero count ancillary code as decoded from the audio signal 14. If the difference between the zero count that it calculates and the zero count contained in the zero count ancillary code is greater than a count threshold (such as 400), the decoder 26 may conclude that the received audio stream has been subjected to compression/decompression. The eight-bit descriptor obtained from the embedded code may be multiplied by five if the zero count determined by the encoder 12 was divided by five prior to encoding. Thus, the calculated zero count must exceed the zero count contained in the zero count ancillary code by a predetermined amount in order for the decoder 26 to conclude that the audio signal 14 has undergone compression/decompression.
Accordingly, if the decoder 26 concludes that the audio signal 14 has undergone compression/decompression, the decoder 26 may be arranged to take some action such as controlling the receiver 20 in a predetermined manner. For example, if the receiver 20 is a player, the decoder 26 may be arranged to prevent the player from playing the audio signal 14.
Certain modifications of the present invention have been discussed above. Other modifications will occur to those practicing in the art of the present invention. For example, the invention has been described above in connection with the transmission of an encoded signal from the transmitter 16 to the receiver 20. Alternatively, the present invention may be used in connection with other types of systems. For example, the transmitter 16 could instead be a recording device arranged to record the encoded signal on a medium, and the receiver 20 could instead be a player arranged to play the encoded signal stored on the medium. As another example, the transmitter 16 could instead be a server, such as a web site, and the receiver 20 could instead be a computer or other receiver such as web compliant device coupled over a network, such as the Internet, to the server in order to download the encoded signal.
Also, as described above, coding a signal with a "1" bit using amplitude modulation involves boosting the frequency f1 and attenuating the frequency f0, and coding a signal with a "0" bit using amplitude modulation involves attenuating the frequency f1 and boosting the frequency f0. Alternatively, coding a signal with a "1" bit using amplitude modulation could instead involve attenuating the frequency f1 and boosting the frequency f0, and coding a signal with a "0" bit using amplitude modulation could involve boosting the frequency f1 and attenuating the frequency f0.
Moreover, a triple tone is used to make a synchronization sequence unique. However, a triple tone need not be used if a unique PN15 sequence is available and is clearly distinguishable from possible data sequences.
Furthermore, as described above, twelve bits are used for the first ancillary code and eight bits are used for the second ancillary code. Instead, the number of bits in the first and/or second ancillary codes may be other than twelve and eight respectively, as long as the total number of bits in the first and second ancillary codes add to a number divisible by five using the PN15 sequences described above. Alternatively, other sequences can be used which would not require the total number of bits in the first and second ancillary codes to be divisible by five. In addition, the zero count (second) ancillary code can be used without the first ancillary code.
Also, as described above, the zeros produced by a transform, which may be an MDCT but which could be any other suitable transform, are counted. However, values other zero count could instead, or in addition, be counted as long as these values occur more often in a transform after compression/decompression than before compression/decompression.
Accordingly, the description of the present invention is to be construed as illustrative only and is for the purpose of teaching those skilled in the art the best mode of carrying out the invention. The details may be varied substantially without departing from the spirit of the invention, and the exclusive use of all modifications which are within the scope of the appended claims is reserved.
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Well folks, it's early July, and that means one thing to obsessive music geeks everywhere: Time to make a list!
The first six months of 2009 have come and gone, and they were a bit of a roller-coaster for me, music-wise. For the first few months of the year, I didn't hear much that I loved and was convinced this would be the weakest year of new music in quite some time. (I'm talking non-local here. Locally, it's been a stellar year for recorded music. Click here or grab a copy of today's GO! Magazine for more on that.)
In the last few months, though, things have picked up considerably. Truth is, I think running this blog has helped a lot. It has forced me to pay closer attention to new music, and to check out more stuff than I think I normally would've. As a result, there are a lot of names on the list below that I wasn't necessarily familiar with on New Year's Day.
And that, really, is what I love about music. I'm constantly on the lookout for what's new. I love the thrill of discovering a new band, even if they're not necessarily making a new sound. I like looking forward, not back. (Never mind the fact that I've listened to Kathleen Edwards' 2008 album "Asking For Flowers" more over the past six months than any of the albums listed below.)
With that in mind, let's look back (haha) at the records that really grabbed me in the first half of 2009. I'm not going to clutter this post up with CD covers or YouTube videos. Just click on the artist names to find out more.
The Best of the Best
Crocodiles, "Crocodiles" (slacker shoegaze)
Japandroids, "Post-Nothing" (noisy garage rock … read more here)
School of Seven Bells, "Alpinisms" (icy electro-pop … read more here)
Shabazz Palaces, "CD1″/"CD2″ (brainy Seattle hip-hop … read more here)
Weinland, "Breaks In The Sun" (graceful folk-pop … read more here and watch video of their live show in Bend here)
The Best of the Rest
Brakes, "Touchdown" (pure, perfect pop-rock)
Dinosaur Jr, "Farm" (guitars on blast)
Truck North, "Truck Jewels" (Philly hip-hop … read more here)
More Good Stuff
Animal Collective, "Merriweather Post Pavilion" (psychedelic Afro-electro-pop)
Arbouretum, "Song of the Pearl" (stoner blues)
Assemble Head in Sunburst Sound, "When Sweet Sleep Returned" (psych-pop)
Black Moth Super Rainbow, "Eating Us" (analog weirdness … read more here)
Gui Boratto, "Take My Breath Away" (twinkling electronica)
Danger Mouse, Sparklehorse and David Lynch, "Dark Night of the Soul" (creepy pop … read more here)
The Decemberists, "The Hazards of Love" (indie-prog)
John Doe & The Sadies, "Country Club" (country)
The Pains of Being Pure at Heart, "The Pains of Being Pure at Heart" (throwback twee)
Sonic Youth, "The Eternal" (noise-rock)
Telekinesis, "Telekinesis!" (indie-pop)
Various artists, "Dark Was the Night" (indie-rock Mt. Rushmore)
Wale, "Back to the Feature" (D.C. hip-hop)
OK. Hopefully I didn't forget something. If I did, remind me in the comments. Think I should've included your favorite album? Tell me in the comments. Think one of those albums stinks? Let me know in the comments! I love nothing more than talking about music…


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The Wellness Working Group (WWG) is charged with helping to connect Colby faculty and staff with College and community wellness opportunities and programs and raise awareness of wellness among College employees. The group currently has three major activities per year:
In addition to these activities, the membership of the committee, drawn from a wide cross-section of Colby employees as well as representative(s) from CIGNA Insurance, passes along information on fitness and wellness opportunities to the entire campus.
For more information, contact Human Resources at extension 5500.


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This story was updated at 11:00 a.m. Tuesday.
A 19-year-old Montgomery College student was taken into police custody Monday afternoon after he threatened to shoot people at the college�s campus in Rockville, according to Rockville police.
Rockville City Police Lt. Eric Over said officers received a call about the incident at about 2:15 p.m. Monday.
�It looks like there was a comment made; the comment was, �I�m going to shoot everyone at this college and enjoy doing it,�� Over said. �... That was actually made in the presence of several witnesses.�
The student was still in custody as of 10 a.m. Tuesday, but has not been charged, said Maj. Bob Rappoport of Rockville Police.
He was taken to a local hospital for an evaluation, Rappoport said.
�We looked at it very seriously to intervene,� he said.
Three students reported the man�s comments to the campus� security office, and after interviewing the witnesses, security staff called police, searched the campus for the student, and found him eating lunch at a bench near the student center, said College Spokeswoman Elizabeth Homan.
Rockville police took the student into custody Monday afternoon and gave him a letter of trespass that bans him from the college campus for three years unless he receives permission from the college, Homan said.
There was no lockdown on the campus, as lockdowns only occur if there is an active security risk that presents an imminent threat to lives, such as an active shooting or hostage situation, Homan said.
The witnesses did the right thing by going to security after talking to the student, Homan said.
�In today�s age you need to take comments seriously and you always should be careful,� Homan said. �Credit should be given to them for taking the steps and being concerned enought to go to the security office.�
Students can also report threats to a trusted faculty member. If it is an active threat, they should call 911 first, Homan said.
Over said it is too early in the investigation to say how serious the man was about carrying out his threat. Police have not determined whether they will charge him and are not releasing his name at this time.
Anyone with information should call Rockville police at 240-314-8900 or Montgomery College Safety and Security at 240-567-5111.