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Featured Article from Software Monetization
Software Monetization: Countering Piracy Threats
As mentioned in previous columns, a software protection solution is necessary to help reduce the use of unlicensed software.For every $100 worth of legitimate software sold, an additional $75 worth of unlicensed software also made its way into the market, generating absolutely no revenue for the original ISV (independent software vendor), according to the Business Software Alliance (BSA)/IDC 2010 Piracy Impact Study.
A recent SafeNet (News - Alert) whitepaper, From Software Hack to Counterattack, examines a variety of the most common application hacking techniques and the best counterattacks for protecting applications from piracy. Patching executable files and dynamically linked libraries is one such anti-hacking tactic.
"The most convenient way to protect a large number of executable and dynamically linked libraries is by using automatic protection tools also known as 'packers,'" the whitepaper stated. "The most effective tools also offer anti-piracy specific measures such as anti-debugging mechanisms, executable file encryption and data file encryption mechanisms allowing the protected application to encrypt and decrypt data files on-the-fly."
SafeNet also offers general software protection strategies for publishers employing hardware-based protection keys, including the following best practices:
Use Both the API-based implementation and automatic protection – Maximize security by using the API (application programming interface) to implement calls to the software protection keys, and protect the program with an automatic protection tool that offers anti-debugging and anti-reverse engineering measures. According to SafeNet, using one protection method does not preclude the use of the other.
Insert multiple calls in your code – Inserting many calls to the protection key throughout the code and binding key data with software functionality frustrates those attempting to crack your software. Multiple calls increase the difficulty in tracing a protection scheme.
Encrypt/decrypt data using the protection key's on-chip encryption engine – The encryption process, which takes place before distributing the software, and the decryption processes, which take place during run-time of the protected application.
Only when both the security infrastructure and its implementation are maximized is the software truly protected, according to SafeNet.
"This may sound like an enormous task to accomplish, and while it is by no means trivial, it is definitely achievable," the whitepaper stated. "Implementing effective protection can not only increase your product's longevity, but it can also significantly increase your company's revenue, making it well worth the effort."
Edited by Peter Bernstein
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I replaced the shifter plate on my empi after market shifter. All the gears work when the car is not running, but when I start the car I can't get any gear. WHY! Thanks Bill
Might try the clutch plate.
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4 items from 2016
23 February 2016 12:57 PM, PST | DailyDead | See recent DailyDead news »
Today, Scream Factory releases Wes Craven's The Serpent and the Rainbow Collector's Edition on Blu-ray, and we've been provided with three copies to give away to Daily Dead readers.
Prize Details: (3) Winners will receive (1) Collector's Edition Blu-ray copy of The Serpent and the Rainbow.
How to Enter: For a chance to win, email email@example.com with the subject "The Serpent and the Rainbow Contest". Be sure to include your name and mailing address.
Entry Details: The contest will end at 12:01am Est on February 29th. This contest is only open to those who are eighteen years of age or older that live in the United States. Only one entry per household will be accepted.
"Wes Craven (A Nightmare On Elm Street, Scream) directs this terrifying story of one man's nightmarish journey into the blood-curdling, deadly world of voodoo.
A Harvard anthropologist (Bill Pullman, »
- Derek Anderson
22 February 2016 5:52 PM, PST | DailyDead | See recent DailyDead news »
February's home entertainment releases are ending on a high note, so genre fans should get their wallets ready in anticipation of all the great horror and sci-fi titles coming our way on the 23rd. Scream Factory has several fantastic releases planned for this Tuesday, including the highly anticipated Blu-ray for Wes Craven's The Serpent and the Rainbow as well as double features of The Curse / Curse II: The Bite and Millennium / R.O.T.O.R.
Candyman director Bernard Rose's latest endeavor—the modern interpretation of the classic Frankenstein tale—hits Blu and DVD this week, and the cult classic Children Shouldn't Play With Dead Things is getting an HD overhaul this Tuesday as well.
Other notable Blu-ray and DVD releases on February 23rd include The Bees, Bigfoot Vs. Zombies, Moonwalkers, Demonoid, American Horror Project: Volume One and Upsidedown Cross.
Children Shouldn't Play With Dead Things (Vci Entertainment, »
- Heather Wixson
20 February 2016 10:23 AM, PST | DailyDead | See recent DailyDead news »
Next Tuesday, Scream Factory will bring viewers to the "forbidden world between life and death" with their Collector's Edition Blu-ray of the late Wes Craven's The Serpent and the Rainbow. To give fans an idea of what to expect from the special release, we have high-definition clips and a trailer from the anticipated Blu-ray.
"Wes Craven (A Nightmare On Elm Street, Scream) directs this terrifying story of one man's nightmarish journey into the blood-curdling, deadly world of voodoo.
A Harvard anthropologist (Bill Pullman, Lake Placid, Independence Day) is sent to Haiti to retrieve a strange powder that is said to have the power to bring human beings back from the dead. In his quest to find the miracle drug, the cynical scientist enters the rarely seen netherworld of walking zombies, blood rites and ancient curses.
Based on the true-to-life experiences of Wade Davis, starring Cathy Tyson (Mona Lisa), Zakes Mokae (Dust Devil, »
- Derek Anderson
6 January 2016 9:57 AM, PST | DailyDead | See recent DailyDead news »
For quite some time now, the folks at Scream Factory have been hard at work on a Collector's Edition Blu-ray of the late, great Wes Craven's The Serpent and the Rainbow. Today, they have rewarded fans' patience by revealing the release's official list of special features, and they are well worth the wait.
In addition to a fresh HD transfer taken from The Serpent and the Rainbow's original inter-positive, this Collector's Edition Blu-ray also includes a new commentary with lead actor Bill Pullman and a "making of" featurette with an impressive number of interviews. Scream Factory will release this special Blu-ray on February 23rd:
Press Release: Loyal fans of legendary director Wes Craven (Scream, The People Under the Stairs, A Nightmare on Elm Street) know well the terrifying story of one man's nightmarish journey into the eerie and deadly world of voodoo from the 1988 horror cult classic The Serpent And The Rainbow, »
- Derek Anderson
4 items from 2016
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3rd Degree has no description.
I Love Music in a Weird way. When i hear the Bass DROP i tend to get kinda cheesy and smile ear to ear. I Love songs that are Brand New and just Dropped hours ago. New to the World Kinda jams. I have Fun with Musick and you should too......…ENJOY! Shout Out to all DJs and Producers on the Come Up. I Dig It !!
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Geely Philippines
One of China's most successful automotive firms, Geely has been around since 1986. But back then, Geely was a refrigerator maker. Founded by Mr. Li Shufu, with financial help from his father, he forged Geely into its current status as a manufacturer of quality and inexpensive cars, targeted to the Chinese masses. As a politically-independent automotive manufacturing privateer, Geely remained as the only automotive firm in China that has no ties with the country's state government, allowing it to grow more rapidly on a global scale.
The '90s saw its success as a motorcycle and small van manufacturer, and in 1998, it was finally granted an approval by the state government to manufacture and develop a competitive line of automobiles.
Global expansion was its next mission, and sure enough, after making ties with Volvo in Europe, its closest Asian neighbour was one of its first destinations. Here in the Philippines, Geely operates under the leadership of the AutoChina Inc. (ACI).
ACI began distributing Geely automobiles in the country in 2011 and almost instantaneously managed to gain a stable stepping stone into the highly competitive Philippine automotive industry.
Geely PH currently has three dealerships nationwide, located along North EDSA, in Cebu, and in Davao.


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|Publication number
|Publication type
|Application number
|Publication date
|Aug 2, 2005
|Filing date
|Oct 8, 2003
|Priority date
|Jun 10, 1997
|Fee status
|Also published as
|Publication number
|Lars Gustaf Liljeryd, Per Rune Albin Ekstrand, Lars Fredrik Henn, Hans Magnus Kristofer Kjorling
|Original Assignee
|Coding Technologies Ab
|Export Citation
|BiBTeX, EndNote, RefMan
|Patent Citations (14), Non-Patent Citations (16), Referenced by (45), Classifications (10), Legal Events (5)
|External Links: USPTO, USPTO Assignment, Espacenet
This application is a divisional of application Ser. No. 09/230,799, filed on Feb. 9, 1999, now U.S. Pat. No. 6,680,972, and for which priority is claimed under 35 U.S.C. � 120. Application Ser. No. 09/230,799 is the national phase of PCT International Application No. PCT/IB98/00893 filed on Jun. 9, 1998 under 35 U.S.C. � 371. The entire contents of each of the above-identified applications are hereby incorporated by reference. This application also claims priority of Application No. 9702213-1; 9704634-6; and 9800268-6 filed in Sweden on Jun. 10, 1997; Dec. 12, 1997; and Jan. 30, 1998, respectively under 35 U.S.C. � 119.
In source coding systems, digital data is compressed before transmission or storage to reduce the required bitrate or storing capacity. The present invention relates to a new method and apparatus for the improvement of source coding systems by means of Spectral Band Replication (SBR). Substantial bitrate reduction is achieved while maintaining the same perceptual quality or conversely, an improvement in perceptual quality is achieved at a given bitrate. This is accomplished by means of spectral bandwidth reduction at the encoder side and subsequent spectral band replication at the decoder, whereby the invention exploits new concepts of signal redundancy in the spectral domain.
Audio source coding techniques can be divided into two classes: natural audio coding and speech coding. Natural audio coding is commonly used for music or arbitrary signals at medium bitrates, and generally offers wide audio bandwidth. Speech coders are basically limited to speech reproduction but can on the other hand be used at very low bitrates, albeit with low audio bandwidth. Wideband speech offers a major subjective quality improvement over narrow band speech. Increasing the bandwidth not only improves intelligibility and naturalness of speech, but also facilitates speaker recognition. Wideband speech coding is thus an important issue in next generation telephone systems. Further, due to the tremendous growth of the multimedia field, transmission of music and other non-speech signals at high quality over telephone systems is a desirable feature.
A high-fidelity linear PCM signal is very inefficient in terms of bitrate versus the perceptual entropy. The CD standard dictates 44.1 kHz sampling frequency, 16 bits per sample resolution and stereo. This equals a bitrate of 1411 kbit/s. To drastically reduce the bitrate, source coding can be performed using split-band perceptual audio codecs. These natural audio codecs exploit perceptual irrelevancy and statistical redundancy in the signal. Using the best codec technology, approximately 90% data reduction can be achieved for a standard CD-format signal with practically no perceptible degradation. Very high sound quality in stereo is thus possible at around 96 kbit/s, i.e. a compression factor of approximately 15:1. Some perceptual codecs offer even higher compression ratios. To achieve this, it is common to reduce the sample-rate and thus the audio bandwidth. It is also common to decrease the number of quantization levels, allowing occasionally audible quantization distortion, and to employ degradation of the stereo field, through intensity coding. Excessive use of such methods results in annoying perceptual degradation. Current codec technology is near saturation and further progress in coding gain is not expected. In order to improve the coding performance further, a new approach is necessary.
The human voice and most musical instruments generate quasistationary signals that emerge from oscillating systems. According to Fourier theory, any periodic signal may be expressed as a sum of sinusoids with the frequencies f, 2f, 3f, 4f, 5f etc. where f is the fundamental frequency. The frequencies form a harmonic series. A bandwidth limitation of such a signal is equivalent to a truncation on of the harmonic series. Such a truncation alters the perceived timbre, tone colour, of a musical instrument or voice, and yields an audio signal that will sound "muffled" or "dull", and intelligibility may be reduced. The high frequencies are thus important for the subjective impression of sound quality.
Prior art methods are mainly intended for improvement of speech codec performance and in particular intended for High Frequency Regeneration (HFR), an issue in speech coding. Such methods employ broadband linear frequency shifts, non-linearities or aliasing [U.S. Pat. No. 5,127,054] generating intermodulation products or other non-harmonic frequency components which cause severe dissonance when applied to music signals. Such dissonance is referred to in the speech coding literature as "harsh" and "rough" sounding. Other synthetic speech HFR methods generate sinusoidal harmonics that are based on fundamental pitch estimation and are thus limited to tonal stationary sounds [U.S. Pat. No. 4,771,465]. Such prior art methods, although useful for low-quality speech applications, do not work for high quality speech or music signals. A few methods attempt to improve the performance of high quality audio source codecs. One uses synthetic noise signals generated at the decoder to substitute noise-like signals in speech or music previously discarded by the encoder ["Improving Audio Codecs by Noise Substitution" D. Schultz, JAES, Vol. 44, No. 7/8, 1996]. This is performed within an otherwise normally transmitted highband at an intermittent basis when noise signals are present. Another method recreates some missing highband harmonics that were lost in the coding process ["Audio Spectral Coder" A. J. S. Ferreira, AES Peeprint 4201, 100th Convention, May 11-14 1996, Copenhagen] and is again dependent on tonal signals and pitch detection. Both methods operate at a low duty-cycle basis offering comparatively limited coding or performance gain.
The present invention provides a new method and an apparatus for substantial improvements of digital source coding systems and more specifically for the improvements of audio codecs. The objective includes bitrate reduction or improved perceptual quality or a combination thereof. The invention is based on new methods exploiting harmonic redundancy, offering the possibility to discard passbands of a signal prior to transmission or storage. No perceptual degradation is perceived if the decoder performs high quality spectral replication according to the invention. The discarded bits represent the coding gain at a fixed perceptual quality. Alternatively, more bits can be allocated for encoding of the lowband information at a fixed bitrate, thereby achieving a higher perceptual quality.
The present invention postulates that a truncated harmonic series can be extended based on the direct relation between lowband and highband spectral components. This extended series resembles the original in a perceptual sense if certain rules are followed: First, the extrapolated spectral components must be harmonically related to the truncated harmonic series, in order to avoid dissonance-related artifacts. The present invention uses transposition as a means for the spectral replication process, which ensures that this criterion is met. It is however not necessary that the lowband spectral components form a harmonic series for successful operation, since new replicated components, harmonically related to those of the lowband, will not alter the noise-like or transient nature of the signal. A transposition is defined as a transfer of partials from one position to another on the musical scale while maintaining the frequency ratios of the partials. Second, the spectral envelope, i.e. the coarse spectral distribution, of the replicated highband, must reasonably well resemble that of the original signal. The present invention offers two modes of operation, SBR-1 and SBR-2, that differ in the way the spectral envelope is adjusted.
SBR-1, intended for the improvement of intermediate quality codec applications, is a single-ended process which relies exclusively on the information contained in a received lowband or lowpass signal at the decoder. The spectral envelope of this signal is determined and extrapolated, for instance using polynomials together with a set of rules or a codebook. This information is used to continuously adjust and equalise the replicated highband. The present SBR-1 method offers the advantage of post-processing, i.e. no modifications are needed at the encoder side. A broadcaster will gain in channel utilisation or will be able to offer improved perceptual quality or a combination of both. Existing bitstream syntax and standards can be used without modification.
SBR-2, intended for the improvement of high quality codec applications, is a double-ended process where, in addition to the transmitted lowband signal according to SBR-1, the spectra envelope of the highband is encoded and transmitted. Since the variations of the spectral envelope has a much lower rate than the highband signal components, only a limited amount of information needs to be transmitted in order to successfully represent the spectral envelope. SBR-2 can be used to improve the performance of current codec technologies with no or minor modifications of existing syntax or protocols, and as a valuable tool for future codec development.
Both SBR-1 and SBR-2 can be used to replicate smaller passbands of the lowband when such bands are shut down by the encoder as stipulated by the psychoacoustic model under bit-starved conditions. This results in improvement of the perceptual quality by spectral replication within the lowband in addition to spectral replication outside the lowband. Further, SBR-1 and SBR-2 can also be used in codecs employing bitrate scalability, where the perceptual quality of the signal at the receiver varies depending on transmission channel conditions. This usually implies annoying variations of the audio bandwidth at the receiver. Under such conditions, the SBR methods can be used successfully in order to maintain a constantly high bandwidth, again improving the perceptual quality.
The present invention operates on a continuous basis, replicating any type of signal contents, i.e. tonal or non-tonal (noise-like and transient signals). In addition, the present spectral replication method creates a perceptually accurate replica of the discarded bands from available frequency bands at the decoder. Hence, the SBR method offers a substantially higher level of coding gain or perceptual quality improvement compared to prior art methods. The invention can be combined with such prior art codec improvement methods; however, no performance gain is expected due to such combinations.
The SBR-method comprises the following steps:
The passbands of the second signal may be set not to overlap or partly overlap the passbands of the first signal and may be set in dependence of the temporal characteristics of the original signal and/or the first signal, or transmission channel conditions. The spectral envelope adjustment is performed based on estimation of the original spectral envelope from said first signal or on transmitted envelope information of the original signal.
The present invention includes to basic types of transposers: multiband transposers and time-variant pattern search prediction transposers, having different properties. A basic multiband transposition may be performed according to the present invention by the following:
Alternatively, this basic multiband transposition may be performed according to the invention by the following:
An improved multiband transposition according to the invention incorporates phase adjustments, enhancing the performance of the basic multiband transposition.
The time-variant pattern search prediction transposition according to the present invention may be performed by the following:
The SBR methods and apparatuses according to the present invention offer the following features:
The most attractive application relates to the improvement of various types of low bitrate codecs, such as MPEG 1/2 Layer I/II/III [U.S. Pat. No. 5,040,217], MPEG 2/4 AAC, Dolby AC-2/3, NTT TwinVQ [U.S. Pat. No. 5,684,920], AT&T/Lucent PAC etc. The invention is also useful in high-quality speech codecs such as wide-band CELP and SB-ADPCM G.722 etc. to improve perceived quality. The above codecs are widely used in multimedia, in the telephone industry, on the internet as well as in professional applications. T-DAB (Terrestrial Digital Audio Broadcasting) systems use low bitrate protocols that will gain in channel utilisation by using the present method, or improve quality in FM and AM DAB. Satellite S-DAB will gain considerably, due to the excessive system costs involved, by using the present method to increase the number of programme channels in the DAB multiplex. Furthermore, for the first time, full bandwidth audio real-time streaming over the Internet is achievable using low bitrate telephone modems.
The present invention will now be described by way of illustrative examples, not limiting the scope or spirit of the invention, with reference to the accompanying drawings, in which:
Throughout the explanation of the embodiments herein, emphasis is given to natural audio source coding applications. However, it should be understood that the present invention is applicable on a range of source coding applications other than that of encoding and decoding audio signals.
Transposition Basics
Transposition as defined according to the present invention, is the ideal method for spectral replication, and has several major advantages over prior art such as: no pitch detection is required, equally high performance for single-pitched and polyphonic programme material is obtained, and the transposition works equally well for tonal and non-tonal signals. Contrary to other methods, the transposition according to the invention can be used in arbitrary audio source coding systems for arbitrary signal types.
An exact transposition a factor M of a discrete time signal x(n) in the form of a sum of cosines with time varying amplitudes, is defined by the relation
where N is the number of sinusoids, hereafter referred to as partials, fi, ei(n), αi are the individual input frequencies, time envelopes and phase constants respectively, βi are the arbitrary output phase constants and fs is the sampling frequency, and 0≦Mfi≦fs/2.
An approximation of an exact transposition may be used. According to the present invention, the quality of such approximations is determined using dissonance theory. A criterion for dissonance is presented by Plomp ["Tonal Consonance and Critical Bandwidth" R. Plomp, W. J. M. Levelt JASA, Vol 38, 1965], and states that two partials are considered dissonant if the frequency difference is within approximately 5 to 50% of the bandwidth of the critical band in which the partials are situated. For reference, the critical bandwidth for a given frequency can be approximated by
with f and cb in Hz. Further, Plomp states that the human auditory system can not discriminate two partials if they differ in frequency by approximately less than five percent of the critical bandwidth in which they are situated. The exact transposition in Eq. 2 is approximated by
where Δfi is the deviation from the exact transposition. If the input partials form a harmonic series, a hypothesis of the invention states that the deviations from the harmonic series of the tranposed partials must not exceed five percent of the critical bandwidth in which they are situated. This would explain why prior art methods give unsatisfactory "harsh" and "rough" results, since broad band linear frequency shifts yields a much larger deviation than acceptable. When prior art methods produce more than one partial for only one input partial, the partials must nevertheless be within the above stated deviation limit, as to be perceived as one partial. This again explains the poor results obtained with prior art methods using nonlinearities etc, since they produce intermodulation partials not within the limit of deviation.
When using the above transposition based method of spectral replication according to the present invention, the following important properties are achieved:
Various ways to design the required transposers exist. Typical time-domain implementations expand the signal in time by duplicating signal segments based on the pitch-period. This signal is subsequently read out at a different rate. Unfortunately such methods are strictly dependent on pitch-detection for accurate time splicing of the signal segments. Furthermore, the constraint to work on pitch-period based signal segments makes them sensitive to transients. Since the detected pitch-period can be much longer than the actual transient, the risk of duplicating the entire transient rather than just expanding it in time is obvious. Another type of time domain algorithms obtains time expansion/compression of speech signals by utilising pattern search prediction of the output signal ["Pattern Search Prediction of Speech" R. Bogner, T. Li, Proc. ICASSP '89, Vol. 1, May 1989, "Time-Scale Modification of Speech based on a nonlinear Oscillator Model" G. Kubin, W. B. Kleijn, IEEE, 1994]. This is a form of granular synthesis, where the input signal is divided into small parts, granules, used to synthesise the output signal. This synthesis is usually done by perforating correlation of signal segments in order to determine the best splicing points. This means that the segments used to form the output signal are not dependent on the pitch period, and thus the nontrivial task of pitch detection is not required. Nevertheless, problems with rapidly changing signal amplitudes remain in these methods, and high quality transposition tends to raise high computational demands. However, an improved time-domain pitch shifter/transposer is now presented, where the use of transient detection and dynamic system parameters produces a more accurate transposition for high transposition factors during both stationary (tonal or non-tonal) and transient sounds, at a low computational cost.
Referring to the drawings wherein like numerals indicate like elements, there is shown in
In order to clarify the explanation, a state space representation is introduced. Here, the state vectors, or granules, represent the input and output signals. The input signal is represented by a statevector x(n):
x(n)=[x(n), x(n−D), x(n−2D), . . . , x(n−(N−1)D)] (5)
which is obtained from N delayed samples of the input signal, where N is the dimension of the state vector and D is the delay between the input samples used to build the vector. The granular mapping yields the sample x(n) following each statevector x(n−1). This gives Eq. 6, where a(•) is the mapping:
x(n)=a(x(n−1)). (6)
In the present method the granular mapping is used to determine the next output based on the former output, using a state transition codebook. The codebook of length L is continuously rebuilt containing the statevectors and the next sample following each statevector. Each statevector is separated from its neighbour by K samples; this enables the system to adjust the time resolution depending on the characteristics of the currently processed signal, where K equal to one represents the finest resolution. The input signal segment used to build the codebook is chosen based on the position of a possible transient and the synchronisation position in the previous codebook.
This means that the mapping a(•), theoretically, is evaluated for all transitions included in the codebook:
With this transition codebook, the new output y(n) is calculated by searching for the statevector in the codebook most similar to the current statevector y(n−1). This nearest-neighbour search is done by calculating the minimum difference and gives the new output sample:
y(n)=a(y(n−1)). (8)
However, the system is not limited to work on a sample by sample basis, but is preferably operated on a segment by segment basis. The new output segment is windowed and added, mixed, with the previous output segment, and subsequently down sampled. The pitch transposition factor is determined by the ratio of the input segment length represented by the codebook and the output segment length read out of the codebook.
Returning to the drawings, in FIG. 5 and
Sync_pos=Sync_pos_old+S�M−S, (9)
where Sync_pos and Sync_pos_old are the new and old synchronisation positions respectively, S is the length of the input segment being processed, and M is the transposition factor. This synchronisation point is used to compare the accuracy of the new splicing point with the accuracy of the old splicing point 603. If the match is as good as or better than the previous 605, this new synchronisation point is returned 607 provided it is within the codebook. If not, a new synchronisation point is searched for in the loop 609. This is performed with a similarity measure, in this case a minimum difference function 611, however, it is also possible to use correlation in the time- or frequency-domain. If the position yields a better match than that of the previous position found 613 the synchronisation position is stored 615. When all positions are tried 617 the system returns 619 to the flowchart in FIG. 5. The new synchronisation point obtained is stored 519 and a new segment is read out from the codebook 521 starting at the given synchronisation point. This segment is windowed and added to the previous 523, down sampled by the transposition factor 525, and stored in the output buffer 527.
Most pitch transposers, or time expanders, based on pattern search prediction give satisfactory results for speech and single-pitched material. However, their performance deteriorates rapidly for high complexity signals, like music, in particular at large transposition factors. The present invention offers several solutions for improved performance, therefore producing excellent results for any type of signal. Contrary to other designs, the system is time-variant and the system parameters are based on the properties of the input signal and the parameters used during the previous operation cycle. The use of a transient detector controlling not only the codebook size and position, but also the properties of the statevectors included, is a very robust and computationally efficient method to avoid audible degradation during rapidly changing signal segments. Furthermore, alteration of the length of the signal segment being processed, which would raise higher computational demands, is not required. Also, the present invention utilises a refined codebook search based on the results from the preceding search This means that contrary to an ordinary correlation of two signal segments, as is usually done in time-domain systems based on pattern search prediction, the most likely synchronisation positions are tried first instead of trying all positions consecutively. This new method for reducing the codebook search drastically reduces the computational complexity of the system. Further, when using several transposers, synchronisation position information can be shared among the transposers for further reduction of the computational complexity, as shown in the following implementation.
The time-domain transposers as explained above are used to implement the SBR-1 and SBR-2 systems according to the following, illustrative but not limiting, example. In
sync_pos2=sync_pos4−n�4�S−sync_offset, for n=1,2,3,4 . . . , (10)
sync_offset=sync_pos4−sync_pos2, for n=0, (11)
and S is the length of the input segment represented by the codebook. This is valid as long as neither of the synchronisation position pointers reaches the end of the codebook. During normal operation n is increased by one for each time-frame processed by the second order harmonic transposer, and when the codebook end inevitably is reached, by either of the pointers, the counter n is set to n=0, and sync_pos2 and sync_pos4 are computed individually. Similar results are obtained for the third order harmonic transposer when connected to the fourth order harmonic transposer.
The above-presented use of several interconnected time-domain transposers, for the creation of higher order harmonics, introduces substantial computational reduction. Furthermore, the proposed use of time-domain transposers in connection with a suitable filterbank, presents the opportunity to adjust the envelope of the created spectrum while maintaining the simplicity and low computational cost of a time domain transposer, since these, more or less, may be implemented using fixed point arithmetic and solely additive/subtractive-operations.
Other, illustrative but not limiting, examples of the present invention are:
It should be recognised that the method outlined above may be advantageously used for timescale modification only, by simply omitting the sample rate conversion. Further it is understood, that although the outlined method focuses on pitch transposing to a higher pitch, i.e. time expansion, the same principles apply when transposing to a lower pitch, i.e. time compression, as is obvious to those skilled in the art.
Filter Bank Based Transposition
Various new and innovative filter bank based transposition techniques will now be described. The signal to be transposed is divided into a series of BP- or subband signals. The subband signals are then transposed, exact or approximately, which is advantageously accomplished by a reconnection of analysis- and synthesis subbands, hereinafter referred to as a "patch". The method is first demonstrated using a Short Time Fourier Transform, STFT.
The N-point STFT of a discrete-time signal x(n) is defined by
where k=0,1, . . . ,N−1 and ωk=2πk/N and h(n) is a window. If the window satisfies the following conditions
an inverse transform exists and is given by
The direct transform may be interpreted as an analyser, see
This yields
where M=2. Eq. 15 may be interpreted as a BP-filtering of the input signal, followed by a linear frequency shift or Upper Side Band (USB) modulation, i.e. single side band modulation using the upper side band, see
According to Eq. 15, a sinusoid with the frequency ωi within the passband of analysis channel k yields a harmonic at the frequency Mωk+(ωi−ωk). Hence the method, referred to as basic multiband transposition, only generates exact harmonics for input signals with frequencies ωi=ωk, where 4≦k≦7. However, if the number of filters is sufficiently large, the deviation from an exact transposition is negligible, see Eq. 4. Further, the transposition is made exact for quasi-stationary tonal signals of arbitrary frequencies by inserting the blocks denoted P 917 (
The harmonic patch of
It is also possible to combine amplitude and phase information from different analyzer channels. The amplitude signals |Xk(rR)| may be connected according to
In the above description it was assumed that the highest frequency contained in the input signal was significantly lower than the Nyqvist frequency. Thus, it was possible to perform a bandwidth expansion without an increase in sample rate. This is however not always the case, why a preceding upsampling may be necessary. When using filter bank methods for transposition, it is possible to integrate upsampling in the process.
Most perceptual codecs employ maximally decimated filter banks in the time to frequency mapping ["Introduction to Perceptual Coding" K Brandenburg, AES, Collected Papers on Digital Audio Bitrate Reduction, 1996].
In the illustrative, but not limiting, descriptions below it is assumed that an L-channel cosine modulated filter bank splits the input signal x(n) into L subband signals. The generic structure of a maximally decimated filter bank is shown in FIG. 19. The analysis filters are denoted Hk(z) 1901, where k=0, 1, . . . , L−1. The subband signals vk(n) are maximally decimated 1903, each of sampling frequency fs/L, where fs is the sampling frequency of x(n). The synthesis section reassembles the subband signals after interpolation 1905 and filtering 1907 to produce {circumflex over (x)}(n). The synthesis filters are denoted Fk(z). In addition, the present invention performs a spectral replication on {circumflex over (x)}(n), giving an enhanced signal y(n).
Synthesising the subband signals with a QL-channel filter bank, where only the L lowband channels are used and the bandwidth expansion factor Q is chosen so that QL is an integer value, will result in an output bit stream with sampling frequency Qfs. Hence, the extended filter bank will act as if it is an L-channel filter bank followed by an upsampler. Since, in this case, the L(Q−1) highband filters are unused (fed with zeros), the audio bandwidth will not change—the filter bank will merely reconstruct an upsampled version of {circumflex over (x)}(n). If, however, the L subband signals are patched to the highband filters, the bandwidth of {circumflex over (x)}(n) will be increased by a factor Q, producing y(n). This is the maximally decimated filter bank version of the basic multiband transposer, according to the invention. Using this scheme, the upsampling process is integrated in the synthesis filtering as explained earlier. It should be noted that any size of the synthesis filter bank may be used, resulting in different sample-rates of the output signal, and hence different bandwidth expansion factors. Performing spectral replication on {circumflex over (x)}(n) according to the present invention of the basic multiband transposition method with an integer transposition factor M, is accomplished by patching the subband signals as
v Mk(n)=e Mk(n)(−1)(M-1)kn v k(n), (16)
where k∈[0,L−1] and chosen so that Mk∈[L,QL−1], eMk(n) is the envelope correction and (−1)(M-1)kn is a correction factor for spectral inverted subbands. Spectral inversion results from decimation of subband signals, and the inverted signals may be reinverted by changing sign on every second sample in those channels. Referring to
Using the basic multiband transposition method according to the present invention, the generated harmonics are in general not exact multiples of the fundamentals. All frequencies but the lowest in every subband differs in some extent from an exact transposition. Further, the replicated spectrum contains zeros since the target interval covers a wider frequency range than the source interval. Moreover, the alias cancellation properties of the cosine modulated filter bank vanishes, since the subband signals are separated in frequency in the target interval. That is, neighbouring subband signals do not overlap in the highland area. However, aliasing reduction methods, known by those skilled in the art, may be used to reduce this type of artifacts. Advantages of this transposition method are ease of implementation, and the very low computational cost.
To achieve perfect transposition of sinusoids, an effective maximally decimated filter bank solution of the improved multiband transposition method is now presented. The system uses an additional modified analysis filter bank, while the synthesis filter bank is cosine modulated as described by Vaidyanathan ["Multirate Systems and Filter Banks" P. P. Vaidyanathan, Prentice Hall, Englewood Cliffs, N.J., 1993, ISBN 0-13-605718-7]. The steps for operation, using the improved multiband transposition method according to the present invention, based on maximally decimated filter banks are shown schematically in FIG. 21 and in the flow-chart of FIG. 22 and are as follows:
Steps 3 to 6 may be repeated for different values of the transposition factor U, thus adding multiple harmonics to x1(n). This mode of operation is illustrated by the dotted figures of
for every applicable k. In the first iteration of the loop of
s k(n″)=s k(n″)+s k (M
where k=K/Q, K/Q+1, . . . , min(K,Ti)−1. The subband signals sk(n″) are synthesised once with a K-channel filter bank according to step 7.
The modified analysis filter bank of step 4, is derived through the theory of cosine modulated filter banks, where the modulated lapped transform (MLT) ["Lapped Transforms for Efficient Transform/Subband Coding" H. S. Malvar, IEEE Trans ASSP, vol. 38, no. 6, 1990] is a special case. The impulse responses hk(n) of the filters in a T-channel cosine modulated filter bank may be written
where k=0, 1, . . . , T−1, N is the length of the lowpass prototype filter p0(n), C is a constant and Φk is a phase-angle that ensures alias cancellation between adjacent channels. The constraints on Φk is
which may be simplified to the closed form expression
With this choice of Φk, perfect reconstruction systems or approximate reconstruction systems (pseudo QMF systems) may be obtained using synthesis filter banks with impulse responses as
Consider the filters
where h′k(n) are sine-modulated versions of the prototype filter p0(n). The filters H′k(z) and Hk(z) have identical passband supports, but the phase responses differ. The passbands of the filters are actually Hilbert transforms of each other (this is not valid for frequencies close to ω=0 and ω=π). Combining Eq. 19 and Eq. 23 according to
yields filters that have the same shape of the magnitude responses as Hk(z) for positive frequencies but are zero for negative frequencies. Using a filter bank with impulse responses as in Eq. 24 gives a set of subband signals that may be interpreted as the analytic (complex) signals corresponding to the subband signals obtained from a filter bank with impulse responses as in Eq. 19. Analytic signals are suitable for manipulation, since the complex-valued samples may be written in a polar form, that is z(n)=r(n)+j i(n)=|z(n)|exp{j arg(z(n))}. However, when using the complex filter bank for transposition, the constraint on Φk has to be generalised to retain the alias cancellation property. The new constraint on Φk, to ensure alias cancellation in combination with a synthesis filter bank with impulse responses as in Eq. 22 is
which simplifies to Eq. 21 when M=1. With this choice, transposed partials will have the same relative phases as they would have when M=1 (no transposition).
Combining Eq. 24 and Eq. 25 results in
which are the filters used in the modified filter bank of step 4, according to the present invention.
Some clarifications concerning step 5: downsampling the complex-valued subband signals by a factor T/M makes them oversampled by M, which is an essential criterion when the phase-angles subsequently are multiplied by the transposition factor M. The oversampling forces the number of subband samples per bandwidth, after transposition to the target range, to equal that of the source range. The individual bandwidths of the transposed subband signals are M times greater than those in the source range, due to the phase-multiplier. This makes the subband signals critically sampled after step 5, and additionally, there will be no zeros in the spectrum when transposing tonal signals.
In order to avoid trigonometric calculations, that is, having to compute the new subband signals as
where |vk (M)(n″)| is the absolute value of vk (M)(n″), the following trigonometric relationship is used:
cos(Mα)=cosM(α)−(2 M)sin2(α)cosM-2(α)+(4 M)sin4(α)cosM-4(α) (28)
and noting that
the computations of step 5 may be accomplished without trigonometric calculations, reducing computational complexity.
When using transpositions where M is even, obstacles with the phase-multiplier may arise, depending on the characteristics of the lowpass prototype filter p0(n). All applicable prototype filters have zeros on the unit circle in the z-plane. A zero on the unit circle imposes a 180� shift in the phase response of the filter. For M even, the phase-multiplier translates thee shifts to 360� shifts; i.e. the phase-shifts vanish. The partials so located in frequency that such phase-shits vanish will give rise to aliasing in the synthesised signal. The worst case scenario is when a partial is located at a point in frequency corresponding to the top of the first side lobe of an analysis filter. Depending on the rejection of this lobe in the magnitude response, the aliasing will be more or less audible. As an example, the first side lobe of the prototype filter used in the ISO/MPEG layer 1 and 2 standard is rejected 96 dB, while the rejection is only 23 dB for the first side lobe of the sine-window used in the MDCT scheme of the ISO/MPEG layer 3 standard. It is clear, that this type of aliasing, using the sine-window, will be audible. A solution to this problem will be presented, and is referred to as relative phase locking.
The filters ha k(n) all have linear phase responses. The phase-angles Φk introduce relative phase differences between adjacent channels, and the zeros on the unit circle introduce 180� phase-shifts at locations in frequency that may differ between channels. By monitoring the phase-difference between neighbouring subband signals, before the phase-multiplier is activated, it is easy to detect the channels that contain phase-inverted information. Considering tonal signals, the phase-difference is approximately π/2M, according to Eq. 25, for non-inverted signals, and consequently approximately π(1−1/2M) for signals, where either of the signals is inverted. The detection of inverted signals may be accomplished simply by computing the dot product of samples in adjacent subbands as
If the product in Eq. 32 is negative, the phase-difference is greater than 90�, and a phase-inversion condition is present. The phase-angles of the complex-valued subband signals are multiplied by M, according to the scheme of step 5, and finally, the inversion-tagged signals are negated. The relative phase locking method thus forces the 180� shifted subband signals to retain this shift after the phase-multiplication, and hence maintain the aliasing cancellation properties.
Spectral Envelope Adjustment
Most sounds, like speech and music, are characterised as products of slowly varying envelopes and rapidly varying carriers with constant amplitude, as described by Stockham ["The Application of Generalized Linearity to Automatic Gain Control" T. G. Stockham, Jr, IEEE Trans. on Audio and Electroacoustics, Vol. AU-16, No. 2, June 1968] and Eq. 1.
In split-band perceptual audio coders, the audio signal is segmented into frames and split into multiple frequency bands using subband filters or a time-to-frequency domain transform. In most codec types, the signal is subsequently separated into two major signal components for transmission or storage, the spectral envelope representation and the normalised subband samples or coefficients. Throughout the following description the term "subband samples" or "coefficients" refers to sample values obtained from subband filters as well as coefficients obtained from a time-to-frequency transform. The term "spectral envelope" or "scale factors" represent values of the subbands on a time-frame basis, such as the average or maximum magnitude in each subband, used for normalisation of the subband samples. However, the spectral envelope may also be obtained using linear prediction LPC, [U.S. Pat. No. 5,684,920]. In a typical codec, the normalised subband samples require coding at a high bitrate (using approximately 90% of the available bitrate), compared to the slowly varying temporal envelopes, and thus the spectral envelopes, that may be coded at a much-reduced rate (using approximately 10% of the available bitrate).
Accurate seal envelope of the replicated bandwidth is important if the timbral qualities of the original signal are to be preserve. The perceived timbre of a musical instrument, or voice, is mainly determined by the spectral distribution below a frequency ftim, located in the highest octaves of hearing. The spectral details above ftim are thus of less importance, and consequently the highband fine structures obtained by the above transposition methods require no adjustment, while the coarse structures generally do. In order to enable such adjustment, it is useful to filter the spectral representation of the signal to separate the envelope coarse structure from the fine structure.
In the SBR-1 implementation according to the present invention, the highband coarse spectral envelope is estimated from the lowband information available at the decoder. This estimation is performed by continuously monitoring the envelope of the lowband and adjusting the highband spectral envelope according to specific rules. A novel method to accomplish the envelope estimation uses asymptotes in a logarithmic frequency-magnitude space, which is equivalent to curve fitting with polynomials of varying order in the linear space. The level and slope of an upper portion of the lowband spectrum are estimated, and the estimates are used to define the level and slope of one or several segments representing the new highband envelope. The asymptote intersections are fixed in frequency and act as pivot points. However not always necessary, it is beneficial to stipulate constraints to keep the highband envelope excursions within realistic boundaries. An alternative approach to estimation of the spectral envelope is to use vector quantization, VQ, of a large number of representative spectral envelopes, and store these in a lookup-table or codebook. Vector quantization is performed by training the desired number of vectors on a vast amount of training data, in this case audio spectral envelopes. The training is usually done with the Generalised Lloyd Algorithm ["Vector Quantization and Signal Compression" A. Gersho, R. M. Gray, Kluwer Academic Publishers, USA 1992, ISBN 0-7923-9181-0], and yields vectors that optimally cover the contents of the training data. Considering a VQ codebook consisting of A spectral envelopes trained by B envelopes (B>>A), then the A envelopes represent the A most likely transitions from the lowband envelope to the highband envelope, based on B observations of a wide variety of sounds. This is, theoretically, the A optimum rules for predicting the envelope based on the B obserations. When estimating a new highband spectral envelope, the original lowband envelope is used to search the codebook and the highband part of the best matching codebook entry is applied to create the new highband spectrum.
In some codecs, it is possible to transmit die scalefactors for the entire spectral envelope while omitting the highband subband samples, as shown in FIG. 24. Other codec standards stipulate that scalefactors and subband samples must cover the same frequency range, i.e. scale-factors cannot be transmitted if the subband samples are omitted. In such cases, there are several solutions; the highband spectral envelope information can be transmitted in separate frames, where the frames have their own headers and optional error protection, followed by the data. Regular decoders, not taking advantage of the present invention, will not recognise the headers and therefore discard the extra frames. In a second solution, the highband spectral envelope information is transmitted as auxiliary data within the encoded bitstream. However, the available auxiliary data field must be large enough to hold the envelope information. In cases where none of the first two solutions are adaptable, a third solution, where the highband spectral envelope information is hidden as subband samples, may be applied. Subband scalefactors cover a large dynamic range, typically exceeding 100 dB. It is thus possible to set an arbitrary number of subband scalefactors, 2505 in
Transient Response Improvements
Transient related artifacts are common problems in audio codecs, and similar artifacts occur in the present invention. In general, patching generates spectral "zero" or notches, corresponding to time domain pre- and post-echoes, i.e. spurious transients before and after "true" transients. Albeit the P-blocks "fill in the zeros" for slowly varying tonal signals, the pre- and post-echoes remain. The improved multiband method is intended to work on discrete sinusoids, where the number of sinusoids is restricted to one per subband. Transients or noise in a subband can be viewed as a large number of discrete sinusoids within that subband. This generates intermodulation distortion. These artifacts are considered as additional quantization-noise sources connected to the replicated highband channels during transient intervals. Traditional methods to avoid pre- and post-echo artifacts in perceptual audio coders, for example adaptive window switching, may hence be used to enhance the subjective quality of the improved multiband method. By using the transient detection provided by the codec or a separate detector and reducing the number of channels under transient conditions the "quantization noise" is forced not to exceed the time-dependent masking threshold. A smaller number of channels is used during transient passages whereas a larger is used during tonal passages. Such adaptive window switching is commonly used in codecs in order to trade frequency resolution for time resolution. Different methods may be used in applications where the filterbank size is fixed. One approach is to shape the "quantization noise" in time via linear prediction in the spectral domain. The transposition is then performed on the residual signal, which is the output of the linear prediction filter. Subsequently, an inverse prediction filter is applied to the original- and spectral replicated channels simultaneously. Another approach employs a compander system i.e. dynamic amplitude compression of the transient signal prior to transposition or coding, and a complementary expansion after transposition. It is also possible to switch between transposition methods in a signal dependent manner, for example, a high resolution filterbank transposition method is used for stationary signals, and a time-variant pattern search prediction method is employed for transient signals.
Practical Implementations
Using a standard signal-processor or a powerful PC, real-time operation of a SBR-enhanced codec is possible. The SBR enhanced codec may also be hard-coded on a custom chip. It may also be implemented in various kinds of systems for storage or transmission of signals, analogue or digital, using arbitrary codecs, FIG. 27 and FIG. 28. The SBR-1 method may be integrated in a decoder or supplied as an add-on hardware or software post-processing module. The SBR-2 method needs additional modification of the encoder. In
The SBR-2 method needs additional modification of the encoder. In
When only very low bitrates are available, (Internet and slow telephone modems, AM-broadcasting etc.) mono coding of the audio program material is unavoidable. In order to improve the perceived quality and make the programme more pleasant sounding, a simple "pseudo-stereo" generator,
The above-described embodiments are merely illustrative for the principles of the present invention for audio source coding improvement. It is understood that modifications and variations of the arrangements and the details described herein will be apparent to others skilled in the art. It is the intent, therefore, to be limited only by the scope of the impending patent claims and not by the specific details presented by way of description and explanation of the embodiments herein.
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|Apr 3, 2006
|Apr 19, 2007
|Vos Koen B
|Systems, methods, and apparatus for speech signal filtering
|Oct 19, 2006
|May 31, 2007
|Lg Electronics Inc.
|Apparatus and method for transmitting and receiving data in a mobile communication terminal
|Jun 15, 2006
|Jul 5, 2007
|Quanta Computer Inc.
|Subband synthesis filtering process and apparatus
|Apr 3, 2006
|May 29, 2008
|Qualcomm Incorporated
|Systems, methods, and apparatus for gain coding
|May 17, 2005
|Oct 23, 2008
|Masahiro Oshikiri
|Encoding Device, Decoding Device, and Method Thereof
|Aug 31, 2007
|Nov 13, 2008
|Japan Advanced Institute Of Science And Technology
|Speech processing method and apparatus, storage medium, and speech system
|Sep 25, 2007
|Nov 20, 2008
|Fujitsu Limited
|Decoding apparatus and decoding method
|May 9, 2007
|Jul 2, 2009
|Panasonic Corporation
|Encoding apparatus and encoding method
|Mar 29, 2007
|Oct 1, 2009
|Panasonic Corporation
|Sound encoder, sound decoder, and their methods
|Jun 27, 2007
|Nov 12, 2009
|Nxp B.V.
|Noise synthesis
|Sep 28, 2011
|Apr 19, 2012
|Motorola Mobility, Inc.
|Audio signal bandwidth extension in celp-based speech coder
|Sep 28, 2011
|Apr 19, 2012
|Motorola Mobility, Inc.
|Audio signal bandwidth extension in celp-based speech coder
|Sep 2, 2011
|May 3, 2012
|Samsung Electronics Co., Ltd.
|Speaker adaptation method and apparatus
|Feb 8, 2013
|Jun 13, 2013
|Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V.
|Resampling output signals of qmf based audio codecs
|Mar 29, 2012
|Oct 3, 2013
|General Electric Company
|Amplitude enhanced frequency modulation
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Well I just emailed my boss. It's an ice skating rink here and I'm not
about to become a statistic.
It's foggy too. Forget about it. That's what personal days are for. And I
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text: Meta Battle Subway PokeBase - Pokemon Q&A Login Register All Activity Questions Unanswered Tags Users Ask a Question Rules Chat Quicker way to get spore? 0 votes 592 views Foongus learns spore at lvl 50, but Amoongus learns it at LEVEL 62, WHY. So basically, if I level grind at E4, will it be quicker to get spore if (I have exp point power levels up to 3 btw) if Foongus holds a lucky egg or an everstone? eggs spore foongus asked Apr 12, 2014 by Hairy Shed retagged Apr 12, 2014 by Sempiternus it has always been like this, pokemon learn moves that both forms get at lower levels than if they're in their evolved forms. you have to choose between power or learning a move earlier. there are also moves that you may miss if you wait to evolve or may miss out on if you evolve early. commented Aug 9, 2014 by Meewec Please log in or register to add a comment. Please log in or register to answer this question. 2 Answers 1 vote That other answer was a little cryptic, so let me help you out. It's easy to figure out which one's faster if you do the math. Let's say you have a level 1 Foongus with an Everstone. To reach level 50, it must gain 125,000 exp. points. To get Amoonguss to level 62, it must gain 238,328 exp. points, which is more, except you have the lucky egg, which makes it get experience at 1.5x the normal rate. So to compare the two, let's divide it: 238328 / 1.5 = 158,885 So even with the Lucky Egg, it would still take longer. So to answer your question, it takes less time if you leave it as a Foonguss. Oh, but there's more! You can give your Foonguss a Lucky Egg and mash the B button every time it tries to evolve. That way, it can reach level 50 at a comparative 83,333 exp. points! Of course, canceling the evolution every time it levels up can be a pain, so it's your choice. And as a side note, going to Cafe le Wow is a ton faster than the Elite Four if you want to level stuff up fast (assuming you're playing XY). Hope I helped! Source answered Aug 9, 2014 by Pimplup Please log in or register to add a comment. 0 votes One word. Yes Yes it will be quicker. Just dont' forget to spam dat B when evolving o3o! Or have everstone. answered Apr 12, 2014 by Sempiternus Please log in or register to add a comment. Related questions What happens when you evolve later? Is there any way to get Spore on my Breloom? Can a steel type get poisoned by Effect Spore? Does Spore affect grass types? Whats the fastest non uber that can learn spore? Does the move Spore make a pokemon uber? what is the mysterey behind foongus pokeball shaped cap Is there a way to make eggs hatch quicker in gen 5? Useful Meta Posts Vote for Sun/Moon moveset questions How to change your username How to add a user avatar Why was my post hidden? The points system Question etiquette Flags vs Downvotes Anime FAQ


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A diamond in the rough
My eyelids fluttered open, allowing the intense sunlight to seep into them. Dancing amongst the furniture, the rays of light lingered in my dusty, cob webbed filled, attic bedroom. For a moment, I felt free and able to lie in bed all day until I realised what day today was. Saturday. For most people, it was a day for them to put their feet up, but such pleasures were reserved for the wealthy. So this morning I would have to slave away to prepare the master and mistress of this immaculate manor for pretty much anything. The things a scullery maid has to do these days.
Suddenly aware of fatigue (I was up late last night, due to a surprise visit from some of the Lord and Lady's posh friends), I splashed my olive brown, freckle nosed face with icy water. As it ran down my cheeks, I began to drag the comb through my mass of red hair. To be honest, a rake would've been better, but for health and safety purposes, I thought better of it. Anyway, there I was, attempting to look remotely human, when I heard a (very) pitchy voice shouting, "Eva! I need some help with the breakfast!"
"Coming, ma'm!" I half-heartedly replied. Mercy, the head maid, was not the kind of woman to be left waiting. Unless you wanted to be turned inside out and upside down. Believe me, I've been there already. You do not want to go there. So I, Eva Faith, would be sent on a voyage of discovery downstairs. . .


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|Message view
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|"Bill Barker" <wbar...@wilshire.com>
|[PROPOSAL] Release the HEAD of j-t as Tomcat 3.3.2
|Mon, 09 Feb 2004 02:48:42 GMT
It has been a long time since the 3.3.1 release, and there are several important bug fixes in the CVS. Therefore, I'm proposing cutting a 3.3.2 release with myself as RM (although I'll step aside if Larry really wants the job :). Time Line: This proposal will run until Thursday, Feb 12 (I was going to do Friday, until I noticed that the date was the 13th :) 6:00 PM PDT. At that point it will become a [VOTE] that will last until Thursday, Feb 19 6:00 PM PDT. If the [VOTE] passes with 3 PMC +1s and no -1s and the enhancements below are complete (with the possible exception of licensing), then we cut a 3.3.2-RC1 release. I don't see that it is likely that we have any major regressions, so I don't see the need for a Beta. I propose to leave RC1 out for two weeks until Friday, Mar 5 (March 4th isn't good for me). At that point we cut a 3.3.2 release. Packaging Changes: I propose that we no longer ship src/native with the src distro (unless it is too hard to modify Larry's scripts, in which case we include a do-not-use in the release notes). I'd also like to include the commons-daemon procrun/w and jsvc (source) with the release, similar to TC 5. Also I'd like to make the CoyoteConnector2 (possible renamed to CoyoteConnector :) the default connector in server.xml. We can add a note to use the old Connectors for people that can't use Java2. Code Changes: 1) I'd like to add a blocking option to o.a.t.startup.Main (e.g. 'startb') so that 3.3 can be used with procrun. Also make sure that the init/start/stop/destroy semantics are there for jsvc. 2) Add fix for Bug #8643 (allow to specify alternate module.xml files) 3) Port the requestURI encoding fixes for TC 4/5 (although, I believe that this is on DecodeInterceptor, instead of the Connector). 4) Update license to Apache-2.0 (non-negotiable, with a March release date :). Bugs: Several of the LATER bugs have been already fixed, so mark them as such. Of the remaining 18 open bugs, they will be marked as one of INVALID/FIXED/WORKSFORME/LATER. Comments/Suggestions/Flames?
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International Centre
this wiki
Stadium Cleanup
To meet Pro Wrestling Wiki's quality standards, this Stadium article or section may require cleanup. Please help by improving the article.
The International Centre is a multi-purpose convention centre in Mississauga, Ontario, Canada. It is located at 6900 Airport Road, near the Toronto Pearson International Airport. The building was opened in 1972 by DeHavilland. Today, the major multi-purpose facility hosts some of Canada's largest and popular events.
The International Centre is one of the few privately owned convention facilities in Canada. Opened in 1972, the centre's exhibit space nearly doubled from the original 260,000 square feet (24,000 m2) following a major expansion and renovation in 2002. The centre now features over 500,000 square feet (46,000 m2) of exhibit space and approximately 1,000,000 square feet (93,000 m2) of gross space which includes exhibit, meeting, office and retail space paying host to a variety of festivities annually.
Roots of the building stem back from Canada's aviation history during the 1950s when the site occupied the production of wings, rear fuselages and empennages to service DC-9 airliners. The area was purchased in 1971 from DeHavilland by a group of investors which eventually led to its purpose as a prominent multi-purpose facility.


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Sexual sensitivity with anchor procedure; is this my only option?
I am 4'10" and my cup size is J-K. My breasts are very heavy. My PS says because of the amount he has to remove to get to a C-D range, he would have to detach the pedicle and do a FNG, which would leave me with no sensitivity. Is this my only option?
Doctor Answers 7
Not necessary
I agree with Dr. Rosenblatt. I rarely do free nipple grafts on patients like you and almost all patients can get a pedicle type procedure, with excellent result. See a plastic surgeon with at least 15 years of experience in these cases. I'm sure you'll do well.Hope this helps.
Leland Deane MD FACSNYC, Garden City, Babylon NY
Leland Deane MD FACSNYC, Garden City, Babylon NY
Breast reduction
Most of the time a breast reduction can be performed leaving the nipple on a pedicle and most of the time the nipple is sensate after surgery. I rarely have to resort to a free nipple graft, but if this is required, sensation of the nipple will be lost.
Web reference: https://stevenwallachmd.com/breast/breast-reduction/
Sexual sensitivity with anchor procedure; is this my only option?
I'm afraid that is your only option. It is not commonly done, but in your case that is the only way your plastic surgeon can reduce your breasts to a size proportional to your body. Maintaining a pedicle and sensation to the nipple would only reduce you to 2-3 cup sizes smaller, and your breasts will still be too large for your body, and you will continue to have your back and shoulder symptoms.
You might also like...
Free nipple grafts
Most patients with your shape and dimension can be treated without FNG, but without a personal examination and consultation, it's not possible for anyone on this site to say this for sure. Your best bet is to have another opinion or two. If everyone seems to think you need the FNG, then odds are that's your only chance.
Free Nipple Graft
No need to do a Free Nipple Graft based on your dimension. I recommend an in-office examination as well as a detailed discussion with a surgeon who you are comfortable with and who is a board-certified Plastic Surgeon certified by the American Board of Plastic Surgery. Best wishes! Dr. Desai
Harvard Educated, Beverly Hills & Miami Beach Trained, Double-Board Certified Plastic Surgeon
Harvard Educated, Beverly Hills & Miami Beach Trained, Double-Board Certified Plastic Surgeon
Web reference: http://www.desaiplasticsurgery.com
Get a second opinion. In the over 30 years I have been doing reductions, I have rarely done FNGs and in those cases the removal of breast tissue was over 5-7 pounds/breast. Those are the Largest reductions. I don't know how big you are because you need to be examined before giving you a final opinion.
Sexual sensitivity with anchor procedure; is this my only option?
I would suggest, before you commit, that you seek additional consultations with board-certified plastic surgeons. Free nipple grafting, although sometimes necessary, is not necessary in the vast majority of cases. Best wishes.
Web reference: http://www.poustiplasticsurgery.com/Procedures/Procedure_breastReduction.htm
These answers are for educational purposes and should not be relied upon as a substitute for medical advice you may receive from your physician. If you have a medical emergency, please call 911. These answers do not constitute or initiate a patient/doctor relationship.


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Meite for some reason looks clunky and her muscles scare me - kind of like the william sisters - there is something awkward or aesthetically unpleasing about the look. I think Yuna is like Kwan they are perfection but in the end they just skate t whatever music in the same style and moves - perfect but rather perfunctory. Virtue and Moir despite their flaws had picked different characters and styles. Which is why i put Davis and white on this lilst too. Really and this is not an insult they are perfect disney chara\cters on ice.
Gracie Gold, for the aforementioned racist personality she exhibited. I spent five years of my childhood in suburbs of Chicago. There is plenty of opportunity to learn diversity there and eighteen is old enough to know better, really. She also moves like a truck driver and is ungraceful physically.


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I've had a Bone Graft to build up the area for an implant. My Bone Graft was 5 years ago, would an implant still work? (photo)
I was born without one lateral incisor. I've had braces to close gaps and make room for implant. My bone graph was done 5 years ago. Is the implant still the beast way to go? There is no other problems with any teeth or gum issues.
Doctor Answers 3
Bone graft lasting
It is hard to say where you stand from the photo alone. Bone grafts as well as natural bone tend to resorb over time if not used. Sometimes this can occur rapidly and sometimes very slowly. An x-ray and/or CT scan would be needed to evaluate the amount of bone available at this time to support an implant. IF there is not enough bone present, additional bone grafting can always be performed.
Bone Graft
Bone grafting will last a long time. However, 5 yrs. maybe too long to assume all the bone is still present. Have the site evaluated to see how much bone is available. If there is, an implant can be your best choice. Once implant is placed, the bone should remain. An Implant will help you avoid having the natural teeth on either side ground down for a bridge.
Yes , to place an implant is your best solution , and its a very easy proceedure ,to be performed.you are still young and you must search to keep that area healthy and good
,try to set the dental implant as soon as possible ,so that way you will preserve the height of that bone ,where the tooth was extracted.
,try to set the dental implant as soon as possible ,so that way you will preserve the height of that bone ,where the tooth was extracted.
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These answers are for educational purposes and should not be relied upon as a substitute for medical advice you may receive from your physician. If you have a medical emergency, please call 911. These answers do not constitute or initiate a patient/doctor relationship.