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The Inevitable Rise and Liberation of NiggyTardust!

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The Inevitable Rise and Liberation of NiggyTardust! is the third solo studio album by Saul Williams . It was released in 2007. It peaked at number 41 on the Billboard Heatseekers Albums chart, as well as number 89 on the Top R&B/Hip-Hop Albums chart. The album is entirely produced by Trent Reznor . The title of the album is a reference to David Bowie 's 1972 album The Rise and Fall of Ziggy Stardust and the Spiders from Mars .

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52-486: The album was available for purchase or free download at NiggyTardust.com. The website allowed users to pay $ 5 to support the artist and be given the choice of downloading a 192 kbit/s MP3 version, 320 kbit/s MP3 version or lossless FLAC version. Digital distribution of the album is provided by Musicane . Reznor publicised the album on the Nine Inch Nails website and mailing list, saying that "Saul's not

104-558: A bandpass signal is sampled slower than its Nyquist rate , the samples are indistinguishable from samples of a low-frequency alias of the high-frequency signal. That is often done purposefully in such a way that the lowest-frequency alias satisfies the Nyquist criterion , because the bandpass signal is still uniquely represented and recoverable. Such undersampling is also known as bandpass sampling , harmonic sampling , IF sampling , and direct IF to digital conversion. Oversampling

156-457: A moiré pattern . The process of volume rendering samples a 3D grid of voxels to produce 3D renderings of sliced (tomographic) data. The 3D grid is assumed to represent a continuous region of 3D space. Volume rendering is common in medical imaging, X-ray computed tomography (CT/CAT), magnetic resonance imaging (MRI), positron emission tomography (PET) are some examples. It is also used for seismic tomography and other applications. When

208-512: A CD. If a CD is read and ripped perfectly to FLAC files, the CUE file allows later burning of an audio CD that is identical in audio data to the original CD, including track order and pregap , but excluding additional data such as lyrics and CD+G graphics. But depending on the burning program used, CD-Text may be recovered from the metadata stored in the CUE sheet and burned back to a new copy on blank CD-R media. The reference implementation of FLAC

260-413: A Nyquist rate of B {\displaystyle B} , because all of its non-zero frequency content is shifted into the interval [ − B / 2 , B / 2 ] {\displaystyle [-B/2,B/2]} . Although complex-valued samples can be obtained as described above, they are also created by manipulating samples of a real-valued waveform. For instance,

312-417: A few GHz, and may be prohibitively expensive at much lower frequencies. Furthermore, while oversampling can reduce quantization error and non-linearity, it cannot eliminate these entirely. Consequently, practical ADCs at audio frequencies typically do not exhibit aliasing, aperture error, and are not limited by quantization error. Instead, analog noise dominates. At RF and microwave frequencies where oversampling

364-427: A much lower rate. For most phonemes , almost all of the energy is contained in the 100 Hz – 4 kHz range, allowing a sampling rate of 8 kHz. This is the sampling rate used by nearly all telephony systems, which use the G.711 sampling and quantization specifications. Standard-definition television (SDTV) uses either 720 by 480 pixels (US NTSC 525-line) or 720 by 576 pixels (UK PAL 625-line) for

416-404: A point in time and/or space; this definition differs from the term's usage in statistics , which refers to a set of such values. A sampler is a subsystem or operation that extracts samples from a continuous signal . A theoretical ideal sampler produces samples equivalent to the instantaneous value of the continuous signal at the desired points. The original signal can be reconstructed from

468-438: A proposed nonlinear function . Digital audio uses pulse-code modulation (PCM) and digital signals for sound reproduction. This includes analog-to-digital conversion (ADC), digital-to-analog conversion (DAC), storage, and transmission. In effect, the system commonly referred to as digital is in fact a discrete-time, discrete-level analog of a previous electrical analog. While modern systems can be quite subtle in their methods,

520-429: A sequence of samples, up to the Nyquist limit , by passing the sequence of samples through a reconstruction filter . Functions of space, time, or any other dimension can be sampled, and similarly in two or more dimensions. For functions that vary with time, let s ( t ) {\displaystyle s(t)} be a continuous function (or "signal") to be sampled, and let sampling be performed by measuring

572-597: A theoretical reconstruction is a common measure of the effectiveness of sampling. That fidelity is reduced when s ( t ) {\displaystyle s(t)} contains frequency components whose cycle length (period) is less than 2 sample intervals (see Aliasing ). The corresponding frequency limit, in cycles per second ( hertz ), is 0.5 {\displaystyle 0.5} cycle/sample × f s {\displaystyle f_{s}} samples/second = f s / 2 {\displaystyle f_{s}/2} , known as

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624-400: A waveform. The result is formed by adding the residual and the calculated waveform. As FLAC compresses losslessly , the decoded waveform is identical to the waveform before encoding. For two-channel stereo, the encoder may choose to joint-encode the audio. The channels are transformed into a side channel, which is the difference between the two input channels, and a mid channel, the sum of

676-530: Is a consequence of the Nyquist theorem . Sampling rates higher than about 50 kHz to 60 kHz cannot supply more usable information for human listeners. Early professional audio equipment manufacturers chose sampling rates in the region of 40 to 50 kHz for this reason. There has been an industry trend towards sampling rates well beyond the basic requirements: such as 96 kHz and even 192 kHz Even though ultrasonic frequencies are inaudible to humans, recording and mixing at higher sampling rates

728-526: Is an audio coding format for lossless compression of digital audio , developed by the Xiph.Org Foundation , and is also the name of the free software project producing the FLAC tools, the reference software package that includes a codec implementation. Digital audio compressed by FLAC's algorithm can typically be reduced to between 50 and 70 percent of its original size and decompresses to an identical copy of

780-502: Is converted to digital video , a different sampling process occurs, this time at the pixel frequency, corresponding to a spatial sampling rate along scan lines . A common pixel sampling rate is: Spatial sampling in the other direction is determined by the spacing of scan lines in the raster . The sampling rates and resolutions in both spatial directions can be measured in units of lines per picture height. Spatial aliasing of high-frequency luma or chroma video components shows up as

832-611: Is currently no multicore support in libFLAC, but utilities such as GNU parallel and various graphical frontends can be used to spin up multiple instances of the encoder. FLAC playback support in portable audio devices and dedicated audio systems is limited compared to formats such as MP3 or uncompressed PCM . FLAC support is included by default in Windows 10 , Android , BlackBerry 10 and Jolla devices. In 2014, several aftermarket mobile electronics companies introduced multimedia solutions that include support for FLAC. These include

884-442: Is effective in eliminating the distortion that can be caused by foldback aliasing . Conversely, ultrasonic sounds may interact with and modulate the audible part of the frequency spectrum ( intermodulation distortion ), degrading the fidelity. One advantage of higher sampling rates is that they can relax the low-pass filter design requirements for ADCs and DACs , but with modern oversampling delta-sigma-converters this advantage

936-560: Is implemented as the libFLAC core encoder & decoder library, with the main distributable program flac being the reference implementation of the libFLAC API. This codec API is also available in C++ as libFLAC++. The reference implementation of FLAC compiles on many platforms, including most Unix (such as Solaris , BSD ) and Unix-like (including Linux ), Windows , BeOS , and OS/2 operating systems. There are build-systems for autoconf / automake , MSVC , Watcom C , and Xcode . There

988-435: Is impractical and filters are expensive, aperture error, quantization error and aliasing can be significant limitations. Jitter, noise, and quantization are often analyzed by modeling them as random errors added to the sample values. Integration and zero-order hold effects can be analyzed as a form of low-pass filtering . The non-linearities of either ADC or DAC are analyzed by replacing the ideal linear function mapping with

1040-431: Is less important. The Audio Engineering Society recommends 48 kHz sampling rate for most applications but gives recognition to 44.1 kHz for CD and other consumer uses, 32 kHz for transmission-related applications, and 96 kHz for higher bandwidth or relaxed anti-aliasing filtering . Both Lavry Engineering and J. Robert Stuart state that the ideal sampling rate would be about 60 kHz, but since this

1092-435: Is not a standard frequency, recommend 88.2 or 96 kHz for recording purposes. A more complete list of common audio sample rates is: Audio is typically recorded at 8-, 16-, and 24-bit depth, which yield a theoretical maximum signal-to-quantization-noise ratio (SQNR) for a pure sine wave of, approximately, 49.93  dB , 98.09 dB and 122.17 dB. CD quality audio uses 16-bit samples. Thermal noise limits

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1144-520: Is optimized for decoding speed at the expense of encoding speed. A benchmark has shown that, while there is little variation in decoding speed as compression level increases, beyond the default compression level 5, the encoding process takes up considerably more time with little space saved compared to level 5. Alongside the format, the FLAC project also contains a free and open-source reference implementation of FLAC called libFLAC. libFLAC contains facilities to encode and decode FLAC data and to manipulate

1196-444: Is sampled using an analog-to-digital converter (ADC), a device with various physical limitations. This results in deviations from the theoretically perfect reconstruction, collectively referred to as distortion . Various types of distortion can occur, including: Although the use of oversampling can completely eliminate aperture error and aliasing by shifting them out of the passband, this technique cannot be practically used above

1248-472: Is the Hilbert transform of the other waveform, s ( t ) {\displaystyle s(t)} , the complex-valued function, s a ( t ) ≜ s ( t ) + i ⋅ s ^ ( t ) {\displaystyle s_{a}(t)\triangleq s(t)+i\cdot {\hat {s}}(t)} , is called an analytic signal , whose Fourier transform

1300-610: Is used in most modern analog-to-digital converters to reduce the distortion introduced by practical digital-to-analog converters , such as a zero-order hold instead of idealizations like the Whittaker–Shannon interpolation formula . Complex sampling (or I/Q sampling ) is the simultaneous sampling of two different, but related, waveforms, resulting in pairs of samples that are subsequently treated as complex numbers . When one waveform, s ^ ( t ) {\displaystyle {\hat {s}}(t)} ,

1352-519: Is zero for all negative values of frequency. In that case, the Nyquist rate for a waveform with no frequencies ≥  B can be reduced to just B (complex samples/sec), instead of 2 B {\displaystyle 2B} (real samples/sec). More apparently, the equivalent baseband waveform , s a ( t ) ⋅ e − i 2 π B 2 t {\displaystyle s_{a}(t)\cdot e^{-i2\pi {\frac {B}{2}}t}} , also has

1404-461: The Files (Google) file manager. Various other containers are supported, independently from used operating system, depending on used playback software. Sample rate In signal processing , sampling is the reduction of a continuous-time signal to a discrete-time signal . A common example is the conversion of a sound wave to a sequence of "samples". A sample is a value of the signal at

1456-470: The Nyquist frequency of the sampler. Therefore, s ( t ) {\displaystyle s(t)} is usually the output of a low-pass filter , functionally known as an anti-aliasing filter . Without an anti-aliasing filter, frequencies higher than the Nyquist frequency will influence the samples in a way that is misinterpreted by the interpolation process. In practice, the continuous signal

1508-809: The Pono music player and streaming service used the FLAC format. Bandcamp insists on a lossless format for uploading, and has FLAC as a download option. The Wikimedia Foundation sponsored a free and open-source online ECMAScript FLAC tool for browsers supporting the required HTML5 features. Support introduced in Windows 10. Windows Media Player (2022) also supports FLAC in an Ogg container for live streams (e.g. Icecast internet radio ). Support introduced in Android 3.1. Android natively supports regular FLAC (.flac), but not Ogg FLAC (.oga). However, support for both regular FLAC and Ogg FLAC were later added to

1560-525: The Xiph.Org Foundation and the FLAC project announced the incorporation of FLAC under the Xiph.org banner. Xiph.org is home to other free compression formats such as Vorbis , Theora , Speex and Opus . Version 1.3.0 was released on 26 May 2013, at which point development was moved to the Xiph.org git repository. In 2019, FLAC was proposed as an IETF standard. FLAC is a lossless encoding of linear pulse-code modulation data. A FLAC file consists of

1612-431: The magic number fLaC , metadata , and encoded audio. The encoded audio is divided into frames, each of which consists of a header, a data block, and a CRC16 checksum. Each frame is encoded independent of each other. A frame header begins with a sync word , used to identify the beginning of a valid frame. The rest of the header contains the number of samples, position of the frame, channel assignment, and optionally

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1664-436: The sample rate and bit depth . The data block contains the audio information. Metadata in FLAC precedes the audio. Properties like the sample rate and the number of channels are always contained in the metadata. It may also contain other information, the album cover for example. FLAC uses Vorbis comments for textual metadata like track title and artist name. The FLAC encoding algorithm consists of multiple stages. In

1716-596: The NEX series from Pioneer Electronics and the VX404 and NX404 from Clarion. The European Broadcasting Union (EBU) has adopted the FLAC format for the distribution of high-quality audio over its Euroradio network. The Windows operating system has supported native FLAC integration since the introduction of Windows 10. The Android operating system has supported native FLAC playback since version 3.1. macOS High Sierra and iOS 11 add native FLAC playback support. Among others

1768-470: The album 4.5 stars out of 5, saying: "This is Williams' finest moment, and interestingly, one of Reznor's, too." Quentin B. Huff of PopMatters placed it at number 12 on the "101 Hip-Hop Albums of 2007" list. All tracks are written by Saul Williams and Trent Reznor , except where noted Credits adapted from liner notes. Musicians Technical personnel Free Lossless Audio Codec FLAC ( / f l æ k / ; Free Lossless Audio Codec )

1820-450: The album. It was announced at nin.com that, as of January 2, 2008, two months since its release, 154,449 people had downloaded NiggyTardust. Of that number, 28,322 people chose to pay the asked price of US$ 5 (US$ 141,610 Total). In comparison, Saul's self-titled album has sold 30,000 copies since its release in 2004. A physical release of the album was released on July 8, 2008. It contained five bonus tracks. Thom Jurek of AllMusic gave

1872-486: The block size. Regardless of the amount of compression, the original data can always be reconstructed perfectly. For the user's convenience, the reference implementation defines 9 compression levels, which are presets of the more technical parameters to the encoding algorithm. The levels are labeled from 0 to 8, with higher numbers resulting in a higher compression ratio, at the cost of compression speed. The meaning of each compression level varies by implementation. FLAC

1924-407: The difference between the approximation and the input, called residual, is encoded using Rice coding . In many cases, a description of the approximation and the encoded residual takes up less space than using pulse-code modulation . The decoding process is the reverse of encoding. The compressed residual is first decoded. The description of the mathematical approximation is then used to calculate

1976-445: The equivalent baseband waveform can be created without explicitly computing s ^ ( t ) {\displaystyle {\hat {s}}(t)} , by processing the product sequence, [ s ( n T ) ⋅ e − i 2 π B 2 T n ] {\displaystyle \left[s(nT)\cdot e^{-i2\pi {\frac {B}{2}}Tn}\right]} , through

2028-413: The first stage, the input audio is split into blocks. If the audio contains multiple channels , each channel is encoded separately as a subblock. The encoder then tries to find a good mathematical approximation of the block, either by fitting a simple polynomial , or through general linear predictive coding . A description of the approximation, which is only a few bytes in length, is then written. Finally,

2080-456: The household name that Radiohead is" and urging fans to support him. This was a reference to Radiohead's In Rainbows , which was released in October on the band's own website with customers choosing how much they want to pay for the album. The free option has since been removed, with a message on the website claiming their intention had always been to remove it after 100,000 free downloads of

2132-400: The integration period may be significantly shorter than the time between repetitions, the sampling frequency can be different from the inverse of the sample time: Video digital-to-analog converters operate in the megahertz range (from ~3 MHz for low quality composite video scalers in early games consoles, to 250 MHz or more for the highest-resolution VGA output). When analog video

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2184-583: The metadata of FLAC files. libFLAC++, an object-oriented wrapper around libFLAC for C++ , and the command-line programs flac and metaflac , are also part of the reference implementation. The FLAC format, along with libFLAC, are not known to be covered by any patents , and anyone is free to write their own implementations of FLAC. FLAC is specifically designed for efficient packing of audio data, unlike general-purpose lossless algorithms such as DEFLATE , which are used in ZIP and gzip . While ZIP may reduce

2236-443: The original audio data. FLAC is an open format with royalty-free licensing and a reference implementation which is free software . FLAC supports metadata tagging, album cover art, and fast seeking. Development was started in 2000 by Josh Coalson. The bitstream format was frozen with the release of version 0.9 of the reference implementation on 31 March 2001. Version 1.0 was released on 20 July 2001. On 29 January 2003,

2288-481: The original media are lost, damaged, or worn out, a FLAC copy of the audio tracks ensures that an exact duplicate of the original data can be recovered at any time. An exact restoration from a lossy copy (e.g., MP3 ) of the same data is impossible. FLAC's being lossless means it is highly suitable for transcoding e.g. to MP3, without the normally associated transcoding quality loss between one lossy format and another. A CUE file can optionally be created when ripping

2340-443: The primary usefulness of a digital system is the ability to store, retrieve and transmit signals without any loss of quality. When it is necessary to capture audio covering the entire 20–20,000 Hz range of human hearing such as when recording music or many types of acoustic events, audio waveforms are typically sampled at 44.1 kHz ( CD ), 48 kHz, 88.2 kHz, or 96 kHz. The approximately double-rate requirement

2392-517: The sample values. When the time interval between adjacent samples is a constant ( T ) {\displaystyle (T)} , the sequence of delta functions is called a Dirac comb . Mathematically, the modulated Dirac comb is equivalent to the product of the comb function with s ( t ) {\displaystyle s(t)} . That mathematical abstraction is sometimes referred to as impulse sampling . Most sampled signals are not simply stored and reconstructed. The fidelity of

2444-469: The size of a CD-quality audio file by 10–20%, FLAC is able to reduce the size of audio data by 40–50% by taking advantage of the characteristics of audio. The technical strengths of FLAC compared to other lossless formats lie in its ability to be streamed and decoded quickly, independent of compression level. Since FLAC is a lossless scheme, it is suitable as an archive format for owners of CDs and other media who wish to preserve their audio collections. If

2496-455: The true number of bits that can be used in quantization. Few analog systems have signal to noise ratios (SNR) exceeding 120 dB. However, digital signal processing operations can have very high dynamic range, consequently it is common to perform mixing and mastering operations at 32-bit precision and then convert to 16- or 24-bit for distribution. Speech signals, i.e., signals intended to carry only human speech , can usually be sampled at

2548-417: The two input channels. In place of a mid channel, the left channel or the right channel may be encoded instead, which is sometimes more space-efficient. Even though the reference encoder uses a single block size for the whole stream, FLAC allows the block size in samples to vary per block. The amount of compression is determined by various parameters, including the order of the linear prediction model and

2600-402: The unit samples per second , sometimes referred to as hertz , for example 48 kHz is 48,000 samples per second . Reconstructing a continuous function from samples is done by interpolation algorithms. The Whittaker–Shannon interpolation formula is mathematically equivalent to an ideal low-pass filter whose input is a sequence of Dirac delta functions that are modulated (multiplied) by

2652-484: The value of the continuous function every T {\displaystyle T} seconds, which is called the sampling interval or sampling period . Then the sampled function is given by the sequence: The sampling frequency or sampling rate , f s {\displaystyle f_{s}} , is the average number of samples obtained in one second, thus f s = 1 / T {\displaystyle f_{s}=1/T} , with

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2704-435: The visible picture area. High-definition television (HDTV) uses 720p (progressive), 1080i (interlaced), and 1080p (progressive, also known as Full-HD). In digital video , the temporal sampling rate is defined as the frame rate  – or rather the field rate  – rather than the notional pixel clock . The image sampling frequency is the repetition rate of the sensor integration period. Since

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