romikgosai/squad_modified huggingface dataset:
Most graduate immunology schools follow the AAI courses immunology which are offered throughout numerous schools in the United States. For example, in New York State, there are several universities that offer the AAI courses immunology: Albany Medical College, Cornell University, Icahn School of Medicine at Mount Sinai, New York University Langone Medical Center, University at Albany (SUNY), University at Buffalo (SUNY), University of Rochester Medical Center and Upstate Medical University (SUNY). The AAI immunology courses include an Introductory Course and an Advance Course. The Introductory Course is a course that gives students an overview of the basics of immunology.

Questions and answers:
["
Q: Most graduate schools specializing in immunology follow what organization's parameters?
A: AAI courses", '
Q: What is the name of the immunology school at Mount Sinai?
A: Icahn School of Medicine', '
Q: AAI immunology courses include what two components?
A: Introductory Course and an Advance Course', '
Q: What does the introductory AAI immunology course do?
A: gives students an overview of the basics of immunology', '
Q: The Langone Medical Center is part of what college?
A: New York University']
In addition, this Introductory Course gives students more information to complement general biology or science training. It also has two different parts: Part I is an introduction to the basic principles of immunology and Part II is a clinically-oriented lecture series. On the other hand, the Advanced Course is another course for those who are willing to expand or update their understanding of immunology. It is advised for students who want to attend the Advanced Course to have a background of the principles of immunology. Most schools require students to take electives in other to complete their degrees. A Master’s degree requires two years of study following the attainment of a bachelor's degree. For a doctoral programme it is required to take two additional years of study.

Questions and answers:
['
Q: Part 1 of the AAI introductory immunology course is meant to be what?
A: an introduction to the basic principles of immunology', '
Q: Part 2 of the AAI introductory immunology course revolves around what?
A: clinically-oriented lecture series', '
Q: What pre-requisite should enrollees in the advanced immunology course have?
A: a background of the principles of immunology', "
Q: How long does it typically take to earn a Master's degree?
A: two years of study", '
Q: What program requires an additional two years of study?
A: a doctoral programme']
MPEG-1 or MPEG-2 Audio Layer III, more commonly referred to as MP3, is an audio coding format for digital audio which uses a form of lossy data compression. It is a common audio format for consumer audio streaming or storage, as well as a de facto standard of digital audio compression for the transfer and playback of music on most digital audio players.

Questions and answers:
['
Q: What is the common name for MPEG-2 Audio Layer III?
A: MP3', '
Q: What kind of audio is the coding format for?
A: digital', '
Q: Other than storage, what is the common audio format used for?
A: consumer audio streaming', "
Q: What does MP3 use to format it's data?
A: lossy data compression", '
Q: Where will MP3 files most commonly be found?
A: digital audio players']
The use of lossy compression is designed to greatly reduce the amount of data required to represent the audio recording and still sound like a faithful reproduction of the original uncompressed audio for most listeners. An MP3 file that is created using the setting of 128 kbit/s will result in a file that is about 1/11 the size of the CD file created from the original audio source (44,100 samples per second × 16 bits per sample × 2 channels = 1,411,200 bit/s; MP3 compressed at 128 kbit/s: 128,000 bit/s [1 k = 1,000, not 1024, because it is a bit rate]. Ratio: 1,411,200/128,000 = 11.025). An MP3 file can also be constructed at higher or lower bit rates, with higher or lower resulting quality.

Questions and answers:
['
Q: What is the main goal aside from reducing the amount of data required to store the audio?
A: sound like a faithful reproduction', '
Q: What kind of compression is used?
A: created', '
Q: If a file is created using 128 kbit/s, what size is the file going to be compared to a CD?
A: 1/11', '
Q: How many total bit/s would a CD have?
A: 1,411,200', '
Q: How many total bit/s would an MP3 compressed at 128 kbit/s have?
A: 128,000']
The compression works by reducing the accuracy of certain parts of a sound that are considered to be beyond the auditory resolution ability of most people. This method is commonly referred to as perceptual coding. It uses psychoacoustic models to discard or reduce precision of components less audible to human hearing, and then records the remaining information in an efficient manner.

Questions and answers:
['
Q: What does the compression target in the song?
A: accuracy', '
Q: The targetting focuses on what aspect that is beyond the ability of most people?
A: auditory resolution', '
Q: What is this method commonly referred to as?
A: perceptual coding', '
Q: What kind of model is used in this kind of compression?
A: psychoacoustic', '
Q: These models ensure that the precision is reduced for the components that are what to human hearing?
A: less audible']
MP3 was designed by the Moving Picture Experts Group (MPEG) as part of its MPEG-1 standard and later extended in the MPEG-2 standard. The first subgroup for audio was formed by several teams of engineers at Fraunhofer IIS, University of Hannover, AT&T-Bell Labs, Thomson-Brandt, CCETT, and others. MPEG-1 Audio (MPEG-1 Part 3), which included MPEG-1 Audio Layer I, II and III was approved as a committee draft of ISO/IEC standard in 1991, finalised in 1992 and published in 1993 (ISO/IEC 11172-3:1993). Backwards compatible MPEG-2 Audio (MPEG-2 Part 3) with additional bit rates and sample rates was published in 1995 (ISO/IEC 13818-3:1995).

Questions and answers:
['
Q: Who designed MP3?
A: Moving Picture Experts Group', '
Q: What was the first standard that this group had?
A: MPEG-1', '
Q: The first standard later developed into which standard?
A: MPEG-2', '
Q: When was the draft passed to have MPEG-1 Audio approved as a standard?
A: 1991', '
Q: When was MPEG-2 Audio finally published?
A: 1995']
The MP3 lossy audio data compression algorithm takes advantage of a perceptual limitation of human hearing called auditory masking. In 1894, the American physicist Alfred M. Mayer reported that a tone could be rendered inaudible by another tone of lower frequency. In 1959, Richard Ehmer described a complete set of auditory curves regarding this phenomenon. Ernst Terhardt et al. created an algorithm describing auditory masking with high accuracy. This work added to a variety of reports from authors dating back to Fletcher, and to the work that initially determined critical ratios and critical bandwidths.

Questions and answers:
['
Q: What does the MP3 compression take advantage of?
A: perceptual limitation', '
Q: What is the limitation in human hearing referred to as?
A: auditory masking', '
Q: Who was the physicist to report that a tone could be rendered inaudible.
A: Alfred M. Mayer', '
Q: When were audio curves described by Richard Ehmer?
A: 1959', '
Q: The initial work was aimed at determining critical ratios and what else?
A: critical bandwidths']
The psychoacoustic masking codec was first proposed in 1979, apparently independently, by Manfred R. Schroeder, et al. from Bell Telephone Laboratories, Inc. in Murray Hill, NJ, and M. A. Krasner both in the United States. Krasner was the first to publish and to produce hardware for speech (not usable as music bit compression), but the publication of his results as a relatively obscure Lincoln Laboratory Technical Report did not immediately influence the mainstream of psychoacoustic codec development. Manfred Schroeder was already a well-known and revered figure in the worldwide community of acoustical and electrical engineers, but his paper was not much noticed, since it described negative results due to the particular nature of speech and the linear predictive coding (LPC) gain present in speech. Both Krasner and Schroeder built upon the work performed by Eberhard F. Zwicker in the areas of tuning and masking of critical bands, that in turn built on the fundamental research in the area from Bell Labs of Harvey Fletcher and his collaborators. A wide variety of (mostly perceptual) audio compression algorithms were reported in IEEE's refereed Journal on Selected Areas in Communications. That journal reported in February 1988 on a wide range of established, working audio bit compression technologies, some of them using auditory masking as part of their fundamental design, and several showing real-time hardware implementations.

Questions and answers:
['
Q: What was first proposed in 1979?
A: The psychoacoustic masking codec', '
Q: Which country were the researchers located in?
A: United States', '
Q: Who was the first to produce hardware for speech?
A: Krasner', '
Q: What does LPC stand for?
A: linear predictive coding', "
Q: What was reported in IEEE's Journal on Selected Areas in Communications?
A: audio compression algorithms"]
The immediate predecessors of MP3 were "Optimum Coding in the Frequency Domain" (OCF), and Perceptual Transform Coding (PXFM). These two codecs, along with block-switching contributions from Thomson-Brandt, were merged into a codec called ASPEC, which was submitted to MPEG, and which won the quality competition, but that was mistakenly rejected as too complex to implement. The first practical implementation of an audio perceptual coder (OCF) in hardware (Krasner's hardware was too cumbersome and slow for practical use), was an implementation of a psychoacoustic transform coder based on Motorola 56000 DSP chips.

Questions and answers:
['
Q: What does OCF stand for?
A: Optimum Coding in the Frequency Domain', '
Q: What does PXFM stand for?
A: Perceptual Transform Coding', '
Q: What was the name of the codec that OCF, PFXM and other contributes merged into?
A: ASPEC', '
Q: Which competition did ASPEC win?
A: quality', '
Q: What was the first practical implementation based on?
A: Motorola 56000 DSP chips']
As a doctoral student at Germany's University of Erlangen-Nuremberg, Karlheinz Brandenburg began working on digital music compression in the early 1980s, focusing on how people perceive music. He completed his doctoral work in 1989. MP3 is directly descended from OCF and PXFM, representing the outcome of the collaboration of Brandenburg—working as a postdoc at AT&T-Bell Labs with James D. Johnston ("JJ") of AT&T-Bell Labs—with the Fraunhofer Institut for Integrated Circuits, Erlangen, with relatively minor contributions from the MP2 branch of psychoacoustic sub-band coders. In 1990, Brandenburg became an assistant professor at Erlangen-Nuremberg. While there, he continued to work on music compression with scientists at the Fraunhofer Society (in 1993 he joined the staff of the Fraunhofer Institute).

Questions and answers:
['
Q: What kind of student was Karlheinz Brandenburg?
A: doctoral', '
Q: When did Karlheinz start working on digitmal music compression?
A: early 1980s', '
Q: Where did Bradenburg become an assistant professor at?
A: Erlangen-Nuremberg', '
Q: Who did Brandenburg work with in music compression?
A: Fraunhofer Society', '
Q: When did Brandenburg join the Fraunhofen Institute?
A: 1993']
The song "Tom's Diner" by Suzanne Vega was the first song used by Karlheinz Brandenburg to develop the MP3. Brandenburg adopted the song for testing purposes, listening to it again and again each time refining the scheme, making sure it did not adversely affect the subtlety of Vega's voice.

Questions and answers:
["
Q: What was the name of the first song used to develop the MP3?
A: Tom's Diner", '
Q: Who was the artist of the first song used?
A: Suzanne Vega', '
Q: Who adjusted the song?
A: Karlheinz Brandenburg', '
Q: The song was listened to multiple times to try and refine what?
A: scheme', '
Q: What particular aspect of the artists voice was he trying to preserve?
A: subtlety']
In 1991, there were only two proposals available that could be completely assessed for an MPEG audio standard: Musicam (Masking pattern adapted Universal Subband Integrated Coding And Multiplexing) and ASPEC (Adaptive Spectral Perceptual Entropy Coding). The Musicam technique, as proposed by Philips (the Netherlands), CCETT (France) and Institut für Rundfunktechnik (Germany) was chosen due to its simplicity and error robustness, as well as its low computational power associated with the encoding of high quality compressed audio. The Musicam format, based on sub-band coding, was the basis of the MPEG Audio compression format (sampling rates, structure of frames, headers, number of samples per frame).

Questions and answers:
['
Q: How many proposals were available in 1991?
A: two', '
Q: Other than ASPEC what was another proposal?
A: Musicam', '
Q: The Musicam format was proposed by Philips, based in which country?
A: Netherlands', '
Q: Which other quality alongside simplicty was key in selecting this proposal?
A: error robustness', '
Q: What was the Musican format based on?
A: sub-band coding']
Much of its technology and ideas were incorporated into the definition of ISO MPEG Audio Layer I and Layer II and the filter bank alone into Layer III (MP3) format as part of the computationally inefficient hybrid filter bank. Under the chairmanship of Professor Musmann (University of Hannover) the editing of the standard was made under the responsibilities of Leon van de Kerkhof (Layer I) and Gerhard Stoll (Layer II).

Questions and answers:
['
Q: What was only incorporated into Layer III and not Layers I or II?
A: filter bank', '
Q: Who was the chair that oversaw the editing of the standard?
A: Professor Musmann', "
Q: Who's responsibility was it to edit the standard for Layer I?
A: Leon van de Kerkhof", "
Q: Who's responsibility was it to edit the standard for Layer II?
A: Gerhard Stoll"]
ASPEC was the joint proposal of AT&T Bell Laboratories, Thomson Consumer Electronics, Fraunhofer Society and CNET. It provided the highest coding efficiency.

Questions and answers:
['
Q: What was the name given to the proposal?
A: ASPEC', '
Q: What did the joint proposal provide?
A: the highest coding efficiency', '
Q: Other than Thomson Consumer Electronics, Fraunhofer Society and CNET, who else was a part of the joint proposal?
A: AT&T Bell Laboratories']
A working group consisting of Leon van de Kerkhof (The Netherlands), Gerhard Stoll (Germany), Leonardo Chiariglione (Italy), Yves-François Dehery (France), Karlheinz Brandenburg (Germany) and James D. Johnston (USA) took ideas from ASPEC, integrated the filter bank from Layer 2, added some of their own ideas and created MP3, which was designed to achieve the same quality at 128 kbit/s as MP2 at 192 kbit/s.

Questions and answers:
['
Q: What did the working group integrate their ideas with?
A: filter bank', '
Q: Where was the filter bank taken from?
A: Layer 2', '
Q: What did the working group take ideas from?
A: ASPEC', '
Q: What did the working group eventually create?
A: MP3', '
Q: What quality were they hoping to match at 128 kbit/s?
A: MP2 at 192 kbit/s']
All algorithms for MPEG-1 Audio Layer I, II and III were approved in 1991 and finalized in 1992 as part of MPEG-1, the first standard suite by MPEG, which resulted in the international standard ISO/IEC 11172-3 (a.k.a. MPEG-1 Audio or MPEG-1 Part 3), published in 1993.

Questions and answers:
['
Q: When were the algorithms approved?
A: 1991', '
Q: When were the approved algorithms finalized?
A: 1992', '
Q: What was the first started suite by MPEG?
A: MPEG-1', '
Q: What was the official name of the internation standard?
A: ISO/IEC 11172-3', '
Q: When was the international standard published?
A: 1993']
Further work on MPEG audio was finalized in 1994 as part of the second suite of MPEG standards, MPEG-2, more formally known as international standard ISO/IEC 13818-3 (a.k.a. MPEG-2 Part 3 or backwards compatible MPEG-2 Audio or MPEG-2 Audio BC), originally published in 1995. MPEG-2 Part 3 (ISO/IEC 13818-3) defined additional bit rates and sample rates for MPEG-1 Audio Layer I, II and III. The new sampling rates are exactly half that of those originally defined in MPEG-1 Audio. This reduction in sampling rate serves to cut the available frequency fidelity in half while likewise cutting the bitrate by 50%. MPEG-2 Part 3 also enhanced MPEG-1's audio by allowing the coding of audio programs with more than two channels, up to 5.1 multichannel.

Questions and answers:
['
Q: Further work on MPEG was finalized as a part of which MPEG standard?
A: second suite', '
Q: What was the official name of ISO/IEC 13818-3 better known as?
A: MPEG-2', '
Q: At what rate were the new sampling rates defined in comparison to MPEG-1 Audio?
A: half', '
Q: Apart from cutting the frequency fidelity in half, what else was cut in half?
A: bitrate', '
Q: Allowing the coding of audio programs with more than two channels was a major aspect of what?
A: MPEG-2 Part 3']
An additional extension to MPEG-2 is named MPEG-2.5 audio, as MPEG-3 already had a different meaning. This extension was developed at Fraunhofer IIS, the registered MP3 patent holders. Like MPEG-2, MPEG-2.5 adds new sampling rates exactly half of that previously possible with MPEG-2. It thus widens the scope of MP3 to include human speech and other applications requiring only 25% of the frequency reproduction possible with MPEG-1. While not an ISO recognized standard, MPEG-2.5 is widely supported by both inexpensive and brand name digital audio players as well as computer software based MP3 encoders and decoders. A sample rate comparison between MPEG-1, 2 and 2.5 is given further down. MPEG-2.5 was not developed by MPEG and was never approved as an international standard. MPEG-2.5 is thus an unofficial or proprietary extension to the MP3 format.

Questions and answers:
['
Q: As MPEG-3 had a different meaning, what was the name given to the extension of MPEG-2?
A: MPEG-2.5 audio', '
Q: Where was this extension developed?
A: Fraunhofer IIS', '
Q: The new sampling rates widened the scope of MP3 to be able to include what?
A: human speech', '
Q: What is not an ISO recognized standard?
A: MPEG-2.5', '
Q: As MPEG-2.5 is unofficial, it is considered what kind of extension to the MP3 format?
A: proprietary']
Compression efficiency of encoders is typically defined by the bit rate, because compression ratio depends on the bit depth and sampling rate of the input signal. Nevertheless, compression ratios are often published. They may use the Compact Disc (CD) parameters as references (44.1 kHz, 2 channels at 16 bits per channel or 2×16 bit), or sometimes the Digital Audio Tape (DAT) SP parameters (48 kHz, 2×16 bit). Compression ratios with this latter reference are higher, which demonstrates the problem with use of the term compression ratio for lossy encoders.

Questions and answers:
['
Q: What defines the compression efficiency of encoders?
A: bit rate', '
Q: Compression ratio depends on the sample rate and bit depth of which signal?
A: input', '
Q: What does CD stand for?
A: Compact Disc', '
Q: Other than CD parameters, what else can be used as parameter references?
A: Digital Audio Tape (DAT) SP', '
Q: What can CD parameters be used as references for?
A: compression ratios']
Karlheinz Brandenburg used a CD recording of Suzanne Vega's song "Tom's Diner" to assess and refine the MP3 compression algorithm. This song was chosen because of its nearly monophonic nature and wide spectral content, making it easier to hear imperfections in the compression format during playbacks. Some refer to Suzanne Vega as "The mother of MP3". This particular track has an interesting property in that the two channels are almost, but not completely, the same, leading to a case where Binaural Masking Level Depression causes spatial unmasking of noise artifacts unless the encoder properly recognizes the situation and applies corrections similar to those detailed in the MPEG-2 AAC psychoacoustic model. Some more critical audio excerpts (glockenspiel, triangle, accordion, etc.) were taken from the EBU V3/SQAM reference compact disc and have been used by professional sound engineers to assess the subjective quality of the MPEG Audio formats.

Questions and answers:
['
Q: What type of recording did Brandenburg use?
A: CD', '
Q: What did Brandenburg use the recording to refine?
A: the MP3 compression algorithm', '
Q: What word describes the nature of the song used?
A: monophonic', '
Q: As a result of her song being used, Suzanne Vega is sometimes referred to as what?
A: The mother of MP3', '
Q: Excerpts were taken from which reference compact disc to assess the subjective quality of the MPEG audio formats?
A: EBU V3/SQAM']
A reference simulation software implementation, written in the C language and later known as ISO 11172-5, was developed (in 1991–1996) by the members of the ISO MPEG Audio committee in order to produce bit compliant MPEG Audio files (Layer 1, Layer 2, Layer 3). It was approved as a committee draft of ISO/IEC technical report in March 1994 and printed as document CD 11172-5 in April 1994. It was approved as a draft technical report (DTR/DIS) in November 1994, finalized in 1996 and published as international standard ISO/IEC TR 11172-5:1998 in 1998. The reference software in C language was later published as a freely available ISO standard. Working in non-real time on a number of operating systems, it was able to demonstrate the first real time hardware decoding (DSP based) of compressed audio. Some other real time implementation of MPEG Audio encoders were available for the purpose of digital broadcasting (radio DAB, television DVB) towards consumer receivers and set top boxes.

Questions and answers:
['
Q: Who developed the reference simulation software?
A: the ISO MPEG Audio committee', '
Q: Which language was the reference simulation software written in?
A: C', '
Q: When was the reference software approved?
A: March 1994', '
Q: When did the reference software become an international standard?
A: 1998', '
Q: What could the reference software demonstrate?
A: real time hardware decoding']
On 7 July 1994, the Fraunhofer Society released the first software MP3 encoder called l3enc. The filename extension .mp3 was chosen by the Fraunhofer team on 14 July 1995 (previously, the files had been named .bit). With the first real-time software MP3 player WinPlay3 (released 9 September 1995) many people were able to encode and play back MP3 files on their PCs. Because of the relatively small hard drives back in that time (~ 500–1000 MB) lossy compression was essential to store non-instrument based (see tracker and MIDI) music for playback on computer.

Questions and answers:
['
Q: When was the first software MP3 encoder released?
A: 7 July 1994', '
Q: What was the name of the first software MP3 encoder?
A: l3enc', '
Q: What was the filename extension?
A: .mp3', '
Q: What was the name of the first real-time software MP3 player?
A: WinPlay3', '
Q: What was the average high end hard drive size?
A: 1000 MB']
As sound scholar Jonathan Sterne notes, "An Australian hacker acquired l3enc using a stolen credit card. The hacker then reverse-engineered the software, wrote a new user interface, and redistributed it for free, naming it "thank you Fraunhofer"".

Questions and answers:
["
Q: What was Jonathan Sterne's profession?
A: sound scholar", '
Q: Which nationality was the hacker who acquired I3enc?
A: Australian', '
Q: What did the hacker use to acquire the software?
A: stolen credit card', '
Q: The hacker re-wrote which element of the program?
A: user interface', '
Q: How much did the new released program by the hacker cost?
A: free']