romikgosai/squad_modified_2 huggingface dataset:
ASPEC was the joint proposal of AT&T Bell Laboratories, Thomson Consumer Electronics, Fraunhofer Society and CNET. It provided the highest coding efficiency.

Q1: What was the name given to the proposal?
A1: ASPEC
Q2: What did the joint proposal provide?
A2: the highest coding efficiency
Q3: Other than Thomson Consumer Electronics, Fraunhofer Society and CNET, who else was a part of the joint proposal?
A3: 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.

Q1: What did the working group integrate their ideas with?
A1: filter bank
Q2: Where was the filter bank taken from?
A2: Layer 2
Q3: What did the working group take ideas from?
A3: ASPEC
Q4: What did the working group eventually create?
A4: MP3
Q5: What quality were they hoping to match at 128 kbit/s?
A5: 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.

Q1: When were the algorithms approved?
A1: 1991
Q2: When were the approved algorithms finalized?
A2: 1992
Q3: What was the first started suite by MPEG?
A3: MPEG-1
Q4: What was the official name of the internation standard?
A4: ISO/IEC 11172-3
Q5: When was the international standard published?
A5: 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.

Q1: Further work on MPEG was finalized as a part of which MPEG standard?
A1: second suite
Q2: What was the official name of ISO/IEC 13818-3 better known as?
A2: MPEG-2
Q3: At what rate were the new sampling rates defined in comparison to MPEG-1 Audio?
A3: half
Q4: Apart from cutting the frequency fidelity in half, what else was cut in half?
A4: bitrate
Q5: Allowing the coding of audio programs with more than two channels was a major aspect of what?
A5: 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.

Q1: As MPEG-3 had a different meaning, what was the name given to the extension of MPEG-2?
A1: MPEG-2.5 audio
Q2: Where was this extension developed?
A2: Fraunhofer IIS
Q3: The new sampling rates widened the scope of MP3 to be able to include what?
A3: human speech
Q4: What is not an ISO recognized standard?
A4: MPEG-2.5
Q5: As MPEG-2.5 is unofficial, it is considered what kind of extension to the MP3 format?
A5: 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.

Q1: What defines the compression efficiency of encoders?
A1: bit rate
Q2: Compression ratio depends on the sample rate and bit depth of which signal?
A2: input
Q3: What does CD stand for?
A3: Compact Disc
Q4: Other than CD parameters, what else can be used as parameter references?
A4: Digital Audio Tape (DAT) SP
Q5: What can CD parameters be used as references for?
A5: 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.

Q1: What type of recording did Brandenburg use?
A1: CD
Q2: What did Brandenburg use the recording to refine?
A2: the MP3 compression algorithm
Q3: What word describes the nature of the song used?
A3: monophonic
Q4: As a result of her song being used, Suzanne Vega is sometimes referred to as what?
A4: The mother of MP3
Q5: Excerpts were taken from which reference compact disc to assess the subjective quality of the MPEG audio formats?
A5: 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.

Q1: Who developed the reference simulation software?
A1: the ISO MPEG Audio committee
Q2: Which language was the reference simulation software written in?
A2: C
Q3: When was the reference software approved?
A3: March 1994
Q4: When did the reference software become an international standard?
A4: 1998
Q5: What could the reference software demonstrate?
A5: 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.

Q1: When was the first software MP3 encoder released?
A1: 7 July 1994
Q2: What was the name of the first software MP3 encoder?
A2: l3enc
Q3: What was the filename extension?
A3: .mp3
Q4: What was the name of the first real-time software MP3 player?
A4: WinPlay3
Q5: What was the average high end hard drive size?
A5: 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"".

Q1: What was Jonathan Sterne's profession?
A1: sound scholar
Q2: Which nationality was the hacker who acquired I3enc?
A2: Australian
Q3: What did the hacker use to acquire the software?
A3: stolen credit card
Q4: The hacker re-wrote which element of the program?
A4: user interface
Q5: How much did the new released program by the hacker cost?
A5: free
In the second half of '90s, MP3 files began to spread on the Internet. The popularity of MP3s began to rise rapidly with the advent of Nullsoft's audio player Winamp, released in 1997. In 1998, the first portable solid state digital audio player MPMan, developed by SaeHan Information Systems which is headquartered in Seoul, South Korea, was released and the Rio PMP300 was sold afterwards in 1998, despite legal suppression efforts by the RIAA.

Q1: Where did MP3s begin to spread?
A1: the Internet
Q2: Which audio player was released in 1997?
A2: Winamp
Q3: Who released the audio player?
A3: Nullsoft
Q4: What was the name of the first portable solid state digital audio player?
A4: MPMan
Q5: In which country was the first portable audio player developed?
A5: South Korea
In November 1997, the website mp3.com was offering thousands of MP3s created by independent artists for free. The small size of MP3 files enabled widespread peer-to-peer file sharing of music ripped from CDs, which would have previously been nearly impossible. The first large peer-to-peer filesharing network, Napster, was launched in 1999.

Q1: Which website offered MP3s for free?
A1: mp3.com
Q2: How many MP3s were offered for free by the website?
A2: thousands
Q3: What kind of file sharing became popular due to the file sizes of MP3s?
A3: peer-to-peer
Q4: How did people come into possession of MP3s?
A4: ripped from CDs
Q5: Which network was the first large peer to peer file sharing network?
A5: Napster
The ease of creating and sharing MP3s resulted in widespread copyright infringement. Major record companies argued that this free sharing of music reduced sales, and called it "music piracy". They reacted by pursuing lawsuits against Napster (which was eventually shut down and later sold) and against individual users who engaged in file sharing.

Q1: What kind of infringement resulted from sharing MP3s?
A1: copyright
Q2: What name did the record companies associate with MP3 file sharing?
A2: music piracy
Q3: Who was the target of the record companies?
A3: Napster
Q4: What happened to Napster?
A4: shut down and later sold
Unauthorized MP3 file sharing continues on next-generation peer-to-peer networks. Some authorized services, such as Beatport, Bleep, Juno Records, eMusic, Zune Marketplace, Walmart.com, Rhapsody, the recording industry approved re-incarnation of Napster, and Amazon.com sell unrestricted music in the MP3 format.

Q1: What kind of MP3 file sharing continues on?
A1: Unauthorized
Q2: Beatport, Bleep and Juno records are examples of what kind of service?
A2: authorized
Q3: What kind of music is sold by these companies?
A3: unrestricted
Q4: Which format is used by the companies that sell the music legally?
A4: MP3
An MP3 file is made up of MP3 frames, which consist of a header and a data block. This sequence of frames is called an elementary stream. Due to the "byte reservoir", frames are not independent items and cannot usually be extracted on arbitrary frame boundaries. The MP3 Data blocks contain the (compressed) audio information in terms of frequencies and amplitudes. The diagram shows that the MP3 Header consists of a sync word, which is used to identify the beginning of a valid frame. This is followed by a bit indicating that this is the MPEG standard and two bits that indicate that layer 3 is used; hence MPEG-1 Audio Layer 3 or MP3. After this, the values will differ, depending on the MP3 file. ISO/IEC 11172-3 defines the range of values for each section of the header along with the specification of the header. Most MP3 files today contain ID3 metadata, which precedes or follows the MP3 frames, as noted in the diagram.

Q1: A header and data block together make up what?
A1: MP3 frames
Q2: What is a sequence of MP3 frames called?
A2: elementary stream
Q3: What is used to identify the begining of a valid frame of an MP3 header?
A3: dard
Q4: How many bits are needed to indicate that layer 3 is used?
A4: two
Q5: MP3 files today contain what kind of metadeta?
A5: ID3
The MPEG-1 standard does not include a precise specification for an MP3 encoder, but does provide example psychoacoustic models, rate loop, and the like in the non-normative part of the original standard. At present, these suggested implementations are quite dated. Implementers of the standard were supposed to devise their own algorithms suitable for removing parts of the information from the audio input. As a result, there are many different MP3 encoders available, each producing files of differing quality. Comparisons are widely available, so it is easy for a prospective user of an encoder to research the best choice. An encoder that is proficient at encoding at higher bit rates (such as LAME) is not necessarily as good at lower bit rates.

Q1: Which standard does not include a specification for an MP3 encoder?
A1: MPEG-1
Q2: Example psychoacoustic models and rate loops can be foundin which part of the original standard?
A2: non-normative
Q3: What were implementers of the standard supposed to devise?
A3: their own algorithms
Q4: It is easy for a prospective user of an encoder to research the best choice because of what factor?
A4: Comparisons are widely available
Q5: An encoder that is proficient at encoding at higher bit rates might be worse at what?
A5: lower bit rates
During encoding, 576 time-domain samples are taken and are transformed to 576 frequency-domain samples.[clarification needed] If there is a transient, 192 samples are taken instead of 576. This is done to limit the temporal spread of quantization noise accompanying the transient. (See psychoacoustics.)

Q1: How many domain samples are taken during encoding?
A1: 576
Q2: Domain samples are transformed into what?
A2: frequency-domain samples
Q3: When are 192 samples taken instead of 576?
A3: there is a transient
Q4: When 192 samples are taken instead of 576, it is done in an effort to limit what?
A4: temporal spread
Q5: What accompanies a transient?
A5: quantization noise
Due to the tree structure of the filter bank, pre-echo problems are made worse, as the combined impulse response of the two filter banks does not, and cannot, provide an optimum solution in time/frequency resolution. Additionally, the combining of the two filter banks' outputs creates aliasing problems that must be handled partially by the "aliasing compensation" stage; however, that creates excess energy to be coded in the frequency domain, thereby decreasing coding efficiency.[citation needed]

Q1: What kind of structure does a filter bank have?
A1: tree
Q2: Due to a specific kind of structure, what kind of problems are made worse?
A2: pre-echo problems
Q3: What kind of resolution are the filter banks unable to provide an optimal solution for?
A3: time/frequency
Q4: Combining two filter banks' outputs creates what kind of problem?
A4: aliasing
Q5: The need to code excess energy in the frequency domain decreases what kind of efficiency?
A5: coding
Decoding, on the other hand, is carefully defined in the standard. Most decoders are "bitstream compliant", which means that the decompressed output that they produce from a given MP3 file will be the same, within a specified degree of rounding tolerance, as the output specified mathematically in the ISO/IEC high standard document (ISO/IEC 11172-3). Therefore, comparison of decoders is usually based on how computationally efficient they are (i.e., how much memory or CPU time they use in the decoding process).

Q1: What is carefully defined in the standard?
A1: Decoding
Q2: What are most decoders?
A2: bitstream compliant
Q3: The ISO/IEC high standard document states that the decompressed output produced from a given MP3 file will be the same within what standards?
A3: specified degree of rounding tolerance
Q4: Decoders are usually compared by examining which factor?
A4: computationally efficient
Q5: The efficiency of decoders is examined by seeing how much memory and what other process they use in the decoding process?
A5: CPU time
Encoder / decoder overall delay is not defined, which means there is no official provision for gapless playback. However, some encoders such as LAME can attach additional metadata that will allow players that can handle it to deliver seamless playback.

Q1: What is there no official provision for?
A1: gapless playback
Q2: What is an example of an encoder that can attach additional metadeta?
A2: LAME
Q3: Adding additional metadata allows players to handle gapless playback and deliver what?
A3: seamless playback
Q4: For encoders and decoders, what is not defined?
A4: overall delay
When performing lossy audio encoding, such as creating an MP3 file, there is a trade-off between the amount of space used and the sound quality of the result. Typically, the creator is allowed to set a bit rate, which specifies how many kilobits the file may use per second of audio. The higher the bit rate, the larger the compressed file will be, and, generally, the closer it will sound to the original file.

Q1: A typical trade off when creating an MP3 file is between the amount of space used and what other factor?
A1: sound quality of the result
Q2: What would an example of lossy audio encoding be?
A2: creating an MP3 file
Q3: Who gets to set the bit rate for an MP3 file?
A3: the creator
Q4: What does the bit rate of the file actually mean?
A4: how many kilobits the file may use per second of audio
Q5: If the bit rate is higher, the size of the compressed file will be what?
A5: larger
With too low a bit rate, compression artifacts (i.e., sounds that were not present in the original recording) may be audible in the reproduction. Some audio is hard to compress because of its randomness and sharp attacks. When this type of audio is compressed, artifacts such as ringing or pre-echo are usually heard. A sample of applause compressed with a relatively low bit rate provides a good example of compression artifacts.

Q1: If the bit rate is too low, what might be audible in the reproduction?
A1: compression artifacts
Q2: Sounds that were not in the original recording are referred to as what?
A2: compression artifacts
Q3: Randomness and sharp attacks are two reasons that may make it harder to to do what to an audio file?
A3: compress
Q4: Other than pre - echo, what is another example of what can happen when audio is compressed?
A4: ringing
Q5: What kind of sample can display a good example of compression artifacts?
A5: A sample of applause
Besides the bit rate of an encoded piece of audio, the quality of MP3 files also depends on the quality of the encoder itself, and the difficulty of the signal being encoded. As the MP3 standard allows quite a bit of freedom with encoding algorithms, different encoders may feature quite different quality, even with identical bit rates. As an example, in a public listening test featuring two different MP3 encoders at about 128 kbit/s, one scored 3.66 on a 1–5 scale, while the other scored only 2.22.

Q1: Other than the bit rate and the difficulty of the signal, what can also affect the quality of an MP3 file?
A1: quality of the encoder
Q2: The MP3 standard allows encoders to have a lot of freedom with which aspect of encoding?
A2: encoding algorithms
Q3: As a result of the freedom, different encoders can result in different quality even if which aspect of the file is the same?
A3: bit rates
Q4: An example shown has a low range quality score of what value?
A4: 2.22
Q5: To compare qualities, sound files are evaluated on a scale that ranges from 1 to what?
A5: 5
The simplest type of MP3 file uses one bit rate for the entire file: this is known as Constant Bit Rate (CBR) encoding. Using a constant bit rate makes encoding simpler and faster. However, it is also possible to create files where the bit rate changes throughout the file. These are known as Variable Bit Rate (VBR) files. The idea behind this is that, in any piece of audio, some parts will be much easier to compress, such as silence or music containing only a few instruments, while others will be more difficult to compress. So, the overall quality of the file may be increased by using a lower bit rate for the less complex passages and a higher one for the more complex parts. With some encoders, it is possible to specify a given quality, and the encoder will vary the bit rate accordingly. Users who know a particular "quality setting" that is transparent to their ears can use this value when encoding all of their music, and generally speaking not need to worry about performing personal listening tests on each piece of music to determine the correct bit rate.

Q1: How many bit rates does the simplest type of MP3 file use?
A1: one
Q2: What does CBR stand for?
A2: Constant Bit Rate
Q3: What does VBR stand for?
A3: Variable Bit Rate
Q4: Aside from silence, which sections of music highlight parts that are easier to compress?
A4: music containing only a few instruments
Q5: What value do users need to know when encoding their music to help avoid them having to do tests on each piece of music?
A5: quality setting
Perceived quality can be influenced by listening environment (ambient noise), listener attention, and listener training and in most cases by listener audio equipment (such as sound cards, speakers and headphones).

Q1: A listening environment is also know by which term?
A1: ambient noise
Q2: Other than speakers and headphones, what other piece of equipment affects the listener's perceived quality?
A2: sound cards
Q3: Listening environment, listener attention, listener training and listener audio equipment can all affect what kind of quality?
A3: Perceived quality
A test given to new students by Stanford University Music Professor Jonathan Berger showed that student preference for MP3-quality music has risen each year. Berger said the students seem to prefer the 'sizzle' sounds that MP3s bring to music.

Q1: Who gave students a test to determine their preferences?
A1: Music Professor Jonathan Berger
Q2: Where was a test performed to collect data on student preference for MP3 quality?
A2: Stanford University
Q3: Data showed that students' preference for MP3 quality music followed which trend?
A3: risen each year
Q4: What kind of sound did students seem to prefer?
A4: sizzle