derek-thomas/squad-v1.1-t5-question-generation huggingface dataset:
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.

Question: As MPEG-3 had a different meaning, what was the name given to the extension of MPEG-2? Question: Where was this extension developed? Question: The new sampling rates widened the scope of MP3 to be able to include what? Question: What is not an ISO recognized standard? Question: As MPEG-2.5 is unofficial, it is considered what kind of extension to the MP3 format?
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.

Question: What defines the compression efficiency of encoders? Question: Compression ratio depends on the sample rate and bit depth of which signal? Question: What can CD parameters be used as references for? Question: What does CD stand for? Question: Other than CD parameters, what else can be used as parameter references?
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.

Question: What type of recording did Brandenburg use? Question: What did Brandenburg use the recording to refine? Question: What word describes the nature of the song used? Question: As a result of her song being used, Suzanne Vega is sometimes referred to as what? Question: Excerpts were taken from which reference compact disc to assess the subjective quality of the MPEG audio formats?
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.

Question: Which language was the reference simulation software written in? Question: Who developed the reference simulation software? Question: When was the reference software approved? Question: When did the reference software become an international standard? Question: What could the reference software demonstrate?
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.

Question: When was the first software MP3 encoder released? Question: What was the name of the first software MP3 encoder? Question: What was the filename extension? Question: What was the name of the first real-time software MP3 player? Question: What was the average high end hard drive size?
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"".

Question: What was Jonathan Sterne's profession? Question: Which nationality was the hacker who acquired I3enc? Question: What did the hacker use to acquire the software? Question: The hacker re-wrote which element of the program? Question: How much did the new released program by the hacker cost?
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.

Question: Where did MP3s begin to spread? Question: Who released the audio player? Question: Which audio player was released in 1997? Question: What was the name of the first portable solid state digital audio player? Question: In which country was the first portable audio player developed?
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.

Question: Which website offered MP3s for free? Question: How many MP3s were offered for free by the website? Question: What kind of file sharing became popular due to the file sizes of MP3s? Question: How did people come into possession of MP3s? Question: Which network was the first large peer to peer file sharing network?
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.

Question: What kind of infringement resulted from sharing MP3s? Question: What name did the record companies associate with MP3 file sharing? Question: Who was the target of the record companies? Question: What happened to Napster?
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.

Question: What kind of MP3 file sharing continues on? Question: Beatport, Bleep and Juno records are examples of what kind of service? Question: What kind of music is sold by these companies? Question: Which format is used by the companies that sell the music legally?
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.

Question: A header and data block together make up what? Question: What is a sequence of MP3 frames called? Question: What is used to identify the begining of a valid frame of an MP3 header? Question: How many bits are needed to indicate that layer 3 is used? Question: MP3 files today contain what kind of metadeta?
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.

Question: Which standard does not include a specification for an MP3 encoder? Question: Example psychoacoustic models and rate loops can be foundin which part of the original standard? Question: What were implementers of the standard supposed to devise? Question: It is easy for a prospective user of an encoder to research the best choice because of what factor? Question: An encoder that is proficient at encoding at higher bit rates might be worse at what?
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.)

Question: How many domain samples are taken during encoding? Question: Domain samples are transformed into what? Question: When are 192 samples taken instead of 576? Question: When 192 samples are taken instead of 576, it is done in an effort to limit what? Question: What accompanies a transient?
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]

Question: What kind of structure does a filter bank have? Question: Due to a specific kind of structure, what kind of problems are made worse? Question: What kind of resolution are the filter banks unable to provide an optimal solution for? Question: Combining two filter banks' outputs creates what kind of problem? Question: The need to code excess energy in the frequency domain decreases what kind of efficiency?
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).

Question: What is carefully defined in the standard? Question: What are most decoders? Question: The ISO/IEC high standard document states that the decompressed output produced from a given MP3 file will be the same within what standards? Question: Decoders are usually compared by examining which factor? Question: The efficiency of decoders is examined by seeing how much memory and what other process they use in the decoding process?
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.

Question: For encoders and decoders, what is not defined? Question: What is there no official provision for? Question: What is an example of an encoder that can attach additional metadeta? Question: Adding additional metadata allows players to handle gapless playback and deliver what?
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.

Question: What would an example of lossy audio encoding be? Question: A typical trade off when creating an MP3 file is between the amount of space used and what other factor? Question: Who gets to set the bit rate for an MP3 file? Question: What does the bit rate of the file actually mean? Question: If the bit rate is higher, the size of the compressed file will be what?
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.

Question: If the bit rate is too low, what might be audible in the reproduction? Question: Sounds that were not in the original recording are referred to as what? Question: Randomness and sharp attacks are two reasons that may make it harder to to do what to an audio file? Question: Other than pre - echo, what is another example of what can happen when audio is compressed? Question: What kind of sample can display a good example of compression artifacts?
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.

Question: Other than the bit rate and the difficulty of the signal, what can also affect the quality of an MP3 file? Question: The MP3 standard allows encoders to have a lot of freedom with which aspect of encoding? Question: As a result of the freedom, different encoders can result in different quality even if which aspect of the file is the same? Question: To compare qualities, sound files are evaluated on a scale that ranges from 1 to what? Question: An example shown has a low range quality score of what value?
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.

Question: How many bit rates does the simplest type of MP3 file use? Question: What does CBR stand for? Question: What does VBR stand for? Question: Aside from silence, which sections of music highlight parts that are easier to compress? Question: What value do users need to know when encoding their music to help avoid them having to do tests on each piece of music?
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).

Question: Listening environment, listener attention, listener training and listener audio equipment can all affect what kind of quality? Question: A listening environment is also know by which term? Question: Other than speakers and headphones, what other piece of equipment affects the listener's 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.

Question: Where was a test performed to collect data on student preference for MP3 quality? Question: Who gave students a test to determine their preferences? Question: Data showed that students' preference for MP3 quality music followed which trend? Question: What kind of sound did students seem to prefer?
An in-depth study of MP3 audio quality, sound artist and composer Ryan Maguire's project "The Ghost in the MP3" isolates the sounds lost during MP3 compression. In 2015, he released the track "moDernisT" (an anagram of "Tom's Diner"), composed exclusively from the sounds deleted during MP3 compression of the song "Tom's Diner", the track originally used in the formulation of the MP3 standard. A detailed account of the techniques used to isolate the sounds deleted during MP3 compression, along with the conceptual motivation for the project, was published in the 2014 Proceedings of the International Computer Music Conference.

Question: What is the name of the study done by composer Ryan Maguire? Question: What did the study aim to isolate? Question: What is the track "moDernisT" an anagram of? Question: Which year was the study published? Question: Where was an account of the study published?
Several bit rates are specified in the MPEG-1 Audio Layer III standard: 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kbit/s, with available sampling frequencies of 32, 44.1 and 48 kHz. MPEG-2 Audio Layer III allows bit rates of 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 kbit/s with sampling frequencies of 16, 22.05 and 24 kHz. MPEG-2.5 Audio Layer III is restricted to bit rates of 8, 16, 24, 32, 40, 48, 56 and 64 kbit/s with sampling frequencies of 8, 11.025, and 12 kHz.[citation needed] Because of the Nyquist–Shannon sampling theorem, frequency reproduction is always strictly less than half of the sampling frequency, and imperfect filters requires a larger margin for error (noise level versus sharpness of filter), so 8 kHz sampling rate limits the maximum frequency to 4 kHz, while 48 kHz maximum sampling rate limits an MP3 to 24 kHz sound reproduction.

Question: What is the name of the sampling theorem? Question: In relation to the sampling frequency, the frequency reproduction is strictly how much less? Question: What kind of filter requires a larger margin for error? Question: Having an 8 kHz sampling rate would limit the maximum frequency to how much? Question: A sound reproduction of 24 kHz would represent which value of a maximum sampling rate?
A sample rate of 44.1 kHz is almost always used, because this is also used for CD audio, the main source used for creating MP3 files. A greater variety of bit rates are used on the Internet. The rate of 128 kbit/s is commonly used, at a compression ratio of 11:1, offering adequate audio quality in a relatively small space. As Internet bandwidth availability and hard drive sizes have increased, higher bit rates up to 320 kbit/s are widespread.

Question: What sampling rate is used for CD audio? Question: What is the main source used for creating MP3 files? Question: Where can one find a greater variety of bit rates? Question: The common bit rate on the internet is 128 kbit/s using which compression ratio? Question: As technology advances, what is another higher bit rate that is beginning to spread?
Uncompressed audio as stored on an audio-CD has a bit rate of 1,411.2 kbit/s, (16 bit/sample × 44100 samples/second × 2 channels / 1000 bits/kilobit), so the bitrates 128, 160 and 192 kbit/s represent compression ratios of approximately 11:1, 9:1 and 7:1 respectively.

Question: What is stored on an audio-CD that has a bit rate of 1,411.2 kbit/s? Question: What does a bit rate of 128 represent in terms of compression ratio? Question: What would a bit rate of 160 correspond to in terms of compression ratio? Question: What would a bit rate of 192 kbit/s have for a compression ratio?
Non-standard bit rates up to 640 kbit/s can be achieved with the LAME encoder and the freeformat option, although few MP3 players can play those files. According to the ISO standard, decoders are only required to be able to decode streams up to 320 kbit/s.

Question: The LAME encoder allows bit rates that are what? Question: What is the highest bit rate the LAME encoder allows? Question: Even though few MP3 players can play it, LAME also offers which option? Question: The ISO standard dictates that decoders only need to be able to decode streams that have how high of a bit rate?
Early MPEG Layer III encoders used what is now called Constant Bit Rate (CBR). The software was only able to use a uniform bitrate on all frames in an MP3 file. Later more sophisticated MP3 encoders were able to use the bit reservoir to target an average bit rate selecting the encoding rate for each frame based on the complexity of the sound in that portion of the recording.

Question: Which early MPEG layer used CBR? Question: What does the term CBR mean? Question: The early software was restricted to which kind of bitrate? Question: What did more sophisticated MP3 encoders use to target an average bit rate? Question: What did encoders base the encoding rate for each frame on?