Question: What program requires an additional two years of study?

Answer: a doctoral programme

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. <hl> For a doctoral programme it is required to take two additional years of study. <hl>

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 <hl> a doctoral programme <hl> it is required to take two additional years of study.

question: What program requires an additional two years of study?, context: 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.


Question: What is the common name for MPEG-2 Audio Layer III?

Answer: MP3

<hl> 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. <hl> 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.

MPEG-1 or MPEG-2 Audio Layer III, more commonly referred to as <hl> MP3 <hl>, 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.

MPEG-1 or MPEG-2 Audio Layer III, more commonly referred to as <hl> MP3 <hl> , is an audio coding format for digital audio which uses a form of lossy data compression.

question: What is the common name for MPEG-2 Audio Layer III?, context: 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.


Question: What kind of audio is the coding format for?

Answer: digital

<hl> 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. <hl> 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.

MPEG-1 or MPEG-2 Audio Layer III, more commonly referred to as MP3, is an audio coding format for <hl> digital <hl> 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.

question: What kind of audio is the coding format for?, context: 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.


Question: Other than storage, what is the common audio format used for?

Answer: consumer audio streaming

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. <hl> 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. <hl>

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 <hl> consumer audio streaming <hl> 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.

question: Other than storage, what is the common audio format used for?, context: 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.


Question: What does MP3 use to format it's data?

Answer: lossy data compression

<hl> 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 . <hl> 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.

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 <hl> lossy data compression <hl>. 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.

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 <hl> lossy data compression <hl> .

question: What does MP3 use to format it's data?, context: 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.


Question: Where will MP3 files most commonly be found?

Answer: digital audio players

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. <hl> 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 . <hl>

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 <hl> digital audio players <hl>.

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 <hl> digital audio players <hl> .

question: Where will MP3 files most commonly be found?, context: 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.


Question: What is the main goal aside from reducing the amount of data required to store the audio?

Answer: sound like a faithful reproduction

<hl> 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. <hl> 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.

The use of lossy compression is designed to greatly reduce the amount of data required to represent the audio recording and still <hl> sound like a faithful reproduction <hl> 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.

question: What is the main goal aside from reducing the amount of data required to store the audio?, context: 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.


Question: What kind of compression is used?

Answer: created

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. <hl> 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 <hl> [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.

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 <hl> created <hl> 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.

question: What kind of compression is used?, context: 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.


Question: If a file is created using 128 kbit/s, what size is the file going to be compared to a CD?

Answer: 1/11

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. <hl> 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 <hl> [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.

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 <hl> 1/11 <hl> 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.

question: If a file is created using 128 kbit/s, what size is the file going to be compared to a CD?, context: 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.


Question: How many total bit/s would a CD have?

Answer: 1,411,200

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. <hl> 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 <hl> [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.

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 = <hl> 1,411,200 <hl> 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.

question: How many total bit/s would a CD have?, context: 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.


Question: How many total bit/s would an MP3 compressed at 128 kbit/s have?

Answer: 128,000

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. <hl> 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]. <hl> 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.

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: <hl> 128,000 <hl> 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.

question: How many total bit/s would an MP3 compressed at 128 kbit/s have?, context: 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.


Question: What does the compression target in the song?

Answer: accuracy

<hl> 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. <hl> 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.

The compression works by reducing the <hl> accuracy <hl> 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.

question: What does the compression target in the song?, context: 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.


Question: The targetting focuses on what aspect that is beyond the ability of most people?

Answer: auditory resolution

<hl> 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. <hl> 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.

The compression works by reducing the accuracy of certain parts of a sound that are considered to be beyond the <hl> auditory resolution <hl> 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.

question: The targetting focuses on what aspect that is beyond the ability of most people?, context: 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.


Question: What is this method commonly referred to as?

Answer: perceptual coding

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. <hl> This method is commonly referred to as perceptual coding . <hl> 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.

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 <hl> perceptual coding <hl>. 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.

This method is commonly referred to as <hl> perceptual coding <hl> .

question: What is this method commonly referred to as?, context: 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.


Question: What kind of model is used in this kind of compression?

Answer: psychoacoustic

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. <hl> 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. <hl>

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 <hl> psychoacoustic <hl> models to discard or reduce precision of components less audible to human hearing, and then records the remaining information in an efficient manner.

question: What kind of model is used in this kind of compression?, context: 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.


Question: These models ensure that the precision is reduced for the components that are what to human hearing?

Answer: less audible

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. <hl> 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. <hl>

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 <hl> less audible <hl> to human hearing, and then records the remaining information in an efficient manner.

question: These models ensure that the precision is reduced for the components that are what to human hearing?, context: 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.


Question: Who designed MP3?

Answer: Moving Picture Experts Group

<hl> 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. <hl> 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).

MP3 was designed by the <hl> Moving Picture Experts Group <hl> (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).

question: Who designed MP3?, context: 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).


Question: What was the first standard that this group had?

Answer: MPEG-1

<hl> 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. <hl> 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).

MP3 was designed by the Moving Picture Experts Group (MPEG) as part of its <hl> MPEG-1 <hl> 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).

question: What was the first standard that this group had?, context: 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).


Question: The first standard later developed into which standard?

Answer: MPEG-2

<hl> 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. <hl> 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).

MP3 was designed by the Moving Picture Experts Group (MPEG) as part of its MPEG-1 standard and later extended in the <hl> MPEG-2 <hl> 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).

question: The first standard later developed into which standard?, context: 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).


Question: When was the draft passed to have MPEG-1 Audio approved as a standard?

Answer: 1991

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. <hl> 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). <hl> 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).

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 <hl> 1991 <hl>, 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).

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 <hl> 1991 <hl> , finalised in 1992 and published in 1993 (ISO/IEC 11172-3:1993).

question: When was the draft passed to have MPEG-1 Audio approved as a standard?, context: 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).


Question: When was MPEG-2 Audio finally published?

Answer: 1995

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). <hl> 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). <hl>

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 <hl> 1995 <hl> (ISO/IEC 13818-3:1995).

question: When was MPEG-2 Audio finally published?, context: 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).


Question: What does the MP3 compression take advantage of?

Answer: perceptual limitation

<hl> The MP3 lossy audio data compression algorithm takes advantage of a perceptual limitation of human hearing called auditory masking. <hl> 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.

The MP3 lossy audio data compression algorithm takes advantage of a <hl> perceptual limitation <hl> 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.

question: What does the MP3 compression take advantage of?, context: 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.


Question: What is the limitation in human hearing referred to as?

Answer: auditory masking

<hl> The MP3 lossy audio data compression algorithm takes advantage of a perceptual limitation of human hearing called auditory masking . <hl> 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.

The MP3 lossy audio data compression algorithm takes advantage of a perceptual limitation of human hearing called <hl> auditory masking <hl>. 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.

The MP3 lossy audio data compression algorithm takes advantage of a perceptual limitation of human hearing called <hl> auditory masking <hl> .

question: What is the limitation in human hearing referred to as?, context: 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.


Question: Who was the physicist to report that a tone could be rendered inaudible.

Answer: Alfred M. Mayer

The MP3 lossy audio data compression algorithm takes advantage of a perceptual limitation of human hearing called auditory masking. <hl> In 1894, the American physicist Alfred M. Mayer reported that a tone could be rendered inaudible by another tone of lower frequency. <hl> 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.

The MP3 lossy audio data compression algorithm takes advantage of a perceptual limitation of human hearing called auditory masking. In 1894, the American physicist <hl> Alfred M. Mayer <hl> 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.

question: Who was the physicist to report that a tone could be rendered inaudible., context: 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.


Question: When were audio curves described by Richard Ehmer?

Answer: 1959

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. <hl> In 1959 , Richard Ehmer described a complete set of auditory curves regarding this phenomenon. <hl> 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.

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 <hl> 1959 <hl>, 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.

In <hl> 1959 <hl> , Richard Ehmer described a complete set of auditory curves regarding this phenomenon.

question: When were audio curves described by Richard Ehmer?, context: 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.


Question: The initial work was aimed at determining critical ratios and what else?

Answer: critical bandwidths

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. <hl> 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 . <hl>

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 <hl> critical bandwidths <hl>.

This work added to a variety of reports from authors dating back to Fletcher, and to the work that initially determined critical ratios and <hl> critical bandwidths <hl> .

question: The initial work was aimed at determining critical ratios and what else?, context: 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.


Question: What was first proposed in 1979?

Answer: The psychoacoustic masking codec

<hl> 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. <hl> 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.

<hl> The psychoacoustic masking codec <hl> 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.

question: What was first proposed in 1979?, context: 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.


Question: Which country were the researchers located in?

Answer: United States

The psychoacoustic masking codec was first proposed in 1979, apparently independently, by Manfred R. Schroeder, et al. <hl> from Bell Telephone Laboratories, Inc. in Murray Hill, NJ, and M. A. Krasner both in the United States . <hl> 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.

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 <hl> United States <hl>. 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.

from Bell Telephone Laboratories, Inc. in Murray Hill, NJ, and M. A. Krasner both in the <hl> United States <hl> .

question: Which country were the researchers located in?, context: 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.


Question: Who was the first to produce hardware for speech?

Answer: Krasner

Krasner both in the United States.

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. <hl> Krasner both in the United States. <hl> 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.

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. <hl> Krasner <hl> 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.

question: Who was the first to produce hardware for speech?, context: 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.


Question: What does LPC stand for?

Answer: linear predictive coding

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 <hl> the linear predictive coding (LPC) gain present in speech. <hl> 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.

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 <hl> linear predictive coding <hl> (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.

question: What does LPC stand for?, context: 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.


Question: What was reported in IEEE's Journal on Selected Areas in Communications?

Answer: audio compression algorithms

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. <hl> A wide variety of (mostly perceptual) audio compression algorithms were reported in IEEE's refereed Journal on Selected Areas in Communications. <hl> 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.

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) <hl> audio compression algorithms <hl> 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.

question: What was reported in IEEE's Journal on Selected Areas in Communications?, context: 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.


Question: What does OCF stand for?

Answer: Optimum Coding in the Frequency Domain

<hl> The immediate predecessors of MP3 were " Optimum Coding in the Frequency Domain " (OCF), and Perceptual Transform Coding (PXFM). <hl> 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.

The immediate predecessors of MP3 were "<hl> Optimum Coding in the Frequency Domain <hl>" (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.

The immediate predecessors of MP3 were " <hl> Optimum Coding in the Frequency Domain <hl> " (OCF), and Perceptual Transform Coding (PXFM).

question: What does OCF stand for?, context: 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.


Question: What does PXFM stand for?

Answer: Perceptual Transform Coding

<hl> The immediate predecessors of MP3 were "Optimum Coding in the Frequency Domain" (OCF), and Perceptual Transform Coding (PXFM). <hl> 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.

The immediate predecessors of MP3 were "Optimum Coding in the Frequency Domain" (OCF), and <hl> Perceptual Transform Coding <hl> (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.

question: What does PXFM stand for?, context: 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.


Question: What was the name of the codec that OCF, PFXM and other contributes merged into?

Answer: ASPEC

The immediate predecessors of MP3 were "Optimum Coding in the Frequency Domain" (OCF), and Perceptual Transform Coding (PXFM). <hl> 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. <hl> 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.

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 <hl> ASPEC <hl>, 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.

These two codecs, along with block-switching contributions from Thomson-Brandt, were merged into a codec called <hl> ASPEC <hl> , which was submitted to MPEG, and which won the quality competition, but that was mistakenly rejected as too complex to implement.

question: What was the name of the codec that OCF, PFXM and other contributes merged into?, context: 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.


Question: Which competition did ASPEC win?

Answer: quality

The immediate predecessors of MP3 were "Optimum Coding in the Frequency Domain" (OCF), and Perceptual Transform Coding (PXFM). <hl> 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. <hl> 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.

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 <hl> quality <hl> 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.

question: Which competition did ASPEC win?, context: 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.


Question: What was the first practical implementation based on?

Answer: Motorola 56000 DSP chips

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. <hl> 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 . <hl>

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 <hl> Motorola 56000 DSP chips <hl>.

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 <hl> Motorola 56000 DSP chips <hl> .

question: What was the first practical implementation based on?, context: 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.


Question: What kind of student was Karlheinz Brandenburg?

Answer: doctoral

<hl> 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. <hl> 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).

As a <hl> doctoral <hl> 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).

question: What kind of student was Karlheinz Brandenburg?, context: 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).


Question: When did Karlheinz start working on digitmal music compression?

Answer: early 1980s

<hl> 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. <hl> 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).

As a doctoral student at Germany's University of Erlangen-Nuremberg, Karlheinz Brandenburg began working on digital music compression in the <hl> early 1980s <hl>, 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).

As a doctoral student at Germany's University of Erlangen-Nuremberg, Karlheinz Brandenburg began working on digital music compression in the <hl> early 1980s <hl> , focusing on how people perceive music.

question: When did Karlheinz start working on digitmal music compression?, context: 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).


Question: Where did Bradenburg become an assistant professor at?

Answer: Erlangen-Nuremberg

<hl> 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. <hl> 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).

As a doctoral student at Germany's University of <hl> Erlangen-Nuremberg <hl>, 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).

As a doctoral student at Germany's University of <hl> Erlangen-Nuremberg <hl> , Karlheinz Brandenburg began working on digital music compression in the early 1980s, focusing on how people perceive music.

question: Where did Bradenburg become an assistant professor at?, context: 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).


Question: Who did Brandenburg work with in music compression?

Answer: Fraunhofer Society

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. <hl> 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). <hl>

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 <hl> Fraunhofer Society <hl> (in 1993 he joined the staff of the Fraunhofer Institute).

question: Who did Brandenburg work with in music compression?, context: 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).


Question: When did Brandenburg join the Fraunhofen Institute?

Answer: 1993

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. <hl> 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). <hl>

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 <hl> 1993 <hl> he joined the staff of the Fraunhofer Institute).

question: When did Brandenburg join the Fraunhofen Institute?, context: 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).


Question: What was the name of the first song used to develop the MP3?

Answer: Tom's Diner

<hl> The song " Tom's Diner " by Suzanne Vega was the first song used by Karlheinz Brandenburg to develop the MP3. <hl> 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.

The song "<hl> Tom's Diner <hl>" 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.

The song " <hl> Tom's Diner <hl> " by Suzanne Vega was the first song used by Karlheinz Brandenburg to develop the MP3.

question: What was the name of the first song used to develop the MP3?, context: 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.


Question: Who was the artist of the first song used?

Answer: Suzanne Vega

<hl> The song "Tom's Diner" by Suzanne Vega was the first song used by Karlheinz Brandenburg to develop the MP3. <hl> 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.

The song "Tom's Diner" by <hl> Suzanne Vega <hl> 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.

question: Who was the artist of the first song used?, context: 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.


Question: Who adjusted the song?

Answer: Karlheinz Brandenburg

<hl> The song "Tom's Diner" by Suzanne Vega was the first song used by Karlheinz Brandenburg to develop the MP3. <hl> 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.

The song "Tom's Diner" by Suzanne Vega was the first song used by <hl> Karlheinz Brandenburg <hl> 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.

question: Who adjusted the song?, context: 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.