Question: The song was listened to multiple times to try and refine what?

Answer: scheme

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. <hl>

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 <hl> scheme <hl>, making sure it did not adversely affect the subtlety of Vega's voice.

Brandenburg adopted the song for testing purposes, listening to it again and again each time refining the <hl> scheme <hl> , making sure it did not adversely affect the subtlety of Vega's voice.

question: The song was listened to multiple times to try and refine what?, 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: What particular aspect of the artists voice was he trying to preserve?

Answer: subtlety

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. <hl>

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 <hl> subtlety <hl> of Vega's voice.

question: What particular aspect of the artists voice was he trying to preserve?, 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: How many proposals were available in 1991?

Answer: two

<hl> In 1991, there were only two proposals available that could be completely assessed for an MPEG audio standard: Musicam (Masking pattern adapted Universal Subband Integrated Coding And Multiplexing) and ASPEC (Adaptive Spectral Perceptual Entropy Coding). <hl> The Musicam technique, as proposed by Philips (the Netherlands), CCETT (France) and Institut für Rundfunktechnik (Germany) was chosen due to its simplicity and error robustness, as well as its low computational power associated with the encoding of high quality compressed audio. The Musicam format, based on sub-band coding, was the basis of the MPEG Audio compression format (sampling rates, structure of frames, headers, number of samples per frame).

In 1991, there were only <hl> two <hl> proposals available that could be completely assessed for an MPEG audio standard: Musicam (Masking pattern adapted Universal Subband Integrated Coding And Multiplexing) and ASPEC (Adaptive Spectral Perceptual Entropy Coding). The Musicam technique, as proposed by Philips (the Netherlands), CCETT (France) and Institut für Rundfunktechnik (Germany) was chosen due to its simplicity and error robustness, as well as its low computational power associated with the encoding of high quality compressed audio. The Musicam format, based on sub-band coding, was the basis of the MPEG Audio compression format (sampling rates, structure of frames, headers, number of samples per frame).

question: How many proposals were available in 1991?, context: In 1991, there were only two proposals available that could be completely assessed for an MPEG audio standard: Musicam (Masking pattern adapted Universal Subband Integrated Coding And Multiplexing) and ASPEC (Adaptive Spectral Perceptual Entropy Coding). The Musicam technique, as proposed by Philips (the Netherlands), CCETT (France) and Institut für Rundfunktechnik (Germany) was chosen due to its simplicity and error robustness, as well as its low computational power associated with the encoding of high quality compressed audio. The Musicam format, based on sub-band coding, was the basis of the MPEG Audio compression format (sampling rates, structure of frames, headers, number of samples per frame).


Question: Other than ASPEC what was another proposal?

Answer: Musicam

<hl> In 1991, there were only two proposals available that could be completely assessed for an MPEG audio standard: Musicam (Masking pattern adapted Universal Subband Integrated Coding And Multiplexing) and ASPEC (Adaptive Spectral Perceptual Entropy Coding). <hl> The Musicam technique, as proposed by Philips (the Netherlands), CCETT (France) and Institut für Rundfunktechnik (Germany) was chosen due to its simplicity and error robustness, as well as its low computational power associated with the encoding of high quality compressed audio. The Musicam format, based on sub-band coding, was the basis of the MPEG Audio compression format (sampling rates, structure of frames, headers, number of samples per frame).

In 1991, there were only two proposals available that could be completely assessed for an MPEG audio standard: <hl> Musicam <hl> (Masking pattern adapted Universal Subband Integrated Coding And Multiplexing) and ASPEC (Adaptive Spectral Perceptual Entropy Coding). The Musicam technique, as proposed by Philips (the Netherlands), CCETT (France) and Institut für Rundfunktechnik (Germany) was chosen due to its simplicity and error robustness, as well as its low computational power associated with the encoding of high quality compressed audio. The Musicam format, based on sub-band coding, was the basis of the MPEG Audio compression format (sampling rates, structure of frames, headers, number of samples per frame).

question: Other than ASPEC what was another proposal?, context: In 1991, there were only two proposals available that could be completely assessed for an MPEG audio standard: Musicam (Masking pattern adapted Universal Subband Integrated Coding And Multiplexing) and ASPEC (Adaptive Spectral Perceptual Entropy Coding). The Musicam technique, as proposed by Philips (the Netherlands), CCETT (France) and Institut für Rundfunktechnik (Germany) was chosen due to its simplicity and error robustness, as well as its low computational power associated with the encoding of high quality compressed audio. The Musicam format, based on sub-band coding, was the basis of the MPEG Audio compression format (sampling rates, structure of frames, headers, number of samples per frame).


Question: The Musicam format was proposed by Philips, based in which country?

Answer: Netherlands

In 1991, there were only two proposals available that could be completely assessed for an MPEG audio standard: Musicam (Masking pattern adapted Universal Subband Integrated Coding And Multiplexing) and ASPEC (Adaptive Spectral Perceptual Entropy Coding). <hl> The Musicam technique, as proposed by Philips (the Netherlands ), CCETT (France) and Institut für Rundfunktechnik (Germany) was chosen due to its simplicity and error robustness, as well as its low computational power associated with the encoding of high quality compressed audio. <hl> The Musicam format, based on sub-band coding, was the basis of the MPEG Audio compression format (sampling rates, structure of frames, headers, number of samples per frame).

In 1991, there were only two proposals available that could be completely assessed for an MPEG audio standard: Musicam (Masking pattern adapted Universal Subband Integrated Coding And Multiplexing) and ASPEC (Adaptive Spectral Perceptual Entropy Coding). The Musicam technique, as proposed by Philips (the <hl> Netherlands <hl>), CCETT (France) and Institut für Rundfunktechnik (Germany) was chosen due to its simplicity and error robustness, as well as its low computational power associated with the encoding of high quality compressed audio. The Musicam format, based on sub-band coding, was the basis of the MPEG Audio compression format (sampling rates, structure of frames, headers, number of samples per frame).

The Musicam technique, as proposed by Philips (the <hl> Netherlands <hl> ), CCETT (France) and Institut für Rundfunktechnik (Germany) was chosen due to its simplicity and error robustness, as well as its low computational power associated with the encoding of high quality compressed audio.

question: The Musicam format was proposed by Philips, based in which country?, context: In 1991, there were only two proposals available that could be completely assessed for an MPEG audio standard: Musicam (Masking pattern adapted Universal Subband Integrated Coding And Multiplexing) and ASPEC (Adaptive Spectral Perceptual Entropy Coding). The Musicam technique, as proposed by Philips (the Netherlands), CCETT (France) and Institut für Rundfunktechnik (Germany) was chosen due to its simplicity and error robustness, as well as its low computational power associated with the encoding of high quality compressed audio. The Musicam format, based on sub-band coding, was the basis of the MPEG Audio compression format (sampling rates, structure of frames, headers, number of samples per frame).


Question: Which other quality alongside simplicty was key in selecting this proposal?

Answer: error robustness

In 1991, there were only two proposals available that could be completely assessed for an MPEG audio standard: Musicam (Masking pattern adapted Universal Subband Integrated Coding And Multiplexing) and ASPEC (Adaptive Spectral Perceptual Entropy Coding). <hl> The Musicam technique, as proposed by Philips (the Netherlands), CCETT (France) and Institut für Rundfunktechnik (Germany) was chosen due to its simplicity and error robustness , as well as its low computational power associated with the encoding of high quality compressed audio. <hl> The Musicam format, based on sub-band coding, was the basis of the MPEG Audio compression format (sampling rates, structure of frames, headers, number of samples per frame).

In 1991, there were only two proposals available that could be completely assessed for an MPEG audio standard: Musicam (Masking pattern adapted Universal Subband Integrated Coding And Multiplexing) and ASPEC (Adaptive Spectral Perceptual Entropy Coding). The Musicam technique, as proposed by Philips (the Netherlands), CCETT (France) and Institut für Rundfunktechnik (Germany) was chosen due to its simplicity and <hl> error robustness <hl>, as well as its low computational power associated with the encoding of high quality compressed audio. The Musicam format, based on sub-band coding, was the basis of the MPEG Audio compression format (sampling rates, structure of frames, headers, number of samples per frame).

The Musicam technique, as proposed by Philips (the Netherlands), CCETT (France) and Institut für Rundfunktechnik (Germany) was chosen due to its simplicity and <hl> error robustness <hl> , as well as its low computational power associated with the encoding of high quality compressed audio.

question: Which other quality alongside simplicty was key in selecting this proposal?, context: In 1991, there were only two proposals available that could be completely assessed for an MPEG audio standard: Musicam (Masking pattern adapted Universal Subband Integrated Coding And Multiplexing) and ASPEC (Adaptive Spectral Perceptual Entropy Coding). The Musicam technique, as proposed by Philips (the Netherlands), CCETT (France) and Institut für Rundfunktechnik (Germany) was chosen due to its simplicity and error robustness, as well as its low computational power associated with the encoding of high quality compressed audio. The Musicam format, based on sub-band coding, was the basis of the MPEG Audio compression format (sampling rates, structure of frames, headers, number of samples per frame).


Question: What was the Musican format based on?

Answer: sub-band coding

In 1991, there were only two proposals available that could be completely assessed for an MPEG audio standard: Musicam (Masking pattern adapted Universal Subband Integrated Coding And Multiplexing) and ASPEC (Adaptive Spectral Perceptual Entropy Coding). The Musicam technique, as proposed by Philips (the Netherlands), CCETT (France) and Institut für Rundfunktechnik (Germany) was chosen due to its simplicity and error robustness, as well as its low computational power associated with the encoding of high quality compressed audio. <hl> The Musicam format, based on sub-band coding , was the basis of the MPEG Audio compression format (sampling rates, structure of frames, headers, number of samples per frame). <hl>

In 1991, there were only two proposals available that could be completely assessed for an MPEG audio standard: Musicam (Masking pattern adapted Universal Subband Integrated Coding And Multiplexing) and ASPEC (Adaptive Spectral Perceptual Entropy Coding). The Musicam technique, as proposed by Philips (the Netherlands), CCETT (France) and Institut für Rundfunktechnik (Germany) was chosen due to its simplicity and error robustness, as well as its low computational power associated with the encoding of high quality compressed audio. The Musicam format, based on <hl> sub-band coding <hl>, was the basis of the MPEG Audio compression format (sampling rates, structure of frames, headers, number of samples per frame).

The Musicam format, based on <hl> sub-band coding <hl> , was the basis of the MPEG Audio compression format (sampling rates, structure of frames, headers, number of samples per frame).

question: What was the Musican format based on?, context: In 1991, there were only two proposals available that could be completely assessed for an MPEG audio standard: Musicam (Masking pattern adapted Universal Subband Integrated Coding And Multiplexing) and ASPEC (Adaptive Spectral Perceptual Entropy Coding). The Musicam technique, as proposed by Philips (the Netherlands), CCETT (France) and Institut für Rundfunktechnik (Germany) was chosen due to its simplicity and error robustness, as well as its low computational power associated with the encoding of high quality compressed audio. The Musicam format, based on sub-band coding, was the basis of the MPEG Audio compression format (sampling rates, structure of frames, headers, number of samples per frame).


Question: What was only incorporated into Layer III and not Layers I or II?

Answer: filter bank

<hl> Much of its technology and ideas were incorporated into the definition of ISO MPEG Audio Layer I and Layer II and the filter bank alone into Layer III (MP3) format as part of the computationally inefficient hybrid filter bank. <hl> Under the chairmanship of Professor Musmann (University of Hannover) the editing of the standard was made under the responsibilities of Leon van de Kerkhof (Layer I) and Gerhard Stoll (Layer II).

Much of its technology and ideas were incorporated into the definition of ISO MPEG Audio Layer I and Layer II and the <hl> filter bank <hl> alone into Layer III (MP3) format as part of the computationally inefficient hybrid filter bank. Under the chairmanship of Professor Musmann (University of Hannover) the editing of the standard was made under the responsibilities of Leon van de Kerkhof (Layer I) and Gerhard Stoll (Layer II).

question: What was only incorporated into Layer III and not Layers I or II?, context: Much of its technology and ideas were incorporated into the definition of ISO MPEG Audio Layer I and Layer II and the filter bank alone into Layer III (MP3) format as part of the computationally inefficient hybrid filter bank. Under the chairmanship of Professor Musmann (University of Hannover) the editing of the standard was made under the responsibilities of Leon van de Kerkhof (Layer I) and Gerhard Stoll (Layer II).


Question: Who was the chair that oversaw the editing of the standard?

Answer: Professor Musmann

Much of its technology and ideas were incorporated into the definition of ISO MPEG Audio Layer I and Layer II and the filter bank alone into Layer III (MP3) format as part of the computationally inefficient hybrid filter bank. <hl> Under the chairmanship of Professor Musmann (University of Hannover) the editing of the standard was made under the responsibilities of Leon van de Kerkhof (Layer I) and Gerhard Stoll (Layer II). <hl>

Much of its technology and ideas were incorporated into the definition of ISO MPEG Audio Layer I and Layer II and the filter bank alone into Layer III (MP3) format as part of the computationally inefficient hybrid filter bank. Under the chairmanship of <hl> Professor Musmann <hl> (University of Hannover) the editing of the standard was made under the responsibilities of Leon van de Kerkhof (Layer I) and Gerhard Stoll (Layer II).

question: Who was the chair that oversaw the editing of the standard?, context: Much of its technology and ideas were incorporated into the definition of ISO MPEG Audio Layer I and Layer II and the filter bank alone into Layer III (MP3) format as part of the computationally inefficient hybrid filter bank. Under the chairmanship of Professor Musmann (University of Hannover) the editing of the standard was made under the responsibilities of Leon van de Kerkhof (Layer I) and Gerhard Stoll (Layer II).


Question: Who's responsibility was it to edit the standard for Layer I?

Answer: Leon van de Kerkhof

Much of its technology and ideas were incorporated into the definition of ISO MPEG Audio Layer I and Layer II and the filter bank alone into Layer III (MP3) format as part of the computationally inefficient hybrid filter bank. <hl> Under the chairmanship of Professor Musmann (University of Hannover) the editing of the standard was made under the responsibilities of Leon van de Kerkhof (Layer I) and Gerhard Stoll (Layer II). <hl>

Much of its technology and ideas were incorporated into the definition of ISO MPEG Audio Layer I and Layer II and the filter bank alone into Layer III (MP3) format as part of the computationally inefficient hybrid filter bank. Under the chairmanship of Professor Musmann (University of Hannover) the editing of the standard was made under the responsibilities of <hl> Leon van de Kerkhof <hl> (Layer I) and Gerhard Stoll (Layer II).

question: Who's responsibility was it to edit the standard for Layer I?, context: Much of its technology and ideas were incorporated into the definition of ISO MPEG Audio Layer I and Layer II and the filter bank alone into Layer III (MP3) format as part of the computationally inefficient hybrid filter bank. Under the chairmanship of Professor Musmann (University of Hannover) the editing of the standard was made under the responsibilities of Leon van de Kerkhof (Layer I) and Gerhard Stoll (Layer II).


Question: Who's responsibility was it to edit the standard for Layer II?

Answer: Gerhard Stoll

Much of its technology and ideas were incorporated into the definition of ISO MPEG Audio Layer I and Layer II and the filter bank alone into Layer III (MP3) format as part of the computationally inefficient hybrid filter bank. <hl> Under the chairmanship of Professor Musmann (University of Hannover) the editing of the standard was made under the responsibilities of Leon van de Kerkhof (Layer I) and Gerhard Stoll (Layer II). <hl>

Much of its technology and ideas were incorporated into the definition of ISO MPEG Audio Layer I and Layer II and the filter bank alone into Layer III (MP3) format as part of the computationally inefficient hybrid filter bank. Under the chairmanship of Professor Musmann (University of Hannover) the editing of the standard was made under the responsibilities of Leon van de Kerkhof (Layer I) and <hl> Gerhard Stoll <hl> (Layer II).

question: Who's responsibility was it to edit the standard for Layer II?, context: Much of its technology and ideas were incorporated into the definition of ISO MPEG Audio Layer I and Layer II and the filter bank alone into Layer III (MP3) format as part of the computationally inefficient hybrid filter bank. Under the chairmanship of Professor Musmann (University of Hannover) the editing of the standard was made under the responsibilities of Leon van de Kerkhof (Layer I) and Gerhard Stoll (Layer II).


Question: What was the name given to the proposal?

Answer: ASPEC

<hl> ASPEC was the joint proposal of AT&T Bell Laboratories, Thomson Consumer Electronics, Fraunhofer Society and CNET. <hl> It provided the highest coding efficiency.

<hl> ASPEC <hl> was the joint proposal of AT&T Bell Laboratories, Thomson Consumer Electronics, Fraunhofer Society and CNET. It provided the highest coding efficiency.

question: What was the name given to the proposal?, context: ASPEC was the joint proposal of AT&T Bell Laboratories, Thomson Consumer Electronics, Fraunhofer Society and CNET. It provided the highest coding efficiency.


Question: What did the joint proposal provide?

Answer: the highest coding efficiency

ASPEC was the joint proposal of AT&T Bell Laboratories, Thomson Consumer Electronics, Fraunhofer Society and CNET. <hl> It provided the highest coding efficiency . <hl>

ASPEC was the joint proposal of AT&T Bell Laboratories, Thomson Consumer Electronics, Fraunhofer Society and CNET. It provided <hl> the highest coding efficiency <hl>.

It provided <hl> the highest coding efficiency <hl> .

question: What did the joint proposal provide?, context: ASPEC was the joint proposal of AT&T Bell Laboratories, Thomson Consumer Electronics, Fraunhofer Society and CNET. It provided the highest coding efficiency.


Question: Other than Thomson Consumer Electronics, Fraunhofer Society and CNET, who else was a part of the joint proposal?

Answer: AT&T Bell Laboratories

<hl> ASPEC was the joint proposal of AT&T Bell Laboratories , Thomson Consumer Electronics, Fraunhofer Society and CNET. <hl> It provided the highest coding efficiency.

ASPEC was the joint proposal of <hl> AT&T Bell Laboratories <hl>, Thomson Consumer Electronics, Fraunhofer Society and CNET. It provided the highest coding efficiency.

ASPEC was the joint proposal of <hl> AT&T Bell Laboratories <hl> , Thomson Consumer Electronics, Fraunhofer Society and CNET.

question: Other than Thomson Consumer Electronics, Fraunhofer Society and CNET, who else was a part of the joint proposal?, context: ASPEC was the joint proposal of AT&T Bell Laboratories, Thomson Consumer Electronics, Fraunhofer Society and CNET. It provided the highest coding efficiency.


Question: What did the working group integrate their ideas with?

Answer: filter bank

<hl> A working group consisting of Leon van de Kerkhof (The Netherlands), Gerhard Stoll (Germany), Leonardo Chiariglione (Italy), Yves-François Dehery (France), Karlheinz Brandenburg (Germany) and James D. Johnston (USA) took ideas from ASPEC, integrated the filter bank from Layer 2, added some of their own ideas and created MP3, which was designed to achieve the same quality at 128 kbit/s as MP2 at 192 kbit/s. <hl>

A working group consisting of Leon van de Kerkhof (The Netherlands), Gerhard Stoll (Germany), Leonardo Chiariglione (Italy), Yves-François Dehery (France), Karlheinz Brandenburg (Germany) and James D. Johnston (USA) took ideas from ASPEC, integrated the <hl> filter bank <hl> from Layer 2, added some of their own ideas and created MP3, which was designed to achieve the same quality at 128 kbit/s as MP2 at 192 kbit/s.

question: What did the working group integrate their ideas with?, context: A working group consisting of Leon van de Kerkhof (The Netherlands), Gerhard Stoll (Germany), Leonardo Chiariglione (Italy), Yves-François Dehery (France), Karlheinz Brandenburg (Germany) and James D. Johnston (USA) took ideas from ASPEC, integrated the filter bank from Layer 2, added some of their own ideas and created MP3, which was designed to achieve the same quality at 128 kbit/s as MP2 at 192 kbit/s.


Question: Where was the filter bank taken from?

Answer: Layer 2

<hl> A working group consisting of Leon van de Kerkhof (The Netherlands), Gerhard Stoll (Germany), Leonardo Chiariglione (Italy), Yves-François Dehery (France), Karlheinz Brandenburg (Germany) and James D. Johnston (USA) took ideas from ASPEC, integrated the filter bank from Layer 2 , added some of their own ideas and created MP3, which was designed to achieve the same quality at 128 kbit/s as MP2 at 192 kbit/s. <hl>

A working group consisting of Leon van de Kerkhof (The Netherlands), Gerhard Stoll (Germany), Leonardo Chiariglione (Italy), Yves-François Dehery (France), Karlheinz Brandenburg (Germany) and James D. Johnston (USA) took ideas from ASPEC, integrated the filter bank from <hl> Layer 2 <hl>, added some of their own ideas and created MP3, which was designed to achieve the same quality at 128 kbit/s as MP2 at 192 kbit/s.

A working group consisting of Leon van de Kerkhof (The Netherlands), Gerhard Stoll (Germany), Leonardo Chiariglione (Italy), Yves-François Dehery (France), Karlheinz Brandenburg (Germany) and James D. Johnston (USA) took ideas from ASPEC, integrated the filter bank from <hl> Layer 2 <hl> , added some of their own ideas and created MP3, which was designed to achieve the same quality at 128 kbit/s as MP2 at 192 kbit/s.

question: Where was the filter bank taken from?, context: A working group consisting of Leon van de Kerkhof (The Netherlands), Gerhard Stoll (Germany), Leonardo Chiariglione (Italy), Yves-François Dehery (France), Karlheinz Brandenburg (Germany) and James D. Johnston (USA) took ideas from ASPEC, integrated the filter bank from Layer 2, added some of their own ideas and created MP3, which was designed to achieve the same quality at 128 kbit/s as MP2 at 192 kbit/s.


Question: What did the working group take ideas from?

Answer: ASPEC

<hl> A working group consisting of Leon van de Kerkhof (The Netherlands), Gerhard Stoll (Germany), Leonardo Chiariglione (Italy), Yves-François Dehery (France), Karlheinz Brandenburg (Germany) and James D. Johnston (USA) took ideas from ASPEC , integrated the filter bank from Layer 2, added some of their own ideas and created MP3, which was designed to achieve the same quality at 128 kbit/s as MP2 at 192 kbit/s. <hl>

A working group consisting of Leon van de Kerkhof (The Netherlands), Gerhard Stoll (Germany), Leonardo Chiariglione (Italy), Yves-François Dehery (France), Karlheinz Brandenburg (Germany) and James D. Johnston (USA) took ideas from <hl> ASPEC <hl>, integrated the filter bank from Layer 2, added some of their own ideas and created MP3, which was designed to achieve the same quality at 128 kbit/s as MP2 at 192 kbit/s.

A working group consisting of Leon van de Kerkhof (The Netherlands), Gerhard Stoll (Germany), Leonardo Chiariglione (Italy), Yves-François Dehery (France), Karlheinz Brandenburg (Germany) and James D. Johnston (USA) took ideas from <hl> ASPEC <hl> , integrated the filter bank from Layer 2, added some of their own ideas and created MP3, which was designed to achieve the same quality at 128 kbit/s as MP2 at 192 kbit/s.

question: What did the working group take ideas from?, context: A working group consisting of Leon van de Kerkhof (The Netherlands), Gerhard Stoll (Germany), Leonardo Chiariglione (Italy), Yves-François Dehery (France), Karlheinz Brandenburg (Germany) and James D. Johnston (USA) took ideas from ASPEC, integrated the filter bank from Layer 2, added some of their own ideas and created MP3, which was designed to achieve the same quality at 128 kbit/s as MP2 at 192 kbit/s.


Question: What did the working group eventually create?

Answer: MP3

<hl> A working group consisting of Leon van de Kerkhof (The Netherlands), Gerhard Stoll (Germany), Leonardo Chiariglione (Italy), Yves-François Dehery (France), Karlheinz Brandenburg (Germany) and James D. Johnston (USA) took ideas from ASPEC, integrated the filter bank from Layer 2, added some of their own ideas and created MP3 , which was designed to achieve the same quality at 128 kbit/s as MP2 at 192 kbit/s. <hl>

A working group consisting of Leon van de Kerkhof (The Netherlands), Gerhard Stoll (Germany), Leonardo Chiariglione (Italy), Yves-François Dehery (France), Karlheinz Brandenburg (Germany) and James D. Johnston (USA) took ideas from ASPEC, integrated the filter bank from Layer 2, added some of their own ideas and created <hl> MP3 <hl>, which was designed to achieve the same quality at 128 kbit/s as MP2 at 192 kbit/s.

A working group consisting of Leon van de Kerkhof (The Netherlands), Gerhard Stoll (Germany), Leonardo Chiariglione (Italy), Yves-François Dehery (France), Karlheinz Brandenburg (Germany) and James D. Johnston (USA) took ideas from ASPEC, integrated the filter bank from Layer 2, added some of their own ideas and created <hl> MP3 <hl> , which was designed to achieve the same quality at 128 kbit/s as MP2 at 192 kbit/s.

question: What did the working group eventually create?, context: A working group consisting of Leon van de Kerkhof (The Netherlands), Gerhard Stoll (Germany), Leonardo Chiariglione (Italy), Yves-François Dehery (France), Karlheinz Brandenburg (Germany) and James D. Johnston (USA) took ideas from ASPEC, integrated the filter bank from Layer 2, added some of their own ideas and created MP3, which was designed to achieve the same quality at 128 kbit/s as MP2 at 192 kbit/s.


Question: What quality were they hoping to match at 128 kbit/s?

Answer: MP2 at 192 kbit/s

<hl> A working group consisting of Leon van de Kerkhof (The Netherlands), Gerhard Stoll (Germany), Leonardo Chiariglione (Italy), Yves-François Dehery (France), Karlheinz Brandenburg (Germany) and James D. Johnston (USA) took ideas from ASPEC, integrated the filter bank from Layer 2, added some of their own ideas and created MP3, which was designed to achieve the same quality at 128 kbit/s as MP2 at 192 kbit/s . <hl>

A working group consisting of Leon van de Kerkhof (The Netherlands), Gerhard Stoll (Germany), Leonardo Chiariglione (Italy), Yves-François Dehery (France), Karlheinz Brandenburg (Germany) and James D. Johnston (USA) took ideas from ASPEC, integrated the filter bank from Layer 2, added some of their own ideas and created MP3, which was designed to achieve the same quality at 128 kbit/s as <hl> MP2 at 192 kbit/s <hl>.

A working group consisting of Leon van de Kerkhof (The Netherlands), Gerhard Stoll (Germany), Leonardo Chiariglione (Italy), Yves-François Dehery (France), Karlheinz Brandenburg (Germany) and James D. Johnston (USA) took ideas from ASPEC, integrated the filter bank from Layer 2, added some of their own ideas and created MP3, which was designed to achieve the same quality at 128 kbit/s as <hl> MP2 at 192 kbit/s <hl> .

question: What quality were they hoping to match at 128 kbit/s?, context: A working group consisting of Leon van de Kerkhof (The Netherlands), Gerhard Stoll (Germany), Leonardo Chiariglione (Italy), Yves-François Dehery (France), Karlheinz Brandenburg (Germany) and James D. Johnston (USA) took ideas from ASPEC, integrated the filter bank from Layer 2, added some of their own ideas and created MP3, which was designed to achieve the same quality at 128 kbit/s as MP2 at 192 kbit/s.


Question: When were the algorithms approved?

Answer: 1991

<hl> All algorithms for MPEG-1 Audio Layer I, II and III were approved in 1991 and finalized in 1992 as part of MPEG-1, the first standard suite by MPEG, which resulted in the international standard ISO/IEC 11172-3 (a.k.a. <hl> MPEG-1 Audio or MPEG-1 Part 3), published in 1993.

All algorithms for MPEG-1 Audio Layer I, II and III were approved in <hl> 1991 <hl> and finalized in 1992 as part of MPEG-1, the first standard suite by MPEG, which resulted in the international standard ISO/IEC 11172-3 (a.k.a. MPEG-1 Audio or MPEG-1 Part 3), published in 1993.

question: When were the algorithms approved?, context: All algorithms for MPEG-1 Audio Layer I, II and III were approved in 1991 and finalized in 1992 as part of MPEG-1, the first standard suite by MPEG, which resulted in the international standard ISO/IEC 11172-3 (a.k.a. MPEG-1 Audio or MPEG-1 Part 3), published in 1993.


Question: When were the approved algorithms finalized?

Answer: 1992

<hl> All algorithms for MPEG-1 Audio Layer I, II and III were approved in 1991 and finalized in 1992 as part of MPEG-1, the first standard suite by MPEG, which resulted in the international standard ISO/IEC 11172-3 (a.k.a. <hl> MPEG-1 Audio or MPEG-1 Part 3), published in 1993.

All algorithms for MPEG-1 Audio Layer I, II and III were approved in 1991 and finalized in <hl> 1992 <hl> as part of MPEG-1, the first standard suite by MPEG, which resulted in the international standard ISO/IEC 11172-3 (a.k.a. MPEG-1 Audio or MPEG-1 Part 3), published in 1993.

question: When were the approved algorithms finalized?, context: All algorithms for MPEG-1 Audio Layer I, II and III were approved in 1991 and finalized in 1992 as part of MPEG-1, the first standard suite by MPEG, which resulted in the international standard ISO/IEC 11172-3 (a.k.a. MPEG-1 Audio or MPEG-1 Part 3), published in 1993.


Question: What was the first started suite by MPEG?

Answer: MPEG-1

<hl> All algorithms for MPEG-1 Audio Layer I, II and III were approved in 1991 and finalized in 1992 as part of MPEG-1, the first standard suite by MPEG, which resulted in the international standard ISO/IEC 11172-3 (a.k.a. <hl> MPEG-1 Audio or MPEG-1 Part 3), published in 1993.

All algorithms for <hl> MPEG-1 <hl> Audio Layer I, II and III were approved in 1991 and finalized in 1992 as part of MPEG-1, the first standard suite by MPEG, which resulted in the international standard ISO/IEC 11172-3 (a.k.a. MPEG-1 Audio or MPEG-1 Part 3), published in 1993.

question: What was the first started suite by MPEG?, context: All algorithms for MPEG-1 Audio Layer I, II and III were approved in 1991 and finalized in 1992 as part of MPEG-1, the first standard suite by MPEG, which resulted in the international standard ISO/IEC 11172-3 (a.k.a. MPEG-1 Audio or MPEG-1 Part 3), published in 1993.


Question: What was the official name of the internation standard?

Answer: ISO/IEC 11172-3

<hl> All algorithms for MPEG-1 Audio Layer I, II and III were approved in 1991 and finalized in 1992 as part of MPEG-1, the first standard suite by MPEG, which resulted in the international standard ISO/IEC 11172-3 (a.k.a. <hl> MPEG-1 Audio or MPEG-1 Part 3), published in 1993.

All algorithms for MPEG-1 Audio Layer I, II and III were approved in 1991 and finalized in 1992 as part of MPEG-1, the first standard suite by MPEG, which resulted in the international standard <hl> ISO/IEC 11172-3 <hl> (a.k.a. MPEG-1 Audio or MPEG-1 Part 3), published in 1993.

question: What was the official name of the internation standard?, context: All algorithms for MPEG-1 Audio Layer I, II and III were approved in 1991 and finalized in 1992 as part of MPEG-1, the first standard suite by MPEG, which resulted in the international standard ISO/IEC 11172-3 (a.k.a. MPEG-1 Audio or MPEG-1 Part 3), published in 1993.


Question: When was the international standard published?

Answer: 1993

All algorithms for MPEG-1 Audio Layer I, II and III were approved in 1991 and finalized in 1992 as part of MPEG-1, the first standard suite by MPEG, which resulted in the international standard ISO/IEC 11172-3 (a.k.a. <hl> MPEG-1 Audio or MPEG-1 Part 3), published in 1993 . <hl>

All algorithms for MPEG-1 Audio Layer I, II and III were approved in 1991 and finalized in 1992 as part of MPEG-1, the first standard suite by MPEG, which resulted in the international standard ISO/IEC 11172-3 (a.k.a. MPEG-1 Audio or MPEG-1 Part 3), published in <hl> 1993 <hl>.

MPEG-1 Audio or MPEG-1 Part 3), published in <hl> 1993 <hl> .

question: When was the international standard published?, context: All algorithms for MPEG-1 Audio Layer I, II and III were approved in 1991 and finalized in 1992 as part of MPEG-1, the first standard suite by MPEG, which resulted in the international standard ISO/IEC 11172-3 (a.k.a. MPEG-1 Audio or MPEG-1 Part 3), published in 1993.


Question: Further work on MPEG was finalized as a part of which MPEG standard?

Answer: second suite

<hl> Further work on MPEG audio was finalized in 1994 as part of the second suite of MPEG standards, MPEG-2, more formally known as international standard ISO/IEC 13818-3 (a.k.a. <hl> MPEG-2 Part 3 or backwards compatible MPEG-2 Audio or MPEG-2 Audio BC), originally published in 1995. MPEG-2 Part 3 (ISO/IEC 13818-3) defined additional bit rates and sample rates for MPEG-1 Audio Layer I, II and III. The new sampling rates are exactly half that of those originally defined in MPEG-1 Audio. This reduction in sampling rate serves to cut the available frequency fidelity in half while likewise cutting the bitrate by 50%. MPEG-2 Part 3 also enhanced MPEG-1's audio by allowing the coding of audio programs with more than two channels, up to 5.1 multichannel.

Further work on MPEG audio was finalized in 1994 as part of the <hl> second suite <hl> of MPEG standards, MPEG-2, more formally known as international standard ISO/IEC 13818-3 (a.k.a. MPEG-2 Part 3 or backwards compatible MPEG-2 Audio or MPEG-2 Audio BC), originally published in 1995. MPEG-2 Part 3 (ISO/IEC 13818-3) defined additional bit rates and sample rates for MPEG-1 Audio Layer I, II and III. The new sampling rates are exactly half that of those originally defined in MPEG-1 Audio. This reduction in sampling rate serves to cut the available frequency fidelity in half while likewise cutting the bitrate by 50%. MPEG-2 Part 3 also enhanced MPEG-1's audio by allowing the coding of audio programs with more than two channels, up to 5.1 multichannel.

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


Question: What was the official name of ISO/IEC 13818-3 better known as?

Answer: MPEG-2

<hl> Further work on MPEG audio was finalized in 1994 as part of the second suite of MPEG standards, MPEG-2 , more formally known as international standard ISO/IEC 13818-3 (a.k.a. <hl> MPEG-2 Part 3 or backwards compatible MPEG-2 Audio or MPEG-2 Audio BC), originally published in 1995. MPEG-2 Part 3 (ISO/IEC 13818-3) defined additional bit rates and sample rates for MPEG-1 Audio Layer I, II and III. The new sampling rates are exactly half that of those originally defined in MPEG-1 Audio. This reduction in sampling rate serves to cut the available frequency fidelity in half while likewise cutting the bitrate by 50%. MPEG-2 Part 3 also enhanced MPEG-1's audio by allowing the coding of audio programs with more than two channels, up to 5.1 multichannel.

Further work on MPEG audio was finalized in 1994 as part of the second suite of MPEG standards, <hl> MPEG-2 <hl>, more formally known as international standard ISO/IEC 13818-3 (a.k.a. MPEG-2 Part 3 or backwards compatible MPEG-2 Audio or MPEG-2 Audio BC), originally published in 1995. MPEG-2 Part 3 (ISO/IEC 13818-3) defined additional bit rates and sample rates for MPEG-1 Audio Layer I, II and III. The new sampling rates are exactly half that of those originally defined in MPEG-1 Audio. This reduction in sampling rate serves to cut the available frequency fidelity in half while likewise cutting the bitrate by 50%. MPEG-2 Part 3 also enhanced MPEG-1's audio by allowing the coding of audio programs with more than two channels, up to 5.1 multichannel.

Further work on MPEG audio was finalized in 1994 as part of the second suite of MPEG standards, <hl> MPEG-2 <hl> , more formally known as international standard ISO/IEC 13818-3 (a.k.a.

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


Question: At what rate were the new sampling rates defined in comparison to MPEG-1 Audio?

Answer: half

Further work on MPEG audio was finalized in 1994 as part of the second suite of MPEG standards, MPEG-2, more formally known as international standard ISO/IEC 13818-3 (a.k.a. MPEG-2 Part 3 or backwards compatible MPEG-2 Audio or MPEG-2 Audio BC), originally published in 1995. MPEG-2 Part 3 (ISO/IEC 13818-3) defined additional bit rates and sample rates for MPEG-1 Audio Layer I, II and III. <hl> The new sampling rates are exactly half that of those originally defined in MPEG-1 Audio. <hl> This reduction in sampling rate serves to cut the available frequency fidelity in half while likewise cutting the bitrate by 50%. MPEG-2 Part 3 also enhanced MPEG-1's audio by allowing the coding of audio programs with more than two channels, up to 5.1 multichannel.

Further work on MPEG audio was finalized in 1994 as part of the second suite of MPEG standards, MPEG-2, more formally known as international standard ISO/IEC 13818-3 (a.k.a. MPEG-2 Part 3 or backwards compatible MPEG-2 Audio or MPEG-2 Audio BC), originally published in 1995. MPEG-2 Part 3 (ISO/IEC 13818-3) defined additional bit rates and sample rates for MPEG-1 Audio Layer I, II and III. The new sampling rates are exactly <hl> half <hl> that of those originally defined in MPEG-1 Audio. This reduction in sampling rate serves to cut the available frequency fidelity in half while likewise cutting the bitrate by 50%. MPEG-2 Part 3 also enhanced MPEG-1's audio by allowing the coding of audio programs with more than two channels, up to 5.1 multichannel.

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


Question: Apart from cutting the frequency fidelity in half, what else was cut in half?

Answer: bitrate

Further work on MPEG audio was finalized in 1994 as part of the second suite of MPEG standards, MPEG-2, more formally known as international standard ISO/IEC 13818-3 (a.k.a. MPEG-2 Part 3 or backwards compatible MPEG-2 Audio or MPEG-2 Audio BC), originally published in 1995. MPEG-2 Part 3 (ISO/IEC 13818-3) defined additional bit rates and sample rates for MPEG-1 Audio Layer I, II and III. The new sampling rates are exactly half that of those originally defined in MPEG-1 Audio. <hl> This reduction in sampling rate serves to cut the available frequency fidelity in half while likewise cutting the bitrate by 50%. <hl> MPEG-2 Part 3 also enhanced MPEG-1's audio by allowing the coding of audio programs with more than two channels, up to 5.1 multichannel.

Further work on MPEG audio was finalized in 1994 as part of the second suite of MPEG standards, MPEG-2, more formally known as international standard ISO/IEC 13818-3 (a.k.a. MPEG-2 Part 3 or backwards compatible MPEG-2 Audio or MPEG-2 Audio BC), originally published in 1995. MPEG-2 Part 3 (ISO/IEC 13818-3) defined additional bit rates and sample rates for MPEG-1 Audio Layer I, II and III. The new sampling rates are exactly half that of those originally defined in MPEG-1 Audio. This reduction in sampling rate serves to cut the available frequency fidelity in half while likewise cutting the <hl> bitrate <hl> by 50%. MPEG-2 Part 3 also enhanced MPEG-1's audio by allowing the coding of audio programs with more than two channels, up to 5.1 multichannel.

question: Apart from cutting the frequency fidelity in half, what else was cut in half?, context: Further work on MPEG audio was finalized in 1994 as part of the second suite of MPEG standards, MPEG-2, more formally known as international standard ISO/IEC 13818-3 (a.k.a. MPEG-2 Part 3 or backwards compatible MPEG-2 Audio or MPEG-2 Audio BC), originally published in 1995. MPEG-2 Part 3 (ISO/IEC 13818-3) defined additional bit rates and sample rates for MPEG-1 Audio Layer I, II and III. The new sampling rates are exactly half that of those originally defined in MPEG-1 Audio. This reduction in sampling rate serves to cut the available frequency fidelity in half while likewise cutting the bitrate by 50%. MPEG-2 Part 3 also enhanced MPEG-1's audio by allowing the coding of audio programs with more than two channels, up to 5.1 multichannel.


Question: Allowing the coding of audio programs with more than two channels was a major aspect of what?

Answer: MPEG-2 Part 3

Further work on MPEG audio was finalized in 1994 as part of the second suite of MPEG standards, MPEG-2, more formally known as international standard ISO/IEC 13818-3 (a.k.a. <hl> MPEG-2 Part 3 or backwards compatible MPEG-2 Audio or MPEG-2 Audio BC), originally published in 1995. <hl> MPEG-2 Part 3 (ISO/IEC 13818-3) defined additional bit rates and sample rates for MPEG-1 Audio Layer I, II and III. The new sampling rates are exactly half that of those originally defined in MPEG-1 Audio. This reduction in sampling rate serves to cut the available frequency fidelity in half while likewise cutting the bitrate by 50%. MPEG-2 Part 3 also enhanced MPEG-1's audio by allowing the coding of audio programs with more than two channels, up to 5.1 multichannel.

Further work on MPEG audio was finalized in 1994 as part of the second suite of MPEG standards, MPEG-2, more formally known as international standard ISO/IEC 13818-3 (a.k.a. <hl> MPEG-2 Part 3 <hl> or backwards compatible MPEG-2 Audio or MPEG-2 Audio BC), originally published in 1995. MPEG-2 Part 3 (ISO/IEC 13818-3) defined additional bit rates and sample rates for MPEG-1 Audio Layer I, II and III. The new sampling rates are exactly half that of those originally defined in MPEG-1 Audio. This reduction in sampling rate serves to cut the available frequency fidelity in half while likewise cutting the bitrate by 50%. MPEG-2 Part 3 also enhanced MPEG-1's audio by allowing the coding of audio programs with more than two channels, up to 5.1 multichannel.

question: Allowing the coding of audio programs with more than two channels was a major aspect of what?, context: Further work on MPEG audio was finalized in 1994 as part of the second suite of MPEG standards, MPEG-2, more formally known as international standard ISO/IEC 13818-3 (a.k.a. MPEG-2 Part 3 or backwards compatible MPEG-2 Audio or MPEG-2 Audio BC), originally published in 1995. MPEG-2 Part 3 (ISO/IEC 13818-3) defined additional bit rates and sample rates for MPEG-1 Audio Layer I, II and III. The new sampling rates are exactly half that of those originally defined in MPEG-1 Audio. This reduction in sampling rate serves to cut the available frequency fidelity in half while likewise cutting the bitrate by 50%. MPEG-2 Part 3 also enhanced MPEG-1's audio by allowing the coding of audio programs with more than two channels, up to 5.1 multichannel.


Question: As MPEG-3 had a different meaning, what was the name given to the extension of MPEG-2?

Answer: MPEG-2.5 audio

<hl> An additional extension to MPEG-2 is named MPEG-2.5 audio , as MPEG-3 already had a different meaning. <hl> This extension was developed at Fraunhofer IIS, the registered MP3 patent holders. Like MPEG-2, MPEG-2.5 adds new sampling rates exactly half of that previously possible with MPEG-2. It thus widens the scope of MP3 to include human speech and other applications requiring only 25% of the frequency reproduction possible with MPEG-1. While not an ISO recognized standard, MPEG-2.5 is widely supported by both inexpensive and brand name digital audio players as well as computer software based MP3 encoders and decoders. A sample rate comparison between MPEG-1, 2 and 2.5 is given further down. MPEG-2.5 was not developed by MPEG and was never approved as an international standard. MPEG-2.5 is thus an unofficial or proprietary extension to the MP3 format.

An additional extension to MPEG-2 is named <hl> MPEG-2.5 audio <hl>, as MPEG-3 already had a different meaning. This extension was developed at Fraunhofer IIS, the registered MP3 patent holders. Like MPEG-2, MPEG-2.5 adds new sampling rates exactly half of that previously possible with MPEG-2. It thus widens the scope of MP3 to include human speech and other applications requiring only 25% of the frequency reproduction possible with MPEG-1. While not an ISO recognized standard, MPEG-2.5 is widely supported by both inexpensive and brand name digital audio players as well as computer software based MP3 encoders and decoders. A sample rate comparison between MPEG-1, 2 and 2.5 is given further down. MPEG-2.5 was not developed by MPEG and was never approved as an international standard. MPEG-2.5 is thus an unofficial or proprietary extension to the MP3 format.

An additional extension to MPEG-2 is named <hl> MPEG-2.5 audio <hl> , as MPEG-3 already had a different meaning.

question: As MPEG-3 had a different meaning, what was the name given to the extension of MPEG-2?, context: An additional extension to MPEG-2 is named MPEG-2.5 audio, as MPEG-3 already had a different meaning. This extension was developed at Fraunhofer IIS, the registered MP3 patent holders. Like MPEG-2, MPEG-2.5 adds new sampling rates exactly half of that previously possible with MPEG-2. It thus widens the scope of MP3 to include human speech and other applications requiring only 25% of the frequency reproduction possible with MPEG-1. While not an ISO recognized standard, MPEG-2.5 is widely supported by both inexpensive and brand name digital audio players as well as computer software based MP3 encoders and decoders. A sample rate comparison between MPEG-1, 2 and 2.5 is given further down. MPEG-2.5 was not developed by MPEG and was never approved as an international standard. MPEG-2.5 is thus an unofficial or proprietary extension to the MP3 format.


Question: Where was this extension developed?

Answer: Fraunhofer IIS

An additional extension to MPEG-2 is named MPEG-2.5 audio, as MPEG-3 already had a different meaning. <hl> This extension was developed at Fraunhofer IIS , the registered MP3 patent holders. <hl> Like MPEG-2, MPEG-2.5 adds new sampling rates exactly half of that previously possible with MPEG-2. It thus widens the scope of MP3 to include human speech and other applications requiring only 25% of the frequency reproduction possible with MPEG-1. While not an ISO recognized standard, MPEG-2.5 is widely supported by both inexpensive and brand name digital audio players as well as computer software based MP3 encoders and decoders. A sample rate comparison between MPEG-1, 2 and 2.5 is given further down. MPEG-2.5 was not developed by MPEG and was never approved as an international standard. MPEG-2.5 is thus an unofficial or proprietary extension to the MP3 format.

An additional extension to MPEG-2 is named MPEG-2.5 audio, as MPEG-3 already had a different meaning. This extension was developed at <hl> Fraunhofer IIS <hl>, the registered MP3 patent holders. Like MPEG-2, MPEG-2.5 adds new sampling rates exactly half of that previously possible with MPEG-2. It thus widens the scope of MP3 to include human speech and other applications requiring only 25% of the frequency reproduction possible with MPEG-1. While not an ISO recognized standard, MPEG-2.5 is widely supported by both inexpensive and brand name digital audio players as well as computer software based MP3 encoders and decoders. A sample rate comparison between MPEG-1, 2 and 2.5 is given further down. MPEG-2.5 was not developed by MPEG and was never approved as an international standard. MPEG-2.5 is thus an unofficial or proprietary extension to the MP3 format.

This extension was developed at <hl> Fraunhofer IIS <hl> , the registered MP3 patent holders.

question: Where was this extension developed?, context: An additional extension to MPEG-2 is named MPEG-2.5 audio, as MPEG-3 already had a different meaning. This extension was developed at Fraunhofer IIS, the registered MP3 patent holders. Like MPEG-2, MPEG-2.5 adds new sampling rates exactly half of that previously possible with MPEG-2. It thus widens the scope of MP3 to include human speech and other applications requiring only 25% of the frequency reproduction possible with MPEG-1. While not an ISO recognized standard, MPEG-2.5 is widely supported by both inexpensive and brand name digital audio players as well as computer software based MP3 encoders and decoders. A sample rate comparison between MPEG-1, 2 and 2.5 is given further down. MPEG-2.5 was not developed by MPEG and was never approved as an international standard. MPEG-2.5 is thus an unofficial or proprietary extension to the MP3 format.


Question: The new sampling rates widened the scope of MP3 to be able to include what?

Answer: human speech

An additional extension to MPEG-2 is named MPEG-2.5 audio, as MPEG-3 already had a different meaning. This extension was developed at Fraunhofer IIS, the registered MP3 patent holders. Like MPEG-2, MPEG-2.5 adds new sampling rates exactly half of that previously possible with MPEG-2. <hl> It thus widens the scope of MP3 to include human speech and other applications requiring only 25% of the frequency reproduction possible with MPEG-1. <hl> While not an ISO recognized standard, MPEG-2.5 is widely supported by both inexpensive and brand name digital audio players as well as computer software based MP3 encoders and decoders. A sample rate comparison between MPEG-1, 2 and 2.5 is given further down. MPEG-2.5 was not developed by MPEG and was never approved as an international standard. MPEG-2.5 is thus an unofficial or proprietary extension to the MP3 format.

An additional extension to MPEG-2 is named MPEG-2.5 audio, as MPEG-3 already had a different meaning. This extension was developed at Fraunhofer IIS, the registered MP3 patent holders. Like MPEG-2, MPEG-2.5 adds new sampling rates exactly half of that previously possible with MPEG-2. It thus widens the scope of MP3 to include <hl> human speech <hl> and other applications requiring only 25% of the frequency reproduction possible with MPEG-1. While not an ISO recognized standard, MPEG-2.5 is widely supported by both inexpensive and brand name digital audio players as well as computer software based MP3 encoders and decoders. A sample rate comparison between MPEG-1, 2 and 2.5 is given further down. MPEG-2.5 was not developed by MPEG and was never approved as an international standard. MPEG-2.5 is thus an unofficial or proprietary extension to the MP3 format.

question: The new sampling rates widened the scope of MP3 to be able to include what?, context: An additional extension to MPEG-2 is named MPEG-2.5 audio, as MPEG-3 already had a different meaning. This extension was developed at Fraunhofer IIS, the registered MP3 patent holders. Like MPEG-2, MPEG-2.5 adds new sampling rates exactly half of that previously possible with MPEG-2. It thus widens the scope of MP3 to include human speech and other applications requiring only 25% of the frequency reproduction possible with MPEG-1. While not an ISO recognized standard, MPEG-2.5 is widely supported by both inexpensive and brand name digital audio players as well as computer software based MP3 encoders and decoders. A sample rate comparison between MPEG-1, 2 and 2.5 is given further down. MPEG-2.5 was not developed by MPEG and was never approved as an international standard. MPEG-2.5 is thus an unofficial or proprietary extension to the MP3 format.


Question: What is not an ISO recognized standard?

Answer: MPEG-2.5

<hl> An additional extension to MPEG-2 is named MPEG-2.5 audio, as MPEG-3 already had a different meaning. <hl> This extension was developed at Fraunhofer IIS, the registered MP3 patent holders. Like MPEG-2, MPEG-2.5 adds new sampling rates exactly half of that previously possible with MPEG-2. It thus widens the scope of MP3 to include human speech and other applications requiring only 25% of the frequency reproduction possible with MPEG-1. While not an ISO recognized standard, MPEG-2.5 is widely supported by both inexpensive and brand name digital audio players as well as computer software based MP3 encoders and decoders. A sample rate comparison between MPEG-1, 2 and 2.5 is given further down. MPEG-2.5 was not developed by MPEG and was never approved as an international standard. MPEG-2.5 is thus an unofficial or proprietary extension to the MP3 format.

An additional extension to MPEG-2 is named <hl> MPEG-2.5 <hl> audio, as MPEG-3 already had a different meaning. This extension was developed at Fraunhofer IIS, the registered MP3 patent holders. Like MPEG-2, MPEG-2.5 adds new sampling rates exactly half of that previously possible with MPEG-2. It thus widens the scope of MP3 to include human speech and other applications requiring only 25% of the frequency reproduction possible with MPEG-1. While not an ISO recognized standard, MPEG-2.5 is widely supported by both inexpensive and brand name digital audio players as well as computer software based MP3 encoders and decoders. A sample rate comparison between MPEG-1, 2 and 2.5 is given further down. MPEG-2.5 was not developed by MPEG and was never approved as an international standard. MPEG-2.5 is thus an unofficial or proprietary extension to the MP3 format.

question: What is not an ISO recognized standard?, context: An additional extension to MPEG-2 is named MPEG-2.5 audio, as MPEG-3 already had a different meaning. This extension was developed at Fraunhofer IIS, the registered MP3 patent holders. Like MPEG-2, MPEG-2.5 adds new sampling rates exactly half of that previously possible with MPEG-2. It thus widens the scope of MP3 to include human speech and other applications requiring only 25% of the frequency reproduction possible with MPEG-1. While not an ISO recognized standard, MPEG-2.5 is widely supported by both inexpensive and brand name digital audio players as well as computer software based MP3 encoders and decoders. A sample rate comparison between MPEG-1, 2 and 2.5 is given further down. MPEG-2.5 was not developed by MPEG and was never approved as an international standard. MPEG-2.5 is thus an unofficial or proprietary extension to the MP3 format.


Question: As MPEG-2.5 is unofficial, it is considered what kind of extension to the MP3 format?

Answer: proprietary

An additional extension to MPEG-2 is named MPEG-2.5 audio, as MPEG-3 already had a different meaning. This extension was developed at Fraunhofer IIS, the registered MP3 patent holders. Like MPEG-2, MPEG-2.5 adds new sampling rates exactly half of that previously possible with MPEG-2. It thus widens the scope of MP3 to include human speech and other applications requiring only 25% of the frequency reproduction possible with MPEG-1. While not an ISO recognized standard, MPEG-2.5 is widely supported by both inexpensive and brand name digital audio players as well as computer software based MP3 encoders and decoders. A sample rate comparison between MPEG-1, 2 and 2.5 is given further down. MPEG-2.5 was not developed by MPEG and was never approved as an international standard. <hl> MPEG-2.5 is thus an unofficial or proprietary extension to the MP3 format. <hl>

An additional extension to MPEG-2 is named MPEG-2.5 audio, as MPEG-3 already had a different meaning. This extension was developed at Fraunhofer IIS, the registered MP3 patent holders. Like MPEG-2, MPEG-2.5 adds new sampling rates exactly half of that previously possible with MPEG-2. It thus widens the scope of MP3 to include human speech and other applications requiring only 25% of the frequency reproduction possible with MPEG-1. While not an ISO recognized standard, MPEG-2.5 is widely supported by both inexpensive and brand name digital audio players as well as computer software based MP3 encoders and decoders. A sample rate comparison between MPEG-1, 2 and 2.5 is given further down. MPEG-2.5 was not developed by MPEG and was never approved as an international standard. MPEG-2.5 is thus an unofficial or <hl> proprietary <hl> extension to the MP3 format.

question: As MPEG-2.5 is unofficial, it is considered what kind of extension to the MP3 format?, context: An additional extension to MPEG-2 is named MPEG-2.5 audio, as MPEG-3 already had a different meaning. This extension was developed at Fraunhofer IIS, the registered MP3 patent holders. Like MPEG-2, MPEG-2.5 adds new sampling rates exactly half of that previously possible with MPEG-2. It thus widens the scope of MP3 to include human speech and other applications requiring only 25% of the frequency reproduction possible with MPEG-1. While not an ISO recognized standard, MPEG-2.5 is widely supported by both inexpensive and brand name digital audio players as well as computer software based MP3 encoders and decoders. A sample rate comparison between MPEG-1, 2 and 2.5 is given further down. MPEG-2.5 was not developed by MPEG and was never approved as an international standard. MPEG-2.5 is thus an unofficial or proprietary extension to the MP3 format.


Question: What defines the compression efficiency of encoders?

Answer: bit rate

<hl> Compression efficiency of encoders is typically defined by the bit rate , because compression ratio depends on the bit depth and sampling rate of the input signal. <hl> Nevertheless, compression ratios are often published. They may use the Compact Disc (CD) parameters as references (44.1 kHz, 2 channels at 16 bits per channel or 2×16 bit), or sometimes the Digital Audio Tape (DAT) SP parameters (48 kHz, 2×16 bit). Compression ratios with this latter reference are higher, which demonstrates the problem with use of the term compression ratio for lossy encoders.

Compression efficiency of encoders is typically defined by the <hl> bit rate <hl>, because compression ratio depends on the bit depth and sampling rate of the input signal. Nevertheless, compression ratios are often published. They may use the Compact Disc (CD) parameters as references (44.1 kHz, 2 channels at 16 bits per channel or 2×16 bit), or sometimes the Digital Audio Tape (DAT) SP parameters (48 kHz, 2×16 bit). Compression ratios with this latter reference are higher, which demonstrates the problem with use of the term compression ratio for lossy encoders.

Compression efficiency of encoders is typically defined by the <hl> bit rate <hl> , because compression ratio depends on the bit depth and sampling rate of the input signal.

question: What defines the compression efficiency of encoders?, context: Compression efficiency of encoders is typically defined by the bit rate, because compression ratio depends on the bit depth and sampling rate of the input signal. Nevertheless, compression ratios are often published. They may use the Compact Disc (CD) parameters as references (44.1 kHz, 2 channels at 16 bits per channel or 2×16 bit), or sometimes the Digital Audio Tape (DAT) SP parameters (48 kHz, 2×16 bit). Compression ratios with this latter reference are higher, which demonstrates the problem with use of the term compression ratio for lossy encoders.


Question: Compression ratio depends on the sample rate and bit depth of which signal?

Answer: input

<hl> Compression efficiency of encoders is typically defined by the bit rate, because compression ratio depends on the bit depth and sampling rate of the input signal. <hl> Nevertheless, compression ratios are often published. They may use the Compact Disc (CD) parameters as references (44.1 kHz, 2 channels at 16 bits per channel or 2×16 bit), or sometimes the Digital Audio Tape (DAT) SP parameters (48 kHz, 2×16 bit). Compression ratios with this latter reference are higher, which demonstrates the problem with use of the term compression ratio for lossy encoders.

Compression efficiency of encoders is typically defined by the bit rate, because compression ratio depends on the bit depth and sampling rate of the <hl> input <hl> signal. Nevertheless, compression ratios are often published. They may use the Compact Disc (CD) parameters as references (44.1 kHz, 2 channels at 16 bits per channel or 2×16 bit), or sometimes the Digital Audio Tape (DAT) SP parameters (48 kHz, 2×16 bit). Compression ratios with this latter reference are higher, which demonstrates the problem with use of the term compression ratio for lossy encoders.

question: Compression ratio depends on the sample rate and bit depth of which signal?, context: Compression efficiency of encoders is typically defined by the bit rate, because compression ratio depends on the bit depth and sampling rate of the input signal. Nevertheless, compression ratios are often published. They may use the Compact Disc (CD) parameters as references (44.1 kHz, 2 channels at 16 bits per channel or 2×16 bit), or sometimes the Digital Audio Tape (DAT) SP parameters (48 kHz, 2×16 bit). Compression ratios with this latter reference are higher, which demonstrates the problem with use of the term compression ratio for lossy encoders.


Question: What does CD stand for?

Answer: Compact Disc

Compression efficiency of encoders is typically defined by the bit rate, because compression ratio depends on the bit depth and sampling rate of the input signal. Nevertheless, compression ratios are often published. <hl> They may use the Compact Disc (CD) parameters as references (44.1 kHz, 2 channels at 16 bits per channel or 2×16 bit), or sometimes the Digital Audio Tape (DAT) SP parameters (48 kHz, 2×16 bit). <hl> Compression ratios with this latter reference are higher, which demonstrates the problem with use of the term compression ratio for lossy encoders.

Compression efficiency of encoders is typically defined by the bit rate, because compression ratio depends on the bit depth and sampling rate of the input signal. Nevertheless, compression ratios are often published. They may use the <hl> Compact Disc <hl> (CD) parameters as references (44.1 kHz, 2 channels at 16 bits per channel or 2×16 bit), or sometimes the Digital Audio Tape (DAT) SP parameters (48 kHz, 2×16 bit). Compression ratios with this latter reference are higher, which demonstrates the problem with use of the term compression ratio for lossy encoders.

question: What does CD stand for?, context: Compression efficiency of encoders is typically defined by the bit rate, because compression ratio depends on the bit depth and sampling rate of the input signal. Nevertheless, compression ratios are often published. They may use the Compact Disc (CD) parameters as references (44.1 kHz, 2 channels at 16 bits per channel or 2×16 bit), or sometimes the Digital Audio Tape (DAT) SP parameters (48 kHz, 2×16 bit). Compression ratios with this latter reference are higher, which demonstrates the problem with use of the term compression ratio for lossy encoders.


Question: Other than CD parameters, what else can be used as parameter references?

Answer: Digital Audio Tape (DAT) SP

Compression efficiency of encoders is typically defined by the bit rate, because compression ratio depends on the bit depth and sampling rate of the input signal. Nevertheless, compression ratios are often published. <hl> They may use the Compact Disc (CD) parameters as references (44.1 kHz, 2 channels at 16 bits per channel or 2×16 bit), or sometimes the Digital Audio Tape (DAT) SP parameters (48 kHz, 2×16 bit). <hl> Compression ratios with this latter reference are higher, which demonstrates the problem with use of the term compression ratio for lossy encoders.

Compression efficiency of encoders is typically defined by the bit rate, because compression ratio depends on the bit depth and sampling rate of the input signal. Nevertheless, compression ratios are often published. They may use the Compact Disc (CD) parameters as references (44.1 kHz, 2 channels at 16 bits per channel or 2×16 bit), or sometimes the <hl> Digital Audio Tape (DAT) SP <hl> parameters (48 kHz, 2×16 bit). Compression ratios with this latter reference are higher, which demonstrates the problem with use of the term compression ratio for lossy encoders.

question: Other than CD parameters, what else can be used as parameter references?, context: Compression efficiency of encoders is typically defined by the bit rate, because compression ratio depends on the bit depth and sampling rate of the input signal. Nevertheless, compression ratios are often published. They may use the Compact Disc (CD) parameters as references (44.1 kHz, 2 channels at 16 bits per channel or 2×16 bit), or sometimes the Digital Audio Tape (DAT) SP parameters (48 kHz, 2×16 bit). Compression ratios with this latter reference are higher, which demonstrates the problem with use of the term compression ratio for lossy encoders.


Question: What can CD parameters be used as references for?

Answer: compression ratios

Compression efficiency of encoders is typically defined by the bit rate, because compression ratio depends on the bit depth and sampling rate of the input signal. <hl> Nevertheless, compression ratios are often published. <hl> They may use the Compact Disc (CD) parameters as references (44.1 kHz, 2 channels at 16 bits per channel or 2×16 bit), or sometimes the Digital Audio Tape (DAT) SP parameters (48 kHz, 2×16 bit). Compression ratios with this latter reference are higher, which demonstrates the problem with use of the term compression ratio for lossy encoders.

Compression efficiency of encoders is typically defined by the bit rate, because compression ratio depends on the bit depth and sampling rate of the input signal. Nevertheless, <hl> compression ratios <hl> are often published. They may use the Compact Disc (CD) parameters as references (44.1 kHz, 2 channels at 16 bits per channel or 2×16 bit), or sometimes the Digital Audio Tape (DAT) SP parameters (48 kHz, 2×16 bit). Compression ratios with this latter reference are higher, which demonstrates the problem with use of the term compression ratio for lossy encoders.

question: What can CD parameters be used as references for?, context: Compression efficiency of encoders is typically defined by the bit rate, because compression ratio depends on the bit depth and sampling rate of the input signal. Nevertheless, compression ratios are often published. They may use the Compact Disc (CD) parameters as references (44.1 kHz, 2 channels at 16 bits per channel or 2×16 bit), or sometimes the Digital Audio Tape (DAT) SP parameters (48 kHz, 2×16 bit). Compression ratios with this latter reference are higher, which demonstrates the problem with use of the term compression ratio for lossy encoders.


Question: What type of recording did Brandenburg use?

Answer: CD

<hl> Karlheinz Brandenburg used a CD recording of Suzanne Vega's song "Tom's Diner" to assess and refine the MP3 compression algorithm. <hl> This song was chosen because of its nearly monophonic nature and wide spectral content, making it easier to hear imperfections in the compression format during playbacks. Some refer to Suzanne Vega as "The mother of MP3". This particular track has an interesting property in that the two channels are almost, but not completely, the same, leading to a case where Binaural Masking Level Depression causes spatial unmasking of noise artifacts unless the encoder properly recognizes the situation and applies corrections similar to those detailed in the MPEG-2 AAC psychoacoustic model. Some more critical audio excerpts (glockenspiel, triangle, accordion, etc.) were taken from the EBU V3/SQAM reference compact disc and have been used by professional sound engineers to assess the subjective quality of the MPEG Audio formats.

Karlheinz Brandenburg used a <hl> CD <hl> recording of Suzanne Vega's song "Tom's Diner" to assess and refine the MP3 compression algorithm. This song was chosen because of its nearly monophonic nature and wide spectral content, making it easier to hear imperfections in the compression format during playbacks. Some refer to Suzanne Vega as "The mother of MP3". This particular track has an interesting property in that the two channels are almost, but not completely, the same, leading to a case where Binaural Masking Level Depression causes spatial unmasking of noise artifacts unless the encoder properly recognizes the situation and applies corrections similar to those detailed in the MPEG-2 AAC psychoacoustic model. Some more critical audio excerpts (glockenspiel, triangle, accordion, etc.) were taken from the EBU V3/SQAM reference compact disc and have been used by professional sound engineers to assess the subjective quality of the MPEG Audio formats.

question: What type of recording did Brandenburg use?, context: Karlheinz Brandenburg used a CD recording of Suzanne Vega's song "Tom's Diner" to assess and refine the MP3 compression algorithm. This song was chosen because of its nearly monophonic nature and wide spectral content, making it easier to hear imperfections in the compression format during playbacks. Some refer to Suzanne Vega as "The mother of MP3". This particular track has an interesting property in that the two channels are almost, but not completely, the same, leading to a case where Binaural Masking Level Depression causes spatial unmasking of noise artifacts unless the encoder properly recognizes the situation and applies corrections similar to those detailed in the MPEG-2 AAC psychoacoustic model. Some more critical audio excerpts (glockenspiel, triangle, accordion, etc.) were taken from the EBU V3/SQAM reference compact disc and have been used by professional sound engineers to assess the subjective quality of the MPEG Audio formats.


Question: What did Brandenburg use the recording to refine?

Answer: the MP3 compression algorithm

<hl> Karlheinz Brandenburg used a CD recording of Suzanne Vega's song "Tom's Diner" to assess and refine the MP3 compression algorithm . <hl> This song was chosen because of its nearly monophonic nature and wide spectral content, making it easier to hear imperfections in the compression format during playbacks. Some refer to Suzanne Vega as "The mother of MP3". This particular track has an interesting property in that the two channels are almost, but not completely, the same, leading to a case where Binaural Masking Level Depression causes spatial unmasking of noise artifacts unless the encoder properly recognizes the situation and applies corrections similar to those detailed in the MPEG-2 AAC psychoacoustic model. Some more critical audio excerpts (glockenspiel, triangle, accordion, etc.) were taken from the EBU V3/SQAM reference compact disc and have been used by professional sound engineers to assess the subjective quality of the MPEG Audio formats.

Karlheinz Brandenburg used a CD recording of Suzanne Vega's song "Tom's Diner" to assess and refine <hl> the MP3 compression algorithm <hl>. This song was chosen because of its nearly monophonic nature and wide spectral content, making it easier to hear imperfections in the compression format during playbacks. Some refer to Suzanne Vega as "The mother of MP3". This particular track has an interesting property in that the two channels are almost, but not completely, the same, leading to a case where Binaural Masking Level Depression causes spatial unmasking of noise artifacts unless the encoder properly recognizes the situation and applies corrections similar to those detailed in the MPEG-2 AAC psychoacoustic model. Some more critical audio excerpts (glockenspiel, triangle, accordion, etc.) were taken from the EBU V3/SQAM reference compact disc and have been used by professional sound engineers to assess the subjective quality of the MPEG Audio formats.

Karlheinz Brandenburg used a CD recording of Suzanne Vega's song "Tom's Diner" to assess and refine <hl> the MP3 compression algorithm <hl> .

question: What did Brandenburg use the recording to refine?, context: Karlheinz Brandenburg used a CD recording of Suzanne Vega's song "Tom's Diner" to assess and refine the MP3 compression algorithm. This song was chosen because of its nearly monophonic nature and wide spectral content, making it easier to hear imperfections in the compression format during playbacks. Some refer to Suzanne Vega as "The mother of MP3". This particular track has an interesting property in that the two channels are almost, but not completely, the same, leading to a case where Binaural Masking Level Depression causes spatial unmasking of noise artifacts unless the encoder properly recognizes the situation and applies corrections similar to those detailed in the MPEG-2 AAC psychoacoustic model. Some more critical audio excerpts (glockenspiel, triangle, accordion, etc.) were taken from the EBU V3/SQAM reference compact disc and have been used by professional sound engineers to assess the subjective quality of the MPEG Audio formats.


Question: What word describes the nature of the song used?

Answer: monophonic

Karlheinz Brandenburg used a CD recording of Suzanne Vega's song "Tom's Diner" to assess and refine the MP3 compression algorithm. <hl> This song was chosen because of its nearly monophonic nature and wide spectral content, making it easier to hear imperfections in the compression format during playbacks. <hl> Some refer to Suzanne Vega as "The mother of MP3". This particular track has an interesting property in that the two channels are almost, but not completely, the same, leading to a case where Binaural Masking Level Depression causes spatial unmasking of noise artifacts unless the encoder properly recognizes the situation and applies corrections similar to those detailed in the MPEG-2 AAC psychoacoustic model. Some more critical audio excerpts (glockenspiel, triangle, accordion, etc.) were taken from the EBU V3/SQAM reference compact disc and have been used by professional sound engineers to assess the subjective quality of the MPEG Audio formats.

Karlheinz Brandenburg used a CD recording of Suzanne Vega's song "Tom's Diner" to assess and refine the MP3 compression algorithm. This song was chosen because of its nearly <hl> monophonic <hl> nature and wide spectral content, making it easier to hear imperfections in the compression format during playbacks. Some refer to Suzanne Vega as "The mother of MP3". This particular track has an interesting property in that the two channels are almost, but not completely, the same, leading to a case where Binaural Masking Level Depression causes spatial unmasking of noise artifacts unless the encoder properly recognizes the situation and applies corrections similar to those detailed in the MPEG-2 AAC psychoacoustic model. Some more critical audio excerpts (glockenspiel, triangle, accordion, etc.) were taken from the EBU V3/SQAM reference compact disc and have been used by professional sound engineers to assess the subjective quality of the MPEG Audio formats.

question: What word describes the nature of the song used?, context: Karlheinz Brandenburg used a CD recording of Suzanne Vega's song "Tom's Diner" to assess and refine the MP3 compression algorithm. This song was chosen because of its nearly monophonic nature and wide spectral content, making it easier to hear imperfections in the compression format during playbacks. Some refer to Suzanne Vega as "The mother of MP3". This particular track has an interesting property in that the two channels are almost, but not completely, the same, leading to a case where Binaural Masking Level Depression causes spatial unmasking of noise artifacts unless the encoder properly recognizes the situation and applies corrections similar to those detailed in the MPEG-2 AAC psychoacoustic model. Some more critical audio excerpts (glockenspiel, triangle, accordion, etc.) were taken from the EBU V3/SQAM reference compact disc and have been used by professional sound engineers to assess the subjective quality of the MPEG Audio formats.


Question: As a result of her song being used, Suzanne Vega is sometimes referred to as what?

Answer: The mother of MP3

Karlheinz Brandenburg used a CD recording of Suzanne Vega's song "Tom's Diner" to assess and refine the MP3 compression algorithm. This song was chosen because of its nearly monophonic nature and wide spectral content, making it easier to hear imperfections in the compression format during playbacks. <hl> Some refer to Suzanne Vega as " The mother of MP3 ". <hl> This particular track has an interesting property in that the two channels are almost, but not completely, the same, leading to a case where Binaural Masking Level Depression causes spatial unmasking of noise artifacts unless the encoder properly recognizes the situation and applies corrections similar to those detailed in the MPEG-2 AAC psychoacoustic model. Some more critical audio excerpts (glockenspiel, triangle, accordion, etc.) were taken from the EBU V3/SQAM reference compact disc and have been used by professional sound engineers to assess the subjective quality of the MPEG Audio formats.

Karlheinz Brandenburg used a CD recording of Suzanne Vega's song "Tom's Diner" to assess and refine the MP3 compression algorithm. This song was chosen because of its nearly monophonic nature and wide spectral content, making it easier to hear imperfections in the compression format during playbacks. Some refer to Suzanne Vega as "<hl> The mother of MP3 <hl>". This particular track has an interesting property in that the two channels are almost, but not completely, the same, leading to a case where Binaural Masking Level Depression causes spatial unmasking of noise artifacts unless the encoder properly recognizes the situation and applies corrections similar to those detailed in the MPEG-2 AAC psychoacoustic model. Some more critical audio excerpts (glockenspiel, triangle, accordion, etc.) were taken from the EBU V3/SQAM reference compact disc and have been used by professional sound engineers to assess the subjective quality of the MPEG Audio formats.

Some refer to Suzanne Vega as " <hl> The mother of MP3 <hl> ".

question: As a result of her song being used, Suzanne Vega is sometimes referred to as what?, context: Karlheinz Brandenburg used a CD recording of Suzanne Vega's song "Tom's Diner" to assess and refine the MP3 compression algorithm. This song was chosen because of its nearly monophonic nature and wide spectral content, making it easier to hear imperfections in the compression format during playbacks. Some refer to Suzanne Vega as "The mother of MP3". This particular track has an interesting property in that the two channels are almost, but not completely, the same, leading to a case where Binaural Masking Level Depression causes spatial unmasking of noise artifacts unless the encoder properly recognizes the situation and applies corrections similar to those detailed in the MPEG-2 AAC psychoacoustic model. Some more critical audio excerpts (glockenspiel, triangle, accordion, etc.) were taken from the EBU V3/SQAM reference compact disc and have been used by professional sound engineers to assess the subjective quality of the MPEG Audio formats.


Question: Excerpts were taken from which reference compact disc to assess the subjective quality of the MPEG audio formats?

Answer: EBU V3/SQAM

Karlheinz Brandenburg used a CD recording of Suzanne Vega's song "Tom's Diner" to assess and refine the MP3 compression algorithm. This song was chosen because of its nearly monophonic nature and wide spectral content, making it easier to hear imperfections in the compression format during playbacks. Some refer to Suzanne Vega as "The mother of MP3". This particular track has an interesting property in that the two channels are almost, but not completely, the same, leading to a case where Binaural Masking Level Depression causes spatial unmasking of noise artifacts unless the encoder properly recognizes the situation and applies corrections similar to those detailed in the MPEG-2 AAC psychoacoustic model. <hl> Some more critical audio excerpts (glockenspiel, triangle, accordion, etc.) were taken from the EBU V3/SQAM reference compact disc and have been used by professional sound engineers to assess the subjective quality of the MPEG Audio formats. <hl>

Karlheinz Brandenburg used a CD recording of Suzanne Vega's song "Tom's Diner" to assess and refine the MP3 compression algorithm. This song was chosen because of its nearly monophonic nature and wide spectral content, making it easier to hear imperfections in the compression format during playbacks. Some refer to Suzanne Vega as "The mother of MP3". This particular track has an interesting property in that the two channels are almost, but not completely, the same, leading to a case where Binaural Masking Level Depression causes spatial unmasking of noise artifacts unless the encoder properly recognizes the situation and applies corrections similar to those detailed in the MPEG-2 AAC psychoacoustic model. Some more critical audio excerpts (glockenspiel, triangle, accordion, etc.) were taken from the <hl> EBU V3/SQAM <hl> reference compact disc and have been used by professional sound engineers to assess the subjective quality of the MPEG Audio formats.

question: Excerpts were taken from which reference compact disc to assess the subjective quality of the MPEG audio formats?, context: Karlheinz Brandenburg used a CD recording of Suzanne Vega's song "Tom's Diner" to assess and refine the MP3 compression algorithm. This song was chosen because of its nearly monophonic nature and wide spectral content, making it easier to hear imperfections in the compression format during playbacks. Some refer to Suzanne Vega as "The mother of MP3". This particular track has an interesting property in that the two channels are almost, but not completely, the same, leading to a case where Binaural Masking Level Depression causes spatial unmasking of noise artifacts unless the encoder properly recognizes the situation and applies corrections similar to those detailed in the MPEG-2 AAC psychoacoustic model. Some more critical audio excerpts (glockenspiel, triangle, accordion, etc.) were taken from the EBU V3/SQAM reference compact disc and have been used by professional sound engineers to assess the subjective quality of the MPEG Audio formats.


Question: Who developed the reference simulation software?

Answer: the ISO MPEG Audio committee

<hl> A reference simulation software implementation, written in the C language and later known as ISO 11172-5, was developed (in 1991–1996) by the members of the ISO MPEG Audio committee in order to produce bit compliant MPEG Audio files (Layer 1, Layer 2, Layer 3). <hl> It was approved as a committee draft of ISO/IEC technical report in March 1994 and printed as document CD 11172-5 in April 1994. It was approved as a draft technical report (DTR/DIS) in November 1994, finalized in 1996 and published as international standard ISO/IEC TR 11172-5:1998 in 1998. The reference software in C language was later published as a freely available ISO standard. Working in non-real time on a number of operating systems, it was able to demonstrate the first real time hardware decoding (DSP based) of compressed audio. Some other real time implementation of MPEG Audio encoders were available for the purpose of digital broadcasting (radio DAB, television DVB) towards consumer receivers and set top boxes.

A reference simulation software implementation, written in the C language and later known as ISO 11172-5, was developed (in 1991–1996) by the members of <hl> the ISO MPEG Audio committee <hl> in order to produce bit compliant MPEG Audio files (Layer 1, Layer 2, Layer 3). It was approved as a committee draft of ISO/IEC technical report in March 1994 and printed as document CD 11172-5 in April 1994. It was approved as a draft technical report (DTR/DIS) in November 1994, finalized in 1996 and published as international standard ISO/IEC TR 11172-5:1998 in 1998. The reference software in C language was later published as a freely available ISO standard. Working in non-real time on a number of operating systems, it was able to demonstrate the first real time hardware decoding (DSP based) of compressed audio. Some other real time implementation of MPEG Audio encoders were available for the purpose of digital broadcasting (radio DAB, television DVB) towards consumer receivers and set top boxes.

question: Who developed the reference simulation software?, context: A reference simulation software implementation, written in the C language and later known as ISO 11172-5, was developed (in 1991–1996) by the members of the ISO MPEG Audio committee in order to produce bit compliant MPEG Audio files (Layer 1, Layer 2, Layer 3). It was approved as a committee draft of ISO/IEC technical report in March 1994 and printed as document CD 11172-5 in April 1994. It was approved as a draft technical report (DTR/DIS) in November 1994, finalized in 1996 and published as international standard ISO/IEC TR 11172-5:1998 in 1998. The reference software in C language was later published as a freely available ISO standard. Working in non-real time on a number of operating systems, it was able to demonstrate the first real time hardware decoding (DSP based) of compressed audio. Some other real time implementation of MPEG Audio encoders were available for the purpose of digital broadcasting (radio DAB, television DVB) towards consumer receivers and set top boxes.


Question: Which language was the reference simulation software written in?

Answer: C

<hl> A reference simulation software implementation, written in the C language and later known as ISO 11172-5, was developed (in 1991–1996) by the members of the ISO MPEG Audio committee in order to produce bit compliant MPEG Audio files (Layer 1, Layer 2, Layer 3). <hl> It was approved as a committee draft of ISO/IEC technical report in March 1994 and printed as document CD 11172-5 in April 1994. It was approved as a draft technical report (DTR/DIS) in November 1994, finalized in 1996 and published as international standard ISO/IEC TR 11172-5:1998 in 1998. The reference software in C language was later published as a freely available ISO standard. Working in non-real time on a number of operating systems, it was able to demonstrate the first real time hardware decoding (DSP based) of compressed audio. Some other real time implementation of MPEG Audio encoders were available for the purpose of digital broadcasting (radio DAB, television DVB) towards consumer receivers and set top boxes.

A reference simulation software implementation, written in the <hl> C <hl> language and later known as ISO 11172-5, was developed (in 1991–1996) by the members of the ISO MPEG Audio committee in order to produce bit compliant MPEG Audio files (Layer 1, Layer 2, Layer 3). It was approved as a committee draft of ISO/IEC technical report in March 1994 and printed as document CD 11172-5 in April 1994. It was approved as a draft technical report (DTR/DIS) in November 1994, finalized in 1996 and published as international standard ISO/IEC TR 11172-5:1998 in 1998. The reference software in C language was later published as a freely available ISO standard. Working in non-real time on a number of operating systems, it was able to demonstrate the first real time hardware decoding (DSP based) of compressed audio. Some other real time implementation of MPEG Audio encoders were available for the purpose of digital broadcasting (radio DAB, television DVB) towards consumer receivers and set top boxes.

question: Which language was the reference simulation software written in?, context: A reference simulation software implementation, written in the C language and later known as ISO 11172-5, was developed (in 1991–1996) by the members of the ISO MPEG Audio committee in order to produce bit compliant MPEG Audio files (Layer 1, Layer 2, Layer 3). It was approved as a committee draft of ISO/IEC technical report in March 1994 and printed as document CD 11172-5 in April 1994. It was approved as a draft technical report (DTR/DIS) in November 1994, finalized in 1996 and published as international standard ISO/IEC TR 11172-5:1998 in 1998. The reference software in C language was later published as a freely available ISO standard. Working in non-real time on a number of operating systems, it was able to demonstrate the first real time hardware decoding (DSP based) of compressed audio. Some other real time implementation of MPEG Audio encoders were available for the purpose of digital broadcasting (radio DAB, television DVB) towards consumer receivers and set top boxes.


Question: When was the reference software approved?

Answer: March 1994

A reference simulation software implementation, written in the C language and later known as ISO 11172-5, was developed (in 1991–1996) by the members of the ISO MPEG Audio committee in order to produce bit compliant MPEG Audio files (Layer 1, Layer 2, Layer 3). <hl> It was approved as a committee draft of ISO/IEC technical report in March 1994 and printed as document CD 11172-5 in April 1994. <hl> It was approved as a draft technical report (DTR/DIS) in November 1994, finalized in 1996 and published as international standard ISO/IEC TR 11172-5:1998 in 1998. The reference software in C language was later published as a freely available ISO standard. Working in non-real time on a number of operating systems, it was able to demonstrate the first real time hardware decoding (DSP based) of compressed audio. Some other real time implementation of MPEG Audio encoders were available for the purpose of digital broadcasting (radio DAB, television DVB) towards consumer receivers and set top boxes.

A reference simulation software implementation, written in the C language and later known as ISO 11172-5, was developed (in 1991–1996) by the members of the ISO MPEG Audio committee in order to produce bit compliant MPEG Audio files (Layer 1, Layer 2, Layer 3). It was approved as a committee draft of ISO/IEC technical report in <hl> March 1994 <hl> and printed as document CD 11172-5 in April 1994. It was approved as a draft technical report (DTR/DIS) in November 1994, finalized in 1996 and published as international standard ISO/IEC TR 11172-5:1998 in 1998. The reference software in C language was later published as a freely available ISO standard. Working in non-real time on a number of operating systems, it was able to demonstrate the first real time hardware decoding (DSP based) of compressed audio. Some other real time implementation of MPEG Audio encoders were available for the purpose of digital broadcasting (radio DAB, television DVB) towards consumer receivers and set top boxes.

question: When was the reference software approved?, context: A reference simulation software implementation, written in the C language and later known as ISO 11172-5, was developed (in 1991–1996) by the members of the ISO MPEG Audio committee in order to produce bit compliant MPEG Audio files (Layer 1, Layer 2, Layer 3). It was approved as a committee draft of ISO/IEC technical report in March 1994 and printed as document CD 11172-5 in April 1994. It was approved as a draft technical report (DTR/DIS) in November 1994, finalized in 1996 and published as international standard ISO/IEC TR 11172-5:1998 in 1998. The reference software in C language was later published as a freely available ISO standard. Working in non-real time on a number of operating systems, it was able to demonstrate the first real time hardware decoding (DSP based) of compressed audio. Some other real time implementation of MPEG Audio encoders were available for the purpose of digital broadcasting (radio DAB, television DVB) towards consumer receivers and set top boxes.