Question: MP3 files today contain what kind of metadeta?

Answer: ID3

An MP3 file is made up of MP3 frames, which consist of a header and a data block. This sequence of frames is called an elementary stream. Due to the "byte reservoir", frames are not independent items and cannot usually be extracted on arbitrary frame boundaries. The MP3 Data blocks contain the (compressed) audio information in terms of frequencies and amplitudes. The diagram shows that the MP3 Header consists of a sync word, which is used to identify the beginning of a valid frame. This is followed by a bit indicating that this is the MPEG standard and two bits that indicate that layer 3 is used; hence MPEG-1 Audio Layer 3 or MP3. After this, the values will differ, depending on the MP3 file. ISO/IEC 11172-3 defines the range of values for each section of the header along with the specification of the header. <hl> Most MP3 files today contain ID3 metadata, which precedes or follows the MP3 frames, as noted in the diagram. <hl>

An MP3 file is made up of MP3 frames, which consist of a header and a data block. This sequence of frames is called an elementary stream. Due to the "byte reservoir", frames are not independent items and cannot usually be extracted on arbitrary frame boundaries. The MP3 Data blocks contain the (compressed) audio information in terms of frequencies and amplitudes. The diagram shows that the MP3 Header consists of a sync word, which is used to identify the beginning of a valid frame. This is followed by a bit indicating that this is the MPEG standard and two bits that indicate that layer 3 is used; hence MPEG-1 Audio Layer 3 or MP3. After this, the values will differ, depending on the MP3 file. ISO/IEC 11172-3 defines the range of values for each section of the header along with the specification of the header. Most MP3 files today contain <hl> ID3 <hl> metadata, which precedes or follows the MP3 frames, as noted in the diagram.


Question: Which standard does not include a specification for an MP3 encoder?

The MPEG-1 standard does not include a precise specification for an MP3 encoder, but does provide example psychoacoustic models, rate loop, and the like in the non-normative part of the original standard. At present, these suggested implementations are quite dated. Implementers of the standard were supposed to devise their own algorithms suitable for removing parts of the information from the audio input. As a result, there are many different MP3 encoders available, each producing files of differing quality. Comparisons are widely available, so it is easy for a prospective user of an encoder to research the best choice. An encoder that is proficient at encoding at higher bit rates (such as LAME) is not necessarily as good at lower bit rates.

Answer: MPEG-1

<hl> The MPEG-1 standard does not include a precise specification for an MP3 encoder, but does provide example psychoacoustic models, rate loop, and the like in the non-normative part of the original standard. <hl> At present, these suggested implementations are quite dated. Implementers of the standard were supposed to devise their own algorithms suitable for removing parts of the information from the audio input. As a result, there are many different MP3 encoders available, each producing files of differing quality. Comparisons are widely available, so it is easy for a prospective user of an encoder to research the best choice. An encoder that is proficient at encoding at higher bit rates (such as LAME) is not necessarily as good at lower bit rates.

The <hl> MPEG-1 <hl> standard does not include a precise specification for an MP3 encoder, but does provide example psychoacoustic models, rate loop, and the like in the non-normative part of the original standard. At present, these suggested implementations are quite dated. Implementers of the standard were supposed to devise their own algorithms suitable for removing parts of the information from the audio input. As a result, there are many different MP3 encoders available, each producing files of differing quality. Comparisons are widely available, so it is easy for a prospective user of an encoder to research the best choice. An encoder that is proficient at encoding at higher bit rates (such as LAME) is not necessarily as good at lower bit rates.


Question: Example psychoacoustic models and rate loops can be foundin which part of the original standard?

Answer: non-normative

<hl> The MPEG-1 standard does not include a precise specification for an MP3 encoder, but does provide example psychoacoustic models, rate loop, and the like in the non-normative part of the original standard. <hl> At present, these suggested implementations are quite dated. Implementers of the standard were supposed to devise their own algorithms suitable for removing parts of the information from the audio input. As a result, there are many different MP3 encoders available, each producing files of differing quality. Comparisons are widely available, so it is easy for a prospective user of an encoder to research the best choice. An encoder that is proficient at encoding at higher bit rates (such as LAME) is not necessarily as good at lower bit rates.

The MPEG-1 standard does not include a precise specification for an MP3 encoder, but does provide example psychoacoustic models, rate loop, and the like in the <hl> non-normative <hl> part of the original standard. At present, these suggested implementations are quite dated. Implementers of the standard were supposed to devise their own algorithms suitable for removing parts of the information from the audio input. As a result, there are many different MP3 encoders available, each producing files of differing quality. Comparisons are widely available, so it is easy for a prospective user of an encoder to research the best choice. An encoder that is proficient at encoding at higher bit rates (such as LAME) is not necessarily as good at lower bit rates.


Question: What were implementers of the standard supposed to devise?

Answer: their own algorithms

The MPEG-1 standard does not include a precise specification for an MP3 encoder, but does provide example psychoacoustic models, rate loop, and the like in the non-normative part of the original standard. At present, these suggested implementations are quite dated. <hl> Implementers of the standard were supposed to devise their own algorithms suitable for removing parts of the information from the audio input. <hl> As a result, there are many different MP3 encoders available, each producing files of differing quality. Comparisons are widely available, so it is easy for a prospective user of an encoder to research the best choice. An encoder that is proficient at encoding at higher bit rates (such as LAME) is not necessarily as good at lower bit rates.

The MPEG-1 standard does not include a precise specification for an MP3 encoder, but does provide example psychoacoustic models, rate loop, and the like in the non-normative part of the original standard. At present, these suggested implementations are quite dated. Implementers of the standard were supposed to devise <hl> their own algorithms <hl> suitable for removing parts of the information from the audio input. As a result, there are many different MP3 encoders available, each producing files of differing quality. Comparisons are widely available, so it is easy for a prospective user of an encoder to research the best choice. An encoder that is proficient at encoding at higher bit rates (such as LAME) is not necessarily as good at lower bit rates.


Question: It is easy for a prospective user of an encoder to research the best choice because of what factor?

Answer: Comparisons are widely available

The MPEG-1 standard does not include a precise specification for an MP3 encoder, but does provide example psychoacoustic models, rate loop, and the like in the non-normative part of the original standard. At present, these suggested implementations are quite dated. Implementers of the standard were supposed to devise their own algorithms suitable for removing parts of the information from the audio input. As a result, there are many different MP3 encoders available, each producing files of differing quality. <hl> Comparisons are widely available , so it is easy for a prospective user of an encoder to research the best choice. <hl> An encoder that is proficient at encoding at higher bit rates (such as LAME) is not necessarily as good at lower bit rates.

The MPEG-1 standard does not include a precise specification for an MP3 encoder, but does provide example psychoacoustic models, rate loop, and the like in the non-normative part of the original standard. At present, these suggested implementations are quite dated. Implementers of the standard were supposed to devise their own algorithms suitable for removing parts of the information from the audio input. As a result, there are many different MP3 encoders available, each producing files of differing quality. <hl> Comparisons are widely available <hl>, so it is easy for a prospective user of an encoder to research the best choice. An encoder that is proficient at encoding at higher bit rates (such as LAME) is not necessarily as good at lower bit rates.

<hl> Comparisons are widely available <hl> , so it is easy for a prospective user of an encoder to research the best choice.


Question: An encoder that is proficient at encoding at higher bit rates might be worse at what?

Answer: lower bit rates

The MPEG-1 standard does not include a precise specification for an MP3 encoder, but does provide example psychoacoustic models, rate loop, and the like in the non-normative part of the original standard. At present, these suggested implementations are quite dated. Implementers of the standard were supposed to devise their own algorithms suitable for removing parts of the information from the audio input. As a result, there are many different MP3 encoders available, each producing files of differing quality. Comparisons are widely available, so it is easy for a prospective user of an encoder to research the best choice. <hl> An encoder that is proficient at encoding at higher bit rates (such as LAME) is not necessarily as good at lower bit rates . <hl>

The MPEG-1 standard does not include a precise specification for an MP3 encoder, but does provide example psychoacoustic models, rate loop, and the like in the non-normative part of the original standard. At present, these suggested implementations are quite dated. Implementers of the standard were supposed to devise their own algorithms suitable for removing parts of the information from the audio input. As a result, there are many different MP3 encoders available, each producing files of differing quality. Comparisons are widely available, so it is easy for a prospective user of an encoder to research the best choice. An encoder that is proficient at encoding at higher bit rates (such as LAME) is not necessarily as good at <hl> lower bit rates <hl>.

An encoder that is proficient at encoding at higher bit rates (such as LAME) is not necessarily as good at <hl> lower bit rates <hl> .


Question: How many domain samples are taken during encoding?

During encoding, 576 time-domain samples are taken and are transformed to 576 frequency-domain samples.[clarification needed] If there is a transient, 192 samples are taken instead of 576. This is done to limit the temporal spread of quantization noise accompanying the transient. (See psychoacoustics.)

Answer: 576

<hl> During encoding, 576 time-domain samples are taken and are transformed to 576 frequency-domain samples.[clarification needed] If there is a transient, 192 samples are taken instead of 576. <hl> This is done to limit the temporal spread of quantization noise accompanying the transient. (See psychoacoustics.)

During encoding, <hl> 576 <hl> time-domain samples are taken and are transformed to 576 frequency-domain samples.[clarification needed] If there is a transient, 192 samples are taken instead of 576. This is done to limit the temporal spread of quantization noise accompanying the transient. (See psychoacoustics.)


Question: Domain samples are transformed into what?

Answer: frequency-domain samples

<hl> During encoding, 576 time-domain samples are taken and are transformed to 576 frequency-domain samples .[clarification needed] If there is a transient, 192 samples are taken instead of 576. <hl> This is done to limit the temporal spread of quantization noise accompanying the transient. (See psychoacoustics.)

During encoding, 576 time-domain samples are taken and are transformed to 576 <hl> frequency-domain samples <hl>.[clarification needed] If there is a transient, 192 samples are taken instead of 576. This is done to limit the temporal spread of quantization noise accompanying the transient. (See psychoacoustics.)

During encoding, 576 time-domain samples are taken and are transformed to 576 <hl> frequency-domain samples <hl> .[clarification needed] If there is a transient, 192 samples are taken instead of 576.


Question: When are 192 samples taken instead of 576?

Answer: there is a transient

<hl> During encoding, 576 time-domain samples are taken and are transformed to 576 frequency-domain samples.[clarification needed] If there is a transient , 192 samples are taken instead of 576. <hl> This is done to limit the temporal spread of quantization noise accompanying the transient. (See psychoacoustics.)

During encoding, 576 time-domain samples are taken and are transformed to 576 frequency-domain samples.[clarification needed] If <hl> there is a transient <hl>, 192 samples are taken instead of 576. This is done to limit the temporal spread of quantization noise accompanying the transient. (See psychoacoustics.)

During encoding, 576 time-domain samples are taken and are transformed to 576 frequency-domain samples.[clarification needed] If <hl> there is a transient <hl> , 192 samples are taken instead of 576.


Question: When 192 samples are taken instead of 576, it is done in an effort to limit what?

Answer: temporal spread

During encoding, 576 time-domain samples are taken and are transformed to 576 frequency-domain samples.[clarification needed] If there is a transient, 192 samples are taken instead of 576. <hl> This is done to limit the temporal spread of quantization noise accompanying the transient. <hl> (See psychoacoustics.)

During encoding, 576 time-domain samples are taken and are transformed to 576 frequency-domain samples.[clarification needed] If there is a transient, 192 samples are taken instead of 576. This is done to limit the <hl> temporal spread <hl> of quantization noise accompanying the transient. (See psychoacoustics.)


Question: What accompanies a transient?

Answer: quantization noise

During encoding, 576 time-domain samples are taken and are transformed to 576 frequency-domain samples.[clarification needed] If there is a transient, 192 samples are taken instead of 576. <hl> This is done to limit the temporal spread of quantization noise accompanying the transient. <hl> (See psychoacoustics.)

During encoding, 576 time-domain samples are taken and are transformed to 576 frequency-domain samples.[clarification needed] If there is a transient, 192 samples are taken instead of 576. This is done to limit the temporal spread of <hl> quantization noise <hl> accompanying the transient. (See psychoacoustics.)


Question: What kind of structure does a filter bank have?

Due to the tree structure of the filter bank, pre-echo problems are made worse, as the combined impulse response of the two filter banks does not, and cannot, provide an optimum solution in time/frequency resolution. Additionally, the combining of the two filter banks' outputs creates aliasing problems that must be handled partially by the "aliasing compensation" stage; however, that creates excess energy to be coded in the frequency domain, thereby decreasing coding efficiency.[citation needed]

Answer: tree

<hl> Due to the tree structure of the filter bank, pre-echo problems are made worse, as the combined impulse response of the two filter banks does not, and cannot, provide an optimum solution in time/frequency resolution. <hl> Additionally, the combining of the two filter banks' outputs creates aliasing problems that must be handled partially by the "aliasing compensation" stage; however, that creates excess energy to be coded in the frequency domain, thereby decreasing coding efficiency.[citation needed]

Due to the <hl> tree <hl> structure of the filter bank, pre-echo problems are made worse, as the combined impulse response of the two filter banks does not, and cannot, provide an optimum solution in time/frequency resolution. Additionally, the combining of the two filter banks' outputs creates aliasing problems that must be handled partially by the "aliasing compensation" stage; however, that creates excess energy to be coded in the frequency domain, thereby decreasing coding efficiency.[citation needed]


Question: Due to a specific kind of structure, what kind of problems are made worse?

Answer: pre-echo problems

<hl> Due to the tree structure of the filter bank, pre-echo problems are made worse, as the combined impulse response of the two filter banks does not, and cannot, provide an optimum solution in time/frequency resolution. <hl> Additionally, the combining of the two filter banks' outputs creates aliasing problems that must be handled partially by the "aliasing compensation" stage; however, that creates excess energy to be coded in the frequency domain, thereby decreasing coding efficiency.[citation needed]

Due to the tree structure of the filter bank, <hl> pre-echo problems <hl> are made worse, as the combined impulse response of the two filter banks does not, and cannot, provide an optimum solution in time/frequency resolution. Additionally, the combining of the two filter banks' outputs creates aliasing problems that must be handled partially by the "aliasing compensation" stage; however, that creates excess energy to be coded in the frequency domain, thereby decreasing coding efficiency.[citation needed]


Question: What kind of resolution are the filter banks unable to provide an optimal solution for?

Answer: time/frequency

<hl> Due to the tree structure of the filter bank, pre-echo problems are made worse, as the combined impulse response of the two filter banks does not, and cannot, provide an optimum solution in time/frequency resolution. <hl> Additionally, the combining of the two filter banks' outputs creates aliasing problems that must be handled partially by the "aliasing compensation" stage; however, that creates excess energy to be coded in the frequency domain, thereby decreasing coding efficiency.[citation needed]

Due to the tree structure of the filter bank, pre-echo problems are made worse, as the combined impulse response of the two filter banks does not, and cannot, provide an optimum solution in <hl> time/frequency <hl> resolution. Additionally, the combining of the two filter banks' outputs creates aliasing problems that must be handled partially by the "aliasing compensation" stage; however, that creates excess energy to be coded in the frequency domain, thereby decreasing coding efficiency.[citation needed]


Question: Combining two filter banks' outputs creates what kind of problem?

Answer: aliasing

Due to the tree structure of the filter bank, pre-echo problems are made worse, as the combined impulse response of the two filter banks does not, and cannot, provide an optimum solution in time/frequency resolution. <hl> Additionally, the combining of the two filter banks' outputs creates aliasing problems that must be handled partially by the "aliasing compensation" stage; however, that creates excess energy to be coded in the frequency domain, thereby decreasing coding efficiency.[citation needed] <hl>

Due to the tree structure of the filter bank, pre-echo problems are made worse, as the combined impulse response of the two filter banks does not, and cannot, provide an optimum solution in time/frequency resolution. Additionally, the combining of the two filter banks' outputs creates <hl> aliasing <hl> problems that must be handled partially by the "aliasing compensation" stage; however, that creates excess energy to be coded in the frequency domain, thereby decreasing coding efficiency.[citation needed]


Question: The need to code excess energy in the frequency domain decreases what kind of efficiency?

Answer: coding

Due to the tree structure of the filter bank, pre-echo problems are made worse, as the combined impulse response of the two filter banks does not, and cannot, provide an optimum solution in time/frequency resolution. <hl> Additionally, the combining of the two filter banks' outputs creates aliasing problems that must be handled partially by the "aliasing compensation" stage; however, that creates excess energy to be coded in the frequency domain, thereby decreasing coding efficiency.[citation needed] <hl>

Due to the tree structure of the filter bank, pre-echo problems are made worse, as the combined impulse response of the two filter banks does not, and cannot, provide an optimum solution in time/frequency resolution. Additionally, the combining of the two filter banks' outputs creates aliasing problems that must be handled partially by the "aliasing compensation" stage; however, that creates excess energy to be coded in the frequency domain, thereby decreasing <hl> coding <hl> efficiency.[citation needed]


Question: What is carefully defined in the standard?

Decoding, on the other hand, is carefully defined in the standard. Most decoders are "bitstream compliant", which means that the decompressed output that they produce from a given MP3 file will be the same, within a specified degree of rounding tolerance, as the output specified mathematically in the ISO/IEC high standard document (ISO/IEC 11172-3). Therefore, comparison of decoders is usually based on how computationally efficient they are (i.e., how much memory or CPU time they use in the decoding process).

Answer: Decoding

<hl> Decoding , on the other hand, is carefully defined in the standard. <hl> Most decoders are "bitstream compliant", which means that the decompressed output that they produce from a given MP3 file will be the same, within a specified degree of rounding tolerance, as the output specified mathematically in the ISO/IEC high standard document (ISO/IEC 11172-3). Therefore, comparison of decoders is usually based on how computationally efficient they are (i.e., how much memory or CPU time they use in the decoding process).

<hl> Decoding <hl>, on the other hand, is carefully defined in the standard. Most decoders are "bitstream compliant", which means that the decompressed output that they produce from a given MP3 file will be the same, within a specified degree of rounding tolerance, as the output specified mathematically in the ISO/IEC high standard document (ISO/IEC 11172-3). Therefore, comparison of decoders is usually based on how computationally efficient they are (i.e., how much memory or CPU time they use in the decoding process).

<hl> Decoding <hl> , on the other hand, is carefully defined in the standard.


Question: What are most decoders?

Answer: bitstream compliant

Decoding, on the other hand, is carefully defined in the standard. <hl> Most decoders are " bitstream compliant ", which means that the decompressed output that they produce from a given MP3 file will be the same, within a specified degree of rounding tolerance, as the output specified mathematically in the ISO/IEC high standard document (ISO/IEC 11172-3). <hl> Therefore, comparison of decoders is usually based on how computationally efficient they are (i.e., how much memory or CPU time they use in the decoding process).

Decoding, on the other hand, is carefully defined in the standard. Most decoders are "<hl> bitstream compliant <hl>", which means that the decompressed output that they produce from a given MP3 file will be the same, within a specified degree of rounding tolerance, as the output specified mathematically in the ISO/IEC high standard document (ISO/IEC 11172-3). Therefore, comparison of decoders is usually based on how computationally efficient they are (i.e., how much memory or CPU time they use in the decoding process).

Most decoders are " <hl> bitstream compliant <hl> ", which means that the decompressed output that they produce from a given MP3 file will be the same, within a specified degree of rounding tolerance, as the output specified mathematically in the ISO/IEC high standard document (ISO/IEC 11172-3).


Question: The ISO/IEC high standard document states that the decompressed output produced from a given MP3 file will be the same within what standards?

Answer: specified degree of rounding tolerance

Decoding, on the other hand, is carefully defined in the standard. <hl> Most decoders are "bitstream compliant", which means that the decompressed output that they produce from a given MP3 file will be the same, within a specified degree of rounding tolerance , as the output specified mathematically in the ISO/IEC high standard document (ISO/IEC 11172-3). <hl> Therefore, comparison of decoders is usually based on how computationally efficient they are (i.e., how much memory or CPU time they use in the decoding process).

Decoding, on the other hand, is carefully defined in the standard. Most decoders are "bitstream compliant", which means that the decompressed output that they produce from a given MP3 file will be the same, within a <hl> specified degree of rounding tolerance <hl>, as the output specified mathematically in the ISO/IEC high standard document (ISO/IEC 11172-3). Therefore, comparison of decoders is usually based on how computationally efficient they are (i.e., how much memory or CPU time they use in the decoding process).

Most decoders are "bitstream compliant", which means that the decompressed output that they produce from a given MP3 file will be the same, within a <hl> specified degree of rounding tolerance <hl> , as the output specified mathematically in the ISO/IEC high standard document (ISO/IEC 11172-3).


Question: Decoders are usually compared by examining which factor?

Answer: computationally efficient

Decoding, on the other hand, is carefully defined in the standard. Most decoders are "bitstream compliant", which means that the decompressed output that they produce from a given MP3 file will be the same, within a specified degree of rounding tolerance, as the output specified mathematically in the ISO/IEC high standard document (ISO/IEC 11172-3). <hl> Therefore, comparison of decoders is usually based on how computationally efficient they are (i.e., how much memory or CPU time they use in the decoding process). <hl>

Decoding, on the other hand, is carefully defined in the standard. Most decoders are "bitstream compliant", which means that the decompressed output that they produce from a given MP3 file will be the same, within a specified degree of rounding tolerance, as the output specified mathematically in the ISO/IEC high standard document (ISO/IEC 11172-3). Therefore, comparison of decoders is usually based on how <hl> computationally efficient <hl> they are (i.e., how much memory or CPU time they use in the decoding process).


Question: The efficiency of decoders is examined by seeing how much memory and what other process they use in the decoding process?

Answer: CPU time

Decoding, on the other hand, is carefully defined in the standard. Most decoders are "bitstream compliant", which means that the decompressed output that they produce from a given MP3 file will be the same, within a specified degree of rounding tolerance, as the output specified mathematically in the ISO/IEC high standard document (ISO/IEC 11172-3). <hl> Therefore, comparison of decoders is usually based on how computationally efficient they are (i.e., how much memory or CPU time they use in the decoding process). <hl>

Decoding, on the other hand, is carefully defined in the standard. Most decoders are "bitstream compliant", which means that the decompressed output that they produce from a given MP3 file will be the same, within a specified degree of rounding tolerance, as the output specified mathematically in the ISO/IEC high standard document (ISO/IEC 11172-3). Therefore, comparison of decoders is usually based on how computationally efficient they are (i.e., how much memory or <hl> CPU time <hl> they use in the decoding process).


Question: What is there no official provision for?

Encoder / decoder overall delay is not defined, which means there is no official provision for gapless playback. However, some encoders such as LAME can attach additional metadata that will allow players that can handle it to deliver seamless playback.

Answer: gapless playback

<hl> Encoder / decoder overall delay is not defined, which means there is no official provision for gapless playback . <hl> However, some encoders such as LAME can attach additional metadata that will allow players that can handle it to deliver seamless playback.

Encoder / decoder overall delay is not defined, which means there is no official provision for <hl> gapless playback <hl>. However, some encoders such as LAME can attach additional metadata that will allow players that can handle it to deliver seamless playback.

Encoder / decoder overall delay is not defined, which means there is no official provision for <hl> gapless playback <hl> .


Question: What is an example of an encoder that can attach additional metadeta?

Answer: LAME

Encoder / decoder overall delay is not defined, which means there is no official provision for gapless playback. <hl> However, some encoders such as LAME can attach additional metadata that will allow players that can handle it to deliver seamless playback. <hl>

Encoder / decoder overall delay is not defined, which means there is no official provision for gapless playback. However, some encoders such as <hl> LAME <hl> can attach additional metadata that will allow players that can handle it to deliver seamless playback.


Question: Adding additional metadata allows players to handle gapless playback and deliver what?

Answer: seamless playback

Encoder / decoder overall delay is not defined, which means there is no official provision for gapless playback. <hl> However, some encoders such as LAME can attach additional metadata that will allow players that can handle it to deliver seamless playback . <hl>

Encoder / decoder overall delay is not defined, which means there is no official provision for gapless playback. However, some encoders such as LAME can attach additional metadata that will allow players that can handle it to deliver <hl> seamless playback <hl>.

However, some encoders such as LAME can attach additional metadata that will allow players that can handle it to deliver <hl> seamless playback <hl> .


Question: For encoders and decoders, what is not defined?

Answer: overall delay

<hl> Encoder / decoder overall delay is not defined, which means there is no official provision for gapless playback. <hl> However, some encoders such as LAME can attach additional metadata that will allow players that can handle it to deliver seamless playback.

Encoder / decoder <hl> overall delay <hl> is not defined, which means there is no official provision for gapless playback. However, some encoders such as LAME can attach additional metadata that will allow players that can handle it to deliver seamless playback.


Question: A typical trade off when creating an MP3 file is between the amount of space used and what other factor?

When performing lossy audio encoding, such as creating an MP3 file, there is a trade-off between the amount of space used and the sound quality of the result. Typically, the creator is allowed to set a bit rate, which specifies how many kilobits the file may use per second of audio. The higher the bit rate, the larger the compressed file will be, and, generally, the closer it will sound to the original file.

Answer: sound quality of the result

<hl> When performing lossy audio encoding, such as creating an MP3 file, there is a trade-off between the amount of space used and the sound quality of the result . <hl> Typically, the creator is allowed to set a bit rate, which specifies how many kilobits the file may use per second of audio. The higher the bit rate, the larger the compressed file will be, and, generally, the closer it will sound to the original file.

When performing lossy audio encoding, such as creating an MP3 file, there is a trade-off between the amount of space used and the <hl> sound quality of the result <hl>. Typically, the creator is allowed to set a bit rate, which specifies how many kilobits the file may use per second of audio. The higher the bit rate, the larger the compressed file will be, and, generally, the closer it will sound to the original file.

When performing lossy audio encoding, such as creating an MP3 file, there is a trade-off between the amount of space used and the <hl> sound quality of the result <hl> .


Question: What would an example of lossy audio encoding be?

Answer: creating an MP3 file

<hl> When performing lossy audio encoding, such as creating an MP3 file , there is a trade-off between the amount of space used and the sound quality of the result. <hl> Typically, the creator is allowed to set a bit rate, which specifies how many kilobits the file may use per second of audio. The higher the bit rate, the larger the compressed file will be, and, generally, the closer it will sound to the original file.

When performing lossy audio encoding, such as <hl> creating an MP3 file <hl>, there is a trade-off between the amount of space used and the sound quality of the result. Typically, the creator is allowed to set a bit rate, which specifies how many kilobits the file may use per second of audio. The higher the bit rate, the larger the compressed file will be, and, generally, the closer it will sound to the original file.

When performing lossy audio encoding, such as <hl> creating an MP3 file <hl> , there is a trade-off between the amount of space used and the sound quality of the result.


Question: Who gets to set the bit rate for an MP3 file?

Answer: the creator

When performing lossy audio encoding, such as creating an MP3 file, there is a trade-off between the amount of space used and the sound quality of the result. <hl> Typically, the creator is allowed to set a bit rate, which specifies how many kilobits the file may use per second of audio. <hl> The higher the bit rate, the larger the compressed file will be, and, generally, the closer it will sound to the original file.

When performing lossy audio encoding, such as creating an MP3 file, there is a trade-off between the amount of space used and the sound quality of the result. Typically, <hl> the creator <hl> is allowed to set a bit rate, which specifies how many kilobits the file may use per second of audio. The higher the bit rate, the larger the compressed file will be, and, generally, the closer it will sound to the original file.


Question: What does the bit rate of the file actually mean?

Answer: how many kilobits the file may use per second of audio

When performing lossy audio encoding, such as creating an MP3 file, there is a trade-off between the amount of space used and the sound quality of the result. <hl> Typically, the creator is allowed to set a bit rate, which specifies how many kilobits the file may use per second of audio . <hl> The higher the bit rate, the larger the compressed file will be, and, generally, the closer it will sound to the original file.

When performing lossy audio encoding, such as creating an MP3 file, there is a trade-off between the amount of space used and the sound quality of the result. Typically, the creator is allowed to set a bit rate, which specifies <hl> how many kilobits the file may use per second of audio <hl>. The higher the bit rate, the larger the compressed file will be, and, generally, the closer it will sound to the original file.

Typically, the creator is allowed to set a bit rate, which specifies <hl> how many kilobits the file may use per second of audio <hl> .


Question: If the bit rate is higher, the size of the compressed file will be what?

Answer: larger

When performing lossy audio encoding, such as creating an MP3 file, there is a trade-off between the amount of space used and the sound quality of the result. Typically, the creator is allowed to set a bit rate, which specifies how many kilobits the file may use per second of audio. <hl> The higher the bit rate, the larger the compressed file will be, and, generally, the closer it will sound to the original file. <hl>

When performing lossy audio encoding, such as creating an MP3 file, there is a trade-off between the amount of space used and the sound quality of the result. Typically, the creator is allowed to set a bit rate, which specifies how many kilobits the file may use per second of audio. The higher the bit rate, the <hl> larger <hl> the compressed file will be, and, generally, the closer it will sound to the original file.


Question: If the bit rate is too low, what might be audible in the reproduction?

With too low a bit rate, compression artifacts (i.e., sounds that were not present in the original recording) may be audible in the reproduction. Some audio is hard to compress because of its randomness and sharp attacks. When this type of audio is compressed, artifacts such as ringing or pre-echo are usually heard. A sample of applause compressed with a relatively low bit rate provides a good example of compression artifacts.

Answer: compression artifacts

<hl> With too low a bit rate, compression artifacts (i.e., sounds that were not present in the original recording) may be audible in the reproduction. <hl> Some audio is hard to compress because of its randomness and sharp attacks. When this type of audio is compressed, artifacts such as ringing or pre-echo are usually heard. A sample of applause compressed with a relatively low bit rate provides a good example of compression artifacts.

With too low a bit rate, <hl> compression artifacts <hl> (i.e., sounds that were not present in the original recording) may be audible in the reproduction. Some audio is hard to compress because of its randomness and sharp attacks. When this type of audio is compressed, artifacts such as ringing or pre-echo are usually heard. A sample of applause compressed with a relatively low bit rate provides a good example of compression artifacts.


Question: Sounds that were not in the original recording are referred to as what?

Answer: compression artifacts

<hl> With too low a bit rate, compression artifacts (i.e., sounds that were not present in the original recording) may be audible in the reproduction. <hl> Some audio is hard to compress because of its randomness and sharp attacks. When this type of audio is compressed, artifacts such as ringing or pre-echo are usually heard. A sample of applause compressed with a relatively low bit rate provides a good example of compression artifacts.

With too low a bit rate, <hl> compression artifacts <hl> (i.e., sounds that were not present in the original recording) may be audible in the reproduction. Some audio is hard to compress because of its randomness and sharp attacks. When this type of audio is compressed, artifacts such as ringing or pre-echo are usually heard. A sample of applause compressed with a relatively low bit rate provides a good example of compression artifacts.


Question: Randomness and sharp attacks are two reasons that may make it harder to to do what to an audio file?

Answer: compress

<hl> With too low a bit rate, compress ion artifacts (i.e., sounds that were not present in the original recording) may be audible in the reproduction. <hl> Some audio is hard to compress because of its randomness and sharp attacks. When this type of audio is compressed, artifacts such as ringing or pre-echo are usually heard. A sample of applause compressed with a relatively low bit rate provides a good example of compression artifacts.

With too low a bit rate, <hl> compress <hl>ion artifacts (i.e., sounds that were not present in the original recording) may be audible in the reproduction. Some audio is hard to compress because of its randomness and sharp attacks. When this type of audio is compressed, artifacts such as ringing or pre-echo are usually heard. A sample of applause compressed with a relatively low bit rate provides a good example of compression artifacts.

With too low a bit rate, <hl> compress <hl> ion artifacts (i.e., sounds that were not present in the original recording) may be audible in the reproduction.


Question: Other than pre - echo, what is another example of what can happen when audio is compressed?

Answer: ringing

With too low a bit rate, compression artifacts (i.e., sounds that were not present in the original recording) may be audible in the reproduction. Some audio is hard to compress because of its randomness and sharp attacks. <hl> When this type of audio is compressed, artifacts such as ringing or pre-echo are usually heard. <hl> A sample of applause compressed with a relatively low bit rate provides a good example of compression artifacts.

With too low a bit rate, compression artifacts (i.e., sounds that were not present in the original recording) may be audible in the reproduction. Some audio is hard to compress because of its randomness and sharp attacks. When this type of audio is compressed, artifacts such as <hl> ringing <hl> or pre-echo are usually heard. A sample of applause compressed with a relatively low bit rate provides a good example of compression artifacts.


Question: What kind of sample can display a good example of compression artifacts?

Answer: A sample of applause

With too low a bit rate, compression artifacts (i.e., sounds that were not present in the original recording) may be audible in the reproduction. Some audio is hard to compress because of its randomness and sharp attacks. When this type of audio is compressed, artifacts such as ringing or pre-echo are usually heard. <hl> A sample of applause compressed with a relatively low bit rate provides a good example of compression artifacts. <hl>

With too low a bit rate, compression artifacts (i.e., sounds that were not present in the original recording) may be audible in the reproduction. Some audio is hard to compress because of its randomness and sharp attacks. When this type of audio is compressed, artifacts such as ringing or pre-echo are usually heard. <hl> A sample of applause <hl> compressed with a relatively low bit rate provides a good example of compression artifacts.


Question: Other than the bit rate and the difficulty of the signal, what can also affect the quality of an MP3 file?

Besides the bit rate of an encoded piece of audio, the quality of MP3 files also depends on the quality of the encoder itself, and the difficulty of the signal being encoded. As the MP3 standard allows quite a bit of freedom with encoding algorithms, different encoders may feature quite different quality, even with identical bit rates. As an example, in a public listening test featuring two different MP3 encoders at about 128 kbit/s, one scored 3.66 on a 1–5 scale, while the other scored only 2.22.

Answer: quality of the encoder

<hl> Besides the bit rate of an encoded piece of audio, the quality of MP3 files also depends on the quality of the encoder itself, and the difficulty of the signal being encoded. <hl> As the MP3 standard allows quite a bit of freedom with encoding algorithms, different encoders may feature quite different quality, even with identical bit rates. As an example, in a public listening test featuring two different MP3 encoders at about 128 kbit/s, one scored 3.66 on a 1–5 scale, while the other scored only 2.22.

Besides the bit rate of an encoded piece of audio, the quality of MP3 files also depends on the <hl> quality of the encoder <hl> itself, and the difficulty of the signal being encoded. As the MP3 standard allows quite a bit of freedom with encoding algorithms, different encoders may feature quite different quality, even with identical bit rates. As an example, in a public listening test featuring two different MP3 encoders at about 128 kbit/s, one scored 3.66 on a 1–5 scale, while the other scored only 2.22.


Question: The MP3 standard allows encoders to have a lot of freedom with which aspect of encoding?

Answer: encoding algorithms

Besides the bit rate of an encoded piece of audio, the quality of MP3 files also depends on the quality of the encoder itself, and the difficulty of the signal being encoded. <hl> As the MP3 standard allows quite a bit of freedom with encoding algorithms , different encoders may feature quite different quality, even with identical bit rates. <hl> As an example, in a public listening test featuring two different MP3 encoders at about 128 kbit/s, one scored 3.66 on a 1–5 scale, while the other scored only 2.22.

Besides the bit rate of an encoded piece of audio, the quality of MP3 files also depends on the quality of the encoder itself, and the difficulty of the signal being encoded. As the MP3 standard allows quite a bit of freedom with <hl> encoding algorithms <hl>, different encoders may feature quite different quality, even with identical bit rates. As an example, in a public listening test featuring two different MP3 encoders at about 128 kbit/s, one scored 3.66 on a 1–5 scale, while the other scored only 2.22.

As the MP3 standard allows quite a bit of freedom with <hl> encoding algorithms <hl> , different encoders may feature quite different quality, even with identical bit rates.


Question: As a result of the freedom, different encoders can result in different quality even if which aspect of the file is the same?

Answer: bit rates

Besides the bit rate of an encoded piece of audio, the quality of MP3 files also depends on the quality of the encoder itself, and the difficulty of the signal being encoded. <hl> As the MP3 standard allows quite a bit of freedom with encoding algorithms, different encoders may feature quite different quality, even with identical bit rates . <hl> As an example, in a public listening test featuring two different MP3 encoders at about 128 kbit/s, one scored 3.66 on a 1–5 scale, while the other scored only 2.22.

Besides the bit rate of an encoded piece of audio, the quality of MP3 files also depends on the quality of the encoder itself, and the difficulty of the signal being encoded. As the MP3 standard allows quite a bit of freedom with encoding algorithms, different encoders may feature quite different quality, even with identical <hl> bit rates <hl>. As an example, in a public listening test featuring two different MP3 encoders at about 128 kbit/s, one scored 3.66 on a 1–5 scale, while the other scored only 2.22.

As the MP3 standard allows quite a bit of freedom with encoding algorithms, different encoders may feature quite different quality, even with identical <hl> bit rates <hl> .


Question: An example shown has a low range quality score of what value?

Answer: 2.22

Besides the bit rate of an encoded piece of audio, the quality of MP3 files also depends on the quality of the encoder itself, and the difficulty of the signal being encoded. As the MP3 standard allows quite a bit of freedom with encoding algorithms, different encoders may feature quite different quality, even with identical bit rates. <hl> As an example, in a public listening test featuring two different MP3 encoders at about 128 kbit/s, one scored 3.66 on a 1–5 scale, while the other scored only 2.22 . <hl>

Besides the bit rate of an encoded piece of audio, the quality of MP3 files also depends on the quality of the encoder itself, and the difficulty of the signal being encoded. As the MP3 standard allows quite a bit of freedom with encoding algorithms, different encoders may feature quite different quality, even with identical bit rates. As an example, in a public listening test featuring two different MP3 encoders at about 128 kbit/s, one scored 3.66 on a 1–5 scale, while the other scored only <hl> 2.22 <hl>.

As an example, in a public listening test featuring two different MP3 encoders at about 128 kbit/s, one scored 3.66 on a 1–5 scale, while the other scored only <hl> 2.22 <hl> .


Question: To compare qualities, sound files are evaluated on a scale that ranges from 1 to what?

Answer: 5

Besides the bit rate of an encoded piece of audio, the quality of MP3 files also depends on the quality of the encoder itself, and the difficulty of the signal being encoded. As the MP3 standard allows quite a bit of freedom with encoding algorithms, different encoders may feature quite different quality, even with identical bit rates. <hl> As an example, in a public listening test featuring two different MP3 encoders at about 128 kbit/s, one scored 3.66 on a 1– 5 scale, while the other scored only 2.22. <hl>

Besides the bit rate of an encoded piece of audio, the quality of MP3 files also depends on the quality of the encoder itself, and the difficulty of the signal being encoded. As the MP3 standard allows quite a bit of freedom with encoding algorithms, different encoders may feature quite different quality, even with identical bit rates. As an example, in a public listening test featuring two different MP3 encoders at about 128 kbit/s, one scored 3.66 on a 1–<hl> 5 <hl> scale, while the other scored only 2.22.

As an example, in a public listening test featuring two different MP3 encoders at about 128 kbit/s, one scored 3.66 on a 1– <hl> 5 <hl> scale, while the other scored only 2.22.


Question: How many bit rates does the simplest type of MP3 file use?

The simplest type of MP3 file uses one bit rate for the entire file: this is known as Constant Bit Rate (CBR) encoding. Using a constant bit rate makes encoding simpler and faster. However, it is also possible to create files where the bit rate changes throughout the file. These are known as Variable Bit Rate (VBR) files. The idea behind this is that, in any piece of audio, some parts will be much easier to compress, such as silence or music containing only a few instruments, while others will be more difficult to compress. So, the overall quality of the file may be increased by using a lower bit rate for the less complex passages and a higher one for the more complex parts. With some encoders, it is possible to specify a given quality, and the encoder will vary the bit rate accordingly. Users who know a particular "quality setting" that is transparent to their ears can use this value when encoding all of their music, and generally speaking not need to worry about performing personal listening tests on each piece of music to determine the correct bit rate.

Answer: one

<hl> The simplest type of MP3 file uses one bit rate for the entire file: this is known as Constant Bit Rate (CBR) encoding. <hl> Using a constant bit rate makes encoding simpler and faster. However, it is also possible to create files where the bit rate changes throughout the file. These are known as Variable Bit Rate (VBR) files. The idea behind this is that, in any piece of audio, some parts will be much easier to compress, such as silence or music containing only a few instruments, while others will be more difficult to compress. So, the overall quality of the file may be increased by using a lower bit rate for the less complex passages and a higher one for the more complex parts. With some encoders, it is possible to specify a given quality, and the encoder will vary the bit rate accordingly. Users who know a particular "quality setting" that is transparent to their ears can use this value when encoding all of their music, and generally speaking not need to worry about performing personal listening tests on each piece of music to determine the correct bit rate.

The simplest type of MP3 file uses <hl> one <hl> bit rate for the entire file: this is known as Constant Bit Rate (CBR) encoding. Using a constant bit rate makes encoding simpler and faster. However, it is also possible to create files where the bit rate changes throughout the file. These are known as Variable Bit Rate (VBR) files. The idea behind this is that, in any piece of audio, some parts will be much easier to compress, such as silence or music containing only a few instruments, while others will be more difficult to compress. So, the overall quality of the file may be increased by using a lower bit rate for the less complex passages and a higher one for the more complex parts. With some encoders, it is possible to specify a given quality, and the encoder will vary the bit rate accordingly. Users who know a particular "quality setting" that is transparent to their ears can use this value when encoding all of their music, and generally speaking not need to worry about performing personal listening tests on each piece of music to determine the correct bit rate.


Question: What does CBR stand for?

Answer: Constant Bit Rate

<hl> The simplest type of MP3 file uses one bit rate for the entire file: this is known as Constant Bit Rate (CBR) encoding. <hl> Using a constant bit rate makes encoding simpler and faster. However, it is also possible to create files where the bit rate changes throughout the file. These are known as Variable Bit Rate (VBR) files. The idea behind this is that, in any piece of audio, some parts will be much easier to compress, such as silence or music containing only a few instruments, while others will be more difficult to compress. So, the overall quality of the file may be increased by using a lower bit rate for the less complex passages and a higher one for the more complex parts. With some encoders, it is possible to specify a given quality, and the encoder will vary the bit rate accordingly. Users who know a particular "quality setting" that is transparent to their ears can use this value when encoding all of their music, and generally speaking not need to worry about performing personal listening tests on each piece of music to determine the correct bit rate.

The simplest type of MP3 file uses one bit rate for the entire file: this is known as <hl> Constant Bit Rate <hl> (CBR) encoding. Using a constant bit rate makes encoding simpler and faster. However, it is also possible to create files where the bit rate changes throughout the file. These are known as Variable Bit Rate (VBR) files. The idea behind this is that, in any piece of audio, some parts will be much easier to compress, such as silence or music containing only a few instruments, while others will be more difficult to compress. So, the overall quality of the file may be increased by using a lower bit rate for the less complex passages and a higher one for the more complex parts. With some encoders, it is possible to specify a given quality, and the encoder will vary the bit rate accordingly. Users who know a particular "quality setting" that is transparent to their ears can use this value when encoding all of their music, and generally speaking not need to worry about performing personal listening tests on each piece of music to determine the correct bit rate.


Question: What does VBR stand for?

Answer: Variable Bit Rate

The simplest type of MP3 file uses one bit rate for the entire file: this is known as Constant Bit Rate (CBR) encoding. Using a constant bit rate makes encoding simpler and faster. However, it is also possible to create files where the bit rate changes throughout the file. <hl> These are known as Variable Bit Rate (VBR) files. <hl> The idea behind this is that, in any piece of audio, some parts will be much easier to compress, such as silence or music containing only a few instruments, while others will be more difficult to compress. So, the overall quality of the file may be increased by using a lower bit rate for the less complex passages and a higher one for the more complex parts. With some encoders, it is possible to specify a given quality, and the encoder will vary the bit rate accordingly. Users who know a particular "quality setting" that is transparent to their ears can use this value when encoding all of their music, and generally speaking not need to worry about performing personal listening tests on each piece of music to determine the correct bit rate.

The simplest type of MP3 file uses one bit rate for the entire file: this is known as Constant Bit Rate (CBR) encoding. Using a constant bit rate makes encoding simpler and faster. However, it is also possible to create files where the bit rate changes throughout the file. These are known as <hl> Variable Bit Rate <hl> (VBR) files. The idea behind this is that, in any piece of audio, some parts will be much easier to compress, such as silence or music containing only a few instruments, while others will be more difficult to compress. So, the overall quality of the file may be increased by using a lower bit rate for the less complex passages and a higher one for the more complex parts. With some encoders, it is possible to specify a given quality, and the encoder will vary the bit rate accordingly. Users who know a particular "quality setting" that is transparent to their ears can use this value when encoding all of their music, and generally speaking not need to worry about performing personal listening tests on each piece of music to determine the correct bit rate.


Question: Aside from silence, which sections of music highlight parts that are easier to compress?

Answer: music containing only a few instruments

The simplest type of MP3 file uses one bit rate for the entire file: this is known as Constant Bit Rate (CBR) encoding. Using a constant bit rate makes encoding simpler and faster. However, it is also possible to create files where the bit rate changes throughout the file. These are known as Variable Bit Rate (VBR) files. <hl> The idea behind this is that, in any piece of audio, some parts will be much easier to compress, such as silence or music containing only a few instruments , while others will be more difficult to compress. <hl> So, the overall quality of the file may be increased by using a lower bit rate for the less complex passages and a higher one for the more complex parts. With some encoders, it is possible to specify a given quality, and the encoder will vary the bit rate accordingly. Users who know a particular "quality setting" that is transparent to their ears can use this value when encoding all of their music, and generally speaking not need to worry about performing personal listening tests on each piece of music to determine the correct bit rate.

The simplest type of MP3 file uses one bit rate for the entire file: this is known as Constant Bit Rate (CBR) encoding. Using a constant bit rate makes encoding simpler and faster. However, it is also possible to create files where the bit rate changes throughout the file. These are known as Variable Bit Rate (VBR) files. The idea behind this is that, in any piece of audio, some parts will be much easier to compress, such as silence or <hl> music containing only a few instruments <hl>, while others will be more difficult to compress. So, the overall quality of the file may be increased by using a lower bit rate for the less complex passages and a higher one for the more complex parts. With some encoders, it is possible to specify a given quality, and the encoder will vary the bit rate accordingly. Users who know a particular "quality setting" that is transparent to their ears can use this value when encoding all of their music, and generally speaking not need to worry about performing personal listening tests on each piece of music to determine the correct bit rate.

The idea behind this is that, in any piece of audio, some parts will be much easier to compress, such as silence or <hl> music containing only a few instruments <hl> , while others will be more difficult to compress.


Question: What value do users need to know when encoding their music to help avoid them having to do tests on each piece of music?

Answer: quality setting

The simplest type of MP3 file uses one bit rate for the entire file: this is known as Constant Bit Rate (CBR) encoding. Using a constant bit rate makes encoding simpler and faster. However, it is also possible to create files where the bit rate changes throughout the file. These are known as Variable Bit Rate (VBR) files. The idea behind this is that, in any piece of audio, some parts will be much easier to compress, such as silence or music containing only a few instruments, while others will be more difficult to compress. So, the overall quality of the file may be increased by using a lower bit rate for the less complex passages and a higher one for the more complex parts. With some encoders, it is possible to specify a given quality, and the encoder will vary the bit rate accordingly. <hl> Users who know a particular " quality setting " that is transparent to their ears can use this value when encoding all of their music, and generally speaking not need to worry about performing personal listening tests on each piece of music to determine the correct bit rate. <hl>

The simplest type of MP3 file uses one bit rate for the entire file: this is known as Constant Bit Rate (CBR) encoding. Using a constant bit rate makes encoding simpler and faster. However, it is also possible to create files where the bit rate changes throughout the file. These are known as Variable Bit Rate (VBR) files. The idea behind this is that, in any piece of audio, some parts will be much easier to compress, such as silence or music containing only a few instruments, while others will be more difficult to compress. So, the overall quality of the file may be increased by using a lower bit rate for the less complex passages and a higher one for the more complex parts. With some encoders, it is possible to specify a given quality, and the encoder will vary the bit rate accordingly. Users who know a particular "<hl> quality setting <hl>" that is transparent to their ears can use this value when encoding all of their music, and generally speaking not need to worry about performing personal listening tests on each piece of music to determine the correct bit rate.

Users who know a particular " <hl> quality setting <hl> " that is transparent to their ears can use this value when encoding all of their music, and generally speaking not need to worry about performing personal listening tests on each piece of music to determine the correct bit rate.


Question: A listening environment is also know by which term?

Perceived quality can be influenced by listening environment (ambient noise), listener attention, and listener training and in most cases by listener audio equipment (such as sound cards, speakers and headphones).

Answer: ambient noise

<hl> Perceived quality can be influenced by listening environment ( ambient noise ), listener attention, and listener training and in most cases by listener audio equipment (such as sound cards, speakers and headphones). <hl>

Perceived quality can be influenced by listening environment (<hl> ambient noise <hl>), listener attention, and listener training and in most cases by listener audio equipment (such as sound cards, speakers and headphones).

Perceived quality can be influenced by listening environment ( <hl> ambient noise <hl> ), listener attention, and listener training and in most cases by listener audio equipment (such as sound cards, speakers and headphones).


Question: Other than speakers and headphones, what other piece of equipment affects the listener's perceived quality?

Answer: sound cards

<hl> Perceived quality can be influenced by listening environment (ambient noise), listener attention, and listener training and in most cases by listener audio equipment (such as sound cards , speakers and headphones). <hl>

Perceived quality can be influenced by listening environment (ambient noise), listener attention, and listener training and in most cases by listener audio equipment (such as <hl> sound cards <hl>, speakers and headphones).

Perceived quality can be influenced by listening environment (ambient noise), listener attention, and listener training and in most cases by listener audio equipment (such as <hl> sound cards <hl> , speakers and headphones).


Question: Listening environment, listener attention, listener training and listener audio equipment can all affect what kind of quality?

Answer: Perceived quality

<hl> Perceived quality can be influenced by listening environment (ambient noise), listener attention, and listener training and in most cases by listener audio equipment (such as sound cards, speakers and headphones). <hl>

<hl> Perceived quality <hl> can be influenced by listening environment (ambient noise), listener attention, and listener training and in most cases by listener audio equipment (such as sound cards, speakers and headphones).


Question: Who gave students a test to determine their preferences?

A test given to new students by Stanford University Music Professor Jonathan Berger showed that student preference for MP3-quality music has risen each year. Berger said the students seem to prefer the 'sizzle' sounds that MP3s bring to music.

Answer: Music Professor Jonathan Berger

<hl> A test given to new students by Stanford University Music Professor Jonathan Berger showed that student preference for MP3-quality music has risen each year. <hl> Berger said the students seem to prefer the 'sizzle' sounds that MP3s bring to music.

A test given to new students by Stanford University <hl> Music Professor Jonathan Berger <hl> showed that student preference for MP3-quality music has risen each year. Berger said the students seem to prefer the 'sizzle' sounds that MP3s bring to music.


Question: Where was a test performed to collect data on student preference for MP3 quality?

Answer: Stanford University

<hl> A test given to new students by Stanford University Music Professor Jonathan Berger showed that student preference for MP3-quality music has risen each year. <hl> Berger said the students seem to prefer the 'sizzle' sounds that MP3s bring to music.

A test given to new students by <hl> Stanford University <hl> Music Professor Jonathan Berger showed that student preference for MP3-quality music has risen each year. Berger said the students seem to prefer the 'sizzle' sounds that MP3s bring to music.


Question: Data showed that students' preference for MP3 quality music followed which trend?

Answer: risen each year

<hl> A test given to new students by Stanford University Music Professor Jonathan Berger showed that student preference for MP3-quality music has risen each year . <hl> Berger said the students seem to prefer the 'sizzle' sounds that MP3s bring to music.

A test given to new students by Stanford University Music Professor Jonathan Berger showed that student preference for MP3-quality music has <hl> risen each year <hl>. Berger said the students seem to prefer the 'sizzle' sounds that MP3s bring to music.

A test given to new students by Stanford University Music Professor Jonathan Berger showed that student preference for MP3-quality music has <hl> risen each year <hl> .


Question: What kind of sound did students seem to prefer?

Answer: sizzle

A test given to new students by Stanford University Music Professor Jonathan Berger showed that student preference for MP3-quality music has risen each year. <hl> Berger said the students seem to prefer the ' sizzle ' sounds that MP3s bring to music. <hl>

A test given to new students by Stanford University Music Professor Jonathan Berger showed that student preference for MP3-quality music has risen each year. Berger said the students seem to prefer the '<hl> sizzle <hl>' sounds that MP3s bring to music.

Berger said the students seem to prefer the ' <hl> sizzle <hl> ' sounds that MP3s bring to music.


Question: What is the name of the study done by composer Ryan Maguire?

An in-depth study of MP3 audio quality, sound artist and composer Ryan Maguire's project "The Ghost in the MP3" isolates the sounds lost during MP3 compression. In 2015, he released the track "moDernisT" (an anagram of "Tom's Diner"), composed exclusively from the sounds deleted during MP3 compression of the song "Tom's Diner", the track originally used in the formulation of the MP3 standard. A detailed account of the techniques used to isolate the sounds deleted during MP3 compression, along with the conceptual motivation for the project, was published in the 2014 Proceedings of the International Computer Music Conference.

Answer: The Ghost in the MP3

<hl> An in-depth study of MP3 audio quality, sound artist and composer Ryan Maguire's project " The Ghost in the MP3 " isolates the sounds lost during MP3 compression. <hl> In 2015, he released the track "moDernisT" (an anagram of "Tom's Diner"), composed exclusively from the sounds deleted during MP3 compression of the song "Tom's Diner", the track originally used in the formulation of the MP3 standard. A detailed account of the techniques used to isolate the sounds deleted during MP3 compression, along with the conceptual motivation for the project, was published in the 2014 Proceedings of the International Computer Music Conference.

An in-depth study of MP3 audio quality, sound artist and composer Ryan Maguire's project "<hl> The Ghost in the MP3 <hl>" isolates the sounds lost during MP3 compression. In 2015, he released the track "moDernisT" (an anagram of "Tom's Diner"), composed exclusively from the sounds deleted during MP3 compression of the song "Tom's Diner", the track originally used in the formulation of the MP3 standard. A detailed account of the techniques used to isolate the sounds deleted during MP3 compression, along with the conceptual motivation for the project, was published in the 2014 Proceedings of the International Computer Music Conference.

An in-depth study of MP3 audio quality, sound artist and composer Ryan Maguire's project " <hl> The Ghost in the MP3 <hl> " isolates the sounds lost during MP3 compression.


Question: What did the study aim to isolate?

Answer: sounds lost during MP3 compression

<hl> An in-depth study of MP3 audio quality, sound artist and composer Ryan Maguire's project "The Ghost in the MP3" isolates the sounds lost during MP3 compression . <hl> In 2015, he released the track "moDernisT" (an anagram of "Tom's Diner"), composed exclusively from the sounds deleted during MP3 compression of the song "Tom's Diner", the track originally used in the formulation of the MP3 standard. A detailed account of the techniques used to isolate the sounds deleted during MP3 compression, along with the conceptual motivation for the project, was published in the 2014 Proceedings of the International Computer Music Conference.

An in-depth study of MP3 audio quality, sound artist and composer Ryan Maguire's project "The Ghost in the MP3" isolates the <hl> sounds lost during MP3 compression <hl>. In 2015, he released the track "moDernisT" (an anagram of "Tom's Diner"), composed exclusively from the sounds deleted during MP3 compression of the song "Tom's Diner", the track originally used in the formulation of the MP3 standard. A detailed account of the techniques used to isolate the sounds deleted during MP3 compression, along with the conceptual motivation for the project, was published in the 2014 Proceedings of the International Computer Music Conference.

An in-depth study of MP3 audio quality, sound artist and composer Ryan Maguire's project "The Ghost in the MP3" isolates the <hl> sounds lost during MP3 compression <hl> .


Question: What is the track "moDernisT" an anagram of?

Answer: Tom's Diner

An in-depth study of MP3 audio quality, sound artist and composer Ryan Maguire's project "The Ghost in the MP3" isolates the sounds lost during MP3 compression. <hl> In 2015, he released the track "moDernisT" (an anagram of " Tom's Diner "), composed exclusively from the sounds deleted during MP3 compression of the song "Tom's Diner", the track originally used in the formulation of the MP3 standard. <hl> A detailed account of the techniques used to isolate the sounds deleted during MP3 compression, along with the conceptual motivation for the project, was published in the 2014 Proceedings of the International Computer Music Conference.

An in-depth study of MP3 audio quality, sound artist and composer Ryan Maguire's project "The Ghost in the MP3" isolates the sounds lost during MP3 compression. In 2015, he released the track "moDernisT" (an anagram of "<hl> Tom's Diner <hl>"), composed exclusively from the sounds deleted during MP3 compression of the song "Tom's Diner", the track originally used in the formulation of the MP3 standard. A detailed account of the techniques used to isolate the sounds deleted during MP3 compression, along with the conceptual motivation for the project, was published in the 2014 Proceedings of the International Computer Music Conference.

In 2015, he released the track "moDernisT" (an anagram of " <hl> Tom's Diner <hl> "), composed exclusively from the sounds deleted during MP3 compression of the song "Tom's Diner", the track originally used in the formulation of the MP3 standard.


Question: Where was an account of the study published?

Answer: International Computer Music Conference

An in-depth study of MP3 audio quality, sound artist and composer Ryan Maguire's project "The Ghost in the MP3" isolates the sounds lost during MP3 compression. In 2015, he released the track "moDernisT" (an anagram of "Tom's Diner"), composed exclusively from the sounds deleted during MP3 compression of the song "Tom's Diner", the track originally used in the formulation of the MP3 standard. <hl> A detailed account of the techniques used to isolate the sounds deleted during MP3 compression, along with the conceptual motivation for the project, was published in the 2014 Proceedings of the International Computer Music Conference . <hl>

An in-depth study of MP3 audio quality, sound artist and composer Ryan Maguire's project "The Ghost in the MP3" isolates the sounds lost during MP3 compression. In 2015, he released the track "moDernisT" (an anagram of "Tom's Diner"), composed exclusively from the sounds deleted during MP3 compression of the song "Tom's Diner", the track originally used in the formulation of the MP3 standard. A detailed account of the techniques used to isolate the sounds deleted during MP3 compression, along with the conceptual motivation for the project, was published in the 2014 Proceedings of the <hl> International Computer Music Conference <hl>.

A detailed account of the techniques used to isolate the sounds deleted during MP3 compression, along with the conceptual motivation for the project, was published in the 2014 Proceedings of the <hl> International Computer Music Conference <hl> .


Question: Which year was the study published?

Answer: 2014

An in-depth study of MP3 audio quality, sound artist and composer Ryan Maguire's project "The Ghost in the MP3" isolates the sounds lost during MP3 compression. In 2015, he released the track "moDernisT" (an anagram of "Tom's Diner"), composed exclusively from the sounds deleted during MP3 compression of the song "Tom's Diner", the track originally used in the formulation of the MP3 standard. <hl> A detailed account of the techniques used to isolate the sounds deleted during MP3 compression, along with the conceptual motivation for the project, was published in the 2014 Proceedings of the International Computer Music Conference. <hl>

An in-depth study of MP3 audio quality, sound artist and composer Ryan Maguire's project "The Ghost in the MP3" isolates the sounds lost during MP3 compression. In 2015, he released the track "moDernisT" (an anagram of "Tom's Diner"), composed exclusively from the sounds deleted during MP3 compression of the song "Tom's Diner", the track originally used in the formulation of the MP3 standard. A detailed account of the techniques used to isolate the sounds deleted during MP3 compression, along with the conceptual motivation for the project, was published in the <hl> 2014 <hl> Proceedings of the International Computer Music Conference.


Question: What is the name of the sampling theorem?

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

Answer: Nyquist–Shannon

Several bit rates are specified in the MPEG-1 Audio Layer III standard: 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kbit/s, with available sampling frequencies of 32, 44.1 and 48 kHz. MPEG-2 Audio Layer III allows bit rates of 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 kbit/s with sampling frequencies of 16, 22.05 and 24 kHz. <hl> MPEG-2.5 Audio Layer III is restricted to bit rates of 8, 16, 24, 32, 40, 48, 56 and 64 kbit/s with sampling frequencies of 8, 11.025, and 12 kHz.[citation needed] Because of the Nyquist–Shannon sampling theorem, frequency reproduction is always strictly less than half of the sampling frequency, and imperfect filters requires a larger margin for error (noise level versus sharpness of filter), so 8 kHz sampling rate limits the maximum frequency to 4 kHz, while 48 kHz maximum sampling rate limits an MP3 to 24 kHz sound reproduction. <hl>

Several bit rates are specified in the MPEG-1 Audio Layer III standard: 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kbit/s, with available sampling frequencies of 32, 44.1 and 48 kHz. MPEG-2 Audio Layer III allows bit rates of 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 kbit/s with sampling frequencies of 16, 22.05 and 24 kHz. MPEG-2.5 Audio Layer III is restricted to bit rates of 8, 16, 24, 32, 40, 48, 56 and 64 kbit/s with sampling frequencies of 8, 11.025, and 12 kHz.[citation needed] Because of the <hl> Nyquist–Shannon <hl> sampling theorem, frequency reproduction is always strictly less than half of the sampling frequency, and imperfect filters requires a larger margin for error (noise level versus sharpness of filter), so 8 kHz sampling rate limits the maximum frequency to 4 kHz, while 48 kHz maximum sampling rate limits an MP3 to 24 kHz sound reproduction.


Question: In relation to the sampling frequency, the frequency reproduction is strictly how much less?

Answer: less than half

Several bit rates are specified in the MPEG-1 Audio Layer III standard: 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kbit/s, with available sampling frequencies of 32, 44.1 and 48 kHz. MPEG-2 Audio Layer III allows bit rates of 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 kbit/s with sampling frequencies of 16, 22.05 and 24 kHz. <hl> MPEG-2.5 Audio Layer III is restricted to bit rates of 8, 16, 24, 32, 40, 48, 56 and 64 kbit/s with sampling frequencies of 8, 11.025, and 12 kHz.[citation needed] Because of the Nyquist–Shannon sampling theorem, frequency reproduction is always strictly less than half of the sampling frequency, and imperfect filters requires a larger margin for error (noise level versus sharpness of filter), so 8 kHz sampling rate limits the maximum frequency to 4 kHz, while 48 kHz maximum sampling rate limits an MP3 to 24 kHz sound reproduction. <hl>

Several bit rates are specified in the MPEG-1 Audio Layer III standard: 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kbit/s, with available sampling frequencies of 32, 44.1 and 48 kHz. MPEG-2 Audio Layer III allows bit rates of 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 kbit/s with sampling frequencies of 16, 22.05 and 24 kHz. MPEG-2.5 Audio Layer III is restricted to bit rates of 8, 16, 24, 32, 40, 48, 56 and 64 kbit/s with sampling frequencies of 8, 11.025, and 12 kHz.[citation needed] Because of the Nyquist–Shannon sampling theorem, frequency reproduction is always strictly <hl> less than half <hl> of the sampling frequency, and imperfect filters requires a larger margin for error (noise level versus sharpness of filter), so 8 kHz sampling rate limits the maximum frequency to 4 kHz, while 48 kHz maximum sampling rate limits an MP3 to 24 kHz sound reproduction.


Question: Having an 8 kHz sampling rate would limit the maximum frequency to how much?

Answer: 4 kHz

Several bit rates are specified in the MPEG-1 Audio Layer III standard: 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kbit/s, with available sampling frequencies of 32, 44.1 and 48 kHz. <hl> MPEG-2 Audio Layer III allows bit rates of 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 kbit/s with sampling frequencies of 16, 22.05 and 2 4 kHz . <hl> MPEG-2.5 Audio Layer III is restricted to bit rates of 8, 16, 24, 32, 40, 48, 56 and 64 kbit/s with sampling frequencies of 8, 11.025, and 12 kHz.[citation needed] Because of the Nyquist–Shannon sampling theorem, frequency reproduction is always strictly less than half of the sampling frequency, and imperfect filters requires a larger margin for error (noise level versus sharpness of filter), so 8 kHz sampling rate limits the maximum frequency to 4 kHz, while 48 kHz maximum sampling rate limits an MP3 to 24 kHz sound reproduction.

Several bit rates are specified in the MPEG-1 Audio Layer III standard: 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kbit/s, with available sampling frequencies of 32, 44.1 and 48 kHz. MPEG-2 Audio Layer III allows bit rates of 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 kbit/s with sampling frequencies of 16, 22.05 and 2<hl> 4 kHz <hl>. MPEG-2.5 Audio Layer III is restricted to bit rates of 8, 16, 24, 32, 40, 48, 56 and 64 kbit/s with sampling frequencies of 8, 11.025, and 12 kHz.[citation needed] Because of the Nyquist–Shannon sampling theorem, frequency reproduction is always strictly less than half of the sampling frequency, and imperfect filters requires a larger margin for error (noise level versus sharpness of filter), so 8 kHz sampling rate limits the maximum frequency to 4 kHz, while 48 kHz maximum sampling rate limits an MP3 to 24 kHz sound reproduction.

MPEG-2 Audio Layer III allows bit rates of 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 kbit/s with sampling frequencies of 16, 22.05 and 2 <hl> 4 kHz <hl> .


Question: What kind of filter requires a larger margin for error?

Answer: imperfect filters

Several bit rates are specified in the MPEG-1 Audio Layer III standard: 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kbit/s, with available sampling frequencies of 32, 44.1 and 48 kHz. MPEG-2 Audio Layer III allows bit rates of 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 kbit/s with sampling frequencies of 16, 22.05 and 24 kHz. <hl> MPEG-2.5 Audio Layer III is restricted to bit rates of 8, 16, 24, 32, 40, 48, 56 and 64 kbit/s with sampling frequencies of 8, 11.025, and 12 kHz.[citation needed] Because of the Nyquist–Shannon sampling theorem, frequency reproduction is always strictly less than half of the sampling frequency, and imperfect filters requires a larger margin for error (noise level versus sharpness of filter), so 8 kHz sampling rate limits the maximum frequency to 4 kHz, while 48 kHz maximum sampling rate limits an MP3 to 24 kHz sound reproduction. <hl>

Several bit rates are specified in the MPEG-1 Audio Layer III standard: 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kbit/s, with available sampling frequencies of 32, 44.1 and 48 kHz. MPEG-2 Audio Layer III allows bit rates of 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 kbit/s with sampling frequencies of 16, 22.05 and 24 kHz. MPEG-2.5 Audio Layer III is restricted to bit rates of 8, 16, 24, 32, 40, 48, 56 and 64 kbit/s with sampling frequencies of 8, 11.025, and 12 kHz.[citation needed] Because of the Nyquist–Shannon sampling theorem, frequency reproduction is always strictly less than half of the sampling frequency, and <hl> imperfect filters <hl> requires a larger margin for error (noise level versus sharpness of filter), so 8 kHz sampling rate limits the maximum frequency to 4 kHz, while 48 kHz maximum sampling rate limits an MP3 to 24 kHz sound reproduction.


Question: A sound reproduction of 24 kHz would represent which value of a maximum sampling rate?

Answer: 48

<hl> Several bit rates are specified in the MPEG-1 Audio Layer III standard: 32, 40, 48 , 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kbit/s, with available sampling frequencies of 32, 44.1 and 48 kHz. <hl> MPEG-2 Audio Layer III allows bit rates of 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 kbit/s with sampling frequencies of 16, 22.05 and 24 kHz. MPEG-2.5 Audio Layer III is restricted to bit rates of 8, 16, 24, 32, 40, 48, 56 and 64 kbit/s with sampling frequencies of 8, 11.025, and 12 kHz.[citation needed] Because of the Nyquist–Shannon sampling theorem, frequency reproduction is always strictly less than half of the sampling frequency, and imperfect filters requires a larger margin for error (noise level versus sharpness of filter), so 8 kHz sampling rate limits the maximum frequency to 4 kHz, while 48 kHz maximum sampling rate limits an MP3 to 24 kHz sound reproduction.

Several bit rates are specified in the MPEG-1 Audio Layer III standard: 32, 40, <hl> 48 <hl>, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kbit/s, with available sampling frequencies of 32, 44.1 and 48 kHz. MPEG-2 Audio Layer III allows bit rates of 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 kbit/s with sampling frequencies of 16, 22.05 and 24 kHz. MPEG-2.5 Audio Layer III is restricted to bit rates of 8, 16, 24, 32, 40, 48, 56 and 64 kbit/s with sampling frequencies of 8, 11.025, and 12 kHz.[citation needed] Because of the Nyquist–Shannon sampling theorem, frequency reproduction is always strictly less than half of the sampling frequency, and imperfect filters requires a larger margin for error (noise level versus sharpness of filter), so 8 kHz sampling rate limits the maximum frequency to 4 kHz, while 48 kHz maximum sampling rate limits an MP3 to 24 kHz sound reproduction.

Several bit rates are specified in the MPEG-1 Audio Layer III standard: 32, 40, <hl> 48 <hl> , 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kbit/s, with available sampling frequencies of 32, 44.1 and 48 kHz.


Question: What sampling rate is used for CD audio?

A sample rate of 44.1 kHz is almost always used, because this is also used for CD audio, the main source used for creating MP3 files. A greater variety of bit rates are used on the Internet. The rate of 128 kbit/s is commonly used, at a compression ratio of 11:1, offering adequate audio quality in a relatively small space. As Internet bandwidth availability and hard drive sizes have increased, higher bit rates up to 320 kbit/s are widespread.

Answer: 44.1 kHz

<hl> A sample rate of 44.1 kHz is almost always used, because this is also used for CD audio, the main source used for creating MP3 files. <hl> A greater variety of bit rates are used on the Internet. The rate of 128 kbit/s is commonly used, at a compression ratio of 11:1, offering adequate audio quality in a relatively small space. As Internet bandwidth availability and hard drive sizes have increased, higher bit rates up to 320 kbit/s are widespread.

A sample rate of <hl> 44.1 kHz <hl> is almost always used, because this is also used for CD audio, the main source used for creating MP3 files. A greater variety of bit rates are used on the Internet. The rate of 128 kbit/s is commonly used, at a compression ratio of 11:1, offering adequate audio quality in a relatively small space. As Internet bandwidth availability and hard drive sizes have increased, higher bit rates up to 320 kbit/s are widespread.


Question: What is the main source used for creating MP3 files?

Answer: CD audio

<hl> A sample rate of 44.1 kHz is almost always used, because this is also used for CD audio , the main source used for creating MP3 files. <hl> A greater variety of bit rates are used on the Internet. The rate of 128 kbit/s is commonly used, at a compression ratio of 11:1, offering adequate audio quality in a relatively small space. As Internet bandwidth availability and hard drive sizes have increased, higher bit rates up to 320 kbit/s are widespread.

A sample rate of 44.1 kHz is almost always used, because this is also used for <hl> CD audio <hl>, the main source used for creating MP3 files. A greater variety of bit rates are used on the Internet. The rate of 128 kbit/s is commonly used, at a compression ratio of 11:1, offering adequate audio quality in a relatively small space. As Internet bandwidth availability and hard drive sizes have increased, higher bit rates up to 320 kbit/s are widespread.

A sample rate of 44.1 kHz is almost always used, because this is also used for <hl> CD audio <hl> , the main source used for creating MP3 files.


Question: Where can one find a greater variety of bit rates?

Answer: the Internet

A sample rate of 44.1 kHz is almost always used, because this is also used for CD audio, the main source used for creating MP3 files. <hl> A greater variety of bit rates are used on the Internet . <hl> The rate of 128 kbit/s is commonly used, at a compression ratio of 11:1, offering adequate audio quality in a relatively small space. As Internet bandwidth availability and hard drive sizes have increased, higher bit rates up to 320 kbit/s are widespread.

A sample rate of 44.1 kHz is almost always used, because this is also used for CD audio, the main source used for creating MP3 files. A greater variety of bit rates are used on <hl> the Internet <hl>. The rate of 128 kbit/s is commonly used, at a compression ratio of 11:1, offering adequate audio quality in a relatively small space. As Internet bandwidth availability and hard drive sizes have increased, higher bit rates up to 320 kbit/s are widespread.

A greater variety of bit rates are used on <hl> the Internet <hl> .


Question: The common bit rate on the internet is 128 kbit/s using which compression ratio?

Answer: 11:1

A sample rate of 44.1 kHz is almost always used, because this is also used for CD audio, the main source used for creating MP3 files. A greater variety of bit rates are used on the Internet. <hl> The rate of 128 kbit/s is commonly used, at a compression ratio of 11:1 , offering adequate audio quality in a relatively small space. <hl> As Internet bandwidth availability and hard drive sizes have increased, higher bit rates up to 320 kbit/s are widespread.

A sample rate of 44.1 kHz is almost always used, because this is also used for CD audio, the main source used for creating MP3 files. A greater variety of bit rates are used on the Internet. The rate of 128 kbit/s is commonly used, at a compression ratio of <hl> 11:1 <hl>, offering adequate audio quality in a relatively small space. As Internet bandwidth availability and hard drive sizes have increased, higher bit rates up to 320 kbit/s are widespread.

The rate of 128 kbit/s is commonly used, at a compression ratio of <hl> 11:1 <hl> , offering adequate audio quality in a relatively small space.


Question: As technology advances, what is another higher bit rate that is beginning to spread?

Answer: 320 kbit/s

A sample rate of 44.1 kHz is almost always used, because this is also used for CD audio, the main source used for creating MP3 files. A greater variety of bit rates are used on the Internet. The rate of 128 kbit/s is commonly used, at a compression ratio of 11:1, offering adequate audio quality in a relatively small space. <hl> As Internet bandwidth availability and hard drive sizes have increased, higher bit rates up to 320 kbit/s are widespread. <hl>

A sample rate of 44.1 kHz is almost always used, because this is also used for CD audio, the main source used for creating MP3 files. A greater variety of bit rates are used on the Internet. The rate of 128 kbit/s is commonly used, at a compression ratio of 11:1, offering adequate audio quality in a relatively small space. As Internet bandwidth availability and hard drive sizes have increased, higher bit rates up to <hl> 320 kbit/s <hl> are widespread.


Question: What is stored on an audio-CD that has a bit rate of 1,411.2 kbit/s?

Answer: Uncompressed audio

<hl> Uncompressed audio as stored on an audio-CD has a bit rate of 1,411.2 kbit/s, (16 bit/sample × 44100 samples/second × 2 channels / 1000 bits/kilobit), so the bitrates 128, 160 and 192 kbit/s represent compression ratios of approximately 11:1, 9:1 and 7:1 respectively. <hl>

<hl> Uncompressed audio <hl> as stored on an audio-CD has a bit rate of 1,411.2 kbit/s, (16 bit/sample × 44100 samples/second × 2 channels / 1000 bits/kilobit), so the bitrates 128, 160 and 192 kbit/s represent compression ratios of approximately 11:1, 9:1 and 7:1 respectively.


Question: What does a bit rate of 128 represent in terms of compression ratio?

Answer: 11:1

<hl> Uncompressed audio as stored on an audio-CD has a bit rate of 1,411.2 kbit/s, (16 bit/sample × 44100 samples/second × 2 channels / 1000 bits/kilobit), so the bitrates 128, 160 and 192 kbit/s represent compression ratios of approximately 11:1 , 9:1 and 7:1 respectively. <hl>

Uncompressed audio as stored on an audio-CD has a bit rate of 1,411.2 kbit/s, (16 bit/sample × 44100 samples/second × 2 channels / 1000 bits/kilobit), so the bitrates 128, 160 and 192 kbit/s represent compression ratios of approximately <hl> 11:1 <hl>, 9:1 and 7:1 respectively.

Uncompressed audio as stored on an audio-CD has a bit rate of 1,411.2 kbit/s, (16 bit/sample × 44100 samples/second × 2 channels / 1000 bits/kilobit), so the bitrates 128, 160 and 192 kbit/s represent compression ratios of approximately <hl> 11:1 <hl> , 9:1 and 7:1 respectively.


Question: What would a bit rate of 160 correspond to in terms of compression ratio?

Answer: 9:1

<hl> Uncompressed audio as stored on an audio-CD has a bit rate of 1,411.2 kbit/s, (16 bit/sample × 44100 samples/second × 2 channels / 1000 bits/kilobit), so the bitrates 128, 160 and 192 kbit/s represent compression ratios of approximately 11:1, 9:1 and 7:1 respectively. <hl>

Uncompressed audio as stored on an audio-CD has a bit rate of 1,411.2 kbit/s, (16 bit/sample × 44100 samples/second × 2 channels / 1000 bits/kilobit), so the bitrates 128, 160 and 192 kbit/s represent compression ratios of approximately 11:1, <hl> 9:1 <hl> and 7:1 respectively.


Question: What would a bit rate of 192 kbit/s have for a compression ratio?

Answer: 7:1

<hl> Uncompressed audio as stored on an audio-CD has a bit rate of 1,411.2 kbit/s, (16 bit/sample × 44100 samples/second × 2 channels / 1000 bits/kilobit), so the bitrates 128, 160 and 192 kbit/s represent compression ratios of approximately 11:1, 9:1 and 7:1 respectively. <hl>

Uncompressed audio as stored on an audio-CD has a bit rate of 1,411.2 kbit/s, (16 bit/sample × 44100 samples/second × 2 channels / 1000 bits/kilobit), so the bitrates 128, 160 and 192 kbit/s represent compression ratios of approximately 11:1, 9:1 and <hl> 7:1 <hl> respectively.


Question: The LAME encoder allows bit rates that are what?

Non-standard bit rates up to 640 kbit/s can be achieved with the LAME encoder and the freeformat option, although few MP3 players can play those files. According to the ISO standard, decoders are only required to be able to decode streams up to 320 kbit/s.

Answer: Non-standard

<hl> Non-standard bit rates up to 640 kbit/s can be achieved with the LAME encoder and the freeformat option, although few MP3 players can play those files. <hl> According to the ISO standard, decoders are only required to be able to decode streams up to 320 kbit/s.

<hl> Non-standard <hl> bit rates up to 640 kbit/s can be achieved with the LAME encoder and the freeformat option, although few MP3 players can play those files. According to the ISO standard, decoders are only required to be able to decode streams up to 320 kbit/s.


Question: Even though few MP3 players can play it, LAME also offers which option?

Answer: freeformat

<hl> Non-standard bit rates up to 640 kbit/s can be achieved with the LAME encoder and the freeformat option, although few MP3 players can play those files. <hl> According to the ISO standard, decoders are only required to be able to decode streams up to 320 kbit/s.

Non-standard bit rates up to 640 kbit/s can be achieved with the LAME encoder and the <hl> freeformat <hl> option, although few MP3 players can play those files. According to the ISO standard, decoders are only required to be able to decode streams up to 320 kbit/s.


Question: What is the highest bit rate the LAME encoder allows?

Answer: 640 kbit/s

<hl> Non-standard bit rates up to 640 kbit/s can be achieved with the LAME encoder and the freeformat option, although few MP3 players can play those files. <hl> According to the ISO standard, decoders are only required to be able to decode streams up to 320 kbit/s.

Non-standard bit rates up to <hl> 640 kbit/s <hl> can be achieved with the LAME encoder and the freeformat option, although few MP3 players can play those files. According to the ISO standard, decoders are only required to be able to decode streams up to 320 kbit/s.


Question: The ISO standard dictates that decoders only need to be able to decode streams that have how high of a bit rate?

Answer: 320

Non-standard bit rates up to 640 kbit/s can be achieved with the LAME encoder and the freeformat option, although few MP3 players can play those files. <hl> According to the ISO standard, decoders are only required to be able to decode streams up to 320 kbit/s. <hl>

Non-standard bit rates up to 640 kbit/s can be achieved with the LAME encoder and the freeformat option, although few MP3 players can play those files. According to the ISO standard, decoders are only required to be able to decode streams up to <hl> 320 <hl> kbit/s.