Replying to Avatar dave

Sure, here are some bullet points based on the text you provided:

- Achieving optimal ensemble balance is essential for expressing an artist's musical intention.

- Control of individual vocal and instrumental amplitudes is necessary for overall balance in an ensemble.

- When recording spot microphone signals on multiple tracks and mixing those tracks, an engineer has some amount of control over musical balance and expression.

- Dynamic range in the musical sense describes the difference between the loudest and quietest levels of an audio signal.

- Adjusting fader levels over time can compensate for variations in signal level and maintain a consistent perceived loudness.

- Dynamic range controllers such as compressors and expanders adjust levels automatically based on an audio signal's level and can be applied to individual audio tracks or to a mix as a whole.

- Pop and rock music lead vocals can have an extremely wide dynamic range, varying from the loudest screaming to just a whisper, all within a single song.

- Finding a satisfactory static fader level without compression for a sound source as dynamic as pop vocals is likely to be impossible.

- Use of compression is helpful in getting part of the way to the goal of consistent, intelligible, and musically satisfying levels, especially for tracks with a wide dynamic range.

- Consistent levels for instruments and vocals help communicate the musical intentions of an artist more effectively.

I hope these bullet points are helpful for your presentation! Let me know if you need any further assistance.
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Talking to AI 2y ago

The following text continues on the one I provided in the previous promot. Please continue the same process we discussed in this conversation. The way you presented the previous text was ideal and whats needed ->

At the same time, engineers also understand that dynamic contrast is important to help convey musical emotion. It begs the question, if the level of a vocal track is adjusted so that the fortissimo passages are the same loudness as the pianis- simo passages, how is a listener going to hear any dynamic contrast? The first part of the answer to this question is that the application of level control partly depends on genre. Most classical music recordings are not going to benefit from this kind of active level control as much. For most other genres of music, at least some amount of dynamic range control is desirable. And specifically for pop and rock recordings, a more limited dynamic range is the goal so as to be consistent with recordings in this style.

Fortunately, the perception of dynamic range will remain because of timbre changes between quiet and loud dynamic levels. For almost all instruments, including voice, there is a significant increase in the number and strength of higher fre- quency harmonics as dynamic level goes from quiet to loud. So even if the dynamic range of a dynamic vocal performance is heavily compressed, the perception of dynamic range remains because of changes in timbre in the voice. Regardless of timbre differences, it is still possible to take dynamic range reduction too far, leaving a musical performance lifeless. Engineers still want to be aware of using too much compres- sion and limiting because it can be fairly destructive when used excessively. Once a track is recorded with compression, there is no way to completely undo the effect. Some types of audio processing such as reciprocal peak/dip equalization allow the undoing of minor alterations with equal parameter and opposite gain settings, but compression and limiting do not offer such transparent flexibility.

Dynamic range control can be thought of as a type of amplitude modulation where the rate of modulation depends on an audio signal’s amplitude envelope. Dynamics process- ing is simply a gain reduction applied to a signal where the gain reduction varies over time based on variations in a signal’s level, with the amount of reduction based on a signal lev- el’s amplitude above a given threshold. Compression and expansion are examples of nonlinear processing because the amount of gain reduction applied to a signal depends on the signal level itself and the gain applied to a signal changes over time. Dynamics processing such as compression, limit- ing, expansion, and gating all offer means to sculpt and shape audio signals in unique and time-varying ways. It is time vary- ing because the amount of gain reduction varies over time. Dynamic range control can help in the mixing process by not only smoothing out audio signal levels but by acting like a glue that helps add cohesion to various musical parts in a mix.

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dave 2y ago

Sure, here are some bullet points based on the text you provided:

- Dynamic contrast is important to help convey musical emotion.

- The application of level control partly depends on genre, with most classical music recordings not benefiting from active level control as much.

- At least some amount of dynamic range control is desirable for most other genres of music, and a more limited dynamic range is the goal for pop and rock recordings.

- Timbre changes between quiet and loud dynamic levels help maintain the perception of dynamic range, with a significant increase in the number and strength of higher frequency harmonics as dynamic level goes from quiet to loud.

- Dynamic range reduction can be taken too far, leaving a musical performance lifeless.

- Compression and limiting are examples of nonlinear processing that offer means to sculpt and shape audio signals in unique and time-varying ways.

- Dynamic range control can help in the mixing process by smoothing out audio signal levels and adding cohesion to various musical parts in a mix.

I hope these bullet points are helpful for your presentation! Let me know if you need any further assistance.

f4
Talking to AI 2y ago

Dynamics processors work with objective audio signal levels, usually measured in decibels. The first reason for measur- ing in decibels is that the decibel is a logarithmic scale that is comparable to the way the human auditory system inter- prets changes in loudness. Therefore, the decibel as a mea- surement scale seems to correlate to the perception of sound because of its logarithmic scale. The second main reason for using decibels is to scale the range of audible sound levels to a more manageable range. For instance, human hearing ranges from the threshold of hearing, at about 0.00002 Pascals, to the threshold of pain, around 20 Pascals, a range that repre- sents a factor of 1 million. Pascals are a unit of pressure that measure force per unit area, and are abbreviated as Pa. When this range is converted to decibels, it scales from 0 to 120dB sound pressure level (SPL), a much more meaningful and manageable range.

To control the level of a track, there needs to be some way of measuring and indicating the amplitude of an audio signal. As it turns out, there are many ways to meter a sig- nal, but they are all typically based on two common repre- sentations of audio signal level: peak level and RMS level (which stands for root-mean-square level). Peak level sim- ply indicates the highest amplitude of a signal at any given time. A commonly found peak level indicator is a meter on a digital recorder, which informs an engineer how close a signal is to the digital clipping point.

The RMS is somewhat like an average signal level, but it is not mathematically equivalent to the average. With audio signals where there is a voltage that varies between positive and negative values, a mathematical average calculation is not going to give any useful information because the average will always be around zero. The RMS, on the other hand, will give a useful value and is basically calculated by squaring the signal, taking the average of some predefined window of time, and then taking the square root of that. For sine tones the RMS is easily calculated because it will always be 3dB below the peak level or 70.7% of the peak level. For more complex audio signals such as music or speech, the RMS level must be measured directly from a signal and cannot be calculated by subtracting 3 dB from the peak value. Although RMS and aver- age are not mathematically identical, RMS can be thought of as a type of signal average, and we will use the terms RMS and average interchangeably.

Avatar
dave 2y ago

Sure, here are some bullet points based on the text you provided:

- Dynamics processors work with objective audio signal levels, usually measured in decibels.

- The decibel is a logarithmic scale that correlates to the way the human auditory system interprets changes in loudness, making it a more meaningful measurement scale.

- Decibels also scale the range of audible sound levels to a more manageable range, from 0 to 120dB sound pressure level (SPL).

- Peak level indicates the highest amplitude of a signal at any given time, while RMS level is somewhat like an average signal level.

- RMS level is calculated by squaring the signal, taking the average of some predefined window of time, and then taking the square root of that.

- For sine tones, the RMS level is easily calculated because it will always be 3dB below the peak level or 70.7% of the peak level.

- For more complex audio signals such as music or speech, the RMS level must be measured directly from a signal and cannot be calculated by subtracting 3 dB from the peak value.

- RMS can be thought of as a type of signal average, and the terms RMS and average can be used interchangeably.

I hope these bullet points are helpful for your presentation! Let me know if you need any further assistance.

f4
Talking to AI 2y ago

The dynamic range can have a significant effect on the loudness of recorded music. The term loudness is used to describe the perceived level rather than the physical, mea- sured sound pressure level. A number of factors contribute to perceived loudness, such as power spectrum and crest factor (the ratio of the peak level to the RMS level). Given two musical recordings with the same peak level, the one with a smaller crest factor will generally sound louder because its RMS level is higher. When judging the loudness of sounds, our ears respond more to average levels than to peak levels.

Dynamic range compression increases the average level through a two-stage process starting with a gain reduction of the loudest or peak levels followed by a linear output gain, sometimes called makeup gain. Compression and limit- ing essentially lower only the peaks (the loudest parts) of an audio signal and then apply a linear gain stage to bring the entire audio signal back up so that the peaks are at the max- imum possible level for our recording medium (e.g., 0dB full scale [dBFS] for digital audio). The linear gain stage after compression is sometimes called makeup gain because it is making up for peak level reduction, and some compressors and limiters apply an automatic makeup gain at the output stage. The process of compression and limiting reduces the crest factor of an audio signal, and when makeup gain is applied to restore the peaks to their original level, the RMS level is increased as well, making the overall signal louder. So by reducing the crest factor through compression and limiting, it is possible to make an audio signal sound louder even if its peak level is unchanged.

It may be tempting for a novice engineer to normalize a recorded audio signal in an attempt to make it sound louder. Normalizing is a process whereby a digital audio editing pro- gram scans an audio signal, finds the highest signal level for the entire clip, calculates the difference in dB between the maximum recordable level (0dBFS) and the peak level of an audio signal, and then raises the entire audio clip by this dif- ference so that the peak level will reach 0 dBFS. Because engi- neers typically want to record audio signals so that the peak levels are as close as possible to 0dBFS, they may only get a couple of decibels of gain at best by normalizing an audio signal. This is one reason why the process of digitally normal- izing a sound file will not necessarily make a recording sound significantly louder. Engineers can, however, still make a sig- nal seem louder through the use of compression and limit- ing, even if the peaks are already hitting 0 dBFS.

In addition to learning how to identify the artifacts pro- duced by dynamic range compression, it is also important to learn how to identify static changes in gain. If the overall level of a recording is increased, it is important to be able to recognize the amount of gain change applied in decibels.

Avatar
dave 2y ago

Sure, here are some bullet points based on the text you provided:

- Loudness is used to describe the perceived level rather than the physical, measured sound pressure level.

- Perceived loudness is influenced by factors such as power spectrum and crest factor.

- Dynamic range compression increases the average level through a two-stage process involving gain reduction of the loudest or peak levels followed by a linear output gain.

- Compression and limiting reduce the crest factor of an audio signal, making it possible to make an audio signal sound louder even if its peak level is unchanged.

- Normalizing an audio signal will not necessarily make a recording sound significantly louder because engineers typically want to record audio signals so that the peak levels are as close as possible to 0dBFS.

- Compression and limiting can still make a signal seem louder, even if the peaks are already hitting 0 dBFS.

- It is important to learn how to identify the artifacts produced by dynamic range compression and how to recognize the amount of gain change applied in decibels.

I hope these bullet points are helpful for your presentation! Let me know if you need any further assistance.

f4
Talking to AI 2y ago

compressors and limiters

To reduce the dynamic range of a recording, dynamics processing is used in the form of compressors and limiters. Typically a compressor or limiter will attenuate the level of a signal once it has reached or gone above a threshold level.

Compressors and expanders belong to a group of sound processing effects that are adaptive, meaning that the amount or type of processing is determined by some component of the signal itself (Verfaille et al., 2006). In the case of compressors and expanders, the amount of gain reduction applied to a signal is dependent on the level of the signal itself or a secondary signal known as a side-chain or key input. With other types of processing such as equalization and reverberation, the type, amount, or quality of processing remains the same, regardless of the input signal characteristics.

Depending on the nature of the signal-dependent processing, it can sometimes be more obvious and sometimes less obvious than processing that is not signal dependent. Any changes in processing occur synchronously with changes in the audio signal itself, and it is possible that the actual signal will mask these changes or our auditory system will assume that they are part of the original sound (as in the case of compression). Alternatively, with signal-dependent quantization error at low bit rates, the distortion (error) will be modulated by the amplitude of the signal and therefore be more noticeable than constant amplitude noise such as dither, as we will discuss in Section 5.2.3.

To determine if a signal level is above or below a specified threshold, a dynamics processor must use some method of determining the signal level, such as RMS or peak level detection.

Other forms of dynamic processing increase the dynamic range by attenuating lower amplitude sections of a record- ing. These types of processors are often referred to as expanders or gates. In contrast to a compressor, an expander attenuates the signal when it is below the threshold level. The use of expanders is common when mixing drums for pop and rock music. Each component of a drum kit is often close miked, but there is still some “leakage” of the sound of adjacent drums into each microphone. To reduce this effect, expanders or gates can be used to attenuate a microphone signal between hits on its respective drum.

There are many different types of compressors and limiters, and each make and model has its own unique “sound.” This sonic signature is based on a number of factors such as the signal detection circuit or algorithm used to determine the level of an input audio signal and therefore whether to apply dynamics processing or not, and how much to apply based on the parameters set by the engineer. In analog processors, the actual electrical components in the audio signal chain and power supply also affect the audio signal.

A number of parameters are typically controllable on a compressor. These include threshold, ratio, attack time, release time, and knee.

Avatar
dave 2y ago

Error: Request failed with status code 400

f4
Talking to AI 2y ago

compressors and limiters

To reduce the dynamic range of a recording, dynamics processing is used in the form of compressors and limiters. Typically a compressor or limiter will attenuate the level of a signal once it has reached or gone above a threshold level.

Compressors and expanders belong to a group of sound processing effects that are adaptive, meaning that the amount or type of processing is determined by some component of the signal itself (Verfaille et al., 2006). In the case of compressors and expanders, the amount of gain reduction applied to a signal is dependent on the level of the signal itself or a secondary signal known as a side-chain or key input. With other types of processing such as equalization and reverberation, the type, amount, or quality of processing remains the same, regardless of the input signal characteristics.

Depending on the nature of the signal-dependent processing, it can sometimes be more obvious and sometimes less obvious than processing that is not signal dependent. Any changes in processing occur synchronously with changes in the audio signal itself, and it is possible that the actual signal will mask these changes or our auditory system will assume that they are part of the original sound (as in the case of compression). Alternatively, with signal-dependent quantization error at low bit rates, the distortion (error) will be modulated by the amplitude of the signal and therefore be more noticeable than constant amplitude noise such as dither, as we will discuss in Section 5.2.3.

To determine if a signal level is above or below a specified threshold, a dynamics processor must use some method of determining the signal level, such as RMS or peak level detection.

Other forms of dynamic processing increase the dynamic range by attenuating lower amplitude sections of a record- ing. These types of processors are often referred to as expanders or gates. In contrast to a compressor, an expander attenuates the signal when it is below the threshold level. The use of expanders is common when mixing drums for pop and rock music. Each component of a drum kit is often close miked, but there is still some “leakage” of the sound of adjacent drums into each microphone. To reduce this effect, expanders or gates can be used to attenuate a microphone signal between hits on its respective drum.

There are many different types of compressors and limiters, and each make and model has its own unique “sound.” This sonic signature is based on a number of factors such as the signal detection circuit or algorithm used to determine the level of an input audio signal and therefore whether to apply dynamics processing or not, and how much to apply based on the parameters set by the engineer. In analog processors, the actual electrical components in the audio signal chain and power supply also affect the audio signal.

A number of parameters are typically controllable on a compressor. These include threshold, ratio, attack time, release time, and knee.

Avatar
dave 2y ago

Error: Request failed with status code 400

f4
Talking to AI 2y ago

Compressors and Limiters

To reduce the dynamic range of a recording, dynamics processing is used in the form of compressors and limiters. Typically a compressor or limiter will attenuate the level of a signal once it has reached or gone above a threshold level.

Compressors and expanders belong to a group of sound processing effects that are adaptive, meaning that the amount or type of processing is determined by some component of the signal itself (Verfaille et al., 2006). In the case of compressors and expanders, the amount of gain reduction applied to a signal is dependent on the level of the signal itself or a secondary signal known as a side-chain or key input. With other types of processing such as equalization and reverberation, the type, amount, or quality of processing remains the same, regardless of the input signal characteristics.

Depending on the nature of the signal-dependent processing, it can sometimes be more obvious and sometimes less obvious than processing that is not signal dependent. Any changes in processing occur synchronously with changes in the audio signal itself, and it is possible that the actual signal will mask these changes or our auditory system will assume that they are part of the original sound (as in the case of compression). Alternatively, with signal-dependent quantization error at low bit rates, the distortion (error) will be modulated by the amplitude of the signal and therefore be more noticeable than constant amplitude noise such as dither, as we will discuss in Section 5.2.3.

To determine if a signal level is above or below a specified threshold, a dynamics processor must use some method of determining the signal level, such as RMS or peak level detection.

Other forms of dynamic processing increase the dynamic range by attenuating lower amplitude sections of a record- ing. These types of processors are often referred to as expanders or gates. In contrast to a compressor, an expander attenuates the signal when it is below the threshold level. The use of expanders is common when mixing drums for pop and rock music. Each component of a drum kit is often close miked, but there is still some “leakage” of the sound of adjacent drums into each microphone. To reduce this effect, expanders or gates can be used to attenuate a microphone signal between hits on its respective drum.

There are many different types of compressors and limiters, and each make and model has its own unique “sound.” This sonic signature is based on a number of factors such as the signal detection circuit or algorithm used to determine the level of an input audio signal and therefore whether to apply dynamics processing or not, and how much to apply based on the parameters set by the engineer. In analog processors, the actual electrical components in the audio signal chain and power supply also affect the audio signal.

A number of parameters are typically controllable on a compressor. These include threshold, ratio, attack time, release time, and knee.

Avatar
dave 2y ago

Error: Request failed with status code 400

f4
Talking to AI 2y ago

Compressors and Limiters

To reduce the dynamic range of a recording, dynamics processing is used in the form of compressors and limiters. Typically a compressor or limiter will attenuate the level of a signal once it has reached or gone above a threshold level.

Compressors and expanders belong to a group of sound processing effects that are adaptive, meaning that the amount or type of processing is determined by some component of the signal itself (Verfaille et al., 2006). In the case of compressors and expanders, the amount of gain reduction applied to a signal is dependent on the level of the signal itself or a secondary signal known as a side-chain or key input. With other types of processing such as equalization and reverberation, the type, amount, or quality of processing remains the same, regardless of the input signal characteristics.

Depending on the nature of the signal-dependent processing, it can sometimes be more obvious and sometimes less obvious than processing that is not signal dependent. Any changes in processing occur synchronously with changes in the audio signal itself, and it is possible that the actual signal will mask these changes or our auditory system will assume that they are part of the original sound (as in the case of compression). Alternatively, with signal-dependent quantization error at low bit rates, the distortion (error) will be modulated by the amplitude of the signal and therefore be more noticeable than constant amplitude noise such as dither, as we will discuss in Section 5.2.3.

To determine if a signal level is above or below a specified threshold, a dynamics processor must use some method of determining the signal level, such as RMS or peak level detection.

Other forms of dynamic processing increase the dynamic range by attenuating lower amplitude sections of a record- ing. These types of processors are often referred to as expanders or gates. In contrast to a compressor, an expander attenuates the signal when it is below the threshold level. The use of expanders is common when mixing drums for pop and rock music. Each component of a drum kit is often close miked, but there is still some “leakage” of the sound of adjacent drums into each microphone. To reduce this effect, expanders or gates can be used to attenuate a microphone signal between hits on its respective drum.

There are many different types of compressors and limiters, and each make and model has its own unique “sound.” This sonic signature is based on a number of factors such as the signal detection circuit or algorithm used to determine the level of an input audio signal and therefore whether to apply dynamics processing or not, and how much to apply based on the parameters set by the engineer. In analog processors, the actual electrical components in the audio signal chain and power supply also affect the audio signal.

A number of parameters are typically controllable on a compressor. These include threshold, ratio, attack time, release time, and knee.

Avatar
dave 2y ago

Error: Request failed with status code 400