Audio Dynamic Range Compression: Limiting and Expansion

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Audio Dynamic Range Compression: Limiting and Expansion

Audio Compression: What It Is and How It Works

As an audio engineer, I’ve spent countless hours working with dynamic range compression. At its core, compression is a tool used to control the dynamic range of an audio signal. This means that it can be used to make quiet sounds louder and loud sounds quieter, resulting in a more consistent overall volume. Compression is an essential tool in the world of audio production, and it’s used in everything from music to film and television.
One of my favorite quotes about compression comes from the legendary producer Quincy Jones. He once said, “Compression is like a good pair of eyeglasses. You don’t notice them, but they make things look better.” And it’s true. When used correctly, compression can make an audio signal sound more polished and professional.

Dynamic Range Compression vs. Limiting

While compression and limiting are often used interchangeably, they are actually two different processes. Compression is used to control the dynamic range of an audio signal, while limiting is used to prevent the signal from exceeding a certain level. In other words, limiting is a more extreme form of compression.
One of the most common uses of limiting is in mastering, which is the final step in the audio production process. During mastering, the goal is to make sure that the audio sounds consistent across different playback systems. Limiting is used to prevent the audio from distorting or clipping when played back on a variety of systems.

Audio Expansion: The Other Side of the Coin

While compression and limiting are used to control the dynamic range of an audio signal, expansion is used to increase it. Expansion is the opposite of compression, and it’s used to make quiet sounds even quieter. This can be useful in situations where you want to bring out the details in a recording.
One of my favorite examples of the use of expansion comes from the film “No Country for Old Men.” In the film, there’s a scene where the main character is walking through a hotel room. As he walks, you can hear the sound of his footsteps echoing off the walls. The sound of the footsteps is very quiet, but it’s still audible. This is a great example of the use of expansion to bring out the details in a recording.
Final Words:
In conclusion, dynamic range compression, limiting, and expansion are all essential tools in the world of audio production. Whether you’re working on music, film, or television, understanding how to use these tools can make a huge difference in the final product. And while there are many different techniques and approaches to using compression, limiting, and expansion, the most important thing is to use your ears and trust your instincts.
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What is Audio Normalization?

Audio normalization is the process of adjusting the volume of an audio file to a desired level without changing its dynamic range, unlike compression that changes volume over time in varying amounts. There are two main reasons to normalize audio: getting the maximum volume and matching volumes. The first reason is when you have a quiet audio file and you want to make it as loud as possible (0 dBFS) without changing its dynamic range, and the second reason is when you have a group of audio files at different volumes, and you want to make them all as close as possible to the same volume.

Peak volume detection is the method of measuring the volume of audio that only considers how loud the peaks of the waveform are for deciding the overall volume of the file. This is the best method if you want to make the audio as loud as possible. RMS volume detection considers the overall loudness of a file, and it takes an average and calls that the volume. This method is closer to how the human ear works and will create more natural results across varying audio files.

The new standard in broadcast audio, EBU R 128 volume detection, is similar to RMS but can be thought of as emulating a human ear. It listens to the volume intelligently and thinks about how we will hear it. It understands that we hear frequencies between 1000 – 6000 Hz as louder and takes that into account.

Normalization can be performed in an audio editor or inside a DAW, but it is a destructive process that can change the sound quality of the file. This was a bigger issue when digital files were all stored as 16 bit. If you turned the volume down, you effectively reduced the bit depth. Your CD-quality 16-bit file could end up 12-bit or less, even if you turned it up with peak normalization. Nowadays, audio editing software works internally at a much higher bit depth, often 32-bit floating point, which means that calculations are done more accurately and affect the sound quality far less. To take advantage of the high quality of high bit depth inside audio editing software, it is essential to keep the file at the higher resolution once it has been processed. Finally, peak normalization to 0 dBFS is a bad idea for any parts to be used in a multi-track recording, as it may overload DAW or plugins.

What is RMS?

RMS stands for Root Mean Square and is a measure of the average power of a signal. It’s commonly used in electrical engineering and other fields that deal with signals, such as audio processing.

To calculate the RMS value of a signal, you first square each value in the signal and then take the average of all the squared values. Finally, you take the square root of that average. Mathematically, it can be expressed as:

RMS = sqrt((1/N) * sum(x^2))

Where N is the number of samples in the signal and x is the value of each sample.

The resulting RMS value represents the equivalent DC voltage that would produce the same amount of heat in a resistor as the original AC signal. In other words, it’s a measure of the signal’s power level.

RMS is particularly useful when dealing with signals that have both positive and negative values, as it takes into account the magnitude of both. It’s also commonly used to specify the power of audio signals, such as in the specification of the power output of an amplifier.

Overall, RMS is a useful tool for understanding the power level of signals and can help in the design and analysis of electrical and audio systems.

What is Bit Depht?

Bit depth refers to the number of bits used to represent the amplitude of an audio signal. In digital audio, the amplitude is quantized into a finite number of levels, which are then represented by binary numbers. The bit depth determines the number of possible levels, and therefore, the resolution of the digital signal.

For example, with a bit depth of 16 bits, there are 2^16, or 65,536 possible levels. With a bit depth of 24 bits, there are 2^24, or 16,777,216 possible levels. This means that a higher bit depth provides a more accurate representation of the original analog signal.

The bit depth of an audio signal affects its dynamic range and signal-to-noise ratio. Dynamic range refers to the difference between the loudest and softest parts of the signal, while signal-to-noise ratio refers to the ratio of the signal to any background noise present.

With a higher bit depth, the dynamic range is increased, allowing for a greater difference between the loudest and softest parts of the signal to be accurately represented. Similarly, a higher bit depth also increases the signal-to-noise ratio, since there are more levels available to represent the signal and less quantization noise is introduced.

However, a higher bit depth also requires a larger data rate and storage space, and may not be necessary for all types of audio signals. For example, speech and other types of less complex signals may not require a high bit depth, while music with a wide dynamic range and complex sounds may benefit from a higher bit depth.

In summary, the bit depth of an audio signal determines the resolution of the digital signal and affects the dynamic range and signal-to-noise ratio. A higher bit depth provides a more accurate representation of the original analog signal, but also requires a larger data rate and storage space. The appropriate bit depth for a given audio signal depends on the complexity of the signal and the desired quality.

Dynamic compression

Dynamic compression

dynamic compression

Dynamic Compression (DRC): narrowing (or expanding in the case of an expander) the dynamic range of a soundtrack. Dynamic range is the difference between the quietest and loudest sound. Sometimes the lowest sound on the soundtrack will be a little louder than the noise level, and sometimes a little lower than the loudest. The hardware devices and programs that perform dynamic compression are called compressors, distinguishing four main groups among them: the compressors themselves, limiters, expanders, and gates.

dynamic compression

Up and down compression

Creep Compression lowers the volume when it exceeds a certain threshold, leaving quieter sounds unchanged. The extreme downward compression option is the limiter. Increase compression (upward compression), conversely, increase volume if it is below a threshold value without affecting louder sounds. In this case, both types of compression reduce the dynamic range of the audio signal.

Expander and gate

If the compressor decreases the dynamic range, the expander increases it. When the signal level rises above the threshold level, the expander raises it further, thus increasing the difference between high and low sounds. These devices are often used when recording drums to separate the sounds of some drums from others.

A type of expander that is used not to amplify loud sounds, but to drown out quiet sounds that do not exceed the threshold level (for example, background noise) is called a noise gate. In such a device, as soon as the sound level falls below the threshold, the signal flow stops. Normally, the door is used to suppress noise during breaks. In some models, you can make sure that the sound does not stop abruptly when the threshold level is reached, but gradually fades away. In this case, the decay rate is set with the Decay knob.

The gate, like other types of compressors, can be frequency dependent (that is, handle certain frequency bands differently) and can operate in side chain mode (see below).

Compressor working principle

The signal entering the compressor is divided into two copies. One copy is sent to an amplifier, in which the degree of amplification is controlled by an external signal, the second copy forms this signal. It goes into a device called a side chain, where the signal is measured and based on this data, an envelope is created that describes the change in its volume.
This is how most modern compressors are organized, this is the so-called feed-forward type. In older devices (feedback type), the signal level is measured after the amplifier.

There are several analog variable gain amplification technologies, each with its own advantages and disadvantages: tube, optics with photoresistors, and transistor. When working with digital sound (in a sound editor or DAW), you can use your own mathematical algorithms or emulate the work of analog technologies.

Basic parameters of compressors
Limit
The compressor reduces the level of the audio signal if its amplitude exceeds a certain threshold value. Typically specified in decibels, a lower threshold (eg -60 dB) means that more sound will be processed than a higher threshold (eg -5 dB).

Proportion
The amount of level reduction is determined by the ratio parameter: the 4: 1 ratio means that if the input level is 4 dB higher than the threshold, the output signal level will be 1 dB higher than the threshold .
For example:
Threshold = −10 dB
Input signal = −6 dB (4 dB above threshold level)
Output signal = −9 dB (1 dB above threshold level)

It is important to note that the suppression of the signal level continues for some time after it falls below the threshold level, and this time is determined by the value of the release parameter.

Compression with a maximum ratio of ∞: 1 is called limiting. This means that any signal above the threshold level is suppressed down to the threshold level (except for a short period after a sudden increase in input volume). For more details, see Limiter below.

Attack and release
The compressor provides some control over how quickly it responds to changes in signal dynamics. The Attack parameter determines the time it takes for the compressor to reduce the gain to the level determined by the Ratio parameter.

Dynamic range normalizer

Dynamic range normalizer

Dynamics Processing

The audio quality of compressed formats such as MP3 or AAC is often perceived as better or at least equivalent to so-called lossless formats or audio CDs even by trained listeners. This is the conclusion reached by the TrustedReview test platform in a blind test with several people.

TrustedReview points out that this is not an experiment with strictly scientific parameters. However, what was remarkable about the results was that the self-reported hi-fi enthusiasts among the test subjects were wrong to recognize the uncompressed piece of music.

Dynamic Compression

Already sophisticated coding

“I am not surprised by this result, because we reached a similar conclusion in our internal experiments. Encoding technologies are now so sophisticated that most people no longer hear any difference even when compressing at a lower data rate,” he explains. Bernhard, MP3 expert at Fraunhofer. Grill in conversation with press release.

“It would even go so far that a properly encoded 192 kbit / s AAC sound file is the best alternative to today’s audio CD, as the 16-bit resolution of the CD can be far exceeded.” For people who are particularly sensitive to audio and want to be absolutely sure, Grill recommends a data rate of 256 kbit / s for AAC and 320 kbit / s for MP3.

The development trend goes in two directions

According to Grill, the current trend in audio encoding is bi-directional. For one thing, it’s about getting even smaller files with low-quality compromises. “This currently plays an important role, especially with streaming services over cellular networks, as bandwidths are still low despite UMTS and retrieval of large amounts of data is still expensive in most countries. “explains Grill. The most promising encoding is the HE – AAC audio codec, also known as AAC +, which is a further development of the AAC standard used by iTunes, for example.

On the other hand, the developers are also targeting the best of music listeners and want to show that with the right encoding technology, even the last noticeable differences from Audio CD can be eliminated. “Meanwhile, record labels and sound engineers have also understood how to encode for excellent sound quality. Another advantage of existing encodings, such as MP3 or AAC, is that the processes behind them are constantly being improved,” he says. Grill. MP3 encoded at 128 kbit / s sounds much better today than 15 years ago.

The term “compression” is misleading

The Fraunhofer MP3 and AAC developers find the term “compression” misleading anyway. “This should not be confused with dynamic compression, that is, the weakening of noisy passages and the elevation of soft passages, which has always been a technological necessity, especially with famous records and in FM broadcasting.” In this regard, MP3 and AAC with dynamic range up to 24 bits are even better than audio CD, which is limited to 16 bits. This restriction is not removed with a lossless copy of an audio CD.

Mp4Gain offers a dynamic range normalize, that is, the weakening of the noisy passages and the elevation of the soft passages, which is the most modern.

Much more important for good sound quality or an attractive sound image is not the medium or file format used, but the speakers used. “These are still the worst adulterers in the entire chain,” says Grill.

Dynamic Range and Bit Depth, those great strangers

Many times we listen to songs and they seem like absolute songs and other times they seem like a waste of time. To some extent it is because of the music that is heard. But today we are going a little further. I dare say that part of the fact that we end up liking it is due to the fact that there are two concepts that have not been taken into account in production lately.

I mean the Dynamic Range.

What is it? Well, it’s just the difference between the highest possible peaks and the noise threshold. Yes, there is always a noise basis, even in the best soundproof studio. And how does that affect what you hear is a great song? Very easy, since this variation in amplitude influences the sound intensity, how the song flows, the difference between strong and weak. The specific nominal amplitude of each equipment must also be taken into account, which is called the dynamic ceiling or Headroom. Normally it stays at 120Db, since it is the limit that integrated circuits allow. What we hear from all this is the SNR, signal noise ratio, that is, the relationship between residual noise and the average signal of the subject.

If we transfer this to our tasks as musicians, it means that no frequency that we are using within our theme should be fixed, which is called average, and that too much compression is bad too, since it can crush the sound, since applying a compressor decrease dynamic differences. This also includes mixing and normalization (which we will talk about later) where some resources can help us to make our theme cover the widest dynamic range in all its frequencies. For example: If the amplitude of the mid frequencies is reduced, approximately the frequencies of 250Hz, it is possible to boost frequencies of 5KHz.

Another resource that sometimes we fail when we are playing a synthesizer, whether analog or digital, or when we are mixing is Bit Depth or Bit Resolution / Quantization.

Bit Depth is the amplitude of variations of a sound wave. Translated into Christian, the number of bits recorded per sample. This affects the previously explained concept, dynamic range, since the greater the bit depth, the greater the possibility of working with wide dynamic ranges, which is called the rule-of-thumb. Since for example, if we work with 8-bit synthesizers, which are so fashionable, we are saying that this synthesizer, the wave that sends the output, only allows to register 256 variations in the amplitude of the wave. Let’s remember that 8 bits = 2 to 8. With this I don’t mean that an 8-bit synthesizer is bad, each one does its job, but there are many cases where pears are asked from the elm, when we all know that you have to go to the pear tree. I would like to emphasize this concept especially for we use analog synthesizers, drum machines, samplers. Since the bit depth within a PCM, the digital representation of an analog sound, recreates the maximum dynamic range, which means that the sound will be more real.

So let us return to the question asked above. How does this affect my theme to sound good? Well the bit depth limits the amount of dynamic range and signal noise ratio. This means that if we cover a higher bit resolution we will be able to interact with more precision within the theme that we are composing and therefore improve the output signal of the set.

Even so, I am sorry to be this party pooper, the bit depth problem comes from the fact that nowadays we use a format that has very low and bad compression, such as MP3 or WMA. Although you export at 24 bits, it does not mean that the MP3 / WMA has a bit depth of 24 bits, since you have mastered them at 24 bits but then it has been compressed and that is where it fails. This also happens even if we copy directly to a CD, since CDs are burned at 16 bits.