Should You Use Compression In Audio Recording?

A very common question I see is “should I use compression in audio recording?” The best answer to this question is, as is so often the case, “Do you need compression in your recordings?” Let’s start with a workable definition of compression in the audio recording sense. When you lower the volume of only SOME of your audio, usually the bits that are clearly louder than most of the rest of the audio you’re working with, you are “compressing” that audio. This is usually done in order to allow you to raise the average loudness of the entire audio file.

Now, this can be done manually, by which I mean you could open your audio in an editor, seek out all the areas where the wave forms (I like to use the term “blobs” instead) are loudest, then turn those bits down. But that can get REALLY tedious and time consuming. So to automate this process, a machine (nowadays done with software) called a “compressor” was invented. This allowed folks who really knew what they were doing to more quickly manipulate volume and loudness dynamics. The dark side of the situation, though, was that it allowed folks who were less expert to mess up their audio, and do it much faster and more efficiently than ever.

There are all kinds of settings on a compressor that are better discussed in other articles. For the moment I’d like to focus on an explanation of basic compression, since I strongly believe that those who use compressors to mess up their audio (usually without actually WANTING to mess up their audio) do it because they don’t have a really good grasp on what compression really is. This should help.

Let’s say you have a voice narration file open in your audio editor. You’ve got your familiar blobs going horizontally across the screen (in its “swim lane”). Now imagine that at two points in the audio there are very loud and quick peaks, perhaps caused by a cough or an overly-excited consonant. In this example, most of the blobs top out about half way between the center line (total silence, remember?) and the top of the swim lane. However, the two loud peaks I mentioned go most of the way up to the top. If I tried to turn up the volume of this audio, all of the audio, including the 2 peaks, would “get bigger.” But there is a problem here. Do you see it? If any of the audio wave forms (blobs) get pushed beyond the top border line, you get nasty, awful digital distortion. Suffice it to say that the border line represents the upper boundary you must not cross.

Knowing that, it should be easier to see why you can’t turn the loudness up very far before the peaks hit that boundary. And when that happens, none of the rest of the audio can get any louder either. Oh no, whatever shall we do?

How about if we turn DOWN the level of JUST those two peaks? Well, the first thing that would happen is that the average level of the audio blobs would be much more even. THAT is what compression is for. Now that we don’t have just a couple of pesky peaks preventing us from turning our audio up without it distorting, we can raise the average level much higher, resulting in louder audio across the entire file. If we now tell our editor to raise the level of the loudest peak to maximum loudness (raising the rest of the audio by the same amount), you will notice that blobs are much wider/bigger across the board. Turning down only the few offending bits of audio was the act of compressing the audio file. Once that was done, and we were left with more even levels, we were able to turn the entire thing up without distortion, which we couldn’t do before.

Being able to increase the overall loudness level of your audio is just one benefit of compression, and perhaps its most common goal. when used wisely, this can also add punch and “up-frontness” to your audio. But don’t forget what compression actually accomplished for us BEFORE we were able to turn it up. It evened out the overall levels first, which gives a more consistent listening level to the user.

I’ve often wished I had an audio compressor attached to my television for this very reason. Have you ever been watching a movie where the action scenes were so loud that you had to turn the volume down on the TV, only to find that now the talking parts are too quiet? Then you have to turn the TV back up to hear those parts. You end up turning the volume up and down throughout the show.

What about when the commercials on TV are much louder than the show itself? Don’t you wish you had a compressor to even out the volume so that it automatically turned the commercials down to the same level as your TV show?

I mentioned earlier that people frequently mess up their audio by over-use or incorrect use of compression. The most common problem is with music. If you compress it too much, say, in order to make your mix louder than everyone else’s, you risk sucking out the dynamic range of the music. Changes in levels are important to the emotion of music. Over-compressing can flatten things so much that there is no more emotional flow to the music. That can also make it sound unnatural. Also, compressors tend to impart certain sonic strangeness to audio when over-done, such as increasing sibilance (the sort of hissy sounds you hear when someone uses the letter “S”) in vocals, or audio “pumping” which is when the compressor is rapidly clamping down on loud audio and then letting up on quiet parts over and over.

So back to the original question. “Should I use compression in audio recording?” Now that you understand what compression is, you can answer your own question. Do you want to even out the loudness of your audio? Then compression can help you. Do you want to raise the overall loudness of your audio without distorting? Compression can help you. Do you want to add some punch and ‘in-your-faceness” to a voice over? Compression can help. Just make sure you remember that it is really easy to overdo compression if you’re not careful.

Now go forth and squash your audio responsibly.

All About DVD Part 4: DVD-Audio and SACD

New sound formats: DVD-Audio and SACD

When Audio-CD format emerged, most music producers and music lovers were absolutely sure that Audio CD provides the best quality sound possible and nothing else will ever be required. However, some audiophiles instantly refused to accept new Audio CD and reverted back to old vinyl disks. At first no-one took them seriously, however, as the time passed an increasing number of listeners started to notice, especially when comparing CD albums with the same records on vinyl, that CDs are not capable to transmit full range of sound. Eventually it becomes clear that there are scientific reasons behind this, namely dynamic range and sampling rate.

Dynamic range is the difference between the loudest and softest sounds in audio record. Sound data is recorded on CD in 16-bit PCM format and therefore its dynamical range does not exceed 96 decibels. And the upper range for human ear is 120Db. The difference is obvious. While the lack of dynamic range might be hard to notice for the pop music, but for something like symphony orchestra sound engineers often have a problem with too soft or too loud instruments which are beyond dynamical range and therefore lost in record.

Apart from dynamic range it also became clear that Audio CDs have insufficient sampling rate. Digital record consists of small pieces called samples, each of them contains information about sound in particular point in time. The more samples are per unit of time, the higher is the quality of record. Basically, sample is a digital analogue of sound oscillation and for that reason they are measured by using the same unit, hertz (Hz). The maximum frequency of sound that humans can hear is around 20 thousands hertz (20 kHz). To reproduce sound accurately, the sound information have to be digitized with sampling rate higher than a human can hear.

During Audio-CD development it was believed that doubling this rate – up to 44.1 kHz would be sufficient. But for some records part of sound was lost and turned into noise. Especially often that happened for classical music and jazz, with high-frequency instruments such as violin and flute being the most unlucky ones. It is scientifically proven now that for the most accurate sound the sampling rate has to be not lower than 64 kHz. This, of course, is far beyond Audio CD capabilities.

To address these issues the DVD-Audio format was developed. It took more time to approve than DVD-Video, as DVD-Forum had to justify between two alternative technologies: one from Toshiba and the other from Sony-Philips (they also developed Audio CD) alliance. The problem was to decide, which format was better. Both of them successfully solved the drawbacks of Audio CD.

Toshiba simply improved existing PCM standard and turned it into 24-bit one. The dynamic range grew to 144db and sampling rate soared up to 192 kHz. These values exceed human ear capabilities by a good margin. This format also had support for multi-channel sound.

Sony and Philips took another route and invented new single-bit audio format called DSD (Direct Stream Digital). The idea was to read data from disk in significantly smaller quantities (one bit at a time, compared to 16 bits for Audio CD and 20-24 bits for DVD-Audio) but with insane sampling rate – 2.8224 MHz. Similar principle is actually employed in the vinyl audio pickup. Due to this the dynamical range for new format grew to 120db. Apart from that multi-channel support was added along with compatibility with Audio CD. The latter was achieved by adding second layer with sound in Audio CD format. If such disk is inserted in generic Audio-CD player, it will be played just like any Audio CD, with all limitations of that format.

At the end, in 1999 Toshiba finally prevailed. This was due to several reasons. First of all, DSD soundtrack is hard to make even in studio environment. Second, when this technology first appeared, there was no special equipment that could support it. All existing devices were able to work with PCM sound only, and DSD required conversion. This eventually reduced the benefits of DSD to zero. Toshiba, however, selected DVD as a medium for the new format, which made it possible to create audio disks in this format even at home. The situation with Sony-Philips creation was completely different. It had to be recorded on special disks, which were similar in appearance and sizes to normal DVD, but totally different otherwise. These disks also required special equipment for printing and recording. All of these were the reasons why Toshiba received rights for the DVD-Audio brand-name.

But that didn’t stop Sony and Philips and they decided to introduce their own format – regardless of DVD-Forum decision. The new format was called Super Audio Compact Disc. As a result, there was no single standard and neither SACD nor DVD-Audio managed to even get close to ordinary Audio-CD sales volumes.

The difference in sound between SACD and DVD-Audio wasn’t spotted even by most “advanced” audiophiles.

Let’s examine each of the technologies in details: DVD-Audio

Since DVD-Audio standard is based on ordinary DVD, such disk can be double-sided or double-layered or a combination of these two. Unlike SACD, a sound can be recorded on DVD-Audio in different quality and in various formats, stereo or multi-channel. Sampling rate for DVD-Audio can vary from 44.1 kHz up to 192 kHz (however sound can only be stereo at 192 kHz, as it is simply not enough space for multi-channel sound of that quality). Generally, DVD-Audio is able to deliver 1000 times more accurate sound when compared with Audio CD!

To be able to fit multi-channel uncompressed sound (which usually takes lots of space) developers have used special lossless compression technology called MLP (Meridian Lossless Packing), which in some ways is similar to PC archiving packages such as Zip or RAR.

In order to be compatible with generic DVD players, DVD-Audio disks can contain copy of the soundtrack encoded in Dolby Digital and DTS on the other side of the disk. Such disks are known as Hybrid DVD-Audio disc – HDAD. Sometimes it is possible to encounter disks that have only Dolby Digital or DTS encoded soundtracks, however, such disks hardly have anything to do with DVD-Audio.

Apart from this, several sound-recording companies have developed in 2004 a new format called DualDisc. Essentially it is a double sided disc, one side of which is single- or double layer DVD and the other side is plain CD. The CD format used in such disk is slightly different form normal CD. Standard thickness of CD or DVD disk is 1.2mm. Audio CD layer occupies all that space, and DVD requires only half of it, or 0.6mm. Therefore it is not possible to create a disk with both proper DVD and CD layers as it would be 2mm thick and won’t fit in any drive. Due to this reason the developers decided to cheat a little bit and reduced thickness of Audio CD layer to 0.9mm. As a result, it is possible to play DualDisc on most players, but it is still too thick and doesn’t fit in most cars CD. In addition to this it can damage some Hi-End equipment, and will also void the warranty. Another interesting effect of making Audio CD layer thinner is the inability to record anything to it via normal CD recording methods due to spherical aberrations. This means that laser is not able to focus on data pits and “see” only blurred image which cannot provide any actual data. The problem was solved my making pits bigger, however their amount reduced. As a result, the capacity of disc has dropped and instead of standard 74 minutes such disks contain only 60, and there are only 525 MB available instead of 650.

Most probably DualDisc technology won’t last long. There actually is a new technology to replace it – Onedisc/DVDplus. German engineers managed to create hybrid medium with standard thickness – 1.2mm. More importantly, the capacity of CD side doesn’t suffer; it still is able to contain 650mb of data or 74 minutes of audio. The developers of this standard, however, do not share their secrets, and how exactly did they managed to overcome obvious problems is still a mystery. Many recording studios have purchased license for this technology and Onedisc will probably soon appear on shop shelves.

Beside audio data, DVD-Audio disc is able to hold any type of DVD-Video data. It can be video, menu, slides, etc (it is also possible to put sound in DVD-Audio format on DVD-Video disk). The standard also provides new technology for storing text information, some sort of replacement for CD-Text. Disk can contain titles, lyrics, information about performers, etc. Text can be viewed on TV or audio player screen. Just as DVD-Video, DVD-Audio can provide a selection of several languages for text info.

DVD-Audio copy protection is also worth mentioning. After famous DVD-Video hack manufacturers decided to postpone DVD-Audio until better protection becomes available. This better protection was CPPM (Content Protection for Pre-recorded Media), which include several different protection measures:

1. According to the new standard, DVD-Audio data is encrypted. Unique set of keys can only be kept in legal player device (that is, DVD-Audio player that comply with CPPM requirements). In case if keys do not match the disk (for example when DVD-Audio player is manufactured without appropriate license), it will not be possible to decrypt the data and DVD-Audio disk simply won’t play.

Theoretically, it is possible to copy such keys by hacking the device, but practically it is rather useless as if stolen keys would be revealed, all later releases of DVD-Audio disks simply won’t support them. And most manufacturers perform monthly revision of keys.

2. Contents of DVD-Audio are also bound to the physical medium. This is achieved by recording “digital watermarks” on the disk. This hidden information contains disk details and the information about how many times this disk can be copied. Watermarks are recorded with intervals of several seconds and if the player discovers that watermarks doesn’t match medium it instantly stops playback.

Apart from this, if DVD-Audio player is equipped with Firewire (iLink) digital out, it won’t be possible to use it for illegal copying too. All outgoing information is encrypted and because of this it only will be possible to connect the cable to an amplifier or receiver that supports encryption. It will not be possible to connect player to the PC and therefore won’t be possible to copy digital output.

SACD disks come in three flavors – single layer, double-layer and hybrid.

In case of SACD hybrid disks have two layers. Bottom layer has Audio CD data and upper semi-transparent layer contains SACD data. If such disk is inserted in normal CD player, there will be no problems with playback – the player simply won’t notice the upper layer.

Single- and double-layered disks contain only SACD data, also known as HD (high definition). Each HD layer takes the same space as standard DVD layer – 4.7 GB. Sound can be recorded in mono, stereo or multi-channel formats. Some SACD player can read only mono or stereo disks, and it will take a special device to read multi-channel disks. In most cases, however, there are both multi-channel and stereo versions present. Each type of soundtrack has its own space on the disk – stereo data is recorded closer to the center of the disk, multi-channel data comes next and additional data (such as photos or lyrics) is recorded close to the edge. Each sound zone contains at least 74 minutes of sound. There are, however, purely multi-channel disks with several hours of audio on them.

In ordered to reduce size of data, SACD developers had used DST (Direct Stream Transfer) compression. This compression is similar to MPL and resembles PC archiving utility. It operates by compressing repeating sequences of bits and can achieve significant reduction in size without any losses.

Just as DVD-Audio does, SACD has complex set of copy protection measures:

1. SACD disks are not compatible with computer DVD drives, which makes illegal copying at home impossible.

2. Disk contents are inaccessible until special lead-in SACD-Mark is read. This label is located in a hidden area of the disk and contains information essential for the playback. Only certified devices can read and decrypt it.

3. Disk contents are also protected by PSP-PDM (Pit Signal Processing-Physical Disc Mark) watermarks. By changing pits (tracks) width it is possible to write additional info on the disk (for example, key from encrypted soundtrack). These watermarks can be reproduced only on special licensed equipment. Without them the playback will stop after only few seconds. Besides, it is possible to create a visible picture on working surface of the disk – for example, company logo, thus the original disk can be recognized even with naked eye. To avoid compatibility issues, the second layer in SACD is protected as much as any normal Audio CD is.

For audio output mainly analog out is used, but encrypted digital out via Firewire (iLink) is also possible, just as in DVD-Audio. Coaxial and optics outputs are not supported at all.


Most probably, both formats will still remain an expensive technology for true music fans and audiophiles. Majority of consumers already made their choice in favor of medium-independent music format – MP3. So it is highly possible that DVD-Audio and SACD will remain the top audio quality available, as any higher quality doesn’t have much sense and will be too expensive to reach.

Audio Interface or Sound Card in Your Home Studio?

In this article you will learn:

  • What a Sound Card/Audio interface does
  • Why you need a Sound Card/Audio interface
  • The differences between a Sound Card and an Audio Interface
  • The internal pieces of a Sound Card/Audio Interface
  • The Connection Types

What is an Audio Interface/Sound Card and what does it do?

A sound card receives audio signals and converts them into digital audio.

A sound card is synonymous in function to an audio interface.

The conventional Sound Card is a chip that is installed into your computers PCI slot.

An Audio Interface does the same thing. It converts input audio signals. It is just in the form of a hardware interface that connects to your DAW computer. An audio interface is an external device that receives an analog signal, and sends it to your music software application in its digital form.

For example; by plugging a microphone into an audio interface with a compatible audio sequencer, an audio interface can convert the analog microphone signal and record a digital audio file onto a track. This can be done with a sound card as well.

Why You Need a Sound Card/Audio Interface?

Music production and intensive audio processing requires more than your stock SoundCard can typically handle. Simple as that.

See, when an audio signal is recorded from your microphone and onto the hard drive of your computer, it goes through a process of conversion from an analog signal into a stream of binary code, which is the digital “representation” or “translation” of that original signal.

The main problem is what is known as latency. Latency occurs when the time it takes for conversion, and the output of the recorded track, along with any effects or signal processing that happens anywhere in between, is delayed. There is a lag, and you hear it late. Thus, “LATE”-ncy.

Clicks, pops, error messages, and other artifacts can result with a cheap Sound Card, or improperly optimizing the settings for your recording platform.

The Differences Between a Sound Card and an Audio Interface

They both have virtually the same function. The difference-primarily lies in the hardware itself. A Sound Card is a “card” that gets internally installed into the back your computer through a PCI slot, while an audio interface is an external piece of hardware that can sit on your desk and offer you the convenience of not having to reach around to the back of your computer to plug stuff in and adjust things.

The audio interface typically has a “breakout box” for all your inputs, as well a preamp, which converts a mic level signal into a line level signal.

The Internal Pieces of a Sound Card/Audio Interface

As described above, the core component of a Sound Card/Audio Interface is the digital audio converters.

The other important piece is the software drivers which manage the “code” of data flow and thus play a critical role in the overall effectiveness of your sound card.

The other piece that can be included with audio interfaces is onboard preamps. Preamps can be the most expensive part of an audio interface, and some don’t have them.

Sound Card and Audio Interface Connection Options:

  • Fire-wire: Speed
  • USB: Plug and play quick
  • PCI: More tracks and no need for attach/unattachment, because it is installed.(Some high-end studios use state of the art HD Sound Cards that are capable of the highest possible sampling rate and bit depth.)

In most cases they all produce similar sound quality, (with exception to the pro HD card) but offer different advantages with each connection option.

There are two components within both of these devices which factor into making a unit – produce superior/inferior audio recordings.

  • Drivers – Software that ships with your product.
  • Digital Audio Converters – The conversion of audio to digital audio, for editing and processing on your PC. (See my Analog to Digital Converter section for more on this subject.)

Some audio interfaces may have built in Preamps, which can be an added benefit and may help produce a better recording. (See my Preamps section for more on this subject.)

As I mentioned earlier there are areas in which both the audio interface and sound card excel. Of course, you must research that the audio interface/card is compatible with your set-up. You should also evaluate whether or not you want to do more portable (on the road) or stationary (in the studio) recordings. (Respectively)

If your just starting out and looking for something with good sound quality, reasonable prices, and can withstand a few accidental BANGs! A portable audio interface will give you many options to start with and expand on.

If you’re looking to record solely from your home or project studio with a generally large track count – A traditional sound card or PCI chip with a breakout box will offer stable conversion and a large track count at very fast speed.

  • Most importantly the digital audio converters, which touches the sound, is the most important component in both. This is the thing that transfers the input audio and transforms it into digital audio.

In Conclusion:

  • Make sure you have a handle on the concepts of both before looking for specifics.
  • Research the compatibility of the interface/card and your PC/Laptop.
  • Keep in mind you are really looking for good: A/D/A/Converter/Preamps and Driver within the unit.

Remember, if you are just starting out: This is one important component among a number of important components involved in a quality home recording studio. So assess your budget/needs carefully.