The Audio Chain

When we want to listen to some music, we click on the desired song, and it starts playing through our speakers or headphones. Have you ever thought more deeply about what happens in the background? What series of events has to occur for a digital audio file—a bunch of ones and zeroes—to transform into physical sound waves our brain then receives through our ears and interprets as sound? Let's go over the process.

Digital-to-Analog Conversion

First, your file has to go through digital-to-analog conversion. The audio signal can't be amplified or played back unless it's in an analog form, and that's where a digital-to-analog converter (DAC) steps in. A DAC is a chip that takes a digital file, "reads" it in its digitally sampled form, and turns it into an analog waveform corresponding to the digital samples contained in that file. The more digital samples an audio file has, the better a digital-to-analog conversion can be achieved. That's why one of the key technical specifications of a DAC chip is its maximum sample rate—the number of samples of data taken in a second. The sample rate is expressed in kilohertz (kHz). When it comes to specifications, we also look for bit depth—the amount of data recorded per sample, expressed in bits. A standard modern-day DAC should be capable of handling at least 24-bit/96 kHz audio files.


Here we're talking about Pulse-Code Modulation (PCM) audio files. PCM is the de facto standard non-proprietary digital sampling method in which samples are taken regularly at uniform intervals. Most if not all music files on your PC (MP3, OGG, FLAC, WAV, and AIFF), as well as almost everything you play from your music streaming service of choice, are sampled using PCM. The story is quite different with Direct Stream Digital (DSD) audio files, which are a result of a pulse-density modulation, as well as other "audiophile" audio compression formats, such as the controversial Master Quality Authenticated (MQA) available to Tidal HiFi Plus subscribers. While PCM digital audio files are supported by any device that can play music, DSD and MQA support is limited and device-specific.


The optimal bit depth and sample rate of PCM audio files is a subject of constant debate between audio enthusiasts (along with just about everything else). Some will easily discard a DAC that can't work with bit depths and sample rates higher than 24-bit/96 kHz. In reality, most users won't benefit from their DAC being capable of working with higher bit depths and sample rates because they'll never play an audio file recorded in such "quality." Even 24-bit/96 kHz is a stretch for most. For example, your Tidal HiFi or Apple Music subscriptions you're probably using because of their high quality max out at 16-bit/44.1 kHz. This isn't a drawback, though, as the fidelity of a 16-bit/44.1 kHz audio file is generally considered "CD quality" and high enough to reproduce all of the frequency and dynamic range an average person can hear.

Bit depth isn't to be confused with bitrate, a value that tells us the number of bits processed per second. Out of all numerical values describing the "quality" of an audio file, bitrate is probably the one average users are most familiar with. This dates back to the glory days of MP3 files when everyone knew that a 128 kbps bitrate song would sound much worse than a 320 kbps one. Nowadays, music streaming services offer 160, 256, or 320 kbps streams in their lossy subscription tiers, with the lossless (HiFi) ones going up to 1,411 kbps.

While we've only scratched the surface of digital audio file conversion, it's time to move on to the next step of the process: analog signal amplification.

Amplification

Once the DAC is done converting ones and zeroes to an analog waveform, that waveform is sent at a line level to the amplifier. The amplifier's job is to amplify the signal (who knew?) to a level where we can listen to it comfortably without adding any distortion in the process. The raw output power of the amplifier and quality of its implementation will determine how loud you can push your headphones and what amount of distortion will occur. From my experience, the amplifier is also a determining factor in the overall sound signature of a sound card. Is the sound card going to come across as snappy or mellow, will it be perfectly clean or somewhat noisy, how much detail will you be able to retrieve from the source material—an amplifier will influence all that and more. Just to be perfectly clear, no amplifier can turn poor-quality headphones into good ones. However, a good amplifier can certainly bring the best out of the headphones connected to it.

The power of a headphone amplifier is expressed in a way that's hard to grasp by an average user. To better understand it, we first need to take into consideration two aspects of headphones themselves: impedance and sensitivity. The impedance of a headphone, directly related to the design of its voice coils, is typically rated between 8 and 600 Ω, with 32–38 Ω being the norm for gaming headsets and many mainstream headphones. In general, high impedance headphones (over 50 Ω) are harder to drive than low impedance ones—they require a more powerful amplifier to reach a comfortable listening volume. In general, but not always because of sensitivity, the other factor I mentioned.

Sensitivity (or efficiency) tells us how much volume (in dB) a headphone can produce at a certain power rating. It is usually measured at 1 kHz, with 1 mW of power being applied to the headphones. Headphone sensitivity can range anywhere from 70 to 110 dB without necessarily being directly linked to impedance. You can have a high sensitivity headphone with high impedance or vice versa. However, when comparing two headphones of identical impedance, the more sensitive ones would play louder at a fixed power level. In other words, they could be driven by a less powerful amplifier.


To make matters slightly more complicated, headphone amplifiers are commonly rated by how much output voltage they can deliver (instead of power), usually at two different impedances, such as 32 and 150 Ω. In order to find out if the headphone amplifier has enough juice to drive your headphones confidently, you don't have to break your teeth on complicated mathematical formulas. Use a decent online calculator instead, such as the one available at digiZoid. After you enter the impedance and sensitivity of your headphones, look for the voltage needed to achieve the listening loudness you're after. It would be best not to go much above 100 dB SPL unless you want to permanently damage your hearing. Finally, look at the specified voltage output of your headphone amplifier and check if it's capable of delivering that voltage for the desired loudness at the impedance level of your headphones.


What about speakers? They still need a DAC to convert a digital file into an analog waveform, but the difference compared to headphones is that PC speakers are usually powered, which means they're equipped with their own amplifier(s), while the headphone analog waveform amplification happens in the sound card itself. For that reason, you want to avoid connecting your powered speakers to the headphone output. Otherwise, the analog signal will get amplified twice, which can result in a lot of distortion or, in extreme cases, damage to speaker membranes. Your powered speakers need a line-level analog signal, so the line-out port of your sound card is the one to use.

As for headphones and headsets, most sound cards use a 3.5-millimeter output, and the connector itself usually comes in two versions: 3-pole TRS or 4-pole TRRS. They look identical, so it's good to be aware of their differences to avoid confusion about why something's not working as expected.


A TRS (Tip – Ring – Sleeve) connector is used for a stereo signal. Most if not all sound cards integrated on the motherboard use 3.5-mm TRS connectors, which is why you'll have to use two separate plugs when connecting your headset: one for the headphone output and another for the microphone input. PC headsets regularly come with a splitter cable which makes this possible.


A TRRS (Tip – Ring – Ring – Sleeve) connector has an extra ring on it, which is used for the microphone input. In other words, you can plug a single 3.5-mm TRRS plug into a TRRS port to get your headset to output stereo sound and record the microphone at the same time. TRRS connectors first started showing up on smartphones and laptops, but are now everywhere: on console gamepads, various mobile devices, and most modern USB sound cards.


It's worth mentioning that two different, mutually incompatible variants of the TRRS connector coexist, with different pin layouts. In the OMTP standard, the top of the connector (tip) is the left channel, first ring the right channel, second ring the microphone, and bottom of the connector (sleeve) the ground line. In the CTIA standard, the ground line and microphone are swapped around. Fortunately, OMTP is dying out, so nowadays, you no longer have to worry about plugging a 3.5-mm TRS plug into a TRRS port (or vice versa). While your headset's microphone won't work in any of these combinations because either the plug or port is missing the aforementioned extra ring, the sound will output normally. If you do happen to run into a TRRS connector based on the OMTP layout, your TRS plug won't work with it properly.