+++ title = "What is digital audio?" description = "What is digital audio?" chapter = false weight = 3 #pre = "1. " +++ **Ardour** is a Digital Audio Workstation (DAW). Beforing using it to record and edit sound, it might be useful to review how digital audio works. ![analogue-digital](en/Ardour4_Digital_Audio_1.png) {{}} graph TD; A(fa:fa-microphone Analog input) --> B(Analog to digital conversion) B --> | digital numeric data, samples | C(Digital system) C --> D(Digital to analog conversion) D --> E(fa:fa-headphones Analog output) {{< /mermaid >}} The diagram above shows how sound travels to and from your computer. The "Analogue to Digital Conversion" (ADC) and the "Digital to Analogue Conversion" (DAC) are done by the sound card or audio interface. The "Digital System" in this case is your computer running **Ardour.** ## Frequency and Gain Imagine a loudspeaker. To move the air in front of it and make sound, the membrane of the speaker must vibrate from its center position (at rest) backwards and forwards. The number of times the membrane vibrates each second determines the **Frequency** (the note, or *pitch*) of the sound you hear. The distance the membrane travels from its resting point determines the **Amplitude** (the volume, or *loudness*) of the sound. Normally, we measure Frequency in **Hertz** (Hz) and Amplitude in **Decibels** (dB). ![speaker](en/Ardour4_Digital_Audio_2.png) Check out the great animation on this page illustrating this process: {{< youtube RxdFP31QYAg >}} A microphone works like a loudspeaker in reverse: vibrations in the air cause its membrane to vibrate. The microphone turns these acoustic vibrations into an electrical current. If you plug this microphone into a computer's sound card and start recording, the sound card makes thousands of measurements of this electric current per second and records them as numbers. The number of **Samples** (i.e. measurements) made per second is called the **Sample Rate**, and the number of possible values each Sample can have is called the **Bit Depth**. The combination of Sample Rate and Bit Depth indicates how closely the digital signal can reproduce the sound it has recorded. ## Peaks and Clipping When Ardour displays the Samples which have been recorded, they appear as the **Waveform** we see below. The center horizontal line indicates the membrane of the speaker at rest, and the **Peaks** of the Waveform indicate the maximum **Amplitude.** ![waveform](en/Ardour4_Digital_Audio_Waveform.png) If we take a Waveform and increase its the Amplitude a lot, some of the Peaks may now fall outside the range that the computer can represent digitally. The computer's inability to represent Peaks outside the range of Amplitude is called **Clipping**, which results in a permanent loss of digital information, as well as a change in the sound quality which is recognizable as **Distortion**. Ardour marks clipped Peaks with the color red, as can be seen in the image below. ![clipping](en/Ardour4_Digital_Audio_Clipping2.png) In the image above, one can also see the **Mixer Strip** on the far left, which gives a running measurement of the Peaks, as well as an indication at the top of the **Peak Meters** showing the maximum Peak so far. The red number indicates Clipping has occurred. {{% notice tip %}} Clipping often can happen at the time of recording if you set your microphone levels too high. {{% /notice %}} The range of decibels between the region's maximum peak and the clipping point is commonly referred to as **Headroom**, and common recording practice is to keep approximately three to six Decibels of Headroom between the maximum of your signal and the clipping point, with the clipping point itself being represented as 0 dB (zero Decibels). In other words, an audio region with a comfortable amount of Headroom would have its maximum peaks between −6 dB and −3 dB. Also, because the Peaks of audio signals add together, care must be taken when **Mixing** several sources together to keep the combined signals from Clipping. ## Sample Rate and Bit Depth To make audio playable on a compact disc, for example, the computer must generate 44,100 Samples per second. The Sample Rate determines the highest frequency which can be recorded or played back by the computer. A sampling rate of 44.1 kHz means that the highest frequency which can be represented is just under 22.05 kHz. Since normal human hearing lies within the range of approximately 20 Hz to 20 kHz, this is commonly accepted as a reasonable Sample Rate. Other commonly used Sample Rates include 48 kHz (DAT recorders) or 96 kHz (DVD audio). Each Sample is recorded as a **16-bit number**. One **Bit** is a piece of information which is either 0 or 1. If there are 16 bits together to make one sample, then there are 2^16 (65,536) possible values for each sample. Thus, we can say that CD-quality audio has a Sample Rate of 44.1 kHz and a **Bit Depth** of 16 bits. Professional music recordings are usually mixed using 24 bits to preserve the highest amount of detail before being mixed down to 16 bits for CD. Older computer games have a distinctively rough sound, using only 8 bits. By increasing the Sample Rate, we are able to record higher sonic frequencies, and by increasing the Bit Depth, we are able to use a greater **Dynamic Range** (the difference between the quietest and the loudest sounds possible to record and play). Here is a great video tutorial explaining sampling rate and bit depth in a lot more detail. {{< youtube zC5KFnSUPNo >}}