The Well-Tempered Computer, an introduction to computer audio


16/44, 24/88
Bit depth and sample rate.
PCM audio is samples and sample rate.
The samples are 16 or 24 bit words and the sample rate is in kHz
16/44.1 is 16 bits words that should be played at a speed of 44.1 kHz or 44100 samples per second.
This is what you will find on a CD (Redbook audio).
There are two series.
44/88/176 and 48/96/192.
Most of the time high sample rates are combined with 24 bit words.

 A common standard in the pro-world to connect audio devices digital.
It was developed by the Audio Engineering Society (AES) and the European Broadcasting Union (EBU) and first published in 1985
It requires a balanced and screened (shielded) cable with nominal characteristic impedance of 110 Ω at frequencies from 0.1 to 128 times the maximum frame rate.
SPDIF is derived from this standard and intended for consumer use.

Bit perfect playback
True bit perfect playback is that the information in audio file (the bits) is send unaltered to the audio device.
Bit depth, sample rate, number of channels should remain unaltered.
This of course requires the hardware to match the properties of the audio source exactly.
Do observe that bit perfect playback is about the samples played without any DSP applied. It says nothing about the accuracy of the time step.
As PCM audio is samples with a fixed sample rate; perfect playback is bit perfect and time step perfect.

Media players often report the bitrate of the audio file playing.
1411 kbits is the value you will see when playing CD quality audio using an uncompressed lossless format like WAV.
If you play FLAC you see most of time a value between 700 – 800 as FLAC compresses 40-50%.
This is the bitrate of the compressed file.
However it is lossless so when expanded to raw PCM you have the full unaltered original 1411 kbits.
The bitrate of a lossless compressed format indicates the amount of lossless compression.
If a MP3 is created as e.g. 320 kbs CBR this indicates how much of the original information is discarded.
Indeed MP3 is like magic, at 320 the bitrate is reduced from 1411 to 320, it throws out ¾ of the information and still most of us won’t hear a difference.
Not to be mistaken for a absence of differences.

Digital to Analog Converter
A chip converting a numerical value (the bits) to an analog signal.
Audiophile speak: a box doing the conversion
PC speak: sound card

Pulse Code modulation as a method to digitize sound has been proposed by Alec Reeves in 1937. It had to wait until the transistor and the IC become available at low cost before this method was successfully introduced in the consumers market. Almost all of today's digital audio is PCM audio.

The principle is simple, the analogue signal is measured at uniform intervals. The magnitude of the signal is translated into a numeric value and as it is a number, it could be represented in bits.

Sony/Philips Digital Interconnect Format is essentially a minor modification of the original AES/EBU standard for consumer use.
It has become the digital equivalent of the analogue RCA connection, in fact it often uses the same RCA connectors.
It is one way communication (unidirectional) from a transmitter to a receiver.
It is very popular in the audio world.
Multi media PCs might have a SPDIF out.
Often this is and optical one (Toslink)

 Toshiba created TOSLINK (1983) to connect their CD players to their receivers. It was soon adopted by manufacturers of most CD players. Using light it is immune RFI and EMI and the devices are completely electrically separated.
Toslink (EIAJ optical) is SPDIF over optical.
The light source is a simple and inexpensive LED.
Toslink cables are made of plastic or glass.
The plastic ones are cheaper but have a higher attenuation coefficients (1 dB/m or higher).
 Due to the high attenuation, reliable transmission distances were limited to 10 meters, in practice often under 5 meters.
Today companies like Lifatec offers length up to 30 m (98 ft) using APF (All Plastic Fiber) with an attenuation (optical loss) less than 0.15dB/m.  

USB audio
The audio is send to the USB audio device in isochronous mode.
The receiver can adapt itself to the send rate of the PC by regulating the speed of the DAC.
This is aptly called adaptive synchronisation.
As USB is bi-directional the receiver can control the rate of the data send by the PC.
This is called asynchronous synchronization.

USB audio limitations
A lot of people think USB audio is limited to 16 bits / 48 kHz max.
This is not true.
A lot of (cheap and sometimes not so cheap) USB DACs are indeed limited to this resolution.
This is because the manufacturer decided to use a simple and cheap of the shelf hardware solution (USB receiver).
USB Audio Class 1 standard (1998) This standard allows for 24 bits / 96 kHz max.
The standard itself doesn't impose any limitation on sample rate.
Class 1 is tied to USB 1 Full Speed = 12 MHz so 24/96 is the highest popular sample rate that fits in.
All OS (Win, OSX, Linux) supports this standard.

USB Audio Class 2 standard (2009)
It is downwards compatible with class 1.
USB Audio Class 2 additionally supports 24 and 32 bit and all common sample rates.
Class 2 need USB High Speed (480 MHz). This requires USB 2 or 3.
From mid-2010 on USB audio class 2 drivers are available in OSX 10.6.4 and Linux.
Both support sample rates up to 384 kHz.
It is unclear if Microsoft is going to support USB Audio 2.
You need a third party USB class 2 driver on Windows.

Source: The Welltempered Computer