Audio Recording and Transmission
The audio chain from microphone and analog-to-digital conversion through sampling, quantization, bit depth, channel layout, codecs, real-time transport, and clock drift.
Audio recording begins with a microphone, analogue front end and analogue-to-digital converter. The resulting sample stream can be stored, compressed or transported over a network in real time.
Analog-to-Digital Conversion
A microphone converts variations in air pressure into an electrical signal. The analog front end applies gain and the required filtering, and the analog-to-digital converter samples the signal at discrete times and quantizes each sample to a finite numerical level.
This chain has two distinct forms of resolution:
- time resolution, determined by sampling rate,
- amplitude resolution, determined by bit depth.
Increasing sampling rate does not directly increase quantization resolution, and increasing bit depth does not directly increase temporal resolution. They affect different error mechanisms.
Analog band limiting before conversion is important. If components above the Nyquist frequency are not sufficiently attenuated, sampling can fold them into the baseband as aliasing; this distortion cannot later be removed completely by digital filtering alone.
Sampling and channel layout
Sampling rate defines how many samples are acquired per second. Bit depth affects quantisation resolution and available dynamic range. Channel count may preserve additional spatial or source-separation information.
Blindly reducing multichannel audio to mono can remove useful information for later processing.
PCM and codecs
WAV often carries PCM, but the file extension alone does not guarantee the actual encoding. MP3, AAC and Opus use different lossy coding strategies and introduce different artefacts.
In machine-learning systems, differences in codec, sample rate and channel configuration between training and deployment data can affect model stability.
Real-time transport
Packet duration, loss recovery and jitter-buffer behaviour determine the balance between continuity and latency. Larger buffers absorb network variation while increasing interaction delay.
Clock drift also matters in long recordings. Two devices operating at nominally identical sampling rates may still progress at slightly different effective rates.
Codec, sampling rate, channel layout and timestamps should therefore be treated as part of the audio data contract.
Parent topic: Image and Audio Processing
References
- Alan V. Oppenheim; Ronald W. Schafer. Discrete-Time Signal Processing, Third Edition. Pearson, 2010.
- Lawrence R. Rabiner; Ronald W. Schafer. Theory and Applications of Digital Speech Processing. Pearson, 2011.