Audio codec review

An audio codec (short for audio encoder/decoder) plays a key role in modern digital audio systems, converting audio signals between analog and digital formats. It efficiently compresses audio data for transmission or storage and decompresses it for playback. Audio codecs are widely used in applications such as VoIP, streaming, video conferencing, and radio broadcasts, where it is important to maintain high audio quality while optimizing bandwidth.

What is an audio codec?

The term “codec” is short for COder/DECoder. It is a software or hardware module (or sometimes both) that performs two main, often sequential, tasks:

Encoding (compression)

Takes a raw, uncompressed digital audio signal (such as a clean WAV or AIFF straight from a studio-grade microphone) and compresses it. This dramatically reduces the file size by removing redundant or less important information for human perception.

Decoding (decompression)

Takes a compressed audio file (such as MP3, AAC, or FLAC) and restores it to a form that your speakers or headphones can play. This happens in real time when you press play.

Why do we need audio codecs?

Uncompressed digital audio, such as CD quality, can generate more than 1 Mbps of data per second. For embedded systems and IP networks, this is a huge burden.

With the right audio codecs, you can:

• Reduce the amount of data, saving bandwidth

• Optimize the storage of recordings

• Ensure real-time transmission, which is especially important for two-way communication

• Improve the sound quality with features such as noise reduction or gain control

How do audio codecs work?

Audio codecs are at the heart of any digital audio communication system. Their main job is to convert an analog audio signal into a compressed digital stream for transmission, and then restore the original sound on the receiving end.

Let's break this process down into stages:

• Analog-to-digital conversion (ADC)

If the audio source is analog (such as a microphone), the signal is first digitized. An ADC (analog-to-digital converter) samples a wave at a fixed frequency (e.g. 44.1 or 48 kHz) and quantizes it with a certain bit depth (e.g. 16 or 24 bits).

• Coding (compression)

The digitized signal passes through an encoder, which applies algorithms to reduce the amount of data.

• Lossy (MP3, AAC, Opus): removes audio components that are less noticeable to the human ear.

• Lossless (FLAC, ALAC): reduces the size without removing information, using entropy coding methods.

• Transmission or storage

The compressed stream is transmitted over the network (VoIP, conferences, streaming) or saved to a file. Due to the reduced size, bandwidth and memory consumption are reduced.

• Decoding (decompression)

During playback, the decoder performs the inverse conversion, restoring the digital signal. For real time (calls, broadcasts), low latency is important.

• Digital to Analog Conversion (DAC)

If the device is playing analog sound (headphones, speakers), the DAC converts the decoded digital signal back into a continuous analog wave.

As a result, the listener hears sound that is as close to the original as possible, even after compression, transmission and restoration.

Common audio codecs and comparison

Tips:

• G.711 - Provides the best compatibility across all VoIP/SIP systems

• G.722 - Provides enhanced voice clarity

• Opus - Ideal for modern low-latency, high-quality conferences

• AAC - Widely used for streaming audio and video

Real-World Applications of Audio Codecs

IP Public Address System

The diagram shows the real-time audio transmission from a SPON server to an IP speaker.

According to the packet data obtained via Wireshark, the MP3 codec is used for audio transmission.

Advantages of using MP3 in notification systems:

• High compression efficiency - MP3 significantly reduces file size without noticeable loss of quality, which is important for stable transmission over the network.

• Low bandwidth consumption - compared to uncompressed formats (e.g. PCM), MP3 requires fewer resources.

• Wide compatibility - MP3 is supported by almost all modern devices, which simplifies integration.

• Mature and stable technology - proven encoding and decoding algorithms.

• Low latency support - when properly configured, suitable for emergency notification systems and live announcements.

• Resource saving - MP3 decoding requires little processor power and memory, which is convenient for embedded devices.

IP intercom system (Intercom System)

The diagram shows real-time audio transmission from SPON Windows APP to an IP intercom.

According to packet data obtained via Wireshark, the G.711μ (PCMU) codec is used for audio transmission.

Advantages of using G.711μ in intercom systems:

• Low latency - minimal latency due to the absence of complex compression.
• High speech clarity - 64 kbps provides excellent quality for dialogues.
• Ease of processing - the codec requires minimal processor resources, ideal for embedded devices.
• Wide compatibility - supported by SIP/VoIP systems, PBX, NVR.
• No license fees - standard, free for manufacturers and users codec.

How to choose the right audio codec?

Criterion	Recommendation
Network bandwidth	Use low-bitrate codecs (G.726, mp3)
Audio quality	Choose Opus, AAC or G.722 for best clarity
Delay sensitivity	G.711 or Opus for minimum delay
System compatibility	G.711 is the most supported in SIP/NVR
Device performance	Avoid "heavy" codecs on weak hardware

Frequently Asked Questions (FAQ):

Q1: Why are AAC recordings smaller than G.711?

A1: AAC is a lossy, highly compressed format, while G.711 is uncompressed PCM, which takes up more space.

Q2: Is G.722 compatible with G.711?

A2: No, they are different codecs. If the device does not support G.722, transcoding will be required.

Q3: Opus is a great codec. Why is it not always used?

A3: Opus requires more CPU and memory, which is not suitable for low-end devices.

Q4: No sound on SIP call?

A4: Check codec negotiation in SIP INVITE/200 OK messages and use Wireshark for diagnostics.

Conclusion

Audio codecs are a key element of modern communication and broadcasting systems. Understanding their features allows you to:

• Improve sound quality

• Reduce latency and network load

• Avoid compatibility issues

• Choose equipment wisely

We are ready to help you choose a codec strategy that is best suited for your system.