Linear predictive coding in the context of Bit rate

Opus is a free and open source lossy audio coding format developed by the Xiph.Org Foundation and standardized by the Internet Engineering Task Force, designed for efficient low-latency encoding of both speech and general audio. Due to its lower latency relative to other standard codecs, Opus finds specific use cases in real-time interactive communication for low-end embedded processors. Opus replaces both Vorbis and Speex for new applications.

Opus combines the speech-oriented LPC-based SILK algorithm and the lower-latency MDCT-based CELT algorithm, switching between or combining them as needed. Bitrate, audio bandwidth, complexity, and algorithm choice can be adjusted for each individual frame. Opus has low algorithmic delay (26.5 ms by default) ideal for use as part of a real-time communication link, networked music performances, and live lip sync; by trading off quality or bitrate, the delay can be further reduced down to 5 ms. Its delay thus is significantly lower compared to competing codecs, which require well over 100 ms. Opus remains competitive with these formats in terms of quality per bitrate.

An audio codec is a device or computer program capable of encoding or decoding a digital data stream (a codec) that encodes or decodes audio. In software, an audio codec is a computer program implementing an algorithm that compresses and decompresses digital audio data according to a given audio file or streaming media audio coding format. The objective of the algorithm is to represent the high-fidelity audio signal with a minimum number of bits while retaining quality. This can effectively reduce the storage space and the bandwidth required for transmission of the stored audio file. Most software codecs are implemented as libraries which interface to one or more multimedia players. Most modern audio compression algorithms are based on modified discrete cosine transform (MDCT) coding and linear predictive coding (LPC).

In hardware, audio codec refers to a single device that encodes analog audio as digital signals and decodes digital back into analog. In other words, it contains both an analog-to-digital converter (ADC) and digital-to-analog converter (DAC) running off the same clock signal. This is used in sound cards that support both audio in and out, for instance. Hardware audio codecs send and receive digital data using buses such as AC'97, SoundWire, I²S, SPI, I²C, etc. Most commonly the digital data is linear PCM, and this is the only format that most codecs support, but some legacy codecs support other formats such as G.711 for telephony.

Speech coding is an application of data compression to digital audio signals containing speech. Speech coding uses speech-specific parameter estimation using audio signal processing techniques to model the speech signal, combined with generic data compression algorithms to represent the resulting modeled parameters in a compact bitstream.

Common applications of speech coding are mobile telephony and voice over IP (VoIP). The most widely used speech coding technique in mobile telephony is linear predictive coding (LPC), while the most widely used in VoIP applications are the LPC and modified discrete cosine transform (MDCT) techniques.

Full Rate (FR), also known as GSM-FR or GSM 06.10 (sometimes simply GSM), was the first digital speech coding standard used in the GSM digital mobile phone system. It uses linear predictive coding (LPC). The bit rate of the codec is 13 kbit/s, or 1.625 bits/audio sample (often padded out to 33 bytes/20 ms or 13.2 kbit/s). The quality of the coded speech is quite poor by modern standards, but at the time of development (early 1990s) it was a good compromise between computational complexity and quality, requiring only on the order of a million additions and multiplications per second. The codec is still widely used in networks around the world. Gradually FR will be replaced by Enhanced Full Rate (EFR) and Adaptive Multi-Rate (AMR) standards, which provide much higher speech quality with lower bit rate.