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A specific software or hardware implementation capable of audio compression and decompression to/from a specific audio coding format is called an audio codec; an example of an audio codec is LAME, which is one of several different codecs which implements encoding and decoding audio in the MP3 audio coding format in software.
As the MP3 standard allows quite a bit of freedom with encoding algorithms, different encoders do feature quite different quality, even with identical bit rates. As an example, in a public listening test featuring two early MP3 encoders set at about 128 kbit/s, [75] one scored 3.66 on a 1–5 scale, while the other scored only 2.22. Quality is ...
The Adaptive Multi-Rate (AMR, AMR-NB or GSM-AMR) audio codec is an audio compression format optimized for speech coding.AMR is a multi-rate narrowband speech codec that encodes narrowband (200–3400 Hz) signals at variable bit rates ranging from 4.75 to 12.2 kbit/s with toll quality [3] speech starting at 7.4 kbit/s.
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.
A classic method is nonlinear PCM, such as the μ-law algorithm. Small signals are digitized with finer granularity than are large ones; the effect is to add noise that is proportional to the signal strength. Sun's Au file format for sound is a popular example of mu-law encoding. Using 8-bit mu-law encoding would cut the per-channel bitrate of ...
For example, MP3 and AAC dominate the personal audio market in terms of market share, though many other formats are comparably well suited to fill this role from a purely technical standpoint. First public release date is first of either specification publishing or source releasing, or in the case of closed-specification, closed-source codecs ...
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.
It uses 4 or 8 subbands, an adaptive bit allocation algorithm in combination with an adaptive block PCM quantizer. [1] Frans de Bont has based the SBC audio codec on his earlier work, [7] and – in parts – on the MPEG-1 Audio Layer II standard. In addition, the SBC is based on the algorithms described in the EP-0400755B1. [8]