MP3

MPEG-2.5 was not developed by MPEG (see above) and was never approved as an international standard. MPEG-2.5 is thus an unofficial or proprietary extension to the MP3 format. It is nonetheless ubiquitous and especially advantageous for low-bit-rate human speech applications.

Bitrate is the product of the sample rate and number of bits per sample used to encode the music. CD audio is 44100 samples per second. The number of bits per sample also depends on the number of audio channels. CD is stereo and 16 bits per channel. So, multiplying 44100 by 32 gives 1411200—the bitrate of uncompressed CD digital audio. MP3 was designed to encode this 1411 kbit/s data at 320 kbit/s or less. As less complex passages are detected by MP3 algorithms then lower bitrates may be employed. When using MPEG-2 instead of MPEG-1, MP3 supports only lower sampling rates (16000, 22050 or 24000 samples per second) and offers choices of bitrate as low as 8 kbit/s but no higher than 160 kbit/s. By lowering the sampling rate, MPEG-2 layer III removes all frequencies above half the new sampling rate that may have been present in the source audio.

For the general field of human speech reproduction, a bandwidth of 5512 Hz is sufficient to produce excellent results (for voice) using the sampling rate of 11025 and VBR encoding from 44100 (standard) WAV file. English speakers average 41–42 kbit/s with -V 9.6 setting but this may vary with amount of silence recorded or the rate of delivery (wpm). Resampling to 12000 (6K bandwidth) is selected by the LAME parameter -V 9.4. Likewise -V 9.2 selects 16000 sample rate and a resultant 8K lowpass filtering. For more information see Nyquist – Shannon. Older versions of LAME and FFmpeg only support integer arguments for the variable bit rate quality selection parameter. The n.nnn quality parameter (-V) is documented at lame.sourceforge.net but is only supported in LAME with the new style VBR variable bit rate quality selector—not average bit rate (ABR).

The ancillary data field can be used to store user defined data. The ancillary data is optional and the number of bits available is not explicitly given. The ancillary data is located after the Huffman code bits and ranges to where the next frame's main_data_begin points to. Encoder mp3PRO used ancillary data to encode extra information which could improve audio quality when decoded with its own algorithm.

The first is uncompressed WAV file. The second is a Vorbis file encoded at 48kbit/s, and third is an MP3 encoded at 48kbit/s using LAME.