Original Audio Formats

ADPCM stands for Adaptive Differential Pulse Code Modulation. ADPCM is a lossy compression mechanism. There are various flavors of ADPCM. This particular algorithm was suggested by Microsoft; its quality is similar to IMA (Interactive Multimedia Association) ADPCM. MS ADPCM compresses data recorded at various sampling rates. Sound is encoded as a succession of 4-bit nibbles. Each nibble represents the difference between the current sampled signal value and the previous value. The compression ratio obtained is relatively modest: 16-bit data samples encoded as 4-bit differences result in 4:1 compression format.

Microsoft ADPCM is directly supported on most Windows implementations as a native format. Although the quality of IMA ADPCM voice files is not great, the files are portable. There is a real advantage in having compact files that can be played on most Windows PCs.

Compressed WAV format. A-Law (or CCITT standard G.711) is an audio compression scheme common in telephony applications. It is a slight variation of the u-Law compression format, and is found in European systems. This encoding format compresses original 16-bit audio down to 8 bits (for a 2:1 compression ratio) with a dynamic range of about 13-bits. Thus, a-law encoded waveforms have a higher s/n ratio than 8-bit PCM, but at the price of a bit more distortion than the original 16-bit audio. The quality is higher than you would get with 4-bit ADPCM formats. Encoding and decoding is rather fast and generally, widely supported.

AIFC is short for AIFF(C) or AIFF-C, i.e. the Audio Interchange File Format with optional compression. AIFC is a newer version of the format that includes the ability to compress the audio data.

DSP Group's TrueSpeech is a family of high quality, low bit rate, speech compression algorithms which compress speech down to as little as 1/40th its original size. Several different versions of TrueSpeech at different compression rates are available for licensing, from 8.5 Kbps through 3.9 Kbps. All offer excellent communications over a 14.4 Kbps or better modem.

TrueSpeech 8.5 is the 8.5 Kbps member of DSP Group's TrueSpeech family of software products. It is a low complexity speech coder, which is an integral component of Microsoft windows and has also been endorsed by Dialogic for computer telephony products. TrueSpeech 8.5 should be used when compatibility with Microsoft is a prerequisite.

Speech information can now be exchanged compatibly between different applications. For example, using TrueSpeech 8.5 for digital simultaneous voice and data applications (DSVD), it may be feasible to utilize the same DSP chip for both speech compression and high speed modem data pump tasks. At the sampling rate of 8 KHz, continuous digital speech is compressed from 128 Kbps to 8.5 Kbps, a 15:1 compression ratio, while maintaining good speech quality. With slightly lower voice quality and lower levels of compression, TrueSpeech 8.5 requires only about half the MIPS and program memory space as TrueSpeech 6.3 and 5.3.

GSM 06.10 stands for Global System for Mobile Communications and is a variant of LPC called RPE-LPC (Regular Pulse Excited - Linear Predictive Coder) and is a European standard originally for use in encoding speech for satellite distribution to mobile phones. It can be found in use in various telephony products such as voice mail applications.

It compresses 160 13-bit samples into 260 bits (or 33 bytes), i.e. 1650 bytes/sec (at 8000 samples/sec). It results in very good compression with good quality output but is very costly in terms of performance.

Used for computer telephony. 32 kbit/s adaptive differential pulse code modulation (ADPCM).
Good for keeping of human speech.

The extensions are *.mp2 or *.mpa.

The mp3 algorithm development started in 1987, with a joint cooperation of Fraunhofer iis-a and the university of erlangen. it is standardized as iso-mpeg audio layer 3. it soon became the de facto standard for lossy audio encoding, due to the high compression rates (1/12 of the original size, still remaining considerable quality), the high availability of decoders and the low cpu requirements for playback. (486 dx2-66 is enough for real-time decoding). it supports multichannel files (although there's no implementation yet), sampling frequencies from 16khz to 24khz (mpeg2 layer 3) and 32khz to 48khz (mpeg1 layer 3). formal and informal listening tests have shown that mp3 at the 192-256 kbps range provide encoded results undistinguishable from the original materials in most of the cases.

mp3 uses the following for compression:

Compression ratio is 1:10...1:12 corresponds to 128..112 kbps for a stereo signal.

MPEG Version 2.5 was added lately to the MPEG 2 standard. It is an extension used for very low bitrate files, allowing the use of lower sampling frequencies. If your decoder does not support this extension, it is recommended for you to use 12 bits for synchronization instead of 11 bits.

MPC is a new space-saving audio format which was formerly known as MPEG Plus (.mp+). Very similar to MPEG Layer 2, but uses subband-based selectable channel coupling, Huffman coding, differential Huffman coding. Typical data rates are between 160 and 200 kbps. MPEGplus encoder uses a frequency range that can reach up to 22KHz. This is because many people can hear sounds above 16KHz even when it is sometimes hard to hear anything but it makes a lot of difference in sound dynamics.

The MPEGplus format's most important technique to reduce the bitrate is the exploitation of psychacoustic effects. The psychacoustic effect is determined by doing sound test and to check which sounds the human ear can hears and which sounds not. This means that when you hear a very hard explosion it is almost impossible to hear a drop of water falling at the same time. This is why the sound of the drop of water is faded out because it's impossible to hear and so it will not be noticed. When this sound of the drop of water is faded away this preserves the bitrate. The M/S Stereo technique uses the stereo field to compare both channels and when the sound on both channels is the same or almost the same it has to be encoded just once, preserving the bitrate.

it has simple stereo support and is limited to a frequency of 44.100hz, although stream version 8 [sv8] will be able to encode 32/48khz streams, as well as multichannel ones.

informal listening tests have demonstrated that mpc is the best publicly available lossy audio encoder at bitrates higher than 160kbps. being a subband encoder and given their inherint nature to be less efficient than transform coders, it is worse than aac and ogg vorbis in bitrates lower than 160kbps.

it uses for compression:

Ogg Vorbis is a fully open, non-proprietary, patent-and-royalty-free, general-purpose compressed audio format for mid to high quality (8kHz-48.0kHz, 16+ bit, polyphonic) audio and music at fixed and variable bitrates from 16 to 128 kbps/channel.

It is the files with .wav extension. But this files can be converted by different codecs. NCTAudioStudio2 supports the following types of WAV files: