Dialogic ADPCM or VOX is an audio file format, optimized for storing digitized voice data at a low sampling rate. VOX files are most commonly found in telephony applications, as well as an occasional arcade redemption game.[ citation needed ] It uses a lossy compression algorithm, optimized for voice, not high fidelity.
Similar to other ADPCM (Adaptive Differential Pulse Code Modulation) formats, Dialogic ADPCM compresses audio data into a series of 4-bit samples. The original Dialogic ADPCM paper (linked to below) does not specify or mention a recording or playback frequency; it may be at the implementer discretion. However, traditionally, files commonly have a sampling rate of 6000 or 8000 samples per second, but 8000 samples per second (8000 Hz) is more common. 8000 Hz matches the sampling rate used in G.711 voice systems such as DS1.
Unlike a WAV file, a VOX file does not contain a header to specify the encoding format or the sampling rate, so this information must be known in order to play the file. If not known, it is normally assumed that a VOX file is encoded with Dialogic ADPCM at a sampling rate of 8000 Hz. It is possible that a VOX file may be encoded in a format other than Dialogic ADPCM, but this is not common.
Dialogic ADPCM is an open file format. It matches ITU-T standard G.721, later superseded by G.726.
The algorithm for Dialogic ADPCM was developed by Oki Electric, which also produced ICs such as the Oki Semiconductor MSM7580 to implement the algorithm in hardware. These ICs were used on popular telephony interface cards manufactured by Dialogic Corporation for use in voicemail and similar systems. As this was the most common use for the file format, it became known as "Dialogic ADPCM."
Some early BlackBerry phones that don't support MP3 format (e.g. 7100) used that codec for sound files which had ADP filetype extension. These must be of max filesize of 128 Kbytes or less.
ADP filetype extension were being just renamed VOX filetype extension. The AD4 extension is also used for files compressed using "Dialogic ADPCM" with a sample rate of 36000 Hz. This means that .ad4 files can be decoded if imported as VOX ADPCM with a sample rate set to 36 kHz on software that allows such settings, like e.g. Audacity.
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MP3 is a coding format for digital audio developed largely by the Fraunhofer Society in Germany under the lead of Karlheinz Brandenburg, with support from other digital scientists in other countries. Originally defined as the third audio format of the MPEG-1 standard, it was retained and further extended — defining additional bit-rates and support for more audio channels — as the third audio format of the subsequent MPEG-2 standard. A third version, known as MPEG-2.5 — extended to better support lower bit rates — is commonly implemented, but is not a recognized standard.
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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.
Waveform Audio File Format is an audio file format standard, developed by IBM and Microsoft, for storing an audio bitstream on personal computers. It is the main format used on Microsoft Windows systems for uncompressed audio. The usual bitstream encoding is the linear pulse-code modulation (LPCM) format.
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Digital audio is a representation of sound recorded in, or converted into, digital form. In digital audio, the sound wave of the audio signal is typically encoded as numerical samples in a continuous sequence. For example, in CD audio, samples are taken 44,100 times per second, each with 16-bit sample depth. Digital audio is also the name for the entire technology of sound recording and reproduction using audio signals that have been encoded in digital form. Following significant advances in digital audio technology during the 1970s and 1980s, it gradually replaced analog audio technology in many areas of audio engineering, record production and telecommunications in the 1990s and 2000s.
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G.711 is a narrowband audio codec originally designed for use in telephony that provides toll-quality audio at 64 kbit/s. It is an ITU-T standard (Recommendation) for audio encoding, titled Pulse code modulation (PCM) of voice frequencies released for use in 1972.
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aptX is a family of proprietary audio codec compression algorithms owned by Qualcomm, with a heavy emphasis on wireless audio applications.
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