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SPE3 – Releases and Changelogs

     Posted on: 2019-12-09

Speech Engine (SPE) is developed as RESTfull API on top of Phonexia BSAPI. SPE was formerly known as BSAPI-rest (up to v2.x) or as Phonexia Server (up to v3.2.x). This page lists changes in SPE releases. Releases Changelogs == SPE v3.18.x == Speech Engine 3.18.3 (12/09/2019) - DB v1300, BSAPI 3.22.2 Fixed: STT on stream may cause assert violation when waiting for stream timeout on no input data Fixed: SPE sends IP address in Host: HTTP header instead of hostname Fixed: SPE sometimes outputs "[ERRFMT]" string to log messages instead of actual value Speech Engine 3.18.2 (10/14/2019) - DB v1300,…

Supported audio formats

     Posted on: 2018-12-10

Supported audio format are: WAVE (*.wav) container including any of: unsigned 8-bit PCM (u8) unsigned 16-bit PCM (u16le) IEEE float 32-bit (f32le) A-law (alaw) µ-law (mulaw) ADPCM FLAC codec inside FLAC (*.flac) container OPUS codec inside OGG (*.opus) container   Other audio formats must be converted using external tools. SPE server can be configured to support automated conversion on background, see SPE configuration hints. Great tools for converting other than supported formats to supported are ffmpeg (http://www.ffmpeg.org) or SoX (http://sox.sourceforge.net/). Both are multiplatform software tools for MS Windows, Linux and Apple OS X. Example of usage: ffmpeg ffmpeg -i <source_audio_file_name>…

Age Estimation

     Posted on: 2018-04-12

Phonexia Age Estimation (AGE) estimates the age of a speaker from audio recording. The process of voiceprint extraction is similar to the extraction of SID, but as a result different features get extracted; therefore, the voiceprints extracted from AGE and SID are not mutually compatible. Technology Trained with emphasis on spontaneous telephony conversation The technology is language-, accent-, text-, and channel- independent Compatibility with the widest range of audio sources possible (applies channel compensation techniques): GSM/CDMA, 3G, VoIP, landlines, etc. Input Input format for processing: WAV or RAW (8 or 16 bits linear coding), A-law or Mu-law, PCM, 8kHz+ sampling…

Voice Activity Detection – Essential

     Posted on: 2018-04-04

Phonexia Voice Activity Detection (VAD) identifies parts of audio recordings with speech content vs. nonspeech content. Technology Trained with emphasis on spontaneous telephony conversation The technology is language-, accent-, text-, and channel- independent Compatibility with the widest range of audio sources possible (applies channel compensation techniques): GSM/CDMA, 3G, VoIP, landlines, etc. Input Input format for processing: WAV or RAW (8 or 16 bits linear coding), A-law or Mu-law, PCM, 8kHz+ sampling Output Log file with processed information (speech vs. nonspeech segments) Segmentation The section Segmentation describes the results of VAD, which are segments of detected voice and silence. Segments are…

Speech Quality Estimator – Essential

     Posted on: 2018-04-04

Phonexia’s Speech Quality Estimator quantifies the acoustic quality of recordings. This helps the user to quickly determine whether the acoustic quality of a recording is good for processing with other speech technologies or not. As an answer for SQE, the SPE returns a json/xml file. This file includes general information about the technology and statistics of all (one or two) channels. The statistics of all channels include the numbers for many aspects of recording quality, and the overall global score. Technology The technology is language-, accent-, text-, and channel- independent Compatibility with the widest range of audio sources possible (applies…

Speaker Diarization (DIAR)

     Posted on: 2017-06-26

About DIAR Phonexia Speaker Diarization (DIAR) enables segmentation of voices in one monochannel audio record. Technology Trained with emphasis on spontaneous telephony conversation The technology is language-, accent-, text-, and channel- independent Compatibility with the widest range of audio sources possible (applies channel compensation techniques): GSM/CDMA, 3G, VoIP, landlines, etc. Input Input format for processing: WAV or RAW (8 or 16 bits linear coding), A-law or Mu-law, PCM, 8kHz+ sampling Output Log file with processed information (segmentation of speech, silence, and technical signals – ie. elimination of phone lines beeps, DTMF tones, music, pauses, etc.) Audio file extracted for each…