37 lines
1.6 KiB
Text
37 lines
1.6 KiB
Text
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GSM 06.10 13 kbit/s RPE/LTP speech compression available
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The Communications and Operating Systems Research Group (KBS) at the
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Technische Universitaet Berlin is currently working on a set of
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UNIX-based tools for computer-mediated telecooperation that will be
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made freely available.
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As part of this effort we are publishing an implementation of the
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European GSM 06.10 provisional standard for full-rate speech
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transcoding, prI-ETS 300 036, which uses RPE/LTP (residual pulse
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excitation/long term prediction) coding at 13 kbit/s.
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GSM 06.10 compresses frames of 160 13-bit samples (8 kHz sampling
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rate, i.e. a frame rate of 50 Hz) into 260 bits; for compatibility
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with typical UNIX applications, our implementation turns frames of 160
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16-bit linear samples into 33-byte frames (1650 Bytes/s).
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The quality of the algorithm is good enough for reliable speaker
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recognition; even music often survives transcoding in recognizable
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form (given the bandwidth limitations of 8 kHz sampling rate).
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The interfaces offered are a front end modelled after compress(1), and
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a library API. Compression and decompression run faster than realtime
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on most SPARCstations. The implementation has been verified against the
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ETSI standard test patterns.
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Jutta Degener (jutta@cs.tu-berlin.de)
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Carsten Bormann (cabo@cs.tu-berlin.de)
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Communications and Operating Systems Research Group, TU Berlin
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Fax: +49.30.31425156, Phone: +49.30.31424315
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--
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Copyright 1992 by Jutta Degener and Carsten Bormann, Technische
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Universitaet Berlin. See the accompanying file "COPYRIGHT" for
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details. THERE IS ABSOLUTELY NO WARRANTY FOR THIS SOFTWARE.
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