パッケージ: libgsm-tools (1.0.13-3)

User binaries for a GSM speech compressor

This package contains user binaries for libgsm, an implementation of the European GSM 06.10 provisional standard for full-rate speech transcoding, prI-ETS 300 036, which uses RPE/LTP (residual pulse excitation/long term prediction) coding at 13 kbit/s.

GSM 06.10 compresses frames of 160 13-bit samples (8 kHz sampling rate, i.e. a frame rate of 50 Hz) into 260 bits; for compatibility with typical UNIX applications, our implementation turns frames of 160 16-bit linear samples into 33-byte frames (1650 Bytes/s). The quality of the algorithm is good enough for reliable speaker recognition; even music often survives transcoding in recognizable form (given the bandwidth limitations of 8 kHz sampling rate).

The interfaces offered are a front end modelled after compress(1), and a library API. Compression and decompression run faster than realtime on most SPARCstations. The implementation has been verified against the ETSI standard test patterns.

タグ: アクセシビリティサポート: 音声合成, ソフトウェア開発: ライブラリ, ユーザインタフェース: コマンドライン, 役割: 開発ライブラリ, プログラム, 対象範囲: ユーティリティ, 音声・音楽: 圧縮, 音声合成