套件：libgsm-tools（1.0.22-1 以及其他的）

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, this 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.

標籤: 無障礙支持: 語音合成, 實做語言: C, 使用者介面: interface::commandline, protocol::voip, 角色: 程式, 範圍: 實用程式, 聲音與音樂: sound::compression, sound::player, 語音合成, Purpose: Playing Media, 支援的格式: works-with-format::TODO, works-with::audio

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