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laforge, 05/29/2017 09:16 AM
fix link to osmocomBB

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h1. gapk
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The *G* SM *A* udio *P* ocket *K* nive (GAPK) is a small command-line tool that supports conversion between the various GSM related codecs (HR/FR/EFR/AMR) and PCM audio. It supports many different formats for the codec frames, including ETSI and IETF standardized formats, as well as vendor-specific formats like those found in the TI Calypso DSP (see [[OsmocomBB:]]) and those of Racal 6103/6113 GSM test equipment.
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h2. Supported Codecs
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|_.Codec|_.Description|
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|pcm|Raw PCM signed 16bits samples|
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|hr|GSM 06.20 Half Rate codec|
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|fr|GSM 06.10 Full Rate codec (classic gsm codec)|
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|efr|GSM 06.60 Enhanced Full Rate codec|
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|amr|GSM 26.061 Adaptive Multi Rate codec|
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h2. Supported Formats
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|_.Format|_.Description|
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|amr-efr             |Classic .amr file containing EFR (=AMR 12.2k) data|
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|gsm                 |Classic .gsm file format (and RTP payload for FR according to RFC3551)|
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|hr-ref-dec          |3GPP HR Reference decoder code parameters file format|
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|hr-ref-enc          |3GPP HR Reference encoder code parameters file format|
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|racal-hr            |Racal HR TCH/H recording|
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|racal-fr            |Racal FR TCH/F recording|
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|racal-efr           |Racal EFR TCH/F recording|
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|rawpcm-s16le        |Raw PCM samples Signed 16 bits little endian|
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|ti-hr               |Texas Instrument HR TCH/H buffer format|
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|ti-fr               |Texas Instrument FR TCH/F buffer format|
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|ti-efr              |Texas Instrument EFR TCH/F buffer format|
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|amr-opencore        |Input format to libopencore-amrnb|
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|rtp-amr             |RTP payload for AMR according to RFC4867|
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|rtp-efr             |RTP payload for EFR according to RFC3551|
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|rtp-hr-etsi         |RTP payload for HR according to ETSI TS 101 318|
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|rtp-hr-ietf         |RTP payload for HR according to IETF RFC5993|
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h2. Common use cases
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h3. RTP sink with audio playback on sound card
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You can run @gapk@ as a _RTP sink_, i.e. listening to a given UDP port for incoming RTP frames, decoding them from their respective audio codec and then playing them back via your computers' sound card.
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<pre>
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$ gapk -I 0.0.0.0/30000 -f rtp-amr -A default -g rawpcm-s16le
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</pre>
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where
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* *@-I 0.0.0.0/30000@* indicates the IP adddress (any) and UDP port (30000) to bind to and receive RTP frames on
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* *@-f rtp-amr@* indicates the codec. Use *@gsm, rtp-efr, rtp-amr, rtp-hr-etsi or rtp-hr-ietf@* depending on your use case
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* *@-A default@* is the alsa hardware device name (default is the default sound card)
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h2. Source Code
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You can find the source code in git:
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* http://git.osmocom.org/gapk/ (cgit web interface)
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* @git clone git://git.osmocom.org/gapk@ for cloning the repository
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h2. Authors
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@gapk@ as written by Sylvain Munaut with contributions from Harald Welte. 
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It uses external libraries for the actual audio codecs, such as @libgsm@, @libopencore-amrnb@ nd the ETSI reference implementation for GSM-HR.
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