Gapk » History » Revision 2
Revision 1 (laforge, 05/29/2017 09:12 AM) → Revision 2/18 (laforge, 05/29/2017 09:16 AM)
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h1. gapk
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:]]) [[OsmocomBB]]) and those of Racal 6103/6113 GSM test equipment.
h2. Supported Codecs
|_.Codec|_.Description|
|pcm|Raw PCM signed 16bits samples|
|hr|GSM 06.20 Half Rate codec|
|fr|GSM 06.10 Full Rate codec (classic gsm codec)|
|efr|GSM 06.60 Enhanced Full Rate codec|
|amr|GSM 26.061 Adaptive Multi Rate codec|
h2. Supported Formats
|_.Format|_.Description|
|amr-efr |Classic .amr file containing EFR (=AMR 12.2k) data|
|gsm |Classic .gsm file format (and RTP payload for FR according to RFC3551)|
|hr-ref-dec |3GPP HR Reference decoder code parameters file format|
|hr-ref-enc |3GPP HR Reference encoder code parameters file format|
|racal-hr |Racal HR TCH/H recording|
|racal-fr |Racal FR TCH/F recording|
|racal-efr |Racal EFR TCH/F recording|
|rawpcm-s16le |Raw PCM samples Signed 16 bits little endian|
|ti-hr |Texas Instrument HR TCH/H buffer format|
|ti-fr |Texas Instrument FR TCH/F buffer format|
|ti-efr |Texas Instrument EFR TCH/F buffer format|
|amr-opencore |Input format to libopencore-amrnb|
|rtp-amr |RTP payload for AMR according to RFC4867|
|rtp-efr |RTP payload for EFR according to RFC3551|
|rtp-hr-etsi |RTP payload for HR according to ETSI TS 101 318|
|rtp-hr-ietf |RTP payload for HR according to IETF RFC5993|
h2. Common use cases
h3. RTP sink with audio playback on sound card
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.
<pre>
$ gapk -I 0.0.0.0/30000 -f rtp-amr -A default -g rawpcm-s16le
</pre>
where
* *@-I 0.0.0.0/30000@* indicates the IP adddress (any) and UDP port (30000) to bind to and receive RTP frames on
* *@-f rtp-amr@* indicates the codec. Use *@gsm, rtp-efr, rtp-amr, rtp-hr-etsi or rtp-hr-ietf@* depending on your use case
* *@-A default@* is the alsa hardware device name (default is the default sound card)
h2. Source Code
You can find the source code in git:
* http://git.osmocom.org/gapk/ (cgit web interface)
* @git clone git://git.osmocom.org/gapk@ for cloning the repository
h2. Authors
@gapk@ as written by Sylvain Munaut with contributions from Harald Welte.
It uses external libraries for the actual audio codecs, such as @libgsm@, @libopencore-amrnb@ nd the ETSI reference implementation for GSM-HR.
