GAPK (GSM Audio Pocket Knife)¶

• Table of contents
• GAPK (GSM Audio Pocket Knife)
  • Supported Codecs
  • Supported Formats
  • Common use cases
    • RTP sink with audio playback on sound card
  • Source Code
  • Contact / Mailing List
  • Authors

The GAPK (GSM Audio Pocket Knife) is a project, the product of which is a core library (libosmogapk) and a command-line tool (osmo-gapk) for conversions 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.

Supported Codecs¶

Codec	Description	Implementation
pcm	Raw PCM signed 16bits samples	native
hr	GSM 06.20 Half Rate codec	libgsmhr* (GSM 06.06 reference code)
fr	GSM 06.10 Full Rate codec (classic gsm codec)	libgsm
efr	GSM 06.60 Enhanced Full Rate codec	opencore-amr
amr	GSM 26.061 Adaptive Multi Rate codec	opencore-amr

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

Common use cases¶

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.

$ gapk -I 0.0.0.0/30000 -f rtp-amr -A default -g rawpcm-s16le

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)

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

Contact / Mailing List¶

The project is too small to have it's own mailing list. Instead, we use the openbsc@lists.osmocom.org mailing list (subscribe). Please observe the Mailing_List_Rules.

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.