Several Gstreamer scripts are used to perform REMOTE QRQ CW KEYING of a RIG(afcw mode) from the REMOTE OP's location using TCP audio over ip network packets through a router and network switches in the pathway(WIRED ETHERNET)
there are 2 Gstreamer BANDPASS filters applied before audio goes to rig for transmit:
1. 680 hz BPF Gstreamer code
gst-launch-1.0 jackaudiosrc ! "audio/x-raw,channels=1,rate=48000,format=F32LE" ! audioconvert ! audiowsincband mode=band-pass lower-frequency=610 upper-frequency=750 length=700 window=hamming ! audioconvert ! audioamplify amplification=1 ! audiorate ! jackaudiosink buffer-time=15000
2. 5555 hz BPF Gstreamer code
gst-launch-1.0 jackaudiosrc ! "audio/x-raw,channels=1,rate=48000,format=F32LE" ! audioconvert ! audiowsincband mode=band-pass lower-frequency=5000 upper-frequency=6000 length=700 window=hamming ! audioconvert ! audioamplify amplification=1 ! audiorate ! jackaudiosink buffer-time=15000
There are 2 sets of 1 channel RTSP SERVER scripts:
1. to send audio from PI::RIG interface to remote OP
./test-launch "(jackaudiosrc ! queue ! "audio/x-raw,channels=1,format=F32LE,rate=48000,layout=interleaved,payload=96" ! audioconvert ! rtpgstpay name=pay0 pt=96 )"
2. to send transmit audio from REMOTE OP to the PI::RIG interface
./test-launch "(jackaudiosrc ! "audio/x-raw,channels=1,format=F32LE,rate=48000,layout=interleaved,payload=96" ! audioconvert ! rtpgstpay name=pay0 pt=96 )"
There are 2 sets of 1 channel RTSP CLIENT scripts:
1. remote OP receives audio from the PI::RIG interface
NOTE: this script also contains the Gstreamer code for a BandReject filter to block out the 5555 hz PTT ACTIVATION TONE from the RIG's transmit input audio CW sidetone monitor
gst-launch-1.0 rtspsrc latency=0 location=rtsp://192.168.1.130:8554/test protocols=tcp ! rtpjitterbuffer latency=15 ! rtpgstdepay ! audioconvert ! audiochebband mode=1 lower-frequency=3000 upper-frequency=9000 poles=13 ! queue ! jackaudiosink buffer-time=16000
2. PI::RIG interface receives audio from remote OP for transmit:
gst-launch-1.0 rtspsrc latency=0 location=rtsp://192.168.1.101:8554/test protocols=tcp ! rtpjitterbuffer latency=15 ! rtpgstdepay ! audioconvert ! jackaudiosink buffer-time=16000
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