NAME

       freedv - Digital Voice for HF

DESCRIPTION

       FreeDV  is a GUI application that allows any SSB radio to be used for low bit rate digital
       voice.

       Speech is compressed down to 700-1600 bit/s then modulated onto a  1.25  kHz  wide  signal
       comprised of 16 QPSK carriers which is sent to the Mic input of a SSB radio. The signal is
       received by an SSB  radio,  then  demodulated  and  decoded  by  FreeDV.  FreeDV  700C  is
       approaching SSB in it's low SNR performance. At high SNRs FreeDV 1600 sounds like FM, with
       no annoying analog HF radio noise.

       FreeDV was built by an international team of Radio Amateurs working  together  on  coding,
       design, user interface and testing. FreeDV is open source software, released under the GNU
       Lesser General Public License version 2.1. The FDMDV modem and Codec 2 Speech  codec  used
       in FreeDV are also open source.

Why FreeDV?

       Amateur  Radio is transitioning from analog to digital, much as it transitioned from AM to
       SSB in the 1950s and 1960s. How would you feel if one or two companies owned  the  patents
       for  SSB,  then  forced you to use their technology, made it illegal to experiment with or
       even understand the technology, and insisted you stay  locked  to  it  for  the  next  100
       years??  That  is  exactly  what  was  happening  with digital voice. But now, hams are in
       control of their technology again.

       FreeDV is unique as it uses 100 percent Open Source Software, including the  audio  codec.
       No  secrets,  nothing  proprietary FreeDV represents a path for 21st century Amateur Radio
       where Hams are free to experiment and innovate, rather than a future locked into a  single
       manufacturers closed technology.

Demo Video

       Watch this video of a FreeDV QSO.

       http://freedv.org/tiki-index.php?page=video

       Here is what you need:

           A SSB receiver or transceiver
           FreeDV software
           A computer with one (receive only) or two sound cards.
           Cables to connect your computer to your SSB radio.

Test your Transmitter Frequency Response

       When  you  play  this 10 second 1 kHz to 2 kHz sweep .wav file(external link) through your
       transmitter, the power level should remain  constant.  If  not,  look  for  filtering  and
       processing to turn off.

Connecting Your Radio

       If  you are lucky enough to have a "9600" input and output on your radio, this is the best
       connection for every digital mode,  even  1200  packet,  and  your  audio  box  should  be
       configured  for  9600  or  "no  pre-emphasis/de-emphasis"  if  it has that setting. If the
       radio's configuration menu has a 1200/9600 setting, leave it permanently on 9600.

       The "9600" and "1200" settings are  misnamed.  "9600"  should  really  be  called  "direct
       connection",  and  "1200" should be called "processed". The audio processing in your radio
       does not help any digital mode.

Configuring Your Radio

       Turn off as much processing as  possible.  In  general  noise  blankers,  DSP  band  limit
       filtering,  and  narrow bandpass filters are likely to hurt rather than help. Compression,
       DSP noise and carrier elimination, and voice processing are definitely wrong  for  Digital
       modes.  FreeDV's FDM modem does its own DSP, and in general this is true for other digital
       programs as well. The only things that we would expect to hurt the signal are intrusion of
       the  opposite sideband, images of out-of-passband signals, and intermodulation distortion.
       You can see the effect of different settings in the S/N display of FreeDV.

       Drive your transmitter and amplifier so that it emits 10%% to  20%%  of  its  rated  power
       continuously.  There  is  a  12  dB peak-to-average power ratio in the FDM modem, and peak
       clipping in  your  amplifier  will  reduce  the  received  S/N.  Modern  transmitters  and
       amplifiers  are  only as linear, and only have as much headroom, as is necessary for voice
       SSB. Ask manufacturers and reviewers to start rating linearity and  headroom  for  digital
       modes.

PTT Configuration

       Tools-PTT Dialog

       Hamlib  comes  with  a  default serial rate for each radio.  If your radio has a different
       serial rate change the Serial Rate drop down box to match your radio.

       When "Test" is pressed, the "Serial Params" field is populated and displayed.   This  will
       help track down any mis-matches between Hamlib and your radio.

       Serial PTT support is complex.  We get many reports that FreeDV Hamlib PTT doesn't work on
       a particular radio, but may work fine with other programs such as Fldigi.  This is  always
       a  mis-match between the serial parameters Hamlib is using with FreeDV and your radio. For
       example you may have changed the default serial rate on your radio.  Carefully  check  the
       serial parameters on your radio match those used by FreeDV in the PTT Dialog.

       If  you  are  really stuck, download Hamlib (Debian package libhamlib-utils) and test your
       radio's PTT using the command line rigctl program.

Voice Keyer

       Voice Keyer Button on Front Page Options-PTT Dialog

       Puts FreeDV and your radio into transmit, reads a wave file of your voice to call CQ, then
       switches  to receive to see if anyone is replying.  If you press space bar the voice keyer
       stops.  If a signal with a valid sync is received for a few seconds the voice keyer stops.

       Options-PTT dialog can be used to select the wave file, set the Rx delay,  and  number  of
       times the tx/rx cycle repeats.

       The  wave  file for the voice keyer should be in 8kHz mono 16 bit sample form.  Use a free
       application such as Audacity to convert a file you have recorded to this format.

Test Frame Histogram

       Test Frame Histogram tab on Front Page

       Displays BER of each carrier when in "test frame" mode.  As each QPSK carrier has  2  bits
       there are 2*Nc histogram points.

       Ideally  all  carriers will have about the same BER (+/- 20% after 5000 total bit errors).
       However problems can occur with filtering in the tx path.  If one carrier has less  power,
       then  it will have a higher BER.  The errors in this carrier will tend to dominate overall
       BER. For example if one carrier is attenuated due to SSB filter ripple in the tx path then
       the BER on that carrier will be higher.  This is bad news for DV.

       Suggested usage:

       i)  Transmit FreeDV in test frame mode.  Use a 2nd rx (or get a friend) to monitor your rx
       signal with FreeDV in test frame mode.

       ii) Adjust your rx SNR to get a BER of a few % (e.g. reduce tx power, use a short  antenna
       for the rx, point your beam away, adjust rx RF gain).

       iii)  Monitor  the  error  histogram  for a few minutes, until you have say 5000 total bit
       errors.  You have a problem if the BER of any carrier is more than 20% different from  the
       rest.

       A  typical  issue  will  be  one  carrier at 1.0, the others at 0.5, indicating the poorer
       carrier BER is twice the larger.

Full Duplex Testing with loopback

       Options - Half Duplex check box

       FreeDV GUI can operate in full duplex mode which is useful for development of listening to
       your own FreeDV signal as only one PC is required.  Normal operation is half duplex.

       Tx and Rx signals can be looped back via an analog connection between the sound cards.

       On Linux, using the Alsa loopback module:

         $ sudo modprobe snd-aloop
           $ ./freedv

         In  Tools  -  Audio  Config - Receive Tab  - From Radio select -> Loopback: Loopback PCM
       (hw:1,0)
                                   - Transmit Tab - To Radio select   -> Loopback:  Loopback  PCM
       (hw:1,1)

Design & Key Features

       Design:

        Codec 2 voice codec and FDMDV/COHPSK modems
        1.25 kHz spectrum bandwidth (half SSB) with 75 Hz carrier spacing
        FreeDV 1600 mode: 1275 bit/s voice coding, 25 bit/s text
           for call sign ID, 300 bit/s FEC, 16x50 baud DQPSK carriers,
           Differential QPSK demodulation
        FreeDV 700(C) mode: 700 bit/s voice coding, no FEC, 14x75
           baud QPSK carriers, frequency diversity to combat fading,
           coherent QPSK demodulation
        No interleaving in time, resulting in low latency, fast
           synchronization and quick recovery from fades.
        44.1 or 48kHz sample rate sound card compatible

       Key Features:

        Cross platform, runs on Linux and Windows.
        Open source, patent free Codec and Modem that anyone can
           experiment with and modify Waterfall, spectrum, scatter and
           audio oscilloscope displays.
        Adjustable squelch
        Fast/slow SNR estimation
        Microphone and Speaker signal audio Equaliser
        Control of Transmitter PTT via RS232 levels
        Works with one (receive only) or two
           (transmit and receive) sound cards, for example a built in
           sound card and USB headphones.

Credits

       FreeDV  is  being  maintained  and  extended  by  David  Rowe, VK5DGR. Richard Shaw KF5OIM
       maintains the Cmake build system, Windows and Fedora packaging. Walter,  K5WH  is  leading
       Windows testing in the USA.

       As  development  continues,  many  people  are  helping  whom we have not credited, but we
       appreciate all of their work.

       This manual page was written by Maitland Bottoms for the Debian project (but may  be  used
       by others).

History

       In  2012  FreeDV was coded from scratch by David Witten (GUI, architecture) and David Rowe
       (Codec 2, modem implementation, integration).

       The FreeDV design and user interface is based on FDMDV, which was developed  by  Francesco
       Lanza,  HB9TLK.  Francesco  received advice on modem design from Peter Martinez G3PLX, who
       has also advised David on the FDMDV modem used in FreeDV.

       Mel Whitten, K0PFX has contributed greatly to the design, testing and promotion of several
       Digital  Voice  systems, including FDMDV. This practical experience has led to the current
       design – a fast sync, no FEC, low  latency  system  that  gives  a  “SSB”  type  feel  for
       operators.  Mel and a team of alpha testers (Gerry, N4DVR; Jim, K3DCC; Rick, WA6NUT; Tony,
       K2MO) provided feedback on usability and design of FreeDV.

       Bruce Perens has been a thought leader on  open  source,  patent  free  voice  codecs  for
       Amateur  Radio.  He  has  inspired, promoted and encouraged the development of Codec 2 and
       FreeDV.

SEE ALSO

       http://freedv.org/

       For casual chat there is a #freedv IRC channel on freenode.net