bionic (1) freedv.1.gz

Provided by: freedv_1.2.2-3_amd64 bug

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