FreeDV GUI (or just FreeDV) is a GUI program for Linux, Windows, and OSX for running FreeDV on a desktop PC or laptop.
This is a live document. Notes on new FreeDV features are being added as they are developed.
FreeDV GUI can be challenging to set up. The easiest way is to find a friend who has set up FreeDV and get them to help. This section contains instructions to help you get started.
For Receive only operation you just need one sound card; this is a great way to get started.
For Tx/Rx operation you need two sound cards. One connects to your radio, and one for the operator. The sound card connecting to the radio can be a rig interface device like a Signalink, RIGblaster, your radio's internal USB sound card, or a home brew rig interface.
The second sound card is often a set of USB headphones or your computer's internal sound card.
Start with just a receive only station. You just need the basic sound hardware in your computer, for example a microphone/speaker on your computer.
When you press Start FreeDV will start decoding any incoming signals on the microphone input, playing the decoded audio out of your speaker. If no valid FreeDV signals are received, no audio will be played.
If you connect the microphone input on your computer to your radio receiver, you can decode off air signals. If you have a rig interface, try configuring that as the From Radio To Computer device, with your computer's sound card as the From Computer To Speaker/Headphone device.
If you don't have anyone to transmit FreeDV signals to you, try the test wave files in the next section.
In the wav directory are audio files containing off-air FreeDV modem signals. There is one for each FreeDV mode. Select a FreeDV mode and press Start. Choose a file using "Tools - Start/Stop Play File From Radio". You should hear decoded FreeDV speech.
These files will give you a feel for what FreeDV signals sound like, and for the basic operation of the FreeDV software.
For Tx/Rx operation you need to configure two sound cards, by setting up Tools - Audio Config Transmit and Receive Tabs.
When receiving, FreeDV off-air signals from your radio are decoded by your computer and sent to your speaker/headphones, where you can listen to them.
When transmitting, FreeDV takes your voice from the microphone, and encodes it to a FreeDV signal in you computer which is sent to your radio for transmission over the air.
Tab | Sound Device | Notes |
---|---|---|
Receive Tab | Input To Computer From Radio | The off air FreeDV signal from your radio rig interface to your computer |
Receive Tab | Output From Computer To Speaker/Headphones | The decoded audio from your computer to your Speaker/Headphones |
Transmit Tab | Input From Microphone To Computer | Your voice from the microphone to your computer |
Transmit Tab | Output From Computer To Radio | The FreeDV signal from your computer sent to your rig interface for Tx |
If you change audio devices (e.g. add or remove sound cards, USB hardware), it's a good idea to check the Tools/Audio Config dialog before pressing Start, to verify the audio devices are as expected. This is particularly important if any audio devices e.g. Headsets, USB Sound Cards, or Virtual Cables have been disconnected since the last time FreeDV was used.
Hitting Refresh in the lower left hand corner of the Tools/Audio Config will normally update the audio devices list. Keeping a screen shot of a known working configuration will be useful for new users. Unexpected audio configuration changes may also occur following a Windows updates.
Another solution is to re-start FreeDV and check Tools/Audio Config again after changing any audio hardware.
If you change/remove USB audio devices without refreshing Tools/Audio Config, FreeDV may crash.
Sound card levels are generally adjusted in the computer's Control Panel or Settings, or in some cases via controls on your rig interface hardware or menus on your radio.
When FreeDV is running, you can observe the sound card signals in the main window tabs (From Radio, From Mic, To Speaker).
On receive, FreeDV is not very sensitive to the From Radio level, adjust so it is mid-range and not clipping. FreeDV uses phase shift keying (PSK) so is not sensitive to amplitude.
The transmit level from your computer to your radio is important. On transmit, adjust your level so that the ALC is just being nudged. More is not better with the FreeDV transmit signal. Overdriving your transmitter will lead to a distorted transit signal, and a poor SNR at the receiver. This is a very common problem.
FreeDV 700D and 700E can drive your transmitter at an average power of 40% of it's peak power rating. For example 40W RMS for a 100W PEP radio. Make sure your transmitter can handle continuous power output at these levels, and reduce the power if necessary.
Adjust the microphone audio so the peaks are not clipping, and the average is about half the maximum.
FreeDV likes a clean path through your radio. Turn all audio processing OFF on transmit and receive:
On receive, DSP noise reduction should be off.
On transmit, speech compression should be off.
Keep the receive audio path as "flat" as possible, no special filters.
FreeDV will not work any better if you band pass filter the off air received signals. It has its own, very tight filters in the demodulator.
The Tools - PTT dialog supports three different ways to control PTT on your radio:
Once you have configured PTT, try the Test button.
Serial PTT support is complex. We get many reports that FreeDV PTT doesn't work on a particular radio, but may work fine with other programs such as Fldigi. This is often 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.
Also see Common Problems section of this manual.
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 mismatches between Hamlib and your radio.
If you are really stuck, download Hamlib and test your radio's PTT using the command line rigctl
program.
If using an Icom radio, Hamlib will use the radio's default CI-V address when connecting. If this has been changed, you can specify the correct address in the "Radio Address" field (valid values are 00 through FF in hexadecimal).
Note that "00" is the "wildcard" CI-V address. Your radio must have the "CI-V Transceive" option enabled in order for it to respond to commands to that address. Otherwise, FreeDV must be configured to use the same CI-V address as configured in the radio. For best results, ensure that there are no other Icom/CI-V capable devices in the chain if "00"/"CI-V Transceive" is used.
If you change the COM port of a USB-Serial device in Device Manager, please unplug and plug back in the USB device. Windows/FreeDV won't recognise the device on the new COM Port until it has been unplugged/plugged.
On bands below 10 MHz, LSB is used for FreeDV. On 10MHz and above, USB is used. After much debate, the FreeDV community has adopted the same conventions as SSB, based on the reasoning that FreeDV is a voice mode.
As an aid to the above, FreeDV will show the current mode on the bottom of the window upon pressing the Start button if Hamlib is enabled and your radio supports retrieving frequency and mode information over CAT. If your radio is using an unexpected mode (e.g. LSB on 20 meters), it will display that mode on the bottom of the window next to the Clear button in red letters. When a session is not active, Hamlib isn't enabled, or if your radio doesn't support retrieving frequency and mode over CAT, it will remain grayed out with "unk" displaying instead of the mode (for "unknown").
This is a very common problem for first time FreeDV users. Adjust your transmit levels so the ALC is just being nudged. More power is not better with FreeDV. An overdriven signal will have poor SNR at the receiver. For FreeDV 700D/700E operation with the clipper, make sure your transmitter can sustain high average power levels without damage (e.g. 40W RMS on a 100W PEP radio).
This can be challenging the first time around:
Try a receive only (one audio card) set up first.
Ask someone who already runs FreeDV for help.
If you don't know anyone local, ask for help on the digital voice mailing list. Be specific about the hardware you have and the exact nature of your problem.
The most common issue with Icom radios is that the CI-V address configured in FreeDV does not match the address configured in the radio. Ensure that the CI-V address in both FreeDV and on the radio are the same. If "00" is used on the FreeDV side, ensure that the "CI-V Transceive" option is enabled on the radio or else the radio will not respond to requests directed to that address.
There are many radios, many computers, and many sound cards. It is impossible to test them all. Many radios have intricate menus with custom settings. It is unreasonable to expect the authors of FreeDV to have special knowledge of your exact hardware.
However someone may have worked through the same problem as you. Ask on the digital voice mailing list.
Many FreeDV modes will not play any audio if there is no valid signal. You may also have squelch set too high. In some modes the Analog button will let you hear the received signal from the SSB radio.
Try the Test Wave Files above to get a feel for what a FreeDV signal looks and sounds like.
You need to be within +/- 60 Hz on the transmit signal. It helps if both the Tx and Rx stations tune to known, exact frequencies such as exactly 7.177MHz. On channels with fast fading sync may take a few seconds.
Many people struggle with initial PTT setup:
Read the PTT Configuration section above.
Try the Tools - PTT Test function.
Check your rig serial settings. Did you change them from defaults for another program?
Linux version: do you have permissions for the serial port? Are you a member of the dialout
group?
Ask someone who already uses FreeDV to help.
Contact the digital voice mailing list. Be specific about your hardware, what you have tried, and the exact nature of the problem.
You must have a modern CPU with AVX support to run FreeDV 2020. If you do not have AVX the FreeDV 2020 mode button will be greyed out. A Microsoft utlity called coreinfo can be also used to determine if your CPU supports AVX. A * means you have AVX, a - means no AVX:
AES - Supports AES extensions
AVX * Supports AVX intruction extensions
FMA - Supports FMA extensions using YMM state
On Linux, you can check for avx
in the flags section of /proc/cpuinfo
or the output of the lscpu
command:
lscpu | grep -o "avx[^ ]*"
will display avx
(or avx2
) if your CPU supports the instructions.
Preliminary testing on ARM Macs has shown that NEON optimizations in LPCNet are sufficient to allow 2020 to be whitelisted on those machines. However, this is definitely experimental. If you are experiencing issues with 2020 mode on these Macs, please let the development team know so that further investigation can be done.
You may need to clean out the previous configuration. Try Tools-Restore Defaults.
Have you removed/changed USB audio devices? If you remove/change USB audio devices without pressing Tools/Audio Config, FreeDV may crash. See Changing Audio Devices above.
From January 2020 Apple is enforcing notarization for all OSX applications. The FreeDV developers do not wish to operate within the Apple ecosystem due to the cost/intrusiveness of this requirement.
notarization
Security & Privacy shows the Open Anyway option for FreeDV:
notarization
notarization
Or you can use command line options:
xattr -d com.apple.quarantine FreeDV.app
or
xattr -d -r com.apple.quarantine FreeDV.app
The Voice Keyer Button on the front page, and the Options-PTT dialog puts FreeDV and your radio into transmit, reads a wave file of your voice to call CQ, and then switches to receive to see if anyone is replying. If you press the space bar the voice keyer stops. If a signal with a valid sync is received for a few seconds the voice keyer stops.
The 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 (16 kHz for 2020). Use a free application such as Audacity to convert a file you have recorded to this format.
FreeDV has the ability to send FreeDV signal reports to PSK Reporter by enabling the option in Tools-Options and specifying your callsign and grid square. When enabled, this causes FreeDV to disable the free form Txt Msg field and only transmit the Callsign field.
FreeDV validates the received information before submitting a position report to PSK Reporter. This is to ensure that FreeDV does not report invalid callsigns to the service (e.g. ones that don't exist or that correspond to real non-FreeDV users). However, all received text will display in the main window even if it has errors.
Reports sent to PSK Reporter will display using the mode "FREEDV" for ease of filtering. The user's current mode (e.g. 700D, 1600, etc.) will also appear in the "Using" field when hovering over or clicking on a reception report.
Note that Hamlib must be enabled so PSK Reporter can read your radio's frequency. A message will appear on pushing Start if this is not the case.
The following table is a guide to the different modes, using analog SSB and Skype as anchors for a rough guide to audio quality:
Mode | Min SNR | Fading | Latency | Speech Bandwidth | Speech Quality |
---|---|---|---|---|---|
SSB | 0 | 8/10 | low | 2600 | 5/10 |
1600 | 4 | 3/10 | low | 4000 | 4/10 |
700C | 2 | 6/10 | low | 4000 | 3/10 |
700D | -2 | 4/10 | high | 4000 | 3/10 |
700E | 1 | 7/10 | medium | 4000 | 3/10 |
2020 | 4 | 4/10 | high | 8000 | 7/10 |
Skype | - | - | medium | 8000 | 8/10 |
The Min SNR is roughly the SNR where you cannot converse without repeating yourself. The numbers above are on channels without fading (AWGN channels like VHF radio). For fading channels the minimum SNR is a few dB higher. The Fading column shows how robust the mode is to HF Fading channels, higher is more robust.
The more advanced 700D and 2020 modes have a high latency due to the use of large Forward Error Correction (FEC) codes. They buffer many frames of speech, which combined with PC sound card buffering results in end-to-end latencies of 1-2 seconds. They may take a few seconds to sync at the start of an over, especially in fading channels.
In mid 2018 FreeDV 700D was released, with a new OFDM modem, powerful Forward Error Correction (FEC) and optional interleaving. It uses the same 700 bit/s speech codec at 700C. It operates at SNRs as low as -2dB, and has good HF channel performance. It is around 10dB better than FreeDV 1600 on fading channels, and is competitive with SSB at low SNRs. The FEC provides some protection from urban HF noise.
FreeDV 700D is sensitive to tuning. To obtain sync you must be within +/- 60Hz of the transmit frequency. This is straightforward with modern radios which are generally accurate to +/-1 Hz, but requires skill and practice when used with older, VFO based radios.
FreeDV 700E was developed in December 2020 using lessons learned from on air operation of 700C and 700D. A variant of 700D, it uses a shorter frame size (80ms) to reduce latency and sync time. It is optimised for fast fading channels channels with up to 4Hz Doppler spread and 6ms delay spread. FreeDV 7000E uses the same 700 bit/s codec as FreeDV 700C and 700D. It requires about 3dB more power than 700D, but can operate reliably on fast fading channels.
The 700E release also includes optional compression (clipping) of the 700D and 700E transmit waveforms to reduce the Peak to Average Power Ratio to about 4dB. For example a 100W PEP transmitter can be driven to about 40W RMS. This is an improvement of 6dB over previous releases of FreeDV 700D. Before enabling the clipper make sure your transmitter is capable of handling sustained high average power without damage.
Clipping can be enabled via Tools-Options.
On good channels with high SNR clipping may actually reduce the SNR of the received signal. This is intentional - we are adding some pre-distortion in order to increase the RMS power. Forward error correction (FEC) will clean up any errors introduced by clipping, and on poor channels the benefits of increased signal power outweigh the slight reduction in SNR on good channels.
FreeDV 2020 was developed in 2019. It uses an experimental codec based on the LPCNet neural net (deep learning) synthesis engine developed by Jean-Marc Valin. It offers 8 kHz audio bandwidth in an RF bandwidth of just 1600 Hz. FreeDV 2020 employs the same OFDM modem and FEC as 700D.
The purpose of FreeDV 2020 is to test neural net speech coding over HF radio. It is highly experimental, and possibly the first use of neural net vocoders in a real world, over the air system.
FreeDV 2020 is designed for slow fading HF channels with a SNR of 10dB or better. It is not designed for fast fading or very low SNRs like 700D. It is designed to be a high quality alternative to SSB in channels where SSB is already an "arm-chair" copy. On an AWGN (non- fading channel), it will deliver reasonable speech quality down to 2dB SNR.
FreeDV 2020 Tips:
This section describes features on Tools-Filter.
Control | Description |
---|---|
Noise Supression | Enable noise supression, dereverberation, AGC of mic signal using the Speex pre-processor |
700C/700D Auto EQ | Automatic equalisation for FreeDV 700C and FreeDV 700D Codec input audio |
Auto EQ (Automatic Equalisation) adjusts the input speech spectrum to best fit the speech codec. It can remove annoying bass artefacts and make the codec speech easier to understand.
Control | Description |
---|---|
Clipping | Increases the average power. Ensure your transmitter can handle high RMS powers before using! |
700C Diversity Combine | Combining of two sets of 700C carriers for better fading channel performance |
Tx Band Pass Filter | Reduces Tx spectrum bandwidth |
Manual Unsync | Forces modem to remain in sync, and not drop sync automatically |
These options apply to the FreeDV 700D and 2020 modes that use the OFDM modem:
The High Bandwidth option gives better performance on channels where the phase changes quickly, for example fast fading HF channels and the Es'Hail 2 satellite. When unchecked, the phase estimator bandwidth is automatically selected. It starts off high to enable fast sync, then switches to low bandwidth to optimise performance for low SNR HF channels.
The DPSK (differential PSK) checkbox has a similar effect - better performance on High SNR channels where the phase changes rapidly. This option converts the OFDM modem to use differential PSK rather than coherent PSK. DPSK is used by earlier FreeDV modes such as FreeDV 1600. It affects the Tx and Rx side, so both sides must select DPSK.
If you have problems with 700D or 2020 sync even though you have a strong signal - try these options.
If you have an interesting test case, for example:
Please send the developers an off air recording of the signal. FreeDV can record files from your radio using Tools-Record File from Radio. A recording of 30 to 60 seconds is most useful.
With a recording we can reproduce your exact problem. If we can reproduce it we can fix it. Recordings are much more useful than anecdotes or subjective reports like "FreeDV doesn't work", "SSB is better", or "On 23 December it didn't work well on grid location XYZ". With subjective reports problems are impossible to reproduce, cannot be fixed, and you are unlikely to get the attention of the developers.
Located on the lower left hand side of the main screen.
Term | Notes |
---|---|
Bits | Number of bits demodulated |
Errs | Number of bit errors detected |
Resyncs | Number of times the demodulator has resynced |
ClkOff | Estimated sample clock offset in parts per million |
FreqOff | Estimated frequency offset in Hz |
Sync | Sync metric (OFDM modes like 700D and 2020) |
Var | Speech encoder distortion for 700C/700D (see Auto EQ) |
The sample clock offset is the estimated difference between the modulator (tx) and demodulator (rx) sample clocks. For example if the transmit station sound card is sampling at 44000 Hz and the receive station sound card 44001 Hz, the sample clock offset would be ((44000-44001)/44000)*1E6 = 22.7 ppm.
This indicates the symbol timing estimate of the demodulator, in the range of +/- 0.5 of a symbol. With off air signals this will have a sawtooth appearance, as the demod tracks the modulator sample clock. The steeper the slope, the greater the sample clock offset.
When FreeDV syncs on a received signal for 5 seconds, it will send a "rx sync" UDP message to a port on your machine (localhost). An external program or script listening on this port can then take some action, for example send "spotting" information to a web server or send an email your phone.
Enable UDP messages on Tools-Options, and test using the "Test" button.
On Linux you can test reception of messages using netcat:
$ nc -ul 3000
A sample script to email you on FreeDV sync: send_email_on_sync.py
Usage for Gmail:
python send_email_on_sync.py --listen_port 3000 --smtp_server smtp.gmail.com \
--smtp_port 587 your@gmail.com your_pass
These features were added for FreeDV 700D, to help diagnose sound card issues during development.
Debug FIFO and PortAudio counters: used for debugging audio problems on 700D. During beta testing there were problems with break up in the 700D Tx and Rx audio on Windows.
The PortAudio counters (PortAudio1 and PortAudio2) should not increment when running in Tx or Rx, as this indicates samples are being lost by the sound driver which will lead to sync problems.
The Fifo counter outempty1 counter should not increment during Tx, as this indicates FreeDV is not supplying samples fast enough to the PortAudio drivers. The results will be resyncs at the receiver.
Check these counters by pressing Start, then Reset them and observe the counters for 30 seconds.
If the PortAudio counters are incrementing on receive try:
Adjusting framesPerBuffer; try 0, 128, 256, 512, 1024.
Shut down other applications that might be using audio, such as Skype or your web browser.
A different sound card rate such as 44.1kHz instead of 48kHz.
If the outempty1 counter is incrementing on transmit try increasing the FifoSize.
The txThreadPriority checkbox reduces the priority of the main txRx thread in FreeDV which may help the sound driver thread process samples.
The txRxDumpTiming check box dumps timing information to a console that is used for debugging the rx break up problem on 700D. Each number is how many ms the txRxThread took to run.
The txRxDumpTiming check box dumps the number of samples free in the tx FIFO sending samples to the Tx. If this hits zero, your tx audio will break up and the rx will lose sync. Tx audio break up will also occur if you see "outfifo1" being incremented on the "Fifo" line during tx. Try increasing the FifoSize.
This feature was developed for testing FreeDV 700C. Select the 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:
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.
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).
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 and the others at 0.5, indicating the poorer carrier BER is twice the larger.
Tools - Options - Half Duplex check box
FreeDV GUI can operate in full duplex mode which is useful for development or 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)
For the Linux inclined:
$ pandoc USER_MANUAL.md -o USER_MANUAL.pdf "-fmarkdown-implicit_figures -o" \
--from=markdown -V geometry:margin=.4in --toc --highlight-style=espresso
Term | Notes |
---|---|
AWGN | Additive White Gaussian Noise - a channel with just noise and no fading (like VHF) |
FEC | Forward Error Correction - extra bits to we send to protect the speech codec bits |
LDPC | Low Density Parity Check Codes - a family of powerful FEC codes |