Hi Dan,

as Codec2 is a digital codec, a drop-down of quality like on analog
transmission seems hard to archieve. Normally, a good squelch-filter
with amplifier would work better instead of trying to modulate it again
as an analog narrowband signal.
Instead of 16QAM, I would prefer 4QAM, as it is much more robust than
16QAM in case of an angular phase shift. That's why 64QAM and 256QAM
only work fine in a stable connection and 16QAM is standard in DVB-T
(64QAM in DVB-C/S).


Am 15.01.2013 15:03, schrieb Daniel Ankers:
> Hi David,
>
>
> On 15 January 2013 05:58, David Rowe <[email protected]
> <mailto:[email protected]>> wrote:
>
>     > I'm in the middle of hacking the code in fdmdv.c to give an
>     indication
>     > of the quality of the received symbols as well as the symbols
>     > themselves.  This means that if the CRC check fails, we can flip the
>     > bits most likely to be incorrect and see if it passes.
>
>     This is also something I would like to look at so perhaps we can
>     collaborate.  There is some other structure in the parameters (e.g the
>     LSPs) that could be treated the same way as you have described above.
>
>
> That would be good.  My approach is to change
> float qpsk_to_bits(int rx_bits[], int *sync_bit, COMP
> phase_difference[], COMP prev_rx_symbols[], COMP rx_symbols[])
> to
> float qpsk_to_bits(int rx_bits[], int *sync_bit, COMP
> phase_difference[], COMP prev_rx_symbols[], COMP rx_symbols[], float
> errors[])
>
> I calculate the mean magnitude avg_r for each frame, and then define
> the error in each symbol as the distance between the symbol and a
> "perfect" symbol at avg_r, theta (where theta is one of pi/4, 3*pi/4,
> 5*pi/4 or 7*pi/4, depending on the quadrant the symbol is in.)
>
> At the moment I'm returning the error on a per-bit basis rather than
> per-symbol.  This has the disadvantage that it ignores the fact that
> errors are going to affect whole symbols not individual bits, but it
> means that the whole approach can be used with more complicated
> modems.  (One idea I had was that the codec2 bits which have a more
> minor effect on the voice quality could be carried in 16QAM carriers
> to reduce the overall transmitted bandwidth and giving a more
> analogue-like drop in quality as signal levels dropped instead of a
> digital all-or-nothing effect.)  I'm hoping to just start simple and
> then possibly improve in the future to a more intelligent approach
> which carries data about which bits come from which symbol and the
> quality of that symbol.
>
> I'm also very deliberately sticking to a frame-by-frame approach
> rather than gathering data across several frames.  I'm trying to avoid
> doing anything which will reduce the sync speed of the system.
>
>
>     To the list in general: over the next few months I will be looking at
>     making FreeDV more robust on HF radio channels.  This will involve
>     work
>     to FreeDV, Codec2, and the fdmdv modem
>
>
> One thing which I think would be valuable is some kind of indication
> of the format of the data being sent over the fdmdv modem.
> At the moment, if we change the data format without changing the modem
> then existing FreeDV users will hear odd noises (since the data will
> demodulate from the modem correctly but will not make sense when
> decoded by the codec.)  It would be nice if in future we are able to
> change the data format and FreeDV mutes the sound and possibly brings
> up a notification that a new modulation scheme is being used and it
> may be time to go online and check for updates.
> Obviously if the modem is changed then this will be shown in the
> waterfall looking different and existing copies of FreeDV would
> probably not be able to decode anything.
>
> Cheers,
> Dan MD1CLV
>
>
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