On 23/04/18 15:39, Alan Beard wrote:
What I read into this PDF was, if one looks at a few Codec2 "packets", one can
deduce where the speech is going ie. transitioning from one sound to
another. One syllable to another. Having "learned" this to recreate what
the original words were, better.
I've been trying to "get my head around" Codec2 encoding, explaining to
others:
One's voice starts with one's vocal chords which give a pulsating air stream,
this being the basic pitch and amplitude. Then, mouth, nose and throat filter
this waveform. Then, some wind noise is added.
So rather than trying to reproduce the waveform, digitise and send the
components.
Is this an adequate description?
Alan VK2ZIW
wikipedia has a good write up on codec2:
"Codec 2 uses sinusoidal coding to model speech. In sinusoidal coding,
spoken audio is recreated by modelling speech as a sum of harmonically
related sine waves with independent amplitudes called Line spectral
pairs, or LSP. The fundamental frequency of the speaker's voice (pitch)
and the amplitude (energy) of the harmonics is encoded, and with the
LSP's are exchanged across a channel in a digital format. The LSP
coefficients represent the Linear Predictive Coding (LPC) model in the
frequency domain, and lend themselves to a robust and efficient
quantisation of the LPC parameters." -
<https://en.wikipedia.org/wiki/Codec_2>
On Mon, 23 Apr 2018 14:34:24 +0930, David Rowe wrote
I don't think they are on this list, but I have had a response via
email so free free to reach out to them.
Their current algorithm needs a GPU to decode, so won't run on a
SM1000
(or even CPU) class machine. This sort of computational complexity
is common for new DSP algorithms. The tend to get more efficient
over time.
- David
On 23/04/18 13:46, Akshay Mishra wrote:
Are the authors available on this list -- I read the paper and if some
trial is available would like to play with this on our setup (similar to
SM1000 but with a different RF front end). A training set for each user
can be incorporated when they first state and made available to all...
thanks,
On 22 April 2018 at 01:32, David Rowe <da...@rowetel.com
<mailto:da...@rowetel.com>> wrote:
Some amazing work with Codec 2 and WavNet ... using a Codec 2
encoder and WaveNet decoder they have speech quality as good as AMB
wideband at just 2400 bit/s
http://www.rowetel.com/?p=5966
- David
On 20/04/18 11:41, David Rowe wrote:
Hi Bruce,
I'll be presenting at our WIA national conference on the same
dates .. will send you the slides. Aiming to release on FreeDV
GUI by then.
- David
On 20/04/18 11:35, Bruce Perens wrote:
Woo hoo! Please tell me about anything you'd like me to
present at Hamvention. This will definitely be in there.
On Thu, Apr 19, 2018 at 6:51 PM, David Rowe
<da...@rowetel.com <mailto:da...@rowetel.com>
<mailto:da...@rowetel.com <mailto:da...@rowetel.com>>> wrote:
Hello,
I've been working steadily on FreeDV 700D, porting from
Octave to C,
and working through some tricky sync problems. I can
now simulate
the end-end system on the command line, and listen to
the results. It's outperforming FreeDV 1600 by about
10dB on fast fading
channels, and is error free at -2dB SNR on AWGN channels.
The main differences from FreeDV 700C are a new OFDM
modem with rate
1/2 LDPC code and the option of a long interleaver
(seconds of
latency). It uses the the 700 bit/s codec, although
with a few
tweaks we could run other codecs and LDPC codes using a
few dB more
SNR. More inf on the FreeDV 700D mode here:
https://www.rowetel.com/?p=5573
<https://www.rowetel.com/?p=5573>
<https://www.rowetel.com/?p=5573
<https://www.rowetel.com/?p=5573>>
For anyone who would like to play along, here's
README_ofdm.txt for
the mode:
https://svn.code.sf.net/p/freetel/code/codec2-
dev/README_ofdm.txt
<https://svn.code.sf.net/p/freetel/code/codec2-
dev/README_ofdm.txt>
<https://svn.code.sf.net/p/freetel/code/codec2-
dev/README_ofdm.txt
<https://svn.code.sf.net/p/freetel/code/codec2-
dev/README_ofdm.txt>>
This includes references to blog posts, lots of command
line
examples, and specification at the end. There's enough
information
for the super keen to run it over the air using the cmd
line tools,
which I haven't done yet.
Next steps are some over the air tests, write a
freedv_api.c "mode"
for it, integration into the FreeDV GUI program, and
lots of tweaking.
- David
On 16/04/18 08:52, Glen English wrote:
I will write more abt this when I return, however
a) the quality
of the capacitor influences notch depth, b )
getting the area
and proximity of the loop to the line matters. for
a 6" filter,
anything over 33db for 600k bp-reject is very good.
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