On Fri, May 15, 2009 at 3:56 AM, Martin Braun
<address@hidden> wrote:
On Wed, May 13, 2009 at 04:25:39PM -0400, Justin G. Eskesen wrote:
> P.S. I've tried both wavfile_source & file_source & neither work. I'm aware of
> the -1.0 to 1.0 normalization in wavfile_* & have commented that portion of the
> code out. If I get this to work, I'll write a wavfile_source & sink that do no
> rescaling & take shorts because I'm doing too many unecessary conversions.
For the start, it probably would have been better to put in a
gr.multiply_const_ff() after the filter with a value of something like
20000. In the setup you mentioned, reading shorts from the WAV file
would not have helped, since you're using a float FIR afterwards, and a
short FIR would introduce too much quantization noise.
I've tried pretty much exactly the same setup before (but with the
multiplier) and it worked. Also, for debugging, I recommend using a
file_sink (not a wavfile_sink) instead of the USRP and check the sample
values in that file scale well in the DAC range. Actually, if you /did/
pipe it to WAV or audio, it should /not/ work, since they rely on data
being between -1 and 1.
Should you do ever do write a short wavfile_* and also want to support 8
Bit, remember that scales differently.
Hope this was what you wanted.
MB
--
Dipl.-Ing. Martin Braun Phone: +49-(0)721-608 3790
Institut fuer Nachrichtentechnik Fax: +49-(0)721-608 6071
Universitaet Karlsruhe (TH) http://www.int.uni-karlsruhe.de/
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