Subject: Re: [ecasound] a few questions
From: Jeremy Hall (jhall@UU.NET)
Date: Mon Apr 17 2000 - 22:18:07 EEST
Hi,
what version of oss are you using? You might have luck with the latest
alsa drivers.
Lame has undergone lots of changes lately, and I'd suggest getting 3.70 or
later. You should be able to edit the bitrate in the .ecasoundrc
file. Newer versions of lame want -r, newer still don't seem to need it,
I haven't enquired about that.
One possible solution to the mandolin problem would be to reencode it with
a bit of tail on it of zeros. Also if you decode it first and don't try
to use mpg123 to play it, you may avoid your looping problem.
can't help ya with ecawave, because I don't run it. I'd suspect it
probably is a simple need to write EOF on the file after removing your
dead space, but then again, that might not be desired. in fact, I'd almost
rather the approach you see here, you can simply trim off the excess junk
at the end of the file.
_J
In the new year, Jan Stary wrote:
> Dear Kai,
>
> I upgraded to ecasound-1.7.5d11-1, ecasound-qt-1.7.5d11-1, ecawave-0.2.0d2-1 and lame-3.51.
> I use RedHat 6.1 (kernel 2.2.12-20) on an i586 (100MHz PII) with 24RAM,
> SB AWE 64, kernel's OSS.
>
> I have a few problems / questions:
>
> 1.
> ecasound -a:1 -f:16,1,44100 -i:guitar.mp3 -erc:1,2 -ea:100 -epp:15 -y:4.8 \
> -a:2 -f:16,1,44100 -i:mandolin.mp3 -erc:1,2 -ea:190 -epp:90 \
> -a:all -f:16,2,44100 -o:song.mp3
>
> guitar.mp3 is about 50s long, mandolin.mp3 about 15s. The output is OK till mandolin.mp3
> finishes in the right channel - after that a noise appears in the right channel (looping
> the very last few miliseconds of mandolin.mp3, I'm sure). If I used ecasound -b:2 the output
> was OK, but it processed REALLY slowly.
>
> 2.
> Is there a way to change the bitrate of mp3-output? Seems like lame makes it 128 kbps by default.
> How can I say (on an ecasound command line) that I want lame to make it something else?
>
> 3.
> Ecawave seems to have this bug: if you cut a 2sec sequence from the beginnig of a 10sec wave,
> (which is a usual thing to do: you cut the silence/noise between i-started-recording-here and
> i-hit-the-strings-here), the rest of the wave (2.00 to 10.00) gets shifted to the (new) beginning,
> BUT the total length remains the same and the end sequence <6.00 to 10.00> is now two times repeating
> what originally was <8.00 to 10.00> (one copy shifted to the new beginnig, one remaining at the end).
> It looks like:
>
> [original]
>
> |to cut| x
> | | xxxxx xxxx
> v v xxxx xxxxxxxxx xxxxxxx xx
> --------xxxxxxxxxxxxxxxxxxxxxx-----xxxxxxxxxxxxxxxxxxxxx
> ^ xxxx xxxxxxxxx xxxxxxx xx ^
> | xxxxx xxxx |
> | x |
> +------------ will get shited ------------------+
>
>
> [after cut]
>
> +------------- got shifted --------------------+
> | |
> | x |
> | xxxxx xxxx |
> v xxxx xxxxxxxxx xxxxxxx xx v xx
> xxxxxxxxxxxxxxxxxxxxxx-----xxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
> xxxx xxxxxxxxx xxxxxxx ^ xx ^ xx ^
> xxxxx xxxx | | |
> x | | |
> | |remained|
> | |
> +-- two times --+
>
>
> 4.
> And finally, one general question. On my system, "ll /dev/dsp*" says
>
> lrwxrwxrwx 1 root root 9 Apr 14 08:32 /dev/dsp -> /dev/dsp0
> crw------- 1 hans sys 14, 3 Apr 14 08:32 /dev/dsp0
> crw------- 1 hans sys 14, 19 Apr 14 08:32 /dev/dsp1
> crw------- 1 hans sys 14, 35 Apr 14 08:32 /dev/dsp2
> crw------- 1 hans sys 14, 51 Apr 14 08:32 /dev/dsp3
>
> What do the numbers (14,3 ...) mean, please? I can't record and playback at the same time,
> not both. It says "Can't open input file '/dev/dsp': device or resource busy".
> SB AWE 64 definitely is full duplex. What's wrong?
> I am still quite a newbie so excuse me please if my questions are stupid.
>
> Thanks a lot.
>
> Jan Stary
>
>
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