Nabble - Ecasound - General

Compiling problems with git

2009-12-14T00:53:47Z

Hi folks.

I do get a compilation error when compiling from git. ( Perhaps I've overdone it with below disable-config?)

g++ -DHAVE_CONFIG_H -I. -I.. -I -I. -I.. -I../libecasound -I../kvutils -D_REENTRANT -D_XOPEN_SOURCE=500 -ffast-math -fstrict-aliasing -DNDEBUG -g -O2 -MT ecasound.o -MD -MP -MF .deps/ecasound.Tpo -c -o ecasound.o ecasound.cpp
ecasound.cpp: In destructor ‘ECASOUND_RUN_STATE::~ECASOUND_RUN_STATE()’:
ecasound.cpp:186: warning: possible problem detected in invocation of delete operator:
ecasound.cpp:186: warning: invalid use of incomplete type ‘struct ECA_OSC_INTERFACE’
ecasound.h:15: warning: forward declaration of ‘struct ECA_OSC_INTERFACE’
ecasound.cpp:186: note: neither the destructor nor the class-specific operator delete will be called, even if they are declared when the class is defined.
ecasound.cpp: In function ‘void ecasound_parse_command_line(ECASOUND_RUN_STATE*, const COMMAND_LINE&, COMMAND_LINE*)’:
ecasound.cpp:594: error: ‘stdout’ was not declared in this scope
ecasound.cpp:595: error: ‘fprintf’ was not declared in this scope
make[2]: *** [ecasound.o] Error 1
make[2]: Leaving directory `/usr/src/ecasound/ecasound'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/usr/src/ecasound'
make: *** [all] Error 2

./configure --prefix=$PREFIXDIR --disable-ncurses --disable-pyecasound --disable-oss --disable-arts --disable-osstrigger \
--disable-largefile --disable-dbc --without-pic --disable-jack --disable-audiofile \
--disable-sndfile --disable-rubyecasound --disable-libsamplerate --enable-liboil >>$LOG 2>&1 || die "Configure failed. Please have a look at logfile $LOG"

BTW: You might ask? Why all the disabled functions? I tried to configure ecasound as slim as possible - just that it is able to playback wav and to do some volume control.

Any ideas.

THX

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Re: chain setup equivalence

2009-12-07T00:58:18Z

On Thu, Dec 03, 2009 at 08:34:04PM -1000, Joel Roth wrote:

> Hi all,
>
> I wonder if these two lines are equivalent:
>
> -a:1,2 -i:something
>
> -a:1 -i:something -a:2 -i:something
>
> Are they equivalent regardless of what 'something' is,
> and whether the direction is -i or -o?

Attempting to answer my own question, the following
is valid:

-a:1 -i:foo.wav
-a:2 -i:bar.wav
-a:1 -o:null
-a:2 -o:null

But not this:

-a:1 -i:foo.wav
-a:2 -i:bar.wav
-a:1 -o:loop,1
-a:2 -o:loop,1

You would need to say:

-a:1 -i:foo.wav
-a:2 -i:bar.wav
-a:1,2 -o:loop,1

Regards,

Joel

--
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Re: multiple input objects

2009-12-07T00:51:09Z

Hi!
Fortunitely the end of the year's nt far now. And then you can hopefully
grade up to 2.7.0. Make it a christmas wish. :-)
Warmly,
Julien

--------
Music was my first love and it will be my last (John Miles)

======== FIND MY WEB-PROJECT AT: ========
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: =======
http://www.juliencoder.de

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Re: multiple input objects

2009-12-06T14:56:52Z

Thanks a ton.

This seems to work, but alas not in version 2.4.6, which is where I'm
stuck on our studio's production machine until the end of the year...
I'll do a workaround for this year.

I appreciate your help.

Matt

On Sun, Dec 6, 2009 at 4:44 PM, Julien Claassen <julien@...> wrote:

> Ah that...
> Youneed to stack them:
> -i resample-hq,96000,sndfile,input -o output
> At least if this works the same as the other input objects. but I don't see
> a reason why not.
> Bye
> Julien
>
> --------
> Music was my first love and it will be my last (John Miles)
>
> ======== FIND MY WEB-PROJECT AT: ========
> http://ltsb.sourceforge.net
> the Linux TextBased Studio guide
> ======= AND MY PERSONAL PAGES AT: =======
> http://www.juliencoder.de
>

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Re: multiple input objects

2009-12-06T13:44:43Z

Ah that...
Youneed to stack them:
-i resample-hq,96000,sndfile,input -o output
At least if this works the same as the other input objects. but I don't see
a reason why not.
Bye
Julien

--------
Music was my first love and it will be my last (John Miles)

======== FIND MY WEB-PROJECT AT: ========
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: =======
http://www.juliencoder.de

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Re: multiple input objects

2009-12-06T13:38:41Z

On Sun, Dec 6, 2009 at 3:56 PM, Julien Claassen <julien@...> wrote:
> Hello Matt!
> If you want to do this with ecasound, it shouldn't be a problem, just use
> several chains.
> ecasound -a:1 -f:16,2,44100 -i resample-hq,96000,input.wav -o
> output44100.wav -a:2 -f:16,2,22050 -i resample-hq,96000,input.wav -o
> output22050.wav
> Or whatever you use for resampling. This should theoretically work. If you
> want to open the file only once... Perhaps you might use loops, but I'm not
> sure on that one.

Thanks! But, what if I would like to open the file using libsndfile
AND resample it within one chain? I'm not sure it's possible since to
do each you need a special input object:

I've tried something like this, but I'm not sure it's possible to get
both things into the chain:

ecasound -f:f32_le,2,44100 -i:sndfile,input.wav resample-hq,96000 -o:output.wav

etc.

I have a feeling that I'd have to have two separate instances and a named pipe:

mkfifo fifo.wav

ecasound -f:f32_le,2,44100 -i:resample-hq,96000,fifo.wav -o:output.wav &
## this process should run until an EOF is piped through

ecasound -f:f32_le,2,96000 -i:sndfile,input.wav -o:fifo.wav

rm fifo.wav

I haven't tried this yet, so I'm not even sure this will work.

The only reason I need this is because I'm writing floats from 16-bit
files, and libsndfile handles the int->float conversion differently
than does ecasound, and all the rest of my eventual processing will be
done using csound which reads in and out via libsndfile.

Thanks again,

Matt

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Re: multiple input objects

2009-12-06T12:56:41Z

Hello Matt!
If you want to do this with ecasound, it shouldn't be a problem, just use
several chains.
ecasound -a:1 -f:16,2,44100 -i resample-hq,96000,input.wav -o output44100.wav
-a:2 -f:16,2,22050 -i resample-hq,96000,input.wav -o output22050.wav
Or whatever you use for resampling. This should theoretically work. If you
want to open the file only once... Perhaps you might use loops, but I'm not
sure on that one.
Hope that helps
Julien

--------
Music was my first love and it will be my last (John Miles)

======== FIND MY WEB-PROJECT AT: ========
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: =======
http://www.juliencoder.de

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multiple input objects

2009-12-05T23:38:36Z

Is there a way to open a file using libsndfile and resample it
simultaneously? Since both require a special object in the input
line, I'm not sure how to get both at once. Or, would I need an
intermediate file or a named pipe for this instead?

Thanks,

Matt

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int to float or float to int conversion

2009-12-05T23:24:32Z

Hello,

How does ecasound convert 16-bit ints to floats? It does not seem to
be a simple divide by 38768, since a file with a range of -32768 to
+32767 seems to map perfectly to -1.0 to +1.0 after conversion. This
is not the case when libsndfile is invoked to open the file -- there
it looks like a division by 38768, leading to a range -1.0 to
+0.999969

I'm also curious about the opposite conversion from float to int.

I assume that the conversion methods are the same for 24-bit integers,
but with different multipliers/divisors etc.

Thanks,

Matt

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chain setup equivalence

2009-12-03T22:34:04Z

Hi all,

I wonder if these two lines are equivalent:

-a:1,2 -i:something

-a:1 -i:something -a:2 -i:something

Are they equivalent regardless of what 'something' is,
and whether the direction is -i or -o?

Thanks,

Joel

--
Joel Roth

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Re: Strange configure error

2009-11-30T02:26:57Z

--- On Mon, 11/30/09, Julien Claassen <julien@...> wrote:

> From: Julien Claassen <julien@...>
> Subject: [ecasound] Strange configure error
> To: "ecasound-list" <ecasound-list@...>
> Date: Monday, November 30, 2009, 1:13 AM
> Hello all!
>    I'm almost 100% sure, that this is not
> Ecasound's and its team's fault, but
> still I better report it here first.
>    I ran ./configure and was notified that
> some requirements were not met. Some
> ac_nonexistent.h wasn't found. As the name suggests, it
> should't be. :-) The
> problem was, that g++ couldn't be found. A Debian problem.
> But still shouldn't
> ./configure say: g++ not found.
>    This was ecasound 2.7.0 and the system
> was a newly installed Debian Squeeze.
> In the ecasound dependencies everything was taken from
> Debian archives.
>    Thanks for reading on upto here. Sorry
> for rambling...
>    Kindest regards
>    Julien
>
> --------
> Music was my first love and it will be my last (John
> Miles)
>
> ======== FIND MY WEB-PROJECT AT: ========
> http://ltsb.sourceforge.net
> the Linux TextBased Studio guide
> ======= AND MY PERSONAL PAGES AT: =======
> http://www.juliencoder.de
>
> ---------------------------------------------------------------------------

I remember at least one other package which doesn't check for g++ presence though it should, i.e. its 'configure' is faulty.

Regards,
Sergei.

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Strange configure error

2009-11-30T01:13:40Z

Hello all!
I'm almost 100% sure, that this is not Ecasound's and its team's fault, but
still I better report it here first.
I ran ./configure and was notified that some requirements were not met. Some
ac_nonexistent.h wasn't found. As the name suggests, it should't be. :-) The
problem was, that g++ couldn't be found. A Debian problem. But still shouldn't
./configure say: g++ not found.
This was ecasound 2.7.0 and the system was a newly installed Debian Squeeze.
In the ecasound dependencies everything was taken from Debian archives.
Thanks for reading on upto here. Sorry for rambling...
Kindest regards
Julien

--------
Music was my first love and it will be my last (John Miles)

======== FIND MY WEB-PROJECT AT: ========
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: =======
http://www.juliencoder.de

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Re: obviously wrong ALSA underruns reported by ecasound-2.6.0

2009-11-25T11:00:52Z

--- On Wed, 11/25/09, Sergei Steshenko <sergstesh@...> wrote:

> From: Sergei Steshenko <sergstesh@...>
> Subject: Re: [ecasound] obviously wrong ALSA underruns reported by ecasound-2.6.0
> To: "Kai Vehmanen" <kvehmanen@...>
> Cc: ecasound-list@...
> Date: Wednesday, November 25, 2009, 10:55 AM
>
>
> --- On Wed, 11/25/09, Kai Vehmanen <kvehmanen@...>
> wrote:
>
> > From: Kai Vehmanen <kvehmanen@...>
> > Subject: Re: [ecasound] obviously wrong ALSA underruns
> reported by ecasound-2.6.0
> > To: "Sergei Steshenko" <sergstesh@...>
> > Cc: ecasound-list@...
> > Date: Wednesday, November 25, 2009, 9:41 AM
> > Hi,
> >
> > On Mon, 23 Nov 2009, Sergei Steshenko wrote:
> >
> > > " WARNING: ALSA playback underrun, glitches in
> audio
> > playback possible! Break was at least 566281884.09 ms
> long.
> > "
> > [...]
> > > In reality the drop was like 1 second, maybe
> less.
> > [...]
> > > Is it a known issue ?
> > >
> > > Which of the two - ecasound or ALSA - reports
> > underruns ?
> >
> > ok, tis is quite interesting? The warning is printed
> by
> > ecasound, but the code has remained the same for who
> now how
> > many years, and is virtually identical to the warning
> > printed by the most recent aplay (of alsa-utils). Only
> thing
> > that's different that ecasound does not use the
> monotonic
> > timestamps even if they are available (reason: these
> are a
> > later addition and were not available when ecasound's
> ALSA
> > code was written).
> >
> > What version of alsa-lib and kernel (cat
> > /proc/asound/version for the latter)
> > do you have...? Also what soundcard/chip do you have
> (cat
> > /proc/asound/cards)...?
> >
> > > I'd say, I've never seen a reasonable number
> since
> > upgrade to both
> > > SUSE-11.1 (i.e. newer ALSA) and ecasound-2.6.0 -
> from
> > ecasound-2.5.2
> > >
> > > Both versions of ecasound are self-built using
> the
> > same tool of mine.
> >
> > So I'm 99+% sure it's caused by the ALSA upgrade (the
> xrun
> > reporting code is exactly the same in 2.5.2 as in
> 2.6.0).
> >
> > Now the actual bug might still be in ecasound (and
> possibly
> > in aplay as well), but it's just been hidden with
> older ALSA
> > versions. Hmm, I wonder how I could reproduce this
> myself...
> > :P
> >
> > > WARNING: ALSA playback underrun, glitches in
> audio
> > playback possible! Break was at least 566281807.81 ms
> long.
> > > WARNING: ALSA playback underrun, glitches in
> audio
> > playback possible! Break was at least 566281802.92 ms
> long.
> >
> > Btw, in any case the XRUN is real in these cases
> (this
> > warning is only printed when the stream goes to XRUN
> state
> > and the length estimation is done as a separate step).
> I.e.
> > the break length is not used to decide whether there
> was a
> > XRUN or not... (this would be a fairly bad bug).
> >
>
> 'uname -a' output:
>
> Linux amdam2 2.6.27.37-0.1-default #1 SMP 2009-10-15
> 14:56:58 +0200 i686 athlon i386 GNU/Linux
>
>
> 'cat /proc/asound/version' output:
>
> "
> Advanced Linux Sound Architecture Driver Version 1.0.17
> ",
>
> but (!) at the same time:
>
> "
> rpm -q alsa
> alsa-1.0.18-8.12.1
> "
>
> Sound chip:
>
> "
> Card: HDA NVidia
> Chip: Realtek ALC883
> ".
>
> I don't know how to answer regarding ALSA library, for
> example, ALSA
> plugins are 1.0.18-6.12, but "Samplerate Plug-In for the
> ALSA Library" is
> 1.0.14-41. alsa-utils is 1.0.18-6.4.
>
> It's SUSE package manager which decided to install these
> versions.
>
> I know XRUNs are real; overall sound feels better - it
> appears now out of
> the box audio applications have realtime priority (or is it
> just a bigger
> buffer ?) - pretty heavy applications which used to skip
> sometimes
> because of file access in parallel with file access of
> other applications
> seem to work much smoother now.
>
> The reported XRUNs were observed during Internet MP3 stream
> playback, and
> as such do not surprise me, but I payed attention to the
> numbers.
>
> Thanks,
> Sergei.
>

Aha, it's called

"libasound2 - Advanced Linux Sound Architecture Library" - 1.0.18-8.12.1.

--Sergei.

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Re: obviously wrong ALSA underruns reported by ecasound-2.6.0

2009-11-25T10:55:29Z

--- On Wed, 11/25/09, Kai Vehmanen <kvehmanen@...> wrote:

> From: Kai Vehmanen <kvehmanen@...>
> Subject: Re: [ecasound] obviously wrong ALSA underruns reported by ecasound-2.6.0
> To: "Sergei Steshenko" <sergstesh@...>
> Cc: ecasound-list@...
> Date: Wednesday, November 25, 2009, 9:41 AM
> Hi,
>
> On Mon, 23 Nov 2009, Sergei Steshenko wrote:
>
> > " WARNING: ALSA playback underrun, glitches in audio
> playback possible! Break was at least 566281884.09 ms long.
> "
> [...]
> > In reality the drop was like 1 second, maybe less.
> [...]
> > Is it a known issue ?
> >
> > Which of the two - ecasound or ALSA - reports
> underruns ?
>
> ok, tis is quite interesting? The warning is printed by
> ecasound, but the code has remained the same for who now how
> many years, and is virtually identical to the warning
> printed by the most recent aplay (of alsa-utils). Only thing
> that's different that ecasound does not use the monotonic
> timestamps even if they are available (reason: these are a
> later addition and were not available when ecasound's ALSA
> code was written).
>
> What version of alsa-lib and kernel (cat
> /proc/asound/version for the latter)
> do you have...? Also what soundcard/chip do you have (cat
> /proc/asound/cards)...?
>
> > I'd say, I've never seen a reasonable number since
> upgrade to both
> > SUSE-11.1 (i.e. newer ALSA) and ecasound-2.6.0 - from
> ecasound-2.5.2
> >
> > Both versions of ecasound are self-built using the
> same tool of mine.
>
> So I'm 99+% sure it's caused by the ALSA upgrade (the xrun
> reporting code is exactly the same in 2.5.2 as in 2.6.0).
>
> Now the actual bug might still be in ecasound (and possibly
> in aplay as well), but it's just been hidden with older ALSA
> versions. Hmm, I wonder how I could reproduce this myself...
> :P
>
> > WARNING: ALSA playback underrun, glitches in audio
> playback possible! Break was at least 566281807.81 ms long.
> > WARNING: ALSA playback underrun, glitches in audio
> playback possible! Break was at least 566281802.92 ms long.
>
> Btw, in any case the XRUN is real in these cases (this
> warning is only printed when the stream goes to XRUN state
> and the length estimation is done as a separate step). I.e.
> the break length is not used to decide whether there was a
> XRUN or not... (this would be a fairly bad bug).
>

'uname -a' output:

Linux amdam2 2.6.27.37-0.1-default #1 SMP 2009-10-15 14:56:58 +0200 i686 athlon i386 GNU/Linux

'cat /proc/asound/version' output:

"
Advanced Linux Sound Architecture Driver Version 1.0.17
",

but (!) at the same time:

"
rpm -q alsa
alsa-1.0.18-8.12.1
"

Sound chip:

"
Card: HDA NVidia
Chip: Realtek ALC883
".

I don't know how to answer regarding ALSA library, for example, ALSA
plugins are 1.0.18-6.12, but "Samplerate Plug-In for the ALSA Library" is
1.0.14-41. alsa-utils is 1.0.18-6.4.

It's SUSE package manager which decided to install these versions.

I know XRUNs are real; overall sound feels better - it appears now out of
the box audio applications have realtime priority (or is it just a bigger
buffer ?) - pretty heavy applications which used to skip sometimes
because of file access in parallel with file access of other applications
seem to work much smoother now.

The reported XRUNs were observed during Internet MP3 stream playback, and
as such do not surprise me, but I payed attention to the numbers.

Thanks,
Sergei.

------------------------------------------------------------------------------
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Re: obviously wrong ALSA underruns reported by ecasound-2.6.0

2009-11-25T09:41:50Z

Hi,

On Mon, 23 Nov 2009, Sergei Steshenko wrote:

> " WARNING: ALSA playback underrun, glitches in audio playback possible!
> Break was at least 566281884.09 ms long. "
[...]
> In reality the drop was like 1 second, maybe less.
[...]
> Is it a known issue ?
>
> Which of the two - ecasound or ALSA - reports underruns ?

ok, tis is quite interesting? The warning is printed by ecasound, but the
code has remained the same for who now how many years, and is virtually
identical to the warning printed by the most recent aplay (of alsa-utils).
Only thing that's different that ecasound does not use the monotonic
timestamps even if they are available (reason: these are a later addition
and were not available when ecasound's ALSA code was written).

What version of alsa-lib and kernel (cat /proc/asound/version for the latter)
do you have...? Also what soundcard/chip do you have (cat
/proc/asound/cards)...?

> I'd say, I've never seen a reasonable number since upgrade to both
> SUSE-11.1 (i.e. newer ALSA) and ecasound-2.6.0 - from ecasound-2.5.2
>
> Both versions of ecasound are self-built using the same tool of mine.

So I'm 99+% sure it's caused by the ALSA upgrade (the xrun reporting code
is exactly the same in 2.5.2 as in 2.6.0).

Now the actual bug might still be in ecasound (and possibly in aplay as
well), but it's just been hidden with older ALSA versions. Hmm, I wonder
how I could reproduce this myself... :P

> WARNING: ALSA playback underrun, glitches in audio playback possible! Break was at least 566281807.81 ms long.
> WARNING: ALSA playback underrun, glitches in audio playback possible! Break was at least 566281802.92 ms long.

Btw, in any case the XRUN is real in these cases (this warning is only
printed when the stream goes to XRUN state and the length estimation is
done as a separate step). I.e. the break length is not used to decide
whether there was a XRUN or not... (this would be a fairly bad bug).

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Snow Leopard and JACK

2009-11-25T02:58:55Z

Hi,

I've just now tried to compile ecasound-2.7.0 with JACK support under
OS X 10.6.1 and had no luck. Someone else posted recently about this
issue, and I thought I'd add my error output.

This is the message conveying the crux of the issue, I think:

ld: warning: in /usr/local/lib/libjack.dylib, missing required
architecture x86_64 in file

... which occurs when it tries the final link. I have downloaded the
latest JACK, version 0.83, and perhaps this is the problem.

I ./configure'd with no extra options, and it automatically found I
had JACK installed. It gets through to the final link, before erroring
out when dealing with the JACK library. Below I've pasted the output
of running 'make' again to re-create the error.

Anyone got any ideas? Perhaps we can specify an architecture somewhere?

Thanks for any feedback,

Julius

$ make
make all-recursive
Making all in kvutils
make[2]: Nothing to be done for `all'.
Making all in libecasound
Making all in plugins
make[3]: Nothing to be done for `all'.
make[3]: Nothing to be done for `all-am'.
Making all in ecasound
/bin/sh ../libtool --tag=CXX --mode=link g++ -ffast-math
-fstrict-aliasing -DNDEBUG -DENABLE_DBC -g -O2 -framework CoreAudio
-o ecasound -export-dynamic ecasound.o eca-curses.o eca-comhelp.o
eca-neteci-server.o eca-plaintext.o textdebug.o -lncurses -lreadline
../libecasound/libecasound.la ../kvutils/libkvutils.la
g++ -ffast-math -fstrict-aliasing -DNDEBUG -DENABLE_DBC -g -O2
-framework CoreAudio -o ecasound ecasound.o eca-curses.o eca-comhelp.o
eca-neteci-server.o eca-plaintext.o textdebug.o -Wl,-bind_at_load
-lncurses -lreadline ../libecasound/.libs/libecasound.a -ljack
../kvutils/.libs/libkvutils.a
ld: warning: in /usr/local/lib/libjack.dylib, missing required
architecture x86_64 in file
Undefined symbols:
"_jack_get_buffer_size", referenced from:
ECA_CHAINSETUP::check_for_locked_buffersize() const in
libecasound.a(eca-chainsetup.o)
AUDIO_IO_JACK_MANAGER::open_server_connection() in
libecasound.a(audioio_jack_manager.o)
"_jack_set_sync_callback", referenced from:
AUDIO_IO_JACK_MANAGER::open_server_connection() in
libecasound.a(audioio_jack_manager.o)
"_jack_set_buffer_size_callback", referenced from:
AUDIO_IO_JACK_MANAGER::open_server_connection() in
libecasound.a(audioio_jack_manager.o)
"_jack_disconnect", referenced from:
JACK_CONNECTIONS::disconnect(char const*, char const*)in
libecasound.a(jack-connections.o)
AUDIO_IO_JACK_MANAGER::set_node_connection(AUDIO_IO_JACK_MANAGER::eca_jack_node*,
bool)in libecasound.a(audioio_jack_manager.o)
"_jack_port_get_total_latency", referenced from:
AUDIO_IO_JACK_MANAGER::get_total_port_latency(_jack_client*,
AUDIO_IO_JACK_MANAGER::eca_jack_port_data*)in
libecasound.a(audioio_jack_manager.o)
"_jack_get_sample_rate", referenced from:
AUDIO_IO_JACK_MANAGER::open_server_connection() in
libecasound.a(audioio_jack_manager.o)
"_jack_port_unregister", referenced from:
AUDIO_IO_JACK_MANAGER::unregister_jack_ports(int) in
libecasound.a(audioio_jack_manager.o)
"_jack_port_name", referenced from:
AUDIO_IO_JACK_MANAGER::get_total_port_latency(_jack_client*,
AUDIO_IO_JACK_MANAGER::eca_jack_port_data*)in
libecasound.a(audioio_jack_manager.o)
AUDIO_IO_JACK_MANAGER::set_node_connection(AUDIO_IO_JACK_MANAGER::eca_jack_node*,
bool)in libecasound.a(audioio_jack_manager.o)
"_jack_client_close", referenced from:
ECA_CHAINSETUP::check_for_locked_buffersize() const in
libecasound.a(eca-chainsetup.o)
ECA_OBJECT_FACTORY::probe_default_output_device() in
libecasound.a(eca-object-factory.o)
priv_cleanup(_jack_client*) in libecasound.a(jack-connections.o)
AUDIO_IO_JACK_MANAGER::close_server_connection() in
libecasound.a(audioio_jack_manager.o)
"_jack_set_sample_rate_callback", referenced from:
AUDIO_IO_JACK_MANAGER::open_server_connection() in
libecasound.a(audioio_jack_manager.o)
"_jack_port_get_buffer", referenced from:
eca_jack_process_mute(unsigned int, void*)in
libecasound.a(audioio_jack_manager.o)
eca_jack_process_engine_iteration(unsigned int, void*)in
libecasound.a(audioio_jack_manager.o)
eca_jack_process_engine_iteration(unsigned int, void*)in
libecasound.a(audioio_jack_manager.o)
"_jack_client_new", referenced from:
ECA_CHAINSETUP::check_for_locked_buffersize() const in
libecasound.a(eca-chainsetup.o)
ECA_OBJECT_FACTORY::probe_default_output_device() in
libecasound.a(eca-object-factory.o)
priv_prepare() in libecasound.a(jack-connections.o)
AUDIO_IO_JACK_MANAGER::open_server_connection() in
libecasound.a(audioio_jack_manager.o)
"_jack_transport_stop", referenced from:
AUDIO_IO_JACK_MANAGER::stop() in
libecasound.a(audioio_jack_manager.o)
"_jack_activate", referenced from:
AUDIO_IO_JACK_MANAGER::activate_server_connection() in
libecasound.a(audioio_jack_manager.o)
"_jack_connect", referenced from:
JACK_CONNECTIONS::connect(char const*, char const*)in
libecasound.a(jack-connections.o)
AUDIO_IO_JACK_MANAGER::set_node_connection(AUDIO_IO_JACK_MANAGER::eca_jack_node*,
bool)in libecasound.a(audioio_jack_manager.o)
"_jack_deactivate", referenced from:
AUDIO_IO_JACK_MANAGER::deactivate_server_connection() in
libecasound.a(audioio_jack_manager.o)
"_jack_port_get_all_connections", referenced from:
JACK_CONNECTIONS::list_connections(std::basic_string<char,
std::char_traits<char>, std::allocator<char> >*)in
libecasound.a(jack-connections.o)
"_jack_set_process_callback", referenced from:
AUDIO_IO_JACK_MANAGER::open_server_connection() in
libecasound.a(audioio_jack_manager.o)
"_jack_transport_locate", referenced from:
AUDIO_IO_JACK_MANAGER::initial_seek() in
libecasound.a(audioio_jack_manager.o)
AUDIO_IO_JACK_MANAGER::exec(ECA_ENGINE*, ECA_CHAINSETUP*)
in libecasound.a(audioio_jack_manager.o)
"_jack_transport_query", referenced from:
AUDIO_IO_JACK_MANAGER::initial_seek() in
libecasound.a(audioio_jack_manager.o)
eca_jack_process_callback(unsigned int, void*)in
libecasound.a(audioio_jack_manager.o)
"_jack_get_ports", referenced from:
JACK_CONNECTIONS::list_connections(std::basic_string<char,
std::char_traits<char>, std::allocator<char> >*)in
libecasound.a(jack-connections.o)
AUDIO_IO_JACK_MANAGER::auto_connect_jack_port_client(int,
std::basic_string<char, std::char_traits<char>, std::allocator<char> >
const&, int)in libecasound.a(audioio_jack_manager.o)
AUDIO_IO_JACK_MANAGER::auto_connect_jack_port_client(int,
std::basic_string<char, std::char_traits<char>, std::allocator<char> >
const&, int)in libecasound.a(audioio_jack_manager.o)
"_jack_port_by_name", referenced from:
JACK_CONNECTIONS::list_connections(std::basic_string<char,
std::char_traits<char>, std::allocator<char> >*)in
libecasound.a(jack-connections.o)
"_jack_port_register", referenced from:
AUDIO_IO_JACK_MANAGER::register_jack_ports(int, int,
std::basic_string<char, std::char_traits<char>, std::allocator<char> >
const&)in libecasound.a(audioio_jack_manager.o)
AUDIO_IO_JACK_MANAGER::register_jack_ports(int, int,
std::basic_string<char, std::char_traits<char>, std::allocator<char> >
const&)in libecasound.a(audioio_jack_manager.o)
"_jack_transport_start", referenced from:
AUDIO_IO_JACK_MANAGER::start() in
libecasound.a(audioio_jack_manager.o)
"_jack_on_shutdown", referenced from:
AUDIO_IO_JACK_MANAGER::open_server_connection() in
libecasound.a(audioio_jack_manager.o)
ld: symbol(s) not found
collect2: ld returned 1 exit status
make[2]: *** [ecasound] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2

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Re: obviously wrong ALSA underruns reported by ecasound-2.6.0

2009-11-24T05:49:27Z

On Tue, Nov 24, 2009 at 01:26:43AM -0800, Sergei Steshenko wrote:

>
>
> --- On Mon, 11/23/09, Joel Roth <joelz@...> wrote:
>
> > From: Joel Roth <joelz@...>
> > Subject: Re: [ecasound] obviously wrong ALSA underruns reported by ecasound-2.6.0
> > To: ecasound-list@...
> > Date: Monday, November 23, 2009, 3:06 PM
> > On Mon, Nov 23, 2009 at 02:46:23PM
> > -0800, Sergei Steshenko wrote:
> > > Hello,
> > >
> > > I've just the following:
> > >
> > > "
> > > WARNING: ALSA playback underrun, glitches in audio
> > playback possible! Break was at least 566281884.09 ms long.
> > > "
> > >
> > > In reality the drop was like 1 second, maybe less.
> > >
> > > Is it a known issue ?
> >
> > I've observed this, and I think posted to the list once
> > before.
> >
> > Regards,
> >
> > Joel
> >
> >
> > > Which of the two - ecasound or ALSA - reports
> > underruns ?
> > >
> > > I'd say, I've never seen a reasonable number since
> > upgrade to both
> > > SUSE-11.1 (i.e. newer ALSA) and ecasound-2.6.0 - from
> > ecasound-2.5.2
> > >
> > > Both versions of ecasound are self-built using the
> > same tool of mine.
> > >
> > > Thanks,
> > > Sergei.
> > >
> > > P.S. A recent bunch:
> > >
> > > WARNING: ALSA playback underrun, glitches in audio
> > playback possible! Break was at least 566281807.81 ms long.
> > > WARNING: ALSA playback underrun, glitches in audio
> > playback possible! Break was at least 566281802.92 ms long.
> > > WARNING: ALSA playback underrun, glitches in audio
> > playback possible! Break was at least 566281849.38 ms long.
> > > WARNING: ALSA playback underrun, glitches in audio
> > playback possible! Break was at least 566281858.54 ms long.
> > >
> > >
> > >
> > >
> > >
> > >
> > ------------------------------------------------------------------------------
> > > Let Crystal Reports handle the reporting - Free
> > Crystal Reports 2008 30-Day
> > > trial. Simplify your report design, integration and
> > deployment - and focus on
> > > what you do best, core application coding. Discover
> > what's new with
> > > Crystal Reports now. http://p.sf.net/sfu/bobj-july
> > > _______________________________________________
> > > Ecasound-list mailing list
> > > Ecasound-list@...
> > > https://lists.sourceforge.net/lists/listinfo/ecasound-list
> > --
> > Joel Roth
> >
>
> Do you know whether this has been fixed in later versions ?
>
> Thanks,
> Sergei.

Not that I know of. Maybe Kai has something to say about it.

>
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--
Joel Roth

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Re: replaygain with ecasound

2009-11-24T05:21:57Z

Hallo everybody,

first of all thanks for the prompt response. I decided to use a less
accurate calculation of the apparent volume of the track, by emulating
the ecasignalview tool. (actually it would be nice if the
ecasignalview would return the average volume after the file has
ended).

In practice I calculate the average of the volume of the track as read
by the bar meters. It's a bit coarse but it is sort of how you would
do it for real with a mixer and a good hear.

Find below the code that I wrote and that seems to give good results.
At least now the volumes don't look to be so different (which is what
I wanted).

I run getFileLengthVolume() on some files and set the volume of the
background music as -20dB with respect to the foreground sound.

Ciao

Edo

#reproduce ecalength and calculates the average signal power
def getFileLengthVolume(filename):
e = ECA_CONTROL_INTERFACE(0)
e.command("cs-add main")
e.command("c-add main")
e.command("ai-add "+filename)
e.command("ao-add null")
e.command("cs-set-audio-format wav")
e.command("cop-add -evp");
e.command("cop-add -ev");
e.command("cop-select 1");
e.command("cs-connect")
e.command("start")
i = 1
dx = 0.
sx = 0.
while 1:
#time.sleep(1)
e.command("copp-select 1");
e.command("copp-get");
sx += e.last_float()
e.command("copp-select 2");
e.command("copp-get");
dx += e.last_float()
i += 1
e.command("engine-status")
if e.last_string() != "running":
print e.last_error()
break
e.command("get-position")
curpos = e.last_float()
e.command("stop")
e.command("cs-disconnect")
e.command("cop-status")
return curpos , sx/i , dx/i

On Mon, Nov 16, 2009 at 6:59 PM, Antonio P. P. Almeida
<perusio@...> wrote:

> On 16 Nov 2009 11h55 WET, edo.paskino@... wrote:
>
>> Hallo Julien,
>>
>> the answer is: in a frequency sense. In particular, I know what is
>> the cut of the filter at each frequency by means of a function. For
>> instance I know that at 100Hz it cuts -3dB and at 110Hz it cuts
>> 2.5dB and so forth.
>>
>> So I can provide what is the cut of the filter in the range 20-20000
>> Hz in steps of, say 100 Hz.
>>
>> I hope I clarified my point
>>
>> Thank you very much
>
> Hello Edoardo,
>
> Like Julien wrote there's always the possibility of using a more
> "granular" tool. You can use Snd
> https://ccrma.stanford.edu/software/snd/snd/snd.html, for instance,
> which allows sample level processing in the frequency domain.
>
> If you're using a Linux distribution there's bound to be a Snd
> package for you to install. In debian is called snd-gtk.
>
> Snd requires using Guile (Scheme), Ruby or Forth.
>
> But if I understand correctly I think that you can get the effect you
> want by using one of the several filter operators it provides.
>
> There's an example in the docs for ecassound interactive where a
> low-pass brickwall filter gets its cuttoff frequency shifted every
> second.
>
> Does this suits you? Or do you need to implement a filter with a
> specific transfer function? If it's the later then you'll have to use Snd,
> or CLM, or csound like Julien suggested.
>
> Regards,
>
> --- appa
>
>

--
Edo
weblog http://www.edo.imanetti.net/wordpress
podcast http://sieegiueeaccaso.blogspot.com
:wq

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Re: obviously wrong ALSA underruns reported by ecasound-2.6.0

2009-11-24T01:26:43Z

--- On Mon, 11/23/09, Joel Roth <joelz@...> wrote:

> From: Joel Roth <joelz@...>
> Subject: Re: [ecasound] obviously wrong ALSA underruns reported by ecasound-2.6.0
> To: ecasound-list@...
> Date: Monday, November 23, 2009, 3:06 PM
> On Mon, Nov 23, 2009 at 02:46:23PM
> -0800, Sergei Steshenko wrote:
> > Hello,
> >
> > I've just the following:
> >
> > "
> > WARNING: ALSA playback underrun, glitches in audio
> playback possible! Break was at least 566281884.09 ms long.
> > "
> >
> > In reality the drop was like 1 second, maybe less.
> >
> > Is it a known issue ?
>
> I've observed this, and I think posted to the list once
> before.
>
> Regards,
>
> Joel
>
>
> > Which of the two - ecasound or ALSA - reports
> underruns ?
> >
> > I'd say, I've never seen a reasonable number since
> upgrade to both
> > SUSE-11.1 (i.e. newer ALSA) and ecasound-2.6.0 - from
> ecasound-2.5.2
> >
> > Both versions of ecasound are self-built using the
> same tool of mine.
> >
> > Thanks,
> > Sergei.
> >
> > P.S. A recent bunch:
> >
> > WARNING: ALSA playback underrun, glitches in audio
> playback possible! Break was at least 566281807.81 ms long.
> > WARNING: ALSA playback underrun, glitches in audio
> playback possible! Break was at least 566281802.92 ms long.
> > WARNING: ALSA playback underrun, glitches in audio
> playback possible! Break was at least 566281849.38 ms long.
> > WARNING: ALSA playback underrun, glitches in audio
> playback possible! Break was at least 566281858.54 ms long.
> >
> >
> >
> >
> >
> >
> ------------------------------------------------------------------------------
> > Let Crystal Reports handle the reporting - Free
> Crystal Reports 2008 30-Day
> > trial. Simplify your report design, integration and
> deployment - and focus on
> > what you do best, core application coding. Discover
> what's new with
> > Crystal Reports now. http://p.sf.net/sfu/bobj-july
> > _______________________________________________
> > Ecasound-list mailing list
> > Ecasound-list@...
> > https://lists.sourceforge.net/lists/listinfo/ecasound-list
> --
> Joel Roth
>

Do you know whether this has been fixed in later versions ?

Thanks,
Sergei.

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Re: obviously wrong ALSA underruns reported by ecasound-2.6.0

2009-11-23T15:06:32Z

On Mon, Nov 23, 2009 at 02:46:23PM -0800, Sergei Steshenko wrote:

> Hello,
>
> I've just the following:
>
> "
> WARNING: ALSA playback underrun, glitches in audio playback possible! Break was at least 566281884.09 ms long.
> "
>
> In reality the drop was like 1 second, maybe less.
>
> Is it a known issue ?

I've observed this, and I think posted to the list once before.

Regards,

Joel

> Which of the two - ecasound or ALSA - reports underruns ?
>
> I'd say, I've never seen a reasonable number since upgrade to both
> SUSE-11.1 (i.e. newer ALSA) and ecasound-2.6.0 - from ecasound-2.5.2
>
> Both versions of ecasound are self-built using the same tool of mine.
>
> Thanks,
> Sergei.
>
> P.S. A recent bunch:
>
> WARNING: ALSA playback underrun, glitches in audio playback possible! Break was at least 566281807.81 ms long.
> WARNING: ALSA playback underrun, glitches in audio playback possible! Break was at least 566281802.92 ms long.
> WARNING: ALSA playback underrun, glitches in audio playback possible! Break was at least 566281849.38 ms long.
> WARNING: ALSA playback underrun, glitches in audio playback possible! Break was at least 566281858.54 ms long.
>
>
>
>
>
> ------------------------------------------------------------------------------
> Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day
> trial. Simplify your report design, integration and deployment - and focus on
> what you do best, core application coding. Discover what's new with
> Crystal Reports now. http://p.sf.net/sfu/bobj-july
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obviously wrong ALSA underruns reported by ecasound-2.6.0

2009-11-23T14:46:23Z

Hello,

I've just the following:

"
WARNING: ALSA playback underrun, glitches in audio playback possible! Break was at least 566281884.09 ms long.
"
.

In reality the drop was like 1 second, maybe less.

Is it a known issue ?

Which of the two - ecasound or ALSA - reports underruns ?

I'd say, I've never seen a reasonable number since upgrade to both
SUSE-11.1 (i.e. newer ALSA) and ecasound-2.6.0 - from ecasound-2.5.2

Both versions of ecasound are self-built using the same tool of mine.

Thanks,
Sergei.

P.S. A recent bunch:

WARNING: ALSA playback underrun, glitches in audio playback possible! Break was at least 566281807.81 ms long.
WARNING: ALSA playback underrun, glitches in audio playback possible! Break was at least 566281802.92 ms long.
WARNING: ALSA playback underrun, glitches in audio playback possible! Break was at least 566281849.38 ms long.
WARNING: ALSA playback underrun, glitches in audio playback possible! Break was at least 566281858.54 ms long.

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pyeca creating extra empty .wav files when recording?

2009-11-21T05:41:40Z

I am using the python interface to ecasound to record short clips from
microphone for a language learning program.

I have just hit a new problem, and I'm not sure why this has just
started happening. After initialising by creating a chainsetup, chain
and input routing, I am using a sequence of commands like this:

cs-select rec_chainsetup
c-select 1st_chain
ao-add myfile1.wav
cs-connect
start
<make the recording>
stop
ao-get-length (because I need to know the length of the recording)
cs-disconnect

All that works fine. After that, I do some post-processing and
convert the file to another format, so myfile1.wav is eventually
removed.

However, when I do the same sequence again for the next recording
(myfile2.wav), ecasound creates an extra, empty (only header: 44
bytes) .wav file with the same name as the first time, i.e.
myfile1.wav. The third time, it creates 2 empty .wav files, and so
on. They keep multiplying.

I wondered if this was because I never actually ao-remove the first
file from the chain. So, I tried doing that immediately after
cs-disconnect. However, now when I try to make the second recording,
I get a 'broken pipe' error.

What am I doing wrong? Why has this problem suddenly appeared? (I
don't think I have changed the sequence of commands recently.) I am
using ecasound 2.4.6.1-2 in Ubuntu 8.10.

Peter Clarke

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ECA Sound on Snow Leopard

2009-11-16T14:59:56Z

I was just wondering if there where any os x ECA Sound users who had ECA Sound successfully running on Snow Leopard?

It seems to have some 32/64bit problems with some of the libraries when I tried to build it..

Justin Maynard
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Re: replaygain with ecasound

2009-11-16T09:59:22Z

On 16 Nov 2009 11h55 WET, edo.paskino@... wrote:

> Hallo Julien,
>
> the answer is: in a frequency sense. In particular, I know what is
> the cut of the filter at each frequency by means of a function. For
> instance I know that at 100Hz it cuts -3dB and at 110Hz it cuts
> 2.5dB and so forth.
>
> So I can provide what is the cut of the filter in the range 20-20000
> Hz in steps of, say 100 Hz.
>
> I hope I clarified my point
>
> Thank you very much

Hello Edoardo,

Like Julien wrote there's always the possibility of using a more
"granular" tool. You can use Snd
https://ccrma.stanford.edu/software/snd/snd/snd.html, for instance,
which allows sample level processing in the frequency domain.

If you're using a Linux distribution there's bound to be a Snd
package for you to install. In debian is called snd-gtk.

Snd requires using Guile (Scheme), Ruby or Forth.

But if I understand correctly I think that you can get the effect you
want by using one of the several filter operators it provides.

There's an example in the docs for ecassound interactive where a
low-pass brickwall filter gets its cuttoff frequency shifted every
second.

Does this suits you? Or do you need to implement a filter with a
specific transfer function? If it's the later then you'll have to use Snd,
or CLM, or csound like Julien suggested.

Regards,

--- appa

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Re: replaygain with ecasound

2009-11-16T04:14:56Z

Hello Edoardo!
OK, so it seems you need s special filter plugin, that can do the job for
you. Would a standard filter be good enough for you. There are the Glame
Filter plugins, LADSPA unique ID around 1890-1894 or 95. There are lp, hp, bp
and so on. You can tell the filter how many poles it should have and you can
define the cutoff.
Another possibility would be an EQ, with finer grained control over certain
bands. There the FIL plugin by Fons Adriaenson. It has four bands with
adjustable bandwith, frequency, gain and you can ask them to be on or off. You
could use this plugin a couple of times to implement the curve you need.
Beyond that, I know that there's JaMIN, which is graphical, but also has a
graphical EQ, which, I thik, you can control very finely.
Or if you want to stay text-based and do batch-processing, you could go on
to csound or CLM. Both synthesizer programming languages. But you can write
instruments - something of a processing script - to get your job done. They
both possess a lot of filters, envelope functions and more to adjust
everything you need.
I know a bit about csound, so if all the ecasound suggestions don't meet
your needs, I might give you a simple example of how to implement something,
that filters a soundfile for you. I't s not too dificult.
I hope that helps a bit.
Kindest regards
Julien

--------
Music was my first love and it will be my last (John Miles)

======== FIND MY WEB-PROJECT AT: ========
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: =======
http://www.juliencoder.de

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Re: replaygain with ecasound

2009-11-16T03:55:55Z

Hallo Julien,

the answer is: in a frequency sense. In particular, I know what is the
cut of the filter at each frequency by means of a function.
For instance I know that at 100Hz it cuts -3dB and at 110Hz it cuts
2.5dB and so forth.

So I can provide what is the cut of the filter in the range 20-20000
Hz in steps of, say 100 Hz.

I hope I clarified my point

Thank you very much

Edoardo

On Mon, Nov 16, 2009 at 12:11 PM, Julien Claassen <julien@...> wrote:

> Hello Edoardo!
> I've worked with Ecasound for years and I've come to see a bit of it. :-0
> But I don't understand your problem description 100%.
> How do you mean: The curve of the filter? In a time sese or in a sense,
> that describes the filter itself (like a special chracteristic of the
> filter)?
> If you could point a bit for me, I might be able to help you.
> Kindest regards
> Julien
>
> --------
> Music was my first love and it will be my last (John Miles)
>
> ======== FIND MY WEB-PROJECT AT: ========
> http://ltsb.sourceforge.net
> the Linux TextBased Studio guide
> ======= AND MY PERSONAL PAGES AT: =======
> http://www.juliencoder.de
>

Re: replaygain with ecasound

2009-11-16T03:11:14Z

Hello Edoardo!
I've worked with Ecasound for years and I've come to see a bit of it. :-0
But I don't understand your problem description 100%.
How do you mean: The curve of the filter? In a time sese or in a sense, that
describes the filter itself (like a special chracteristic of the filter)?
If you could point a bit for me, I might be able to help you.
Kindest regards
Julien

--------
Music was my first love and it will be my last (John Miles)

======== FIND MY WEB-PROJECT AT: ========
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: =======
http://www.juliencoder.de

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replaygain with ecasound

2009-11-15T04:46:43Z

Dear All,

I use ecasound to edit my podcast and I'm very satisfied. The
equalization of the tracks is nevertheless a bit tricky and sometimes
I don't manage to setup a good volume balance.

I wanted to implement a program (with pyeca) to calculate the ReplayGain.

As a first step I have to apply a digital filter to the audio data;
let's say I have a curve describing the transfer function of the
filter, how do I apply this to the signal with ecasound?

Thanks

Edoardo

--
Edo
weblog http://www.edo.imanetti.net/wordpress
podcast http://sieegiueeaccaso.blogspot.com
:wq

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Re: AAC batch conversion with Ecasound?

2009-11-01T03:46:54Z

Hi,

On Thu, 29 Oct 2009, Xavier G wrote:

> Our website members are uploading large mp3 audio files days after days on our servers.
> We currently try to convert on the fly these mp3 files once uploaded with ecasound to aac files
>
> We use this command
> for i in mp3/*; do ecasound -i resample,96000,$i -o aac/`basename $i .mp3`.aac; done

btw, one thing to note that you lose some audio quality in this conversion
(from one lossy encoding to another, e.g. from mp3 to aac). You are
probably already aware of this, but just as a reminder to others.

> The result is fine, but it seems we get aac files encoded with vbr
> http://1clickcd.com/convert/aac/file-88815-1256688222.aac

Ecasound uses 'faac' utility [1] to encode aac files. You can alter the
bitrate settings by modifying the 'ext-cmd-aac-output' setting in
ecasoundrc. E.g. changing...

> ext-cmd-aac-output = faac -P -o %f -R %s -b 96 -C %c -

... unfortunately, it would seem faac doesn't support CBR mode at all. So
your best bet is to play with '-q' and '-b' to reduce the variance. E.g.
faac 1.24 has options:

Quality-related options:
-q <quality> Set default variable bitrate (VBR) quantizer quality in percent
(default: 100, averages at approx. 120 kbps VBR for a normal
stereo input file with 16 bit and 44.1 kHz sample rate; max.
value 500, min. 10).Set quantizer quality.
-b <bitrate> Set average bitrate (ABR) to approximately <bitrate> kbps.
-c <freq> Set the bandwidth in Hz (default: automatic, i.e. adapts
maximum value to input sample rate).

One option would be to plug in another AAC codec to ecasound, but there
are at least no open-source options. One closed-source option is:

http://www.audiocoding.com/nero_aacenc.html

... that requires a license for commercial use, but works in Linux. You
can't plug that to ecasound (almost works, but it require a RIFF header
written to its stdin which ecasound cannot do), so you have to run it
separately.

[1] http://www.audiocoding.com/

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Re: DBC warnings with ecasound v2.7.0+dev-20091011

2009-11-01T03:21:05Z

On Mon, Oct 26, 2009 at 09:12:36PM +0100, Julien Claassen wrote:
> Hello all!
> I got the following warnings, I know, that they are not deadly, but I think
> they aren't there just for fun. So I hope this can help in improving stability
> and the like.
>
> Warning: type DBC_CHECK soft-assert 'is_prepared() != true' failed at
> -> audioio-device.cpp:38 [(null)]
> Warning: type DBC_CHECK soft-assert 'shutdown_request_rep != true' failed at
> -> audioio_jack_manager.cpp:1546 [void AUDIO_IO_JACK_MANAGER::open(int)]

I get similar warnings using Ecasound version 2.6.0+pre via Net-ECI

Warning: DBC_CHECK failed - "is_prepared() != true",
audioio-device.cpp, 38.
Warning: DBC_CHECK failed - "is_prepared() != true",
audioio-device.cpp, 38.
Warning: DBC_CHECK failed - "shutdown_request_rep != true",
audioio_jack_manager.cpp, 1534.

Here is the setup, as stored by cs-save.

# general
-b:1024 -r:50 -z:nointbuf -z:db,100000 -n:"/otherroot/home/jroth/nama/untitled/Setup.ecs" -X -z:noxruns -z:psr -z:mixmode,sum

# audio inputs
-a:1 -f:f32_le,2,44100 -i:loop,Master_in
-a:3,R3 -f:f32_le,2,44100 -i:jack_multi,system:capture_1,,

# audio outputs
-a:1 -f:f32_le,2,44100 -o:jack,system,
-a:3 -f:f32_le,2,44100 -o:loop,Master_in
-a:R3 -f:s16_le,1,44100 -o:/otherroot/home/jroth/nama/untitled/.wav/brass_21.wav,

# chain operators and controllers
-a:1 -epp:50.00 -ea:100.00
-a:3 -chcopy:1.00,2.00 -epp:50.00 -ea:100.00

Also, I notice there are a lot of format statements that aren't in
my original setup (below), probably some defaults whose influence
I should consider.

# general

-B auto -r -z:mixmode,sum -z:psr

# audio inputs

-a:1 -i:loop,Master_in
-a:3,R3 -i:jack_multi,system:capture_1

# post-input processing

-a:3 -chcopy:1,2

# pre-output processing

-a:R3 -f:s16_le,1,44100,i

# audio outputs

-a:1 -f:f32_le,2,44100 -o:jack,system
-a:3 -o:loop,Master_in
-a:R3 -f:s16_le,1,44100,i -o:/otherroot/home/jroth/nama/untitled/.wav/brass_21.wav

Regards,

Joel
--
Joel Roth

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Re: AAC batch conversion with Ecasound?

2009-10-29T04:50:43Z

Hi,

We are using ecasound for 4 yeras now and love it :)

We have a question :

Our website members are uploading large mp3 audio files days after days on our servers.
We currently try to convert on the fly these mp3 files once uploaded with ecasound to aac files

We use this command
for i in mp3/*; do ecasound -i resample,96000,$i -o aac/`basename $i .mp3`.aac; done

The result is fine, but it seems we get aac files encoded with vbr

http://1clickcd.com/convert/aac/file-88815-1256688222.aac

Is there anything to do in the command or in the settings to get constant bitratre instead of variable?

Thanks a lot

Xavier Galliot
+33682833615

find below the content of out ecasounrc file :

ecasound-version = 2.4.6.1
midi-device = rawmidi,/dev/midi
default-output = autodetect
default-audio-format = s16_le,2,44100,i
default-to-precise-sample-

rates = false
default-to-interactive-mode = false
default-mix-mode = avg
bmode-defaults-nonrt = 1024,false,50,false,100000,true
bmode-defaults-rt = 1024,true,50,true,100000,true
bmode-defaults-rtlowlatency = 256,true,50,true,100000,false
resource-directory = /usr/share/ecasound2.2/ecasound
resource-file-genosc-envelopes = generic_oscillators
resource-file-effect-presets = effect_presets
ladspa-plugin-directory = /usr/lib/ladspa
ext-cmd-text-editor = pico
ext-cmd-text-editor-use-getenv = true
ext-cmd-wave-editor = ecawave
ext-cmd-mp3-input = mpg123 --stereo -r %s -b 0 -q -s -k %o %f
ext-cmd-mp3-output = lame -b %B -s %S -x -S - %f
ext-cmd-ogg-input = ogg123 -d raw -o byteorder:%E --file=- %f
ext-cmd-ogg-output = oggenc -b %B --raw --raw-bits=%b --raw-chan=%c --raw-rate=%s --raw-endianness 0 --output=%f -
ext-cmd-mikmod = mikmod -d stdout -o 16s -q -f %s -p 0 --noloops %f
ext-cmd-timidity = timidity -Or1S -id -s %s -o - %f
ext-cmd-flac-input = flac -d -c %f
ext-cmd-flac-output = flac -o %f -f --force-raw-format --channels=%c --bps=%b --sample-rate=%s --sign=%I --endian=%E -
ext-cmd-aac-input = faad -w -b 1 -f 2 -d %f
ext-cmd-aac-output = faac -P -o %f -R %s -b 96 -C %c -

2009/7/15 Xavier G <j@...>

Hi, we have ecasound installed on our file server with 1 T of mp3 files.
We need to convert all our mp3 files (mostely 100 Mo mp3 files) in AAC 96kbps.

Is there a way to do it in batch with ecasound?

If not does anybody have a alternate solution?

Thanks a lot for your help.

Javier

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A short addition

2009-10-26T13:18:45Z

Hi again!
With the same version and neteci again, I do get this:
Warning: type DBC_ENSURE soft-assert 'db_clients_rep > 0' failed at
-> eca-chainsetup.cpp:677 [void
ECA_CHAINSETUP::switch_to_db_mode_helper(std::v
ector<AUDIO_IO*, std::allocator<AUDIO_IO*> >*, const std::vector<AUDIO_IO*,
std:
:allocator<AUDIO_IO*> >&)]
Warning: type DBC_ENSURE soft-assert 'db_clients_rep > 0' failed at
-> eca-chainsetup.cpp:677 [void
ECA_CHAINSETUP::switch_to_db_mode_helper(std::v
ector<AUDIO_IO*, std::allocator<AUDIO_IO*> >*, const std::vector<AUDIO_IO*,
std:
:allocator<AUDIO_IO*> >&)]
With this setup:
# ecasound chainsetup file

# general

-b:64 -z:mixmode,sum

# audio inputs

-a:1 -i:loop,Master_in
-a:3 -i:jack_multi,system:capture_5,system:capture_6

# post-input processing

# pre-output processing

# audio outputs

-a:1 -f:f32_le,10,48000 -o:jack,system
-a:3 -o:loop,Master_in

It is a bit disturbing to see all those messages coming through on the
console. This again happend, just while readying the engine. I suppose it's
engine-launch that's used.
Best regards
Julien

--------
Music was my first love and it will be my last (John Miles)

======== FIND MY WEB-PROJECT AT: ========
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: =======
http://www.juliencoder.de

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DBC warnings with ecasound v2.7.0+dev-20091011

2009-10-26T13:12:36Z

Hello all!
I got the following warnings, I know, that they are not deadly, but I think
they aren't there just for fun. So I hope this can help in improving stability
and the like.
=============================================================================
1 Master -- mon MON -- -- 100 50
2 Mixdown -- mon OFF -- -- 100 50
3 piano 1 rec MON [5] -- 100 50

[ live input disabled, playback ]

Warning: type DBC_CHECK soft-assert 'is_prepared() != true' failed at
-> audioio-device.cpp:38 [(null)]
Warning: type DBC_CHECK soft-assert 'shutdown_request_rep != true' failed at
-> audioio_jack_manager.cpp:1546 [void AUDIO_IO_JACK_MANAGER::open(int)]

I got these warnings while readying the engine using neteci and the
following chainsetup:
Setup.ecs;
# ecasound chainsetup file

# general

-b:64 -z:mixmode,sum

# audio inputs

-a:1 -i:loop,Master_in
-a:3 -i:/home/songs/long/nama/test/.wav/piano_1.wav
-a:4,5 -i:loop,piano_insert
-a:6 -f:f32_le,2,48000 -i:jack,jconv

# post-input processing

-a:3 -chcopy:1,2

# pre-output processing

# audio outputs

-a:1 -f:f32_le,10,48000 -o:jack,system
-a:3 -o:loop,piano_insert
-a:5 -f:f32_le,2,48000 -o:jack,jconv
-a:6,4 -o:loop,Master_in

This didn't happen without the additional chains for the jconv-insert. that
setup should have looked like this:
Old Setup.ecs:
# ecasound chainsetup file

# general

-b:64 -z:mixmode,sum

# audio inputs

-a:1 -i:loop,Master_in
-a:3 -i:/home/songs/long/nama/test/.wav/piano_1.wav

# post-input processing

-a:3 -chcopy:1,2

# pre-output processing

# audio outputs

-a:1 -f:f32_le,10,48000 -o:jack,system
-a:3 -o:loop,Master_in
If there's more, that I can do to hlep here, just let me know.
I'm running all these setups from Nama.
Kindest regards
Julien

--------
Music was my first love and it will be my last (John Miles)

======== FIND MY WEB-PROJECT AT: ========
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: =======
http://www.juliencoder.de

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Re: Problem with 2.7.0

2009-10-19T22:11:34Z

Hi,

Thanks first for having fixed this problem and for this precise
and convincing explanation !

Regards

Pierre

From: Kai Vehmanen <kvehmanen@...>
Subject: Re: [ecasound] Problem with 2.7.0
Date: Tue, 20 Oct 2009 00:47:11 +0300 (EEST)

> Hi,
>
> On Mon, 19 Oct 2009, Pierre Lorenzon wrote:
>
>>> hmm, ok, I have one hypothesis. Here's one snapshot, let me
>>> know if it helps:
>>
>> Yes it does ! The problem no longer exists with this version.
>
> ok, great, that is really good news. I now wrote a longer mail
> about this to ALSA list:
> http://permalink.gmane.org/gmane.linux.alsa.devel/67752
>
> The good news is that this bug should not hit too many
> people. Most ALSA PCMs support power-of-two period-sizes
> (-b:1024, -b:128, ...), and in these cases no glitches should
> not occur.
>
> In your specific case, I think the PCM doesn't support -b:1024
> but something close, as the block/period size. Now due to the
> bug this results in discontinuity at every block, leading to
> glitches at a rate of ~20Hz when playing 44.1kHz audio. This
> matches pretty closely what you described.
>
>>> Also, I'd be interested to know which version of alsa-lib
>> 1.0.13
>>> (libasound) do you have, and which kernel version?
>> 2.6.22.5
>
> These match as well. These still have the old implementation of
> snd_pcm_sw_params_set_xfer_align both in alsa-lib and in
> kernel.
>
> Thanks for helping to track this down! I'll release 2.7.1
> soonish with this fix (and possibly a few others as well)
> included.

Re: Problem with 2.7.0

2009-10-19T14:47:11Z

Hi,

On Mon, 19 Oct 2009, Pierre Lorenzon wrote:

>> hmm, ok, I have one hypothesis. Here's one snapshot, let me
>> know if it helps:
>
> Yes it does ! The problem no longer exists with this version.

ok, great, that is really good news. I now wrote a longer mail about this
to ALSA list:
http://permalink.gmane.org/gmane.linux.alsa.devel/67752

The good news is that this bug should not hit too many people. Most ALSA
PCMs support power-of-two period-sizes (-b:1024, -b:128, ...), and in
these cases no glitches should not occur.

In your specific case, I think the PCM doesn't support -b:1024 but
something close, as the block/period size. Now due to the bug this results
in discontinuity at every block, leading to glitches at a rate of ~20Hz
when playing 44.1kHz audio. This matches pretty closely what you
described.

>> Also, I'd be interested to know which version of alsa-lib
> 1.0.13
>> (libasound) do you have, and which kernel version?
> 2.6.22.5

These match as well. These still have the old implementation of
snd_pcm_sw_params_set_xfer_align both in alsa-lib and in kernel.

Thanks for helping to track this down! I'll release 2.7.1 soonish with
this fix (and possibly a few others as well) included.

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