there's a hidden link somewhere to the tech docs we have on various
chips. i'd like to find the one for the trident.
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>background: afaik, with the ice1712 for example it is possible to
>run two or more cards in sync. will this be as simple as calling
>snd_pcm_open for n streams, and snd_pcm_link (stream[0]) for
>streams [1, n - 1]?
no, i don't think that's the plan. you use the "multi" PCM device type
in an ~/.
It looks like that the capture direction receives an interrupt before the
>capture pointer has reached the period size boundary. You can increase the
>insterrupt delay, something like this:
i'll check on that, but i don't think thats what happened above. this
should make it clearer (i processed t
In message <[EMAIL PROTECTED]>you write:
>dave willis wrote:
>>
>> On Thu, 29 Nov 2001, Abramo Bagnara wrote:
>>
>> > Q: with the ice1712 it is possible to run two or more cards in sync
>> > A: Yes, if you're not thinking to share the same clock chip (i.e. some
>> > drift is possible).
>>
>> wh
>Q: with the ice1712 it is possible to run two or more cards in sync
>A: Yes, if you're not thinking to share the same clock chip (i.e. some
>drift is possible).
My interpretation: two cards will drift. or something. can you be more
clear about what you mean? they will drift if you do use the sam
In message <[EMAIL PROTECTED]>you write:
>Andy Wingo wrote:
>>
>> "hw:N,M": the Nth card, with the Mth pcm device. N and M start with 0.
>> this device does not use the software plugin layer.
>
>"hwplug:2,0" was the setting was the one I was looking for. Dunno what it
>means, but seems to work. "
>it's been the same with MOTU, some vendors are just ignorant of a
>large potential customer base.
lets not kid ourselves. there is a *tiny* potential customer base for
the next several years at least, and the existing customer base is
even smaller.
the customer base for audio chipsets in thing
>i seriously doubt that. perhaps it's true for additional companies, as
>hammerfall is well established in the high-end and m-audio (and terratec)
how many hammerfall purchasers do you think run linux? even the fact
that the most impressive and forward-thinking demo of the last 2 years
(mcgill's
>If hardware manufactures wanted their products to have good support on
>Linux, all they have to do is publish the hardware programming details, and
>the linux community will do the actual driver development.
thats why i wrote "inertia-bound companies". they don't see this. they
think that linux
>I made such circuit on AT89C52 microcontroller, but chip is too slow for
>that job.
>So I've diecided to use PC as a "fast microcontroller".
>But why not to write kernel module - the PC would stand on Linux.
>
>So I have to write special driver for this amplifier.
>I think that writting it in ass
>> Linux runs on at least a dozen h/w platforms. AFAIK, no device driver
>> directly includes assembler, and if they do, they are unlikely to be
>> part of the mainstream kernel. They certainly wouldn't be part of
>> ALSA, I would hope.
>>
>
>a part of assembly-howto :)
[ ... example elided ...
>About GNU assembler I wanted to say only that it's multiplatform...
yes it is. but "mov eax, edx" is not, and never will be, regardless of
what assembler you use.
--p
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In message <[EMAIL PROTECTED]>you write:
>Hi. I'm writing the alsa plugin for gstreamer. When I go to capture audio, my
>loop eats the entire cpu. I set avail_min but still sometimes get out 32 frame
>avail values. Here's the relevant portion:
>
>/* shamelessly stolen from pbd's audioengine. thank
>there are two of these loops running separately, one for each
>element, with separate handles, etc. I was thinking for some reason
>that snd_pcm_update_avai l would block until avail_min frames were
>available, but that's not the case. If I run the following pipeline,
>I get:
snd_pcm_avail_updat
>I just came to a pseudo-epiphany, but I can't test it atm. Do I poll
>on the fd , but set pfd.events = POLLIN | POLLERR?
there now, that wasn't too hard, was it? :))
the rest of the code should look identical.
note that audioengine (and now JACK) is technically in error because
it only polls
Don't laugh. This is from a review of the Creamware Luna II card, from
Remix magazine:
--
With some direction from CreamWare's technical support staff, I found
that Fruityloops works best with DirectSound drivers, whereas Acid Pro
>Don't make this mistake: poll *have* to return immediately in *all*
>cases where to wait is useless (i.e. when no non-user driven event may
>happen).
>
>This is the rule you need to remember.
where does this rule come from? i was under the impression that
poll(2) should timeout in those cases, n
>> well, if there are > 2 threads, one only polling and others doing
>> management work -- and this is probably the most useful approach to low-
>> latency PCM IO -- it is imaginable that a non-polling thread starts and
>> stops the device.
>
>In this case pthread_cond_wait and pthread_cond_signal
>It's really discarded as spurious? All voices has separate spurious
>interrupt checking based on the global frame counter (stimer). Please, can
>you verify it? I still think, that it would be better to call
>snd_trident_capture_pointer() and see the actual ptr. My patch should
>fix the problem, i
>>From the RME documentation it appears that it is possible to detect when a
>source is connected to the RME 9652 card, and whether the source has the
>sampling rate expected. But when I try to read from the card using ALSA, I
>get data whether or not the card has any input. Is there a way to dete
>I would like to use the sound card as a clock source for a DVD player.
>I have a SB Live with alsa drivers 0.9.x.
>
>Which API call can I make to the sound card which is equivalent to the
>gettimeofday call?
>Also, what will be the units of time received from the call?
there is no direct API cal
well, jaroslav's patch (to change the ESO value) was most of the
solution. in the diff below, i have also cleaned up the spurious
interrupt detection. with these two changes, i can run my trident with
a configuration that previously caused endless xruns in full duplex
mode (44.1kHz, 64 frame perio
>Your patch doesn't take care about frequencies other than 48000Hz, because
>stimer is absolute 48000Hz timer, so the spurious threshold should be
>adjusted for other sample rates as well. I've put your code with my
>modifications to the CVS tree.
my attempt was to make it independent of srate by
>OK. I modified that full duplex program to store the # of available
>samples for input and output in a history buffer. This history is then
>printed back at the end. The numbers confirm that the input and output
>are drifting and are not synchronous. The OUTPUT stream always has
>period size samp
>#ifdef __KERNEL__
>
>#define ATOMIC_INIT(i) { (i) }
>
>#define atomic_read(v) ((v)->counter)
>#define atomic_set(v,i) (((v)->counter) = (i))
>
>
>etc.
>
>Not using atomic reads where they are necessary would seem like a bad
>idea. Are the ones in pcm_meter (and maybe they are u
>i'm doing full-duplex mmapped io on emu8k. the good news is the streams
>stay in sync, the bad news is how the processing occurs is not optimal if
>one follows the 'schoolbook approach'.
>
>remember the alsa schoolbook:
>
>while (running)
>{
> poll();
> available = min (streams->update_availabl
in doing lots of testing on JACK, i've noticed that although the
trident driver now works (there were some patches from jaroslav and
myself), in general i still get xruns with the lowest possible latency
setting for that card (1.3msec per interrupt, 2.6msec buffer). with
the same settings on my ha
>Perhaps only because error will be returned on next call and taken for
>granted that a partial transfer definitely imply a next call, they have
>considered the early return of error a little benefit.
not that we have that problem, but this design really screwed up
multithreaded programs too, bec
>i'd make snd_pcm_mmap_commit return -errno in case of error and number of
>frames commited otherwise.
>that's approximately what read/write(2) do, and that's what i'd prefer.
i'm with tim.
--p
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>but sometimes i also get inexplicable corruption within ordinary
>dynamically allocated memory of the process before it exits.
>
>i have spent considerable time on verifying that these are indeed
>caused by changing the period_size, and not by my own code.
i would be too certain of this conclus
>I'm thinking for the benefits that low latency audio applications will
>have from preemption of running task in favour of the waiting task just
>at end of interrupt handler execution (instead of end of time slice).
>
>I don't believe that such a mechanism is already in place in linux
>kernel but
In message <[EMAIL PROTECTED]>you write:
>anyone know why when viewing a flash plugin in netscape with audio and
>alsa the sound is super-choppy? i don't get this problem with oss. using
>ice1712 and alsa-cvs-10/20/01.
i'm using a much older version of ALSA with the trident and viewing
madblast
>Nothing I said was new of course (as I tried to note several times), I
>was simply trying to add my own enthusiasm to this topic. Perhaps you
>didn't get that though, at least I gather from your response.
sorry about the tone of my response. days of grappling with the total
fsck-up that is autoc
>What is a difference between an ASCII identifier "plughw:0,0"
>and "hw:0,0"? And what the output of 'aplay -L' means?
a "hw" device is totally constrained by the hardware characteristics
of the underlying audio interface. if the device has 26 channels, then
it must be used with 26 channels; if i
>Could you explain to me how to do the following: the user has a TV capture
>card with audio-out that is plugged in into one of the sound cards
>installed in the system. I need to ask user which card and which input
>s/he plugged the TV audio into. How do I do that ? (Preferably without the
>user
>> >Could you explain to me how to do the following: the user has a TV capture
>> >card with audio-out that is plugged in into one of the sound cards
>> >installed in the system. I need to ask user which card and which input
>> >s/he plugged the TV audio into. How do I do that ? (Preferably withou
i'm pleased to announce that, with the continuing collaboration of
RME, work on the driver for the RME Hammerfall DSP will be beginning
soon after the start of 2002.
--p
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>I think that it's nonsense to force average users to study the
>alsa driver architecture and learn a format of the configuration
>file (they just don't know about the existence of such file) to
"average" users have 1 sound card. "default" will work for them.
>perform such trivial task as to poi
>I would be delighted if someone could tell me what the name semantics are.
there are none. just use the string provided the user, or "default" if
they don't provide any (or "hw:0,0" if you prefer and have good
reasons for doing so).
>In BruteFIR I only want to be as close to the hardware as pos
>Have you ever used/seen some good audio application in MacOS
>or M$ Win world? Each application from this category has
>a config dialog for these things. ...or do you think that
>musicians have to be programmers or administrators?
Windows doesn't have ALSA. they have a two-bit device driver syst
>As I don't own a Hammerfall card I'd like to know, how windows presents
>your hammerfall hardware? How do you use the features of the Hammerfall
>with Windows. It's certainly possible somehow to do multitrack recording
>with windows, isn't it? They probably live of windows user, not of linux
>sou
i would much prefer a clear macro than the code that is found in the
existing low level drivers. whether snd_mwait() is the right answer is
one question, and whether ALSA should provide it when the kernel
itself does not is another.
--p
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ok, suppose i merge two PCM devices into one "multi" device. then
suppose i do the same for the corresponding control device.
how does an application access the control switches for the multi
device? in the underlying "hw" devices, there are duplicate sets of
control switch names. how can i refer
>1) Is it necessary to open the device twice, as opposed to opening
>it once, passing SND_PCM_STREAM_PLAYBACK|SND_PCM_STREAM_CAPTURE instead?
yes it is. ALSA PCM streams are uni-directional, always. we've played
with it the other way, and there were some problems.
>2) Doesn't the link function d
>I'm interested in whether there has been any large scale discussion
>about a unified approach to sound support. Right now supporting
>oss/alsa/arts/esd see ms a bit much and I'm curious as to whether
>there has been any talk of establishi ng something like sound
>server(arts)->sound driver(alsa)
>Is there any real time application or applications, that have been made
>directly to ALSA?
please be a bit more descriptive when you use the term "real time". it
has many different meanings, and so its impossible to answer your
question clearly without more definitions of what you mean.
--p
__
>Paul, could you spare some more keystrokes on what you
>think are the best steps to take to solve this problem ?
actually, i don't see a way forward.
neither jaroslav nor abramo have indicated that they accept the
desirability of imposing a synchronously executed API (SE-API; this
being the he
>I do know that ALSA is going to be replacing OSS in the kernel although from a
>system standpoint it makes more sense to be a little general with thing.
>
>It seems like high end applications require low latency. Would it be possible
>to create a sound server(arts) that fulfilled latency and oth
>On Fri, 11 Jan 2002, Paul Davis wrote:
>
>> ok, suppose i merge two PCM devices into one "multi" device. then
>> suppose i do the same for the corresponding control device.
>>
>> how does an application access the control switches for the multi
>>
>Most sound apps have to turn the PCM into the frequency domain before
>applying a sound effect anyway, why not just stay in the frequency domain.
this is not true. most audio FX are carried out in the time domain. i
don't know many sound apps that do what you describe unless they are
doing somet
>I'm ommiting discussion about questionable efficiency of a callback based
>API in unix environment here.
abramo and i have already disagreed, then measured, then agreed that:
although an IPC-based callback system is not ideal, on today's
processors, it is fast enough to be useful. the gains this
>I don't think this is relevant wrt Jaroslav objection. He was not
>proposing a *all-in-a-process* solution.
i don't see what other issue surrounds "questionable efficiency of a
callback based API in unix environment". can you enlighten me? i also
note that CoreAudio runs in a Unix environment ..
> I meant mainly streaming applications for full duplex usage.
ecasound and ardour both do this, and JACK provides an API for
applications that do this. i think GLAME might do this to, though i am
not certain of its full duplex-ness.
i am the author of ardour and the primary author of JACK. if
>1/ the ability to do the cleanup (nothing more is needed)
>2/ reliability and fidelity to the original
>3/ ease of use (i don't mind command line, but won't be able to figure out
>too tech savvy parameters)
check dave phillips sound+MIDI pages for linux. there are at least 2
or 3 denoising utili
>[snip]
>> what you're missing is that high end applications need *two* things:
>>
>> 1) low latency
>> 2) synchronous execution
>>
>> the latter is extremely important so that different elements in a
>> system do not drift in and out of sync with each other at any time.
>>
>If it is pos
>I have looked at the Jack web page (http://jackit.sourceforge.net/)
>
>It would help more if jack.h had more documentation for all api function,
>and not just a few of them.
well, we're not quite finished with the API yet. Once its really set
in stone (for v1.0), something that i imagine will ha
>Here there is no requirement for low latency or synchronous execution.
>The requirement is just that the app is told exactly how long it will be
>between the next samples written to the driver, and
>the sound actually coming out of the speakers.
there's another very important i forgot here. beca
>I'm not at all sure why the callback mechanism is such an issue. Windows uses
>callbacks for their standard sound layer as well as with DirectSound. I'm not
>sure why the callback model is so difficult to incorporate into an application
imagine a standard tracker-style program. it has a UI of
>I'm not sure if I understand why this would help to position jack as the
>standard linux sound server. It seems like we need to get some kind of
>discussion going between arts and jack developers as arts is in the position
>that jack would like to share. No doubt that jack has something to offe
>API which pretends to be "one and only" for linux must work not only with
JACK is not an API pretending to be the one and only anything. Its an
API designed to fill a niche that nothing else is doing for Linux. It
has been written with 98% of the attention paid to high end apps. If
it happens to
>Paul,
>As you obviously know more about Jack than I do, can you explain how an API
>like Jack could provide information to the APP so that Audio and Video can
>be kept in sync.
JACK doesn't do that. JACK is run by a single timing source (called a
"driver"). that timing source can come from anyth
i wrote to lexicon when they released the core2. they demonstrated
extreme antipathy to linux in general and to open source drivers in
particular. they seem to believe, as do many companies, that their
basically off-the-shelf hardware contains some important intellectual
property that an open sour
>Shucks. I was afraid of that. I'll bug them some more, and if I get really
>ambitious (I know nothing about it), perhaps try and reverse engineer
>something. Is this technically possible?
yes, but it might take you months and months without a logic analyzer,
and months with one :)
>On Tue, Jan 15, 2002 at 08:06:43 -0500, Wm. Josiah Erikson wrote:
>> Shucks. I was afraid of that. I'll bug them some more, and if I get really
>> ambitious (I know nothing about it), perhaps try and reverse engineer
>> something. Is this technically possible?
>> On another note, are there multit
jaroslav -
for some time now, i've had a version of the wavefront driver that
doesn't require any module parameters - it just uses isapnp to figure
out the configuration of the card. i would have submitted the changes
to you already, except for one problem. after a reboot (cold or warm),
the firs
>At least the driver handling the standard USB Audio Device Class is
>located in the USB kernel directory, ie. linux/drivers/usb. All code is in
>the big (~4000loc) audio.c file. It implements the OSS ioctls, plus
>OSS-style mmap() (and of course read()/write()).
>
>But it does register itself to
>>> But it does register itself to the OSS subsystem (to
>>> drivers/sound/sound_core.c) like all other sound drivers, so it _is_
>>> part of OSS.
>> no. sound_core is NOT part of OSS. ALSA attaches to it as well. Alan
>> Cox wrote that so that OSS and ALSA could (theoretically) co-exist.
>
>Now
>NO one has specifically sent a copy of the file for the Delta 66 that I
>can seem to understand and there seem to be a few diferent versions
>floating about on the mailing lists. (ardour-dev,alsa-dev, and
>alsa-user).
>
>If anyone could help here by sending an asoundrc file configured for a
>si
>Hello problem like that:
>I got ardour workin with my motherboard via686a driver card but it
>hanged bad in some fase of using ardour.(no use anyway for that noisy
>card)
>
>I got Xwave 6000 Pro too but here comes up some errors.
>I use 0.9.0beta10 drivers and libs 2.4.8 kernel mdk 8.1.
>
>what'
So, how do we ever get anyone who makes money to recognize that
everything in this description is true of Linux/ALSA as well?
(Rhetorical question, I think).
--p
--
Propellerhead Software, the Swedish company behind acclaimed pr
>I am offering my machine and some of my time for finding
>and fixing issues with the via686a. AMD 1Ghz, Kernel 2.4.14+
do you know about the athlon paging h/w bug?
--p
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why would snd_pcm_mmap_commit() take an amount of time to execute
that is linear with the number of frames committed (on h/w that
supports mmap mode without any games) ?
with 64 frames, and a hammerfall, it takes 36 usecs; with 4096 it
takes 2.3msecs. the code doesn't suggest anything to me ...
>Paul Davis wrote:
>>
>> why would snd_pcm_mmap_commit() take an amount of time to execute
>> that is linear with the number of frames committed (on h/w that
>> supports mmap mode without any games) ?
>>
>> with 64 frames, and a hammerfall, it takes 36 use
>Why not use (and possibly improve) an existing one:
seconded.
> - http://www.eca.cx/ecasound
> - http://www.wakkanet.fi/kv/ecasound/Documentation/examples.html
>
>...? ;)
... not to mention that ardour can be run in "passthru" mode, allowing
it to function as a multichannel, GUI-fronted realt
>Hello all,
>
> it seems that I've finished an initial version of documentation
>for alsa-lib's configuration files and PCM interface. Here is URL:
looks like a decent start. i hope i can find time to improve some
parts of this. in reading the section on the syntax of config files,
however,
>> Ah, but it's not "another small language", it's the same language that
>> X11 and java resource files use, with two extensions that are well
>> chosen:
>> 1) a compound node syntax so that you don't need to type the
>> entire a.b.c.d.etc... path name for each resource.
>> 2) an arra
>> in addition, i've already reported a bug in the config parsing whereby
>> if something could be a string or an integer, and the string contains
>> an integer value, it assumes the item was an integer.
>
>It depends on the code which uses a configuration node what information it
>expects. It's n
>Are you serious about xml? Do you have smoked something weird?
xml was just an example. i happen to like it, because there are
standard parsers, and standard presentation tools. but i wasn't
seriously suggesting it as an alternative. i would just feel more
comfortable with a language that wasn't
>We check the type of value in the single configuration parsers (which is
>different from the ascii load/save code). I don't see the problem you
>described.
i have a card named "9652". the configuration code fails to work on
such a card unless i refer to it by its ALSA ID. it tries to find a
card
Jeremy has correctly answered many of the points/questions you
raise. I will try to finish off the story :)
>problem is that I cannot get the mixer to have any "elements" (or
>"elems" as the alsamixer calls them) which thus makes the card
>un-openable even with the alsamixer.
there is no mixer.
>just writing a recording app with alsa lib i found i need
>to use
>
>snd_pcm_start(chandle);
>
>Why? What does it do?
the hardware of audio interfaces doesn't activate all by itself - it
needs to be told to start processing i/o and generating interrupts.
when you open a handle on either the pl
>1) I understand now that there is no mixer. Yet, I am a bit baffled how then d
>oes ardour (for instance) control this app's output? Couldn't there be a some
>kind of generic mixer app derrived from the way Ardour addresses volume issue
>in this card? Could it be that the app only attenuates th
>> 1) I understand now that there is no mixer. Yet, I am a bit baffled
>> how then d oes ardour (for instance) control this app's output?
>> Couldn't there be a some kind of generic mixer app derrived from
>> the way Ardour addresses volume issue in this card? Could it be
>> that the app only att
>You want the ctl entry so that ardour will work properly. When you use
>aplay or any other application, use dsp1 as your alsa device, and when you
>use ardour, use rme9652, because ardour wants the raw access.
3 quick notes:
1) i prefer:
pcm.dsp1 {
type plughw
card 1
}
as
>Also, could someone please be so kind to send me their asoundrc file
>that they are using for the hammerfall, so that I can learn from it as
>to how to configure it since the one that comes with alsa-driver package
>is rather esotheric for my understanding? Thank you very much!
this is mine.
pc
>I noticed your post regarding the IRQ issue. How did this issue manifest
>itself when you tried to use the Hammerfall card? Could you still
>install (modprobe) the driver without any problems, or was it preventing
>you from doing even that? Could you please give me more info so that I
>can see if
>Hello
>I am thinking of writing a low level sound card driver for alsa.
>The card in question is the dxr3 hardware DVD player, which has analogue and
>SPDIF out.
>It currently works fine with OSS drivers.
>The card has no mixers, or recording function, just write and GETOPTR,
>SETSPEED etc functi
>It seems I have been celebrating too early.
>
>Here's the scoop on my current issues:
>
>I have finally managed to produce sound with the darn thing, but every
>so often now I get sporadic "crackle" out of the output, as well as
>complete distortion of the sound, almost as if the pitch is constan
>> i have had this problem from time to time as well. i am still, after 2
>> years of working with the hammerfall, unsure of what causes it.
>
>Could it be a PCI problem? Latency timer issue? Busmastering problem?
it seems to be related to the condition of the h/w when it was stopped
and/or resta
>Well, I tried restarting the alsasound and that did not fix it at all.
>Granted, I was also trying to use cmixplay command which most likely
>invokes rme card via oss emulation, and I've heard that oss emulation is
>a bit flaky. So, I did end up having to restart the machine to get the
>sound bac
>> the hammerfall cannot do this. the hammerfall-DSP can do matrix
>> routing of signals between channels, but it does not mix software
>> streams down to a channel.
>
>Crap! :-) So does this mean that RME card cannot be accessed from
>multiple applications at the same time? This is one of the mos
>It probably wouldn't be too hard to write a driver that presented a
>/dev/dsp* interface to the world and was attached to jack at the other
>end. You could then run two and attach rtcmix or whatever to them.
>
>On the other hand it might encourage poeple to put off supporting jack
>nativly and th
>I have seen that alsa provides nice pcm devices in /dev/snd directory
>
>*They are split in playback and capture devices :) nice idea.
>In Oss I was able to pipe data from and to them and therfore was
>recording/playing back at a certain sample rate.
>With alsa this is not possible anymore. it sa
>Anyway, I believe going with the sequencer is the better solution anyway;
IMHO, its not. i had to do this to download a huge firmware update via
sysex to our Alesis M20 ADAT's last year, and ran into all kinds of
problems. Unless your Sysex messages are all rather short, the
sequencer is not you
should it work for me to call snd_pcm_drop() for both playback and
capture streams if i'm in the middle of a snd_pcm_mmap_{begin,commit}
pair?
unless i use RT timing, on both the trident and hammerfall, doing do
invariably causes an effect upon restarting the streams that suggests
an out-of-sync
>> should it work for me to call snd_pcm_drop() for both playback and
>> capture streams if i'm in the middle of a snd_pcm_mmap_{begin,commit}
>> pair?
>
>Nope. Do you think that we need to lock these related code blocks in
>alsa-lib for threaded applications or leave this locking for application
>> poll
>> snd_pcm_mmap_begin();
>> snd_pcm_mmap_commit();
>> poll
>> snd_pcm_mmap_begin();
>> ... realize things have gone wrong ...
>> snd_pcm_drop();
>> snd_pcm_prepare();
>> snd_pcm_start ();
>> poll
>> snd_pcm_mmap_begin();
>> snd_pcm_mmap_commi
> the big moment in ALSA existence is here. Linus integrated ALSA
>into 2.5 kernel tree!! The first pre-patch with ALSA will be available
>soon. For impatient hackers: Check http://openlogging.org, click on
>'Sorted by number of lines' and choose 'Linux kernel tree' with domain
>'transmeta.c
>I've now added a few debugging points to jackd's alsa_driver.c:
> - five checks for xruns in various parts
> of alsa_driver_wait() (prints an error if xrun
> is detected; check with snd_pcm_state())
> - prints an error if playback_avail or capture_avail
> is no
>Normal 'snd_pcm_drop() -> snd_pcm_prepare() -> snd_pcm_start()' sequence
>doesn't solve the problem. Audio is still garbled. I didn't try full
>'close() -> ... -> open() -> ..' cycle yet, as that would involve quite a
>lot of work (jackd's alsa_driver.c is a long piece of code).
its my feelin
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