>ok paul i had another look at the driver playback/capture tutorials. I have on
>e question.Do applications initialize and use a pcm_handle of the alsa driver
>when they begin interaction with it?Surely not
surely yes.
ALSA isn't structured the way windows seems to concieve of audio. We
don
> --- Paul Davis <[EMAIL PROTECTED]> wrote: >
>>Hi ,
>> >i'm trying to change the fragment size that the
>> alsa
>> >driver is capturing/playing out. It's currently set
>> at
>> >32 msecs(256 bytes).
>> >They are set in
>Hi ,
>i'm trying to change the fragment size that the alsa
>driver is capturing/playing out. It's currently set at
>32 msecs(256 bytes).
>They are set in the runtime structure...
> runtime->period_max ...etc
>Could anybody tell me which file these parameters are
>set in the alsa-driver directory
>When using OSS you can just do mixer ioctl's on the opened PCM fd.
When using OSS, you can't model a lot of the functionality present in
contemporary audio interface hardware mixers.
So there's a choice: a limited, simple API that fails to support card
features, or a complex API that over time g
>On Thu, May 06, 2004 at 09:49:31PM -0400, Paul Davis wrote:
>> at the risk of endlessly repeating myself,
>
>If you're being asked this frequently, I'd recommend adding some notes to
I'm not. I'm just a big mouth who always pipes up when SIGIO is mentioned
>Kernel people: is poll() less effective than using SND_PCM_ASYNC and a
>signal handler for low-latency sound? I'm guessing it is, but there
at the risk of endlessly repeating myself, SIGIO is basically
useless. your handler executes in signal-handling context, and can do
very, very little. not e
>in says under "known bugs":
>
> - 96kHz and 88.2kHz not accessible via PCM interface
>
>What does that mean? The card and driver does work in 96 and 88.2 kHz, I
>know that since I wrote the driver... but if there is some sort of bug,
>I'd like to know.
anders, i don't know, but it might mean t
>Do you plan any driver for M-Audio Firewire Audiophile? I have contacted =
>M-Audio=20
>French Support and post this request, they told me that M-Audio driver te=
>am=20
>(USA) had request you for this card to have a driver under alsa. Can you =
>confirm=20
>that?
Please note: since NAMM this yea
>1) Open my Player, tune to my favorite station. Use my new device for
>audio output.
>2) do something like: dd if=/dev/audiocapturepcmout count=xxx | lame
>--output myfile.mp3
streamripper is a much better option for this. its specially designed
for this task. you are also talking about mp3->P
>Does this mean that now one can channel ALSA-only aware apps directly to JACK
Yes.
>and if so, are there any penalties of doing it this way as oposed to using JAC
>K-aware apps (i.e. Sample-sync?)?
Yes, you lose sample-synchronous guarantees. Also, there *might* be glitches
in the audio output
>hope this is the right list to post: I installed to RME Pci Cards with
>a Multiface each. Running hdsploader finds and initializes both and
>hdspmixer can also get used with both cards using the card select
>buttons for card1 and card2. However after startup of jack, I only see
>18 alsa_pcm in and
>Well, if there is a way of monitoring these hex registers for various
>hardware in Linux I could try to compare their state upon initialization in
>Linux to that one in Windows (since in both situations they share the same
>IRQ, at least on my notebook) and forward this info to you guys.
>
>So my
>Given that the alsa driv interrupts at period
>boundaries(which are defined in bytes), the tests i
>have carried out prove otherwise..i.e that the driver
>operates in millisecond power-of-two based periods.
on most PCI-based hardware, the audio interface interrupts every N
audio frames, where N i
>I can't find any way to detect the running ALSA version, for diagnostic
cat /proc/asound/version
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>and it HAS do be a software issue, since at least on ico's and my
>machine, the hdsp works flawless with winxp ...
agreed. but its much more likely to be something about the cardbus
support under linux than the hdsp driver, and neither side (the
cardbus people or us hdsp people) knows the other
>mainly the alsa guys who wrote the driver and know, how they access the
>bus system and the hardware, have to tell the pcmcia guys, who know,
>what's going on in the bus ... i CC'd this mail to the thomas
>charbonnel, paul davis and winfried ritsch, who wrote or maintain thi
>The problem is that I cannot get the mixer to have any "elements" (or=20
>"elems" as the alsamixer calls them) which thus makes the card un-openable=
>=20
>even with the alsamixer.
thats correct. the hdsp has no mixer that can be represented using
conventional mixer elements. there is nothing cor
>> select is generally deprecated in linux (linus says so!). but you can
>> use the same pfds in select as in poll (select is implemented in the
>> kernel using the poll code). the problem is interpreting the results
>> you get back (as noted recently for the dmix plugin).
>
>Nope, the application
>> >open pcm, and get a handle.
>> >
>> >snd_pcm_poll_descriptors(handle, &pfd, err);
>> >
>> >Get a poll file scriptor in pfd.
>> >
>> >select(nfds, rfds, wfds, efds, tvp);
>> >
>> >Is it possible to use this call with alsa ?
>>
>> select is generally deprecated in linux (linus says so!). but you
>open pcm, and get a handle.
>
>snd_pcm_poll_descriptors(handle, &pfd, err);
>
>Get a poll file scriptor in pfd.
>
>select(nfds, rfds, wfds, efds, tvp);
>
>Is it possible to use this call with alsa ?
select is generally deprecated in linux (linus says so!). but you can
use the same pfds in select
>$ measure -p ./out/x11.png -o ./out/x11.out -c 0 -f 1024 -n 2 -t 2
>I get this error message: "error setting freq 1024"
probably a permissions problem. try it as root.
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>> Where can I get the complete list of controls which an audio codec has to
>> support.
there is no such list.
and lets get the terminology straightened out here before this goes
any further.
"codec" comes (in this context) from "coder/decoder". that maps
roughly to what is more properly referr
>What I basically did is investigated the current driver in Windoze for the
>carbus and it came up as a "generic cardbus interface." I was by now aware
>that the exact cardbus chipset is/was ENE C1410. So I did some searching on
>the Google for the driver and came up with HP's driver download page
>In any case, most of this can all be handled by software. The only stuff
>the hardware really need is the volume, frequency, direction to sound
>source, and any additional effects (such as reverb). Application
>software can deal with the rest.
i can certainly see the utility of such an API. howev
>At 05 Mar 2004 11:23:35 +0100,
>Simon Schampijer wrote:
>>
>> Hello,
>> I've got a Maudio Firewire 410.
>> Are you going to make also drivers
>> for this one ?
>
>so far, no ALSA developer works on it.
>
>> Or are there Problems with
>> Maudio or firewire???
>
>unfortunately, no hardware, no dat
>And by that time, games will be even more demanding than before, and
>still want to use those cycles for itself rather than software sound
>processing, so just sitting idle waiting for the ultimate
>infinite-megahertz cpu, that can do everything in no time, gains nobody
>anything... at least our b
>Are you planning to standardize such features or design a
>device-independent API for it? (And my manager wants to know which ALSA
>release this might get implemented in...)
Just a thought on your overall issues with ALSA over the last week.
The kinds of features you seem to expect from hardware
>Taken for granted that configuration space on entry of function is not
>empty
can this be taken for granted?
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>On Wed, 2004-02-25 at 13:53, Takashi Iwai wrote:
>
>> - PCM mixing and software MIDI rendering in the kernel space are
>> evil. if you doubt it, ask on LKML :)
>
>So is low-latency, LKML people made that clear also about year or two
>back...
on the contrary. they've made it very clear that its
>It would still be nice to have a system clock based interval process for
>applications that require activity at specific intervals, but are not to
>do with sound or video.
have any actual use cases?
>An audio hardware interrupt can interrupt any process at any time.
>Is there already a procedu
>It would still be nice to have a system clock based interval process for
>applications that require activity at specific intervals, but are not to
>do with sound or video.
i'd be interested to see a realistic usage case. its mostly related to
RT devices of some kind, most of which provide their
>The ideal scheduler for realtime apps would be one that has an api that
>allows for a call like "schedule me at exactly 10ms intervals+-1ms".
no, thats not true.
the system clock does not run in sync with the sample clock. the drift
in this would become noticeable in a few minutes.
the only ti
>On Wed, Feb 25, 2004 at 09:17:54AM -0500, Paul Davis wrote:
>> There are only 2 differences associated with running the code you are
>> talking about in the kernel:
>>
>> a) it runs deterministically in interrupt context
>> b) it avoids a context swit
>I don't think so. If an sound app is swapped out or another app is doing
>intensive disk IO we could observe - hear - the difference.
and how can OSS help with that? ok, so we know that non-SCHED_FIFO
apps (and occasionally, even them) can be delayed by kernel-side
issues. but not keeping up with
>Interface Descriptor:
> bLength 9
> bDescriptorType 4
> bInterfaceNumber0
> bAlternateSetting 0
> bNumEndpoints 6
> bInterfaceClass 255 Vendor Specific Class
^^^
>I have a few usability comments with regards to the drivers that I would
>like to share, will draft a fuller email later, but basically I am using it
>as a hifi input for music, nothing else, but I'm frequently getting very
>jittery & broken up audio (guessing that the default alsa buffer/latency
>Anyone using the latest RME HDSP 9632 driver in ALSA, while also using
>the expansion cards? I mean, the newer AI4S-192 and/or AO4S-192 which
>are required to get the 192kHz sampling on all analogs.
>
>Do the latest expansion cards (192kHz) work with the latest ALSA driver?
i doubt it. RME have n
>it's not working on my system (i tried 64, 128, 256) ... but playing
then i guess there is more than one problem. i would also recommend
checking into whatever is sharing the interrupt that the hdsp is
on. on my laptop, way too many things are on IRQ10 (USB, CardBus,
Wifi, HDSP). this seems hard
>Yes, we're aware that our APIs are rather too much complicated for simple
>applications. We are designing the very simple (current name ordinary)
>APIs to let more developers join us. But as Kai noticed, these APIs will
>be a bit limited, so they won't be intended for more complex applications.
> I am currently looking into rewriting our current OSS sound routines to nat
>ive ALSA, as it seems OSS will invariably be phased out now that the ALSA driv
>er is distrubuted with the Linux kernel, plus ALSA seems to have a great numbe
>r of benefits for us.
>
> Our current sound routines per
>> >I have application A that needs to open payback and capture streams
>> >because it's a two-way communications program. Application B is a game
>> >that uses playback only. I want to use these both at the same time.
>>
>> but i don't think you can do this with OSS. why should ALSA's OSS
>>
>I have application A that needs to open payback and capture streams
>because it's a two-way communications program. Application B is a game
>that uses playback only. I want to use these both at the same time.
but i don't think you can do this with OSS. why should ALSA's OSS
emulation make it
>Using latest cvs I am unable to run the hdsp as a normal user. It works
>as root user however.
>
>Any ideas for fixing this?
more specifics about what doesn't work would be helpful.
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>However, trying to use either xmms, mplayer or mythtv with the same alsa
>device just seems to leave the app "hanging" and not doing anything, and
>attempts to fastforward in mplayer and mythtv (which presumably causes the
>alsa layer in each app to try and pause then reopen the sound layer) cause
>I'm working on changing DMIX to allow clients to open the capture stream.
i don't get it. dmix is for playback, not capture. what would be the
semantics of this?
--p
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>That kind of argument would just corroborate the TransGaming report that
>ALSA is no good overhead-wise, which is probably not what the ALSA
>developers wish to do here.
i wasn't talking about ALSA. i was talking about audio hardware and
programming models in general. ALSA will allow pretty much
>although i agree that from a pragmatic perspective, it would be good
>to understand how to use ALSA to do this.
>
>however, this model of audio programming is not portable, and has
>gotten people into trouble before, not just under Linux.
>
>making the assumption that direct access at all times to
>> We have been unable to find a way to tell ALSA, through
>> snd_pcm_mmap_begin(), where we want to write data when using the mmap
>> api. OSS doesn't have this restriction and this is the only documented
>> way we can see to support sound engines which mix in new sounds using
>> looping buffe
>meanwhile, i have found that the plughw layer doesn't work with JACK
>anymore. it appears that any attempt to set the period size fails. any
>ideas about that?
to clarify: any attempt to set the period size if the SR is not 48kHz
will fail.
--p
-
>yOn Sat, 27 Dec 2003, Paul Davis wrote:
>
>> >i've been getting my laptop with an ICH soundchip up to speed, and i
>> >just noticed for the first time yesterday that ALSA fails to get the
>> >speed to the rate that JACK requests. a quick google reveals that
>i've been getting my laptop with an ICH soundchip up to speed, and i
>just noticed for the first time yesterday that ALSA fails to get the
>speed to the rate that JACK requests. a quick google reveals that the
>OSS drivers only support 1 h/w rate, but there are suggestions that
>the ALSA driver wo
i've been getting my laptop with an ICH soundchip up to speed, and i
just noticed for the first time yesterday that ALSA fails to get the
speed to the rate that JACK requests. a quick google reveals that the
OSS drivers only support 1 h/w rate, but there are suggestions that
the ALSA driver works b
>Justin,
> I'm running Alsa-1.0.0rc2. How much more up to date could I be?
>
> Also, this is an HDSP 9652 which has the firmware on the board. Why
>is a firmware loader required at all?
its not. jaroslav used that phrase because the bug manifested itself
most clearly when loading firmware. it
i'm a bit late to the party, but ...
>> > --- rme9652.c.ORIGINAL Thu Dec 18 23:43:36 2003
>> > +++ rme9652.c Thu Dec 18 23:45:12 2003
>> > @@ -1618,7 +1618,6 @@
>> > RME9652_SPDIF_RATE("IEC958 Sample Rate", 0),
>> > RME9652_ADAT_SYNC("ADAT1 Sync Check", 0, 0),
>> > RME9652_ADAT_SYNC("A
>Since there is almost nothing else to do to support the Aureal Vortex
>3D processor on Linux, just as i announced some time ago i started
>designing a OpenAL interface for ALSA. The design is meant to be
>applicable to other hardware too.
<
>I made a preliminary description, from what i have done
>My opinion is that a simple function could be included in alsactl which
>scans for available devices, makes a list of their abilities. Everyone
>uses "post-insert alsactl restore" in the modules.conf file so it would
>be essentially a non issue from a user perspective.
i think it needs to be s
>I wanted to check my knowledge of something: does an xrun necessarily
>correspond to a "drop-out" in the audio stream? In other words, could
>you have a drop-out WITHOUT an xrun, or an xrun WITHOUT a drop-out? Is
>there a strict one-to-one correspondence between the two?
a dropout occurs when t
>Tim Goetze wrote:
>> * what kind(s) of sync-to-external will you need?
>> * do you want to receive/send MIDI sysex data?
>>
>> last time i checked, the latter was impossible to do via the sequencer
>> API (please correct if things have changed).
>
>Sending/receiving sysex has always been possible
>I hope ALSA supports hardware features in certain sound cards (like hardware
>mixing, sound buffer memory, and so on) like DirectSound does in Windows...
all that and a lot more.
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>I don't agree. The control API (usually) is for things that don't affect
>the way data is transferred between the card the the computer.
it is *now*. i was just imagining a different conception of what it
could be used for.
> Sample
>We all think in the same way, but there's no simple solution for this
>problem. I prefer to have such configuration information in an user-space
>database accessed via an alsa-lib API. It's nothing for the kernel space.
i'm not sure i agree with that. a user-space config DB could be used
to desc
>>> I'd like to be able to query the capabilities (number of channels,=20
>>> buffer
>>> size etc.) of ALSA devices in the system, even if they should be in us=
>e=20
>>> by
>>> some other process. The only current way to probe device capabilities =
>is
>>> to open a pcm, and use snd_pcm_hw_params,
>
>> Actually, I'm not completely opposed to the idea. But I am totally
>> clueless when it comes to writing device drivers, and would have no
>> idea where to start.
>
>Start here: http://www.alsa-project.org/documentation.php3#Driver
>
>You don't have to start writing the driver from scratch. You
>> be overjoyed that was if and when it finally does. Don't expect anythin=
>g
>> soon, unless you're willing to be that developer.
>Actually, I'm not completely opposed to the idea. But I am totally=20
>clueless when it comes to writing device drivers, and would have no idea=20
>where to start. Do
>On Thu, 4 Dec 2003 06:41:56 -0800, Mark Knecht <[EMAIL PROTECTED]>=20
>wrote:
>
>>>
>>> Whenever I try to activate double speed (88.2/96kHz) mode with my RME
>>> Hammerfall Lite (DIGI 9636), snd_pcm_hw_params fails with a 'Device or
>>> resource busy' message. This is with the number of channels s
>has anyone contact with yamaha, btw?
i've spoken in person with guys inside the mLAN development group (at
NAMM, last februrary), and they put me in touch with a professor in
south africa who has been acting as a consultant for them on a number
of aspects of mLAN. i talked with him a few times to
>> I can't help you much, however, unless JACK has
>> recently been worked on I think there's a 32 track
>> limitation. It's known to be technically trivial.
still in place at this time. there is a simple 1-2 line solution; i'm
not sure if its the right one, since it simply substitutes a 64
channe
>On Tue, 25 Nov 2003, Paul Davis wrote:
>
>> at what point, if any, did alsa-lib start returning positive E
>> values (e.g. EBUSY) rather than -EBUSY?
>
>I'm not aware. Which functions?
none. its OK, i was checking errno rather than the returned value. my
mistake. so
at what point, if any, did alsa-lib start returning positive E
values (e.g. EBUSY) rather than -EBUSY?
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>
>If my hammerfall (rme9652) sound card is locked slave to 48 kHz, and I put
>44.1 kHz into snd_pcm_hw_params_set_rate_near for playback it accepts it,
>should it really be that way? I expected that it would set the rate to 48kHz,
>since the card does not change the sample rate to 44.1 kHz (loo
>> 5) I think that this sound mixing problem might be better served by
>> sound servers like jack.
>
>Hmmm,
>
>actualy I did have a very close look at jack, but the problem that I saw=20
>(correct me if Im wrong here), is the fact that jack only works well if the=
>=20
>applications are specificaly
>> the RME hardware has 16-26+ channels, but only one stream.
>
>Ok,
>
>here we come to my ignorance on sound drivers/hardware/terminology. If I ta=
>lk=20
>about a stream I mean one application --> soundcard link. What would a=20
>channel be ? And what a stream if not what I defined a line up?=20
>On Mon, 17 Nov 2003, Paul Davis wrote:
>
>> >what you say seems to be valid to me, but not if you implement smart dmix t
>he
>> >way I said =). The way I suggested *every* application would connect to sma
>rt
>> >dmix, and none directly to alsa lib
>> well, my concern is that with the high-end cards, people tend to stick
>> with the quality of sounds. that means, any reason to reduce the
>> quality wouldn't be acceptable for some people. since dmix will do it
>> silently (if needed), it might be unacceptable.
>
>what ? are you suggesting th
>what you say seems to be valid to me, but not if you implement smart dmix the
>way I said =). The way I suggested *every* application would connect to smart
>dmix, and none directly to alsa lib (except those that use devices like hw: -
>and those should never be mixed). Since every stream is conne
>So, these are two numbers - and basicly all is fine as long as you dont wan=
>t=20
>to excede them, but if you do, you need to use dmix or dsnoop. Why not use=
>=20
>dmix and dsnoop automaticly when necessary ? Wouldnt it be possible to have=
because it would be catastrophic, or, well, at least v
1) just a quick note to point out that whether you know it or not, the
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>Hi,
>it seems to me that with alsa 0.9.8 drivers it's not possible to load
>snd-rme9652. With 0.9.7c there is no problem. The snd-hdsp driver works
>in both versions (i tried both cards with both alsa versions).
>
>error message:
>modprobe snd-rme9652
>/lib/modules/2.4.22/kernel/sound/pci/rme9652/
>Many thanks for the info.!
just keep in mind what jaroslav noted about signal handling
context. please don't build a serious application around this
mechanism. sooner or later, you may want to make a system call from
the handler that violates the POSIX guarantees about what can be
done. at that p
>I am trying to write an application that uses ALSA interrupt-driven
>(which I guess is obtained by adding PCM handler(s) in ALSA?),
>full-duplex (simultaneous capture/playback) using the ALSA PCM API
>directly (i.e. not using JACK).
>
>My question: do I only need to register one callback/handler f
>> anyway, it'd be helpful for debugging if someone can reproduce this
>> certainly under a fixed condition...
>
>Well, about 3 times out of 7 I get the problem, so it's not hard to
>reproduce at all. What I'd have to know is where to place the debugging
>printk()'s or something like that :(
just
>again I've got a question regarding the ALSA midi sequencer. I tried to
>generate Midi time code.
you cannot send 100% valid MIDI time code on a Linux system without
the high resolution timers patch or some other interrupt source that provides
a suitable timing resolution.
MIDI time code is no
>At Mon, 03 Nov 2003 14:41:51 -0500,
>Jonathan Kraut wrote:
>>
>>
>> Hello again. Further research:
>>
>> I tried an earlier version of jack (0.71.0) with alsa-driver-0.9.8, accessin
>g hw:0,2 via my .asoundrc. I get the same result as before, but in addition th
>is particular version of jackd
>This problem happens randomly but is very annoying, specially if I
>program my PVR (MythTV) to automatically record a show - the show can
>get recorded with metallic sound, which makes it very hard to understand
>during playback.
>
>This is not MythTV-specific since I have been able to reproduce t
>I use kernel 2.6 with Alsa (2.6 kernel internal ALSA). And some program is
>freezee or going to zombie.
>Example KDE artsd use alsa and some program want to acces it (example
>MozillaFirebird flash plugin) the accessing program going to zombie. (I try
>to desable artsD because if I use mplayer
>I am working on a voice over IP application. We would like delays from
>the write to soundcard to the actual playout lower than 50ms. Do you
>think it is impossible with ALSA ? With OSS free the delay we got were
>quite good, without any tweaking. But ALSA is said to have better
>support for full-
>My second idea was to have a rather big hw buffer (500ms), and then set
>the start_threshold to a low value (32 frames for instance). But whatever
>my parameters were, I always got a playout delay of about the hw buffer
>size.
the output latency is always roughly the size of the hardware
buffer.
>I followed the discussion: "optical SPDIF output on Abit NF7 nforce2
>main board".
>
>I am using a similar Epox Mainboard with SPDIF in/out. Currently i am
>using the todays cvs alsa version and have two questions:
>
>Using gqmgeg which uses ogg123 to play ogg files, ogg123 seems to have
>the samp
>I've not managed to fix this bug into my program... Do you see the error on my
>
>code (bad soundcard initialization or other) or have you got another way to
>propose me to capture the two channels of my soundcard separately ?
try increasing BUF_SIZE to 4096 and see what happens. 128
frames/per
>It sounds to me that the problem Nick Arnold is describing is that in
>single-speed (48kS/s) mode, channels 0, 8, and 16 have a 1-sample delay
>with respect to all the other channels (using 0-based indexing for
>channel numbers here). This is irrelevant when recording uncorrelated
>signals, and s
>it's not related with threads, but it invokes a fork for a server
>process (a main control only, doesn't do mixing stuffs).
>it looks like there is something wrong with this together with xmms.
>i've seen that Xlib got a spurious async.
>magically enough, the attached patch seems to fix.
>From ht
>Well, i am not author of this program actually. It is xmms plugin
>allowing to use ALSA for sound output (alsa-xmms-0.9). I only wanted to
>modify it, to be able to use dmix. I see, that it is out of my abilities
>:-( and, so i will have to conctact the author of it and ask him to
>correct i
> Starting to open device
> Opening user device: --default--
> Xlib: unexpected async reply (sequence 0x963)!
your program uses threads, right? or it forks at some point? and your
alsa_error() function involves calls to GUI functions? this error is from
Xlib, it has nothing
>Hi. I'm developping a signal processing program which needs to compute data
>from each channel of soundcards separately.
>After searches I found that I must open the soundcard with the
>SND_PCM_ACCESS_RW_NONINTERLEAVED flag but it doesn't works well : for each
>readn operation i get the -32 er
>I experimented with xmms ALSA output plugin and i forced "default"
>instead of "hw:0:0" to the audio opening command:
>
>snd_pcm_open ( &handle, "default", SND_PCM_STREAM_PLAYBACK,
>SND_PCM_NONBLOCK );
>
>but a error occured. I would like to ask you, if it is sufficient to
perhaps you know thi
>I mentioned my ham radio realtime program with mmap
>in a posting on 3. Oct and before.
>It works now with ALSA 0.9.7 with AC97 and maybe some other cards, thanks!!!
>With ESS ES1969 mmap(ibuf, isize, PROT_READ, MAP_FILE | MAP_SHARED |
>MAP_FIXED, fd_audio, 0)
>(and I tried also all other MAP_ op
>(in hdsp double speed channels is 13 and ss speed are 26,
>in hdspm they are 32 and 64)
>
>Why ? Is there a reason to let channels min not to be 1 ?
ALSA kernel drivers mirror hardware capabilities. its not possible to
configure the hardware to use just 1 channel. user space (alsa-lib)
allows us
>I don't know, if the term "mixer" is used correctly, because i am not
>expert on it. I have in mind to use ALSA by more than one programs at
>the same time (simultaneously), and ALSA itself should mix these audio
>streams to one and play it via soundcard.
just so you know, although this functi
>On Tue, 2003-10-07 at 06:01, Paul Davis wrote:
>> > I agree that they may be related. Possibly my noise only happens
>> >continuously when trying to sync to an external 48K source, and possibly
>> >this is just a sign of it never syncing. When I set the Pref. S
> I agree that they may be related. Possibly my noise only happens
>continuously when trying to sync to an external 48K source, and possibly
>this is just a sign of it never syncing. When I set the Pref. Sync. Ref.
>to ADAT1 and use AutoSync I get the noise continuously and the Sync
>indicator ju
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