> > Is the application supposed to always add silence at the end to ensure that
> > a multiple of period_size has been written?
>
> It looks like a bug. Could you send me a little C code which triggers this
> problem?
Sure, the code is below. Everything is alright until the I drain() before
clos
Hi all,
I have come to realize that, whenever I drain() on a playback_handle to which a
number of sample non-multiple of the period_size has been written, my
application waits for several seconds to finally recover with an error EIO.
Is the driver supposed to play audio if less than period_size i
Hi all,
Could anyone tell me the purpose of the "index" field in struct snd_kcontrol_t
?
Thanks,
Guilhem.
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> This is very good question. We should make a decision. If we use 50Hz, we
> can easily calculate frames per second, otherwise we have to translate
> sampling rate to physical rate.. But it's the only good point which I see.
I am not sure how you'd translate between the two? To me, one option i
Hi,
There are some frame-based formats defined in asound.h and pcm.h. For full-rate
GSM, frames are 260 bits every 20 ms. So my question is:
In such a case, should an application use (and the driver define) a rate of
50Hz, or the standard 8000Hz?
Thanks,
Guilhem.
___
> -d is duration, use -D 'hw:0,1' or 'plughw:0,1'
Damn out-dated MAN pages! Thanks for the hint.
Guilhem.
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Hi all,
I have tried playing audio on device 1 of a "pci2103" card through aplay, which
correctly displayed 2 devices and 1 subdevice each (aplay -l). The result was
that audio was still played on device 0, regardless of the -d parameter.
So my questions are:
- are you aware of any bug in aplay
This card can open / close N audio streams independently. Each audio stream can
be matched to a different port (e.g. mike in, line in) or the same, with
different formats if necessary.
Following your earlier message, I'd guess that a) is the solution. Then, only
one mixer is enough to handle all
Hi,
I am unsure what makes the most sense for multiple pcm channels on the same
card:
a) N devices, 1 pcm channel each
b) 1 device, N pcm channels each
In the same vein, is the mixer always one per card?
Thanks,
Guilhem.
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> > I modified my previous patch for supporting a period_size (rather than
> > period_bytes) constraint with Takashi's suggestion that the fields be
> > non-zero for either constraint to kick in. See file in attachment.
> > It has worked fine on my hardware for the past few days.
>
> you don't ne
Hi all,
I modified my previous patch for supporting a period_size (rather than
period_bytes) constraint with Takashi's suggestion that the fields be non-zero
for either constraint to kick in. See file in attachment. It has worked fine on
my hardware for the past few days.
Guilhem.
Hi all,
I understand that prepare() is used to restart playout or capture from a known
state in the driver, and wonder if it is OK to have it block until buffers
being currently played out have completed?
This came up because our app is experiencing -EPIPE quite often, most likely
because real-t
Jaroslav,
I am working on a conferencing application, and need to deal with audio packets
of various sizes (in sample).
> > It's software only limit for read/write transfers. The direct transfers
> > (mmap) don't take care about this limit. The idea is to avoid using very
> > small transfer chun
> It's software only limit for read/write transfers. The direct transfers
> (mmap) don't take care about this limit. The idea is to avoid using very
> small transfer chunks which are inefficient. Application might change
> this value from 1 (no control) to buffer_size (in samples), but values
> bi
Hi,
I have implemented some hwdep support in a driver, and would like to test it
from an app. As I couldn't find any example of hwdep use in an app either in
the utils or on the mail archive, I figured maybe one of you could help and
point me to some example in source code?
Thanks,
Guilhem.
__
Hi all,
While trying to understand snd_pcm_lib_write1(), I found that
runtime->xfer_align is set to period_size in pcm_native.c, but wonder if it is
just a default?
Paul Davis' example (at http://www.op.net/~pbd/alsa-audio.html#playex) doesn't
check for a particular transfer size, and my card su
> > snd_pcm_hardware_t is NOT pre-initialized, and drivers are responsible to
> > define ALL fields necessary. This is too bad, because it essentially means
> > that a new constraint (i.e. period_size_min + period_size_max) will need
> > to be supported by all drivers.
>
> well, but drivers suppo
Hi,
snd_pcm_hardware_t is NOT pre-initialized, and drivers are responsible to
define ALL fields necessary. This is too bad, because it essentially means that
a new constraint (i.e. period_size_min + period_size_max) will need to be
supported by all drivers.
I have tested successfully such a new
Hi,
I have a problem with supporting multiple audio formats in that my hardware
defines its buffers in a number of frames rather than bytes.
I evaluated the changes required to support description of a hardware's period
contraints in frame AND bytes. Essentially, this would mean adding 'size_t
p
Hi,
I have enquired last Friday about the preferred way for adding formats, since
I'd prefer not follow one option and have someone differ later on. Since I
received no answer, I guess no one as an opinion on the topic and there won't
be any argument. ;)
Anyway, I found out that there's not so m
Hi,
Is this a common occurence that rc3 doesn't compile?
In my case, there are all sorts of isa-pnp related functions that have already
been defined. Note that I did NOT turn isa-pnp ON (the INSTALL files implies
default is OFF).
Regards,
Guilhem.
__
Hi,
It is my understanding that the SNDRV_PCM_FORMAT_IMA_ADPCM format supported by
ALSA is the 32kbps variant of G.726 standard in telephony (vs. propriatory
variants like MS-ADPCM).
I would like to add support for at least the 16kbps variant of G.726 (24kbps
and 40kbps are other possibilities),
Hi,
There are 3 standard PCM ioctls, and a driver could potentially support others
via its own ioctl function registered in the snd_pcm_ops_t structure. Could
anyone give me a hint regarding how an application could then call such
standard and other ioctls?
Thanks,
Guilhem.
___
Takashi,
I have now the playout AND recording alright, thanks to your explanation. Now,
I reset my pointers in prepare(), which makes pointer() return a value
meaningful to the rest of the ALSA framework.
A big THANK YOU!
Guilhem.
PS: Regarding ADPCM, I hope that we'll manage to get other bit r
Is there any way for an application developer to select among those
sub-formats?
Guilhem.
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This sf.net emai
Hi,
I think that I got around understanding snd_pcm_update_hw_ptr_interrupt() now.
The reason of my enquiries is that I get a very strange behaviour while playing
back audio on a sound card: the playback_copy() function is called too many
times before audio is actually played out, thus overwritin
I am getting to know all the internals (although I didn't really want to)...
the ops->pointer() is called only from snd_pcm_update_hw_ptr(), which updates
the runtime->status->hw_ptr. But it doesn't:
playback_copy() on sound dev0, hwoff:0 pos:0 ->160
playback_pointer() on sound dev0, las
> > When does the pointer to the next playback/capture audio period rewind? I
> > noticed that for more than 2 periods_max, this can happen anytime
> > (eg. after 2, 3, ...) even before we reached the end of the buffer.
>
> it must be after the hw_ptr reaches the end of the buffer.
> please check
> > OK, then I know how to solve my problem, but it would have been nice to
> > avoid copy and silence callbacks, and just get from the substream or
> > runtime structure the hw offset where the next period is expected to be
> > written to.
>
> sorry, i don't understand your question.
>
> regar
Hi all,
I would have a few questions regarding buffer mgt:
When is substream->dma_area used for capture or playback? (probably never, as
snd_pcm_lib_malloc_pages() seems to imply that one should only use
substream->runtime->dma_area)
Where should captured audio data be copied into runtime->dma_
Hi,
I wonder if I am right to write audio packets to substream->dma_addr ???
Should I define copy() and silence() functions as well (like for playback)?
Thanks,
Guilhem.
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Hi,
I am starting to write an sound driver for a card with no previous support and
would like to know if DMA is necessary as per the ALSA framework itself. My
card doesn't support DMA, yet. So if your answer is no, I would need to
fallback on OSS for the time being.
Thanks,
Guilhem.
__
Hi,
Now, I have this new driver which manages to open/close a pcm channel whenever
one attempts to play a file with "play test.au", but complains (Sound protocol
it not compatible) with "aplay test.au". Any idea what's wrong?
Besides, I have no mixer defined (I would just to allow open/close a pl
Hi,
I am writing a driver for a new card, and wonder how the ALSA server will
recognize my module as one of its own? Besides, is there anything I should know
regarding debugging ALSA drivers?
Thanks,
Guilhem.
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Hi all,
Could anyone point to me a reference for the following (or explain to me
directly what is the purpose of each of those functions) in the PCM module:
prepare:snd__capture_prepare,
trigger:snd__trigger,
pointer:snd__pointer,
> > What is required from my part to support memory mapping from the driver
> > to the application? Would this be supported through the OSS compatibility
> > layer, too?
>
> hmm... mmap without dma?
> how do you transfer the data on buffer to hardware?
The card doesn't support DMA, yet. Data is
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