[Alsa-devel] Rate Conversion
I am writing a PCM application and wish to play a .wav file that has a
sampling rate of 8kHz on a hardware setup that only supports sampling
frequencies of 44.1 or 48kHz.
Does anyone know how to implement this?
I have tried setting up a rate plugin in my .asoundrc file
pcm.r8 {
type rate # Rate PCM
slave slave_rate8000Hz # Slave name
}
pcm_slave.slave_rate8000Hz {
pcm "hw:0,0"
rate 8000
}
But when using this device I am unable to set the rate to 8000 using
snd_pcm_hw_params_set_rate_near ();
I tried using the r8 device with aplay (version 0.9.0beta12) and get the
following error
> aplay -Dr8 s1.wav
Playing WAVE 's1.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono
aplay: set_params:717: Broken configuration for this PCM: no
configurations available
I know it is possible to play the file because it works when I play it
from a netscape window.
Thanks in advance
Kris Modrak
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Re: [Alsa-devel] Rate Conversion
Jaroslav Kysela wrote: > > On Thu, 27 Jun 2002, Kris Modrak wrote: > > > I am writing a PCM application and wish to play a .wav file that has a > > sampling rate of 8kHz on a hardware setup that only supports sampling > > frequencies of 44.1 or 48kHz. > > > > Does anyone know how to implement this? > > You don't need to do this. Use the 'plughw' device which should do all > conversions from you. I am a little confused about your advice. I am not sure how to access the 'plughw' device from my application. I have a full duplex application that can play a file at 44.1kHz and record at the same rate but I want it to operate at lower sampling frequencies. I am not sure of how I should be setting up my PCM devices. How do I tell them to convert, say, 8kHz audio data to 44.1kHz audio data so my hardware can play it at the correct speed? I tried using aplay with plughw to play an 8kHz file and got the following results >aplay -Dplughw s1.wav Playing WAVE 's1.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono Segmentation fault However it worked when using a 441.kHz file. Regards Kris Modrak --- This sf.net email is sponsored by:ThinkGeek Bringing you mounds of caffeinated joy. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Rate Conversion
Paul Davis wrote: > > >I am a little confused about your advice. I am not sure how to access > >the 'plughw' device from my application. > > snd_pcm_open (&handle, "plughw:N,M", ...) > > where N and M are the card number and subdevice number. Preferably, > this is a string supplied by the user, not hard coded into the application. > > >I am not sure of how I should be setting up my PCM devices. How do I > >tell them to convert, say, 8kHz audio data to 44.1kHz audio data so my > >hardware can play it at the correct speed? > > just as in the few token API examples. using > snd_pcm_hw_params_*(). there is no difference setting the parameters > no matter what "type" of PCM device you are using. Apart from rates or 44.1 or 48kHz I was unable to find a rate/period size combination that I was able to use with plughw:0,0. snd_pcm_hw_params_set_period_size () failed for every period size I tried (64, 128, 256, 512, 1024, 2048 and 4096) when I had the rate set to 8, 16 or 32 kHz. I managed to set the following combinations of hardware parameters: -22050Hz with period sizes of 2048, 1024, 512, 256, 128 and 64 -11025Hz with period sizes of 1024, 512, 256, 128 and 64 but when I ran my application I encountered the following error ALSA lib pcm_hw.c:380:(snd_pcm_hw_start) SNDRV_PCM_IOCTL_START failed: Broken pipe Does anyone have any further suggestions of things that I could try or why this rate conversion isn't working for me? I am using ALSA 0.9.0beta12 with a Hammerfall Light (RME Digi9636) soundcard. Kris Modrak --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Rate Conversion
Paul Davis wrote: > > >> You don't need to do this. Use the 'plughw' device which should do all > >> conversions from you. > > > >I am a little confused about your advice. I am not sure how to access > >the 'plughw' device from my application. > > snd_pcm_open (&handle, "plughw:N,M", ...) > > where N and M are the card number and subdevice number. Preferably, > this is a string supplied by the user, not hard coded into the application. > > >I am not sure of how I should be setting up my PCM devices. How do I > >tell them to convert, say, 8kHz audio data to 44.1kHz audio data so my > >hardware can play it at the correct speed? > > just as in the few token API examples. using > snd_pcm_hw_params_*(). there is no difference setting the parameters > no matter what "type" of PCM device you are using. If use the virtual plughw:0,0 device and set the rate to 8kHz, at what rate is my actual hardware device set to? Where are these settings made and how are they decided? Do I need to, or is it possible to, tell my soundcard to operate at 44.1kHz? Kris Modrak --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] .asoundrc file
The start of my .asoundrc file is as follows:
pcm.rme9652 {
type hw
card 0
device 0
}
ctl.rme9652 {
type hw
card 0
}
What is the purpose of the ctl.rme9652 declaration? My plugin devices
seem to work without it.
Regards
Kris Modrak
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Re: [Alsa-devel] .asoundrc file
Patrick Shirkey wrote:
>
> Kris Modrak wrote:
> > The start of my .asoundrc file is as follows:
> >
> > pcm.rme9652 {
> > type hw
> > card 0
> > device 0
> > }
> >
> > ctl.rme9652 {
> > type hw
> > card 0
> > }
> >
> > What is the purpose of the ctl.rme9652 declaration? My plugin devices
> > seem to work without it.
> >
>
> It is a configuration for the onboard mixer. Very little is known about
> why we use it. Some programs (JACK) won't work without it.
The Hammerfall Light doesn't have a mixer I guess I don't need it?
>
> > Regards
> > Kris Modrak
> >
> >
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[Alsa-devel] RME 9636 lockup
I am writing an application that uses an RME9636 Hammerfall light with the ALSA API. Occasionally (and randomly) the card is left in a locked up state by my application as the next time I run the program the card is unable to signal to the computer when there is capture or playback data to be transferred and doesn't signal xruns. At this point I am unable to use the card with any ALSA applications, aplay etc., until after rebooting the computer. Why might this be happening? Could the program be exiting when the card is temporarily is a state of being locked? I am using threads, could this have something to do with it? And is there any way of resetting the card in software, ie. is there any ALSA API function that does this? I would prefer not to reboot the computer every time that this happens. Thanks Kris Modrak --- This SF.net email is sponsored by: SF.net Giveback Program. Does SourceForge.net help you be more productive? Does it help you create better code? SHARE THE LOVE, and help us help YOU! Click Here: http://sourceforge.net/donate/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
