Re: [Discuss-gnuradio] sound card sampling rate question

that was it indeed: selecting out of the aplay -L options the default setup
yields the behaviour I described in my email, while selecting hw as shown in
the output of aplay -L

hw:CARD=PCH,DEV=0
HDA Intel PCH, ALC269VC Analog
Direct hardware device without any conversions

allows me to generate a clean 32768 Hz sine wave.

Thanks, JM

> guessing in the dark: You are likely running a modern linux
> distribution. Many of these tend to emulate an Alsa device, which in
> fact is but an Pulseaudio server, which in turn speaks to the real
> device. They set that as the default audio device in the system.
> Have you tried using the explicit device name (aplay -L) of your
> physical device in the audio sink?
>
> Greetings,
> Marcus
>
> On 25.06.2014 10:40, jmfriedt wrote:
> > I am facing a funny issue: I want to use the sound card for analyzing the spectral response
> > of a quartz tuning fork at 32768 Hz. I just happened to discover that my laptop (Panasonic CF-19)
> > has a sound card able to sample a signal at 192 kHz. I checked with audacity (Generate -> Tone at
> > 32768 Hz and a frequency counter gives the right output frequency while an oscilloscope displays
> > a clean sine wave).
> > Now I want to do the same with gnuradio-companion: Signal Source sampled at 192 kHz, output frequency
> > at 32768 Hz, directly connected to the audio sink manually set to 192 kHz. To make a long story short:
> > at low frequency (<20 kHz output) the output is at the right frequency, so the sampling rate is
> > properly understood. Above 24 kHz I get a clean sine wave output at f-24 kHz, so it looks like an
> > aliasing effect with a sampling frequency of 48 kHz, which is not consistent with my first observation.
> > And setting an output frequency of 24.xx kHz (xx=300 or 400 Hz) generates on the oscilloscope a funny
> > low frequency beat signal which must be related to the antialiasing filters of the card.
> >
> > What I cannot understand is where gnuradio fails to initialize the sound card the way audacity does.
> > Reading the source code, I find in gr-audio/lib/alsa/alsa_sink.cc the following intialization
> >
> > // sampling rate
> > unsigned int orig_sampling_rate = d_sampling_rate;
> > if((error = snd_pcm_hw_params_set_rate_near(d_pcm_handle, d_hw_params,
> > &d_sampling_rate, 0)) < 0)
> > bail("failed to set rate near", error);
> >
> > if(orig_sampling_rate != d_sampling_rate) {
> > fprintf(stderr, "audio_alsa_sink[%s]: unable to support sampling rate %d\n",
> > snd_pcm_name(d_pcm_handle), orig_sampling_rate);
> > fprintf(stderr, " card requested %d instead.\n", d_sampling_rate);
> > }
> >
> > which does not seem to test whether the sampling rate is above or below 48 kHz (as found in the
> > pull down menu of the Audio Sink block of gnuradio-companion), and I get no error message when
> > running my application.
> > Any idea what could be going wrong ?
> >
> > Thanks, JM
> >
>
>
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--
JM Friedt, FEMTO-ST Time & Frequency/SENSeOR, 32 av. observatoire, 25044 Besancon, France

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