Hi Vanush,
remove the gui sink. It also limits the sample processing speed and thus is equally bad as throttle.

Am 09.08.2013 19:33, schrieb Vanush Vaswani:
Hi,

It doesn't help, I still consistently drop samples.

I tried increasing the buffer size in udp_source_impl.cc. I can record to .wav perfectly for about two minutes before the flow graph suddenly exits with (I'm assuming) a segfault.

For reference, I have attached images of the sender and receiver. Surely, this sort of streaming has been done before? I am using a BeagleBone Black running Ubuntu 12.04 (which runs fine - no overflows). The correctness of the FM receiver is less of a concern than the dropped samples.

Regards,
Vanush

On Sat, Aug 10, 2013 at 3:11 AM, Marcus Müller <mar...@hostalia.de <mailto:mar...@hostalia.de>> wrote:

    Hi Vanush,

    ok. Throttle does not _do anything_ with the samples. It just
    slows down the computational speed that they are processed with.
    Remove it, it breaks your flowgraph.
    As said before, the WARN:... warnings stem from your UDP source.
    Data does not get processed fast enough; that's throttle's fault
    (before, the audio sink was limiting
    your sample processing speed, which is in its nature).
    Your flowgraphs are _not_ running at the same rate; they _may_
    process samples at the same average rate (that's what throttle
    does, stop the sample flow when it's faster
    than specified), but throttle does not convert sample rates.
    Refer to existing Receiver flowgraphs from the internet on how to
    do things right, and please remember: 192kHz input MUST be
    downsampled for audio usage.

    Have a nice day,
    Marcus Müller

    Ok, I still don't understand why I'm getting so many dropped packets.

    sender
    FCDPP -> UDP Sink

    receiver
    UDP Src -> _Throttle_ (at 192KHz) -> WBFM Receive -> Wav File Sink

    Note: no audio sinks.
    I constantly get "WARN: Too much data; dropping packet."

    Wave file is full of skips, yet both flowgraphs are running at
    the same rate?


    On Fri, Aug 9, 2013 at 10:37 AM, Iain Young, G7III
    <g7...@g7iii.net <mailto:g7...@g7iii.net>> wrote:

        On 09/08/13 00:22, Vanush Vaswani wrote:

            Hi,

            If i add a rational resampler to match the audio rate
            before the UDP
            sink, I can get a continuous output, but it's of terrible
            quality due to
            the loss of information.


        You are always going to have to throw information away. You
        have 192k
        that you are trying to fit into human hearing, so you *must*
        filter
        and then down sample.

        How wide is your BPF/LPF ? Is it more than your soundcard can
        handle ?


            How does the flowgraph work having a sample rate
            'mismatch' when the
            FCDPP block is in the same graph?!


        Your soundcard (probably) expects 44k1. The resampler takes
        care of
        this, but of course, 192k is not going to go into 44k1. You
        need to
        filter first.


            Could you give me a hint in working around this issue?


        As I said before, google has plenty of examples of FM
        receivers with grc
        (Google for gnuradio FM receiver grc, I'm sure you'll find loads)

        Here are two working flowgraphs that start at 192k and 250k
        respectively:

        http://hal.g7iii.net/GRC/Examples/Simple_Multimode_RX.png
        http://hal.g7iii.net/GRC/Examples/GUI_TRX_JACK.
        <http://hal.g7iii.net/GRC/Examples/GUI_TRX_JACK.png>
        They are actually a multimode receiver I wrote as an example
        for a
        friend, and a multimode transceiver for ham radio use, but
        they show
        the FM receive chain.

        With both, you'll note I have a variable BPF so I can listen to
        broadcast stations on MW/LW, and FM, SSB, or CW stations on
        the amateur
        radio bands.

        For pure FM you can drop the selector blocks, and just
        concentrate on
        the NBFM receiver chain

        Note if you need to be tuning with the FIR filter, you'll
        need to define
        filter taps. I used a simple LPF.


        Hope that helps

        Iain





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