Thank you very much. I’ll see what I can do based on your suggestions. Best regards, Lannan Jiang > On Jul 15, 2020, at 6:38 AM, Jeff Long <[email protected]> wrote: > > The signal source is outputting unsigned bytes. The sample rate is 48k and > the tone is 1k. Something I missed before that helps explain your plot ... > the signal is rounded down to zero for all but the peak values, since abs(x) > < 1 can not be represented without scaling. Packed/unpacked refer to bits of > a digital signal. The tone is "analog", but you could think of it a "packed" > if your audio codec is PCM (raw samples). PCM is a valid codec. It's what you > find in a wav file. The problem you will run into is that any lost or > corrupted symbols will ruin the audio. So, you would need to add > framing/packetizing and error correction. Maybe someone else has a link to an > example that shows how this works in GR. The concepts are not specific to GR. > > On Tue, Jul 14, 2020 at 10:50 PM lannan jiang <[email protected] > <mailto:[email protected]>> wrote: > Hi Jeff Long, > Thank you so much for your reply. > > I understand the plot of the signal source now. I have the mpsk_stage6.grc > running properly from the tutorial, and was able to compare the transmitting > and receiving bit streams. I attached the grc file to this email. > Additionally, could you please elaborate more on the byte output of the > signal source? Are they packed? Unpacked? > Moreover, as you stated that i should encode an analog signal to data > before transmission, so does that mean I also have to use codecs in order to > transmit a tone? > My last question would be: if I were to transmit an mp3 file, which is > already encoded, will i be able to recover the audio using audio decoders? > > Thanks again for your help! > > Lannan Jiang > > ps: I apologize for my many questions as they may seem very basic. I am an > engineering student and I am greatly thankful for your advice. > > > > From: Discuss-gnuradio <[email protected] > <mailto:[email protected]>> on behalf of Jeff Long <[email protected] > <mailto:[email protected]>> > Sent: Tuesday, July 14, 2020 9:57 PM > To: GNURadio Discussion List <[email protected] > <mailto:[email protected]>> > Subject: Re: Question regarding transmission of a tone using QPSK > > A better explanation of why that plot is correct: if you sample a tone twice > per cycle, you see [-1,1,-1,1,...]. Four times per cycle, looks like > [-1,0,1,0,...]. Even though it looks discontinuous, it will sound like a tone > when played through your sound card due to filtering in the audio software > and/or hardware. > > That tutorial goes through the low level portions of the digital chain, > including timing recovery. Framing, error correction and (optionally) an > audio codec would all be in addition to the blocks shown in the tutorial. > > On Tue, Jul 14, 2020 at 9:03 PM Jeff Long <[email protected] > <mailto:[email protected]>> wrote: > Depending on your sample rate and tone frequency, that plot would be correct. > > The analog signal needs to be encoded somehow as data before transmission. > While you could feed an audio file 2 bits at a time into a QPSK modulator, > it's pretty unlikely that you will be able to recover the audio. If you're > thinking of "transmitting audio", look into audio codecs. If you're thinking > of sending a wav file, you're really just sending packets. Either way, you > will need a complete chain that includes error correction, clock recovery, > etc. > > On Tue, Jul 14, 2020 at 3:58 PM lannan jiang <[email protected] > <mailto:[email protected]>> wrote: > Hi all, > I have been following the PSK guided tutorial > https://wiki.gnuradio.org/index.php/Guided_Tutorial_PSK_Demodulation > <https://wiki.gnuradio.org/index.php/Guided_Tutorial_PSK_Demodulation> . I am > on the mpsk_stage6.grc, but I want to transmit a simple tone instead of a > random source, so I added a signal source which generates a sine wave. > However, here are my questions: > > 1. I select the output of the signal source as bytes, and the time plot > of it is attached. As you can see, the plot looks like bursts. But if I add > an audio sink after signal source directly, I hear a constant tone. This does > not make sense to me, as I thought I should hear discontinuous sound as the > plot shows, could someone explain this? > > 2. With the first question being said, I am using a constellation > modulator (QPSK) that takes 2 bits/symbol. > How can I feed the output of signal source ( a 16-bit audio file later > on) to the constellation modulator properly? > > Thanks in advance! > > Regards, > Lannan Jiang >
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