with the use of DLL (delay locked loop).
Sent: Wednesday, September 27, 2017 10:45 PM
To: Marcus Müller; GNURadio Discussion List
Subject: Re: [USRP-users] Audio Control loop testing
>>Could you maybe elaborate how you're planning to solve all a),b),c) instead of asking for new feedback?
For a) & b) will use the sound card clock and using micro seconds timer.
And for c) run the decoded PCM through a FIFO buffer this is a local buffer which is not part of gnu-radio connect buffers, between the SRC and the play-out stage. The trade-off for this approach of course is increased latency.
This way any variable work-load length is not going to affect and the local fifo will have fixed length.
Timing errors needs to be filtered using DLL which is the same used in JACK.
-ben
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And as also said earlier, I don't believe very much that it will work that easily, since the CPU clock is a) worse than the typical SDR and sound card clocks, b) has different resolutions, c) and needs to still be sufficiently interpolatable for the jittery, variable-workload-length that GNU Radio has. The point c) is what's different for Jack internally, because that can work on fixed-length buffers.
This is a comment that you've gotten from me (and by the way, Fons, too) multiple times now. Could you maybe elaborate how you're planning to solve all a),b),c) instead of asking for new feedback?
Sent: Wednesday, September 27, 2017 6:50 AM
To: Marcus Müller; GNURadio Discussion List
Subject: Re: [USRP-users] Audio Control loop testing
Hi Benny,
you're, again, missing the core problem: it's hard to measure the time deviation between two symbols without a better reference clock. And you don't have that. And thus, we're back at the start of all our email chain.
Best regards,
MarcusSent: Tuesday, September 26, 2017 10:56 PM
To: Marcus Müller; GNURadio Discussion List
Subject: Re: [USRP-users] Audio Control loop testing
Sent: Friday, September 22, 2017 10:26 PM
To: Marcus Müller; GNURadio Discussion List
Subject: Re: [USRP-users] Audio Control loop testing
Sent: Tuesday, September 19, 2017 10:47 PM
To: Benny Alexandar; GNURadio Discussion List
Subject: Re: [USRP-users] Audio Control loop testing
Hi Ben,
May I know why not with JACK ?
From the very same email you're referring to:
(not much sense writing it for the Jack sink, if Jack can already do it internally)Also,
Here, I need your inputs.I spent around 5 hrs on input on this topic already. I don't feel like you need more input, it feels more like you haven't had the chance yet to understand all the input that there is on the GNU Radio mailing list. We should also not be having this discussion on usrp-users, as your approach doesn't involve USRPs directly!
Can you please state the requirements. How it has to be in GNU radio chain etc.
Please re-read my previous email. I explicitly say I'm not even convinced this will reliably work in software. GNU Radio is software.
What about you just start by trying to implement a control loop, and read as much on theory of discrete-time control systems as you'll need for this? I'm afraid I can't take that burden off your shoulder if you want to implement a control loop. It is hard stuff.
Best regards,
Marcus
Hi Marcus,
Yes its true I couldn' t make much progress on this. Not able to find time as I have a full time job. If I remember correctly, you mentioned that no-one has implemented audio control loop within GNU Radio. And you were suggesting to write it for ALSA and not with JACK.
May I know why not with JACK ? If I need to make it with JACK, GNU radio should run as a client and output to JACK input port and another client which does the audio control loop and send the output for playback. May be its not required, if we can make a sink block with ALSA and implement the audio control loop.
Here, I need your inputs. Can you please state the requirements. How it has to be in GNU radio chain etc.
-ben
From: USRP-users <usrp-users-bounces@lists.ettus.com> on behalf of Marcus Müller via USRP-users <usrp-users@lists.ettus.com>
Sent: Tuesday, September 19, 2017 2:10 AM
To: usrp-users@lists.ettus.com; GNURadio Discussion List
Subject: Re: [USRP-users] Audio Control loop testingHi Ben,
that's the old multi-clock problem we've been talking about multiple times – it's hard to even define what the "correct" clock is, so you usually just settle on recovering the transmitter clock and, if you were doing this in hardware, would derive the audio DAC's clock from that.
In a software receiver, you need to estimate the offset of the audio DAC clock from the sender's audio clock. That's hard to do properly, because these clock offsets might be to fine to do it with general purpose PC CPU software. But we've talked about all that before on the Discuss-gnuradio list!
As a way around that, you might use the same clock to derive the RF receiver's sampling clock and the audio DAC's sampling clock. You then get a direct relation between RF sampling and audio playback, for example "every 1 million RF samples, I need to produce one audio sample". Fons and I really tried to explain that in about 20 emails on discuss-gnuradio. So, I think we've covered the stage of "any suggestions on this would be helpful" pretty well. It is a hard problem, and there's a solid chance you can't solve it for all use cases in software. There's also a solid chance you might be able to solve it for a specific use case, but that would require you to become an expert on multi-rate processing and clock matching, and frankly, you're not showing much progress at that over last 10 months.
Best regards,
Marcus
On 09/16/2017 05:38 AM, Benny Alexandar via USRP-users wrote:
Hi,
I want to create an artificial audio drift in transmitter side and test it using my audio control loop in receiver. This is what I'm planning.
Take an audio wav file which is sampled at 12 kHz. Re sample it such that the sample rate is now having a drift of 100 ppm, ie with sample frequencies with an error up to 12000*100e-6 is 1.2Hz in case of 12kHz sample frequency. Now transmit this audio file using Gnu radio and USRP.
Receiver does the channel decoding and audio decoding.
So in this most extreme case the receiver drifts with more than one sample per second, so after an hour it is drifted by 1.2*3600 = 4320 samples
If the receiver doesn't have an audio control loop then it will go into under run. By enabling the audio control loop i can check the drift compensation.
Any suggestions on this method of testing.
-ben
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