Re: [Discuss-gnuradio] Discuss-gnuradio Digest, Vol 187, Issue 12

[F salem]

Dear all,

I have mini project I need to implement it with SDR-RTL the idea is gathering 
sensors data by SDR.

if you have any resources that can help me please share it

Sincerely
Fahad

From: Discuss-gnuradio  
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Sent: Thursday, May 10, 2018 11:49:46 PM
To: discuss-gnuradio@gnu.org
Subject: Discuss-gnuradio Digest, Vol 187, Issue 12

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Today's Topics:

   1. Re: RTP block / Opus vocoder (Albin Stig?)
   2. Audio File transmission using Qpsk modulation (JONNY LEYIKUN)
   3. Re: Audio File transmission using Qpsk modulation (Jeff Long)
   4. Re: RTP block / Opus vocoder (Adrian Musceac)
   5. Re: RTP block / Opus vocoder (Albin Stig?)


--

Message: 1
Date: Thu, 10 May 2018 18:49:24 +0200
From: Albin Stig? 
To: Michelle Thompson 
Cc: GNURadio Discussion List 
Subject: Re: [Discuss-gnuradio] RTP block / Opus vocoder
Message-ID:

Content-Type: text/plain; charset="UTF-8"

Hi Michelle,

Well my need for RTP/Opus is because of a Raspberry Pi FPGA based
shortwave transceiver for remote ham radio I'm building together with
Daniel, sm6vfz.

Today I sat down and wrote an RTP/Opus sink block and I have to say
it's working really well. I'm using libopus and jrtplib. My original
plan was to create two blocks: an Opus vocoder and and an RTP block
but after a while I found that that didn't make a lot of sense. RTP is
not designed to be a standalone protocol and is very application
specific. So in the end I wrote a combined block. It's probably not
more than 100 lines of C++ more than the boilerplate. Of course I will
have to add a bit more to make it fully configurable. I also haven't
written the xml for grc yet.

I will of course open source my work when I'm a bit further...

Next on my list is to add clock drift compensation on the alsasink.


--Albin SM6WJM

On Thu, May 10, 2018 at 4:54 PM, Michelle Thompson
 wrote:
> Hi Albin,
>
> We have a similar interest in getting RTP functionality in GNU Radio.
>
> Phil Karn recently wrote and published an SDR package for Phase 4 Ground
> that includes RTP multicast functionality. We have been talking about
> getting this functionality into GNU Radio. Another person interested in
> seeing this happen is David Rowetel from FreeDV. He and Phil have started
> talking about this.
>
> Ron and Marcus are of course correct in that RTP streams are already handled
> pretty well in GNU Radio. No, we do not want to reinvent the wheel.
>
> We would like to combine Opus/CODEC2 and RTP multicast to have stereo field
> audio. The sources of the audio appear at different points in the stereo
> field, so that a roundtable conversation feels more like a roundtable, or so
> that two streams from two different SDRs are distinct.
>
> Everyone is welcome to comment and contribute. The repository with the RTP
> multicast code from Phil Karn is here:
> https://github.com/phase4ground/ka9q-sdr
>
> If it can be done without a new block, then great. If there is value in
> handling bookkeeping, or functions in RTP multicast that would be much
> easier or much more useful in flowgraphs with a new block, then that's what
> we want to do. I have advice that it would be useful to get it into blocks.
> We want to do the right thing and we want comment and critique.
>
> Phase 4 Ground will have a "block party" at GNU Radio Conference 2018. The
> overall goal of Phase 4 Ground is to get DVB-S2 and DVB-S2X receive in GNU
> Radio. We have a dedicated room/lab for the week of the conference and will
> bring as much test equipment as we can fit into a minivan. Several
> satellite/modem/DVB docents have signed up and we're working on swag! If you
> have interest or expertise in any area of DVB-S2 receiver design, then we
> enthusiastically welcome you. We have a Slack and a mailing list.
> https://lists.openresearch.institute/
>
> One of the hard parts of the receiver is the LDPC FEC decode. We have one
> working implementation from Charles Brain, written for a GPU. It needs some
> work to get adaptive coding and modulation (we want ACM) working due to the
> architectural choices. It's a big achievement 

Re: [Discuss-gnuradio] Discuss-gnuradio Digest, Vol 187, Issue 13

[F salem]

Dear all

I made alot of post but it didn't shown in the list why?

sincerely
Fahad Alqurashi


From: Discuss-gnuradio  
on behalf of discuss-gnuradio-requ...@gnu.org 
Sent: Friday, May 11, 2018 7:00:10 PM
To: discuss-gnuradio@gnu.org
Subject: Discuss-gnuradio Digest, Vol 187, Issue 13

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When replying, please edit your Subject line so it is more specific
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Today's Topics:

   1. Re: Audio File transmission using Qpsk modulation (JONNY LEYIKUN)
   2. Re: RTP block / Opus vocoder (Adrian Musceac)
   3. Re: Tx Burst for Chirp signals (Yeo Jin Kuang Alvin (IA))
   4. Re: Tx Burst for Chirp signals (M?ller)
   5. Re: Tx Burst for Chirp signals (Yeo Jin Kuang Alvin (IA))
   6. Rx to tx GNU Radio (Yeo Jin Kuang Alvin (IA))
   7. Re: Issue while adding streaming data : Integrating MRC in
  gr-ieee 80211 (Sumit Kumar)


--

Message: 1
Date: Thu, 10 May 2018 17:29:25 -0400
From: JONNY LEYIKUN 
To: discuss-gnuradio@gnu.org
Subject: Re: [Discuss-gnuradio] Audio File transmission using Qpsk
modulation
Message-ID:

Content-Type: text/plain; charset="utf-8"

HI  Jeff Long

Yaah 15K is very minimum to use in USRP sink
I use 15K sample rate in order to get clear  Audio File. but I also use
large sample rate in USRP sink but the result is more or less the same
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Message: 2
Date: Fri, 11 May 2018 00:34:06 +0300
From: Adrian Musceac 
To: Albin Stig? 
Cc: GNURadio Discussion List 
Subject: Re: [Discuss-gnuradio] RTP block / Opus vocoder
Message-ID:

Content-Type: text/plain; charset="UTF-8"

Hi Albin,

Sorry, I was under the impression that RTP would be transmitted over
the air, which in some cases might make sense... like IP multicast
over MPEG-TS over DVB-S. But for Codec2 it does seem like raw audio
transmission is better suited. So then why the need for a Codec2/Opus
RTP block? I would think it would be better left to the application
layer outside GNU Radio to handle IP routing of audio streams?

My previous approach was to use GNU Radio blocks only for the PHY
layer, and use a simple and robust VOIP protocol (Mumble) to
distribute audio over TCP/IP, with some transcoding involved in case
of FM and Codec2 digital voice. With a single GNU Radio instance I
could then serve up to 4 voice channels and one control channel using
FDMA, with a mix of analog and digital channels.
Then again, this is just an amateur radio perspective, where network
capacity is not such a big issue and infrastructure density or SNR
performance are the limiting factor.

Regards,
Adrian YO8RZZ

On 5/10/18, Albin Stig?  wrote:
> Hi Adrian,
>
> UDP and RTP adds a lot of overhead to a codec like Codec2 and doesn't
> make any sense at all unless you wan't to route your packets over an
> IP WAN like the internet. Then it makes a lot of sense.
>
> I imagine the only use case for an RTP/Codec2 or RTP/Opus block is
> streaming audio from a remote receiver/transceiver over an existing IP
> network. If you only stream over LAN, UDP is good enough but the
> Internet an stack RTP is better. All streaming software like skype,
> facetime, hangouts etc use RTP.
>
>
> --Albin SM6WJM
>
> On Thu, May 10, 2018 at 10:05 PM, Adrian Musceac 
> wrote:
>>>
>>> We would like to combine Opus/CODEC2 and RTP multicast to have stereo
>>> field
>>> audio. The sources of the audio appear at different points in the stereo
>>> field, so that a roundtable conversation feels more like a roundtable,
>>> or
>>> so that two streams from two different SDRs are distinct.
>>>
>>
>> Hi Michelle,
>>
>> This is very interesting to me. I did some work with Codec2, Opus and
>> GNU Radio myself.
>> I'm curious though: RTP and UDP + IP encapsulation, isn't that a bit
>> too much overhead for a low bitrate codec like Codec2? A 40 msec audio
>> frame encoded with Codec2 1400 is only 48 bits, so the encapsulation
>> overhead is quite large. How does that affect low SNR