Receiving the right value when transmitting .dat file using FM radio - matlab

I am new to GNU Radio and I'm trying to transmit a value using it and the USRP B210 board.
I used Matlab to convert the value 0.121 to wav format then convert the wav file to .dat file using audio_to_file example in GNU Radio.
When I transmit the .dat file using the B210 and GNU Radio, I received a wav file but when I read the wav using matlab function (audioread()) I get a different value.
P.S.
Sample rate for the converted .dat file was 44100 Hz and 16 bits per sample.
The receiver and transmitter sampling rate is 400K Hz.
I used fm_tx4.py example from the GNU Radio package for my transmitter.
I used uhd_nbfm_receiver.grc for the receiver.


If you're wondering why your received signal doesn't have the same amplitude as your sent signal, you're not getting the very basics of radio communications: as there is no digital line between your transmitter and your receiver, power can go anywhere, and how much reaches the receiver depends on a lot of factors, including gain, antennas, distance, matching...
There will be a lot more things that are different on the RX side than they were on the TX side: Your reception has not been time-synchronized, so you might see a phase shift. You don't mention whether the receiver is the same, a clock-synchronized or an clock-independent B210, which means you have the general case, where no two physical clocks can be identical (yes, that's impossible, but you can reduce errors), so you'll generally see some frequency offset, too.
I recommend reading up a bit on basic radio comm theory, I often recommend GNU Radio's pictured introduction, and GNU Radio's suggested Reading Page. Michael Ossmann gets some recognition for his courses, too, so you should definitely have a look at them.
Also, all your data->Wav->transmit conversion is totally unnecessary. Matlabs fread/fwrite functions can read/store the native machine float format that GNU Radio's file_sink/file_source can store/read. See the FAQ entry.

Related

Matlab noise source discontinuity

Using Matlab, I've made some random noise, filtered it and then successfully saved it as a gnuradio readable file for a file source. Once used in gnuradio, I set the file source to repeat and then viewed it using QT Gui Frequency Sink. I can see the filtered noise fine, but every now and then (every 10 seconds or so), the spectrum will drop in power and jump around for around a tenth of a second, then return back to normal power. My sample rate for the matlab filter is 320k and same with my gnuradio sample rate if that matters.
I think it may have to do with the fact that the noise generated on matlab is going to be a sequence that is repeated on gnuradio. I think the discontinuity happens right when the sequence repeats. Any idea how I can stop this discontinuity so I can transmit without having to worry about it? If I'm missing any info, please let me know and I'll edit the question. Thanks in advance.
NOTE: I needed to create a matlab binary file to be able to read it on GNU Radio. GNU Radio reads the binary file from my desktop, then uses the information as the file source.

Output 4 channels of audio in MATLAB

I'm looking to output four channels of audio simultaneously from MATLAB using an external soundcard (Creative Soundblaster X-Fi Surround 5.1 Pro USB) and haven't yet found a working solution.
As far as I understand it, MATLAB's audioplayer object can only output a stereo signal, so I've tried two alternatives: playrec and pa_wavplay. Both appear to do precisely what I need, but seem to recognize the soundcard as a two-channel device only.
Any advice would be terrific. Thanks for reading.
(The MATLAB version is R2007b and the only available toolbox is the Signal Processing Toolbox.)

I've got a bit of experience of pa_wavplay and found it dealt with large numbers of inputs/output without any problems. I'd suspect the problem is with your audio interface.
While it can output 5.1, it's quite possibly producing those "additional" channels itself by decoding a Dolby Digital stream once in the device. This suggests the interface won't allow you to output 6 six channels of PCM audio as such.
If you're determined to use this device and prepared to get your hands dirty you could always try encoding your audio as ac3 yourself, but I guess you'd have to do this outside Matlab.

Transferring data using ultrasound

Yamaha InfoSound and ShopKick application use technologies that allow to transfer data using ultrasound. That is playing an inaudible signal (>18kHz) that can be picked up by modern mobile phones (iOS, Android).
What is the approach used in such technologies? What kind of modulation they use?

I see several problems with this approach. First, 18kHz is not inaudible. Many people cannot hear it, especially as they age, but I know I certainly can (I do regular hearing tests, work-related). Also, most phones have different low-pass filters on their A/D converters, and many devices, especially older Android ones (I've personally seen that happen), filter everything below 16 kHz or so. Your app therefore is not guaranteed to work on any hardware. The iPhone should probably be able to do it.
In terms of modulation, it could be anything really, but I would definitely rule out AM. Sound has next to zero robustness when it comes to volume. If I were to implement something like that, I would go with FSK. I would think that PSK would fail due to acoustic reflections and such. The difficulty is that you're working with non-robust energy transfer within a very narrow bandwidth. I certainly do not doubt that it can be achieved, but I don't see something like this proving reliable. Just IMHO, that is.
Update: Now that i think about it, a plain on-off would work with a single tone if you're not transferring any data, just some short signals.

Can't say for Yamaha InfoSound and ShopKick, but what we used in our project was a variation of frequency modulation: the frequency of the carrier is modulated by a digital binary signal, where 0 and 1 correspond to 17 kHz and 18 kHz respectively. As for demodulator, we tried heterodyne. More details you could find here: http://rnd.azoft.com/mobile-app-transering-data-using-ultrasound/

There's nothing special in being ultrasound, the principle is the same as data transmission through a modem, so any digital modulation is -in principle- feasible. You only have a specific frequency band (above 18khz) and some practical requisites (the medium is very unreliable, I guess) that suggest to use a simple-robust scheme with low-bit rate.

I don't know how they do it but this is how I do it:
If it is a string then make sure it's not a long one (the longer the higher is the error probability ). Lets assume we're working with the vital part of the ASCII code, namely up to character number 127, then all you need is 7 bits per character. Transform this character into bits and modulate those bits using QFSK (there are several modulations to choose from, frequency shift based ones have turned out to be the most robust I've tried from the conventional ones... I've created my own modulation scheme for this use case). Select the carrier frequencies as 18.5,19,19.5, and 20 kHz (if you want to be mathematically strict in your design, select frequency values that assure you both orthogonality and phase continuity at symbol transitions, if you can't, a good workaround to avoid abrupt symbols transitions is to multiply your symbols by a window of the same size, eg. a Gaussian or Bartlet ). In my experience you can move this values in the range from 17.5 to 20.5 kHz (if you go lower it will start to bother people using your app, if you go higher the average type microphone frequency response will attenuate your transmission and induce unwanted errors).
On the receiver side implement a correlation or matched filter receiver (an FFT receiver works as well, specially a zero padded one but it might be a little bit slower, I wouldn't recommend Goertzel because frequency shift due to Doppler effect or speaker-microphone non-linearities could affect your reception). Once you have received the bit stream make characters with them and you will recover your message
If you face too many broadcasting errors, try selecting a higher amount of samples per symbol or band-pass filtering each frequency value before giving them to the demodulator, using an error correction code such as BCH or Reed Solomon is sometimes the only way to assure an error free communication.
One topic everybody always forgets to talk about is synchronization (to know on the receiver side when the transmission has begun), you have to be creative here and make a lot a tests with a lot of phones before you can derive an actual detection threshold that works on all, notice that this might also be distance dependent
If you are unfamiliar with these subjects I would recommend a couple of great books:
Digital Modulation Techniques from Fuqin Xiong
DIGITAL COMMUNICATIONS Fundamentals and Applications from BERNARD SKLAR
Digital Communications from John G. Proakis

You might have luck with a library I created for sound-based modems, libquiet. It gives you a handful of profiles to work from, including a slow "Ultrasonic whisper" profile with spectral content above 19kHz. The library is written in C but would require some work to interface with iOS.

Watermarking sound, reading through iPhone

I want to add a few bytes of data to a sound file (for example a song). The sound file will be transmitted via radio to a received who uses for example the iPhone microphone to pick up the sound, and an application will show the original bytes of data. Preferably it should not be hearable for humans.
What is such technology called? Are there any applications that can do this?
Libraries/apps that can be used on iPhone?

It's audio steganography. There are algorithms to do it. Refer to here.

I've done some research, and it seems the way to go is:
Use low audio frequencies.
Spread the "bits" around randomly - do not use a pattern as it will be picked up by the listener. "White noise" is a good clue. The random pattern is known by the sender and receiver.
Use Fourier transform to pick up frequency and amplitude
Clean up input data.
Use checksum/redundancy-algorithms to compensate for loss.
I'm writing a prototype and am having a bit difficulty in picking up the right frequency as if has a ~4 Hz offset (100 Hz becomes 96.x Hz when played and picked up by the microphone).
This is not the answer, but I hope it helps.

iPhone audio and AFSK

Here is a question for all you iPhone experts:
If you guys remember the sounds that modems used to make, or when one was trying to load a program from a cassette tape – I am trying to replicate this in an iPhone for a ham radio application. I have a stream of data (ASCII) and I need to encode it as AFSK at 1200 baud. So basically everything in the stream is converted to a series of 1200 and 2200 Hz tones. It needs to sound something like this: http://upload.wikimedia.org/wikipedia/commons/2/27/AFSK_1200_baud.ogg
I successfully built a bit array out of the string, but when I try to assign tones to each bit I get gaps in the sound, therefore it doesn’t demodulate correctly.
Any thought of how one should tackle this problem? Thank you.

The mobilesynth project is open-source. You might be able to scan that for code that generates the tones you need.

How are you assigning tones to the bits? Remember, a digital audio signal is just a stream of samples with values between -1 and 1. Perhaps there is a clipping issue between tone assignments. This can happen if the signal dives below -1 or above 1. If it stays above or below this range at a constant value, there will be no sound. Maybe you could output your stream of samples to check if this is the case. Or plug the output into an oscilloscope...
Also note that clicking can occur between "uneven" transitions of signals. For example if i output a sample with value 1 followed immediately by a sample with value -1, a click or pop will be produced.