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.
Related
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.
I've implemented a PIC32 as a USB sound card, using USB Audio Class 1. I'm sending a sawtooth signal from the microcontroller to the PC(windows 7, 64 bit), as 16-bit samples:
in decimal:
000
800
1600
2400
.. so on
then i try recording the received audio using Audacity, with MME -driver, as .wav or .raw.
I use MATLAB to open and inspect the data, and there i see data like:
000
799
1599
2400
..
The distortion varies from -1 to +1 bit pr sample..
Anyone have any idea where the problem might be.?
Windows-audio drivers.?
Since you receive the audio signal on PC, playback it, and record it using SW, the audio signal is converted from digital to analog, and to digital again. These introduce quantization error and noise, and you see the little difference between two signals.
I solved my problem..
The problem was caused by the application i used to record the data, and the method i used.. I used Audacity, which supports the old windows MME audio API, and the DirectSound API. These are relatively high-level API's apparently, and are the cause of the distortion.
About the Windows Core Audio APIs
Instead i used another program, called Reaper, it has an option to record using ASIO og WASAPI. This solves my problem. I've checked every sample in an 2 hour .wav file, using MATLAB, and it is completely bit-perfect.
I was probably some quantization error, but it was caused by the API.
ASIO and WASAPI gave me bit-perfect sound, MME and DirectSound gave me a distorted signal.
I have a matlab stereo audio input ala:
aud=analoginput('winsound',0);
addchannel(aud,1:2)
When I'm done running this and ask for
aud_data=getdata(aud);
I get an array showing all the data from both channels. I have verified that this data is, in fact, the valid stereo signal that I want.
However, if I run
aud_peek=peekdata(aud,some_number_of_samples);
whilst data is being collected, I only seem to get one channel's worth of data, though MathWorks says I should get two.
Any thoughts as to why this is happening?
I am using Matlab 7.
Did you try
aud_peek=peekdata(aud,some_number_of_samples,'native');
or
aud_peek=peekdata(aud,some_number_of_samples,'double');
How do I force MATLAB functions to use a different codec when I need to play an AVI file?
I am using windows7 and I find that indeo5 codec is not working properly because of the OS.
All code works fine on XP.
Thanks!
If your question is about how to play an avi using a codec other than the one used to create it, you'll have to use some kind of image conversion software ( RAD video tools aka bink and smacker is popular and free ) to recompress the movie using the new codec. There may be some loss of video quality.
If your question is how to tell MATLAB to create an avi using a different codec, type help avifile to see the full documentation.
If your problem is that MATLAB is choking on a codec it says it supports and if you're using 64 bit MATLAB, I've had this problem. The short answer, unfortunately, is that there are no 64 bit codecs that work with MATLAB. There are two workarounds, and they both suck. One is to install 32 bit MATLAB and use that when you need to access an avi. The other is to use some kind of image conversion software to convert your avi to an image stack, then load that with MATLAB.
(EDIT: In newer versions of MATLAB, you can use the VideoWriter object to write MP4 and other compressed formats in 64 bit windows)
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.