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');
Related
I've recently started working with audioworklets and am trying to figure out how to determine the pitch(s) from the input. I found a simple algorithm to use for a script processor, but the input values are different than a script processor and doesn't work. Plus each input array is only 128 units. So, how can I determine pitch using an audioworklet? As a bonus question, how do the values relate to the actual audio going in?
If it worked with a ScriptProcessorNode, it will work in an AudioWorklet, but you'll have to buffer the data in the worklet because, as you noted, you only get 128 frames per call. The ScriptProcessor gets anywhere from 256 to 16384.
The values going to the worklet are the actual values that are produced from the graph connected to the input. These are exactly the same values that would go to the script processor, except you get them in chunks of 128.
I'm trying to read a lot of data coming from my Arduino, I've set my input buffer to 500000 to make sure that it can handle all these data. My data are 4 sensors readings each samples at 250 Hz. With the default buffer size (712), I used to get snags when I plot the readings in real time and the samples get disordered which makes the plot go crazy. I solved this by increasing the buffer size to 50000. But now, this will work for a while but if I want to run it for 15 minutes, I get the same misbehavior after 5 minutes, with the addition that the plotting gets slower. I do have some processing code along with the live plotting but it shouldn't be like this with such a bi buffer. I want to know whether the buffer will contain all the data from the beginning until it's full or will it keep erasing older data when it gets full (knowing that I already saved it in another vector and plotted it). I truly don't understand why this keeps happening.
kind regards
I.H
When the buffer gets full, once you get new data it erases the old data. The behavior you are seeing is because your processing and your plotting is slower than the flow of the data.
Try to make sure that you optimize you processing
Make sure that for plotting is done by "drawnow". Like this you are sure that if there is anything in the queue it is not executed
Try to avoid saving and keeping all the data
If the problem is still there, you can try to implement a timer to make sure that you are consistent with reading your data
I've been trying to to read an MP4 file using VideoReader. Matlab is able to read the images, but the further the frame is along the video, the more time it takes.
tic;I=read(v,1);toc
Elapsed time is 0.264011 seconds.
tic;I=read(v,2000);toc
Elapsed time is 32.859614 seconds.
Also, I'm not sure if this is related, but Matlab cannot determine the number of frames in the file:
v=VideoReader('S1140007 (~200 cubes, large).MP4');
Warning: Unable to determine the number of frames in this file.
I've tried using two versions R2012b and R2015a, and the problem persists.
On a different machine, however, the number of frames can be determined and the reading times don't get longer, so obviously there's something configured wrong on my machine.
I there a known solution for this problem (can this be related to codecs somehow?), or maybe an alternative method of reading one image at a time (readFrame is not relevant for my needs).
Any help would be appreciated,
Aviram
OK, so this is not exactly an answer, but a workaround...
It seems that to set the NumberofFrames property in the videoreader object created for a video with an undetermined number of frames, one needs to read the last frame using the following code (as mentioned in the documentation of VideoReader):
v=VideoReader('path.mp4');
l=read(v,inf);
This sets the number of frames in the video, and allows for indexing and quick reading of single frames from the video. However, this only works in matlab r2012b. In 2015a, the NumberofFrames property is set by the read(v,inf) trick, but the reading is still very time-consuming, for some reason.
I'm not sure why this happens, and as I've said, some of the other machines I've checked were able to read my files properly (but some didn't), so this is far from completed. It is not clear why it cannot determine the number of frames, or why there's any variability between computers and why in some versions the last(v, inf) works and in others only partially.
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.
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.