I'm regularly testing smartphones for my blog and I'd like to measure their audio quality (file playback but also call quality).
I thought of connecting the jack port from the smartphone to the line-in input of my soundcard and play some sounds to measure the quality through a software.
I'd like to measure sound quality based on THD, SNR, Crosstalk, ...
Is there a software I could use to do this? Would you recommend another method to achieve the same?
Thanks
Laurent
I found found something that might do the trick. It's a program called RightMark Audio Analyzer, it can use a test signal on external devices to be re-analyzed through line-in.
Related
We want to test sound characteristics of lets say an android device emitting audio out of the speaker, is there any Audio Testing Hardware which can tell in real time the heard sound pitch, frequency, loudness , etc?
I have searched for this Audio Testing Hardware
If there is a single steady tone, a Tuner (that musicians use to practice with to check their intonation) would give you the frequency.
The free software Audacity has some nice analysis tools, including fourier analysis.
For more advanced tools, I'd check at the Signal Processing or Sound Design forums.
is there a scenario where we can use the Google Resonance Audio SDK not with headphones, but with real speakers (e.g. mounted in a 360° cyrcle setting)?
Or are all algorithms not working for real speaker outputs?
Thank you!
Currently, Resonance Audio is optimized for headphone playback. For example, HRTF processing is done in the Ambisonics domain, without generating (virtual) speaker signals - this is because it is a much more efficient way of generating binaural output.
However, in the Resonance Audio open source release, the Ambisonic Codec class can readily be used to decode Ambisonics to any arbitrary loudspeaker array. To use that with the rest of the Resonance Audio system, however, it would be necessary to modify/extend the audio processing graph by adding a new decoder node.
Please, feel free to add a feature request and, depending on popularity, we might consider adding that in the future!
I'm trying to make a video tutorial, so i decided to record the speeches using a TTS online service.
I use Audacity to capture the sound, and the sound was clear !
After dinning, i wanted to finish the last speeches, but the sound wasn't the same anymore, there is a background noise(parasite) which is disturbing, i removed it with Audacity, but despite this, the voice isn't the same ...
You can see here the difference between the soundtrack of the same speech before and after the occurrence of the problem.
The codec used by the stereo mix peripheral is "IDT High Definition Codec".
Thank you.
Perhaps some cable or plug got loose? Do check for this!
If you are using really cheap gear (built-in soundcard and the likes) it might very well also be a problem of electrical interference, anything from ...
Switching on some device emitting a electro magnetic field (e.g. another monitor close by)
Repositioning electrical devices on your desk
Changes in CPU load on your computer (yes i'm serious!)
... could very well cause some kinds of noises with low-fi sound hardware.
Generally, if you need help on audio sounding wrong make sure that you provide a way to LISTEN to the files, not just a visual representation.
Also in your posted waveform graphics i can see that the latter signal is more compressed, which may point to some kind of automated levelling going on somewhere in the audio chain.
I've been looking online for solutions on how to generate a simple beep with an DE2 Altera board using VHDL but I can not seem to find anything.
I've seen some things that are talking about Audio Codec but I haven't been able to get a firm grasp of what I would need to do with it.
Thanks a lot!
-Will
The "audio codec" is a separate chip (a Wolfson WM8731 in your case) which can turn information on digital form into analog information to be amplified/played through a standard sound system with amplifier/speaker/etc.
You'll need to find out how to interface to the audio codec chip from the FPGA (google a datasheet for the chip), and then send it the necessary information to generate you beep. Depending on how advanced the audio chip is, this task can vary rather much in complexity. As it's an educational board, you might be able to find people who've already done something similar, and then build on their experience.
If all you need is a simple beep though, you might also be able to create a simple circuit with a piezo-buzzer or similar yourself - that can be controlled without having to use an audio codec...
Having just witnessed Sound Load technology on the Nintendo DS game Bangai-O Spritis. I was curious as to how this technology works? Does anyone have any links, documentation or sample code on implementing such a feature, that would allow the state of an application to be saved and loaded via audio?
Its the same old thing used in ZX Spectrum era. You load programs/games from tape.Only the sound quality and the filters are probably better.
In my opinion something like Bluetooth or WiFi is better. You can also send files that can be put on some storage and then load them. I find these methods much easier than sound because if there is a lot of noise around you cannot do much.
It is just a conversion of data to audio and then back from audio to data.
Search for Zotyocopy and Copy86M on google - these are the utilities used for saving a game to tape after loading it into memory on zx spectrum.
If you want to pass data as audio through the air there are a few things you need to be aware of though, such as how the speaker and microphone interact for example. It is important that they don't distort or alter the sound too much as what you are sending are in fact the raw bytes.
Some audio software will let you open any file as audio so that you may listen to it. If you record audio as data do not use lossy compression such as mp3 on the audio file!