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.
Related
Out of curiosity. Not sure whether this is the right place within the Stack universe to ask this.
I know that different bluetooth audio codecs provide different sound quality, and that the quality also depends on a codec's configuration. Overview for example from Wirecutter. What I'd like to know is whether audio devices adjust the codec settings on the fly to compensate for bad connections.
For example, if I play a sound file on my Android device on my car stereo, and something interferes with the bluetooth signal, causing package loss or delay, would the mobile reduce the bitrate in order to ensure that the audio goes on without stopping/stuttering?
Bluetooth will address these types of issues at the radio layer. Therefore it uses Adaptive Frequency Hopping which then avoids the disturbed frequencies.
The audio codec negotiation takes place at the beginning of a connection and is not repeated during the connection.
I work in the field of phonetics and often need to record human speech for acoustic analysis. I have two questions that I couldn't find answers:
If I record in stereo channels, I need to convert to mono later on to proceed with annotation. So in principle mono signal is good enough. Are there reasons that stereo sound should be used (e.g. the signal would be better?)
Also, we were warned that the gain level should be kept small so that the recording level shouldn't exceed the maximum, which leads to signal cuttoff. However, I was also criticised when the recording file shows too low an amplitude (it's still very clear though), for that leads to a low SNR. How do people choose an appropriate gain level?
As the act of recording is involved, the Sound Design forum might be your best bet.
I can't think anything that might be gained, in terms of frequency analysis, by having a stereo signal. Stereo is more about locating the source of a sound in 3D space. Does the source of sound emit different frequency profiles in different directions? Does the environment filter the sound differently over the course of the two paths to the stereo inputs? If the the answer is "not significantly" then mono should be fine.
Choosing an appropriate gain level is mostly a matter of knowing your equipment. Ideally, your recording setup will provide feedback (usually a visual meter of some sort) that shows the signal strength. The "best" would be (theoretically) the loudest level that does not distort. So you have to know at what level distortion happens on all the elements of the recording chain.
There can be some fudging on this, given that the loudest peak on a recorded segment may be an outlier.
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.
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.