Develop Reference

How to use noise meter package in flutter to give only few decibel readings per second

I am using noise meter to read noise in decibels. When I run the app it is recording almost 120 readings per second. I don't want those many recordings. Is there any way to specify that I want only one or two recordings per second like that. Thanks in advance. noise_meter package.
I am using code from git hub which is already written using noise_meter github repo noise_meter example
I tried to calculate no. of samples using sample rate which is 40100 in the package. but I can't understand it.

As you see in the source code , audio streamer uses a fixed size buffer of a new thousand and an audio sample rate of 41000, and includes this comment Uses a buffer array of size 512. Whenever buffer is full, the content is sent to Flutter. So, small audio blocks will arrive at the consumer frequently (as you might expect from a streamer). It doesn't seem possible to adjust this.
The noise meter package simply takes each block of audio and calculates the noise level, so the rate of arrival of those is exactly the same as rate of arrival of audio blocks from the underlying package.
Given the simplicity of the noise meter calculation, you could replace it with your own code directly on top of audio streamer. You just need to collect multiple blocks of audio together before performing the simple decibel calculation.
Alternatively you could simply discard N out of each N+1 samples.

Need help in using EGLImageKHR instead of glReadPixels

We're building a simple Android app where user can create a collage of images and video and then export it as a single video. Our current code on 16, 17 is very slow. So imagine if we have 6 videos 1 min long (each) it take us about 15min to combine them -- when they are playing one after another -- and about 7 mins when they are working simultaneously. I want to bring it to maximum of twice the total length of the video. So, for last example 2 min when playing simultaneously and 12 mins when playing one after another.
We've tried some software libraries like FFMPEG without any help. Probably we should go hardware decoding route. For which probably we should mess up with NDK. I found this article https://vec.io/posts/faster-alternatives-to-glreadpixels-and-glteximage2d-in-opengl-es that is helpful. But I need some help on it ....

audio, balance the sound from 2 sources

I am working a phone recording software (android) which record a conversation between 2 people on a phone call. The output of each phone call is an audio file of which contains the sound from both the caller and callee.
However, most of the time, the voice from the phone that this software run on is clearer than the other. Users request me to make the 2 sound equally clear.
So the problem I have now is: I have a sound file containing voices from 2 sources with different volume, what should I do make the volume of voice from those 2 sources equally regarding the noise should not be increased. Given that this is a phone call so at a specific time there is only one person speaking.
I see at least 1 straight solution for this: making a program analyzing the wave form of the sound file, identifying parts of the sound file coming from the source having smaller voice and increase it to a level seemingly balance with the another. However this will be not an easy one to implement and I also hope that there would be better solution out there. Do you have any suggestion for me?
Thank you.

Well, the first thing to do is to get rid of all of the noise that you do not care about.
The spectrum that you would want to use is: 300 Hz to 3500 Hz
You can cut all of the other frequencies which would substantially cut your noise. You can then apply an autoequalization gain profile or even tap into the DSP profiles available on several devices.
I would also take a look at this whitepaper if you have a chance. (IEEE or ACM membership required).
An Auto-Equalization System Based on DirectShow Technology and Its Application in Audio Broadcast System of Radio Station
http://ieeexplore.ieee.org/xpl/articleDetails.jsp?tp=&arnumber=5384659&contentType=Conference+Publications&searchWithin%3Dp_Authors%3A.QT.Bai+Xinyue.QT.

This is how I have solved this problem:
1. I decode the audio into a series of Integer value thank to the storing WAV format.
The result be [xi] ; 0 < xi < 255
2. Then I have to decide 2 custom value:
- Noise threshold? if xi > threshold => it is not noise (pretty naive!)
- How long should sound be a chunk of human voice?
I myself choose the first value to 5 and the second value to 100ms
3. My algorithm will analyze the [xi] in to [Yi] with each Y is an array of x and each Y represent a chunk of human sound.
After that, I apply k-mean with k=2 and got 2 different cluster of Y, one belongs to the person whose voice is louder and the other belongs to the one with softer voice.
4. What left is pretty straight forward, I have to decide a parameter M, each x belong to a Y of the softer voice will multiply with M and I get the final result.

Android display refresh rate

I'm developing an Android app which acts like a movie clapperboard/clapboard/slate. Is there any way in which I can set the display's refresh rate?
It is very important because when you edit the movie it's necessary to "land" on specific frames. The point is that if the timer is set to 25 frames per second, I need the display to update exactly 25 times per second, when the timer changes its value.
The problem on physical devices is that, let's say my Samsung Spica GT-I5700 returns a refresh rate of 62.016 which is totally inappropriate for a 25 fps timecode, and when editing you can see Frame1-Frame1-Frame2-Frame2 etc. or intermediaries, when you should see exactly Frame1-Frame2 etc.
The point is that I would need the refresh rate to be in sync with the timecode. If the user sets 25 fps, then the display should refresh exactly 25 times per second.
Any ideas, please? Thank you!

Unfortunately, there is no way to set an arbitrary refresh rate for any given device, whether it be android or not. Video circuitry is limited to the fidelity of its timers, and like all discrete systems, cannot operate with continuous precision. However, this shouldn't be too much of a problem. If you're performing a dirty render (on demand), simply base your frame build around a 25Hz system timer, and the video hardware will render it during the next raster pass - a 60Hz raster pass will almost certainly occur within your 25Hz interval. You'll also have to consider frame-dropping when the system is too busy to honour a 25Hz interval. If you're using fixed rate rendering, simply use the elapsed time between renders in order to determine the appropriate frame for a given frame rate.

how to set playback speed for android generated tones

I am using the AudioTrack class and have generated my own tones in my Android application. However, I want to be able to control the speed of playback, and I can't figure out how.
I see the setLoopPoints method but that doesn't seem to do what I want (if anyone has used it and can explain that method to me that would be great, the api documentation doesn't help me much).
What I want to do:
As a point (here, a touch on the screen) gets closer to a target on the screen, I want to increase speed of the tones I'm generating. For example, farther away I would, say, have the tone playing 1 time in 1 second, but very close to the target, 5 times in 1 second. I am struggling to find out the best way to do this with Android sounds (generated tones or even .wav files saved to my res/raw).
Any help would be much appreciated!
Shani

You want to use the setPlaybackRate method for this:
http://developer.android.com/reference/android/media/AudioTrack.html
in conjunction with setLoopPoints. However, I believe there is probably a limit to how much you can speed up the file's "natural" playback rate, and the limit is probably 48 kHz (I'm not sure, though, and it may be device-dependent).
So, if you have a file that was recorded at, say, 8000 Hz, to get the effect you want you would set the loop count to 4 (so that it plays 5 times in a row) and set the playback rate to 40,000 (5 * 8000).
Since there is (probably) an upper limit to playback rate, your best approach might be to instead record the original sound at a high frequency, and slow down the playback as necessary to achieve the effect you want.
Update: setLoopPoints lets you specify two arbitrary locations within the file, such that when playback reaches the end looppoint the audio engine will wrap back around to the start looppoint. To loop the entire file, you would set the start looppoint to 0 and the end looppoint to the last frame in the file (the size of each frame is dependent upon the file's format - so a stereo file using 2 bytes per sample would have a frame size of 4, so the last frame is just the size of the audio data in bytes divided by 4).
To get 5 consecutive plays of your file, you would set the loop count to 4 (loopcount of 0 means the file plays once; -1 means it will loop forever).
Update 2: just read the docs some more - the upper limit for setPlaybackRate is documented as twice the rate returned by getNativeOutputSampleRate, which for most devices is probably 44,100 or 48,000 Hz. This means that a standard CD-quality WAV file can only be played back at twice its normal speed. A 22,050 Hz file could be played back at up to 4 times its normal speed, etc.