I want to measure the RMS value of the input audio signal over periods of 100-200 ms on an Android device without recording the audio to a file. This is not to measure the noise level, I only need the RMS value of the ADC. Please suggest an appropriate approach. Should I look into the AudioRecord class, the Oboe library or something completely different? I am an amateur with very basic Android/Kotlin knowledge. Thank you!)
I tried to find examples of Kotlin code regarding recording audio from a microphone to a byte array, but I did not succeed.
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I’m struggling since days trying to obtain a raw audio stream from the microphone. I am trying different ways: the low-level JNI way with Oboe Library (either AAudio and OpenSL ES implementations) and the Android’s AudioRecord Java classes.
The problem I am facing is that I am not able to retrieve amplitudes near -/+1.0 while being sure of saturating the microphone input with a calibrated pure tone with such a high amplitude.
I think that the problem is that I am not able to effectively disable the signal preprocessing from AndroidOS (Automatic Gain Control or Noise Cancelling).
AutomaticGainControl.create(id).setEnabled(false)
(not working!)
Also, it seems that it is not possible also to disable any additional microphone rather than the one "selected" (done that as selecting the setPreferredDevice on AudioRecord instance). Used as audio source: unprocessed, mic, voice_recognition.
Is there anyway doing this or am I missing something?
Thank you
Which audio source are you using for your recording? VOICE_RECOGNITION or UNPROCESSED are mandated to not have any pre-processing enabled by default (i.e. see https://source.android.com/compatibility/10/android-10-cdd#5_11_capture_for_unprocessed) and therefore would allow you to check your signal path.
I need to record 10 seconds of audio and than to perform convolution with some other signal. I need to record audio with sampling rate of 512hz. As my phone(i guess it's hard that any phone supports sample rate of 512hz) doesn't support that sampling rate, i need to record audio in higher sampling rate and than downsample to 512hz. For recording audio I use AudioRecord and the only frequency that is guaranteed to work is 44100hz. Every lib or code that i found perform downsampling by reading and writing to file. As i need it to be very fast, i need to perform this action couple times in one second(at least two), is any way to perform dowsampling on raw PCM data written in byte array and to be very fast?
Any reason you are aiming for a sample rate of 512hz? Seems a bit low to be useful!
I am not sure what language you are using but I use libsoxr within C++. A quick google comes up with libresample4j for java. Both these will let you do the resampling in real time, without have to save a file first.
I am developing a sound related application. I am trying to change the audio sound in to completely different like robot sound or make the audio echo. I tried with soundpool , but no any idea, anyone knows how to achieve that? i need only a basic idea to achieve this, please help. many thanks.
Pitch and echo are 2 different things.
Pitch:
You can alter the pitch by modifing the playback rate. You can do it in 2 ways, with audioTrack and setPlayBackRate or with SoundPool and setRate. Depends on your needs, AudioTrack allow a larger range of pitch (from 1hz to x2) on large files and SoundPool for sound effects and picth can vary between x0.5 and x2.
Echo/reverb:
You can archive this with AudioEffect since API lvl 9 by attaching it to an AudioTrack or MediaPlayer instance.
For a robot effect you want to set a constant pitch for the audio. I.e. do a FFT, move everything into a single frequency bin, and then do an inverse FFT to get back into the time domain.
For an echo effect you could keep a separate buffer which is as long as your desired echo delay. And for every sample do something like the following (pseudo-code):
output = mix(currentSample, echoBuffer[echoPos]*echoVolume)
echoBuffer[echoPos] = mix(currentSample, echoBuffer[echoPos]*echofeedback)
echoPos += 1
Im working on a similar project and i can say that you need to look into DSP (digital signal processing), PCM 16 format and preferably fourier transforms.
It is possible to loopback audio with the AudioRecord class (running a thread constantly filling the buffer on a AudioTrack)
But the delay might be too big for what you are trying to accomplish.
Best of luck in your endevours!
Some really good pointers:
Android AudioRecord class - process live mic audio quickly, set up callback function
I want to develop an android app that takes in the audio signal and generates the time and frequency domain values.
I have the audio in a buffer which is obtained from the android MIC. I want to use this buffer values to generate a frequency domain graph. Is there some code that would help me find the FFT of an audio signal??
I have seen Moonblink's Audalyzer code and there are some missing components in the code. I could find lots of missing modules. I want to find a better logic that would take in audio and perform some calculations on it.
I found these for you using duckduckgo:
Android audio FFT to retrieve specific frequency magnitude using audiorecord
http://www.digiphd.com/android-java-reconstruction-fast-fourier-transform-real-signal-libgdx-fft/
This should help
I am new to android and presently doing android voice recording application. I want top know which format is best for saving audio file in android. (i.e RAW-AMR or 3gp or mp4).So rhat we can hear playback sound loudly in device.
Is there any alternative way to increase audio sound through voice processing in android.
Thanks in advance.
Question: Which bear is best? Answer: Black Bear
Seriously though, you would need to state your criteria for the audio file for us to make a codec recommendation. Does it need to be portable? Best compression? Highest fidelity?
The codec that you choose has no affect on the loudness of audio that will be played over the device, so this should not factor into your criteria.
Is there an alternative way to increase audio?
Yes, if you are recording audio from the microphone then you can amplify the audio data before you save it to a file.
Let an audio sample from the microphone be represented by the function:
f(t)
Amplification is achieved by multiplying the audio sample by some factor A
A * f(t)
You can use AGC(Automatic Gain Control) module from WebRTC to increase sound level.
I didn't find any simple Java API yet. You can use C++ API via JNI.
Have a look here, WebRTC AGC (Automatic Gain Control) .
I want top know which format is best for saving audio file in android.
To save voice audio on Android (or any other platform), take a look at Opus. It's a free, state-of-the-art audio codec that also supports voice mode.