Develop Reference

Difference between VOICE_COMMUNICATION and MIC in MediaRecorder.AudioSource

I am using media recorder for recording call in android using VOICE_COMMUNICATION & MIC mode alternatively.
RECORD_SOURCE = MediaRecorder.AudioSource.VOICE_COMMUNICATION;
//RECORD_SOURCE = MediaRecorder.AudioSource.MIC;
iAudioRecorder = new AudioRecord(RECORD_SOURCE, NATIVE_SAMPLE_RATE, AudioFormat.CHANNEL_IN_MONO, AudioFormat.ENCODING_PCM_16BIT, RECORD_BUF_UNIT);
I saw that MIC recorded audio data has gain much greater than VOICE_COMMUNICATION (about 2/3 times) for some devices. Also background music captured by VOICE_COMMUNICATION is not as good as MIC.
Why do audio quality (like gain, responsiveness) differs for this two recording modes?

According to the Android Developer Reference for MIC and VOICE_COMMUNICATION, some pre-processing like echo cancellation, noise suppression is applied on the audio captured using VOICE_COMMUNICATION which, in turn, causes some attenuation on the data. Moreover, on doing such pre-processing, audio signals with low amplitude becomes even more weak due to this attenuation. So we feel like background audio is sort of gone away for some devices when it is too low.
Another point was varied characteristics from device to device. From this link, it solely depends on device manufacturer and they are free to decide which pre-processing should work on which modes and how.

Android AudioTrack latency with playback

I am trying to play raw sound data using AudioTrack class in Android, I am using the write method, but I noticed that there is a latency between the write method returns and the actual sound is played, to make it simple let us use AudioRecord class as the following psedu code:
//init AudioTrack
//init AudioRecord
while(true){
byte [] buffer = new byte[1000];
int read = audioRecord(buffer,0,1000);
audioTrack.write(buffer,0,read);
}
I expect to get latency that is read / sample rate seconds but the actual sound is played after and extra of about 0.5 seconds, I really need the audio to be played with minimum latency, so does anyone has an explanation of what is going on and is there any available solution or should I accept this as it is a hardware issue?

I'm assuming your goal is to come up with some interactive audio solution (that is, where sound is played in response to some user action), because in this scenario low latency really matters.
On Android, to achieve the lowest latency you need to use Open SL ES API which is available to native (C++) code via NDK. The only Java side mechanism that can achieve low latency is SoundPool class, but it has limitations in what kind of sounds you can play.
For more information, see the page on high-performance audio, and also check out this SO answer: Low-latency audio playback on Android

Audio quality issue when recorded with MediaRecorder

The audio recorded by MediaRecorder in android App having so much noise.How can I use noise suppression to remove noice while recording.

I had the same problem of low audio quality while using MediaRecorder and finally figured out the correct working solution. Here are few modifications you need to do for good quality audio recordings:
save the file using .m4a extention.
and
mRecorder.setOutputFormat(MediaRecorder.OutputFormat.MPEG_4);
mRecorder.setAudioEncoder(MediaRecorder.OutputFormat.AMR_NB);
mRecorder.setAudioEncodingBitRate(16*44100);
mRecorder.setAudioSamplingRate(44100);
Many solutions on stackoverflow would suggest .setAudioEncodingBiteRate(16) but 16 is too low to be considered meaningless .
Source: #Grant answer on stackoverflow very poor quality of audio recorded on my droidx using MediaRecorder, why?

Are you testing this with the emulator, or on an actual device (if so, which device)? The acoustic tuning (which includes gain control, noise reduction, etc) will be specific to a given platform and product, and is not something you can change.
Jellybean includes APIs to let applications apply certain acoustic filters on recordings, and a noise suppressor is one of those. However, by using that API you're limiting your app to only function correctly on devices running Jellybean or later (and not even all of those devices might actually implement this functionality).
Another possibility would be to include a noise suppressor in your app. I think e.g. Speex includes noise supressing functionality, but it's geared towards low-bitrate speech encoding.
https://developer.android.com/reference/android/media/audiofx/NoiseSuppressor.html
android voice recording - voice with background noice issue

Voice communication at 8KHz sampling rate for all android device using OpenSL

I need to create a VOIP app and I'm using OpenSL ES. I need to capture and play pcm audio data at 8KHz sampling rate for all android devices. But, when i capture audio at sampling rate 8KHz and play it at the same time (voice communication), it produces noise and the audio is distorted for some devices like Samsung Galaxy S3, S4 etc. I know, there's a specific preferred sampling rate for each device and I want to know is there any workaround or any way to work with 8KHz sampling rate only without any distortion?
I tried increasing buffer size and many other things but failed to find an optimum and generic solution. I need audio data sampled at 8KHz for my encoder and decoder. I took re-sampling audio data before it is passed to my encoder or decoder as my second thought, but its not the solution i'm looking for.
I found CSipSimple used OpenSL and I went through some of their codes too. But, yet I couldn't find a solution and may be I failed to understand where to concentrate.
I'm stuck here!

Here's how I solved my problem:
I was working on audio streaming for Android using OpenSL ES and this tutorial helped me a lot. I followed the instructions here and got the thing working. Then i found audio streaming with this approach doesn't work very well for some devices (mostly samsung devices). I tried many things like increasing buffer size, disabling environmental reverb etc etc. I found this answer very useful for improving streaming performance.
Finally, I found the audio is distorted because of theadlocks I had to use for synchronizing the buffer switches. Using lock free structure is suggested for better audio performance. Then I went with another approach of Victor Lazzarini which is a lock free Audio IO. This article of Lock-free audio IO with OpenSL ES on Android helped a lot to implement a lock free structure along with a better audio performance.

In android, how can I boost the input from the microphone for audio recording?

I wrote an app that records audio. Everything works. However, I am going to be using this app to record class room notes. How can I boost the input of the microphone to better capture all the noise? I wouldn't mind using root if I must. But wasn't sure if there was an API to do this.
Thanks all for reading!

If you are asking how to make the microphone more sensitive, I'm not sure. That would involve either operating the microphone at a higher voltage and/or hacking the drivers, neither of which are doable programatically, AFAIK. However, you could try amplifying the output by multiplying the output by some value (say 1.1 for 10% volume boost). Of course, the more you "amplify" the output, the more you will saturate the speaker (aka distort the audio). There are some signal processing techniques you can try to remove background noise and to isolate the paticular audio of interest, however, these things are merely processing improvements, not hardware upgrades. You can always try plugging in an external microphone into the headphone jack and using that to record the audio.
I know this isn't the answer you were hoping for, but I hope it helps.