B4A Library Audio library v1.5 - New AudioStreamer object

This version includes a new AudioStreamer type. Note that this is a beta version (for this type only).

The purpose of AudioStreamer is to make it simple to stream audio from the microphone and to the speakers. Internally it is based on AudioTrack and AudioRecord.

Next week I plan to create an example of streaming audio between two devices (walkie-talkie app).

Recording is done by calling StartRecording. The RecordBuffer event will be raised multiple times during the recording:

Sub streamer_RecordBuffer (Buffer() As Byte)
   'collect the recording data
   buffers.Add(Buffer)
End Sub

You can write this data to a file or as done in this example, collect the buffers in a list. Later we will use this list to play the recording:

Sub btnPlay_Click
   btnStartRecording.Enabled = False
   streamer.StartPlaying
   For Each b() As Byte In buffers
      streamer.Write(b)
   Next
   streamer.Write(Null) 'when this "message" will be processed, the player will stop.
End Sub

Note that the player has an internal queue. The write method will not block the main thread. The data is just added to the queue. It is processed by another thread.

 

[NJDude]

Have you refreshed the libraries? (Right click on the Libs panel and click REFRESH) or close and reopen B4A.

 

[js1234]

Yes! thanks!! All is ok!

 

[Erel]

This is strange. The Audio library is included in the installation file. It should be there.

Where is Basic4android installed?

 

[js1234]

you are right but not version 1.5 !
Only version (I think) have AudioStreamer reference!
or?

Can I with this library Play stereo? I want synth some raw data and make stereo sound.....and put to audio output! (headspeakers left and right chanel)!
(plaing cca 4s) ...... then after cca 2s record cca 1s long period "music" (with microphone)!?

 

[Erel]

You can play stereo with AudioStreamer.

 

[birdwes]

Wish:

It would be really useful if you could specify as smaller output buffer size for realtime apps. 22050 Hz sample rate outputs 21.5(ish) 1024 16 bit buffer word chunks per second on a Galaxy S3 which does not look slick on the UI, even though it is processing as fast as the data comes in (no lost buffers). I have to collect the "mini buffers" and put them in an 8Kb circular buffer for my purpose. I need to update the screen more frequently.

I am using agraham's thread library for backgroung processing (Like very much).

 

[Erel]

There is no good reason to use the Threading library with AudioStreamer. AudioStreamer handles the buffers with two background threads.

I'm not sure that I understand the request. The array size can be smaller than PlayerBufferSize.

 

[birdwes]

Maybe I didn't get that across clearly. I'm doing lots of intense floating point calculations in the thread that does the math. If I do the math in the GUI thread, it's very slow, won't run realtime and makes the view animations jerky.

Back to my main question though; PlayerBufferSize is read-only - I understand and have noticed that sometimes the buffer is smaller anyway, but what I'm wondering is, if it would be possible to set the streamer to intentionally send smaller buffers of some specified desired size (obviously a power of 2) into the streamer record event.

The S3 and S4 send 2048 bytes (or less when you finally shut down the stream). The Allwinner A13 tablet (very cheap test device) sends 1920 byes. Could there be a way in future say to request buffers smaller e.g. if I want 1024 default bytes per buffer instead of 2048. That way, I can pump the data to the other thread even closer to realtime.

Thanks for all you work so far though! B4A is way quicker and easier than developing using native Java.

 

[Erel]

The value is: AudioRecord.getMinBufferSize(SampleRate,channelConfig, audioFormat)

This is the minimum possible size.

 

[birdwes]

What I'm hoping for is AudioStreamer.setMaxBufferSize(size).

 

[Erel]

There is no such native API.

 

[tigrot]

Hi Erel,
I have written a two ways, full duplex audio communicator for Android using Audio 1.5. We tested it on some device and we found the sound clear and very "real". We ranged from simple Samsung pocket to Samsung Galaxy III and we got no issue. Yesterday I was showing to a spanish customer and after some time of bad audio the APP crashed for three times, always in the same way. Did you have any issue with Note(quad-core CPU)?

Thank for your nice support.

Mauro

 

[Erel]

I haven't tested it on Galaxy Note. Which error do you get?

 

[ComposerB4A]

Hi to all, one question please (because i want access to internal data of arrays), ¿what is the format of data in array(s) "b()"?
***code**
*

...
ForEach b() AsByte In buffers
streamer.Write(b)Next

...

** comments ***
i see in array b() in first cycle (approx.)
"-80,2,-100,2,-120,2,-127,2,-110,2,123,2,122..."
after approx. 1000 or 2000 datas example approx.
"72/1/50/1/20/1/..."



with max value for array 4096 datas
¿what format of data indicates "volume or frecuency or time"?

 

[Erel]

Each item in this element is a Byte (value between -128 to 127).

 

[ComposerB4A]

Thanks, but i referring to the meaning of the number
But the moment i found the volume in other library

 

[birdwes]

Thanks, but i referring to the meaning of the number
But the moment i found the volume in other library

To convert the byte buffer to signed ints do something like this (for 16 bit):

j = buffer.Length / 2

For i = 0 To j - 1
s = Bit.AND ( buffer( 2 * i ), 255)
t = Bit.AND ( buffer( 2 * i + 1), 255)
filterBuffer(i) = ( s + Bit.ShiftLeft(t, 8) + Bit.AND(t,128) * -512 )
Next

 

[ComposerB4A]

Thanks birdwes, one wuestion in "filterBuffer", the values are "volumne" or "frecuency" of sounds??

 

[birdwes]

You're welcome,

my destination filterbuffer (double() in my case) is the raw waveform of amplitude (i.e. +ve and -ve). If you want the frequency you'll need to use the FFT library or an alternative algorithm. If you use FFT, then this link may be of interest: http://cds.cern.ch/record/738182/files/ab-2004-023 . See equation 15 on page 6.

Be aware that different devices put out different byte array sizes. I'm sampling at 22050 Hz 16 bit. The galaxy s3 puts out 2048 byte at a time. The Allwinner A13 puts out 1920 bytes at a time, so you'll have to place your data into a circular buffer first if you want a consistent FFT window size of say e.g. 4096 samples. Then copy from the circular buffer into an array of double as you apply your window function, before you call the FF transform.

 

[ComposerB4A]

Thanks you very much birdwes, excellent all information