How to see the selected hardware audio device

[MidimasterSoft]

Hi, I'm working on the Binding for BlitzMax and now I added the context-feature to select hardware devices. I already can do the enumeration and also understand, that i have to send the STRUCT pPlaybackInfos[ID_Playback].id received from the context to switch between devices. but where can I see, which device is really selected. Or which is "default"?

[mackron]

Use these members from the ma_device object:

device.playback.id
device.playback.name
device.capture.id
device.capture.name

If you used a specific device ID to initialize the device, that's what you'll get, or you'll get an error when you try to initialize the device. When you pass in NULL for the device ID, you should get the device that has it's isDefault member of ma_device_info set to true during device initialization.

[MidimasterSoft]

thank you for the help. I did not see this in the source code. Now I could continue. I was able to connect a ZOOM R16 USB AUDIO DEVICE to the windows pc and was able to playback and capture via MiniAudio.

But this is only working as long as I select MONO for capture. When I try to select 2 or more channels I get a bad noise on the duplex device (computer speaker)

EDIT
I had a progress... It looks like the device is able to process 1 channel or all 8. When I use 8 capture-channels with FORMAT_S32 and 48kHz and only one or two playback channels the noise disappears and I can hear the microphones connected to the interface.

This was the report, when the noise was still there. I leave it here, although it works now:

length of  context 544  
: Attempting to initialize WASAPI backend...
[miniaudio] Endian:  LE
[miniaudio] SSE2:    YES
[miniaudio] AVX2:    NO
[miniaudio] AVX512F: NO
[miniaudio] NEON:    NO
list of playback devices: 
0 - 1 - 32W_LCD_TV (AMD High Definition Audio Device)  
1 - Realtek Digital Output (Realtek High Definition Audio)  
2 - Line (ZOOM R16_R24 Audio Interface)  
list of capture devices: 
0 - Stereo Mix (Realtek High Definition Audio)
1 - Line (ZOOM R16_R24 Audio Interface)
** end of context ************************* 
selected ID: P=0 C=1
START_DEVICE_GLUE 
selected playback device 0 1 - 32W_LCD_TV (AMD High Definition Audio Device) 
------ 
FILL_VALUE_B 
------ 
CREATE A DEVICE 
[WASAPI] Trying IAudioClient3_InitializeSharedAudioStream(actualPeriodInFrames=480)
    defaultPeriodInFrames=480
    fundamentalPeriodInFrames=480
    minPeriodInFrames=480
    maxPeriodInFrames=480
[WASAPI] Using IAudioClient3
    periodSizeInFramesOut=480
INFO: [WASAPI]
INFO:   Line (ZOOM R16_R24 Audio Interface) (Capture)
INFO:     Format:      32-bit IEEE Floating Point -> 16-bit Signed Integer
INFO:     Channels:    8 -> 2
INFO:     Sample Rate: 48000 -> 48000
INFO:     Buffer Size: 480*3 (1440)
INFO:     Conversion:
INFO:       Pre Format Conversion:    YES
INFO:       Post Format Conversion:   NO
INFO:       Channel Routing:          YES
INFO:       Resampling:               NO
INFO:       Passthrough:              NO
INFO:   1 - 32W_LCD_TV (AMD High Definition Audio Device) (Playback)
INFO:     Format:      16-bit Signed Integer -> 32-bit IEEE Floating Point
INFO:     Channels:    2 -> 2
INFO:     Sample Rate: 48000 -> 48000
INFO:     Buffer Size: 480*3 (1440)
INFO:     Conversion:
INFO:       Pre Format Conversion:    NO
INFO:       Post Format Conversion:   YES
INFO:       Channel Routing:          NO
INFO:       Resampling:               NO
INFO:       Passthrough:              NO
------ 
RECORD
....
[WASAPI] Data discontinuity (possible overrun). framesAvailableCapture=929, mappedBufferSizeInFramesCapture=449
PLAY

(the error message and the end appears always when I stop the device, think this is normal...)

[mackron]

Can you post your device initialization code? It looks like your formats are inconsistent. If you're using a duplex device it's best to keep the formats the same to avoid the need for data conversion.

[MidimasterSoft]

As I wrote... now it works At first I thought it had perhaps to do with restictions of channels.. So I switched to FORMAT_S32 and 8 IN (USB) , but 2 OUT (speakers) . And suddenly it worked. I think it had to do with my own forwarding of the samples in the Callback()

Now it looks like I'm able to also use 3-8 OUT without any problems. Also FORMAT_S16 seems to be possible, but with a poor quality. I did not expect, that it is more noise compared to the use of this interface in a 16bit studio- equippment.

I will continue with more experiments. And I need to add the WAV-saving finally. Will the 8 tracks still be independent in the WAV file? That is what i have to find out next.

this is what I'm using now:
callback in BlitzMax:

MiniAudio.GetDevice_II( MiniAudio.DUPLEX, Miniaudio.FORMAT_S32, 2, Miniaudio.FORMAT_S32, 8, HERTZ, MyCallBack)

//void MM_SetDeviceConfig (struct ma_device_config *config, int playformat, int playchannels, int captformat, int captchannels, int hertz, ma_device_callback_proc Callback) {

Function MyCallBack(a%, PlayBackBuffer:Int Ptr, RecordingBuffer:Int Ptr, Frames%)
	For Local i%=0 To frames-1
		Local valueL%=RecordingBuffer[8*i]+RecordingBuffer[8*i+1]+RecordingBuffer[8*i+2]+RecordingBuffer[8*i+3]
		Local valueR%=RecordingBuffer[8*i+7]+RecordingBuffer[8*i+7]'+RecordingBuffer[8*i+6]+RecordingBuffer[8*i+7]
		PlayBackBuffer[2*i]= valueR
		PlayBackBuffer[2*i+1]=valueR
               ' needed for transfering to 16bit blitzmax-world:
		SoundBank.PokeShort(WritePointer+2*i , valueR/65536)
	Next
    WritePointer = (WritePointer + 2*Frames) Mod (HERTZ*60)
End Function 

MM CREATE CONTEXT 
length of  context 544  
: Attempting to initialize WASAPI backend...
[miniaudio] Endian:  LE
[miniaudio] SSE2:    YES
[miniaudio] AVX2:    NO
[miniaudio] AVX512F: NO
[miniaudio] NEON:    NO
list of playback devices: 
0 - 1 - 32W_LCD_TV (AMD High Definition Audio Device)  
1 - Realtek Digital Output (Realtek High Definition Audio)  
2 - Line (ZOOM R16_R24 Audio Interface)  
list of capture devices: 
0 - Stereo Mix (Realtek High Definition Audio)
1 - Line (ZOOM R16_R24 Audio Interface)
** end of context ************************* 
selected ID: P=0 C=1
START_DEVICE_GLUE 
selected playback device 0 1 - 32W_LCD_TV (AMD High Definition Audio Device) 
------ 
FILL_VALUE_B 
------ 
CREATE A DEVICE 
[WASAPI] Trying IAudioClient3_InitializeSharedAudioStream(actualPeriodInFrames=480)
    defaultPeriodInFrames=480
    fundamentalPeriodInFrames=480
    minPeriodInFrames=480
    maxPeriodInFrames=480
[WASAPI] Using IAudioClient3
    periodSizeInFramesOut=480
INFO: [WASAPI]
INFO:   Line (ZOOM R16_R24 Audio Interface) (Capture)
INFO:     Format:      32-bit IEEE Floating Point -> 32-bit Signed Integer
INFO:     Channels:    8 -> 8
INFO:     Sample Rate: 48000 -> 48000
INFO:     Buffer Size: 480*3 (1440)
INFO:     Conversion:
INFO:       Pre Format Conversion:    NO
INFO:       Post Format Conversion:   YES
INFO:       Channel Routing:          NO
INFO:       Resampling:               NO
INFO:       Passthrough:              NO
INFO:   1 - 32W_LCD_TV (AMD High Definition Audio Device) (Playback)
INFO:     Format:      32-bit Signed Integer -> 32-bit IEEE Floating Point
INFO:     Channels:    2 -> 2
INFO:     Sample Rate: 48000 -> 48000
INFO:     Buffer Size: 480*3 (1440)
INFO:     Conversion:
INFO:       Pre Format Conversion:    NO
INFO:       Post Format Conversion:   YES
INFO:       Channel Routing:          NO
INFO:       Resampling:               NO
INFO:       Passthrough:              NO
------ 
RECORD

[mackron]

I've just realised that I've misread your debug log. Your formats are correct. It's strange that 16 bit is poor quality. It should be barely perceptible. Ideally it would be good to just use FORMAT_S16 rather than having to do your own conversion.

Is this code here used for converting from 8 channels to 2?

Local valueL%=RecordingBuffer[8*i]+RecordingBuffer[8*i+1]+RecordingBuffer[8*i+2]+RecordingBuffer[8*i+3]
Local valueR%=RecordingBuffer[8*i+7]+RecordingBuffer[8*i+7]'+RecordingBuffer[8*i+6]+RecordingBuffer[8*i+7]

If you try using FORMAT_S16 and RecordingBuffer is a buffer of 16-bit integers (which it would need to be), that math won't work - you'll need to accumulate as 32-bit, then clamp it to [-32767, +32768] before writing it to the output buffer. The reason is that your summing will overflow a 16-bit value and result in wrapping and bad glitching. Example in C for how you should accumulate 16-bit samples:

int32 valueL = (int32)RecordingBuffer[0] + (int32)RecordingBuffer[1] + (int32)RecordingBuffer[2] + (int32)RecordingBuffer[3]
...
PlayBackBuffer[2*i] = (short)ma_clamp(valueL, -32767, 32768);

That could be the cause of your quality issues, but I'm not 100% sure.

WAV should support all 8 channels no problem.

[MidimasterSoft]

dont worry.. in this example code the buffer was 32bit, you can see it, because the buffer is defined INT

Function MyCallBack(a%, PlayBackBuffer:Int Ptr, RecordingBuffer:Int Ptr, Frames%)

so I added 32bit integers. If I need to bring it back to the blitzMax's 16bit world i devided it:

		SoundBank.PokeShort(WritePointer+2*i , valueR/65536)

And of cource I know that i have to care about Clipping and distortion (with real audio) when adding 8 values to keep below -3dB.

by the way... is the 32bit integer audio format defined as SHORT*65536 when converting? This would mean again no headroom? In the 24bit studio world we try to use ony 21bit and stay -16dB away from clipping

[mackron]

Yes, you're right - there wouldn't be much headroom for 32-bit either. I don't have an answer for your quality issues.