Help with design decision (using sysvad virtual driver as base)

Also remember that you will need one private circular buffer in each direction. The stream will need to know which direction it is going, and it’s easy to get those confused. You need to think about “am I speaker/renderer here, or am I microphone/capture here?” Each stream only worries about one of them, but your dispatcher will have access to both. You’ll also probably need a chart to remind you whether you a reading from or writing to the buffer. ReadBytes, for example, is called in the speaker/renderer path. It reads from the DMA buffer, and writes to the speaker circular buffer. Your corresponding ReadFile dispatcher, then, needs to read from the speaker/renderer circular buffer.

Thanks Tim for all those detailed information!
I was going to a completely wrong direction, I was trying to create the private buffer inside CMiniportWaveRTStream object and access it from
the dispatcher.
Now the process is a bit more clear for me. I don’t know the details
to implement this approach yet but you gave the overall idea and a good starting point. I’ll try to apply what you said and go deeper into this.
If I have more specific questions (certainly I will) I’ll post here with more details.
Thanks a lot!

Thanks for your help, I was able to copy dma buffer to the private buffer :slight_smile:
I created a private buffer with 1MB size and from ReadBytes function, I’m filling it with data until it’s completely full.
To verify if the buffer it’s OK, I’m passing it to m_SaveData.WriteData function and, from the dispatcher, I’m copying the
data to client app through DeviceIoControl function.
The data saved by m_SaveData.WriteData function is a little bigger then 1MB and I can play the wav file generated in some
players like VLC, for example. The file I’m copying to client app has exactly 1MB size and is not possible to play it.
I think the data is ok because WriteData function performs extra processing by saving frames but the data copied
directly from DMA buffer is raw, that’s why is not correctly encoded to be played, is that correct?

A megabyte is 5 seconds worth of 48k stereo data. You don’t want to introduce 5 seconds of latency in your audio. For streaming, any buffer bigger than 8k is too much.

I’m guessing you don’t understand the format of a wave file. You can’t just save a bunch of bytes to a file and pass it to a player. The file has to have a specific set of headers that identify the type of file and the exact format of the audio data. The sysvad CSaveData class has code to create those headers. It’s easy. The only tricky part is that you have to go fill in the length of the file in a couple of places when the capture is complete.

got it! Thanks!

Hi Tim, how are you doing?


I finally managed to send some audio data to sysvad microphone, but the sound
has some ditortion. Let me tell what I’m doing in details:

1 - I’m using a sample program from MS CaptureSharedEventDriven, to capture audio
using the real microphone from my system.

2 - The mix information from the real microphone is:

Num channels: 2
bits per sample: 32
samples per second: 48000
avg bytes per second: 384000
Block align: 8

3 - The MS sample captures audio and saves it in a wave file so, I needed to change it a little to not add
the wave header and only save the raw data in a file.

4 - From my client app, I’m sending the captured raw buffer to sysvad
via IOCTL.

5 - From sysvad, I needed to set the default format of the virtual mic I’m using to same as the raw format:
{
WAVE_FORMAT_EXTENSIBLE,
2,
48000,
384000,
8,
32,
sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX)
},

 STATIC_AUDIO_SIGNALPROCESSINGMODE_DEFAULT,
 &MicArray2PinSupportedDeviceFormats[SIZEOF_ARRAY(MicArray2PinSupportedDeviceFormats)-1].DataFormat

5 - From WriteBytes, I’m copying data from private buffer to dmaBuffer on this way:

  ULONG runWrite = min(ByteDisplacement, m_ulDmaBufferSize - bufferOffset);
    if (m_pExtensionData != NULL && m_pDmaBuffer != NULL) {
        ULONG ulRemainingOffset = m_ulSpaceAvailable - m_ulExtensionDataPosition;
        UINT32 frameSize = ((UINT32)m_pWfExt->Format.wBitsPerSample / 8) * m_pWfExt->Format.nChannels;
        UINT32 framesToCopy = ulRemainingOffset / frameSize;
        UINT32 bytesToCopy = min(runWrite, framesToCopy * frameSize);
        if (bytesToCopy > 0 && ulRemainingOffset >= bytesToCopy) {
            RtlCopyMemory(m_pDmaBuffer + bufferOffset, m_pExtensionData, bytesToCopy);
            m_pExtensionData += bytesToCopy;
            m_ulExtensionDataPosition += bytesToCopy;
        }
        else {
            DPF(D_TERSE, ("***No room to advance ulRemainingOffset  %lu", ulRemainingOffset));
            //Call tone generator instead copy data from audio file
            m_ToneGenerator.GenerateSine(m_pDmaBuffer + bufferOffset, runWrite);
        }
    }
    bufferOffset = (bufferOffset + runWrite) % m_ulDmaBufferSize;
    ByteDisplacement -= runWrite;

6 - After data is loaded to dmaBuffer, I run the CaptureSharedEventDriven program again and capture audio from sysvad virtual microphone.


My intention with this test is to copy captured audio data to dmaBuffer and after it reaches the end, fill it
wiht noise by calling GenerateSine function.
I could achieve the goal, partially :slight_smile: The problem is, the audio is there but it’s very distorced.
I’m not sure if the problem is in the logic to copy the captured data into the buffer, the process to generate the raw buffer or some misconfiguration caused by the changes I did on the microphone format pin.


Do you have any idea about what can be causing this distortion? Any information or documentation you have would help me a lot!


Thanks

How did you set the format? Are you dumping the format data to make sure Audio Engine has really agreed to your format? Is your wave data floating point data or integer? Since it’s 32 bits per sample, it could be either. The sine generator always generates integer data. If you want to send floats, you have to advertise that in the WAVEFORMATEXTENSIBLE, using KSDATAFORMAT_SUBTYPE_IEEE_FLOAT instead of KSDATAFORMAT_SUBTYPE_PCM.

Hi Tim,


Finally I could manage to get audio buffers from sysvad in real time and get notification on client app when other apps send data to sysvad speaker.

I’m using the following approach to capture audio data and save the audio data into a file:

From the client app:

1 - Create a thread and wait in a loop na event being signaled by the driver when a 8k buffer is completely filled with data from dmaBuffer
2 - Event signed, send a async IOCTL to get the data filled by the driver.
3 - Get the data received in a completion port thread with GetQueuedCompletionStatus function
4 - Copy buffer received into a final bigger buffer
5 - When the final buffer is totally full, save it in a wave file.


From the driver:

1 - ReadBytes calls a aux function I created, CopyBytes, to copy data from DmaBuffer to the private cyclic buffer.
2 - CopyBytes copies data from dmaBuffer until the cyclic buffer is full
3 - When cyclic buffer is full, sign the event to wake up the thread from client app and, get the data through async IOCTL
4 - If there’s any remaining bytes, copy them into the beginning of the cyclic buffer, increment the bytes copied
on buffer’s position and start the process over again.


The client app is getting buffers in the correct order but, when I join the buffers received together and save them in a wave file, there's a kind o "pop" sound between the buffers that deteriorates the final sound quality.
Strangely, I'm saving the cyclic buffer from driver's side by using the m_SaveData.WriteData() function and the sound it's OK.
Do you have any suggestion about what can be causing this problem?
Thanks,

The “pop” indicates that the data is not contiguous. I’m concerned about your “from the driver” steps there. You said you only signal the app when the cyclic buffer is full, but if there’s something remaining, you copy it into the beginning of the cyclic buffer. Doesn’t that mean you’ve now destroyed early bytes in that buffer that the application hasn’t read yet? The purpose of the circular buffer is to allow the river to continue to accumulate data until the app can pull its chunk out. Perhaps you should signal the app when the buffer is half-full or 3/4-full, so you have room to add additional data.

Thanks for your help, worked perfectly!

Hello gentlemen,

We are using Sysvad example to implement our noise remover.
We made a POC where a user program receives the Sysvad speaker stream and save in a file (insted of the file being saved by the Sysvad itself).
This is working fine.

However we are facing some difficulties to implement the microphone part. We tried to use the same logic as speaker but the result is not working as it should.

What we are doing initially is:

1 - the application sends a 800k buffer to the Sysvad (the WAV file data with 1 channel, 16 bits, 48000hz which is the same as the Sysvad microphone)
2 - in the WriteBytes method we use the ByteDisplacement as the number of bytes we copy from the buffer received to Sysvad m_pDmaBuffer
3 - start the Voice Recorder and record the audio received from the Sysvad microphone

But the recorded sound has poor sound quality, is full of gagging and glitches and at a certain moment, it looks like the noise of a modem connection.
Is there anything else we need to control?

Our WriteBytes implementation:

VOID CMiniportWaveRTStream::WriteBytes
(
    _In_ ULONG ByteDisplacement
)
/*++

Routine Description:
  This function writes the audio buffer using a received buffer from user application

Arguments: 
  ByteDisplacement - # of bytes to process.

--*/
{
	ULONG bufferOffset = m_ullLinearPosition % m_ulDmaBufferSize;
	ULONG runWrite = 0;

	if (m_pExtensionData != NULL && m_pDmaBuffer != NULL) {

		// Is a pointer to buffer's start that persists over each iteration
		// and maps the audio starting point to be sent to the system
		BYTE* sendIni = m_pExtensionData->MicBufferSendIni;
		
		// Is a pointer to buffer's end that persists over each iteration
		// and maps the audio ending point to be sent to the system
		BYTE* sendEnd = m_pExtensionData->MicBufferSendEnd;

		// maps how many bytes left in the buffer to be sent
		runWrite = (ULONG)(sendEnd - sendIni);

		while (ByteDisplacement > 0)
		{
			// ensure we never read beyond our buffer
			LONG minWrite = min(ByteDisplacement, min(m_ulDmaBufferSize - bufferOffset, runWrite));
			
			if (minWrite <= 0) {
				break;
			}

			RtlCopyMemory(m_pDmaBuffer + bufferOffset, sendIni, minWrite);
			bufferOffset = (bufferOffset + minWrite) % m_ulDmaBufferSize;
			ByteDisplacement -= minWrite;
			runWrite -= minWrite;
			
			// slide the pointer to the next piece of buffer to be sent
			sendIni += minWrite;
		}

		// slide the initial position to be sent in the next run 
		m_pExtensionData->MicBufferSendIni = sendIni;
	}
}

Here is the MJ function part that receive the buffer from the application:

{

.....

		// Reading client app input/output buffer information 
		systemBuffer = (BYTE*)_Irp->AssociatedIrp.SystemBuffer;
		
		// the buffer size informed by the application
		inputBufferLength = stack->Parameters.DeviceIoControl.InputBufferLength;

		//Get pointer to the device extension reserved to store our data (copy of cyclic buffers) 
		ExtensionData = (PNOISE_DATA_STRUCTURE)((PCHAR)_DeviceObject->DeviceExtension + PORT_CLASS_DEVICE_EXTENSION_SIZE);

		if (inputBufferLength > 0 && ExtensionData != NULL && systemBuffer != NULL) {
			DPF(D_TERSE, ("***Input received of %lu bytes", inputBufferLength));
			
			// copy the received buffer to our structure
			RtlCopyMemory(ExtensionData->MicBuffer, systemBuffer, inputBufferLength);
			
			// set the relative buffer end (is less or equal buffer size on our example)
			ExtensionData->MicBufferEnd += inputBufferLength;
			
			// maps the buffer's end position to be read on the WriteBytes function
			ExtensionData->MicBufferSendEnd = ExtensionData->MicBufferEnd;
			
			bufLen = inputBufferLength;
		}

		_Irp->IoStatus.Information = bufLen;
		_Irp->IoStatus.Status = STATUS_SUCCESS;
		IoCompleteRequest(_Irp, IO_NO_INCREMENT);

		return STATUS_SUCCESS;
		break;

.....
}

Is there any other thing we need to take care?

Thanks in advance!
André Fellows

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@andrefellows what is PNOISE_DATA_STRUCTURE

That is his device context structure, custom to his driver. It has all the data he needs to keep track of to do his work.

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@Tim_Roberts How can i create it.Is the structure the same as _PortClassDeviceContext. I have seen above and you call the GetDeviceContext but where is GetDeviceContext function

Do you have any driver experience at all? These are very fundamental questions. EVERY driver has a context structure that holds all of the data for each device instance. In the case of a port class driver, things are more complicated because the port (from Microsoft) and the miniport (provided by you) act as one device and share one context. Port class creates the context, but it lets you tack on extra space for your own use, in PcAddAdapterDevice.

The device context is stored in the DEVICE_OBJECT in the DeviceExtension field. In the case of a port class driver, your context starts after the port class section, and we know that part is PORT_CLASS_DEVICE_EXTENSION_SIZE bytes long. So, this line from above finds his part of the extension:

ExtensionData = (PNOISE_DATA_STRUCTURE)((PCHAR)_DeviceObject->DeviceExtension + PORT_CLASS_DEVICE_EXTENSION_SIZE);

In my port class drivers, I create a function called GetDeviceContext to do exactly that so I don’t have to type that repeatedly.

1 Like

@Tim_Roberts.I have no previous driver knowledge.Sorry for my bad English. In school i am not learning about this. I’m groping for it. As a beginner i am trying to follow the available examples to better understand this problem. Sorry for bothering you. Where can I see more examples.Is there any other way to send audio data to sysvad and write it to writebytes without using ioctl does i get an advice to use this

…to better understand this problem.

What problem? You haven’t told us anything about what goal you’re trying to achieve.

… send audio data to sysvad and write it to writebytes without using ioctl …

The link you included shows how to send data to a speaker endpoint. WriteBytes is used to manage data for the microphone endpoint. Totally separate paths. You need to think about what you have. Sysvad is a fake speaker that writes the speaker data to file, and a fake microphone that generates a sine wave. That’s what it does. To do anything else, you have to write the code to do it, and that means inventing some kind of “back door” to get data in and out.

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@Tim_Roberts I have read many of your answers to make the audio transition from user mode to application (ex: skype), I am actually copying the exact same code contained in the related questions and I don’t understand Concepts or constructs, how to implement it into code as I asked above. Can you please give me some sample projects. Thank you for the answer.

You can see from the rather good chart above that there a lot of pieces to this, and they all have to work together. It’s complicated, and it has to run in real-time. If you don’t have experience writing audio applications AND experience writing drivers, then you will never make this work. Sorry to be blunt. Even the big companies hire people to do this kind of thing.

There are no samples. Because so many people want to do this, I’ve suggested for the last 15 years that the Microsoft audio team create a much simplified version of SysVad that has external hooks to circular buffers, but so far they’ve been busy doing real work.

You need to put circular buffers in SysVad. You need to add ioctls that allow you pull data in and out. You need to write an application to do the “in and out” by calling those ioctls. You need to write a test application to take the place of Skype by reading and writing using the WASAPI APIs. You need to decide how to handle volume and mute controls. You need to figure out how much of SysVad you can delete. None of those pieces are easy.

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@Tim_Roberts Thanks sir for advice.