Developing Directshow source filter

STK STK 61

I have created (C++, Win10, VS2022) a simple source DirectShow filter. It’s purpose is to read audio stream from the external source (file – for testing, network – in future). It has one output pin, which I connect to soundspeaker.

In order to do it I have implemented FillBuffer method for an output pin of the filter. Media type - MEDIATYPE_Stream/MEDIASUBTYPE_PCM.
Before being connected the pin gets info about media type via SetMediaType (WAVEFORMATEX) and remembers it (wBitsPerSample; nSamplesPerSec; nChannels). Audio stream comes from the external source (file.net) to FillBuffer with (wBitsPerSample; nSamplesPerSec; nChannels)

Could you help me – which actions and calls should I make in FillBuffer() method if I will receive audio stream with changed wBitsPerSample or nSamplesPerSec or nChannels ?

The fact is that these parameters have already been agreed between my pin and the input pin of the soundspeaker and I need to change these agreement correctly.

Jeanine Zhang-MSFT 10,356 Reputation points Microsoft Vendor

2022-06-09T03:09:14.97+00:00

@STK STK

Sorry, I couldn't undenstand what do you want to achieve. What you are looking for is the CSourceStream.DoBufferProcessingLoop method? I suggest you could refer to the Doc: How to Write a Source Filter for DirectShow
STK STK 61 Reputation points

2022-06-09T19:29:05.857+00:00

JeanineZhang-MSFT. thank you for your reply.

If I understood correctly, DoBufferProcessingLoop method calls FillBuffer().
but my FillBuffer() implementation works correctly, it fills buffer in IMediaSample by audio data, then a graph passes it to a downstream filter (speakers). It works fine while the key parameters of media samples don't changed.

I would like to handle situation, when my filter while executing FillBuffer() gets data with other parameters ((): for example - wBitsPerSample old value (previous sample) was 11025, new value = 22050). Therefore, it must produce media sample in FillBuffer() with the new () parameters wBitsPerSample = 22050.

But previously my filter (upstream) and it's downstream (speakers) agreed 11025.

My question is : how in FillBuffer() to change agreement about bits per sample to 22050 ?

Are you sure, that DoBufferProcessingLoop can help me ? If it can - how ?
Jeanine Zhang-MSFT 10,356 Reputation points Microsoft Vendor

2022-06-13T08:23:58.787+00:00

@STK STK

According to the Doc: WAVEFORMATEX structure

wBitsPerSample
Bits per sample for the wFormatTag format type. If wFormatTag is WAVE_FORMAT_PCM, then wBitsPerSample should be equal to 8 or 16. For non-PCM formats, this member must be set according to the manufacturer's specification of the format tag. If wFormatTag is WAVE_FORMAT_EXTENSIBLE, this value can be any integer multiple of 8 and represents the container size, not necessarily the sample size; for example, a 20-bit sample size is in a 24-bit container.

I suggest you could try to use WAVE_FORMAT_EXTENSIBLE, and set the value of wBitsPerSample to 22056.

1 answer

STK STK 61 Reputation points

2022-06-13T12:45:28.837+00:00

@Jeanine Zhang-MSFT . I made a mistake in my example. Certainly the values 11025 or 22050 are the values of nSamplesPerSec (not bits per sample), but it does not matter which parameter of audio stream was changed - any of them could be changed. I was advised to call QueryAccept or DynamicQueryAccept, but both failed when my graph was not stopped. It seems to be the only solution - to stop the graph and reconnect the pins.
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Jeanine Zhang-MSFT 10,356 Reputation points Microsoft Vendor

2022-06-14T01:21:30.54+00:00

@STK STK

I am glad you have got your solution and thanks for your sharing, I would appreciate it if you mark them as answer and this will be beneficial to other community.
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Developing Directshow source filter

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