[wdmaudiodev] Re: buffering samples in an APO
- From: Matthew van Eerde <Matthew.van.Eerde@xxxxxxxxxxxxx>
- To: "wdmaudiodev@xxxxxxxxxxxxx" <wdmaudiodev@xxxxxxxxxxxxx>
- Date: Wed, 17 Dec 2014 19:10:27 +0000
Hmm... to err on the side of caution, let me make sure I understand correctly.
The following is a naïve approach:
APOProcess call
You are handed
You do
You output
1
448 samples
Append 448 samples to incoming queue
Incoming queue now has 448 samples
Outgoing queue is empty
448 samples of silence
2
448 samples
Append 448 samples to incoming queue
Process 768 samples
Output 448 samples
Incoming queue now has 128 samples
Outgoing queue now has 320 samples
448 samples of data
3
448 samples
Append 448 samples to incoming queue
Incoming queue now has 576 samples
Outgoing queue now has 320 samples
448 samples of silence
4
448 samples
Append 448 samples to incoming queue
Process 768 samples
Output 448 samples
Incoming queue now has 256 samples
Outgoing queue now has 640 samples
448 samples of data
This approach won't work; you're alternating between outputting data and
silence, which will cause glitchiness. Instead, adopt a fixed delay equal to
your internal processing buffer size by pre-filling your output queue with 768
samples of silence.
APOProcess call
You are handed
You do
You output
1
448 samples
Append 448 samples to incoming queue
Incoming queue now has 448 samples
Output 448 samples of silence
Outgoing queue now has 320 samples of silence
448 samples of silence
2
448 samples
Append 448 samples to incoming queue
Process 768 samples
Output 448 samples
320 of silence
128 of processed data
Incoming queue now has 128 samples
Outgoing queue now has 640 samples
320 samples of silence
128 of processed data
3
448 samples
Append 448 samples to incoming queue
Incoming queue now has 576 samples
Output 448 samples
Outgoing queue now has 192 samples
448 samples of processed data
4
448 samples
Append 448 samples to incoming queue
Process 768 samples
Output 448 samples
Incoming queue now has 256 samples
Outgoing queue now has 512 samples
448 samples of processed data
Eventually you will get a final APOProcess call (though you won't know it at
the time.) Your incoming queue and your outgoing queue will still have data in
them. That's fine; the APO is under no obligation to flush its internal queues,
or to let Windows know what the state of its internal queues are, other than by
implementing IAudioProcessingObject::GetLatency().
Since your latency is 768 samples, your GetLatency implementation should return
1.0 * 768 * HNS_PER_SECOND / WAVEFORMATEX.nSamplesPerSec; for 44.1 data, this
is 174150 hundred-nanoseconds, or 17.4 ms.
From: wdmaudiodev-bounce@xxxxxxxxxxxxx
[mailto:wdmaudiodev-bounce@xxxxxxxxxxxxx] On Behalf Of Abhinav Singh
Sent: Wednesday, December 17, 2014 10:41 AM
To: wdmaudiodev@xxxxxxxxxxxxx
Subject: [wdmaudiodev] Re: buffering samples in an APO
I totally forgot to implement GetLatency(..)..will add it to the APO.
However, my problem is slightly different. I have tried to explain it clearer
with an example below:
* I am maintaining my own internal input and output buffer(separate from
the FLOAT32 buffer point passed to the APOProcess(..) function.
* At 44.1 kHz, I will get 448 samples per call to APOProcess().
* However, since my DSP algorithm only works with say exactly 768 samples,
i will internally buffer the input received from APOProcess(...) until i have
enough samples to process. I am also maintaining an internal output buffer
where i will store the processed audio samples.
* I process audio samples when the samples count in my internal buffer
exceeds 767 and store the processed audio stream in my internal output buffer.
* If i have enough samples in my internal output buffer(>=448),i will copy
to APOProcess(..)'s output buffer otherwise i just set the first 448 samples in
the APOProcess(..)'s output buffer with zero.
* So the very first call to APOProces(...), i will just buffer the input
and send zeros as the output.
* However during the second call to APOProcess(..) i will have enough data
to process and also send valid output(not zeros).
* Now, at the very end , my own internal input and output buffer might have
some samples leftover and i was wondering if there are any workaround around
this buffering problem without changing the DSP algorithm.
________________________________
From: soccerl@xxxxxxxxxxxxx<mailto:soccerl@xxxxxxxxxxxxx>
To: wdmaudiodev@xxxxxxxxxxxxx<mailto:wdmaudiodev@xxxxxxxxxxxxx>
Subject: [wdmaudiodev] Re: buffering samples in an APO
Date: Tue, 16 Dec 2014 00:13:35 +0000
Is this a 100% repro? Which OS version is it?
An APO has no idea on when the stream will end. If an audio application stops
its streaming before the end of stream, I am not sure whether APO could do
anything with this situation.
From: wdmaudiodev-bounce@xxxxxxxxxxxxx<mailto:wdmaudiodev-bounce@xxxxxxxxxxxxx>
[mailto:wdmaudiodev-bounce@xxxxxxxxxxxxx] On Behalf Of Matthew van Eerde
Sent: Monday, December 15, 2014 10:23 AM
To: wdmaudiodev@xxxxxxxxxxxxx<mailto:wdmaudiodev@xxxxxxxxxxxxx>
Subject: [wdmaudiodev] Re: buffering samples in an APO
Make sure you implement IAudioProcessingObject::GetLatency(...), and report the
delay between your input and output.
If an app cares about its sound playing all the way through to the very last
byte, the design is for the IAudioClient to pad silence onto the end of the
stream, and keep pumping silence until IAudioClock indicates that the very last
byte has made it all the way through the speaker.
From: wdmaudiodev-bounce@xxxxxxxxxxxxx<mailto:wdmaudiodev-bounce@xxxxxxxxxxxxx>
[mailto:wdmaudiodev-bounce@xxxxxxxxxxxxx] On Behalf Of Abhinav Singh
Sent: Monday, December 15, 2014 10:12 AM
To: wdmaudiodev@xxxxxxxxxxxxx<mailto:wdmaudiodev@xxxxxxxxxxxxx>
Subject: [wdmaudiodev] buffering samples in an APO
I am trying to integrate a DSP algorithm with my GFX APO which requires a fixed
number of samples greater than the number of samples i get every time the Audio
Engine calls the APOProcess() function of my APO. So i buffer the samples until
i have the required number of samples and feed the audio engine zeros until i
have valid output to feed to the engine. This works but there is a loss of few
samples at the very end(some samples remain unprocessed in the input buffer
while some are left over in the output buffer ). Is there any workaround to
this problem?
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