Thanks for sharing your experience.

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• The device/driver sends per urb a constant number of packets (as you

already also discovered). The number of samples per packet differ however (+- 1 sample per packet). I accuse the asynchronousity between the urb interval and the sound card's internal clock for this fact. The device expects the driver to send back an urb with exactly the same structure. The device produced crackling noise if the samples per packet were not equal. I cannot say to what extend this is true (i.e. if you could wait some packets until catching up the difference without producing crackles) but due to this fact it's at least not a simple task to send urbs exactly corresponding to the period limit.

I am still unclear on what makes the number of samples per packet vary, I didn't see this in the code. I understand that it could vary for 44.1 but for 48kHz (my case) I'd think it should be constant.

The variation is not directly in the code. I remember that the caiaq driver enlightened me since this behaviour is commented there. The variation comes from the device and these steps lead to this variation: 1. the driver sends n urbs containing m packets with size 0 (i.e. no audio) to the device 2. the device sends back each of the n urbs one-by-one, filled with the captured audio; the urbs will contain m packets, each packet having an individual number of samples s_p ("sample" being defined here as: 1 sample contains one value for each channel) 3. for each urb returned from the device, the driver sends another urb containing again m packets, but this time with s_p samples per packet (corresponding to the packet-dependent s_p received from the device)

I indeed found out that when using 48kHz, I first thought that the number of samples per packet were identical every time. When using this heuristic, crackling noise occured in the sound output and I looked at the number of samples per packet. They varied not very often but sometimes - around once per 2 seconds.

My explanation for this problem is, that the sound card has its own clock, that will not oscillate at exactly an integer multiple as the computer's/usb's internal clock. Even temperature will change the frequency a little bit, meaning that 48kHz is not always exactly 48kHz. I would have to start speculating about which clock drives the usb urb scheduling, so this question should be left for the more experienced USB programmers.

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• Another thing came into my mind while reading your post, but I'm not

sure if that would apply in praxis: If sending an urb is scheduled with the alsa buffer, it might be possible that the buffer is overwritten before the urb is sent, leading to complete audio-rubbish.

Indeed there could be some issues with pulseaudio rewinds, but the fact is that rewinds are broken anyway when the data is copied...

I gave this another thought and came to the conclusion: if you'd call snd_pcm_period_elapsed in the output urb retire callback, the theoretical problem wouldn't occur.

Torsten