[LAD] Linux Audio Architecture

Hi, I started working with Linux audio very recently. Primarily, I wanted to understand the Linux audio stack and audio processing flow. As a simple exercise, - I write a simple capture/playback application for PCM interface and store the captured audio data in a .wav or .raw file - The device parameters I played with are: hw:2,0/plughw:2,0 (for USB headset) , 1/2 channel, 44100 sample rate, SND_PCM_ACCESS_RW_INTERLEAVED mode, period size 32, S16_LE format. - Use Ubuntu, kernel 3.9.4 and Logitech USB headset for development/testing To understand the control flow, I inspected the ALSA driver source code and got an understanding of the flow in kernel space. However, I am kind of lost in the user-space ALSA library. - What happens after ALSA handovers the data i.e. the audio data from mic is copied from USB transfer buffer to userspace buffer? What are the data processing steps? - Where is Pulseaudio working here? - Is there any software mixing happening? - In the USB sound driver, I changed the data in urb->transferbuffer to a pattern like 'abcdefghijklmnop', but in the capture application when I store the audio data in a .raw file, I don't completely get the data back. If I set 1 channel in the hw params, I get almost 'abcdefghijklmnop' pattern but sometimes get 'abcdefghijklm' i.e. the pattern incomplete and another pattern starts over. For 2 channels, the data interleaves in a pattern like 'ababcdcd...' but there are also some incomplete patterns and also I see some unexpected characters. I know it's a long post, but even if you can help with a part of it, I'll be greatly benefited. Thanks! -- View this message in context: http://linux-audio.4202.n7.nabble.com/Linux-Audio-Architecture-tp85571.html Sent from the linux-audio-dev mailing list archive at Nabble.com.