Hello, I am developing a driver for a DAC on raspberry pi 3 by mostly making use of existing PCM5102A, and bcm2708 I2S driver code. What I've done is just configuring the cpu dai link, and for the sampling rate changes, I am setting some gpios which configures the clock of an ADC (PCM1804) which is a master. The main CPU and the DAC are slaves to the ADC and I have a 24.576MHz oscillator. The sampling rates I want to support are 48000, 96000 and 192000. I am able to react just fine to the requested rate in hw_params callback. Trying out the different rates in audacity works just fine. However, if I try to playback any wav files using mplayer or aplay or youtube, all of their playback requests for 48000, but they get played way too fast. Also if I request mplayer to use alsa output, it gets a buffer underrun and gets stuck. If I set audacity to use some other output device than alsa directly, it plays corrupted audio and hangs on stopping playback. How does audacity manage to play the audio at the correct speed? Is it doing software resampling to the 'project rate'? I've tried getting resampling to work using 'plug', but completely unsuccessfully - I can't get any sound out of it, it says it failed setting the requested parameters. Any pointers to what I should be investigating? I am new to the linux kernel module development, as well as sound drivers. Thank you in advance, Giedrius. _______________________________________________ Alsa-devel mailing list Alsa-devel@xxxxxxxxxxxxxxxx http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
