Hi All,
I am trying to read data from a usb microphone and using the pretty standard
method of using ioctl's to setup the sampling rate, channels, bits and block
size . This all works so the device is correctly setup. I then use "read" to
read samples from the device which shows up as /dev/dsp1. I get a lot more
samples from this read command in one second of recording than the set
sample rate. E.g. if i set 10Khz on one run i got 269312 samples. Looking at
the raw data it looks like there is a lot of duplication of data? is this
common for the audio input device? if so what kind of encoding is it (e.g
with some specific redundancy built in)?
thanks
farhan
Hi all,
I am working on a system which includes a connexant AD1989A HDA codec
connected to a ATOM processor.
I have four microphones connected to the B and C ports of this codec.
If I change one of the 3 capture gains present in the alsamixer (Capture,
Capture 1 or Capture 3), I can't have any more signal on the B and C ports
It seems to be due to the fact that ALSA breaks (for example for the Capture
gain) the link between the ADC selector 0 and ADC_0 widgets of the codec,
what can be easily seen with codecgraph.
Have you ever had this kind of problem?
Do you know how can I resolve it ?
Many thanks for your asnwers and best regards
Vincent
