The Sigmatel chip is capable of recording 16 and 24-bit samples. However, due to restrictions of the HDA spec (only 8, 16, and 32 bit sample formats), the codec zero-pads 24 bit samples to 32 bits. As a consequence, it is not possible to use -f S24, but -f S32. Therefore, this will record 24 bit sound (zero padded to 32 bits).
Typically, there are three ADCs (Analog to Digital Converters), and each ADC has three controls:
In general, all three ADCs have the same mixer control names, which means
that you will see three 'Input source' controls, three 'Capture volume'
controls, and so on. In amixer, they can be distinguished by a
control index, such as:
#set capture source of second ADC to line-in
amixer -c 0 sset 'Input Source',1 'Line'
In alsamixer, on the other hand, there are controls like "Input
source, Input source 1, Input source 2", meaning that
ADC0's controls have no suffix, ADC1's controls have suffix "1", and
ADC2's controls have suffix "2". The "PF3" and "PF4" keys switch between
the Playback and Capture views, and the "space" key makes a selection. The
"m" key toggles between mute/unmute.
kmix and gnome-alsamixer do not use indices at all.
Therefore, one should rely on the fact that the first control is for ADC0,
and so on.
There has been implemented support for using all of the three ADCs at the same time, and recording from different sources. Besides, there is an Analog Loopback, which works as follow:
1) First, sound enters through input of ADC0.
2) Second, it can be muted and/or amplified by the analog part of ADC0 (on
that codec, ADC can amplify the sound, but can not decrease it)
3) Then, it is mixed with output of ADC0, after its volume is applied.
Therefore, the only way to control the volume is to increase it with the
capture volume / capture boost.
Furthermore, since ADC0 is connected to the headphone/line out, the loopback will not be heard on the other output channels, and this is the reason for the loud volume: headphone outputs are amplified more than the other outputs.
Finally, there are three input source mixer controls for each ADC, and a Master Volume, which will affect the output of all DACs at the same time.
If you have 5.1/7.1 speakers, and the connector for the Center/Sub-woofer is non-standard, it is possible to swap the Center/Sub-woofer channels.
What is good, is the fact that recording from different input sources is not supported by the Windows driver! Note that the Windows driver has an analog loopback (they call it a Monitor Input), and its volume can be decreased to zero, but it seems that this loopback is fake (records first and then plays the sound back).
The format of ALSA (Advanced Linux Sound Architecture) hw devices is shown below, and the capture device is a stream with three substreams:
hw:card:stream:substream
To record from an specific ADC, one should specify hw:0,0,0 ; hw:0,0,1 ; hw:0,0,2 as a record device (and use an audio rate up to 192000 Hz):
# to record from 2nd ADC and play on the first sound card
arecord -D hw:0,0,1 -f S16_LE -c2 -r48000 | aplay -D hw:0 &
# to record from 1st ADC and play via pulseaudio
arecord -D hw:0,0,0 -f S16_LE -c2 -r48000 | aplay -D pulse &
# to record from 3rd ADC and play on the third sound card
arecord -D hw:0,0,2 -f S16_LE -c2 -r48000 | aplay -D hw:3 &
Note that this can be used to avoid the high volume from the Analog
Loopback (a "software" loopback).
For controlling the channel levels, it is possible to use a graphical tool, such as gnome-alsamixer. However, amixer can also be used for that purpose.
For instance:
Therefore, bash scripts can be associated to buttons (application launchers) on the Desktop, thus providing a fast way of setting recording levels for a Microphone, Line in, or activating the software loopback.
OSS was used back to the 2.4 kernel series. However, ALSA can emulate its API, because several applications still use it.
Basically, there are two methods to emulate OSS:
1) In-kernel emulation: It is not possible to tell the programs which sub-stream to use, but they will pick the first available. Just launch, say, skype, then xvidcap, and skype will use ADC0, and xvidcap the ADC1.
For example, you can open /dev/dsp three times, and each time a free ADC
will be used. This can be tested by doing:
# will record from 1st ADC
cat /dev/dsp > /dev/null &
# will record from 2nd ADC
cat /dev/dsp > /dev/null &
# will record from 3rd ADC
cat /dev/dsp > /dev/null &
# will fail, since all ADCs are already in use
cat /dev/dsp > /dev/null &
Note: remember to configure the ADCs for your needs, that is, set input
sources, increase/decrease gain if necessary, etc.
2) Using
aoss
(alsa-oss-1.0.14):
(configurable through ~/.asoundrc)
#
# for using with aoss
# e.g., aoss xvidcap
#
pcm.dsp0 {
type plug
slave.pcm "hw:0,0,1"
}
Jack is a low-latency audio server based on ALSA, and allows performing low-latency professional-grade audio editing and mixing. To avoid xruns when using jack (callback messages and statistics reports, which occur when the delayed_usecs reported by the driver is "too large"), the following option can be added to /etc/modprobe.conf:
options snd-hda-intel position_fix=1 model=ref
There are several models available to choose from (/usr/share/doc/alsa-driver-1.0.15/Documentation/ALSA-Configuration.txt):
| Model | Layout |
|---|---|
| ref | Reference board |
| 3stack | D965 3stack |
| 5stack | D965 5stack + SPDIF |
| dell-3stack | Dell Dimension E520 |
Skype is supporting ALSA natively. Just add the appropriate configuration to ~/.asoundrc:
#
# for using with skype-1.4.0.118
# set card0 as the input device
# ADC1 (subdevice 1)
#
pcm.card0 {
type hw
card 0
device 0
subdevice 1
}
This module (snd-hda-intel) is working just fine with
MythTV (a PVR - Personal Video
Recorder). Just set the Audio Device to /dev/dsp in the Capture
Card Setup, turn off the Analog Loopback and set the recording
level of ADC0. For being able to use
pulseaudio, the Audio output device must be set to ALSA:default in
the frontend. In my case, I have to turn on the sound of my capture card
first, by using v4lctl from
xawtv:
v4lctl -c /dev/tvtuner volume mute off
This howto is applicable only to the STAC922x and STAC927x codec family, because some STACs have two ADCs, some have just one, some may not have an analog loopback, and some may have significantly different mixer controls.
This is the list of supported codecs by the patch_sigmatel.c driver:
| Name | #DACs | #ADCs | Analog loopback | Maximum sampling rate | Notes |
|---|---|---|---|---|---|
| 9200 | 1 | 1 | yes(?) | 96000 | (1) |
| 9202/9202D | 1 | 1 | yes(?) | 192000 | (1) |
| 9250(D)/1(D) | 1 | 1 | yes(?) | 192000 | (1) |
| 9204(D)/5(D) | 2 | 2 | yes | 192000 | |
| 9254(D)/5(D) | 2 | 2 | yes(?) | 192000(?) | (2) |
| 9220(D)/1(D)/3D | 4 | 2 | no | 192000 | |
| 9227/8/9 | 4 | 3 | yes | 192000 | |
| 9271(D)/2(D)/3(D)/4(D) | 5 | 3 | yes | 192000 | |
| 9872 | ? | ? | ? | 192000 | (3) |
Notes:
(1) - these chips are very different from what is described here.
(2) - no data-sheet available. Master volume + analog loopback may not be
available.
(3) - no data-sheet available. Driver currently partially supports codecs
found on Sony Vaio laptops.
Written by: Maxim Levitsky and Paulo Roma Cavalcanti.
#alsa on freenode: MaximLevitsky (nick)