ALSA: 2 input channels, only one is recording

So here is my pickle. I am very new to linux and to make matters worse i am working on an embedded system and not a PC. i have been searching for this topic but almost everything i find shows just 1 recording device.
we have a codec/sound card that has 2 input channels and 2 output channels and i need all of them at the same time. when i run my code to capture the audio samples i am only getting 1 channel and not the other. when i do arecord it is the same result.
my first series of questions are: Does ALSA only do 1 input at a time by default? if so is there a way to change that? if not does it mix the 2 inputs when doing an arecord or do i have to specify which channel i want?
my other thought about my issue is that it isn't being routed properly and it is taking the samples off an input that is not connected to anything. the thing is i can't figure out, in the OS, which port it thinks it is connected to. i think that is too deep into the hardware for linux to have an idea about that.
thanks for any help you can give
Answers
-
By default ALSA has all channels muted. Those have to be unmuted manually.
Source - https://wiki.archlinux.org/index.php/Advanced_Linux_Sound_Architecture
0 -
so here is more information:
when i do "amixer" i get:
...
Simple mixer control 'Left PGA Mixer Line1L',0
Capabilities: pswitch pswitch-joined
Playback channels: Mono
Mono: Playback [on]
...which, if "Mono: Playback" means the samples are being collected is right
and
...
Simple mixer control 'Right PGA Mixer Line1L',0
Capabilities: pswitch pswitch-joined
Playback channels: Mono
Mono: Playback [off]
Simple mixer control 'Right PGA Mixer Line1R',0
Capabilities: pswitch pswitch-joined
Playback channels: Mono
Mono: Playback [on]
Simple mixer control 'Right PGA Mixer Line2R',0
Capabilities: pswitch pswitch-joined
Playback channels: Mono
Mono: Playback [off]
...which looks like it is pulling from Line1R, which is wrong
even though in my DTS file i have:
simple-audio-card,widgets =
"Speaker", "POTS Line Out",
"Speaker", "Handset Speaker",
"Line", "POTS Line In",
"Microphone", "Handset Mic";
simple-audio-card,routing =
"POTS Line Out", "MONO_LOUT",
"Handset Speaker", "LLOUT",
"LINE2R", "POTS Line In",
"LINE1L", "Handset Mic",
"Handset Mic", "Mic Bias";it would appear to me it is pulling the samples in from the wrong port
so when i do:
amixer set 'Right PGA Mixer Line1R' offamixer set 'Right PGA Mixer Line2R' on
and then do an arecord, i get the audio i was looking for. so it is routing the audio incorrectly
hopefully somebody can help me with this: in code (c) is there a way to do the input switching like above? i have been using snd_mixer_* commands but i am not sure how to use something like that to achieve what i want. or am i stuck using "system" calls?
0 -
well i might have figured it out so nevermind:
/****************************************************************************************
* found @ https://alsa.opensrc.org/HowTo_access_a_mixer_control
*/
void SetInputToLine2R(void)
{- const char *card = "default";
- const char *selem_name_line1r = "Right PGA Mixer Line1R Switch";
- const char *selem_name_line2r = "Right PGA Mixer Line2R Switch";
- int err;
- snd_hctl_t *hctl;
- err = snd_hctl_open(&hctl, card, 0);
- err = snd_hctl_load(hctl);
- snd_ctl_elem_id_t *id;
- snd_ctl_elem_id_alloca(&id);
- snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_MIXER);
- //snd_ctl_elem_id_set_id(id, 1);
- snd_ctl_elem_id_set_name(id, selem_name_line1r);
- snd_hctl_elem_t *elem = snd_hctl_find_elem(hctl, id);
- if(elem != 0)
- {
- snd_ctl_elem_value_t *control;
- snd_ctl_elem_value_alloca(&control);
- snd_ctl_elem_value_set_id(control, id);
- snd_ctl_elem_value_set_integer(control, 0, 0);
- err = snd_hctl_elem_write(elem, control);
- //turn on line2r
- snd_ctl_elem_id_set_name(id, selem_name_line2r);
- elem = snd_hctl_find_elem(hctl, id);
- if(elem != 0)
- {
- snd_ctl_elem_value_set_integer(control, 0, 1);
- err = snd_hctl_elem_write(elem, control);
- }
- }
- snd_hctl_close(hctl);
}
0
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