Hello I am trying to check if a Beaglebone AI is getting correctly I2S from a TIDA-1454 so if I try to arecord test.wav I get:
debian#beaglebone:/etc$ arecord -d 10 ALSA lib
pcm_dsnoop.c:638:(snd_pcm_dsnoop_open) unable to open slave arecord:
main:828: audio open error: No such file or directory
But the card is detected since i get this output from arecord -l:
**** List of CAPTURE Hardware Devices ****
card 1: PCM5102a [PCM5102a], device 0: davinci-mcasp.0-pcm5102a-hifi pcm5102a-hifi-0 []
Subdevices: 1/1
Subdevice #0: subdevice #0
or arecord -L:
debian#beaglebone:/var/lib/cloud9$ arecord -L
null
Discard all samples (playback) or generate zero samples (capture)
onboard
onboard_capture_left
onboard_playback_left
onboard_capture_right
onboard_playback_right
plug_onboard_capture_left
plug_onboard_playback_left
default
sysdefault:CARD=PCM5102a
PCM5102a,
Default Audio Device
dmix:CARD=PCM5102a,DEV=0
PCM5102a,
Direct sample mixing device
dsnoop:CARD=PCM5102a,DEV=0
PCM5102a,
Direct sample snooping device
hw:CARD=PCM5102a,DEV=0
PCM5102a,
Direct hardware device without any conversions
plughw:CARD=PCM5102a,DEV=0
PCM5102a,
Hardware device with all software conversions
and ALSA configuration:
pcm.onboard{
type hw
card 0
}
ctl.onboard {
type hw
card 0
}
### Dsnoop splited channels
pcm.onboard_capture_left {
type dsnoop
ipc_key 32
slave {
pcm "onboard"
channels 2
rate 48000
format S32_LE
}
bindings.0 0
}
pcm.onboard_playback_left {
type dmix
ipc_key 33
slave {
pcm "onboard"
channels 2
}
bindings.0 0
}
pcm.onboard_capture_right {
type dsnoop
ipc_key 32
slave {
pcm "onboard"
channels 2
rate 48000
format S32_LE
}
bindings.0 1
}
pcm.onboard_playback_right {
type dmix
ipc_key 33
slave {
pcm "onboard"
channels 2
}
bindings.0 1
}
### PLUGS ##
### used with darkice
### device = plug:plug_onboard_left
pcm.plug_onboard_capture_left{
type plug
slave.pcm "onboard_capture_left"
}
pcm.plug_onboard_playback_left{
type plug
slave.pcm "onboard_playback_left"
}
pcm.!default {
type asym
playback.pcm "plug_onboard_playback_left"
capture.pcm "plug_onboard_capture_left"
}
So what can it be the problem?
When recording you should tell alsa what device to use:
**** List of CAPTURE Hardware Devices ****
card 1: PCM5102a [PCM5102a], device 0: davinci-mcasp.0-pcm5102a-hifi >pcm5102a-hifi-0 []
Subdevices: 1/1
Subdevice #0: subdevice #0
Your device address is : hw:1,0 (card 1, device 0 )
So :
arecord --device="hw:1,0" test.wav
He should ask you for the format, for instance:
Available format:
- S8
- S16_LE
Then you can add the format :
arecord --device="hw:1,0" -f S16_LE test.wav
I have written a small program to play PCM audio files using a reference from github with slight modifications for my application. The program is compiling fine, but I am having a few runtime issues regarding dependencies and ALSA devices.
The errors I am receiving are:
ERROR: Failed to load plugin /usr/lib/x86_64-linux-gnu/ao/plugins-4/libsndio.so => dlopen() failed
ERROR: Failed to load plugin /usr/lib/x86_64-linux-gnu/ao/plugins-4/libnas.so => dlopen() failed
ao_alsa WARNING: Unable to open surround playback. Trying default device...
ALSA lib pcm_dmix.c:1089:(snd_pcm_dmix_open) unable to open slave
ao_alsa ERROR: Unable to open ALSA device 'default' for playback => No such file or directory
I have been researching many different solutions and none seem to be working, including adding a line to the alsa.conf file in /etc/modprobe.d (however, these solutions suggest adding a line in reference to intel when I am using an AMD setup.
Here is the full code:
/*
*
* ao_example.c
*
* Written by Stan Seibert - July 2001
*
* Legal Terms:
*
* This source file is released into the public domain. It is
* distributed without any warranty; without even the implied
* warranty * of merchantability or fitness for a particular
* purpose.
*
* Function:
*
* This program opens the default driver and plays a 440 Hz tone for
* one second.
*
* Compilation command line (for Linux systems):
*
* gcc -o ao_example ao_example.c -lao -ldl -lm -lsndfile
*
*/
#include <stdio.h>
#include <string.h>
#include <ao/ao.h>
#include <math.h>
#include <sndfile.h>
#define BUF_SIZE 4096
static void clean(ao_device *device, SNDFILE *file){
ao_close(device);
sf_close(file);
ao_shutdown();
}
int main(int argc, char **argv)
{
ao_device *device;
ao_sample_format format;
unsigned long count;
int default_driver;
short *buffer;
int buf_size;
int sample;
int i;
//FILE *fp;
SF_INFO sfinfo;
if (argc != 2){
printf("usage: %s <filename>\n", argv[0]);
exit(1);
}
/* -- Initialize -- */
fprintf(stderr, "libao example program\n");
SNDFILE *fp = sf_open(argv[1], SFM_READ, &sfinfo);
// fp = fopen(argv[1], "rb");
if (fp == NULL){
printf("Cannot open %s.\n", argv[1]);
exit(1);
}
printf("samples: %d\n", sfinfo.frames);
printf("sample rate: %d\n", sfinfo.samplerate);
printf("Channels: %d\n", sfinfo.channels);
ao_initialize();
/* -- Setup for default driver -- */
default_driver = ao_default_driver_id();
switch(sfinfo.format & SF_FORMAT_SUBMASK){
case SF_FORMAT_PCM_16:
format.bits = 16;
break;
case SF_FORMAT_PCM_24:
format.bits = 24;
break;
case SF_FORMAT_PCM_32:
format.bits = 32;
break;
case SF_FORMAT_PCM_S8:
format.bits = 8;
break;
case SF_FORMAT_PCM_U8:
format.bits = 8;
break;
default:
format.bits = 24;
break;
}
format.channels = sfinfo.channels;
format.rate = sfinfo.samplerate;
format.byte_format = AO_FMT_LITTLE;
// format.byte_format = AO_FMT_NATIVE;
format.matrix = 0;
// memset(&format, 0, sizeof(format));
// format.bits = 24;
// format.channels = 16;
// format.rate = 48000;
// format.byte_format = AO_FMT_LITTLE;
/* -- Open driver -- */
device = ao_open_live(default_driver, &format, NULL /* no options */);
if (device == NULL) {
fprintf(stderr, "Error opening device.\n");
return 1;
}
// fseek(fp, 0, SEEK_END);
// count = ftell(fp);
// fseek(fp, 0, SEEK_SET);
// // printf("count: %ld\n", count);
buffer = calloc(BUF_SIZE, sizeof(short));
while(1){
int read = sf_read_short(fp, buffer, BUF_SIZE);
if (ao_play(device, (char *) buffer, (uint_32)(read * sizeof(short))) == 0){
printf("ao_play: failed\n");
clean(device, fp);
break;
}
}
clean(device, fp);
return 0;
}
I hope someone else has had the same trouble and can shed light on the solution. Thank you.
To summarise the discussion:
The problem does not seem to lie within the code per se but in the libao configuration.
According to the libao documentation, the library tries to determine a default driver as follows:
In the absence of configuration files to explicit identify a default
driver, the library will try to detect a suitable default driver. It
does this by testing every available live output driver (using
ao_plugin_test()) and finding the driver with the highest priority
(see the ao_info struct) that works. Drivers with priority 0, such as
the null and file output drivers, are never selected as the default.
The error messages indicate that several drivers including nas, sndio and alsa are tried. Choosing the driver manually with ao_driver_id(...) instead of using ao_default_driver_id() fixes the issue.
Further problems with opening the device with ao_open_live(...) can be investigated by getting the corresponding error number with printf("errno %d\n", errno);. The output can be interpreted as follows:
AO_ENODRIVER (1) - No driver corresponds to driver_id.
AO_ENOTLIVE (3) - This driver is not a live output device.
AO_EBADOPTION (4) - A valid option key has an invalid value.
AO_EOPENDEVICE (5) - Cannot open the device (for example, if /dev/dsp cannot be opened for writing).
AO_EFAIL (100) - Any other cause of failure.
Besides that debugging can be enabled in the ~/.libao configuration file to get further information.
Here is the lsmod output:
Module Size Used by
rfcomm 81920 4
cmac 16384 2
algif_hash 16384 1
algif_skcipher 16384 1
af_alg 28672 6 algif_hash,algif_skcipher
bnep 24576 2
nls_iso8859_1 16384 1
hid_logitech_hidpp 45056 0
input_leds 16384 0
joydev 24576 0
hid_logitech_dj 28672 0
hid_generic 16384 0
snd_hda_codec_realtek 131072 1
snd_hda_codec_generic 81920 1 snd_hda_codec_realtek
ledtrig_audio 16384 1 snd_hda_codec_generic
snd_hda_codec_hdmi 61440 1
snd_hda_intel 53248 4
edac_mce_amd 32768 0
snd_intel_dspcfg 28672 1 snd_hda_intel
iwlmvm 393216 0
snd_hda_codec 139264 4 snd_hda_codec_generic,snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec_realtek
snd_hda_core 94208 5 snd_hda_codec_generic,snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec,snd_hda_codec_realtek
mac80211 905216 1 iwlmvm
snd_hwdep 20480 1 snd_hda_codec
snd_pcm 114688 4 snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec,snd_hda_core
amdgpu 5218304 11
libarc4 16384 1 mac80211
snd_seq_midi 20480 0
snd_seq_midi_event 16384 1 snd_seq_midi
kvm 712704 0
snd_rawmidi 36864 1 snd_seq_midi
snd_seq 69632 2 snd_seq_midi,snd_seq_midi_event
crct10dif_pclmul 16384 1
snd_seq_device 16384 3 snd_seq,snd_seq_midi,snd_rawmidi
ghash_clmulni_intel 16384 0
iwlwifi 352256 1 iwlmvm
snd_timer 40960 2 snd_seq,snd_pcm
aesni_intel 372736 3
iommu_v2 20480 1 amdgpu
crypto_simd 16384 1 aesni_intel
gpu_sched 36864 1 amdgpu
cryptd 24576 3 crypto_simd,ghash_clmulni_intel
glue_helper 16384 1 aesni_intel
usbhid 57344 1 hid_logitech_dj
ttm 102400 1 amdgpu
wmi_bmof 16384 0
hid 135168 4 usbhid,hid_generic,hid_logitech_dj,hid_logitech_hidpp
rapl 20480 0
drm_kms_helper 217088 1 amdgpu
efi_pstore 16384 0
cfg80211 778240 3 iwlmvm,iwlwifi,mac80211
cec 53248 1 drm_kms_helper
rc_core 61440 1 cec
snd 94208 19 snd_hda_codec_generic,snd_seq,snd_seq_device,snd_hda_codec_hdmi,snd_hwdep,snd_hda_intel,snd_hda_codec,snd_hda_codec_realtek,snd_timer,snd_pcm,snd_rawmidi
fb_sys_fops 16384 1 drm_kms_helper
syscopyarea 16384 1 drm_kms_helper
sysfillrect 16384 1 drm_kms_helper
sysimgblt 16384 1 drm_kms_helper
k10temp 16384 0
ccp 98304 0
soundcore 16384 1 snd
btusb 57344 0
btrtl 24576 1 btusb
btbcm 16384 1 btusb
btintel 28672 1 btusb
mac_hid 16384 0
bluetooth 581632 31 btrtl,btintel,btbcm,bnep,btusb,rfcomm
ecdh_generic 16384 1 bluetooth
ecc 32768 1 ecdh_generic
sch_fq_codel 20480 3
hwmon_vid 16384 0
parport_pc 45056 0
ppdev 24576 0
lp 20480 0
drm 552960 9 gpu_sched,drm_kms_helper,amdgpu,ttm
parport 65536 3 parport_pc,lp,ppdev
ip_tables 32768 0
x_tables 49152 1 ip_tables
autofs4 45056 2
crc32_pclmul 16384 0
xhci_pci 20480 0
nvme 49152 2
i2c_piix4 28672 0
igb 221184 0
ahci 40960 0
libahci 36864 1 ahci
xhci_pci_renesas 20480 1 xhci_pci
i2c_algo_bit 16384 2 igb,amdgpu
nvme_core 114688 4 nvme
dca 16384 1 igb
wmi 32768 1 wmi_bmof
video 49152 0
and here is the aplay -L output:
default
Playback/recording through the PulseAudio sound server
surround21
2.1 Surround output to Front and Subwoofer speakers
surround40
4.0 Surround output to Front and Rear speakers
surround41
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50
5.0 Surround output to Front, Center and Rear speakers
surround51
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
null
Discard all samples (playback) or generate zero samples (capture)
samplerate
Rate Converter Plugin Using Samplerate Library
speexrate
Rate Converter Plugin Using Speex Resampler
jack
JACK Audio Connection Kit
oss
Open Sound System
pulse
PulseAudio Sound Server
upmix
Plugin for channel upmix (4,6,8)
vdownmix
Plugin for channel downmix (stereo) with a simple spacialization
hdmi:CARD=Generic,DEV=0
HD-Audio Generic, HDMI 0
HDMI Audio Output
hdmi:CARD=Generic,DEV=1
HD-Audio Generic, HDMI 1
HDMI Audio Output
hdmi:CARD=Generic,DEV=2
HD-Audio Generic, HDMI 2
HDMI Audio Output
dmix:CARD=Generic,DEV=3
HD-Audio Generic, HDMI 0
Direct sample mixing device
dmix:CARD=Generic,DEV=7
HD-Audio Generic, HDMI 1
Direct sample mixing device
dmix:CARD=Generic,DEV=8
HD-Audio Generic, HDMI 2
Direct sample mixing device
dsnoop:CARD=Generic,DEV=3
HD-Audio Generic, HDMI 0
Direct sample snooping device
dsnoop:CARD=Generic,DEV=7
HD-Audio Generic, HDMI 1
Direct sample snooping device
dsnoop:CARD=Generic,DEV=8
HD-Audio Generic, HDMI 2
Direct sample snooping device
hw:CARD=Generic,DEV=3
HD-Audio Generic, HDMI 0
Direct hardware device without any conversions
hw:CARD=Generic,DEV=7
HD-Audio Generic, HDMI 1
Direct hardware device without any conversions
hw:CARD=Generic,DEV=8
HD-Audio Generic, HDMI 2
Direct hardware device without any conversions
plughw:CARD=Generic,DEV=3
HD-Audio Generic, HDMI 0
Hardware device with all software conversions
plughw:CARD=Generic,DEV=7
HD-Audio Generic, HDMI 1
Hardware device with all software conversions
plughw:CARD=Generic,DEV=8
HD-Audio Generic, HDMI 2
Hardware device with all software conversions
usbstream:CARD=Generic
HD-Audio Generic
USB Stream Output
sysdefault:CARD=Generic_1
HD-Audio Generic, ALC1220 Analog
Default Audio Device
front:CARD=Generic_1,DEV=0
HD-Audio Generic, ALC1220 Analog
Front speakers
surround21:CARD=Generic_1,DEV=0
HD-Audio Generic, ALC1220 Analog
2.1 Surround output to Front and Subwoofer speakers
surround40:CARD=Generic_1,DEV=0
HD-Audio Generic, ALC1220 Analog
4.0 Surround output to Front and Rear speakers
surround41:CARD=Generic_1,DEV=0
HD-Audio Generic, ALC1220 Analog
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=Generic_1,DEV=0
HD-Audio Generic, ALC1220 Analog
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=Generic_1,DEV=0
HD-Audio Generic, ALC1220 Analog
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=Generic_1,DEV=0
HD-Audio Generic, ALC1220 Analog
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
dmix:CARD=Generic_1,DEV=0
HD-Audio Generic, ALC1220 Analog
Direct sample mixing device
dsnoop:CARD=Generic_1,DEV=0
HD-Audio Generic, ALC1220 Analog
Direct sample snooping device
hw:CARD=Generic_1,DEV=0
HD-Audio Generic, ALC1220 Analog
Direct hardware device without any conversions
plughw:CARD=Generic_1,DEV=0
HD-Audio Generic, ALC1220 Analog
Hardware device with all software conversions
usbstream:CARD=Generic_1
HD-Audio Generic
USB Stream Output
I am trying to use the multi ALSA plugin to copy my desktop audio to the Loopback device created by snd-aloop. However, when I try to use the .asoundrc below, I get this error:
> aplay -vv test.wav
Playing WAVE 'test.wav' : Signed 32 bit Little Endian, Rate 48000 Hz, Stereo
ALSA lib pcm_params.c:2226:(snd1_pcm_hw_refine_slave) Slave PCM not usable
aplay: set_params:1314: Broken configuration for this PCM: no configurations available
.asoundrc:
pcm.both {
type multi
slaves.a.pcm "dmixer"
slaves.b.pcm "loop"
slaves.a.channels 2
slaves.b.channels 2
bindings.0 { slave a; channel 0; }
bindings.1 { slave a; channel 1; }
bindings.2 { slave b; channel 0; }
bindings.3 { slave b; channel 1; }
}
pcm.dmixer {
type dmix
ipc_key 1231
ipc_perm 0666
slave.pcm "hw:PCH,0"
slave.channels 2
slave.rate 48000
slave.format S32_LE
bindings {
0 0
1 1
}
}
pcm.loop {
type dmix
ipc_key 29637
slave {
pcm "hw:Loopback,0"
rate 48000
format S32_LE
channels 2
period_size 1024
buffer_size 9182
}
}
pcm.hubcap {
type plug
slave {
pcm "hw:Loopback,1,0"
rate 48000
format S32_LE
}
}
pcm.!default {
type asym
playback.pcm {
type plug
slave.pcm "both"
ttable [
[ 1 0 1 0 ]
[ 0 1 0 1 ]
]
}
capture.pcm "plug:dsnoop"
}
defaults.pcm.surround21.card off
defaults.pcm.surround21.device off
defaults.pcm.surround40.card off
defaults.pcm.surround40.device off
defaults.pcm.surround41.card off
defaults.pcm.surround41.device off
defaults.pcm.surround50.card off
defaults.pcm.surround50.device off
defaults.pcm.surround51.card off
defaults.pcm.surround51.device off
defaults.pcm.surround71.card off
defaults.pcm.surround71.device off
Things to note:
When I remove both dmixes, it works fine.
In the multi plugin, when I change both slaves to the same pcm, it works fine for that one pcm.
Ex.
...
type multi
slaves.a.pcm "dmixer"
slaves.b.pcm "dmixer"
...
Try setting slave.period_size 1024 in pcm.dmixer block.
I am trying to add a dix to my Alsa conf file, but when I playback using it the sample rate is wrong (it's too slow)
I have the following:
pcm.dmix0 {
type dmix
ipc_key 34521
slave {
pcm "hw:0,0"
format S16_LE
rate 48000
}
hint {
description "DMix of Card0"
}
}
pcm.asym51 {
type asym
playback.pcm "dmix0"
capture.pcm "hw:0,0"
}
pcm.dsp0 {
type plug
slave.pcm "asym51"
}
When I play the audio back directly to the hardware, it plays fine, but I do miss a bit at the start.aplay -D hw:0,0 /usr/share/sounds/alsa/Front_Center.wav
When I use my dsp0 / dmix the rate is much slower than it should be:aplay -D dsp0 /usr/share/sounds/alsa/Front_Center.wav
What am I doing wrong?