I trying to write ALSA application for recording audio, and when I try to set some parameters and then print them to the screen I getting some default numbers that i cant change
#include <alsa/asoundlib.h>
using namespace std;
typedef struct {
int audio;
int recording;
void *cons;
snd_pcm_t *inhandle;
snd_pcm_t *outhandle;
unsigned long sampleIndex;
unsigned long inlen;
unsigned long sampleRate;
} audio_t;
static audio_t aud;
void aboutAlsa(snd_pcm_t *handle,snd_pcm_hw_params_t *params) {
unsigned int val, val2;
snd_pcm_format_t val3;
int dir;
snd_pcm_uframes_t frames;
printf("ALSA library version: %s\n",SND_LIB_VERSION_STR);
printf("PCM handle name = '%s'\n",snd_pcm_name(handle));
printf("PCM state = %s\n",snd_pcm_state_name(snd_pcm_state(handle)));
snd_pcm_hw_params_get_access(params,(snd_pcm_access_t *) &val);
printf("access type = %s\n",snd_pcm_access_name((snd_pcm_access_t)val));
snd_pcm_hw_params_get_format(params, &val3);
printf("format = '%s' (%s)\n",snd_pcm_format_name(val3),
snd_pcm_format_description(val3));
snd_pcm_hw_params_get_subformat(params,(snd_pcm_subformat_t *)&val);
printf("subformat = '%s' (%s)\n",snd_pcm_subformat_name((snd_pcm_subformat_t)val),
snd_pcm_subformat_description((snd_pcm_subformat_t)val));
snd_pcm_hw_params_get_channels(params, &val);
printf("channels = %d\n", val);
snd_pcm_hw_params_get_rate(params, &val, &dir);
printf("rate = %d bps\n", val);
snd_pcm_hw_params_get_period_time(params,&val, &dir);
printf("period time = %d us\n", val);
snd_pcm_hw_params_get_period_size(params,&frames, &dir);
printf("period size = %d frames\n", (int)frames);
snd_pcm_hw_params_get_buffer_time(params,&val, &dir);
printf("buffer time = %d us\n", val);
snd_pcm_hw_params_get_buffer_size(params,(snd_pcm_uframes_t *) &val);
printf("buffer size = %d frames\n", val);
snd_pcm_hw_params_get_periods(params, &val, &dir);
printf("periods per buffer = %d frames\n", val);
snd_pcm_hw_params_get_rate_numden(params,&val, &val2);
printf("exact rate = %d/%d bps\n", val, val2);
val = snd_pcm_hw_params_get_sbits(params);
printf("significant bits = %d\n", val);
return;
}
static int openKnownAudio(int record) {
int rc;
int SAMPLERATE = 16000;
unsigned int val;
int dir=0;
snd_pcm_t *handle;
snd_pcm_hw_params_t *hw_params=NULL;
snd_pcm_uframes_t frames;
size_t esz = 256;
char err[esz];
/* Open PCM device for recording (capture). */
if (record) {
if ((rc=snd_pcm_open(&aud.inhandle, "default",SND_PCM_STREAM_CAPTURE, 0))<0) {
snprintf(err, esz, "unable to open pcm device for recording: %s\n",snd_strerror(rc));
}
handle=aud.inhandle;
} else {
if ((rc=snd_pcm_open(&aud.outhandle, "default",SND_PCM_STREAM_PLAYBACK, 0))<0) {
snprintf(err, esz, "unable to open pcm device for playback: %s\n",snd_strerror(rc));
}
handle=aud.outhandle;
}
/* Configure hardware parameters */
if((rc=snd_pcm_hw_params_malloc(&hw_params)) < 0) {
snprintf(err, esz, "unable to malloc hw_params: %s\n",snd_strerror(rc));
}
if((rc=snd_pcm_hw_params_any(handle, hw_params))<0) {
snprintf(err, esz, "unable to setup hw_params: %s\n",snd_strerror(rc));
}
if((rc=snd_pcm_hw_params_set_access(handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED))<0) {
snprintf(err, esz, "unable to set access mode: %s\n",snd_strerror(rc));
}
if((rc=snd_pcm_hw_params_set_format(handle, hw_params, SND_PCM_FORMAT_S16_LE))<0) {
snprintf(err, esz, "unable to set format: %s\n",snd_strerror(rc));
}
if((rc=snd_pcm_hw_params_set_channels(handle, hw_params, 1))<0) {
snprintf(err, esz, "unable to set channels: %s\n",snd_strerror(rc));
}
val = SAMPLERATE;
dir = 0;
if((rc=snd_pcm_hw_params_set_rate(handle, hw_params, SAMPLERATE,0))<0) {
snprintf(err, esz, "unable to set samplerate: %s\n",snd_strerror(rc));
}
if (val!=SAMPLERATE) {
snprintf(err, esz, "unable to set requested samplerate: requested=%i got=%i\n",SAMPLERATE,val);
}
frames = 64;
if ((rc=snd_pcm_hw_params_set_period_size_near(handle,hw_params, &frames, &dir))<0) {
snprintf(err, esz, "unable to set period size: %s\n",snd_strerror(rc));
}
frames = 4096;
if ((rc=snd_pcm_hw_params_set_buffer_size_near(handle,hw_params, &frames))<0) {
snprintf(err, esz, "unable to set buffer size: %s\n",snd_strerror(rc));
}
if ((rc = snd_pcm_hw_params(handle, hw_params))<0) {
snprintf(err, esz, "unable to set hw parameters: %s\n",snd_strerror(rc));
}
aboutAlsa(handle,hw_params);
snd_pcm_hw_params_free(hw_params);
aud.recording = (record)? 1:0;
aud.audio=1;
return 1;
}
This what I get on raspberry pi when I run it:
ALSA library version: 1.0.28
PCM handle name = 'default'
PCM state = PREPARED
access type = RW_INTERLEAVED
format = 'S16_LE' (Signed 16 bit Little Endian)
subformat = 'STD' (Standard)
channels = 1
rate = 16000 bps
period time = 21333 us
period size = 341 frames
buffer time = 256000 us
buffer size = 4096 frames
periods per buffer = 4096 frames
exact rate = 16000/1 bps
significant bits = 16
And this is what I get when I run it on desktop pc:
ALSA library version: 1.0.28
PCM handle name = 'default'
PCM state = PREPARED
access type = RW_INTERLEAVED
format = 'S16_LE' (Signed 16 bit Little Endian)
subformat = 'STD' (Standard)
channels = 1
rate = 16000 bps
period time = 4000 us
period size = 64 frames
buffer time = 256000 us
buffer size = 4096 frames
periods per buffer = 64 frames
exact rate = 16000/1 bps
significant bits = 16
As you can see I'm trying to set the period size to 64 and getting back 341, this value only changes when I change the rate, lets say I set the rate to 44100 and this what I getting back:
rate = 44100 bps
period time = 21333 us
period size = 940 frames
buffer time = 85328 us
buffer size = 3763 frames
periods per buffer = 3763 frames
On desktop pc this doesn't happens I tried to trace down this functions in alsa-lib but I getting lost there also tried different sound cards and still getting same result .
In case of PulseAudio you did set the PulseAudio device , not the real device. The real HW can have the limitation, you must correctly react to. If you'd like to see min/max boundary of some parameter, you can do the next:
The same using min/max functions
I faced with the same issue, when tried to set the period size. There are my boundary (two different pcm devices):
log #1
log#2
Here we can't change the period size due to period time limitation.