Hi, and thanks for the reply.
>I think that you must link .so to -lasound, too.
This was done and shows in the command line example below, IE,
samiam@samiam-home-pc:~/projects/petit_ami$ gcc -fPIC -c test2.c -o test2.o
> samiam@samiam-home-pc:~/projects/petit_ami$ gcc -g3 -Iinclude linux/playwav.c ./test2.so -lasound -o playwav
> samiam@samiam-home-pc:~/projeclufthansa check reservationts/petit_ami$ ./playwav
> alsaplaywave: rate: 1
> alsaplaywave: rate: 0
> samiam@samiam-home-pc:~/projects/petit_ami$ gcc -g3 -Iinclude linux/playwav.c ./test2.o -lasound -o playwav
> samiam@samiam-home-pc:~/projects/petit_ami$ ./playwav
> alsaplaywave: rate: 1
> alsaplaywave: rate: 1
Note that in both the "succeeds" case and the "fails" case carry the -lasound option. It would not compile
if not.
Scott Franco
--------- Original Message --------- Subject: Re: Serious bug calling ALSA lib functions from .so vs. .o file
From: "Jaroslav Kysela" <perex@xxxxxxxx>
Date: 6/18/19 11:18 am
To: "scott andrew franco" <samiam@xxxxxxxxxxxx>
Cc: alsa-devel@xxxxxxxxxxxxxxxx
Dne 18. 06. 19 v 17:16 scott andrew franco napsal(a):
> Hello,
>
> The issue:
>
> calling across .so (dynamic linking) produces different behavior in ALSA library calls than normal,
> and causes ALSA to malfunction.
>
> I have minimized my issue to as small a code section as possible. Note the code is from commonly available
> internet examples, including Free Electrons "Audio in embedded linux systems".
It seems like a linker issue. I think that you must link .so to -lasound, too.
(add -lasound to the command producing your .so library to satisfy the
versioned symbols)
Jaroslav
>
> The run is:
>
> samiam@samiam-home-pc:~/projects/petit_ami$ gcc -fPIC -c test2.c -o test2.o
> samiam@samiam-home-pc:~/projects/petit_ami$ gcc -g3 -Iinclude linux/playwav.c ./test2.so -lasound -o playwav
> samiam@samiam-home-pc:~/projeclufthansa check reservationts/petit_ami$ ./playwav
> alsaplaywave: rate: 1
> alsaplaywave: rate: 0
> samiam@samiam-home-pc:~/projects/petit_ami$ gcc -g3 -Iinclude linux/playwav.c ./test2.o -lasound -o playwav
> samiam@samiam-home-pc:~/projects/petit_ami$ ./playwav
> alsaplaywave: rate: 1
> alsaplaywave: rate: 1
>
> Note the only difference between the two runs of "playwave" is if the second module, test2, is linked as a .so or
> linked as a .o.
>
> The code is:
> =======================================================================================
> playwav.c
> #include <alsa/asoundlib.h>
> #include <stdio.h>
> void alsaplaywave1(void)
> {
> snd_pcm_t *pcm_handle;
> snd_pcm_hw_params_t *params;
> unsigned int val;
> unsigned int rate;
> int r;
> /* open pcm device */
> r = snd_pcm_open(&pcm_handle, "default", SND_PCM_STREAM_PLAYBACK, 0);
> if (r < 0) printf("Cannot open PCM output device");
> snd_pcm_hw_params_alloca(¶ms); /* get hw parameter block */
> snd_pcm_hw_params_any(pcm_handle, params);
> r = snd_pcm_hw_params_set_access(pcm_handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
> if (r < 0) printf("Cannot set interleaved mode");
> r = snd_pcm_hw_params_set_format(pcm_handle, params, SND_PCM_FORMAT_S16_LE);
> if (r < 0) printf("Cannot set format");
> r = snd_pcm_hw_params_set_channels(pcm_handle, params, 2);
> if (r < 0) printf("Cannot set channels number");
> val = 44100;
> r = snd_pcm_hw_params_set_rate_near(pcm_handle, params, &rate, 0);
> if (r < 0) printf("Cannot set rate");
> snd_pcm_hw_params_get_rate(params, &rate, 0);
> printf("alsaplaywave: rate: %d\n", rate);
> snd_pcm_close(pcm_handle);
> }
> extern void alsaplaywave2(void);
> int main(int argc, char **argv)
> {
> alsaplaywave1();
> alsaplaywave2();
> return 0;
> }
>
> ==================================================================================
> test2.c
> #include <alsa/asoundlib.h>
> #include <stdio.h>
> void alsaplaywave2(void)
> {
> snd_pcm_t *pcm_handle;
> snd_pcm_hw_params_t *params;
> unsigned int val;
> unsigned int rate;
> int r;
> /* open pcm device */
> r = snd_pcm_open(&pcm_handle, "default", SND_PCM_STREAM_PLAYBACK, 0);
> if (r < 0) printf("Cannot open PCM output device");
> snd_pcm_hw_params_alloca(¶ms); /* get hw parameter block */
> snd_pcm_hw_params_any(pcm_handle, params);
> r = snd_pcm_hw_params_set_access(pcm_handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
> if (r < 0) printf("Cannot set interleaved mode");
> r = snd_pcm_hw_params_set_format(pcm_handle, params, SND_PCM_FORMAT_S16_LE);
> if (r < 0) printf("Cannot set format");
> r = snd_pcm_hw_params_set_channels(pcm_handle, params, 2);
> if (r < 0) printf("Cannot set channels number");
> val = 44100;
> r = snd_pcm_hw_params_set_rate_near(pcm_handle, params, &rate, 0);
> if (r < 0) printf("Cannot set rate");
> snd_pcm_hw_params_get_rate(params, &rate, 0);
> printf("alsaplaywave: rate: %d\n", rate);
> snd_pcm_close(pcm_handle);
> }
> _______________________________________________
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> https://mailman.alsa-project.org/mailman/listinfo/alsa-devel
>
--
Jaroslav Kysela <perex@xxxxxxxx>
Linux Sound Maintainer; ALSA Project; Red Hat, Inc.
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