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/* | |
* Simple sound playback using ALSA API and libasound. | |
* | |
* Compile: | |
* $ cc -o play sound_playback.c -lasound | |
* | |
* Usage: | |
* $ ./play <sample_rate> <channels> <seconds> < <file> | |
* | |
* Examples: | |
* $ ./play 44100 2 5 < /dev/urandom | |
* $ ./play 22050 1 8 < /path/to/file.wav | |
* | |
* Copyright (C) 2009 Alessandro Ghedini <[email protected]> | |
* -------------------------------------------------------------- | |
* "THE BEER-WARE LICENSE" (Revision 42): | |
* Alessandro Ghedini wrote this file. As long as you retain this | |
* notice you can do whatever you want with this stuff. If we | |
* meet some day, and you think this stuff is worth it, you can | |
* buy me a beer in return. | |
* -------------------------------------------------------------- | |
*/ | |
#include <alsa/asoundlib.h> | |
#include <stdio.h> | |
#define PCM_DEVICE "default" | |
int main(int argc, char **argv) { | |
unsigned int pcm, tmp, dir; | |
int rate, channels, seconds; | |
snd_pcm_t *pcm_handle; | |
snd_pcm_hw_params_t *params; | |
snd_pcm_uframes_t frames; | |
char *buff; | |
int buff_size, loops; | |
if (argc < 4) { | |
printf("Usage: %s <sample_rate> <channels> <seconds>\n", | |
argv[0]); | |
return -1; | |
} | |
rate = atoi(argv[1]); | |
channels = atoi(argv[2]); | |
seconds = atoi(argv[3]); | |
/* Open the PCM device in playback mode */ | |
if (pcm = snd_pcm_open(&pcm_handle, PCM_DEVICE, | |
SND_PCM_STREAM_PLAYBACK, 0) < 0) | |
printf("ERROR: Can't open \"%s\" PCM device. %s\n", | |
PCM_DEVICE, snd_strerror(pcm)); | |
/* Allocate parameters object and fill it with default values*/ | |
snd_pcm_hw_params_alloca(¶ms); | |
snd_pcm_hw_params_any(pcm_handle, params); | |
/* Set parameters */ | |
if (pcm = snd_pcm_hw_params_set_access(pcm_handle, params, | |
SND_PCM_ACCESS_RW_INTERLEAVED) < 0) | |
printf("ERROR: Can't set interleaved mode. %s\n", snd_strerror(pcm)); | |
if (pcm = snd_pcm_hw_params_set_format(pcm_handle, params, | |
SND_PCM_FORMAT_S16_LE) < 0) | |
printf("ERROR: Can't set format. %s\n", snd_strerror(pcm)); | |
if (pcm = snd_pcm_hw_params_set_channels(pcm_handle, params, channels) < 0) | |
printf("ERROR: Can't set channels number. %s\n", snd_strerror(pcm)); | |
if (pcm = snd_pcm_hw_params_set_rate_near(pcm_handle, params, &rate, 0) < 0) | |
printf("ERROR: Can't set rate. %s\n", snd_strerror(pcm)); | |
/* Write parameters */ | |
if (pcm = snd_pcm_hw_params(pcm_handle, params) < 0) | |
printf("ERROR: Can't set harware parameters. %s\n", snd_strerror(pcm)); | |
/* Resume information */ | |
printf("PCM name: '%s'\n", snd_pcm_name(pcm_handle)); | |
printf("PCM state: %s\n", snd_pcm_state_name(snd_pcm_state(pcm_handle))); | |
snd_pcm_hw_params_get_channels(params, &tmp); | |
printf("channels: %i ", tmp); | |
if (tmp == 1) | |
printf("(mono)\n"); | |
else if (tmp == 2) | |
printf("(stereo)\n"); | |
snd_pcm_hw_params_get_rate(params, &tmp, 0); | |
printf("rate: %d bps\n", tmp); | |
printf("seconds: %d\n", seconds); | |
/* Allocate buffer to hold single period */ | |
snd_pcm_hw_params_get_period_size(params, &frames, 0); | |
buff_size = frames * channels * 2 /* 2 -> sample size */; | |
buff = (char *) malloc(buff_size); | |
snd_pcm_hw_params_get_period_time(params, &tmp, NULL); | |
for (loops = (seconds * 1000000) / tmp; loops > 0; loops--) { | |
if (pcm = read(0, buff, buff_size) == 0) { | |
printf("Early end of file.\n"); | |
return 0; | |
} | |
if (pcm = snd_pcm_writei(pcm_handle, buff, frames) == -EPIPE) { | |
printf("XRUN.\n"); | |
snd_pcm_prepare(pcm_handle); | |
} else if (pcm < 0) { | |
printf("ERROR. Can't write to PCM device. %s\n", snd_strerror(pcm)); | |
} | |
} | |
snd_pcm_drain(pcm_handle); | |
snd_pcm_close(pcm_handle); | |
free(buff); | |
return 0; | |
} |
Question: Will this code run if the target device is a mixer (dmix)?
Question: Will this code run if the target device is a mixer (dmix)?
Yep, it seems helpuf to use with Soundpipe.
Can confirm the code of OP is a quite good way to use alsa pcm.
I dont' like the idea of pcm
being an unsigned int
when later, we have pcm = snd_pcm_open(...
.
snd_pcm_open
can return a negative number, so we should have int pcm;
This worked, This is a great start for the beginners.
This is the first working program i found online.
Please share a program to capture the audio as well.
Thank you.
Where in the program the data read from the .wav file ?
Where in the program the data read from the .wav file ?
It reads from the standard input stream (stdint = 0):
if (pcm = read(0, buff, buff_size) == 0) {
printf("Early end of file.\n");
return 0;
}
Where stdint is the content of .wav file you pass using <
character in command line (./play 22050 1 8 < /path/to/file.wav)
did anyone encounter the same problem as me? the playback quit after 3-4s with "Early end of file." Tried different files:
//./playwave 32000 1 20 /audio/11_en.wav...loops=1932/2500, tmp=8000, quit after 4s
//./playwave 16000 1 20 /audio/test1.raw...loops=2007/2500, tmp=8000, quit after 3s
I modified a little bit to allow taking argv[4] as filename.
Ps. Remove 'return 0' will finish the broadcast
Thank you for this.
if (pcm = snd_pcm_writei(pcm_handle, buff, frames) == -EPIPE) {
printf("XRUN.\n");
In this code won't it always be a buffer overrun if you are trying to write frames out and can't?
The application can run in command line, but when added to startup script (eg. rc.local) so after power on it can run automatically, it crashes. The core dump file show it crashed at following:
root@linaro-ubuntu-desktop:/test# gdb ./playback /opt/core.playback.6238.1604316560.6
GNU gdb (Ubuntu/Linaro 7.3-0ubuntu2) 7.3-2011.08
Copyright (C) 2011 Free Software Foundation, Inc.
License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html>
This is free software: you are free to change and redistribute it.
There is NO WARRANTY, to the extent permitted by law. Type "show copying"
and "show warranty" for details.
This GDB was configured as "arm-linux-gnueabi".
For bug reporting instructions, please see:
<http://bugs.launchpad.net/gdb-linaro/>...
Reading symbols from /test/playback...(no debugging symbols found)...done.
[New LWP 6238]
[Thread debugging using libthread_db enabled]
Core was generated by `/test/playback'.
Program terminated with signal 6, Aborted.
#0 0x2ad93ed6 in ?? () from /lib/arm-linux-gnueabi/libc.so.6
(gdb) bt full
#0 0x2ad93ed6 in ?? () from /lib/arm-linux-gnueabi/libc.so.6
No symbol table info available.
#1 0x2ada20da in raise () from /lib/arm-linux-gnueabi/libc.so.6
No symbol table info available.
#2 0x2ada4506 in abort () from /lib/arm-linux-gnueabi/libc.so.6
No symbol table info available.
#3 0x2ad9d1ec in __assert_fail () from /lib/arm-linux-gnueabi/libc.so.6
No symbol table info available.
#4 0x2ad1fb72 in snd_pcm_hw_refine () from /usr/lib/arm-linux-gnueabi/libasound.so.2
No symbol table info available.
#5 0x00008cd4 in main ()
No symbol table info available.
Note snd_pcm_hw_refine() is not directly called by this application.
I have hardcoded all parameters to 32000hz / 1chn / 30s and "/opt/test1.wav". The read changed to
int fd = open("/opt/test1.wav", O_RDONLY); ///added
for (loops = (seconds * 1000000) / tmp; loops > 0; loops--) {
if (pcm = read(fd, buff, buff_size) == 0) { ///changed to fd from 0
printf("Early end of file.\n");
return 0;
}
Further tracing found it crashed at snd_pcm_hw_params_any() after snd_pcm_open(), snd_pcm_hw_params_alloca().
Note: The application only crashes after auto-run from power on; if run the script in putty (command line), no problem.
Anyone encounter this problem? How have you solved it? Thank you.
Got another issue, XRUN in alternative cycle (this would happen after the playback from 2nd time onwards):
2020-11-02 21:03:42 [audio]XRUN:0,1 --- cycle 1, 0=previous is ok
2020-11-02 21:03:42 [audio]XRUN:0,3 --- cycle 3, 0=previous is ok
2020-11-02 21:03:42 [audio]XRUN:0,5 --- cycle 5, 0=previous is ok
2020-11-02 21:03:42 [audio]XRUN:0,7
2020-11-02 21:03:42 [audio]XRUN:0,9
The effect of XRUN is heard either partially or no sound.
Looks easier than I thought
./play 44100 2 2 < hello.wav
Just gives out a very short click sound and then Im getting early end of file. Can anyone help ?
pi@raspberrypi:~/tinyalsa $ ./play 44100 2 2 < hello.wav
PCM name: 'default'
PCM state: PREPARED
channels: 2 (stereo)
rate: 44100 bps
seconds: 2
Early end of file.
@rsingh2083 The click sound is probably because it plays the metadata of the wav file. In a real player you would need to skip the header of the wav file.
How to play 24-bit audio?
How to play 24-bit audio?
You can use SND_PCM_FORMAT_S24_LE instead of SND_PCM_FORMAT_S16_LE
It's nice to see an attempt at a minimal ALSA program. There are some bugs that should probably be cleaned up, though. Examples:
-
&rate
should be justrate
, -
variable
pcm
should be callederr
as in the ALSA docs, -
the audio file should be played to the end instead of for a certain number of seconds,
-
buff_size should not presume sample width is 2 bytes. For calculating buff_size, one can use
_physical_width
snd_pcm_format_t sampfmt; snd_pcm_hw_params_get_format(params, &sampfmt); printf("sample format: %s\n", snd_pcm_format_description(sampfmt)); snd_pcm_format_t sampwidth = snd_pcm_format_physical_width(sampfmt); printf("sample width: %d bits\n", sampwidth);
Hey! Thanks for sharing. It does run flawless but i'm running into memory corruption when wrapping logic into functions. I probably am thinking about it wrong but i don't see a problem with my code, but i want to rule out ALSA is doing something odd. If anyone could take a look it would be greatly appreciated.
https://gist.github.com/dedobbin/29adb1dae10932b4a88721bb00a1fc45
edit: oh, already found it. very silly, i used snd_pcm_hw_params_alloca which allocated on the stack obviously
should have used snd_pcm_hw_params_malloc
I have modified the code to play wav files instead of stdin, I find that the wav files are played at a faster tempo. The problem is described here https://stackoverflow.com/questions/77920231/wav-files-are-played-by-alsa-at-a-faster-tempo-is-there-a-way-to-fix-the-tempo
Could anyone kindly point me my mistake?
./play 44100 2 2 < hello.wav Just gives out a very short click sound and then Im getting early end of file. Can anyone help ?
pi@raspberrypi:~/tinyalsa $ ./play 44100 2 2 < hello.wav PCM name: 'default' PCM state: PREPARED channels: 2 (stereo) rate: 44100 bps seconds: 2 Early end of file.
I know I'm extremely late but since no one has replied to you, I figured I could at least answer this qn.
The reason no sound is played when it finishes reading early is because we return as well after finish reading, which destroys everything. Whereas what we really want is just to break out of the for loop and stop reading and writing.
So arnd line 108, inside the if(pcm = read(0...))
Replace return 0; to break;
Oh I just realised someone else alr found the issue but it was commented b4 you
Я змінив код для відтворення файлів wav замість stdin, я виявив, що файли wav відтворюються у швидшому темпі. Проблема описана тут https://stackoverflow.com/questions/77920231/wav-files-are-played-by-alsa-at-a-faster-tempo-is-there-a-way-to-fix-the -темп
Хтось може вказати мені на мою помилку?
Your code is fine and it works fine for me, so I can conclude that something is wrong with the program startup parameters.
For your wav file, the startup command should be as follows:
./<program name> 22050 1 30 PinkPanther30.wav default
Edit:
I would recommend that you do not skip the first 44 bytes of the metadata. Some files have more than 44 bytes of metadata if there is some additional information. You need to search for the keyword “data” + 4 bytes and after that only the audio data comes.
I tried using this code it works for /dev/urandom but it doesn't work for a wave file.
I tried using fseek(fp,44, SEEK_SET); but it returns core dumped.
I am just a beginner at programming with C and I might have misused the fseek if anyone can tell me how to use it or how can I make it work for wave file, thanks.