Record & Play Audio on Linux

Audio on Linux

Record & Play Audio on Command Line

$ aplay -h
Usage: aplay [OPTION]... [FILE]...
-h, --help help
--version print current version
-l, --list-devices list all soundcards and digital audio devices
-L, --list-pcms list device names
-D, --device=NAME select PCM by name
-q, --quiet quiet mode
-t, --file-type TYPE file type (voc, wav, raw or au)
-c, --channels=# channels
-f, --format=FORMAT sample format (case insensitive)
-r, --rate=# sample rate
-d, --duration=# interrupt after # seconds
-M, --mmap mmap stream
-N, --nonblock nonblocking mode
-F, --period-time=# distance between interrupts is # microseconds
-B, --buffer-time=# buffer duration is # microseconds
--period-size=# distance between interrupts is # frames
--buffer-size=# buffer duration is # frames
-A, --avail-min=# min available space for wakeup is # microseconds
-R, --start-delay=# delay for automatic PCM start is # microseconds
(relative to buffer size if <= 0)
-T, --stop-delay=# delay for automatic PCM stop is # microseconds from xrun
-v, --verbose show PCM structure and setup (accumulative)
-V, --vumeter=TYPE enable VU meter (TYPE: mono or stereo)
-I, --separate-channels one file for each channel
-i, --interactive allow interactive operation from stdin
-m, --chmap=ch1,ch2,.. Give the channel map to override or follow
--disable-resample disable automatic rate resample
--disable-channels disable automatic channel conversions
--disable-format disable automatic format conversions
--disable-softvol disable software volume control (softvol)
--test-position test ring buffer position
--test-coef=# test coefficient for ring buffer position (default 8)
expression for validation is: coef * (buffer_size / 2)
--test-nowait do not wait for ring buffer - eats whole CPU
--max-file-time=# start another output file when the old file has recorded
for this many seconds
--process-id-file write the process ID here
--use-strftime apply the strftime facility to the output file name
--dump-hw-params dump hw_params of the device
--fatal-errors treat all errors as fatal
Recognized sample formats are: S8 U8 S16_LE S16_BE U16_LE U16_BE S24_LE S24_BE U24_LE U24_BE S32_LE S32_BE U32_LE U32_BE FLOAT_LE FLOAT_BE FLOAT64_LE FLOAT64_BE IEC958_SUBFRAME_LE IEC958_SUBFRAME_BE MU_LAW A_LAW IMA_ADPCM MPEG GSM SPECIAL S24_3LE S24_3BE U24_3LE U24_3BE S20_3LE S20_3BE U20_3LE U20_3BE S18_3LE S18_3BE U18_3LE U18_3BE G723_24 G723_24_1B G723_40 G723_40_1B DSD_U8 DSD_U16_LE DSD_U32_LE DSD_U16_BE DSD_U32_BE
Some of these may not be available on selected hardware
The available format shortcuts are:
-f cd (16 bit little endian, 44100, stereo)
-f cdr (16 bit big endian, 44100, stereo)
-f dat (16 bit little endian, 48000, stereo)
arecord -d 10 -r 48000 -c 2 -f S16_LE audio.wav   # or arecord -d 10 -f dat audio.wav
arecord -v -f dat - | aplay -v -Vstereo -
arecord -v -D plug:dsnoop:0 -f dat -V stereo /dev/null
aplay -v -D plug:dmix:0 music.wav

Record & Play Audio in the Code

#include <stdio.h>
int main(int argc, char **argv) {
char *cmd = "arecord -D plughw:0 -f S16_LE -c 1 -r 16000 -t raw -q -";
char buf[256];
FILE *fp = popen(cmd, "r");
for (int i=0; i<16; i++) {
int result = fread(buf, 1, sizeof(buf), fp);
printf("read %d bytes\n", result);
}
pclose(fp);
return 0;
}
#include <stdio.h>
int main(int argc, char **argv) {
char *cmd = "aplay -D plughw:0 -f S16_LE -c 1 -r 16000 -t raw -q -";
char buf[256] = {0,};
FILE *fp = popen(cmd, "r");
for (int i=0; i<16; i++) {
int result = fwrite(buf, 1, sizeof(buf), fp);
printf("write %d bytes\n", result);
}
pclose(fp);
return 0;
}
import subprocess
import audioop
cmd = ['arecord', '-D', 'plughw:0', '-f', 'S16_LE', '-c', '1', '-r', '16000', '-t', 'raw', '-q', '-']
process = subprocess.Popen(cmd, stdout=subprocess.PIPE)
for i in range(16):
audio = process.stdout.read(160 * 2)
print(audioop.rms(audio, 2))
process.stdout.close()
process.terminate()
"""PyAudio example: Record a few seconds of audio and save to a WAVE file."""import pyaudio
import wave
CHUNK = 1024
FORMAT = pyaudio.paInt16
CHANNELS = 2
RATE = 44100
RECORD_SECONDS = 5
WAVE_OUTPUT_FILENAME = "output.wav"
p = pyaudio.PyAudio()stream = p.open(format=FORMAT,
channels=CHANNELS,
rate=RATE,
input=True,
frames_per_buffer=CHUNK)
print("* recording")frames = []for i in range(0, int(RATE / CHUNK * RECORD_SECONDS)):
data = stream.read(CHUNK)
frames.append(data)
print("* done recording")stream.stop_stream()
stream.close()
p.terminate()
wf = wave.open(WAVE_OUTPUT_FILENAME, 'wb')
wf.setnchannels(CHANNELS)
wf.setsampwidth(p.get_sample_size(FORMAT))
wf.setframerate(RATE)
wf.writeframes(b''.join(frames))
wf.close()

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