I am exploring sound generation using C++ in Ubuntu Linux. Here is my code: #include <iostream> #include <cmath> #include <stdint.h> #include <ncurses.h> //to compile: make [file_name] && ./[file_name]|aplay int main() { initscr(); cbreak(); noecho(); nodelay(stdscr, TRUE); scrollok(stdscr, TRUE); timeout(0); for ( int t=0;; t++ ) { int ch = getch(); if (ch == ‘q’) { ..
Category : alsa
So Im trying to play a wave file on raspberry pi in C++ with this code #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: ..
I am working on a Game Library for C++, now I am on audio part. After a lots of topic read, I obtained for using PortAudio with libsndfile. Header: #ifndef AUDIO_H #define AUDIO_H #include "sndfile.h" #include "portaudio.h" #include <iostream> #define FRAMES_PER_BUFFER (512) typedef struct { SNDFILE *file; SF_INFO info; } callback_data_s; static int callback( const ..
I am following the sample code here to generate a pure tone sinewave using ALSA API in C++ in Linux: http://equalarea.com/paul/alsa-audio.html The code I am using is under "A Minimal Interrupt-Driven Program". The code as is does not compile and my modified version of the code that actually compiles makes some sound is given below: ..
I am using a raspberry pi zero W with the latest raspberry pi OS I have installed both libasound2 and libasound2-dev, and this code that I got off the raspberry pi forums for testing the alsa library compiles and runs correctly: #include <stdio.h> #include <stdlib.h> #include <alsa/asoundlib.h> #include <alsa/pcm.h> int main() { int val; printf("ALSA ..
The question says it all. I am going in circles here. I set snd_pcm_sw_params_set_stop_threshold to boundary (and zero too just for fun) and I am still getting buffer underrun errors on snd_pcm_writei. I cannot understand why. The documentation is pretty clear on this: If the stop threshold is equal to boundary (also software parameter – ..
i use three mics to link a SOC codec(i2s_8ch)，i use cammand: "arecord -D hw:0,0 -c 4 -r 44100 -f S16_LE test.wav" two mics work well. when i use cammand: "arecord -D hw:0,0 -c 4 -r 44100 -f S16_LE –period-size=1024 –buffer-size=4096 test.wav" all the mics work. when i set sample rate 16k except period-size and buffer-size, ..
My environment is Raspberry Pi 3B/Raspberry Pi OS, and I’m building remotely from Visual Studio 2019 on Windows 10. I want to play PCM audio decoded by FFmpeg in ALSA(C language API). But this code returns Underrun error(EPIPE) at "snd_pcm_writei" and plays a cracky sound. Probably the way the resources are managed is wrong, but ..
I am currently trying to make the following code work, however when trying to get access to the device with portaudio the function I use a linux computer, and the cards use ALSA. And both code use the same version of portaudio The following code which is pa_dev.c is provided with portaudio. This code has ..
I am new to alsa, pulseAudio, needed help with this issue. This is the truncated output of pacmd command on my system pacmd list-sources: name: <alsa_input.pci-0000_00_1f.3.analog-stereo> properties: alsa.resolution_bits = "16" device.profile.name = "analog-stereo" device.profile.description = "Analog Stereo" device.description = "Built-in Audio Analog Stereo" module-udev-detect.discovered = "1" device.icon_name = "audio-card-pci" I want to fetch the field: ..
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