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 – ..
Category : alsa
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: ..
I am using a USB sound card with 16 microphones on my Raspberry Pi (the device is the UMA 16). Currently, I am collecting all 16 channels data and using up to 8 of them by separating their contents programmatically for an application that needs real time audio input, according to a configuration file which ..
I have a buffer of length 1024 coming over the network filled with single channel audio data at 48000Hz in S32LE format. I’m able to play this back using SDL2 however I need change this to use the ALSA API. I’ve followed countless tutorials but all I’m getting is static. This is the setup code ..
I am trying to make a call using PJSUA2, play a wav file to the callee, and callee should record this file to wav. I created two programs for that (caller and callee). And when caller starts playing wav file into AudioMedia object, console of callee program just spams out "alsa.c pb_thread_func: underrun!" and nothing ..
I am writing a C++ project in which I want to get frequences from surrandings. It is based on Raspberry Pi. I run some code using Alsa to record sound to .pcm. I am stuck in getting data which I have to put in FFT function. My idea was to collect data: `snd_pcm_readi (capture_handle, buffer, ..
Here is code: if (!BASS_Init(-1, 44100, 0, NULL, NULL)) { return 0; } DWORD chan = BASS_StreamCreateFile(FALSE, argv[1] , 0, 0, BASS_SAMPLE_LOOP); BASS_ChannelPlay(chan, FALSE); After compiling and running program i get following error: ALSA lib pcm_dmix.c:1075:(snd_pcm_dmix_open) unable to open slave How can i fix it? Source: Windows Que..
Web Development Company
Recent Comments