My app plays raw PCM audio data through various channels using ALSA. I’m allocating a new audio channel by using snd_pcm_open(), then setting the PCM format via the snd_pcm_hw_params_xxx() calls and finally feeding raw PCM audio data to ALSA by using the snd_pcm_writei() API. This is all working fine so far but I haven’t found any way to tell ALSA
Tag: audio
ALSA vs PulseAudio – Latency Concerns
Good day, I have been debating some details with a colleague about ALSA vs PulseAudio, and need some help coming to a conclusion with it. It’s to my understanding that ALSA is relatively low-level, and talks directly to the hardware, while PulseAudio sits on top of ALSA as a service. Additionally, it’s to my understanding that ALSA is tied to
ALSA: Ways to prevent underrun for speaker
I am playing a single channel audio in non-interleaved mode. I am getting underrun when I am writing audio data into speaker : ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred Here is how I write: What are the different ways/parameter configurations to prevent ALSA under run ? (I am using Linux 3.0, ARM ) Edit: Here is a buffer measurement using snd_pcm_avail()