On 02/06/2014 12:36 AM, Vesa wrote: > On 02/06/2014 12:31 AM, Tobias Doerffel wrote: >> Hi, >> >> I don't think there's a bug but simple loss of precision when doing >> the conversions. Furthermore the sample is 48 KHz so there are >> situations where LMMS operates at 44,1 KHz and thus has to downsample >> the sample using libsamplerate. We possibly loose data here as well. I >> can't think of an easy fix as long as we have different sample rates >> in the audio backends, the samples and while rendering. >> >> When using JACK, the internally rendered float buffers get passed to >> JACK without any conversions. In the PulseAudio backend clipping >> happens if samples are outside of allowed range [-1,1]. When using >> ALSA, currently a conversion to 16 bit integer happens. As ALSA seems >> to support SND_PCM_FORMAT_FLOAT as well, maybe we should try to switch >> to that. However I fear we break things before the 1.0.0 release so >> maybe we should wait with that change until 1.0.0 is out? >> > Does the selected backend affect rendering also?
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