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? >
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