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? Toby ------------------------------------------------------------------------------ Managing the Performance of Cloud-Based Applications Take advantage of what the Cloud has to offer - Avoid Common Pitfalls. Read the Whitepaper. http://pubads.g.doubleclick.net/gampad/clk?id=121051231&iu=/4140/ostg.clktrk _______________________________________________ LMMS-devel mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/lmms-devel
