bpa wrote: > my basic understanding. > > PCM Audio is sent via ALSA interface to ALSA device driver. > > Audio samples are put into a ring buffer. > The ring buffer size is controlled by the buffer parameter. > The ring buffer is processed in terms of periods (i.e there x periods in > the whole buffer). > A period the number of frames per h/w interrupts (edit: corrected from > first post) > I don't know how samples are fitted into frames. > > The audio samples can either be copied into buffer using API Read/write > or the buffer can be memory mapped into calling process memory so no > additional copying of audio data. > > These are all hardware paramaters and so valid values are dependent on > the audio device. > > edit: > > Found an explanation of frames and periods. > > > https://www.alsa-project.org/main/index.php/FramesPeriods > > edit 2: > > Working link to writing-an-alsa-driver.pdf > https://www.ibr.cs.tu-bs.de/theses/kurtisi/NMP-EmbeddedClient/Abgabe/Quellen/writing-an-alsa-driver.pdf
That's great and it makes a lot more sense now. Thanks very much! SB3 -> Quad 909 -> Quad Electrostatic speakers, Quad 405 -> TBI subwoofer 3 x SB3s + SB Radio LMS on Raspbian on a Pi4 Rotel RSP1068 surround processor, Quad 707 -> B&W surround ------------------------------------------------------------------------ rbl's Profile: http://forums.slimdevices.com/member.php?userid=4517 View this thread: http://forums.slimdevices.com/showthread.php?t=114868 _______________________________________________ plugins mailing list plugins@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/plugins
