[LAU] What's the point of that? - Was: rtirq

As usual madorf is spouting crap. Sync-to-timecode is complex and many DAWs don't do as good a job as some others. The accuracy of the sync has ABSOLUTELY NOTHING to do with the speed of MIDI transmission, since all such synchronization requires a DLL/PLL anyway. Ardour can sync to an MTC or LTC source within 1 sample of the master, but does have the defect that for MTC it does not take the input latency from the master into consideration (this is in the process of being fixed). Problems with synchronization are problems with synchronization NOT problems with MIDI (in general). There are additional problems, however, with madorf's claims. Recording while synced to a timecode source is problematic UNLESS you know for sure that the timecode source and the audio clock being used to drive the DAW share the same word clock. I know of no DAW that can accurately record audio if this is not the case, because the time periods required for varispeeding to track the master are too big for audio (they may even make a difference with just MIDI data). The whole design of synchronization MUST smooth out jitter in the incoming timecode signal and this necessarily means ignoring (for now) minor variations in the apparent speed of the master. If you tried to throw away or add extra samples in an attempt to precisely track the master, you'd just be going crazy and accomplishing nothing. On Thu, Mar 19, 2015 at 3:37 PM, Len Ovens <len(a)ovenwerks.net> wrote: > On Thu, 19 Mar 2015, Ralf Mardorf wrote: > > On Thu, 19 Mar 2015 09:07:34 -0700 (PDT), Len Ovens wrote: >> >>> So I have to ask myself if what you are hearing is just the >>> effects of a slow standard MIDI transport before the info even >>> gets to the computer. >>> >> >> You can do an experiment, assumed you still own old computers and >> tapes. >> >> Tape synced to the Computer by SMPTE (Atari) or by Click (C64). >> Record a MIDI synth and after that record the same synth on another >> tape track. This will double the synth sound, all you get is a >> phasing, that doesn't move. >> >> Do the same with a Linux or Windows PC. Record a track with >> Qtractor or Cubase and after that record the same external synth on >> another Qtractor or Cubase track. Sounds do not start at the same >> time, there's always audible shift, comparable to slow early >> reflections and the phasing is moving. >> > > That is the first explanation of this that makes sense to me. Thank > you. I do not know if it is possible to fix this in Linux or at > least in the sequencing SW we have. In a machine that only deals > with MIDI (Atari or C64), each midi event has it's own time stamp or > possition based on the OS clock (whatever the sequencing program is > using for a time base). In a machine that deals with audio, that > time base is an audio buffer length which may contain more than one > midi event, but may not contain all midi events that are meant to be > together. Not only that, but when the same midi goes in a cycle, > there is no guarentee that the events that were within one buffer > length with again remain within the same buffer length and this may > go in and out of sync as midi and it's time signature may form a > beat with the audio media clock. This would be the moving phase you > hear. > > Perhaps setting jack up for 16/3 at 48k would solve that. 16 samples > seems to be close to one MIDI byte and most events are three > bytes... though with a chord running status would take 9 bytes and > make seven or even possibly 6 (I don't remember if active sensing > resets running status). In any case each midi byte should be aligned > with the number of samples that best fits that one byte. I don't > know is 16/2 would be better or not (read I don't wish to spend the > brain power thinking about it). > > It's not a limitation of MIDI. >> > > It is some of both. A faster MIDI would not solve things unless Linux > audio was done differently. Assuming time stamping had to be done at > each byte would fix this. Using media clock still seems like the > right thing to do because in general, the media clock goes with a > project from computer to computer. Basically, you are telling me > that an audio buffer of 128 samples is too long for good MIDI sync. > This can be fixed in two ways: Use a short audio buffer or decouple > midi from the audio buffer completely and run midi processing and > time stamping separately. The first can be done by anyone who has a > machine that can run at 16/2 or 16/3 (maybe even 32/2 would be ok) > xrun free. The second would require redoing the sequencer SW and > possibly the ALSA midi drivers (I don't know enough about them to > say). > > However, I wont discuss this again, I just want to know if RTC is >> needed in the rtirq config for audio (ALSA/jackd), assumed jackd >> doens't start with "--clocksource OR -c h(pet)". And should rtirq >> config include an entry for HRTIMER/HPET? Is it possible to add >> HRTIMER/HPET to the rtirq config (e.g. in addition to RTC) and what >> is the name of such an entry? >> > > I can't answer if RTC is needed. HRTIMER is probably snd_hrtimer, > but I don't know if elevating the priority of that alone would help > because of all we discused above. Also the priority of > snd_mpu401_uart and the snd_seq* group of modules may suffer as > well... But none of that matters if the midi read/write > clock/timestamp is related to the audio buffer which is how any jack > connected application would do things. Jack does allow a midi port > to set which sample a midi event belongs to, but does that timing > make it past jack? Do sequencers use this possibility or just send > all the midi stuff right now for each graph knowing there are lots > of delays in there anyway? > > Maybe those fussy people who decided aes67 should support at least > 1ms latency are right. > > > -- > Len Ovens > www.ovenwerks.net > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user(a)lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user >