Re: Enhanced PLAY.CMD with play list support

From: Michael Baldwin Bruce (mbbruce_at_mighty.co.za)
Date: 11/06/04 

Date: 5 Nov 2004 19:07:17 -0800

bruce aka tholen@antispam.ham the troll wrote in message news:<kcxid.63356$Kl3.36017@twister.socal.rr.com>...
> Marty writes:
>
> >>>>>>> Peter Weilbacher wrote:
>
> >>>>>>>> Michael DeBusk wrote:
>
> >>>>>>>>> Marty wrote:
>
> >>>>>>> The most important aspect of the script for me was the resource leak
> >>>>>>> fix. Something inside of Timidity_MCD does not get cleaned up correctly
> >>>>>>> without the call to mciRxExit(). Strangely, missing this call doesn't
> >>>>>>> seem to cause a problem with any other codecs that I've found, just
> >>>>>>> Timidity. I guess the other codecs are smarter about cleaning up after
> >>>>>>> sloppy processes.
>
> >>>>>> I've probably encountered that problem. There have been times when
> >>>>>> PLAY.CMD would fail, while other audio file playing utilities would
> >>>>>> continue to work, including some Windows-based ones. But I have also
> >>>>>> encountered one instance in which PLAY.CMD worked from inside my editing
> >>>>>> program but failed as a standalone utility in another windowed command
> >>>>>> prompt session. Haven't been able to figure that one out.
>
> >>>>> Did you get any messages from PLAY about a "REXX Alias" or something to
> >>>>> that effect?
>
> >>>> Unfortunately, I cannot reproduce the problem on demand, so I have to
> >>>> rely on memory for the error message, but it was something like
> >>>> "mciRxInit routine not found".
>
> >>> I've seen this type of thing before with other DLLs (most prominently
> >>> REXXUTIL). The call to SysLoadFuncs returns a success, but REXX later
> >>> claims that the functions which were allegedly just loaded are not
> >>> available. I think this is a separate issue from the ones that I
> >>> addressed with my changes.
>
> >> A separate issue involving a bug in REXX?
>
> > I believe so, since I've seen it outside of the context of multimedia.
>
> >>>>> Instead of maintaining handles with binary data and the
> >>>>> like, in REXX you create a name or "alias" to associate with your MCI
> >>>>> device. Every multimedia operation you do in REXX is done against this
> >>>>> alias so that the state is maintained by the MMPM libraries, but your
> >>>>> context is maintained.
> >>>>>
> >>>>> Unfortunately, PLAY.CMD hard-codes a REXX alias name, so that if you
> >>>>> attempt to run two of them at the same time, the same alias is used and
> >>>>> the context is applied from your original running PLAY.CMD. Likewise,
> >>>>> if you fail to close out and free the REXX alias (due to killing the CMD
> >>>>> process, for example), its state continues to be maintained by the
> >>>>> system. The alias is global. You have to free up the REXX alias before
> >>>>> it can be used again.
>
> >>>> Well, that seems consistent with one occasion in which I was repeatedly
> >>>> playing a very short audio clip, pressing the key to repeat the play
> >>>> very rapidly, such that it seems possible the system started up the next
> >>>> PLAY before completely cleaning up after the previous PLAY.
>
> >>>>> My version of PLAY checks for this condition and performs the free if
> >>>>> needed.
>
> >>>> I'm not sure under what conditions my audio subsystem supports the
> >>>> simultaneous playing of two audio streams. I'm using the SoundMax
> >>>> audio built in to the Intel 850 motherboard (ADI 1885). I know that
> >>>> if I'm playing a .wav file, then fire up Mahjongg, it complains about
> >>>> the device being busy, even though I've disabled sounds in Mahjongg.
>
> >>> Even if your audio drivers are capable of audio stream mixing, an
> >>> application can still open the device in exclusive mode. What are you
> >>> using to play the wave file?
>
> >> I've tried a variety of different utilities over the years. I probably
> >> use PLAY.CMD more often than anything else. I like the simplicity of
> >> the command line interface.
>
> >>> I don't think the system will default to
> >>> opening it exclusively if you double-click a wave file or use PLAY.
>
> >> Well, Mahjongg complains that the device is locked when PLAY is
> >> being used to play a .wav file. When it's finished playing, no
> >> such message occurs. Maybe Mahjongg wants exclusive access and
> >> refuses to share, but then it shouldn't say that the device is
> >> locked when it really isn't (if that is indeed the case).
>
> > Usually when a program requests exclusive access, as long as no other
> > applications have the device open exclusively, it will be able to rip
> > the device away from programs that open it in shared mode.
>
> And I have encountered situations in which music being played was
> abruptly stopped while some other application took over, and then
> resumed when that other application was done, but I don't remember
> which applications were involved. In addition to PLAY.CMD, I have
> also utilized z! for .mp3 files and Internet audio streams.

those apps probably don't like Barnes either, bruce.

>
> >>>> On the other hand, the driver came with WarpMix, so it looks like it
> >>>> should be able to mix at least a few audio sources.
>
> >>> I think WarpMix is tool used to set volume levels of various audio
> >>> channels (CD, line in, etc.), correct?
>
> >> Yes.
>
> >>> That's different from stream
> >>> "mixing" where different audio streams are combined, in software or
> >>> hardware in some cases, into one stream that is played. The former
> >>> process is an analog one and only happens among a fixed set of channels.
> >>> The latter is digital and can take place over a variable number of
> >>> audio streams.
>
> >> Odd to use "mix" in the name if it doesn't do any mixing.
>
> > It does mixing of a sort, but I certainly never liked the fact that the
> > same word describes two very different scenarios. It can be quite
> > confusing.
>
> >>>>>> And I've
> >>>>>> also seen cases where the first invocation of PLAY.CMD would return an
> >>>>>> error message but a second invocation would cause the entire windowed
> >>>>>> command prompt session to die.
>
> >>>>> This one sounds like the MM libraries ran out of resources somewhere or
> >>>>> were left in an unstable state. This can be caused by not cleaning up
> >>>>> properly. I've seen this happen on my own system but haven't been able
> >>>>> to nail down the exact spark that triggered it. Since modifying
> >>>>> PLAY.CMD, I haven't seen this happen, but I'm not convinced that the
> >>>>> problem is gone yet.
>
> >>>>>> I'm curious as to how much digging others did to try and find the
> >>>>>> kinds of TIMEFMT supported by PLAY.CMD. I know it supports "samples"
> >>>>>> and "ms" (for milliseconds), but the default units are something like
> >>>>>> 15 samples (for 44.1 kHz audio). Haven't dug enough to understand
> >>>>>> the origin of that choice. Any other time formats supported?
>
> >>>>> 15 samples / 44100 samples/sec = .00034 secs of audio
>
> >>>>> Seems like a strange choice indeed. It is approximately 1/100th of a
> >>>>> normal process time slice (32ms). That's a bit of a stretch, though.
>
> >>>> Sufficiently odd that I never use the default (other than to
> >>>> investigate it), which means I always have to use the TIMEFMT keyword,
> >>>> which is a bit annoying. On the other hand, my editing program works
> >>>> with the audio samples in memory, so to play an edited clip using
> >>>> PLAY.CMD, I need to write it to disk, so as long as I write just the
> >>>> clip I want to hear, I don't have to bother with the FROM, TO, and
> >>>> TIMEFMT keywords. Would be nicer to simply play a clip of audio
> >>>> samples directly from memory and thus avoid hitting the disk every
> >>>> time I want to repeat a short clip, but then I'd have to learn an
> >>>> entirely new part of the multimedia system. One of the editing
> >>>> tools plays a short clip whose end points I can move in 0.01 sec
> >>>> increments using the cursor keys; it has to write a new .wav file
> >>>> every time I press the key so as to utilize PLAY.CMD.
>
> >>> The learning curve for the multimedia subsystem proper can be quite
> >>> steep. It would allow you the ability to play from memory, volume
> >>> control, and data format independence (you could use codecs to read and
> >>> write your raw audio data), but getting there is tricky.
>
> >> Reading .wav files is sufficiently simple that I can avoid the overhead
> >> of a codec. On the other hand, handling every possible data type is a
> >> headache for the source code. There are digital audio units becoming
> >> available that can record 24-bit samples. Judging from the recording
> >> time available, they're not wasting space by padding the samples to
> >> 32-bit. If I wanted to handle files like that, I'd need to rewrite
> >> the code to read three 8-bit samples, because there is no native 24-bit
> >> data type.
>
> > I may be mistaken, but I believe some sound cards support 24-bit samples
> > in hardware (which could be why such WAV files might be appearing). I
> > don't know if the OS/2 codec supports such a thing. I'd imagine not.
>
> I expect to be finding out over the next few months if reviews of the
> Edirol R-1 coax me into buying one.

lol!!!!! Wanker!

>
> > But allowing the use of a codec could open up other possibilities, such
> > as MP3 or Ogg-Vorbis.
>
> How about extracting the audio stream from a video DVD?

Which does about as much as pissing on your keyboard, bruce.

>
> >>> In what language is your main project written?
>
> >> I rely heavily on a library of routines that I've written over the last
> >> two decades, largely in Fortran. The Fourier transform is an example.
> >> A lot of the "magic" my editor can perform (fixing wrong notes, for
> >> example) uses the Fourier transform.
>
> > Understandable to hold on to such code. Certainly not the easiest thing
> > to code accurately in C.
>
> >>> I ask because I'm
> >>> trudging my way through the same territory right now in one of my C
> >>> projects, and maybe I can provide you with a simplified code snippet to
> >>> get the system going for you. I have DART and raw MMPM code from past
> >>> projects, but they were muddled down in the mechanics of synchronization
> >>> with a video stream, so they don't serve as good examples.
>
> >> They might make good examples to the WarpVision people. The only time
> >> I tried WarpVision, I never could get decent synchronization with the
> >> video, even with the adjustment keys provided.
>
> > I think the problem that Warpvision addresses is a problem of accuracy
> > within the header information, as much as its own timing accuracy. Some
> > video clips just plain have the wrong information for the playback speed
> > of audio versus video. Other players can adjust usually by sync'ing up
> > on "key" frames that occur periodically in the playback. At least the
> > sync problem with emulation (the source of my code) is one where the
> > sound has to be *created* at the right rate. The issue gets muddied
> > when there are skipped frames, but it's generally an easier problem
> > because it's more controllable.
>
> >>> Also, they
> >>> only dealt in pumping raw data out to the sound card. My new code will
> >>> need to be a bit more robust (being format-independent, for example).
>
> >> But API calls are API calls, regardless of the language used to call
> >> them.
>
> > True. Translation from one language to another is relatively trivial
> > when writing code based largely on API calls.

irrelevant and illogical, bruce.

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