This will be because the sample rate is wrong. solaris has a call to set this
"Nikolaus J. Sucher" wrote:
> Stefan,
>
> While what I said below is certainly true, I also found that after a reboot, artsd
>and knotify do start up, and
> the kde theme as well as the sound effects are playing. HOWEVER, the sound is like
>in extreme slow motion and I
> can't even recognize the sounds anymore. Any suggestions??
>
> Tx,
> Nikolaus
>
> "Nikolaus J. Sucher" wrote:
>
> > Hi Stefan,
> >
> > The problem describe below appears to be linked to knotify. It appears to crash on
>startup. Once I start it up
> > manually and then start up artsd, it appears to work. However, the sounds appear
>to be delayed and the system's
> > response is very, very slow. The testaudiosubsys is working fine.
> >
> > Nikolaus
> >
> > "Nikolaus J. Sucher" wrote:
> >
> > > Hi Stefan,
> > >
> > > So I finally got the sound to work on my Sparc Ultra 5! There is still a lot to
>be done though! Artsd does
> > > not seem to start up automatically on starting kde. When I start it manually as
>artsd, the sound effects
> > > work, when I start artswrapper they don't work. Do you have any suggestions
>what I might to have to look
> > > for???
> > >
> > > Tx again, things are looking up!
> > >
> > > Nikolaus
> > >
> > > Stefan Westerfeld wrote:
> > >
> > > > Hi!
> > > >
> > > > The routine that does float -> integer conversion is defined in convert.cc
> > > > and looks like this (it's called from Synth_PLAY):
> > > >
> > > > void Arts::convert_stereo_2float_i16le(unsigned long samples, float *left,
>float *right, unsigned char *to)
> > > > {
> > > > float *end = left+samples;
> > > > long syn;
> > > >
> > > > while(left < end)
> > > > {
> > > > syn = QRound((*left++)*32767);
> > > >
> > > > if(syn < -32768) syn = -32768; /* clipping */
> > > > if(syn > 32767) syn = 32767;
> > > >
> > > > *to++ = syn & 0xff;
> > > > *to++ = (syn >> 8) & 0xff;
> > > > [...]
> > > >
> > > > The signal as float is between -1.0 and 1.0 - after the multiplication
> > > > and conversion to integer it is between -32767 and 32767. You see that it
> > > > is signed there, i.e.
> > > >
> > > > 0x7fff (represents 1.0)
> > > > 0x7ffe
> > > > ...
> > > > 0x0001
> > > > 0x0000 (represents 0.0)
> > > > 0xffff
> > > > ...
> > > > 0x8002
> > > > 0x8001 (represents -1.0)
> > > >
> > > > Then, the *to++ lines take the least significant byte first, and then the
> > > > most significant byte - no further conversion is done. That means that
> > > > a value which is a tad bit larger than -1.0 would be written as
> > > >
> > > > fragment_buffer[i] = 0x02; fragment_buffer[i+1] = 0x80;
> > > >
> > > > This is called little endian because the number 0x1234 would be represented
> > > > as 0x34 0x12, that is, the endian gets little - it is confusing, because the
> > > > endian still holds the most significant byte.
> > > >
> > > > Cu... Stefan
> > > >
> > > > On Sat, Dec 02, 2000 at 12:29:07AM +0800, Nikolaus J. Sucher wrote:
> > > > > Stefan,
> > > > >
> > > > > Ok, after having read through the OSS programmers guide
>(http://www.opensound.com/pguide/oss.pdf),
> > > > > I am now convinced that the problems stem from the big endian nature of the
>Sparc while you send
> > > > > little endian data down to audiosubsystem. If I understand this right, you
>add the data from
> > > > > wBuffer to the fragment buffer. The type of the fragment buffer is char.
>Does this mean that you
> > > > > split the 16 bit data in two parts that you add to the fragment_buffer (i.e.
>big byte into
> > > > > fragment_buffer[i], small byte into fragment_buffer[i+1])? What is
>happening with sign? Can you
> > > > > elaborate on this, please?
> > > > >
> > > > > Tx,
> > > > > Nikolaus
> > > > >
> > > > >
> > > > > "Nikolaus J. Sucher" wrote:
> > > > >
> > > > > > Hi Stefan,
> > > > > >
> > > > > > I did what ;you suggested and compiled the testasubsystem program. Using
>this program, Ihave
> > > > > > ascertained that the audiosubsystem gets data, it reads the data, puts the
>data into the
> > > > > > fragment buffer and writes the data to the sound card. HOWEVER, for some
>mysterious reason, I
> > > > > > don't get the sine wave sound on my audio output but more like a
>hammering. I checked all the
> > > > > > parameters (encoding, sample rate, bits/sample etc), everything appears to
>be allright. Byte
> > > > > > swapping doesn't make any difference either. Do you have any suggestion or
>idea what might be
> > > > > > missing here???
> > > > > >
> > > > > > Have a great weekend,
> > > > > > Nikolaus
> > > > > >
> > > > > > Stefan Westerfeld wrote:
> > > > > >
> > > > > > > Hi!
> > > > > > >
> > > > > > > On Fri, Dec 01, 2000 at 12:55:26AM +0800, Nikolaus J. Sucher wrote:
> > > > > > > > Stefan Westerfeld wrote:
> > > > > > > >
> > > > > > > > > Hi!
> > > > > > > > >
> > > > > > > > > On Thu, Nov 30, 2000 at 11:00:25AM +0800, Nikolaus J. Sucher wrote:
> > > > > > > > > > > 2. reimplement handleIO(), call needMore() exactly once
> > > > > > > > > >
> > > > > > > > > > OK!
> > > > > > > > > >
> > > > > > > > > > > and write through
> > > > > > > > > > > the result to the audio device (the encoding aRts produces is
>signed,
> > > > > > > > > > > 16bit, little endian)
> > > > > > > > > >
> > > > > > > > > > I am very close. One more question: what is the name of the
>buffer where the data goes
> > > > > > > > > > that comes from the producer (producer->needMore();). How do I
>"write through"? I am
> > > > > > > > > > not sufe yet if I have to convert from little endian to bigendian
>or not. I will check
> > > > > > > > > > on that.
> > > > > > > > >
> > > > > > > > > producer->needMore will lead to a call of AudioSubSystem::write -
>the OSS
> > > > > > > > > implementation puts the data in wBuffer, and later (in handleIO)
>gets it
> > > > > > > > > from there and writes it to the audio device.
> > > > > > > > >
> > > > > > > > > So you can either keep that mechanism, then you can write what is in
> > > > > > > > > wBuffer after calling needMore, or reimplement AudioSubSystem::write
>to
> > > > > > > > > "write through" directly.
> > > > > > > >
> > > > > > > > Sorry, that I keep bothering about this but I am still learning...
> > > > > > > > I am stuck here. I don't know how to actually read what is in the
>wBuffer. I only get
> > > > > > > > click, click, click from my audio, no matter what I try. When I
>check the OSS
> > > > > > > > implementation, it appears as if I never get anything into the
>fragment_buffer except
> > > > > > > > zeroes????
> > > > > > > > I very much appreciate your help and patience!!
> > > > > > >
> > > > > > > What might be helpful is if you don't try to get artsd working at first,
>but
> > > > > > > testasubsys (which is in kdelibs/arts/examples - you need to do make
>check
> > > > > > > to get it compiled). It plays a 2 seconds sine wave. That should be
>suitable
> > > > > > > for testing.
> > > > > > >
> > > > > > > With artsd, the normal thing you should expect when you start it is
>getting
> > > > > > > zeros, and only if you use applications, like artsplay, kaiman,
>artsbuilder
> > > > > > > or similar, you'll get some sound.
> > > > > > >
> > > > > > > Cu... Stefan
> > > > > > > --
> > > > > > > -* Stefan Westerfeld, [EMAIL PROTECTED] (PGP!), Hamburg/Germany
> > > > > > > KDE Developer, project infos at http://space.twc.de/~stefan/kde *-
> > > >
> > > > Content-Description: Card for Nikolaus J. Sucher
> > > >
> > > > --
> > > > -* Stefan Westerfeld, [EMAIL PROTECTED] (PGP!), Hamburg/Germany
> > > > KDE Developer, project infos at http://space.twc.de/~stefan/kde *-
