malc,
Other than the lack of error handling and blocking mode, what is the
problem with using QEMU's VM clock as the audio clock source? In my
experience from older projects using esd (not related to QEMU), it is
almost impossible to get reliable timing from ESD, especially if you are
transporting over a network. The timing will certainly not be accurate
enough for example for Windows guests using Windows Media Player, etc.
Plus, you have to factor in socket buffering and underruns due to
network latency if you expect anything synchronous on the socket. So
what is the reason for objecting to the QEMU vm clock, much like the
mute audio driver uses? I just want to know your reasoning for it just
to further my own understanding.
Thank you,
Leo Reiter
P.S. I can attest that using the mute audio driver you used, if the host
clock resolution is well above 100Hz such as on Linux and FreeBSD,
Windows Media Player guests have no problem with multimedia streams. I
keep referring to WMP because it is extremely picky about timing in
general, more so than other applications even. It seems that QEMU's vm
clock is accurate enough for this task, just FYI.
malc wrote:
<snip>
There are several problems with this patch:
a. esd_play_stream returns negative values on error (not zero)
b. esd_play_stream can return with EINTR
c. no error checking whatsoever on write to the socket
d. the socket is most likely opened in blocking mode
e. i don't think it's wise to use anything other than esd itself
as a audio-clock source
So basically the only way it can work (given the circumstances)
is to have a thread that would perform a blocking write to the
esd, and the rest as per coreaudio and sdladuio.
P.S. Oh yeah, mixing tabs and spaces is also not a great idea.
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Leonardo E. Reiter
Vice President of Product Development, CTO
Win4Lin, Inc.
Virtual Computing that means Business
Main: +1 512 339 7979
Fax: +1 512 532 6501
http://www.win4lin.com
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