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This is necessary for uaudio devices, for instance to start aucat
before the device is plugged. Or to unplug a device whithout
having to restart aucat when another device is plugged. This is
controlled with the new -a option.
Allow multiple audio devices to be used concurently, i.e.
multiple ``-f devname'' options to be used; -f options must follow
per-device options, which is what we do for other options.
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- make the ``-m mode'' option per subdevice, allowing
subdevices to be play-only or rec-only even if the
server is full-duplex
- add ``monitoring'' mode (with ``-m mon''). This is a
record-only stream from which played streams can be
recorded (kind of ``record what you hear'').
- allow MIDI devices to be subscribed to the controlling
MIDI port of the server, ie what midicat does (with -f
option), but using the -q option.
- add flow control to the protocol, and serialize clock
ticks (sio_onmove() calls) and data chunks. This should
fix certain full-duplex programs, broken with ``magic''
block/buffer size combinations.
- enable 3 block latency which is the minimum theoretical.
Make -z and -b options correspond to device parameters.
- make sio_getcap(3) the same for aucat and sun backends,
ie return whatever is supported (``everything'' in the
aucat case, since everything is actulally supported).
aucat (player):
- enable ``-m mode'' option to select between monitoring
and recording when ``-o file'' is used.
- plug MIDI code to non-server codepath. The MIDI control
device is specified with the ``-q device'' option, as in
server mode.
- impliment lseek()'ing within files (controlled
through MIDI). Necessary to use aucat with a MIDI
sequencer.
midicat (thrubox):
- rename ``-f'' flag to ``-q'', so it has the
same name as in aucat (-f is still working)
ok jakemsr@, tweaks from jmc@
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otherwise the client would underrun by more than its own buffer
size, which currently is not allowed (causes the client to
deadlock).
Set the minimum client buffer size to two blocks. Single block
buffers don't work properly yet.
with help from jakemsr
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underuns and overruns; such bugs are hard to debug with ktrace
or a debugger. They are also handy to debug or monitor code
using aucat.
To enable traces, compile aucat with ``make DEBUG=-DDEBUG'' and
use the -d option multiple times.
ok jakemsr
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- put aproc-specific parameters into unions since they are never
used together
- remove constant ``data'' pointer always pointing the end of the
abuf structure
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realtime. For now only the playback volume of individual streams can be
changed/monitored. To each stream is assigned a MIDI channel; the volume
is changed/monitored using the standard controller number 7.
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from Thomas Pfaff
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way, when in server mode, it consumes no CPU if there are no clients.
Later, this will allow to start aucat(1) at session or system startup.
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inside the mixer and the demultiplexer. This way, aucat will not
trigger the heavy conversion code when only channel conversions are
required. Cuts ~50% of the CPU usage on envy(4) devices, can improve
surround 4.0, 5.1 and 7.1 capable devices. No functionnal change.
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mode, it listens on an unix socket and mixes/demultiplexes any number
of full-duplex streams, doing necessary format conversions and
resampling on the fly.
programs can use the new libsa(3) library to play and record audio.
The library provides a very simple API to connect to the audio server;
if aucat(1) isn't running, it uses the audio(4) driver transparently
instead.
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end of the buffer because periodic boundary conditions of the FIFO are
not met
ok jakemsr
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bytes that entered the FIFO. The counter may overflow, so it
should be used with "modulo 2^32" arithmetic. The counter will
be used later to synchronize playback to record. No behaviour
change.
ok jakemsr
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generic abuf structure, so it can reused. Required for an audio
server. No behaviour change.
ok jakemsr
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abuf_rdiscard() routine. Similarly add abuf_wcommit() routine for
writing. The purpose is to stop manupulating FIFO pointers in
various places outiside abuf.c, its too error prone.
No behaviour change.
ok jakemsr
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field of structures.
from deraadt@
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- recording, full-duplex operation
- format conversions and resampling on the fly
- mixing on the fly of multiple inputs of different formats
- up to 16 channels, simplistic "routing" of channel ranges
- more linear encodings (in raw and wav files)
the old behaviour is fully preserved if none of the new -i and -o
options are used.
code and fixes from jakemsr@ and eric@, suggestions by others.
ok "go ahead" deraadt@
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