is explicitly set. Without this, the machine would crash in the audio interupt when the driver
needs to divide by the block size (e.g., cs4281.c/cs4280.c).
Idea for the fix by yamt.
to open(). Previously only one call to open() was allowed.
This change should allow some Linux derived programs to work unaltered.
XXX This isn't really possible to implement correctly in a simple way.
So this implementation is incorrect: It is not until device has been closed
by both the reader and writer that it is possible to open it at all again.
underlying audio device was opened read/write. This is consistent with
Linux OSS behaviour and fixes a bug with certain applications (including
Skype) that assume this behaviour.
Fixes PR# 30044.
call audio_clear() only when blocksizes are actually changed.
This fixes a problem on some audio devices and applications which
use ossaudio and call SNDCTL_DSP_GETOSPACE repeatedly while playing.
* audiostartp()
add some debug prints.
audio framework
Summary of changes:
* struct audio_params
- remove sw_code, factor, factor_denom, hw_sample_rate,
hw_encoding ,hw_precision, and hw_channels. Conversion information
is conveyed by stream_filter_list_t.
- change the type of sample_rate: u_long -> u_int
- add `validbits,' which represents the valid data size in
precision bits. It is required in order to distinguish 24/32bit
from 24/24bit or 32/32bit.
* audio_hw_if
- add two parameters to set_params()
stream_filter_list_t *pfil, stream_filter_list *rfil
A HW driver should set filter recipes for requested formats
- constify audio_params parameters of trigger_output() and
trigger_input(). They represent audio formats for the hardware.
- make open() and close() optional
- add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters
to round_blocksize()
* sw_code is replaced with stream_filter_t.
stream_filer_t converts audio data in an input buffer and writes
into another output buffer unlike sw_code, which converts data in
single buffer.
converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c,
dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are
reimplemented as stream_filter_t
* MI audio
- audiosetinfo() builds filter pipelines from stream_filter_list_t
filled by audio_hw_if::set_params()
- audiosetinfo() returns with EINVAL if mmapped and set_params()
requests filters
- audio_write(), audio_pint(), and audio_rint() invoke a filter
pipeline.
- ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS,
AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for
AUDIO_GETINFO handle values for a buffer nearest to userland.
* add `struct device *' parameter to ac97_attach()
* all of audio HW drivers follow audio_hw_if and ac97 changes
present, do an extra wakeup() on the write channel. This happens when we
change parameters during playback, at which point we flush the buffer, and
would stall.
audio device interface:
1) When attempting to match the appropriate mixer control, we weren't
checking the class label, but only the second level label, so for
devices that had both an "inputs.cd" and a "record.cd", for example,
we could never do the right thing except by chance. For this reason,
evidently, all the record masters were labeled (by the underlying
drivers) either "record.record" or "record.volume", to distinguish
from "outputs.master". We'll now accept "record.master", and document
that as the the new preferred way.
2) More importantly, the model was deficient. Selecting a port on many
chips completely disables most of the level controls, which doesn't play
nice with other applications which are trying to use the interface. Now,
selecting a port simply sets which mixer input control shall be changed,
setting state in the audio layer. In other words, the "mixerout" port
is really selected all the time, enabling the final stage mixer, and
setting "gain" sets the level of the appropriate input. It should be
possible for separate applications to each control the mic, dac, and cd
inputs at the same time using this interface, simply by reiterating their
port selections with each change, but applications that don't bother to
do that aren't any worse off than they were before.
The user is expected to set the master output with another application,
dedicated to that task. Though it is now meaningful to select "no port"
with the audio device interface, to manipulate the master output, there's
no particular reason for an audio device consumer to do that. (I added
the capability in order to restore the initial state of the audio device,
for testing purposes. It might also be useful to users of broken binary-
only applications.)
Observe that the mixer device interface (and so, "mixerctl") still
retains all capabilities, including the ability to set the actual input
port on the chip, overriding the level controls. No change is being made
to the mixer device interface. The mixer device simply presents all the
controls on the chip, with no attempt at abstraction, so there are no
bugs there.
The upshot is, that applications that have been trying to use the audio
device interface to change the volume, such as mplayer, now "just work".
I've tested these changes extensively with "eso" and "eap" since first
proposing them on tech-kern last January, and somewhat with "esm" and a
few others. This closes both PR kern/10221, and PR kern/17159.
* Separate the code to set the default parameters into a new function,
audio_set_defaults(). Make it use audiosetinfo(), which properly initializes
the block size and whatnot. Use this in both audioattach() and the
/dev/audio case of audio_open().
* Do not force a reinitialization when /dev/sound is opened.
* Do all of the block size sanity checks in auto_init_ringbuffer(), not in
both audio_calc_blksize() and audiosetinfo().
* Fix a bug in audiosetinfo() that caused the block size to not be recalculated
immediately if we set it to 0.
* For AUDIO_GET[IO]OFFS, modify the deltablks calculation so that it gives us
the number of block boundaries crossed.
be inserted into ktrace records. The general change has been to replace
"struct proc *" with "struct lwp *" in various function prototypes, pass
the lwp through and use l_proc to get the process pointer when needed.
Bump the kernel rev up to 1.6V
Back-out revision 1.170 and and 1.166. Revision 1.170 fixes some problems
introduced in revision 1.166. It isn't clear what problem revision 1.166
was attempting to address. It seems bogus anyway, since we later check
the modes in audio_check_params().
cannot change the recording or playback mode of the device, it can
only change other mode-like values (like AUMODE_PLAY_ALL). Be very
explicit about fixing up the user's mode value based on the mode of
the device. Before, giving AUMODE_PLAY_ALL could cause AUMODE_PLAY
to become set on the device, and once AUMODE_PLAY_ALL was set it
was impossible to clear.
kqueue provides a stateful and efficient event notification framework
currently supported events include socket, file, directory, fifo,
pipe, tty and device changes, and monitoring of processes and signals
kqueue is supported by all writable filesystems in NetBSD tree
(with exception of Coda) and all device drivers supporting poll(2)
based on work done by Jonathan Lemon for FreeBSD
initial NetBSD port done by Luke Mewburn and Jason Thorpe
