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
malloc types into a structure, a pointer to which is passed around,
instead of an int constant. Allow the limit to be adjusted when the
malloc type is defined, or with a function call, as suggested by
Jonathan Stone.
"off_t" and the return value is a "paddr_t" to allow mappings
at offsets past 2^31 bytes. Somewhat inspired by FreeBSD, which
only changed the offset to a "vm_offset_t".
Includes updates for the i386, pc532 and sh3 mmmmap from Jason Thorpe.
* Rearrange the speed mapping table and adjust the code so that the highest
rate can actually be used. Previously we ended up rounding up slightly
lower speeds and then losing because set_params couldn't set the mode
back to the current one.
* Allow 260 as a valid I/O address, since the SB1 can be jumpered to this.
* Change the MPU-401 code so it can be attached as a separate device.
(XXX Really, the SB code ought to just attach a subdevice itself.)
* Do not attach an OPL on the SB1. Writing to the OPL registers at
SB_base+0 on this card wedges my machine.
(XXX Should we access it at 388 instead? The Creative web site claims
that this board *does* have an OPL2, but I haven't played with this
extensively.)
* Allocate the SB DMA channels at open time, rather than attach time, so
that a single DRQ can be used for multiple cards (if only one is in use
at a given time).
(XXX Let me tell you why this is a horrible hack. If the ISA DMA code
tries to allocate a bounce buffer after boot time, it will generally fail,
because there is no contiguous memory below 16MB and the code to allocate
contiguous pages doesn't know how to move things around. Now, we
shouldn't ever be using bounce buffers here, because we use
isa_dmamem_alloc(). So we just turn off BUS_DMA_ALLOCNOW and we don't
actually try to. That's cool, and it even works, but isa_dmamem_alloc()
has the same problem. It just happens that we allocate the ring buffers
at boot time, and whenever we reallocate them (due to the buffer size
changing), we just deallocated the previous (contiguous) buffer, so we get
lucky. This is absolutely disgusting and needs to be fixed.)
Also, even on the SB1, we can leave the DMA controller in auto-initalize
mode and just send a command to the board for each block. This may help
prevent FIFO underruns.
the /dev/rmidiN devices, or with a sequencer interface via /dev/music.
So far the only supported MIDI device is the MPU401 port on SoundBlaster
(and only on SB on isapnp, since we do not have locators with multiple
values yet).
The changes is to allow some limited mixer manipulation through
the audio device (instead of the mixer device).
This rendered 4 methods in audio_hw_if unused so garbage collect these.
at the same time instead by using two different calls. This enables
it to check more easily if the combined mode is all right.
- Improve the error checking in audio.c.
- Add a new audio property, AUDIO_PROP_INDEPENDENT, show if the
play and record settings are independent.
- Fix some buglets in audio.c.
* Make it possible to use software assisted encodings that expand the
sample size.
* Use 16 bits per sample when emulating mulaw coding on the SB.
* Prepare for SB16 without CT1745 mixer.
- Change the way attach and open works to allow multiple audio
devices.
- Split the mulaw.c file into two to avoid dragging in mulaw
convertsion when they are not needed. Add 16 bit alaw/mulaw tables.
- Change the way audio properties are gotten.
- Recognize more versions os SoundBlaster.
- It is now possible to handle devices that want "looping" DMA,
e.g. the SoundBlaster correctly. The WSS and SB drivers use this.
To do this several new methods were introduced in audio_hw_if.
- Different silence handling (forced by previous change).
- The audio driver can now be mmap()-ed, but due to problems in
the VM system only for writing for now.
- The OSS (Linux) audio emulation takes advantage of some of the
new features.
Set the encoding parameters slightly differently.
Remove the SW encoding/decodinf functions from this interface
and move them to the audio_parameter struct; this is both more efficient
and flexible.
'flags 1' on the sb? kernel configuration file line (because it frobs a
noncontiguous IO port to configure the Jazz16 extensions).
Also, remove static sb_device structure and fill in user's buffer on
each request.
