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.
* Snap the sample rate when setting it, and remember only the time constant.
* Set the time constant when changing between play/record.
* Always return the actual sample rate with AUDIO_GETINFO.
* Add more delays while writing registers.
* Replace sc_dma{in,out}_inprogress with sc_dmadir.
* Eliminate the need for sc_locked.
* Add more DPRINTF()s.
mycroft