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
esa_round_blocksize().
- Fix esa_trigger_output() and esa_trigger_input() to initialise the
channel's buffer/block size using the supplied parameters.
- While here, simplify esa_intr().
This gets kphone working on my esa(4)-equipped laptop.
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.
would either:
1. Cause the machine to stop responding, or
2. Cause the currently playing voices to stop output.
With this change, voices are stopped on suspend, and continue from where they
left off on resume.
audio accelerators.
Mixing is done in hardware by the ASSP, but is limited to 4 simultaneous
channels due to the restricted "minisrc" image that we are currently
working with.
Due to limitations in the audio subsystem, I'm currently attaching multiple
'audio' devices to 'esa', one for each voice. Because of this hack, the
default ESA_NUM_VOICES is 1.
to the mixer. Ie:
$ mixerctl -w outputs.master=0,191
Would result in the _right_ speaker being turned off.
So, we will swap the left and right mixer channels to compensate
for this.
fancy, works for the basic output case, but does not support programs
which use playback while going into suspend.
Sketched after code found in FreeBSD and Linux drivers.
jmcneill