freebsd-nq/sys/dev/sound/usb/uaudio_pcm.c
Ariff Abdullah 90da2b2859 Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
	[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .

Summary of changes includes:

1 Volume Per-Channel (vpc).  Provides private / standalone volume control
  unique per-stream pcm channel without touching master volume / pcm.
  Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
  backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
  instead of /dev/mixer.  Special "bypass" mode is enabled through
  /dev/mixer which will automatically detect if the adjustment is made
  through /dev/mixer and forward its request to this private volume
  controller.  Changes to this volume object will not interfere with
  other channels.

  Requirements:
    - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
      require specific application modifications (preferred).
    - No modifications required for using bypass mode, so applications
      like mplayer or xmms should work out of the box.

  Kernel hints:
    - hint.pcm.%d.vpc (0 = disable vpc).

  Kernel sysctls:
    - hw.snd.vpc_mixer_bypass (default: 1).  Enable or disable /dev/mixer
      bypass mode.
    - hw.snd.vpc_autoreset (default: 1).  By default, closing/opening
      /dev/dsp will reset the volume back to 0 db gain/attenuation.
      Setting this to 0 will preserve its settings across device
      closing/opening.
    - hw.snd.vpc_reset (default: 0).  Panic/reset button to reset all
      volume settings back to 0 db.
    - hw.snd.vpc_0db (default: 45).  0 db relative to linear mixer value.

2 High quality fixed-point Bandlimited SINC sampling rate converter,
  based on Julius O'Smith's Digital Audio Resampling -
  http://ccrma.stanford.edu/~jos/resample/.  It includes a filter design
  script written in awk (the clumsiest joke I've ever written)
    - 100% 32bit fixed-point, 64bit accumulator.
    - Possibly among the fastest (if not fastest) of its kind.
    - Resampling quality is tunable, either runtime or during kernel
      compilation (FEEDER_RATE_PRESETS).
    - Quality can be further customized during kernel compilation by
      defining FEEDER_RATE_PRESETS in /etc/make.conf.

  Kernel sysctls:
    - hw.snd.feeder_rate_quality.
      0 - Zero-order Hold (ZOH).  Fastest, bad quality.
      1 - Linear Interpolation (LINEAR).  Slightly slower than ZOH,
          better quality but still does not eliminate aliasing.
      2 - (and above) - Sinc Interpolation(SINC).  Best quality.  SINC
          quality always start from 2 and above.

  Rough quality comparisons:
    - http://people.freebsd.org/~ariff/z_comparison/

3 Bit-perfect mode.  Bypasses all feeder/dsp effects.  Pure sound will be
  directly fed into the hardware.

4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
  be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.

5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
  vchans in order to make digital format pass through.  It also makes
  vchans more dynamic by choosing a better format/rate among all the
  concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
  becomes sort of optional.

6 Exclusive Stream, with special open() mode O_EXCL.  This will "mute"
  other concurrent vchan streams and only allow a single channel with
  O_EXCL set to keep producing sound.

Other Changes:
    * most feeder_* stuffs are compilable in userland. Let's not
      speculate whether we should go all out for it (save that for
      FreeBSD 16.0-RELEASE).
    * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
    * pull out channel mixing logic out of vchan.c and create its own
      feeder_mixer for world justice.
    * various refactoring here and there, for good or bad.
    * activation of few more OSSv4 ioctls() (see [1] above).
    * opt_snd.h for possible compile time configuration:
      (mostly for debugging purposes, don't try these at home)
        SND_DEBUG
        SND_DIAGNOSTIC
        SND_FEEDER_MULTIFORMAT
        SND_FEEDER_FULL_MULTIFORMAT
        SND_FEEDER_RATE_HP
        SND_PCM_64
        SND_OLDSTEREO

Manual page updates are on the way.

Tested by:	joel, Olivier SMEDTS <olivier at gid0 d org>, too many
          	unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00

243 lines
5.9 KiB
C

/* $FreeBSD$ */
/*-
* Copyright (c) 2000-2002 Hiroyuki Aizu <aizu@navi.org>
* Copyright (c) 2006 Hans Petter Selasky
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND
* ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
* ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE
* FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
* OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
* LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
* SUCH DAMAGE.
*/
#ifdef HAVE_KERNEL_OPTION_HEADERS
#include "opt_snd.h"
#endif
#include <dev/sound/pcm/sound.h>
#include <dev/sound/chip.h>
#include <dev/sound/usb/uaudio.h>
#include "mixer_if.h"
/************************************************************/
static void *
ua_chan_init(kobj_t obj, void *devinfo, struct snd_dbuf *b, struct pcm_channel *c, int dir)
{
return (uaudio_chan_init(devinfo, b, c, dir));
}
static int
ua_chan_free(kobj_t obj, void *data)
{
return (uaudio_chan_free(data));
}
static int
ua_chan_setformat(kobj_t obj, void *data, uint32_t format)
{
/*
* At this point, no need to query as we
* shouldn't select an unsorted format
*/
return (uaudio_chan_set_param_format(data, format));
}
static uint32_t
ua_chan_setspeed(kobj_t obj, void *data, uint32_t speed)
{
return (uaudio_chan_set_param_speed(data, speed));
}
static uint32_t
ua_chan_setblocksize(kobj_t obj, void *data, uint32_t blocksize)
{
return (uaudio_chan_set_param_blocksize(data, blocksize));
}
static int
ua_chan_setfragments(kobj_t obj, void *data, uint32_t blocksize, uint32_t blockcount)
{
return (uaudio_chan_set_param_fragments(data, blocksize, blockcount));
}
static int
ua_chan_trigger(kobj_t obj, void *data, int go)
{
if (!PCMTRIG_COMMON(go)) {
return (0);
}
if (go == PCMTRIG_START) {
return (uaudio_chan_start(data));
} else {
return (uaudio_chan_stop(data));
}
}
static uint32_t
ua_chan_getptr(kobj_t obj, void *data)
{
return (uaudio_chan_getptr(data));
}
static struct pcmchan_caps *
ua_chan_getcaps(kobj_t obj, void *data)
{
return (uaudio_chan_getcaps(data));
}
static struct pcmchan_matrix *
ua_chan_getmatrix(kobj_t obj, void *data, uint32_t format)
{
return (uaudio_chan_getmatrix(data, format));
}
static kobj_method_t ua_chan_methods[] = {
KOBJMETHOD(channel_init, ua_chan_init),
KOBJMETHOD(channel_free, ua_chan_free),
KOBJMETHOD(channel_setformat, ua_chan_setformat),
KOBJMETHOD(channel_setspeed, ua_chan_setspeed),
KOBJMETHOD(channel_setblocksize, ua_chan_setblocksize),
KOBJMETHOD(channel_setfragments, ua_chan_setfragments),
KOBJMETHOD(channel_trigger, ua_chan_trigger),
KOBJMETHOD(channel_getptr, ua_chan_getptr),
KOBJMETHOD(channel_getcaps, ua_chan_getcaps),
KOBJMETHOD(channel_getmatrix, ua_chan_getmatrix),
KOBJMETHOD_END
};
CHANNEL_DECLARE(ua_chan);
/************************************************************/
static int
ua_mixer_init(struct snd_mixer *m)
{
return (uaudio_mixer_init_sub(mix_getdevinfo(m), m));
}
static int
ua_mixer_set(struct snd_mixer *m, unsigned type, unsigned left, unsigned right)
{
struct mtx *mtx = mixer_get_lock(m);
uint8_t do_unlock;
if (mtx_owned(mtx)) {
do_unlock = 0;
} else {
do_unlock = 1;
mtx_lock(mtx);
}
uaudio_mixer_set(mix_getdevinfo(m), type, left, right);
if (do_unlock) {
mtx_unlock(mtx);
}
return (left | (right << 8));
}
static uint32_t
ua_mixer_setrecsrc(struct snd_mixer *m, uint32_t src)
{
struct mtx *mtx = mixer_get_lock(m);
int retval;
uint8_t do_unlock;
if (mtx_owned(mtx)) {
do_unlock = 0;
} else {
do_unlock = 1;
mtx_lock(mtx);
}
retval = uaudio_mixer_setrecsrc(mix_getdevinfo(m), src);
if (do_unlock) {
mtx_unlock(mtx);
}
return (retval);
}
static int
ua_mixer_uninit(struct snd_mixer *m)
{
return (uaudio_mixer_uninit_sub(mix_getdevinfo(m)));
}
static kobj_method_t ua_mixer_methods[] = {
KOBJMETHOD(mixer_init, ua_mixer_init),
KOBJMETHOD(mixer_uninit, ua_mixer_uninit),
KOBJMETHOD(mixer_set, ua_mixer_set),
KOBJMETHOD(mixer_setrecsrc, ua_mixer_setrecsrc),
KOBJMETHOD_END
};
MIXER_DECLARE(ua_mixer);
/************************************************************/
static int
ua_probe(device_t dev)
{
struct sndcard_func *func;
/* the parent device has already been probed */
func = device_get_ivars(dev);
if ((func == NULL) ||
(func->func != SCF_PCM)) {
return (ENXIO);
}
device_set_desc(dev, "USB audio");
return (BUS_PROBE_DEFAULT);
}
static int
ua_attach(device_t dev)
{
return (uaudio_attach_sub(dev, &ua_mixer_class, &ua_chan_class));
}
static int
ua_detach(device_t dev)
{
return (uaudio_detach_sub(dev));
}
/************************************************************/
static device_method_t ua_pcm_methods[] = {
/* Device interface */
DEVMETHOD(device_probe, ua_probe),
DEVMETHOD(device_attach, ua_attach),
DEVMETHOD(device_detach, ua_detach),
{0, 0}
};
static driver_t ua_pcm_driver = {
"pcm",
ua_pcm_methods,
PCM_SOFTC_SIZE,
};
DRIVER_MODULE(ua_pcm, uaudio, ua_pcm_driver, pcm_devclass, 0, 0);
MODULE_DEPEND(ua_pcm, uaudio, 1, 1, 1);
MODULE_DEPEND(ua_pcm, sound, SOUND_MINVER, SOUND_PREFVER, SOUND_MAXVER);
MODULE_VERSION(ua_pcm, 1);