2006-07-15 19:36:28 +00:00
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/*-
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* Copyright (c) 1999 Cameron Grant <gandalf@vilnya.demon.co.uk>
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2007-09-12 07:43:43 +00:00
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* Copyright (c) 2003-2007 Yuriy Tsibizov <yuriy.tsibizov@gfk.ru>
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2006-07-15 19:36:28 +00:00
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* All rights reserved.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions
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* are met:
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* 1. Redistributions of source code must retain the above copyright
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* notice, this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright
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* notice, this list of conditions and the following disclaimer in the
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* documentation and/or other materials provided with the distribution.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND
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* ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
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* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
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* ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE
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* FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
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* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
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* OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
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* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHERIN CONTRACT, STRICT
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* LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
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* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
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* SUCH DAMAGE.
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*
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* $FreeBSD$
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*/
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#include <sys/param.h>
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#include <sys/types.h>
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#include <sys/bus.h>
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#include <machine/bus.h>
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#include <sys/rman.h>
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#include <sys/systm.h>
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#include <sys/sbuf.h>
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#include <sys/queue.h>
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#include <sys/systm.h>
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#include <sys/lock.h>
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#include <sys/mutex.h>
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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
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#ifdef HAVE_KERNEL_OPTION_HEADERS
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#include "opt_snd.h"
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#endif
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2006-07-15 19:36:28 +00:00
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#include <dev/sound/chip.h>
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#include <dev/sound/pcm/sound.h>
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#include <dev/sound/pcm/ac97.h>
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#include "mixer_if.h"
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2012-01-11 21:17:14 +00:00
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#include <dev/sound/pci/emuxkireg.h>
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2006-07-15 19:36:28 +00:00
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#include <dev/sound/pci/emu10kx.h>
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struct emu_pcm_pchinfo {
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int spd;
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int fmt;
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2007-09-12 07:43:43 +00:00
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unsigned int blksz;
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2006-07-15 19:36:28 +00:00
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int run;
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struct emu_voice *master;
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struct emu_voice *slave;
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struct snd_dbuf *buffer;
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struct pcm_channel *channel;
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struct emu_pcm_info *pcm;
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int timer;
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};
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struct emu_pcm_rchinfo {
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int spd;
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int fmt;
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2007-09-12 07:43:43 +00:00
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unsigned int blksz;
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2006-07-15 19:36:28 +00:00
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int run;
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uint32_t idxreg;
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uint32_t basereg;
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uint32_t sizereg;
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uint32_t setupreg;
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uint32_t irqmask;
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uint32_t iprmask;
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int ihandle;
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struct snd_dbuf *buffer;
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struct pcm_channel *channel;
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struct emu_pcm_info *pcm;
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2007-09-12 07:43:43 +00:00
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int timer;
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2006-07-15 19:36:28 +00:00
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};
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2007-01-06 18:59:35 +00:00
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/* XXX Hardware playback channels */
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2006-07-15 19:36:28 +00:00
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#define MAX_CHANNELS 4
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#if MAX_CHANNELS > 13
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#error Too many hardware channels defined. 13 is the maximum
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#endif
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2007-01-06 18:59:35 +00:00
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2006-07-15 19:36:28 +00:00
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struct emu_pcm_info {
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struct mtx *lock;
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device_t dev; /* device information */
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struct emu_sc_info *card;
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struct emu_pcm_pchinfo pch[MAX_CHANNELS]; /* hardware channels */
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int pnum; /* next free channel number */
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2007-01-06 18:59:35 +00:00
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struct emu_pcm_rchinfo rch_adc;
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struct emu_pcm_rchinfo rch_efx;
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2006-07-15 19:36:28 +00:00
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struct emu_route rt;
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2006-08-07 22:55:21 +00:00
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struct emu_route rt_mono;
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2006-07-15 19:36:28 +00:00
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int route;
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int ihandle; /* interrupt handler */
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unsigned int bufsz;
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int is_emu10k1;
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struct ac97_info *codec;
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uint32_t ac97_state[0x7F];
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2007-09-12 07:43:43 +00:00
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kobj_class_t ac97_mixerclass;
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uint32_t ac97_recdevs;
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uint32_t ac97_playdevs;
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struct snd_mixer *sm;
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int mch_disabled;
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unsigned int emu10k1_volcache[2][2];
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2006-07-15 19:36:28 +00:00
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};
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2007-01-06 18:59:35 +00:00
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static uint32_t emu_rfmt_adc[] = {
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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
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SND_FORMAT(AFMT_S16_LE, 1, 0),
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SND_FORMAT(AFMT_S16_LE, 2, 0),
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2006-07-15 19:36:28 +00:00
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0
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};
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2007-01-06 18:59:35 +00:00
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static struct pcmchan_caps emu_reccaps_adc = {
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8000, 48000, emu_rfmt_adc, 0
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};
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static uint32_t emu_rfmt_efx[] = {
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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
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SND_FORMAT(AFMT_S16_LE, 1, 0),
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2007-01-06 18:59:35 +00:00
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0
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};
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static struct pcmchan_caps emu_reccaps_efx_live = {
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48000*32, 48000*32, emu_rfmt_efx, 0
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};
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static struct pcmchan_caps emu_reccaps_efx_audigy = {
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48000*64, 48000*64, emu_rfmt_efx, 0
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2006-07-15 19:36:28 +00:00
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};
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2007-09-12 07:43:43 +00:00
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static int emu_rates_live[] = {
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48000*32
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};
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static int emu_rates_audigy[] = {
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48000*64
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};
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2006-07-15 19:36:28 +00:00
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static uint32_t emu_pfmt[] = {
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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
|
|
|
SND_FORMAT(AFMT_U8, 1, 0),
|
|
|
|
SND_FORMAT(AFMT_U8, 2, 0),
|
|
|
|
SND_FORMAT(AFMT_S16_LE, 1, 0),
|
|
|
|
SND_FORMAT(AFMT_S16_LE, 2, 0),
|
2006-07-15 19:36:28 +00:00
|
|
|
0
|
|
|
|
};
|
|
|
|
static uint32_t emu_pfmt_mono[] = {
|
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
|
|
|
SND_FORMAT(AFMT_U8, 1, 0),
|
|
|
|
SND_FORMAT(AFMT_S16_LE, 1, 0),
|
2006-07-15 19:36:28 +00:00
|
|
|
0
|
|
|
|
};
|
|
|
|
|
|
|
|
static struct pcmchan_caps emu_playcaps = {4000, 48000, emu_pfmt, 0};
|
|
|
|
static struct pcmchan_caps emu_playcaps_mono = {4000, 48000, emu_pfmt_mono, 0};
|
|
|
|
|
|
|
|
static int emu10k1_adcspeed[8] = {48000, 44100, 32000, 24000, 22050, 16000, 11025, 8000};
|
|
|
|
/* audigy supports 12kHz. */
|
|
|
|
static int emu10k2_adcspeed[9] = {48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000};
|
|
|
|
|
|
|
|
static uint32_t emu_pcm_intr(void *pcm, uint32_t stat);
|
|
|
|
|
2007-09-12 07:43:43 +00:00
|
|
|
static const struct emu_dspmix_props_k1 {
|
|
|
|
uint8_t present;
|
|
|
|
uint8_t recdev;
|
|
|
|
int8_t input;
|
|
|
|
} dspmix_k1 [SOUND_MIXER_NRDEVICES] = {
|
|
|
|
/* no mixer device for ac97 */ /* in0 AC97 */
|
|
|
|
[SOUND_MIXER_DIGITAL1] = {1, 1, 1}, /* in1 CD SPDIF */
|
|
|
|
/* not connected */ /* in2 (zoom) */
|
|
|
|
[SOUND_MIXER_DIGITAL2] = {1, 1, 3}, /* in3 toslink */
|
|
|
|
[SOUND_MIXER_LINE2] = {1, 1, 4}, /* in4 Line-In2 */
|
|
|
|
[SOUND_MIXER_DIGITAL3] = {1, 1, 5}, /* in5 on-card SPDIF */
|
|
|
|
[SOUND_MIXER_LINE3] = {1, 1, 6}, /* in6 AUX2 */
|
|
|
|
/* not connected */ /* in7 */
|
|
|
|
};
|
|
|
|
static const struct emu_dspmix_props_k2 {
|
|
|
|
uint8_t present;
|
|
|
|
uint8_t recdev;
|
|
|
|
int8_t input;
|
|
|
|
} dspmix_k2 [SOUND_MIXER_NRDEVICES] = {
|
|
|
|
[SOUND_MIXER_VOLUME] = {1, 0, (-1)},
|
|
|
|
[SOUND_MIXER_PCM] = {1, 0, (-1)},
|
|
|
|
|
|
|
|
/* no mixer device */ /* in0 AC97 */
|
|
|
|
[SOUND_MIXER_DIGITAL1] = {1, 1, 1}, /* in1 CD SPDIF */
|
|
|
|
[SOUND_MIXER_DIGITAL2] = {1, 1, 2}, /* in2 COAX SPDIF */
|
|
|
|
/* not connected */ /* in3 */
|
|
|
|
[SOUND_MIXER_LINE2] = {1, 1, 4}, /* in4 Line-In2 */
|
|
|
|
[SOUND_MIXER_DIGITAL3] = {1, 1, 5}, /* in5 on-card SPDIF */
|
|
|
|
[SOUND_MIXER_LINE3] = {1, 1, 6}, /* in6 AUX2 */
|
|
|
|
/* not connected */ /* in7 */
|
2006-07-15 19:36:28 +00:00
|
|
|
};
|
|
|
|
|
|
|
|
static int
|
|
|
|
emu_dspmixer_init(struct snd_mixer *m)
|
|
|
|
{
|
2007-09-12 07:43:43 +00:00
|
|
|
struct emu_pcm_info *sc;
|
2006-07-15 19:36:28 +00:00
|
|
|
int i;
|
2007-09-12 07:43:43 +00:00
|
|
|
int p, r;
|
|
|
|
|
|
|
|
p = 0;
|
|
|
|
r = 0;
|
|
|
|
|
|
|
|
sc = mix_getdevinfo(m);
|
|
|
|
|
|
|
|
if (sc->route == RT_FRONT) {
|
|
|
|
/* create submixer for AC97 codec */
|
|
|
|
if ((sc->ac97_mixerclass != NULL) && (sc->codec != NULL)) {
|
|
|
|
sc->sm = mixer_create(sc->dev, sc->ac97_mixerclass, sc->codec, "ac97");
|
|
|
|
if (sc->sm != NULL) {
|
|
|
|
p = mix_getdevs(sc->sm);
|
|
|
|
r = mix_getrecdevs(sc->sm);
|
|
|
|
}
|
|
|
|
}
|
2006-07-15 19:36:28 +00:00
|
|
|
|
2007-09-12 07:43:43 +00:00
|
|
|
sc->ac97_playdevs = p;
|
|
|
|
sc->ac97_recdevs = r;
|
2006-07-15 19:36:28 +00:00
|
|
|
}
|
|
|
|
|
2007-09-12 07:43:43 +00:00
|
|
|
/* This two are always here */
|
|
|
|
p |= (1 << SOUND_MIXER_PCM);
|
|
|
|
p |= (1 << SOUND_MIXER_VOLUME);
|
|
|
|
|
|
|
|
if (sc->route == RT_FRONT) {
|
|
|
|
if (sc->is_emu10k1) {
|
|
|
|
for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) {
|
|
|
|
if (dspmix_k1[i].present)
|
|
|
|
p |= (1 << i);
|
|
|
|
if (dspmix_k1[i].recdev)
|
|
|
|
r |= (1 << i);
|
|
|
|
}
|
|
|
|
} else {
|
|
|
|
for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) {
|
|
|
|
if (dspmix_k2[i].present)
|
|
|
|
p |= (1 << i);
|
|
|
|
if (dspmix_k2[i].recdev)
|
|
|
|
r |= (1 << i);
|
|
|
|
}
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
mix_setdevs(m, p);
|
|
|
|
mix_setrecdevs(m, r);
|
|
|
|
|
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
emu_dspmixer_uninit(struct snd_mixer *m)
|
|
|
|
{
|
|
|
|
struct emu_pcm_info *sc;
|
|
|
|
int err = 0;
|
|
|
|
|
|
|
|
/* drop submixer for AC97 codec */
|
|
|
|
sc = mix_getdevinfo(m);
|
|
|
|
if (sc->sm != NULL)
|
|
|
|
err = mixer_delete(sc->sm);
|
|
|
|
if (err)
|
|
|
|
return (err);
|
|
|
|
sc->sm = NULL;
|
2006-07-15 19:36:28 +00:00
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
emu_dspmixer_set(struct snd_mixer *m, unsigned dev, unsigned left, unsigned right)
|
|
|
|
{
|
|
|
|
struct emu_pcm_info *sc;
|
|
|
|
|
|
|
|
sc = mix_getdevinfo(m);
|
|
|
|
|
|
|
|
switch (dev) {
|
|
|
|
case SOUND_MIXER_VOLUME:
|
|
|
|
switch (sc->route) {
|
2007-09-12 07:43:43 +00:00
|
|
|
case RT_FRONT:
|
|
|
|
if (sc->sm != NULL)
|
|
|
|
mix_set(sc->sm, dev, left, right);
|
|
|
|
if (sc->mch_disabled) {
|
|
|
|
/* In emu10k1 case PCM volume does not affect
|
|
|
|
sound routed to rear & center/sub (it is connected
|
|
|
|
to AC97 codec). Calculate it manually. */
|
|
|
|
/* This really should belong to emu10kx.c */
|
|
|
|
if (sc->is_emu10k1) {
|
|
|
|
sc->emu10k1_volcache[0][0] = left;
|
|
|
|
left = left * sc->emu10k1_volcache[1][0] / 100;
|
|
|
|
sc->emu10k1_volcache[0][1] = right;
|
|
|
|
right = right * sc->emu10k1_volcache[1][1] / 100;
|
|
|
|
}
|
|
|
|
|
|
|
|
emumix_set_volume(sc->card, M_MASTER_REAR_L, left);
|
|
|
|
emumix_set_volume(sc->card, M_MASTER_REAR_R, right);
|
|
|
|
if (!sc->is_emu10k1) {
|
|
|
|
emumix_set_volume(sc->card, M_MASTER_CENTER, (left+right)/2);
|
|
|
|
emumix_set_volume(sc->card, M_MASTER_SUBWOOFER, (left+right)/2);
|
|
|
|
/* XXX side */
|
|
|
|
}
|
|
|
|
} /* mch disabled */
|
|
|
|
break;
|
2006-07-15 19:36:28 +00:00
|
|
|
case RT_REAR:
|
|
|
|
emumix_set_volume(sc->card, M_MASTER_REAR_L, left);
|
|
|
|
emumix_set_volume(sc->card, M_MASTER_REAR_R, right);
|
|
|
|
break;
|
|
|
|
case RT_CENTER:
|
|
|
|
emumix_set_volume(sc->card, M_MASTER_CENTER, (left+right)/2);
|
|
|
|
break;
|
|
|
|
case RT_SUB:
|
|
|
|
emumix_set_volume(sc->card, M_MASTER_SUBWOOFER, (left+right)/2);
|
|
|
|
break;
|
|
|
|
}
|
|
|
|
break;
|
|
|
|
case SOUND_MIXER_PCM:
|
|
|
|
switch (sc->route) {
|
2007-09-12 07:43:43 +00:00
|
|
|
case RT_FRONT:
|
|
|
|
if (sc->sm != NULL)
|
|
|
|
mix_set(sc->sm, dev, left, right);
|
|
|
|
if (sc->mch_disabled) {
|
|
|
|
/* See SOUND_MIXER_VOLUME case */
|
|
|
|
if (sc->is_emu10k1) {
|
|
|
|
sc->emu10k1_volcache[1][0] = left;
|
|
|
|
left = left * sc->emu10k1_volcache[0][0] / 100;
|
|
|
|
sc->emu10k1_volcache[1][1] = right;
|
|
|
|
right = right * sc->emu10k1_volcache[0][1] / 100;
|
|
|
|
}
|
|
|
|
emumix_set_volume(sc->card, M_MASTER_REAR_L, left);
|
|
|
|
emumix_set_volume(sc->card, M_MASTER_REAR_R, right);
|
|
|
|
|
|
|
|
if (!sc->is_emu10k1) {
|
|
|
|
emumix_set_volume(sc->card, M_MASTER_CENTER, (left+right)/2);
|
|
|
|
emumix_set_volume(sc->card, M_MASTER_SUBWOOFER, (left+right)/2);
|
|
|
|
/* XXX side */
|
|
|
|
}
|
|
|
|
} /* mch_disabled */
|
|
|
|
break;
|
2006-07-15 19:36:28 +00:00
|
|
|
case RT_REAR:
|
|
|
|
emumix_set_volume(sc->card, M_FX2_REAR_L, left);
|
|
|
|
emumix_set_volume(sc->card, M_FX3_REAR_R, right);
|
|
|
|
break;
|
|
|
|
case RT_CENTER:
|
|
|
|
emumix_set_volume(sc->card, M_FX4_CENTER, (left+right)/2);
|
|
|
|
break;
|
|
|
|
case RT_SUB:
|
|
|
|
emumix_set_volume(sc->card, M_FX5_SUBWOOFER, (left+right)/2);
|
|
|
|
break;
|
|
|
|
}
|
|
|
|
break;
|
2007-09-12 07:43:43 +00:00
|
|
|
case SOUND_MIXER_DIGITAL1: /* CD SPDIF, in1 */
|
|
|
|
emumix_set_volume(sc->card, M_IN1_FRONT_L, left);
|
|
|
|
emumix_set_volume(sc->card, M_IN1_FRONT_R, right);
|
|
|
|
break;
|
|
|
|
case SOUND_MIXER_DIGITAL2:
|
|
|
|
if (sc->is_emu10k1) {
|
|
|
|
/* TOSLink, in3 */
|
|
|
|
emumix_set_volume(sc->card, M_IN3_FRONT_L, left);
|
|
|
|
emumix_set_volume(sc->card, M_IN3_FRONT_R, right);
|
|
|
|
} else {
|
|
|
|
/* COAX SPDIF, in2 */
|
|
|
|
emumix_set_volume(sc->card, M_IN2_FRONT_L, left);
|
|
|
|
emumix_set_volume(sc->card, M_IN2_FRONT_R, right);
|
|
|
|
}
|
|
|
|
break;
|
|
|
|
case SOUND_MIXER_LINE2: /* Line-In2, in4 */
|
|
|
|
emumix_set_volume(sc->card, M_IN4_FRONT_L, left);
|
|
|
|
emumix_set_volume(sc->card, M_IN4_FRONT_R, right);
|
|
|
|
break;
|
|
|
|
case SOUND_MIXER_DIGITAL3: /* on-card SPDIF, in5 */
|
|
|
|
emumix_set_volume(sc->card, M_IN5_FRONT_L, left);
|
|
|
|
emumix_set_volume(sc->card, M_IN5_FRONT_R, right);
|
|
|
|
break;
|
|
|
|
case SOUND_MIXER_LINE3: /* AUX2, in6 */
|
|
|
|
emumix_set_volume(sc->card, M_IN6_FRONT_L, left);
|
|
|
|
emumix_set_volume(sc->card, M_IN6_FRONT_R, right);
|
|
|
|
break;
|
2006-07-15 19:36:28 +00:00
|
|
|
default:
|
2007-09-12 07:43:43 +00:00
|
|
|
if (sc->sm != NULL) {
|
|
|
|
/* XXX emumix_set_volume is not required here */
|
|
|
|
emumix_set_volume(sc->card, M_IN0_FRONT_L, 100);
|
|
|
|
emumix_set_volume(sc->card, M_IN0_FRONT_R, 100);
|
|
|
|
mix_set(sc->sm, dev, left, right);
|
|
|
|
} else
|
|
|
|
device_printf(sc->dev, "mixer error: unknown device %d\n", dev);
|
2006-07-15 19:36:28 +00:00
|
|
|
}
|
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
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
|
|
|
static u_int32_t
|
2007-09-12 07:43:43 +00:00
|
|
|
emu_dspmixer_setrecsrc(struct snd_mixer *m, u_int32_t src)
|
2006-07-15 19:36:28 +00:00
|
|
|
{
|
2007-09-12 07:43:43 +00:00
|
|
|
struct emu_pcm_info *sc;
|
|
|
|
int i;
|
|
|
|
u_int32_t recmask;
|
|
|
|
int input[8];
|
|
|
|
|
|
|
|
sc = mix_getdevinfo(m);
|
|
|
|
recmask = 0;
|
|
|
|
for (i=0; i < 8; i++)
|
|
|
|
input[i]=0;
|
|
|
|
|
|
|
|
if (sc->sm != NULL)
|
|
|
|
if ((src & sc->ac97_recdevs) !=0)
|
|
|
|
if (mix_setrecsrc(sc->sm, src & sc->ac97_recdevs) == 0) {
|
|
|
|
recmask |= (src & sc->ac97_recdevs);
|
|
|
|
/* Recording from AC97 codec.
|
|
|
|
Enable AC97 route to rec on DSP */
|
|
|
|
input[0] = 1;
|
|
|
|
}
|
|
|
|
if (sc->is_emu10k1) {
|
|
|
|
for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) {
|
|
|
|
if (dspmix_k1[i].recdev)
|
|
|
|
if ((src & (1 << i)) == ((uint32_t)1 << i)) {
|
|
|
|
recmask |= (1 << i);
|
|
|
|
/* enable device i */
|
|
|
|
input[dspmix_k1[i].input] = 1;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
} else {
|
|
|
|
for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) {
|
|
|
|
if (dspmix_k2[i].recdev)
|
|
|
|
if ((src & (1 << i)) == ((uint32_t)1 << i)) {
|
|
|
|
recmask |= (1 << i);
|
|
|
|
/* enable device i */
|
|
|
|
input[dspmix_k2[i].input] = 1;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
}
|
|
|
|
emumix_set_volume(sc->card, M_IN0_REC_L, input[0] == 1 ? 100 : 0);
|
|
|
|
emumix_set_volume(sc->card, M_IN0_REC_R, input[0] == 1 ? 100 : 0);
|
|
|
|
|
|
|
|
emumix_set_volume(sc->card, M_IN1_REC_L, input[1] == 1 ? 100 : 0);
|
|
|
|
emumix_set_volume(sc->card, M_IN1_REC_R, input[1] == 1 ? 100 : 0);
|
|
|
|
|
|
|
|
if (!sc->is_emu10k1) {
|
|
|
|
emumix_set_volume(sc->card, M_IN2_REC_L, input[2] == 1 ? 100 : 0);
|
|
|
|
emumix_set_volume(sc->card, M_IN2_REC_R, input[2] == 1 ? 100 : 0);
|
|
|
|
}
|
|
|
|
|
|
|
|
if (sc->is_emu10k1) {
|
|
|
|
emumix_set_volume(sc->card, M_IN3_REC_L, input[3] == 1 ? 100 : 0);
|
|
|
|
emumix_set_volume(sc->card, M_IN3_REC_R, input[3] == 1 ? 100 : 0);
|
|
|
|
}
|
|
|
|
|
|
|
|
emumix_set_volume(sc->card, M_IN4_REC_L, input[4] == 1 ? 100 : 0);
|
|
|
|
emumix_set_volume(sc->card, M_IN4_REC_R, input[4] == 1 ? 100 : 0);
|
|
|
|
|
|
|
|
emumix_set_volume(sc->card, M_IN5_REC_L, input[5] == 1 ? 100 : 0);
|
|
|
|
emumix_set_volume(sc->card, M_IN5_REC_R, input[5] == 1 ? 100 : 0);
|
|
|
|
|
|
|
|
emumix_set_volume(sc->card, M_IN6_REC_L, input[6] == 1 ? 100 : 0);
|
|
|
|
emumix_set_volume(sc->card, M_IN6_REC_R, input[6] == 1 ? 100 : 0);
|
|
|
|
|
|
|
|
/* XXX check for K1/k2 differences? */
|
|
|
|
if ((src & (1 << SOUND_MIXER_PCM)) == (1 << SOUND_MIXER_PCM)) {
|
|
|
|
emumix_set_volume(sc->card, M_FX0_REC_L, emumix_get_volume(sc->card, M_FX0_FRONT_L));
|
|
|
|
emumix_set_volume(sc->card, M_FX1_REC_R, emumix_get_volume(sc->card, M_FX1_FRONT_R));
|
|
|
|
} else {
|
|
|
|
emumix_set_volume(sc->card, M_FX0_REC_L, 0);
|
|
|
|
emumix_set_volume(sc->card, M_FX1_REC_R, 0);
|
|
|
|
}
|
|
|
|
|
|
|
|
return (recmask);
|
2006-07-15 19:36:28 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
static kobj_method_t emudspmixer_methods[] = {
|
|
|
|
KOBJMETHOD(mixer_init, emu_dspmixer_init),
|
2007-09-12 07:43:43 +00:00
|
|
|
KOBJMETHOD(mixer_uninit, emu_dspmixer_uninit),
|
2006-07-15 19:36:28 +00:00
|
|
|
KOBJMETHOD(mixer_set, emu_dspmixer_set),
|
|
|
|
KOBJMETHOD(mixer_setrecsrc, emu_dspmixer_setrecsrc),
|
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
|
|
|
KOBJMETHOD_END
|
2006-07-15 19:36:28 +00:00
|
|
|
};
|
|
|
|
MIXER_DECLARE(emudspmixer);
|
|
|
|
|
2007-09-12 07:43:43 +00:00
|
|
|
static int
|
|
|
|
emu_efxmixer_init(struct snd_mixer *m)
|
|
|
|
{
|
|
|
|
mix_setdevs(m, SOUND_MASK_VOLUME);
|
|
|
|
mix_setrecdevs(m, SOUND_MASK_MONITOR);
|
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
emu_efxmixer_set(struct snd_mixer *m, unsigned dev, unsigned left, unsigned right)
|
|
|
|
{
|
|
|
|
if (left + right == 200) return (0);
|
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
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
|
|
|
static u_int32_t
|
2007-09-12 07:43:43 +00:00
|
|
|
emu_efxmixer_setrecsrc(struct snd_mixer *m __unused, u_int32_t src __unused)
|
|
|
|
{
|
|
|
|
return (SOUND_MASK_MONITOR);
|
|
|
|
}
|
|
|
|
|
|
|
|
static kobj_method_t emuefxmixer_methods[] = {
|
|
|
|
KOBJMETHOD(mixer_init, emu_efxmixer_init),
|
|
|
|
KOBJMETHOD(mixer_set, emu_efxmixer_set),
|
|
|
|
KOBJMETHOD(mixer_setrecsrc, emu_efxmixer_setrecsrc),
|
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
|
|
|
KOBJMETHOD_END
|
2007-09-12 07:43:43 +00:00
|
|
|
};
|
|
|
|
MIXER_DECLARE(emuefxmixer);
|
|
|
|
|
2006-07-15 19:36:28 +00:00
|
|
|
/*
|
|
|
|
* AC97 emulation code for Audigy and later cards.
|
|
|
|
* Some parts of AC97 codec are not used by hardware, but can be used
|
|
|
|
* to change some DSP controls via AC97 mixer interface. This includes:
|
|
|
|
* - master volume controls MASTER_FRONT_[R|L]
|
|
|
|
* - pcm volume controls FX[0|1]_FRONT_[R|L]
|
|
|
|
* - rec volume controls MASTER_REC_[R|L]
|
|
|
|
* We do it because we need to put it under user control....
|
|
|
|
* We also keep some parts of AC97 disabled to get better sound quality
|
|
|
|
*/
|
|
|
|
|
|
|
|
#define AC97LEFT(x) ((x & 0x7F00)>>8)
|
|
|
|
#define AC97RIGHT(x) (x & 0x007F)
|
|
|
|
#define AC97MUTE(x) ((x & 0x8000)>>15)
|
|
|
|
#define BIT4_TO100(x) (100-(x)*100/(0x0f))
|
|
|
|
#define BIT6_TO100(x) (100-(x)*100/(0x3f))
|
|
|
|
#define BIT4_TO255(x) (255-(x)*255/(0x0f))
|
|
|
|
#define BIT6_TO255(x) (255-(x)*255/(0x3f))
|
|
|
|
#define V100_TOBIT6(x) (0x3f*(100-x)/100)
|
|
|
|
#define V100_TOBIT4(x) (0x0f*(100-x)/100)
|
2007-09-12 07:43:43 +00:00
|
|
|
#define AC97ENCODE(x_muted, x_left, x_right) (((x_muted & 1)<<15) | ((x_left & 0x3f)<<8) | (x_right & 0x3f))
|
2006-07-15 19:36:28 +00:00
|
|
|
|
|
|
|
static int
|
|
|
|
emu_ac97_read_emulation(struct emu_pcm_info *sc, int regno)
|
|
|
|
{
|
|
|
|
int use_ac97;
|
|
|
|
int emulated;
|
|
|
|
int tmp;
|
|
|
|
|
|
|
|
use_ac97 = 1;
|
|
|
|
emulated = 0;
|
|
|
|
|
|
|
|
switch (regno) {
|
|
|
|
case AC97_MIX_MASTER:
|
|
|
|
emulated = sc->ac97_state[AC97_MIX_MASTER];
|
|
|
|
use_ac97 = 0;
|
|
|
|
break;
|
|
|
|
case AC97_MIX_PCM:
|
|
|
|
emulated = sc->ac97_state[AC97_MIX_PCM];
|
|
|
|
use_ac97 = 0;
|
|
|
|
break;
|
|
|
|
case AC97_REG_RECSEL:
|
|
|
|
emulated = 0x0505;
|
|
|
|
use_ac97 = 0;
|
|
|
|
break;
|
|
|
|
case AC97_MIX_RGAIN:
|
|
|
|
emulated = sc->ac97_state[AC97_MIX_RGAIN];
|
|
|
|
use_ac97 = 0;
|
|
|
|
break;
|
|
|
|
}
|
|
|
|
|
2012-01-11 21:17:14 +00:00
|
|
|
emu_wr(sc->card, EMU_AC97ADDR, regno, 1);
|
|
|
|
tmp = emu_rd(sc->card, EMU_AC97DATA, 2);
|
2006-07-15 19:36:28 +00:00
|
|
|
|
|
|
|
if (use_ac97)
|
|
|
|
emulated = tmp;
|
|
|
|
|
|
|
|
return (emulated);
|
|
|
|
}
|
|
|
|
|
|
|
|
static void
|
|
|
|
emu_ac97_write_emulation(struct emu_pcm_info *sc, int regno, uint32_t data)
|
|
|
|
{
|
|
|
|
int write_ac97;
|
|
|
|
int left, right;
|
|
|
|
uint32_t emu_left, emu_right;
|
|
|
|
int is_mute;
|
|
|
|
|
|
|
|
write_ac97 = 1;
|
|
|
|
|
|
|
|
left = AC97LEFT(data);
|
|
|
|
emu_left = BIT6_TO100(left); /* We show us as 6-bit AC97 mixer */
|
|
|
|
right = AC97RIGHT(data);
|
|
|
|
emu_right = BIT6_TO100(right);
|
|
|
|
is_mute = AC97MUTE(data);
|
|
|
|
if (is_mute)
|
|
|
|
emu_left = emu_right = 0;
|
|
|
|
|
|
|
|
switch (regno) {
|
|
|
|
/* TODO: reset emulator on AC97_RESET */
|
|
|
|
case AC97_MIX_MASTER:
|
|
|
|
emumix_set_volume(sc->card, M_MASTER_FRONT_L, emu_left);
|
|
|
|
emumix_set_volume(sc->card, M_MASTER_FRONT_R, emu_right);
|
|
|
|
sc->ac97_state[AC97_MIX_MASTER] = data & (0x8000 | 0x3f3f);
|
|
|
|
data = 0x8000; /* Mute AC97 main out */
|
|
|
|
break;
|
|
|
|
case AC97_MIX_PCM: /* PCM OUT VOL */
|
|
|
|
emumix_set_volume(sc->card, M_FX0_FRONT_L, emu_left);
|
|
|
|
emumix_set_volume(sc->card, M_FX1_FRONT_R, emu_right);
|
|
|
|
sc->ac97_state[AC97_MIX_PCM] = data & (0x8000 | 0x3f3f);
|
|
|
|
data = 0x8000; /* Mute AC97 PCM out */
|
|
|
|
break;
|
|
|
|
case AC97_REG_RECSEL:
|
|
|
|
/*
|
|
|
|
* PCM recording source is set to "stereo mix" (labeled "vol"
|
2007-09-12 07:43:43 +00:00
|
|
|
* in mixer). There is no 'playback' from AC97 codec -
|
|
|
|
* if you want to hear anything from AC97 you have to _record_
|
|
|
|
* it. Keep things simple and record "stereo mix".
|
2006-07-15 19:36:28 +00:00
|
|
|
*/
|
|
|
|
data = 0x0505;
|
|
|
|
break;
|
|
|
|
case AC97_MIX_RGAIN: /* RECORD GAIN */
|
|
|
|
emu_left = BIT4_TO100(left); /* rgain is 4-bit */
|
|
|
|
emu_right = BIT4_TO100(right);
|
|
|
|
emumix_set_volume(sc->card, M_MASTER_REC_L, 100-emu_left);
|
|
|
|
emumix_set_volume(sc->card, M_MASTER_REC_R, 100-emu_right);
|
|
|
|
/*
|
|
|
|
* Record gain on AC97 should stay zero to get AC97 sound on
|
|
|
|
* AC97_[RL] connectors on EMU10K2 chip. AC97 on Audigy is not
|
|
|
|
* directly connected to any output, only to EMU10K2 chip Use
|
|
|
|
* this control to set AC97 mix volume inside EMU10K2 chip
|
|
|
|
*/
|
|
|
|
sc->ac97_state[AC97_MIX_RGAIN] = data & (0x8000 | 0x0f0f);
|
|
|
|
data = 0x0000;
|
|
|
|
break;
|
|
|
|
}
|
|
|
|
if (write_ac97) {
|
2012-01-11 21:17:14 +00:00
|
|
|
emu_wr(sc->card, EMU_AC97ADDR, regno, 1);
|
|
|
|
emu_wr(sc->card, EMU_AC97DATA, data, 2);
|
2006-07-15 19:36:28 +00:00
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
emu_erdcd(kobj_t obj __unused, void *devinfo, int regno)
|
|
|
|
{
|
|
|
|
struct emu_pcm_info *sc = (struct emu_pcm_info *)devinfo;
|
|
|
|
|
|
|
|
return (emu_ac97_read_emulation(sc, regno));
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
emu_ewrcd(kobj_t obj __unused, void *devinfo, int regno, uint32_t data)
|
|
|
|
{
|
|
|
|
struct emu_pcm_info *sc = (struct emu_pcm_info *)devinfo;
|
|
|
|
|
|
|
|
emu_ac97_write_emulation(sc, regno, data);
|
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
|
|
|
static kobj_method_t emu_eac97_methods[] = {
|
|
|
|
KOBJMETHOD(ac97_read, emu_erdcd),
|
|
|
|
KOBJMETHOD(ac97_write, emu_ewrcd),
|
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
|
|
|
KOBJMETHOD_END
|
2006-07-15 19:36:28 +00:00
|
|
|
};
|
|
|
|
AC97_DECLARE(emu_eac97);
|
|
|
|
|
|
|
|
/* real ac97 codec */
|
|
|
|
static int
|
|
|
|
emu_rdcd(kobj_t obj __unused, void *devinfo, int regno)
|
|
|
|
{
|
|
|
|
int rd;
|
|
|
|
struct emu_pcm_info *sc = (struct emu_pcm_info *)devinfo;
|
|
|
|
|
|
|
|
KASSERT(sc->card != NULL, ("emu_rdcd: no soundcard"));
|
2012-01-11 21:17:14 +00:00
|
|
|
emu_wr(sc->card, EMU_AC97ADDR, regno, 1);
|
|
|
|
rd = emu_rd(sc->card, EMU_AC97DATA, 2);
|
2006-07-15 19:36:28 +00:00
|
|
|
return (rd);
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
emu_wrcd(kobj_t obj __unused, void *devinfo, int regno, uint32_t data)
|
|
|
|
{
|
|
|
|
struct emu_pcm_info *sc = (struct emu_pcm_info *)devinfo;
|
|
|
|
|
|
|
|
KASSERT(sc->card != NULL, ("emu_wrcd: no soundcard"));
|
2012-01-11 21:17:14 +00:00
|
|
|
emu_wr(sc->card, EMU_AC97ADDR, regno, 1);
|
|
|
|
emu_wr(sc->card, EMU_AC97DATA, data, 2);
|
2006-07-15 19:36:28 +00:00
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
|
|
|
static kobj_method_t emu_ac97_methods[] = {
|
|
|
|
KOBJMETHOD(ac97_read, emu_rdcd),
|
|
|
|
KOBJMETHOD(ac97_write, emu_wrcd),
|
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
|
|
|
KOBJMETHOD_END
|
2006-07-15 19:36:28 +00:00
|
|
|
};
|
|
|
|
AC97_DECLARE(emu_ac97);
|
|
|
|
|
|
|
|
|
|
|
|
static int
|
|
|
|
emu_k1_recval(int speed)
|
|
|
|
{
|
|
|
|
int val;
|
|
|
|
|
|
|
|
val = 0;
|
|
|
|
while ((val < 7) && (speed < emu10k1_adcspeed[val]))
|
|
|
|
val++;
|
|
|
|
return (val);
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
emu_k2_recval(int speed)
|
|
|
|
{
|
|
|
|
int val;
|
|
|
|
|
|
|
|
val = 0;
|
|
|
|
while ((val < 8) && (speed < emu10k2_adcspeed[val]))
|
|
|
|
val++;
|
|
|
|
return (val);
|
|
|
|
}
|
|
|
|
|
|
|
|
static void *
|
|
|
|
emupchan_init(kobj_t obj __unused, void *devinfo, struct snd_dbuf *b, struct pcm_channel *c, int dir __unused)
|
|
|
|
{
|
|
|
|
struct emu_pcm_info *sc = devinfo;
|
|
|
|
struct emu_pcm_pchinfo *ch;
|
|
|
|
void *r;
|
|
|
|
|
|
|
|
KASSERT(dir == PCMDIR_PLAY, ("emupchan_init: bad direction"));
|
|
|
|
KASSERT(sc->card != NULL, ("empchan_init: no soundcard"));
|
|
|
|
|
|
|
|
|
|
|
|
if (sc->pnum >= MAX_CHANNELS)
|
|
|
|
return (NULL);
|
|
|
|
ch = &(sc->pch[sc->pnum++]);
|
|
|
|
ch->buffer = b;
|
|
|
|
ch->pcm = sc;
|
|
|
|
ch->channel = c;
|
2007-01-06 18:59:35 +00:00
|
|
|
ch->blksz = sc->bufsz;
|
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
|
|
|
ch->fmt = SND_FORMAT(AFMT_U8, 1, 0);
|
2006-07-15 19:36:28 +00:00
|
|
|
ch->spd = 8000;
|
|
|
|
ch->master = emu_valloc(sc->card);
|
|
|
|
/*
|
|
|
|
* XXX we have to allocate slave even for mono channel until we
|
|
|
|
* fix emu_vfree to handle this case.
|
|
|
|
*/
|
|
|
|
ch->slave = emu_valloc(sc->card);
|
|
|
|
ch->timer = emu_timer_create(sc->card);
|
2006-08-07 22:44:01 +00:00
|
|
|
r = (emu_vinit(sc->card, ch->master, ch->slave, EMU_PLAY_BUFSZ, ch->buffer)) ? NULL : ch;
|
2006-07-15 19:36:28 +00:00
|
|
|
return (r);
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
emupchan_free(kobj_t obj __unused, void *c_devinfo)
|
|
|
|
{
|
|
|
|
struct emu_pcm_pchinfo *ch = c_devinfo;
|
|
|
|
struct emu_pcm_info *sc = ch->pcm;
|
|
|
|
|
|
|
|
emu_timer_clear(sc->card, ch->timer);
|
|
|
|
if (ch->slave != NULL)
|
|
|
|
emu_vfree(sc->card, ch->slave);
|
|
|
|
emu_vfree(sc->card, ch->master);
|
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
emupchan_setformat(kobj_t obj __unused, void *c_devinfo, uint32_t format)
|
|
|
|
{
|
|
|
|
struct emu_pcm_pchinfo *ch = c_devinfo;
|
|
|
|
|
|
|
|
ch->fmt = format;
|
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
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
|
|
|
static uint32_t
|
2006-07-15 19:36:28 +00:00
|
|
|
emupchan_setspeed(kobj_t obj __unused, void *c_devinfo, uint32_t speed)
|
|
|
|
{
|
|
|
|
struct emu_pcm_pchinfo *ch = c_devinfo;
|
|
|
|
|
|
|
|
ch->spd = speed;
|
|
|
|
return (ch->spd);
|
|
|
|
}
|
|
|
|
|
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
|
|
|
static uint32_t
|
2006-07-15 19:36:28 +00:00
|
|
|
emupchan_setblocksize(kobj_t obj __unused, void *c_devinfo, uint32_t blocksize)
|
|
|
|
{
|
|
|
|
struct emu_pcm_pchinfo *ch = c_devinfo;
|
|
|
|
struct emu_pcm_info *sc = ch->pcm;
|
|
|
|
|
|
|
|
if (blocksize > ch->pcm->bufsz)
|
|
|
|
blocksize = ch->pcm->bufsz;
|
|
|
|
snd_mtxlock(sc->lock);
|
|
|
|
ch->blksz = blocksize;
|
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
|
|
|
emu_timer_set(sc->card, ch->timer, ch->blksz / sndbuf_getalign(ch->buffer));
|
2006-07-15 19:36:28 +00:00
|
|
|
snd_mtxunlock(sc->lock);
|
2007-09-12 07:43:43 +00:00
|
|
|
return (ch->blksz);
|
2006-07-15 19:36:28 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
emupchan_trigger(kobj_t obj __unused, void *c_devinfo, int go)
|
|
|
|
{
|
|
|
|
struct emu_pcm_pchinfo *ch = c_devinfo;
|
|
|
|
struct emu_pcm_info *sc = ch->pcm;
|
|
|
|
|
2007-06-11 00:49:46 +00:00
|
|
|
if (!PCMTRIG_COMMON(go))
|
2006-07-15 19:36:28 +00:00
|
|
|
return (0);
|
2007-06-11 00:49:46 +00:00
|
|
|
|
2006-07-15 19:36:28 +00:00
|
|
|
snd_mtxlock(sc->lock); /* XXX can we trigger on parallel threads ? */
|
|
|
|
if (go == PCMTRIG_START) {
|
|
|
|
emu_vsetup(ch->master, ch->fmt, ch->spd);
|
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
|
|
|
if (AFMT_CHANNEL(ch->fmt) > 1)
|
2006-08-07 22:55:21 +00:00
|
|
|
emu_vroute(sc->card, &(sc->rt), ch->master);
|
|
|
|
else
|
|
|
|
emu_vroute(sc->card, &(sc->rt_mono), ch->master);
|
2006-07-15 19:36:28 +00:00
|
|
|
emu_vwrite(sc->card, ch->master);
|
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
|
|
|
emu_timer_set(sc->card, ch->timer, ch->blksz / sndbuf_getalign(ch->buffer));
|
2006-07-15 19:36:28 +00:00
|
|
|
emu_timer_enable(sc->card, ch->timer, 1);
|
|
|
|
}
|
|
|
|
/* PCM interrupt handler will handle PCMTRIG_STOP event */
|
|
|
|
ch->run = (go == PCMTRIG_START) ? 1 : 0;
|
|
|
|
emu_vtrigger(sc->card, ch->master, ch->run);
|
|
|
|
snd_mtxunlock(sc->lock);
|
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
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
|
|
|
static uint32_t
|
2006-07-15 19:36:28 +00:00
|
|
|
emupchan_getptr(kobj_t obj __unused, void *c_devinfo)
|
|
|
|
{
|
|
|
|
struct emu_pcm_pchinfo *ch = c_devinfo;
|
|
|
|
struct emu_pcm_info *sc = ch->pcm;
|
|
|
|
int r;
|
|
|
|
|
|
|
|
r = emu_vpos(sc->card, ch->master);
|
|
|
|
|
|
|
|
return (r);
|
|
|
|
}
|
|
|
|
|
|
|
|
static struct pcmchan_caps *
|
|
|
|
emupchan_getcaps(kobj_t obj __unused, void *c_devinfo __unused)
|
|
|
|
{
|
|
|
|
struct emu_pcm_pchinfo *ch = c_devinfo;
|
|
|
|
struct emu_pcm_info *sc = ch->pcm;
|
|
|
|
|
|
|
|
switch (sc->route) {
|
|
|
|
case RT_FRONT:
|
|
|
|
/* FALLTHROUGH */
|
|
|
|
case RT_REAR:
|
|
|
|
/* FALLTHROUGH */
|
|
|
|
case RT_SIDE:
|
|
|
|
return (&emu_playcaps);
|
|
|
|
break;
|
|
|
|
case RT_CENTER:
|
|
|
|
/* FALLTHROUGH */
|
|
|
|
case RT_SUB:
|
|
|
|
return (&emu_playcaps_mono);
|
|
|
|
break;
|
|
|
|
}
|
|
|
|
return (NULL);
|
|
|
|
}
|
|
|
|
|
|
|
|
static kobj_method_t emupchan_methods[] = {
|
|
|
|
KOBJMETHOD(channel_init, emupchan_init),
|
|
|
|
KOBJMETHOD(channel_free, emupchan_free),
|
|
|
|
KOBJMETHOD(channel_setformat, emupchan_setformat),
|
|
|
|
KOBJMETHOD(channel_setspeed, emupchan_setspeed),
|
|
|
|
KOBJMETHOD(channel_setblocksize, emupchan_setblocksize),
|
|
|
|
KOBJMETHOD(channel_trigger, emupchan_trigger),
|
|
|
|
KOBJMETHOD(channel_getptr, emupchan_getptr),
|
|
|
|
KOBJMETHOD(channel_getcaps, emupchan_getcaps),
|
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
|
|
|
KOBJMETHOD_END
|
2006-07-15 19:36:28 +00:00
|
|
|
};
|
|
|
|
CHANNEL_DECLARE(emupchan);
|
|
|
|
|
|
|
|
static void *
|
|
|
|
emurchan_init(kobj_t obj __unused, void *devinfo, struct snd_dbuf *b, struct pcm_channel *c, int dir __unused)
|
|
|
|
{
|
|
|
|
struct emu_pcm_info *sc = devinfo;
|
|
|
|
struct emu_pcm_rchinfo *ch;
|
|
|
|
|
|
|
|
KASSERT(dir == PCMDIR_REC, ("emurchan_init: bad direction"));
|
2007-01-06 18:59:35 +00:00
|
|
|
ch = &sc->rch_adc;
|
2006-07-15 19:36:28 +00:00
|
|
|
ch->buffer = b;
|
|
|
|
ch->pcm = sc;
|
|
|
|
ch->channel = c;
|
2007-01-06 18:59:35 +00:00
|
|
|
ch->blksz = sc->bufsz / 2; /* We rise interrupt for half-full buffer */
|
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
|
|
|
ch->fmt = SND_FORMAT(AFMT_U8, 1, 0);
|
2007-01-06 18:59:35 +00:00
|
|
|
ch->spd = 8000;
|
2012-01-11 21:17:14 +00:00
|
|
|
ch->idxreg = sc->is_emu10k1 ? EMU_ADCIDX : EMU_A_ADCIDX;
|
|
|
|
ch->basereg = EMU_ADCBA;
|
|
|
|
ch->sizereg = EMU_ADCBS;
|
|
|
|
ch->setupreg = EMU_ADCCR;
|
|
|
|
ch->irqmask = EMU_INTE_ADCBUFENABLE;
|
|
|
|
ch->iprmask = EMU_IPR_ADCBUFFULL | EMU_IPR_ADCBUFHALFFULL;
|
2006-07-15 19:36:28 +00:00
|
|
|
|
2007-04-18 18:26:41 +00:00
|
|
|
if (sndbuf_alloc(ch->buffer, emu_gettag(sc->card), 0, sc->bufsz) != 0)
|
2006-07-15 19:36:28 +00:00
|
|
|
return (NULL);
|
|
|
|
else {
|
2007-09-12 07:43:43 +00:00
|
|
|
ch->timer = emu_timer_create(sc->card);
|
2006-07-15 19:36:28 +00:00
|
|
|
emu_wrptr(sc->card, 0, ch->basereg, sndbuf_getbufaddr(ch->buffer));
|
|
|
|
emu_wrptr(sc->card, 0, ch->sizereg, 0); /* off */
|
|
|
|
return (ch);
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
2007-09-12 07:43:43 +00:00
|
|
|
static int
|
|
|
|
emurchan_free(kobj_t obj __unused, void *c_devinfo)
|
|
|
|
{
|
|
|
|
struct emu_pcm_rchinfo *ch = c_devinfo;
|
|
|
|
struct emu_pcm_info *sc = ch->pcm;
|
|
|
|
|
|
|
|
emu_timer_clear(sc->card, ch->timer);
|
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
2006-07-15 19:36:28 +00:00
|
|
|
static int
|
|
|
|
emurchan_setformat(kobj_t obj __unused, void *c_devinfo, uint32_t format)
|
|
|
|
{
|
|
|
|
struct emu_pcm_rchinfo *ch = c_devinfo;
|
|
|
|
|
|
|
|
ch->fmt = format;
|
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
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
|
|
|
static uint32_t
|
2006-07-15 19:36:28 +00:00
|
|
|
emurchan_setspeed(kobj_t obj __unused, void *c_devinfo, uint32_t speed)
|
|
|
|
{
|
|
|
|
struct emu_pcm_rchinfo *ch = c_devinfo;
|
|
|
|
|
|
|
|
if (ch->pcm->is_emu10k1) {
|
|
|
|
speed = emu10k1_adcspeed[emu_k1_recval(speed)];
|
|
|
|
} else {
|
|
|
|
speed = emu10k2_adcspeed[emu_k2_recval(speed)];
|
|
|
|
}
|
|
|
|
ch->spd = speed;
|
|
|
|
return (ch->spd);
|
|
|
|
}
|
|
|
|
|
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
|
|
|
static uint32_t
|
2006-07-15 19:36:28 +00:00
|
|
|
emurchan_setblocksize(kobj_t obj __unused, void *c_devinfo, uint32_t blocksize)
|
|
|
|
{
|
|
|
|
struct emu_pcm_rchinfo *ch = c_devinfo;
|
2007-09-12 07:43:43 +00:00
|
|
|
struct emu_pcm_info *sc = ch->pcm;
|
2006-07-15 19:36:28 +00:00
|
|
|
|
|
|
|
ch->blksz = blocksize;
|
2007-09-12 07:43:43 +00:00
|
|
|
/*
|
|
|
|
* If blocksize is less than half of buffer size we will not get
|
|
|
|
* BUFHALFFULL interrupt in time and channel will need to generate
|
|
|
|
* (and use) timer interrupts. Otherwise channel will be marked dead.
|
|
|
|
*/
|
|
|
|
if (ch->blksz < (ch->pcm->bufsz / 2)) {
|
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
|
|
|
emu_timer_set(sc->card, ch->timer, ch->blksz / sndbuf_getalign(ch->buffer));
|
2007-09-12 07:43:43 +00:00
|
|
|
emu_timer_enable(sc->card, ch->timer, 1);
|
|
|
|
} else {
|
|
|
|
emu_timer_enable(sc->card, ch->timer, 0);
|
|
|
|
}
|
2007-01-06 18:59:35 +00:00
|
|
|
return (ch->blksz);
|
2006-07-15 19:36:28 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
emurchan_trigger(kobj_t obj __unused, void *c_devinfo, int go)
|
|
|
|
{
|
|
|
|
struct emu_pcm_rchinfo *ch = c_devinfo;
|
|
|
|
struct emu_pcm_info *sc = ch->pcm;
|
|
|
|
uint32_t val, sz;
|
|
|
|
|
2007-06-11 00:49:46 +00:00
|
|
|
if (!PCMTRIG_COMMON(go))
|
|
|
|
return (0);
|
|
|
|
|
2006-07-15 19:36:28 +00:00
|
|
|
switch (sc->bufsz) {
|
|
|
|
case 4096:
|
2012-01-11 21:17:14 +00:00
|
|
|
sz = EMU_RECBS_BUFSIZE_4096;
|
2006-07-15 19:36:28 +00:00
|
|
|
break;
|
|
|
|
case 8192:
|
2012-01-11 21:17:14 +00:00
|
|
|
sz = EMU_RECBS_BUFSIZE_8192;
|
2006-07-15 19:36:28 +00:00
|
|
|
break;
|
|
|
|
case 16384:
|
2012-01-11 21:17:14 +00:00
|
|
|
sz = EMU_RECBS_BUFSIZE_16384;
|
2006-07-15 19:36:28 +00:00
|
|
|
break;
|
|
|
|
case 32768:
|
2012-01-11 21:17:14 +00:00
|
|
|
sz = EMU_RECBS_BUFSIZE_32768;
|
2006-07-15 19:36:28 +00:00
|
|
|
break;
|
|
|
|
case 65536:
|
2012-01-11 21:17:14 +00:00
|
|
|
sz = EMU_RECBS_BUFSIZE_65536;
|
2006-07-15 19:36:28 +00:00
|
|
|
break;
|
|
|
|
default:
|
2012-01-11 21:17:14 +00:00
|
|
|
sz = EMU_RECBS_BUFSIZE_4096;
|
2006-07-15 19:36:28 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
snd_mtxlock(sc->lock);
|
|
|
|
switch (go) {
|
|
|
|
case PCMTRIG_START:
|
|
|
|
ch->run = 1;
|
|
|
|
emu_wrptr(sc->card, 0, ch->sizereg, sz);
|
2012-01-11 21:17:14 +00:00
|
|
|
val = sc->is_emu10k1 ? EMU_ADCCR_LCHANENABLE : EMU_A_ADCCR_LCHANENABLE;
|
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
|
|
|
if (AFMT_CHANNEL(ch->fmt) > 1)
|
2012-01-11 21:17:14 +00:00
|
|
|
val |= sc->is_emu10k1 ? EMU_ADCCR_RCHANENABLE : EMU_A_ADCCR_RCHANENABLE;
|
2006-07-15 19:36:28 +00:00
|
|
|
val |= sc->is_emu10k1 ? emu_k1_recval(ch->spd) : emu_k2_recval(ch->spd);
|
|
|
|
emu_wrptr(sc->card, 0, ch->setupreg, 0);
|
|
|
|
emu_wrptr(sc->card, 0, ch->setupreg, val);
|
|
|
|
ch->ihandle = emu_intr_register(sc->card, ch->irqmask, ch->iprmask, &emu_pcm_intr, sc);
|
|
|
|
break;
|
|
|
|
case PCMTRIG_STOP:
|
|
|
|
/* FALLTHROUGH */
|
|
|
|
case PCMTRIG_ABORT:
|
|
|
|
ch->run = 0;
|
|
|
|
emu_wrptr(sc->card, 0, ch->sizereg, 0);
|
|
|
|
if (ch->setupreg)
|
|
|
|
emu_wrptr(sc->card, 0, ch->setupreg, 0);
|
|
|
|
(void)emu_intr_unregister(sc->card, ch->ihandle);
|
|
|
|
break;
|
|
|
|
case PCMTRIG_EMLDMAWR:
|
|
|
|
/* FALLTHROUGH */
|
|
|
|
case PCMTRIG_EMLDMARD:
|
|
|
|
/* FALLTHROUGH */
|
|
|
|
default:
|
|
|
|
break;
|
|
|
|
}
|
|
|
|
snd_mtxunlock(sc->lock);
|
|
|
|
|
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
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
|
|
|
static uint32_t
|
2006-07-15 19:36:28 +00:00
|
|
|
emurchan_getptr(kobj_t obj __unused, void *c_devinfo)
|
|
|
|
{
|
|
|
|
struct emu_pcm_rchinfo *ch = c_devinfo;
|
|
|
|
struct emu_pcm_info *sc = ch->pcm;
|
|
|
|
int r;
|
|
|
|
|
|
|
|
r = emu_rdptr(sc->card, 0, ch->idxreg) & 0x0000ffff;
|
|
|
|
|
|
|
|
return (r);
|
|
|
|
}
|
|
|
|
|
|
|
|
static struct pcmchan_caps *
|
|
|
|
emurchan_getcaps(kobj_t obj __unused, void *c_devinfo __unused)
|
|
|
|
{
|
2007-01-06 18:59:35 +00:00
|
|
|
return (&emu_reccaps_adc);
|
2006-07-15 19:36:28 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
static kobj_method_t emurchan_methods[] = {
|
|
|
|
KOBJMETHOD(channel_init, emurchan_init),
|
2007-09-12 07:43:43 +00:00
|
|
|
KOBJMETHOD(channel_free, emurchan_free),
|
2006-07-15 19:36:28 +00:00
|
|
|
KOBJMETHOD(channel_setformat, emurchan_setformat),
|
|
|
|
KOBJMETHOD(channel_setspeed, emurchan_setspeed),
|
|
|
|
KOBJMETHOD(channel_setblocksize, emurchan_setblocksize),
|
|
|
|
KOBJMETHOD(channel_trigger, emurchan_trigger),
|
|
|
|
KOBJMETHOD(channel_getptr, emurchan_getptr),
|
|
|
|
KOBJMETHOD(channel_getcaps, emurchan_getcaps),
|
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
|
|
|
KOBJMETHOD_END
|
2006-07-15 19:36:28 +00:00
|
|
|
};
|
|
|
|
CHANNEL_DECLARE(emurchan);
|
|
|
|
|
2007-01-06 18:59:35 +00:00
|
|
|
static void *
|
|
|
|
emufxrchan_init(kobj_t obj __unused, void *devinfo, struct snd_dbuf *b, struct pcm_channel *c, int dir __unused)
|
|
|
|
{
|
|
|
|
struct emu_pcm_info *sc = devinfo;
|
|
|
|
struct emu_pcm_rchinfo *ch;
|
|
|
|
|
|
|
|
KASSERT(dir == PCMDIR_REC, ("emurchan_init: bad direction"));
|
|
|
|
|
|
|
|
if (sc == NULL) return (NULL);
|
|
|
|
|
|
|
|
ch = &(sc->rch_efx);
|
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
|
|
|
ch->fmt = SND_FORMAT(AFMT_S16_LE, 1, 0);
|
2007-01-06 18:59:35 +00:00
|
|
|
ch->spd = sc->is_emu10k1 ? 48000*32 : 48000 * 64;
|
2012-01-11 21:17:14 +00:00
|
|
|
ch->idxreg = EMU_FXIDX;
|
|
|
|
ch->basereg = EMU_FXBA;
|
|
|
|
ch->sizereg = EMU_FXBS;
|
|
|
|
ch->irqmask = EMU_INTE_EFXBUFENABLE;
|
|
|
|
ch->iprmask = EMU_IPR_EFXBUFFULL | EMU_IPR_EFXBUFHALFFULL;
|
2007-01-06 18:59:35 +00:00
|
|
|
ch->buffer = b;
|
|
|
|
ch->pcm = sc;
|
|
|
|
ch->channel = c;
|
2007-09-12 07:43:43 +00:00
|
|
|
ch->blksz = sc->bufsz / 2;
|
2007-01-06 18:59:35 +00:00
|
|
|
|
2007-04-18 18:26:41 +00:00
|
|
|
if (sndbuf_alloc(ch->buffer, emu_gettag(sc->card), 0, sc->bufsz) != 0)
|
2007-01-06 18:59:35 +00:00
|
|
|
return (NULL);
|
|
|
|
else {
|
|
|
|
emu_wrptr(sc->card, 0, ch->basereg, sndbuf_getbufaddr(ch->buffer));
|
|
|
|
emu_wrptr(sc->card, 0, ch->sizereg, 0); /* off */
|
|
|
|
return (ch);
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
emufxrchan_setformat(kobj_t obj __unused, void *c_devinfo __unused, uint32_t format)
|
|
|
|
{
|
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
|
|
|
if (format == SND_FORMAT(AFMT_S16_LE, 1, 0)) return (0);
|
2007-09-12 07:43:43 +00:00
|
|
|
return (EINVAL);
|
2007-01-06 18:59:35 +00:00
|
|
|
}
|
|
|
|
|
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
|
|
|
static uint32_t
|
2007-01-06 18:59:35 +00:00
|
|
|
emufxrchan_setspeed(kobj_t obj __unused, void *c_devinfo, uint32_t speed)
|
|
|
|
{
|
|
|
|
struct emu_pcm_rchinfo *ch = c_devinfo;
|
|
|
|
|
|
|
|
/* FIXED RATE CHANNEL */
|
|
|
|
return (ch->spd);
|
|
|
|
}
|
|
|
|
|
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
|
|
|
static uint32_t
|
2007-01-06 18:59:35 +00:00
|
|
|
emufxrchan_setblocksize(kobj_t obj __unused, void *c_devinfo, uint32_t blocksize)
|
|
|
|
{
|
|
|
|
struct emu_pcm_rchinfo *ch = c_devinfo;
|
|
|
|
|
|
|
|
ch->blksz = blocksize;
|
2007-09-12 07:43:43 +00:00
|
|
|
/*
|
|
|
|
* XXX If blocksize is less than half of buffer size we will not get
|
|
|
|
* interrupt in time and channel will die due to interrupt timeout.
|
|
|
|
* This should not happen with FX rchan, because it will fill buffer
|
|
|
|
* very fast (64K buffer is 0.021seconds on Audigy).
|
|
|
|
*/
|
|
|
|
if (ch->blksz < (ch->pcm->bufsz / 2))
|
2007-01-06 18:59:35 +00:00
|
|
|
ch->blksz = ch->pcm->bufsz / 2;
|
|
|
|
return (ch->blksz);
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
emufxrchan_trigger(kobj_t obj __unused, void *c_devinfo, int go)
|
|
|
|
{
|
|
|
|
struct emu_pcm_rchinfo *ch = c_devinfo;
|
|
|
|
struct emu_pcm_info *sc = ch->pcm;
|
|
|
|
uint32_t sz;
|
|
|
|
|
2007-06-11 00:49:46 +00:00
|
|
|
if (!PCMTRIG_COMMON(go))
|
|
|
|
return (0);
|
|
|
|
|
2007-01-06 18:59:35 +00:00
|
|
|
switch (sc->bufsz) {
|
|
|
|
case 4096:
|
2012-01-11 21:17:14 +00:00
|
|
|
sz = EMU_RECBS_BUFSIZE_4096;
|
2007-01-06 18:59:35 +00:00
|
|
|
break;
|
|
|
|
case 8192:
|
2012-01-11 21:17:14 +00:00
|
|
|
sz = EMU_RECBS_BUFSIZE_8192;
|
2007-01-06 18:59:35 +00:00
|
|
|
break;
|
|
|
|
case 16384:
|
2012-01-11 21:17:14 +00:00
|
|
|
sz = EMU_RECBS_BUFSIZE_16384;
|
2007-01-06 18:59:35 +00:00
|
|
|
break;
|
|
|
|
case 32768:
|
2012-01-11 21:17:14 +00:00
|
|
|
sz = EMU_RECBS_BUFSIZE_32768;
|
2007-01-06 18:59:35 +00:00
|
|
|
break;
|
|
|
|
case 65536:
|
2012-01-11 21:17:14 +00:00
|
|
|
sz = EMU_RECBS_BUFSIZE_65536;
|
2007-01-06 18:59:35 +00:00
|
|
|
break;
|
|
|
|
default:
|
2012-01-11 21:17:14 +00:00
|
|
|
sz = EMU_RECBS_BUFSIZE_4096;
|
2007-01-06 18:59:35 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
snd_mtxlock(sc->lock);
|
|
|
|
switch (go) {
|
|
|
|
case PCMTRIG_START:
|
|
|
|
ch->run = 1;
|
|
|
|
emu_wrptr(sc->card, 0, ch->sizereg, sz);
|
|
|
|
ch->ihandle = emu_intr_register(sc->card, ch->irqmask, ch->iprmask, &emu_pcm_intr, sc);
|
2007-09-12 07:43:43 +00:00
|
|
|
/*
|
|
|
|
* SB Live! is limited to 32 mono channels. Audigy
|
|
|
|
* has 64 mono channels. Channels are enabled
|
2012-01-11 21:17:14 +00:00
|
|
|
* by setting a bit in EMU_A_FXWC[1|2] registers.
|
2007-01-06 18:59:35 +00:00
|
|
|
*/
|
|
|
|
/* XXX there is no way to demultiplex this streams for now */
|
2007-09-12 07:43:43 +00:00
|
|
|
if (sc->is_emu10k1) {
|
2012-01-11 21:17:14 +00:00
|
|
|
emu_wrptr(sc->card, 0, EMU_FXWC, 0xffffffff);
|
2007-01-06 18:59:35 +00:00
|
|
|
} else {
|
2012-01-11 21:17:14 +00:00
|
|
|
emu_wrptr(sc->card, 0, EMU_A_FXWC1, 0xffffffff);
|
|
|
|
emu_wrptr(sc->card, 0, EMU_A_FXWC2, 0xffffffff);
|
2007-01-06 18:59:35 +00:00
|
|
|
}
|
|
|
|
break;
|
|
|
|
case PCMTRIG_STOP:
|
|
|
|
/* FALLTHROUGH */
|
|
|
|
case PCMTRIG_ABORT:
|
|
|
|
ch->run = 0;
|
2007-09-12 07:43:43 +00:00
|
|
|
if (sc->is_emu10k1) {
|
2012-01-11 21:17:14 +00:00
|
|
|
emu_wrptr(sc->card, 0, EMU_FXWC, 0x0);
|
2007-01-06 18:59:35 +00:00
|
|
|
} else {
|
2012-01-11 21:17:14 +00:00
|
|
|
emu_wrptr(sc->card, 0, EMU_A_FXWC1, 0x0);
|
|
|
|
emu_wrptr(sc->card, 0, EMU_A_FXWC2, 0x0);
|
2007-01-06 18:59:35 +00:00
|
|
|
}
|
|
|
|
emu_wrptr(sc->card, 0, ch->sizereg, 0);
|
|
|
|
(void)emu_intr_unregister(sc->card, ch->ihandle);
|
|
|
|
break;
|
|
|
|
case PCMTRIG_EMLDMAWR:
|
|
|
|
/* FALLTHROUGH */
|
|
|
|
case PCMTRIG_EMLDMARD:
|
|
|
|
/* FALLTHROUGH */
|
|
|
|
default:
|
|
|
|
break;
|
|
|
|
}
|
|
|
|
snd_mtxunlock(sc->lock);
|
|
|
|
|
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
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
|
|
|
static uint32_t
|
2007-01-06 18:59:35 +00:00
|
|
|
emufxrchan_getptr(kobj_t obj __unused, void *c_devinfo)
|
|
|
|
{
|
|
|
|
struct emu_pcm_rchinfo *ch = c_devinfo;
|
|
|
|
struct emu_pcm_info *sc = ch->pcm;
|
|
|
|
int r;
|
|
|
|
|
|
|
|
r = emu_rdptr(sc->card, 0, ch->idxreg) & 0x0000ffff;
|
|
|
|
|
|
|
|
return (r);
|
|
|
|
}
|
|
|
|
|
|
|
|
static struct pcmchan_caps *
|
|
|
|
emufxrchan_getcaps(kobj_t obj __unused, void *c_devinfo)
|
|
|
|
{
|
|
|
|
struct emu_pcm_rchinfo *ch = c_devinfo;
|
|
|
|
struct emu_pcm_info *sc = ch->pcm;
|
|
|
|
|
2007-09-12 07:43:43 +00:00
|
|
|
if (sc->is_emu10k1)
|
2007-01-06 18:59:35 +00:00
|
|
|
return (&emu_reccaps_efx_live);
|
|
|
|
return (&emu_reccaps_efx_audigy);
|
|
|
|
|
|
|
|
}
|
|
|
|
|
2007-09-12 07:43:43 +00:00
|
|
|
static int
|
|
|
|
emufxrchan_getrates(kobj_t obj __unused, void *c_devinfo, int **rates)
|
|
|
|
{
|
|
|
|
struct emu_pcm_rchinfo *ch = c_devinfo;
|
|
|
|
struct emu_pcm_info *sc = ch->pcm;
|
|
|
|
|
|
|
|
if (sc->is_emu10k1)
|
|
|
|
*rates = emu_rates_live;
|
|
|
|
else
|
|
|
|
*rates = emu_rates_audigy;
|
|
|
|
|
|
|
|
return 1;
|
|
|
|
}
|
|
|
|
|
2007-01-06 18:59:35 +00:00
|
|
|
static kobj_method_t emufxrchan_methods[] = {
|
|
|
|
KOBJMETHOD(channel_init, emufxrchan_init),
|
|
|
|
KOBJMETHOD(channel_setformat, emufxrchan_setformat),
|
|
|
|
KOBJMETHOD(channel_setspeed, emufxrchan_setspeed),
|
|
|
|
KOBJMETHOD(channel_setblocksize, emufxrchan_setblocksize),
|
|
|
|
KOBJMETHOD(channel_trigger, emufxrchan_trigger),
|
|
|
|
KOBJMETHOD(channel_getptr, emufxrchan_getptr),
|
|
|
|
KOBJMETHOD(channel_getcaps, emufxrchan_getcaps),
|
2007-09-12 07:43:43 +00:00
|
|
|
KOBJMETHOD(channel_getrates, emufxrchan_getrates),
|
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
|
|
|
KOBJMETHOD_END
|
2007-01-06 18:59:35 +00:00
|
|
|
};
|
|
|
|
CHANNEL_DECLARE(emufxrchan);
|
|
|
|
|
2006-07-15 19:36:28 +00:00
|
|
|
|
|
|
|
static uint32_t
|
|
|
|
emu_pcm_intr(void *pcm, uint32_t stat)
|
|
|
|
{
|
|
|
|
struct emu_pcm_info *sc = (struct emu_pcm_info *)pcm;
|
|
|
|
uint32_t ack;
|
|
|
|
int i;
|
|
|
|
|
|
|
|
ack = 0;
|
|
|
|
|
2007-09-12 07:43:43 +00:00
|
|
|
snd_mtxlock(sc->lock);
|
|
|
|
|
2012-01-11 21:17:14 +00:00
|
|
|
if (stat & EMU_IPR_INTERVALTIMER) {
|
|
|
|
ack |= EMU_IPR_INTERVALTIMER;
|
2006-07-15 19:36:28 +00:00
|
|
|
for (i = 0; i < MAX_CHANNELS; i++)
|
|
|
|
if (sc->pch[i].channel) {
|
2007-09-12 07:43:43 +00:00
|
|
|
if (sc->pch[i].run == 1) {
|
|
|
|
snd_mtxunlock(sc->lock);
|
2006-07-15 19:36:28 +00:00
|
|
|
chn_intr(sc->pch[i].channel);
|
2007-09-12 07:43:43 +00:00
|
|
|
snd_mtxlock(sc->lock);
|
|
|
|
} else
|
2006-07-15 19:36:28 +00:00
|
|
|
emu_timer_enable(sc->card, sc->pch[i].timer, 0);
|
|
|
|
}
|
2007-09-12 07:43:43 +00:00
|
|
|
/* ADC may install timer to get low-latency interrupts */
|
|
|
|
if ((sc->rch_adc.channel) && (sc->rch_adc.run)) {
|
|
|
|
snd_mtxunlock(sc->lock);
|
|
|
|
chn_intr(sc->rch_adc.channel);
|
|
|
|
snd_mtxlock(sc->lock);
|
|
|
|
}
|
|
|
|
/*
|
|
|
|
* EFX does not use timer, because it will fill
|
|
|
|
* buffer at least 32x times faster than ADC.
|
|
|
|
*/
|
2006-07-15 19:36:28 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
|
2012-01-11 21:17:14 +00:00
|
|
|
if (stat & (EMU_IPR_ADCBUFFULL | EMU_IPR_ADCBUFHALFFULL)) {
|
|
|
|
ack |= stat & (EMU_IPR_ADCBUFFULL | EMU_IPR_ADCBUFHALFFULL);
|
2007-09-12 07:43:43 +00:00
|
|
|
if (sc->rch_adc.channel) {
|
|
|
|
snd_mtxunlock(sc->lock);
|
2007-01-06 18:59:35 +00:00
|
|
|
chn_intr(sc->rch_adc.channel);
|
2007-09-12 07:43:43 +00:00
|
|
|
snd_mtxlock(sc->lock);
|
|
|
|
}
|
2007-01-06 18:59:35 +00:00
|
|
|
}
|
|
|
|
|
2012-01-11 21:17:14 +00:00
|
|
|
if (stat & (EMU_IPR_EFXBUFFULL | EMU_IPR_EFXBUFHALFFULL)) {
|
|
|
|
ack |= stat & (EMU_IPR_EFXBUFFULL | EMU_IPR_EFXBUFHALFFULL);
|
2007-09-12 07:43:43 +00:00
|
|
|
if (sc->rch_efx.channel) {
|
|
|
|
snd_mtxunlock(sc->lock);
|
2007-01-06 18:59:35 +00:00
|
|
|
chn_intr(sc->rch_efx.channel);
|
2007-09-12 07:43:43 +00:00
|
|
|
snd_mtxlock(sc->lock);
|
|
|
|
}
|
2006-07-15 19:36:28 +00:00
|
|
|
}
|
2007-09-12 07:43:43 +00:00
|
|
|
snd_mtxunlock(sc->lock);
|
|
|
|
|
2006-07-15 19:36:28 +00:00
|
|
|
return (ack);
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
emu_pcm_init(struct emu_pcm_info *sc)
|
|
|
|
{
|
2006-08-07 22:44:01 +00:00
|
|
|
sc->bufsz = pcm_getbuffersize(sc->dev, EMUPAGESIZE, EMU_REC_BUFSZ, EMU_MAX_BUFSZ);
|
2006-07-15 19:36:28 +00:00
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
emu_pcm_uninit(struct emu_pcm_info *sc __unused)
|
|
|
|
{
|
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
emu_pcm_probe(device_t dev)
|
|
|
|
{
|
|
|
|
uintptr_t func, route, r;
|
|
|
|
const char *rt;
|
|
|
|
char buffer[255];
|
|
|
|
|
|
|
|
r = BUS_READ_IVAR(device_get_parent(dev), dev, EMU_VAR_FUNC, &func);
|
|
|
|
|
|
|
|
if (func != SCF_PCM)
|
|
|
|
return (ENXIO);
|
|
|
|
|
|
|
|
rt = "UNKNOWN";
|
|
|
|
r = BUS_READ_IVAR(device_get_parent(dev), dev, EMU_VAR_ROUTE, &route);
|
|
|
|
switch (route) {
|
|
|
|
case RT_FRONT:
|
2007-01-06 18:59:35 +00:00
|
|
|
rt = "front";
|
2006-07-15 19:36:28 +00:00
|
|
|
break;
|
|
|
|
case RT_REAR:
|
2007-01-06 18:59:35 +00:00
|
|
|
rt = "rear";
|
2006-07-15 19:36:28 +00:00
|
|
|
break;
|
|
|
|
case RT_CENTER:
|
2007-01-06 18:59:35 +00:00
|
|
|
rt = "center";
|
2006-07-15 19:36:28 +00:00
|
|
|
break;
|
|
|
|
case RT_SUB:
|
2007-01-06 18:59:35 +00:00
|
|
|
rt = "subwoofer";
|
2006-07-15 19:36:28 +00:00
|
|
|
break;
|
|
|
|
case RT_SIDE:
|
2007-01-06 18:59:35 +00:00
|
|
|
rt = "side";
|
|
|
|
break;
|
|
|
|
case RT_MCHRECORD:
|
|
|
|
rt = "multichannel recording";
|
2006-07-15 19:36:28 +00:00
|
|
|
break;
|
|
|
|
}
|
|
|
|
|
2007-01-06 18:59:35 +00:00
|
|
|
snprintf(buffer, 255, "EMU10Kx DSP %s PCM interface", rt);
|
2006-07-15 19:36:28 +00:00
|
|
|
device_set_desc_copy(dev, buffer);
|
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
emu_pcm_attach(device_t dev)
|
|
|
|
{
|
|
|
|
struct emu_pcm_info *sc;
|
|
|
|
unsigned int i;
|
|
|
|
char status[SND_STATUSLEN];
|
|
|
|
uint32_t inte, ipr;
|
2007-09-12 07:43:43 +00:00
|
|
|
uintptr_t route, r, ivar;
|
2006-07-15 19:36:28 +00:00
|
|
|
|
2007-06-17 06:10:43 +00:00
|
|
|
sc = malloc(sizeof(*sc), M_DEVBUF, M_WAITOK | M_ZERO);
|
2006-07-15 19:36:28 +00:00
|
|
|
sc->card = (struct emu_sc_info *)(device_get_softc(device_get_parent(dev)));
|
|
|
|
if (sc->card == NULL) {
|
|
|
|
device_printf(dev, "cannot get bridge conf\n");
|
2007-06-17 06:10:43 +00:00
|
|
|
free(sc, M_DEVBUF);
|
2006-07-15 19:36:28 +00:00
|
|
|
return (ENXIO);
|
|
|
|
}
|
|
|
|
|
2007-09-12 07:43:43 +00:00
|
|
|
sc->lock = snd_mtxcreate(device_get_nameunit(dev), "snd_emu10kx pcm softc");
|
2006-07-15 19:36:28 +00:00
|
|
|
sc->dev = dev;
|
|
|
|
|
2007-09-12 07:43:43 +00:00
|
|
|
r = BUS_READ_IVAR(device_get_parent(dev), dev, EMU_VAR_ISEMU10K1, &ivar);
|
|
|
|
sc->is_emu10k1 = ivar ? 1 : 0;
|
|
|
|
|
|
|
|
r = BUS_READ_IVAR(device_get_parent(dev), dev, EMU_VAR_MCH_DISABLED, &ivar);
|
|
|
|
sc->mch_disabled = ivar ? 1 : 0;
|
2006-07-15 19:36:28 +00:00
|
|
|
|
|
|
|
sc->codec = NULL;
|
|
|
|
|
|
|
|
for (i = 0; i < 8; i++) {
|
|
|
|
sc->rt.routing_left[i] = i;
|
|
|
|
sc->rt.amounts_left[i] = 0x00;
|
|
|
|
sc->rt.routing_right[i] = i;
|
|
|
|
sc->rt.amounts_right[i] = 0x00;
|
|
|
|
}
|
|
|
|
|
2006-08-07 22:55:21 +00:00
|
|
|
for (i = 0; i < 8; i++) {
|
|
|
|
sc->rt_mono.routing_left[i] = i;
|
|
|
|
sc->rt_mono.amounts_left[i] = 0x00;
|
|
|
|
sc->rt_mono.routing_right[i] = i;
|
|
|
|
sc->rt_mono.amounts_right[i] = 0x00;
|
|
|
|
}
|
|
|
|
|
2007-09-12 07:43:43 +00:00
|
|
|
sc->emu10k1_volcache[0][0] = 75;
|
|
|
|
sc->emu10k1_volcache[1][0] = 75;
|
|
|
|
sc->emu10k1_volcache[0][1] = 75;
|
|
|
|
sc->emu10k1_volcache[1][1] = 75;
|
2006-07-15 19:36:28 +00:00
|
|
|
r = BUS_READ_IVAR(device_get_parent(dev), dev, EMU_VAR_ROUTE, &route);
|
|
|
|
sc->route = route;
|
|
|
|
switch (route) {
|
|
|
|
case RT_FRONT:
|
|
|
|
sc->rt.amounts_left[0] = 0xff;
|
|
|
|
sc->rt.amounts_right[1] = 0xff;
|
2006-08-07 22:55:21 +00:00
|
|
|
sc->rt_mono.amounts_left[0] = 0xff;
|
|
|
|
sc->rt_mono.amounts_left[1] = 0xff;
|
2006-07-15 19:36:28 +00:00
|
|
|
if (sc->is_emu10k1)
|
|
|
|
sc->codec = AC97_CREATE(dev, sc, emu_ac97);
|
|
|
|
else
|
|
|
|
sc->codec = AC97_CREATE(dev, sc, emu_eac97);
|
2007-09-12 07:43:43 +00:00
|
|
|
sc->ac97_mixerclass = NULL;
|
|
|
|
if (sc->codec != NULL)
|
|
|
|
sc->ac97_mixerclass = ac97_getmixerclass();
|
|
|
|
if (mixer_init(dev, &emudspmixer_class, sc)) {
|
|
|
|
device_printf(dev, "failed to initialize DSP mixer\n");
|
|
|
|
goto bad;
|
|
|
|
}
|
2006-07-15 19:36:28 +00:00
|
|
|
break;
|
|
|
|
case RT_REAR:
|
|
|
|
sc->rt.amounts_left[2] = 0xff;
|
|
|
|
sc->rt.amounts_right[3] = 0xff;
|
2006-08-07 22:55:21 +00:00
|
|
|
sc->rt_mono.amounts_left[2] = 0xff;
|
|
|
|
sc->rt_mono.amounts_left[3] = 0xff;
|
2006-07-15 19:36:28 +00:00
|
|
|
if (mixer_init(dev, &emudspmixer_class, sc)) {
|
|
|
|
device_printf(dev, "failed to initialize mixer\n");
|
|
|
|
goto bad;
|
|
|
|
}
|
|
|
|
break;
|
|
|
|
case RT_CENTER:
|
|
|
|
sc->rt.amounts_left[4] = 0xff;
|
2006-08-07 22:55:21 +00:00
|
|
|
sc->rt_mono.amounts_left[4] = 0xff;
|
2006-07-15 19:36:28 +00:00
|
|
|
if (mixer_init(dev, &emudspmixer_class, sc)) {
|
|
|
|
device_printf(dev, "failed to initialize mixer\n");
|
|
|
|
goto bad;
|
|
|
|
}
|
|
|
|
break;
|
|
|
|
case RT_SUB:
|
|
|
|
sc->rt.amounts_left[5] = 0xff;
|
2006-08-07 22:55:21 +00:00
|
|
|
sc->rt_mono.amounts_left[5] = 0xff;
|
2006-07-15 19:36:28 +00:00
|
|
|
if (mixer_init(dev, &emudspmixer_class, sc)) {
|
|
|
|
device_printf(dev, "failed to initialize mixer\n");
|
|
|
|
goto bad;
|
|
|
|
}
|
|
|
|
break;
|
|
|
|
case RT_SIDE:
|
|
|
|
sc->rt.amounts_left[6] = 0xff;
|
|
|
|
sc->rt.amounts_right[7] = 0xff;
|
2006-08-07 22:55:21 +00:00
|
|
|
sc->rt_mono.amounts_left[6] = 0xff;
|
|
|
|
sc->rt_mono.amounts_left[7] = 0xff;
|
2006-07-15 19:36:28 +00:00
|
|
|
if (mixer_init(dev, &emudspmixer_class, sc)) {
|
|
|
|
device_printf(dev, "failed to initialize mixer\n");
|
|
|
|
goto bad;
|
|
|
|
}
|
|
|
|
break;
|
2007-01-06 18:59:35 +00:00
|
|
|
case RT_MCHRECORD:
|
2007-09-12 07:43:43 +00:00
|
|
|
if (mixer_init(dev, &emuefxmixer_class, sc)) {
|
|
|
|
device_printf(dev, "failed to initialize EFX mixer\n");
|
|
|
|
goto bad;
|
|
|
|
}
|
2007-01-06 18:59:35 +00:00
|
|
|
break;
|
2006-07-15 19:36:28 +00:00
|
|
|
default:
|
|
|
|
device_printf(dev, "invalid default route\n");
|
|
|
|
goto bad;
|
|
|
|
}
|
|
|
|
|
2012-01-11 21:17:14 +00:00
|
|
|
inte = EMU_INTE_INTERTIMERENB;
|
|
|
|
ipr = EMU_IPR_INTERVALTIMER; /* Used by playback & ADC */
|
2006-07-15 19:36:28 +00:00
|
|
|
sc->ihandle = emu_intr_register(sc->card, inte, ipr, &emu_pcm_intr, sc);
|
|
|
|
|
|
|
|
if (emu_pcm_init(sc) == -1) {
|
|
|
|
device_printf(dev, "unable to initialize PCM part of the card\n");
|
|
|
|
goto bad;
|
|
|
|
}
|
|
|
|
|
2007-09-12 07:43:43 +00:00
|
|
|
/*
|
|
|
|
* We don't register interrupt handler with snd_setup_intr
|
|
|
|
* in pcm device. Mark pcm device as MPSAFE manually.
|
|
|
|
*/
|
|
|
|
pcm_setflags(dev, pcm_getflags(dev) | SD_F_MPSAFE);
|
|
|
|
|
2006-07-15 19:36:28 +00:00
|
|
|
/* XXX we should better get number of available channels from parent */
|
|
|
|
if (pcm_register(dev, sc, (route == RT_FRONT) ? MAX_CHANNELS : 1, (route == RT_FRONT) ? 1 : 0)) {
|
|
|
|
device_printf(dev, "can't register PCM channels!\n");
|
|
|
|
goto bad;
|
|
|
|
}
|
|
|
|
sc->pnum = 0;
|
2007-01-06 18:59:35 +00:00
|
|
|
if (route != RT_MCHRECORD)
|
|
|
|
pcm_addchan(dev, PCMDIR_PLAY, &emupchan_class, sc);
|
2006-07-15 19:36:28 +00:00
|
|
|
if (route == RT_FRONT) {
|
|
|
|
for (i = 1; i < MAX_CHANNELS; i++)
|
|
|
|
pcm_addchan(dev, PCMDIR_PLAY, &emupchan_class, sc);
|
|
|
|
pcm_addchan(dev, PCMDIR_REC, &emurchan_class, sc);
|
|
|
|
}
|
2007-01-06 18:59:35 +00:00
|
|
|
if (route == RT_MCHRECORD)
|
|
|
|
pcm_addchan(dev, PCMDIR_REC, &emufxrchan_class, sc);
|
|
|
|
|
2006-07-15 19:36:28 +00:00
|
|
|
snprintf(status, SND_STATUSLEN, "on %s", device_get_nameunit(device_get_parent(dev)));
|
|
|
|
pcm_setstatus(dev, status);
|
|
|
|
|
|
|
|
return (0);
|
|
|
|
|
|
|
|
bad:
|
|
|
|
if (sc->codec)
|
|
|
|
ac97_destroy(sc->codec);
|
|
|
|
if (sc->lock)
|
|
|
|
snd_mtxfree(sc->lock);
|
|
|
|
free(sc, M_DEVBUF);
|
|
|
|
return (ENXIO);
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
emu_pcm_detach(device_t dev)
|
|
|
|
{
|
|
|
|
int r;
|
|
|
|
struct emu_pcm_info *sc;
|
|
|
|
|
|
|
|
sc = pcm_getdevinfo(dev);
|
|
|
|
|
|
|
|
r = pcm_unregister(dev);
|
|
|
|
|
|
|
|
if (r) return (r);
|
|
|
|
|
|
|
|
emu_pcm_uninit(sc);
|
|
|
|
|
|
|
|
if (sc->lock)
|
|
|
|
snd_mtxfree(sc->lock);
|
|
|
|
free(sc, M_DEVBUF);
|
|
|
|
|
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
|
|
|
static device_method_t emu_pcm_methods[] = {
|
|
|
|
DEVMETHOD(device_probe, emu_pcm_probe),
|
|
|
|
DEVMETHOD(device_attach, emu_pcm_attach),
|
|
|
|
DEVMETHOD(device_detach, emu_pcm_detach),
|
|
|
|
{0, 0}
|
|
|
|
};
|
|
|
|
|
|
|
|
static driver_t emu_pcm_driver = {
|
|
|
|
"pcm",
|
|
|
|
emu_pcm_methods,
|
|
|
|
PCM_SOFTC_SIZE,
|
|
|
|
NULL,
|
|
|
|
0,
|
|
|
|
NULL
|
|
|
|
};
|
|
|
|
DRIVER_MODULE(snd_emu10kx_pcm, emu10kx, emu_pcm_driver, pcm_devclass, 0, 0);
|
|
|
|
MODULE_DEPEND(snd_emu10kx_pcm, snd_emu10kx, SND_EMU10KX_MINVER, SND_EMU10KX_PREFVER, SND_EMU10KX_MAXVER);
|
|
|
|
MODULE_DEPEND(snd_emu10kx_pcm, sound, SOUND_MINVER, SOUND_PREFVER, SOUND_MAXVER);
|
|
|
|
MODULE_VERSION(snd_emu10kx_pcm, SND_EMU10KX_PREFVER);
|