2005-01-06 01:43:34 +00:00
|
|
|
/*-
|
2001-04-23 21:53:12 +00:00
|
|
|
* Copyright (c) 2001 Orion Hodson <oho@acm.org>
|
|
|
|
* All rights reserved.
|
|
|
|
*
|
|
|
|
* Redistribution and use in source and binary forms, with or without
|
|
|
|
* modification, are permitted provided that the following conditions
|
|
|
|
* are met:
|
|
|
|
* 1. Redistributions of source code must retain the above copyright
|
|
|
|
* notice, this list of conditions and the following disclaimer.
|
|
|
|
* 2. Redistributions in binary form must reproduce the above copyright
|
|
|
|
* notice, this list of conditions and the following disclaimer in the
|
|
|
|
* documentation and/or other materials provided with the distribution.
|
|
|
|
*
|
|
|
|
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND
|
|
|
|
* ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
|
|
|
|
* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
|
|
|
|
* ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE
|
|
|
|
* FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
|
|
|
|
* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
|
|
|
|
* OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
|
|
|
|
* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHERIN CONTRACT, STRICT
|
|
|
|
* LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
|
|
|
|
* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THEPOSSIBILITY OF
|
|
|
|
* SUCH DAMAGE.
|
|
|
|
*/
|
|
|
|
|
|
|
|
/*
|
|
|
|
* als4000.c - driver for the Avance Logic ALS 4000 chipset.
|
|
|
|
*
|
2003-01-01 18:49:04 +00:00
|
|
|
* The ALS4000 is effectively an SB16 with a PCI interface.
|
2001-04-23 21:53:12 +00:00
|
|
|
*
|
|
|
|
* This driver derives from ALS4000a.PDF, Bart Hartgers alsa driver, and
|
2001-06-16 21:25:10 +00:00
|
|
|
* SB16 register descriptions.
|
2001-04-23 21:53:12 +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
|
|
|
#ifdef HAVE_KERNEL_OPTION_HEADERS
|
|
|
|
#include "opt_snd.h"
|
|
|
|
#endif
|
|
|
|
|
2001-04-23 21:53:12 +00:00
|
|
|
#include <dev/sound/pcm/sound.h>
|
|
|
|
#include <dev/sound/isa/sb.h>
|
|
|
|
#include <dev/sound/pci/als4000.h>
|
|
|
|
|
2003-08-22 07:08:17 +00:00
|
|
|
#include <dev/pci/pcireg.h>
|
|
|
|
#include <dev/pci/pcivar.h>
|
2001-04-23 21:53:12 +00:00
|
|
|
|
|
|
|
#include "mixer_if.h"
|
|
|
|
|
2001-08-23 11:30:52 +00:00
|
|
|
SND_DECLARE_FILE("$FreeBSD$");
|
|
|
|
|
2001-04-23 21:53:12 +00:00
|
|
|
/* Debugging macro's */
|
|
|
|
#undef DEB
|
|
|
|
#ifndef DEB
|
|
|
|
#define DEB(x) /* x */
|
|
|
|
#endif /* DEB */
|
|
|
|
|
2001-10-10 17:56:35 +00:00
|
|
|
#define ALS_DEFAULT_BUFSZ 16384
|
|
|
|
|
2001-04-23 21:53:12 +00:00
|
|
|
/* ------------------------------------------------------------------------- */
|
|
|
|
/* Structures */
|
|
|
|
|
|
|
|
struct sc_info;
|
|
|
|
|
|
|
|
struct sc_chinfo {
|
|
|
|
struct sc_info *parent;
|
|
|
|
struct pcm_channel *channel;
|
|
|
|
struct snd_dbuf *buffer;
|
|
|
|
u_int32_t format, speed, phys_buf, bps;
|
|
|
|
u_int32_t dma_active:1, dma_was_active:1;
|
|
|
|
u_int8_t gcr_fifo_status;
|
|
|
|
int dir;
|
|
|
|
};
|
|
|
|
|
|
|
|
struct sc_info {
|
|
|
|
device_t dev;
|
|
|
|
bus_space_tag_t st;
|
|
|
|
bus_space_handle_t sh;
|
|
|
|
bus_dma_tag_t parent_dmat;
|
|
|
|
struct resource *reg, *irq;
|
|
|
|
int regid, irqid;
|
|
|
|
void *ih;
|
2005-07-31 11:01:13 +00:00
|
|
|
struct mtx *lock;
|
2001-10-10 17:56:35 +00:00
|
|
|
|
|
|
|
unsigned int bufsz;
|
2001-04-23 21:53:12 +00:00
|
|
|
struct sc_chinfo pch, rch;
|
|
|
|
};
|
|
|
|
|
|
|
|
/* Channel caps */
|
|
|
|
|
|
|
|
static u_int32_t als_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
|
|
|
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),
|
2001-04-23 21:53:12 +00:00
|
|
|
0
|
|
|
|
};
|
|
|
|
|
2005-07-31 11:01:13 +00:00
|
|
|
/*
|
|
|
|
* I don't believe this rotten soundcard can do 48k, really,
|
|
|
|
* trust me.
|
|
|
|
*/
|
|
|
|
static struct pcmchan_caps als_caps = { 4000, 44100, als_format, 0 };
|
2001-04-23 21:53:12 +00:00
|
|
|
|
|
|
|
/* ------------------------------------------------------------------------- */
|
|
|
|
/* Register Utilities */
|
|
|
|
|
2001-06-16 21:25:10 +00:00
|
|
|
static u_int32_t
|
2001-04-23 21:53:12 +00:00
|
|
|
als_gcr_rd(struct sc_info *sc, int index)
|
|
|
|
{
|
|
|
|
bus_space_write_1(sc->st, sc->sh, ALS_GCR_INDEX, index);
|
|
|
|
return bus_space_read_4(sc->st, sc->sh, ALS_GCR_DATA);
|
|
|
|
}
|
|
|
|
|
|
|
|
static void
|
|
|
|
als_gcr_wr(struct sc_info *sc, int index, int data)
|
|
|
|
{
|
|
|
|
bus_space_write_1(sc->st, sc->sh, ALS_GCR_INDEX, index);
|
|
|
|
bus_space_write_4(sc->st, sc->sh, ALS_GCR_DATA, data);
|
|
|
|
}
|
|
|
|
|
|
|
|
static u_int8_t
|
|
|
|
als_intr_rd(struct sc_info *sc)
|
|
|
|
{
|
|
|
|
return bus_space_read_1(sc->st, sc->sh, ALS_SB_MPU_IRQ);
|
|
|
|
}
|
|
|
|
|
|
|
|
static void
|
|
|
|
als_intr_wr(struct sc_info *sc, u_int8_t data)
|
|
|
|
{
|
|
|
|
bus_space_write_1(sc->st, sc->sh, ALS_SB_MPU_IRQ, data);
|
|
|
|
}
|
|
|
|
|
|
|
|
static u_int8_t
|
|
|
|
als_mix_rd(struct sc_info *sc, u_int8_t index)
|
|
|
|
{
|
|
|
|
bus_space_write_1(sc->st, sc->sh, ALS_MIXER_INDEX, index);
|
|
|
|
return bus_space_read_1(sc->st, sc->sh, ALS_MIXER_DATA);
|
|
|
|
}
|
|
|
|
|
|
|
|
static void
|
|
|
|
als_mix_wr(struct sc_info *sc, u_int8_t index, u_int8_t data)
|
|
|
|
{
|
|
|
|
bus_space_write_1(sc->st, sc->sh, ALS_MIXER_INDEX, index);
|
|
|
|
bus_space_write_1(sc->st, sc->sh, ALS_MIXER_DATA, data);
|
|
|
|
}
|
|
|
|
|
|
|
|
static void
|
|
|
|
als_esp_wr(struct sc_info *sc, u_int8_t data)
|
|
|
|
{
|
|
|
|
u_int32_t tries, v;
|
|
|
|
|
|
|
|
tries = 1000;
|
|
|
|
do {
|
|
|
|
v = bus_space_read_1(sc->st, sc->sh, ALS_ESP_WR_STATUS);
|
|
|
|
if (~v & 0x80)
|
|
|
|
break;
|
|
|
|
DELAY(20);
|
|
|
|
} while (--tries != 0);
|
|
|
|
|
2001-06-16 21:25:10 +00:00
|
|
|
if (tries == 0)
|
2001-04-23 21:53:12 +00:00
|
|
|
device_printf(sc->dev, "als_esp_wr timeout");
|
2001-06-16 21:25:10 +00:00
|
|
|
|
2001-04-23 21:53:12 +00:00
|
|
|
bus_space_write_1(sc->st, sc->sh, ALS_ESP_WR_DATA, data);
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
als_esp_reset(struct sc_info *sc)
|
|
|
|
{
|
|
|
|
u_int32_t tries, u, v;
|
|
|
|
|
|
|
|
bus_space_write_1(sc->st, sc->sh, ALS_ESP_RST, 1);
|
|
|
|
DELAY(10);
|
|
|
|
bus_space_write_1(sc->st, sc->sh, ALS_ESP_RST, 0);
|
|
|
|
DELAY(30);
|
|
|
|
|
|
|
|
tries = 1000;
|
|
|
|
do {
|
|
|
|
u = bus_space_read_1(sc->st, sc->sh, ALS_ESP_RD_STATUS8);
|
|
|
|
if (u & 0x80) {
|
|
|
|
v = bus_space_read_1(sc->st, sc->sh, ALS_ESP_RD_DATA);
|
2001-06-16 21:25:10 +00:00
|
|
|
if (v == 0xaa)
|
2001-04-23 21:53:12 +00:00
|
|
|
return 0;
|
2001-06-16 21:25:10 +00:00
|
|
|
else
|
2001-04-23 21:53:12 +00:00
|
|
|
break;
|
|
|
|
}
|
|
|
|
DELAY(20);
|
|
|
|
} while (--tries != 0);
|
|
|
|
|
2001-06-16 21:25:10 +00:00
|
|
|
if (tries == 0)
|
2001-04-23 21:53:12 +00:00
|
|
|
device_printf(sc->dev, "als_esp_reset timeout");
|
|
|
|
return 1;
|
|
|
|
}
|
|
|
|
|
|
|
|
static u_int8_t
|
|
|
|
als_ack_read(struct sc_info *sc, u_int8_t addr)
|
|
|
|
{
|
|
|
|
u_int8_t r = bus_space_read_1(sc->st, sc->sh, addr);
|
|
|
|
return r;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* ------------------------------------------------------------------------- */
|
|
|
|
/* Common pcm channel implementation */
|
|
|
|
|
|
|
|
static void *
|
2001-06-16 21:25:10 +00:00
|
|
|
alschan_init(kobj_t obj, void *devinfo,
|
2001-04-23 21:53:12 +00:00
|
|
|
struct snd_dbuf *b, struct pcm_channel *c, int dir)
|
|
|
|
{
|
|
|
|
struct sc_info *sc = devinfo;
|
|
|
|
struct sc_chinfo *ch;
|
|
|
|
|
2005-07-31 11:01:13 +00:00
|
|
|
snd_mtxlock(sc->lock);
|
2001-04-23 21:53:12 +00:00
|
|
|
if (dir == PCMDIR_PLAY) {
|
|
|
|
ch = &sc->pch;
|
|
|
|
ch->gcr_fifo_status = ALS_GCR_FIFO0_STATUS;
|
|
|
|
} else {
|
|
|
|
ch = &sc->rch;
|
|
|
|
ch->gcr_fifo_status = ALS_GCR_FIFO1_STATUS;
|
|
|
|
}
|
|
|
|
ch->dir = dir;
|
|
|
|
ch->parent = sc;
|
|
|
|
ch->channel = c;
|
|
|
|
ch->bps = 1;
|
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->format = SND_FORMAT(AFMT_U8, 1, 0);
|
2001-04-23 21:53:12 +00:00
|
|
|
ch->speed = DSP_DEFAULT_SPEED;
|
|
|
|
ch->buffer = b;
|
2005-07-31 11:01:13 +00:00
|
|
|
snd_mtxunlock(sc->lock);
|
2005-07-31 13:22:48 +00:00
|
|
|
|
2007-04-18 18:26:41 +00:00
|
|
|
if (sndbuf_alloc(ch->buffer, sc->parent_dmat, 0, sc->bufsz) != 0)
|
2005-07-31 13:22:48 +00:00
|
|
|
return NULL;
|
|
|
|
|
2001-04-23 21:53:12 +00:00
|
|
|
return ch;
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
2001-06-16 21:25:10 +00:00
|
|
|
alschan_setformat(kobj_t obj, void *data, u_int32_t format)
|
2001-04-23 21:53:12 +00:00
|
|
|
{
|
|
|
|
struct sc_chinfo *ch = data;
|
|
|
|
|
|
|
|
ch->format = 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 u_int32_t
|
2001-04-23 21:53:12 +00:00
|
|
|
alschan_setspeed(kobj_t obj, void *data, u_int32_t speed)
|
|
|
|
{
|
|
|
|
struct sc_chinfo *ch = data, *other;
|
|
|
|
struct sc_info *sc = ch->parent;
|
|
|
|
|
|
|
|
other = (ch->dir == PCMDIR_PLAY) ? &sc->rch : &sc->pch;
|
2001-06-16 21:25:10 +00:00
|
|
|
|
2001-04-23 21:53:12 +00:00
|
|
|
/* Deny request if other dma channel is active */
|
|
|
|
if (other->dma_active) {
|
|
|
|
ch->speed = other->speed;
|
|
|
|
return other->speed;
|
|
|
|
}
|
|
|
|
|
|
|
|
ch->speed = speed;
|
|
|
|
return speed;
|
|
|
|
}
|
|
|
|
|
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
|
2001-04-23 21:53:12 +00:00
|
|
|
alschan_setblocksize(kobj_t obj, void *data, u_int32_t blocksize)
|
|
|
|
{
|
|
|
|
struct sc_chinfo *ch = data;
|
2001-10-10 17:56:35 +00:00
|
|
|
struct sc_info *sc = ch->parent;
|
2001-04-23 21:53:12 +00:00
|
|
|
|
2001-10-10 17:56:35 +00:00
|
|
|
if (blocksize > sc->bufsz / 2) {
|
|
|
|
blocksize = sc->bufsz / 2;
|
2001-04-23 21:53:12 +00:00
|
|
|
}
|
|
|
|
sndbuf_resize(ch->buffer, 2, blocksize);
|
2001-09-03 00:45:00 +00:00
|
|
|
return blocksize;
|
2001-04-23 21:53:12 +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 u_int32_t
|
2001-04-23 21:53:12 +00:00
|
|
|
alschan_getptr(kobj_t obj, void *data)
|
|
|
|
{
|
|
|
|
struct sc_chinfo *ch = data;
|
2005-10-05 20:05:52 +00:00
|
|
|
struct sc_info *sc = ch->parent;
|
2001-04-23 21:53:12 +00:00
|
|
|
int32_t pos, sz;
|
|
|
|
|
2005-10-05 20:05:52 +00:00
|
|
|
snd_mtxlock(sc->lock);
|
2001-04-23 21:53:12 +00:00
|
|
|
pos = als_gcr_rd(ch->parent, ch->gcr_fifo_status) & 0xffff;
|
2005-10-05 20:05:52 +00:00
|
|
|
snd_mtxunlock(sc->lock);
|
2001-04-23 21:53:12 +00:00
|
|
|
sz = sndbuf_getsize(ch->buffer);
|
|
|
|
return (2 * sz - pos - 1) % sz;
|
|
|
|
}
|
|
|
|
|
|
|
|
static struct pcmchan_caps*
|
|
|
|
alschan_getcaps(kobj_t obj, void *data)
|
|
|
|
{
|
|
|
|
return &als_caps;
|
|
|
|
}
|
|
|
|
|
|
|
|
static void
|
|
|
|
als_set_speed(struct sc_chinfo *ch)
|
|
|
|
{
|
|
|
|
struct sc_info *sc = ch->parent;
|
|
|
|
struct sc_chinfo *other;
|
|
|
|
|
|
|
|
other = (ch->dir == PCMDIR_PLAY) ? &sc->rch : &sc->pch;
|
|
|
|
if (other->dma_active == 0) {
|
|
|
|
als_esp_wr(sc, ALS_ESP_SAMPLE_RATE);
|
|
|
|
als_esp_wr(sc, ch->speed >> 8);
|
|
|
|
als_esp_wr(sc, ch->speed & 0xff);
|
|
|
|
} else {
|
2001-06-16 21:25:10 +00:00
|
|
|
DEB(printf("speed locked at %d (tried %d)\n",
|
2001-04-23 21:53:12 +00:00
|
|
|
other->speed, ch->speed));
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
/* ------------------------------------------------------------------------- */
|
|
|
|
/* Playback channel implementation */
|
|
|
|
|
|
|
|
#define ALS_8BIT_CMD(x, y) { (x), (y), DSP_DMA8, DSP_CMD_DMAPAUSE_8 }
|
|
|
|
#define ALS_16BIT_CMD(x, y) { (x), (y), DSP_DMA16, DSP_CMD_DMAPAUSE_16 }
|
|
|
|
|
|
|
|
struct playback_command {
|
|
|
|
u_int32_t pcm_format; /* newpcm format */
|
|
|
|
u_int8_t format_val; /* sb16 format value */
|
|
|
|
u_int8_t dma_prog; /* sb16 dma program */
|
|
|
|
u_int8_t dma_stop; /* sb16 stop register */
|
|
|
|
} static const playback_cmds[] = {
|
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
|
|
|
ALS_8BIT_CMD(SND_FORMAT(AFMT_U8, 1, 0), DSP_MODE_U8MONO),
|
|
|
|
ALS_8BIT_CMD(SND_FORMAT(AFMT_U8, 2, 0), DSP_MODE_U8STEREO),
|
|
|
|
ALS_16BIT_CMD(SND_FORMAT(AFMT_S16_LE, 1, 0), DSP_MODE_S16MONO),
|
|
|
|
ALS_16BIT_CMD(SND_FORMAT(AFMT_S16_LE, 2, 0), DSP_MODE_S16STEREO),
|
2001-04-23 21:53:12 +00:00
|
|
|
};
|
|
|
|
|
|
|
|
static const struct playback_command*
|
|
|
|
als_get_playback_command(u_int32_t format)
|
|
|
|
{
|
|
|
|
u_int32_t i, n;
|
|
|
|
|
|
|
|
n = sizeof(playback_cmds) / sizeof(playback_cmds[0]);
|
|
|
|
for (i = 0; i < n; i++) {
|
|
|
|
if (playback_cmds[i].pcm_format == format) {
|
|
|
|
return &playback_cmds[i];
|
|
|
|
}
|
|
|
|
}
|
2001-06-16 21:25:10 +00:00
|
|
|
DEB(printf("als_get_playback_command: invalid format 0x%08x\n",
|
2001-04-23 21:53:12 +00:00
|
|
|
format));
|
|
|
|
return &playback_cmds[0];
|
|
|
|
}
|
|
|
|
|
|
|
|
static void
|
|
|
|
als_playback_start(struct sc_chinfo *ch)
|
|
|
|
{
|
|
|
|
const struct playback_command *p;
|
|
|
|
struct sc_info *sc = ch->parent;
|
|
|
|
u_int32_t buf, bufsz, count, dma_prog;
|
|
|
|
|
2003-02-20 17:31:12 +00:00
|
|
|
buf = sndbuf_getbufaddr(ch->buffer);
|
2001-04-23 21:53:12 +00:00
|
|
|
bufsz = sndbuf_getsize(ch->buffer);
|
|
|
|
count = bufsz / 2;
|
|
|
|
if (ch->format & AFMT_16BIT)
|
|
|
|
count /= 2;
|
|
|
|
count--;
|
|
|
|
|
|
|
|
als_esp_wr(sc, DSP_CMD_SPKON);
|
|
|
|
als_set_speed(ch);
|
|
|
|
|
|
|
|
als_gcr_wr(sc, ALS_GCR_DMA0_START, buf);
|
|
|
|
als_gcr_wr(sc, ALS_GCR_DMA0_MODE, (bufsz - 1) | 0x180000);
|
|
|
|
|
|
|
|
p = als_get_playback_command(ch->format);
|
|
|
|
dma_prog = p->dma_prog | DSP_F16_DAC | DSP_F16_AUTO | DSP_F16_FIFO_ON;
|
|
|
|
|
|
|
|
als_esp_wr(sc, dma_prog);
|
|
|
|
als_esp_wr(sc, p->format_val);
|
|
|
|
als_esp_wr(sc, count & 0xff);
|
|
|
|
als_esp_wr(sc, count >> 8);
|
2001-06-16 21:25:10 +00:00
|
|
|
|
2001-04-23 21:53:12 +00:00
|
|
|
ch->dma_active = 1;
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
als_playback_stop(struct sc_chinfo *ch)
|
|
|
|
{
|
|
|
|
const struct playback_command *p;
|
|
|
|
struct sc_info *sc = ch->parent;
|
|
|
|
u_int32_t active;
|
|
|
|
|
|
|
|
active = ch->dma_active;
|
|
|
|
if (active) {
|
|
|
|
p = als_get_playback_command(ch->format);
|
|
|
|
als_esp_wr(sc, p->dma_stop);
|
|
|
|
}
|
|
|
|
ch->dma_active = 0;
|
|
|
|
return active;
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
alspchan_trigger(kobj_t obj, void *data, int go)
|
|
|
|
{
|
|
|
|
struct sc_chinfo *ch = data;
|
2005-10-05 20:05:52 +00:00
|
|
|
struct sc_info *sc = ch->parent;
|
2001-04-23 21:53:12 +00:00
|
|
|
|
2007-06-11 00:49:46 +00:00
|
|
|
if (!PCMTRIG_COMMON(go))
|
|
|
|
return 0;
|
|
|
|
|
2005-10-05 20:05:52 +00:00
|
|
|
snd_mtxlock(sc->lock);
|
2001-04-23 21:53:12 +00:00
|
|
|
switch(go) {
|
|
|
|
case PCMTRIG_START:
|
|
|
|
als_playback_start(ch);
|
|
|
|
break;
|
2007-06-11 00:49:46 +00:00
|
|
|
case PCMTRIG_STOP:
|
2001-04-23 21:53:12 +00:00
|
|
|
case PCMTRIG_ABORT:
|
|
|
|
als_playback_stop(ch);
|
|
|
|
break;
|
2007-06-11 00:49:46 +00:00
|
|
|
default:
|
|
|
|
break;
|
2001-04-23 21:53:12 +00:00
|
|
|
}
|
2005-10-05 20:05:52 +00:00
|
|
|
snd_mtxunlock(sc->lock);
|
2001-04-23 21:53:12 +00:00
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
static kobj_method_t alspchan_methods[] = {
|
|
|
|
KOBJMETHOD(channel_init, alschan_init),
|
|
|
|
KOBJMETHOD(channel_setformat, alschan_setformat),
|
|
|
|
KOBJMETHOD(channel_setspeed, alschan_setspeed),
|
|
|
|
KOBJMETHOD(channel_setblocksize, alschan_setblocksize),
|
|
|
|
KOBJMETHOD(channel_trigger, alspchan_trigger),
|
|
|
|
KOBJMETHOD(channel_getptr, alschan_getptr),
|
|
|
|
KOBJMETHOD(channel_getcaps, alschan_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
|
2001-04-23 21:53:12 +00:00
|
|
|
};
|
|
|
|
CHANNEL_DECLARE(alspchan);
|
|
|
|
|
|
|
|
/* ------------------------------------------------------------------------- */
|
|
|
|
/* Capture channel implementation */
|
|
|
|
|
|
|
|
static u_int8_t
|
|
|
|
als_get_fifo_format(struct sc_info *sc, u_int32_t format)
|
|
|
|
{
|
|
|
|
switch (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
|
|
|
case SND_FORMAT(AFMT_U8, 1, 0):
|
2001-04-23 21:53:12 +00:00
|
|
|
return ALS_FIFO1_8BIT;
|
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
|
|
|
case SND_FORMAT(AFMT_U8, 2, 0):
|
2001-04-23 21:53:12 +00:00
|
|
|
return ALS_FIFO1_8BIT | ALS_FIFO1_STEREO;
|
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
|
|
|
case SND_FORMAT(AFMT_S16_LE, 1, 0):
|
2001-04-23 21:53:12 +00:00
|
|
|
return ALS_FIFO1_SIGNED;
|
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
|
|
|
case SND_FORMAT(AFMT_S16_LE, 2, 0):
|
2001-04-23 21:53:12 +00:00
|
|
|
return ALS_FIFO1_SIGNED | ALS_FIFO1_STEREO;
|
|
|
|
}
|
|
|
|
device_printf(sc->dev, "format not found: 0x%08x\n", format);
|
|
|
|
return ALS_FIFO1_8BIT;
|
|
|
|
}
|
|
|
|
|
|
|
|
static void
|
|
|
|
als_capture_start(struct sc_chinfo *ch)
|
|
|
|
{
|
|
|
|
struct sc_info *sc = ch->parent;
|
|
|
|
u_int32_t buf, bufsz, count, dma_prog;
|
|
|
|
|
2003-02-20 17:31:12 +00:00
|
|
|
buf = sndbuf_getbufaddr(ch->buffer);
|
2001-04-23 21:53:12 +00:00
|
|
|
bufsz = sndbuf_getsize(ch->buffer);
|
|
|
|
count = bufsz / 2;
|
|
|
|
if (ch->format & AFMT_16BIT)
|
|
|
|
count /= 2;
|
|
|
|
count--;
|
|
|
|
|
|
|
|
als_esp_wr(sc, DSP_CMD_SPKON);
|
|
|
|
als_set_speed(ch);
|
|
|
|
|
|
|
|
als_gcr_wr(sc, ALS_GCR_FIFO1_START, buf);
|
|
|
|
als_gcr_wr(sc, ALS_GCR_FIFO1_COUNT, (bufsz - 1));
|
|
|
|
|
|
|
|
als_mix_wr(sc, ALS_FIFO1_LENGTH_LO, count & 0xff);
|
|
|
|
als_mix_wr(sc, ALS_FIFO1_LENGTH_HI, count >> 8);
|
|
|
|
|
|
|
|
dma_prog = ALS_FIFO1_RUN | als_get_fifo_format(sc, ch->format);
|
|
|
|
als_mix_wr(sc, ALS_FIFO1_CONTROL, dma_prog);
|
|
|
|
|
|
|
|
ch->dma_active = 1;
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
als_capture_stop(struct sc_chinfo *ch)
|
|
|
|
{
|
|
|
|
struct sc_info *sc = ch->parent;
|
|
|
|
u_int32_t active;
|
|
|
|
|
|
|
|
active = ch->dma_active;
|
|
|
|
if (active) {
|
|
|
|
als_mix_wr(sc, ALS_FIFO1_CONTROL, ALS_FIFO1_STOP);
|
|
|
|
}
|
|
|
|
ch->dma_active = 0;
|
|
|
|
return active;
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
alsrchan_trigger(kobj_t obj, void *data, int go)
|
|
|
|
{
|
|
|
|
struct sc_chinfo *ch = data;
|
2005-10-05 20:05:52 +00:00
|
|
|
struct sc_info *sc = ch->parent;
|
2001-04-23 21:53:12 +00:00
|
|
|
|
2005-10-05 20:05:52 +00:00
|
|
|
snd_mtxlock(sc->lock);
|
2001-04-23 21:53:12 +00:00
|
|
|
switch(go) {
|
|
|
|
case PCMTRIG_START:
|
|
|
|
als_capture_start(ch);
|
|
|
|
break;
|
2007-06-11 00:49:46 +00:00
|
|
|
case PCMTRIG_STOP:
|
2001-04-23 21:53:12 +00:00
|
|
|
case PCMTRIG_ABORT:
|
|
|
|
als_capture_stop(ch);
|
|
|
|
break;
|
|
|
|
}
|
2005-10-05 20:05:52 +00:00
|
|
|
snd_mtxunlock(sc->lock);
|
2001-04-23 21:53:12 +00:00
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
static kobj_method_t alsrchan_methods[] = {
|
|
|
|
KOBJMETHOD(channel_init, alschan_init),
|
|
|
|
KOBJMETHOD(channel_setformat, alschan_setformat),
|
|
|
|
KOBJMETHOD(channel_setspeed, alschan_setspeed),
|
|
|
|
KOBJMETHOD(channel_setblocksize, alschan_setblocksize),
|
|
|
|
KOBJMETHOD(channel_trigger, alsrchan_trigger),
|
|
|
|
KOBJMETHOD(channel_getptr, alschan_getptr),
|
|
|
|
KOBJMETHOD(channel_getcaps, alschan_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
|
2001-04-23 21:53:12 +00:00
|
|
|
};
|
|
|
|
CHANNEL_DECLARE(alsrchan);
|
|
|
|
|
|
|
|
/* ------------------------------------------------------------------------- */
|
|
|
|
/* Mixer related */
|
|
|
|
|
|
|
|
/*
|
|
|
|
* ALS4000 has an sb16 mixer, with some additional controls that we do
|
|
|
|
* not yet a means to support.
|
|
|
|
*/
|
|
|
|
|
|
|
|
struct sb16props {
|
|
|
|
u_int8_t lreg;
|
|
|
|
u_int8_t rreg;
|
|
|
|
u_int8_t bits;
|
|
|
|
u_int8_t oselect;
|
|
|
|
u_int8_t iselect; /* left input mask */
|
|
|
|
} static const amt[SOUND_MIXER_NRDEVICES] = {
|
|
|
|
[SOUND_MIXER_VOLUME] = { 0x30, 0x31, 5, 0x00, 0x00 },
|
|
|
|
[SOUND_MIXER_PCM] = { 0x32, 0x33, 5, 0x00, 0x00 },
|
|
|
|
[SOUND_MIXER_SYNTH] = { 0x34, 0x35, 5, 0x60, 0x40 },
|
|
|
|
[SOUND_MIXER_CD] = { 0x36, 0x37, 5, 0x06, 0x04 },
|
|
|
|
[SOUND_MIXER_LINE] = { 0x38, 0x39, 5, 0x18, 0x10 },
|
|
|
|
[SOUND_MIXER_MIC] = { 0x3a, 0x00, 5, 0x01, 0x01 },
|
|
|
|
[SOUND_MIXER_SPEAKER] = { 0x3b, 0x00, 2, 0x00, 0x00 },
|
|
|
|
[SOUND_MIXER_IGAIN] = { 0x3f, 0x40, 2, 0x00, 0x00 },
|
|
|
|
[SOUND_MIXER_OGAIN] = { 0x41, 0x42, 2, 0x00, 0x00 },
|
|
|
|
/* The following have register values but no h/w implementation */
|
|
|
|
[SOUND_MIXER_TREBLE] = { 0x44, 0x45, 4, 0x00, 0x00 },
|
|
|
|
[SOUND_MIXER_BASS] = { 0x46, 0x47, 4, 0x00, 0x00 }
|
|
|
|
};
|
|
|
|
|
|
|
|
static int
|
2001-06-16 21:25:10 +00:00
|
|
|
alsmix_init(struct snd_mixer *m)
|
2001-04-23 21:53:12 +00:00
|
|
|
{
|
|
|
|
u_int32_t i, v;
|
|
|
|
|
|
|
|
for (i = v = 0; i < SOUND_MIXER_NRDEVICES; i++) {
|
|
|
|
if (amt[i].bits) v |= 1 << i;
|
|
|
|
}
|
|
|
|
mix_setdevs(m, v);
|
|
|
|
|
|
|
|
for (i = v = 0; i < SOUND_MIXER_NRDEVICES; i++) {
|
|
|
|
if (amt[i].iselect) v |= 1 << i;
|
2001-06-16 21:25:10 +00:00
|
|
|
}
|
2001-04-23 21:53:12 +00:00
|
|
|
mix_setrecdevs(m, v);
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
alsmix_set(struct snd_mixer *m, unsigned dev, unsigned left, unsigned right)
|
|
|
|
{
|
|
|
|
struct sc_info *sc = mix_getdevinfo(m);
|
|
|
|
u_int32_t r, l, v, mask;
|
|
|
|
|
|
|
|
/* Fill upper n bits in mask with 1's */
|
|
|
|
mask = ((1 << amt[dev].bits) - 1) << (8 - amt[dev].bits);
|
|
|
|
|
|
|
|
l = (left * mask / 100) & mask;
|
|
|
|
v = als_mix_rd(sc, amt[dev].lreg) & ~mask;
|
|
|
|
als_mix_wr(sc, amt[dev].lreg, l | v);
|
2001-06-16 21:25:10 +00:00
|
|
|
|
2001-04-23 21:53:12 +00:00
|
|
|
if (amt[dev].rreg) {
|
|
|
|
r = (right * mask / 100) & mask;
|
|
|
|
v = als_mix_rd(sc, amt[dev].rreg) & ~mask;
|
|
|
|
als_mix_wr(sc, amt[dev].rreg, r | v);
|
|
|
|
} else {
|
|
|
|
r = 0;
|
|
|
|
}
|
2001-06-16 21:25:10 +00:00
|
|
|
|
2001-04-23 21:53:12 +00:00
|
|
|
/* Zero gain does not mute channel from output, but this does. */
|
|
|
|
v = als_mix_rd(sc, SB16_OMASK);
|
|
|
|
if (l == 0 && r == 0) {
|
|
|
|
v &= ~amt[dev].oselect;
|
|
|
|
} else {
|
|
|
|
v |= amt[dev].oselect;
|
|
|
|
}
|
|
|
|
als_mix_wr(sc, SB16_OMASK, v);
|
|
|
|
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
|
2001-04-23 21:53:12 +00:00
|
|
|
alsmix_setrecsrc(struct snd_mixer *m, u_int32_t src)
|
|
|
|
{
|
|
|
|
struct sc_info *sc = mix_getdevinfo(m);
|
2005-11-07 09:26:17 +00:00
|
|
|
u_int32_t i, l, r;
|
2001-04-23 21:53:12 +00:00
|
|
|
|
|
|
|
for (i = l = r = 0; i < SOUND_MIXER_NRDEVICES; i++) {
|
|
|
|
if (src & (1 << i)) {
|
2005-11-07 09:26:17 +00:00
|
|
|
if (amt[i].iselect == 1) { /* microphone */
|
|
|
|
l |= amt[i].iselect;
|
|
|
|
r |= amt[i].iselect;
|
|
|
|
} else {
|
|
|
|
l |= amt[i].iselect;
|
|
|
|
r |= amt[i].iselect >> 1;
|
|
|
|
}
|
2001-04-23 21:53:12 +00:00
|
|
|
}
|
|
|
|
}
|
|
|
|
|
2005-11-07 09:26:17 +00:00
|
|
|
als_mix_wr(sc, SB16_IMASK_L, l);
|
|
|
|
als_mix_wr(sc, SB16_IMASK_R, r);
|
2001-04-23 21:53:12 +00:00
|
|
|
return src;
|
|
|
|
}
|
|
|
|
|
|
|
|
static kobj_method_t als_mixer_methods[] = {
|
|
|
|
KOBJMETHOD(mixer_init, alsmix_init),
|
|
|
|
KOBJMETHOD(mixer_set, alsmix_set),
|
|
|
|
KOBJMETHOD(mixer_setrecsrc, alsmix_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
|
2001-04-23 21:53:12 +00:00
|
|
|
};
|
|
|
|
MIXER_DECLARE(als_mixer);
|
|
|
|
|
|
|
|
/* ------------------------------------------------------------------------- */
|
|
|
|
/* Interrupt Handler */
|
|
|
|
|
|
|
|
static void
|
|
|
|
als_intr(void *p)
|
|
|
|
{
|
|
|
|
struct sc_info *sc = (struct sc_info *)p;
|
|
|
|
u_int8_t intr, sb_status;
|
2001-06-16 21:25:10 +00:00
|
|
|
|
2005-07-31 11:01:13 +00:00
|
|
|
snd_mtxlock(sc->lock);
|
2001-04-23 21:53:12 +00:00
|
|
|
intr = als_intr_rd(sc);
|
2001-06-16 21:25:10 +00:00
|
|
|
|
2005-07-31 11:01:13 +00:00
|
|
|
if (intr & 0x80) {
|
|
|
|
snd_mtxunlock(sc->lock);
|
2001-04-23 21:53:12 +00:00
|
|
|
chn_intr(sc->pch.channel);
|
2005-07-31 11:01:13 +00:00
|
|
|
snd_mtxlock(sc->lock);
|
|
|
|
}
|
2001-06-16 21:25:10 +00:00
|
|
|
|
2005-07-31 11:01:13 +00:00
|
|
|
if (intr & 0x40) {
|
|
|
|
snd_mtxunlock(sc->lock);
|
2001-04-23 21:53:12 +00:00
|
|
|
chn_intr(sc->rch.channel);
|
2005-07-31 11:01:13 +00:00
|
|
|
snd_mtxlock(sc->lock);
|
|
|
|
}
|
2001-06-16 21:25:10 +00:00
|
|
|
|
2001-04-23 21:53:12 +00:00
|
|
|
/* ACK interrupt in PCI core */
|
|
|
|
als_intr_wr(sc, intr);
|
|
|
|
|
|
|
|
/* ACK interrupt in SB core */
|
|
|
|
sb_status = als_mix_rd(sc, IRQ_STAT);
|
|
|
|
|
|
|
|
if (sb_status & ALS_IRQ_STATUS8)
|
|
|
|
als_ack_read(sc, ALS_ESP_RD_STATUS8);
|
|
|
|
if (sb_status & ALS_IRQ_STATUS16)
|
|
|
|
als_ack_read(sc, ALS_ESP_RD_STATUS16);
|
|
|
|
if (sb_status & ALS_IRQ_MPUIN)
|
|
|
|
als_ack_read(sc, ALS_MIDI_DATA);
|
|
|
|
if (sb_status & ALS_IRQ_CR1E)
|
|
|
|
als_ack_read(sc, ALS_CR1E_ACK_PORT);
|
2005-07-31 11:01:13 +00:00
|
|
|
|
|
|
|
snd_mtxunlock(sc->lock);
|
2001-04-23 21:53:12 +00:00
|
|
|
return;
|
2001-06-16 21:25:10 +00:00
|
|
|
}
|
2001-04-23 21:53:12 +00:00
|
|
|
|
|
|
|
/* ------------------------------------------------------------------------- */
|
|
|
|
/* H/W initialization */
|
|
|
|
|
|
|
|
static int
|
|
|
|
als_init(struct sc_info *sc)
|
|
|
|
{
|
|
|
|
u_int32_t i, v;
|
|
|
|
|
|
|
|
/* Reset Chip */
|
|
|
|
if (als_esp_reset(sc)) {
|
|
|
|
return 1;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Enable write on DMA_SETUP register */
|
|
|
|
v = als_mix_rd(sc, ALS_SB16_CONFIG);
|
|
|
|
als_mix_wr(sc, ALS_SB16_CONFIG, v | 0x80);
|
2001-06-16 21:25:10 +00:00
|
|
|
|
2001-04-23 21:53:12 +00:00
|
|
|
/* Select DMA0 */
|
|
|
|
als_mix_wr(sc, ALS_SB16_DMA_SETUP, 0x01);
|
2001-06-16 21:25:10 +00:00
|
|
|
|
2001-04-23 21:53:12 +00:00
|
|
|
/* Disable write on DMA_SETUP register */
|
|
|
|
als_mix_wr(sc, ALS_SB16_CONFIG, v & 0x7f);
|
|
|
|
|
|
|
|
/* Enable interrupts */
|
|
|
|
v = als_gcr_rd(sc, ALS_GCR_MISC);
|
|
|
|
als_gcr_wr(sc, ALS_GCR_MISC, v | 0x28000);
|
|
|
|
|
|
|
|
/* Black out GCR DMA registers */
|
|
|
|
for (i = 0x91; i <= 0x96; i++) {
|
|
|
|
als_gcr_wr(sc, i, 0);
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Emulation mode */
|
|
|
|
v = als_gcr_rd(sc, ALS_GCR_DMA_EMULATION);
|
|
|
|
als_gcr_wr(sc, ALS_GCR_DMA_EMULATION, v);
|
|
|
|
DEB(printf("GCR_DMA_EMULATION 0x%08x\n", v));
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
static void
|
|
|
|
als_uninit(struct sc_info *sc)
|
|
|
|
{
|
|
|
|
/* Disable interrupts */
|
|
|
|
als_gcr_wr(sc, ALS_GCR_MISC, 0);
|
|
|
|
}
|
|
|
|
|
|
|
|
/* ------------------------------------------------------------------------- */
|
|
|
|
/* Probe and attach card */
|
|
|
|
|
|
|
|
static int
|
|
|
|
als_pci_probe(device_t dev)
|
|
|
|
{
|
|
|
|
if (pci_get_devid(dev) == ALS_PCI_ID0) {
|
|
|
|
device_set_desc(dev, "Avance Logic ALS4000");
|
2005-03-01 08:58:06 +00:00
|
|
|
return BUS_PROBE_DEFAULT;
|
2001-04-23 21:53:12 +00:00
|
|
|
}
|
|
|
|
return ENXIO;
|
|
|
|
}
|
|
|
|
|
|
|
|
static void
|
|
|
|
als_resource_free(device_t dev, struct sc_info *sc)
|
|
|
|
{
|
|
|
|
if (sc->reg) {
|
|
|
|
bus_release_resource(dev, SYS_RES_IOPORT, sc->regid, sc->reg);
|
|
|
|
sc->reg = 0;
|
|
|
|
}
|
|
|
|
if (sc->ih) {
|
|
|
|
bus_teardown_intr(dev, sc->irq, sc->ih);
|
|
|
|
sc->ih = 0;
|
|
|
|
}
|
|
|
|
if (sc->irq) {
|
|
|
|
bus_release_resource(dev, SYS_RES_IRQ, sc->irqid, sc->irq);
|
|
|
|
sc->irq = 0;
|
|
|
|
}
|
|
|
|
if (sc->parent_dmat) {
|
|
|
|
bus_dma_tag_destroy(sc->parent_dmat);
|
|
|
|
sc->parent_dmat = 0;
|
|
|
|
}
|
2005-07-31 11:01:13 +00:00
|
|
|
if (sc->lock) {
|
|
|
|
snd_mtxfree(sc->lock);
|
|
|
|
sc->lock = NULL;
|
|
|
|
}
|
2001-04-23 21:53:12 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
als_resource_grab(device_t dev, struct sc_info *sc)
|
|
|
|
{
|
2003-09-02 17:30:40 +00:00
|
|
|
sc->regid = PCIR_BAR(0);
|
2001-04-23 21:53:12 +00:00
|
|
|
sc->reg = bus_alloc_resource(dev, SYS_RES_IOPORT, &sc->regid, 0, ~0,
|
|
|
|
ALS_CONFIG_SPACE_BYTES, RF_ACTIVE);
|
|
|
|
if (sc->reg == 0) {
|
|
|
|
device_printf(dev, "unable to allocate register space\n");
|
|
|
|
goto bad;
|
|
|
|
}
|
|
|
|
sc->st = rman_get_bustag(sc->reg);
|
|
|
|
sc->sh = rman_get_bushandle(sc->reg);
|
|
|
|
|
2004-03-17 17:50:55 +00:00
|
|
|
sc->irq = bus_alloc_resource_any(dev, SYS_RES_IRQ, &sc->irqid,
|
|
|
|
RF_ACTIVE | RF_SHAREABLE);
|
2001-04-23 21:53:12 +00:00
|
|
|
if (sc->irq == 0) {
|
|
|
|
device_printf(dev, "unable to allocate interrupt\n");
|
|
|
|
goto bad;
|
|
|
|
}
|
|
|
|
|
2005-07-31 11:01:13 +00:00
|
|
|
if (snd_setup_intr(dev, sc->irq, INTR_MPSAFE, als_intr,
|
2001-04-23 21:53:12 +00:00
|
|
|
sc, &sc->ih)) {
|
|
|
|
device_printf(dev, "unable to setup interrupt\n");
|
|
|
|
goto bad;
|
|
|
|
}
|
|
|
|
|
2001-10-10 17:56:35 +00:00
|
|
|
sc->bufsz = pcm_getbuffersize(dev, 4096, ALS_DEFAULT_BUFSZ, 65536);
|
|
|
|
|
2007-02-23 13:47:34 +00:00
|
|
|
if (bus_dma_tag_create(/*parent*/bus_get_dma_tag(dev),
|
2001-04-23 21:53:12 +00:00
|
|
|
/*alignment*/2, /*boundary*/0,
|
|
|
|
/*lowaddr*/BUS_SPACE_MAXADDR_24BIT,
|
|
|
|
/*highaddr*/BUS_SPACE_MAXADDR,
|
|
|
|
/*filter*/NULL, /*filterarg*/NULL,
|
2001-10-10 17:56:35 +00:00
|
|
|
/*maxsize*/sc->bufsz,
|
2001-04-23 21:53:12 +00:00
|
|
|
/*nsegments*/1, /*maxsegz*/0x3ffff,
|
2005-07-31 11:01:13 +00:00
|
|
|
/*flags*/0, /*lockfunc*/NULL,
|
|
|
|
/*lockarg*/NULL, &sc->parent_dmat) != 0) {
|
2001-04-23 21:53:12 +00:00
|
|
|
device_printf(dev, "unable to create dma tag\n");
|
|
|
|
goto bad;
|
|
|
|
}
|
|
|
|
return 0;
|
|
|
|
bad:
|
|
|
|
als_resource_free(dev, sc);
|
|
|
|
return ENXIO;
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
als_pci_attach(device_t dev)
|
|
|
|
{
|
|
|
|
struct sc_info *sc;
|
|
|
|
u_int32_t data;
|
|
|
|
char status[SND_STATUSLEN];
|
|
|
|
|
2007-06-17 06:10:43 +00:00
|
|
|
sc = malloc(sizeof(*sc), M_DEVBUF, M_WAITOK | M_ZERO);
|
Fix severe out-of-bound mtx "type" pointer, causing WITNESS refcount
confusions and panic provided that the following conditions are met:
1) WITNESS is enabled (watch/trace).
2) Using modules, instead of statically linked (Not a strict
requirement, but easier to reproduce this way).
3) 2 or more modules share the same mtx type ("sound softc").
- They might share the same name (strcmp() == 0), but it always
point to different address.
4) Repetitive kldunload/load on any module that shares the same mtx
type (Not a strict requirement, but easier to reproduce this way).
Consider module A and module B:
- From enroll() - subr_witness.c:
* Load module A. Everything seems fine right now.
wA-w_refcount == 1 ; wA-w_name = "sound softc"
* Load module B.
* w->w_name == description will always fail.
("sound softc" from A and B point to different address).
* wA->w_refcount > 0 && strcmp(description, wA->w_name) == 0
* enroll() will return wA instead of returning (possibly unique)
wB.
wA->w_refcount++ , == 2.
* Unload module A, mtx_destroy(), wA->w_name become invalid,
but wA->w_refcount-- become 1 instead of 0. wA will not be
removed from witness list.
* Some other places call mtx_init(), iterating witness list,
found wA, failed on wA->w_name == description
* wA->w_refcount > 0 && strcmp(description, wA->w_name)
* Panic on strcmp() since wA->w_name no longer point to valid
address.
Note that this could happened in other places as well, not just sound
(eg. consider lots of drivers that share simmilar MTX_NETWORK_LOCK).
Solutions (for sound case):
1) Provide unique mtx type string for each mutex creation (chosen)
or
2) Put "sound softc" global variable somewhere and use it.
2007-03-15 16:41:27 +00:00
|
|
|
sc->lock = snd_mtxcreate(device_get_nameunit(dev), "snd_als4000 softc");
|
2001-04-23 21:53:12 +00:00
|
|
|
sc->dev = dev;
|
|
|
|
|
|
|
|
data = pci_read_config(dev, PCIR_COMMAND, 2);
|
|
|
|
data |= (PCIM_CMD_PORTEN | PCIM_CMD_MEMEN | PCIM_CMD_BUSMASTEREN);
|
|
|
|
pci_write_config(dev, PCIR_COMMAND, data, 2);
|
|
|
|
/*
|
|
|
|
* By default the power to the various components on the
|
|
|
|
* ALS4000 is entirely controlled by the pci powerstate. We
|
|
|
|
* could attempt finer grained control by setting GCR6.31.
|
|
|
|
*/
|
|
|
|
#if __FreeBSD_version > 500000
|
|
|
|
if (pci_get_powerstate(dev) != PCI_POWERSTATE_D0) {
|
|
|
|
/* Reset the power state. */
|
|
|
|
device_printf(dev, "chip is in D%d power mode "
|
|
|
|
"-- setting to D0\n", pci_get_powerstate(dev));
|
|
|
|
pci_set_powerstate(dev, PCI_POWERSTATE_D0);
|
|
|
|
}
|
|
|
|
#else
|
|
|
|
data = pci_read_config(dev, ALS_PCI_POWERREG, 2);
|
|
|
|
if ((data & 0x03) != 0) {
|
|
|
|
device_printf(dev, "chip is in D%d power mode "
|
|
|
|
"-- setting to D0\n", data & 0x03);
|
|
|
|
data &= ~0x03;
|
|
|
|
pci_write_config(dev, ALS_PCI_POWERREG, data, 2);
|
|
|
|
}
|
|
|
|
#endif
|
|
|
|
|
|
|
|
if (als_resource_grab(dev, sc)) {
|
|
|
|
device_printf(dev, "failed to allocate resources\n");
|
|
|
|
goto bad_attach;
|
|
|
|
}
|
|
|
|
|
|
|
|
if (als_init(sc)) {
|
|
|
|
device_printf(dev, "failed to initialize hardware\n");
|
|
|
|
goto bad_attach;
|
|
|
|
}
|
|
|
|
|
|
|
|
if (mixer_init(dev, &als_mixer_class, sc)) {
|
|
|
|
device_printf(dev, "failed to initialize mixer\n");
|
|
|
|
goto bad_attach;
|
|
|
|
}
|
|
|
|
|
|
|
|
if (pcm_register(dev, sc, 1, 1)) {
|
|
|
|
device_printf(dev, "failed to register pcm entries\n");
|
|
|
|
goto bad_attach;
|
|
|
|
}
|
|
|
|
|
|
|
|
pcm_addchan(dev, PCMDIR_PLAY, &alspchan_class, sc);
|
|
|
|
pcm_addchan(dev, PCMDIR_REC, &alsrchan_class, sc);
|
|
|
|
|
2004-03-06 15:52:42 +00:00
|
|
|
snprintf(status, SND_STATUSLEN, "at io 0x%lx irq %ld %s",
|
|
|
|
rman_get_start(sc->reg), rman_get_start(sc->irq),PCM_KLDSTRING(snd_als4000));
|
2001-04-23 21:53:12 +00:00
|
|
|
pcm_setstatus(dev, status);
|
|
|
|
return 0;
|
|
|
|
|
|
|
|
bad_attach:
|
|
|
|
als_resource_free(dev, sc);
|
|
|
|
free(sc, M_DEVBUF);
|
|
|
|
return ENXIO;
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
als_pci_detach(device_t dev)
|
|
|
|
{
|
|
|
|
struct sc_info *sc;
|
|
|
|
int r;
|
|
|
|
|
|
|
|
r = pcm_unregister(dev);
|
|
|
|
if (r)
|
|
|
|
return r;
|
|
|
|
|
|
|
|
sc = pcm_getdevinfo(dev);
|
|
|
|
als_uninit(sc);
|
|
|
|
als_resource_free(dev, sc);
|
|
|
|
free(sc, M_DEVBUF);
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
als_pci_suspend(device_t dev)
|
|
|
|
{
|
|
|
|
struct sc_info *sc = pcm_getdevinfo(dev);
|
|
|
|
|
2005-10-05 20:05:52 +00:00
|
|
|
snd_mtxlock(sc->lock);
|
2001-04-23 21:53:12 +00:00
|
|
|
sc->pch.dma_was_active = als_playback_stop(&sc->pch);
|
|
|
|
sc->rch.dma_was_active = als_capture_stop(&sc->rch);
|
|
|
|
als_uninit(sc);
|
2005-10-05 20:05:52 +00:00
|
|
|
snd_mtxunlock(sc->lock);
|
2001-04-23 21:53:12 +00:00
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
als_pci_resume(device_t dev)
|
|
|
|
{
|
|
|
|
struct sc_info *sc = pcm_getdevinfo(dev);
|
2001-06-16 21:25:10 +00:00
|
|
|
|
2005-07-31 11:01:13 +00:00
|
|
|
|
2005-10-05 20:05:52 +00:00
|
|
|
snd_mtxlock(sc->lock);
|
2001-04-23 21:53:12 +00:00
|
|
|
if (als_init(sc) != 0) {
|
|
|
|
device_printf(dev, "unable to reinitialize the card\n");
|
2005-10-05 20:05:52 +00:00
|
|
|
snd_mtxunlock(sc->lock);
|
2001-04-23 21:53:12 +00:00
|
|
|
return ENXIO;
|
2001-06-16 21:25:10 +00:00
|
|
|
}
|
2001-04-23 21:53:12 +00:00
|
|
|
|
|
|
|
if (mixer_reinit(dev) != 0) {
|
|
|
|
device_printf(dev, "unable to reinitialize the mixer\n");
|
2005-10-05 20:05:52 +00:00
|
|
|
snd_mtxunlock(sc->lock);
|
2001-04-23 21:53:12 +00:00
|
|
|
return ENXIO;
|
|
|
|
}
|
|
|
|
|
|
|
|
if (sc->pch.dma_was_active) {
|
|
|
|
als_playback_start(&sc->pch);
|
|
|
|
}
|
|
|
|
|
|
|
|
if (sc->rch.dma_was_active) {
|
|
|
|
als_capture_start(&sc->rch);
|
|
|
|
}
|
2005-10-05 20:05:52 +00:00
|
|
|
snd_mtxunlock(sc->lock);
|
2005-07-31 11:01:13 +00:00
|
|
|
|
2001-04-23 21:53:12 +00:00
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
static device_method_t als_methods[] = {
|
|
|
|
/* Device interface */
|
|
|
|
DEVMETHOD(device_probe, als_pci_probe),
|
|
|
|
DEVMETHOD(device_attach, als_pci_attach),
|
|
|
|
DEVMETHOD(device_detach, als_pci_detach),
|
|
|
|
DEVMETHOD(device_suspend, als_pci_suspend),
|
|
|
|
DEVMETHOD(device_resume, als_pci_resume),
|
|
|
|
{ 0, 0 }
|
|
|
|
};
|
|
|
|
|
|
|
|
static driver_t als_driver = {
|
|
|
|
"pcm",
|
|
|
|
als_methods,
|
2001-08-23 11:30:52 +00:00
|
|
|
PCM_SOFTC_SIZE,
|
2001-04-23 21:53:12 +00:00
|
|
|
};
|
|
|
|
|
2001-10-24 21:42:06 +00:00
|
|
|
DRIVER_MODULE(snd_als4000, pci, als_driver, pcm_devclass, 0, 0);
|
2004-07-16 04:00:08 +00:00
|
|
|
MODULE_DEPEND(snd_als4000, sound, SOUND_MINVER, SOUND_PREFVER, SOUND_MAXVER);
|
2001-10-24 21:42:06 +00:00
|
|
|
MODULE_VERSION(snd_als4000, 1);
|