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freebsd-dev/sys/dev/sound/isa/ad1816.c

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Start each of the license/copyright comments with /*-, minor shuffle of lines
2005-01-06 01:43:34 +00:00
/*-
update my email address.
2003-09-07 16:28:03 +00:00
* Copyright (c) 1999 Cameron Grant <cg@freebsd.org>
Clean up the BSD license to match the preferred license in /usr/share/examples/etc/bsd-style-copyright. I've fixed a few minor wording and formatting differences. Approved by: luigi, Hannu Savolainen <hannu@opensound.com>
2007-02-02 13:39:20 +00:00
* Copyright (c) 1997,1998 Luigi Rizzo
* Copyright (c) 1994,1995 Hannu Savolainen
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
* 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, WHETHER IN CONTRACT, STRICT
* LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
* SUCH DAMAGE.
*/
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
repo-copied to make way for newmidi, this commit updates include paths
1999-11-20 16:50:33 +00:00
#include <dev/sound/pcm/sound.h>
#include <dev/sound/isa/ad1816.h>
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
- Clean up ISA DMA supports. - Rename all sndbuf_isadma* functions to sndbuf_dma* and move them into sys/dev/sound/isa/sndbuf_dma.c. No response from: sound
2003-02-07 14:05:34 +00:00
#include <isa/isavar.h>
kobjify. this gives us several benefits, including: * easier extensibility- new optional methods can be added to ac97/mixer/channel classes without having to fixup every driver. * forward compatibility for drivers, provided no new mandatory methods are added.
2000-12-18 01:36:41 +00:00
#include "mixer_if.h"
many changes: * add new channels to the end of the list so channels used in order of addition * de-globalise definition of struct snddev_info and provide accessor functions where necessary. * move the $FreeBSD$ tag in each .c file into a macro and allow the /dev/sndstat handler to display these when set to maximum verbosity to aid debugging. * allow each device to register its own sndstat handler to reduce the amount of groping sndstat must do in foreign structs.
2001-08-23 11:30:52 +00:00
SND_DECLARE_FILE("$FreeBSD$");
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
struct ad1816_info;
struct ad1816_chinfo {
struct ad1816_info *parent;
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
struct pcm_channel *channel;
struct snd_dbuf *buffer;
update code dealing with snd_dbuf objects to do so using a functional interface modify chn_setblocksize() to pick a default soft-blocksize appropriate to the sample rate and format in use. it will aim for a power of two size small enough to generate block sizes of at most 20ms. it will also set the hard-blocksize taking into account rate/format conversions in use. update drivers to implement setblocksize correctly: updated, tested: sb16, emu10k1, maestro, solo updated, untested: ad1816, ess, mss, sb8, csa not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686 i lack hardware to test: ad1816, csa, fm801, neomagic others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
int dir, blksz;
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
};
struct ad1816_info {
allow the hardware buffer size to be controlled with hints release isa dma channels on unload (ad1816, ess, sb8)
2001-09-29 07:57:07 +00:00
struct resource *io_base; /* primary I/O address for the board */
int io_rid;
struct resource *irq;
int irq_rid;
struct resource *drq1; /* play */
int drq1_rid;
struct resource *drq2; /* rec */
int drq2_rid;
void *ih;
bus_dma_tag_t parent_dmat;
(hopefully) fix build breakage some people are seeing Approved by: re
2002-11-26 18:16:27 +00:00
struct mtx *lock;
allow the hardware buffer size to be controlled with hints release isa dma channels on unload (ad1816, ess, sb8)
2001-09-29 07:57:07 +00:00
unsigned int bufsize;
struct ad1816_chinfo pch, rch;
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
};
rework feeder sytem to allow feeders in klds modify driver capability reporting format to list every audio format seperately- required for above and because we could not previously indicate that mono was unsupported. there should be no functional impact.
2000-08-20 22:18:56 +00:00
static u_int32_t ad1816_fmt[] = {
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),
SND_FORMAT(AFMT_MU_LAW, 1, 0),
SND_FORMAT(AFMT_MU_LAW, 2, 0),
SND_FORMAT(AFMT_A_LAW, 1, 0),
SND_FORMAT(AFMT_A_LAW, 2, 0),
rework feeder sytem to allow feeders in klds modify driver capability reporting format to list every audio format seperately- required for above and because we could not previously indicate that mono was unsupported. there should be no functional impact.
2000-08-20 22:18:56 +00:00
0
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
};
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
static struct pcmchan_caps ad1816_caps = {4000, 55200, ad1816_fmt, 0};
rework feeder sytem to allow feeders in klds modify driver capability reporting format to list every audio format seperately- required for above and because we could not previously indicate that mono was unsupported. there should be no functional impact.
2000-08-20 22:18:56 +00:00
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
#define AD1816_MUTE 31 /* value for mute */
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
static void
ad1816_lock(struct ad1816_info *ad1816)
{
lock the mutex, not the softc pointer.
2001-04-10 13:52:26 +00:00
snd_mtxlock(ad1816->lock);
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
}
static void
ad1816_unlock(struct ad1816_info *ad1816)
{
lock the mutex, not the softc pointer.
2001-04-10 13:52:26 +00:00
snd_mtxunlock(ad1816->lock);
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
}
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
static int
port_rd(struct resource *port, int off)
{
if (port)
return bus_space_read_1(rman_get_bustag(port),
rman_get_bushandle(port),
off);
else
return -1;
}
static void
port_wr(struct resource *port, int off, u_int8_t data)
{
if (port)
Do not return(foo()) in void function. Submitted by: marius@alchemy.franken.de MFC after: 3 days
2002-12-18 22:53:24 +00:00
bus_space_write_1(rman_get_bustag(port),
rman_get_bushandle(port),
off, data);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
}
static int
io_rd(struct ad1816_info *ad1816, int reg)
{
return port_rd(ad1816->io_base, reg);
}
static void
io_wr(struct ad1816_info *ad1816, int reg, u_int8_t data)
{
Do not return(foo()) in void function. Submitted by: marius@alchemy.franken.de MFC after: 3 days
2002-12-18 22:53:24 +00:00
port_wr(ad1816->io_base, reg, data);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
}
static void
ad1816_intr(void *arg)
{
struct ad1816_info *ad1816 = (struct ad1816_info *)arg;
unsigned char c, served = 0;
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
ad1816_lock(ad1816);
Spelling fix for interupt -> interrupt
2007-10-12 06:03:46 +00:00
/* get interrupt status */
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
c = io_rd(ad1816, AD1816_INT);
Spelling fix for interupt -> interrupt
2007-10-12 06:03:46 +00:00
/* check for stray interrupts */
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
if (c & ~(AD1816_INTRCI | AD1816_INTRPI)) {
printf("pcm: stray int (%x)\n", c);
c &= AD1816_INTRCI | AD1816_INTRPI;
}
Spelling fix for interupt -> interrupt
2007-10-12 06:03:46 +00:00
/* check for capture interrupt */
update code dealing with snd_dbuf objects to do so using a functional interface modify chn_setblocksize() to pick a default soft-blocksize appropriate to the sample rate and format in use. it will aim for a power of two size small enough to generate block sizes of at most 20ms. it will also set the hard-blocksize taking into account rate/format conversions in use. update drivers to implement setblocksize correctly: updated, tested: sb16, emu10k1, maestro, solo updated, untested: ad1816, ess, mss, sb8, csa not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686 i lack hardware to test: ad1816, csa, fm801, neomagic others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
if (sndbuf_runsz(ad1816->rch.buffer) && (c & AD1816_INTRCI)) {
- Fixup the locking. - Don't mark MPSAFE (yet). Submitted by: Ariff Abdullah <skywizard@MyBSD.org.my>
2005-07-31 13:51:04 +00:00
ad1816_unlock(ad1816);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
chn_intr(ad1816->rch.channel);
- Fixup the locking. - Don't mark MPSAFE (yet). Submitted by: Ariff Abdullah <skywizard@MyBSD.org.my>
2005-07-31 13:51:04 +00:00
ad1816_lock(ad1816);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
served |= AD1816_INTRCI; /* cp served */
}
Spelling fix for interupt -> interrupt
2007-10-12 06:03:46 +00:00
/* check for playback interrupt */
update code dealing with snd_dbuf objects to do so using a functional interface modify chn_setblocksize() to pick a default soft-blocksize appropriate to the sample rate and format in use. it will aim for a power of two size small enough to generate block sizes of at most 20ms. it will also set the hard-blocksize taking into account rate/format conversions in use. update drivers to implement setblocksize correctly: updated, tested: sb16, emu10k1, maestro, solo updated, untested: ad1816, ess, mss, sb8, csa not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686 i lack hardware to test: ad1816, csa, fm801, neomagic others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
if (sndbuf_runsz(ad1816->pch.buffer) && (c & AD1816_INTRPI)) {
- Fixup the locking. - Don't mark MPSAFE (yet). Submitted by: Ariff Abdullah <skywizard@MyBSD.org.my>
2005-07-31 13:51:04 +00:00
ad1816_unlock(ad1816);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
chn_intr(ad1816->pch.channel);
- Fixup the locking. - Don't mark MPSAFE (yet). Submitted by: Ariff Abdullah <skywizard@MyBSD.org.my>
2005-07-31 13:51:04 +00:00
ad1816_lock(ad1816);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
served |= AD1816_INTRPI; /* pb served */
}
if (served == 0) {
/* this probably means this is not a (working) ad1816 chip, */
/* or an error in dma handling */
printf("pcm: int without reason (%x)\n", c);
c = 0;
} else c &= ~served;
io_wr(ad1816, AD1816_INT, c);
c = io_rd(ad1816, AD1816_INT);
if (c != 0) printf("pcm: int clear failed (%x)\n", c);
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
ad1816_unlock(ad1816);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
}
static int
ad1816_wait_init(struct ad1816_info *ad1816, int x)
{
int n = 0; /* to shut up the compiler... */
for (; x--;)
if ((n = (io_rd(ad1816, AD1816_ALE) & AD1816_BUSY)) == 0) DELAY(10);
else return n;
printf("ad1816_wait_init failed 0x%02x.\n", n);
return -1;
}
static unsigned short
ad1816_read(struct ad1816_info *ad1816, unsigned int reg)
{
u_short x = 0;
if (ad1816_wait_init(ad1816, 100) == -1) return 0;
io_wr(ad1816, AD1816_ALE, 0);
io_wr(ad1816, AD1816_ALE, (reg & AD1816_ALEMASK));
if (ad1816_wait_init(ad1816, 100) == -1) return 0;
x = (io_rd(ad1816, AD1816_HIGH) << 8) | io_rd(ad1816, AD1816_LOW);
return x;
}
static void
ad1816_write(struct ad1816_info *ad1816, unsigned int reg, unsigned short data)
{
if (ad1816_wait_init(ad1816, 100) == -1) return;
io_wr(ad1816, AD1816_ALE, (reg & AD1816_ALEMASK));
io_wr(ad1816, AD1816_LOW, (data & 0x000000ff));
io_wr(ad1816, AD1816_HIGH, (data & 0x0000ff00) >> 8);
}
kobjify. this gives us several benefits, including: * easier extensibility- new optional methods can be added to ac97/mixer/channel classes without having to fixup every driver. * forward compatibility for drivers, provided no new mandatory methods are added.
2000-12-18 01:36:41 +00:00
/* -------------------------------------------------------------------- */
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
static int
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
ad1816mix_init(struct snd_mixer *m)
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
{
mix_setdevs(m, AD1816_MIXER_DEVICES);
mix_setrecdevs(m, AD1816_REC_DEVICES);
return 0;
}
static int
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
ad1816mix_set(struct snd_mixer *m, unsigned dev, unsigned left, unsigned right)
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
{
struct ad1816_info *ad1816 = mix_getdevinfo(m);
u_short reg = 0;
/* Scale volumes */
left = AD1816_MUTE - (AD1816_MUTE * left) / 100;
right = AD1816_MUTE - (AD1816_MUTE * right) / 100;
reg = (left << 8) | right;
/* do channel selective muting if volume is zero */
if (left == AD1816_MUTE) reg |= 0x8000;
if (right == AD1816_MUTE) reg |= 0x0080;
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
ad1816_lock(ad1816);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
switch (dev) {
case SOUND_MIXER_VOLUME: /* Register 14 master volume */
ad1816_write(ad1816, 14, reg);
break;
case SOUND_MIXER_CD: /* Register 15 cd */
case SOUND_MIXER_LINE1:
ad1816_write(ad1816, 15, reg);
break;
case SOUND_MIXER_SYNTH: /* Register 16 synth */
ad1816_write(ad1816, 16, reg);
break;
case SOUND_MIXER_PCM: /* Register 4 pcm */
ad1816_write(ad1816, 4, reg);
break;
case SOUND_MIXER_LINE:
case SOUND_MIXER_LINE3: /* Register 18 line in */
ad1816_write(ad1816, 18, reg);
break;
case SOUND_MIXER_MIC: /* Register 19 mic volume */
ad1816_write(ad1816, 19, reg & ~0xff); /* mic is mono */
break;
case SOUND_MIXER_IGAIN:
/* and now to something completely different ... */
ad1816_write(ad1816, 20, ((ad1816_read(ad1816, 20) & ~0x0f0f)
| (((AD1816_MUTE - left) / 2) << 8) /* four bits of adc gain */
| ((AD1816_MUTE - right) / 2)));
break;
default:
printf("ad1816_mixer_set(): unknown device.\n");
break;
}
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
ad1816_unlock(ad1816);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
fix mixer to return properly scaled values
1999-11-22 16:20:48 +00:00
left = ((AD1816_MUTE - left) * 100) / AD1816_MUTE;
right = ((AD1816_MUTE - right) * 100) / AD1816_MUTE;
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
return left | (right << 8);
}
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
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
ad1816mix_setrecsrc(struct snd_mixer *m, u_int32_t src)
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
{
struct ad1816_info *ad1816 = mix_getdevinfo(m);
int dev;
switch (src) {
case SOUND_MASK_LINE:
case SOUND_MASK_LINE3:
dev = 0x00;
break;
case SOUND_MASK_CD:
case SOUND_MASK_LINE1:
dev = 0x20;
break;
case SOUND_MASK_MIC:
default:
dev = 0x50;
src = SOUND_MASK_MIC;
}
dev |= dev << 8;
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
ad1816_lock(ad1816);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
ad1816_write(ad1816, 20, (ad1816_read(ad1816, 20) & ~0x7070) | dev);
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
ad1816_unlock(ad1816);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
return src;
}
kobjify. this gives us several benefits, including: * easier extensibility- new optional methods can be added to ac97/mixer/channel classes without having to fixup every driver. * forward compatibility for drivers, provided no new mandatory methods are added.
2000-12-18 01:36:41 +00:00
static kobj_method_t ad1816mixer_methods[] = {
KOBJMETHOD(mixer_init, ad1816mix_init),
KOBJMETHOD(mixer_set, ad1816mix_set),
KOBJMETHOD(mixer_setrecsrc, ad1816mix_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
kobjify. this gives us several benefits, including: * easier extensibility- new optional methods can be added to ac97/mixer/channel classes without having to fixup every driver. * forward compatibility for drivers, provided no new mandatory methods are added.
2000-12-18 01:36:41 +00:00
};
MIXER_DECLARE(ad1816mixer);
/* -------------------------------------------------------------------- */
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
/* channel interface */
static void *
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
ad1816chan_init(kobj_t obj, void *devinfo, struct snd_dbuf *b, struct pcm_channel *c, int dir)
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
{
struct ad1816_info *ad1816 = devinfo;
struct ad1816_chinfo *ch = (dir == PCMDIR_PLAY)? &ad1816->pch : &ad1816->rch;
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->dir = dir;
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
ch->parent = ad1816;
ch->channel = c;
ch->buffer = b;
sndbuf_alloc() now accept dmaflags argument which will be forwarded to internal bus_dmammem_alloc() for greater flexibility on setting up DMA / page attributes.
2007-04-18 18:26:41 +00:00
if (sndbuf_alloc(ch->buffer, ad1816->parent_dmat, 0, ad1816->bufsize) != 0)
Audio drivers failed to detect failure condition and attempted to assign DMA address to the wrong address. It can cause system lockup or other mysterious errors. Since most sound cards requires low DMA address(BUS_SPACE_MAXADDR_24BIT) sndbuf_alloc() would fail when the audio driver is loaded after long running of operations. Approved by: jake (mentor) Reviewed by: truckman, matk
2004-10-13 05:45:16 +00:00
return NULL;
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +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
sndbuf_dmasetup(ch->buffer, (dir == PCMDIR_PLAY) ? ad1816->drq1 :
ad1816->drq2);
if (SND_DMA(ch->buffer))
sndbuf_dmasetdir(ch->buffer, dir);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +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
return ch;
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
}
static int
kobjify. this gives us several benefits, including: * easier extensibility- new optional methods can be added to ac97/mixer/channel classes without having to fixup every driver. * forward compatibility for drivers, provided no new mandatory methods are added.
2000-12-18 01:36:41 +00:00
ad1816chan_setformat(kobj_t obj, void *data, u_int32_t format)
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
{
struct ad1816_chinfo *ch = data;
struct ad1816_info *ad1816 = ch->parent;
int fmt = AD1816_U8, reg;
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
ad1816_lock(ad1816);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
if (ch->dir == PCMDIR_PLAY) {
reg = AD1816_PLAY;
ad1816_write(ad1816, 8, 0x0000); /* reset base and current counter */
ad1816_write(ad1816, 9, 0x0000); /* for playback and capture */
} else {
reg = AD1816_CAPT;
ad1816_write(ad1816, 10, 0x0000);
ad1816_write(ad1816, 11, 0x0000);
}
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
switch (AFMT_ENCODING(format)) {
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
case AFMT_A_LAW:
fmt = AD1816_ALAW;
break;
case AFMT_MU_LAW:
fmt = AD1816_MULAW;
break;
case AFMT_S16_LE:
fmt = AD1816_S16LE;
break;
case AFMT_S16_BE:
fmt = AD1816_S16BE;
break;
case AFMT_U8:
fmt = AD1816_U8;
break;
}
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
if (AFMT_CHANNEL(format) > 1) fmt |= AD1816_STEREO;
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
io_wr(ad1816, reg, fmt);
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
ad1816_unlock(ad1816);
Fix driver to re-enable sound output on AD1816 based cards caused by an obviously bogous return value of ad1816chan_setformat(). PR: 37932 Submitted by: Martin Kaeske <Martin.Kaeske@Stud.TU-Ilmenau.DE> Reviewed by: hm MFC after: 10 days
2002-06-09 14:20:17 +00:00
#if 0
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
return format;
Fix driver to re-enable sound output on AD1816 based cards caused by an obviously bogous return value of ad1816chan_setformat(). PR: 37932 Submitted by: Martin Kaeske <Martin.Kaeske@Stud.TU-Ilmenau.DE> Reviewed by: hm MFC after: 10 days
2002-06-09 14:20:17 +00:00
#else
return 0;
(hopefully) fix build breakage some people are seeing Approved by: re
2002-11-26 18:16:27 +00:00
#endif
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +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
kobjify. this gives us several benefits, including: * easier extensibility- new optional methods can be added to ac97/mixer/channel classes without having to fixup every driver. * forward compatibility for drivers, provided no new mandatory methods are added.
2000-12-18 01:36:41 +00:00
ad1816chan_setspeed(kobj_t obj, void *data, u_int32_t speed)
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
{
struct ad1816_chinfo *ch = data;
struct ad1816_info *ad1816 = ch->parent;
RANGE(speed, 4000, 55200);
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
ad1816_lock(ad1816);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
ad1816_write(ad1816, (ch->dir == PCMDIR_PLAY)? 2 : 3, speed);
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
ad1816_unlock(ad1816);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
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
kobjify. this gives us several benefits, including: * easier extensibility- new optional methods can be added to ac97/mixer/channel classes without having to fixup every driver. * forward compatibility for drivers, provided no new mandatory methods are added.
2000-12-18 01:36:41 +00:00
ad1816chan_setblocksize(kobj_t obj, void *data, u_int32_t blocksize)
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
{
update code dealing with snd_dbuf objects to do so using a functional interface modify chn_setblocksize() to pick a default soft-blocksize appropriate to the sample rate and format in use. it will aim for a power of two size small enough to generate block sizes of at most 20ms. it will also set the hard-blocksize taking into account rate/format conversions in use. update drivers to implement setblocksize correctly: updated, tested: sb16, emu10k1, maestro, solo updated, untested: ad1816, ess, mss, sb8, csa not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686 i lack hardware to test: ad1816, csa, fm801, neomagic others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
struct ad1816_chinfo *ch = data;
ch->blksz = blocksize;
return ch->blksz;
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
}
static int
kobjify. this gives us several benefits, including: * easier extensibility- new optional methods can be added to ac97/mixer/channel classes without having to fixup every driver. * forward compatibility for drivers, provided no new mandatory methods are added.
2000-12-18 01:36:41 +00:00
ad1816chan_trigger(kobj_t obj, void *data, int go)
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
{
struct ad1816_chinfo *ch = data;
struct ad1816_info *ad1816 = ch->parent;
int wr, reg;
Filter/compress the amount of channel trigger. This should reduce much of lock/unlock contentions within the interrupt handler. Most of these drivers only need PCMTRIG_START or STOP (ABORT). Discussed with: scottl
2007-06-11 00:49:46 +00:00
if (!PCMTRIG_COMMON(go))
handle emulated dma reads don't try to get sample size from snd_dbuf
2000-05-26 21:15:47 +00:00
return 0;
- Clean up ISA DMA supports. - Rename all sndbuf_isadma* functions to sndbuf_dma* and move them into sys/dev/sound/isa/sndbuf_dma.c. No response from: sound
2003-02-07 14:05:34 +00:00
sndbuf_dma(ch->buffer, go);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
wr = (ch->dir == PCMDIR_PLAY);
reg = wr? AD1816_PLAY : AD1816_CAPT;
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
ad1816_lock(ad1816);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
switch (go) {
case PCMTRIG_START:
/* start only if not already running */
if (!(io_rd(ad1816, reg) & AD1816_ENABLE)) {
update code dealing with snd_dbuf objects to do so using a functional interface modify chn_setblocksize() to pick a default soft-blocksize appropriate to the sample rate and format in use. it will aim for a power of two size small enough to generate block sizes of at most 20ms. it will also set the hard-blocksize taking into account rate/format conversions in use. update drivers to implement setblocksize correctly: updated, tested: sb16, emu10k1, maestro, solo updated, untested: ad1816, ess, mss, sb8, csa not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686 i lack hardware to test: ad1816, csa, fm801, neomagic others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
int cnt = ((ch->blksz) >> 2) - 1;
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
ad1816_write(ad1816, wr? 8 : 10, cnt); /* count */
make drivers start at beginning of buffer when triggered - improves mmap. not all tested. not sure about aureal.c or csapcm.c
2000-04-17 16:57:12 +00:00
ad1816_write(ad1816, wr? 9 : 11, 0); /* reset cur cnt */
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
ad1816_write(ad1816, 1, ad1816_read(ad1816, 1) |
(wr? 0x8000 : 0x4000)); /* enable int */
/* enable playback */
io_wr(ad1816, reg, io_rd(ad1816, reg) | AD1816_ENABLE);
if (!(io_rd(ad1816, reg) & AD1816_ENABLE))
printf("ad1816: failed to start %s DMA!\n",
wr? "play" : "rec");
}
break;
case PCMTRIG_STOP:
case PCMTRIG_ABORT: /* XXX check this... */
/* we don't test here if it is running... */
if (wr) {
ad1816_write(ad1816, 1, ad1816_read(ad1816, 1) &
~(wr? 0x8000 : 0x4000));
/* disable int */
io_wr(ad1816, reg, io_rd(ad1816, reg) & ~AD1816_ENABLE);
/* disable playback */
if (io_rd(ad1816, reg) & AD1816_ENABLE)
printf("ad1816: failed to stop %s DMA!\n",
wr? "play" : "rec");
ad1816_write(ad1816, wr? 8 : 10, 0); /* reset base cnt */
ad1816_write(ad1816, wr? 9 : 11, 0); /* reset cur cnt */
}
break;
}
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
ad1816_unlock(ad1816);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
return 0;
}
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
static u_int32_t
kobjify. this gives us several benefits, including: * easier extensibility- new optional methods can be added to ac97/mixer/channel classes without having to fixup every driver. * forward compatibility for drivers, provided no new mandatory methods are added.
2000-12-18 01:36:41 +00:00
ad1816chan_getptr(kobj_t obj, void *data)
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
{
struct ad1816_chinfo *ch = data;
- Clean up ISA DMA supports. - Rename all sndbuf_isadma* functions to sndbuf_dma* and move them into sys/dev/sound/isa/sndbuf_dma.c. No response from: sound
2003-02-07 14:05:34 +00:00
return sndbuf_dmaptr(ch->buffer);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
}
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
static struct pcmchan_caps *
kobjify. this gives us several benefits, including: * easier extensibility- new optional methods can be added to ac97/mixer/channel classes without having to fixup every driver. * forward compatibility for drivers, provided no new mandatory methods are added.
2000-12-18 01:36:41 +00:00
ad1816chan_getcaps(kobj_t obj, void *data)
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
{
return &ad1816_caps;
}
kobjify. this gives us several benefits, including: * easier extensibility- new optional methods can be added to ac97/mixer/channel classes without having to fixup every driver. * forward compatibility for drivers, provided no new mandatory methods are added.
2000-12-18 01:36:41 +00:00
static kobj_method_t ad1816chan_methods[] = {
KOBJMETHOD(channel_init, ad1816chan_init),
KOBJMETHOD(channel_setformat, ad1816chan_setformat),
KOBJMETHOD(channel_setspeed, ad1816chan_setspeed),
KOBJMETHOD(channel_setblocksize, ad1816chan_setblocksize),
KOBJMETHOD(channel_trigger, ad1816chan_trigger),
KOBJMETHOD(channel_getptr, ad1816chan_getptr),
KOBJMETHOD(channel_getcaps, ad1816chan_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
kobjify. this gives us several benefits, including: * easier extensibility- new optional methods can be added to ac97/mixer/channel classes without having to fixup every driver. * forward compatibility for drivers, provided no new mandatory methods are added.
2000-12-18 01:36:41 +00:00
};
CHANNEL_DECLARE(ad1816chan);
/* -------------------------------------------------------------------- */
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
static void
ad1816_release_resources(struct ad1816_info *ad1816, device_t dev)
{
if (ad1816->irq) {
detach support remove un-needed setdir functions add bus_teardown_intr calls where necessary destroy our dma tags where necessary destroy ac97 before releasing resources
2000-09-09 19:21:04 +00:00
if (ad1816->ih)
bus_teardown_intr(dev, ad1816->irq, ad1816->ih);
allow the hardware buffer size to be controlled with hints release isa dma channels on unload (ad1816, ess, sb8)
2001-09-29 07:57:07 +00:00
bus_release_resource(dev, SYS_RES_IRQ, ad1816->irq_rid, ad1816->irq);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
ad1816->irq = 0;
}
if (ad1816->drq1) {
allow the hardware buffer size to be controlled with hints release isa dma channels on unload (ad1816, ess, sb8)
2001-09-29 07:57:07 +00:00
isa_dma_release(rman_get_start(ad1816->drq1));
bus_release_resource(dev, SYS_RES_DRQ, ad1816->drq1_rid, ad1816->drq1);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
ad1816->drq1 = 0;
}
if (ad1816->drq2) {
allow the hardware buffer size to be controlled with hints release isa dma channels on unload (ad1816, ess, sb8)
2001-09-29 07:57:07 +00:00
isa_dma_release(rman_get_start(ad1816->drq2));
bus_release_resource(dev, SYS_RES_DRQ, ad1816->drq2_rid, ad1816->drq2);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
ad1816->drq2 = 0;
}
if (ad1816->io_base) {
allow the hardware buffer size to be controlled with hints release isa dma channels on unload (ad1816, ess, sb8)
2001-09-29 07:57:07 +00:00
bus_release_resource(dev, SYS_RES_IOPORT, ad1816->io_rid, ad1816->io_base);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
ad1816->io_base = 0;
}
detach support remove un-needed setdir functions add bus_teardown_intr calls where necessary destroy our dma tags where necessary destroy ac97 before releasing resources
2000-09-09 19:21:04 +00:00
if (ad1816->parent_dmat) {
bus_dma_tag_destroy(ad1816->parent_dmat);
ad1816->parent_dmat = 0;
}
allow the hardware buffer size to be controlled with hints release isa dma channels on unload (ad1816, ess, sb8)
2001-09-29 07:57:07 +00:00
if (ad1816->lock)
snd_mtxfree(ad1816->lock);
detach support remove un-needed setdir functions add bus_teardown_intr calls where necessary destroy our dma tags where necessary destroy ac97 before releasing resources
2000-09-09 19:21:04 +00:00
free(ad1816, M_DEVBUF);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
}
static int
ad1816_alloc_resources(struct ad1816_info *ad1816, device_t dev)
{
int ok = 1, pdma, rdma;
allow the hardware buffer size to be controlled with hints release isa dma channels on unload (ad1816, ess, sb8)
2001-09-29 07:57:07 +00:00
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
if (!ad1816->io_base)
Convert callers to the new bus_alloc_resource_any(9) API. Submitted by: Mark Santcroos <marks@ripe.net> Reviewed by: imp, dfr, bde
2004-03-17 17:50:55 +00:00
ad1816->io_base = bus_alloc_resource_any(dev,
SYS_RES_IOPORT, &ad1816->io_rid, RF_ACTIVE);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
if (!ad1816->irq)
Convert callers to the new bus_alloc_resource_any(9) API. Submitted by: Mark Santcroos <marks@ripe.net> Reviewed by: imp, dfr, bde
2004-03-17 17:50:55 +00:00
ad1816->irq = bus_alloc_resource_any(dev, SYS_RES_IRQ,
&ad1816->irq_rid, RF_ACTIVE);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
if (!ad1816->drq1)
Convert callers to the new bus_alloc_resource_any(9) API. Submitted by: Mark Santcroos <marks@ripe.net> Reviewed by: imp, dfr, bde
2004-03-17 17:50:55 +00:00
ad1816->drq1 = bus_alloc_resource_any(dev, SYS_RES_DRQ,
&ad1816->drq1_rid, RF_ACTIVE);
fix a bug in sndstat output, tidy up a few bits
1999-12-19 15:07:56 +00:00
if (!ad1816->drq2)
Convert callers to the new bus_alloc_resource_any(9) API. Submitted by: Mark Santcroos <marks@ripe.net> Reviewed by: imp, dfr, bde
2004-03-17 17:50:55 +00:00
ad1816->drq2 = bus_alloc_resource_any(dev, SYS_RES_DRQ,
&ad1816->drq2_rid, RF_ACTIVE);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
if (!ad1816->io_base || !ad1816->drq1 || !ad1816->irq) ok = 0;
if (ok) {
pdma = rman_get_start(ad1816->drq1);
isa_dma_acquire(pdma);
allow the hardware buffer size to be controlled with hints release isa dma channels on unload (ad1816, ess, sb8)
2001-09-29 07:57:07 +00:00
isa_dmainit(pdma, ad1816->bufsize);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
if (ad1816->drq2) {
rdma = rman_get_start(ad1816->drq2);
isa_dma_acquire(rdma);
allow the hardware buffer size to be controlled with hints release isa dma channels on unload (ad1816, ess, sb8)
2001-09-29 07:57:07 +00:00
isa_dmainit(rdma, ad1816->bufsize);
} else
rdma = pdma;
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
if (pdma == rdma)
pcm_setflags(dev, pcm_getflags(dev) | SD_F_SIMPLEX);
}
allow the hardware buffer size to be controlled with hints release isa dma channels on unload (ad1816, ess, sb8)
2001-09-29 07:57:07 +00:00
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
return ok;
}
static int
ad1816_init(struct ad1816_info *ad1816, device_t dev)
{
ad1816_write(ad1816, 1, 0x2); /* disable interrupts */
ad1816_write(ad1816, 32, 0x90F0); /* SoundSys Mode, split fmt */
ad1816_write(ad1816, 5, 0x8080); /* FM volume mute */
ad1816_write(ad1816, 6, 0x8080); /* I2S1 volume mute */
ad1816_write(ad1816, 7, 0x8080); /* I2S0 volume mute */
ad1816_write(ad1816, 17, 0x8888); /* VID Volume mute */
ad1816_write(ad1816, 20, 0x5050); /* recsrc mic, agc off */
/* adc gain is set to 0 */
return 0;
}
static int
ad1816_probe(device_t dev)
{
char *s = NULL;
u_int32_t logical_id = isa_get_logicalid(dev);
switch (logical_id) {
case 0x80719304: /* ADS7180 */
fix a bug in sndstat output, tidy up a few bits
1999-12-19 15:07:56 +00:00
s = "AD1816";
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
break;
Recover (?) support for AD1815 based ISA soundcards. PR: kern/94388 Submitted by: Krzysztof Kotlenga <piernik at gmail dot com> MFC after: 3 days
2006-03-21 03:47:25 +00:00
case 0x50719304: /* ADS7150 */
s = "AD1815";
break;
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
}
if (s) {
device_set_desc(dev, s);
Use BUS_PROBE_DEFAULT in preference to 0 and BUS_PROBE_LOW_PRIORITY in preference to some random negative number to allow other drivers a bite at the apple.
2005-03-01 08:58:06 +00:00
return BUS_PROBE_DEFAULT;
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
}
return ENXIO;
}
static int
ad1816_attach(device_t dev)
{
struct ad1816_info *ad1816;
allow the hardware buffer size to be controlled with hints release isa dma channels on unload (ad1816, ess, sb8)
2001-09-29 07:57:07 +00:00
char status[SND_STATUSLEN], status2[SND_STATUSLEN];
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
Flush remaining malloc() cleanups (M_NOWAIT -> M_WAITOK).
2007-06-17 06:10:43 +00:00
ad1816 = malloc(sizeof(*ad1816), 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
ad1816->lock = snd_mtxcreate(device_get_nameunit(dev),
"snd_ad1816 softc");
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
ad1816->io_rid = 2;
ad1816->irq_rid = 0;
ad1816->drq1_rid = 0;
ad1816->drq2_rid = 1;
allow the hardware buffer size to be controlled with hints release isa dma channels on unload (ad1816, ess, sb8)
2001-09-29 07:57:07 +00:00
ad1816->bufsize = pcm_getbuffersize(dev, 4096, DSP_BUFFSIZE, 65536);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
if (!ad1816_alloc_resources(ad1816, dev)) goto no;
ad1816_init(ad1816, dev);
kobjify. this gives us several benefits, including: * easier extensibility- new optional methods can be added to ac97/mixer/channel classes without having to fixup every driver. * forward compatibility for drivers, provided no new mandatory methods are added.
2000-12-18 01:36:41 +00:00
if (mixer_init(dev, &ad1816mixer_class, ad1816)) goto no;
The newpcm headers currently #define away INTR_MPSAFE and INTR_TYPE_AV because they bogusly check for defined(INTR_MPSAFE) -- something which never was a #define. Correct the definitions. This make INTR_TYPE_AV finally get used instead of the lower-priority INTR_TYPE_TTY, so it's quite possible some improvement will be had on sound driver performance. It would also make all the drivers marked INTR_MPSAFE actually run without Giant (which does seem to work for me), but: INTR_MPSAFE HAS BEEN REMOVED FROM EVERY SOUND DRIVER! It needs to be re-added on a case-by-case basis since there is no one who will vouch for which sound drivers, if any, willy actually operate correctly without Giant, since there hasn't been testing because of this bug disabling INTR_MPSAFE. Found by: "Yuriy Tsibizov" <Yuriy.Tsibizov@gfk.ru>
2004-04-14 14:57:49 +00:00
snd_setup_intr(dev, ad1816->irq, 0, ad1816_intr, ad1816, &ad1816->ih);
MFp4 (114068): Use bus_get_dma_tag() to obtain the parent DMA tag to make the drivers a little bit more non-ia32/amd64 friendly. There is no man page for bus_get_dma_tag, so this is modelled after rev. 1.62 of src/sys/dev/sound/pci/es137x.c by marius. Inspired by: commit by marius
2007-02-23 13:47:34 +00:00
if (bus_dma_tag_create(/*parent*/bus_get_dma_tag(dev), /*alignment*/2,
/*boundary*/0,
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
/*lowaddr*/BUS_SPACE_MAXADDR_24BIT,
/*highaddr*/BUS_SPACE_MAXADDR,
/*filter*/NULL, /*filterarg*/NULL,
allow the hardware buffer size to be controlled with hints release isa dma channels on unload (ad1816, ess, sb8)
2001-09-29 07:57:07 +00:00
/*maxsize*/ad1816->bufsize, /*nsegments*/1,
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
/*maxsegz*/0x3ffff,
Mega busdma API commit. Add two new arguments to bus_dma_tag_create(): lockfunc and lockfuncarg. Lockfunc allows a driver to provide a function for managing its locking semantics while using busdma. At the moment, this is used for the asynchronous busdma_swi and callback mechanism. Two lockfunc implementations are provided: busdma_lock_mutex() performs standard mutex operations on the mutex that is specified from lockfuncarg. dftl_lock() is a panic implementation and is defaulted to when NULL, NULL are passed to bus_dma_tag_create(). The only time that NULL, NULL should ever be used is when the driver ensures that bus_dmamap_load() will not be deferred. Drivers that do not provide their own locking can pass busdma_lock_mutex,&Giant args in order to preserve the former behaviour. sparc64 and powerpc do not provide real busdma_swi functions, so this is largely a noop on those platforms. The busdma_swi on is64 is not properly locked yet, so warnings will be emitted on this platform when busdma callback deferrals happen. If anyone gets panics or warnings from dflt_lock() being called, please let me know right away. Reviewed by: tmm, gibbs
2003-07-01 15:52:06 +00:00
/*flags*/0, /*lockfunc*/busdma_lock_mutex,
/*lockarg*/ &Giant, &ad1816->parent_dmat) != 0) {
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
device_printf(dev, "unable to create dma tag\n");
goto no;
}
allow the hardware buffer size to be controlled with hints release isa dma channels on unload (ad1816, ess, sb8)
2001-09-29 07:57:07 +00:00
if (ad1816->drq2)
snprintf(status2, SND_STATUSLEN, ":%ld", rman_get_start(ad1816->drq2));
else
status2[0] = '\0';
Augment /dev/sndstat with the module names, if applicable. Approved by: tanimura (mentor)
2004-03-06 15:52:42 +00:00
snprintf(status, SND_STATUSLEN, "at io 0x%lx irq %ld drq %ld%s bufsz %u %s",
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
rman_get_start(ad1816->io_base),
rman_get_start(ad1816->irq),
allow the hardware buffer size to be controlled with hints release isa dma channels on unload (ad1816, ess, sb8)
2001-09-29 07:57:07 +00:00
rman_get_start(ad1816->drq1),
status2,
Augment /dev/sndstat with the module names, if applicable. Approved by: tanimura (mentor)
2004-03-06 15:52:42 +00:00
ad1816->bufsize,
PCM_KLDSTRING(snd_ad1816));
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
if (pcm_register(dev, ad1816, 1, 1)) goto no;
kobjify. this gives us several benefits, including: * easier extensibility- new optional methods can be added to ac97/mixer/channel classes without having to fixup every driver. * forward compatibility for drivers, provided no new mandatory methods are added.
2000-12-18 01:36:41 +00:00
pcm_addchan(dev, PCMDIR_REC, &ad1816chan_class, ad1816);
pcm_addchan(dev, PCMDIR_PLAY, &ad1816chan_class, ad1816);
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
pcm_setstatus(dev, status);
return 0;
no:
ad1816_release_resources(ad1816, dev);
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
return ENXIO;
}
detach support remove un-needed setdir functions add bus_teardown_intr calls where necessary destroy our dma tags where necessary destroy ac97 before releasing resources
2000-09-09 19:21:04 +00:00
static int
ad1816_detach(device_t dev)
{
int r;
struct ad1816_info *ad1816;
r = pcm_unregister(dev);
if (r)
return r;
ad1816 = pcm_getdevinfo(dev);
ad1816_release_resources(ad1816, dev);
return 0;
}
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
static device_method_t ad1816_methods[] = {
/* Device interface */
DEVMETHOD(device_probe, ad1816_probe),
DEVMETHOD(device_attach, ad1816_attach),
detach support remove un-needed setdir functions add bus_teardown_intr calls where necessary destroy our dma tags where necessary destroy ac97 before releasing resources
2000-09-09 19:21:04 +00:00
DEVMETHOD(device_detach, ad1816_detach),
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
{ 0, 0 }
};
static driver_t ad1816_driver = {
"pcm",
ad1816_methods,
many changes: * add new channels to the end of the list so channels used in order of addition * de-globalise definition of struct snddev_info and provide accessor functions where necessary. * move the $FreeBSD$ tag in each .c file into a macro and allow the /dev/sndstat handler to display these when set to maximum verbosity to aid debugging. * allow each device to register its own sndstat handler to reduce the amount of groping sndstat must do in foreign structs.
2001-08-23 11:30:52 +00:00
PCM_SOFTC_SIZE,
seperate the ad1816 driver from the mss driver since it shares no code
1999-09-28 20:00:06 +00:00
};
add module metadata. this is a hack, sound drivers will eventually present a bus to which pcm, mixer, etc will attach.
2000-07-03 20:52:27 +00:00
DRIVER_MODULE(snd_ad1816, isa, ad1816_driver, pcm_devclass, 0, 0);
Add acpi attachments for ISA sound drivers. This is needed so they'll probe and attach when ACPI is enabled. Submitted by: takawata (sbc fix) MFC after: 1 day
2004-10-11 19:52:31 +00:00
DRIVER_MODULE(snd_ad1816, acpi, ad1816_driver, pcm_devclass, 0, 0);
Rename the sound device drivers: - `sound' The generic sound driver, always required. - `snd_*' Device-dependent drivers, named after the sound module names. Configure accordingly to your hardware. In addition, rename the `snd_pcm' module to `sound' in order to sync with the driver names. Suggested by: cg
2004-07-16 04:00:08 +00:00
MODULE_DEPEND(snd_ad1816, sound, SOUND_MINVER, SOUND_PREFVER, SOUND_MAXVER);
add module metadata. this is a hack, sound drivers will eventually present a bus to which pcm, mixer, etc will attach.
2000-07-03 20:52:27 +00:00
MODULE_VERSION(snd_ad1816, 1);
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