freebsd-skq/sys/dev/sound/pci/ds1.c

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/*-
* SPDX-License-Identifier: BSD-2-Clause-FreeBSD
*
2000-06-06 22:34:09 +00:00
* Copyright (c) 2000 Cameron Grant <cg@freebsd.org>
* All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND
* ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
* ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE
* FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
* OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHERIN CONTRACT, STRICT
* LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THEPOSSIBILITY OF
* SUCH DAMAGE.
*/
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
#include <dev/sound/pcm/sound.h>
#include <dev/sound/pcm/ac97.h>
#include <dev/pci/pcireg.h>
#include <dev/pci/pcivar.h>
#include <dev/sound/pci/ds1.h>
#include <dev/sound/pci/ds1-fw.h>
SND_DECLARE_FILE("$FreeBSD$");
/* -------------------------------------------------------------------- */
#define DS1_CHANS 4
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#define DS1_RECPRIMARY 0
#define DS1_IRQHZ ((48000 << 8) / 256)
#define DS1_BUFFSIZE 4096
struct pbank {
volatile u_int32_t Format;
volatile u_int32_t LoopDefault;
volatile u_int32_t PgBase;
volatile u_int32_t PgLoop;
volatile u_int32_t PgLoopEnd;
volatile u_int32_t PgLoopFrac;
volatile u_int32_t PgDeltaEnd;
volatile u_int32_t LpfKEnd;
volatile u_int32_t EgGainEnd;
volatile u_int32_t LchGainEnd;
volatile u_int32_t RchGainEnd;
volatile u_int32_t Effect1GainEnd;
volatile u_int32_t Effect2GainEnd;
volatile u_int32_t Effect3GainEnd;
volatile u_int32_t LpfQ;
volatile u_int32_t Status;
volatile u_int32_t NumOfFrames;
volatile u_int32_t LoopCount;
volatile u_int32_t PgStart;
volatile u_int32_t PgStartFrac;
volatile u_int32_t PgDelta;
volatile u_int32_t LpfK;
volatile u_int32_t EgGain;
volatile u_int32_t LchGain;
volatile u_int32_t RchGain;
volatile u_int32_t Effect1Gain;
volatile u_int32_t Effect2Gain;
volatile u_int32_t Effect3Gain;
volatile u_int32_t LpfD1;
volatile u_int32_t LpfD2;
};
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struct rbank {
volatile u_int32_t PgBase;
volatile u_int32_t PgLoopEnd;
volatile u_int32_t PgStart;
volatile u_int32_t NumOfLoops;
};
struct sc_info;
/* channel registers */
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struct sc_pchinfo {
int run, spd, dir, fmt;
struct snd_dbuf *buffer;
struct pcm_channel *channel;
volatile struct pbank *lslot, *rslot;
int lsnum, rsnum;
struct sc_info *parent;
};
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struct sc_rchinfo {
int run, spd, dir, fmt, num;
struct snd_dbuf *buffer;
struct pcm_channel *channel;
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volatile struct rbank *slot;
struct sc_info *parent;
};
/* device private data */
struct sc_info {
device_t dev;
u_int32_t type, rev;
u_int32_t cd2id, ctrlbase;
bus_space_tag_t st;
bus_space_handle_t sh;
bus_dma_tag_t buffer_dmat, control_dmat;
bus_dmamap_t map;
struct resource *reg, *irq;
int regid, irqid;
void *ih;
struct mtx *lock;
void *regbase;
u_int32_t *pbase, pbankbase, pbanksize;
volatile struct pbank *pbank[2 * 64];
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volatile struct rbank *rbank;
int pslotfree, currbank, pchn, rchn;
unsigned int bufsz;
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struct sc_pchinfo pch[DS1_CHANS];
struct sc_rchinfo rch[2];
};
struct {
u_int32_t dev, subdev;
char *name;
u_int32_t *mcode;
} ds_devs[] = {
{0x00041073, 0, "Yamaha DS-1 (YMF724)", CntrlInst},
{0x000d1073, 0, "Yamaha DS-1E (YMF724F)", CntrlInst1E},
{0x00051073, 0, "Yamaha DS-1? (YMF734)", CntrlInst},
{0x00081073, 0, "Yamaha DS-1? (YMF737)", CntrlInst},
{0x00201073, 0, "Yamaha DS-1? (YMF738)", CntrlInst},
{0x00061073, 0, "Yamaha DS-1? (YMF738_TEG)", CntrlInst},
{0x000a1073, 0x00041073, "Yamaha DS-1 (YMF740)", CntrlInst},
{0x000a1073, 0x000a1073, "Yamaha DS-1 (YMF740B)", CntrlInst},
{0x000a1073, 0x53328086, "Yamaha DS-1 (YMF740I)", CntrlInst},
{0x000a1073, 0, "Yamaha DS-1 (YMF740?)", CntrlInst},
{0x000c1073, 0, "Yamaha DS-1E (YMF740C)", CntrlInst1E},
{0x00101073, 0, "Yamaha DS-1E (YMF744)", CntrlInst1E},
{0x00121073, 0, "Yamaha DS-1E (YMF754)", CntrlInst1E},
{0, 0, NULL, NULL}
};
/* -------------------------------------------------------------------- */
/*
* prototypes
*/
/* stuff */
static int ds_init(struct sc_info *);
static void ds_intr(void *);
/* talk to the card */
static u_int32_t ds_rd(struct sc_info *, int, int);
static void ds_wr(struct sc_info *, int, u_int32_t, int);
/* -------------------------------------------------------------------- */
static u_int32_t ds_recfmt[] = {
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_S8, 1, 0),
SND_FORMAT(AFMT_S8, 2, 0),
SND_FORMAT(AFMT_S16_LE, 1, 0),
SND_FORMAT(AFMT_S16_LE, 2, 0),
SND_FORMAT(AFMT_U16_LE, 1, 0),
SND_FORMAT(AFMT_U16_LE, 2, 0),
0
};
static struct pcmchan_caps ds_reccaps = {4000, 48000, ds_recfmt, 0};
static u_int32_t ds_playfmt[] = {
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),
0
};
static struct pcmchan_caps ds_playcaps = {4000, 96000, ds_playfmt, 0};
/* -------------------------------------------------------------------- */
/* Hardware */
static u_int32_t
ds_rd(struct sc_info *sc, int regno, int size)
{
switch (size) {
case 1:
return bus_space_read_1(sc->st, sc->sh, regno);
case 2:
return bus_space_read_2(sc->st, sc->sh, regno);
case 4:
return bus_space_read_4(sc->st, sc->sh, regno);
default:
return 0xffffffff;
}
}
static void
ds_wr(struct sc_info *sc, int regno, u_int32_t data, int size)
{
switch (size) {
case 1:
bus_space_write_1(sc->st, sc->sh, regno, data);
break;
case 2:
bus_space_write_2(sc->st, sc->sh, regno, data);
break;
case 4:
bus_space_write_4(sc->st, sc->sh, regno, data);
break;
}
}
static void
wrl(struct sc_info *sc, u_int32_t *ptr, u_int32_t val)
{
*(volatile u_int32_t *)ptr = val;
bus_space_barrier(sc->st, sc->sh, 0, 0, BUS_SPACE_BARRIER_WRITE);
}
/* -------------------------------------------------------------------- */
/* ac97 codec */
static int
ds_cdbusy(struct sc_info *sc, int sec)
{
int i, reg;
reg = sec? YDSXGR_SECSTATUSADR : YDSXGR_PRISTATUSADR;
i = YDSXG_AC97TIMEOUT;
while (i > 0) {
if (!(ds_rd(sc, reg, 2) & 0x8000))
return 0;
i--;
}
return ETIMEDOUT;
}
static u_int32_t
ds_initcd(kobj_t obj, void *devinfo)
{
struct sc_info *sc = (struct sc_info *)devinfo;
u_int32_t x;
x = pci_read_config(sc->dev, PCIR_DSXGCTRL, 1);
if (x & 0x03) {
pci_write_config(sc->dev, PCIR_DSXGCTRL, x & ~0x03, 1);
pci_write_config(sc->dev, PCIR_DSXGCTRL, x | 0x03, 1);
pci_write_config(sc->dev, PCIR_DSXGCTRL, x & ~0x03, 1);
/*
* The YMF740 on some Intel motherboards requires a pretty
* hefty delay after this reset for some reason... Otherwise:
* "pcm0: ac97 codec init failed"
* Maybe this is needed for all YMF740's?
* 400ms and 500ms here seem to work, 300ms does not.
*
* do it for all chips -cg
*/
DELAY(500000);
}
return ds_cdbusy(sc, 0)? 0 : 1;
}
static int
ds_rdcd(kobj_t obj, void *devinfo, int regno)
{
struct sc_info *sc = (struct sc_info *)devinfo;
int sec, cid, i;
u_int32_t cmd, reg;
sec = regno & 0x100;
regno &= 0xff;
cid = sec? (sc->cd2id << 8) : 0;
reg = sec? YDSXGR_SECSTATUSDATA : YDSXGR_PRISTATUSDATA;
if (sec && cid == 0)
return 0xffffffff;
cmd = YDSXG_AC97READCMD | cid | regno;
ds_wr(sc, YDSXGR_AC97CMDADR, cmd, 2);
if (ds_cdbusy(sc, sec))
return 0xffffffff;
if (sc->type == 11 && sc->rev < 2)
for (i = 0; i < 600; i++)
ds_rd(sc, reg, 2);
return ds_rd(sc, reg, 2);
}
static int
ds_wrcd(kobj_t obj, void *devinfo, int regno, u_int32_t data)
{
struct sc_info *sc = (struct sc_info *)devinfo;
int sec, cid;
u_int32_t cmd;
sec = regno & 0x100;
regno &= 0xff;
cid = sec? (sc->cd2id << 8) : 0;
if (sec && cid == 0)
return ENXIO;
cmd = YDSXG_AC97WRITECMD | cid | regno;
cmd <<= 16;
cmd |= data;
ds_wr(sc, YDSXGR_AC97CMDDATA, cmd, 4);
return ds_cdbusy(sc, sec);
}
static kobj_method_t ds_ac97_methods[] = {
KOBJMETHOD(ac97_init, ds_initcd),
KOBJMETHOD(ac97_read, ds_rdcd),
KOBJMETHOD(ac97_write, ds_wrcd),
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
KOBJMETHOD_END
};
AC97_DECLARE(ds_ac97);
/* -------------------------------------------------------------------- */
static void
ds_enadsp(struct sc_info *sc, int on)
{
u_int32_t v, i;
v = on? 1 : 0;
if (on) {
ds_wr(sc, YDSXGR_CONFIG, 0x00000001, 4);
} else {
if (ds_rd(sc, YDSXGR_CONFIG, 4))
ds_wr(sc, YDSXGR_CONFIG, 0x00000000, 4);
i = YDSXG_WORKBITTIMEOUT;
while (i > 0) {
if (!(ds_rd(sc, YDSXGR_CONFIG, 4) & 0x00000002))
break;
i--;
}
}
}
static volatile struct pbank *
ds_allocpslot(struct sc_info *sc)
{
int slot;
if (sc->pslotfree > 63)
return NULL;
slot = sc->pslotfree++;
return sc->pbank[slot * 2];
}
static int
ds_initpbank(volatile struct pbank *pb, int ch, int stereo, int b16, u_int32_t rate, bus_addr_t base, u_int32_t len)
{
u_int32_t lv[] = {1, 1, 0, 0, 0};
u_int32_t rv[] = {1, 0, 1, 0, 0};
u_int32_t e1[] = {0, 0, 0, 0, 0};
u_int32_t e2[] = {1, 0, 0, 1, 0};
u_int32_t e3[] = {1, 0, 0, 0, 1};
int ss, i;
u_int32_t delta;
struct {
int rate, fK, fQ;
} speedinfo[] = {
{ 100, 0x00570000, 0x35280000},
{ 2000, 0x06aa0000, 0x34a70000},
{ 8000, 0x18b20000, 0x32020000},
{11025, 0x20930000, 0x31770000},
{16000, 0x2b9a0000, 0x31390000},
{22050, 0x35a10000, 0x31c90000},
{32000, 0x3eaa0000, 0x33d00000},
/* {44100, 0x04646000, 0x370a0000},
*/ {48000, 0x40000000, 0x40000000},
};
ss = b16? 1 : 0;
ss += stereo? 1 : 0;
delta = (65536 * rate) / 48000;
i = 0;
while (i < 7 && speedinfo[i].rate < rate)
i++;
pb->Format = stereo? 0x00010000 : 0;
pb->Format |= b16? 0 : 0x80000000;
pb->Format |= (stereo && (ch == 2 || ch == 4))? 0x00000001 : 0;
pb->LoopDefault = 0;
pb->PgBase = base;
pb->PgLoop = 0;
pb->PgLoopEnd = len >> ss;
pb->PgLoopFrac = 0;
pb->Status = 0;
pb->NumOfFrames = 0;
pb->LoopCount = 0;
pb->PgStart = 0;
pb->PgStartFrac = 0;
pb->PgDelta = pb->PgDeltaEnd = delta << 12;
pb->LpfQ = speedinfo[i].fQ;
pb->LpfK = pb->LpfKEnd = speedinfo[i].fK;
pb->LpfD1 = pb->LpfD2 = 0;
pb->EgGain = pb->EgGainEnd = 0x40000000;
pb->LchGain = pb->LchGainEnd = lv[ch] * 0x40000000;
pb->RchGain = pb->RchGainEnd = rv[ch] * 0x40000000;
pb->Effect1Gain = pb->Effect1GainEnd = e1[ch] * 0x40000000;
pb->Effect2Gain = pb->Effect2GainEnd = e2[ch] * 0x40000000;
pb->Effect3Gain = pb->Effect3GainEnd = e3[ch] * 0x40000000;
return 0;
}
static void
ds_enapslot(struct sc_info *sc, int slot, int go)
{
wrl(sc, &sc->pbase[slot + 1], go? (sc->pbankbase + 2 * slot * sc->pbanksize) : 0);
/* printf("pbase[%d] = 0x%x\n", slot + 1, go? (sc->pbankbase + 2 * slot * sc->pbanksize) : 0); */
}
static void
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ds_setuppch(struct sc_pchinfo *ch)
{
2000-06-06 22:34:09 +00:00
int stereo, b16, c, sz;
bus_addr_t addr;
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
stereo = (AFMT_CHANNEL(ch->fmt) > 1)? 1 : 0;
b16 = (ch->fmt & AFMT_16BIT)? 1 : 0;
c = stereo? 1 : 0;
addr = sndbuf_getbufaddr(ch->buffer);
sz = sndbuf_getsize(ch->buffer);
ds_initpbank(ch->lslot, c, stereo, b16, ch->spd, addr, sz);
ds_initpbank(ch->lslot + 1, c, stereo, b16, ch->spd, addr, sz);
ds_initpbank(ch->rslot, 2, stereo, b16, ch->spd, addr, sz);
ds_initpbank(ch->rslot + 1, 2, stereo, b16, ch->spd, addr, sz);
}
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static void
ds_setuprch(struct sc_rchinfo *ch)
{
struct sc_info *sc = ch->parent;
int stereo, b16, i, sz, pri;
u_int32_t x, y;
bus_addr_t addr;
2000-06-06 22:34:09 +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
stereo = (AFMT_CHANNEL(ch->fmt) > 1)? 1 : 0;
2000-06-06 22:34:09 +00:00
b16 = (ch->fmt & AFMT_16BIT)? 1 : 0;
addr = sndbuf_getbufaddr(ch->buffer);
sz = sndbuf_getsize(ch->buffer);
2000-06-06 22:34:09 +00:00
pri = (ch->num == DS1_RECPRIMARY)? 1 : 0;
for (i = 0; i < 2; i++) {
ch->slot[i].PgBase = addr;
2000-06-06 22:34:09 +00:00
ch->slot[i].PgLoopEnd = sz;
ch->slot[i].PgStart = 0;
ch->slot[i].NumOfLoops = 0;
}
x = (b16? 0x00 : 0x01) | (stereo? 0x02 : 0x00);
y = (48000 * 4096) / ch->spd;
y--;
/* printf("pri = %d, x = %d, y = %d\n", pri, x, y); */
ds_wr(sc, pri? YDSXGR_ADCFORMAT : YDSXGR_RECFORMAT, x, 4);
ds_wr(sc, pri? YDSXGR_ADCSLOTSR : YDSXGR_RECSLOTSR, y, 4);
}
/* -------------------------------------------------------------------- */
2000-06-06 22:34:09 +00:00
/* play channel interface */
static void *
ds1pchan_init(kobj_t obj, void *devinfo, struct snd_dbuf *b, struct pcm_channel *c, int dir)
{
struct sc_info *sc = devinfo;
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struct sc_pchinfo *ch;
KASSERT(dir == PCMDIR_PLAY, ("ds1pchan_init: bad direction"));
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ch = &sc->pch[sc->pchn++];
ch->buffer = b;
ch->parent = sc;
ch->channel = c;
2000-06-06 22:34:09 +00:00
ch->dir = dir;
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
ch->fmt = SND_FORMAT(AFMT_U8, 1, 0);
ch->spd = 8000;
ch->run = 0;
if (sndbuf_alloc(ch->buffer, sc->buffer_dmat, 0, sc->bufsz) != 0)
return NULL;
else {
ch->lsnum = sc->pslotfree;
ch->lslot = ds_allocpslot(sc);
ch->rsnum = sc->pslotfree;
ch->rslot = ds_allocpslot(sc);
2000-06-06 22:34:09 +00:00
ds_setuppch(ch);
return ch;
}
}
static int
ds1pchan_setformat(kobj_t obj, void *data, u_int32_t format)
{
2000-06-06 22:34:09 +00:00
struct sc_pchinfo *ch = data;
ch->fmt = format;
return 0;
}
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
static u_int32_t
ds1pchan_setspeed(kobj_t obj, void *data, u_int32_t speed)
{
2000-06-06 22:34:09 +00:00
struct sc_pchinfo *ch = data;
ch->spd = speed;
return speed;
}
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
static u_int32_t
ds1pchan_setblocksize(kobj_t obj, void *data, u_int32_t blocksize)
{
struct sc_pchinfo *ch = data;
struct sc_info *sc = ch->parent;
int drate;
/* irq rate is fixed at 187.5hz */
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
drate = ch->spd * sndbuf_getalign(ch->buffer);
blocksize = roundup2((drate << 8) / DS1_IRQHZ, 4);
sndbuf_resize(ch->buffer, sc->bufsz / blocksize, blocksize);
return blocksize;
}
/* semantic note: must start at beginning of buffer */
static int
ds1pchan_trigger(kobj_t obj, void *data, int go)
{
2000-06-06 22:34:09 +00:00
struct sc_pchinfo *ch = data;
struct sc_info *sc = ch->parent;
int stereo;
if (!PCMTRIG_COMMON(go))
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
stereo = (AFMT_CHANNEL(ch->fmt) > 1)? 1 : 0;
if (go == PCMTRIG_START) {
ch->run = 1;
2000-06-06 22:34:09 +00:00
ds_setuppch(ch);
ds_enapslot(sc, ch->lsnum, 1);
ds_enapslot(sc, ch->rsnum, stereo);
snd_mtxlock(sc->lock);
ds_wr(sc, YDSXGR_MODE, 0x00000003, 4);
snd_mtxunlock(sc->lock);
} else {
ch->run = 0;
2000-06-06 22:34:09 +00:00
/* ds_setuppch(ch); */
ds_enapslot(sc, ch->lsnum, 0);
ds_enapslot(sc, ch->rsnum, 0);
}
return 0;
}
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
static u_int32_t
ds1pchan_getptr(kobj_t obj, void *data)
{
2000-06-06 22:34:09 +00:00
struct sc_pchinfo *ch = data;
struct sc_info *sc = ch->parent;
volatile struct pbank *bank;
int ss;
u_int32_t ptr;
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
ss = (AFMT_CHANNEL(ch->fmt) > 1)? 1 : 0;
ss += (ch->fmt & AFMT_16BIT)? 1 : 0;
bank = ch->lslot + sc->currbank;
/* printf("getptr: %d\n", bank->PgStart << ss); */
ptr = bank->PgStart;
ptr <<= ss;
return ptr;
}
static struct pcmchan_caps *
ds1pchan_getcaps(kobj_t obj, void *data)
{
2000-06-06 22:34:09 +00:00
return &ds_playcaps;
}
static kobj_method_t ds1pchan_methods[] = {
KOBJMETHOD(channel_init, ds1pchan_init),
KOBJMETHOD(channel_setformat, ds1pchan_setformat),
KOBJMETHOD(channel_setspeed, ds1pchan_setspeed),
KOBJMETHOD(channel_setblocksize, ds1pchan_setblocksize),
KOBJMETHOD(channel_trigger, ds1pchan_trigger),
KOBJMETHOD(channel_getptr, ds1pchan_getptr),
KOBJMETHOD(channel_getcaps, ds1pchan_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
};
CHANNEL_DECLARE(ds1pchan);
/* -------------------------------------------------------------------- */
2000-06-06 22:34:09 +00:00
/* record channel interface */
static void *
ds1rchan_init(kobj_t obj, void *devinfo, struct snd_dbuf *b, struct pcm_channel *c, int dir)
2000-06-06 22:34:09 +00:00
{
struct sc_info *sc = devinfo;
struct sc_rchinfo *ch;
2000-06-06 22:34:09 +00:00
KASSERT(dir == PCMDIR_REC, ("ds1rchan_init: bad direction"));
ch = &sc->rch[sc->rchn];
ch->num = sc->rchn++;
ch->buffer = b;
ch->parent = sc;
ch->channel = c;
ch->dir = dir;
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
ch->fmt = SND_FORMAT(AFMT_U8, 1, 0);
2000-06-06 22:34:09 +00:00
ch->spd = 8000;
if (sndbuf_alloc(ch->buffer, sc->buffer_dmat, 0, sc->bufsz) != 0)
2000-06-06 22:34:09 +00:00
return NULL;
else {
ch->slot = (ch->num == DS1_RECPRIMARY)? sc->rbank + 2: sc->rbank;
ds_setuprch(ch);
return ch;
}
}
static int
ds1rchan_setformat(kobj_t obj, void *data, u_int32_t format)
2000-06-06 22:34:09 +00:00
{
struct sc_rchinfo *ch = data;
ch->fmt = format;
return 0;
}
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
static u_int32_t
ds1rchan_setspeed(kobj_t obj, void *data, u_int32_t speed)
2000-06-06 22:34:09 +00:00
{
struct sc_rchinfo *ch = data;
ch->spd = speed;
return speed;
}
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
static u_int32_t
ds1rchan_setblocksize(kobj_t obj, void *data, u_int32_t blocksize)
2000-06-06 22:34:09 +00:00
{
struct sc_rchinfo *ch = data;
struct sc_info *sc = ch->parent;
int drate;
/* irq rate is fixed at 187.5hz */
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
drate = ch->spd * sndbuf_getalign(ch->buffer);
blocksize = roundup2((drate << 8) / DS1_IRQHZ, 4);
sndbuf_resize(ch->buffer, sc->bufsz / blocksize, blocksize);
2000-06-06 22:34:09 +00:00
return blocksize;
}
/* semantic note: must start at beginning of buffer */
static int
ds1rchan_trigger(kobj_t obj, void *data, int go)
2000-06-06 22:34:09 +00:00
{
struct sc_rchinfo *ch = data;
struct sc_info *sc = ch->parent;
u_int32_t x;
if (!PCMTRIG_COMMON(go))
2000-06-06 22:34:09 +00:00
return 0;
if (go == PCMTRIG_START) {
ch->run = 1;
ds_setuprch(ch);
snd_mtxlock(sc->lock);
2000-06-06 22:34:09 +00:00
x = ds_rd(sc, YDSXGR_MAPOFREC, 4);
x |= (ch->num == DS1_RECPRIMARY)? 0x02 : 0x01;
ds_wr(sc, YDSXGR_MAPOFREC, x, 4);
ds_wr(sc, YDSXGR_MODE, 0x00000003, 4);
snd_mtxunlock(sc->lock);
2000-06-06 22:34:09 +00:00
} else {
ch->run = 0;
snd_mtxlock(sc->lock);
2000-06-06 22:34:09 +00:00
x = ds_rd(sc, YDSXGR_MAPOFREC, 4);
x &= ~((ch->num == DS1_RECPRIMARY)? 0x02 : 0x01);
ds_wr(sc, YDSXGR_MAPOFREC, x, 4);
snd_mtxunlock(sc->lock);
2000-06-06 22:34:09 +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
ds1rchan_getptr(kobj_t obj, void *data)
2000-06-06 22:34:09 +00:00
{
struct sc_rchinfo *ch = data;
struct sc_info *sc = ch->parent;
return ch->slot[sc->currbank].PgStart;
}
static struct pcmchan_caps *
ds1rchan_getcaps(kobj_t obj, void *data)
2000-06-06 22:34:09 +00:00
{
return &ds_reccaps;
}
static kobj_method_t ds1rchan_methods[] = {
KOBJMETHOD(channel_init, ds1rchan_init),
KOBJMETHOD(channel_setformat, ds1rchan_setformat),
KOBJMETHOD(channel_setspeed, ds1rchan_setspeed),
KOBJMETHOD(channel_setblocksize, ds1rchan_setblocksize),
KOBJMETHOD(channel_trigger, ds1rchan_trigger),
KOBJMETHOD(channel_getptr, ds1rchan_getptr),
KOBJMETHOD(channel_getcaps, ds1rchan_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
};
CHANNEL_DECLARE(ds1rchan);
/* -------------------------------------------------------------------- */
/* The interrupt handler */
static void
ds_intr(void *p)
{
struct sc_info *sc = (struct sc_info *)p;
u_int32_t i, x;
snd_mtxlock(sc->lock);
i = ds_rd(sc, YDSXGR_STATUS, 4);
2000-06-06 22:34:09 +00:00
if (i & 0x00008000)
device_printf(sc->dev, "timeout irq\n");
if (i & 0x80008000) {
ds_wr(sc, YDSXGR_STATUS, i & 0x80008000, 4);
sc->currbank = ds_rd(sc, YDSXGR_CTRLSELECT, 4) & 0x00000001;
x = 0;
2000-06-06 22:34:09 +00:00
for (i = 0; i < DS1_CHANS; i++) {
if (sc->pch[i].run) {
x = 1;
snd_mtxunlock(sc->lock);
chn_intr(sc->pch[i].channel);
snd_mtxlock(sc->lock);
}
2000-06-06 22:34:09 +00:00
}
for (i = 0; i < 2; i++) {
if (sc->rch[i].run) {
x = 1;
snd_mtxunlock(sc->lock);
2000-06-06 22:34:09 +00:00
chn_intr(sc->rch[i].channel);
snd_mtxlock(sc->lock);
2000-06-06 22:34:09 +00:00
}
}
i = ds_rd(sc, YDSXGR_MODE, 4);
if (x)
ds_wr(sc, YDSXGR_MODE, i | 0x00000002, 4);
}
snd_mtxunlock(sc->lock);
}
/* -------------------------------------------------------------------- */
/*
* Probe and attach the card
*/
static void
ds_setmap(void *arg, bus_dma_segment_t *segs, int nseg, int error)
{
struct sc_info *sc = arg;
sc->ctrlbase = error? 0 : (u_int32_t)segs->ds_addr;
if (bootverbose) {
printf("ds1: setmap (%lx, %lx), nseg=%d, error=%d\n",
(unsigned long)segs->ds_addr, (unsigned long)segs->ds_len,
nseg, error);
}
}
static int
ds_init(struct sc_info *sc)
{
int i;
u_int32_t *ci, r, pcs, rcs, ecs, ws, memsz, cb;
u_int8_t *t;
void *buf;
ci = ds_devs[sc->type].mcode;
ds_wr(sc, YDSXGR_NATIVEDACOUTVOL, 0x00000000, 4);
ds_enadsp(sc, 0);
ds_wr(sc, YDSXGR_MODE, 0x00010000, 4);
ds_wr(sc, YDSXGR_MODE, 0x00000000, 4);
ds_wr(sc, YDSXGR_MAPOFREC, 0x00000000, 4);
ds_wr(sc, YDSXGR_MAPOFEFFECT, 0x00000000, 4);
ds_wr(sc, YDSXGR_PLAYCTRLBASE, 0x00000000, 4);
ds_wr(sc, YDSXGR_RECCTRLBASE, 0x00000000, 4);
ds_wr(sc, YDSXGR_EFFCTRLBASE, 0x00000000, 4);
r = ds_rd(sc, YDSXGR_GLOBALCTRL, 2);
ds_wr(sc, YDSXGR_GLOBALCTRL, r & ~0x0007, 2);
for (i = 0; i < YDSXG_DSPLENGTH; i += 4)
ds_wr(sc, YDSXGR_DSPINSTRAM + i, DspInst[i >> 2], 4);
for (i = 0; i < YDSXG_CTRLLENGTH; i += 4)
ds_wr(sc, YDSXGR_CTRLINSTRAM + i, ci[i >> 2], 4);
ds_enadsp(sc, 1);
pcs = 0;
for (i = 100; i > 0; i--) {
pcs = ds_rd(sc, YDSXGR_PLAYCTRLSIZE, 4) << 2;
if (pcs == sizeof(struct pbank))
break;
DELAY(1000);
}
if (pcs != sizeof(struct pbank)) {
device_printf(sc->dev, "preposterous playctrlsize (%d)\n", pcs);
return -1;
}
rcs = ds_rd(sc, YDSXGR_RECCTRLSIZE, 4) << 2;
ecs = ds_rd(sc, YDSXGR_EFFCTRLSIZE, 4) << 2;
ws = ds_rd(sc, YDSXGR_WORKSIZE, 4) << 2;
memsz = 64 * 2 * pcs + 2 * 2 * rcs + 5 * 2 * ecs + ws;
memsz += (64 + 1) * 4;
if (sc->regbase == NULL) {
if (bus_dma_tag_create(bus_get_dma_tag(sc->dev), 2, 0,
BUS_SPACE_MAXADDR_32BIT,
BUS_SPACE_MAXADDR,
NULL, NULL, memsz, 1, memsz, 0, NULL,
NULL, &sc->control_dmat))
return -1;
if (bus_dmamem_alloc(sc->control_dmat, &buf, BUS_DMA_NOWAIT, &sc->map))
return -1;
if (bus_dmamap_load(sc->control_dmat, sc->map, buf, memsz, ds_setmap, sc, 0) || !sc->ctrlbase) {
device_printf(sc->dev, "pcs=%d, rcs=%d, ecs=%d, ws=%d, memsz=%d\n",
pcs, rcs, ecs, ws, memsz);
return -1;
}
sc->regbase = buf;
} else
buf = sc->regbase;
cb = 0;
t = buf;
ds_wr(sc, YDSXGR_WORKBASE, sc->ctrlbase + cb, 4);
cb += ws;
sc->pbase = (u_int32_t *)(t + cb);
/* printf("pbase = %p -> 0x%x\n", sc->pbase, sc->ctrlbase + cb); */
ds_wr(sc, YDSXGR_PLAYCTRLBASE, sc->ctrlbase + cb, 4);
cb += (64 + 1) * 4;
2000-06-06 22:34:09 +00:00
sc->rbank = (struct rbank *)(t + cb);
ds_wr(sc, YDSXGR_RECCTRLBASE, sc->ctrlbase + cb, 4);
cb += 2 * 2 * rcs;
ds_wr(sc, YDSXGR_EFFCTRLBASE, sc->ctrlbase + cb, 4);
cb += 5 * 2 * ecs;
sc->pbankbase = sc->ctrlbase + cb;
sc->pbanksize = pcs;
for (i = 0; i < 64; i++) {
wrl(sc, &sc->pbase[i + 1], 0);
sc->pbank[i * 2] = (struct pbank *)(t + cb);
/* printf("pbank[%d] = %p -> 0x%x; ", i * 2, (struct pbank *)(t + cb), sc->ctrlbase + cb - vtophys(t + cb)); */
cb += pcs;
sc->pbank[i * 2 + 1] = (struct pbank *)(t + cb);
/* printf("pbank[%d] = %p -> 0x%x\n", i * 2 + 1, (struct pbank *)(t + cb), sc->ctrlbase + cb - vtophys(t + cb)); */
cb += pcs;
}
wrl(sc, &sc->pbase[0], DS1_CHANS * 2);
sc->pchn = sc->rchn = 0;
ds_wr(sc, YDSXGR_NATIVEDACOUTVOL, 0x3fff3fff, 4);
2000-06-06 22:34:09 +00:00
ds_wr(sc, YDSXGR_NATIVEADCINVOL, 0x3fff3fff, 4);
ds_wr(sc, YDSXGR_NATIVEDACINVOL, 0x3fff3fff, 4);
return 0;
}
static int
ds_uninit(struct sc_info *sc)
{
ds_wr(sc, YDSXGR_NATIVEDACOUTVOL, 0x00000000, 4);
ds_wr(sc, YDSXGR_NATIVEADCINVOL, 0, 4);
ds_wr(sc, YDSXGR_NATIVEDACINVOL, 0, 4);
ds_enadsp(sc, 0);
ds_wr(sc, YDSXGR_MODE, 0x00010000, 4);
ds_wr(sc, YDSXGR_MAPOFREC, 0x00000000, 4);
ds_wr(sc, YDSXGR_MAPOFEFFECT, 0x00000000, 4);
ds_wr(sc, YDSXGR_PLAYCTRLBASE, 0x00000000, 4);
ds_wr(sc, YDSXGR_RECCTRLBASE, 0x00000000, 4);
ds_wr(sc, YDSXGR_EFFCTRLBASE, 0x00000000, 4);
ds_wr(sc, YDSXGR_GLOBALCTRL, 0, 2);
bus_dmamap_unload(sc->control_dmat, sc->map);
bus_dmamem_free(sc->control_dmat, sc->regbase, sc->map);
return 0;
}
static int
ds_finddev(u_int32_t dev, u_int32_t subdev)
{
int i;
for (i = 0; ds_devs[i].dev; i++) {
if (ds_devs[i].dev == dev &&
(ds_devs[i].subdev == subdev || ds_devs[i].subdev == 0))
return i;
}
return -1;
}
static int
ds_pci_probe(device_t dev)
{
int i;
u_int32_t subdev;
subdev = (pci_get_subdevice(dev) << 16) | pci_get_subvendor(dev);
i = ds_finddev(pci_get_devid(dev), subdev);
if (i >= 0) {
device_set_desc(dev, ds_devs[i].name);
return BUS_PROBE_DEFAULT;
} else
return ENXIO;
}
static int
ds_pci_attach(device_t dev)
{
u_int32_t subdev, i;
struct sc_info *sc;
struct ac97_info *codec = NULL;
char status[SND_STATUSLEN];
sc = malloc(sizeof(*sc), M_DEVBUF, M_WAITOK | M_ZERO);
Fix severe out-of-bound mtx "type" pointer, causing WITNESS refcount confusions and panic provided that the following conditions are met: 1) WITNESS is enabled (watch/trace). 2) Using modules, instead of statically linked (Not a strict requirement, but easier to reproduce this way). 3) 2 or more modules share the same mtx type ("sound softc"). - They might share the same name (strcmp() == 0), but it always point to different address. 4) Repetitive kldunload/load on any module that shares the same mtx type (Not a strict requirement, but easier to reproduce this way). Consider module A and module B: - From enroll() - subr_witness.c: * Load module A. Everything seems fine right now. wA-w_refcount == 1 ; wA-w_name = "sound softc" * Load module B. * w->w_name == description will always fail. ("sound softc" from A and B point to different address). * wA->w_refcount > 0 && strcmp(description, wA->w_name) == 0 * enroll() will return wA instead of returning (possibly unique) wB. wA->w_refcount++ , == 2. * Unload module A, mtx_destroy(), wA->w_name become invalid, but wA->w_refcount-- become 1 instead of 0. wA will not be removed from witness list. * Some other places call mtx_init(), iterating witness list, found wA, failed on wA->w_name == description * wA->w_refcount > 0 && strcmp(description, wA->w_name) * Panic on strcmp() since wA->w_name no longer point to valid address. Note that this could happened in other places as well, not just sound (eg. consider lots of drivers that share simmilar MTX_NETWORK_LOCK). Solutions (for sound case): 1) Provide unique mtx type string for each mutex creation (chosen) or 2) Put "sound softc" global variable somewhere and use it.
2007-03-15 16:41:27 +00:00
sc->lock = snd_mtxcreate(device_get_nameunit(dev), "snd_ds1 softc");
sc->dev = dev;
subdev = (pci_get_subdevice(dev) << 16) | pci_get_subvendor(dev);
sc->type = ds_finddev(pci_get_devid(dev), subdev);
sc->rev = pci_get_revid(dev);
pci_enable_busmaster(dev);
sc->regid = PCIR_BAR(0);
sc->reg = bus_alloc_resource_any(dev, SYS_RES_MEMORY, &sc->regid,
RF_ACTIVE);
if (!sc->reg) {
device_printf(dev, "unable to map register space\n");
goto bad;
}
sc->st = rman_get_bustag(sc->reg);
sc->sh = rman_get_bushandle(sc->reg);
sc->bufsz = pcm_getbuffersize(dev, 4096, DS1_BUFFSIZE, 65536);
if (bus_dma_tag_create(/*parent*/bus_get_dma_tag(dev), /*alignment*/2,
/*boundary*/0,
/*lowaddr*/BUS_SPACE_MAXADDR_32BIT,
/*highaddr*/BUS_SPACE_MAXADDR,
/*filter*/NULL, /*filterarg*/NULL,
/*maxsize*/sc->bufsz, /*nsegments*/1, /*maxsegz*/0x3ffff,
/*flags*/0, /*lockfunc*/NULL,
/*lockarg*/NULL, &sc->buffer_dmat) != 0) {
device_printf(dev, "unable to create dma tag\n");
goto bad;
}
sc->regbase = NULL;
if (ds_init(sc) == -1) {
device_printf(dev, "unable to initialize the card\n");
goto bad;
}
codec = AC97_CREATE(dev, sc, ds_ac97);
if (codec == NULL)
goto bad;
/*
* Turn on inverted external amplifier sense flags for few
* 'special' boards.
*/
switch (subdev) {
case 0x81171033: /* NEC ValueStar (VT550/0) */
ac97_setflags(codec, ac97_getflags(codec) | AC97_F_EAPD_INV);
break;
default:
break;
}
mixer_init(dev, ac97_getmixerclass(), codec);
sc->irqid = 0;
sc->irq = bus_alloc_resource_any(dev, SYS_RES_IRQ, &sc->irqid,
RF_ACTIVE | RF_SHAREABLE);
if (!sc->irq || snd_setup_intr(dev, sc->irq, INTR_MPSAFE, ds_intr, sc, &sc->ih)) {
device_printf(dev, "unable to map interrupt\n");
goto bad;
}
Use uintmax_t (typedef'd to rman_res_t type) for rman ranges. On some architectures, u_long isn't large enough for resource definitions. Particularly, powerpc and arm allow 36-bit (or larger) physical addresses, but type `long' is only 32-bit. This extends rman's resources to uintmax_t. With this change, any resource can feasibly be placed anywhere in physical memory (within the constraints of the driver). Why uintmax_t and not something machine dependent, or uint64_t? Though it's possible for uintmax_t to grow, it's highly unlikely it will become 128-bit on 32-bit architectures. 64-bit architectures should have plenty of RAM to absorb the increase on resource sizes if and when this occurs, and the number of resources on memory-constrained systems should be sufficiently small as to not pose a drastic overhead. That being said, uintmax_t was chosen for source clarity. If it's specified as uint64_t, all printf()-like calls would either need casts to uintmax_t, or be littered with PRI*64 macros. Casts to uintmax_t aren't horrible, but it would also bake into the API for resource_list_print_type() either a hidden assumption that entries get cast to uintmax_t for printing, or these calls would need the PRI*64 macros. Since source code is meant to be read more often than written, I chose the clearest path of simply using uintmax_t. Tested on a PowerPC p5020-based board, which places all device resources in 0xfxxxxxxxx, and has 8GB RAM. Regression tested on qemu-system-i386 Regression tested on qemu-system-mips (malta profile) Tested PAE and devinfo on virtualbox (live CD) Special thanks to bz for his testing on ARM. Reviewed By: bz, jhb (previous) Relnotes: Yes Sponsored by: Alex Perez/Inertial Computing Differential Revision: https://reviews.freebsd.org/D4544
2016-03-18 01:28:41 +00:00
snprintf(status, SND_STATUSLEN, "at memory 0x%jx irq %jd %s",
rman_get_start(sc->reg), rman_get_start(sc->irq),PCM_KLDSTRING(snd_ds1));
2000-06-06 22:34:09 +00:00
if (pcm_register(dev, sc, DS1_CHANS, 2))
goto bad;
for (i = 0; i < DS1_CHANS; i++)
pcm_addchan(dev, PCMDIR_PLAY, &ds1pchan_class, sc);
2000-06-06 22:34:09 +00:00
for (i = 0; i < 2; i++)
pcm_addchan(dev, PCMDIR_REC, &ds1rchan_class, sc);
pcm_setstatus(dev, status);
return 0;
bad:
if (codec)
ac97_destroy(codec);
if (sc->reg)
bus_release_resource(dev, SYS_RES_MEMORY, sc->regid, sc->reg);
if (sc->ih)
bus_teardown_intr(dev, sc->irq, sc->ih);
if (sc->irq)
bus_release_resource(dev, SYS_RES_IRQ, sc->irqid, sc->irq);
if (sc->buffer_dmat)
bus_dma_tag_destroy(sc->buffer_dmat);
if (sc->control_dmat)
bus_dma_tag_destroy(sc->control_dmat);
if (sc->lock)
snd_mtxfree(sc->lock);
free(sc, M_DEVBUF);
return ENXIO;
}
static int
ds_pci_resume(device_t dev)
{
struct sc_info *sc;
sc = pcm_getdevinfo(dev);
if (ds_init(sc) == -1) {
device_printf(dev, "unable to reinitialize the card\n");
return ENXIO;
}
if (mixer_reinit(dev) == -1) {
device_printf(dev, "unable to reinitialize the mixer\n");
return ENXIO;
}
return 0;
}
static int
ds_pci_detach(device_t dev)
{
int r;
struct sc_info *sc;
r = pcm_unregister(dev);
if (r)
return r;
sc = pcm_getdevinfo(dev);
ds_uninit(sc);
bus_release_resource(dev, SYS_RES_MEMORY, sc->regid, sc->reg);
bus_teardown_intr(dev, sc->irq, sc->ih);
bus_release_resource(dev, SYS_RES_IRQ, sc->irqid, sc->irq);
bus_dma_tag_destroy(sc->buffer_dmat);
bus_dma_tag_destroy(sc->control_dmat);
snd_mtxfree(sc->lock);
free(sc, M_DEVBUF);
return 0;
}
static device_method_t ds1_methods[] = {
/* Device interface */
DEVMETHOD(device_probe, ds_pci_probe),
DEVMETHOD(device_attach, ds_pci_attach),
DEVMETHOD(device_detach, ds_pci_detach),
DEVMETHOD(device_resume, ds_pci_resume),
{ 0, 0 }
};
static driver_t ds1_driver = {
"pcm",
ds1_methods,
PCM_SOFTC_SIZE,
};
DRIVER_MODULE(snd_ds1, pci, ds1_driver, pcm_devclass, 0, 0);
MODULE_DEPEND(snd_ds1, sound, SOUND_MINVER, SOUND_PREFVER, SOUND_MAXVER);
MODULE_VERSION(snd_ds1, 1);