freebsd-dev/sys/dev/sound/pci/t4dwave.c

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/*-
* SPDX-License-Identifier: BSD-2-Clause-FreeBSD
*
2003-09-07 16:28:03 +00:00
* Copyright (c) 1999 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/sound/pci/t4dwave.h>
#include <dev/pci/pcireg.h>
#include <dev/pci/pcivar.h>
SND_DECLARE_FILE("$FreeBSD$");
/* -------------------------------------------------------------------- */
#define TDX_PCI_ID 0x20001023
#define TNX_PCI_ID 0x20011023
#define ALI_PCI_ID 0x545110b9
#define SPA_PCI_ID 0x70181039
2001-10-08 05:59:54 +00:00
#define TR_DEFAULT_BUFSZ 0x1000
/* For ALi M5451 the DMA transfer size appears to be fixed to 64k. */
#define ALI_BUFSZ 0x10000
#define TR_BUFALGN 0x8
#define TR_TIMEOUT_CDC 0xffff
#define TR_MAXHWCH 64
#define ALI_MAXHWCH 32
#define TR_MAXPLAYCH 4
#define ALI_MAXPLAYCH 1
/*
* Though, it's not clearly documented in the 4DWAVE datasheet, the
* DX and NX chips can't handle DMA addresses located above 1GB as the
* LBA (loop begin address) register which holds the DMA base address
* is 32-bit, but the two MSBs are used for other purposes.
*/
#define TR_MAXADDR ((1U << 30) - 1)
#define ALI_MAXADDR ((1U << 31) - 1)
struct tr_info;
/* channel registers */
struct tr_chinfo {
u_int32_t cso, alpha, fms, fmc, ec;
u_int32_t lba;
u_int32_t eso, delta;
u_int32_t rvol, cvol;
u_int32_t gvsel, pan, vol, ctrl;
u_int32_t active:1, was_active:1;
int index, bufhalf;
struct snd_dbuf *buffer;
struct pcm_channel *channel;
struct tr_info *parent;
};
struct tr_rchinfo {
u_int32_t delta;
u_int32_t active:1, was_active:1;
struct snd_dbuf *buffer;
struct pcm_channel *channel;
struct tr_info *parent;
};
/* device private data */
struct tr_info {
u_int32_t type;
u_int32_t rev;
bus_space_tag_t st;
bus_space_handle_t sh;
bus_dma_tag_t parent_dmat;
struct resource *reg, *irq;
2001-10-08 05:59:54 +00:00
int regtype, regid, irqid;
void *ih;
struct mtx *lock;
u_int32_t hwchns;
u_int32_t playchns;
2001-10-08 05:59:54 +00:00
unsigned int bufsz;
struct tr_chinfo chinfo[TR_MAXPLAYCH];
struct tr_rchinfo recchinfo;
};
/* -------------------------------------------------------------------- */
static u_int32_t tr_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 tr_reccaps = {4000, 48000, tr_recfmt, 0};
static u_int32_t tr_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_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 tr_playcaps = {4000, 48000, tr_playfmt, 0};
/* -------------------------------------------------------------------- */
/* Hardware */
static u_int32_t
tr_rd(struct tr_info *tr, int regno, int size)
{
switch(size) {
case 1:
return bus_space_read_1(tr->st, tr->sh, regno);
case 2:
return bus_space_read_2(tr->st, tr->sh, regno);
case 4:
return bus_space_read_4(tr->st, tr->sh, regno);
default:
return 0xffffffff;
}
}
static void
tr_wr(struct tr_info *tr, int regno, u_int32_t data, int size)
{
switch(size) {
case 1:
bus_space_write_1(tr->st, tr->sh, regno, data);
break;
case 2:
bus_space_write_2(tr->st, tr->sh, regno, data);
break;
case 4:
bus_space_write_4(tr->st, tr->sh, regno, data);
break;
}
}
/* -------------------------------------------------------------------- */
/* ac97 codec */
static int
tr_rdcd(kobj_t obj, void *devinfo, int regno)
{
struct tr_info *tr = (struct tr_info *)devinfo;
int i, j, treg, trw;
switch (tr->type) {
case SPA_PCI_ID:
treg=SPA_REG_CODECRD;
trw=SPA_CDC_RWSTAT;
break;
case ALI_PCI_ID:
if (tr->rev > 0x01)
treg=TDX_REG_CODECWR;
else
treg=TDX_REG_CODECRD;
trw=TDX_CDC_RWSTAT;
break;
case TDX_PCI_ID:
treg=TDX_REG_CODECRD;
trw=TDX_CDC_RWSTAT;
break;
case TNX_PCI_ID:
treg=(regno & 0x100)? TNX_REG_CODEC2RD : TNX_REG_CODEC1RD;
trw=TNX_CDC_RWSTAT;
break;
default:
printf("!!! tr_rdcd defaulted !!!\n");
return -1;
}
i = j = 0;
regno &= 0x7f;
snd_mtxlock(tr->lock);
if (tr->type == ALI_PCI_ID) {
u_int32_t chk1, chk2;
j = trw;
for (i = TR_TIMEOUT_CDC; (i > 0) && (j & trw); i--)
j = tr_rd(tr, treg, 4);
if (i > 0) {
chk1 = tr_rd(tr, 0xc8, 4);
chk2 = tr_rd(tr, 0xc8, 4);
for (i = TR_TIMEOUT_CDC; (i > 0) && (chk1 == chk2);
i--)
chk2 = tr_rd(tr, 0xc8, 4);
}
}
if (tr->type != ALI_PCI_ID || i > 0) {
tr_wr(tr, treg, regno | trw, 4);
j=trw;
for (i=TR_TIMEOUT_CDC; (i > 0) && (j & trw); i--)
j=tr_rd(tr, treg, 4);
}
snd_mtxunlock(tr->lock);
if (i == 0) printf("codec timeout during read of register %x\n", regno);
return (j >> TR_CDC_DATA) & 0xffff;
}
static int
tr_wrcd(kobj_t obj, void *devinfo, int regno, u_int32_t data)
{
struct tr_info *tr = (struct tr_info *)devinfo;
int i, j, treg, trw;
switch (tr->type) {
case SPA_PCI_ID:
treg=SPA_REG_CODECWR;
trw=SPA_CDC_RWSTAT;
break;
case ALI_PCI_ID:
case TDX_PCI_ID:
treg=TDX_REG_CODECWR;
trw=TDX_CDC_RWSTAT;
break;
case TNX_PCI_ID:
treg=TNX_REG_CODECWR;
trw=TNX_CDC_RWSTAT | ((regno & 0x100)? TNX_CDC_SEC : 0);
break;
default:
printf("!!! tr_wrcd defaulted !!!");
return -1;
}
i = 0;
regno &= 0x7f;
#if 0
printf("tr_wrcd: reg %x was %x", regno, tr_rdcd(devinfo, regno));
#endif
j=trw;
snd_mtxlock(tr->lock);
if (tr->type == ALI_PCI_ID) {
j = trw;
for (i = TR_TIMEOUT_CDC; (i > 0) && (j & trw); i--)
j = tr_rd(tr, treg, 4);
if (i > 0) {
u_int32_t chk1, chk2;
chk1 = tr_rd(tr, 0xc8, 4);
chk2 = tr_rd(tr, 0xc8, 4);
for (i = TR_TIMEOUT_CDC; (i > 0) && (chk1 == chk2);
i--)
chk2 = tr_rd(tr, 0xc8, 4);
}
}
if (tr->type != ALI_PCI_ID || i > 0) {
for (i=TR_TIMEOUT_CDC; (i>0) && (j & trw); i--)
j=tr_rd(tr, treg, 4);
if (tr->type == ALI_PCI_ID && tr->rev > 0x01)
trw |= 0x0100;
tr_wr(tr, treg, (data << TR_CDC_DATA) | regno | trw, 4);
}
#if 0
printf(" - wrote %x, now %x\n", data, tr_rdcd(devinfo, regno));
#endif
snd_mtxunlock(tr->lock);
if (i==0) printf("codec timeout writing %x, data %x\n", regno, data);
return (i > 0)? 0 : -1;
}
static kobj_method_t tr_ac97_methods[] = {
KOBJMETHOD(ac97_read, tr_rdcd),
KOBJMETHOD(ac97_write, tr_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(tr_ac97);
/* -------------------------------------------------------------------- */
/* playback channel interrupts */
#if 0
static u_int32_t
tr_testint(struct tr_chinfo *ch)
{
struct tr_info *tr = ch->parent;
int bank, chan;
bank = (ch->index & 0x20) ? 1 : 0;
chan = ch->index & 0x1f;
return tr_rd(tr, bank? TR_REG_ADDRINTB : TR_REG_ADDRINTA, 4) & (1 << chan);
}
#endif
static void
tr_clrint(struct tr_chinfo *ch)
{
struct tr_info *tr = ch->parent;
int bank, chan;
bank = (ch->index & 0x20) ? 1 : 0;
chan = ch->index & 0x1f;
tr_wr(tr, bank? TR_REG_ADDRINTB : TR_REG_ADDRINTA, 1 << chan, 4);
}
static void
tr_enaint(struct tr_chinfo *ch, int enable)
{
struct tr_info *tr = ch->parent;
u_int32_t i, reg;
int bank, chan;
snd_mtxlock(tr->lock);
bank = (ch->index & 0x20) ? 1 : 0;
chan = ch->index & 0x1f;
reg = bank? TR_REG_INTENB : TR_REG_INTENA;
i = tr_rd(tr, reg, 4);
i &= ~(1 << chan);
i |= (enable? 1 : 0) << chan;
tr_clrint(ch);
tr_wr(tr, reg, i, 4);
snd_mtxunlock(tr->lock);
}
/* playback channels */
static void
tr_selch(struct tr_chinfo *ch)
{
struct tr_info *tr = ch->parent;
int i;
i = tr_rd(tr, TR_REG_CIR, 4);
i &= ~TR_CIR_MASK;
i |= ch->index & 0x3f;
tr_wr(tr, TR_REG_CIR, i, 4);
}
static void
tr_startch(struct tr_chinfo *ch)
{
struct tr_info *tr = ch->parent;
int bank, chan;
bank = (ch->index & 0x20) ? 1 : 0;
chan = ch->index & 0x1f;
tr_wr(tr, bank? TR_REG_STARTB : TR_REG_STARTA, 1 << chan, 4);
}
static void
tr_stopch(struct tr_chinfo *ch)
{
struct tr_info *tr = ch->parent;
int bank, chan;
bank = (ch->index & 0x20) ? 1 : 0;
chan = ch->index & 0x1f;
tr_wr(tr, bank? TR_REG_STOPB : TR_REG_STOPA, 1 << chan, 4);
}
static void
tr_wrch(struct tr_chinfo *ch)
{
struct tr_info *tr = ch->parent;
u_int32_t cr[TR_CHN_REGS], i;
ch->gvsel &= 0x00000001;
ch->fmc &= 0x00000003;
ch->fms &= 0x0000000f;
ch->ctrl &= 0x0000000f;
ch->pan &= 0x0000007f;
ch->rvol &= 0x0000007f;
ch->cvol &= 0x0000007f;
ch->vol &= 0x000000ff;
ch->ec &= 0x00000fff;
ch->alpha &= 0x00000fff;
ch->delta &= 0x0000ffff;
if (tr->type == ALI_PCI_ID)
ch->lba &= ALI_MAXADDR;
else
ch->lba &= TR_MAXADDR;
cr[1]=ch->lba;
cr[3]=(ch->fmc<<14) | (ch->rvol<<7) | (ch->cvol);
cr[4]=(ch->gvsel<<31) | (ch->pan<<24) | (ch->vol<<16) | (ch->ctrl<<12) | (ch->ec);
switch (tr->type) {
case SPA_PCI_ID:
case ALI_PCI_ID:
case TDX_PCI_ID:
ch->cso &= 0x0000ffff;
ch->eso &= 0x0000ffff;
cr[0]=(ch->cso<<16) | (ch->alpha<<4) | (ch->fms);
cr[2]=(ch->eso<<16) | (ch->delta);
break;
case TNX_PCI_ID:
ch->cso &= 0x00ffffff;
ch->eso &= 0x00ffffff;
cr[0]=((ch->delta & 0xff)<<24) | (ch->cso);
cr[2]=((ch->delta>>8)<<24) | (ch->eso);
cr[3]|=(ch->alpha<<20) | (ch->fms<<16) | (ch->fmc<<14);
break;
}
snd_mtxlock(tr->lock);
tr_selch(ch);
for (i=0; i<TR_CHN_REGS; i++)
tr_wr(tr, TR_REG_CHNBASE+(i<<2), cr[i], 4);
snd_mtxunlock(tr->lock);
}
static void
tr_rdch(struct tr_chinfo *ch)
{
struct tr_info *tr = ch->parent;
u_int32_t cr[5], i;
snd_mtxlock(tr->lock);
tr_selch(ch);
for (i=0; i<5; i++)
cr[i]=tr_rd(tr, TR_REG_CHNBASE+(i<<2), 4);
snd_mtxunlock(tr->lock);
if (tr->type == ALI_PCI_ID)
ch->lba=(cr[1] & ALI_MAXADDR);
else
ch->lba=(cr[1] & TR_MAXADDR);
ch->fmc= (cr[3] & 0x0000c000) >> 14;
ch->rvol= (cr[3] & 0x00003f80) >> 7;
ch->cvol= (cr[3] & 0x0000007f);
ch->gvsel= (cr[4] & 0x80000000) >> 31;
ch->pan= (cr[4] & 0x7f000000) >> 24;
ch->vol= (cr[4] & 0x00ff0000) >> 16;
ch->ctrl= (cr[4] & 0x0000f000) >> 12;
ch->ec= (cr[4] & 0x00000fff);
switch(tr->type) {
case SPA_PCI_ID:
case ALI_PCI_ID:
case TDX_PCI_ID:
ch->cso= (cr[0] & 0xffff0000) >> 16;
ch->alpha= (cr[0] & 0x0000fff0) >> 4;
ch->fms= (cr[0] & 0x0000000f);
ch->eso= (cr[2] & 0xffff0000) >> 16;
ch->delta= (cr[2] & 0x0000ffff);
break;
case TNX_PCI_ID:
ch->cso= (cr[0] & 0x00ffffff);
ch->eso= (cr[2] & 0x00ffffff);
ch->delta= ((cr[2] & 0xff000000) >> 16) | ((cr[0] & 0xff000000) >> 24);
ch->alpha= (cr[3] & 0xfff00000) >> 20;
ch->fms= (cr[3] & 0x000f0000) >> 16;
break;
}
}
static u_int32_t
tr_fmttobits(u_int32_t fmt)
{
u_int32_t bits;
bits = 0;
bits |= (fmt & AFMT_SIGNED)? 0x2 : 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
bits |= (AFMT_CHANNEL(fmt) > 1)? 0x4 : 0;
bits |= (fmt & AFMT_16BIT)? 0x8 : 0;
return bits;
}
/* -------------------------------------------------------------------- */
/* channel interface */
static void *
trpchan_init(kobj_t obj, void *devinfo, struct snd_dbuf *b, struct pcm_channel *c, int dir)
{
struct tr_info *tr = devinfo;
struct tr_chinfo *ch;
KASSERT(dir == PCMDIR_PLAY, ("trpchan_init: bad direction"));
ch = &tr->chinfo[tr->playchns];
ch->index = tr->playchns++;
ch->buffer = b;
ch->parent = tr;
ch->channel = c;
if (sndbuf_alloc(ch->buffer, tr->parent_dmat, 0, tr->bufsz) != 0)
return NULL;
return ch;
}
static int
trpchan_setformat(kobj_t obj, void *data, u_int32_t format)
{
struct tr_chinfo *ch = data;
ch->ctrl = tr_fmttobits(format) | 0x01;
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
trpchan_setspeed(kobj_t obj, void *data, u_int32_t speed)
{
struct tr_chinfo *ch = data;
ch->delta = (speed << 12) / 48000;
return (ch->delta * 48000) >> 12;
}
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
trpchan_setblocksize(kobj_t obj, void *data, u_int32_t blocksize)
{
struct tr_chinfo *ch = data;
sndbuf_resize(ch->buffer, 2, blocksize);
return blocksize;
}
static int
trpchan_trigger(kobj_t obj, void *data, int go)
{
struct tr_chinfo *ch = data;
if (!PCMTRIG_COMMON(go))
return 0;
if (go == PCMTRIG_START) {
ch->fmc = 3;
ch->fms = 0;
ch->ec = 0;
ch->alpha = 0;
ch->lba = sndbuf_getbufaddr(ch->buffer);
ch->cso = 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
ch->eso = (sndbuf_getsize(ch->buffer) / sndbuf_getalign(ch->buffer)) - 1;
ch->rvol = ch->cvol = 0x7f;
ch->gvsel = 0;
ch->pan = 0;
ch->vol = 0;
ch->bufhalf = 0;
tr_wrch(ch);
tr_enaint(ch, 1);
tr_startch(ch);
ch->active = 1;
} else {
tr_stopch(ch);
ch->active = 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
trpchan_getptr(kobj_t obj, void *data)
{
struct tr_chinfo *ch = data;
tr_rdch(ch);
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
return ch->cso * sndbuf_getalign(ch->buffer);
}
static struct pcmchan_caps *
trpchan_getcaps(kobj_t obj, void *data)
{
return &tr_playcaps;
}
static kobj_method_t trpchan_methods[] = {
KOBJMETHOD(channel_init, trpchan_init),
KOBJMETHOD(channel_setformat, trpchan_setformat),
KOBJMETHOD(channel_setspeed, trpchan_setspeed),
KOBJMETHOD(channel_setblocksize, trpchan_setblocksize),
KOBJMETHOD(channel_trigger, trpchan_trigger),
KOBJMETHOD(channel_getptr, trpchan_getptr),
KOBJMETHOD(channel_getcaps, trpchan_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(trpchan);
/* -------------------------------------------------------------------- */
/* rec channel interface */
static void *
trrchan_init(kobj_t obj, void *devinfo, struct snd_dbuf *b, struct pcm_channel *c, int dir)
{
struct tr_info *tr = devinfo;
struct tr_rchinfo *ch;
KASSERT(dir == PCMDIR_REC, ("trrchan_init: bad direction"));
ch = &tr->recchinfo;
ch->buffer = b;
ch->parent = tr;
ch->channel = c;
if (sndbuf_alloc(ch->buffer, tr->parent_dmat, 0, tr->bufsz) != 0)
return NULL;
return ch;
}
static int
trrchan_setformat(kobj_t obj, void *data, u_int32_t format)
{
struct tr_rchinfo *ch = data;
struct tr_info *tr = ch->parent;
u_int32_t i, bits;
bits = tr_fmttobits(format);
/* set # of samples between interrupts */
i = (sndbuf_runsz(ch->buffer) >> ((bits & 0x08)? 1 : 0)) - 1;
tr_wr(tr, TR_REG_SBBL, i | (i << 16), 4);
/* set sample format */
i = 0x18 | (bits << 4);
tr_wr(tr, TR_REG_SBCTRL, i, 1);
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
trrchan_setspeed(kobj_t obj, void *data, u_int32_t speed)
{
struct tr_rchinfo *ch = data;
struct tr_info *tr = ch->parent;
/* setup speed */
ch->delta = (48000 << 12) / speed;
tr_wr(tr, TR_REG_SBDELTA, ch->delta, 2);
/* return closest possible speed */
return (48000 << 12) / ch->delta;
}
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
trrchan_setblocksize(kobj_t obj, void *data, u_int32_t blocksize)
{
struct tr_rchinfo *ch = data;
sndbuf_resize(ch->buffer, 2, blocksize);
return blocksize;
}
static int
trrchan_trigger(kobj_t obj, void *data, int go)
{
struct tr_rchinfo *ch = data;
struct tr_info *tr = ch->parent;
u_int32_t i;
if (!PCMTRIG_COMMON(go))
return 0;
if (go == PCMTRIG_START) {
/* set up dma mode regs */
tr_wr(tr, TR_REG_DMAR15, 0, 1);
i = tr_rd(tr, TR_REG_DMAR11, 1) & 0x03;
tr_wr(tr, TR_REG_DMAR11, i | 0x54, 1);
/* set up base address */
tr_wr(tr, TR_REG_DMAR0, sndbuf_getbufaddr(ch->buffer), 4);
/* set up buffer size */
i = tr_rd(tr, TR_REG_DMAR4, 4) & ~0x00ffffff;
tr_wr(tr, TR_REG_DMAR4, i | (sndbuf_runsz(ch->buffer) - 1), 4);
/* start */
tr_wr(tr, TR_REG_SBCTRL, tr_rd(tr, TR_REG_SBCTRL, 1) | 1, 1);
ch->active = 1;
} else {
tr_wr(tr, TR_REG_SBCTRL, tr_rd(tr, TR_REG_SBCTRL, 1) & ~7, 1);
ch->active = 0;
}
/* return 0 if ok */
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
trrchan_getptr(kobj_t obj, void *data)
{
struct tr_rchinfo *ch = data;
struct tr_info *tr = ch->parent;
/* return current byte offset of channel */
return tr_rd(tr, TR_REG_DMAR0, 4) - sndbuf_getbufaddr(ch->buffer);
}
static struct pcmchan_caps *
trrchan_getcaps(kobj_t obj, void *data)
{
return &tr_reccaps;
}
static kobj_method_t trrchan_methods[] = {
KOBJMETHOD(channel_init, trrchan_init),
KOBJMETHOD(channel_setformat, trrchan_setformat),
KOBJMETHOD(channel_setspeed, trrchan_setspeed),
KOBJMETHOD(channel_setblocksize, trrchan_setblocksize),
KOBJMETHOD(channel_trigger, trrchan_trigger),
KOBJMETHOD(channel_getptr, trrchan_getptr),
KOBJMETHOD(channel_getcaps, trrchan_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(trrchan);
/* -------------------------------------------------------------------- */
/* The interrupt handler */
static void
tr_intr(void *p)
{
struct tr_info *tr = (struct tr_info *)p;
struct tr_chinfo *ch;
u_int32_t active, mask, bufhalf, chnum, intsrc;
int tmp;
intsrc = tr_rd(tr, TR_REG_MISCINT, 4);
if (intsrc & TR_INT_ADDR) {
chnum = 0;
while (chnum < tr->hwchns) {
mask = 0x00000001;
active = tr_rd(tr, (chnum < 32)? TR_REG_ADDRINTA : TR_REG_ADDRINTB, 4);
bufhalf = tr_rd(tr, (chnum < 32)? TR_REG_CSPF_A : TR_REG_CSPF_B, 4);
if (active) {
do {
if (active & mask) {
tmp = (bufhalf & mask)? 1 : 0;
if (chnum < tr->playchns) {
ch = &tr->chinfo[chnum];
/* printf("%d @ %d, ", chnum, trpchan_getptr(NULL, ch)); */
if (ch->bufhalf != tmp) {
chn_intr(ch->channel);
ch->bufhalf = tmp;
}
}
}
chnum++;
mask <<= 1;
} while (chnum & 31);
} else
chnum += 32;
tr_wr(tr, (chnum <= 32)? TR_REG_ADDRINTA : TR_REG_ADDRINTB, active, 4);
}
}
if (intsrc & TR_INT_SB) {
chn_intr(tr->recchinfo.channel);
tr_rd(tr, TR_REG_SBR9, 1);
tr_rd(tr, TR_REG_SBR10, 1);
}
}
/* -------------------------------------------------------------------- */
/*
* Probe and attach the card
*/
static int
tr_init(struct tr_info *tr)
{
switch (tr->type) {
case SPA_PCI_ID:
tr_wr(tr, SPA_REG_GPIO, 0, 4);
tr_wr(tr, SPA_REG_CODECST, SPA_RST_OFF, 4);
break;
case TDX_PCI_ID:
tr_wr(tr, TDX_REG_CODECST, TDX_CDC_ON, 4);
break;
case TNX_PCI_ID:
tr_wr(tr, TNX_REG_CODECST, TNX_CDC_ON, 4);
break;
}
tr_wr(tr, TR_REG_CIR, TR_CIR_MIDENA | TR_CIR_ADDRENA, 4);
return 0;
}
static int
tr_pci_probe(device_t dev)
{
switch (pci_get_devid(dev)) {
case SPA_PCI_ID:
device_set_desc(dev, "SiS 7018");
return BUS_PROBE_DEFAULT;
case ALI_PCI_ID:
device_set_desc(dev, "Acer Labs M5451");
return BUS_PROBE_DEFAULT;
case TDX_PCI_ID:
device_set_desc(dev, "Trident 4DWave DX");
return BUS_PROBE_DEFAULT;
case TNX_PCI_ID:
device_set_desc(dev, "Trident 4DWave NX");
return BUS_PROBE_DEFAULT;
}
return ENXIO;
}
static int
tr_pci_attach(device_t dev)
{
struct tr_info *tr;
struct ac97_info *codec = NULL;
bus_addr_t lowaddr;
int i, dacn;
char status[SND_STATUSLEN];
tr = malloc(sizeof(*tr), M_DEVBUF, M_WAITOK | M_ZERO);
tr->type = pci_get_devid(dev);
tr->rev = pci_get_revid(dev);
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
tr->lock = snd_mtxcreate(device_get_nameunit(dev), "snd_t4dwave softc");
if (resource_int_value(device_get_name(dev), device_get_unit(dev),
"dac", &i) == 0) {
if (i < 1)
dacn = 1;
else if (i > TR_MAXPLAYCH)
dacn = TR_MAXPLAYCH;
else
dacn = i;
} else {
switch (tr->type) {
case ALI_PCI_ID:
dacn = ALI_MAXPLAYCH;
break;
default:
dacn = TR_MAXPLAYCH;
break;
}
}
pci_enable_busmaster(dev);
tr->regid = PCIR_BAR(0);
tr->regtype = SYS_RES_IOPORT;
tr->reg = bus_alloc_resource_any(dev, tr->regtype, &tr->regid,
RF_ACTIVE);
if (tr->reg) {
tr->st = rman_get_bustag(tr->reg);
tr->sh = rman_get_bushandle(tr->reg);
} else {
device_printf(dev, "unable to map register space\n");
goto bad;
}
if (tr_init(tr) == -1) {
device_printf(dev, "unable to initialize the card\n");
goto bad;
}
tr->playchns = 0;
codec = AC97_CREATE(dev, tr, tr_ac97);
if (codec == NULL) goto bad;
if (mixer_init(dev, ac97_getmixerclass(), codec) == -1) goto bad;
tr->irqid = 0;
tr->irq = bus_alloc_resource_any(dev, SYS_RES_IRQ, &tr->irqid,
RF_ACTIVE | RF_SHAREABLE);
if (!tr->irq || snd_setup_intr(dev, tr->irq, 0, tr_intr, tr, &tr->ih)) {
device_printf(dev, "unable to map interrupt\n");
goto bad;
}
if (tr->type == ALI_PCI_ID) {
/*
* The M5451 generates 31 bit of DMA and in order to do
* 32-bit DMA, the 31st bit can be set via its accompanying
* ISA bridge. Note that we can't predict whether bus_dma(9)
* will actually supply us with a 32-bit buffer and even when
* using a low address of BUS_SPACE_MAXADDR_32BIT for both
* we might end up with the play buffer being in the 32-bit
* range while the record buffer isn't or vice versa. So we
* don't enabling the 31st bit.
*/
lowaddr = ALI_MAXADDR;
tr->hwchns = ALI_MAXHWCH;
tr->bufsz = ALI_BUFSZ;
} else {
lowaddr = TR_MAXADDR;
tr->hwchns = TR_MAXHWCH;
tr->bufsz = pcm_getbuffersize(dev, 4096, TR_DEFAULT_BUFSZ,
65536);
}
if (bus_dma_tag_create(/*parent*/bus_get_dma_tag(dev),
/*alignment*/TR_BUFALGN,
/*boundary*/0,
/*lowaddr*/lowaddr,
/*highaddr*/BUS_SPACE_MAXADDR,
/*filter*/NULL, /*filterarg*/NULL,
/*maxsize*/tr->bufsz, /*nsegments*/1, /*maxsegz*/tr->bufsz,
/*flags*/0, /*lockfunc*/busdma_lock_mutex,
/*lockarg*/&Giant, &tr->parent_dmat) != 0) {
device_printf(dev, "unable to create dma tag\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, 64, "at io 0x%jx irq %jd %s",
rman_get_start(tr->reg), rman_get_start(tr->irq),PCM_KLDSTRING(snd_t4dwave));
if (pcm_register(dev, tr, dacn, 1))
goto bad;
pcm_addchan(dev, PCMDIR_REC, &trrchan_class, tr);
for (i = 0; i < dacn; i++)
pcm_addchan(dev, PCMDIR_PLAY, &trpchan_class, tr);
pcm_setstatus(dev, status);
return 0;
bad:
if (codec) ac97_destroy(codec);
if (tr->reg) bus_release_resource(dev, tr->regtype, tr->regid, tr->reg);
if (tr->ih) bus_teardown_intr(dev, tr->irq, tr->ih);
if (tr->irq) bus_release_resource(dev, SYS_RES_IRQ, tr->irqid, tr->irq);
if (tr->parent_dmat) bus_dma_tag_destroy(tr->parent_dmat);
if (tr->lock) snd_mtxfree(tr->lock);
free(tr, M_DEVBUF);
return ENXIO;
}
static int
tr_pci_detach(device_t dev)
{
int r;
struct tr_info *tr;
r = pcm_unregister(dev);
if (r)
return r;
tr = pcm_getdevinfo(dev);
bus_release_resource(dev, tr->regtype, tr->regid, tr->reg);
bus_teardown_intr(dev, tr->irq, tr->ih);
bus_release_resource(dev, SYS_RES_IRQ, tr->irqid, tr->irq);
bus_dma_tag_destroy(tr->parent_dmat);
snd_mtxfree(tr->lock);
free(tr, M_DEVBUF);
return 0;
}
static int
tr_pci_suspend(device_t dev)
{
int i;
struct tr_info *tr;
tr = pcm_getdevinfo(dev);
for (i = 0; i < tr->playchns; i++) {
tr->chinfo[i].was_active = tr->chinfo[i].active;
if (tr->chinfo[i].active) {
trpchan_trigger(NULL, &tr->chinfo[i], PCMTRIG_STOP);
}
}
tr->recchinfo.was_active = tr->recchinfo.active;
if (tr->recchinfo.active) {
trrchan_trigger(NULL, &tr->recchinfo, PCMTRIG_STOP);
}
return 0;
}
static int
tr_pci_resume(device_t dev)
{
int i;
struct tr_info *tr;
tr = pcm_getdevinfo(dev);
if (tr_init(tr) == -1) {
device_printf(dev, "unable to initialize the card\n");
return ENXIO;
}
if (mixer_reinit(dev) == -1) {
device_printf(dev, "unable to initialize the mixer\n");
return ENXIO;
}
for (i = 0; i < tr->playchns; i++) {
if (tr->chinfo[i].was_active) {
trpchan_trigger(NULL, &tr->chinfo[i], PCMTRIG_START);
}
}
if (tr->recchinfo.was_active) {
trrchan_trigger(NULL, &tr->recchinfo, PCMTRIG_START);
}
return 0;
}
static device_method_t tr_methods[] = {
/* Device interface */
DEVMETHOD(device_probe, tr_pci_probe),
DEVMETHOD(device_attach, tr_pci_attach),
DEVMETHOD(device_detach, tr_pci_detach),
DEVMETHOD(device_suspend, tr_pci_suspend),
DEVMETHOD(device_resume, tr_pci_resume),
{ 0, 0 }
};
static driver_t tr_driver = {
"pcm",
tr_methods,
PCM_SOFTC_SIZE,
};
DRIVER_MODULE(snd_t4dwave, pci, tr_driver, pcm_devclass, 0, 0);
MODULE_DEPEND(snd_t4dwave, sound, SOUND_MINVER, SOUND_PREFVER, SOUND_MAXVER);
MODULE_VERSION(snd_t4dwave, 1);