freebsd-dev/sys/dev/sound/pci/fm801.c

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/*-
* SPDX-License-Identifier: BSD-2-Clause-FreeBSD
*
* Copyright (c) 2000 Dmitry Dicky diwil@dataart.com
* All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS `AS IS'' AND
* ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
* ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE
* FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
* OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
* LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
* SUCH DAMAGE.
*/
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
#ifdef HAVE_KERNEL_OPTION_HEADERS
#include "opt_snd.h"
#endif
#include <dev/sound/pcm/sound.h>
#include <dev/sound/pcm/ac97.h>
#include <dev/pci/pcireg.h>
#include <dev/pci/pcivar.h>
SND_DECLARE_FILE("$FreeBSD$");
#define PCI_VENDOR_FORTEMEDIA 0x1319
#define PCI_DEVICE_FORTEMEDIA1 0x08011319 /* Audio controller */
#define PCI_DEVICE_FORTEMEDIA2 0x08021319 /* Joystick controller */
#define FM_PCM_VOLUME 0x00
#define FM_FM_VOLUME 0x02
#define FM_I2S_VOLUME 0x04
#define FM_RECORD_SOURCE 0x06
#define FM_PLAY_CTL 0x08
#define FM_PLAY_RATE_MASK 0x0f00
#define FM_PLAY_BUF1_LAST 0x0001
#define FM_PLAY_BUF2_LAST 0x0002
#define FM_PLAY_START 0x0020
#define FM_PLAY_PAUSE 0x0040
#define FM_PLAY_STOPNOW 0x0080
#define FM_PLAY_16BIT 0x4000
#define FM_PLAY_STEREO 0x8000
#define FM_PLAY_DMALEN 0x0a
#define FM_PLAY_DMABUF1 0x0c
#define FM_PLAY_DMABUF2 0x10
#define FM_REC_CTL 0x14
#define FM_REC_RATE_MASK 0x0f00
#define FM_REC_BUF1_LAST 0x0001
#define FM_REC_BUF2_LAST 0x0002
#define FM_REC_START 0x0020
#define FM_REC_PAUSE 0x0040
#define FM_REC_STOPNOW 0x0080
#define FM_REC_16BIT 0x4000
#define FM_REC_STEREO 0x8000
#define FM_REC_DMALEN 0x16
#define FM_REC_DMABUF1 0x18
#define FM_REC_DMABUF2 0x1c
#define FM_CODEC_CTL 0x22
#define FM_VOLUME 0x26
#define FM_VOLUME_MUTE 0x8000
#define FM_CODEC_CMD 0x2a
#define FM_CODEC_CMD_READ 0x0080
#define FM_CODEC_CMD_VALID 0x0100
#define FM_CODEC_CMD_BUSY 0x0200
#define FM_CODEC_DATA 0x2c
#define FM_IO_CTL 0x52
#define FM_CARD_CTL 0x54
#define FM_INTMASK 0x56
#define FM_INTMASK_PLAY 0x0001
#define FM_INTMASK_REC 0x0002
#define FM_INTMASK_VOL 0x0040
#define FM_INTMASK_MPU 0x0080
#define FM_INTSTATUS 0x5a
#define FM_INTSTATUS_PLAY 0x0100
#define FM_INTSTATUS_REC 0x0200
#define FM_INTSTATUS_VOL 0x4000
#define FM_INTSTATUS_MPU 0x8000
2001-10-08 05:59:54 +00:00
#define FM801_DEFAULT_BUFSZ 4096 /* Other values do not work!!! */
/* debug purposes */
#define DPRINT if(0) printf
/*
static int fm801ch_setup(struct pcm_channel *c);
*/
static u_int32_t fmts[] = {
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 fm801ch_caps = {
5500, 48000,
fmts, 0
};
struct fm801_info;
struct fm801_chinfo {
struct fm801_info *parent;
struct pcm_channel *channel;
struct snd_dbuf *buffer;
u_int32_t spd, dir, fmt; /* speed, direction, format */
u_int32_t shift;
};
struct fm801_info {
int type;
bus_space_tag_t st;
bus_space_handle_t sh;
bus_dma_tag_t parent_dmat;
device_t dev;
int num;
u_int32_t unit;
struct resource *reg, *irq;
int regtype, regid, irqid;
void *ih;
u_int32_t play_flip,
play_nextblk,
play_start,
play_blksize,
play_fmt,
play_shift,
play_size;
u_int32_t rec_flip,
rec_nextblk,
rec_start,
rec_blksize,
rec_fmt,
rec_shift,
rec_size;
unsigned int bufsz;
2001-10-08 05:59:54 +00:00
struct fm801_chinfo pch, rch;
device_t radio;
};
/* Bus Read / Write routines */
static u_int32_t
fm801_rd(struct fm801_info *fm801, int regno, int size)
{
switch(size) {
case 1:
return (bus_space_read_1(fm801->st, fm801->sh, regno));
case 2:
return (bus_space_read_2(fm801->st, fm801->sh, regno));
case 4:
return (bus_space_read_4(fm801->st, fm801->sh, regno));
default:
return 0xffffffff;
}
}
static void
fm801_wr(struct fm801_info *fm801, int regno, u_int32_t data, int size)
{
switch(size) {
case 1:
bus_space_write_1(fm801->st, fm801->sh, regno, data);
break;
case 2:
bus_space_write_2(fm801->st, fm801->sh, regno, data);
break;
case 4:
bus_space_write_4(fm801->st, fm801->sh, regno, data);
break;
}
}
/* -------------------------------------------------------------------- */
/*
* ac97 codec routines
*/
#define TIMO 50
static int
fm801_rdcd(kobj_t obj, void *devinfo, int regno)
{
struct fm801_info *fm801 = (struct fm801_info *)devinfo;
int i;
for (i = 0; i < TIMO && fm801_rd(fm801,FM_CODEC_CMD,2) & FM_CODEC_CMD_BUSY; i++) {
DELAY(10000);
DPRINT("fm801 rdcd: 1 - DELAY\n");
}
if (i >= TIMO) {
printf("fm801 rdcd: codec busy\n");
return 0;
}
fm801_wr(fm801,FM_CODEC_CMD, regno|FM_CODEC_CMD_READ,2);
for (i = 0; i < TIMO && !(fm801_rd(fm801,FM_CODEC_CMD,2) & FM_CODEC_CMD_VALID); i++)
{
DELAY(10000);
DPRINT("fm801 rdcd: 2 - DELAY\n");
}
if (i >= TIMO) {
printf("fm801 rdcd: write codec invalid\n");
return 0;
}
return fm801_rd(fm801,FM_CODEC_DATA,2);
}
static int
fm801_wrcd(kobj_t obj, void *devinfo, int regno, u_int32_t data)
{
struct fm801_info *fm801 = (struct fm801_info *)devinfo;
int i;
DPRINT("fm801_wrcd reg 0x%x val 0x%x\n",regno, data);
/*
if(regno == AC97_REG_RECSEL) return;
*/
/* Poll until codec is ready */
for (i = 0; i < TIMO && fm801_rd(fm801,FM_CODEC_CMD,2) & FM_CODEC_CMD_BUSY; i++) {
DELAY(10000);
DPRINT("fm801 rdcd: 1 - DELAY\n");
}
if (i >= TIMO) {
printf("fm801 wrcd: read codec busy\n");
return -1;
}
fm801_wr(fm801,FM_CODEC_DATA,data, 2);
fm801_wr(fm801,FM_CODEC_CMD, regno,2);
/* wait until codec is ready */
for (i = 0; i < TIMO && fm801_rd(fm801,FM_CODEC_CMD,2) & FM_CODEC_CMD_BUSY; i++) {
DELAY(10000);
DPRINT("fm801 wrcd: 2 - DELAY\n");
}
if (i >= TIMO) {
printf("fm801 wrcd: read codec busy\n");
return -1;
}
DPRINT("fm801 wrcd release reg 0x%x val 0x%x\n",regno, data);
return 0;
}
static kobj_method_t fm801_ac97_methods[] = {
KOBJMETHOD(ac97_read, fm801_rdcd),
KOBJMETHOD(ac97_write, fm801_wrcd),
DEVMETHOD_END
};
AC97_DECLARE(fm801_ac97);
/* -------------------------------------------------------------------- */
/*
* The interrupt handler
*/
static void
fm801_intr(void *p)
{
struct fm801_info *fm801 = (struct fm801_info *)p;
u_int32_t intsrc = fm801_rd(fm801, FM_INTSTATUS, 2);
DPRINT("\nfm801_intr intsrc 0x%x ", intsrc);
if(intsrc & FM_INTSTATUS_PLAY) {
fm801->play_flip++;
if(fm801->play_flip & 1) {
fm801_wr(fm801, FM_PLAY_DMABUF1, fm801->play_start,4);
} else
fm801_wr(fm801, FM_PLAY_DMABUF2, fm801->play_nextblk,4);
chn_intr(fm801->pch.channel);
}
if(intsrc & FM_INTSTATUS_REC) {
fm801->rec_flip++;
if(fm801->rec_flip & 1) {
fm801_wr(fm801, FM_REC_DMABUF1, fm801->rec_start,4);
} else
fm801_wr(fm801, FM_REC_DMABUF2, fm801->rec_nextblk,4);
chn_intr(fm801->rch.channel);
}
if ( intsrc & FM_INTSTATUS_MPU ) {
/* This is a TODOish thing... */
fm801_wr(fm801, FM_INTSTATUS, intsrc & FM_INTSTATUS_MPU,2);
}
if ( intsrc & FM_INTSTATUS_VOL ) {
/* This is a TODOish thing... */
fm801_wr(fm801, FM_INTSTATUS, intsrc & FM_INTSTATUS_VOL,2);
}
DPRINT("fm801_intr clear\n\n");
fm801_wr(fm801, FM_INTSTATUS, intsrc & (FM_INTSTATUS_PLAY | FM_INTSTATUS_REC), 2);
}
/* -------------------------------------------------------------------- */
/* channel interface */
static void *
fm801ch_init(kobj_t obj, void *devinfo, struct snd_dbuf *b, struct pcm_channel *c, int dir)
{
struct fm801_info *fm801 = (struct fm801_info *)devinfo;
struct fm801_chinfo *ch = (dir == PCMDIR_PLAY)? &fm801->pch : &fm801->rch;
DPRINT("fm801ch_init, direction = %d\n", dir);
ch->parent = fm801;
ch->channel = c;
ch->buffer = b;
ch->dir = dir;
if (sndbuf_alloc(ch->buffer, fm801->parent_dmat, 0, fm801->bufsz) != 0)
return NULL;
return (void *)ch;
}
static int
fm801ch_setformat(kobj_t obj, void *data, u_int32_t format)
{
struct fm801_chinfo *ch = data;
struct fm801_info *fm801 = ch->parent;
DPRINT("fm801ch_setformat 0x%x : %s, %s, %s, %s\n", format,
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
(AFMT_CHANNEL(format) > 1)?"stereo":"mono",
(format & AFMT_16BIT) ? "16bit":"8bit",
(format & AFMT_SIGNED)? "signed":"unsigned",
(format & AFMT_BIGENDIAN)?"bigendiah":"littleendian" );
if(ch->dir == PCMDIR_PLAY) {
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
fm801->play_fmt =
(AFMT_CHANNEL(format) > 1)? FM_PLAY_STEREO : 0;
fm801->play_fmt |= (format & AFMT_16BIT) ? FM_PLAY_16BIT : 0;
return 0;
}
if(ch->dir == PCMDIR_REC ) {
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
fm801->rec_fmt = (AFMT_CHANNEL(format) > 1)? FM_REC_STEREO:0;
fm801->rec_fmt |= (format & AFMT_16BIT) ? FM_PLAY_16BIT : 0;
return 0;
}
return 0;
}
struct {
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
u_int32_t limit;
u_int32_t rate;
} fm801_rates[11] = {
{ 6600, 5500 },
{ 8750, 8000 },
{ 10250, 9600 },
{ 13200, 11025 },
{ 17500, 16000 },
{ 20500, 19200 },
{ 26500, 22050 },
{ 35000, 32000 },
{ 41000, 38400 },
{ 46000, 44100 },
{ 48000, 48000 },
/* anything above -> 48000 */
};
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
fm801ch_setspeed(kobj_t obj, void *data, u_int32_t speed)
{
struct fm801_chinfo *ch = data;
struct fm801_info *fm801 = ch->parent;
int i;
for (i = 0; i < 10 && fm801_rates[i].limit <= speed; i++) ;
if(ch->dir == PCMDIR_PLAY) {
fm801->pch.spd = fm801_rates[i].rate;
fm801->play_shift = (i<<8);
fm801->play_shift &= FM_PLAY_RATE_MASK;
}
if(ch->dir == PCMDIR_REC ) {
fm801->rch.spd = fm801_rates[i].rate;
fm801->rec_shift = (i<<8);
fm801->rec_shift &= FM_REC_RATE_MASK;
}
ch->spd = fm801_rates[i].rate;
return fm801_rates[i].rate;
}
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
fm801ch_setblocksize(kobj_t obj, void *data, u_int32_t blocksize)
{
struct fm801_chinfo *ch = data;
struct fm801_info *fm801 = ch->parent;
/*
* Don't mind for play_flip, set the blocksize to the
* desired values in any case - otherwise sound playback
* breaks here.
*/
if(ch->dir == PCMDIR_PLAY)
fm801->play_blksize = blocksize;
if(ch->dir == PCMDIR_REC)
fm801->rec_blksize = blocksize;
DPRINT("fm801ch_setblocksize %d (dir %d)\n",blocksize, ch->dir);
return blocksize;
}
static int
fm801ch_trigger(kobj_t obj, void *data, int go)
{
struct fm801_chinfo *ch = data;
struct fm801_info *fm801 = ch->parent;
u_int32_t baseaddr = sndbuf_getbufaddr(ch->buffer);
u_int32_t k1;
DPRINT("fm801ch_trigger go %d , ", go);
if (!PCMTRIG_COMMON(go))
return 0;
if (ch->dir == PCMDIR_PLAY) {
if (go == PCMTRIG_START) {
fm801->play_start = baseaddr;
fm801->play_nextblk = fm801->play_start + fm801->play_blksize;
fm801->play_flip = 0;
fm801_wr(fm801, FM_PLAY_DMALEN, fm801->play_blksize - 1, 2);
fm801_wr(fm801, FM_PLAY_DMABUF1,fm801->play_start,4);
fm801_wr(fm801, FM_PLAY_DMABUF2,fm801->play_nextblk,4);
fm801_wr(fm801, FM_PLAY_CTL,
FM_PLAY_START | FM_PLAY_STOPNOW | fm801->play_fmt | fm801->play_shift,
2 );
} else {
fm801->play_flip = 0;
k1 = fm801_rd(fm801, FM_PLAY_CTL,2);
fm801_wr(fm801, FM_PLAY_CTL,
(k1 & ~(FM_PLAY_STOPNOW | FM_PLAY_START)) |
FM_PLAY_BUF1_LAST | FM_PLAY_BUF2_LAST, 2 );
}
} else if(ch->dir == PCMDIR_REC) {
if (go == PCMTRIG_START) {
fm801->rec_start = baseaddr;
fm801->rec_nextblk = fm801->rec_start + fm801->rec_blksize;
fm801->rec_flip = 0;
fm801_wr(fm801, FM_REC_DMALEN, fm801->rec_blksize - 1, 2);
fm801_wr(fm801, FM_REC_DMABUF1,fm801->rec_start,4);
fm801_wr(fm801, FM_REC_DMABUF2,fm801->rec_nextblk,4);
fm801_wr(fm801, FM_REC_CTL,
FM_REC_START | FM_REC_STOPNOW | fm801->rec_fmt | fm801->rec_shift,
2 );
} else {
fm801->rec_flip = 0;
k1 = fm801_rd(fm801, FM_REC_CTL,2);
fm801_wr(fm801, FM_REC_CTL,
(k1 & ~(FM_REC_STOPNOW | FM_REC_START)) |
FM_REC_BUF1_LAST | FM_REC_BUF2_LAST, 2);
}
}
return 0;
}
/* Almost ALSA copy */
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
fm801ch_getptr(kobj_t obj, void *data)
{
struct fm801_chinfo *ch = data;
struct fm801_info *fm801 = ch->parent;
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
u_int32_t result = 0;
if (ch->dir == PCMDIR_PLAY) {
result = fm801_rd(fm801,
(fm801->play_flip&1) ?
FM_PLAY_DMABUF2:FM_PLAY_DMABUF1, 4) - fm801->play_start;
}
if (ch->dir == PCMDIR_REC) {
result = fm801_rd(fm801,
(fm801->rec_flip&1) ?
FM_REC_DMABUF2:FM_REC_DMABUF1, 4) - fm801->rec_start;
}
return result;
}
static struct pcmchan_caps *
fm801ch_getcaps(kobj_t obj, void *data)
{
return &fm801ch_caps;
}
static kobj_method_t fm801ch_methods[] = {
KOBJMETHOD(channel_init, fm801ch_init),
KOBJMETHOD(channel_setformat, fm801ch_setformat),
KOBJMETHOD(channel_setspeed, fm801ch_setspeed),
KOBJMETHOD(channel_setblocksize, fm801ch_setblocksize),
KOBJMETHOD(channel_trigger, fm801ch_trigger),
KOBJMETHOD(channel_getptr, fm801ch_getptr),
KOBJMETHOD(channel_getcaps, fm801ch_getcaps),
DEVMETHOD_END
};
CHANNEL_DECLARE(fm801ch);
/* -------------------------------------------------------------------- */
/*
* Init routine is taken from an original NetBSD driver
*/
static int
fm801_init(struct fm801_info *fm801)
{
u_int32_t k1;
/* reset codec */
fm801_wr(fm801, FM_CODEC_CTL, 0x0020,2);
DELAY(100000);
fm801_wr(fm801, FM_CODEC_CTL, 0x0000,2);
DELAY(100000);
fm801_wr(fm801, FM_PCM_VOLUME, 0x0808,2);
fm801_wr(fm801, FM_FM_VOLUME, 0x0808,2);
fm801_wr(fm801, FM_I2S_VOLUME, 0x0808,2);
fm801_wr(fm801, 0x40,0x107f,2); /* enable legacy audio */
fm801_wr((void *)fm801, FM_RECORD_SOURCE, 0x0000,2);
/* Unmask playback, record and mpu interrupts, mask the rest */
k1 = fm801_rd((void *)fm801, FM_INTMASK,2);
fm801_wr(fm801, FM_INTMASK,
(k1 & ~(FM_INTMASK_PLAY | FM_INTMASK_REC | FM_INTMASK_MPU)) |
FM_INTMASK_VOL,2);
fm801_wr(fm801, FM_INTSTATUS,
FM_INTSTATUS_PLAY | FM_INTSTATUS_REC | FM_INTSTATUS_MPU |
FM_INTSTATUS_VOL,2);
DPRINT("FM801 init Ok\n");
return 0;
}
static int
fm801_pci_attach(device_t dev)
{
struct ac97_info *codec = NULL;
struct fm801_info *fm801;
int i;
int mapped = 0;
char status[SND_STATUSLEN];
fm801 = malloc(sizeof(*fm801), M_DEVBUF, M_WAITOK | M_ZERO);
fm801->type = pci_get_devid(dev);
pci_enable_busmaster(dev);
for (i = 0; (mapped == 0) && (i < PCI_MAXMAPS_0); i++) {
fm801->regid = PCIR_BAR(i);
fm801->regtype = SYS_RES_MEMORY;
fm801->reg = bus_alloc_resource_any(dev, fm801->regtype,
&fm801->regid, RF_ACTIVE);
if(!fm801->reg)
{
fm801->regtype = SYS_RES_IOPORT;
fm801->reg = bus_alloc_resource_any(dev,
fm801->regtype,
&fm801->regid,
RF_ACTIVE);
}
if(fm801->reg) {
fm801->st = rman_get_bustag(fm801->reg);
fm801->sh = rman_get_bushandle(fm801->reg);
mapped++;
}
}
if (mapped == 0) {
device_printf(dev, "unable to map register space\n");
goto oops;
}
2001-10-08 05:59:54 +00:00
fm801->bufsz = pcm_getbuffersize(dev, 4096, FM801_DEFAULT_BUFSZ, 65536);
fm801_init(fm801);
codec = AC97_CREATE(dev, fm801, fm801_ac97);
if (codec == NULL) goto oops;
if (mixer_init(dev, ac97_getmixerclass(), codec) == -1) goto oops;
fm801->irqid = 0;
fm801->irq = bus_alloc_resource_any(dev, SYS_RES_IRQ, &fm801->irqid,
RF_ACTIVE | RF_SHAREABLE);
if (!fm801->irq ||
snd_setup_intr(dev, fm801->irq, 0, fm801_intr, fm801, &fm801->ih)) {
device_printf(dev, "unable to map interrupt\n");
goto oops;
}
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,
2001-10-08 05:59:54 +00:00
/*maxsize*/fm801->bufsz, /*nsegments*/1, /*maxsegz*/0x3ffff,
/*flags*/0, /*lockfunc*/NULL, /*lockarg*/NULL,
&fm801->parent_dmat) != 0) {
device_printf(dev, "unable to create dma tag\n");
goto oops;
}
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 %s 0x%jx irq %jd %s",
(fm801->regtype == SYS_RES_IOPORT)? "io" : "memory",
rman_get_start(fm801->reg), rman_get_start(fm801->irq),PCM_KLDSTRING(snd_fm801));
#define FM801_MAXPLAYCH 1
if (pcm_register(dev, fm801, FM801_MAXPLAYCH, 1)) goto oops;
pcm_addchan(dev, PCMDIR_PLAY, &fm801ch_class, fm801);
pcm_addchan(dev, PCMDIR_REC, &fm801ch_class, fm801);
pcm_setstatus(dev, status);
fm801->radio = device_add_child(dev, "radio", -1);
bus_generic_attach(dev);
return 0;
oops:
if (codec) ac97_destroy(codec);
if (fm801->reg) bus_release_resource(dev, fm801->regtype, fm801->regid, fm801->reg);
if (fm801->ih) bus_teardown_intr(dev, fm801->irq, fm801->ih);
if (fm801->irq) bus_release_resource(dev, SYS_RES_IRQ, fm801->irqid, fm801->irq);
if (fm801->parent_dmat) bus_dma_tag_destroy(fm801->parent_dmat);
free(fm801, M_DEVBUF);
return ENXIO;
}
static int
fm801_pci_detach(device_t dev)
{
int r;
struct fm801_info *fm801;
DPRINT("Forte Media FM801 detach\n");
fm801 = pcm_getdevinfo(dev);
r = bus_generic_detach(dev);
if (r)
return r;
if (fm801->radio != NULL) {
r = device_delete_child(dev, fm801->radio);
if (r)
return r;
fm801->radio = NULL;
}
r = pcm_unregister(dev);
if (r)
return r;
bus_release_resource(dev, fm801->regtype, fm801->regid, fm801->reg);
bus_teardown_intr(dev, fm801->irq, fm801->ih);
bus_release_resource(dev, SYS_RES_IRQ, fm801->irqid, fm801->irq);
bus_dma_tag_destroy(fm801->parent_dmat);
free(fm801, M_DEVBUF);
return 0;
}
static int
fm801_pci_probe( device_t dev )
{
int id;
if ((id = pci_get_devid(dev)) == PCI_DEVICE_FORTEMEDIA1 ) {
device_set_desc(dev, "Forte Media FM801 Audio Controller");
return BUS_PROBE_DEFAULT;
}
/*
if ((id = pci_get_devid(dev)) == PCI_DEVICE_FORTEMEDIA2 ) {
device_set_desc(dev, "Forte Media FM801 Joystick (Not Supported)");
return ENXIO;
}
*/
return ENXIO;
}
static struct resource *
fm801_alloc_resource(device_t bus, device_t child, int type, int *rid,
rman_res_t start, rman_res_t end, rman_res_t count,
u_int flags)
{
struct fm801_info *fm801;
fm801 = pcm_getdevinfo(bus);
if (type == SYS_RES_IOPORT && *rid == PCIR_BAR(0))
return (fm801->reg);
return (NULL);
}
static int
fm801_release_resource(device_t bus, device_t child, int type, int rid,
struct resource *r)
{
return (0);
}
static device_method_t fm801_methods[] = {
/* Device interface */
DEVMETHOD(device_probe, fm801_pci_probe),
DEVMETHOD(device_attach, fm801_pci_attach),
DEVMETHOD(device_detach, fm801_pci_detach),
DEVMETHOD(device_shutdown, bus_generic_shutdown),
DEVMETHOD(device_suspend, bus_generic_suspend),
DEVMETHOD(device_resume, bus_generic_resume),
/* Bus interface */
DEVMETHOD(bus_alloc_resource, fm801_alloc_resource),
DEVMETHOD(bus_release_resource, fm801_release_resource),
DEVMETHOD(bus_activate_resource, bus_generic_activate_resource),
DEVMETHOD(bus_deactivate_resource, bus_generic_deactivate_resource),
DEVMETHOD_END
};
static driver_t fm801_driver = {
"pcm",
fm801_methods,
PCM_SOFTC_SIZE,
};
2001-07-01 18:59:15 +00:00
DRIVER_MODULE(snd_fm801, pci, fm801_driver, pcm_devclass, 0, 0);
MODULE_DEPEND(snd_fm801, sound, SOUND_MINVER, SOUND_PREFVER, SOUND_MAXVER);
2001-07-01 18:59:15 +00:00
MODULE_VERSION(snd_fm801, 1);