freebsd-dev/sys/dev/sound/macio/tumbler.c

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/*-
* SPDX-License-Identifier: BSD-2-Clause-FreeBSD AND BSD-3-Clause
*
* Copyright 2008 by Marco Trillo. 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 ``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 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.
*
* $FreeBSD$
*/
/*-
* Copyright (c) 2002, 2003 Tsubai Masanari. 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.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``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 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.
*
* NetBSD: tumbler.c,v 1.28 2008/05/16 03:49:54 macallan Exp
* Id: tumbler.c,v 1.11 2002/10/31 17:42:13 tsubai Exp
*/
/*
* Apple I2S audio controller.
*/
#include <sys/param.h>
#include <sys/systm.h>
#include <sys/kernel.h>
#include <sys/module.h>
#include <sys/bus.h>
#include <sys/malloc.h>
#include <sys/lock.h>
#include <sys/mutex.h>
#include <machine/dbdma.h>
#include <machine/intr_machdep.h>
#include <machine/resource.h>
#include <machine/bus.h>
#include <machine/pio.h>
#include <sys/rman.h>
#include <dev/iicbus/iicbus.h>
#include <dev/iicbus/iiconf.h>
#include <dev/ofw/ofw_bus.h>
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>
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
#include "mixer_if.h"
extern kobj_class_t i2s_mixer_class;
extern device_t i2s_mixer;
struct tumbler_softc
{
device_t sc_dev;
uint32_t sc_addr;
};
static int tumbler_probe(device_t);
static int tumbler_attach(device_t);
static int tumbler_init(struct snd_mixer *m);
static int tumbler_uninit(struct snd_mixer *m);
static int tumbler_reinit(struct snd_mixer *m);
static int tumbler_set(struct snd_mixer *m, unsigned dev, unsigned left,
unsigned right);
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 tumbler_setrecsrc(struct snd_mixer *m, u_int32_t src);
static device_method_t tumbler_methods[] = {
/* Device interface. */
DEVMETHOD(device_probe, tumbler_probe),
DEVMETHOD(device_attach, tumbler_attach),
{ 0, 0 }
};
static driver_t tumbler_driver = {
"tumbler",
tumbler_methods,
sizeof(struct tumbler_softc)
};
static devclass_t tumbler_devclass;
DRIVER_MODULE(tumbler, iicbus, tumbler_driver, tumbler_devclass, 0, 0);
MODULE_VERSION(tumbler, 1);
MODULE_DEPEND(tumbler, iicbus, 1, 1, 1);
static kobj_method_t tumbler_mixer_methods[] = {
KOBJMETHOD(mixer_init, tumbler_init),
KOBJMETHOD(mixer_uninit, tumbler_uninit),
KOBJMETHOD(mixer_reinit, tumbler_reinit),
KOBJMETHOD(mixer_set, tumbler_set),
KOBJMETHOD(mixer_setrecsrc, tumbler_setrecsrc),
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
KOBJMETHOD_END
};
MIXER_DECLARE(tumbler_mixer);
#define TUMBLER_IICADDR 0x68 /* Tumbler I2C slave address */
/* Tumbler (Texas Instruments TAS3001) registers. */
#define TUMBLER_MCR 0x01 /* Main control register (1byte) */
#define TUMBLER_DRC 0x02 /* Dynamic Range Compression (2bytes) */
#define TUMBLER_VOLUME 0x04 /* Volume (6bytes) */
#define TUMBLER_TREBLE 0x05 /* Treble control (1byte) */
#define TUMBLER_BASS 0x06 /* Bass control (1byte) */
#define TUMBLER_MIXER1 0x07 /* Mixer1 (3bytes) */
#define TUMBLER_MIXER2 0x08 /* Mixer2 (3bytes) */
#define TUMBLER_LB0 0x0a /* Left biquad 0 (15bytes) */
#define TUMBLER_LB1 0x0b /* Left biquad 1 (15bytes) */
#define TUMBLER_LB2 0x0c /* Left biquad 2 (15bytes) */
#define TUMBLER_LB3 0x0d /* Left biquad 3 (15bytes) */
#define TUMBLER_LB4 0x0e /* Left biquad 4 (15bytes) */
#define TUMBLER_LB5 0x0f /* Left biquad 5 (15bytes) */
#define TUMBLER_RB0 0x13 /* Right biquad 0 (15bytes) */
#define TUMBLER_RB1 0x14 /* Right biquad 1 (15bytes) */
#define TUMBLER_RB2 0x15 /* Right biquad 2 (15bytes) */
#define TUMBLER_RB3 0x16 /* Right biquad 3 (15bytes) */
#define TUMBLER_RB4 0x17 /* Right biquad 4 (15bytes) */
#define TUMBLER_RB5 0x18 /* Right biquad 5 (15bytes) */
#define TUMBLER_MCR_FL 0x80 /* Fast load */
#define TUMBLER_MCR_SC 0x40 /* SCLK frequency */
#define TUMBLER_MCR_SC_32 0x00 /* 32fs */
#define TUMBLER_MCR_SC_64 0x40 /* 64fs */
#define TUMBLER_MCR_SM 0x30 /* Output serial port mode */
#define TUMBLER_MCR_SM_L 0x00 /* Left justified */
#define TUMBLER_MCR_SM_R 0x10 /* Right justified */
#define TUMBLER_MCR_SM_I2S 0x20 /* I2S */
#define TUMBLER_MCR_ISM 0x0C /* Input serial mode */
#define TUMBLER_MCR_ISM_L 0x00
#define TUMBLER_MCR_ISM_R 0x04
#define TUMBLER_MCR_ISM_I2S 0x08
#define TUMBLER_MCR_W 0x03 /* Serial port word length */
#define TUMBLER_MCR_W_16 0x00 /* 16 bit */
#define TUMBLER_MCR_W_18 0x01 /* 18 bit */
#define TUMBLER_MCR_W_20 0x02 /* 20 bit */
#define TUMBLER_DRC_COMP_31 0xc0 /* 3:1 compression */
#define TUMBLER_DRC_ENABLE 0x01 /* enable DRC */
#define TUMBLER_DRC_DEFL_TH 0xa0 /* default compression threshold */
/*
* Tumbler codec.
*/
struct tumbler_reg {
u_char MCR[1];
u_char DRC[2];
u_char VOLUME[6];
u_char TREBLE[1];
u_char BASS[1];
u_char MIXER1[3];
u_char MIXER2[3];
u_char LB0[15];
u_char LB1[15];
u_char LB2[15];
u_char LB3[15];
u_char LB4[15];
u_char LB5[15];
u_char RB0[15];
u_char RB1[15];
u_char RB2[15];
u_char RB3[15];
u_char RB4[15];
u_char RB5[15];
};
const struct tumbler_reg tumbler_initdata = {
{ TUMBLER_MCR_SC_64 | TUMBLER_MCR_SM_I2S |
TUMBLER_MCR_ISM_I2S | TUMBLER_MCR_W_16 }, /* MCR */
{ TUMBLER_DRC_COMP_31, TUMBLER_DRC_DEFL_TH }, /* DRC */
{ 0, 0, 0, 0, 0, 0 }, /* VOLUME */
{ 0x72 }, /* TREBLE */
{ 0x3e }, /* BASS */
{ 0x10, 0x00, 0x00 }, /* MIXER1 */
{ 0x00, 0x00, 0x00 }, /* MIXER2 */
{ 0x10, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, /* BIQUAD */
{ 0x10, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, /* BIQUAD */
{ 0x10, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, /* BIQUAD */
{ 0x10, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, /* BIQUAD */
{ 0x10, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, /* BIQUAD */
{ 0x10, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, /* BIQUAD */
{ 0x10, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, /* BIQUAD */
{ 0x10, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, /* BIQUAD */
{ 0x10, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, /* BIQUAD */
{ 0x10, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, /* BIQUAD */
{ 0x10, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, /* BIQUAD */
{ 0x10, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 } /* BIQUAD */
};
const char tumbler_regsize[] = {
0, /* 0x00 */
sizeof tumbler_initdata.MCR, /* 0x01 */
sizeof tumbler_initdata.DRC, /* 0x02 */
0, /* 0x03 */
sizeof tumbler_initdata.VOLUME, /* 0x04 */
sizeof tumbler_initdata.TREBLE, /* 0x05 */
sizeof tumbler_initdata.BASS, /* 0x06 */
sizeof tumbler_initdata.MIXER1, /* 0x07 */
sizeof tumbler_initdata.MIXER2, /* 0x08 */
0, /* 0x09 */
sizeof tumbler_initdata.LB0, /* 0x0a */
sizeof tumbler_initdata.LB1, /* 0x0b */
sizeof tumbler_initdata.LB2, /* 0x0c */
sizeof tumbler_initdata.LB3, /* 0x0d */
sizeof tumbler_initdata.LB4, /* 0x0e */
sizeof tumbler_initdata.LB5, /* 0x0f */
0, /* 0x10 */
0, /* 0x11 */
0, /* 0x12 */
sizeof tumbler_initdata.RB0, /* 0x13 */
sizeof tumbler_initdata.RB1, /* 0x14 */
sizeof tumbler_initdata.RB2, /* 0x15 */
sizeof tumbler_initdata.RB3, /* 0x16 */
sizeof tumbler_initdata.RB4, /* 0x17 */
sizeof tumbler_initdata.RB5 /* 0x18 */
};
/* dB = 20 * log (x) table. */
static u_int tumbler_volume_table[100] = {
0x00000148, 0x0000015C, 0x00000171, 0x00000186, // -46.0, -45.5, -45.0, -44.5,
0x0000019E, 0x000001B6, 0x000001D0, 0x000001EB, // -44.0, -43.5, -43.0, -42.5,
0x00000209, 0x00000227, 0x00000248, 0x0000026B, // -42.0, -41.5, -41.0, -40.5,
0x0000028F, 0x000002B6, 0x000002DF, 0x0000030B, // -40.0, -39.5, -39.0, -38.5,
0x00000339, 0x0000036A, 0x0000039E, 0x000003D5, // -38.0, -37.5, -37.0, -36.5,
0x0000040F, 0x0000044C, 0x0000048D, 0x000004D2, // -36.0, -35.5, -35.0, -34.5,
0x0000051C, 0x00000569, 0x000005BB, 0x00000612, // -34.0, -33.5, -33.0, -32.5,
0x0000066E, 0x000006D0, 0x00000737, 0x000007A5, // -32.0, -31.5, -31.0, -30.5,
0x00000818, 0x00000893, 0x00000915, 0x0000099F, // -30.0, -29.5, -29.0, -28.5,
0x00000A31, 0x00000ACC, 0x00000B6F, 0x00000C1D, // -28.0, -27.5, -27.0, -26.5,
0x00000CD5, 0x00000D97, 0x00000E65, 0x00000F40, // -26.0, -25.5, -25.0, -24.5,
0x00001027, 0x0000111C, 0x00001220, 0x00001333, // -24.0, -23.5, -23.0, -22.5,
0x00001456, 0x0000158A, 0x000016D1, 0x0000182B, // -22.0, -21.5, -21.0, -20.5,
0x0000199A, 0x00001B1E, 0x00001CB9, 0x00001E6D, // -20.0, -19.5, -19.0, -18.5,
0x0000203A, 0x00002223, 0x00002429, 0x0000264E, // -18.0, -17.5, -17.0, -16.5,
0x00002893, 0x00002AFA, 0x00002D86, 0x00003039, // -16.0, -15.5, -15.0, -14.5,
0x00003314, 0x0000361B, 0x00003950, 0x00003CB5, // -14.0, -13.5, -13.0, -12.5,
0x0000404E, 0x0000441D, 0x00004827, 0x00004C6D, // -12.0, -11.5, -11.0, -10.5,
0x000050F4, 0x000055C0, 0x00005AD5, 0x00006037, // -10.0, -9.5, -9.0, -8.5,
0x000065EA, 0x00006BF4, 0x0000725A, 0x00007920, // -8.0, -7.5, -7.0, -6.5,
0x0000804E, 0x000087EF, 0x00008FF6, 0x0000987D, // -6.0, -5.5, -5.0, -4.5,
0x0000A186, 0x0000AB19, 0x0000B53C, 0x0000BFF9, // -4.0, -3.5, -3.0, -2.5,
0x0000CB59, 0x0000D766, 0x0000E429, 0x0000F1AE, // -2.0, -1.5, -1.0, -0.5,
0x00010000, 0x00010F2B, 0x00011F3D, 0x00013042, // 0.0, +0.5, +1.0, +1.5,
0x00014249, 0x00015562, 0x0001699C, 0x00017F09 // 2.0, +2.5, +3.0, +3.5,
};
static int
tumbler_write(struct tumbler_softc *sc, uint8_t reg, const void *data)
{
u_int size;
uint8_t buf[16];
struct iic_msg msg[] = {
{ sc->sc_addr, IIC_M_WR, 0, buf }
};
KASSERT(reg < sizeof(tumbler_regsize), ("bad reg"));
size = tumbler_regsize[reg];
msg[0].len = size + 1;
buf[0] = reg;
memcpy(&buf[1], data, size);
iicbus_transfer(sc->sc_dev, msg, 1);
return (0);
}
static int
tumbler_probe(device_t dev)
{
const char *name;
name = ofw_bus_get_name(dev);
if (name == NULL)
return (ENXIO);
if (strcmp(name, "deq") == 0 && iicbus_get_addr(dev) ==
TUMBLER_IICADDR) {
device_set_desc(dev, "Texas Instruments TAS3001 Audio Codec");
return (0);
}
return (ENXIO);
}
static int
tumbler_attach(device_t dev)
{
struct tumbler_softc *sc;
sc = device_get_softc(dev);
sc->sc_dev = dev;
sc->sc_addr = iicbus_get_addr(dev);
i2s_mixer_class = &tumbler_mixer_class;
i2s_mixer = dev;
return (0);
}
static int
tumbler_init(struct snd_mixer *m)
{
struct tumbler_softc *sc;
u_int x = 0;
sc = device_get_softc(mix_getdevinfo(m));
tumbler_write(sc, TUMBLER_LB0, tumbler_initdata.LB0);
tumbler_write(sc, TUMBLER_LB1, tumbler_initdata.LB1);
tumbler_write(sc, TUMBLER_LB2, tumbler_initdata.LB2);
tumbler_write(sc, TUMBLER_LB3, tumbler_initdata.LB3);
tumbler_write(sc, TUMBLER_LB4, tumbler_initdata.LB4);
tumbler_write(sc, TUMBLER_LB5, tumbler_initdata.LB5);
tumbler_write(sc, TUMBLER_RB0, tumbler_initdata.RB0);
tumbler_write(sc, TUMBLER_RB1, tumbler_initdata.RB1);
tumbler_write(sc, TUMBLER_RB1, tumbler_initdata.RB1);
tumbler_write(sc, TUMBLER_RB2, tumbler_initdata.RB2);
tumbler_write(sc, TUMBLER_RB3, tumbler_initdata.RB3);
tumbler_write(sc, TUMBLER_RB4, tumbler_initdata.RB4);
tumbler_write(sc, TUMBLER_RB5, tumbler_initdata.RB5);
tumbler_write(sc, TUMBLER_MCR, tumbler_initdata.MCR);
tumbler_write(sc, TUMBLER_DRC, tumbler_initdata.DRC);
tumbler_write(sc, TUMBLER_VOLUME, tumbler_initdata.VOLUME);
tumbler_write(sc, TUMBLER_TREBLE, tumbler_initdata.TREBLE);
tumbler_write(sc, TUMBLER_BASS, tumbler_initdata.BASS);
tumbler_write(sc, TUMBLER_MIXER1, tumbler_initdata.MIXER1);
tumbler_write(sc, TUMBLER_MIXER2, tumbler_initdata.MIXER2);
x |= SOUND_MASK_VOLUME;
mix_setdevs(m, x);
return (0);
}
static int
tumbler_uninit(struct snd_mixer *m)
{
return (0);
}
static int
tumbler_reinit(struct snd_mixer *m)
{
return (0);
}
static int
tumbler_set(struct snd_mixer *m, unsigned dev, unsigned left, unsigned right)
{
struct tumbler_softc *sc;
struct mtx *mixer_lock;
int locked;
u_int l, r;
u_char reg[6];
sc = device_get_softc(mix_getdevinfo(m));
mixer_lock = mixer_get_lock(m);
locked = mtx_owned(mixer_lock);
switch (dev) {
case SOUND_MIXER_VOLUME:
if (left > 100 || right > 100)
return (0);
l = (left == 0 ? 0 : tumbler_volume_table[left - 1]);
r = (right == 0 ? 0 : tumbler_volume_table[right - 1]);
reg[0] = (l & 0xff0000) >> 16;
reg[1] = (l & 0x00ff00) >> 8;
reg[2] = l & 0x0000ff;
reg[3] = (r & 0xff0000) >> 16;
reg[4] = (r & 0x00ff00) >> 8;
reg[5] = r & 0x0000ff;
/*
* We need to unlock the mixer lock because iicbus_transfer()
* may sleep. The mixer lock itself is unnecessary here
* because it is meant to serialize hardware access, which
* is taken care of by the I2C layer, so this is safe.
*/
if (locked)
mtx_unlock(mixer_lock);
tumbler_write(sc, TUMBLER_VOLUME, reg);
if (locked)
mtx_lock(mixer_lock);
return (left | (right << 8));
}
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
tumbler_setrecsrc(struct snd_mixer *m, u_int32_t src)
{
return (0);
}