NetBSD/sys/dev/pci/esm.c

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/* $NetBSD: esm.c,v 1.41 2006/11/16 01:33:08 christos Exp $ */
/*-
* Copyright (c) 2002, 2003 Matt Fredette
* All rights reserved.
*
* Copyright (c) 2000, 2001 Rene Hexel <rh@NetBSD.org>
* All rights reserved.
*
* Copyright (c) 2000 Taku YAMAMOTO <taku@cent.saitama-u.ac.jp>
* 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.
*
* Taku Id: maestro.c,v 1.12 2000/09/06 03:32:34 taku Exp
* FreeBSD: /c/ncvs/src/sys/dev/sound/pci/maestro.c,v 1.4 2000/12/18 01:36:35 cg Exp
*/
/*
* TODO:
* - hardware volume support
* - fix 16-bit stereo recording, add 8-bit recording
* - MIDI support
* - joystick support
*
*
* Credits:
*
* This code is based on the FreeBSD driver written by Taku YAMAMOTO
*
*
* Original credits from the FreeBSD driver:
*
* Part of this code (especially in many magic numbers) was heavily inspired
* by the Linux driver originally written by
* Alan Cox <alan.cox@linux.org>, modified heavily by
* Zach Brown <zab@zabbo.net>.
*
* busdma()-ize and buffer size reduction were suggested by
* Cameron Grant <gandalf@vilnya.demon.co.uk>.
* Also he showed me the way to use busdma() suite.
*
* Internal speaker problems on NEC VersaPro's and Dell Inspiron 7500
* were looked at by
* Munehiro Matsuda <haro@tk.kubota.co.jp>,
* who brought patches based on the Linux driver with some simplification.
*/
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#include <sys/cdefs.h>
__KERNEL_RCSID(0, "$NetBSD: esm.c,v 1.41 2006/11/16 01:33:08 christos Exp $");
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#include <sys/param.h>
#include <sys/systm.h>
#include <sys/kernel.h>
#include <sys/malloc.h>
#include <sys/device.h>
#include <machine/bus.h>
#include <sys/audioio.h>
#include <dev/audio_if.h>
#include <dev/mulaw.h>
#include <dev/auconv.h>
#include <dev/ic/ac97var.h>
#include <dev/ic/ac97reg.h>
#include <dev/pci/pcidevs.h>
#include <dev/pci/pcivar.h>
#include <dev/pci/esmreg.h>
#include <dev/pci/esmvar.h>
#define PCI_CBIO 0x10 /* Configuration Base I/O Address */
/* Debug */
#ifdef AUDIO_DEBUG
#define DPRINTF(l,x) do { if (esm_debug & (l)) printf x; } while(0)
#define DUMPREG(x) do { if (esm_debug & ESM_DEBUG_REG) \
esm_dump_regs(x); } while(0)
int esm_debug = 0xfffc;
#define ESM_DEBUG_CODECIO 0x0001
#define ESM_DEBUG_IRQ 0x0002
#define ESM_DEBUG_DMA 0x0004
#define ESM_DEBUG_TIMER 0x0008
#define ESM_DEBUG_REG 0x0010
#define ESM_DEBUG_PARAM 0x0020
#define ESM_DEBUG_APU 0x0040
#define ESM_DEBUG_CODEC 0x0080
#define ESM_DEBUG_PCI 0x0100
#define ESM_DEBUG_RESUME 0x0200
#else
#define DPRINTF(x,y) /* nothing */
#define DUMPREG(x) /* nothing */
#endif
#ifdef DIAGNOSTIC
#define RANGE(n, l, h) if ((n) < (l) || (n) >= (h)) \
printf (#n "=%d out of range (%d, %d) in " \
__FILE__ ", line %d\n", (n), (l), (h), __LINE__)
#else
#define RANGE(x,y,z) /* nothing */
#endif
#define inline inline
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static inline void ringbus_setdest(struct esm_softc *, int, int);
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static inline uint16_t wp_rdreg(struct esm_softc *, uint16_t);
static inline void wp_wrreg(struct esm_softc *, uint16_t, uint16_t);
static inline uint16_t wp_rdapu(struct esm_softc *, int, uint16_t);
static inline void wp_wrapu(struct esm_softc *, int, uint16_t,
uint16_t);
static inline void wp_settimer(struct esm_softc *, u_int);
static inline void wp_starttimer(struct esm_softc *);
static inline void wp_stoptimer(struct esm_softc *);
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static inline uint16_t wc_rdreg(struct esm_softc *, uint16_t);
static inline void wc_wrreg(struct esm_softc *, uint16_t, uint16_t);
static inline uint16_t wc_rdchctl(struct esm_softc *, int);
static inline void wc_wrchctl(struct esm_softc *, int, uint16_t);
static inline u_int calc_timer_freq(struct esm_chinfo*);
static void set_timer(struct esm_softc *);
static void esmch_set_format(struct esm_chinfo *,
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
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const audio_params_t *);
static void esmch_combine_input(struct esm_softc *,
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
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struct esm_chinfo *);
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/* Power Management */
void esm_powerhook(int, void *);
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CFATTACH_DECL(esm, sizeof(struct esm_softc),
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esm_match, esm_attach, NULL, NULL);
const struct audio_hw_if esm_hw_if = {
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
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NULL, /* open */
NULL, /* close */
NULL, /* drain */
esm_query_encoding,
esm_set_params,
esm_round_blocksize,
NULL, /* commit_settings */
esm_init_output,
esm_init_input,
NULL, /* start_output */
NULL, /* start_input */
esm_halt_output,
esm_halt_input,
NULL, /* speaker_ctl */
esm_getdev,
NULL, /* getfd */
esm_set_port,
esm_get_port,
esm_query_devinfo,
esm_malloc,
esm_free,
esm_round_buffersize,
esm_mappage,
esm_get_props,
esm_trigger_output,
esm_trigger_input,
NULL,
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NULL,
};
struct audio_device esm_device = {
"ESS Maestro",
"",
"esm"
};
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#define MAESTRO_NENCODINGS 8
static audio_encoding_t esm_encoding[MAESTRO_NENCODINGS] = {
{ 0, AudioEulinear, AUDIO_ENCODING_ULINEAR, 8, 0 },
{ 1, AudioEmulaw, AUDIO_ENCODING_ULAW, 8,
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AUDIO_ENCODINGFLAG_EMULATED },
{ 2, AudioEalaw, AUDIO_ENCODING_ALAW, 8, AUDIO_ENCODINGFLAG_EMULATED },
{ 3, AudioEslinear, AUDIO_ENCODING_SLINEAR, 8, 0 },
{ 4, AudioEslinear_le, AUDIO_ENCODING_SLINEAR_LE, 16, 0 },
{ 5, AudioEulinear_le, AUDIO_ENCODING_ULINEAR_LE, 16,
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AUDIO_ENCODINGFLAG_EMULATED },
{ 6, AudioEslinear_be, AUDIO_ENCODING_SLINEAR_BE, 16,
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AUDIO_ENCODINGFLAG_EMULATED },
{ 7, AudioEulinear_be, AUDIO_ENCODING_ULINEAR_BE, 16,
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AUDIO_ENCODINGFLAG_EMULATED },
};
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
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#define ESM_NFORMATS 4
static const struct audio_format esm_formats[ESM_NFORMATS] = {
{NULL, AUMODE_PLAY | AUMODE_RECORD, AUDIO_ENCODING_SLINEAR_LE, 16, 16,
2, AUFMT_STEREO, 0, {4000, 48000}},
{NULL, AUMODE_PLAY | AUMODE_RECORD, AUDIO_ENCODING_SLINEAR_LE, 16, 16,
1, AUFMT_MONAURAL, 0, {4000, 48000}},
{NULL, AUMODE_PLAY | AUMODE_RECORD, AUDIO_ENCODING_ULINEAR_LE, 8, 8,
2, AUFMT_STEREO, 0, {4000, 48000}},
{NULL, AUMODE_PLAY | AUMODE_RECORD, AUDIO_ENCODING_ULINEAR_LE, 8, 8,
1, AUFMT_MONAURAL, 0, {4000, 48000}},
};
static const struct esm_quirks esm_quirks[] = {
/* COMPAL 38W2 OEM Notebook, e.g. Dell INSPIRON 5000e */
{ PCI_VENDOR_COMPAL, PCI_PRODUCT_COMPAL_38W2, ESM_QUIRKF_SWAPPEDCH },
/* COMPAQ Armada M700 Notebook */
{ PCI_VENDOR_COMPAQ, PCI_PRODUCT_COMPAQ_M700, ESM_QUIRKF_SWAPPEDCH },
/* NEC Versa Pro LX VA26D */
{ PCI_VENDOR_NEC, PCI_PRODUCT_NEC_VA26D, ESM_QUIRKF_GPIO },
/* NEC Versa LX */
{ PCI_VENDOR_NEC, PCI_PRODUCT_NEC_VERSALX, ESM_QUIRKF_GPIO },
/* Toshiba Portege */
{ PCI_VENDOR_TOSHIBA2, PCI_PRODUCT_TOSHIBA2_PORTEGE, ESM_QUIRKF_SWAPPEDCH }
};
enum esm_quirk_flags
esm_get_quirks(pcireg_t subid)
{
int i;
for (i = 0; i < (sizeof esm_quirks / sizeof esm_quirks[0]); i++) {
if (PCI_VENDOR(subid) == esm_quirks[i].eq_vendor &&
PCI_PRODUCT(subid) == esm_quirks[i].eq_product) {
return esm_quirks[i].eq_quirks;
}
}
return 0;
}
#ifdef AUDIO_DEBUG
struct esm_reg_info {
int offset; /* register offset */
int width; /* 1/2/4 bytes */
} dump_regs[] = {
{ PORT_WAVCACHE_CTRL, 2 },
{ PORT_HOSTINT_CTRL, 2 },
{ PORT_HOSTINT_STAT, 2 },
{ PORT_HWVOL_VOICE_SHADOW, 1 },
{ PORT_HWVOL_VOICE, 1 },
{ PORT_HWVOL_MASTER_SHADOW, 1 },
{ PORT_HWVOL_MASTER, 1 },
{ PORT_RINGBUS_CTRL, 4 },
{ PORT_GPIO_DATA, 2 },
{ PORT_GPIO_MASK, 2 },
{ PORT_GPIO_DIR, 2 },
{ PORT_ASSP_CTRL_A, 1 },
{ PORT_ASSP_CTRL_B, 1 },
{ PORT_ASSP_CTRL_C, 1 },
{ PORT_ASSP_INT_STAT, 1 }
};
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static void
esm_dump_regs(struct esm_softc *ess)
{
int i;
printf("%s registers:", ess->sc_dev.dv_xname);
for (i = 0; i < (sizeof dump_regs / sizeof dump_regs[0]); i++) {
if (i % 5 == 0)
printf("\n");
printf("0x%2.2x: ", dump_regs[i].offset);
switch(dump_regs[i].width) {
case 4:
printf("%8.8x, ", bus_space_read_4(ess->st, ess->sh,
dump_regs[i].offset));
break;
case 2:
printf("%4.4x, ", bus_space_read_2(ess->st, ess->sh,
dump_regs[i].offset));
break;
default:
printf("%2.2x, ",
bus_space_read_1(ess->st, ess->sh,
dump_regs[i].offset));
}
}
printf("\n");
}
#endif
/* -----------------------------
* Subsystems.
*/
/* Codec/Ringbus */
/* -------------------------------------------------------------------- */
int
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esm_read_codec(void *sc, uint8_t regno, uint16_t *result)
{
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struct esm_softc *ess;
unsigned t;
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ess = sc;
/* We have to wait for a SAFE time to write addr/data */
for (t = 0; t < 20; t++) {
if ((bus_space_read_1(ess->st, ess->sh, PORT_CODEC_STAT)
& CODEC_STAT_MASK) != CODEC_STAT_PROGLESS)
break;
delay(2); /* 20.8us / 13 */
}
if (t == 20)
printf("%s: esm_read_codec() PROGLESS timed out.\n",
ess->sc_dev.dv_xname);
bus_space_write_1(ess->st, ess->sh, PORT_CODEC_CMD,
CODEC_CMD_READ | regno);
delay(21); /* AC97 cycle = 20.8usec */
/* Wait for data retrieve */
for (t = 0; t < 20; t++) {
if ((bus_space_read_1(ess->st, ess->sh, PORT_CODEC_STAT)
& CODEC_STAT_MASK) == CODEC_STAT_RW_DONE)
break;
delay(2); /* 20.8us / 13 */
}
if (t == 20)
/* Timed out, but perform dummy read. */
printf("%s: esm_read_codec() RW_DONE timed out.\n",
ess->sc_dev.dv_xname);
*result = bus_space_read_2(ess->st, ess->sh, PORT_CODEC_REG);
return 0;
}
int
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esm_write_codec(void *sc, uint8_t regno, uint16_t data)
{
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struct esm_softc *ess;
unsigned t;
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ess = sc;
/* We have to wait for a SAFE time to write addr/data */
for (t = 0; t < 20; t++) {
if ((bus_space_read_1(ess->st, ess->sh, PORT_CODEC_STAT)
& CODEC_STAT_MASK) != CODEC_STAT_PROGLESS)
break;
delay(2); /* 20.8us / 13 */
}
if (t == 20) {
/* Timed out. Abort writing. */
printf("%s: esm_write_codec() PROGLESS timed out.\n",
ess->sc_dev.dv_xname);
return -1;
}
bus_space_write_2(ess->st, ess->sh, PORT_CODEC_REG, data);
bus_space_write_1(ess->st, ess->sh, PORT_CODEC_CMD,
CODEC_CMD_WRITE | regno);
return 0;
}
/* -------------------------------------------------------------------- */
static inline void
ringbus_setdest(struct esm_softc *ess, int src, int dest)
{
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uint32_t data;
data = bus_space_read_4(ess->st, ess->sh, PORT_RINGBUS_CTRL);
data &= ~(0xfU << src);
data |= (0xfU & dest) << src;
bus_space_write_4(ess->st, ess->sh, PORT_RINGBUS_CTRL, data);
}
/* Wave Processor */
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static inline uint16_t
wp_rdreg(struct esm_softc *ess, uint16_t reg)
{
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bus_space_write_2(ess->st, ess->sh, PORT_DSP_INDEX, reg);
return bus_space_read_2(ess->st, ess->sh, PORT_DSP_DATA);
}
static inline void
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wp_wrreg(struct esm_softc *ess, uint16_t reg, uint16_t data)
{
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bus_space_write_2(ess->st, ess->sh, PORT_DSP_INDEX, reg);
bus_space_write_2(ess->st, ess->sh, PORT_DSP_DATA, data);
}
static inline void
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apu_setindex(struct esm_softc *ess, uint16_t reg)
{
int t;
wp_wrreg(ess, WPREG_CRAM_PTR, reg);
/* Sometimes WP fails to set apu register index. */
for (t = 0; t < 1000; t++) {
if (bus_space_read_2(ess->st, ess->sh, PORT_DSP_DATA) == reg)
break;
bus_space_write_2(ess->st, ess->sh, PORT_DSP_DATA, reg);
}
if (t == 1000)
printf("%s: apu_setindex() timed out.\n", ess->sc_dev.dv_xname);
}
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static inline uint16_t
wp_rdapu(struct esm_softc *ess, int ch, uint16_t reg)
{
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uint16_t ret;
apu_setindex(ess, ((unsigned)ch << 4) + reg);
ret = wp_rdreg(ess, WPREG_DATA_PORT);
return ret;
}
static inline void
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wp_wrapu(struct esm_softc *ess, int ch, uint16_t reg, uint16_t data)
{
int t;
DPRINTF(ESM_DEBUG_APU,
("wp_wrapu(%p, ch=%d, reg=0x%x, data=0x%04x)\n",
ess, ch, reg, data));
apu_setindex(ess, ((unsigned)ch << 4) + reg);
wp_wrreg(ess, WPREG_DATA_PORT, data);
for (t = 0; t < 1000; t++) {
if (bus_space_read_2(ess->st, ess->sh, PORT_DSP_DATA) == data)
break;
bus_space_write_2(ess->st, ess->sh, PORT_DSP_DATA, data);
}
if (t == 1000)
printf("%s: wp_wrapu() timed out.\n", ess->sc_dev.dv_xname);
}
static inline void
wp_settimer(struct esm_softc *ess, u_int freq)
{
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u_int clock;
u_int prescale, divide;
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clock = 48000 << 2;
prescale = 0;
divide = (freq != 0) ? (clock / freq) : ~0;
RANGE(divide, WPTIMER_MINDIV, WPTIMER_MAXDIV);
for (; divide > 32 << 1; divide >>= 1)
prescale++;
divide = (divide + 1) >> 1;
for (; prescale < 7 && divide > 2 && !(divide & 1); divide >>= 1)
prescale++;
DPRINTF(ESM_DEBUG_TIMER,
("wp_settimer(%p, %u): clock = %u, prescale = %u, divide = %u\n",
ess, freq, clock, prescale, divide));
wp_wrreg(ess, WPREG_TIMER_ENABLE, 0);
wp_wrreg(ess, WPREG_TIMER_FREQ,
(prescale << WP_TIMER_FREQ_PRESCALE_SHIFT) | (divide - 1));
wp_wrreg(ess, WPREG_TIMER_ENABLE, 1);
}
static inline void
wp_starttimer(struct esm_softc *ess)
{
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wp_wrreg(ess, WPREG_TIMER_START, 1);
}
static inline void
wp_stoptimer(struct esm_softc *ess)
{
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wp_wrreg(ess, WPREG_TIMER_START, 0);
bus_space_write_2(ess->st, ess->sh, PORT_INT_STAT, 1);
}
/* WaveCache */
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static inline uint16_t
wc_rdreg(struct esm_softc *ess, uint16_t reg)
{
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bus_space_write_2(ess->st, ess->sh, PORT_WAVCACHE_INDEX, reg);
return bus_space_read_2(ess->st, ess->sh, PORT_WAVCACHE_DATA);
}
static inline void
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wc_wrreg(struct esm_softc *ess, uint16_t reg, uint16_t data)
{
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bus_space_write_2(ess->st, ess->sh, PORT_WAVCACHE_INDEX, reg);
bus_space_write_2(ess->st, ess->sh, PORT_WAVCACHE_DATA, data);
}
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static inline uint16_t
wc_rdchctl(struct esm_softc *ess, int ch)
{
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return wc_rdreg(ess, ch << 3);
}
static inline void
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wc_wrchctl(struct esm_softc *ess, int ch, uint16_t data)
{
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wc_wrreg(ess, ch << 3, data);
}
/* -----------------------------
* Controller.
*/
int
esm_attach_codec(void *sc, struct ac97_codec_if *codec_if)
{
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struct esm_softc *ess;
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ess = sc;
ess->codec_if = codec_if;
return 0;
}
int
esm_reset_codec(void *sc)
{
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return 0;
}
enum ac97_host_flags
esm_flags_codec(void *sc)
{
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struct esm_softc *ess;
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ess = sc;
return ess->codec_flags;
}
void
esm_initcodec(struct esm_softc *ess)
{
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uint16_t data;
DPRINTF(ESM_DEBUG_CODEC, ("esm_initcodec(%p)\n", ess));
if (bus_space_read_4(ess->st, ess->sh, PORT_RINGBUS_CTRL)
& RINGBUS_CTRL_ACLINK_ENABLED) {
bus_space_write_4(ess->st, ess->sh, PORT_RINGBUS_CTRL, 0);
delay(104); /* 20.8us * (4 + 1) */
}
/* XXX - 2nd codec should be looked at. */
bus_space_write_4(ess->st, ess->sh, PORT_RINGBUS_CTRL,
RINGBUS_CTRL_AC97_SWRESET);
delay(2);
bus_space_write_4(ess->st, ess->sh, PORT_RINGBUS_CTRL,
RINGBUS_CTRL_ACLINK_ENABLED);
delay(21);
esm_read_codec(ess, 0, &data);
if (bus_space_read_1(ess->st, ess->sh, PORT_CODEC_STAT)
& CODEC_STAT_MASK) {
bus_space_write_4(ess->st, ess->sh, PORT_RINGBUS_CTRL, 0);
delay(21);
/* Try cold reset. */
printf("%s: will perform cold reset.\n", ess->sc_dev.dv_xname);
data = bus_space_read_2(ess->st, ess->sh, PORT_GPIO_DIR);
if (pci_conf_read(ess->pc, ess->tag, 0x58) & 1)
data |= 0x10;
data |= 0x009 &
~bus_space_read_2(ess->st, ess->sh, PORT_GPIO_DATA);
bus_space_write_2(ess->st, ess->sh, PORT_GPIO_MASK, 0xff6);
bus_space_write_2(ess->st, ess->sh, PORT_GPIO_DIR,
data | 0x009);
bus_space_write_2(ess->st, ess->sh, PORT_GPIO_DATA, 0x000);
delay(2);
bus_space_write_2(ess->st, ess->sh, PORT_GPIO_DATA, 0x001);
delay(1);
bus_space_write_2(ess->st, ess->sh, PORT_GPIO_DATA, 0x009);
delay(500000);
bus_space_write_2(ess->st, ess->sh, PORT_GPIO_DIR, data);
delay(84); /* 20.8us * 4 */
bus_space_write_4(ess->st, ess->sh, PORT_RINGBUS_CTRL,
RINGBUS_CTRL_ACLINK_ENABLED);
delay(21);
}
}
void
esm_init(struct esm_softc *ess)
{
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/* Reset direct sound. */
bus_space_write_2(ess->st, ess->sh, PORT_HOSTINT_CTRL,
HOSTINT_CTRL_DSOUND_RESET);
delay(10000);
bus_space_write_2(ess->st, ess->sh, PORT_HOSTINT_CTRL, 0);
delay(10000);
/* Enable direct sound interruption. */
bus_space_write_2(ess->st, ess->sh, PORT_HOSTINT_CTRL,
HOSTINT_CTRL_DSOUND_INT_ENABLED);
/* Setup Wave Processor. */
/* Enable WaveCache */
wp_wrreg(ess, WPREG_WAVE_ROMRAM,
WP_WAVE_VIRTUAL_ENABLED | WP_WAVE_DRAM_ENABLED);
bus_space_write_2(ess->st, ess->sh, PORT_WAVCACHE_CTRL,
WAVCACHE_ENABLED | WAVCACHE_WTSIZE_4MB);
/* Setup Codec/Ringbus. */
esm_initcodec(ess);
bus_space_write_4(ess->st, ess->sh, PORT_RINGBUS_CTRL,
RINGBUS_CTRL_RINGBUS_ENABLED | RINGBUS_CTRL_ACLINK_ENABLED);
/* Undocumented registers from the Linux driver. */
wp_wrreg(ess, 0x8, 0xB004);
wp_wrreg(ess, 0x9, 0x001B);
wp_wrreg(ess, 0xA, 0x8000);
wp_wrreg(ess, 0xB, 0x3F37);
wp_wrreg(ess, 0xD, 0x7632);
wp_wrreg(ess, WPREG_BASE, 0x8598); /* Parallel I/O */
ringbus_setdest(ess, RINGBUS_SRC_ADC,
RINGBUS_DEST_STEREO | RINGBUS_DEST_DSOUND_IN);
ringbus_setdest(ess, RINGBUS_SRC_DSOUND,
RINGBUS_DEST_STEREO | RINGBUS_DEST_DAC);
/* Setup ASSP. Needed for Dell Inspiron 7500? */
bus_space_write_1(ess->st, ess->sh, PORT_ASSP_CTRL_B, 0x00);
bus_space_write_1(ess->st, ess->sh, PORT_ASSP_CTRL_A, 0x03);
bus_space_write_1(ess->st, ess->sh, PORT_ASSP_CTRL_C, 0x00);
/*
* Setup GPIO.
* There seems to be speciality with NEC systems.
*/
if (esm_get_quirks(ess->subid) & ESM_QUIRKF_GPIO) {
bus_space_write_2(ess->st, ess->sh, PORT_GPIO_MASK,
0x9ff);
bus_space_write_2(ess->st, ess->sh, PORT_GPIO_DIR,
bus_space_read_2(ess->st, ess->sh, PORT_GPIO_DIR) |
0x600);
bus_space_write_2(ess->st, ess->sh, PORT_GPIO_DATA,
0x200);
}
DUMPREG(ess);
}
/* Channel controller. */
int
esm_init_output (void *sc, void *start, int size)
{
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struct esm_softc *ess;
struct esm_dma *p;
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ess = sc;
p = &ess->sc_dma;
if ((caddr_t)start != p->addr + MAESTRO_PLAYBUF_OFF) {
printf("%s: esm_init_output: bad addr %p\n",
ess->sc_dev.dv_xname, start);
return EINVAL;
}
ess->pch.base = DMAADDR(p) + MAESTRO_PLAYBUF_OFF;
DPRINTF(ESM_DEBUG_DMA, ("%s: pch.base = 0x%x\n",
ess->sc_dev.dv_xname, ess->pch.base));
return 0;
}
int
esm_init_input (void *sc, void *start, int size)
{
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struct esm_softc *ess;
struct esm_dma *p;
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ess = sc;
p = &ess->sc_dma;
if ((caddr_t)start != p->addr + MAESTRO_RECBUF_OFF) {
printf("%s: esm_init_input: bad addr %p\n",
ess->sc_dev.dv_xname, start);
return EINVAL;
}
switch (ess->rch.aputype) {
case APUTYPE_16BITSTEREO:
ess->rch.base = DMAADDR(p) + MAESTRO_RECBUF_L_OFF;
break;
default:
ess->rch.base = DMAADDR(p) + MAESTRO_RECBUF_OFF;
break;
}
DPRINTF(ESM_DEBUG_DMA, ("%s: rch.base = 0x%x\n",
ess->sc_dev.dv_xname, ess->rch.base));
return 0;
}
int
esm_trigger_output(void *sc, void *start, void *end, int blksize,
void (*intr)(void *), void *arg, const audio_params_t *param)
{
size_t size;
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struct esm_softc *ess;
struct esm_chinfo *ch;
struct esm_dma *p;
int pan, choffset;
int i, nch;
unsigned speed, offset, wpwa, dv;
uint16_t apuch;
DPRINTF(ESM_DEBUG_DMA,
("esm_trigger_output(%p, %p, %p, 0x%x, %p, %p, %p)\n",
sc, start, end, blksize, intr, arg, param));
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ess = sc;
ch = &ess->pch;
pan = 0;
nch = 1;
speed = ch->sample_rate;
apuch = ch->num << 1;
#ifdef DIAGNOSTIC
if (ess->pactive) {
printf("%s: esm_trigger_output: already running",
ess->sc_dev.dv_xname);
return EINVAL;
}
#endif
ess->sc_pintr = intr;
ess->sc_parg = arg;
p = &ess->sc_dma;
if ((caddr_t)start != p->addr + MAESTRO_PLAYBUF_OFF) {
printf("%s: esm_trigger_output: bad addr %p\n",
ess->sc_dev.dv_xname, start);
return EINVAL;
}
ess->pch.blocksize = blksize;
ess->pch.apublk = blksize >> 1;
ess->pactive = 1;
size = (size_t)(((caddr_t)end - (caddr_t)start) >> 1);
choffset = MAESTRO_PLAYBUF_OFF;
offset = choffset >> 1;
wpwa = APU_USE_SYSMEM | ((offset >> 8) & APU_64KPAGE_MASK);
DPRINTF(ESM_DEBUG_DMA,
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("choffs=0x%x, wpwa=0x%x, size=0x%lx words\n",
choffset, wpwa, (unsigned long int)size));
switch (ch->aputype) {
case APUTYPE_16BITSTEREO:
ess->pch.apublk >>= 1;
wpwa >>= 1;
size >>= 1;
offset >>= 1;
/* FALLTHROUGH */
case APUTYPE_8BITSTEREO:
if (ess->codec_flags & AC97_HOST_SWAPPED_CHANNELS)
pan = 8;
else
pan = -8;
nch++;
break;
case APUTYPE_8BITLINEAR:
ess->pch.apublk <<= 1;
speed >>= 1;
break;
}
ess->pch.apubase = offset;
ess->pch.apubuf = size;
ess->pch.nextirq = ess->pch.apublk;
set_timer(ess);
wp_starttimer(ess);
dv = (((speed % 48000) << 16) + 24000) / 48000
+ ((speed / 48000) << 16);
for (i = nch-1; i >= 0; i--) {
wp_wrapu(ess, apuch + i, APUREG_WAVESPACE, wpwa & 0xff00);
wp_wrapu(ess, apuch + i, APUREG_CURPTR, offset);
wp_wrapu(ess, apuch + i, APUREG_ENDPTR, offset + size);
wp_wrapu(ess, apuch + i, APUREG_LOOPLEN, size - 1);
wp_wrapu(ess, apuch + i, APUREG_AMPLITUDE, 0xe800);
wp_wrapu(ess, apuch + i, APUREG_POSITION, 0x8f00
| (RADIUS_CENTERCIRCLE << APU_RADIUS_SHIFT)
| ((PAN_FRONT + pan) << APU_PAN_SHIFT));
wp_wrapu(ess, apuch + i, APUREG_FREQ_LOBYTE, APU_plus6dB
| ((dv & 0xff) << APU_FREQ_LOBYTE_SHIFT));
wp_wrapu(ess, apuch + i, APUREG_FREQ_HIWORD, dv >> 8);
if (ch->aputype == APUTYPE_16BITSTEREO)
wpwa |= APU_STEREO >> 1;
pan = -pan;
}
wc_wrchctl(ess, apuch, ch->wcreg_tpl);
if (nch > 1)
wc_wrchctl(ess, apuch + 1, ch->wcreg_tpl);
wp_wrapu(ess, apuch, APUREG_APUTYPE,
(ch->aputype << APU_APUTYPE_SHIFT) | APU_DMA_ENABLED | 0xf);
if (ch->wcreg_tpl & WAVCACHE_CHCTL_STEREO)
wp_wrapu(ess, apuch + 1, APUREG_APUTYPE,
(ch->aputype << APU_APUTYPE_SHIFT) | APU_DMA_ENABLED | 0xf);
return 0;
}
int
esm_trigger_input(void *sc, void *start, void *end, int blksize,
void (*intr)(void *), void *arg, const audio_params_t *param)
{
size_t size;
size_t mixsize;
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struct esm_softc *ess;
struct esm_chinfo *ch;
struct esm_dma *p;
uint32_t chctl, choffset;
uint32_t speed, offset, wpwa, dv;
uint32_t mixoffset, mixdv;
int i, nch;
uint16_t apuch;
uint16_t reg;
DPRINTF(ESM_DEBUG_DMA,
("esm_trigger_input(%p, %p, %p, 0x%x, %p, %p, %p)\n",
sc, start, end, blksize, intr, arg, param));
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ess = sc;
ch = &ess->rch;
nch = 1;
speed = ch->sample_rate;
apuch = ch->num << 1;
#ifdef DIAGNOSTIC
if (ess->ractive) {
printf("%s: esm_trigger_input: already running",
ess->sc_dev.dv_xname);
return EINVAL;
}
#endif
ess->sc_rintr = intr;
ess->sc_rarg = arg;
p = &ess->sc_dma;
if ((caddr_t)start != p->addr + MAESTRO_RECBUF_OFF) {
printf("%s: esm_trigger_input: bad addr %p\n",
ess->sc_dev.dv_xname, start);
return EINVAL;
}
ess->rch.buffer = (caddr_t)start;
ess->rch.offset = 0;
ess->rch.blocksize = blksize;
ess->rch.bufsize = ((caddr_t)end - (caddr_t)start);
ess->rch.apublk = blksize >> 1;
ess->ractive = 1;
size = (size_t)(((caddr_t)end - (caddr_t)start) >> 1);
choffset = MAESTRO_RECBUF_OFF;
switch (ch->aputype) {
case APUTYPE_16BITSTEREO:
size >>= 1;
choffset = MAESTRO_RECBUF_L_OFF;
ess->rch.apublk >>= 1;
nch++;
break;
case APUTYPE_16BITLINEAR:
break;
default:
ess->ractive = 0;
return EINVAL;
}
mixsize = (MAESTRO_MIXBUF_SZ >> 1) >> 1;
mixoffset = MAESTRO_MIXBUF_OFF;
ess->rch.apubase = (choffset >> 1);
ess->rch.apubuf = size;
ess->rch.nextirq = ess->rch.apublk;
set_timer(ess);
wp_starttimer(ess);
if (speed > 47999) speed = 47999;
if (speed < 4000) speed = 4000;
dv = (((speed % 48000) << 16) + 24000) / 48000
+ ((speed / 48000) << 16);
mixdv = 65536; /* 48 kHz */
for (i = 0; i < nch; i++) {
/* Clear all rate conversion WP channel registers first. */
for (reg = 0; reg < 15; reg++)
wp_wrapu(ess, apuch + i, reg, 0);
/* Program the WaveCache for the rate conversion WP channel. */
chctl = (DMAADDR(p) + choffset - 0x10) &
WAVCACHE_CHCTL_ADDRTAG_MASK;
wc_wrchctl(ess, apuch + i, chctl);
/* Program the rate conversion WP channel. */
wp_wrapu(ess, apuch + i, APUREG_FREQ_LOBYTE, APU_plus6dB
| ((dv & 0xff) << APU_FREQ_LOBYTE_SHIFT) | 0x08);
wp_wrapu(ess, apuch + i, APUREG_FREQ_HIWORD, dv >> 8);
offset = choffset >> 1;
wpwa = APU_USE_SYSMEM | ((offset >> 8) & APU_64KPAGE_MASK);
wp_wrapu(ess, apuch + i, APUREG_WAVESPACE, wpwa);
wp_wrapu(ess, apuch + i, APUREG_CURPTR, offset);
wp_wrapu(ess, apuch + i, APUREG_ENDPTR, offset + size);
wp_wrapu(ess, apuch + i, APUREG_LOOPLEN, size - 1);
wp_wrapu(ess, apuch + i, APUREG_EFFECTS_ENV, 0x00f0);
wp_wrapu(ess, apuch + i, APUREG_AMPLITUDE, 0xe800);
wp_wrapu(ess, apuch + i, APUREG_POSITION, 0x8f00
| (RADIUS_CENTERCIRCLE << APU_RADIUS_SHIFT)
| (PAN_FRONT << APU_PAN_SHIFT));
wp_wrapu(ess, apuch + i, APUREG_ROUTE, apuch + 2 + i);
DPRINTF(ESM_DEBUG_DMA,
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("choffs=0x%x, wpwa=0x%x, offset=0x%x words, size=0x%lx words\n",
choffset, wpwa, offset, (unsigned long int)size));
/* Clear all mixer WP channel registers first. */
for (reg = 0; reg < 15; reg++)
wp_wrapu(ess, apuch + 2 + i, reg, 0);
/* Program the WaveCache for the mixer WP channel. */
chctl = (ess->rch.base + mixoffset - 0x10) &
WAVCACHE_CHCTL_ADDRTAG_MASK;
wc_wrchctl(ess, apuch + 2 + i, chctl);
/* Program the mixer WP channel. */
wp_wrapu(ess, apuch + 2 + i, APUREG_FREQ_LOBYTE, APU_plus6dB
| ((mixdv & 0xff) << APU_FREQ_LOBYTE_SHIFT) | 0x08);
wp_wrapu(ess, apuch + 2 + i, APUREG_FREQ_HIWORD, mixdv >> 8);
offset = mixoffset >> 1;
wpwa = APU_USE_SYSMEM | ((offset >> 8) & APU_64KPAGE_MASK);
wp_wrapu(ess, apuch + 2 + i, APUREG_WAVESPACE, wpwa);
wp_wrapu(ess, apuch + 2 + i, APUREG_CURPTR, offset);
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wp_wrapu(ess, apuch + 2 + i, APUREG_ENDPTR,
offset + mixsize);
wp_wrapu(ess, apuch + 2 + i, APUREG_LOOPLEN, mixsize);
wp_wrapu(ess, apuch + 2 + i, APUREG_EFFECTS_ENV, 0x00f0);
wp_wrapu(ess, apuch + 2 + i, APUREG_AMPLITUDE, 0xe800);
wp_wrapu(ess, apuch + 2 + i, APUREG_POSITION, 0x8f00
| (RADIUS_CENTERCIRCLE << APU_RADIUS_SHIFT)
| (PAN_FRONT << APU_PAN_SHIFT));
wp_wrapu(ess, apuch + 2 + i, APUREG_ROUTE,
ROUTE_PARALLEL + i);
DPRINTF(ESM_DEBUG_DMA,
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("mixoffs=0x%x, wpwa=0x%x, offset=0x%x words, size=0x%lx words\n",
mixoffset, wpwa, offset, (unsigned long int)mixsize));
/* Assume we're going to loop to do the right channel. */
choffset += MAESTRO_RECBUF_L_SZ;
mixoffset += MAESTRO_MIXBUF_SZ >> 1;
}
wp_wrapu(ess, apuch, APUREG_APUTYPE,
(APUTYPE_RATECONV << APU_APUTYPE_SHIFT) |
APU_DMA_ENABLED | 0xf);
if (nch > 1)
wp_wrapu(ess, apuch + 1, APUREG_APUTYPE,
(APUTYPE_RATECONV << APU_APUTYPE_SHIFT) |
APU_DMA_ENABLED | 0xf);
wp_wrapu(ess, apuch + 2, APUREG_APUTYPE,
(APUTYPE_INPUTMIXER << APU_APUTYPE_SHIFT) |
APU_DMA_ENABLED | 0xf);
if (nch > 1)
wp_wrapu(ess, apuch + 3, APUREG_APUTYPE,
(APUTYPE_RATECONV << APU_APUTYPE_SHIFT) |
APU_DMA_ENABLED | 0xf);
return 0;
}
int
esm_halt_output(void *sc)
{
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struct esm_softc *ess;
struct esm_chinfo *ch;
DPRINTF(ESM_DEBUG_PARAM, ("esm_halt_output(%p)\n", sc));
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ess = sc;
ch = &ess->pch;
wp_wrapu(ess, (ch->num << 1), APUREG_APUTYPE,
APUTYPE_INACTIVE << APU_APUTYPE_SHIFT);
wp_wrapu(ess, (ch->num << 1) + 1, APUREG_APUTYPE,
APUTYPE_INACTIVE << APU_APUTYPE_SHIFT);
ess->pactive = 0;
if (!ess->ractive)
wp_stoptimer(ess);
return 0;
}
int
esm_halt_input(void *sc)
{
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struct esm_softc *ess;
struct esm_chinfo *ch;
DPRINTF(ESM_DEBUG_PARAM, ("esm_halt_input(%p)\n", sc));
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ess = sc;
ch = &ess->rch;
wp_wrapu(ess, (ch->num << 1), APUREG_APUTYPE,
APUTYPE_INACTIVE << APU_APUTYPE_SHIFT);
wp_wrapu(ess, (ch->num << 1) + 1, APUREG_APUTYPE,
APUTYPE_INACTIVE << APU_APUTYPE_SHIFT);
wp_wrapu(ess, (ch->num << 1) + 2, APUREG_APUTYPE,
APUTYPE_INACTIVE << APU_APUTYPE_SHIFT);
wp_wrapu(ess, (ch->num << 1) + 3, APUREG_APUTYPE,
APUTYPE_INACTIVE << APU_APUTYPE_SHIFT);
ess->ractive = 0;
if (!ess->pactive)
wp_stoptimer(ess);
return 0;
}
static inline u_int
calc_timer_freq(struct esm_chinfo *ch)
{
u_int freq;
freq = (ch->sample_rate + ch->apublk - 1) / ch->apublk;
DPRINTF(ESM_DEBUG_TIMER,
("calc_timer_freq(%p): rate = %u, blk = 0x%x (0x%x): freq = %u\n",
ch, ch->sample_rate, ch->apublk, ch->blocksize, freq));
return freq;
}
static void
set_timer(struct esm_softc *ess)
{
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unsigned freq, freq2;
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freq = 0;
if (ess->pactive)
freq = calc_timer_freq(&ess->pch);
if (ess->ractive) {
freq2 = calc_timer_freq(&ess->rch);
if (freq2 > freq)
freq = freq2;
}
KASSERT(freq != 0);
for (; freq < MAESTRO_MINFREQ; freq <<= 1)
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continue;
if (freq > 0)
wp_settimer(ess, freq);
}
static void
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
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esmch_set_format(struct esm_chinfo *ch, const audio_params_t *p)
{
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uint16_t wcreg_tpl;
uint16_t aputype;
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wcreg_tpl = (ch->base - 16) & WAVCACHE_CHCTL_ADDRTAG_MASK;
aputype = APUTYPE_16BITLINEAR;
if (p->channels == 2) {
wcreg_tpl |= WAVCACHE_CHCTL_STEREO;
aputype++;
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
if (p->precision == 8) {
aputype += 2;
switch (p->encoding) {
case AUDIO_ENCODING_ULINEAR:
case AUDIO_ENCODING_ULINEAR_BE:
case AUDIO_ENCODING_ULINEAR_LE:
wcreg_tpl |= WAVCACHE_CHCTL_U8;
break;
}
}
ch->wcreg_tpl = wcreg_tpl;
ch->aputype = aputype;
ch->sample_rate = p->sample_rate;
DPRINTF(ESM_DEBUG_PARAM, ("esmch_set_format: "
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
"numch=%u, prec=%u, tpl=0x%x, aputype=%d, rate=%u\n",
p->channels, p->precision, wcreg_tpl, aputype, p->sample_rate));
}
/*
* Since we can't record in true stereo, this function combines
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* the separately recorded left and right channels into the final
* buffer for the upper layer.
*/
static void
esmch_combine_input(struct esm_softc *ess, struct esm_chinfo *ch)
{
size_t offset, resid, count;
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uint32_t *dst32s;
const uint32_t *left32s, *right32s;
uint32_t left32, right32;
/* The current offset into the upper layer buffer. */
offset = ch->offset;
/* The number of bytes left to combine. */
resid = ch->blocksize;
while (resid > 0) {
/* The 32-bit words for the left channel. */
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left32s = (const uint32_t *)(ess->sc_dma.addr +
MAESTRO_RECBUF_L_OFF + offset / 2);
/* The 32-bit words for the right channel. */
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right32s = (const uint32_t *)(ess->sc_dma.addr +
MAESTRO_RECBUF_R_OFF + offset / 2);
/* The pointer to the 32-bit words we will write. */
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dst32s = (uint32_t *)(ch->buffer + offset);
/* Get the number of bytes we will combine now. */
count = ch->bufsize - offset;
if (count > resid)
count = resid;
resid -= count;
offset += count;
if (offset == ch->bufsize)
offset = 0;
/* Combine, writing two 32-bit words at a time. */
KASSERT((count & (sizeof(uint32_t) * 2 - 1)) == 0);
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count /= (sizeof(uint32_t) * 2);
while (count > 0) {
left32 = *(left32s++);
right32 = *(right32s++);
/* XXX this endian handling is half-baked at best */
#if BYTE_ORDER == LITTLE_ENDIAN
*(dst32s++) = (left32 & 0xFFFF) | (right32 << 16);
*(dst32s++) = (left32 >> 16) | (right32 & 0xFFFF0000);
#else /* BYTE_ORDER == BIG_ENDIAN */
*(dst32s++) = (left32 & 0xFFFF0000) | (right32 >> 16);
*(dst32s++) = (left32 << 16) | (right32 & 0xFFFF);
#endif /* BYTE_ORDER == BIG_ENDIAN */
count--;
}
}
/* Update the offset. */
ch->offset = offset;
}
/*
* Audio interface glue functions
*/
int
esm_getdev (void *sc, struct audio_device *adp)
{
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*adp = esm_device;
return 0;
}
int
esm_round_blocksize(void *sc, int blk, int mode,
const audio_params_t *param)
{
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DPRINTF(ESM_DEBUG_PARAM,
("esm_round_blocksize(%p, 0x%x)", sc, blk));
blk &= ~0x3f; /* keep good alignment */
DPRINTF(ESM_DEBUG_PARAM, (" = 0x%x\n", blk));
return blk;
}
int
esm_query_encoding(void *sc, struct audio_encoding *fp)
{
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DPRINTF(ESM_DEBUG_PARAM,
("esm_query_encoding(%p, %d)\n", sc, fp->index));
if (fp->index < 0 || fp->index >= MAESTRO_NENCODINGS)
return EINVAL;
*fp = esm_encoding[fp->index];
return 0;
}
int
esm_set_params(void *sc, int setmode, int usemode,
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
audio_params_t *play, audio_params_t *rec,
stream_filter_list_t *pfil, stream_filter_list_t *rfil)
{
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struct esm_softc *ess;
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
audio_params_t *p;
const audio_params_t *hw_play, *hw_rec;
stream_filter_list_t *fil;
int mode, i;
DPRINTF(ESM_DEBUG_PARAM,
("esm_set_params(%p, 0x%x, 0x%x, %p, %p)\n",
sc, setmode, usemode, play, rec));
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ess = sc;
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
hw_play = NULL;
hw_rec = NULL;
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for (mode = AUMODE_RECORD; mode != -1;
mode = mode == AUMODE_RECORD ? AUMODE_PLAY : -1) {
if ((setmode & mode) == 0)
continue;
p = mode == AUMODE_PLAY ? play : rec;
if (p->sample_rate < 4000 || p->sample_rate > 48000 ||
(p->precision != 8 && p->precision != 16) ||
(p->channels != 1 && p->channels != 2))
return EINVAL;
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
fil = mode == AUMODE_PLAY ? pfil : rfil;
i = auconv_set_converter(esm_formats, ESM_NFORMATS,
mode, p, FALSE, fil);
if (i < 0)
return EINVAL;
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
if (fil->req_size > 0)
p = &fil->filters[0].param;
if (mode == AUMODE_PLAY)
hw_play = p;
else
hw_rec = p;
}
if (hw_play)
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
esmch_set_format(&ess->pch, hw_play);
if (hw_rec)
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
esmch_set_format(&ess->rch, hw_rec);
return 0;
}
int
esm_set_port(void *sc, mixer_ctrl_t *cp)
{
2005-01-15 18:19:51 +03:00
struct esm_softc *ess;
2005-01-15 18:19:51 +03:00
ess = sc;
return ess->codec_if->vtbl->mixer_set_port(ess->codec_if, cp);
}
int
esm_get_port(void *sc, mixer_ctrl_t *cp)
{
2005-01-15 18:19:51 +03:00
struct esm_softc *ess;
2005-01-15 18:19:51 +03:00
ess = sc;
return ess->codec_if->vtbl->mixer_get_port(ess->codec_if, cp);
}
int
esm_query_devinfo(void *sc, mixer_devinfo_t *dip)
{
2005-01-15 18:19:51 +03:00
struct esm_softc *ess;
2005-01-15 18:19:51 +03:00
ess = sc;
return ess->codec_if->vtbl->query_devinfo(ess->codec_if, dip);
}
void *
esm_malloc(void *sc, int direction, size_t size,
struct malloc_type *pool, int flags)
{
2005-01-15 18:19:51 +03:00
struct esm_softc *ess;
int off;
DPRINTF(ESM_DEBUG_DMA,
2004-07-08 22:08:58 +04:00
("esm_malloc(%p, %d, 0x%lx, %p, 0x%x)",
sc, direction, (unsigned long int)size, pool, flags));
2005-01-15 18:19:51 +03:00
ess = sc;
/*
* Each buffer can only be allocated once.
*/
if (ess->rings_alloced & direction) {
DPRINTF(ESM_DEBUG_DMA, (" = 0 (ENOMEM)\n"));
return 0;
}
/*
* Mark this buffer as allocated and return its
* kernel virtual address.
*/
ess->rings_alloced |= direction;
off = (direction == AUMODE_PLAY ?
MAESTRO_PLAYBUF_OFF : MAESTRO_RECBUF_OFF);
DPRINTF(ESM_DEBUG_DMA, (" = %p (DMAADDR 0x%x)\n",
ess->sc_dma.addr + off,
(int)DMAADDR(&ess->sc_dma) + off));
2005-01-15 18:19:51 +03:00
return ess->sc_dma.addr + off;
}
void
esm_free(void *sc, void *ptr, struct malloc_type *pool)
{
2005-01-15 18:19:51 +03:00
struct esm_softc *ess;
DPRINTF(ESM_DEBUG_DMA,
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("esm_free(%p, %p, %p)\n",
sc, ptr, pool));
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ess = sc;
if ((caddr_t)ptr == ess->sc_dma.addr + MAESTRO_PLAYBUF_OFF)
ess->rings_alloced &= ~AUMODE_PLAY;
else if ((caddr_t)ptr == ess->sc_dma.addr + MAESTRO_RECBUF_OFF)
ess->rings_alloced &= ~AUMODE_RECORD;
}
size_t
esm_round_buffersize(void *sc, int direction, size_t size)
{
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if (size > MAESTRO_PLAYBUF_SZ)
size = MAESTRO_PLAYBUF_SZ;
if (size > MAESTRO_RECBUF_SZ)
size = MAESTRO_RECBUF_SZ;
return size;
}
paddr_t
esm_mappage(void *sc, void *mem, off_t off, int prot)
{
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struct esm_softc *ess;
DPRINTF(ESM_DEBUG_DMA,
("esm_mappage(%p, %p, 0x%lx, 0x%x)\n",
sc, mem, (unsigned long)off, prot));
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ess = sc;
if (off < 0)
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return -1;
if ((caddr_t)mem == ess->sc_dma.addr + MAESTRO_PLAYBUF_OFF)
off += MAESTRO_PLAYBUF_OFF;
else if ((caddr_t)mem == ess->sc_dma.addr + MAESTRO_RECBUF_OFF)
off += MAESTRO_RECBUF_OFF;
else
return -1;
return bus_dmamem_mmap(ess->dmat, ess->sc_dma.segs, ess->sc_dma.nsegs,
off, prot, BUS_DMA_WAITOK);
}
int
esm_get_props(void *sc)
{
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return AUDIO_PROP_MMAP | AUDIO_PROP_INDEPENDENT | AUDIO_PROP_FULLDUPLEX;
}
/* -----------------------------
* Bus space.
*/
int
esm_intr(void *sc)
{
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struct esm_softc *ess;
uint16_t status;
uint16_t pos;
int ret;
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ess = sc;
ret = 0;
status = bus_space_read_1(ess->st, ess->sh, PORT_HOSTINT_STAT);
if (!status)
return 0;
/* Acknowledge all. */
bus_space_write_2(ess->st, ess->sh, PORT_INT_STAT, 1);
bus_space_write_1(ess->st, ess->sh, PORT_HOSTINT_STAT, 0);
#if 0 /* XXX - HWVOL */
if (status & HOSTINT_STAT_HWVOL) {
u_int delta;
delta = bus_space_read_1(ess->st, ess->sh, PORT_HWVOL_MASTER)
- 0x88;
if (delta & 0x11)
mixer_set(device_get_softc(ess->dev),
SOUND_MIXER_VOLUME, 0);
else {
mixer_set(device_get_softc(ess->dev),
SOUND_MIXER_VOLUME,
mixer_get(device_get_softc(ess->dev),
SOUND_MIXER_VOLUME)
+ ((delta >> 5) & 0x7) - 4
+ ((delta << 7) & 0x700) - 0x400);
}
bus_space_write_1(ess->st, ess->sh, PORT_HWVOL_MASTER, 0x88);
ret++;
}
#endif /* XXX - HWVOL */
if (ess->pactive) {
pos = wp_rdapu(ess, ess->pch.num << 1, APUREG_CURPTR);
DPRINTF(ESM_DEBUG_IRQ, (" %4.4x/%4.4x ", pos,
wp_rdapu(ess, (ess->pch.num<<1)+1, APUREG_CURPTR)));
pos -= ess->pch.apubase;
if (pos >= ess->pch.nextirq &&
pos - ess->pch.nextirq < ess->pch.apubuf / 2) {
ess->pch.nextirq += ess->pch.apublk;
if (ess->pch.nextirq >= ess->pch.apubuf)
ess->pch.nextirq = 0;
if (ess->sc_pintr) {
DPRINTF(ESM_DEBUG_IRQ, ("P\n"));
ess->sc_pintr(ess->sc_parg);
}
}
ret++;
}
if (ess->ractive) {
pos = wp_rdapu(ess, ess->rch.num << 1, APUREG_CURPTR);
DPRINTF(ESM_DEBUG_IRQ, (" %4.4x/%4.4x ", pos,
wp_rdapu(ess, (ess->rch.num<<1)+1, APUREG_CURPTR)));
pos -= ess->rch.apubase;
if (pos >= ess->rch.nextirq &&
pos - ess->rch.nextirq < ess->rch.apubuf / 2) {
ess->rch.nextirq += ess->rch.apublk;
if (ess->rch.nextirq >= ess->rch.apubuf)
ess->rch.nextirq = 0;
if (ess->sc_rintr) {
DPRINTF(ESM_DEBUG_IRQ, ("R\n"));
switch(ess->rch.aputype) {
case APUTYPE_16BITSTEREO:
esmch_combine_input(ess, &ess->rch);
break;
}
ess->sc_rintr(ess->sc_rarg);
}
}
ret++;
}
return ret;
}
int
esm_allocmem(struct esm_softc *sc, size_t size, size_t align,
struct esm_dma *p)
{
int error;
p->size = size;
error = bus_dmamem_alloc(sc->dmat, p->size, align, 0,
p->segs, sizeof(p->segs)/sizeof(p->segs[0]),
&p->nsegs, BUS_DMA_NOWAIT);
if (error)
return error;
error = bus_dmamem_map(sc->dmat, p->segs, p->nsegs, p->size,
&p->addr, BUS_DMA_NOWAIT|BUS_DMA_COHERENT);
if (error)
goto free;
error = bus_dmamap_create(sc->dmat, p->size, 1, p->size,
0, BUS_DMA_NOWAIT, &p->map);
if (error)
goto unmap;
error = bus_dmamap_load(sc->dmat, p->map, p->addr, p->size, NULL,
BUS_DMA_NOWAIT);
if (error)
goto destroy;
return 0;
destroy:
bus_dmamap_destroy(sc->dmat, p->map);
unmap:
bus_dmamem_unmap(sc->dmat, p->addr, p->size);
free:
bus_dmamem_free(sc->dmat, p->segs, p->nsegs);
return error;
}
int
esm_match(struct device *dev, struct cfdata *match, void *aux)
{
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struct pci_attach_args *pa;
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pa = (struct pci_attach_args *)aux;
switch (PCI_VENDOR(pa->pa_id)) {
case PCI_VENDOR_ESSTECH:
switch (PCI_PRODUCT(pa->pa_id)) {
case PCI_PRODUCT_ESSTECH_MAESTRO1:
case PCI_PRODUCT_ESSTECH_MAESTRO2:
case PCI_PRODUCT_ESSTECH_MAESTRO2E:
return 1;
}
case PCI_VENDOR_ESSTECH2:
switch (PCI_PRODUCT(pa->pa_id)) {
case PCI_PRODUCT_ESSTECH2_MAESTRO1:
return 1;
}
}
return 0;
}
void
esm_attach(struct device *parent, struct device *self, void *aux)
{
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char devinfo[256];
struct esm_softc *ess;
struct pci_attach_args *pa;
const char *intrstr;
pci_chipset_tag_t pc;
pcitag_t tag;
pci_intr_handle_t ih;
pcireg_t csr, data;
int revision;
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uint16_t codec_data;
uint16_t pcmbar;
int error;
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ess = (struct esm_softc *)self;
pa = (struct pci_attach_args *)aux;
pc = pa->pa_pc;
tag = pa->pa_tag;
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aprint_naive(": Audio controller\n");
pci_devinfo(pa->pa_id, pa->pa_class, 0, devinfo, sizeof(devinfo));
revision = PCI_REVISION(pa->pa_class);
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aprint_normal(": %s (rev. 0x%02x)\n", devinfo, revision);
/* Enable the device. */
csr = pci_conf_read(pc, tag, PCI_COMMAND_STATUS_REG);
pci_conf_write(pc, tag, PCI_COMMAND_STATUS_REG,
csr | PCI_COMMAND_MASTER_ENABLE | PCI_COMMAND_IO_ENABLE);
/* Map I/O register */
if (pci_mapreg_map(pa, PCI_CBIO, PCI_MAPREG_TYPE_IO, 0,
&ess->st, &ess->sh, NULL, NULL)) {
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aprint_error("%s: can't map i/o space\n", ess->sc_dev.dv_xname);
return;
}
/* Initialize softc */
ess->pch.num = 0;
ess->rch.num = 1;
ess->dmat = pa->pa_dmat;
ess->tag = tag;
ess->pc = pc;
ess->subid = pci_conf_read(pc, tag, PCI_SUBSYS_ID_REG);
DPRINTF(ESM_DEBUG_PCI,
("%s: sub-system vendor 0x%4.4x, product 0x%4.4x\n",
ess->sc_dev.dv_xname,
PCI_VENDOR(ess->subid), PCI_PRODUCT(ess->subid)));
/* Map and establish the interrupt. */
if (pci_intr_map(pa, &ih)) {
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aprint_error("%s: can't map interrupt\n", ess->sc_dev.dv_xname);
return;
}
intrstr = pci_intr_string(pc, ih);
ess->ih = pci_intr_establish(pc, ih, IPL_AUDIO, esm_intr, self);
if (ess->ih == NULL) {
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aprint_error("%s: can't establish interrupt",
ess->sc_dev.dv_xname);
if (intrstr != NULL)
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aprint_normal(" at %s", intrstr);
aprint_normal("\n");
return;
}
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aprint_normal("%s: interrupting at %s\n",
ess->sc_dev.dv_xname, intrstr);
/*
* Setup PCI config registers
*/
/* power up chip */
if ((error = pci_activate(pa->pa_pc, pa->pa_tag, ess,
pci_activate_null)) && error != EOPNOTSUPP) {
aprint_error("%s: cannot activate %d\n", ess->sc_dev.dv_xname,
error);
return;
}
delay(100000);
/* Disable all legacy emulations. */
data = pci_conf_read(pc, tag, CONF_LEGACY);
pci_conf_write(pc, tag, CONF_LEGACY, data | LEGACY_DISABLED);
/* Disconnect from CHI. (Makes Dell inspiron 7500 work?)
* Enable posted write.
* Prefer PCI timing rather than that of ISA.
* Don't swap L/R. */
data = pci_conf_read(pc, tag, CONF_MAESTRO);
data |= MAESTRO_CHIBUS | MAESTRO_POSTEDWRITE | MAESTRO_DMA_PCITIMING;
data &= ~MAESTRO_SWAP_LR;
pci_conf_write(pc, tag, CONF_MAESTRO, data);
/* initialize sound chip */
esm_init(ess);
esm_read_codec(ess, 0, &codec_data);
if (codec_data == 0x80) {
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aprint_error("%s: PT101 codec detected!\n",
ess->sc_dev.dv_xname);
return;
}
/*
* Some cards and Notebooks appear to have left and right channels
* reversed. Check if there is a corresponding quirk entry for
* the subsystem vendor and product and if so, set the appropriate
* codec flag.
*/
if (esm_get_quirks(ess->subid) & ESM_QUIRKF_SWAPPEDCH) {
ess->codec_flags |= AC97_HOST_SWAPPED_CHANNELS;
}
ess->codec_flags |= AC97_HOST_DONT_READ;
/* initialize AC97 host interface */
ess->host_if.arg = self;
ess->host_if.attach = esm_attach_codec;
ess->host_if.read = esm_read_codec;
ess->host_if.write = esm_write_codec;
ess->host_if.reset = esm_reset_codec;
ess->host_if.flags = esm_flags_codec;
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
if (ac97_attach(&ess->host_if, self) != 0)
return;
/* allocate our DMA region */
if (esm_allocmem(ess, MAESTRO_DMA_SZ, MAESTRO_DMA_ALIGN,
&ess->sc_dma)) {
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aprint_error("%s: couldn't allocate memory!\n",
ess->sc_dev.dv_xname);
return;
}
ess->rings_alloced = 0;
/* set DMA base address */
for (pcmbar = WAVCACHE_PCMBAR; pcmbar < WAVCACHE_PCMBAR + 4; pcmbar++)
wc_wrreg(ess, pcmbar,
DMAADDR(&ess->sc_dma) >> WAVCACHE_BASEADDR_SHIFT);
audio_attach_mi(&esm_hw_if, self, &ess->sc_dev);
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ess->esm_suspend = PWR_RESUME;
ess->esm_powerhook = powerhook_establish(ess->sc_dev.dv_xname,
esm_powerhook, ess);
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}
/* Power Hook */
void
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esm_powerhook(int why, void *v)
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{
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struct esm_softc *ess;
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ess = (struct esm_softc *)v;
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DPRINTF(ESM_DEBUG_PARAM,
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("%s: ESS maestro 2E why=%d\n", ess->sc_dev.dv_xname, why));
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switch (why) {
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case PWR_SUSPEND:
case PWR_STANDBY:
ess->esm_suspend = why;
esm_suspend(ess);
DPRINTF(ESM_DEBUG_RESUME, ("esm_suspend\n"));
break;
case PWR_RESUME:
ess->esm_suspend = why;
esm_resume(ess);
DPRINTF(ESM_DEBUG_RESUME, ("esm_resumed\n"));
break;
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}
}
int
esm_suspend(struct esm_softc *ess)
{
int x;
x = splaudio();
wp_stoptimer(ess);
bus_space_write_2(ess->st, ess->sh, PORT_HOSTINT_CTRL, 0);
esm_halt_output(ess);
esm_halt_input(ess);
splx(x);
/* Power down everything except clock. */
esm_write_codec(ess, AC97_REG_POWER, 0xdf00);
delay(20);
bus_space_write_4(ess->st, ess->sh, PORT_RINGBUS_CTRL, 0);
delay(1);
return 0;
}
int
esm_resume(struct esm_softc *ess)
{
int x;
uint16_t pcmbar;
delay(100000);
esm_init(ess);
/* set DMA base address */
for (pcmbar = WAVCACHE_PCMBAR; pcmbar < WAVCACHE_PCMBAR + 4; pcmbar++)
wc_wrreg(ess, pcmbar,
DMAADDR(&ess->sc_dma) >> WAVCACHE_BASEADDR_SHIFT);
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ess->codec_if->vtbl->restore_ports(ess->codec_if);
#if 0
if (mixer_reinit(dev)) {
printf("%s: unable to reinitialize the mixer\n",
ess->sc_dev.dv_xname);
return ENXIO;
}
#endif
x = splaudio();
#if TODO
if (ess->pactive)
esm_start_output(ess);
if (ess->ractive)
esm_start_input(ess);
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#endif
if (ess->pactive || ess->ractive) {
set_timer(ess);
wp_starttimer(ess);
}
splx(x);
return 0;
}
#if 0
int
esm_shutdown(struct esm_softc *ess)
{
int i;
wp_stoptimer(ess);
bus_space_write_2(ess->st, ess->sh, PORT_HOSTINT_CTRL, 0);
esm_halt_output(ess);
esm_halt_input(ess);
return 0;
}
#endif