NetBSD/sys/dev/ic/am7930.c

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/* $NetBSD: am7930.c,v 1.47 2005/01/15 15:19:52 kent Exp $ */
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/*
* Copyright (c) 1995 Rolf Grossmann
* 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. All advertising materials mentioning features or use of this software
* must display the following acknowledgement:
* This product includes software developed by Rolf Grossmann.
* 4. 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.
*/
/*
* Front-end attachment independent layer for AMD 79c30
* audio driver. No ISDN support.
*/
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#include <sys/cdefs.h>
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__KERNEL_RCSID(0, "$NetBSD: am7930.c,v 1.47 2005/01/15 15:19:52 kent Exp $");
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#include "audio.h"
#if NAUDIO > 0
#include <sys/param.h>
#include <sys/systm.h>
#include <sys/errno.h>
#include <sys/ioctl.h>
#include <sys/device.h>
#include <sys/proc.h>
#include <machine/bus.h>
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#include <machine/autoconf.h>
#include <machine/cpu.h>
#include <sys/audioio.h>
#include <dev/audio_if.h>
#include <dev/ic/am7930reg.h>
#include <dev/ic/am7930var.h>
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#ifdef AUDIO_DEBUG
int am7930debug = 0;
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#define DPRINTF(x) if (am7930debug) printf x
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#else
#define DPRINTF(x)
#endif
/* The following tables stolen from former (4.4Lite's) sys/sparc/bsd_audio.c */
/*
* gx, gr & stg gains. this table must contain 256 elements with
* the 0th being "infinity" (the magic value 9008). The remaining
* elements match sun's gain curve (but with higher resolution):
* -18 to 0dB in .16dB steps then 0 to 12dB in .08dB steps.
*/
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static const uint16_t gx_coeff[256] = {
0x9008, 0x8e7c, 0x8e51, 0x8e45, 0x8d42, 0x8d3b, 0x8c36, 0x8c33,
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0x8b32, 0x8b2a, 0x8b2b, 0x8b2c, 0x8b25, 0x8b23, 0x8b22, 0x8b22,
0x9122, 0x8b1a, 0x8aa3, 0x8aa3, 0x8b1c, 0x8aa6, 0x912d, 0x912b,
0x8aab, 0x8b12, 0x8aaa, 0x8ab2, 0x9132, 0x8ab4, 0x913c, 0x8abb,
0x9142, 0x9144, 0x9151, 0x8ad5, 0x8aeb, 0x8a79, 0x8a5a, 0x8a4a,
0x8b03, 0x91c2, 0x91bb, 0x8a3f, 0x8a33, 0x91b2, 0x9212, 0x9213,
0x8a2c, 0x921d, 0x8a23, 0x921a, 0x9222, 0x9223, 0x922d, 0x9231,
0x9234, 0x9242, 0x925b, 0x92dd, 0x92c1, 0x92b3, 0x92ab, 0x92a4,
0x92a2, 0x932b, 0x9341, 0x93d3, 0x93b2, 0x93a2, 0x943c, 0x94b2,
0x953a, 0x9653, 0x9782, 0x9e21, 0x9d23, 0x9cd2, 0x9c23, 0x9baa,
0x9bde, 0x9b33, 0x9b22, 0x9b1d, 0x9ab2, 0xa142, 0xa1e5, 0x9a3b,
0xa213, 0xa1a2, 0xa231, 0xa2eb, 0xa313, 0xa334, 0xa421, 0xa54b,
0xada4, 0xac23, 0xab3b, 0xaaab, 0xaa5c, 0xb1a3, 0xb2ca, 0xb3bd,
0xbe24, 0xbb2b, 0xba33, 0xc32b, 0xcb5a, 0xd2a2, 0xe31d, 0x0808,
0x72ba, 0x62c2, 0x5c32, 0x52db, 0x513e, 0x4cce, 0x43b2, 0x4243,
0x41b4, 0x3b12, 0x3bc3, 0x3df2, 0x34bd, 0x3334, 0x32c2, 0x3224,
0x31aa, 0x2a7b, 0x2aaa, 0x2b23, 0x2bba, 0x2c42, 0x2e23, 0x25bb,
0x242b, 0x240f, 0x231a, 0x22bb, 0x2241, 0x2223, 0x221f, 0x1a33,
0x1a4a, 0x1acd, 0x2132, 0x1b1b, 0x1b2c, 0x1b62, 0x1c12, 0x1c32,
0x1d1b, 0x1e71, 0x16b1, 0x1522, 0x1434, 0x1412, 0x1352, 0x1323,
0x1315, 0x12bc, 0x127a, 0x1235, 0x1226, 0x11a2, 0x1216, 0x0a2a,
0x11bc, 0x11d1, 0x1163, 0x0ac2, 0x0ab2, 0x0aab, 0x0b1b, 0x0b23,
0x0b33, 0x0c0f, 0x0bb3, 0x0c1b, 0x0c3e, 0x0cb1, 0x0d4c, 0x0ec1,
0x079a, 0x0614, 0x0521, 0x047c, 0x0422, 0x03b1, 0x03e3, 0x0333,
0x0322, 0x031c, 0x02aa, 0x02ba, 0x02f2, 0x0242, 0x0232, 0x0227,
0x0222, 0x021b, 0x01ad, 0x0212, 0x01b2, 0x01bb, 0x01cb, 0x01f6,
0x0152, 0x013a, 0x0133, 0x0131, 0x012c, 0x0123, 0x0122, 0x00a2,
0x011b, 0x011e, 0x0114, 0x00b1, 0x00aa, 0x00b3, 0x00bd, 0x00ba,
0x00c5, 0x00d3, 0x00f3, 0x0062, 0x0051, 0x0042, 0x003b, 0x0033,
0x0032, 0x002a, 0x002c, 0x0025, 0x0023, 0x0022, 0x001a, 0x0021,
0x001b, 0x001b, 0x001d, 0x0015, 0x0013, 0x0013, 0x0012, 0x0012,
0x000a, 0x000a, 0x0011, 0x0011, 0x000b, 0x000b, 0x000c, 0x000e,
};
/*
* second stage play gain.
*/
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static const uint16_t ger_coeff[] = {
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0x431f, /* 5. dB */
0x331f, /* 5.5 dB */
0x40dd, /* 6. dB */
0x11dd, /* 6.5 dB */
0x440f, /* 7. dB */
0x411f, /* 7.5 dB */
0x311f, /* 8. dB */
0x5520, /* 8.5 dB */
0x10dd, /* 9. dB */
0x4211, /* 9.5 dB */
0x410f, /* 10. dB */
0x111f, /* 10.5 dB */
0x600b, /* 11. dB */
0x00dd, /* 11.5 dB */
0x4210, /* 12. dB */
0x110f, /* 13. dB */
0x7200, /* 14. dB */
0x2110, /* 15. dB */
0x2200, /* 15.9 dB */
0x000b, /* 16.9 dB */
0x000f /* 18. dB */
#define NGER (sizeof(ger_coeff) / sizeof(ger_coeff[0]))
};
/*
* Reset chip and set boot-time softc defaults.
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*/
void
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am7930_init(struct am7930_softc *sc, int flag)
{
DPRINTF(("am7930_init()\n"));
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/* set boot defaults */
sc->sc_rlevel = 128;
sc->sc_plevel = 128;
sc->sc_mlevel = 0;
sc->sc_out_port = AUDIOAMD_SPEAKER_VOL;
sc->sc_mic_mute = 0;
/* disable sample interrupts */
AM7930_IWRITE(sc, AM7930_IREG_MUX_MCR4, 0);
/* initialise voice and data, and disable interrupts */
AM7930_IWRITE(sc, AM7930_IREG_INIT,
AM7930_INIT_PMS_ACTIVE | AM7930_INIT_INT_DISABLE);
if (flag == AUDIOAMD_DMA_MODE) {
/* configure PP for serial (SBP) mode */
AM7930_IWRITE(sc, AM7930_IREG_PP_PPCR1, AM7930_PPCR1_SBP);
/*
* Initialise the MUX unit - route the MAP to the PP
*/
AM7930_IWRITE(sc, AM7930_IREG_MUX_MCR1,
(AM7930_MCRCHAN_BA << 4) | AM7930_MCRCHAN_BD);
AM7930_IWRITE(sc, AM7930_IREG_MUX_MCR2, AM7930_MCRCHAN_NC);
AM7930_IWRITE(sc, AM7930_IREG_MUX_MCR3, AM7930_MCRCHAN_NC);
} else {
/*
* Initialize the MUX unit. We use MCR3 to route the MAP
* through channel Bb. MCR1 and MCR2 are unused.
* Setting the INT enable bit in MCR4 will generate an
* interrupt on each converted audio sample.
*/
AM7930_IWRITE(sc, AM7930_IREG_MUX_MCR1, 0);
AM7930_IWRITE(sc, AM7930_IREG_MUX_MCR2, 0);
AM7930_IWRITE(sc, AM7930_IREG_MUX_MCR3,
(AM7930_MCRCHAN_BB << 4) | AM7930_MCRCHAN_BA);
AM7930_IWRITE(sc, AM7930_IREG_MUX_MCR4,
AM7930_MCR4_INT_ENABLE);
}
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}
int
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am7930_open(void *addr, int flags)
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{
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struct am7930_softc *sc;
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sc = addr;
DPRINTF(("sa_open: unit %p\n", sc));
sc->sc_glue->onopen(sc);
DPRINTF(("saopen: ok -> sc=0x%p\n",sc));
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return 0;
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}
void
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am7930_close(void *addr)
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{
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struct am7930_softc *sc;
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sc = addr;
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DPRINTF(("sa_close: sc=%p\n", sc));
sc->sc_glue->onclose(sc);
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DPRINTF(("sa_close: closed.\n"));
}
/*
* XXX should be extended to handle a few of the more common formats.
*/
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int
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am7930_set_params(void *addr, int setmode, int usemode, audio_params_t *p,
audio_params_t *r, stream_filter_list_t *pfil, stream_filter_list_t *rfil)
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{
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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audio_params_t hw;
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struct am7930_softc *sc;
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sc = addr;
if ((usemode & AUMODE_PLAY) == AUMODE_PLAY) {
if (p->sample_rate < 7500 || p->sample_rate > 8500 ||
p->encoding != AUDIO_ENCODING_ULAW ||
p->precision != 8 ||
p->channels != 1)
return EINVAL;
p->sample_rate = 8000;
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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if (sc->sc_glue->output_conv != NULL) {
hw = *p;
hw.encoding = AUDIO_ENCODING_NONE;
hw.precision *= sc->sc_glue->factor;
pfil->append(pfil, sc->sc_glue->output_conv, &hw);
}
}
if ((usemode & AUMODE_RECORD) == AUMODE_RECORD) {
if (r->sample_rate < 7500 || r->sample_rate > 8500 ||
r->encoding != AUDIO_ENCODING_ULAW ||
r->precision != 8 ||
r->channels != 1)
return EINVAL;
r->sample_rate = 8000;
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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if (sc->sc_glue->input_conv != NULL) {
hw = *r;
hw.encoding = AUDIO_ENCODING_NONE;
hw.precision *= sc->sc_glue->factor;
pfil->append(rfil, sc->sc_glue->input_conv, &hw);
}
}
return 0;
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}
int
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am7930_query_encoding(void *addr, struct audio_encoding *fp)
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{
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switch (fp->index) {
case 0:
strcpy(fp->name, AudioEmulaw);
fp->encoding = AUDIO_ENCODING_ULAW;
fp->precision = 8;
fp->flags = 0;
break;
default:
return EINVAL;
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/*NOTREACHED*/
}
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return 0;
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}
int
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am7930_round_blocksize(void *addr, int blk,
int mode, const audio_params_t *param)
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{
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return blk;
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}
int
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am7930_commit_settings(void *addr)
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{
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struct am7930_softc *sc;
uint16_t ger, gr, gx, stgr;
uint8_t mmr2, mmr3;
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int s, level;
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DPRINTF(("sa_commit.\n"));
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sc = addr;
gx = gx_coeff[sc->sc_rlevel];
stgr = gx_coeff[sc->sc_mlevel];
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level = (sc->sc_plevel * (256 + NGER)) >> 8;
if (level >= 256) {
ger = ger_coeff[level - 256];
gr = gx_coeff[255];
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} else {
ger = ger_coeff[0];
gr = gx_coeff[level];
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}
s = splaudio();
mmr2 = AM7930_IREAD(sc, AM7930_IREG_MAP_MMR2);
if (sc->sc_out_port == AUDIOAMD_SPEAKER_VOL)
mmr2 |= AM7930_MMR2_LS;
else
mmr2 &= ~AM7930_MMR2_LS;
AM7930_IWRITE(sc, AM7930_IREG_MAP_MMR2, mmr2);
mmr3 = AM7930_IREAD(sc, AM7930_IREG_MAP_MMR3);
if (sc->sc_mic_mute)
mmr3 |= AM7930_MMR3_MUTE;
else
mmr3 &= ~AM7930_MMR3_MUTE;
AM7930_IWRITE(sc, AM7930_IREG_MAP_MMR3, mmr3);
AM7930_IWRITE(sc, AM7930_IREG_MAP_MMR1,
AM7930_MMR1_GX | AM7930_MMR1_GER |
AM7930_MMR1_GR | AM7930_MMR1_STG);
AM7930_IWRITE16(sc, AM7930_IREG_MAP_GX, gx);
AM7930_IWRITE16(sc, AM7930_IREG_MAP_STG, stgr);
AM7930_IWRITE16(sc, AM7930_IREG_MAP_GR, gr);
AM7930_IWRITE16(sc, AM7930_IREG_MAP_GER, ger);
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splx(s);
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return 0;
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}
int
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am7930_halt_output(void *addr)
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{
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struct am7930_softc *sc;
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sc = addr;
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/* XXX only halt, if input is also halted ?? */
AM7930_IWRITE(sc, AM7930_IREG_INIT,
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AM7930_INIT_PMS_ACTIVE | AM7930_INIT_INT_DISABLE);
return 0;
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}
int
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am7930_halt_input(void *addr)
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{
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struct am7930_softc *sc;
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sc = addr;
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/* XXX only halt, if output is also halted ?? */
AM7930_IWRITE(sc, AM7930_IREG_INIT,
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AM7930_INIT_PMS_ACTIVE | AM7930_INIT_INT_DISABLE);
return 0;
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}
/*
* XXX chip is full-duplex, but really attach-dependent.
* For now we know of no half-duplex attachments.
*/
int
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am7930_get_props(void *addr)
{
return AUDIO_PROP_FULLDUPLEX;
}
/*
* Attach-dependent channel set/query
*/
int
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am7930_set_port(void *addr, mixer_ctrl_t *cp)
{
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struct am7930_softc *sc;
DPRINTF(("am7930_set_port: port=%d", cp->dev));
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sc = addr;
if (cp->dev == AUDIOAMD_RECORD_SOURCE ||
cp->dev == AUDIOAMD_MONITOR_OUTPUT ||
cp->dev == AUDIOAMD_MIC_MUTE) {
if (cp->type != AUDIO_MIXER_ENUM)
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return EINVAL;
} else if (cp->type != AUDIO_MIXER_VALUE ||
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cp->un.value.num_channels != 1) {
return EINVAL;
}
switch(cp->dev) {
case AUDIOAMD_MIC_VOL:
sc->sc_rlevel = cp->un.value.level[AUDIO_MIXER_LEVEL_MONO];
break;
case AUDIOAMD_SPEAKER_VOL:
case AUDIOAMD_HEADPHONES_VOL:
sc->sc_plevel = cp->un.value.level[AUDIO_MIXER_LEVEL_MONO];
break;
case AUDIOAMD_MONITOR_VOL:
sc->sc_mlevel = cp->un.value.level[AUDIO_MIXER_LEVEL_MONO];
break;
case AUDIOAMD_RECORD_SOURCE:
if (cp->un.ord != AUDIOAMD_MIC_VOL)
return EINVAL;
break;
case AUDIOAMD_MIC_MUTE:
sc->sc_mic_mute = cp->un.ord;
break;
case AUDIOAMD_MONITOR_OUTPUT:
if (cp->un.ord != AUDIOAMD_SPEAKER_VOL &&
cp->un.ord != AUDIOAMD_HEADPHONES_VOL)
return EINVAL;
sc->sc_out_port = cp->un.ord;
break;
default:
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return EINVAL;
/* NOTREACHED */
}
return 0;
}
int
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am7930_get_port(void *addr, mixer_ctrl_t *cp)
{
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struct am7930_softc *sc;
DPRINTF(("am7930_get_port: port=%d\n", cp->dev));
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sc = addr;
if (cp->dev == AUDIOAMD_RECORD_SOURCE ||
cp->dev == AUDIOAMD_MONITOR_OUTPUT ||
cp->dev == AUDIOAMD_MIC_MUTE) {
if (cp->type != AUDIO_MIXER_ENUM)
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return EINVAL;
} else if (cp->type != AUDIO_MIXER_VALUE ||
cp->un.value.num_channels != 1) {
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return EINVAL;
}
switch(cp->dev) {
case AUDIOAMD_MIC_VOL:
cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] = sc->sc_rlevel;
break;
case AUDIOAMD_SPEAKER_VOL:
case AUDIOAMD_HEADPHONES_VOL:
cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] = sc->sc_plevel;
break;
case AUDIOAMD_MONITOR_VOL:
cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] = sc->sc_mlevel;
break;
case AUDIOAMD_RECORD_SOURCE:
cp->un.ord = AUDIOAMD_MIC_VOL;
break;
case AUDIOAMD_MIC_MUTE:
cp->un.ord = sc->sc_mic_mute;
break;
case AUDIOAMD_MONITOR_OUTPUT:
cp->un.ord = sc->sc_out_port;
break;
default:
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return EINVAL;
/* NOTREACHED */
}
return 0;
}
/*
* Define mixer control facilities.
*/
int
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am7930_query_devinfo(void *addr, mixer_devinfo_t *dip)
{
DPRINTF(("am7930_query_devinfo()\n"));
switch(dip->index) {
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case AUDIOAMD_MIC_VOL:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = AUDIOAMD_INPUT_CLASS;
dip->prev = AUDIO_MIXER_LAST;
dip->next = AUDIOAMD_MIC_MUTE;
strcpy(dip->label.name, AudioNmicrophone);
dip->un.v.num_channels = 1;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
case AUDIOAMD_SPEAKER_VOL:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = AUDIOAMD_OUTPUT_CLASS;
dip->prev = dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNspeaker);
dip->un.v.num_channels = 1;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
case AUDIOAMD_HEADPHONES_VOL:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = AUDIOAMD_OUTPUT_CLASS;
dip->prev = dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNheadphone);
dip->un.v.num_channels = 1;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
case AUDIOAMD_MONITOR_VOL:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = AUDIOAMD_MONITOR_CLASS;
dip->prev = dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNmonitor);
dip->un.v.num_channels = 1;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
case AUDIOAMD_RECORD_SOURCE:
dip->type = AUDIO_MIXER_ENUM;
dip->mixer_class = AUDIOAMD_RECORD_CLASS;
dip->next = dip->prev = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNsource);
dip->un.e.num_mem = 1;
strcpy(dip->un.e.member[0].label.name, AudioNmicrophone);
dip->un.e.member[0].ord = AUDIOAMD_MIC_VOL;
break;
case AUDIOAMD_MONITOR_OUTPUT:
dip->type = AUDIO_MIXER_ENUM;
dip->mixer_class = AUDIOAMD_MONITOR_CLASS;
dip->next = dip->prev = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNoutput);
dip->un.e.num_mem = 2;
strcpy(dip->un.e.member[0].label.name, AudioNspeaker);
dip->un.e.member[0].ord = AUDIOAMD_SPEAKER_VOL;
strcpy(dip->un.e.member[1].label.name, AudioNheadphone);
dip->un.e.member[1].ord = AUDIOAMD_HEADPHONES_VOL;
break;
case AUDIOAMD_MIC_MUTE:
dip->type = AUDIO_MIXER_ENUM;
dip->mixer_class = AUDIOAMD_INPUT_CLASS;
dip->prev = AUDIOAMD_MIC_VOL;
dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNmute);
dip->un.e.num_mem = 2;
strcpy(dip->un.e.member[0].label.name, AudioNoff);
dip->un.e.member[0].ord = 0;
strcpy(dip->un.e.member[1].label.name, AudioNon);
dip->un.e.member[1].ord = 1;
break;
case AUDIOAMD_INPUT_CLASS:
dip->type = AUDIO_MIXER_CLASS;
dip->mixer_class = AUDIOAMD_INPUT_CLASS;
dip->next = dip->prev = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioCinputs);
break;
case AUDIOAMD_OUTPUT_CLASS:
dip->type = AUDIO_MIXER_CLASS;
dip->mixer_class = AUDIOAMD_OUTPUT_CLASS;
dip->next = dip->prev = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioCoutputs);
break;
case AUDIOAMD_RECORD_CLASS:
dip->type = AUDIO_MIXER_CLASS;
dip->mixer_class = AUDIOAMD_RECORD_CLASS;
dip->next = dip->prev = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioCrecord);
break;
case AUDIOAMD_MONITOR_CLASS:
dip->type = AUDIO_MIXER_CLASS;
dip->mixer_class = AUDIOAMD_MONITOR_CLASS;
dip->next = dip->prev = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioCmonitor);
break;
default:
return ENXIO;
/*NOTREACHED*/
}
DPRINTF(("AUDIO_MIXER_DEVINFO: name=%s\n", dip->label.name));
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return 0;
}
#endif /* NAUDIO */