NetBSD/sys/dev/pci/esmvar.h

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/* $NetBSD: esmvar.h,v 1.13 2005/12/11 12:22:49 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
*
*/
/*
* 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.
*/
/* IRQ timer fequency limits */
#define MAESTRO_MINFREQ 24
#define MAESTRO_MAXFREQ 48000
/*
* This driver allocates a contiguous 256KB region of memory.
* The Maestro's DMA interface, called the WaveCache, is weak
* (or at least incorrectly documented), and forces us to keep
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* things very simple. This region is very carefully divided up
* into 64KB quarters, making 64KB a fundamental constant for
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* this implementation - and this is as large as we can allow
* the upper-layer playback and record buffers to become.
*/
#define MAESTRO_QUARTER_SZ (64 * 1024)
/*
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* The first quarter of memory is used while recording. The
* first 512 bytes of it is reserved as a scratch area for the
* APUs that want to write (uninteresting, to us) FIFO status
* information. After some guard space, another 512 bytes is
* reserved for the APUs doing mixing. The remainder of this
* quarter of memory is wasted.
*/
#define MAESTRO_FIFO_OFF (MAESTRO_QUARTER_SZ * 0)
#define MAESTRO_FIFO_SZ (512)
#define MAESTRO_MIXBUF_OFF (MAESTRO_FIFO_OFF + 4096)
#define MAESTRO_MIXBUF_SZ (512)
/*
* The second quarter of memory is the playback buffer.
*/
#define MAESTRO_PLAYBUF_OFF (MAESTRO_QUARTER_SZ * 1)
#define MAESTRO_PLAYBUF_SZ MAESTRO_QUARTER_SZ
/*
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* The third quarter of memory is the mono record buffer.
* This is the only record buffer that the upper layer knows.
* When recording in stereo, our driver combines (in software)
* separately recorded left and right buffers here.
*/
#define MAESTRO_RECBUF_OFF (MAESTRO_QUARTER_SZ * 2)
#define MAESTRO_RECBUF_SZ MAESTRO_QUARTER_SZ
/*
* The fourth quarter of memory is the stereo record buffer.
* When recording in stereo, the left and right channels are
* recorded separately into the two halves of this buffer.
*/
#define MAESTRO_RECBUF_L_OFF (MAESTRO_QUARTER_SZ * 3)
#define MAESTRO_RECBUF_L_SZ (MAESTRO_QUARTER_SZ / 2)
#define MAESTRO_RECBUF_R_OFF (MAESTRO_RECBUF_L_OFF + MAESTRO_RECBUF_L_SZ)
#define MAESTRO_RECBUF_R_SZ (MAESTRO_QUARTER_SZ / 2)
/*
* The size and alignment of the entire region. We keep
* the region aligned to a 128KB boundary, since this should
* force A16..A0 on all chip-generated addresses to correspond
* exactly to APU register contents.
*/
#define MAESTRO_DMA_SZ (MAESTRO_QUARTER_SZ * 4)
#define MAESTRO_DMA_ALIGN (128 * 1024)
struct esm_dma {
bus_dmamap_t map;
caddr_t addr;
bus_dma_segment_t segs[1];
int nsegs;
size_t size;
struct esm_dma *next;
};
#define DMAADDR(p) ((p)->map->dm_segs[0].ds_addr)
#define KERNADDR(p) ((void *)((p)->addr))
struct esm_chinfo {
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uint32_t base; /* DMA base */
caddr_t buffer; /* upper layer buffer */
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uint32_t offset; /* offset into buffer */
uint32_t blocksize; /* block size in bytes */
uint32_t bufsize; /* buffer size in bytes */
unsigned num; /* logical channel number */
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uint16_t aputype; /* APU channel type */
uint16_t apubase; /* first sample number */
uint16_t apublk; /* blk size in samples per ch */
uint16_t apubuf; /* buf size in samples per ch */
uint16_t nextirq; /* pos to trigger next IRQ at */
uint16_t wcreg_tpl; /* wavecache tag and format */
uint16_t sample_rate;
};
struct esm_softc {
struct device sc_dev;
bus_space_tag_t st;
bus_space_handle_t sh;
pcitag_t tag;
pci_chipset_tag_t pc;
bus_dma_tag_t dmat;
pcireg_t subid;
void *ih;
struct ac97_codec_if *codec_if;
struct ac97_host_if host_if;
enum ac97_host_flags codec_flags;
struct esm_dma sc_dma;
int rings_alloced;
int pactive, ractive;
struct esm_chinfo pch;
struct esm_chinfo rch;
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void (*sc_pintr)(void *);
void *sc_parg;
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void (*sc_rintr)(void *);
void *sc_rarg;
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/* Power Management */
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char esm_suspend;
void *esm_powerhook;
};
enum esm_quirk_flags {
ESM_QUIRKF_GPIO = 0x1, /* needs GPIO operation */
ESM_QUIRKF_SWAPPEDCH = 0x2, /* left/right is reversed */
};
struct esm_quirks {
pci_vendor_id_t eq_vendor; /* subsystem vendor */
pci_product_id_t eq_product; /* and product */
enum esm_quirk_flags eq_quirks; /* needed quirks */
};
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int esm_read_codec(void *, uint8_t, uint16_t *);
int esm_write_codec(void *, uint8_t, uint16_t);
int esm_attach_codec(void *, struct ac97_codec_if *);
int esm_reset_codec(void *);
enum ac97_host_flags esm_flags_codec(void *);
void esm_power(struct esm_softc *, int);
void esm_init(struct esm_softc *);
void esm_initcodec(struct esm_softc *);
int esm_init_output(void *, void *, int);
int esm_init_input(void *, void *, int);
int esm_trigger_output(void *, void *, void *, int, void (*)(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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void *, const audio_params_t *);
int esm_trigger_input(void *, void *, void *, int, void (*)(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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void *, const audio_params_t *);
int esm_halt_output(void *);
int esm_halt_input(void *);
int esm_getdev(void *, struct audio_device *);
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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int esm_round_blocksize(void *, int, int, const audio_params_t *);
int esm_query_encoding(void *, struct audio_encoding *);
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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int esm_set_params(void *, int, int, audio_params_t *, audio_params_t *,
stream_filter_list_t *, stream_filter_list_t *);
int esm_set_port(void *, mixer_ctrl_t *);
int esm_get_port(void *, mixer_ctrl_t *);
int esm_query_devinfo(void *, mixer_devinfo_t *);
void *esm_malloc(void *, int, size_t, struct malloc_type *, int);
void esm_free(void *, void *, struct malloc_type *);
size_t esm_round_buffersize(void *, int, size_t);
paddr_t esm_mappage(void *, void *, off_t, int);
int esm_get_props(void *);
int esm_match(struct device *, struct cfdata *, void *);
void esm_attach(struct device *, struct device *, void *);
int esm_intr(void *);
int esm_allocmem(struct esm_softc *, size_t, size_t,
struct esm_dma *);
int esm_suspend(struct esm_softc *);
int esm_resume(struct esm_softc *);
int esm_shutdown(struct esm_softc *);
enum esm_quirk_flags esm_get_quirks(pcireg_t);