2005-12-11 15:16:03 +03:00
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/* $NetBSD: esmvar.h,v 1.13 2005/12/11 12:22:49 christos Exp $ */
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2001-01-08 22:54:31 +03:00
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/*-
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2002-12-30 08:23:27 +03:00
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* Copyright (c) 2002, 2003 Matt Fredette
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* All rights reserved.
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*
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2003-12-04 16:57:30 +03:00
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* Copyright (c) 2000, 2001 Rene Hexel <rh@NetBSD.org>
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2001-01-08 22:54:31 +03:00
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* All rights reserved.
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*
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* Copyright (c) 2000 Taku YAMAMOTO <taku@cent.saitama-u.ac.jp>
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* All rights reserved.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions
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* are met:
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* 1. Redistributions of source code must retain the above copyright
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* notice, this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright
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* notice, this list of conditions and the following disclaimer in the
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* documentation and/or other materials provided with the distribution.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND
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* ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
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* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
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* ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE
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* FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
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* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
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* OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
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* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
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* LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
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* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
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* SUCH DAMAGE.
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*
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* Taku Id: maestro.c,v 1.12 2000/09/06 03:32:34 taku Exp
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* FreeBSD: /c/ncvs/src/sys/dev/sound/pci/maestro.c,v 1.4 2000/12/18 01:36:35 cg Exp
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*
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*/
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/*
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* Credits:
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*
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* This code is based on the FreeBSD driver written by Taku YAMAMOTO
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*
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*
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* Original credits from the FreeBSD driver:
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*
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* Part of this code (especially in many magic numbers) was heavily inspired
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* by the Linux driver originally written by
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* Alan Cox <alan.cox@linux.org>, modified heavily by
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* Zach Brown <zab@zabbo.net>.
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*
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* busdma()-ize and buffer size reduction were suggested by
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* Cameron Grant <gandalf@vilnya.demon.co.uk>.
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* Also he showed me the way to use busdma() suite.
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*
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* Internal speaker problems on NEC VersaPro's and Dell Inspiron 7500
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* were looked at by
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* Munehiro Matsuda <haro@tk.kubota.co.jp>,
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* who brought patches based on the Linux driver with some simplification.
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*/
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/* IRQ timer fequency limits */
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#define MAESTRO_MINFREQ 24
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#define MAESTRO_MAXFREQ 48000
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2002-12-30 08:23:27 +03:00
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/*
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* This driver allocates a contiguous 256KB region of memory.
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* The Maestro's DMA interface, called the WaveCache, is weak
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* (or at least incorrectly documented), and forces us to keep
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2005-01-15 18:19:51 +03:00
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* things very simple. This region is very carefully divided up
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2002-12-30 08:23:27 +03:00
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* into 64KB quarters, making 64KB a fundamental constant for
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2005-01-15 18:19:51 +03:00
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* this implementation - and this is as large as we can allow
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2002-12-30 08:23:27 +03:00
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* the upper-layer playback and record buffers to become.
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*/
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#define MAESTRO_QUARTER_SZ (64 * 1024)
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/*
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2005-01-15 18:19:51 +03:00
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* The first quarter of memory is used while recording. The
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2002-12-30 08:23:27 +03:00
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* first 512 bytes of it is reserved as a scratch area for the
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* APUs that want to write (uninteresting, to us) FIFO status
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* information. After some guard space, another 512 bytes is
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* reserved for the APUs doing mixing. The remainder of this
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* quarter of memory is wasted.
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*/
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#define MAESTRO_FIFO_OFF (MAESTRO_QUARTER_SZ * 0)
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#define MAESTRO_FIFO_SZ (512)
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#define MAESTRO_MIXBUF_OFF (MAESTRO_FIFO_OFF + 4096)
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#define MAESTRO_MIXBUF_SZ (512)
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/*
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* The second quarter of memory is the playback buffer.
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*/
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#define MAESTRO_PLAYBUF_OFF (MAESTRO_QUARTER_SZ * 1)
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#define MAESTRO_PLAYBUF_SZ MAESTRO_QUARTER_SZ
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/*
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2005-01-15 18:19:51 +03:00
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* The third quarter of memory is the mono record buffer.
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2002-12-30 08:23:27 +03:00
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* This is the only record buffer that the upper layer knows.
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* When recording in stereo, our driver combines (in software)
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* separately recorded left and right buffers here.
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*/
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#define MAESTRO_RECBUF_OFF (MAESTRO_QUARTER_SZ * 2)
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#define MAESTRO_RECBUF_SZ MAESTRO_QUARTER_SZ
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/*
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* The fourth quarter of memory is the stereo record buffer.
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* When recording in stereo, the left and right channels are
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* recorded separately into the two halves of this buffer.
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*/
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#define MAESTRO_RECBUF_L_OFF (MAESTRO_QUARTER_SZ * 3)
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#define MAESTRO_RECBUF_L_SZ (MAESTRO_QUARTER_SZ / 2)
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#define MAESTRO_RECBUF_R_OFF (MAESTRO_RECBUF_L_OFF + MAESTRO_RECBUF_L_SZ)
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#define MAESTRO_RECBUF_R_SZ (MAESTRO_QUARTER_SZ / 2)
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/*
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* The size and alignment of the entire region. We keep
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* the region aligned to a 128KB boundary, since this should
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* force A16..A0 on all chip-generated addresses to correspond
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* exactly to APU register contents.
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*/
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#define MAESTRO_DMA_SZ (MAESTRO_QUARTER_SZ * 4)
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#define MAESTRO_DMA_ALIGN (128 * 1024)
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2001-01-08 22:54:31 +03:00
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struct esm_dma {
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bus_dmamap_t map;
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caddr_t addr;
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bus_dma_segment_t segs[1];
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int nsegs;
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size_t size;
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struct esm_dma *next;
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};
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#define DMAADDR(p) ((p)->map->dm_segs[0].ds_addr)
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#define KERNADDR(p) ((void *)((p)->addr))
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struct esm_chinfo {
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2005-01-15 18:19:51 +03:00
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uint32_t base; /* DMA base */
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2002-12-30 08:23:27 +03:00
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caddr_t buffer; /* upper layer buffer */
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2005-01-15 18:19:51 +03:00
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uint32_t offset; /* offset into buffer */
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uint32_t blocksize; /* block size in bytes */
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uint32_t bufsize; /* buffer size in bytes */
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2001-01-08 22:54:31 +03:00
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unsigned num; /* logical channel number */
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2005-01-15 18:19:51 +03:00
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uint16_t aputype; /* APU channel type */
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uint16_t apubase; /* first sample number */
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uint16_t apublk; /* blk size in samples per ch */
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uint16_t apubuf; /* buf size in samples per ch */
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uint16_t nextirq; /* pos to trigger next IRQ at */
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uint16_t wcreg_tpl; /* wavecache tag and format */
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uint16_t sample_rate;
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2001-01-08 22:54:31 +03:00
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};
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struct esm_softc {
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struct device sc_dev;
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bus_space_tag_t st;
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bus_space_handle_t sh;
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pcitag_t tag;
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pci_chipset_tag_t pc;
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bus_dma_tag_t dmat;
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pcireg_t subid;
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void *ih;
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struct ac97_codec_if *codec_if;
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struct ac97_host_if host_if;
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2001-01-10 02:27:07 +03:00
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enum ac97_host_flags codec_flags;
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2001-01-08 22:54:31 +03:00
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2002-12-30 08:23:27 +03:00
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struct esm_dma sc_dma;
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int rings_alloced;
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2001-01-08 22:54:31 +03:00
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int pactive, ractive;
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struct esm_chinfo pch;
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struct esm_chinfo rch;
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2005-01-15 18:19:51 +03:00
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void (*sc_pintr)(void *);
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void *sc_parg;
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2001-01-08 22:54:31 +03:00
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2005-01-15 18:19:51 +03:00
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void (*sc_rintr)(void *);
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void *sc_rarg;
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2001-02-13 02:56:40 +03:00
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/* Power Management */
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2005-01-15 18:19:51 +03:00
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char esm_suspend;
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void *esm_powerhook;
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2001-01-08 22:54:31 +03:00
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};
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2002-01-01 01:06:47 +03:00
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enum esm_quirk_flags {
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ESM_QUIRKF_GPIO = 0x1, /* needs GPIO operation */
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ESM_QUIRKF_SWAPPEDCH = 0x2, /* left/right is reversed */
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};
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2001-01-10 02:27:07 +03:00
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struct esm_quirks {
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pci_vendor_id_t eq_vendor; /* subsystem vendor */
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pci_product_id_t eq_product; /* and product */
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enum esm_quirk_flags eq_quirks; /* needed quirks */
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};
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2005-01-15 18:19:51 +03:00
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int esm_read_codec(void *, uint8_t, uint16_t *);
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int esm_write_codec(void *, uint8_t, uint16_t);
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2001-01-08 22:54:31 +03:00
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int esm_attach_codec(void *, struct ac97_codec_if *);
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2004-09-22 16:20:24 +04:00
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int esm_reset_codec(void *);
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2001-01-10 02:27:07 +03:00
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enum ac97_host_flags esm_flags_codec(void *);
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2001-01-08 22:54:31 +03:00
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void esm_power(struct esm_softc *, int);
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void esm_init(struct esm_softc *);
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void esm_initcodec(struct esm_softc *);
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int esm_init_output(void *, void *, int);
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2002-12-30 08:23:27 +03:00
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int esm_init_input(void *, void *, int);
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2001-01-08 22:54:31 +03:00
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int esm_trigger_output(void *, void *, void *, int, void (*)(void *),
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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
2005-01-11 01:01:36 +03:00
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void *, const audio_params_t *);
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2001-01-08 22:54:31 +03:00
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int esm_trigger_input(void *, void *, void *, int, void (*)(void *),
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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
2005-01-11 01:01:36 +03:00
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void *, const audio_params_t *);
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2001-01-08 22:54:31 +03:00
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int esm_halt_output(void *);
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int esm_halt_input(void *);
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int esm_getdev(void *, struct audio_device *);
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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
2005-01-11 01:01:36 +03:00
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int esm_round_blocksize(void *, int, int, const audio_params_t *);
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2001-01-08 22:54:31 +03:00
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int esm_query_encoding(void *, struct audio_encoding *);
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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
2005-01-11 01:01:36 +03:00
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int esm_set_params(void *, int, int, audio_params_t *, audio_params_t *,
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stream_filter_list_t *, stream_filter_list_t *);
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2001-01-08 22:54:31 +03:00
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int esm_set_port(void *, mixer_ctrl_t *);
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int esm_get_port(void *, mixer_ctrl_t *);
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int esm_query_devinfo(void *, mixer_devinfo_t *);
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2003-02-01 09:23:35 +03:00
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void *esm_malloc(void *, int, size_t, struct malloc_type *, int);
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void esm_free(void *, void *, struct malloc_type *);
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2001-01-08 22:54:31 +03:00
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size_t esm_round_buffersize(void *, int, size_t);
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paddr_t esm_mappage(void *, void *, off_t, int);
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int esm_get_props(void *);
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int esm_match(struct device *, struct cfdata *, void *);
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void esm_attach(struct device *, struct device *, void *);
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int esm_intr(void *);
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int esm_allocmem(struct esm_softc *, size_t, size_t,
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struct esm_dma *);
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int esm_suspend(struct esm_softc *);
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int esm_resume(struct esm_softc *);
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int esm_shutdown(struct esm_softc *);
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2001-01-10 02:27:07 +03:00
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enum esm_quirk_flags esm_get_quirks(pcireg_t);
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