/* $NetBSD: sv.c,v 1.29 2005/04/30 15:24:51 hannken Exp $ */ /* $OpenBSD: sv.c,v 1.2 1998/07/13 01:50:15 csapuntz Exp $ */ /* * Copyright (c) 1999 The NetBSD Foundation, Inc. * All rights reserved. * * This code is derived from software contributed to The NetBSD Foundation * by Charles M. Hannum. * * 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 the NetBSD * Foundation, Inc. and its contributors. * 4. Neither the name of The NetBSD Foundation nor the names of its * contributors may be used to endorse or promote products derived * from this software without specific prior written permission. * * THIS SOFTWARE IS PROVIDED BY THE NETBSD FOUNDATION, INC. 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 FOUNDATION 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. */ /* * Copyright (c) 1998 Constantine Paul Sapuntzakis * All rights reserved * * Author: Constantine Paul Sapuntzakis (csapuntz@cvs.openbsd.org) * * Redistribution and use in source and binary forms, with or without * modification, are permitted provided that the following conditions * are met: * 1. Redistributions of source code must retain the above copyright * notice, this list of conditions and the following disclaimer. * 2. Redistributions in binary form must reproduce the above copyright * notice, this list of conditions and the following disclaimer in the * documentation and/or other materials provided with the distribution. * 3. The author's name or those of the contributors may be used to * endorse or promote products derived from this software without * specific prior written permission. * * THIS SOFTWARE IS PROVIDED BY THE AUTHOR(S) 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 FOUNDATION 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. */ /* * S3 SonicVibes driver * Heavily based on the eap driver by Lennart Augustsson */ #include __KERNEL_RCSID(0, "$NetBSD: sv.c,v 1.29 2005/04/30 15:24:51 hannken Exp $"); #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include /* XXX * The SonicVibes DMA is broken and only works on 24-bit addresses. * As long as bus_dmamem_alloc_range() is missing we use the ISA * DMA tag on i386. */ #if defined(i386) #include "isa.h" #if NISA > 0 #include #endif #endif #ifdef AUDIO_DEBUG #define DPRINTF(x) if (svdebug) printf x #define DPRINTFN(n,x) if (svdebug>(n)) printf x int svdebug = 0; #else #define DPRINTF(x) #define DPRINTFN(n,x) #endif int sv_match(struct device *, struct cfdata *, void *); void sv_attach(struct device *, struct device *, void *); int sv_intr(void *); struct sv_dma { bus_dmamap_t map; caddr_t addr; bus_dma_segment_t segs[1]; int nsegs; size_t size; struct sv_dma *next; }; #define DMAADDR(p) ((p)->map->dm_segs[0].ds_addr) #define KERNADDR(p) ((void *)((p)->addr)) CFATTACH_DECL(sv, sizeof(struct sv_softc), sv_match, sv_attach, NULL, NULL); struct audio_device sv_device = { "S3 SonicVibes", "", "sv" }; #define ARRAY_SIZE(foo) ((sizeof(foo)) / sizeof(foo[0])) int sv_allocmem(struct sv_softc *, size_t, size_t, int, struct sv_dma *); int sv_freemem(struct sv_softc *, struct sv_dma *); int sv_open(void *, int); int sv_query_encoding(void *, struct audio_encoding *); int sv_set_params(void *, int, int, audio_params_t *, audio_params_t *, stream_filter_list_t *, stream_filter_list_t *); int sv_round_blocksize(void *, int, int, const audio_params_t *); int sv_trigger_output(void *, void *, void *, int, void (*)(void *), void *, const audio_params_t *); int sv_trigger_input(void *, void *, void *, int, void (*)(void *), void *, const audio_params_t *); int sv_halt_output(void *); int sv_halt_input(void *); int sv_getdev(void *, struct audio_device *); int sv_mixer_set_port(void *, mixer_ctrl_t *); int sv_mixer_get_port(void *, mixer_ctrl_t *); int sv_query_devinfo(void *, mixer_devinfo_t *); void *sv_malloc(void *, int, size_t, struct malloc_type *, int); void sv_free(void *, void *, struct malloc_type *); size_t sv_round_buffersize(void *, int, size_t); paddr_t sv_mappage(void *, void *, off_t, int); int sv_get_props(void *); #ifdef AUDIO_DEBUG void sv_dumpregs(struct sv_softc *sc); #endif const struct audio_hw_if sv_hw_if = { sv_open, NULL, /* close */ NULL, sv_query_encoding, sv_set_params, sv_round_blocksize, NULL, NULL, NULL, NULL, NULL, sv_halt_output, sv_halt_input, NULL, sv_getdev, NULL, sv_mixer_set_port, sv_mixer_get_port, sv_query_devinfo, sv_malloc, sv_free, sv_round_buffersize, sv_mappage, sv_get_props, sv_trigger_output, sv_trigger_input, NULL, }; #define SV_NFORMATS 4 static const struct audio_format sv_formats[SV_NFORMATS] = { {NULL, AUMODE_PLAY | AUMODE_RECORD, AUDIO_ENCODING_SLINEAR_LE, 16, 16, 2, AUFMT_STEREO, 0, {2000, 48000}}, {NULL, AUMODE_PLAY | AUMODE_RECORD, AUDIO_ENCODING_SLINEAR_LE, 16, 16, 1, AUFMT_MONAURAL, 0, {2000, 48000}}, {NULL, AUMODE_PLAY | AUMODE_RECORD, AUDIO_ENCODING_ULINEAR_LE, 8, 8, 2, AUFMT_STEREO, 0, {2000, 48000}}, {NULL, AUMODE_PLAY | AUMODE_RECORD, AUDIO_ENCODING_ULINEAR_LE, 8, 8, 1, AUFMT_MONAURAL, 0, {2000, 48000}}, }; static uint8_t sv_read(struct sv_softc *, uint8_t); static uint8_t sv_read_indirect(struct sv_softc *, uint8_t); static void sv_write(struct sv_softc *, uint8_t, uint8_t); static void sv_write_indirect(struct sv_softc *, uint8_t, uint8_t); static void sv_init_mixer(struct sv_softc *); static void sv_defer(struct device *self); static void sv_write(struct sv_softc *sc, uint8_t reg, uint8_t val) { DPRINTFN(8,("sv_write(0x%x, 0x%x)\n", reg, val)); bus_space_write_1(sc->sc_iot, sc->sc_ioh, reg, val); } static uint8_t sv_read(struct sv_softc *sc, uint8_t reg) { uint8_t val; val = bus_space_read_1(sc->sc_iot, sc->sc_ioh, reg); DPRINTFN(8,("sv_read(0x%x) = 0x%x\n", reg, val)); return val; } static uint8_t sv_read_indirect(struct sv_softc *sc, uint8_t reg) { uint8_t val; int s; s = splaudio(); sv_write(sc, SV_CODEC_IADDR, reg & SV_IADDR_MASK); val = sv_read(sc, SV_CODEC_IDATA); splx(s); return val; } static void sv_write_indirect(struct sv_softc *sc, uint8_t reg, uint8_t val) { uint8_t iaddr; int s; iaddr = reg & SV_IADDR_MASK; s = splaudio(); if (reg == SV_DMA_DATA_FORMAT) iaddr |= SV_IADDR_MCE; sv_write(sc, SV_CODEC_IADDR, iaddr); sv_write(sc, SV_CODEC_IDATA, val); splx(s); } int sv_match(struct device *parent, struct cfdata *match, void *aux) { struct pci_attach_args *pa; pa = aux; if (PCI_VENDOR(pa->pa_id) == PCI_VENDOR_S3 && PCI_PRODUCT(pa->pa_id) == PCI_PRODUCT_S3_SONICVIBES) return 1; return 0; } int pci_alloc_io(pci_chipset_tag_t, pcitag_t, int, bus_space_tag_t, bus_size_t, bus_size_t, bus_size_t, int, bus_space_handle_t *); static pcireg_t pci_io_alloc_low, pci_io_alloc_high; int pci_alloc_io(pci_chipset_tag_t pc, pcitag_t pt, int pcioffs, bus_space_tag_t iot, bus_size_t size, bus_size_t align, bus_size_t bound, int flags, bus_space_handle_t *ioh) { bus_addr_t addr; int error; error = bus_space_alloc(iot, pci_io_alloc_low, pci_io_alloc_high, size, align, bound, flags, &addr, ioh); if (error) return error; pci_conf_write(pc, pt, pcioffs, addr); return 0; } /* * Allocate IO addresses when all other configuration is done. */ void sv_defer(struct device *self) { struct sv_softc *sc; pci_chipset_tag_t pc; pcitag_t pt; pcireg_t dmaio; sc = (struct sv_softc *)self; pc = sc->sc_pa.pa_pc; pt = sc->sc_pa.pa_tag; DPRINTF(("sv_defer: %p\n", sc)); /* XXX * Get a reasonable default for the I/O range. * Assume the range around SB_PORTBASE is valid on this PCI bus. */ pci_io_alloc_low = pci_conf_read(pc, pt, SV_SB_PORTBASE_SLOT); pci_io_alloc_high = pci_io_alloc_low + 0x1000; if (pci_alloc_io(pc, pt, SV_DMAA_CONFIG_OFF, sc->sc_iot, SV_DMAA_SIZE, SV_DMAA_ALIGN, 0, 0, &sc->sc_dmaa_ioh)) { printf("sv_attach: cannot allocate DMA A range\n"); return; } dmaio = pci_conf_read(pc, pt, SV_DMAA_CONFIG_OFF); DPRINTF(("sv_attach: addr a dmaio=0x%lx\n", (u_long)dmaio)); pci_conf_write(pc, pt, SV_DMAA_CONFIG_OFF, dmaio | SV_DMA_CHANNEL_ENABLE | SV_DMAA_EXTENDED_ADDR); if (pci_alloc_io(pc, pt, SV_DMAC_CONFIG_OFF, sc->sc_iot, SV_DMAC_SIZE, SV_DMAC_ALIGN, 0, 0, &sc->sc_dmac_ioh)) { printf("sv_attach: cannot allocate DMA C range\n"); return; } dmaio = pci_conf_read(pc, pt, SV_DMAC_CONFIG_OFF); DPRINTF(("sv_attach: addr c dmaio=0x%lx\n", (u_long)dmaio)); pci_conf_write(pc, pt, SV_DMAC_CONFIG_OFF, dmaio | SV_DMA_CHANNEL_ENABLE); sc->sc_dmaset = 1; } void sv_attach(struct device *parent, struct device *self, void *aux) { struct sv_softc *sc; struct pci_attach_args *pa; pci_chipset_tag_t pc; pcitag_t pt; pci_intr_handle_t ih; pcireg_t csr; char const *intrstr; uint8_t reg; struct audio_attach_args arg; sc = (struct sv_softc *)self; pa = aux; pc = pa->pa_pc; pt = pa->pa_tag; printf ("\n"); /* Map I/O registers */ if (pci_mapreg_map(pa, SV_ENHANCED_PORTBASE_SLOT, PCI_MAPREG_TYPE_IO, 0, &sc->sc_iot, &sc->sc_ioh, NULL, NULL)) { printf("%s: can't map enhanced i/o space\n", sc->sc_dev.dv_xname); return; } if (pci_mapreg_map(pa, SV_FM_PORTBASE_SLOT, PCI_MAPREG_TYPE_IO, 0, &sc->sc_opliot, &sc->sc_oplioh, NULL, NULL)) { printf("%s: can't map FM i/o space\n", sc->sc_dev.dv_xname); return; } if (pci_mapreg_map(pa, SV_MIDI_PORTBASE_SLOT, PCI_MAPREG_TYPE_IO, 0, &sc->sc_midiiot, &sc->sc_midiioh, NULL, NULL)) { printf("%s: can't map MIDI i/o space\n", sc->sc_dev.dv_xname); return; } DPRINTF(("sv: IO ports: enhanced=0x%x, OPL=0x%x, MIDI=0x%x\n", (int)sc->sc_ioh, (int)sc->sc_oplioh, (int)sc->sc_midiioh)); #if defined(alpha) /* XXX Force allocation through the SGMAP. */ sc->sc_dmatag = alphabus_dma_get_tag(pa->pa_dmat, ALPHA_BUS_ISA); #elif defined(i386) && NISA > 0 /* XXX * The SonicVibes DMA is broken and only works on 24-bit addresses. * As long as bus_dmamem_alloc_range() is missing we use the ISA * DMA tag on i386. */ sc->sc_dmatag = &isa_bus_dma_tag; #else sc->sc_dmatag = pa->pa_dmat; #endif pci_conf_write(pc, pt, SV_DMAA_CONFIG_OFF, SV_DMAA_EXTENDED_ADDR); pci_conf_write(pc, pt, SV_DMAC_CONFIG_OFF, 0); /* Enable the device. */ csr = pci_conf_read(pc, pt, PCI_COMMAND_STATUS_REG); pci_conf_write(pc, pt, PCI_COMMAND_STATUS_REG, csr | PCI_COMMAND_MASTER_ENABLE); sv_write_indirect(sc, SV_ANALOG_POWER_DOWN_CONTROL, 0); sv_write_indirect(sc, SV_DIGITAL_POWER_DOWN_CONTROL, 0); /* initialize codec registers */ reg = sv_read(sc, SV_CODEC_CONTROL); reg |= SV_CTL_RESET; sv_write(sc, SV_CODEC_CONTROL, reg); delay(50); reg = sv_read(sc, SV_CODEC_CONTROL); reg &= ~SV_CTL_RESET; reg |= SV_CTL_INTA | SV_CTL_ENHANCED; /* This write clears the reset */ sv_write(sc, SV_CODEC_CONTROL, reg); delay(50); /* This write actually shoves the new values in */ sv_write(sc, SV_CODEC_CONTROL, reg); DPRINTF(("sv_attach: control=0x%x\n", sv_read(sc, SV_CODEC_CONTROL))); /* Enable DMA interrupts */ reg = sv_read(sc, SV_CODEC_INTMASK); reg &= ~(SV_INTMASK_DMAA | SV_INTMASK_DMAC); reg |= SV_INTMASK_UD | SV_INTMASK_SINT | SV_INTMASK_MIDI; sv_write(sc, SV_CODEC_INTMASK, reg); sv_read(sc, SV_CODEC_STATUS); /* Map and establish the interrupt. */ if (pci_intr_map(pa, &ih)) { printf("%s: couldn't map interrupt\n", sc->sc_dev.dv_xname); return; } intrstr = pci_intr_string(pc, ih); sc->sc_ih = pci_intr_establish(pc, ih, IPL_AUDIO, sv_intr, sc); if (sc->sc_ih == NULL) { printf("%s: couldn't establish interrupt", sc->sc_dev.dv_xname); if (intrstr != NULL) printf(" at %s", intrstr); printf("\n"); return; } printf("%s: interrupting at %s\n", sc->sc_dev.dv_xname, intrstr); printf("%s: rev %d", sc->sc_dev.dv_xname, sv_read_indirect(sc, SV_REVISION_LEVEL)); if (sv_read(sc, SV_CODEC_CONTROL) & SV_CTL_MD1) printf(", reverb SRAM present"); if (!(sv_read_indirect(sc, SV_WAVETABLE_SOURCE_SELECT) & SV_WSS_WT0)) printf(", wavetable ROM present"); printf("\n"); sv_init_mixer(sc); audio_attach_mi(&sv_hw_if, sc, &sc->sc_dev); arg.type = AUDIODEV_TYPE_OPL; arg.hwif = 0; arg.hdl = 0; (void)config_found(&sc->sc_dev, &arg, audioprint); sc->sc_pa = *pa; /* for deferred setup */ config_defer(self, sv_defer); } #ifdef AUDIO_DEBUG void sv_dumpregs(struct sv_softc *sc) { int idx; #if 0 for (idx = 0; idx < 0x50; idx += 4) printf ("%02x = %x\n", idx, pci_conf_read(pa->pa_pc, pa->pa_tag, idx)); #endif for (idx = 0; idx < 6; idx++) printf ("REG %02x = %02x\n", idx, sv_read(sc, idx)); for (idx = 0; idx < 0x32; idx++) printf ("IREG %02x = %02x\n", idx, sv_read_indirect(sc, idx)); for (idx = 0; idx < 0x10; idx++) printf ("DMA %02x = %02x\n", idx, bus_space_read_1(sc->sc_iot, sc->sc_dmaa_ioh, idx)); } #endif int sv_intr(void *p) { struct sv_softc *sc; uint8_t intr; sc = p; intr = sv_read(sc, SV_CODEC_STATUS); DPRINTFN(5,("sv_intr: intr=0x%x\n", intr)); if (!(intr & (SV_INTSTATUS_DMAA | SV_INTSTATUS_DMAC))) return 0; if (intr & SV_INTSTATUS_DMAA) { if (sc->sc_pintr) sc->sc_pintr(sc->sc_parg); } if (intr & SV_INTSTATUS_DMAC) { if (sc->sc_rintr) sc->sc_rintr(sc->sc_rarg); } return 1; } int sv_allocmem(struct sv_softc *sc, size_t size, size_t align, int direction, struct sv_dma *p) { int error; p->size = size; error = bus_dmamem_alloc(sc->sc_dmatag, p->size, align, 0, p->segs, ARRAY_SIZE(p->segs), &p->nsegs, BUS_DMA_NOWAIT); if (error) return error; error = bus_dmamem_map(sc->sc_dmatag, p->segs, p->nsegs, p->size, &p->addr, BUS_DMA_NOWAIT|BUS_DMA_COHERENT); if (error) goto free; error = bus_dmamap_create(sc->sc_dmatag, p->size, 1, p->size, 0, BUS_DMA_NOWAIT, &p->map); if (error) goto unmap; error = bus_dmamap_load(sc->sc_dmatag, p->map, p->addr, p->size, NULL, BUS_DMA_NOWAIT | (direction == AUMODE_RECORD) ? BUS_DMA_READ : BUS_DMA_WRITE); if (error) goto destroy; DPRINTF(("sv_allocmem: pa=%lx va=%lx pba=%lx\n", (long)p->segs[0].ds_addr, (long)KERNADDR(p), (long)DMAADDR(p))); return 0; destroy: bus_dmamap_destroy(sc->sc_dmatag, p->map); unmap: bus_dmamem_unmap(sc->sc_dmatag, p->addr, p->size); free: bus_dmamem_free(sc->sc_dmatag, p->segs, p->nsegs); return error; } int sv_freemem(struct sv_softc *sc, struct sv_dma *p) { bus_dmamap_unload(sc->sc_dmatag, p->map); bus_dmamap_destroy(sc->sc_dmatag, p->map); bus_dmamem_unmap(sc->sc_dmatag, p->addr, p->size); bus_dmamem_free(sc->sc_dmatag, p->segs, p->nsegs); return 0; } int sv_open(void *addr, int flags) { struct sv_softc *sc; sc = addr; DPRINTF(("sv_open\n")); if (!sc->sc_dmaset) return ENXIO; return 0; } int sv_query_encoding(void *addr, struct audio_encoding *fp) { switch (fp->index) { case 0: strcpy(fp->name, AudioEulinear); fp->encoding = AUDIO_ENCODING_ULINEAR; fp->precision = 8; fp->flags = 0; return 0; case 1: strcpy(fp->name, AudioEmulaw); fp->encoding = AUDIO_ENCODING_ULAW; fp->precision = 8; fp->flags = AUDIO_ENCODINGFLAG_EMULATED; return 0; case 2: strcpy(fp->name, AudioEalaw); fp->encoding = AUDIO_ENCODING_ALAW; fp->precision = 8; fp->flags = AUDIO_ENCODINGFLAG_EMULATED; return 0; case 3: strcpy(fp->name, AudioEslinear); fp->encoding = AUDIO_ENCODING_SLINEAR; fp->precision = 8; fp->flags = AUDIO_ENCODINGFLAG_EMULATED; return 0; case 4: strcpy(fp->name, AudioEslinear_le); fp->encoding = AUDIO_ENCODING_SLINEAR_LE; fp->precision = 16; fp->flags = 0; return 0; case 5: strcpy(fp->name, AudioEulinear_le); fp->encoding = AUDIO_ENCODING_ULINEAR_LE; fp->precision = 16; fp->flags = AUDIO_ENCODINGFLAG_EMULATED; return 0; case 6: strcpy(fp->name, AudioEslinear_be); fp->encoding = AUDIO_ENCODING_SLINEAR_BE; fp->precision = 16; fp->flags = AUDIO_ENCODINGFLAG_EMULATED; return 0; case 7: strcpy(fp->name, AudioEulinear_be); fp->encoding = AUDIO_ENCODING_ULINEAR_BE; fp->precision = 16; fp->flags = AUDIO_ENCODINGFLAG_EMULATED; return 0; default: return EINVAL; } } int sv_set_params(void *addr, int setmode, int usemode, audio_params_t *play, audio_params_t *rec, stream_filter_list_t *pfil, stream_filter_list_t *rfil) { struct sv_softc *sc; audio_params_t *p; uint32_t val; sc = addr; p = NULL; /* * This device only has one clock, so make the sample rates match. */ if (play->sample_rate != rec->sample_rate && usemode == (AUMODE_PLAY | AUMODE_RECORD)) { if (setmode == AUMODE_PLAY) { rec->sample_rate = play->sample_rate; setmode |= AUMODE_RECORD; } else if (setmode == AUMODE_RECORD) { play->sample_rate = rec->sample_rate; setmode |= AUMODE_PLAY; } else return EINVAL; } if (setmode & AUMODE_RECORD) { p = rec; if (auconv_set_converter(sv_formats, SV_NFORMATS, AUMODE_RECORD, rec, FALSE, rfil) < 0) return EINVAL; } if (setmode & AUMODE_PLAY) { p = play; if (auconv_set_converter(sv_formats, SV_NFORMATS, AUMODE_PLAY, play, FALSE, pfil) < 0) return EINVAL; } if (p == NULL) return 0; val = p->sample_rate * 65536 / 48000; /* * If the sample rate is exactly 48KHz, the fraction would overflow the * register, so we have to bias it. This causes a little clock drift. * The drift is below normal crystal tolerance (.0001%), so although * this seems a little silly, we can pretty much ignore it. * (I tested the output speed with values of 1-20, just to be sure this * register isn't *supposed* to have a bias. It isn't.) * - mycroft */ if (val > 65535) val = 65535; sv_write_indirect(sc, SV_PCM_SAMPLE_RATE_0, val & 0xff); sv_write_indirect(sc, SV_PCM_SAMPLE_RATE_1, val >> 8); #define F_REF 24576000 #define ABS(x) (((x) < 0) ? (-x) : (x)) if (setmode & AUMODE_RECORD) { /* The ADC reference frequency (f_out) is 512 * sample rate */ /* f_out is dervied from the 24.576MHz crystal by three values: M & N & R. The equation is as follows: f_out = (m + 2) * f_ref / ((n + 2) * (2 ^ a)) with the constraint that: 80 MHz < (m + 2) / (n + 2) * f_ref <= 150MHz and n, m >= 1 */ int goal_f_out; int a, n, m, best_n, best_m, best_error; int pll_sample; int error; goal_f_out = 512 * rec->sample_rate; best_n = 0; best_m = 0; best_error = 10000000; for (a = 0; a < 8; a++) { if ((goal_f_out * (1 << a)) >= 80000000) break; } /* a != 8 because sample_rate >= 2000 */ for (n = 33; n > 2; n--) { m = (goal_f_out * n * (1 << a)) / F_REF; if ((m > 257) || (m < 3)) continue; pll_sample = (m * F_REF) / (n * (1 << a)); pll_sample /= 512; /* Threshold might be good here */ error = pll_sample - rec->sample_rate; error = ABS(error); if (error < best_error) { best_error = error; best_n = n; best_m = m; if (error == 0) break; } } best_n -= 2; best_m -= 2; sv_write_indirect(sc, SV_ADC_PLL_M, best_m); sv_write_indirect(sc, SV_ADC_PLL_N, best_n | (a << SV_PLL_R_SHIFT)); } return 0; } int sv_round_blocksize(void *addr, int blk, int mode, const audio_params_t *param) { return blk & -32; /* keep good alignment */ } int sv_trigger_output(void *addr, void *start, void *end, int blksize, void (*intr)(void *), void *arg, const audio_params_t *param) { struct sv_softc *sc; struct sv_dma *p; uint8_t mode; int dma_count; DPRINTFN(1, ("sv_trigger_output: sc=%p start=%p end=%p blksize=%d " "intr=%p(%p)\n", addr, start, end, blksize, intr, arg)); sc = addr; sc->sc_pintr = intr; sc->sc_parg = arg; mode = sv_read_indirect(sc, SV_DMA_DATA_FORMAT); mode &= ~(SV_DMAA_FORMAT16 | SV_DMAA_STEREO); if (param->precision == 16) mode |= SV_DMAA_FORMAT16; if (param->channels == 2) mode |= SV_DMAA_STEREO; sv_write_indirect(sc, SV_DMA_DATA_FORMAT, mode); for (p = sc->sc_dmas; p && KERNADDR(p) != start; p = p->next) continue; if (p == NULL) { printf("sv_trigger_output: bad addr %p\n", start); return EINVAL; } dma_count = ((char *)end - (char *)start) - 1; DPRINTF(("sv_trigger_output: DMA start loop input addr=%x cc=%d\n", (int)DMAADDR(p), dma_count)); bus_space_write_4(sc->sc_iot, sc->sc_dmaa_ioh, SV_DMA_ADDR0, DMAADDR(p)); bus_space_write_4(sc->sc_iot, sc->sc_dmaa_ioh, SV_DMA_COUNT0, dma_count); bus_space_write_1(sc->sc_iot, sc->sc_dmaa_ioh, SV_DMA_MODE, DMA37MD_READ | DMA37MD_LOOP); DPRINTF(("sv_trigger_output: current addr=%x\n", bus_space_read_4(sc->sc_iot, sc->sc_dmaa_ioh, SV_DMA_ADDR0))); dma_count = blksize - 1; sv_write_indirect(sc, SV_DMAA_COUNT1, dma_count >> 8); sv_write_indirect(sc, SV_DMAA_COUNT0, dma_count & 0xFF); mode = sv_read_indirect(sc, SV_PLAY_RECORD_ENABLE); sv_write_indirect(sc, SV_PLAY_RECORD_ENABLE, mode | SV_PLAY_ENABLE); return 0; } int sv_trigger_input(void *addr, void *start, void *end, int blksize, void (*intr)(void *), void *arg, const audio_params_t *param) { struct sv_softc *sc; struct sv_dma *p; uint8_t mode; int dma_count; DPRINTFN(1, ("sv_trigger_input: sc=%p start=%p end=%p blksize=%d " "intr=%p(%p)\n", addr, start, end, blksize, intr, arg)); sc = addr; sc->sc_rintr = intr; sc->sc_rarg = arg; mode = sv_read_indirect(sc, SV_DMA_DATA_FORMAT); mode &= ~(SV_DMAC_FORMAT16 | SV_DMAC_STEREO); if (param->precision == 16) mode |= SV_DMAC_FORMAT16; if (param->channels == 2) mode |= SV_DMAC_STEREO; sv_write_indirect(sc, SV_DMA_DATA_FORMAT, mode); for (p = sc->sc_dmas; p && KERNADDR(p) != start; p = p->next) continue; if (!p) { printf("sv_trigger_input: bad addr %p\n", start); return EINVAL; } dma_count = (((char *)end - (char *)start) >> 1) - 1; DPRINTF(("sv_trigger_input: DMA start loop input addr=%x cc=%d\n", (int)DMAADDR(p), dma_count)); bus_space_write_4(sc->sc_iot, sc->sc_dmac_ioh, SV_DMA_ADDR0, DMAADDR(p)); bus_space_write_4(sc->sc_iot, sc->sc_dmac_ioh, SV_DMA_COUNT0, dma_count); bus_space_write_1(sc->sc_iot, sc->sc_dmac_ioh, SV_DMA_MODE, DMA37MD_WRITE | DMA37MD_LOOP); DPRINTF(("sv_trigger_input: current addr=%x\n", bus_space_read_4(sc->sc_iot, sc->sc_dmac_ioh, SV_DMA_ADDR0))); dma_count = (blksize >> 1) - 1; sv_write_indirect(sc, SV_DMAC_COUNT1, dma_count >> 8); sv_write_indirect(sc, SV_DMAC_COUNT0, dma_count & 0xFF); mode = sv_read_indirect(sc, SV_PLAY_RECORD_ENABLE); sv_write_indirect(sc, SV_PLAY_RECORD_ENABLE, mode | SV_RECORD_ENABLE); return 0; } int sv_halt_output(void *addr) { struct sv_softc *sc; uint8_t mode; DPRINTF(("sv: sv_halt_output\n")); sc = addr; mode = sv_read_indirect(sc, SV_PLAY_RECORD_ENABLE); sv_write_indirect(sc, SV_PLAY_RECORD_ENABLE, mode & ~SV_PLAY_ENABLE); sc->sc_pintr = 0; return 0; } int sv_halt_input(void *addr) { struct sv_softc *sc; uint8_t mode; DPRINTF(("sv: sv_halt_input\n")); sc = addr; mode = sv_read_indirect(sc, SV_PLAY_RECORD_ENABLE); sv_write_indirect(sc, SV_PLAY_RECORD_ENABLE, mode & ~SV_RECORD_ENABLE); sc->sc_rintr = 0; return 0; } int sv_getdev(void *addr, struct audio_device *retp) { *retp = sv_device; return 0; } /* * Mixer related code is here * */ #define SV_INPUT_CLASS 0 #define SV_OUTPUT_CLASS 1 #define SV_RECORD_CLASS 2 #define SV_LAST_CLASS 2 static const char *mixer_classes[] = { AudioCinputs, AudioCoutputs, AudioCrecord }; static const struct { uint8_t l_port; uint8_t r_port; uint8_t mask; uint8_t class; const char *audio; } ports[] = { { SV_LEFT_AUX1_INPUT_CONTROL, SV_RIGHT_AUX1_INPUT_CONTROL, SV_AUX1_MASK, SV_INPUT_CLASS, "aux1" }, { SV_LEFT_CD_INPUT_CONTROL, SV_RIGHT_CD_INPUT_CONTROL, SV_CD_MASK, SV_INPUT_CLASS, AudioNcd }, { SV_LEFT_LINE_IN_INPUT_CONTROL, SV_RIGHT_LINE_IN_INPUT_CONTROL, SV_LINE_IN_MASK, SV_INPUT_CLASS, AudioNline }, { SV_MIC_INPUT_CONTROL, 0, SV_MIC_MASK, SV_INPUT_CLASS, AudioNmicrophone }, { SV_LEFT_SYNTH_INPUT_CONTROL, SV_RIGHT_SYNTH_INPUT_CONTROL, SV_SYNTH_MASK, SV_INPUT_CLASS, AudioNfmsynth }, { SV_LEFT_AUX2_INPUT_CONTROL, SV_RIGHT_AUX2_INPUT_CONTROL, SV_AUX2_MASK, SV_INPUT_CLASS, "aux2" }, { SV_LEFT_PCM_INPUT_CONTROL, SV_RIGHT_PCM_INPUT_CONTROL, SV_PCM_MASK, SV_INPUT_CLASS, AudioNdac }, { SV_LEFT_MIXER_OUTPUT_CONTROL, SV_RIGHT_MIXER_OUTPUT_CONTROL, SV_MIXER_OUT_MASK, SV_OUTPUT_CLASS, AudioNmaster } }; static const struct { int idx; const char *name; } record_sources[] = { { SV_REC_CD, AudioNcd }, { SV_REC_DAC, AudioNdac }, { SV_REC_AUX2, "aux2" }, { SV_REC_LINE, AudioNline }, { SV_REC_AUX1, "aux1" }, { SV_REC_MIC, AudioNmicrophone }, { SV_REC_MIXER, AudioNmixerout } }; #define SV_DEVICES_PER_PORT 2 #define SV_FIRST_MIXER (SV_LAST_CLASS + 1) #define SV_LAST_MIXER (SV_DEVICES_PER_PORT * (ARRAY_SIZE(ports)) + SV_LAST_CLASS) #define SV_RECORD_SOURCE (SV_LAST_MIXER + 1) #define SV_MIC_BOOST (SV_LAST_MIXER + 2) #define SV_RECORD_GAIN (SV_LAST_MIXER + 3) #define SV_SRS_MODE (SV_LAST_MIXER + 4) int sv_query_devinfo(void *addr, mixer_devinfo_t *dip) { int i; /* It's a class */ if (dip->index <= SV_LAST_CLASS) { dip->type = AUDIO_MIXER_CLASS; dip->mixer_class = dip->index; dip->next = dip->prev = AUDIO_MIXER_LAST; strcpy(dip->label.name, mixer_classes[dip->index]); return 0; } if (dip->index >= SV_FIRST_MIXER && dip->index <= SV_LAST_MIXER) { int off, mute ,idx; off = dip->index - SV_FIRST_MIXER; mute = (off % SV_DEVICES_PER_PORT); idx = off / SV_DEVICES_PER_PORT; dip->mixer_class = ports[idx].class; strcpy(dip->label.name, ports[idx].audio); if (!mute) { dip->type = AUDIO_MIXER_VALUE; dip->prev = AUDIO_MIXER_LAST; dip->next = dip->index + 1; if (ports[idx].r_port != 0) dip->un.v.num_channels = 2; else dip->un.v.num_channels = 1; strcpy(dip->un.v.units.name, AudioNvolume); } else { dip->type = AUDIO_MIXER_ENUM; dip->prev = dip->index - 1; 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; } return 0; } switch (dip->index) { case SV_RECORD_SOURCE: dip->mixer_class = SV_RECORD_CLASS; dip->prev = AUDIO_MIXER_LAST; dip->next = SV_RECORD_GAIN; strcpy(dip->label.name, AudioNsource); dip->type = AUDIO_MIXER_ENUM; dip->un.e.num_mem = ARRAY_SIZE(record_sources); for (i = 0; i < ARRAY_SIZE(record_sources); i++) { strcpy(dip->un.e.member[i].label.name, record_sources[i].name); dip->un.e.member[i].ord = record_sources[i].idx; } return 0; case SV_RECORD_GAIN: dip->mixer_class = SV_RECORD_CLASS; dip->prev = SV_RECORD_SOURCE; dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, "gain"); dip->type = AUDIO_MIXER_VALUE; dip->un.v.num_channels = 1; strcpy(dip->un.v.units.name, AudioNvolume); return 0; case SV_MIC_BOOST: dip->mixer_class = SV_RECORD_CLASS; dip->prev = AUDIO_MIXER_LAST; dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, "micboost"); goto on_off; case SV_SRS_MODE: dip->mixer_class = SV_OUTPUT_CLASS; dip->prev = dip->next = AUDIO_MIXER_LAST; strcpy(dip->label.name, AudioNspatial); on_off: dip->type = AUDIO_MIXER_ENUM; 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; return 0; } return ENXIO; } int sv_mixer_set_port(void *addr, mixer_ctrl_t *cp) { struct sv_softc *sc; uint8_t reg; int idx; sc = addr; if (cp->dev >= SV_FIRST_MIXER && cp->dev <= SV_LAST_MIXER) { int off, mute; off = cp->dev - SV_FIRST_MIXER; mute = (off % SV_DEVICES_PER_PORT); idx = off / SV_DEVICES_PER_PORT; if (mute) { if (cp->type != AUDIO_MIXER_ENUM) return EINVAL; reg = sv_read_indirect(sc, ports[idx].l_port); if (cp->un.ord) reg |= SV_MUTE_BIT; else reg &= ~SV_MUTE_BIT; sv_write_indirect(sc, ports[idx].l_port, reg); if (ports[idx].r_port) { reg = sv_read_indirect(sc, ports[idx].r_port); if (cp->un.ord) reg |= SV_MUTE_BIT; else reg &= ~SV_MUTE_BIT; sv_write_indirect(sc, ports[idx].r_port, reg); } } else { int lval, rval; if (cp->type != AUDIO_MIXER_VALUE) return EINVAL; if (cp->un.value.num_channels != 1 && cp->un.value.num_channels != 2) return (EINVAL); if (ports[idx].r_port == 0) { if (cp->un.value.num_channels != 1) return (EINVAL); lval = cp->un.value.level[AUDIO_MIXER_LEVEL_MONO]; rval = 0; /* shut up GCC */ } else { if (cp->un.value.num_channels != 2) return (EINVAL); lval = cp->un.value.level[AUDIO_MIXER_LEVEL_LEFT]; rval = cp->un.value.level[AUDIO_MIXER_LEVEL_RIGHT]; } reg = sv_read_indirect(sc, ports[idx].l_port); reg &= ~(ports[idx].mask); lval = (AUDIO_MAX_GAIN - lval) * ports[idx].mask / AUDIO_MAX_GAIN; reg |= lval; sv_write_indirect(sc, ports[idx].l_port, reg); if (ports[idx].r_port != 0) { reg = sv_read_indirect(sc, ports[idx].r_port); reg &= ~(ports[idx].mask); rval = (AUDIO_MAX_GAIN - rval) * ports[idx].mask / AUDIO_MAX_GAIN; reg |= rval; sv_write_indirect(sc, ports[idx].r_port, reg); } sv_read_indirect(sc, ports[idx].l_port); } return 0; } switch (cp->dev) { case SV_RECORD_SOURCE: if (cp->type != AUDIO_MIXER_ENUM) return EINVAL; for (idx = 0; idx < ARRAY_SIZE(record_sources); idx++) { if (record_sources[idx].idx == cp->un.ord) goto found; } return EINVAL; found: reg = sv_read_indirect(sc, SV_LEFT_ADC_INPUT_CONTROL); reg &= ~SV_REC_SOURCE_MASK; reg |= (((cp->un.ord) << SV_REC_SOURCE_SHIFT) & SV_REC_SOURCE_MASK); sv_write_indirect(sc, SV_LEFT_ADC_INPUT_CONTROL, reg); reg = sv_read_indirect(sc, SV_RIGHT_ADC_INPUT_CONTROL); reg &= ~SV_REC_SOURCE_MASK; reg |= (((cp->un.ord) << SV_REC_SOURCE_SHIFT) & SV_REC_SOURCE_MASK); sv_write_indirect(sc, SV_RIGHT_ADC_INPUT_CONTROL, reg); return 0; case SV_RECORD_GAIN: { int val; if (cp->type != AUDIO_MIXER_VALUE) return EINVAL; if (cp->un.value.num_channels != 1) return EINVAL; val = (cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] * SV_REC_GAIN_MASK) / AUDIO_MAX_GAIN; reg = sv_read_indirect(sc, SV_LEFT_ADC_INPUT_CONTROL); reg &= ~SV_REC_GAIN_MASK; reg |= val; sv_write_indirect(sc, SV_LEFT_ADC_INPUT_CONTROL, reg); reg = sv_read_indirect(sc, SV_RIGHT_ADC_INPUT_CONTROL); reg &= ~SV_REC_GAIN_MASK; reg |= val; sv_write_indirect(sc, SV_RIGHT_ADC_INPUT_CONTROL, reg); } return (0); case SV_MIC_BOOST: if (cp->type != AUDIO_MIXER_ENUM) return EINVAL; reg = sv_read_indirect(sc, SV_LEFT_ADC_INPUT_CONTROL); if (cp->un.ord) { reg |= SV_MIC_BOOST_BIT; } else { reg &= ~SV_MIC_BOOST_BIT; } sv_write_indirect(sc, SV_LEFT_ADC_INPUT_CONTROL, reg); return 0; case SV_SRS_MODE: if (cp->type != AUDIO_MIXER_ENUM) return EINVAL; reg = sv_read_indirect(sc, SV_SRS_SPACE_CONTROL); if (cp->un.ord) { reg &= ~SV_SRS_SPACE_ONOFF; } else { reg |= SV_SRS_SPACE_ONOFF; } sv_write_indirect(sc, SV_SRS_SPACE_CONTROL, reg); return 0; } return EINVAL; } int sv_mixer_get_port(void *addr, mixer_ctrl_t *cp) { struct sv_softc *sc; int val; uint8_t reg; sc = addr; if (cp->dev >= SV_FIRST_MIXER && cp->dev <= SV_LAST_MIXER) { int off = cp->dev - SV_FIRST_MIXER; int mute = (off % 2); int idx = off / 2; off = cp->dev - SV_FIRST_MIXER; mute = (off % 2); idx = off / 2; if (mute) { if (cp->type != AUDIO_MIXER_ENUM) return EINVAL; reg = sv_read_indirect(sc, ports[idx].l_port); cp->un.ord = ((reg & SV_MUTE_BIT) ? 1 : 0); } else { if (cp->type != AUDIO_MIXER_VALUE) return EINVAL; if (cp->un.value.num_channels != 1 && cp->un.value.num_channels != 2) return EINVAL; if ((ports[idx].r_port == 0 && cp->un.value.num_channels != 1) || (ports[idx].r_port != 0 && cp->un.value.num_channels != 2)) return EINVAL; reg = sv_read_indirect(sc, ports[idx].l_port); reg &= ports[idx].mask; val = AUDIO_MAX_GAIN - ((reg * AUDIO_MAX_GAIN) / ports[idx].mask); if (ports[idx].r_port != 0) { cp->un.value.level[AUDIO_MIXER_LEVEL_LEFT] = val; reg = sv_read_indirect(sc, ports[idx].r_port); reg &= ports[idx].mask; val = AUDIO_MAX_GAIN - ((reg * AUDIO_MAX_GAIN) / ports[idx].mask); cp->un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = val; } else cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] = val; } return 0; } switch (cp->dev) { case SV_RECORD_SOURCE: if (cp->type != AUDIO_MIXER_ENUM) return EINVAL; reg = sv_read_indirect(sc, SV_LEFT_ADC_INPUT_CONTROL); cp->un.ord = ((reg & SV_REC_SOURCE_MASK) >> SV_REC_SOURCE_SHIFT); return 0; case SV_RECORD_GAIN: if (cp->type != AUDIO_MIXER_VALUE) return EINVAL; if (cp->un.value.num_channels != 1) return EINVAL; reg = sv_read_indirect(sc, SV_LEFT_ADC_INPUT_CONTROL) & SV_REC_GAIN_MASK; cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] = (((unsigned int)reg) * AUDIO_MAX_GAIN) / SV_REC_GAIN_MASK; return 0; case SV_MIC_BOOST: if (cp->type != AUDIO_MIXER_ENUM) return EINVAL; reg = sv_read_indirect(sc, SV_LEFT_ADC_INPUT_CONTROL); cp->un.ord = ((reg & SV_MIC_BOOST_BIT) ? 1 : 0); return 0; case SV_SRS_MODE: if (cp->type != AUDIO_MIXER_ENUM) return EINVAL; reg = sv_read_indirect(sc, SV_SRS_SPACE_CONTROL); cp->un.ord = ((reg & SV_SRS_SPACE_ONOFF) ? 0 : 1); return 0; } return EINVAL; } static void sv_init_mixer(struct sv_softc *sc) { mixer_ctrl_t cp; int i; cp.type = AUDIO_MIXER_ENUM; cp.dev = SV_SRS_MODE; cp.un.ord = 0; sv_mixer_set_port(sc, &cp); for (i = 0; i < ARRAY_SIZE(ports); i++) { if (ports[i].audio == AudioNdac) { cp.type = AUDIO_MIXER_ENUM; cp.dev = SV_FIRST_MIXER + i * SV_DEVICES_PER_PORT + 1; cp.un.ord = 0; sv_mixer_set_port(sc, &cp); break; } } } void * sv_malloc(void *addr, int direction, size_t size, struct malloc_type *pool, int flags) { struct sv_softc *sc; struct sv_dma *p; int error; sc = addr; p = malloc(sizeof(*p), pool, flags); if (p == NULL) return NULL; error = sv_allocmem(sc, size, 16, direction, p); if (error) { free(p, pool); return 0; } p->next = sc->sc_dmas; sc->sc_dmas = p; return KERNADDR(p); } void sv_free(void *addr, void *ptr, struct malloc_type *pool) { struct sv_softc *sc; struct sv_dma **pp, *p; sc = addr; for (pp = &sc->sc_dmas; (p = *pp) != NULL; pp = &p->next) { if (KERNADDR(p) == ptr) { sv_freemem(sc, p); *pp = p->next; free(p, pool); return; } } } size_t sv_round_buffersize(void *addr, int direction, size_t size) { return size; } paddr_t sv_mappage(void *addr, void *mem, off_t off, int prot) { struct sv_softc *sc; struct sv_dma *p; sc = addr; if (off < 0) return -1; for (p = sc->sc_dmas; p && KERNADDR(p) != mem; p = p->next) continue; if (p == NULL) return -1; return bus_dmamem_mmap(sc->sc_dmatag, p->segs, p->nsegs, off, prot, BUS_DMA_WAITOK); } int sv_get_props(void *addr) { return AUDIO_PROP_MMAP | AUDIO_PROP_INDEPENDENT | AUDIO_PROP_FULLDUPLEX; }