e3faadaafe
For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
752 lines
18 KiB
C
752 lines
18 KiB
C
/*-
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* Copyright 2008 by Marco Trillo. 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 ``AS IS'' AND ANY EXPRESS OR
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* IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
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* OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED.
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* IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
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* INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING,
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* BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
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* LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED
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* AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
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* 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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* $FreeBSD$
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*/
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/*-
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* Copyright (c) 2002, 2003 Tsubai Masanari. 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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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
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* IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
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* OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED.
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* IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
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* INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
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* NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
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* DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
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* THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
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* (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
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* OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*
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* NetBSD: snapper.c,v 1.28 2008/05/16 03:49:54 macallan Exp
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* Id: snapper.c,v 1.11 2002/10/31 17:42:13 tsubai Exp
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*/
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/*
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* Apple I2S audio controller.
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*/
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#include <sys/param.h>
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#include <sys/systm.h>
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#include <sys/kernel.h>
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#include <sys/module.h>
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#include <sys/bus.h>
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#include <sys/malloc.h>
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#include <sys/lock.h>
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#include <sys/mutex.h>
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#include <machine/dbdma.h>
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#include <machine/intr_machdep.h>
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#include <machine/resource.h>
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#include <machine/bus.h>
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#include <machine/pio.h>
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#include <sys/rman.h>
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#include <dev/ofw/ofw_bus.h>
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#ifdef HAVE_KERNEL_OPTION_HEADERS
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#include "opt_snd.h"
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#endif
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#include <dev/sound/pcm/sound.h>
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#include <dev/sound/macio/aoa.h>
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#include <powerpc/powermac/macgpiovar.h>
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struct i2s_softc {
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struct aoa_softc aoa;
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phandle_t node;
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phandle_t soundnode;
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struct resource *reg;
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u_int output_mask;
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struct mtx port_mtx;
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};
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static int i2s_probe(device_t);
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static int i2s_attach(device_t);
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static void i2s_postattach(void *);
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static int i2s_setup(struct i2s_softc *, u_int, u_int, u_int);
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static void i2s_mute_headphone (struct i2s_softc *, int);
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static void i2s_mute_lineout (struct i2s_softc *, int);
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static void i2s_mute_speaker (struct i2s_softc *, int);
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static void i2s_set_outputs(void *, u_int);
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static struct intr_config_hook *i2s_delayed_attach = NULL;
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kobj_class_t i2s_mixer_class = NULL;
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device_t i2s_mixer = NULL;
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static device_method_t pcm_i2s_methods[] = {
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/* Device interface. */
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DEVMETHOD(device_probe, i2s_probe),
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DEVMETHOD(device_attach, i2s_attach),
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{ 0, 0 }
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};
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static driver_t pcm_i2s_driver = {
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"pcm",
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pcm_i2s_methods,
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PCM_SOFTC_SIZE
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};
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DRIVER_MODULE(pcm_i2s, macio, pcm_i2s_driver, pcm_devclass, 0, 0);
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MODULE_DEPEND(pcm_i2s, sound, SOUND_MINVER, SOUND_PREFVER, SOUND_MAXVER);
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static int aoagpio_probe(device_t);
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static int aoagpio_attach(device_t);
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static device_method_t aoagpio_methods[] = {
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/* Device interface. */
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DEVMETHOD(device_probe, aoagpio_probe),
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DEVMETHOD(device_attach, aoagpio_attach),
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{ 0, 0 }
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};
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struct aoagpio_softc {
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device_t dev;
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int ctrl;
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int detect_active; /* for extint-gpio */
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int level; /* for extint-gpio */
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struct i2s_softc *i2s; /* for extint-gpio */
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};
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static driver_t aoagpio_driver = {
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"aoagpio",
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aoagpio_methods,
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sizeof(struct aoagpio_softc)
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};
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static devclass_t aoagpio_devclass;
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DRIVER_MODULE(aoagpio, macgpio, aoagpio_driver, aoagpio_devclass, 0, 0);
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/*****************************************************************************
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Probe and attachment routines.
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*****************************************************************************/
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static int
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i2s_probe(device_t self)
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{
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const char *name;
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name = ofw_bus_get_name(self);
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if (!name)
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return (ENXIO);
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if (strcmp(name, "i2s") != 0)
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return (ENXIO);
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device_set_desc(self, "Apple I2S Audio Controller");
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return (0);
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}
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static phandle_t of_find_firstchild_byname(phandle_t, const char *);
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static int
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i2s_attach(device_t self)
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{
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struct i2s_softc *sc;
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struct resource *dbdma_irq;
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void *dbdma_ih;
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int rid, oirq, err;
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phandle_t port;
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sc = malloc(sizeof(*sc), M_DEVBUF, M_WAITOK | M_ZERO);
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sc->aoa.sc_dev = self;
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sc->node = ofw_bus_get_node(self);
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port = of_find_firstchild_byname(sc->node, "i2s-a");
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if (port == -1)
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return (ENXIO);
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sc->soundnode = of_find_firstchild_byname(port, "sound");
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if (sc->soundnode == -1)
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return (ENXIO);
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mtx_init(&sc->port_mtx, "port_mtx", NULL, MTX_DEF);
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/* Map the controller register space. */
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rid = 0;
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sc->reg = bus_alloc_resource_any(self, SYS_RES_MEMORY, &rid, RF_ACTIVE);
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if (sc->reg == NULL)
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return ENXIO;
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/* Map the DBDMA channel register space. */
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rid = 1;
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sc->aoa.sc_odma = bus_alloc_resource_any(self, SYS_RES_MEMORY, &rid,
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RF_ACTIVE);
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if (sc->aoa.sc_odma == NULL)
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return ENXIO;
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/* Establish the DBDMA channel edge-triggered interrupt. */
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rid = 1;
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dbdma_irq = bus_alloc_resource_any(self, SYS_RES_IRQ,
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&rid, RF_SHAREABLE | RF_ACTIVE);
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if (dbdma_irq == NULL)
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return (ENXIO);
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/* Now initialize the controller. */
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err = i2s_setup(sc, 44100, 16, 64);
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if (err != 0)
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return (err);
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snd_setup_intr(self, dbdma_irq, INTR_MPSAFE, aoa_interrupt,
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sc, &dbdma_ih);
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oirq = rman_get_start(dbdma_irq);
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err = powerpc_config_intr(oirq, INTR_TRIGGER_EDGE, INTR_POLARITY_LOW);
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if (err != 0)
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return (err);
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/*
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* Register a hook for delayed attach in order to allow
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* the I2C controller to attach.
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*/
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if ((i2s_delayed_attach = malloc(sizeof(struct intr_config_hook),
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M_TEMP, M_WAITOK | M_ZERO)) == NULL)
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return (ENOMEM);
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i2s_delayed_attach->ich_func = i2s_postattach;
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i2s_delayed_attach->ich_arg = sc;
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if (config_intrhook_establish(i2s_delayed_attach) != 0)
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return (ENOMEM);
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return (aoa_attach(sc));
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}
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/*****************************************************************************
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GPIO routines.
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*****************************************************************************/
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enum gpio_ctrl {
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AMP_MUTE,
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HEADPHONE_MUTE,
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LINEOUT_MUTE,
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AUDIO_HW_RESET,
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HEADPHONE_DETECT,
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LINEOUT_DETECT,
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GPIO_CTRL_NUM
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};
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#define GPIO_CTRL_EXTINT_SET \
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((1 << HEADPHONE_DETECT) | \
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(1 << LINEOUT_DETECT))
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static struct aoagpio_softc *gpio_ctrls[GPIO_CTRL_NUM] =
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{NULL, NULL, NULL, NULL, NULL, NULL};
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static struct gpio_match {
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const char *name;
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enum gpio_ctrl ctrl;
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} gpio_controls[] = {
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{"headphone-mute", HEADPHONE_MUTE},
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{"lineout-mute", LINEOUT_MUTE},
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{"amp-mute", AMP_MUTE},
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{"headphone-detect", HEADPHONE_DETECT},
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{"lineout-detect", LINEOUT_DETECT},
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{"line-output-detect", LINEOUT_DETECT},
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{"audio-hw-reset", AUDIO_HW_RESET},
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{"hw-reset", AUDIO_HW_RESET},
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{NULL, GPIO_CTRL_NUM}
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};
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static void i2s_cint(struct i2s_softc *);
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static void
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aoagpio_int(void *cookie)
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{
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device_t self = cookie;
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struct aoagpio_softc *sc;
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sc = device_get_softc(self);
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if (macgpio_read(self) & GPIO_LEVEL_RO)
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sc->level = sc->detect_active;
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else
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sc->level = !(sc->detect_active);
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if (sc->i2s)
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i2s_cint(sc->i2s);
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}
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static int
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aoagpio_probe(device_t gpio)
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{
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phandle_t node;
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char bname[32];
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const char *name;
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struct gpio_match *m;
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struct aoagpio_softc *sc;
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node = ofw_bus_get_node(gpio);
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if (node == 0 || node == -1)
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return (EINVAL);
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bzero(bname, sizeof(bname));
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if (OF_getprop(node, "audio-gpio", bname, sizeof(bname)) > 2)
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name = bname;
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else
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name = ofw_bus_get_name(gpio);
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/* Try to find a match. */
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for (m = gpio_controls; m->name != NULL; m++) {
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if (strcmp(name, m->name) == 0) {
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sc = device_get_softc(gpio);
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gpio_ctrls[m->ctrl] = sc;
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sc->dev = gpio;
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sc->ctrl = m->ctrl;
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sc->level = 0;
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sc->detect_active = 0;
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sc->i2s = NULL;
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OF_getprop(node, "audio-gpio-active-state",
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&sc->detect_active, sizeof(sc->detect_active));
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if ((1 << m->ctrl) & GPIO_CTRL_EXTINT_SET)
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aoagpio_int(gpio);
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device_set_desc(gpio, m->name);
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device_quiet(gpio);
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return (0);
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}
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}
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return (ENXIO);
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}
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static int
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aoagpio_attach(device_t gpio)
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{
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struct aoagpio_softc *sc;
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struct resource *r;
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void *cookie;
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int irq, rid = 0;
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sc = device_get_softc(gpio);
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if ((1 << sc->ctrl) & GPIO_CTRL_EXTINT_SET) {
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r = bus_alloc_resource_any(gpio, SYS_RES_IRQ, &rid, RF_ACTIVE);
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if (r == NULL)
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return (ENXIO);
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irq = rman_get_start(r);
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DPRINTF(("interrupting at irq %d\n", irq));
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if (powerpc_config_intr(irq, INTR_TRIGGER_EDGE,
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INTR_POLARITY_LOW) != 0)
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return (ENXIO);
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bus_setup_intr(gpio, r, INTR_TYPE_MISC | INTR_MPSAFE |
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INTR_ENTROPY, NULL, aoagpio_int, gpio, &cookie);
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}
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return (0);
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}
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/*
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* I2S module registers
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*/
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#define I2S_INT 0x00
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#define I2S_FORMAT 0x10
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#define I2S_FRAMECOUNT 0x40
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#define I2S_FRAMEMATCH 0x50
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#define I2S_WORDSIZE 0x60
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/* I2S_INT register definitions */
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#define I2S_INT_CLKSTOPPEND 0x01000000 /* clock-stop interrupt pending */
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/* I2S_FORMAT register definitions */
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#define CLKSRC_49MHz 0x80000000 /* Use 49152000Hz Osc. */
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#define CLKSRC_45MHz 0x40000000 /* Use 45158400Hz Osc. */
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#define CLKSRC_18MHz 0x00000000 /* Use 18432000Hz Osc. */
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#define MCLK_DIV_MASK 0x1f000000 /* MCLK = SRC / DIV */
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#define SCLK_DIV_MASK 0x00f00000 /* SCLK = MCLK / DIV */
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#define SCLK_MASTER 0x00080000 /* Master mode */
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#define SCLK_SLAVE 0x00000000 /* Slave mode */
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#define SERIAL_FORMAT 0x00070000
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#define SERIAL_SONY 0x00000000
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#define SERIAL_64x 0x00010000
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#define SERIAL_32x 0x00020000
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#define SERIAL_DAV 0x00040000
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#define SERIAL_SILICON 0x00050000
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/* I2S_WORDSIZE register definitions */
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#define INPUT_STEREO (2 << 24)
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#define INPUT_MONO (1 << 24)
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#define INPUT_16BIT (0 << 16)
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#define INPUT_24BIT (3 << 16)
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#define OUTPUT_STEREO (2 << 8)
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#define OUTPUT_MONO (1 << 8)
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#define OUTPUT_16BIT (0 << 0)
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#define OUTPUT_24BIT (3 << 0)
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/* Master clock, needed by some codecs. We hardcode this
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to 256 * fs as this is valid for most codecs. */
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#define MCLK_FS 256
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/* Number of clock sources we can use. */
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#define NCLKS 3
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static const struct i2s_clksrc {
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u_int cs_clock;
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u_int cs_reg;
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} clksrc[NCLKS] = {
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{49152000, CLKSRC_49MHz},
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{45158400, CLKSRC_45MHz},
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{18432000, CLKSRC_18MHz}
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};
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/* Configure the I2S controller for the required settings.
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'rate' is the frame rate.
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'wordsize' is the sample size (usually 16 bits).
|
|
'sclk_fs' is the SCLK/framerate ratio, which needs to be equal
|
|
or greater to the number of bits per frame. */
|
|
|
|
static int
|
|
i2s_setup(struct i2s_softc *sc, u_int rate, u_int wordsize, u_int sclk_fs)
|
|
{
|
|
u_int mclk, mdiv, sdiv;
|
|
u_int reg = 0, x, wordformat;
|
|
u_int i;
|
|
|
|
/* Make sure the settings are consistent... */
|
|
if ((wordsize * 2) > sclk_fs)
|
|
return (EINVAL);
|
|
|
|
if (sclk_fs != 32 && sclk_fs != 64)
|
|
return (EINVAL);
|
|
|
|
/*
|
|
* Find a clock source to derive the master clock (MCLK)
|
|
* and the I2S bit block (SCLK) and set the divisors as
|
|
* appropriate.
|
|
*/
|
|
mclk = rate * MCLK_FS;
|
|
sdiv = MCLK_FS / sclk_fs;
|
|
|
|
for (i = 0; i < NCLKS; ++i) {
|
|
if ((clksrc[i].cs_clock % mclk) == 0) {
|
|
reg = clksrc[i].cs_reg;
|
|
mdiv = clksrc[i].cs_clock / mclk;
|
|
break;
|
|
}
|
|
}
|
|
if (reg == 0)
|
|
return (EINVAL);
|
|
|
|
switch (mdiv) {
|
|
/* exception cases */
|
|
case 1:
|
|
x = 14;
|
|
break;
|
|
case 3:
|
|
x = 13;
|
|
break;
|
|
case 5:
|
|
x = 12;
|
|
break;
|
|
default:
|
|
x = (mdiv / 2) - 1;
|
|
break;
|
|
}
|
|
reg |= (x << 24) & MCLK_DIV_MASK;
|
|
|
|
switch (sdiv) {
|
|
case 1:
|
|
x = 8;
|
|
break;
|
|
case 3:
|
|
x = 9;
|
|
break;
|
|
default:
|
|
x = (sdiv / 2) - 1;
|
|
break;
|
|
}
|
|
reg |= (x << 20) & SCLK_DIV_MASK;
|
|
|
|
/*
|
|
* XXX use master mode for now. This needs to be
|
|
* revisited if we want to add recording from SPDIF some day.
|
|
*/
|
|
reg |= SCLK_MASTER;
|
|
|
|
switch (sclk_fs) {
|
|
case 64:
|
|
reg |= SERIAL_64x;
|
|
break;
|
|
case 32:
|
|
reg |= SERIAL_32x;
|
|
break;
|
|
}
|
|
|
|
/* stereo input and output */
|
|
wordformat = INPUT_STEREO | OUTPUT_STEREO;
|
|
|
|
switch (wordsize) {
|
|
case 16:
|
|
wordformat |= INPUT_16BIT | OUTPUT_16BIT;
|
|
break;
|
|
case 24:
|
|
wordformat |= INPUT_24BIT | OUTPUT_24BIT;
|
|
break;
|
|
default:
|
|
return (EINVAL);
|
|
}
|
|
|
|
x = bus_read_4(sc->reg, I2S_WORDSIZE);
|
|
if (x != wordformat)
|
|
bus_write_4(sc->reg, I2S_WORDSIZE, wordformat);
|
|
|
|
x = bus_read_4(sc->reg, I2S_FORMAT);
|
|
if (x != reg) {
|
|
/*
|
|
* XXX to change the format we need to stop the clock
|
|
* via the FCR registers. For now, rely on the firmware
|
|
* to set sane defaults (44100).
|
|
*/
|
|
printf("i2s_setup: changing format not supported yet.\n");
|
|
return (ENOTSUP);
|
|
|
|
#ifdef notyet
|
|
if (obio_fcr_isset(OBIO_FCR1, I2S0CLKEN)) {
|
|
|
|
bus_space_write_4(sc->sc_tag, sc->sc_bsh, I2S_INT,
|
|
I2S_INT_CLKSTOPPEND);
|
|
|
|
obio_fcr_clear(OBIO_FCR1, I2S0CLKEN);
|
|
|
|
for (timo = 1000; timo > 0; timo--) {
|
|
if (bus_space_read_4(sc->sc_tag, sc->sc_bsh,
|
|
I2S_INT) & I2S_INT_CLKSTOPPEND)
|
|
break;
|
|
|
|
DELAY(10);
|
|
}
|
|
|
|
if (timo == 0)
|
|
printf("%s: timeout waiting for clock to stop\n",
|
|
sc->sc_dev.dv_xname);
|
|
}
|
|
|
|
bus_space_write_4(sc->sc_tag, sc->sc_bsh, I2S_FORMAT, reg);
|
|
|
|
obio_fcr_set(OBIO_FCR1, I2S0CLKEN);
|
|
#endif
|
|
}
|
|
|
|
return (0);
|
|
}
|
|
|
|
|
|
/* XXX this does not belong here. */
|
|
static phandle_t
|
|
of_find_firstchild_byname(phandle_t node, const char *req_name)
|
|
{
|
|
char name[32]; /* max name len per OF spec. */
|
|
phandle_t n;
|
|
|
|
for (n = OF_child(node); n != -1; n = OF_peer(n)) {
|
|
bzero(name, sizeof(name));
|
|
OF_getprop(n, "name", name, sizeof(name));
|
|
|
|
if (strcmp(name, req_name) == 0)
|
|
return (n);
|
|
}
|
|
|
|
return (-1);
|
|
}
|
|
|
|
|
|
static u_int
|
|
gpio_read(enum gpio_ctrl ctrl)
|
|
{
|
|
struct aoagpio_softc *sc;
|
|
|
|
if ((sc = gpio_ctrls[ctrl]) == NULL)
|
|
return (0);
|
|
|
|
return (macgpio_read(sc->dev) & GPIO_DATA);
|
|
}
|
|
|
|
static void
|
|
gpio_write(enum gpio_ctrl ctrl, u_int x)
|
|
{
|
|
struct aoagpio_softc *sc;
|
|
u_int reg;
|
|
|
|
if ((sc = gpio_ctrls[ctrl]) == NULL)
|
|
return;
|
|
|
|
reg = GPIO_DDR_OUTPUT;
|
|
if (x)
|
|
reg |= GPIO_DATA;
|
|
|
|
macgpio_write(sc->dev, reg);
|
|
}
|
|
|
|
static void
|
|
i2s_cint(struct i2s_softc *sc)
|
|
{
|
|
u_int mask = 0;
|
|
|
|
if (gpio_ctrls[HEADPHONE_DETECT] &&
|
|
gpio_ctrls[HEADPHONE_DETECT]->level)
|
|
mask |= 1 << 1;
|
|
|
|
if (gpio_ctrls[LINEOUT_DETECT] &&
|
|
gpio_ctrls[LINEOUT_DETECT]->level)
|
|
mask |= 1 << 2;
|
|
|
|
if (mask == 0)
|
|
mask = 1 << 0; /* fall back to speakers. */
|
|
|
|
i2s_set_outputs(sc, mask);
|
|
}
|
|
|
|
#define reset_active 0
|
|
|
|
/* these values are in microseconds */
|
|
#define RESET_SETUP_TIME 5000
|
|
#define RESET_HOLD_TIME 20000
|
|
#define RESET_RELEASE_TIME 10000
|
|
|
|
static void
|
|
i2s_audio_hw_reset(struct i2s_softc *sc)
|
|
{
|
|
if (gpio_ctrls[AUDIO_HW_RESET]) {
|
|
DPRINTF(("resetting codec\n"));
|
|
|
|
gpio_write(AUDIO_HW_RESET, !reset_active); /* Negate RESET */
|
|
DELAY(RESET_SETUP_TIME);
|
|
|
|
gpio_write(AUDIO_HW_RESET, reset_active); /* Assert RESET */
|
|
DELAY(RESET_HOLD_TIME);
|
|
|
|
gpio_write(AUDIO_HW_RESET, !reset_active); /* Negate RESET */
|
|
DELAY(RESET_RELEASE_TIME);
|
|
|
|
} else {
|
|
DPRINTF(("no audio_hw_reset\n"));
|
|
}
|
|
}
|
|
|
|
#define AMP_ACTIVE 0 /* XXX OF */
|
|
#define HEADPHONE_ACTIVE 0 /* XXX OF */
|
|
#define LINEOUT_ACTIVE 0 /* XXX OF */
|
|
|
|
#define MUTE_CONTROL(xxx, yyy) \
|
|
static void \
|
|
i2s_mute_##xxx(struct i2s_softc *sc, int mute) \
|
|
{ \
|
|
int x; \
|
|
\
|
|
if (gpio_ctrls[yyy##_MUTE] == NULL) \
|
|
return; \
|
|
if (mute) \
|
|
x = yyy##_ACTIVE; \
|
|
else \
|
|
x = ! yyy##_ACTIVE; \
|
|
\
|
|
if (x != gpio_read(yyy##_MUTE)) \
|
|
gpio_write(yyy##_MUTE, x); \
|
|
}
|
|
|
|
MUTE_CONTROL(speaker, AMP)
|
|
MUTE_CONTROL(headphone, HEADPHONE)
|
|
MUTE_CONTROL(lineout, LINEOUT)
|
|
|
|
static void
|
|
i2s_set_outputs(void *ptr, u_int mask)
|
|
{
|
|
struct i2s_softc *sc = ptr;
|
|
|
|
if (mask == sc->output_mask)
|
|
return;
|
|
|
|
mtx_lock(&sc->port_mtx);
|
|
|
|
i2s_mute_speaker(sc, 1);
|
|
i2s_mute_headphone(sc, 1);
|
|
i2s_mute_lineout(sc, 1);
|
|
|
|
DPRINTF(("enabled outputs: "));
|
|
|
|
if (mask & (1 << 0)) {
|
|
DPRINTF(("SPEAKER "));
|
|
i2s_mute_speaker(sc, 0);
|
|
}
|
|
if (mask & (1 << 1)) {
|
|
DPRINTF(("HEADPHONE "));
|
|
i2s_mute_headphone(sc, 0);
|
|
}
|
|
if (mask & (1 << 2)) {
|
|
DPRINTF(("LINEOUT "));
|
|
i2s_mute_lineout(sc, 0);
|
|
}
|
|
|
|
DPRINTF(("\n"));
|
|
sc->output_mask = mask;
|
|
|
|
mtx_unlock(&sc->port_mtx);
|
|
}
|
|
|
|
static void
|
|
i2s_postattach(void *xsc)
|
|
{
|
|
struct i2s_softc *sc = xsc;
|
|
device_t self;
|
|
int i;
|
|
|
|
self = sc->aoa.sc_dev;
|
|
|
|
/* Reset the codec. */
|
|
i2s_audio_hw_reset(sc);
|
|
|
|
/* If we have a codec, initialize it. */
|
|
if (i2s_mixer)
|
|
mixer_init(self, i2s_mixer_class, i2s_mixer);
|
|
|
|
/* Read initial port status. */
|
|
i2s_cint(sc);
|
|
|
|
/* Enable GPIO interrupt callback. */
|
|
for (i = 0; i < GPIO_CTRL_NUM; i++)
|
|
if (gpio_ctrls[i])
|
|
gpio_ctrls[i]->i2s = sc;
|
|
|
|
config_intrhook_disestablish(i2s_delayed_attach);
|
|
free(i2s_delayed_attach, M_TEMP);
|
|
}
|
|
|