NetBSD/sys/arch/arm/iomd/vidcaudio.c

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merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
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/* $NetBSD: vidcaudio.c,v 1.41 2005/01/10 22:01:36 kent Exp $ */
/*
* Copyright (c) 1995 Melvin Tang-Richardson
*
* 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 RiscBSD team.
* 4. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
* IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
* OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED.
* IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
* INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
* NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
* DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
* THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
* (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
* THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
/*-
* Copyright (c) 2003 Ben Harris
* All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
* IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
* OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED.
* IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
* INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
* NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
* DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
* THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
* (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
* THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
/*
* audio driver for the RiscPC 8/16 bit sound
*
* Interfaces with the NetBSD generic audio driver to provide SUN
* /dev/audio (partial) compatibility.
*/
#include <sys/param.h> /* proc.h */
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
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__KERNEL_RCSID(0, "$NetBSD: vidcaudio.c,v 1.41 2005/01/10 22:01:36 kent Exp $");
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#include <sys/audioio.h>
#include <sys/conf.h> /* autoconfig functions */
#include <sys/device.h> /* device calls */
#include <sys/errno.h>
#include <sys/malloc.h>
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#include <sys/proc.h> /* device calls */
#include <sys/systm.h>
#include <uvm/uvm_extern.h>
#include <dev/audio_if.h>
#include <dev/audiobellvar.h>
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
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#include <dev/auconv.h>
#include <dev/mulaw.h>
#include <machine/intr.h>
#include <machine/machdep.h>
#include <arm/arm32/katelib.h>
#include <arm/iomd/vidcaudiovar.h>
#include <arm/iomd/iomdreg.h>
#include <arm/iomd/iomdvar.h>
#include <arm/iomd/vidc.h>
#include <arm/mainbus/mainbus.h>
#include "pckbd.h"
#if NPCKBD > 0
#include <dev/pckbport/pckbdvar.h>
#endif
#include "rpckbd.h"
#if NRPCKBD > 0
#include <arm/iomd/rpckbdvar.h>
#endif
#include "vt.h"
#if NVT > 0
extern void vt_hookup_bell(void (*)(void *, u_int, u_int, u_int, int), void *);
#endif
extern int *vidc_base;
#ifdef VIDCAUDIO_DEBUG
#define DPRINTF(x) printf x
#else
#define DPRINTF(x)
#endif
struct vidcaudio_softc {
struct device sc_dev;
irqhandler_t sc_ih;
int sc_dma_intr;
int sc_is16bit;
size_t sc_pblksize;
vm_offset_t sc_poffset;
vm_offset_t sc_pbufsize;
paddr_t *sc_ppages;
void (*sc_pintr)(void *);
void *sc_parg;
int sc_pcountdown;
};
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static int vidcaudio_probe(struct device *, struct cfdata *, void *);
static void vidcaudio_attach(struct device *, struct device *, void *);
static void vidcaudio_close(void *);
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static int vidcaudio_intr(void *);
static void vidcaudio_rate(int);
static void vidcaudio_ctrl(int);
static void vidcaudio_stereo(int, int);
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
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static stream_filter_factory_t mulaw_to_vidc;
static stream_filter_factory_t mulaw_to_vidc_stereo;
static int mulaw_to_vidc_fetch_to(stream_fetcher_t *, audio_stream_t *, int);
static int mulaw_to_vidc_stereo_fetch_to(stream_fetcher_t *,
audio_stream_t *, int);
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CFATTACH_DECL(vidcaudio, sizeof(struct vidcaudio_softc),
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vidcaudio_probe, vidcaudio_attach, NULL, NULL);
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static int vidcaudio_query_encoding(void *, struct audio_encoding *);
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
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static int vidcaudio_set_params(void *, int, int, audio_params_t *,
audio_params_t *, stream_filter_list_t *, stream_filter_list_t *);
static int vidcaudio_round_blocksize(void *, int, int, const audio_params_t *);
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static int vidcaudio_trigger_output(void *, void *, void *, int,
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
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void (*)(void *), void *, const audio_params_t *);
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static int vidcaudio_trigger_input(void *, void *, void *, int,
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
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void (*)(void *), void *, const audio_params_t *);
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static int vidcaudio_halt_output(void *);
static int vidcaudio_halt_input(void *);
static int vidcaudio_getdev(void *, struct audio_device *);
static int vidcaudio_set_port(void *, mixer_ctrl_t *);
static int vidcaudio_get_port(void *, mixer_ctrl_t *);
static int vidcaudio_query_devinfo(void *, mixer_devinfo_t *);
static int vidcaudio_get_props(void *);
static struct audio_device vidcaudio_device = {
"ARM VIDC",
"",
"vidcaudio"
};
static const struct audio_hw_if vidcaudio_hw_if = {
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
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NULL, /* open */
vidcaudio_close,
NULL,
vidcaudio_query_encoding,
vidcaudio_set_params,
vidcaudio_round_blocksize,
NULL,
NULL,
NULL,
NULL,
NULL,
vidcaudio_halt_output,
vidcaudio_halt_input,
NULL,
vidcaudio_getdev,
NULL,
vidcaudio_set_port,
vidcaudio_get_port,
vidcaudio_query_devinfo,
NULL,
NULL,
NULL,
NULL,
vidcaudio_get_props,
vidcaudio_trigger_output,
vidcaudio_trigger_input,
NULL,
};
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static int
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vidcaudio_probe(struct device *parent, struct cfdata *cf, void *aux)
{
int id;
id = IOMD_ID;
/* So far I only know about this IOMD */
switch (id) {
case RPC600_IOMD_ID:
case ARM7500_IOC_ID:
case ARM7500FE_IOC_ID:
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return 1;
default:
aprint_error("vidcaudio: Unknown IOMD id=%04x", id);
return 0;
}
}
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static void
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vidcaudio_attach(struct device *parent, struct device *self, void *aux)
{
struct vidcaudio_softc *sc = (void *)self;
struct device *beepdev;
switch (IOMD_ID) {
#ifndef EB7500ATX
case RPC600_IOMD_ID:
sc->sc_is16bit = (cmos_read(0xc4) >> 5) & 1;
sc->sc_dma_intr = IRQ_DMASCH0;
break;
#endif
case ARM7500_IOC_ID:
case ARM7500FE_IOC_ID:
sc->sc_is16bit = TRUE;
sc->sc_dma_intr = IRQ_SDMA;
break;
default:
aprint_error(": strange IOMD\n");
return;
}
if (sc->sc_is16bit)
aprint_normal(": 16-bit external DAC\n");
else
aprint_normal(": 8-bit internal DAC\n");
/* Install the irq handler for the DMA interrupt */
sc->sc_ih.ih_func = vidcaudio_intr;
sc->sc_ih.ih_arg = sc;
sc->sc_ih.ih_level = IPL_AUDIO;
sc->sc_ih.ih_name = self->dv_xname;
if (irq_claim(sc->sc_dma_intr, &sc->sc_ih) != 0) {
aprint_error("%s: couldn't claim IRQ %d\n",
self->dv_xname, sc->sc_dma_intr);
return;
}
disable_irq(sc->sc_dma_intr);
beepdev = audio_attach_mi(&vidcaudio_hw_if, sc, self);
#if NPCKBD > 0
pckbd_hookup_bell(audiobell, beepdev);
#endif
#if NRPCKBD > 0
rpckbd_hookup_bell(audiobell, beepdev);
#endif
#if NVT > 0
vt_hookup_bell(audiobell, beepdev);
#endif
}
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static void
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vidcaudio_close(void *addr)
{
struct vidcaudio_softc *sc = addr;
DPRINTF(("DEBUG: vidcaudio_close called\n"));
/*
* We do this here rather than in vidcaudio_halt_output()
* because the latter can be called from interrupt context
* (audio_pint()->audio_clear()->vidcaudio_halt_output()).
*/
if (sc->sc_ppages != NULL) {
free(sc->sc_ppages, M_DEVBUF);
sc->sc_ppages = NULL;
}
}
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/*
* Interface to the generic audio driver
*/
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static int
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vidcaudio_query_encoding(void *addr, struct audio_encoding *fp)
{
struct vidcaudio_softc *sc = addr;
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switch (fp->index) {
case 0:
strcpy(fp->name, AudioEmulaw);
fp->encoding = AUDIO_ENCODING_ULAW;
fp->precision = 8;
fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
break;
case 1:
if (sc->sc_is16bit) {
strcpy(fp->name, AudioEslinear_le);
fp->encoding = AUDIO_ENCODING_SLINEAR_LE;
fp->precision = 16;
fp->flags = 0;
break;
}
/* FALLTHROUGH */
default:
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return EINVAL;
}
return 0;
}
#define MULAW_TO_VIDC(m) (~((m) << 1 | (m) >> 7))
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
static stream_filter_t *
mulaw_to_vidc(struct audio_softc *sc, const audio_params_t *from,
const audio_params_t *to)
{
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
return auconv_nocontext_filter_factory(mulaw_to_vidc_fetch_to);
}
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
static int
mulaw_to_vidc_fetch_to(stream_fetcher_t *self, audio_stream_t *dst, int max_used)
{
stream_filter_t *this;
int m, err;
this = (stream_filter_t *)self;
if ((err = this->prev->fetch_to(this->prev, this->src, max_used)))
return err;
m = dst->end - dst->start;
m = min(m, max_used);
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 1, m) {
*d = MULAW_TO_VIDC(*s);
} FILTER_LOOP_EPILOGUE(this->src, dst);
return 0;
}
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
static stream_filter_t *
mulaw_to_vidc_stereo(struct audio_softc *sc, const audio_params_t *from,
const audio_params_t *to)
{
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
return auconv_nocontext_filter_factory(mulaw_to_vidc_stereo_fetch_to);
}
static int
mulaw_to_vidc_stereo_fetch_to(stream_fetcher_t *self, audio_stream_t *dst,
int max_used)
{
stream_filter_t *this;
int m, err;
this = (stream_filter_t *)self;
max_used = (max_used + 1) & ~1;
if ((err = this->prev->fetch_to(this->prev, this->src, max_used / 2)))
return err;
m = (dst->end - dst->start) & ~1;
m = min(m, max_used);
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) {
d[0] = d[1] = MULAW_TO_VIDC(*s);
} FILTER_LOOP_EPILOGUE(this->src, dst);
return 0;
}
2003-12-31 18:40:31 +03:00
static int
2003-12-29 19:11:38 +03:00
vidcaudio_set_params(void *addr, int setmode, int usemode,
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
audio_params_t *p, audio_params_t *r,
stream_filter_list_t *pfil, stream_filter_list_t *rfil)
{
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
audio_params_t hw;
struct vidcaudio_softc *sc = addr;
int sample_period, ch;
if ((setmode & AUMODE_PLAY) == 0)
return 0;
if (sc->sc_is16bit) {
/* ARM7500ish, 16-bit, two-channel */
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
hw = *p;
if (p->encoding == AUDIO_ENCODING_ULAW && p->precision == 8) {
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
hw.encoding = AUDIO_ENCODING_SLINEAR_LE;
hw.precision = hw.validbits = 16;
pfil->append(pfil, mulaw_to_linear16, &hw);
} else if (p->encoding != AUDIO_ENCODING_SLINEAR_LE ||
p->precision != 16)
return EINVAL;
sample_period = 705600 / 4 / p->sample_rate;
if (sample_period < 3) sample_period = 3;
vidcaudio_rate(sample_period - 2);
vidcaudio_ctrl(SCR_SERIAL);
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
hw.sample_rate = 705600 / 4 / sample_period;
hw.channels = 2;
pfil->append(pfil, aurateconv, &hw);
} else {
/* VIDC20ish, u-law, 8-channel */
if (p->encoding != AUDIO_ENCODING_ULAW || p->precision != 8)
return EINVAL;
/*
* We always use two hardware channels, because using
* one at 8kHz gives a nasty whining sound from the
* speaker. The aurateconv mechanism doesn't support
* ulaw, so we do the channel duplication ourselves,
* and don't try to do rate conversion.
*/
sample_period = 1000000 / 2 / p->sample_rate;
if (sample_period < 3) sample_period = 3;
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
p->sample_rate = 1000000 / 2 / sample_period;
hw = *p;
hw.encoding = AUDIO_ENCODING_NONE;
hw.precision = 8;
vidcaudio_rate(sample_period - 2);
vidcaudio_ctrl(SCR_SDAC | SCR_CLKSEL);
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
if (p->channels == 1) {
pfil->append(pfil, mulaw_to_vidc_stereo, &hw);
for (ch = 0; ch < 8; ch++)
vidcaudio_stereo(ch, SIR_CENTRE);
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
} else {
pfil->append(pfil, mulaw_to_vidc, &hw);
for (ch = 0; ch < 8; ch += 2)
vidcaudio_stereo(ch, SIR_LEFT_100);
for (ch = 1; ch < 8; ch += 2)
vidcaudio_stereo(ch, SIR_RIGHT_100);
}
}
return 0;
}
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static int
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
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vidcaudio_round_blocksize(void *addr, int wantblk,
int mode, const audio_params_t *param)
{
int blk;
/*
* Find the smallest power of two that's larger than the
* requested block size, but don't allow < 32 (DMA burst is 16
* bytes, and single bursts are tricky) or > PAGE_SIZE (DMA is
* confined to a page).
*/
for (blk = 32; blk < PAGE_SIZE; blk <<= 1)
if (blk >= wantblk) return blk;
return blk;
}
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static int
vidcaudio_trigger_output(void *addr, void *start, void *end, int blksize,
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
void (*intr)(void *), void *arg, const audio_params_t *params)
{
struct vidcaudio_softc *sc = addr;
size_t npages, i;
DPRINTF(("vidcaudio_trigger_output %p-%p/0x%x\n",
start, end, blksize));
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
KASSERT(blksize == vidcaudio_round_blocksize(addr, blksize, 0, NULL));
KASSERT((vaddr_t)start % blksize == 0);
sc->sc_pblksize = blksize;
sc->sc_pbufsize = (char *)end - (char *)start;
npages = sc->sc_pbufsize >> PGSHIFT;
if (sc->sc_ppages != NULL)
free(sc->sc_ppages, M_DEVBUF);
sc->sc_ppages = malloc(npages * sizeof(paddr_t), M_DEVBUF, M_WAITOK);
if (sc->sc_ppages == NULL) return ENOMEM;
for (i = 0; i < npages; i++)
if (!pmap_extract(pmap_kernel(),
(vaddr_t)start + i * PAGE_SIZE, &sc->sc_ppages[i]))
return EIO;
sc->sc_poffset = 0;
sc->sc_pintr = intr;
sc->sc_parg = arg;
IOMD_WRITE_WORD(IOMD_SD0CR, IOMD_DMACR_CLEAR | IOMD_DMACR_QUADWORD);
IOMD_WRITE_WORD(IOMD_SD0CR, IOMD_DMACR_ENABLE | IOMD_DMACR_QUADWORD);
/*
* Queue up the first two blocks, but don't tell audio code
* we're finished with them yet.
*/
sc->sc_pcountdown = 2;
enable_irq(sc->sc_dma_intr);
return 0;
}
2003-12-31 18:40:31 +03:00
static int
vidcaudio_trigger_input(void *addr, void *start, void *end, int blksize,
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
void (*intr)(void *), void *arg, const audio_params_t *params)
{
return ENODEV;
}
2003-12-31 18:40:31 +03:00
static int
2003-12-29 19:11:38 +03:00
vidcaudio_halt_output(void *addr)
{
struct vidcaudio_softc *sc = addr;
2003-12-29 19:20:46 +03:00
DPRINTF(("vidcaudio_halt_output\n"));
disable_irq(sc->sc_dma_intr);
IOMD_WRITE_WORD(IOMD_SD0CR, IOMD_DMACR_CLEAR | IOMD_DMACR_QUADWORD);
return 0;
}
2003-12-31 18:40:31 +03:00
static int
2003-12-29 19:11:38 +03:00
vidcaudio_halt_input(void *addr)
{
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return ENODEV;
}
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static int
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vidcaudio_getdev(void *addr, struct audio_device *retp)
{
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*retp = vidcaudio_device;
return 0;
}
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static int
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vidcaudio_set_port(void *addr, mixer_ctrl_t *cp)
{
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return EINVAL;
}
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static int
2003-12-29 19:11:38 +03:00
vidcaudio_get_port(void *addr, mixer_ctrl_t *cp)
{
2003-12-29 19:20:46 +03:00
return EINVAL;
}
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static int
2003-12-29 19:11:38 +03:00
vidcaudio_query_devinfo(void *addr, mixer_devinfo_t *dip)
{
2003-12-29 19:20:46 +03:00
return ENXIO;
}
2003-12-31 18:40:31 +03:00
static int
2003-12-29 19:11:38 +03:00
vidcaudio_get_props(void *addr)
{
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return 0;
}
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static void
2003-12-29 19:11:38 +03:00
vidcaudio_rate(int rate)
{
2003-12-29 19:20:46 +03:00
WriteWord(vidc_base, VIDC_SFR | rate);
}
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static void
vidcaudio_ctrl(int ctrl)
{
WriteWord(vidc_base, VIDC_SCR | ctrl);
}
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static void
vidcaudio_stereo(int channel, int position)
{
2003-12-29 19:20:46 +03:00
channel = channel << 24 | VIDC_SIR0;
WriteWord(vidc_base, channel | position);
}
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static int
2003-12-29 19:11:38 +03:00
vidcaudio_intr(void *arg)
{
struct vidcaudio_softc *sc = arg;
int status;
paddr_t pnext, pend;
status = IOMD_READ_BYTE(IOMD_SD0ST);
DPRINTF(("I[%x]", status));
if ((status & IOMD_DMAST_INT) == 0)
return 0;
pnext = sc->sc_ppages[sc->sc_poffset >> PGSHIFT] |
(sc->sc_poffset & PGOFSET);
pend = (pnext + sc->sc_pblksize - 16) & IOMD_DMAEND_OFFSET;
switch (status &
(IOMD_DMAST_OVERRUN | IOMD_DMAST_INT | IOMD_DMAST_BANKB)) {
case (IOMD_DMAST_INT | IOMD_DMAST_BANKA):
case (IOMD_DMAST_OVERRUN | IOMD_DMAST_INT | IOMD_DMAST_BANKB):
DPRINTF(("B<0x%08lx,0x%03lx>", pnext, pend));
IOMD_WRITE_WORD(IOMD_SD0CURB, pnext);
IOMD_WRITE_WORD(IOMD_SD0ENDB, pend);
break;
case (IOMD_DMAST_INT | IOMD_DMAST_BANKB):
case (IOMD_DMAST_OVERRUN | IOMD_DMAST_INT | IOMD_DMAST_BANKA):
DPRINTF(("A<0x%08lx,0x%03lx>", pnext, pend));
IOMD_WRITE_WORD(IOMD_SD0CURA, pnext);
IOMD_WRITE_WORD(IOMD_SD0ENDA, pend);
break;
}
sc->sc_poffset += sc->sc_pblksize;
if (sc->sc_poffset >= sc->sc_pbufsize)
sc->sc_poffset = 0;
if (sc->sc_pcountdown > 0)
sc->sc_pcountdown--;
else
(*sc->sc_pintr)(sc->sc_parg);
return 1;
}