NetBSD/sys/dev/audiovar.h

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/* $NetBSD: audiovar.h,v 1.68 2017/11/15 04:28:45 nat Exp $ */
/*-
* Copyright (c) 2002 The NetBSD Foundation, Inc.
* All rights reserved.
*
* This code is derived from software contributed to The NetBSD Foundation
* by TAMURA Kent
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* THIS SOFTWARE IS PROVIDED BY THE NETBSD FOUNDATION, INC. AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED
* TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
* PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR CONTRIBUTORS
* BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
* CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
* SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
* INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
* CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
* ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
* POSSIBILITY OF SUCH DAMAGE.
*/
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/*
* Copyright (c) 1991-1993 Regents of the University of California.
* 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. All advertising materials mentioning features or use of this software
* must display the following acknowledgement:
* This product includes software developed by the Computer Systems
* Engineering Group at Lawrence Berkeley Laboratory.
* 4. Neither the name of the University nor of the Laboratory may be used
* to endorse or promote products derived from this software without
* specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND
* ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
* ARE DISCLAIMED. IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
* FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
* OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
* LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
* SUCH DAMAGE.
*
* From: Header: audiovar.h,v 1.3 93/07/18 14:07:25 mccanne Exp (LBL)
*/
#ifndef _SYS_DEV_AUDIOVAR_H_
#define _SYS_DEV_AUDIOVAR_H_
#include <sys/condvar.h>
#include <sys/proc.h>
#include <sys/queue.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/audio_if.h>
#include <dev/auconv.h>
/* Interfaces for audiobell. */
int audiobellopen(dev_t, int, int, struct lwp *, struct file **);
int audiobellclose(struct file *);
int audiobellwrite(struct file *, off_t *, struct uio *, kauth_cred_t, int);
int audiobellioctl(struct file *, u_long, void *);
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/*
* Initial/default block duration is both configurable and patchable.
*/
#ifndef AUDIO_BLK_MS
#define AUDIO_BLK_MS 50 /* 50 ms */
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#endif
#ifndef AU_RING_SIZE
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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#define AU_RING_SIZE 65536
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#endif
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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#define AUMINBUF 512
#define AUMINBLK 32
#define AUMINNOBLK 3
struct audio_ringbuffer {
struct uvm_object *uobj;
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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audio_stream_t s;
int blksize; /* I/O block size (bytes) */
int maxblks; /* no of blocks in ring */
int usedlow; /* start writer when used falls below this */
int usedhigh; /* stop writer when used goes above this */
u_long stamp; /* bytes transferred */
u_long stamp_last; /* old value of bytes transferred */
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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u_long fstamp; /* bytes transferred from/to the buffer near to userland */
u_long drops; /* missed samples from over/underrun */
u_long pdrops; /* paused samples */
bool pause; /* transfer is paused */
bool copying; /* data is being copied */
bool needfill; /* buffer needs filling when copying is done */
bool mmapped; /* device is mmap()-ed */
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};
struct audio_chan {
dev_t dev;
struct virtual_channel *vc;
int chan; /* virtual channel */
int deschan; /* desired channel for ioctls*/
SIMPLEQ_ENTRY(audio_chan) entries;
};
struct virtual_channel {
u_char sc_open; /* multiple use device */
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#define AUOPEN_READ 0x01
#define AUOPEN_WRITE 0x02
u_char sc_mode; /* bitmask for RECORD/PLAY */
uint8_t *sc_sil_start; /* start of silence in buffer */
bool sc_pbus; /* output DMA in progress */
audio_params_t sc_pparams; /* play encoding parameters */
audio_stream_t *sc_pustream; /* the first buffer */
int sc_npfilters; /* number of filters */
audio_stream_t sc_pstreams[AUDIO_MAX_FILTERS];
stream_filter_t *sc_pfilters[AUDIO_MAX_FILTERS];
struct audio_ringbuffer sc_mpr; /* Play ring to mix */
u_long sc_wstamp; /* # of bytes read with read(2) */
u_long sc_playdrop;
bool sc_rbus; /* input DMA in progress */
struct audio_ringbuffer sc_mrr; /* Virtual record ring */
int sc_nrfilters; /* number of filters */
stream_filter_t *sc_rfilters[AUDIO_MAX_FILTERS];
audio_stream_t sc_rstreams[AUDIO_MAX_FILTERS];
audio_stream_t *sc_rustream; /* the last buffer */
audio_params_t sc_rparams; /* record encoding parameters */
int sc_full_duplex; /* device in full duplex mode */
struct audio_info sc_lastinfo;
bool sc_lastinfovalid;
bool sc_draining;
uint8_t sc_swvol;
uint8_t sc_recswvol;
};
#define AUDIO_N_PORTS 4
struct au_mixer_ports {
Fix a couple of long-standing bugs in the volume control(s) part of the audio device interface: 1) When attempting to match the appropriate mixer control, we weren't checking the class label, but only the second level label, so for devices that had both an "inputs.cd" and a "record.cd", for example, we could never do the right thing except by chance. For this reason, evidently, all the record masters were labeled (by the underlying drivers) either "record.record" or "record.volume", to distinguish from "outputs.master". We'll now accept "record.master", and document that as the the new preferred way. 2) More importantly, the model was deficient. Selecting a port on many chips completely disables most of the level controls, which doesn't play nice with other applications which are trying to use the interface. Now, selecting a port simply sets which mixer input control shall be changed, setting state in the audio layer. In other words, the "mixerout" port is really selected all the time, enabling the final stage mixer, and setting "gain" sets the level of the appropriate input. It should be possible for separate applications to each control the mic, dac, and cd inputs at the same time using this interface, simply by reiterating their port selections with each change, but applications that don't bother to do that aren't any worse off than they were before. The user is expected to set the master output with another application, dedicated to that task. Though it is now meaningful to select "no port" with the audio device interface, to manipulate the master output, there's no particular reason for an audio device consumer to do that. (I added the capability in order to restore the initial state of the audio device, for testing purposes. It might also be useful to users of broken binary- only applications.) Observe that the mixer device interface (and so, "mixerctl") still retains all capabilities, including the ability to set the actual input port on the chip, overriding the level controls. No change is being made to the mixer device interface. The mixer device simply presents all the controls on the chip, with no attempt at abstraction, so there are no bugs there. The upshot is, that applications that have been trying to use the audio device interface to change the volume, such as mplayer, now "just work". I've tested these changes extensively with "eso" and "eap" since first proposing them on tech-kern last January, and somewhat with "esm" and a few others. This closes both PR kern/10221, and PR kern/17159.
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int index; /* index of port-selector mixerctl */
int master; /* index of master mixerctl */
int nports; /* number of selectable ports */
bool isenum; /* selector is enum type */
Fix a couple of long-standing bugs in the volume control(s) part of the audio device interface: 1) When attempting to match the appropriate mixer control, we weren't checking the class label, but only the second level label, so for devices that had both an "inputs.cd" and a "record.cd", for example, we could never do the right thing except by chance. For this reason, evidently, all the record masters were labeled (by the underlying drivers) either "record.record" or "record.volume", to distinguish from "outputs.master". We'll now accept "record.master", and document that as the the new preferred way. 2) More importantly, the model was deficient. Selecting a port on many chips completely disables most of the level controls, which doesn't play nice with other applications which are trying to use the interface. Now, selecting a port simply sets which mixer input control shall be changed, setting state in the audio layer. In other words, the "mixerout" port is really selected all the time, enabling the final stage mixer, and setting "gain" sets the level of the appropriate input. It should be possible for separate applications to each control the mic, dac, and cd inputs at the same time using this interface, simply by reiterating their port selections with each change, but applications that don't bother to do that aren't any worse off than they were before. The user is expected to set the master output with another application, dedicated to that task. Though it is now meaningful to select "no port" with the audio device interface, to manipulate the master output, there's no particular reason for an audio device consumer to do that. (I added the capability in order to restore the initial state of the audio device, for testing purposes. It might also be useful to users of broken binary- only applications.) Observe that the mixer device interface (and so, "mixerctl") still retains all capabilities, including the ability to set the actual input port on the chip, overriding the level controls. No change is being made to the mixer device interface. The mixer device simply presents all the controls on the chip, with no attempt at abstraction, so there are no bugs there. The upshot is, that applications that have been trying to use the audio device interface to change the volume, such as mplayer, now "just work". I've tested these changes extensively with "eso" and "eap" since first proposing them on tech-kern last January, and somewhat with "esm" and a few others. This closes both PR kern/10221, and PR kern/17159.
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u_int allports; /* all aumasks or'd */
u_int aumask[AUDIO_N_PORTS]; /* exposed value of "ports" */
int misel [AUDIO_N_PORTS]; /* ord of port, for selector */
int miport[AUDIO_N_PORTS]; /* index of port's mixerctl */
bool isdual; /* has working mixerout */
Fix a couple of long-standing bugs in the volume control(s) part of the audio device interface: 1) When attempting to match the appropriate mixer control, we weren't checking the class label, but only the second level label, so for devices that had both an "inputs.cd" and a "record.cd", for example, we could never do the right thing except by chance. For this reason, evidently, all the record masters were labeled (by the underlying drivers) either "record.record" or "record.volume", to distinguish from "outputs.master". We'll now accept "record.master", and document that as the the new preferred way. 2) More importantly, the model was deficient. Selecting a port on many chips completely disables most of the level controls, which doesn't play nice with other applications which are trying to use the interface. Now, selecting a port simply sets which mixer input control shall be changed, setting state in the audio layer. In other words, the "mixerout" port is really selected all the time, enabling the final stage mixer, and setting "gain" sets the level of the appropriate input. It should be possible for separate applications to each control the mic, dac, and cd inputs at the same time using this interface, simply by reiterating their port selections with each change, but applications that don't bother to do that aren't any worse off than they were before. The user is expected to set the master output with another application, dedicated to that task. Though it is now meaningful to select "no port" with the audio device interface, to manipulate the master output, there's no particular reason for an audio device consumer to do that. (I added the capability in order to restore the initial state of the audio device, for testing purposes. It might also be useful to users of broken binary- only applications.) Observe that the mixer device interface (and so, "mixerctl") still retains all capabilities, including the ability to set the actual input port on the chip, overriding the level controls. No change is being made to the mixer device interface. The mixer device simply presents all the controls on the chip, with no attempt at abstraction, so there are no bugs there. The upshot is, that applications that have been trying to use the audio device interface to change the volume, such as mplayer, now "just work". I've tested these changes extensively with "eso" and "eap" since first proposing them on tech-kern last January, and somewhat with "esm" and a few others. This closes both PR kern/10221, and PR kern/17159.
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int mixerout; /* ord of mixerout, for dual case */
int cur_port; /* the port that gain actually controls when
mixerout is selected, for dual case */
};
SIMPLEQ_HEAD(chan_queue, audio_chan);
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/*
* Software state, per audio device.
*/
struct audio_softc {
device_t dev;
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 *hw_hdl; /* Hardware driver handle */
const struct audio_hw_if *hw_if; /* Hardware interface */
device_t sc_dev; /* Hardware device struct */
struct chan_queue sc_audiochan; /* queue of open audio chans */
struct virtual_channel *sc_hwvc;
struct virtual_channel sc_mixring; /* Play/rec ring to mix into */
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struct audio_encoding_set *sc_encodings;
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struct selinfo sc_wsel; /* write selector */
struct selinfo sc_rsel; /* read selector */
pid_t sc_async_audio; /* process who wants audio SIGIO */
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void *sc_sih_rd;
void *sc_sih_wr;
struct mixer_asyncs {
struct mixer_asyncs *next;
pid_t pid;
} *sc_async_mixer; /* processes who want mixer SIGIO */
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/* Locks and sleep channels for reading, writing and draining. */
kmutex_t *sc_intr_lock;
kmutex_t *sc_lock;
kcondvar_t sc_rchan;
kcondvar_t sc_wchan;
kcondvar_t sc_lchan;
bool sc_trigger_started;
bool sc_rec_started;
bool sc_writeme;
bool sc_usemixer;
bool sc_ready; /* audio hw configured properly */
unsigned int sc_latency;
int sc_opens;
int sc_recopens;
bool sc_dying;
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
/**
* userland
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* | write(2) & uiomove(9)
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
* sc_pstreams[0] <sc_pparams> == sc_pustream;
* | sc_pfilters[0]
* sc_pstreams[1] <list_t::filters[n-1].param>
* :
* sc_pstreams[n-1] <list_t::filters[1].param>
* | sc_pfilters[n-1]
* sc_pr <list_t::filters[0].param>
* (vchans mixed into sc_pr)
*
* play_thread
* sc_mixring.sc_mpr
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
* |
* sc_hwvc->sc_pustream
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* |
* sc_hwvc->sc_mpr
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* |
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
* hardware
*/
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
/**
* hardware
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* |
* sc_hwvc->sc_mrr
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* : Transform though filters same process as each
* : vc to IF
* sc_hwvc->sc_rustream Audio now in intermediate format (IF)
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* | mix_read();
* sc_mixring.sc_mrr
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* | audio_upmix vc = sc->sc_vchan[n]
* vc->sc_mrr <list_t::filters[0].param>
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* | vc->sc_rfilters[0]
* vc->sc_rstreams[0] <list_t::filters[1].param>
* | vc->sc_rfilters[1]
* vc->sc_rstreams[1] <list_t::filters[2].param>
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
* :
2017-01-15 10:46:57 +03:00
* | vc->sc_rfilters[n-1]
* vc->sc_rstreams[n-1] <vc->sc_rparams> == vc->sc_rustream
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
* | uiomove(9) & read(2)
* userland
*/
ulong sc_last_drops; /* Drops from mix ring */
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
int sc_eof; /* EOF, i.e. zero sized write, counter */
struct au_mixer_ports sc_inports, sc_outports;
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
int sc_monitor_port;
#ifdef AUDIO_INTR_TIME
int sc_pnintr; /* number of interrupts */
int64_t sc_pfirstintr; /* first time we saw a play interrupt */
int64_t sc_plastintr; /* last time we saw a play interrupt */
int64_t sc_pblktime; /* nominal time between interrupts */
int sc_rnintr; /* number of interrupts */
int64_t sc_rfirstintr; /* first time we saw a rec interrupt */
int64_t sc_rlastintr; /* last time we saw a rec interrupt */
int64_t sc_rblktime; /* nominal time between interrupts */
#endif
2007-12-09 23:27:42 +03:00
u_int sc_lastgain;
mixer_ctrl_t *sc_mixer_state;
int sc_nmixer_states;
int sc_static_nmixer_states;
bool schedule_wih;
bool schedule_rih;
lwp_t *sc_playthread;
kcondvar_t sc_condvar;
lwp_t *sc_recthread;
kcondvar_t sc_rcondvar;
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/* These are changeable by sysctl to set the vchan common format */
struct sysctllog *sc_log; /* sysctl log */
struct audio_info sc_ai; /* Recent info for dev sound */
bool sc_aivalid;
#define VAUDIO_NFORMATS 1
struct audio_format sc_format[VAUDIO_NFORMATS];
struct audio_params sc_vchan_params;
bool sc_multiuser;
kauth_cred_t sc_credentials; /* audio user's credentials */
2002-03-18 02:29:55 +03:00
};
#endif /* _SYS_DEV_AUDIOVAR_H_ */