NetBSD/sys/dev/audiovar.h

231 lines
8.7 KiB
C
Raw Normal View History

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.
2004-01-31 03:07:56 +03:00
/* $NetBSD: audiovar.h,v 1.30 2004/01/31 00:07:56 fredb 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.
* 3. All advertising materials mentioning features or use of this software
* must display the following acknowledgement:
* This product includes software developed by the NetBSD
* Foundation, Inc. and its contributors.
* 4. Neither the name of The NetBSD Foundation nor the names of its
* contributors may be used to endorse or promote products derived
* from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE NETBSD FOUNDATION, INC. AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED
* TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
* PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR CONTRIBUTORS
* BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
* CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
* SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
* INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
* CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
* ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
* POSSIBILITY OF SUCH DAMAGE.
*/
1995-02-21 04:35:58 +03:00
/*
* 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_
1995-02-21 04:35:58 +03:00
#include "aurateconv.h" /* generated by config(8) */
1995-02-21 04:35:58 +03:00
/*
* Initial/default block duration is both configurable and patchable.
*/
#ifndef AUDIO_BLK_MS
#define AUDIO_BLK_MS 50 /* 50 ms */
1995-02-21 04:35:58 +03:00
#endif
#ifndef AU_RING_SIZE
#define AU_RING_SIZE 65536
1995-02-21 04:35:58 +03:00
#endif
#define AUMINBUF 512
1997-07-28 03:06:04 +04:00
#define AUMINBLK 32
#define AUMINNOBLK 3
struct audio_ringbuffer {
size_t bufsize; /* allocated memory */
int blksize; /* I/O block size (bytes) */
int maxblks; /* no of blocks in ring */
u_char *start; /* start of buffer area */
u_char *end; /* end of buffer area */
u_char *inp; /* input pointer (to buffer) */
u_char *outp; /* output pointer (from buffer) */
int used; /* no of used bytes */
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 */
u_long drops; /* missed samples from over/underrun */
u_long pdrops; /* paused samples */
char pause; /* transfer is paused */
char copying; /* data is being copied */
char needfill; /* buffer needs filling when copying is done */
char mmapped; /* device is mmap()-ed */
1995-02-21 04:35:58 +03:00
};
#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.
2004-01-31 03:07:56 +03:00
int index; /* index of port-selector mixerctl */
int master; /* index of master mixerctl */
int nports; /* number of selectable ports */
u_char isenum; /* selector is enum type */
u_int allports; /* all aumasks or'd */
u_int aumask[AUDIO_N_PORTS]; /* exposed value of "ports" */
u_int misel [AUDIO_N_PORTS]; /* ord of port, for selector */
u_int miport[AUDIO_N_PORTS]; /* index of port's mixerctl */
u_char isdual; /* has working mixerout */
int mixerout; /* ord of mixerout, for dual case */
int cur_port; /* the port that gain actually controls when
mixerout is selected, for dual case */
};
#if NAURATECONV > 0
struct auconv_context {
long count;
int32_t prev[AUDIO_MAX_CHANNELS];
uint8_t *ring_start;
uint8_t *ring_end;
};
#endif
1995-02-21 04:35:58 +03:00
/*
* Software state, per audio device.
*/
struct audio_softc {
struct device dev;
void *hw_hdl; /* Hardware driver handle */
1995-02-21 04:35:58 +03:00
struct audio_hw_if *hw_if; /* Hardware interface */
struct device *sc_dev; /* Hardware device struct */
1995-02-21 04:35:58 +03:00
u_char sc_open; /* single use device */
#define AUOPEN_READ 0x01
#define AUOPEN_WRITE 0x02
u_char sc_mode; /* bitmask for RECORD/PLAY */
struct selinfo sc_wsel; /* write selector */
struct selinfo sc_rsel; /* read selector */
struct proc *sc_async_audio; /* process who wants audio SIGIO */
struct mixer_asyncs {
struct mixer_asyncs *next;
struct proc *proc;
} *sc_async_mixer; /* processes who want mixer SIGIO */
1995-02-21 04:35:58 +03:00
/* Sleep channels for reading and writing. */
int sc_rchan;
int sc_wchan;
/* Ring buffers, separate for record and play. */
struct audio_ringbuffer sc_rr; /* Record ring */
struct audio_ringbuffer sc_pr; /* Play ring */
#define MAX_SAMPLE_SIZE (AUDIO_MAX_CHANNELS * 24 / 8) /* 6channels x 24bit precision */
int sc_input_fragment_length;
u_char sc_input_fragment[MAX_SAMPLE_SIZE];
int sc_pconvbuffer_size;
u_char *sc_pconvbuffer;
#if NAURATECONV > 0
struct auconv_context sc_pconv;
#endif
int sc_rconvbuffer_size;
int sc_rconvbuffer_begin;
int sc_rconvbuffer_end;
u_char *sc_rconvbuffer;
#if NAURATECONV > 0
struct auconv_context sc_rconv;
#endif
u_char sc_blkset; /* Blocksize has been set */
u_char *sc_sil_start; /* start of silence in buffer */
int sc_sil_count; /* # of silence bytes */
2003-05-03 22:10:37 +04:00
u_char sc_rbus; /* input DMA in progress */
u_char sc_pbus; /* output DMA in progress */
1995-02-21 04:35:58 +03:00
struct audio_params sc_pparams; /* play encoding parameters */
struct audio_params sc_rparams; /* record encoding parameters */
int sc_eof; /* EOF, i.e. zero sized write, counter */
u_long sc_wstamp;
u_long sc_playdrop;
int sc_full_duplex; /* device in full duplex mode */
struct au_mixer_ports sc_inports, sc_outports;
int sc_monitor_port;
int sc_refcnt;
u_char sc_dying;
#ifdef AUDIO_INTR_TIME
u_long sc_pfirstintr; /* first time we saw a play interrupt */
int sc_pnintr; /* number of interrupts */
u_long sc_plastintr; /* last time we saw a play interrupt */
long sc_pblktime; /* nominal time between interrupts */
u_long sc_rfirstintr; /* first time we saw a rec interrupt */
int sc_rnintr; /* number of interrupts */
u_long sc_rlastintr; /* last time we saw a rec interrupt */
long sc_rblktime; /* nominal time between interrupts */
#endif
2002-03-18 02:29:55 +03:00
};
extern int auconv_check_params(const struct audio_params *);
#if NAURATECONV > 0
extern void auconv_init_context(struct auconv_context *, long, long, uint8_t *,
uint8_t *);
extern int auconv_play(struct auconv_context *, const struct audio_params *,
uint8_t *, const uint8_t *, int);
extern int auconv_record(struct auconv_context *, const struct audio_params *,
uint8_t *, const uint8_t *, int);
#endif /* NAURATECONV */
#endif /* _SYS_DEV_AUDIOVAR_H_ */