ON Fri, 24 Aug 2007 01:02:32 +0300
Kai Vehmanen <kvehmanen@email-addr-hidden> Wrote:
>> We have a problem running ecasound on FreeBSD CURRENT (7.0).
>> I tried on a laptop, and a desktop, with two different interfaces,
>> (& two drivers).
> hmm, it would seem something odd is happening with the OSS interface.
> Can anyone else with FreeBSD 7.0 verify this problem?
I will try to investigate, but to help, wavplay works fine or
play (from the sox package). see below, i put the freebsd ports code
of oss.c below.
>> 2. Ecasound does not play anything.
> the dsp is detected as OSS card, but no sound is produced.
> And I think that it's no
>> ecasound -C -i:so52.cdr -ddd -o:/dev/dsp0
> Hmm, you could try passing "-z:nodb" to ecasound (disabled disk i/o
> buffering) and see if that helps.
yes, the warnigs has gone away, but still no sound.
> Hmm, if "-z:nodb" helps to your problem, then it might not be FreeBSD OSS
> causing the problems after all, but instead something goes wrong in the
> disk i/o subsystem. The above errors indicate that the disk i/o buffers
> have run out of data. As the buffer are very larger (multiple seconds
> worth of audio), this should never happen in normal circumstances. And the
> fact that..
> (audioio-db-client) There were total 4140 xruns.
> ... you got this many xruns from disk i/o, means that it's not working at
> all.
in case you want to examine the oss.c from (sox), here it is with our sys/soundcard.h.
many thanks for your help.
Raoul
rmgls@email-addr-hidden
----------------------------------
oss.c from the sox package:
----------------------------------
/*
* Copyright 1997 Chris Bagwell And Sundry Contributors
* This source code is freely redistributable and may be used for
* any purpose. This copyright notice must be maintained.
* Chris Bagwell And Sundry Contributors are not
* responsible for the consequences of using this software.
*
* Direct to Open Sound System (OSS) sound driver
* OSS is a popular unix sound driver for Intel x86 unices (eg. Linux)
* and several other unixes (such as SunOS/Solaris).
* This driver is compatible with OSS original source that was called
* USS, Voxware and TASD.
*
* added by Chris Bagwell (cbagwell@email-addr-hidden) on 2/19/96
* based on info grabed from vplay.c in Voxware snd-utils-3.5 package.
* and on LINUX_PLAYER patches added by Greg Lee
* which was originally from Directo to Sound Blaster device driver (sbdsp.c).
* SBLAST patches by John T. Kohl.
*
* Changes:
*
* Nov. 26, 1999 Stan Brooks <stabro@email-addr-hidden>
* Moved initialization code common to startread and startwrite
* into a single function ossdspinit().
*
*/
#include "st_i.h"
#ifdef HAVE_OSS
#include <unistd.h>
#include <stdlib.h>
#include <stdio.h>
#include <fcntl.h>
#ifdef HAVE_SYS_SOUNDCARD_H /* Please see the end of the mail */
#include <sys/soundcard.h>
#endif
#ifdef HAVE_MACHINE_SOUNDCARD_H /* no */
#include <machine/soundcard.h>
#endif
#include <sys/ioctl.h>
/* common r/w initialization code */
static int ossdspinit(ft_t ft)
{
int sampletype, samplesize, dsp_stereo;
int tmp, rc;
st_fileinfo_t *file = (st_fileinfo_t *)ft->priv;
if (ft->signal.rate == 0.0) ft->signal.rate = 8000;
if (ft->signal.size == -1) ft->signal.size = ST_SIZE_BYTE;
if (ft->signal.size == ST_SIZE_BYTE) {
sampletype = AFMT_U8;
samplesize = 8;
if (ft->signal.encoding == ST_ENCODING_UNKNOWN)
ft->signal.encoding = ST_ENCODING_UNSIGNED;
if (ft->signal.encoding != ST_ENCODING_UNSIGNED) {
st_report("OSS driver only supports unsigned with bytes");
st_report("Forcing to unsigned");
ft->signal.encoding = ST_ENCODING_UNSIGNED;
}
}
else if (ft->signal.size == ST_SIZE_16BIT) {
sampletype = (ST_IS_BIGENDIAN) ? AFMT_S16_BE : AFMT_S16_LE;
samplesize = 16;
if (ft->signal.encoding == ST_ENCODING_UNKNOWN)
ft->signal.encoding = ST_ENCODING_SIGN2;
if (ft->signal.encoding != ST_ENCODING_SIGN2) {
st_report("OSS driver only supports signed with words");
st_report("Forcing to signed linear");
ft->signal.encoding = ST_ENCODING_SIGN2;
}
}
else {
sampletype = (ST_IS_BIGENDIAN) ? AFMT_S16_BE : AFMT_S16_LE;
samplesize = 16;
ft->signal.size = ST_SIZE_16BIT;
ft->signal.encoding = ST_ENCODING_SIGN2;
st_report("OSS driver only supports bytes and words");
st_report("Forcing to signed linear word");
}
if (ft->signal.channels == 0) ft->signal.channels = 1;
else if (ft->signal.channels > 2) ft->signal.channels = 2;
if (ioctl(fileno(ft->fp), SNDCTL_DSP_RESET, 0) < 0)
{
st_fail_errno(ft,ST_EOF,"Unable to reset OSS driver. Possibly accessing an invalid file/device");
return(ST_EOF);
}
/* Query the supported formats and find the best match
*/
rc = ioctl(fileno(ft->fp), SNDCTL_DSP_GETFMTS, &tmp);
if (rc == 0) {
if ((tmp & sampletype) == 0)
{
/* is 16-bit supported? */
if (samplesize == 16 && (tmp & (AFMT_S16_LE|AFMT_S16_BE)) == 0)
{
/* Must not like 16-bits, try 8-bits */
ft->signal.size = ST_SIZE_BYTE;
ft->signal.encoding = ST_ENCODING_UNSIGNED;
st_report("OSS driver doesn't like signed words");
st_report("Forcing to unsigned bytes");
tmp = sampletype = AFMT_U8;
samplesize = 8;
}
/* is 8-bit supported */
else if (samplesize == 8 && (tmp & AFMT_U8) == 0)
{
ft->signal.size = ST_SIZE_16BIT;
ft->signal.encoding = ST_ENCODING_SIGN2;
st_report("OSS driver doesn't like unsigned bytes");
st_report("Forcing to signed words");
sampletype = (ST_IS_BIGENDIAN) ? AFMT_S16_BE : AFMT_S16_LE;
samplesize = 16;
}
/* determine which 16-bit format to use */
if (samplesize == 16 && (tmp & sampletype) == 0)
sampletype = (ST_IS_BIGENDIAN) ? AFMT_S16_LE : AFMT_S16_BE;
}
tmp = sampletype;
rc = ioctl(fileno(ft->fp), SNDCTL_DSP_SETFMT, &tmp);
}
/* Give up and exit */
if (rc < 0 || tmp != sampletype)
{
st_fail_errno(ft,ST_EOF,"Unable to set the sample size to %d", samplesize);
return (ST_EOF);
}
if (samplesize == 16)
ft->signal.reverse_bytes = ST_IS_BIGENDIAN != (sampletype == AFMT_S16_BE);
if (ft->signal.channels == 2) dsp_stereo = 1;
else dsp_stereo = 0;
tmp = dsp_stereo;
if (ioctl(fileno(ft->fp), SNDCTL_DSP_STEREO, &tmp) < 0)
{
st_warn("Couldn't set to %s", dsp_stereo? "stereo":"mono");
dsp_stereo = 0;
}
if (tmp != dsp_stereo)
{
st_warn("Sound card appears to only support %d channels. Overriding format", tmp+1);
ft->signal.channels = tmp + 1;
}
tmp = ft->signal.rate;
if (ioctl (fileno(ft->fp), SNDCTL_DSP_SPEED, &tmp) < 0 ||
(int)ft->signal.rate != tmp) {
/* If the rate the sound card is using is not within 1% of what
* the user specified then override the user setting.
* The only reason not to always override this is because of
* clock-rounding problems. Sound cards will sometimes use
* things like 44101 when you ask for 44100. No need overriding
* this and having strange output file rates for something that
* we can't hear anyways.
*/
if ((int)ft->signal.rate - tmp > (tmp * .01) ||
tmp - (int)ft->signal.rate > (tmp * .01)) {
st_warn("Unable to set audio speed to %d (set to %d)",
ft->signal.rate, tmp);
ft->signal.rate = tmp;
}
}
/* Find out block size to use last because the driver could compute
* its size based on specific rates/formats.
*/
file->size = 0;
ioctl (fileno(ft->fp), SNDCTL_DSP_GETBLKSIZE, &file->size);
if (file->size < 4 || file->size > 65536) {
st_fail_errno(ft,ST_EOF,"Invalid audio buffer size %d", file->size);
return (ST_EOF);
}
file->count = 0;
file->pos = 0;
file->eof = 0;
file->buf = (char *)xmalloc(file->size);
if (ioctl(fileno(ft->fp), SNDCTL_DSP_SYNC, NULL) < 0) {
st_fail_errno(ft,ST_EOF,"Unable to sync dsp");
return (ST_EOF);
}
/* Change to non-buffered I/O */
setvbuf(ft->fp, NULL, _IONBF, sizeof(char) * file->size);
return(ST_SUCCESS);
}
/*
* Do anything required before you start reading samples.
* Read file header.
* Find out sampling rate,
* size and encoding of samples,
* mono/stereo/quad.
*/
static int st_ossdspstartread(ft_t ft)
{
int rc;
rc = ossdspinit(ft);
return rc;
}
static int st_ossdspstartwrite(ft_t ft)
{
return ossdspinit(ft);
}
/* OSS /dev/dsp player */
static const char *ossdspnames[] = {
"ossdsp",
NULL
};
static st_format_t st_ossdsp_format = {
ossdspnames,
NULL,
ST_FILE_DEVICE,
st_ossdspstartread,
st_rawread,
st_rawstopread,
st_ossdspstartwrite,
st_rawwrite,
st_rawstopwrite,
st_format_nothing_seek
};
const st_format_t *st_ossdsp_format_fn(void)
{
return &st_ossdsp_format;
}
#endif
-------------------------------
/*
* soundcard.h
*/
/*-
* Copyright by Hannu Savolainen 1993 / 4Front Technologies 1993-2006
* Modified for the new FreeBSD sound driver by Luigi Rizzo, 1997
*
* 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 AUTHOR 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 AUTHOR
* 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.
*
* $FreeBSD: src/sys/sys/soundcard.h,v 1.48 2006/11/26 11:55:48 netchild Exp $
*/
/*
* Unless coordinating changes with 4Front Technologies, do NOT make any
* modifications to ioctl commands, types, etc. that would break
* compatibility with the OSS API.
*/
#ifndef _SYS_SOUNDCARD_H_
#define _SYS_SOUNDCARD_H_
/*
* If you make modifications to this file, please contact me before
* distributing the modified version. There is already enough
* diversity in the world.
*
* Regards,
* Hannu Savolainen
* hannu@email-addr-hidden
*
**********************************************************************
* PS. The Hacker's Guide to VoxWare available from
* nic.funet.fi:pub/Linux/ALPHA/sound. The file is
* snd-sdk-doc-0.1.ps.gz (gzipped postscript). It contains
* some useful information about programming with VoxWare.
* (NOTE! The pub/Linux/ALPHA/ directories are hidden. You have
* to cd inside them before the files are accessible.)
**********************************************************************
*/
/*
* SOUND_VERSION is only used by the voxware driver. Hopefully apps
* should not depend on it, but rather look at the capabilities
* of the driver in the kernel!
*/
#define SOUND_VERSION 301
#define VOXWARE /* does this have any use ? */
/*
* Supported card ID numbers (Should be somewhere else? We keep
* them here just for compativility with the old driver, but these
* constants are of little or no use).
*/
#define SNDCARD_ADLIB 1
#define SNDCARD_SB 2
#define SNDCARD_PAS 3
#define SNDCARD_GUS 4
#define SNDCARD_MPU401 5
#define SNDCARD_SB16 6
#define SNDCARD_SB16MIDI 7
#define SNDCARD_UART6850 8
#define SNDCARD_GUS16 9
#define SNDCARD_MSS 10
#define SNDCARD_PSS 11
#define SNDCARD_SSCAPE 12
#define SNDCARD_PSS_MPU 13
#define SNDCARD_PSS_MSS 14
#define SNDCARD_SSCAPE_MSS 15
#define SNDCARD_TRXPRO 16
#define SNDCARD_TRXPRO_SB 17
#define SNDCARD_TRXPRO_MPU 18
#define SNDCARD_MAD16 19
#define SNDCARD_MAD16_MPU 20
#define SNDCARD_CS4232 21
#define SNDCARD_CS4232_MPU 22
#define SNDCARD_MAUI 23
#define SNDCARD_PSEUDO_MSS 24
#define SNDCARD_AWE32 25
#define SNDCARD_NSS 26
#define SNDCARD_UART16550 27
#define SNDCARD_OPL 28
#include <sys/types.h>
#include <machine/endian.h>
#ifndef _IOWR
#include <sys/ioccom.h>
#endif /* !_IOWR */
/*
* The first part of this file contains the new FreeBSD sound ioctl
* interface. Tries to minimize the number of different ioctls, and
* to be reasonably general.
*
* 970821: some of the new calls have not been implemented yet.
*/
/*
* the following three calls extend the generic file descriptor
* interface. AIONWRITE is the dual of FIONREAD, i.e. returns the max
* number of bytes for a write operation to be non-blocking.
*
* AIOGSIZE/AIOSSIZE are used to change the behaviour of the device,
* from a character device (default) to a block device. In block mode,
* (not to be confused with blocking mode) the main difference for the
* application is that select() will return only when a complete
* block can be read/written to the device, whereas in character mode
* select will return true when one byte can be exchanged. For audio
* devices, character mode makes select almost useless since one byte
* will always be ready by the next sample time (which is often only a
* handful of microseconds away).
* Use a size of 0 or 1 to return to character mode.
*/
#define AIONWRITE _IOR('A', 10, int) /* get # bytes to write */
struct snd_size {
int play_size;
int rec_size;
};
#define AIOGSIZE _IOR('A', 11, struct snd_size)/* read current blocksize */
#define AIOSSIZE _IOWR('A', 11, struct snd_size) /* sets blocksize */
/*
* The following constants define supported audio formats. The
* encoding follows voxware conventions, i.e. 1 bit for each supported
* format. We extend it by using bit 31 (RO) to indicate full-duplex
* capability, and bit 29 (RO) to indicate that the card supports/
* needs different formats on capture & playback channels.
* Bit 29 (RW) is used to indicate/ask stereo.
*
* The number of bits required to store the sample is:
* o 4 bits for the IDA ADPCM format,
* o 8 bits for 8-bit formats, mu-law and A-law,
* o 16 bits for the 16-bit formats, and
* o 32 bits for the 24/32-bit formats.
* o undefined for the MPEG audio format.
*/
#define AFMT_QUERY 0x00000000 /* Return current format */
#define AFMT_MU_LAW 0x00000001 /* Logarithmic mu-law */
#define AFMT_A_LAW 0x00000002 /* Logarithmic A-law */
#define AFMT_IMA_ADPCM 0x00000004 /* A 4:1 compressed format where 16-bit
* squence represented using the
* the average 4 bits per sample */
#define AFMT_U8 0x00000008 /* Unsigned 8-bit */
#define AFMT_S16_LE 0x00000010 /* Little endian signed 16-bit */
#define AFMT_S16_BE 0x00000020 /* Big endian signed 16-bit */
#define AFMT_S8 0x00000040 /* Signed 8-bit */
#define AFMT_U16_LE 0x00000080 /* Little endian unsigned 16-bit */
#define AFMT_U16_BE 0x00000100 /* Big endian unsigned 16-bit */
#define AFMT_MPEG 0x00000200 /* MPEG MP2/MP3 audio */
#define AFMT_AC3 0x00000400 /* Dolby Digital AC3 */
#if _BYTE_ORDER == _LITTLE_ENDIAN
#define AFMT_S16_NE AFMT_S16_LE /* native endian signed 16 */
#else
#define AFMT_S16_NE AFMT_S16_BE
#endif
/*
* 32-bit formats below used for 24-bit audio data where the data is stored
* in the 24 most significant bits and the least significant bits are not used
* (should be set to 0).
*/
#define AFMT_S32_LE 0x00001000 /* Little endian signed 32-bit */
#define AFMT_S32_BE 0x00002000 /* Big endian signed 32-bit */
#define AFMT_U32_LE 0x00004000 /* Little endian unsigned 32-bit */
#define AFMT_U32_BE 0x00008000 /* Big endian unsigned 32-bit */
#define AFMT_S24_LE 0x00010000 /* Little endian signed 24-bit */
#define AFMT_S24_BE 0x00020000 /* Big endian signed 24-bit */
#define AFMT_U24_LE 0x00040000 /* Little endian unsigned 24-bit */
#define AFMT_U24_BE 0x00080000 /* Big endian unsigned 24-bit */
#define AFMT_STEREO 0x10000000 /* can do/want stereo */
/*
* the following are really capabilities
*/
#define AFMT_WEIRD 0x20000000 /* weird hardware... */
/*
* AFMT_WEIRD reports that the hardware might need to operate
* with different formats in the playback and capture
* channels when operating in full duplex.
* As an example, SoundBlaster16 cards only support U8 in one
* direction and S16 in the other one, and applications should
* be aware of this limitation.
*/
#define AFMT_FULLDUPLEX 0x80000000 /* can do full duplex */
/*
* The following structure is used to get/set format and sampling rate.
* While it would be better to have things such as stereo, bits per
* sample, endiannes, etc split in different variables, it turns out
* that formats are not that many, and not all combinations are possible.
* So we followed the Voxware approach of associating one bit to each
* format.
*/
typedef struct _snd_chan_param {
u_long play_rate; /* sampling rate */
u_long rec_rate; /* sampling rate */
u_long play_format; /* everything describing the format */
u_long rec_format; /* everything describing the format */
} snd_chan_param;
#define AIOGFMT _IOR('f', 12, snd_chan_param) /* get format */
#define AIOSFMT _IOWR('f', 12, snd_chan_param) /* sets format */
/*
* The following structure is used to get/set the mixer setting.
* Up to 32 mixers are supported, each one with up to 32 channels.
*/
typedef struct _snd_mix_param {
u_char subdev; /* which output */
u_char line; /* which input */
u_char left,right; /* volumes, 0..255, 0 = mute */
} snd_mix_param ;
/* XXX AIOGMIX, AIOSMIX not implemented yet */
#define AIOGMIX _IOWR('A', 13, snd_mix_param) /* return mixer status */
#define AIOSMIX _IOWR('A', 14, snd_mix_param) /* sets mixer status */
/*
* channel specifiers used in AIOSTOP and AIOSYNC
*/
#define AIOSYNC_PLAY 0x1 /* play chan */
#define AIOSYNC_CAPTURE 0x2 /* capture chan */
/* AIOSTOP stop & flush a channel, returns the residual count */
#define AIOSTOP _IOWR ('A', 15, int)
/* alternate method used to notify the sync condition */
#define AIOSYNC_SIGNAL 0x100
#define AIOSYNC_SELECT 0x200
/* what the 'pos' field refers to */
#define AIOSYNC_READY 0x400
#define AIOSYNC_FREE 0x800
typedef struct _snd_sync_parm {
long chan ; /* play or capture channel, plus modifier */
long pos;
} snd_sync_parm;
#define AIOSYNC _IOWR ('A', 15, snd_sync_parm) /* misc. synchronization */
/*
* The following is used to return device capabilities. If the structure
* passed to the ioctl is zeroed, default values are returned for rate
* and formats, a bitmap of available mixers is returned, and values
* (inputs, different levels) for the first one are returned.
*
* If formats, mixers, inputs are instantiated, then detailed info
* are returned depending on the call.
*/
typedef struct _snd_capabilities {
u_long rate_min, rate_max; /* min-max sampling rate */
u_long formats;
u_long bufsize; /* DMA buffer size */
u_long mixers; /* bitmap of available mixers */
u_long inputs; /* bitmap of available inputs (per mixer) */
u_short left, right; /* how many levels are supported */
} snd_capabilities;
#define AIOGCAP _IOWR('A', 15, snd_capabilities) /* get capabilities */
/*
* here is the old (Voxware) ioctl interface
*/
/*
* IOCTL Commands for /dev/sequencer
*/
#define SNDCTL_SEQ_RESET _IO ('Q', 0)
#define SNDCTL_SEQ_SYNC _IO ('Q', 1)
#define SNDCTL_SYNTH_INFO _IOWR('Q', 2, struct synth_info)
#define SNDCTL_SEQ_CTRLRATE _IOWR('Q', 3, int) /* Set/get timer res.(hz) */
#define SNDCTL_SEQ_GETOUTCOUNT _IOR ('Q', 4, int)
#define SNDCTL_SEQ_GETINCOUNT _IOR ('Q', 5, int)
#define SNDCTL_SEQ_PERCMODE _IOW ('Q', 6, int)
#define SNDCTL_FM_LOAD_INSTR _IOW ('Q', 7, struct sbi_instrument) /* Valid for FM only */
#define SNDCTL_SEQ_TESTMIDI _IOW ('Q', 8, int)
#define SNDCTL_SEQ_RESETSAMPLES _IOW ('Q', 9, int)
#define SNDCTL_SEQ_NRSYNTHS _IOR ('Q',10, int)
#define SNDCTL_SEQ_NRMIDIS _IOR ('Q',11, int)
#define SNDCTL_MIDI_INFO _IOWR('Q',12, struct midi_info)
#define SNDCTL_SEQ_THRESHOLD _IOW ('Q',13, int)
#define SNDCTL_SEQ_TRESHOLD SNDCTL_SEQ_THRESHOLD /* there was once a typo */
#define SNDCTL_SYNTH_MEMAVL _IOWR('Q',14, int) /* in=dev#, out=memsize */
#define SNDCTL_FM_4OP_ENABLE _IOW ('Q',15, int) /* in=dev# */
#define SNDCTL_PMGR_ACCESS _IOWR('Q',16, struct patmgr_info)
#define SNDCTL_SEQ_PANIC _IO ('Q',17)
#define SNDCTL_SEQ_OUTOFBAND _IOW ('Q',18, struct seq_event_rec)
#define SNDCTL_SEQ_GETTIME _IOR ('Q',19, int)
struct seq_event_rec {
u_char arr[8];
};
#define SNDCTL_TMR_TIMEBASE _IOWR('T', 1, int)
#define SNDCTL_TMR_START _IO ('T', 2)
#define SNDCTL_TMR_STOP _IO ('T', 3)
#define SNDCTL_TMR_CONTINUE _IO ('T', 4)
#define SNDCTL_TMR_TEMPO _IOWR('T', 5, int)
#define SNDCTL_TMR_SOURCE _IOWR('T', 6, int)
# define TMR_INTERNAL 0x00000001
# define TMR_EXTERNAL 0x00000002
# define TMR_MODE_MIDI 0x00000010
# define TMR_MODE_FSK 0x00000020
# define TMR_MODE_CLS 0x00000040
# define TMR_MODE_SMPTE 0x00000080
#define SNDCTL_TMR_METRONOME _IOW ('T', 7, int)
#define SNDCTL_TMR_SELECT _IOW ('T', 8, int)
/*
* Endian aware patch key generation algorithm.
*/
#if defined(_AIX) || defined(AIX)
# define _PATCHKEY(id) (0xfd00|id)
#else
# define _PATCHKEY(id) ((id<<8)|0xfd)
#endif
/*
* Sample loading mechanism for internal synthesizers (/dev/sequencer)
* The following patch_info structure has been designed to support
* Gravis UltraSound. It tries to be universal format for uploading
* sample based patches but is probably too limited.
*/
struct patch_info {
/* u_short key; Use GUS_PATCH here */
short key; /* Use GUS_PATCH here */
#define GUS_PATCH _PATCHKEY(0x04)
#define OBSOLETE_GUS_PATCH _PATCHKEY(0x02)
short device_no; /* Synthesizer number */
short instr_no; /* Midi pgm# */
u_long mode;
/*
* The least significant byte has the same format than the GUS .PAT
* files
*/
#define WAVE_16_BITS 0x01 /* bit 0 = 8 or 16 bit wave data. */
#define WAVE_UNSIGNED 0x02 /* bit 1 = Signed - Unsigned data. */
#define WAVE_LOOPING 0x04 /* bit 2 = looping enabled-1. */
#define WAVE_BIDIR_LOOP 0x08 /* bit 3 = Set is bidirectional looping. */
#define WAVE_LOOP_BACK 0x10 /* bit 4 = Set is looping backward. */
#define WAVE_SUSTAIN_ON 0x20 /* bit 5 = Turn sustaining on. (Env. pts. 3)*/
#define WAVE_ENVELOPES 0x40 /* bit 6 = Enable envelopes - 1 */
/* (use the env_rate/env_offs fields). */
/* Linux specific bits */
#define WAVE_VIBRATO 0x00010000 /* The vibrato info is valid */
#define WAVE_TREMOLO 0x00020000 /* The tremolo info is valid */
#define WAVE_SCALE 0x00040000 /* The scaling info is valid */
/* Other bits must be zeroed */
long len; /* Size of the wave data in bytes */
long loop_start, loop_end; /* Byte offsets from the beginning */
/*
* The base_freq and base_note fields are used when computing the
* playback speed for a note. The base_note defines the tone frequency
* which is heard if the sample is played using the base_freq as the
* playback speed.
*
* The low_note and high_note fields define the minimum and maximum note
* frequencies for which this sample is valid. It is possible to define
* more than one samples for an instrument number at the same time. The
* low_note and high_note fields are used to select the most suitable one.
*
* The fields base_note, high_note and low_note should contain
* the note frequency multiplied by 1000. For example value for the
* middle A is 440*1000.
*/
u_int base_freq;
u_long base_note;
u_long high_note;
u_long low_note;
int panning; /* -128=left, 127=right */
int detuning;
/* New fields introduced in version 1.99.5 */
/* Envelope. Enabled by mode bit WAVE_ENVELOPES */
u_char env_rate[ 6 ]; /* GUS HW ramping rate */
u_char env_offset[ 6 ]; /* 255 == 100% */
/*
* The tremolo, vibrato and scale info are not supported yet.
* Enable by setting the mode bits WAVE_TREMOLO, WAVE_VIBRATO or
* WAVE_SCALE
*/
u_char tremolo_sweep;
u_char tremolo_rate;
u_char tremolo_depth;
u_char vibrato_sweep;
u_char vibrato_rate;
u_char vibrato_depth;
int scale_frequency;
u_int scale_factor; /* from 0 to 2048 or 0 to 2 */
int volume;
int spare[4];
char data[1]; /* The waveform data starts here */
};
struct sysex_info {
short key; /* Use GUS_PATCH here */
#define SYSEX_PATCH _PATCHKEY(0x05)
#define MAUI_PATCH _PATCHKEY(0x06)
short device_no; /* Synthesizer number */
long len; /* Size of the sysex data in bytes */
u_char data[1]; /* Sysex data starts here */
};
/*
* Patch management interface (/dev/sequencer, /dev/patmgr#)
* Don't use these calls if you want to maintain compatibility with
* the future versions of the driver.
*/
#define PS_NO_PATCHES 0 /* No patch support on device */
#define PS_MGR_NOT_OK 1 /* Plain patch support (no mgr) */
#define PS_MGR_OK 2 /* Patch manager supported */
#define PS_MANAGED 3 /* Patch manager running */
#define SNDCTL_PMGR_IFACE _IOWR('P', 1, struct patmgr_info)
/*
* The patmgr_info is a fixed size structure which is used for two
* different purposes. The intended use is for communication between
* the application using /dev/sequencer and the patch manager daemon
* associated with a synthesizer device (ioctl(SNDCTL_PMGR_ACCESS)).
*
* This structure is also used with ioctl(SNDCTL_PGMR_IFACE) which allows
* a patch manager daemon to read and write device parameters. This
* ioctl available through /dev/sequencer also. Avoid using it since it's
* extremely hardware dependent. In addition access trough /dev/sequencer
* may confuse the patch manager daemon.
*/
struct patmgr_info { /* Note! size must be < 4k since kmalloc() is used */
u_long key; /* Don't worry. Reserved for communication
between the patch manager and the driver. */
#define PM_K_EVENT 1 /* Event from the /dev/sequencer driver */
#define PM_K_COMMAND 2 /* Request from an application */
#define PM_K_RESPONSE 3 /* From patmgr to application */
#define PM_ERROR 4 /* Error returned by the patmgr */
int device;
int command;
/*
* Commands 0x000 to 0xfff reserved for patch manager programs
*/
#define PM_GET_DEVTYPE 1 /* Returns type of the patch mgr interface of dev */
#define PMTYPE_FM2 1 /* 2 OP fm */
#define PMTYPE_FM4 2 /* Mixed 4 or 2 op FM (OPL-3) */
#define PMTYPE_WAVE 3 /* Wave table synthesizer (GUS) */
#define PM_GET_NRPGM 2 /* Returns max # of midi programs in parm1 */
#define PM_GET_PGMMAP 3 /* Returns map of loaded midi programs in data8 */
#define PM_GET_PGM_PATCHES 4 /* Return list of patches of a program (parm1) */
#define PM_GET_PATCH 5 /* Return patch header of patch parm1 */
#define PM_SET_PATCH 6 /* Set patch header of patch parm1 */
#define PM_READ_PATCH 7 /* Read patch (wave) data */
#define PM_WRITE_PATCH 8 /* Write patch (wave) data */
/*
* Commands 0x1000 to 0xffff are for communication between the patch manager
* and the client
*/
#define _PM_LOAD_PATCH 0x100
/*
* Commands above 0xffff reserved for device specific use
*/
long parm1;
long parm2;
long parm3;
union {
u_char data8[4000];
u_short data16[2000];
u_long data32[1000];
struct patch_info patch;
} data;
};
/*
* When a patch manager daemon is present, it will be informed by the
* driver when something important happens. For example when the
* /dev/sequencer is opened or closed. A record with key == PM_K_EVENT is
* returned. The command field contains the event type:
*/
#define PM_E_OPENED 1 /* /dev/sequencer opened */
#define PM_E_CLOSED 2 /* /dev/sequencer closed */
#define PM_E_PATCH_RESET 3 /* SNDCTL_RESETSAMPLES called */
#define PM_E_PATCH_LOADED 4 /* A patch has been loaded by appl */
/*
* /dev/sequencer input events.
*
* The data written to the /dev/sequencer is a stream of events. Events
* are records of 4 or 8 bytes. The first byte defines the size.
* Any number of events can be written with a write call. There
* is a set of macros for sending these events. Use these macros if you
* want to maximize portability of your program.
*
* Events SEQ_WAIT, SEQ_MIDIPUTC and SEQ_ECHO. Are also input events.
* (All input events are currently 4 bytes long. Be prepared to support
* 8 byte events also. If you receive any event having first byte >= 128,
* it's a 8 byte event.
*
* The events are documented at the end of this file.
*
* Normal events (4 bytes)
* There is also a 8 byte version of most of the 4 byte events. The
* 8 byte one is recommended.
*/
#define SEQ_NOTEOFF 0
#define SEQ_FMNOTEOFF SEQ_NOTEOFF /* Just old name */
#define SEQ_NOTEON 1
#define SEQ_FMNOTEON SEQ_NOTEON
#define SEQ_WAIT TMR_WAIT_ABS
#define SEQ_PGMCHANGE 3
#define SEQ_FMPGMCHANGE SEQ_PGMCHANGE
#define SEQ_SYNCTIMER TMR_START
#define SEQ_MIDIPUTC 5
#define SEQ_DRUMON 6 /*** OBSOLETE ***/
#define SEQ_DRUMOFF 7 /*** OBSOLETE ***/
#define SEQ_ECHO TMR_ECHO /* For synching programs with output */
#define SEQ_AFTERTOUCH 9
#define SEQ_CONTROLLER 10
/*
* Midi controller numbers
*
* Controllers 0 to 31 (0x00 to 0x1f) and 32 to 63 (0x20 to 0x3f)
* are continuous controllers.
* In the MIDI 1.0 these controllers are sent using two messages.
* Controller numbers 0 to 31 are used to send the MSB and the
* controller numbers 32 to 63 are for the LSB. Note that just 7 bits
* are used in MIDI bytes.
*/
#define CTL_BANK_SELECT 0x00
#define CTL_MODWHEEL 0x01
#define CTL_BREATH 0x02
/* undefined 0x03 */
#define CTL_FOOT 0x04
#define CTL_PORTAMENTO_TIME 0x05
#define CTL_DATA_ENTRY 0x06
#define CTL_MAIN_VOLUME 0x07
#define CTL_BALANCE 0x08
/* undefined 0x09 */
#define CTL_PAN 0x0a
#define CTL_EXPRESSION 0x0b
/* undefined 0x0c - 0x0f */
#define CTL_GENERAL_PURPOSE1 0x10
#define CTL_GENERAL_PURPOSE2 0x11
#define CTL_GENERAL_PURPOSE3 0x12
#define CTL_GENERAL_PURPOSE4 0x13
/* undefined 0x14 - 0x1f */
/* undefined 0x20 */
/*
* The controller numbers 0x21 to 0x3f are reserved for the
* least significant bytes of the controllers 0x00 to 0x1f.
* These controllers are not recognised by the driver.
*
* Controllers 64 to 69 (0x40 to 0x45) are on/off switches.
* 0=OFF and 127=ON (intermediate values are possible)
*/
#define CTL_DAMPER_PEDAL 0x40
#define CTL_SUSTAIN CTL_DAMPER_PEDAL /* Alias */
#define CTL_HOLD CTL_DAMPER_PEDAL /* Alias */
#define CTL_PORTAMENTO 0x41
#define CTL_SOSTENUTO 0x42
#define CTL_SOFT_PEDAL 0x43
/* undefined 0x44 */
#define CTL_HOLD2 0x45
/* undefined 0x46 - 0x4f */
#define CTL_GENERAL_PURPOSE5 0x50
#define CTL_GENERAL_PURPOSE6 0x51
#define CTL_GENERAL_PURPOSE7 0x52
#define CTL_GENERAL_PURPOSE8 0x53
/* undefined 0x54 - 0x5a */
#define CTL_EXT_EFF_DEPTH 0x5b
#define CTL_TREMOLO_DEPTH 0x5c
#define CTL_CHORUS_DEPTH 0x5d
#define CTL_DETUNE_DEPTH 0x5e
#define CTL_CELESTE_DEPTH CTL_DETUNE_DEPTH /* Alias for the above one */
#define CTL_PHASER_DEPTH 0x5f
#define CTL_DATA_INCREMENT 0x60
#define CTL_DATA_DECREMENT 0x61
#define CTL_NONREG_PARM_NUM_LSB 0x62
#define CTL_NONREG_PARM_NUM_MSB 0x63
#define CTL_REGIST_PARM_NUM_LSB 0x64
#define CTL_REGIST_PARM_NUM_MSB 0x65
/* undefined 0x66 - 0x78 */
/* reserved 0x79 - 0x7f */
/* Pseudo controllers (not midi compatible) */
#define CTRL_PITCH_BENDER 255
#define CTRL_PITCH_BENDER_RANGE 254
#define CTRL_EXPRESSION 253 /* Obsolete */
#define CTRL_MAIN_VOLUME 252 /* Obsolete */
#define SEQ_BALANCE 11
#define SEQ_VOLMODE 12
/*
* Volume mode decides how volumes are used
*/
#define VOL_METHOD_ADAGIO 1
#define VOL_METHOD_LINEAR 2
/*
* Note! SEQ_WAIT, SEQ_MIDIPUTC and SEQ_ECHO are used also as
* input events.
*/
/*
* Event codes 0xf0 to 0xfc are reserved for future extensions.
*/
#define SEQ_FULLSIZE 0xfd /* Long events */
/*
* SEQ_FULLSIZE events are used for loading patches/samples to the
* synthesizer devices. These events are passed directly to the driver
* of the associated synthesizer device. There is no limit to the size
* of the extended events. These events are not queued but executed
* immediately when the write() is called (execution can take several
* seconds of time).
*
* When a SEQ_FULLSIZE message is written to the device, it must
* be written using exactly one write() call. Other events cannot
* be mixed to the same write.
*
* For FM synths (YM3812/OPL3) use struct sbi_instrument and write
* it to the /dev/sequencer. Don't write other data together with
* the instrument structure Set the key field of the structure to
* FM_PATCH. The device field is used to route the patch to the
* corresponding device.
*
* For Gravis UltraSound use struct patch_info. Initialize the key field
* to GUS_PATCH.
*/
#define SEQ_PRIVATE 0xfe /* Low level HW dependent events (8 bytes) */
#define SEQ_EXTENDED 0xff /* Extended events (8 bytes) OBSOLETE */
/*
* Record for FM patches
*/
typedef u_char sbi_instr_data[32];
struct sbi_instrument {
u_short key; /* FM_PATCH or OPL3_PATCH */
#define FM_PATCH _PATCHKEY(0x01)
#define OPL3_PATCH _PATCHKEY(0x03)
short device; /* Synth# (0-4) */
int channel; /* Program# to be initialized */
sbi_instr_data operators; /* Reg. settings for operator cells
* (.SBI format) */
};
struct synth_info { /* Read only */
char name[30];
int device; /* 0-N. INITIALIZE BEFORE CALLING */
int synth_type;
#define SYNTH_TYPE_FM 0
#define SYNTH_TYPE_SAMPLE 1
#define SYNTH_TYPE_MIDI 2 /* Midi interface */
int synth_subtype;
#define FM_TYPE_ADLIB 0x00
#define FM_TYPE_OPL3 0x01
#define MIDI_TYPE_MPU401 0x401
#define SAMPLE_TYPE_BASIC 0x10
#define SAMPLE_TYPE_GUS SAMPLE_TYPE_BASIC
#define SAMPLE_TYPE_AWE32 0x20
int perc_mode; /* No longer supported */
int nr_voices;
int nr_drums; /* Obsolete field */
int instr_bank_size;
u_long capabilities;
#define SYNTH_CAP_PERCMODE 0x00000001 /* No longer used */
#define SYNTH_CAP_OPL3 0x00000002 /* Set if OPL3 supported */
#define SYNTH_CAP_INPUT 0x00000004 /* Input (MIDI) device */
int dummies[19]; /* Reserve space */
};
struct sound_timer_info {
char name[32];
int caps;
};
struct midi_info {
char name[30];
int device; /* 0-N. INITIALIZE BEFORE CALLING */
u_long capabilities; /* To be defined later */
int dev_type;
int dummies[18]; /* Reserve space */
};
/*
* ioctl commands for the /dev/midi##
*/
typedef struct {
u_char cmd;
char nr_args, nr_returns;
u_char data[30];
} mpu_command_rec;
#define SNDCTL_MIDI_PRETIME _IOWR('m', 0, int)
#define SNDCTL_MIDI_MPUMODE _IOWR('m', 1, int)
#define SNDCTL_MIDI_MPUCMD _IOWR('m', 2, mpu_command_rec)
#define MIOSPASSTHRU _IOWR('m', 3, int)
#define MIOGPASSTHRU _IOWR('m', 4, int)
/*
* IOCTL commands for /dev/dsp and /dev/audio
*/
#define SNDCTL_DSP_RESET _IO ('P', 0)
#define SNDCTL_DSP_SYNC _IO ('P', 1)
#define SNDCTL_DSP_SPEED _IOWR('P', 2, int)
#define SNDCTL_DSP_STEREO _IOWR('P', 3, int)
#define SNDCTL_DSP_GETBLKSIZE _IOR('P', 4, int)
#define SNDCTL_DSP_SETBLKSIZE _IOW('P', 4, int)
#define SNDCTL_DSP_SETFMT _IOWR('P',5, int) /* Selects ONE fmt*/
/*
* SOUND_PCM_WRITE_CHANNELS is not that different
* from SNDCTL_DSP_STEREO
*/
#define SOUND_PCM_WRITE_CHANNELS _IOWR('P', 6, int)
#define SNDCTL_DSP_CHANNELS SOUND_PCM_WRITE_CHANNELS
#define SOUND_PCM_WRITE_FILTER _IOWR('P', 7, int)
#define SNDCTL_DSP_POST _IO ('P', 8)
/*
* SNDCTL_DSP_SETBLKSIZE and the following two calls mostly do
* the same thing, i.e. set the block size used in DMA transfers.
*/
#define SNDCTL_DSP_SUBDIVIDE _IOWR('P', 9, int)
#define SNDCTL_DSP_SETFRAGMENT _IOWR('P',10, int)
#define SNDCTL_DSP_GETFMTS _IOR ('P',11, int) /* Returns a mask */
/*
* Buffer status queries.
*/
typedef struct audio_buf_info {
int fragments; /* # of avail. frags (partly used ones not counted) */
int fragstotal; /* Total # of fragments allocated */
int fragsize; /* Size of a fragment in bytes */
int bytes; /* Avail. space in bytes (includes partly used fragments) */
/* Note! 'bytes' could be more than fragments*fragsize */
} audio_buf_info;
#define SNDCTL_DSP_GETOSPACE _IOR ('P',12, audio_buf_info)
#define SNDCTL_DSP_GETISPACE _IOR ('P',13, audio_buf_info)
/*
* SNDCTL_DSP_NONBLOCK is the same (but less powerful, since the
* action cannot be undone) of FIONBIO. The same can be achieved
* by opening the device with O_NDELAY
*/
#define SNDCTL_DSP_NONBLOCK _IO ('P',14)
#define SNDCTL_DSP_GETCAPS _IOR ('P',15, int)
#define DSP_CAP_REVISION 0x000000ff /* revision level (0 to 255) */
#define DSP_CAP_DUPLEX 0x00000100 /* Full duplex record/playback */
#define DSP_CAP_REALTIME 0x00000200 /* Real time capability */
#define DSP_CAP_BATCH 0x00000400
/*
* Device has some kind of internal buffers which may
* cause some delays and decrease precision of timing
*/
#define DSP_CAP_COPROC 0x00000800
/* Has a coprocessor, sometimes it's a DSP but usually not */
#define DSP_CAP_TRIGGER 0x00001000 /* Supports SETTRIGGER */
#define DSP_CAP_MMAP 0x00002000 /* Supports mmap() */
/*
* What do these function do ?
*/
#define SNDCTL_DSP_GETTRIGGER _IOR ('P',16, int)
#define SNDCTL_DSP_SETTRIGGER _IOW ('P',16, int)
#define PCM_ENABLE_INPUT 0x00000001
#define PCM_ENABLE_OUTPUT 0x00000002
typedef struct count_info {
int bytes; /* Total # of bytes processed */
int blocks; /* # of fragment transitions since last time */
int ptr; /* Current DMA pointer value */
} count_info;
/*
* GETIPTR and GETISPACE are not that different... same for out.
*/
#define SNDCTL_DSP_GETIPTR _IOR ('P',17, count_info)
#define SNDCTL_DSP_GETOPTR _IOR ('P',18, count_info)
typedef struct buffmem_desc {
caddr_t buffer;
int size;
} buffmem_desc;
#define SNDCTL_DSP_MAPINBUF _IOR ('P', 19, buffmem_desc)
#define SNDCTL_DSP_MAPOUTBUF _IOR ('P', 20, buffmem_desc)
#define SNDCTL_DSP_SETSYNCRO _IO ('P', 21)
#define SNDCTL_DSP_SETDUPLEX _IO ('P', 22)
#define SNDCTL_DSP_GETODELAY _IOR ('P', 23, int)
/*
* I guess these are the readonly version of the same
* functions that exist above as SNDCTL_DSP_...
*/
#define SOUND_PCM_READ_RATE _IOR ('P', 2, int)
#define SOUND_PCM_READ_CHANNELS _IOR ('P', 6, int)
#define SOUND_PCM_READ_BITS _IOR ('P', 5, int)
#define SOUND_PCM_READ_FILTER _IOR ('P', 7, int)
/*
* ioctl calls to be used in communication with coprocessors and
* DSP chips.
*/
typedef struct copr_buffer {
int command; /* Set to 0 if not used */
int flags;
#define CPF_NONE 0x0000
#define CPF_FIRST 0x0001 /* First block */
#define CPF_LAST 0x0002 /* Last block */
int len;
int offs; /* If required by the device (0 if not used) */
u_char data[4000]; /* NOTE! 4000 is not 4k */
} copr_buffer;
typedef struct copr_debug_buf {
int command; /* Used internally. Set to 0 */
int parm1;
int parm2;
int flags;
int len; /* Length of data in bytes */
} copr_debug_buf;
typedef struct copr_msg {
int len;
u_char data[4000];
} copr_msg;
#define SNDCTL_COPR_RESET _IO ('C', 0)
#define SNDCTL_COPR_LOAD _IOWR('C', 1, copr_buffer)
#define SNDCTL_COPR_RDATA _IOWR('C', 2, copr_debug_buf)
#define SNDCTL_COPR_RCODE _IOWR('C', 3, copr_debug_buf)
#define SNDCTL_COPR_WDATA _IOW ('C', 4, copr_debug_buf)
#define SNDCTL_COPR_WCODE _IOW ('C', 5, copr_debug_buf)
#define SNDCTL_COPR_RUN _IOWR('C', 6, copr_debug_buf)
#define SNDCTL_COPR_HALT _IOWR('C', 7, copr_debug_buf)
#define SNDCTL_COPR_SENDMSG _IOW ('C', 8, copr_msg)
#define SNDCTL_COPR_RCVMSG _IOR ('C', 9, copr_msg)
/*
* IOCTL commands for /dev/mixer
*/
/*
* Mixer devices
*
* There can be up to 20 different analog mixer channels. The
* SOUND_MIXER_NRDEVICES gives the currently supported maximum.
* The SOUND_MIXER_READ_DEVMASK returns a bitmask which tells
* the devices supported by the particular mixer.
*/
#define SOUND_MIXER_NRDEVICES 25
#define SOUND_MIXER_VOLUME 0 /* Master output level */
#define SOUND_MIXER_BASS 1 /* Treble level of all output channels */
#define SOUND_MIXER_TREBLE 2 /* Bass level of all output channels */
#define SOUND_MIXER_SYNTH 3 /* Volume of synthesier input */
#define SOUND_MIXER_PCM 4 /* Output level for the audio device */
#define SOUND_MIXER_SPEAKER 5 /* Output level for the PC speaker
* signals */
#define SOUND_MIXER_LINE 6 /* Volume level for the line in jack */
#define SOUND_MIXER_MIC 7 /* Volume for the signal coming from
* the microphone jack */
#define SOUND_MIXER_CD 8 /* Volume level for the input signal
* connected to the CD audio input */
#define SOUND_MIXER_IMIX 9 /* Recording monitor. It controls the
* output volume of the selected
* recording sources while recording */
#define SOUND_MIXER_ALTPCM 10 /* Volume of the alternative codec
* device */
#define SOUND_MIXER_RECLEV 11 /* Global recording level */
#define SOUND_MIXER_IGAIN 12 /* Input gain */
#define SOUND_MIXER_OGAIN 13 /* Output gain */
/*
* The AD1848 codec and compatibles have three line level inputs
* (line, aux1 and aux2). Since each card manufacturer have assigned
* different meanings to these inputs, it's inpractical to assign
* specific meanings (line, cd, synth etc.) to them.
*/
#define SOUND_MIXER_LINE1 14 /* Input source 1 (aux1) */
#define SOUND_MIXER_LINE2 15 /* Input source 2 (aux2) */
#define SOUND_MIXER_LINE3 16 /* Input source 3 (line) */
#define SOUND_MIXER_DIGITAL1 17 /* Digital (input) 1 */
#define SOUND_MIXER_DIGITAL2 18 /* Digital (input) 2 */
#define SOUND_MIXER_DIGITAL3 19 /* Digital (input) 3 */
#define SOUND_MIXER_PHONEIN 20 /* Phone input */
#define SOUND_MIXER_PHONEOUT 21 /* Phone output */
#define SOUND_MIXER_VIDEO 22 /* Video/TV (audio) in */
#define SOUND_MIXER_RADIO 23 /* Radio in */
#define SOUND_MIXER_MONITOR 24 /* Monitor (usually mic) volume */
/*
* Some on/off settings (SOUND_SPECIAL_MIN - SOUND_SPECIAL_MAX)
* Not counted to SOUND_MIXER_NRDEVICES, but use the same number space
*/
#define SOUND_ONOFF_MIN 28
#define SOUND_ONOFF_MAX 30
#define SOUND_MIXER_MUTE 28 /* 0 or 1 */
#define SOUND_MIXER_ENHANCE 29 /* Enhanced stereo (0, 40, 60 or 80) */
#define SOUND_MIXER_LOUD 30 /* 0 or 1 */
/* Note! Number 31 cannot be used since the sign bit is reserved */
#define SOUND_MIXER_NONE 31
#define SOUND_DEVICE_LABELS { \
"Vol ", "Bass ", "Trebl", "Synth", "Pcm ", "Spkr ", "Line ", \
"Mic ", "CD ", "Mix ", "Pcm2 ", "Rec ", "IGain", "OGain", \
"Line1", "Line2", "Line3", "Digital1", "Digital2", "Digital3", \
"PhoneIn", "PhoneOut", "Video", "Radio", "Monitor"}
#define SOUND_DEVICE_NAMES { \
"vol", "bass", "treble", "synth", "pcm", "speaker", "line", \
"mic", "cd", "mix", "pcm2", "rec", "igain", "ogain", \
"line1", "line2", "line3", "dig1", "dig2", "dig3", \
"phin", "phout", "video", "radio", "monitor"}
/* Device bitmask identifiers */
#define SOUND_MIXER_RECSRC 0xff /* 1 bit per recording source */
#define SOUND_MIXER_DEVMASK 0xfe /* 1 bit per supported device */
#define SOUND_MIXER_RECMASK 0xfd /* 1 bit per supp. recording source */
#define SOUND_MIXER_CAPS 0xfc
#define SOUND_CAP_EXCL_INPUT 0x00000001 /* Only 1 rec. src at a time */
#define SOUND_MIXER_STEREODEVS 0xfb /* Mixer channels supporting stereo */
/* Device mask bits */
#define SOUND_MASK_VOLUME (1 << SOUND_MIXER_VOLUME)
#define SOUND_MASK_BASS (1 << SOUND_MIXER_BASS)
#define SOUND_MASK_TREBLE (1 << SOUND_MIXER_TREBLE)
#define SOUND_MASK_SYNTH (1 << SOUND_MIXER_SYNTH)
#define SOUND_MASK_PCM (1 << SOUND_MIXER_PCM)
#define SOUND_MASK_SPEAKER (1 << SOUND_MIXER_SPEAKER)
#define SOUND_MASK_LINE (1 << SOUND_MIXER_LINE)
#define SOUND_MASK_MIC (1 << SOUND_MIXER_MIC)
#define SOUND_MASK_CD (1 << SOUND_MIXER_CD)
#define SOUND_MASK_IMIX (1 << SOUND_MIXER_IMIX)
#define SOUND_MASK_ALTPCM (1 << SOUND_MIXER_ALTPCM)
#define SOUND_MASK_RECLEV (1 << SOUND_MIXER_RECLEV)
#define SOUND_MASK_IGAIN (1 << SOUND_MIXER_IGAIN)
#define SOUND_MASK_OGAIN (1 << SOUND_MIXER_OGAIN)
#define SOUND_MASK_LINE1 (1 << SOUND_MIXER_LINE1)
#define SOUND_MASK_LINE2 (1 << SOUND_MIXER_LINE2)
#define SOUND_MASK_LINE3 (1 << SOUND_MIXER_LINE3)
#define SOUND_MASK_DIGITAL1 (1 << SOUND_MIXER_DIGITAL1)
#define SOUND_MASK_DIGITAL2 (1 << SOUND_MIXER_DIGITAL2)
#define SOUND_MASK_DIGITAL3 (1 << SOUND_MIXER_DIGITAL3)
#define SOUND_MASK_PHONEIN (1 << SOUND_MIXER_PHONEIN)
#define SOUND_MASK_PHONEOUT (1 << SOUND_MIXER_PHONEOUT)
#define SOUND_MASK_RADIO (1 << SOUND_MIXER_RADIO)
#define SOUND_MASK_VIDEO (1 << SOUND_MIXER_VIDEO)
#define SOUND_MASK_MONITOR (1 << SOUND_MIXER_MONITOR)
/* Obsolete macros */
#define SOUND_MASK_MUTE (1 << SOUND_MIXER_MUTE)
#define SOUND_MASK_ENHANCE (1 << SOUND_MIXER_ENHANCE)
#define SOUND_MASK_LOUD (1 << SOUND_MIXER_LOUD)
#define MIXER_READ(dev) _IOR('M', dev, int)
#define SOUND_MIXER_READ_VOLUME MIXER_READ(SOUND_MIXER_VOLUME)
#define SOUND_MIXER_READ_BASS MIXER_READ(SOUND_MIXER_BASS)
#define SOUND_MIXER_READ_TREBLE MIXER_READ(SOUND_MIXER_TREBLE)
#define SOUND_MIXER_READ_SYNTH MIXER_READ(SOUND_MIXER_SYNTH)
#define SOUND_MIXER_READ_PCM MIXER_READ(SOUND_MIXER_PCM)
#define SOUND_MIXER_READ_SPEAKER MIXER_READ(SOUND_MIXER_SPEAKER)
#define SOUND_MIXER_READ_LINE MIXER_READ(SOUND_MIXER_LINE)
#define SOUND_MIXER_READ_MIC MIXER_READ(SOUND_MIXER_MIC)
#define SOUND_MIXER_READ_CD MIXER_READ(SOUND_MIXER_CD)
#define SOUND_MIXER_READ_IMIX MIXER_READ(SOUND_MIXER_IMIX)
#define SOUND_MIXER_READ_ALTPCM MIXER_READ(SOUND_MIXER_ALTPCM)
#define SOUND_MIXER_READ_RECLEV MIXER_READ(SOUND_MIXER_RECLEV)
#define SOUND_MIXER_READ_IGAIN MIXER_READ(SOUND_MIXER_IGAIN)
#define SOUND_MIXER_READ_OGAIN MIXER_READ(SOUND_MIXER_OGAIN)
#define SOUND_MIXER_READ_LINE1 MIXER_READ(SOUND_MIXER_LINE1)
#define SOUND_MIXER_READ_LINE2 MIXER_READ(SOUND_MIXER_LINE2)
#define SOUND_MIXER_READ_LINE3 MIXER_READ(SOUND_MIXER_LINE3)
#define SOUND_MIXER_READ_DIGITAL1 MIXER_READ(SOUND_MIXER_DIGITAL1)
#define SOUND_MIXER_READ_DIGITAL2 MIXER_READ(SOUND_MIXER_DIGITAL2)
#define SOUND_MIXER_READ_DIGITAL3 MIXER_READ(SOUND_MIXER_DIGITAL3)
#define SOUND_MIXER_READ_PHONEIN MIXER_READ(SOUND_MIXER_PHONEIN)
#define SOUND_MIXER_READ_PHONEOUT MIXER_READ(SOUND_MIXER_PHONEOUT)
#define SOUND_MIXER_READ_RADIO MIXER_READ(SOUND_MIXER_RADIO)
#define SOUND_MIXER_READ_VIDEO MIXER_READ(SOUND_MIXER_VIDEO)
#define SOUND_MIXER_READ_MONITOR MIXER_READ(SOUND_MIXER_MONITOR)
/* Obsolete macros */
#define SOUND_MIXER_READ_MUTE MIXER_READ(SOUND_MIXER_MUTE)
#define SOUND_MIXER_READ_ENHANCE MIXER_READ(SOUND_MIXER_ENHANCE)
#define SOUND_MIXER_READ_LOUD MIXER_READ(SOUND_MIXER_LOUD)
#define SOUND_MIXER_READ_RECSRC MIXER_READ(SOUND_MIXER_RECSRC)
#define SOUND_MIXER_READ_DEVMASK MIXER_READ(SOUND_MIXER_DEVMASK)
#define SOUND_MIXER_READ_RECMASK MIXER_READ(SOUND_MIXER_RECMASK)
#define SOUND_MIXER_READ_STEREODEVS MIXER_READ(SOUND_MIXER_STEREODEVS)
#define SOUND_MIXER_READ_CAPS MIXER_READ(SOUND_MIXER_CAPS)
#define MIXER_WRITE(dev) _IOWR('M', dev, int)
#define SOUND_MIXER_WRITE_VOLUME MIXER_WRITE(SOUND_MIXER_VOLUME)
#define SOUND_MIXER_WRITE_BASS MIXER_WRITE(SOUND_MIXER_BASS)
#define SOUND_MIXER_WRITE_TREBLE MIXER_WRITE(SOUND_MIXER_TREBLE)
#define SOUND_MIXER_WRITE_SYNTH MIXER_WRITE(SOUND_MIXER_SYNTH)
#define SOUND_MIXER_WRITE_PCM MIXER_WRITE(SOUND_MIXER_PCM)
#define SOUND_MIXER_WRITE_SPEAKER MIXER_WRITE(SOUND_MIXER_SPEAKER)
#define SOUND_MIXER_WRITE_LINE MIXER_WRITE(SOUND_MIXER_LINE)
#define SOUND_MIXER_WRITE_MIC MIXER_WRITE(SOUND_MIXER_MIC)
#define SOUND_MIXER_WRITE_CD MIXER_WRITE(SOUND_MIXER_CD)
#define SOUND_MIXER_WRITE_IMIX MIXER_WRITE(SOUND_MIXER_IMIX)
#define SOUND_MIXER_WRITE_ALTPCM MIXER_WRITE(SOUND_MIXER_ALTPCM)
#define SOUND_MIXER_WRITE_RECLEV MIXER_WRITE(SOUND_MIXER_RECLEV)
#define SOUND_MIXER_WRITE_IGAIN MIXER_WRITE(SOUND_MIXER_IGAIN)
#define SOUND_MIXER_WRITE_OGAIN MIXER_WRITE(SOUND_MIXER_OGAIN)
#define SOUND_MIXER_WRITE_LINE1 MIXER_WRITE(SOUND_MIXER_LINE1)
#define SOUND_MIXER_WRITE_LINE2 MIXER_WRITE(SOUND_MIXER_LINE2)
#define SOUND_MIXER_WRITE_LINE3 MIXER_WRITE(SOUND_MIXER_LINE3)
#define SOUND_MIXER_WRITE_DIGITAL1 MIXER_WRITE(SOUND_MIXER_DIGITAL1)
#define SOUND_MIXER_WRITE_DIGITAL2 MIXER_WRITE(SOUND_MIXER_DIGITAL2)
#define SOUND_MIXER_WRITE_DIGITAL3 MIXER_WRITE(SOUND_MIXER_DIGITAL3)
#define SOUND_MIXER_WRITE_PHONEIN MIXER_WRITE(SOUND_MIXER_PHONEIN)
#define SOUND_MIXER_WRITE_PHONEOUT MIXER_WRITE(SOUND_MIXER_PHONEOUT)
#define SOUND_MIXER_WRITE_RADIO MIXER_WRITE(SOUND_MIXER_RADIO)
#define SOUND_MIXER_WRITE_VIDEO MIXER_WRITE(SOUND_MIXER_VIDEO)
#define SOUND_MIXER_WRITE_MONITOR MIXER_WRITE(SOUND_MIXER_MONITOR)
#define SOUND_MIXER_WRITE_MUTE MIXER_WRITE(SOUND_MIXER_MUTE)
#define SOUND_MIXER_WRITE_ENHANCE MIXER_WRITE(SOUND_MIXER_ENHANCE)
#define SOUND_MIXER_WRITE_LOUD MIXER_WRITE(SOUND_MIXER_LOUD)
#define SOUND_MIXER_WRITE_RECSRC MIXER_WRITE(SOUND_MIXER_RECSRC)
typedef struct mixer_info {
char id[16];
char name[32];
int modify_counter;
int fillers[10];
} mixer_info;
#define SOUND_MIXER_INFO _IOR('M', 101, mixer_info)
#define LEFT_CHN 0
#define RIGHT_CHN 1
/*
* Level 2 event types for /dev/sequencer
*/
/*
* The 4 most significant bits of byte 0 specify the class of
* the event:
*
* 0x8X = system level events,
* 0x9X = device/port specific events, event[1] = device/port,
* The last 4 bits give the subtype:
* 0x02 = Channel event (event[3] = chn).
* 0x01 = note event (event[4] = note).
* (0x01 is not used alone but always with bit 0x02).
* event[2] = MIDI message code (0x80=note off etc.)
*
*/
#define EV_SEQ_LOCAL 0x80
#define EV_TIMING 0x81
#define EV_CHN_COMMON 0x92
#define EV_CHN_VOICE 0x93
#define EV_SYSEX 0x94
/*
* Event types 200 to 220 are reserved for application use.
* These numbers will not be used by the driver.
*/
/*
* Events for event type EV_CHN_VOICE
*/
#define MIDI_NOTEOFF 0x80
#define MIDI_NOTEON 0x90
#define MIDI_KEY_PRESSURE 0xA0
/*
* Events for event type EV_CHN_COMMON
*/
#define MIDI_CTL_CHANGE 0xB0
#define MIDI_PGM_CHANGE 0xC0
#define MIDI_CHN_PRESSURE 0xD0
#define MIDI_PITCH_BEND 0xE0
#define MIDI_SYSTEM_PREFIX 0xF0
/*
* Timer event types
*/
#define TMR_WAIT_REL 1 /* Time relative to the prev time */
#define TMR_WAIT_ABS 2 /* Absolute time since TMR_START */
#define TMR_STOP 3
#define TMR_START 4
#define TMR_CONTINUE 5
#define TMR_TEMPO 6
#define TMR_ECHO 8
#define TMR_CLOCK 9 /* MIDI clock */
#define TMR_SPP 10 /* Song position pointer */
#define TMR_TIMESIG 11 /* Time signature */
/*
* Local event types
*/
#define LOCL_STARTAUDIO 1
#if (!defined(_KERNEL) && !defined(INKERNEL)) || defined(USE_SEQ_MACROS)
/*
* Some convenience macros to simplify programming of the
* /dev/sequencer interface
*
* These macros define the API which should be used when possible.
*/
#ifndef USE_SIMPLE_MACROS
void seqbuf_dump(void); /* This function must be provided by programs */
/* Sample seqbuf_dump() implementation:
*
* SEQ_DEFINEBUF (2048); -- Defines a buffer for 2048 bytes
*
* int seqfd; -- The file descriptor for /dev/sequencer.
*
* void
* seqbuf_dump ()
* {
* if (_seqbufptr)
* if (write (seqfd, _seqbuf, _seqbufptr) == -1)
* {
* perror ("write /dev/sequencer");
* exit (-1);
* }
* _seqbufptr = 0;
* }
*/
#define SEQ_DEFINEBUF(len) \
u_char _seqbuf[len]; int _seqbuflen = len;int _seqbufptr = 0
#define SEQ_USE_EXTBUF() \
extern u_char _seqbuf[]; \
extern int _seqbuflen;extern int _seqbufptr
#define SEQ_DECLAREBUF() SEQ_USE_EXTBUF()
#define SEQ_PM_DEFINES struct patmgr_info _pm_info
#define _SEQ_NEEDBUF(len) \
if ((_seqbufptr+(len)) > _seqbuflen) \
seqbuf_dump()
#define _SEQ_ADVBUF(len) _seqbufptr += len
#define SEQ_DUMPBUF seqbuf_dump
#else
/*
* This variation of the sequencer macros is used just to format one event
* using fixed buffer.
*
* The program using the macro library must define the following macros before
* using this library.
*
* #define _seqbuf name of the buffer (u_char[])
* #define _SEQ_ADVBUF(len) If the applic needs to know the exact
* size of the event, this macro can be used.
* Otherwise this must be defined as empty.
* #define _seqbufptr Define the name of index variable or 0 if
* not required.
*/
#define _SEQ_NEEDBUF(len) /* empty */
#endif
#define PM_LOAD_PATCH(dev, bank, pgm) \
(SEQ_DUMPBUF(), _pm_info.command = _PM_LOAD_PATCH, \
_pm_info.device=dev, _pm_info.data.data8[0]=pgm, \
_pm_info.parm1 = bank, _pm_info.parm2 = 1, \
ioctl(seqfd, SNDCTL_PMGR_ACCESS, &_pm_info))
#define PM_LOAD_PATCHES(dev, bank, pgm) \
(SEQ_DUMPBUF(), _pm_info.command = _PM_LOAD_PATCH, \
_pm_info.device=dev, bcopy( pgm, _pm_info.data.data8, 128), \
_pm_info.parm1 = bank, _pm_info.parm2 = 128, \
ioctl(seqfd, SNDCTL_PMGR_ACCESS, &_pm_info))
#define SEQ_VOLUME_MODE(dev, mode) { \
_SEQ_NEEDBUF(8);\
_seqbuf[_seqbufptr] = SEQ_EXTENDED;\
_seqbuf[_seqbufptr+1] = SEQ_VOLMODE;\
_seqbuf[_seqbufptr+2] = (dev);\
_seqbuf[_seqbufptr+3] = (mode);\
_seqbuf[_seqbufptr+4] = 0;\
_seqbuf[_seqbufptr+5] = 0;\
_seqbuf[_seqbufptr+6] = 0;\
_seqbuf[_seqbufptr+7] = 0;\
_SEQ_ADVBUF(8);}
/*
* Midi voice messages
*/
#define _CHN_VOICE(dev, event, chn, note, parm) { \
_SEQ_NEEDBUF(8);\
_seqbuf[_seqbufptr] = EV_CHN_VOICE;\
_seqbuf[_seqbufptr+1] = (dev);\
_seqbuf[_seqbufptr+2] = (event);\
_seqbuf[_seqbufptr+3] = (chn);\
_seqbuf[_seqbufptr+4] = (note);\
_seqbuf[_seqbufptr+5] = (parm);\
_seqbuf[_seqbufptr+6] = (0);\
_seqbuf[_seqbufptr+7] = 0;\
_SEQ_ADVBUF(8);}
#define SEQ_START_NOTE(dev, chn, note, vol) \
_CHN_VOICE(dev, MIDI_NOTEON, chn, note, vol)
#define SEQ_STOP_NOTE(dev, chn, note, vol) \
_CHN_VOICE(dev, MIDI_NOTEOFF, chn, note, vol)
#define SEQ_KEY_PRESSURE(dev, chn, note, pressure) \
_CHN_VOICE(dev, MIDI_KEY_PRESSURE, chn, note, pressure)
/*
* Midi channel messages
*/
#define _CHN_COMMON(dev, event, chn, p1, p2, w14) { \
_SEQ_NEEDBUF(8);\
_seqbuf[_seqbufptr] = EV_CHN_COMMON;\
_seqbuf[_seqbufptr+1] = (dev);\
_seqbuf[_seqbufptr+2] = (event);\
_seqbuf[_seqbufptr+3] = (chn);\
_seqbuf[_seqbufptr+4] = (p1);\
_seqbuf[_seqbufptr+5] = (p2);\
*(short *)&_seqbuf[_seqbufptr+6] = (w14);\
_SEQ_ADVBUF(8);}
/*
* SEQ_SYSEX permits sending of sysex messages. (It may look that it permits
* sending any MIDI bytes but it's absolutely not possible. Trying to do
* so _will_ cause problems with MPU401 intelligent mode).
*
* Sysex messages are sent in blocks of 1 to 6 bytes. Longer messages must be
* sent by calling SEQ_SYSEX() several times (there must be no other events
* between them). First sysex fragment must have 0xf0 in the first byte
* and the last byte (buf[len-1] of the last fragment must be 0xf7. No byte
* between these sysex start and end markers cannot be larger than 0x7f. Also
* lengths of each fragments (except the last one) must be 6.
*
* Breaking the above rules may work with some MIDI ports but is likely to
* cause fatal problems with some other devices (such as MPU401).
*/
#define SEQ_SYSEX(dev, buf, len) { \
int i, l=(len); if (l>6)l=6;\
_SEQ_NEEDBUF(8);\
_seqbuf[_seqbufptr] = EV_SYSEX;\
for(i=0;i<l;i++)_seqbuf[_seqbufptr+i+1] = (buf)[i];\
for(i=l;i<6;i++)_seqbuf[_seqbufptr+i+1] = 0xff;\
_SEQ_ADVBUF(8);}
#define SEQ_CHN_PRESSURE(dev, chn, pressure) \
_CHN_COMMON(dev, MIDI_CHN_PRESSURE, chn, pressure, 0, 0)
#define SEQ_SET_PATCH(dev, chn, patch) \
_CHN_COMMON(dev, MIDI_PGM_CHANGE, chn, patch, 0, 0)
#define SEQ_CONTROL(dev, chn, controller, value) \
_CHN_COMMON(dev, MIDI_CTL_CHANGE, chn, controller, 0, value)
#define SEQ_BENDER(dev, chn, value) \
_CHN_COMMON(dev, MIDI_PITCH_BEND, chn, 0, 0, value)
#define SEQ_V2_X_CONTROL(dev, voice, controller, value) { \
_SEQ_NEEDBUF(8);\
_seqbuf[_seqbufptr] = SEQ_EXTENDED;\
_seqbuf[_seqbufptr+1] = SEQ_CONTROLLER;\
_seqbuf[_seqbufptr+2] = (dev);\
_seqbuf[_seqbufptr+3] = (voice);\
_seqbuf[_seqbufptr+4] = (controller);\
*(short *)&_seqbuf[_seqbufptr+5] = (value);\
_seqbuf[_seqbufptr+7] = 0;\
_SEQ_ADVBUF(8);}
/*
* The following 5 macros are incorrectly implemented and obsolete.
* Use SEQ_BENDER and SEQ_CONTROL (with proper controller) instead.
*/
#define SEQ_PITCHBEND(dev, voice, value) \
SEQ_V2_X_CONTROL(dev, voice, CTRL_PITCH_BENDER, value)
#define SEQ_BENDER_RANGE(dev, voice, value) \
SEQ_V2_X_CONTROL(dev, voice, CTRL_PITCH_BENDER_RANGE, value)
#define SEQ_EXPRESSION(dev, voice, value) \
SEQ_CONTROL(dev, voice, CTL_EXPRESSION, value*128)
#define SEQ_MAIN_VOLUME(dev, voice, value) \
SEQ_CONTROL(dev, voice, CTL_MAIN_VOLUME, (value*16383)/100)
#define SEQ_PANNING(dev, voice, pos) \
SEQ_CONTROL(dev, voice, CTL_PAN, (pos+128) / 2)
/*
* Timing and syncronization macros
*/
#define _TIMER_EVENT(ev, parm) { \
_SEQ_NEEDBUF(8);\
_seqbuf[_seqbufptr+0] = EV_TIMING; \
_seqbuf[_seqbufptr+1] = (ev); \
_seqbuf[_seqbufptr+2] = 0;\
_seqbuf[_seqbufptr+3] = 0;\
*(u_int *)&_seqbuf[_seqbufptr+4] = (parm); \
_SEQ_ADVBUF(8); \
}
#define SEQ_START_TIMER() _TIMER_EVENT(TMR_START, 0)
#define SEQ_STOP_TIMER() _TIMER_EVENT(TMR_STOP, 0)
#define SEQ_CONTINUE_TIMER() _TIMER_EVENT(TMR_CONTINUE, 0)
#define SEQ_WAIT_TIME(ticks) _TIMER_EVENT(TMR_WAIT_ABS, ticks)
#define SEQ_DELTA_TIME(ticks) _TIMER_EVENT(TMR_WAIT_REL, ticks)
#define SEQ_ECHO_BACK(key) _TIMER_EVENT(TMR_ECHO, key)
#define SEQ_SET_TEMPO(value) _TIMER_EVENT(TMR_TEMPO, value)
#define SEQ_SONGPOS(pos) _TIMER_EVENT(TMR_SPP, pos)
#define SEQ_TIME_SIGNATURE(sig) _TIMER_EVENT(TMR_TIMESIG, sig)
/*
* Local control events
*/
#define _LOCAL_EVENT(ev, parm) { \
_SEQ_NEEDBUF(8);\
_seqbuf[_seqbufptr+0] = EV_SEQ_LOCAL; \
_seqbuf[_seqbufptr+1] = (ev); \
_seqbuf[_seqbufptr+2] = 0;\
_seqbuf[_seqbufptr+3] = 0;\
*(u_int *)&_seqbuf[_seqbufptr+4] = (parm); \
_SEQ_ADVBUF(8); \
}
#define SEQ_PLAYAUDIO(devmask) _LOCAL_EVENT(LOCL_STARTAUDIO, devmask)
/*
* Events for the level 1 interface only
*/
#define SEQ_MIDIOUT(device, byte) { \
_SEQ_NEEDBUF(4);\
_seqbuf[_seqbufptr] = SEQ_MIDIPUTC;\
_seqbuf[_seqbufptr+1] = (byte);\
_seqbuf[_seqbufptr+2] = (device);\
_seqbuf[_seqbufptr+3] = 0;\
_SEQ_ADVBUF(4);}
/*
* Patch loading.
*/
#define SEQ_WRPATCH(patchx, len) { \
if (_seqbufptr) seqbuf_dump(); \
if (write(seqfd, (char*)(patchx), len)==-1) \
perror("Write patch: /dev/sequencer"); \
}
#define SEQ_WRPATCH2(patchx, len) \
( seqbuf_dump(), write(seqfd, (char*)(patchx), len) )
#endif
/*
* Here I have moved all the aliases for ioctl names.
*/
#define SNDCTL_DSP_SAMPLESIZE SNDCTL_DSP_SETFMT
#define SOUND_PCM_WRITE_BITS SNDCTL_DSP_SETFMT
#define SOUND_PCM_SETFMT SNDCTL_DSP_SETFMT
#define SOUND_PCM_WRITE_RATE SNDCTL_DSP_SPEED
#define SOUND_PCM_POST SNDCTL_DSP_POST
#define SOUND_PCM_RESET SNDCTL_DSP_RESET
#define SOUND_PCM_SYNC SNDCTL_DSP_SYNC
#define SOUND_PCM_SUBDIVIDE SNDCTL_DSP_SUBDIVIDE
#define SOUND_PCM_SETFRAGMENT SNDCTL_DSP_SETFRAGMENT
#define SOUND_PCM_GETFMTS SNDCTL_DSP_GETFMTS
#define SOUND_PCM_GETOSPACE SNDCTL_DSP_GETOSPACE
#define SOUND_PCM_GETISPACE SNDCTL_DSP_GETISPACE
#define SOUND_PCM_NONBLOCK SNDCTL_DSP_NONBLOCK
#define SOUND_PCM_GETCAPS SNDCTL_DSP_GETCAPS
#define SOUND_PCM_GETTRIGGER SNDCTL_DSP_GETTRIGGER
#define SOUND_PCM_SETTRIGGER SNDCTL_DSP_SETTRIGGER
#define SOUND_PCM_SETSYNCRO SNDCTL_DSP_SETSYNCRO
#define SOUND_PCM_GETIPTR SNDCTL_DSP_GETIPTR
#define SOUND_PCM_GETOPTR SNDCTL_DSP_GETOPTR
#define SOUND_PCM_MAPINBUF SNDCTL_DSP_MAPINBUF
#define SOUND_PCM_MAPOUTBUF SNDCTL_DSP_MAPOUTBUF
/***********************************************************************/
/**
* XXX OSSv4 defines -- some bits taken straight out of the new
* sys/soundcard.h bundled with recent OSS releases.
*
* NB: These macros and structures will be reorganized and inserted
* in appropriate places throughout this file once the code begins
* to take shape.
*
* @todo reorganize layout more like the 4Front version
* @todo ask about maintaining __SIOWR vs. _IOWR ioctl cmd defines
*/
/**
* @note The @c OSSV4_EXPERIMENT macro is meant to wrap new development code
* in the sound system relevant to adopting 4Front's OSSv4 specification.
* Users should not enable this! Really!
*/
#if 0
# define OSSV4_EXPERIMENT 1
#else
# undef OSSV4_EXPERIMENT
#endif
#ifdef SOUND_VERSION
# undef SOUND_VERSION
# define SOUND_VERSION 0x040000
#endif /* !SOUND_VERSION */
#define OSS_LONGNAME_SIZE 64
#define OSS_LABEL_SIZE 16
#define OSS_DEVNODE_SIZE 32
typedef char oss_longname_t[OSS_LONGNAME_SIZE];
typedef char oss_label_t[OSS_LABEL_SIZE];
typedef char oss_devnode_t[OSS_DEVNODE_SIZE];
typedef struct audio_errinfo
{
int play_underruns;
int rec_overruns;
unsigned int play_ptradjust;
unsigned int rec_ptradjust;
int play_errorcount;
int rec_errorcount;
int play_lasterror;
int rec_lasterror;
long play_errorparm;
long rec_errorparm;
int filler[16];
} audio_errinfo;
#define SNDCTL_DSP_GETPLAYVOL _IOR ('P', 24, int)
#define SNDCTL_DSP_SETPLAYVOL _IOWR('P', 24, int)
#define SNDCTL_DSP_GETERROR _IOR ('P', 25, audio_errinfo)
/*
****************************************************************************
* Sync groups for audio devices
*/
typedef struct oss_syncgroup
{
int id;
int mode;
int filler[16];
} oss_syncgroup;
#define SNDCTL_DSP_SYNCGROUP _IOWR('P', 28, oss_syncgroup)
#define SNDCTL_DSP_SYNCSTART _IOW ('P', 29, int)
/*
**************************************************************************
* "cooked" mode enables software based conversions for sample rate, sample
* format (bits) and number of channels (mono/stereo). These conversions are
* required with some devices that support only one sample rate or just stereo
* to let the applications to use other formats. The cooked mode is enabled by
* default. However it's necessary to disable this mode when mmap() is used or
* when very deterministic timing is required. SNDCTL_DSP_COOKEDMODE is an
* optional call introduced in OSS 3.9.6f. It's _error return must be ignored_
* since normally this call will return erno=EINVAL.
*
* SNDCTL_DSP_COOKEDMODE must be called immediately after open before doing
* anything else. Otherwise the call will not have any effect.
*/
#define SNDCTL_DSP_COOKEDMODE _IOW ('P', 30, int)
/*
**************************************************************************
* SNDCTL_DSP_SILENCE and SNDCTL_DSP_SKIP are new calls in OSS 3.99.0
* that can be used to implement pause/continue during playback (no effect
* on recording).
*/
#define SNDCTL_DSP_SILENCE _IO ('P', 31)
#define SNDCTL_DSP_SKIP _IO ('P', 32)
/*
****************************************************************************
* Abort transfer (reset) functions for input and output
*/
#define SNDCTL_DSP_HALT_INPUT _IO ('P', 33)
#define SNDCTL_DSP_RESET_INPUT SNDCTL_DSP_HALT_INPUT /* Old name */
#define SNDCTL_DSP_HALT_OUTPUT _IO ('P', 34)
#define SNDCTL_DSP_RESET_OUTPUT SNDCTL_DSP_HALT_OUTPUT /* Old name */
/*
****************************************************************************
* Low water level control
*/
#define SNDCTL_DSP_LOW_WATER _IOW ('P', 34, int)
/** @todo Get rid of OSS_NO_LONG_LONG references? */
/*
****************************************************************************
* 64 bit pointer support. Only available in environments that support
* the 64 bit (long long) integer type.
*/
#ifndef OSS_NO_LONG_LONG
typedef struct
{
long long samples;
int fifo_samples;
int filler[32]; /* For future use */
} oss_count_t;
#define SNDCTL_DSP_CURRENT_IPTR _IOR ('P', 35, oss_count_t)
#define SNDCTL_DSP_CURRENT_OPTR _IOR ('P', 36, oss_count_t)
#endif
/*
****************************************************************************
* Interface for selecting recording sources and playback output routings.
*/
#define SNDCTL_DSP_GET_RECSRC_NAMES _IOR ('P', 37, oss_mixer_enuminfo)
#define SNDCTL_DSP_GET_RECSRC _IOR ('P', 38, int)
#define SNDCTL_DSP_SET_RECSRC _IOWR('P', 38, int)
#define SNDCTL_DSP_GET_PLAYTGT_NAMES _IOR ('P', 39, oss_mixer_enuminfo)
#define SNDCTL_DSP_GET_PLAYTGT _IOR ('P', 40, int)
#define SNDCTL_DSP_SET_PLAYTGT _IOWR('P', 40, int)
#define SNDCTL_DSP_GETRECVOL _IOR ('P', 41, int)
#define SNDCTL_DSP_SETRECVOL _IOWR('P', 41, int)
/*
***************************************************************************
* Some calls for setting the channel assignment with multi channel devices
* (see the manual for details). */
#define SNDCTL_DSP_GET_CHNORDER _IOR ('P', 42, unsigned long long)
#define SNDCTL_DSP_SET_CHNORDER _IOWR('P', 42, unsigned long long)
# define CHID_UNDEF 0
# define CHID_L 1 # define CHID_R 2
# define CHID_C 3
# define CHID_LFE 4
# define CHID_LS 5
# define CHID_RS 6
# define CHID_LR 7
# define CHID_RR 8
#define CHNORDER_UNDEF 0x0000000000000000ULL
#define CHNORDER_NORMAL 0x0000000087654321ULL
#define MAX_PEAK_CHANNELS 128
typedef unsigned short oss_peaks_t[MAX_PEAK_CHANNELS];
#define SNDCTL_DSP_GETIPEAKS _IOR('P', 43, oss_peaks_t)
#define SNDCTL_DSP_GETOPEAKS _IOR('P', 44, oss_peaks_t)
#define SNDCTL_DSP_POLICY _IOW('P', 45, int) /* See the manual */
/*
* OSS_SYSIFO is obsolete. Use SNDCTL_SYSINFO insteads.
*/
#define OSS_GETVERSION _IOR ('M', 118, int)
/**
* @brief Argument for SNDCTL_SYSINFO ioctl.
*
* For use w/ the SNDCTL_SYSINFO ioctl available on audio (/dev/dsp*),
* mixer, and MIDI devices.
*/
typedef struct oss_sysinfo
{
char product[32]; /* For example OSS/Free, OSS/Linux or
OSS/Solaris */
char version[32]; /* For example 4.0a */
int versionnum; /* See OSS_GETVERSION */
char options[128]; /* Reserved */
int numaudios; /* # of audio/dsp devices */
int openedaudio[8]; /* Bit mask telling which audio devices
are busy */
int numsynths; /* # of availavle synth devices */
int nummidis; /* # of available MIDI ports */
int numtimers; /* # of available timer devices */
int nummixers; /* # of mixer devices */
int openedmidi[8]; /* Bit mask telling which midi devices
are busy */
int numcards; /* Number of sound cards in the system */
int filler[241]; /* For future expansion (set to -1) */
} oss_sysinfo;
typedef struct oss_mixext
{
int dev; /* Mixer device number */
int ctrl; /* Controller number */
int type; /* Entry type */
# define MIXT_DEVROOT 0 /* Device root entry */
# define MIXT_GROUP 1 /* Controller group */
# define MIXT_ONOFF 2 /* OFF (0) or ON (1) */
# define MIXT_ENUM 3 /* Enumerated (0 to maxvalue) */
# define MIXT_MONOSLIDER 4 /* Mono slider (0 to 100) */
# define MIXT_STEREOSLIDER 5 /* Stereo slider (dual 0 to 100) */
# define MIXT_MESSAGE 6 /* (Readable) textual message */
# define MIXT_MONOVU 7 /* VU meter value (mono) */
# define MIXT_STEREOVU 8 /* VU meter value (stereo) */
# define MIXT_MONOPEAK 9 /* VU meter peak value (mono) */
# define MIXT_STEREOPEAK 10 /* VU meter peak value (stereo) */
# define MIXT_RADIOGROUP 11 /* Radio button group */
# define MIXT_MARKER 12 /* Separator between normal and extension entries */
# define MIXT_VALUE 13 /* Decimal value entry */
# define MIXT_HEXVALUE 14 /* Hexadecimal value entry */
# define MIXT_MONODB 15 /* Mono atten. slider (0 to -144) */
# define MIXT_STEREODB 16 /* Stereo atten. slider (dual 0 to -144) */
# define MIXT_SLIDER 17 /* Slider (mono) with full integer range */
# define MIXT_3D 18
/* Possible value range (minvalue to maxvalue) */
/* Note that maxvalue may also be smaller than minvalue */
int maxvalue;
int minvalue;
int flags;
# define MIXF_READABLE 0x00000001 /* Has readable value */
# define MIXF_WRITEABLE 0x00000002 /* Has writeable value */
# define MIXF_POLL 0x00000004 /* May change itself */
# define MIXF_HZ 0x00000008 /* Herz scale */
# define MIXF_STRING 0x00000010 /* Use dynamic extensions for value */
# define MIXF_DYNAMIC 0x00000010 /* Supports dynamic extensions */
# define MIXF_OKFAIL 0x00000020 /* Interpret value as 1=OK, 0=FAIL */
# define MIXF_FLAT 0x00000040 /* Flat vertical space requirements */
# define MIXF_LEGACY 0x00000080 /* Legacy mixer control group */
char id[16]; /* Mnemonic ID (mainly for internal use) */
int parent; /* Entry# of parent (group) node (-1 if root) */
int dummy; /* Internal use */
int timestamp;
char data[64]; /* Misc data (entry type dependent) */
unsigned char enum_present[32]; /* Mask of allowed enum values */
int control_no; /* SOUND_MIXER_VOLUME..SOUND_MIXER_MIDI */
/* (-1 means not indicated) */
/*
* The desc field is reserved for internal purposes of OSS. It should not be
* used by applications.
*/
unsigned int desc;
#define MIXEXT_SCOPE_MASK 0x0000003f
#define MIXEXT_SCOPE_OTHER 0x00000000
#define MIXEXT_SCOPE_INPUT 0x00000001
#define MIXEXT_SCOPE_OUTPUT 0x00000002
#define MIXEXT_SCOPE_MONITOR 0x00000003
#define MIXEXT_SCOPE_RECSWITCH 0x00000004
char extname[32];
int update_counter;
int filler[7];
} oss_mixext;
typedef struct oss_mixext_root
{
char id[16];
char name[48];
} oss_mixext_root;
typedef struct oss_mixer_value
{
int dev;
int ctrl;
int value;
int flags; /* Reserved for future use. Initialize to 0 */
int timestamp; /* Must be set to oss_mixext.timestamp */
int filler[8]; /* Reserved for future use. Initialize to 0 */
} oss_mixer_value;
#define OSS_ENUM_MAXVALUE 255
typedef struct oss_mixer_enuminfo
{
int dev;
int ctrl;
int nvalues;
int version; /* Read the manual */
short strindex[OSS_ENUM_MAXVALUE];
char strings[3000];
} oss_mixer_enuminfo;
#define OPEN_READ PCM_ENABLE_INPUT
#define OPEN_WRITE PCM_ENABLE_OUTPUT
#define OPEN_READWRITE (OPEN_READ|OPEN_WRITE)
/**
* @brief Argument for SNDCTL_AUDIOINFO ioctl.
*
* For use w/ the SNDCTL_AUDIOINFO ioctl available on audio (/dev/dsp*)
* devices.
*/
typedef struct oss_audioinfo
{
int dev; /* Audio device number */
char name[64];
int busy; /* 0, OPEN_READ, OPEN_WRITE or OPEN_READWRITE */
int pid;
int caps; /* DSP_CAP_INPUT, DSP_CAP_OUTPUT */
int iformats;
int oformats;
int magic; /* Reserved for internal use */
char cmd[64]; /* Command using the device (if known) */
int card_number;
int port_number;
int mixer_dev;
int real_device; /* Obsolete field. Replaced by devnode */
int enabled; /* 1=enabled, 0=device not ready at this
moment */
int flags; /* For internal use only - no practical
meaning */
int min_rate; /* Sample rate limits */
int max_rate;
int min_channels; /* Number of channels supported */
int max_channels;
int binding; /* DSP_BIND_FRONT, etc. 0 means undefined */
int rate_source;
char handle[32];
#define OSS_MAX_SAMPLE_RATES 20 /* Cannot be changed */
unsigned int nrates;
unsigned int rates[OSS_MAX_SAMPLE_RATES]; /* Please read the manual before using these */
oss_longname_t song_name; /* Song name (if given) */
oss_label_t label; /* Device label (if given) */
int latency; /* In usecs, -1=unknown */
oss_devnode_t devnode; /* Device special file name (inside
/dev) */
int filler[186];
} oss_audioinfo;
typedef struct oss_mixerinfo
{
int dev;
char id[16];
char name[32];
int modify_counter;
int card_number;
int port_number;
char handle[32];
int magic; /* Reserved */
int enabled; /* Reserved */
int caps;
#define MIXER_CAP_VIRTUAL 0x00000001
int flags; /* Reserved */
int nrext;
/*
* The priority field can be used to select the default (motherboard)
* mixer device. The mixer with the highest priority is the
* most preferred one. -2 or less means that this device cannot be used
* as the default mixer.
*/
int priority;
int filler[254]; /* Reserved */
} oss_mixerinfo;
typedef struct oss_midi_info
{
int dev; /* Midi device number */
char name[64];
int busy; /* 0, OPEN_READ, OPEN_WRITE or OPEN_READWRITE */
int pid;
char cmd[64]; /* Command using the device (if known) */
int caps;
#define MIDI_CAP_MPU401 0x00000001 /**** OBSOLETE ****/
#define MIDI_CAP_INPUT 0x00000002
#define MIDI_CAP_OUTPUT 0x00000004
#define MIDI_CAP_INOUT (MIDI_CAP_INPUT|MIDI_CAP_OUTPUT)
#define MIDI_CAP_VIRTUAL 0x00000008 /* Pseudo device */
#define MIDI_CAP_MTCINPUT 0x00000010 /* Supports SNDCTL_MIDI_MTCINPUT */
#define MIDI_CAP_CLIENT 0x00000020 /* Virtual client side device */
#define MIDI_CAP_SERVER 0x00000040 /* Virtual server side device */
#define MIDI_CAP_INTERNAL 0x00000080 /* Internal (synth) device */
#define MIDI_CAP_EXTERNAL 0x00000100 /* external (MIDI port) device */
#define MIDI_CAP_PTOP 0x00000200 /* Point to point link to one device */
#define MIDI_CAP_MTC 0x00000400 /* MTC/SMPTE (control) device */
int magic; /* Reserved for internal use */
int card_number;
int port_number;
int enabled; /* 1=enabled, 0=device not ready at this moment */
int flags; /* For internal use only - no practical meaning */
char handle[32];
oss_longname_t song_name; /* Song name (if known) */
oss_label_t label; /* Device label (if given) */
int latency; /* In usecs, -1=unknown */
int filler[244];
} oss_midi_info;
typedef struct oss_card_info
{
int card;
char shortname[16];
char longname[128];
int flags;
int filler[256];
} oss_card_info;
#define SNDCTL_SYSINFO _IOR ('X', 1, oss_sysinfo)
#define OSS_SYSINFO SNDCTL_SYSINFO /* Old name */
#define SNDCTL_MIX_NRMIX _IOR ('X', 2, int)
#define SNDCTL_MIX_NREXT _IOWR('X', 3, int)
#define SNDCTL_MIX_EXTINFO _IOWR('X', 4, oss_mixext)
#define SNDCTL_MIX_READ _IOWR('X', 5, oss_mixer_value)
#define SNDCTL_MIX_WRITE _IOWR('X', 6, oss_mixer_value)
#define SNDCTL_AUDIOINFO _IOWR('X', 7, oss_audioinfo)
#define SNDCTL_MIX_ENUMINFO _IOWR('X', 8, oss_mixer_enuminfo)
#define SNDCTL_MIDIINFO _IOWR('X', 9, oss_midi_info)
#define SNDCTL_MIXERINFO _IOWR('X',10, oss_mixerinfo)
#define SNDCTL_CARDINFO _IOWR('X',11, oss_card_info)
/*
* Few more "globally" available ioctl calls.
*/
#define SNDCTL_SETSONG _IOW ('Y', 2, oss_longname_t)
#define SNDCTL_GETSONG _IOR ('Y', 2, oss_longname_t)
#define SNDCTL_SETNAME _IOW ('Y', 3, oss_longname_t)
#define SNDCTL_SETLABEL _IOW ('Y', 4, oss_label_t)
#define SNDCTL_GETLABEL _IOR ('Y', 4, oss_label_t)
#endif /* !_SYS_SOUNDCARD_H_ */
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Received on Sat Aug 25 20:15:09 2007
This archive was generated by hypermail 2.1.8 : Sat Aug 25 2007 - 20:15:11 EEST