UAPI: (Scripted) Disintegrate include/sound

Signed-off-by: David Howells <dhowells@redhat.com>
Acked-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Thomas Gleixner <tglx@linutronix.de>
Acked-by: Michael Kerrisk <mtk.manpages@gmail.com>
Acked-by: Paul E. McKenney <paulmck@linux.vnet.ibm.com>
Acked-by: Dave Jones <davej@redhat.com>
This commit is contained in:
David Howells
2012-10-09 09:49:13 +01:00
parent 9e2d8656f5
commit 674e95ca44
16 changed files with 2084 additions and 1990 deletions
+10
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@@ -1 +1,11 @@
# UAPI Header export list
header-y += asequencer.h
header-y += asound.h
header-y += asound_fm.h
header-y += compress_offload.h
header-y += compress_params.h
header-y += emu10k1.h
header-y += hdsp.h
header-y += hdspm.h
header-y += sb16_csp.h
header-y += sfnt_info.h
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#ifndef __SOUND_ASOUND_FM_H
#define __SOUND_ASOUND_FM_H
/*
* Advanced Linux Sound Architecture - ALSA
*
* Interface file between ALSA driver & user space
* Copyright (c) 1994-98 by Jaroslav Kysela <perex@perex.cz>,
* 4Front Technologies
*
* Direct FM control
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
*/
#define SNDRV_DM_FM_MODE_OPL2 0x00
#define SNDRV_DM_FM_MODE_OPL3 0x01
struct snd_dm_fm_info {
unsigned char fm_mode; /* OPL mode, see SNDRV_DM_FM_MODE_XXX */
unsigned char rhythm; /* percussion mode flag */
};
/*
* Data structure composing an FM "note" or sound event.
*/
struct snd_dm_fm_voice {
unsigned char op; /* operator cell (0 or 1) */
unsigned char voice; /* FM voice (0 to 17) */
unsigned char am; /* amplitude modulation */
unsigned char vibrato; /* vibrato effect */
unsigned char do_sustain; /* sustain phase */
unsigned char kbd_scale; /* keyboard scaling */
unsigned char harmonic; /* 4 bits: harmonic and multiplier */
unsigned char scale_level; /* 2 bits: decrease output freq rises */
unsigned char volume; /* 6 bits: volume */
unsigned char attack; /* 4 bits: attack rate */
unsigned char decay; /* 4 bits: decay rate */
unsigned char sustain; /* 4 bits: sustain level */
unsigned char release; /* 4 bits: release rate */
unsigned char feedback; /* 3 bits: feedback for op0 */
unsigned char connection; /* 0 for serial, 1 for parallel */
unsigned char left; /* stereo left */
unsigned char right; /* stereo right */
unsigned char waveform; /* 3 bits: waveform shape */
};
/*
* This describes an FM note by its voice, octave, frequency number (10bit)
* and key on/off.
*/
struct snd_dm_fm_note {
unsigned char voice; /* 0-17 voice channel */
unsigned char octave; /* 3 bits: what octave to play */
unsigned int fnum; /* 10 bits: frequency number */
unsigned char key_on; /* set for active, clear for silent */
};
/*
* FM parameters that apply globally to all voices, and thus are not "notes"
*/
struct snd_dm_fm_params {
unsigned char am_depth; /* amplitude modulation depth (1=hi) */
unsigned char vib_depth; /* vibrato depth (1=hi) */
unsigned char kbd_split; /* keyboard split */
unsigned char rhythm; /* percussion mode select */
/* This block is the percussion instrument data */
unsigned char bass;
unsigned char snare;
unsigned char tomtom;
unsigned char cymbal;
unsigned char hihat;
};
/*
* FM mode ioctl settings
*/
#define SNDRV_DM_FM_IOCTL_INFO _IOR('H', 0x20, struct snd_dm_fm_info)
#define SNDRV_DM_FM_IOCTL_RESET _IO ('H', 0x21)
#define SNDRV_DM_FM_IOCTL_PLAY_NOTE _IOW('H', 0x22, struct snd_dm_fm_note)
#define SNDRV_DM_FM_IOCTL_SET_VOICE _IOW('H', 0x23, struct snd_dm_fm_voice)
#define SNDRV_DM_FM_IOCTL_SET_PARAMS _IOW('H', 0x24, struct snd_dm_fm_params)
#define SNDRV_DM_FM_IOCTL_SET_MODE _IOW('H', 0x25, int)
/* for OPL3 only */
#define SNDRV_DM_FM_IOCTL_SET_CONNECTION _IOW('H', 0x26, int)
/* SBI patch management */
#define SNDRV_DM_FM_IOCTL_CLEAR_PATCHES _IO ('H', 0x40)
#define SNDRV_DM_FM_OSS_IOCTL_RESET 0x20
#define SNDRV_DM_FM_OSS_IOCTL_PLAY_NOTE 0x21
#define SNDRV_DM_FM_OSS_IOCTL_SET_VOICE 0x22
#define SNDRV_DM_FM_OSS_IOCTL_SET_PARAMS 0x23
#define SNDRV_DM_FM_OSS_IOCTL_SET_MODE 0x24
#define SNDRV_DM_FM_OSS_IOCTL_SET_OPL 0x25
/*
* Patch Record - fixed size for write
*/
#define FM_KEY_SBI "SBI\032"
#define FM_KEY_2OP "2OP\032"
#define FM_KEY_4OP "4OP\032"
struct sbi_patch {
unsigned char prog;
unsigned char bank;
char key[4];
char name[25];
char extension[7];
unsigned char data[32];
};
#endif /* __SOUND_ASOUND_FM_H */
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/*
* compress_offload.h - compress offload header definations
*
* Copyright (C) 2011 Intel Corporation
* Authors: Vinod Koul <vinod.koul@linux.intel.com>
* Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; version 2 of the License.
*
* This program is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
*
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
*
*/
#ifndef __COMPRESS_OFFLOAD_H
#define __COMPRESS_OFFLOAD_H
#include <linux/types.h>
#include <sound/asound.h>
#include <sound/compress_params.h>
#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 0)
/**
* struct snd_compressed_buffer: compressed buffer
* @fragment_size: size of buffer fragment in bytes
* @fragments: number of such fragments
*/
struct snd_compressed_buffer {
__u32 fragment_size;
__u32 fragments;
};
/**
* struct snd_compr_params: compressed stream params
* @buffer: buffer description
* @codec: codec parameters
* @no_wake_mode: dont wake on fragment elapsed
*/
struct snd_compr_params {
struct snd_compressed_buffer buffer;
struct snd_codec codec;
__u8 no_wake_mode;
};
/**
* struct snd_compr_tstamp: timestamp descriptor
* @byte_offset: Byte offset in ring buffer to DSP
* @copied_total: Total number of bytes copied from/to ring buffer to/by DSP
* @pcm_frames: Frames decoded or encoded by DSP. This field will evolve by
* large steps and should only be used to monitor encoding/decoding
* progress. It shall not be used for timing estimates.
* @pcm_io_frames: Frames rendered or received by DSP into a mixer or an audio
* output/input. This field should be used for A/V sync or time estimates.
* @sampling_rate: sampling rate of audio
*/
struct snd_compr_tstamp {
__u32 byte_offset;
__u32 copied_total;
snd_pcm_uframes_t pcm_frames;
snd_pcm_uframes_t pcm_io_frames;
__u32 sampling_rate;
};
/**
* struct snd_compr_avail: avail descriptor
* @avail: Number of bytes available in ring buffer for writing/reading
* @tstamp: timestamp infomation
*/
struct snd_compr_avail {
__u64 avail;
struct snd_compr_tstamp tstamp;
};
enum snd_compr_direction {
SND_COMPRESS_PLAYBACK = 0,
SND_COMPRESS_CAPTURE
};
/**
* struct snd_compr_caps: caps descriptor
* @codecs: pointer to array of codecs
* @direction: direction supported. Of type snd_compr_direction
* @min_fragment_size: minimum fragment supported by DSP
* @max_fragment_size: maximum fragment supported by DSP
* @min_fragments: min fragments supported by DSP
* @max_fragments: max fragments supported by DSP
* @num_codecs: number of codecs supported
* @reserved: reserved field
*/
struct snd_compr_caps {
__u32 num_codecs;
__u32 direction;
__u32 min_fragment_size;
__u32 max_fragment_size;
__u32 min_fragments;
__u32 max_fragments;
__u32 codecs[MAX_NUM_CODECS];
__u32 reserved[11];
};
/**
* struct snd_compr_codec_caps: query capability of codec
* @codec: codec for which capability is queried
* @num_descriptors: number of codec descriptors
* @descriptor: array of codec capability descriptor
*/
struct snd_compr_codec_caps {
__u32 codec;
__u32 num_descriptors;
struct snd_codec_desc descriptor[MAX_NUM_CODEC_DESCRIPTORS];
};
/**
* compress path ioctl definitions
* SNDRV_COMPRESS_GET_CAPS: Query capability of DSP
* SNDRV_COMPRESS_GET_CODEC_CAPS: Query capability of a codec
* SNDRV_COMPRESS_SET_PARAMS: Set codec and stream parameters
* Note: only codec params can be changed runtime and stream params cant be
* SNDRV_COMPRESS_GET_PARAMS: Query codec params
* SNDRV_COMPRESS_TSTAMP: get the current timestamp value
* SNDRV_COMPRESS_AVAIL: get the current buffer avail value.
* This also queries the tstamp properties
* SNDRV_COMPRESS_PAUSE: Pause the running stream
* SNDRV_COMPRESS_RESUME: resume a paused stream
* SNDRV_COMPRESS_START: Start a stream
* SNDRV_COMPRESS_STOP: stop a running stream, discarding ring buffer content
* and the buffers currently with DSP
* SNDRV_COMPRESS_DRAIN: Play till end of buffers and stop after that
* SNDRV_COMPRESS_IOCTL_VERSION: Query the API version
*/
#define SNDRV_COMPRESS_IOCTL_VERSION _IOR('C', 0x00, int)
#define SNDRV_COMPRESS_GET_CAPS _IOWR('C', 0x10, struct snd_compr_caps)
#define SNDRV_COMPRESS_GET_CODEC_CAPS _IOWR('C', 0x11,\
struct snd_compr_codec_caps)
#define SNDRV_COMPRESS_SET_PARAMS _IOW('C', 0x12, struct snd_compr_params)
#define SNDRV_COMPRESS_GET_PARAMS _IOR('C', 0x13, struct snd_codec)
#define SNDRV_COMPRESS_TSTAMP _IOR('C', 0x20, struct snd_compr_tstamp)
#define SNDRV_COMPRESS_AVAIL _IOR('C', 0x21, struct snd_compr_avail)
#define SNDRV_COMPRESS_PAUSE _IO('C', 0x30)
#define SNDRV_COMPRESS_RESUME _IO('C', 0x31)
#define SNDRV_COMPRESS_START _IO('C', 0x32)
#define SNDRV_COMPRESS_STOP _IO('C', 0x33)
#define SNDRV_COMPRESS_DRAIN _IO('C', 0x34)
/*
* TODO
* 1. add mmap support
*
*/
#define SND_COMPR_TRIGGER_DRAIN 7 /*FIXME move this to pcm.h */
#endif
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/*
* compress_params.h - codec types and parameters for compressed data
* streaming interface
*
* Copyright (C) 2011 Intel Corporation
* Authors: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
* Vinod Koul <vinod.koul@linux.intel.com>
*
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; version 2 of the License.
*
* This program is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
*
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
*
* The definitions in this file are derived from the OpenMAX AL version 1.1
* and OpenMAX IL v 1.1.2 header files which contain the copyright notice below.
*
* Copyright (c) 2007-2010 The Khronos Group Inc.
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and/or associated documentation files (the
* "Materials "), to deal in the Materials without restriction, including
* without limitation the rights to use, copy, modify, merge, publish,
* distribute, sublicense, and/or sell copies of the Materials, and to
* permit persons to whom the Materials are furnished to do so, subject to
* the following conditions:
*
* The above copyright notice and this permission notice shall be included
* in all copies or substantial portions of the Materials.
*
* THE MATERIALS ARE PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS
* OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY
* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
* MATERIALS OR THE USE OR OTHER DEALINGS IN THE MATERIALS.
*
*/
#ifndef __SND_COMPRESS_PARAMS_H
#define __SND_COMPRESS_PARAMS_H
#include <linux/types.h>
/* AUDIO CODECS SUPPORTED */
#define MAX_NUM_CODECS 32
#define MAX_NUM_CODEC_DESCRIPTORS 32
#define MAX_NUM_BITRATES 32
/* Codecs are listed linearly to allow for extensibility */
#define SND_AUDIOCODEC_PCM ((__u32) 0x00000001)
#define SND_AUDIOCODEC_MP3 ((__u32) 0x00000002)
#define SND_AUDIOCODEC_AMR ((__u32) 0x00000003)
#define SND_AUDIOCODEC_AMRWB ((__u32) 0x00000004)
#define SND_AUDIOCODEC_AMRWBPLUS ((__u32) 0x00000005)
#define SND_AUDIOCODEC_AAC ((__u32) 0x00000006)
#define SND_AUDIOCODEC_WMA ((__u32) 0x00000007)
#define SND_AUDIOCODEC_REAL ((__u32) 0x00000008)
#define SND_AUDIOCODEC_VORBIS ((__u32) 0x00000009)
#define SND_AUDIOCODEC_FLAC ((__u32) 0x0000000A)
#define SND_AUDIOCODEC_IEC61937 ((__u32) 0x0000000B)
#define SND_AUDIOCODEC_G723_1 ((__u32) 0x0000000C)
#define SND_AUDIOCODEC_G729 ((__u32) 0x0000000D)
#define SND_AUDIOCODEC_MAX SND_AUDIOCODEC_G729
/*
* Profile and modes are listed with bit masks. This allows for a
* more compact representation of fields that will not evolve
* (in contrast to the list of codecs)
*/
#define SND_AUDIOPROFILE_PCM ((__u32) 0x00000001)
/* MP3 modes are only useful for encoders */
#define SND_AUDIOCHANMODE_MP3_MONO ((__u32) 0x00000001)
#define SND_AUDIOCHANMODE_MP3_STEREO ((__u32) 0x00000002)
#define SND_AUDIOCHANMODE_MP3_JOINTSTEREO ((__u32) 0x00000004)
#define SND_AUDIOCHANMODE_MP3_DUAL ((__u32) 0x00000008)
#define SND_AUDIOPROFILE_AMR ((__u32) 0x00000001)
/* AMR modes are only useful for encoders */
#define SND_AUDIOMODE_AMR_DTX_OFF ((__u32) 0x00000001)
#define SND_AUDIOMODE_AMR_VAD1 ((__u32) 0x00000002)
#define SND_AUDIOMODE_AMR_VAD2 ((__u32) 0x00000004)
#define SND_AUDIOSTREAMFORMAT_UNDEFINED ((__u32) 0x00000000)
#define SND_AUDIOSTREAMFORMAT_CONFORMANCE ((__u32) 0x00000001)
#define SND_AUDIOSTREAMFORMAT_IF1 ((__u32) 0x00000002)
#define SND_AUDIOSTREAMFORMAT_IF2 ((__u32) 0x00000004)
#define SND_AUDIOSTREAMFORMAT_FSF ((__u32) 0x00000008)
#define SND_AUDIOSTREAMFORMAT_RTPPAYLOAD ((__u32) 0x00000010)
#define SND_AUDIOSTREAMFORMAT_ITU ((__u32) 0x00000020)
#define SND_AUDIOPROFILE_AMRWB ((__u32) 0x00000001)
/* AMRWB modes are only useful for encoders */
#define SND_AUDIOMODE_AMRWB_DTX_OFF ((__u32) 0x00000001)
#define SND_AUDIOMODE_AMRWB_VAD1 ((__u32) 0x00000002)
#define SND_AUDIOMODE_AMRWB_VAD2 ((__u32) 0x00000004)
#define SND_AUDIOPROFILE_AMRWBPLUS ((__u32) 0x00000001)
#define SND_AUDIOPROFILE_AAC ((__u32) 0x00000001)
/* AAC modes are required for encoders and decoders */
#define SND_AUDIOMODE_AAC_MAIN ((__u32) 0x00000001)
#define SND_AUDIOMODE_AAC_LC ((__u32) 0x00000002)
#define SND_AUDIOMODE_AAC_SSR ((__u32) 0x00000004)
#define SND_AUDIOMODE_AAC_LTP ((__u32) 0x00000008)
#define SND_AUDIOMODE_AAC_HE ((__u32) 0x00000010)
#define SND_AUDIOMODE_AAC_SCALABLE ((__u32) 0x00000020)
#define SND_AUDIOMODE_AAC_ERLC ((__u32) 0x00000040)
#define SND_AUDIOMODE_AAC_LD ((__u32) 0x00000080)
#define SND_AUDIOMODE_AAC_HE_PS ((__u32) 0x00000100)
#define SND_AUDIOMODE_AAC_HE_MPS ((__u32) 0x00000200)
/* AAC formats are required for encoders and decoders */
#define SND_AUDIOSTREAMFORMAT_MP2ADTS ((__u32) 0x00000001)
#define SND_AUDIOSTREAMFORMAT_MP4ADTS ((__u32) 0x00000002)
#define SND_AUDIOSTREAMFORMAT_MP4LOAS ((__u32) 0x00000004)
#define SND_AUDIOSTREAMFORMAT_MP4LATM ((__u32) 0x00000008)
#define SND_AUDIOSTREAMFORMAT_ADIF ((__u32) 0x00000010)
#define SND_AUDIOSTREAMFORMAT_MP4FF ((__u32) 0x00000020)
#define SND_AUDIOSTREAMFORMAT_RAW ((__u32) 0x00000040)
#define SND_AUDIOPROFILE_WMA7 ((__u32) 0x00000001)
#define SND_AUDIOPROFILE_WMA8 ((__u32) 0x00000002)
#define SND_AUDIOPROFILE_WMA9 ((__u32) 0x00000004)
#define SND_AUDIOPROFILE_WMA10 ((__u32) 0x00000008)
#define SND_AUDIOMODE_WMA_LEVEL1 ((__u32) 0x00000001)
#define SND_AUDIOMODE_WMA_LEVEL2 ((__u32) 0x00000002)
#define SND_AUDIOMODE_WMA_LEVEL3 ((__u32) 0x00000004)
#define SND_AUDIOMODE_WMA_LEVEL4 ((__u32) 0x00000008)
#define SND_AUDIOMODE_WMAPRO_LEVELM0 ((__u32) 0x00000010)
#define SND_AUDIOMODE_WMAPRO_LEVELM1 ((__u32) 0x00000020)
#define SND_AUDIOMODE_WMAPRO_LEVELM2 ((__u32) 0x00000040)
#define SND_AUDIOMODE_WMAPRO_LEVELM3 ((__u32) 0x00000080)
#define SND_AUDIOSTREAMFORMAT_WMA_ASF ((__u32) 0x00000001)
/*
* Some implementations strip the ASF header and only send ASF packets
* to the DSP
*/
#define SND_AUDIOSTREAMFORMAT_WMA_NOASF_HDR ((__u32) 0x00000002)
#define SND_AUDIOPROFILE_REALAUDIO ((__u32) 0x00000001)
#define SND_AUDIOMODE_REALAUDIO_G2 ((__u32) 0x00000001)
#define SND_AUDIOMODE_REALAUDIO_8 ((__u32) 0x00000002)
#define SND_AUDIOMODE_REALAUDIO_10 ((__u32) 0x00000004)
#define SND_AUDIOMODE_REALAUDIO_SURROUND ((__u32) 0x00000008)
#define SND_AUDIOPROFILE_VORBIS ((__u32) 0x00000001)
#define SND_AUDIOMODE_VORBIS ((__u32) 0x00000001)
#define SND_AUDIOPROFILE_FLAC ((__u32) 0x00000001)
/*
* Define quality levels for FLAC encoders, from LEVEL0 (fast)
* to LEVEL8 (best)
*/
#define SND_AUDIOMODE_FLAC_LEVEL0 ((__u32) 0x00000001)
#define SND_AUDIOMODE_FLAC_LEVEL1 ((__u32) 0x00000002)
#define SND_AUDIOMODE_FLAC_LEVEL2 ((__u32) 0x00000004)
#define SND_AUDIOMODE_FLAC_LEVEL3 ((__u32) 0x00000008)
#define SND_AUDIOMODE_FLAC_LEVEL4 ((__u32) 0x00000010)
#define SND_AUDIOMODE_FLAC_LEVEL5 ((__u32) 0x00000020)
#define SND_AUDIOMODE_FLAC_LEVEL6 ((__u32) 0x00000040)
#define SND_AUDIOMODE_FLAC_LEVEL7 ((__u32) 0x00000080)
#define SND_AUDIOMODE_FLAC_LEVEL8 ((__u32) 0x00000100)
#define SND_AUDIOSTREAMFORMAT_FLAC ((__u32) 0x00000001)
#define SND_AUDIOSTREAMFORMAT_FLAC_OGG ((__u32) 0x00000002)
/* IEC61937 payloads without CUVP and preambles */
#define SND_AUDIOPROFILE_IEC61937 ((__u32) 0x00000001)
/* IEC61937 with S/PDIF preambles+CUVP bits in 32-bit containers */
#define SND_AUDIOPROFILE_IEC61937_SPDIF ((__u32) 0x00000002)
/*
* IEC modes are mandatory for decoders. Format autodetection
* will only happen on the DSP side with mode 0. The PCM mode should
* not be used, the PCM codec should be used instead.
*/
#define SND_AUDIOMODE_IEC_REF_STREAM_HEADER ((__u32) 0x00000000)
#define SND_AUDIOMODE_IEC_LPCM ((__u32) 0x00000001)
#define SND_AUDIOMODE_IEC_AC3 ((__u32) 0x00000002)
#define SND_AUDIOMODE_IEC_MPEG1 ((__u32) 0x00000004)
#define SND_AUDIOMODE_IEC_MP3 ((__u32) 0x00000008)
#define SND_AUDIOMODE_IEC_MPEG2 ((__u32) 0x00000010)
#define SND_AUDIOMODE_IEC_AACLC ((__u32) 0x00000020)
#define SND_AUDIOMODE_IEC_DTS ((__u32) 0x00000040)
#define SND_AUDIOMODE_IEC_ATRAC ((__u32) 0x00000080)
#define SND_AUDIOMODE_IEC_SACD ((__u32) 0x00000100)
#define SND_AUDIOMODE_IEC_EAC3 ((__u32) 0x00000200)
#define SND_AUDIOMODE_IEC_DTS_HD ((__u32) 0x00000400)
#define SND_AUDIOMODE_IEC_MLP ((__u32) 0x00000800)
#define SND_AUDIOMODE_IEC_DST ((__u32) 0x00001000)
#define SND_AUDIOMODE_IEC_WMAPRO ((__u32) 0x00002000)
#define SND_AUDIOMODE_IEC_REF_CXT ((__u32) 0x00004000)
#define SND_AUDIOMODE_IEC_HE_AAC ((__u32) 0x00008000)
#define SND_AUDIOMODE_IEC_HE_AAC2 ((__u32) 0x00010000)
#define SND_AUDIOMODE_IEC_MPEG_SURROUND ((__u32) 0x00020000)
#define SND_AUDIOPROFILE_G723_1 ((__u32) 0x00000001)
#define SND_AUDIOMODE_G723_1_ANNEX_A ((__u32) 0x00000001)
#define SND_AUDIOMODE_G723_1_ANNEX_B ((__u32) 0x00000002)
#define SND_AUDIOMODE_G723_1_ANNEX_C ((__u32) 0x00000004)
#define SND_AUDIOPROFILE_G729 ((__u32) 0x00000001)
#define SND_AUDIOMODE_G729_ANNEX_A ((__u32) 0x00000001)
#define SND_AUDIOMODE_G729_ANNEX_B ((__u32) 0x00000002)
/* <FIXME: multichannel encoders aren't supported for now. Would need
an additional definition of channel arrangement> */
/* VBR/CBR definitions */
#define SND_RATECONTROLMODE_CONSTANTBITRATE ((__u32) 0x00000001)
#define SND_RATECONTROLMODE_VARIABLEBITRATE ((__u32) 0x00000002)
/* Encoder options */
struct snd_enc_wma {
__u32 super_block_align; /* WMA Type-specific data */
};
/**
* struct snd_enc_vorbis
* @quality: Sets encoding quality to n, between -1 (low) and 10 (high).
* In the default mode of operation, the quality level is 3.
* Normal quality range is 0 - 10.
* @managed: Boolean. Set bitrate management mode. This turns off the
* normal VBR encoding, but allows hard or soft bitrate constraints to be
* enforced by the encoder. This mode can be slower, and may also be
* lower quality. It is primarily useful for streaming.
* @max_bit_rate: Enabled only if managed is TRUE
* @min_bit_rate: Enabled only if managed is TRUE
* @downmix: Boolean. Downmix input from stereo to mono (has no effect on
* non-stereo streams). Useful for lower-bitrate encoding.
*
* These options were extracted from the OpenMAX IL spec and Gstreamer vorbisenc
* properties
*
* For best quality users should specify VBR mode and set quality levels.
*/
struct snd_enc_vorbis {
__s32 quality;
__u32 managed;
__u32 max_bit_rate;
__u32 min_bit_rate;
__u32 downmix;
};
/**
* struct snd_enc_real
* @quant_bits: number of coupling quantization bits in the stream
* @start_region: coupling start region in the stream
* @num_regions: number of regions value
*
* These options were extracted from the OpenMAX IL spec
*/
struct snd_enc_real {
__u32 quant_bits;
__u32 start_region;
__u32 num_regions;
};
/**
* struct snd_enc_flac
* @num: serial number, valid only for OGG formats
* needs to be set by application
* @gain: Add replay gain tags
*
* These options were extracted from the FLAC online documentation
* at http://flac.sourceforge.net/documentation_tools_flac.html
*
* To make the API simpler, it is assumed that the user will select quality
* profiles. Additional options that affect encoding quality and speed can
* be added at a later stage if needed.
*
* By default the Subset format is used by encoders.
*
* TAGS such as pictures, etc, cannot be handled by an offloaded encoder and are
* not supported in this API.
*/
struct snd_enc_flac {
__u32 num;
__u32 gain;
};
struct snd_enc_generic {
__u32 bw; /* encoder bandwidth */
__s32 reserved[15];
};
union snd_codec_options {
struct snd_enc_wma wma;
struct snd_enc_vorbis vorbis;
struct snd_enc_real real;
struct snd_enc_flac flac;
struct snd_enc_generic generic;
};
/** struct snd_codec_desc - description of codec capabilities
* @max_ch: Maximum number of audio channels
* @sample_rates: Sampling rates in Hz, use SNDRV_PCM_RATE_xxx for this
* @bit_rate: Indexed array containing supported bit rates
* @num_bitrates: Number of valid values in bit_rate array
* @rate_control: value is specified by SND_RATECONTROLMODE defines.
* @profiles: Supported profiles. See SND_AUDIOPROFILE defines.
* @modes: Supported modes. See SND_AUDIOMODE defines
* @formats: Supported formats. See SND_AUDIOSTREAMFORMAT defines
* @min_buffer: Minimum buffer size handled by codec implementation
* @reserved: reserved for future use
*
* This structure provides a scalar value for profiles, modes and stream
* format fields.
* If an implementation supports multiple combinations, they will be listed as
* codecs with different descriptors, for example there would be 2 descriptors
* for AAC-RAW and AAC-ADTS.
* This entails some redundancy but makes it easier to avoid invalid
* configurations.
*
*/
struct snd_codec_desc {
__u32 max_ch;
__u32 sample_rates;
__u32 bit_rate[MAX_NUM_BITRATES];
__u32 num_bitrates;
__u32 rate_control;
__u32 profiles;
__u32 modes;
__u32 formats;
__u32 min_buffer;
__u32 reserved[15];
};
/** struct snd_codec
* @id: Identifies the supported audio encoder/decoder.
* See SND_AUDIOCODEC macros.
* @ch_in: Number of input audio channels
* @ch_out: Number of output channels. In case of contradiction between
* this field and the channelMode field, the channelMode field
* overrides.
* @sample_rate: Audio sample rate of input data
* @bit_rate: Bitrate of encoded data. May be ignored by decoders
* @rate_control: Encoding rate control. See SND_RATECONTROLMODE defines.
* Encoders may rely on profiles for quality levels.
* May be ignored by decoders.
* @profile: Mandatory for encoders, can be mandatory for specific
* decoders as well. See SND_AUDIOPROFILE defines.
* @level: Supported level (Only used by WMA at the moment)
* @ch_mode: Channel mode for encoder. See SND_AUDIOCHANMODE defines
* @format: Format of encoded bistream. Mandatory when defined.
* See SND_AUDIOSTREAMFORMAT defines.
* @align: Block alignment in bytes of an audio sample.
* Only required for PCM or IEC formats.
* @options: encoder-specific settings
* @reserved: reserved for future use
*/
struct snd_codec {
__u32 id;
__u32 ch_in;
__u32 ch_out;
__u32 sample_rate;
__u32 bit_rate;
__u32 rate_control;
__u32 profile;
__u32 level;
__u32 ch_mode;
__u32 format;
__u32 align;
union snd_codec_options options;
__u32 reserved[3];
};
#endif
+373
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@@ -0,0 +1,373 @@
/*
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
* Creative Labs, Inc.
* Definitions for EMU10K1 (SB Live!) chips
*
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
*/
#ifndef _UAPI__SOUND_EMU10K1_H
#define _UAPI__SOUND_EMU10K1_H
#include <linux/types.h>
/*
* ---- FX8010 ----
*/
#define EMU10K1_CARD_CREATIVE 0x00000000
#define EMU10K1_CARD_EMUAPS 0x00000001
#define EMU10K1_FX8010_PCM_COUNT 8
/* instruction set */
#define iMAC0 0x00 /* R = A + (X * Y >> 31) ; saturation */
#define iMAC1 0x01 /* R = A + (-X * Y >> 31) ; saturation */
#define iMAC2 0x02 /* R = A + (X * Y >> 31) ; wraparound */
#define iMAC3 0x03 /* R = A + (-X * Y >> 31) ; wraparound */
#define iMACINT0 0x04 /* R = A + X * Y ; saturation */
#define iMACINT1 0x05 /* R = A + X * Y ; wraparound (31-bit) */
#define iACC3 0x06 /* R = A + X + Y ; saturation */
#define iMACMV 0x07 /* R = A, acc += X * Y >> 31 */
#define iANDXOR 0x08 /* R = (A & X) ^ Y */
#define iTSTNEG 0x09 /* R = (A >= Y) ? X : ~X */
#define iLIMITGE 0x0a /* R = (A >= Y) ? X : Y */
#define iLIMITLT 0x0b /* R = (A < Y) ? X : Y */
#define iLOG 0x0c /* R = linear_data, A (log_data), X (max_exp), Y (format_word) */
#define iEXP 0x0d /* R = log_data, A (linear_data), X (max_exp), Y (format_word) */
#define iINTERP 0x0e /* R = A + (X * (Y - A) >> 31) ; saturation */
#define iSKIP 0x0f /* R = A (cc_reg), X (count), Y (cc_test) */
/* GPRs */
#define FXBUS(x) (0x00 + (x)) /* x = 0x00 - 0x0f */
#define EXTIN(x) (0x10 + (x)) /* x = 0x00 - 0x0f */
#define EXTOUT(x) (0x20 + (x)) /* x = 0x00 - 0x0f physical outs -> FXWC low 16 bits */
#define FXBUS2(x) (0x30 + (x)) /* x = 0x00 - 0x0f copies of fx buses for capture -> FXWC high 16 bits */
/* NB: 0x31 and 0x32 are shared with Center/LFE on SB live 5.1 */
#define C_00000000 0x40
#define C_00000001 0x41
#define C_00000002 0x42
#define C_00000003 0x43
#define C_00000004 0x44
#define C_00000008 0x45
#define C_00000010 0x46
#define C_00000020 0x47
#define C_00000100 0x48
#define C_00010000 0x49
#define C_00080000 0x4a
#define C_10000000 0x4b
#define C_20000000 0x4c
#define C_40000000 0x4d
#define C_80000000 0x4e
#define C_7fffffff 0x4f
#define C_ffffffff 0x50
#define C_fffffffe 0x51
#define C_c0000000 0x52
#define C_4f1bbcdc 0x53
#define C_5a7ef9db 0x54
#define C_00100000 0x55 /* ?? */
#define GPR_ACCU 0x56 /* ACCUM, accumulator */
#define GPR_COND 0x57 /* CCR, condition register */
#define GPR_NOISE0 0x58 /* noise source */
#define GPR_NOISE1 0x59 /* noise source */
#define GPR_IRQ 0x5a /* IRQ register */
#define GPR_DBAC 0x5b /* TRAM Delay Base Address Counter */
#define GPR(x) (FXGPREGBASE + (x)) /* free GPRs: x = 0x00 - 0xff */
#define ITRAM_DATA(x) (TANKMEMDATAREGBASE + 0x00 + (x)) /* x = 0x00 - 0x7f */
#define ETRAM_DATA(x) (TANKMEMDATAREGBASE + 0x80 + (x)) /* x = 0x00 - 0x1f */
#define ITRAM_ADDR(x) (TANKMEMADDRREGBASE + 0x00 + (x)) /* x = 0x00 - 0x7f */
#define ETRAM_ADDR(x) (TANKMEMADDRREGBASE + 0x80 + (x)) /* x = 0x00 - 0x1f */
#define A_ITRAM_DATA(x) (TANKMEMDATAREGBASE + 0x00 + (x)) /* x = 0x00 - 0xbf */
#define A_ETRAM_DATA(x) (TANKMEMDATAREGBASE + 0xc0 + (x)) /* x = 0x00 - 0x3f */
#define A_ITRAM_ADDR(x) (TANKMEMADDRREGBASE + 0x00 + (x)) /* x = 0x00 - 0xbf */
#define A_ETRAM_ADDR(x) (TANKMEMADDRREGBASE + 0xc0 + (x)) /* x = 0x00 - 0x3f */
#define A_ITRAM_CTL(x) (A_TANKMEMCTLREGBASE + 0x00 + (x)) /* x = 0x00 - 0xbf */
#define A_ETRAM_CTL(x) (A_TANKMEMCTLREGBASE + 0xc0 + (x)) /* x = 0x00 - 0x3f */
#define A_FXBUS(x) (0x00 + (x)) /* x = 0x00 - 0x3f FX buses */
#define A_EXTIN(x) (0x40 + (x)) /* x = 0x00 - 0x0f physical ins */
#define A_P16VIN(x) (0x50 + (x)) /* x = 0x00 - 0x0f p16v ins (A2 only) "EMU32 inputs" */
#define A_EXTOUT(x) (0x60 + (x)) /* x = 0x00 - 0x1f physical outs -> A_FXWC1 0x79-7f unknown */
#define A_FXBUS2(x) (0x80 + (x)) /* x = 0x00 - 0x1f extra outs used for EFX capture -> A_FXWC2 */
#define A_EMU32OUTH(x) (0xa0 + (x)) /* x = 0x00 - 0x0f "EMU32_OUT_10 - _1F" - ??? */
#define A_EMU32OUTL(x) (0xb0 + (x)) /* x = 0x00 - 0x0f "EMU32_OUT_1 - _F" - ??? */
#define A3_EMU32IN(x) (0x160 + (x)) /* x = 0x00 - 0x3f "EMU32_IN_00 - _3F" - Only when .device = 0x0008 */
#define A3_EMU32OUT(x) (0x1E0 + (x)) /* x = 0x00 - 0x0f "EMU32_OUT_00 - _3F" - Only when .device = 0x0008 */
#define A_GPR(x) (A_FXGPREGBASE + (x))
/* cc_reg constants */
#define CC_REG_NORMALIZED C_00000001
#define CC_REG_BORROW C_00000002
#define CC_REG_MINUS C_00000004
#define CC_REG_ZERO C_00000008
#define CC_REG_SATURATE C_00000010
#define CC_REG_NONZERO C_00000100
/* FX buses */
#define FXBUS_PCM_LEFT 0x00
#define FXBUS_PCM_RIGHT 0x01
#define FXBUS_PCM_LEFT_REAR 0x02
#define FXBUS_PCM_RIGHT_REAR 0x03
#define FXBUS_MIDI_LEFT 0x04
#define FXBUS_MIDI_RIGHT 0x05
#define FXBUS_PCM_CENTER 0x06
#define FXBUS_PCM_LFE 0x07
#define FXBUS_PCM_LEFT_FRONT 0x08
#define FXBUS_PCM_RIGHT_FRONT 0x09
#define FXBUS_MIDI_REVERB 0x0c
#define FXBUS_MIDI_CHORUS 0x0d
#define FXBUS_PCM_LEFT_SIDE 0x0e
#define FXBUS_PCM_RIGHT_SIDE 0x0f
#define FXBUS_PT_LEFT 0x14
#define FXBUS_PT_RIGHT 0x15
/* Inputs */
#define EXTIN_AC97_L 0x00 /* AC'97 capture channel - left */
#define EXTIN_AC97_R 0x01 /* AC'97 capture channel - right */
#define EXTIN_SPDIF_CD_L 0x02 /* internal S/PDIF CD - onboard - left */
#define EXTIN_SPDIF_CD_R 0x03 /* internal S/PDIF CD - onboard - right */
#define EXTIN_ZOOM_L 0x04 /* Zoom Video I2S - left */
#define EXTIN_ZOOM_R 0x05 /* Zoom Video I2S - right */
#define EXTIN_TOSLINK_L 0x06 /* LiveDrive - TOSLink Optical - left */
#define EXTIN_TOSLINK_R 0x07 /* LiveDrive - TOSLink Optical - right */
#define EXTIN_LINE1_L 0x08 /* LiveDrive - Line/Mic 1 - left */
#define EXTIN_LINE1_R 0x09 /* LiveDrive - Line/Mic 1 - right */
#define EXTIN_COAX_SPDIF_L 0x0a /* LiveDrive - Coaxial S/PDIF - left */
#define EXTIN_COAX_SPDIF_R 0x0b /* LiveDrive - Coaxial S/PDIF - right */
#define EXTIN_LINE2_L 0x0c /* LiveDrive - Line/Mic 2 - left */
#define EXTIN_LINE2_R 0x0d /* LiveDrive - Line/Mic 2 - right */
/* Outputs */
#define EXTOUT_AC97_L 0x00 /* AC'97 playback channel - left */
#define EXTOUT_AC97_R 0x01 /* AC'97 playback channel - right */
#define EXTOUT_TOSLINK_L 0x02 /* LiveDrive - TOSLink Optical - left */
#define EXTOUT_TOSLINK_R 0x03 /* LiveDrive - TOSLink Optical - right */
#define EXTOUT_AC97_CENTER 0x04 /* SB Live 5.1 - center */
#define EXTOUT_AC97_LFE 0x05 /* SB Live 5.1 - LFE */
#define EXTOUT_HEADPHONE_L 0x06 /* LiveDrive - Headphone - left */
#define EXTOUT_HEADPHONE_R 0x07 /* LiveDrive - Headphone - right */
#define EXTOUT_REAR_L 0x08 /* Rear channel - left */
#define EXTOUT_REAR_R 0x09 /* Rear channel - right */
#define EXTOUT_ADC_CAP_L 0x0a /* ADC Capture buffer - left */
#define EXTOUT_ADC_CAP_R 0x0b /* ADC Capture buffer - right */
#define EXTOUT_MIC_CAP 0x0c /* MIC Capture buffer */
#define EXTOUT_AC97_REAR_L 0x0d /* SB Live 5.1 (c) 2003 - Rear Left */
#define EXTOUT_AC97_REAR_R 0x0e /* SB Live 5.1 (c) 2003 - Rear Right */
#define EXTOUT_ACENTER 0x11 /* Analog Center */
#define EXTOUT_ALFE 0x12 /* Analog LFE */
/* Audigy Inputs */
#define A_EXTIN_AC97_L 0x00 /* AC'97 capture channel - left */
#define A_EXTIN_AC97_R 0x01 /* AC'97 capture channel - right */
#define A_EXTIN_SPDIF_CD_L 0x02 /* digital CD left */
#define A_EXTIN_SPDIF_CD_R 0x03 /* digital CD left */
#define A_EXTIN_OPT_SPDIF_L 0x04 /* audigy drive Optical SPDIF - left */
#define A_EXTIN_OPT_SPDIF_R 0x05 /* right */
#define A_EXTIN_LINE2_L 0x08 /* audigy drive line2/mic2 - left */
#define A_EXTIN_LINE2_R 0x09 /* right */
#define A_EXTIN_ADC_L 0x0a /* Philips ADC - left */
#define A_EXTIN_ADC_R 0x0b /* right */
#define A_EXTIN_AUX2_L 0x0c /* audigy drive aux2 - left */
#define A_EXTIN_AUX2_R 0x0d /* - right */
/* Audigiy Outputs */
#define A_EXTOUT_FRONT_L 0x00 /* digital front left */
#define A_EXTOUT_FRONT_R 0x01 /* right */
#define A_EXTOUT_CENTER 0x02 /* digital front center */
#define A_EXTOUT_LFE 0x03 /* digital front lfe */
#define A_EXTOUT_HEADPHONE_L 0x04 /* headphone audigy drive left */
#define A_EXTOUT_HEADPHONE_R 0x05 /* right */
#define A_EXTOUT_REAR_L 0x06 /* digital rear left */
#define A_EXTOUT_REAR_R 0x07 /* right */
#define A_EXTOUT_AFRONT_L 0x08 /* analog front left */
#define A_EXTOUT_AFRONT_R 0x09 /* right */
#define A_EXTOUT_ACENTER 0x0a /* analog center */
#define A_EXTOUT_ALFE 0x0b /* analog LFE */
#define A_EXTOUT_ASIDE_L 0x0c /* analog side left - Audigy 2 ZS */
#define A_EXTOUT_ASIDE_R 0x0d /* right - Audigy 2 ZS */
#define A_EXTOUT_AREAR_L 0x0e /* analog rear left */
#define A_EXTOUT_AREAR_R 0x0f /* right */
#define A_EXTOUT_AC97_L 0x10 /* AC97 left (front) */
#define A_EXTOUT_AC97_R 0x11 /* right */
#define A_EXTOUT_ADC_CAP_L 0x16 /* ADC capture buffer left */
#define A_EXTOUT_ADC_CAP_R 0x17 /* right */
#define A_EXTOUT_MIC_CAP 0x18 /* Mic capture buffer */
/* Audigy constants */
#define A_C_00000000 0xc0
#define A_C_00000001 0xc1
#define A_C_00000002 0xc2
#define A_C_00000003 0xc3
#define A_C_00000004 0xc4
#define A_C_00000008 0xc5
#define A_C_00000010 0xc6
#define A_C_00000020 0xc7
#define A_C_00000100 0xc8
#define A_C_00010000 0xc9
#define A_C_00000800 0xca
#define A_C_10000000 0xcb
#define A_C_20000000 0xcc
#define A_C_40000000 0xcd
#define A_C_80000000 0xce
#define A_C_7fffffff 0xcf
#define A_C_ffffffff 0xd0
#define A_C_fffffffe 0xd1
#define A_C_c0000000 0xd2
#define A_C_4f1bbcdc 0xd3
#define A_C_5a7ef9db 0xd4
#define A_C_00100000 0xd5
#define A_GPR_ACCU 0xd6 /* ACCUM, accumulator */
#define A_GPR_COND 0xd7 /* CCR, condition register */
#define A_GPR_NOISE0 0xd8 /* noise source */
#define A_GPR_NOISE1 0xd9 /* noise source */
#define A_GPR_IRQ 0xda /* IRQ register */
#define A_GPR_DBAC 0xdb /* TRAM Delay Base Address Counter - internal */
#define A_GPR_DBACE 0xde /* TRAM Delay Base Address Counter - external */
/* definitions for debug register */
#define EMU10K1_DBG_ZC 0x80000000 /* zero tram counter */
#define EMU10K1_DBG_SATURATION_OCCURED 0x02000000 /* saturation control */
#define EMU10K1_DBG_SATURATION_ADDR 0x01ff0000 /* saturation address */
#define EMU10K1_DBG_SINGLE_STEP 0x00008000 /* single step mode */
#define EMU10K1_DBG_STEP 0x00004000 /* start single step */
#define EMU10K1_DBG_CONDITION_CODE 0x00003e00 /* condition code */
#define EMU10K1_DBG_SINGLE_STEP_ADDR 0x000001ff /* single step address */
/* tank memory address line */
#ifndef __KERNEL__
#define TANKMEMADDRREG_ADDR_MASK 0x000fffff /* 20 bit tank address field */
#define TANKMEMADDRREG_CLEAR 0x00800000 /* Clear tank memory */
#define TANKMEMADDRREG_ALIGN 0x00400000 /* Align read or write relative to tank access */
#define TANKMEMADDRREG_WRITE 0x00200000 /* Write to tank memory */
#define TANKMEMADDRREG_READ 0x00100000 /* Read from tank memory */
#endif
struct snd_emu10k1_fx8010_info {
unsigned int internal_tram_size; /* in samples */
unsigned int external_tram_size; /* in samples */
char fxbus_names[16][32]; /* names of FXBUSes */
char extin_names[16][32]; /* names of external inputs */
char extout_names[32][32]; /* names of external outputs */
unsigned int gpr_controls; /* count of GPR controls */
};
#define EMU10K1_GPR_TRANSLATION_NONE 0
#define EMU10K1_GPR_TRANSLATION_TABLE100 1
#define EMU10K1_GPR_TRANSLATION_BASS 2
#define EMU10K1_GPR_TRANSLATION_TREBLE 3
#define EMU10K1_GPR_TRANSLATION_ONOFF 4
struct snd_emu10k1_fx8010_control_gpr {
struct snd_ctl_elem_id id; /* full control ID definition */
unsigned int vcount; /* visible count */
unsigned int count; /* count of GPR (1..16) */
unsigned short gpr[32]; /* GPR number(s) */
unsigned int value[32]; /* initial values */
unsigned int min; /* minimum range */
unsigned int max; /* maximum range */
unsigned int translation; /* translation type (EMU10K1_GPR_TRANSLATION*) */
const unsigned int *tlv;
};
/* old ABI without TLV support */
struct snd_emu10k1_fx8010_control_old_gpr {
struct snd_ctl_elem_id id;
unsigned int vcount;
unsigned int count;
unsigned short gpr[32];
unsigned int value[32];
unsigned int min;
unsigned int max;
unsigned int translation;
};
struct snd_emu10k1_fx8010_code {
char name[128];
DECLARE_BITMAP(gpr_valid, 0x200); /* bitmask of valid initializers */
__u32 __user *gpr_map; /* initializers */
unsigned int gpr_add_control_count; /* count of GPR controls to add/replace */
struct snd_emu10k1_fx8010_control_gpr __user *gpr_add_controls; /* GPR controls to add/replace */
unsigned int gpr_del_control_count; /* count of GPR controls to remove */
struct snd_ctl_elem_id __user *gpr_del_controls; /* IDs of GPR controls to remove */
unsigned int gpr_list_control_count; /* count of GPR controls to list */
unsigned int gpr_list_control_total; /* total count of GPR controls */
struct snd_emu10k1_fx8010_control_gpr __user *gpr_list_controls; /* listed GPR controls */
DECLARE_BITMAP(tram_valid, 0x100); /* bitmask of valid initializers */
__u32 __user *tram_data_map; /* data initializers */
__u32 __user *tram_addr_map; /* map initializers */
DECLARE_BITMAP(code_valid, 1024); /* bitmask of valid instructions */
__u32 __user *code; /* one instruction - 64 bits */
};
struct snd_emu10k1_fx8010_tram {
unsigned int address; /* 31.bit == 1 -> external TRAM */
unsigned int size; /* size in samples (4 bytes) */
unsigned int *samples; /* pointer to samples (20-bit) */
/* NULL->clear memory */
};
struct snd_emu10k1_fx8010_pcm_rec {
unsigned int substream; /* substream number */
unsigned int res1; /* reserved */
unsigned int channels; /* 16-bit channels count, zero = remove this substream */
unsigned int tram_start; /* ring buffer position in TRAM (in samples) */
unsigned int buffer_size; /* count of buffered samples */
unsigned short gpr_size; /* GPR containing size of ringbuffer in samples (host) */
unsigned short gpr_ptr; /* GPR containing current pointer in the ring buffer (host = reset, FX8010) */
unsigned short gpr_count; /* GPR containing count of samples between two interrupts (host) */
unsigned short gpr_tmpcount; /* GPR containing current count of samples to interrupt (host = set, FX8010) */
unsigned short gpr_trigger; /* GPR containing trigger (activate) information (host) */
unsigned short gpr_running; /* GPR containing info if PCM is running (FX8010) */
unsigned char pad; /* reserved */
unsigned char etram[32]; /* external TRAM address & data (one per channel) */
unsigned int res2; /* reserved */
};
#define SNDRV_EMU10K1_VERSION SNDRV_PROTOCOL_VERSION(1, 0, 1)
#define SNDRV_EMU10K1_IOCTL_INFO _IOR ('H', 0x10, struct snd_emu10k1_fx8010_info)
#define SNDRV_EMU10K1_IOCTL_CODE_POKE _IOW ('H', 0x11, struct snd_emu10k1_fx8010_code)
#define SNDRV_EMU10K1_IOCTL_CODE_PEEK _IOWR('H', 0x12, struct snd_emu10k1_fx8010_code)
#define SNDRV_EMU10K1_IOCTL_TRAM_SETUP _IOW ('H', 0x20, int)
#define SNDRV_EMU10K1_IOCTL_TRAM_POKE _IOW ('H', 0x21, struct snd_emu10k1_fx8010_tram)
#define SNDRV_EMU10K1_IOCTL_TRAM_PEEK _IOWR('H', 0x22, struct snd_emu10k1_fx8010_tram)
#define SNDRV_EMU10K1_IOCTL_PCM_POKE _IOW ('H', 0x30, struct snd_emu10k1_fx8010_pcm_rec)
#define SNDRV_EMU10K1_IOCTL_PCM_PEEK _IOWR('H', 0x31, struct snd_emu10k1_fx8010_pcm_rec)
#define SNDRV_EMU10K1_IOCTL_PVERSION _IOR ('H', 0x40, int)
#define SNDRV_EMU10K1_IOCTL_STOP _IO ('H', 0x80)
#define SNDRV_EMU10K1_IOCTL_CONTINUE _IO ('H', 0x81)
#define SNDRV_EMU10K1_IOCTL_ZERO_TRAM_COUNTER _IO ('H', 0x82)
#define SNDRV_EMU10K1_IOCTL_SINGLE_STEP _IOW ('H', 0x83, int)
#define SNDRV_EMU10K1_IOCTL_DBG_READ _IOR ('H', 0x84, int)
/* typedefs for compatibility to user-space */
typedef struct snd_emu10k1_fx8010_info emu10k1_fx8010_info_t;
typedef struct snd_emu10k1_fx8010_control_gpr emu10k1_fx8010_control_gpr_t;
typedef struct snd_emu10k1_fx8010_code emu10k1_fx8010_code_t;
typedef struct snd_emu10k1_fx8010_tram emu10k1_fx8010_tram_t;
typedef struct snd_emu10k1_fx8010_pcm_rec emu10k1_fx8010_pcm_t;
#endif /* _UAPI__SOUND_EMU10K1_H */
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#ifndef __SOUND_HDSP_H
#define __SOUND_HDSP_H
/*
* Copyright (C) 2003 Thomas Charbonnel (thomas@undata.org)
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#include <linux/types.h>
#define HDSP_MATRIX_MIXER_SIZE 2048
enum HDSP_IO_Type {
Digiface,
Multiface,
H9652,
H9632,
RPM,
Undefined,
};
struct hdsp_peak_rms {
__u32 input_peaks[26];
__u32 playback_peaks[26];
__u32 output_peaks[28];
__u64 input_rms[26];
__u64 playback_rms[26];
/* These are only used for H96xx cards */
__u64 output_rms[26];
};
#define SNDRV_HDSP_IOCTL_GET_PEAK_RMS _IOR('H', 0x40, struct hdsp_peak_rms)
struct hdsp_config_info {
unsigned char pref_sync_ref;
unsigned char wordclock_sync_check;
unsigned char spdif_sync_check;
unsigned char adatsync_sync_check;
unsigned char adat_sync_check[3];
unsigned char spdif_in;
unsigned char spdif_out;
unsigned char spdif_professional;
unsigned char spdif_emphasis;
unsigned char spdif_nonaudio;
unsigned int spdif_sample_rate;
unsigned int system_sample_rate;
unsigned int autosync_sample_rate;
unsigned char system_clock_mode;
unsigned char clock_source;
unsigned char autosync_ref;
unsigned char line_out;
unsigned char passthru;
unsigned char da_gain;
unsigned char ad_gain;
unsigned char phone_gain;
unsigned char xlr_breakout_cable;
unsigned char analog_extension_board;
};
#define SNDRV_HDSP_IOCTL_GET_CONFIG_INFO _IOR('H', 0x41, struct hdsp_config_info)
struct hdsp_firmware {
void __user *firmware_data; /* 24413 x 4 bytes */
};
#define SNDRV_HDSP_IOCTL_UPLOAD_FIRMWARE _IOW('H', 0x42, struct hdsp_firmware)
struct hdsp_version {
enum HDSP_IO_Type io_type;
unsigned short firmware_rev;
};
#define SNDRV_HDSP_IOCTL_GET_VERSION _IOR('H', 0x43, struct hdsp_version)
struct hdsp_mixer {
unsigned short matrix[HDSP_MATRIX_MIXER_SIZE];
};
#define SNDRV_HDSP_IOCTL_GET_MIXER _IOR('H', 0x44, struct hdsp_mixer)
struct hdsp_9632_aeb {
int aebi;
int aebo;
};
#define SNDRV_HDSP_IOCTL_GET_9632_AEB _IOR('H', 0x45, struct hdsp_9632_aeb)
/* typedefs for compatibility to user-space */
typedef enum HDSP_IO_Type HDSP_IO_Type;
typedef struct hdsp_peak_rms hdsp_peak_rms_t;
typedef struct hdsp_config_info hdsp_config_info_t;
typedef struct hdsp_firmware hdsp_firmware_t;
typedef struct hdsp_version hdsp_version_t;
typedef struct hdsp_mixer hdsp_mixer_t;
typedef struct hdsp_9632_aeb hdsp_9632_aeb_t;
#endif /* __SOUND_HDSP_H */
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#ifndef __SOUND_HDSPM_H
#define __SOUND_HDSPM_H
/*
* Copyright (C) 2003 Winfried Ritsch (IEM)
* based on hdsp.h from Thomas Charbonnel (thomas@undata.org)
*
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
/* Maximum channels is 64 even on 56Mode you have 64playbacks to matrix */
#define HDSPM_MAX_CHANNELS 64
enum hdspm_io_type {
MADI,
MADIface,
AIO,
AES32,
RayDAT
};
enum hdspm_speed {
ss,
ds,
qs
};
/* -------------------- IOCTL Peak/RMS Meters -------------------- */
struct hdspm_peak_rms {
uint32_t input_peaks[64];
uint32_t playback_peaks[64];
uint32_t output_peaks[64];
uint64_t input_rms[64];
uint64_t playback_rms[64];
uint64_t output_rms[64];
uint8_t speed; /* enum {ss, ds, qs} */
int status2;
};
#define SNDRV_HDSPM_IOCTL_GET_PEAK_RMS \
_IOR('H', 0x42, struct hdspm_peak_rms)
/* ------------ CONFIG block IOCTL ---------------------- */
struct hdspm_config {
unsigned char pref_sync_ref;
unsigned char wordclock_sync_check;
unsigned char madi_sync_check;
unsigned int system_sample_rate;
unsigned int autosync_sample_rate;
unsigned char system_clock_mode;
unsigned char clock_source;
unsigned char autosync_ref;
unsigned char line_out;
unsigned int passthru;
unsigned int analog_out;
};
#define SNDRV_HDSPM_IOCTL_GET_CONFIG \
_IOR('H', 0x41, struct hdspm_config)
/**
* If there's a TCO (TimeCode Option) board installed,
* there are further options and status data available.
* The hdspm_ltc structure contains the current SMPTE
* timecode and some status information and can be
* obtained via SNDRV_HDSPM_IOCTL_GET_LTC or in the
* hdspm_status struct.
**/
enum hdspm_ltc_format {
format_invalid,
fps_24,
fps_25,
fps_2997,
fps_30
};
enum hdspm_ltc_frame {
frame_invalid,
drop_frame,
full_frame
};
enum hdspm_ltc_input_format {
ntsc,
pal,
no_video
};
struct hdspm_ltc {
unsigned int ltc;
enum hdspm_ltc_format format;
enum hdspm_ltc_frame frame;
enum hdspm_ltc_input_format input_format;
};
#define SNDRV_HDSPM_IOCTL_GET_LTC _IOR('H', 0x46, struct hdspm_mixer_ioctl)
/**
* The status data reflects the device's current state
* as determined by the card's configuration and
* connection status.
**/
enum hdspm_sync {
hdspm_sync_no_lock = 0,
hdspm_sync_lock = 1,
hdspm_sync_sync = 2
};
enum hdspm_madi_input {
hdspm_input_optical = 0,
hdspm_input_coax = 1
};
enum hdspm_madi_channel_format {
hdspm_format_ch_64 = 0,
hdspm_format_ch_56 = 1
};
enum hdspm_madi_frame_format {
hdspm_frame_48 = 0,
hdspm_frame_96 = 1
};
enum hdspm_syncsource {
syncsource_wc = 0,
syncsource_madi = 1,
syncsource_tco = 2,
syncsource_sync = 3,
syncsource_none = 4
};
struct hdspm_status {
uint8_t card_type; /* enum hdspm_io_type */
enum hdspm_syncsource autosync_source;
uint64_t card_clock;
uint32_t master_period;
union {
struct {
uint8_t sync_wc; /* enum hdspm_sync */
uint8_t sync_madi; /* enum hdspm_sync */
uint8_t sync_tco; /* enum hdspm_sync */
uint8_t sync_in; /* enum hdspm_sync */
uint8_t madi_input; /* enum hdspm_madi_input */
uint8_t channel_format; /* enum hdspm_madi_channel_format */
uint8_t frame_format; /* enum hdspm_madi_frame_format */
} madi;
} card_specific;
};
#define SNDRV_HDSPM_IOCTL_GET_STATUS \
_IOR('H', 0x47, struct hdspm_status)
/**
* Get information about the card and its add-ons.
**/
#define HDSPM_ADDON_TCO 1
struct hdspm_version {
uint8_t card_type; /* enum hdspm_io_type */
char cardname[20];
unsigned int serial;
unsigned short firmware_rev;
int addons;
};
#define SNDRV_HDSPM_IOCTL_GET_VERSION _IOR('H', 0x48, struct hdspm_version)
/* ------------- get Matrix Mixer IOCTL --------------- */
/* MADI mixer: 64inputs+64playback in 64outputs = 8192 => *4Byte =
* 32768 Bytes
*/
/* organisation is 64 channelfader in a continuous memory block */
/* equivalent to hardware definition, maybe for future feature of mmap of
* them
*/
/* each of 64 outputs has 64 infader and 64 outfader:
Ins to Outs mixer[out].in[in], Outstreams to Outs mixer[out].pb[pb] */
#define HDSPM_MIXER_CHANNELS HDSPM_MAX_CHANNELS
struct hdspm_channelfader {
unsigned int in[HDSPM_MIXER_CHANNELS];
unsigned int pb[HDSPM_MIXER_CHANNELS];
};
struct hdspm_mixer {
struct hdspm_channelfader ch[HDSPM_MIXER_CHANNELS];
};
struct hdspm_mixer_ioctl {
struct hdspm_mixer *mixer;
};
/* use indirect access due to the limit of ioctl bit size */
#define SNDRV_HDSPM_IOCTL_GET_MIXER _IOR('H', 0x44, struct hdspm_mixer_ioctl)
/* typedefs for compatibility to user-space */
typedef struct hdspm_peak_rms hdspm_peak_rms_t;
typedef struct hdspm_config_info hdspm_config_info_t;
typedef struct hdspm_version hdspm_version_t;
typedef struct hdspm_channelfader snd_hdspm_channelfader_t;
typedef struct hdspm_mixer hdspm_mixer_t;
#endif
+122
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/*
* Copyright (c) 1999 by Uros Bizjak <uros@kss-loka.si>
* Takashi Iwai <tiwai@suse.de>
*
* SB16ASP/AWE32 CSP control
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
*/
#ifndef _UAPI__SOUND_SB16_CSP_H
#define _UAPI__SOUND_SB16_CSP_H
/* CSP modes */
#define SNDRV_SB_CSP_MODE_NONE 0x00
#define SNDRV_SB_CSP_MODE_DSP_READ 0x01 /* Record from DSP */
#define SNDRV_SB_CSP_MODE_DSP_WRITE 0x02 /* Play to DSP */
#define SNDRV_SB_CSP_MODE_QSOUND 0x04 /* QSound */
/* CSP load flags */
#define SNDRV_SB_CSP_LOAD_FROMUSER 0x01
#define SNDRV_SB_CSP_LOAD_INITBLOCK 0x02
/* CSP sample width */
#define SNDRV_SB_CSP_SAMPLE_8BIT 0x01
#define SNDRV_SB_CSP_SAMPLE_16BIT 0x02
/* CSP channels */
#define SNDRV_SB_CSP_MONO 0x01
#define SNDRV_SB_CSP_STEREO 0x02
/* CSP rates */
#define SNDRV_SB_CSP_RATE_8000 0x01
#define SNDRV_SB_CSP_RATE_11025 0x02
#define SNDRV_SB_CSP_RATE_22050 0x04
#define SNDRV_SB_CSP_RATE_44100 0x08
#define SNDRV_SB_CSP_RATE_ALL 0x0f
/* CSP running state */
#define SNDRV_SB_CSP_ST_IDLE 0x00
#define SNDRV_SB_CSP_ST_LOADED 0x01
#define SNDRV_SB_CSP_ST_RUNNING 0x02
#define SNDRV_SB_CSP_ST_PAUSED 0x04
#define SNDRV_SB_CSP_ST_AUTO 0x08
#define SNDRV_SB_CSP_ST_QSOUND 0x10
/* maximum QSound value (180 degrees right) */
#define SNDRV_SB_CSP_QSOUND_MAX_RIGHT 0x20
/* maximum microcode RIFF file size */
#define SNDRV_SB_CSP_MAX_MICROCODE_FILE_SIZE 0x3000
/* microcode header */
struct snd_sb_csp_mc_header {
char codec_name[16]; /* id name of codec */
unsigned short func_req; /* requested function */
};
/* microcode to be loaded */
struct snd_sb_csp_microcode {
struct snd_sb_csp_mc_header info;
unsigned char data[SNDRV_SB_CSP_MAX_MICROCODE_FILE_SIZE];
};
/* start CSP with sample_width in mono/stereo */
struct snd_sb_csp_start {
int sample_width; /* sample width, look above */
int channels; /* channels, look above */
};
/* CSP information */
struct snd_sb_csp_info {
char codec_name[16]; /* id name of codec */
unsigned short func_nr; /* function number */
unsigned int acc_format; /* accepted PCM formats */
unsigned short acc_channels; /* accepted channels */
unsigned short acc_width; /* accepted sample width */
unsigned short acc_rates; /* accepted sample rates */
unsigned short csp_mode; /* CSP mode, see above */
unsigned short run_channels; /* current channels */
unsigned short run_width; /* current sample width */
unsigned short version; /* version id: 0x10 - 0x1f */
unsigned short state; /* state bits */
};
/* HWDEP controls */
/* get CSP information */
#define SNDRV_SB_CSP_IOCTL_INFO _IOR('H', 0x10, struct snd_sb_csp_info)
/* load microcode to CSP */
/* NOTE: struct snd_sb_csp_microcode overflows the max size (13 bits)
* defined for some architectures like MIPS, and it leads to build errors.
* (x86 and co have 14-bit size, thus it's valid, though.)
* As a workaround for skipping the size-limit check, here we don't use the
* normal _IOW() macro but _IOC() with the manual argument.
*/
#define SNDRV_SB_CSP_IOCTL_LOAD_CODE \
_IOC(_IOC_WRITE, 'H', 0x11, sizeof(struct snd_sb_csp_microcode))
/* unload microcode from CSP */
#define SNDRV_SB_CSP_IOCTL_UNLOAD_CODE _IO('H', 0x12)
/* start CSP */
#define SNDRV_SB_CSP_IOCTL_START _IOW('H', 0x13, struct snd_sb_csp_start)
/* stop CSP */
#define SNDRV_SB_CSP_IOCTL_STOP _IO('H', 0x14)
/* pause CSP and DMA transfer */
#define SNDRV_SB_CSP_IOCTL_PAUSE _IO('H', 0x15)
/* restart CSP and DMA transfer */
#define SNDRV_SB_CSP_IOCTL_RESTART _IO('H', 0x16)
#endif /* _UAPI__SOUND_SB16_CSP_H */
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#ifndef __SOUND_SFNT_INFO_H
#define __SOUND_SFNT_INFO_H
/*
* Patch record compatible with AWE driver on OSS
*
* Copyright (C) 1999-2000 Takashi Iwai
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
*/
#include <sound/asound.h>
/*
* patch information record
*/
#ifdef SNDRV_BIG_ENDIAN
#define SNDRV_OSS_PATCHKEY(id) (0xfd00|id)
#else
#define SNDRV_OSS_PATCHKEY(id) ((id<<8)|0xfd)
#endif
/* patch interface header: 16 bytes */
struct soundfont_patch_info {
unsigned short key; /* use the key below */
#define SNDRV_OSS_SOUNDFONT_PATCH SNDRV_OSS_PATCHKEY(0x07)
short device_no; /* synthesizer number */
unsigned short sf_id; /* file id (should be zero) */
short optarg; /* optional argument */
int len; /* data length (without this header) */
short type; /* patch operation type */
#define SNDRV_SFNT_LOAD_INFO 0 /* awe_voice_rec */
#define SNDRV_SFNT_LOAD_DATA 1 /* awe_sample_info */
#define SNDRV_SFNT_OPEN_PATCH 2 /* awe_open_parm */
#define SNDRV_SFNT_CLOSE_PATCH 3 /* none */
/* 4 is obsolete */
#define SNDRV_SFNT_REPLACE_DATA 5 /* awe_sample_info (optarg=#channels)*/
#define SNDRV_SFNT_MAP_PRESET 6 /* awe_voice_map */
/* 7 is not used */
#define SNDRV_SFNT_PROBE_DATA 8 /* optarg=sample */
#define SNDRV_SFNT_REMOVE_INFO 9 /* optarg=(bank<<8)|instr */
short reserved; /* word alignment data */
/* the actual patch data begins after this */
};
/*
* open patch
*/
#define SNDRV_SFNT_PATCH_NAME_LEN 32
struct soundfont_open_parm {
unsigned short type; /* sample type */
#define SNDRV_SFNT_PAT_TYPE_MISC 0
#define SNDRV_SFNT_PAT_TYPE_GUS 6
#define SNDRV_SFNT_PAT_TYPE_MAP 7
#define SNDRV_SFNT_PAT_LOCKED 0x100 /* lock the samples */
#define SNDRV_SFNT_PAT_SHARED 0x200 /* sample is shared */
short reserved;
char name[SNDRV_SFNT_PATCH_NAME_LEN];
};
/*
* raw voice information record
*/
/* wave table envelope & effect parameters to control EMU8000 */
struct soundfont_voice_parm {
unsigned short moddelay; /* modulation delay (0x8000) */
unsigned short modatkhld; /* modulation attack & hold time (0x7f7f) */
unsigned short moddcysus; /* modulation decay & sustain (0x7f7f) */
unsigned short modrelease; /* modulation release time (0x807f) */
short modkeyhold, modkeydecay; /* envelope change per key (not used) */
unsigned short voldelay; /* volume delay (0x8000) */
unsigned short volatkhld; /* volume attack & hold time (0x7f7f) */
unsigned short voldcysus; /* volume decay & sustain (0x7f7f) */
unsigned short volrelease; /* volume release time (0x807f) */
short volkeyhold, volkeydecay; /* envelope change per key (not used) */
unsigned short lfo1delay; /* LFO1 delay (0x8000) */
unsigned short lfo2delay; /* LFO2 delay (0x8000) */
unsigned short pefe; /* modulation pitch & cutoff (0x0000) */
unsigned short fmmod; /* LFO1 pitch & cutoff (0x0000) */
unsigned short tremfrq; /* LFO1 volume & freq (0x0000) */
unsigned short fm2frq2; /* LFO2 pitch & freq (0x0000) */
unsigned char cutoff; /* initial cutoff (0xff) */
unsigned char filterQ; /* initial filter Q [0-15] (0x0) */
unsigned char chorus; /* chorus send (0x00) */
unsigned char reverb; /* reverb send (0x00) */
unsigned short reserved[4]; /* not used */
};
/* wave table parameters: 92 bytes */
struct soundfont_voice_info {
unsigned short sf_id; /* file id (should be zero) */
unsigned short sample; /* sample id */
int start, end; /* sample offset correction */
int loopstart, loopend; /* loop offset correction */
short rate_offset; /* sample rate pitch offset */
unsigned short mode; /* sample mode */
#define SNDRV_SFNT_MODE_ROMSOUND 0x8000
#define SNDRV_SFNT_MODE_STEREO 1
#define SNDRV_SFNT_MODE_LOOPING 2
#define SNDRV_SFNT_MODE_NORELEASE 4 /* obsolete */
#define SNDRV_SFNT_MODE_INIT_PARM 8
short root; /* midi root key */
short tune; /* pitch tuning (in cents) */
unsigned char low, high; /* key note range */
unsigned char vellow, velhigh; /* velocity range */
signed char fixkey, fixvel; /* fixed key, velocity */
signed char pan, fixpan; /* panning, fixed panning */
short exclusiveClass; /* exclusive class (0 = none) */
unsigned char amplitude; /* sample volume (127 max) */
unsigned char attenuation; /* attenuation (0.375dB) */
short scaleTuning; /* pitch scale tuning(%), normally 100 */
struct soundfont_voice_parm parm; /* voice envelope parameters */
unsigned short sample_mode; /* sample mode_flag (set by driver) */
};
/* instrument info header: 4 bytes */
struct soundfont_voice_rec_hdr {
unsigned char bank; /* midi bank number */
unsigned char instr; /* midi preset number */
char nvoices; /* number of voices */
char write_mode; /* write mode; normally 0 */
#define SNDRV_SFNT_WR_APPEND 0 /* append anyway */
#define SNDRV_SFNT_WR_EXCLUSIVE 1 /* skip if already exists */
#define SNDRV_SFNT_WR_REPLACE 2 /* replace if already exists */
};
/*
* sample wave information
*/
/* wave table sample header: 32 bytes */
struct soundfont_sample_info {
unsigned short sf_id; /* file id (should be zero) */
unsigned short sample; /* sample id */
int start, end; /* start & end offset */
int loopstart, loopend; /* loop start & end offset */
int size; /* size (0 = ROM) */
short dummy; /* not used */
unsigned short mode_flags; /* mode flags */
#define SNDRV_SFNT_SAMPLE_8BITS 1 /* wave data is 8bits */
#define SNDRV_SFNT_SAMPLE_UNSIGNED 2 /* wave data is unsigned */
#define SNDRV_SFNT_SAMPLE_NO_BLANK 4 /* no blank loop is attached */
#define SNDRV_SFNT_SAMPLE_SINGLESHOT 8 /* single-shot w/o loop */
#define SNDRV_SFNT_SAMPLE_BIDIR_LOOP 16 /* bidirectional looping */
#define SNDRV_SFNT_SAMPLE_STEREO_LEFT 32 /* stereo left sound */
#define SNDRV_SFNT_SAMPLE_STEREO_RIGHT 64 /* stereo right sound */
#define SNDRV_SFNT_SAMPLE_REVERSE_LOOP 128 /* reverse looping */
unsigned int truesize; /* used memory size (set by driver) */
};
/*
* voice preset mapping (aliasing)
*/
struct soundfont_voice_map {
int map_bank, map_instr, map_key; /* key = -1 means all keys */
int src_bank, src_instr, src_key;
};
/*
* ioctls for hwdep
*/
#define SNDRV_EMUX_HWDEP_NAME "Emux WaveTable"
#define SNDRV_EMUX_VERSION ((1 << 16) | (0 << 8) | 0) /* 1.0.0 */
struct snd_emux_misc_mode {
int port; /* -1 = all */
int mode;
int value;
int value2; /* reserved */
};
#define SNDRV_EMUX_IOCTL_VERSION _IOR('H', 0x80, unsigned int)
#define SNDRV_EMUX_IOCTL_LOAD_PATCH _IOWR('H', 0x81, struct soundfont_patch_info)
#define SNDRV_EMUX_IOCTL_RESET_SAMPLES _IO('H', 0x82)
#define SNDRV_EMUX_IOCTL_REMOVE_LAST_SAMPLES _IO('H', 0x83)
#define SNDRV_EMUX_IOCTL_MEM_AVAIL _IOW('H', 0x84, int)
#define SNDRV_EMUX_IOCTL_MISC_MODE _IOWR('H', 0x84, struct snd_emux_misc_mode)
#endif /* __SOUND_SFNT_INFO_H */