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author | Linus Torvalds <torvalds@linux-foundation.org> | 2012-12-13 11:51:23 -0800 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2012-12-13 11:51:23 -0800 |
commit | 046e7d685bc370fd4c879ab6635ad3f69e6673d1 (patch) | |
tree | 36b981f8d1f2bfd348c1479acbe3a9426d35c377 /include/uapi/sound/sb16_csp.h | |
parent | fe504c5c745aeb767d978fbedeb94775fd4cb69c (diff) | |
parent | 6eb827d23577a4efec2b10a9c4cc9ded268a1d1c (diff) | |
download | blackbird-op-linux-046e7d685bc370fd4c879ab6635ad3f69e6673d1.tar.gz blackbird-op-linux-046e7d685bc370fd4c879ab6635ad3f69e6673d1.zip |
Merge tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This update contains a fairly wide range of changes all over in sound
subdirectory, mainly because of UAPI header moves by David and __dev*
annotation removals by Bill. Other highlights are:
- Introduced the support for wallclock timestamps in ALSA PCM core
- Add the poll loop implementation for HD-audio jack detection
- Yet more VGA-switcheroo fixes for HD-audio
- New VIA HD-audio codec support
- More fixes on resource management in USB audio and MIDI drivers
- More quirks for USB-audio ASUS Xonar U3, Reloop Play, Focusrite,
Roland VG-99, etc
- Add support for FastTrack C400 usb-audio
- Clean ups in many drivers regarding firmware loading
- Add PSC724 Ultiimate Edge support to ice1712
- A few hdspm driver updates
- New Stanton SCS.1d/1m FireWire driver
- Standardisation of the logging in ASoC codes
- DT and dmaengine support for ASoC Atmel
- Support for Wolfson ADSP cores
- New drivers for Freescale/iVeia P1022 and Maxim MAX98090
- Lots of other ASoC driver fixes and developments"
Fix up trivial conflicts. And go out on a limb and assume the dts file
'status' field of one of the conflicting things was supposed to be
"disabled", not "disable" like in pretty much all other cases.
* tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (341 commits)
ALSA: hda - Move runtime PM check to runtime_idle callback
ALSA: hda - Add stereo-dmic fixup for Acer Aspire One 522
ALSA: hda - Avoid doubly suspend after vga switcheroo
ALSA: usb-audio: Enable S/PDIF on the ASUS Xonar U3
ALSA: hda - Check validity of CORB/RIRB WP reads
ALSA: hda - use usleep_range in link reset and change timeout check
ALSA: HDA: VIA: Add support for codec VT1808.
ALSA: HDA: VIA Add support for codec VT1705CF.
ASoC: codecs: remove __dev* attributes
ASoC: utils: remove __dev* attributes
ASoC: ux500: remove __dev* attributes
ASoC: txx9: remove __dev* attributes
ASoC: tegra: remove __dev* attributes
ASoC: spear: remove __dev* attributes
ASoC: sh: remove __dev* attributes
ASoC: s6000: remove __dev* attributes
ASoC: OMAP: remove __dev* attributes
ASoC: nuc900: remove __dev* attributes
ASoC: mxs: remove __dev* attributes
ASoC: kirkwood: remove __dev* attributes
...
Diffstat (limited to 'include/uapi/sound/sb16_csp.h')
-rw-r--r-- | include/uapi/sound/sb16_csp.h | 122 |
1 files changed, 122 insertions, 0 deletions
diff --git a/include/uapi/sound/sb16_csp.h b/include/uapi/sound/sb16_csp.h new file mode 100644 index 000000000000..3b96907e2afb --- /dev/null +++ b/include/uapi/sound/sb16_csp.h @@ -0,0 +1,122 @@ +/* + * 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 */ |