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authorLinus Torvalds <torvalds@linux-foundation.org>2012-12-13 11:51:23 -0800
committerLinus Torvalds <torvalds@linux-foundation.org>2012-12-13 11:51:23 -0800
commit046e7d685bc370fd4c879ab6635ad3f69e6673d1 (patch)
tree36b981f8d1f2bfd348c1479acbe3a9426d35c377 /include/uapi/sound/sb16_csp.h
parentfe504c5c745aeb767d978fbedeb94775fd4cb69c (diff)
parent6eb827d23577a4efec2b10a9c4cc9ded268a1d1c (diff)
downloadblackbird-op-linux-046e7d685bc370fd4c879ab6635ad3f69e6673d1.tar.gz
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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.h122
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 */
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