x86, platforms: Remove SGI Visual Workstation

The SGI Visual Workstation seems to be dead; remove support so we
don't have to continue maintaining it.

Cc: Andrey Panin <pazke@donpac.ru>
Cc: Michael Reed <mdr@sgi.com>
Link: http://lkml.kernel.org/r/530CFD6C.7040705@zytor.com
Signed-off-by: H. Peter Anvin <hpa@linux.intel.com>

1 files changed, 0 insertions, 293 deletions

diff --git a/Documentation/sound/oss/vwsnd b/Documentation/sound/oss/vwsnd
deleted file mode 100644
index 4c6cbdb3c548..000000000000
--- a/Documentation/sound/oss/vwsnd
+++ /dev/null
@@ -1,293 +0,0 @@
-vwsnd - Sound driver for the Silicon Graphics 320 and 540 Visual
-Workstations' onboard audio.
-
-Copyright 1999 Silicon Graphics, Inc.  All rights reserved.
-
-
-At the time of this writing, March 1999, there are two models of
-Visual Workstation, the 320 and the 540.  This document only describes
-those models.  Future Visual Workstation models may have different
-sound capabilities, and this driver will probably not work on those
-boxes.
-
-The Visual Workstation has an Analog Devices AD1843 "SoundComm" audio
-codec chip.  The AD1843 is accessed through the Cobalt I/O ASIC, also
-known as Lithium.  This driver programs both chips.
-
-==============================================================================
-QUICK CONFIGURATION
-
-	# insmod soundcore
-	# insmod vwsnd
-
-==============================================================================
-I/O CONNECTIONS
-
-On the Visual Workstation, only three of the AD1843 inputs are hooked
-up.  The analog line in jacks are connected to the AD1843's AUX1
-input.  The CD audio lines are connected to the AD1843's AUX2 input.
-The microphone jack is connected to the AD1843's MIC input.  The mic
-jack is mono, but the signal is delivered to both the left and right
-MIC inputs.  You can record in stereo from the mic input, but you will
-get the same signal on both channels (within the limits of A/D
-accuracy).  Full scale on the Line input is +/- 2.0 V.  Full scale on
-the MIC input is 20 dB less, or +/- 0.2 V.
-
-The AD1843's LOUT1 outputs are connected to the Line Out jacks.  The
-AD1843's HPOUT outputs are connected to the speaker/headphone jack.
-LOUT2 is not connected.  Line out's maximum level is +/- 2.0 V peak to
-peak.  The speaker/headphone out's maximum is +/- 4.0 V peak to peak.
-
-The AD1843's PCM input channel and one of its output channels (DAC1)
-are connected to Lithium.  The other output channel (DAC2) is not
-connected.
-
-==============================================================================
-CAPABILITIES
-
-The AD1843 has PCM input and output (Pulse Code Modulation, also known
-as wavetable).  PCM input and output can be mono or stereo in any of
-four formats.  The formats are 16 bit signed and 8 bit unsigned,
-u-Law, and A-Law format.  Any sample rate from 4 KHz to 49 KHz is
-available, in 1 Hz increments.
-
-The AD1843 includes an analog mixer that can mix all three input
-signals (line, mic and CD) into the analog outputs.  The mixer has a
-separate gain control and mute switch for each input.
-
-There are two outputs, line out and speaker/headphone out.  They
-always produce the same signal, and the speaker always has 3 dB more
-gain than the line out.  The speaker/headphone output can be muted,
-but this driver does not export that function.
-
-The hardware can sync audio to the video clock, but this driver does
-not have a way to specify syncing to video.
-
-==============================================================================
-PROGRAMMING
-
-This section explains the API supported by the driver.  Also see the
-Open Sound Programming Guide at http://www.opensound.com/pguide/ .
-This section assumes familiarity with that document.
-
-The driver has two interfaces, an I/O interface and a mixer interface.
-There is no MIDI or sequencer capability.
-
-==============================================================================
-PROGRAMMING PCM I/O
-
-The I/O interface is usually accessed as /dev/audio or /dev/dsp.
-Using the standard Open Sound System (OSS) ioctl calls, the sample
-rate, number of channels, and sample format may be set within the
-limitations described above.  The driver supports triggering.  It also
-supports getting the input and output pointers with one-sample
-accuracy.
-
-The SNDCTL_DSP_GETCAP ioctl returns these capabilities.
-
-	DSP_CAP_DUPLEX - driver supports full duplex.
-
-	DSP_CAP_TRIGGER - driver supports triggering.
-
-	DSP_CAP_REALTIME - values returned by SNDCTL_DSP_GETIPTR
-	and SNDCTL_DSP_GETOPTR are accurate to a few samples.
-
-Memory mapping (mmap) is not implemented.
-
-The driver permits subdivided fragment sizes from 64 to 4096 bytes.
-The number of fragments can be anything from 3 fragments to however
-many fragments fit into 124 kilobytes.  It is up to the user to
-determine how few/small fragments can be used without introducing
-glitches with a given workload.  Linux is not realtime, so we can't
-promise anything.  (sigh...)
-
-When this driver is switched into or out of mu-Law or A-Law mode on
-output, it may produce an audible click.  This is unavoidable.  To
-prevent clicking, use signed 16-bit mode instead, and convert from
-mu-Law or A-Law format in software.
-
-==============================================================================
-PROGRAMMING THE MIXER INTERFACE
-
-The mixer interface is usually accessed as /dev/mixer.  It is accessed
-through ioctls.  The mixer allows the application to control gain or
-mute several audio signal paths, and also allows selection of the
-recording source.
-
-Each of the constants described here can be read using the
-MIXER_READ(SOUND_MIXER_xxx) ioctl.  Those that are not read-only can
-also be written using the MIXER_WRITE(SOUND_MIXER_xxx) ioctl.  In most
-cases, <sys/soundcard.h> defines constants SOUND_MIXER_READ_xxx and
-SOUND_MIXER_WRITE_xxx which work just as well.
-
-SOUND_MIXER_CAPS	Read-only
-
-This is a mask of optional driver capabilities that are implemented.
-This driver's only capability is SOUND_CAP_EXCL_INPUT, which means
-that only one recording source can be active at a time.
-
-SOUND_MIXER_DEVMASK	Read-only
-
-This is a mask of the sound channels.  This driver's channels are PCM,
-LINE, MIC, CD, and RECLEV.
-
-SOUND_MIXER_STEREODEVS	Read-only
-
-This is a mask of which sound channels are capable of stereo.  All
-channels are capable of stereo.  (But see caveat on MIC input in I/O
-CONNECTIONS section above).
-
-SOUND_MIXER_OUTMASK	Read-only
-
-This is a mask of channels that route inputs through to outputs.
-Those are LINE, MIC, and CD.
-
-SOUND_MIXER_RECMASK	Read-only
-
-This is a mask of channels that can be recording sources.  Those are
-PCM, LINE, MIC, CD.
-
-SOUND_MIXER_PCM		Default: 0x5757 (0 dB)
-
-This is the gain control for PCM output.  The left and right channel
-gain are controlled independently.  This gain control has 64 levels,
-which range from -82.5 dB to +12.0 dB in 1.5 dB steps.  Those 64
-levels are mapped onto 100 levels at the ioctl, see below.
-
-SOUND_MIXER_LINE	Default: 0x4a4a (0 dB)
-
-This is the gain control for mixing the Line In source into the
-outputs.  The left and right channel gain are controlled
-independently.  This gain control has 32 levels, which range from
--34.5 dB to +12.0 dB in 1.5 dB steps.  Those 32 levels are mapped onto
-100 levels at the ioctl, see below.
-
-SOUND_MIXER_MIC		Default: 0x4a4a (0 dB)
-
-This is the gain control for mixing the MIC source into the outputs.
-The left and right channel gain are controlled independently.  This
-gain control has 32 levels, which range from -34.5 dB to +12.0 dB in
-1.5 dB steps.  Those 32 levels are mapped onto 100 levels at the
-ioctl, see below.
-
-SOUND_MIXER_CD		Default: 0x4a4a (0 dB)
-
-This is the gain control for mixing the CD audio source into the
-outputs.  The left and right channel gain are controlled
-independently.  This gain control has 32 levels, which range from
--34.5 dB to +12.0 dB in 1.5 dB steps.  Those 32 levels are mapped onto
-100 levels at the ioctl, see below.
-
-SOUND_MIXER_RECLEV	 Default: 0 (0 dB)
-
-This is the gain control for PCM input (RECording LEVel).  The left
-and right channel gain are controlled independently.  This gain
-control has 16 levels, which range from 0 dB to +22.5 dB in 1.5 dB
-steps.  Those 16 levels are mapped onto 100 levels at the ioctl, see
-below.
-
-SOUND_MIXER_RECSRC	 Default: SOUND_MASK_LINE
-
-This is a mask of currently selected PCM input sources (RECording
-SouRCes).  Because the AD1843 can only have a single recording source
-at a time, only one bit at a time can be set in this mask.  The
-allowable values are SOUND_MASK_PCM, SOUND_MASK_LINE, SOUND_MASK_MIC,
-or SOUND_MASK_CD.  Selecting SOUND_MASK_PCM sets up internal
-resampling which is useful for loopback testing and for hardware
-sample rate conversion.  But software sample rate conversion is
-probably faster, so I don't know how useful that is.
-
-SOUND_MIXER_OUTSRC	DEFAULT: SOUND_MASK_LINE|SOUND_MASK_MIC|SOUND_MASK_CD
-
-This is a mask of sources that are currently passed through to the
-outputs.  Those sources whose bits are not set are muted.
-
-==============================================================================
-GAIN CONTROL
-
-There are five gain controls listed above.  Each has 16, 32, or 64
-steps.  Each control has 1.5 dB of gain per step.  Each control is
-stereo.
-
-The OSS defines the argument to a channel gain ioctl as having two
-components, left and right, each of which ranges from 0 to 100.  The
-two components are packed into the same word, with the left side gain
-in the least significant byte, and the right side gain in the second
-least significant byte.  In C, we would say this.
-
-	#include <assert.h>
-
-	...
-
-	 	assert(leftgain >= 0 && leftgain <= 100);
-		assert(rightgain >= 0 && rightgain <= 100);
-		arg = leftgain | rightgain << 8;
-
-So each OSS gain control has 101 steps.  But the hardware has 16, 32,
-or 64 steps.  The hardware steps are spread across the 101 OSS steps
-nearly evenly.  The conversion formulas are like this, given N equals
-16, 32, or 64.
-
-	int round = N/2 - 1;
-	OSS_gain_steps = (hw_gain_steps * 100 + round) / (N - 1);
-	hw_gain_steps = (OSS_gain_steps * (N - 1) + round) / 100;
-
-Here is a snippet of C code that will return the left and right gain
-of any channel in dB.  Pass it one of the predefined gain_desc_t
-structures to access any of the five channels' gains.
-
-	typedef struct gain_desc {
-		float min_gain;
-		float gain_step;
-		int nbits;
-		int chan;
-	} gain_desc_t;
-
-	const gain_desc_t gain_pcm    = { -82.5, 1.5, 6, SOUND_MIXER_PCM    };
-	const gain_desc_t gain_line   = { -34.5, 1.5, 5, SOUND_MIXER_LINE   };
-	const gain_desc_t gain_mic    = { -34.5, 1.5, 5, SOUND_MIXER_MIC    };
-	const gain_desc_t gain_cd     = { -34.5, 1.5, 5, SOUND_MIXER_CD     };
-	const gain_desc_t gain_reclev = {   0.0, 1.5, 4, SOUND_MIXER_RECLEV };
-
-	int get_gain_dB(int fd, const gain_desc_t *gp,
-			float *left, float *right)
-	{
-		int word;
-		int lg, rg;
-		int mask = (1 << gp->nbits) - 1;
-
-		if (ioctl(fd, MIXER_READ(gp->chan), &word) != 0)
-			return -1;	/* fail */
-		lg = word & 0xFF;
-		rg = word >> 8 & 0xFF;
-		lg = (lg * mask + mask / 2) / 100;
-		rg = (rg * mask + mask / 2) / 100;
-		*left = gp->min_gain + gp->gain_step * lg;
-		*right = gp->min_gain + gp->gain_step * rg;
-		return 0;
-	}	
-
-And here is the corresponding routine to set a channel's gain in dB.
-
-	int set_gain_dB(int fd, const gain_desc_t *gp, float left, float right)
-	{
-		float max_gain =
-			gp->min_gain + (1 << gp->nbits) * gp->gain_step;
-		float round = gp->gain_step / 2;
-		int mask = (1 << gp->nbits) - 1;
-		int word;
-		int lg, rg;
-
-		if (left < gp->min_gain || right < gp->min_gain)
-			return EINVAL;
-		lg = (left - gp->min_gain + round) / gp->gain_step;
-		rg = (right - gp->min_gain + round) / gp->gain_step;
-		if (lg >= (1 << gp->nbits) || rg >= (1 << gp->nbits))
-			return EINVAL;
-		lg = (100 * lg + mask / 2) / mask;
-		rg = (100 * rg + mask / 2) / mask;
-		word = lg | rg << 8;
-
-		return ioctl(fd, MIXER_WRITE(gp->chan), &word);
-	}
-