Merge branch 'x86-nuke-platforms-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/tip

Pull x86 old platform removal from Peter Anvin:
 "This patchset removes support for several completely obsolete
  platforms, where the maintainers either have completely vanished or
  acked the removal.  For some of them it is questionable if there even
  exists functional specimens of the hardware"

Geert Uytterhoeven apparently thought this was a April Fool's pull request ;)

* 'x86-nuke-platforms-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/tip:
  x86, platforms: Remove NUMAQ
  x86, platforms: Remove SGI Visual Workstation
  x86, apic: Remove support for IBM Summit/EXA chipset
  x86, apic: Remove support for ia32-based Unisys ES7000

3 files changed, 0 insertions, 308 deletions

diff --git a/Documentation/00-INDEX b/Documentation/00-INDEX
index 07de7e19b4ce..27e67a98b7be 100644
--- a/Documentation/00-INDEX
+++ b/Documentation/00-INDEX
@@ -413,8 +413,6 @@ serial-console.txt
 	- how to set up Linux with a serial line console as the default.
 sgi-ioc4.txt
 	- description of the SGI IOC4 PCI (multi function) device.
-sgi-visws.txt
-	- short blurb on the SGI Visual Workstations.
 sh/
 	- directory with info on porting Linux to a new architecture.
 smsc_ece1099.txt
diff --git a/Documentation/sgi-visws.txt b/Documentation/sgi-visws.txt
deleted file mode 100644
index 7ff0811ca2ba..000000000000
--- a/Documentation/sgi-visws.txt
+++ /dev/null
@@ -1,13 +0,0 @@
-
-The SGI Visual Workstations (models 320 and 540) are based around
-the Cobalt, Lithium, and Arsenic ASICs.  The Cobalt ASIC is the
-main system ASIC which interfaces the 1-4 IA32 cpus, the memory
-system, and the I/O system in the Lithium ASIC.  The Cobalt ASIC
-also contains the 3D gfx rendering engine which renders to main
-system memory -- part of which is used as the frame buffer which
-is DMA'ed to a video connector using the Arsenic ASIC.  A PIIX4
-chip and NS87307 are used to provide legacy device support (IDE,
-serial, floppy, and parallel).
-
-The Visual Workstation chipset largely conforms to the PC architecture
-with some notable exceptions such as interrupt handling.
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);
-	}
-