Linux-2.6.12-rc2v2.6.12-rc2

Initial git repository build. I'm not bothering with the full history,
even though we have it. We can create a separate "historical" git
archive of that later if we want to, and in the meantime it's about
3.2GB when imported into git - space that would just make the early
git days unnecessarily complicated, when we don't have a lot of good
infrastructure for it.

Let it rip!

1 files changed, 293 insertions, 0 deletions

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+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 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);
+	}
+