vwsnd - Sound driver for the Silicon Graphics 320 and 540 VisualWorkstations' onboard audio.Copyright 1999 Silicon Graphics, Inc.  All rights reserved.At the time of this writing, March 1999, there are two models ofVisual Workstation, the 320 and the 540.  This document only describesthose models.  Future Visual Workstation models may have differentsound capabilities, and this driver will probably not work on thoseboxes.The Visual Workstation has an Analog Devices AD1843 "SoundComm" audiocodec chip.  The AD1843 is accessed through the Cobalt I/O ASIC, alsoknown as Lithium.  This driver programs both both chips.==============================================================================QUICK CONFIGURATION	# insmod soundcore	# insmod vwsnd==============================================================================I/O CONNECTIONSOn the Visual Workstation, only three of the AD1843 inputs are hookedup.  The analog line in jacks are connected to the AD1843's AUX1input.  The CD audio lines are connected to the AD1843's AUX2 input.The microphone jack is connected to the AD1843's MIC input.  The micjack is mono, but the signal is delivered to both the left and rightMIC inputs.  You can record in stereo from the mic input, but you willget the same signal on both channels (within the limits of A/Daccuracy).  Full scale on the Line input is +/- 2.0 V.  Full scale onthe MIC input is 20 dB less, or +/- 0.2 V.The AD1843's LOUT1 outputs are connected to the Line Out jacks.  TheAD1843's HPOUT outputs are connected to the speaker/headphone jack.LOUT2 is not connected.  Line out's maximum level is +/- 2.0 V peak topeak.  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 notconnected.==============================================================================CAPABILITIESThe AD1843 has PCM input and output (Pulse Code Modulation, also knownas wavetable).  PCM input and output can be mono or stereo in any offour 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 isavailable, in 1 Hz increments.The AD1843 includes an analog mixer that can mix all three inputsignals (line, mic and CD) into the analog outputs.  The mixer has aseparate gain control and mute switch for each input.There are two outputs, line out and speaker/headphone out.  Theyalways produce the same signal, and the speaker always has 3 dB moregain 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 doesnot have a way to specify syncing to video.==============================================================================PROGRAMMINGThis section explains the API supported by the driver.  Also see theOpen 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/OThe I/O interface is usually accessed as /dev/audio or /dev/dsp.Using the standard Open Sound System (OSS) ioctl calls, the samplerate, number of channels, and sample format may be set within thelimitations described above.  The driver supports triggering.  It alsosupports getting the input and output pointers with one-sampleaccuracy.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 howevermany fragments fit into 124 kilobytes.  It is up to the user todetermine how few/small fragments can be used without introducingglitches with a given workload.  Linux is not realtime, so we can'tpromise anything.  (sigh...)When this driver is switched into or out of mu-Law or A-Law mode onoutput, it may produce an audible click.  This is unavoidable.  Toprevent clicking, use signed 16-bit mode instead, and convert frommu-Law or A-Law format in software.==============================================================================PROGRAMMING THE MIXER INTERFACEThe mixer interface is usually accessed as /dev/mixer.  It is accessedthrough ioctls.  The mixer allows the application to control gain ormute several audio signal paths, and also allows selection of therecording source.Each of the constants described here can be read using theMIXER_READ(SOUND_MIXER_xxx) ioctl.  Those that are not read-only canalso be written using the MIXER_WRITE(SOUND_MIXER_xxx) ioctl.  In mostcases, <sys/soundcard.h> defines constants SOUND_MIXER_READ_xxx andSOUND_MIXER_WRITE_xxx which work just as well.SOUND_MIXER_CAPS	Read-onlyThis is a mask of optional driver capabilities that are implemented.This driver's only capability is SOUND_CAP_EXCL_INPUT, which meansthat only one recording source can be active at a time.SOUND_MIXER_DEVMASK	Read-onlyThis is a mask of the sound channels.  This driver's channels are PCM,LINE, MIC, CD, and RECLEV.SOUND_MIXER_STEREODEVS	Read-onlyThis is a mask of which sound channels are capable of stereo.  Allchannels are capable of stereo.  (But see caveat on MIC input in I/OCONNECTIONS section above).SOUND_MIXER_OUTMASK	Read-onlyThis is a mask of channels that route inputs through to outputs.Those are LINE, MIC, and CD.SOUND_MIXER_RECMASK	Read-onlyThis is a mask of channels that can be recording sources.  Those arePCM, LINE, MIC, CD.SOUND_MIXER_PCM		Default: 0x5757 (0 dB)This is the gain control for PCM output.  The left and right channelgain 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 64levels 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 theoutputs.  The left and right channel gain are controlledindependently.  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 onto100 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.  Thisgain control has 32 levels, which range from -34.5 dB to +12.0 dB in1.5 dB steps.  Those 32 levels are mapped onto 100 levels at theioctl, see below.SOUND_MIXER_CD		Default: 0x4a4a (0 dB)This is the gain control for mixing the CD audio source into theoutputs.  The left and right channel gain are controlledindependently.  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 onto100 levels at the ioctl, see below.SOUND_MIXER_RECLEV	 Default: 0 (0 dB)This is the gain control for PCM input (RECording LEVel).  The leftand right channel gain are controlled independently.  This gaincontrol has 16 levels, which range from 0 dB to +22.5 dB in 1.5 dBsteps.  Those 16 levels are mapped onto 100 levels at the ioctl, seebelow.SOUND_MIXER_RECSRC	 Default: SOUND_MASK_LINEThis is a mask of currently selected PCM input sources (RECordingSouRCes).  Because the AD1843 can only have a single recording sourceat a time, only one bit at a time can be set in this mask.  Theallowable values are SOUND_MASK_PCM, SOUND_MASK_LINE, SOUND_MASK_MIC,or SOUND_MASK_CD.  Selecting SOUND_MASK_PCM sets up internalresampling which is useful for loopback testing and for hardwaresample rate conversion.  But software sample rate conversion isprobably faster, so I don't know how useful that is.SOUND_MIXER_OUTSRC	DEFAULT: SOUND_MASK_LINE|SOUND_MASK_MIC|SOUND_MASK_CDThis is a mask of sources that are currently passed through to theoutputs.  Those sources whose bits are not set are muted.==============================================================================GAIN CONTROLThere are five gain controls listed above.  Each has 16, 32, or 64steps.  Each control has 1.5 dB of gain per step.  Each control isstereo.The OSS defines the argument to a channel gain ioctl as having twocomponents, left and right, each of which ranges from 0 to 100.  Thetwo components are packed into the same word, with the left side gainin the least significant byte, and the right side gain in the secondleast 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 stepsnearly evenly.  The conversion formulas are like this, given N equals16, 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 gainof any channel in dB.  Pass it one of the predefined gain_desc_tstructures 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);	}