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