+
+
+Front/Rear Multi-channel Playback
+---------------------------------
+
+CM8x38 chip can use ADC as the second DAC so that two different stereo
+channels can be used for front/rear playbacks. Since there are two
+DACs, both streams are handled independently unlike the 4/6ch multi-
+channel playbacks in the section below.
+
+As default, ALSA driver assigns the first PCM device (i.e. hw:0,0 for
+card#0) for front and 4/6ch playbacks, while the second PCM device
+(hw:0,1) is assigned to the second DAC for rear playback.
+
+There are slight difference between two DACs.
+
+- The first DAC supports U8 and S16LE formats, while the second DAC
+ supports only S16LE.
+- The second DAC supports only two channel stereo.
+
+Please note that the CM8x38 DAC doesn't support continuous playback
+rate but only fixed rates: 5512, 8000, 11025, 16000, 22050, 32000,
+44100 and 48000 Hz.
+
+The rear output can be heard only when "Four Channel Mode" switch is
+disabled. Otherwise no signal will be routed to the rear speakers.
+As default it's turned on.
+
+*** WARNING ***
+When "Four Channel Mode" switch is off, the output from rear speakers
+will be FULL VOLUME regardless of Master and PCM volumes.
+This might damage your audio equipment. Please disconnect speakers
+before your turn off this switch.
+*** WARNING ***
+
+[ Well.. I once got the output with correct volume (i.e. same with the
+ front one) and was so excited. It was even with "Four Channel" bit
+ on and "double DAC" mode. Actually I could hear separate 4 channels
+ from front and rear speakers! But.. after reboot, all was gone.
+ It's a very pity that I didn't save the register dump at that
+ time.. Maybe there is an unknown register to achieve this... ]
+
+If your card has an extra output jack for the rear output, the rear
+playback should be routed there as default. If not, there is a
+control switch in the driver "Line-In As Rear", which you can change
+via alsamixer or somewhat else. When this switch is on, line-in jack
+is used as rear output.
+
+There are two more controls regarding to the rear output.
+The "Exchange DAC" switch is used to exchange front and rear playback
+routes, i.e. the 2nd DAC is output from front output.
+
+
+4/6 Multi-Channel Playback
+--------------------------
+
+The recent CM8738 chips support for the 4/6 multi-channel playback
+function. This is useful especially for AC3 decoding.
+
+When the multi-channel is supported, the driver name has a suffix
+"-MC" such like "CMI8738-MC6". You can check this name from
+/proc/asound/cards.
+
+When the 4/6-ch output is enabled, the front DAC accepts up to 6 (or
+4) channels. This is different from the dual DACs described in the
+previous section. While the dual DAC supports two different rates or
+formats, the 4/6-ch playback supports only the same condition for all
+channels.
+
+For using 4/6 channel playback, you need to specify the PCM channels
+as you like and set the format S16LE. For example, for playback with
+4 channels,
+
+ snd_pcm_hw_params_set_access(pcm, hw, SND_PCM_ACCESS_RW_INTERLEAVED);
+ // or mmap if you like
+ snd_pcm_hw_params_set_format(pcm, hw, SND_PCM_FORMAT_S16_LE);
+ snd_pcm_hw_params_set_channels(pcm, hw, 4);
+
+and use the interleaved 4 channel data.
+
+There is a control switch, "Line-In As Bass". As you can imagine from
+its name, the line-in jack is used for the bass (5th and 6th channels)
+output.
+
+
+Digital I/O
+-----------
+
+The CM8x38 provides the excellent SPDIF capability with very chip
+price (yes, that's the reason I bought the card :)
+
+The SPDIF playback and capture are done via the third PCM device
+(hw:0,2). Usually this is assigned to the PCM device "spdif".
+The available rates are 44100 and 48000 Hz.
+For playback with aplay, you can run like below:
+
+ % aplay -Dhw:0,2 foo.wav
+
+or
+
+ % aplay -Dspdif foo.wav
+
+So far, only S16LE format is supported. Still no 24bit. Sorry, not
+enough info for this.
+
+The playback and capture over SPDIF use normal DAC and ADC,
+respectively, so you cannot playback both analog and digital streams
+simultaneously.
+
+To enable SPDIF output, you need to turn on "IEC958 Output Switch"
+control via mixer or alsactl. Then you'll see the red light on from
+the card so you know that's working obviously :)
+The SPDIF input is always enabled, so you can hear SPDIF input data
+from line-out with "IEC958 In Monitor" switch at any time (see
+below).
+
+You can play via SPDIF even with the first device (hw:0,0),
+but SPDIF is enabled only when the proper format (S16LE), sample rate
+(441100 or 48000) and channels (2) are used. Otherwise it's turned
+off. (Also don't forget to turn on "IEC958 Output Switch", too.)
+
+
+Additionally there are relevant control switches:
+
+"IEC958 Mix Analog" - Mix analog PCM playback and FM-OPL/3 streams and
+ output through SPDIF. This switch appears only on old chip
+ models (CM8738 033 and 037).
+ Note: without this control you can output PCM to SPDIF.
+ This is "mixing" of streams, so e.g. it's not for AC3 output
+ (see the next section).
+
+"IEC958 In Select" - Select SPDIF input, the internal CD-in (false)
+ and the external input (true). This switch appears only on
+ the chip models 039 or later.
+
+"IEC958 Loop" - SPDIF input data is loop back into SPDIF
+ output (aka bypass)
+
+"IEC958 Copyright" - Set the copyright bit.
+
+"IEC958 5V" - Select 0.5V (coax) or 5V (optical) interface.
+ On some cards this doesn't work and you need to change the
+ configuration with hardware dip-switch.
+
+"IEC958 In Monitor" - SPDIF input is routed to DAC.
+
+"IEC958 In Phase Inverse" - Set SPDIF input format as inverse.
+ [FIXME: this doesn't work on all chips..]
+
+"IEC958 In Valid" - Set input validity flag detection.
+
+Note: When "PCM Playback Switch" is on, you'll hear the digital output
+stream through analog line-out.
+
+
+The AC3 (RAW DIGITAL) OUTPUT
+----------------------------
+
+The driver supports raw digital (typically AC3) i/o over SPDIF. This
+can be toggled via IEC958 playback control, but usually you need to
+access it via alsa-lib. See alsa-lib documents for more details.
+
+On the raw digital mode, the "PCM Playback Switch" is automatically
+turned off so that non-audio data is heard from the analog line-out.
+Similarly the following switches are off: "IEC958 Mix Analog" and
+"IEC958 Loop". The switches are resumed after closing the SPDIF PCM
+device automatically to the previous state.
+
+
+ANALOG MIXER INTERFACE
+----------------------
+
+The mixer interface on CM8x38 is similar to SB16.
+There are Master, PCM, Synth, CD, Line, Mic and PC Speaker playback
+volumes. Synth, CD, Line and Mic have playback and capture switches,
+too, as well as SB16.
+
+In addition to the standard SB mixer, CM8x38 provides more functions.
+- PCM playback switch
+- PCM capture switch (to capture the data sent to DAC)
+- Mic Boost switch
+- Mic capture volume
+- Aux playback volume/switch and capture switch
+- 3D control switch
+
+
+MIDI CONTROLLER
+---------------
+
+The MPU401-UART interface is enabled as default only for the first
+(CMIPCI) card. You need to set module option "snd_midi_port" properly
+for the 2nd (CMIPCI) card.
+
+There is _no_ hardware wavetable function on this chip (except for
+OPL3 synth below).
+What's said as MIDI synth on Windows is a software synthesizer
+emulation. On Linux use TiMidity or other softsynth program for
+playing MIDI music.
+
+
+FM OPL/3 Synth
+--------------
+
+The FM OPL/3 is also enabled as default only for the first card.
+Set "snd_fm_port" module option for more cards.
+
+The output quality of FM OPL/3 is, however, very weird.
+I don't know why..
+
+
+Joystick and Modem
+------------------
+
+The joystick and modem should be available by enabling the control
+switch "Joystick" and "Modem" respectively. But I myself have never
+tested them yet.
+
+
+Debugging Information
+---------------------
+
+The registers are shown in /proc/asound/cardX/cmipci. If you have any
+problem (especially unexpected behavior of mixer), please attach the
+output of this proc file together with the bug report.
diff -urN linux-2.5.6-pre3/Documentation/sound/alsa/SB-Live-mixer.txt linux-2.5.6/Documentation/sound/alsa/SB-Live-mixer.txt
--- linux-2.5.6-pre3/Documentation/sound/alsa/SB-Live-mixer.txt Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/alsa/SB-Live-mixer.txt Thu Mar 7 18:24:41 2002
@@ -0,0 +1,356 @@
+
+ Sound Blaster Live mixer / default DSP code
+ ===========================================
+
+
+The EMU10K1 chips have a DSP part which can be programmed to support
+various ways of sample processing, which is described here.
+(This acticle does not deal with the overall functionality of the
+EMU10K1 chips. See the manuals section for further details.)
+
+The ALSA driver programs this portion of chip by default code
+(can be altered later) which offers the following functionality:
+
+
+1) IEC958 (S/PDIF) raw PCM
+--------------------------
+
+This PCM device (it's the 4th PCM device (index 3!) and first subdevice
+(index 0) for a given card) allows to forward 48kHz, stereo, 16-bit
+little endian streams without any modifications to the digital output
+(coaxial or optical). The universal interface allows the creation of up
+to 8 raw PCM devices operating at 48kHz, 16-bit little endian. It would
+be easy to add support for multichannel devices to the current code,
+but the conversion routines exist only for stereo (2-channel streams)
+at the time.
+
+Look to tram_poke routines in lowlevel/emu10k1/emufx.c for more details.
+
+
+2) Digital mixer controls
+-------------------------
+
+These controls are built using the DSP instructions. They offer extended
+functionality. Only the default build-in code in the ALSA driver is described
+here. Note that the controls work as attenuators: the maximum value is the
+neutral position leaving the signal unchanged. Note that if the same destination
+is mentioned in multiple controls, the signal is accumulated and can be wrapped
+(set to maximal or minimal value without checking of overflow).
+
+
+Explanation of used abbreviations:
+
+DAC - digital to analog converter
+ADC - analog to digital converter
+I2S - one-way three wire serial bus for digital sound by Philips Semiconductors
+ (this standard is used for connecting standalone DAC and ADC converters)
+LFE - low frequency effects (subwoofer signal)
+AC97 - a chip containing an analog mixer, DAC and ADC converters
+IEC958 - S/PDIF
+FX-bus - the EMU10K1 chip has an effect bus containing 16 accumulators.
+ Each of the synthesizer voices can feed its output to these accumulators
+ and the DSP microcontroller can operate with the resulting sum.
+
+
+name='Wave Playback Volume',index=0
+
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples.
+The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
+
+name='Wave Surround Playback Volume',index=0
+
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples.
+The result samples are forwarded to the rear I2S DACs. These DACs operates
+separately (they are not inside the AC97 codec).
+
+name='Wave Center Playback Volume',index=0
+
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples.
+The result is mixed to mono signal (single channel) and forwarded to
+the ??rear?? right DAC PCM slot of the AC97 codec.
+
+name='Wave LFE Playback Volume',index=0
+
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM.
+The result is mixed to mono signal (single channel) and forwarded to
+the ??rear?? left DAC PCM slot of the AC97 codec.
+
+name='Wave Capture Volume',index=0
+name='Wave Capture Switch',index=0
+
+These controls are used to attenuate samples for left and right PCM FX-bus
+accumulator. ALSA uses accumulators 0 and 1 for left and right PCM.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Music Playback Volume',index=0
+
+This control is used to attenuate samples for left and right MIDI FX-bus
+accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples.
+The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
+
+name='Music Capture Volume',index=0
+name='Music Capture Switch',index=0
+
+These controls are used to attenuate samples for left and right MIDI FX-bus
+accumulator. ALSA uses accumulators 4 and 5 for left and right PCM.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Surround Digital Playback Volume',index=0
+
+This control is used to attenuate samples for left and right rear PCM FX-bus
+accumulators. ALSA uses accumulators 2 and 3 for left and right rear PCM samples.
+The result samples are forwarded to the rear I2S DACs. These DACs operate
+separately (they are not inside the AC97 codec).
+
+name='Surround Digital Capture Volume',index=0
+name='Surround Digital Capture Switch',index=0
+
+These controls are used to attenuate samples for left and right rear PCM FX-bus
+accumulators. ALSA uses accumulators 2 and 3 for left and right rear PCM samples.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Center Playback Volume',index=0
+
+This control is used to attenuate sample for center PCM FX-bus accumulator.
+ALSA uses accumulator 6 for center PCM sample. The result sample is forwarded
+to the ??rear?? right DAC PCM slot of the AC97 codec.
+
+name='LFE Playback Volume',index=0
+
+This control is used to attenuate sample for center PCM FX-bus accumulator.
+ALSA uses accumulator 6 for center PCM sample. The result sample is forwarded
+to the ??rear?? left DAC PCM slot of the AC97 codec.
+
+name='AC97 Playback Volume',index=0
+
+This control is used to attenuate samples for left and right front ADC PCM slots
+of the AC97 codec. The result samples are forwarded to the front DAC PCM
+slots of the AC97 codec.
+********************************************************************************
+*** Note: This control should be zero for the standard operations, otherwise ***
+*** a digital loopback is activated. ***
+********************************************************************************
+
+name='AC97 Capture Volume',index=0
+
+This control is used to attenuate samples for left and right front ADC PCM slots
+of the AC97 codec. The result is forwarded to the ADC capture FIFO (thus to
+the standard capture PCM device).
+********************************************************************************
+*** Note: This control should be 100 (maximal value), otherwise no analog ***
+*** inputs of the AC97 codec can be captured (recorded). ***
+********************************************************************************
+
+name='IEC958 TTL Playback Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 TTL
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the front DAC PCM slots of the AC97 codec.
+
+name='IEC958 TTL Capture Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 TTL
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the ADC capture FIFO (thus to the standard capture PCM device).
+
+name='Zoom Video Playback Volume',index=0
+
+This control is used to attenuate samples from left and right zoom video
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the front DAC PCM slots of the AC97 codec.
+
+name='Zoom Video Capture Volume',index=0
+
+This control is used to attenuate samples from left and right zoom video
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the ADC capture FIFO (thus to the standard capture PCM device).
+
+name='IEC958 Optical Playback Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 optical
+digital input. The result samples are forwarded to the front DAC PCM slots
+of the AC97 codec.
+
+name='IEC958 Optical Capture Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 optical
+digital inputs. The result samples are forwarded to the ADC capture FIFO
+(thus to the standard capture PCM device).
+
+name='IEC958 Coaxial Playback Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 coaxial
+digital inputs. The result samples are forwarded to the front DAC PCM slots
+of the AC97 codec.
+
+name='IEC958 Coaxial Capture Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 coaxial
+digital inputs. The result samples are forwarded to the ADC capture FIFO
+(thus to the standard capture PCM device).
+
+name='Line LiveDrive Playback Volume',index=0
+name='Line LiveDrive Playback Volume',index=1
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the LiveDrive). The result samples are forwarded to the front
+DAC PCM slots of the AC97 codec.
+
+name='Line LiveDrive Capture Volume',index=1
+name='Line LiveDrive Capture Volume',index=1
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the LiveDrive). The result samples are forwarded to the ADC
+capture FIFO (thus to the standard capture PCM device).
+
+name='Tone Control - Switch',index=0
+
+This control turns the tone control on or off. The samples for front, rear
+and center / LFE outputs are affected.
+
+name='Tone Control - Bass',index=0
+
+This control sets the bass intensity. There is no neutral value!!
+When the tone control code is activated, the samples are always modified.
+The closest value to pure signal is 20.
+
+name='Tone Control - Treble',index=0
+
+This control sets the treble intensity. There is no neutral value!!
+When the tone control code is activated, the samples are always modified.
+The closest value to pure signal is 20.
+
+name='IEC958 Optical Raw Playback Switch',index=0
+
+If this switch is on, then the samples for the IEC958 (S/PDIF) digital
+output are taken only from the raw FX8010 PCM, otherwise standard front
+PCM samples are taken.
+
+name='Headphone Playback Volume',index=1
+
+This control attenuates the samples for the headphone output.
+
+name='Headphone Center Playback Switch',index=1
+
+If this switch is on, then the sample for the center PCM is put to the
+left headphone output (useful for SB Live cards without separate center/LFE
+output).
+
+name='Headphone LFE Playback Switch',index=1
+
+If this switch is on, then the sample for the center PCM is put to the
+right headphone output (useful for SB Live cards without separate center/LFE
+output).
+
+
+3) PCM stream related controls
+------------------------------
+
+name='EMU10K1 PCM Volume',index 0-31
+
+Channel volume attenuation in range 0-0xffff. The maximum value (no
+attenuation) is default. The channel mapping for three values is
+as follows:
+
+ 0 - mono, default 0xffff (no attenuation)
+ 1 - left, default 0xffff (no attenuation)
+ 2 - right, default 0xffff (no attenuation)
+
+name='EMU10K1 PCM Send Routing',index 0-31
+
+This control specifies the destination - FX-bus accumulators. There are
+twelve values with this mapping:
+
+ 0 - mono, A destination (FX-bus 0-15), default 0
+ 1 - mono, B destination (FX-bus 0-15), default 1
+ 2 - mono, C destination (FX-bus 0-15), default 2
+ 3 - mono, D destination (FX-bus 0-15), default 3
+ 4 - left, A destination (FX-bus 0-15), default 0
+ 5 - left, B destination (FX-bus 0-15), default 1
+ 6 - left, C destination (FX-bus 0-15), default 2
+ 7 - left, D destination (FX-bus 0-15), default 3
+ 8 - right, A destination (FX-bus 0-15), default 0
+ 9 - right, B destination (FX-bus 0-15), default 1
+ 10 - right, C destination (FX-bus 0-15), default 2
+ 11 - right, D destination (FX-bus 0-15), default 3
+
+Don't forget that it's illegal to assign a channel to the same FX-bus accumulator
+more than once (it means 0=0 && 1=0 is an invalid combination).
+
+name='EMU10K1 PCM Send Volume',index 0-31
+
+It specifies the attenuation (amount) for given destination in range 0-255.
+The channel mapping is following:
+
+ 0 - mono, A destination attn, default 255 (no attenuation)
+ 1 - mono, B destination attn, default 255 (no attenuation)
+ 2 - mono, C destination attn, default 0 (mute)
+ 3 - mono, D destination attn, default 0 (mute)
+ 4 - left, A destination attn, default 255 (no attenuation)
+ 5 - left, B destination attn, default 0 (mute)
+ 6 - left, C destination attn, default 0 (mute)
+ 7 - left, D destination attn, default 0 (mute)
+ 8 - right, A destination attn, default 0 (mute)
+ 9 - right, B destination attn, default 255 (no attenuation)
+ 10 - right, C destination attn, default 0 (mute)
+ 11 - right, D destination attn, default 0 (mute)
+
+
+
+4) MANUALS/PATENTS:
+-------------------
+
+ftp://opensource.creative.com/pub/doc
+-------------------------------------
+
+ Files:
+ LM4545.pdf AC97 Codec
+
+ m2049.pdf The EMU10K1 Digital Audio Processor
+
+ hog63.ps FX8010 - A DSP Chip Architecture for Audio Effects
+
+
+WIPO Patents
+------------
+ Patent numbers:
+ WO 9901813 (A1) Audio Effects Processor with multiple asynchronous (Jan. 14, 1999)
+ streams
+
+ WO 9901814 (A1) Processor with Instruction Set for Audio Effects (Jan. 14, 1999)
+
+ WO 9901953 (A1) Audio Effects Processor having Decoupled Instruction
+ Execution and Audio Data Sequencing (Jan. 14, 1999)
+
+
+US Patents (http://www.uspto.gov/)
+----------------------------------
+
+ US 5925841 Digital Sampling Instrument employing cache memory (Jul. 20, 1999)
+
+ US 5928342 Audio Effects Processor integrated on a single chip (Jul. 27, 1999)
+ with a multiport memory onto which multiple asynchronous
+ digital sound samples can be concurrently loaded
+
+ US 5930158 Processor with Instruction Set for Audio Effects (Jul. 27, 1999)
+
+ US 6032235 Memory initialization circuit (Tram) (Feb. 29, 2000)
+
+ US 6138207 Interpolation looping of audio samples in cache connected to (Oct. 24, 2000)
+ system bus with prioritization and modification of bus transfers
+ in accordance with loop ends and minimum block sizes
+
+ US 6151670 Method for conserving memory storage using a (Nov. 21, 2000)
+ pool of short term memory registers
+
+ US 6195715 Interrupt control for multiple programs communicating with (Feb. 27, 2001)
+ a common interrupt by associating programs to GP registers,
+ defining interrupt register, polling GP registers, and invoking
+ callback routine associated with defined interrupt register
diff -urN linux-2.5.6-pre3/Documentation/sound/alsa/seq_oss.html linux-2.5.6/Documentation/sound/alsa/seq_oss.html
--- linux-2.5.6-pre3/Documentation/sound/alsa/seq_oss.html Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/alsa/seq_oss.html Thu Mar 7 18:24:41 2002
@@ -0,0 +1,409 @@
+
+
+
+ OSS Sequencer Emulation on ALSA
+
+
+
+
+
+
+
+
+
+
+OSS Sequencer Emulation on ALSA
+
+
+Copyright (c) 1998,1999 by Takashi Iwai
+<iwai@ww.uni-erlangen.de>
+
ver.0.1.8; Nov. 16, 1999
+
+
+
+
+
+1. Description
+This directory contains the OSS sequencer emulation driver on ALSA. Note
+that this program is still in the development state.
+What this does - it provides the emulation of the OSS sequencer, access
+via
+/dev/sequencer and /dev/music devices.
+The most of applications using OSS can run if the appropriate ALSA
+sequencer is prepared.
+
The following features are emulated by this driver:
+
+-
+Normal sequencer and MIDI events:
+
+
They are converted to the ALSA sequencer events, and sent to the corresponding
+port.
+-
+Timer events:
+
+
The timer is not selectable by ioctl. The control rate is fixed to
+100 regardless of HZ. That is, even on Alpha system, a tick is always
+1/100 second. The base rate and tempo can be changed in /dev/music.
+
+-
+Patch loading:
+
+
It purely depends on the synth drivers whether it's supported since
+the patch loading is realized by callback to the synth driver.
+-
+I/O controls:
+
+
Most of controls are accepted. Some controls
+are dependent on the synth driver, as well as even on original OSS.
+Furthermore, you can find the following advanced features:
+
+-
+Better queue mechanism:
+
+
The events are queued before processing them.
+-
+Multiple applications:
+
+
You can run two or more applications simultaneously (even for OSS sequencer)!
+However, each MIDI device is exclusive - that is, if a MIDI device is opened
+once by some application, other applications can't use it. No such a restriction
+in synth devices.
+-
+Real-time event processing:
+
+
The events can be processed in real time without using out of bound
+ioctl. To switch to real-time mode, send ABSTIME 0 event. The followed
+events will be processed in real-time without queued. To switch off the
+real-time mode, send RELTIME 0 event.
+-
+/proc interface:
+
+
The status of applications and devices can be shown via /proc/asound/seq/oss
+at any time. In the later version, configuration will be changed via /proc
+interface, too.
+
+
+2. Installation
+Run configure script with both sequencer support (--with-sequencer=yes)
+and OSS emulation (--with-oss=yes) options. A module snd-seq-oss.o
+will be created. If the synth module of your sound card supports for OSS
+emulation (so far, only Emu8000 driver), this module will be loaded automatically.
+Otherwise, you need to load this module manually.
+At beginning, this module probes all the MIDI ports which have been
+already connected to the sequencer. Once after that, the creation and deletion
+of ports are watched by announcement mechanism of ALSA sequencer.
+
The available synth and MIDI devices can be found in proc interface.
+Run "cat /proc/asound/seq/oss", and check the devices. For example,
+if you use an AWE64 card, you'll see like the following:
+
OSS sequencer emulation version 0.1.8
+ ALSA client number 63
+ ALSA receiver port 0
+
+ Number of applications: 0
+
+ Number of synth devices: 1
+
+ synth 0: [EMU8000]
+ type 0x1 : subtype 0x20 : voices 32
+ capabilties : ioctl enabled / load_patch enabled
+
+ Number of MIDI devices: 3
+
+ midi 0: [Emu8000 Port-0] ALSA port 65:0
+ capability write / opened none
+
+ midi 1: [Emu8000 Port-1] ALSA port 65:1
+ capability write / opened none
+
+ midi 2: [0: MPU-401 (UART)] ALSA port 64:0
+ capability read/write / opened none
+Note that the device number may be different from the information of
+/proc/asound/oss-devices
+or ones of the original OSS driver. Use the device number listed in /proc/asound/seq/oss
+to play via OSS sequencer emulation.
+
+3. Using Synthesizer Devices
+Run your favorite program. I've tested playmidi-2.4, awemidi-0.4.3, gmod-3.1
+and xmp-1.1.5. You can load samples via /dev/sequencer like sfxload,
+too.
+If the lowlevel driver supports multiple access to synth devices (like
+Emu8000 driver), two or more applications are allowed to run at the same
+time.
+
+4. Using MIDI Devices
+So far, only MIDI output was tested. MIDI input was not checked at all,
+but hopefully it will work. Use the device number listed in /proc/asound/seq/oss.
+Be aware that these numbers are mostly different from the list in
+/proc/asound/oss-devices.
+
+5. Module Options
+The following module options are available:
+
+-
+maxqlen
+
+
specifies the maximum read/write queue length. This queue is private
+for OSS sequencer, so that it is independent from the queue length of ALSA
+sequencer. Default value is 1024.
+-
+seq_oss_debug
+
+
specifies the debug level and accepts zero (= no debug message) or
+positive integer. Default value is 0.
+
+
+6. Queue Mechanism
+OSS sequencer emulation uses an ALSA priority queue. The
+events from /dev/sequencer are processed and put onto the queue
+specified by module option.
+All the events from /dev/sequencer are parsed at beginning.
+The timing events are also parsed at this moment, so that the events may
+be processed in real-time. Sending an event ABSTIME 0 switches the operation
+mode to real-time mode, and sending an event RELTIME 0 switches it off.
+In the real-time mode, all events are dispatched immediately.
+
The queued events are dispatched to the corresponding ALSA sequencer
+ports after scheduled time by ALSA sequencer dispatcher.
+
If the write-queue is full, the application sleeps until a certain amount
+(as default one half) becomes empty in blocking mode. The synchronization
+to write timing was implemented, too.
+
The input from MIDI devices or echo-back events are stored on read FIFO
+queue. If application reads /dev/sequencer in blocking mode, the
+process will be awaked.
+
+
+7. Interface to Synthesizer Device
+
+
+7.1. Registration
+To register an OSS synthesizer device, use snd_seq_oss_synth_register
+function.
+int snd_seq_oss_synth_register(char *name, int type, int subtype, int nvoices,
+ snd_seq_oss_callback_t *oper, void *private_data)
+The arguments name, type, subtype and
+nvoices
+are used for making the appropriate synth_info structure for ioctl. The
+return value is an index number of this device. This index must be remembered
+for unregister. If registration is failed, -errno will be returned.
+To release this device, call snd_seq_oss_synth_unregister function:
+
int snd_seq_oss_synth_unregister(int index),
+where the index is the index number returned by register function.
+
+7.2. Callbacks
+OSS synthesizer devices have capability for sample downloading and ioctls
+like sample reset. In OSS emulation, these special features are realized
+by using callbacks. The registration argument oper is used to specify these
+callbacks. The following callback functions must be defined:
+snd_seq_oss_callback_t:
+ int (*open)(snd_seq_oss_arg_t *p, void *closure);
+ int (*close)(snd_seq_oss_arg_t *p);
+ int (*ioctl)(snd_seq_oss_arg_t *p, unsigned int cmd, unsigned long arg);
+ int (*load_patch)(snd_seq_oss_arg_t *p, int format, const char *buf, int offs, int count);
+ int (*reset)(snd_seq_oss_arg_t *p);
+Except for open and close callbacks, they are allowed
+to be NULL.
+Each callback function takes the argument type snd_seq_oss_arg_t as the
+first argument.
+
struct snd_seq_oss_arg_t {
+ int app_index;
+ int file_mode;
+ int seq_mode;
+ snd_seq_addr_t addr;
+ void *private_data;
+ int event_passing;
+};
+The first three fields, app_index, file_mode and
+seq_mode
+are initialized by OSS sequencer. The app_index is the application
+index which is unique to each application opening OSS sequencer. The
+file_mode
+is bit-flags indicating the file operation mode. See
+seq_oss.h
+for its meaning. The seq_mode is sequencer operation mode. In
+the current version, only SND_OSSSEQ_MODE_SYNTH is used.
+The next two fields, addr and private_data, must be
+filled by the synth driver at open callback. The addr contains
+the address of ALSA sequencer port which is assigned to this device. If
+the driver allocates memory for private_data, it must be released
+in close callback by itself.
+
The last field, event_passing, indicates how to translate note-on
+/ off events. In PROCESS_EVENTS mode, the note 255 is regarded
+as velocity change, and key pressure event is passed to the port. In PASS_EVENTS
+mode, all note on/off events are passed to the port without modified. PROCESS_KEYPRESS
+mode checks the note above 128 and regards it as key pressure event (mainly
+for Emu8000 driver).
+
+7.2.1. Open Callback
+The open is called at each time this device is opened by an application
+using OSS sequencer. This must not be NULL. Typically, the open callback
+does the following procedure:
+
+-
+Allocate private data record.
+
+-
+Create an ALSA sequencer port.
+
+-
+Set the new port address on arg->addr.
+
+-
+Set the private data record pointer on arg->private_data.
+
+Note that the type bit-flags in port_info of this synth port must NOT contain
+TYPE_MIDI_GENERIC
+bit. Instead, TYPE_SPECIFIC should be used. Also, CAP_SUBSCRIPTION
+bit should NOT be included, too. This is necessary to tell it from other
+normal MIDI devices. If the open procedure succeeded, return zero. Otherwise,
+return -errno.
+
+7.2.2 Ioctl Callback
+The ioctl callback is called when the sequencer receives device-specific
+ioctls. The following two ioctls should be processed by this callback:
+
+-
+IOCTL_SEQ_RESET_SAMPLES
+
+
reset all samples on memory -- return 0
+-
+IOCTL_SYNTH_MEMAVL
+
+
return the available memory size
+-
+FM_4OP_ENABLE
+
+
can be ignored usually
+The other ioctls are processed inside the sequencer without passing to
+the lowlevel driver.
+
+7.2.3 Load_Patch Callback
+The load_patch callback is used for sample-downloading. This callback
+must read the data on user-space and transfer to each device. Return 0
+if succeeded, and -errno if failed. The format argument is the patch key
+in patch_info record. The buf is user-space pointer where patch_info record
+is stored. The offs can be ignored. The count is total data size of this
+sample data.
+
+7.2.4 Close Callback
+The close callback is called when this device is closed by the
+applicaion. If any private data was allocated in open callback, it must
+be released in the close callback. The deletion of ALSA port should be
+done here, too. This callback must not be NULL.
+
+7.2.5 Reset Callback
+The reset callback is called when sequencer device is reset or
+closed by applications. The callback should turn off the sounds on the
+relevant port immediately, and initialize the status of the port. If this
+callback is undefined, OSS seq sends a HEARTBEAT event to the
+port.
+
+7.3 Events
+Most of the events are processed by sequencer and translated to the adequate
+ALSA sequencer events, so that each synth device can receive by input_event
+callback of ALSA sequencer port. The following ALSA events should be implemented
+by the driver:
+
+
+
+| ALSA event |
+
+Original OSS events |
+
+
+
+| NOTEON |
+
+SEQ_NOTEON
+
MIDI_NOTEON |
+
+
+
+| NOTE |
+
+SEQ_NOTEOFF
+
MIDI_NOTEOFF |
+
+
+
+| KEYPRESS |
+
+MIDI_KEY_PRESSURE |
+
+
+
+| CHANPRESS |
+
+SEQ_AFTERTOUCH
+
MIDI_CHN_PRESSURE |
+
+
+
+| PGMCHANGE |
+
+SEQ_PGMCHANGE
+
MIDI_PGM_CHANGE |
+
+
+
+| PITCHBEND |
+
+SEQ_CONTROLLER(CTRL_PITCH_BENDER)
+
MIDI_PITCH_BEND |
+
+
+
+| CONTROLLER |
+
+MIDI_CTL_CHANGE
+
SEQ_BALANCE (with CTL_PAN) |
+
+
+
+| CONTROL14 |
+
+SEQ_CONTROLLER |
+
+
+
+| REGPARAM |
+
+SEQ_CONTROLLER(CTRL_PITCH_BENDER_RANGE) |
+
+
+
+| SYSEX |
+
+SEQ_SYSEX |
+
+
+
+The most of these behavior can be realized by MIDI emulation driver
+included in the Emu8000 lowlevel driver. In the future release, this module
+will be independent.
+
Some OSS events (SEQ_PRIVATE and SEQ_VOLUME events) are passed as event
+type SND_SEQ_OSS_PRIVATE. The OSS sequencer passes these event 8 byte
+packets without any modification. The lowlevel driver should process these
+events appropriately.
+
+8. Interface to MIDI Device
+Since the OSS emulation probes the creation and deletion of ALSA MIDI sequencer
+ports automatically by receiving announcement from ALSA sequencer, the
+MIDI devices don't need to be registered explicitly like synth devices.
+However, the MIDI port_info registered to ALSA sequencer must include a group
+name SND_SEQ_GROUP_DEVICE and a capability-bit CAP_READ or
+CAP_WRITE. Also, subscription capabilities, CAP_SUBS_READ or CAP_SUBS_WRITE,
+must be defined, too. If these conditions are not satisfied, the port is not
+registered as OSS sequencer MIDI device.
+The events via MIDI devices are parsed in OSS sequencer and converted
+to the corresponding ALSA sequencer events. The input from MIDI sequencer
+is also converted to MIDI byte events by OSS sequencer. This works just
+a reverse way of seq_midi module.
+
+9. Known Problems / TODO's
+
+
+-
+Patch loading via ALSA instrument layer is not implemented yet.
+
+
+
+
diff -urN linux-2.5.6-pre3/Documentation/sound/btaudio linux-2.5.6/Documentation/sound/btaudio
--- linux-2.5.6-pre3/Documentation/sound/btaudio Tue Feb 19 18:10:57 2002
+++ linux-2.5.6/Documentation/sound/btaudio Wed Dec 31 16:00:00 1969
@@ -1,92 +0,0 @@
-
-Intro
-=====
-
-people start bugging me about this with questions, looks like I
-should write up some documentation for this beast. That way I
-don't have to answer that much mails I hope. Yes, I'm lazy...
-
-
-You might have noticed that the bt878 grabber cards have actually
-_two_ PCI functions:
-
-$ lspci
-[ ... ]
-00:0a.0 Multimedia video controller: Brooktree Corporation Bt878 (rev 02)
-00:0a.1 Multimedia controller: Brooktree Corporation Bt878 (rev 02)
-[ ... ]
-
-The first does video, it is backward compatible to the bt848. The second
-does audio. btaudio is a driver for the second function. It's a sound
-driver which can be used for recording sound (and _only_ recording, no
-playback). As most TV cards come with a short cable which can be plugged
-into your sound card's line-in you probably don't need this driver if all
-you want to do is just watching TV...
-
-
-Driver Status
-=============
-
-Still somewhat experimental. The driver should work stable, i.e. it
-should'nt crash your box. It might not work as expected, have bugs,
-not being fully OSS API compilant, ...
-
-Latest versions are available from http://bytesex.org/bttv/, the
-driver is in the bttv tarball. Kernel patches might be available too,
-have a look at http://bytesex.org/bttv/listing.html.
-
-The chip knows two different modes. btaudio registers two dsp
-devices, one for each mode. They can not be used at the same time.
-
-
-Digital audio mode
-==================
-
-The chip gives you 16 bit stereo sound. The sample rate depends on
-the external source which feeds the bt878 with digital sound via I2S
-interface. There is a insmod option (rate) to tell the driver which
-sample rate the hardware uses (32000 is the default).
-
-One possible source for digital sound is the msp34xx audio processor
-chip which provides digital sound via I2S with 32 kHz sample rate. My
-Hauppauge board works this way.
-
-The Osprey-200 reportly gives you digital sound with 44100 Hz sample
-rate. It is also possible that you get no sound at all.
-
-
-analog mode (A/D)
-=================
-
-You can tell the driver to use this mode with the insmod option "analog=1".
-The chip has three analog inputs. Consequently you'll get a mixer device
-to control these.
-
-The analog mode supports mono only. Both 8 + 16 bit. Both are _signed_
-int, which is uncommon for the 8 bit case. Sample rate range is 119 kHz
-to 448 kHz. Yes, the number of digits is correct. The driver supports
-downsampling by powers of two, so you can ask for more usual sample rates
-like 44 kHz too.
-
-With my Hauppauge I get noisy sound on the second input (mapped to line2
-by the mixer device). Others get a useable signal on line1.
-
-
-some examples
-=============
-
-* read audio data from btaudio (dsp2), send to es1730 (dsp,dsp1):
- $ sox -w -r 32000 -t ossdsp /dev/dsp2 -t ossdsp /dev/dsp
-
-* read audio data from btaudio, send to esound daemon (which might be
- running on another host):
- $ sox -c 2 -w -r 32000 -t ossdsp /dev/dsp2 -t sw - | esdcat -r 32000
- $ sox -c 1 -w -r 32000 -t ossdsp /dev/dsp2 -t sw - | esdcat -m -r 32000
-
-
-Have fun,
-
- Gerd
-
---
-Gerd Knorr
diff -urN linux-2.5.6-pre3/Documentation/sound/cs46xx linux-2.5.6/Documentation/sound/cs46xx
--- linux-2.5.6-pre3/Documentation/sound/cs46xx Tue Feb 19 18:10:56 2002
+++ linux-2.5.6/Documentation/sound/cs46xx Wed Dec 31 16:00:00 1969
@@ -1,138 +0,0 @@
-
-Documentation for the Cirrus Logic/Crystal SoundFusion cs46xx/cs4280 audio
-controller chips (2001/05/11)
-
-The cs46xx audio driver supports the DSP line of Cirrus controllers.
-Specifically, the cs4610, cs4612, cs4614, cs4622, cs4624, cs4630 and the cs4280
-products. This driver uses the generic ac97_codec driver for AC97 codec
-support.
-
-
-Features:
-
-Full Duplex Playback/Capture supported from 8k-48k.
-16Bit Signed LE & 8Bit Unsigned, with Mono or Stereo supported.
-
-APM/PM - 2.2.x PM is enabled and functional. APM can also
-be enabled for 2.4.x by modifying the CS46XX_ACPI_SUPPORT macro
-definition.
-
-DMA playback buffer size is configurable from 16k (defaultorder=2) up to 2Meg
-(defaultorder=11). DMA capture buffer size is fixed at a single 4k page as
-two 2k fragments.
-
-MMAP seems to work well with QuakeIII, and test XMMS plugin.
-
-Myth2 works, but the polling logic is not fully correct, but is functional.
-
-The 2.4.4-ac6 gameport code in the cs461x joystick driver has been tested
-with a Microsoft Sidewinder joystick (cs461x.o and sidewinder.o). This
-audio driver must be loaded prior to the joystick driver to enable the
-DSP task image supporting the joystick device.
-
-
-Limitations:
-
-SPDIF is currently not supported.
-
-Primary codec support only. No secondary codec support is implemented.
-
-
-
-NOTES:
-
-Hercules Game Theatre XP - the EGPIO2 pin controls the external Amp,
-and has been tested.
-Module parameter hercules_egpio_disable set to 1, will force a 0 to EGPIODR
-to disable the external amplifier.
-
-VTB Santa Cruz - the GPIO7/GPIO8 on the Secondary Codec control
-the external amplifier for the "back" speakers, since we do not
-support the secondary codec then this external amp is not
-turned on. The primary codec external amplifier is supported but
-note that the AC97 EAPD bit is inverted logic (amp_voyetra()).
-
-DMA buffer size - there are issues with many of the Linux applications
-concerning the optimal buffer size. Several applications request a
-certain fragment size and number and then do not verify that the driver
-has the ability to support the requested configuration.
-SNDCTL_DSP_SETFRAGMENT ioctl is used to request a fragment size and
-number of fragments. Some applications exit if an error is returned
-on this particular ioctl. Therefore, in alignment with the other OSS audio
-drivers, no error is returned when a SETFRAGs IOCTL is received, but the
-values passed from the app are not used in any buffer calculation
-(ossfragshift/ossmaxfrags are not used).
-Use the "defaultorder=N" module parameter to change the buffer size if
-you have an application that requires a specific number of fragments
-or a specific buffer size (see below).
-
-Debug Interface
----------------
-There is an ioctl debug interface to allow runtime modification of the
-debug print levels. This debug interface code can be disabled from the
-compilation process with commenting the following define:
-#define CSDEBUG_INTERFACE 1
-There is also a debug print methodolgy to select printf statements from
-different areas of the driver. A debug print level is also used to allow
-additional printfs to be active. Comment out the following line in the
-driver to disable compilation of the CS_DBGOUT print statements:
-#define CSDEBUG 1
-
-Please see the defintions for cs_debuglevel and cs_debugmask for additional
-information on the debug levels and sections.
-
-There is also a csdbg executable to allow runtime manipulation of these
-parameters. for a copy email: twoller@crystal.cirrus.com
-
-
-
-MODULE_PARMS definitions
-------------------------
-MODULE_PARM(defaultorder, "i");
-defaultorder=N
-where N is a value from 1 to 12
-The buffer order determines the size of the dma buffer for the driver.
-under Linux, a smaller buffer allows more responsiveness from many of the
-applications (e.g. games). A larger buffer allows some of the apps (esound)
-to not underrun the dma buffer as easily. As default, use 32k (order=3)
-rather than 64k as some of the games work more responsively.
-(2^N) * PAGE_SIZE = allocated buffer size
-
-MODULE_PARM(cs_debuglevel, "i");
-MODULE_PARM(cs_debugmask, "i");
-cs_debuglevel=N
-cs_debugmask=0xMMMMMMMM
-where N is a value from 0 (no debug printfs), to 9 (maximum)
-0xMMMMMMMM is a debug mask corresponding to the CS_xxx bits (see driver source).
-
-MODULE_PARM(hercules_egpio_disable, "i");
-hercules_egpio_disable=N
-where N is a 0 (enable egpio), or a 1 (disable egpio support)
-
-MODULE_PARM(initdelay, "i");
-initdelay=N
-This value is used to determine the millescond delay during the initialization
-code prior to powering up the PLL. On laptops this value can be used to
-assist with errors on resume, mostly with IBM laptops. Basically, if the
-system is booted under battery power then the mdelay()/udelay() functions fail to
-properly delay the required time. Also, if the system is booted under AC power
-and then the power removed, the mdelay()/udelay() functions will not delay properly.
-
-MODULE_PARM(powerdown, "i");
-powerdown=N
-where N is 0 (disable any powerdown of the internal blocks) or 1 (enable powerdown)
-
-
-MODULE_PARM(external_amp, "i");
-external_amp=1
-if N is set to 1, then force enabling the EAPD support in the primary AC97 codec.
-override the detection logic and force the external amp bit in the AC97 0x26 register
-to be reset (0). EAPD should be 0 for powerup, and 1 for powerdown. The VTB Santa Cruz
-card has inverted logic, so there is a special function for these cards.
-
-MODULE_PARM(thinkpad, "i");
-thinkpad=1
-if N is set to 1, then force enabling the clkrun functionality.
-Currently, when the part is being used, then clkrun is disabled for the entire system,
-but re-enabled when the driver is released or there is no outstanding open count.
-
diff -urN linux-2.5.6-pre3/Documentation/sound/es1370 linux-2.5.6/Documentation/sound/es1370
--- linux-2.5.6-pre3/Documentation/sound/es1370 Tue Feb 19 18:10:58 2002
+++ linux-2.5.6/Documentation/sound/es1370 Wed Dec 31 16:00:00 1969
@@ -1,70 +0,0 @@
-/proc/sound, /dev/sndstat
--------------------------
-
-/proc/sound and /dev/sndstat is not supported by the
-driver. To find out whether the driver succeeded loading,
-check the kernel log (dmesg).
-
-
-ALaw/uLaw sample formats
-------------------------
-
-This driver does not support the ALaw/uLaw sample formats.
-ALaw is the default mode when opening a sound device
-using OSS/Free. The reason for the lack of support is
-that the hardware does not support these formats, and adding
-conversion routines to the kernel would lead to very ugly
-code in the presence of the mmap interface to the driver.
-And since xquake uses mmap, mmap is considered important :-)
-and no sane application uses ALaw/uLaw these days anyway.
-In short, playing a Sun .au file as follows:
-
-cat my_file.au > /dev/dsp
-
-does not work. Instead, you may use the play script from
-Chris Bagwell's sox-12.14 package (available from the URL
-below) to play many different audio file formats.
-The script automatically determines the audio format
-and does do audio conversions if necessary.
-http://home.sprynet.com/sprynet/cbagwell/projects.html
-
-
-Blocking vs. nonblocking IO
----------------------------
-
-Unlike OSS/Free this driver honours the O_NONBLOCK file flag
-not only during open, but also during read and write.
-This is an effort to make the sound driver interface more
-regular. Timidity has problems with this; a patch
-is available from http://www.ife.ee.ethz.ch/~sailer/linux/pciaudio.html.
-(Timidity patched will also run on OSS/Free).
-
-
-MIDI UART
----------
-
-The driver supports a simple MIDI UART interface, with
-no ioctl's supported.
-
-
-MIDI synthesizer
-----------------
-
-This soundcard does not have any hardware MIDI synthesizer;
-MIDI synthesis has to be done in software. To allow this
-the driver/soundcard supports two PCM (/dev/dsp) interfaces.
-The second one goes to the mixer "synth" setting and supports
-only a limited set of sampling rates (44100, 22050, 11025, 5512).
-By setting lineout to 1 on the driver command line
-(eg. insmod es1370 lineout=1) it is even possible on some
-cards to convert the LINEIN jack into a second LINEOUT jack, thus
-making it possible to output four independent audio channels!
-
-There is a freely available software package that allows
-MIDI file playback on this soundcard called Timidity.
-See http://www.cgs.fi/~tt/timidity/.
-
-
-
-Thomas Sailer
-t.sailer@alumni.ethz.ch
diff -urN linux-2.5.6-pre3/Documentation/sound/es1371 linux-2.5.6/Documentation/sound/es1371
--- linux-2.5.6-pre3/Documentation/sound/es1371 Tue Feb 19 18:10:59 2002
+++ linux-2.5.6/Documentation/sound/es1371 Wed Dec 31 16:00:00 1969
@@ -1,64 +0,0 @@
-/proc/sound, /dev/sndstat
--------------------------
-
-/proc/sound and /dev/sndstat is not supported by the
-driver. To find out whether the driver succeeded loading,
-check the kernel log (dmesg).
-
-
-ALaw/uLaw sample formats
-------------------------
-
-This driver does not support the ALaw/uLaw sample formats.
-ALaw is the default mode when opening a sound device
-using OSS/Free. The reason for the lack of support is
-that the hardware does not support these formats, and adding
-conversion routines to the kernel would lead to very ugly
-code in the presence of the mmap interface to the driver.
-And since xquake uses mmap, mmap is considered important :-)
-and no sane application uses ALaw/uLaw these days anyway.
-In short, playing a Sun .au file as follows:
-
-cat my_file.au > /dev/dsp
-
-does not work. Instead, you may use the play script from
-Chris Bagwell's sox-12.14 package (available from the URL
-below) to play many different audio file formats.
-The script automatically determines the audio format
-and does do audio conversions if necessary.
-http://home.sprynet.com/sprynet/cbagwell/projects.html
-
-
-Blocking vs. nonblocking IO
----------------------------
-
-Unlike OSS/Free this driver honours the O_NONBLOCK file flag
-not only during open, but also during read and write.
-This is an effort to make the sound driver interface more
-regular. Timidity has problems with this; a patch
-is available from http://www.ife.ee.ethz.ch/~sailer/linux/pciaudio.html.
-(Timidity patched will also run on OSS/Free).
-
-
-MIDI UART
----------
-
-The driver supports a simple MIDI UART interface, with
-no ioctl's supported.
-
-
-MIDI synthesizer
-----------------
-
-This soundcard does not have any hardware MIDI synthesizer;
-MIDI synthesis has to be done in software. To allow this
-the driver/soundcard supports two PCM (/dev/dsp) interfaces.
-
-There is a freely available software package that allows
-MIDI file playback on this soundcard called Timidity.
-See http://www.cgs.fi/~tt/timidity/.
-
-
-
-Thomas Sailer
-t.sailer@alumni.ethz.ch
diff -urN linux-2.5.6-pre3/Documentation/sound/mwave linux-2.5.6/Documentation/sound/mwave
--- linux-2.5.6-pre3/Documentation/sound/mwave Tue Feb 19 18:11:00 2002
+++ linux-2.5.6/Documentation/sound/mwave Wed Dec 31 16:00:00 1969
@@ -1,185 +0,0 @@
- How to try to survive an IBM Mwave under Linux SB drivers
-
-
-+ IBM have now released documentation of sorts and Torsten is busy
- trying to make the Mwave work. This is not however a trivial task.
-
-----------------------------------------------------------------------------
-
-OK, first thing - the IRQ problem IS a problem, whether the test is bypassed or
-not. It is NOT a Linux problem, but an MWAVE problem that is fixed with the
-latest MWAVE patches. So, in other words, don't bypass the test for MWAVES!
-
-I have Windows 95 on /dev/hda1, swap on /dev/hda2, and Red Hat 5 on /dev/hda3.
-
-The steps, then:
-
- Boot to Linux.
- Mount Windows 95 file system (assume mount point = /dos95).
- mkdir /dos95/linux
- mkdir /dos95/linux/boot
- mkdir /dos95/linux/boot/parms
-
- Copy the kernel, any initrd image, and loadlin to /dos95/linux/boot/.
-
- Reboot to Windows 95.
-
- Edit C:/msdos.sys and add or change the following:
-
- Logo=0
- BootGUI=0
-
- Note that msdos.sys is a text file but it needs to be made 'unhidden',
- readable and writable before it can be edited. This can be done with
- DOS' "attrib" command.
-
- Edit config.sys to have multiple config menus. I have one for windows 95 and
- five for Linux, like this:
-------------
-[menu]
-menuitem=W95, Windows 95
-menuitem=LINTP, Linux - ThinkPad
-menuitem=LINTP3, Linux - ThinkPad Console
-menuitem=LINDOC, Linux - Docked
-menuitem=LINDOC3, Linux - Docked Console
-menuitem=LIN1, Linux - Single User Mode
-REM menudefault=W95,10
-
-[W95]
-
-[LINTP]
-
-[LINDOC]
-
-[LINTP3]
-
-[LINDOC3]
-
-[LIN1]
-
-[COMMON]
-FILES=30
-REM Please read README.TXT in C:\MWW subdirectory before changing the DOS= statement.
-DOS=HIGH,UMB
-DEVICE=C:\MWW\MANAGER\MWD50430.EXE
-SHELL=c:\command.com /e:2048
--------------------
-
-The important things are the SHELL and DEVICE statements.
-
- Then change autoexec.bat. Basically everything in there originally should be
- done ONLY when Windows 95 is booted. Then you add new things specifically
- for Linux. Mine is as follows
-
----------------
-@ECHO OFF
-if "%CONFIG%" == "W95" goto W95
-
-REM
-REM Linux stuff
-REM
-SET MWPATH=C:\MWW\DLL;C:\MWW\MWGAMES;C:\MWW\DSP
-SET BLASTER=A220 I5 D1
-SET MWROOT=C:\MWW
-SET LIBPATH=C:\MWW\DLL
-SET PATH=C:\WINDOWS;C:\MWW\DLL;
-CALL MWAVE START NOSHOW
-c:\linux\boot\loadlin.exe @c:\linux\boot\parms\%CONFIG%.par
-
-:W95
-REM
-REM Windows 95 stuff
-REM
-c:\toolkit\guard
-SET MSINPUT=C:\MSINPUT
-SET MWPATH=C:\MWW\DLL;C:\MWW\MWGAMES;C:\MWW\DSP
-REM The following is used by DOS games to recognize Sound Blaster hardware.
-REM If hardware settings are changed, please change this line as well.
-REM See the Mwave README file for instructions.
-SET BLASTER=A220 I5 D1
-SET MWROOT=C:\MWW
-SET LIBPATH=C:\MWW\DLL
-SET PATH=C:\WINDOWS;C:\WINDOWS\COMMAND;E:\ORAWIN95\BIN;f:\msdev\bin;e:\v30\bin.dbg;v:\devt\v30\bin;c:\JavaSDK\Bin;C:\MWW\DLL;
-SET INCLUDE=f:\MSDEV\INCLUDE;F:\MSDEV\MFC\INCLUDE
-SET LIB=F:\MSDEV\LIB;F:\MSDEV\MFC\LIB
-win
-
-------------------------
-
-Now build a file in c:\linux\boot\parms for each Linux config that you have.
-
-For example, my LINDOC3 config is for a docked Thinkpad at runlevel 3 with no
-initrd image, and has a parameter file named LINDOC3.PAR in c:\linux\boot\parms:
-
------------------------
-# LOADLIN @param_file image=other_image root=/dev/other
-#
-# Linux Console in docking station
-#
-c:\linux\boot\zImage.krn # First value must be filename of Linux kernel.
-root=/dev/hda3 # device which gets mounted as root FS
-ro # Other kernel arguments go here.
-apm=off
-doc=yes
-3
------------------------
-
-The doc=yes parameter is an environment variable used by my init scripts, not
-a kernel argument.
-
-However, the apm=off parameter IS a kernel argument! APM, at least in my setup,
-causes the kernel to crash when loaded via loadlin (but NOT when loaded via
-LILO). The APM stuff COULD be forced out of the kernel via the kernel compile
-options. Instead, I got an unofficial patch to the APM drivers that allows them
-to be dynamically deactivated via kernel arguments. Whatever you chose to
-document, APM, it seems, MUST be off for setups like mine.
-
-Now make sure C:\MWW\MWCONFIG.REF looks like this:
-
-----------------------
-[NativeDOS]
-Default=SB1.5
-SBInputSource=CD
-SYNTH=FM
-QSound=OFF
-Reverb=OFF
-Chorus=OFF
-ReverbDepth=5
-ChorusDepth=5
-SBInputVolume=5
-SBMainVolume=10
-SBWaveVolume=10
-SBSynthVolume=10
-WaveTableVolume=10
-AudioPowerDriver=ON
-
-[FastCFG]
-Show=No
-HideOption=Off
------------------------------
-
-OR the Default= line COULD be
-
-Default=SBPRO
-
-Reboot to Windows 95 and choose Linux. When booted, use sndconfig to configure
-the sound modules and voilà - ThinkPad sound with Linux.
-
-Now the gotchas - you can either have CD sound OR Mixers but not both. That's a
-problem with the SB1.5 (CD sound) or SBPRO (Mixers) settings. No one knows why
-this is!
-
-For some reason MPEG3 files, when played through mpg123, sound like they
-are playing at 1/8th speed - not very useful! If you have ANY insight
-on why this second thing might be happening, I would be grateful.
-
-===========================================================
- _/ _/_/_/_/
- _/_/ _/_/ _/
- _/ _/_/ _/_/_/_/ Martin John Bartlett
- _/ _/ _/ _/ (martin@nitram.demon.co.uk)
-_/ _/_/_/_/
- _/
-_/ _/
- _/_/
-===========================================================
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/AD1816 linux-2.5.6/Documentation/sound/oss/AD1816
--- linux-2.5.6-pre3/Documentation/sound/oss/AD1816 Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/AD1816 Thu Mar 7 18:24:41 2002
@@ -0,0 +1,84 @@
+Documentation for the AD1816(A) sound driver
+============================================
+
+Installation:
+-------------
+
+To get your AD1816(A) based sound card work, you'll have to enable support for
+experimental code ("Prompt for development and/or incomplete code/drivers")
+and isapnp ("Plug and Play support", "ISA Plug and Play support"). Enable
+"Sound card support", "OSS modules support" and "Support for AD1816(A) based
+cards (EXPERIMENTAL)" in the sound configuration menu, too. Now build, install
+and reboot the new kernel as usual.
+
+Features:
+---------
+
+List of features supported by this driver:
+- full-duplex support
+- supported audio formats: unsigned 8bit, signed 16bit little endian,
+ signed 16bit big endian, µ-law, A-law
+- supported channels: mono and stereo
+- supported recording sources: Master, CD, Line, Line1, Line2, Mic
+- supports phat 3d stereo circuit (Line 3)
+
+
+Supported cards:
+----------------
+
+The following cards are known to work with this driver:
+- Terratec Base 1
+- Terratec Base 64
+- HP Kayak
+- Acer FX-3D
+- SY-1816
+- Highscreen Sound-Boostar 32 Wave 3D
+- Highscreen Sound-Boostar 16
+- AVM Apex Pro card
+- (Aztech SC-16 3D)
+- (Newcom SC-16 3D)
+- (Terratec EWS64S)
+
+Cards listed in brackets are not supported reliable. If you have such a card
+you should add the extra parameter:
+ options=1
+when loading the ad1816 module via modprobe.
+
+
+Troubleshooting:
+----------------
+
+First of all you should check, if the driver has been loaded
+properly.
+
+If loading of the driver succeeds, but playback/capture fails, check
+if you used the correct values for irq, dma and dma2 when loading the module.
+If one of them is wrong you usually get the following error message:
+
+Nov 6 17:06:13 tek01 kernel: Sound: DMA (output) timed out - IRQ/DRQ config error?
+
+If playback/capture is too fast or to slow, you should have a look at
+the clock chip of your sound card. The AD1816 was designed for a 33MHz
+oscillator, however most sound card manufacturer use slightly
+different oscillators as they are cheaper than 33MHz oscillators. If
+you have such a card you have to adjust the ad1816_clockfreq parameter
+above. For example: For a card using a 32.875MHz oscillator use
+ad1816_clockfreq=32875 instead of ad1816_clockfreq=33000.
+
+
+Updates, bugfixes and bugreports:
+--------------------------------
+
+As the driver is still experimental and under development, you should
+watch out for updates. Updates of the driver are available on the
+Internet from one of my home pages:
+ http://www.student.informatik.tu-darmstadt.de/~tek/projects/linux.html
+or:
+ http://www.tu-darmstadt.de/~tek01/projects/linux.html
+
+Bugreports, bugfixes and related questions should be sent via E-Mail to:
+ tek@rbg.informatik.tu-darmstadt.de
+
+Thorsten Knabe
+Christoph Hellwig
+ Last modified: 2000/09/20
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/ALS linux-2.5.6/Documentation/sound/oss/ALS
--- linux-2.5.6-pre3/Documentation/sound/oss/ALS Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/ALS Thu Mar 7 18:24:41 2002
@@ -0,0 +1,66 @@
+ALS-007/ALS-100/ALS-200 based sound cards
+=========================================
+
+Support for sound cards based around the Avance Logic
+ALS-007/ALS-100/ALS-200 chip is included. These chips are a single
+chip PnP sound solution which is mostly hardware compatible with the
+Sound Blaster 16 card, with most differences occurring in the use of
+the mixer registers. For this reason the ALS code is integrated
+as part of the Sound Blaster 16 driver (adding only 800 bytes to the
+SB16 driver).
+
+To use an ALS sound card under Linux, enable the following options as
+modules in the sound configuration section of the kernel config:
+ - 100% Sound Blaster compatibles (SB16/32/64, ESS, Jazz16) support
+ - FM synthesizer (YM3812/OPL-3) support
+ - standalone MPU401 support may be required for some cards; for the
+ ALS-007, when using isapnptools, it is required
+Since the ALS-007/100/200 are PnP cards, ISAPnP support should probably be
+compiled in. If kernel level PnP support is not included, isapnptools will
+be required to configure the card before the sound modules are loaded.
+
+When using kernel level ISAPnP, the kernel should correctly identify and
+configure all resources required by the card when the "sb" module is
+inserted. Note that the ALS-007 does not have a 16 bit DMA channel and that
+the MPU401 interface on this card uses a different interrupt to the audio
+section. This should all be correctly configured by the kernel; if problems
+with the MPU401 interface surface, try using the standalone MPU401 module,
+passing "0" as the "sb" module's "mpu_io" module parameter to prevent the
+soundblaster driver attempting to register the MPU401 itself. The onboard
+synth device can be accessed using the "opl3" module.
+
+If isapnptools is used to wake up the sound card (as in 2.2.x), the settings
+of the card's resources should be passed to the kernel modules ("sb", "opl3"
+and "mpu401") using the module parameters. When configuring an ALS-007, be
+sure to specify different IRQs for the audio and MPU401 sections - this card
+requires they be different. For "sb", "io", "irq" and "dma" should be set
+to the same values used to configure the audio section of the card with
+isapnp. "dma16" should be explicitly set to "-1" for an ALS-007 since this
+card does not have a 16 bit dma channel; if not specified the kernel will
+default to using channel 5 anyway which will cause audio not to work.
+"mpu_io" should be set to 0. The "io" parameter of the "opl3" module should
+also agree with the setting used by isapnp. To get the MPU401 interface
+working on an ALS-007 card, the "mpu401" module will be required since this
+card uses separate IRQs for the audio and MPU401 sections and there is no
+parameter available to pass a different IRQ to the "sb" driver (whose
+inbuilt MPU401 driver would otherwise be fine). Insert the mpu401 module
+passing appropriate values using the "io" and "irq" parameters.
+
+The resulting sound driver will provide the following capabilities:
+ - 8 and 16 bit audio playback
+ - 8 and 16 bit audio recording
+ - Software selection of record source (line in, CD, FM, mic, master)
+ - Record and playback of midi data via the external MPU-401
+ - Playback of midi data using inbuilt FM synthesizer
+ - Control of the ALS-007 mixer via any OSS-compatible mixer programs.
+ Controls available are Master (L&R), Line in (L&R), CD (L&R),
+ DSP/PCM/audio out (L&R), FM (L&R) and Mic in (mono).
+
+Jonathan Woithe
+jwoithe@physics.adelaide.edu.au
+30 March 1998
+
+Modified 2000-02-26 by Dave Forrest, drf5n@virginia.edu to add ALS100/ALS200
+Modified 2000-04-10 by Paul Laufer, pelaufer@csupomona.edu to add ISAPnP info.
+Modified 2000-11-19 by Jonathan Woithe, jwoithe@physics.adelaide.edu.au
+ - updated information for kernel 2.4.x.
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/AWE32 linux-2.5.6/Documentation/sound/oss/AWE32
--- linux-2.5.6-pre3/Documentation/sound/oss/AWE32 Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/AWE32 Thu Mar 7 18:24:41 2002
@@ -0,0 +1,76 @@
+ Installing and using Creative AWE midi sound under Linux.
+
+This documentation is devoted to the Creative Sound Blaster AWE32, AWE64 and
+SB32.
+
+1) Make sure you have an ORIGINAL Creative SB32, AWE32 or AWE64 card. This
+ is important, because the driver works only with real Creative cards.
+
+2) The first thing you need to do is re-compile your kernel with support for
+ your sound card. Run your favourite tool to configure the kernel and when
+ you get to the "Sound" menu you should enable support for the following:
+
+ Sound card support,
+ OSS sound modules,
+ 100% Sound Blaster compatibles (SB16/32/64, ESS, Jazz16) support,
+ AWE32 synth
+
+ If your card is "Plug and Play" you will also need to enable these two
+ options, found under the "Plug and Play configuration" menu:
+
+ Plug and Play support
+ ISA Plug and Play support
+
+ Now compile and install the kernel in normal fashion. If you don't know
+ how to do this you can find instructions for this in the README file
+ located in the root directory of the kernel source.
+
+3) Before you can start playing midi files you will have to load a sound
+ bank file. The utility needed for doing this is called "sfxload", and it
+ is one of the utilities found in a package called "awesfx". If this
+ package is not available in your distribution you can download the AWE
+ snapshot from Creative Labs Open Source website:
+
+ http://www.opensource.creative.com/snapshot.html
+
+ Once you have unpacked the AWE snapshot you will see a "awesfx"
+ directory. Follow the instructions in awesfx/docs/INSTALL to install the
+ utilities in this package. After doing this, sfxload should be installed
+ as:
+
+ /usr/local/bin/sfxload
+
+ To enable AWE general midi synthesis you should also get the sound bank
+ file for general midi from:
+
+ http://members.xoom.com/yar/synthgm.sbk.gz
+
+ Copy it to a directory of your choice, and unpack it there.
+
+4) Edit /etc/modules.conf, and insert the following lines at the end of the
+ file:
+
+ alias sound-slot-0 sb
+ alias sound-service-0-1 awe_wave
+ post-install awe_wave /usr/local/bin/sfxload PATH_TO_SOUND_BANK_FILE
+
+ You will of course have to change "PATH_TO_SOUND_BANK_FILE" to the full
+ path of of the sound bank file. That will enable the Sound Blaster and AWE
+ wave synthesis. To play midi files you should get one of these programs if
+ you don't already have them:
+
+ Playmidi: http://playmidi.openprojects.net
+
+ AWEMidi Player (drvmidi) Included in the previously mentioned AWE
+ snapshot.
+
+ You will probably have to pass the "-e" switch to playmidi to have it use
+ your midi device. drvmidi should work without switches.
+
+ If something goes wrong please e-mail me. All comments and suggestions are
+ welcome.
+
+ Yaroslav Rosomakho (alons55@dialup.ptt.ru)
+ http://www.yar.opennet.ru
+
+Last Updated: Feb 3 2001
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/AudioExcelDSP16 linux-2.5.6/Documentation/sound/oss/AudioExcelDSP16
--- linux-2.5.6-pre3/Documentation/sound/oss/AudioExcelDSP16 Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/AudioExcelDSP16 Thu Mar 7 18:24:41 2002
@@ -0,0 +1,101 @@
+Driver
+------
+
+Informations about Audio Excel DSP 16 driver can be found in the source
+file lowlevel/aedsp16.c
+Please, read the head of the source before using it. It contain useful
+informations.
+
+Configuration
+-------------
+
+The Audio Excel configuration, is now done with the standard Linux setup.
+You have to configure the sound card (Sound Blaster or Microsoft Sound System)
+and, if you want it, the Roland MPU-401 (do not use the Sound Blaster MPU-401,
+SB-MPU401) in the main driver menu. Activate the lowlevel drivers then select
+the Audio Excel hardware that you want to initialize. Check the IRQ/DMA/MIRQ
+of the Audio Excel initialization: it must be the same as the SBPRO (or MSS)
+setup. If the parameters are different, correct it.
+I you own a Gallant's audio card based on SC-6600, activate the SC-6600 support.
+If you want to change the configuration of the sound board, be sure to
+check off all the configuration items before re-configure it.
+
+Module parameters
+-----------------
+To use this driver as a module, you must configure some module parameters, to
+set up I/O addresses, IRQ lines and DMA channels. Some parameters are
+mandatory while some others are optional. Here a list of parameters you can
+use with this module:
+
+Name Description
+==== ===========
+MANDATORY
+io I/O base address (0x220 or 0x240)
+irq irq line (5, 7, 9, 10 or 11)
+dma dma channel (0, 1 or 3)
+
+OPTIONAL
+mss_base I/O base address for activate MSS mode (default SBPRO)
+ (0x530 or 0xE80)
+mpu_base I/O base address for activate MPU-401 mode
+ (0x300, 0x310, 0x320 or 0x330)
+mpu_irq MPU-401 irq line (5, 7, 9, 10 or 0)
+
+The /etc/modules.conf will have lines like this:
+
+options opl3 io=0x388
+options ad1848 io=0x530 irq=11 dma=3
+options aedsp16 io=0x220 irq=11 dma=3 mss_base=0x530
+
+Where the aedsp16 options are the options for this driver while opl3 and
+ad1848 are the corresponding options for the MSS and OPL3 modules.
+
+Loading MSS and OPL3 needs to pre load the aedsp16 module to set up correctly
+the sound card. Installation dependencies must be written in the modules.conf
+file:
+
+pre-install ad1848 modprobe aedsp16
+pre-install opl3 modprobe aedsp16
+
+Then you must load the sound modules stack in this order:
+sound -> aedsp16 -> [ ad1848, opl3 ]
+
+With the above configuration, loading ad1848 or opl3 modules, will
+automatically load all the sound stack.
+
+Sound cards supported
+---------------------
+This driver supports the SC-6000 and SC-6600 based Gallant's sound card.
+It don't support the Audio Excel DSP 16 III (try the SC-6600 code).
+I'm working on the III version of the card: if someone have useful
+informations about it, please let me know.
+For all the non-supported audio cards, you have to boot MS-DOS (or WIN95)
+activating the audio card with the MS-DOS device driver, then you have to
+-- and boot Linux.
+Follow these steps:
+
+1) Compile Linux kernel with standard sound driver, using the emulation
+ you want, with the parameters of your audio card,
+ e.g. Microsoft Sound System irq10 dma3
+2) Install your new kernel as the default boot kernel.
+3) Boot MS-DOS and configure the audio card with the boot time device
+ driver, for MSS irq10 dma3 in our example.
+4) -- and boot Linux. This will maintain the DOS configuration
+ and will boot the new kernel with sound driver. The sound driver will find
+ the audio card and will recognize and attach it.
+
+Reports on User successes
+-------------------------
+
+> Date: Mon, 29 Jul 1996 08:35:40 +0100
+> From: Mr S J Greenaway
+> To: riccardo@cdc8g5.cdc.polimi.it (Riccardo Facchetti)
+> Subject: Re: Audio Excel DSP 16 initialization code
+>
+> Just to let you know got my Audio Excel (emulating a MSS) working
+> with my original SB16, thanks for the driver!
+
+
+Last revised: 20 August 1998
+Riccardo Facchetti
+fizban@tin.it
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/CMI8330 linux-2.5.6/Documentation/sound/oss/CMI8330
--- linux-2.5.6-pre3/Documentation/sound/oss/CMI8330 Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/CMI8330 Thu Mar 7 18:24:41 2002
@@ -0,0 +1,153 @@
+Documentation for CMI 8330 (SoundPRO)
+-------------------------------------
+Alessandro Zummo
+
+( Be sure to read Documentation/sound/SoundPro too )
+
+
+This adapter is now directly supported by the sb driver.
+
+ The only thing you have to do is to compile the kernel sound
+support as a module and to enable kernel ISAPnP support,
+as shown below.
+
+
+CONFIG_SOUND=m
+CONFIG_SOUND_SB=m
+
+CONFIG_PNP=y
+CONFIG_ISAPNP=y
+
+
+and optionally:
+
+
+CONFIG_SOUND_MPU401=m
+
+ for MPU401 support.
+
+
+(I suggest you to use "make menuconfig" or "make xconfig"
+ for a more comfortable configuration editing)
+
+
+
+Then you can do
+
+ modprobe sb
+
+and everything will be (hopefully) configured.
+
+You should get something similar in syslog:
+
+sb: CMI8330 detected.
+sb: CMI8330 sb base located at 0x220
+sb: CMI8330 mpu base located at 0x330
+sb: CMI8330 mail reports to Alessandro Zummo
+sb: ISAPnP reports CMI 8330 SoundPRO at i/o 0x220, irq 7, dma 1,5
+
+
+
+
+The old documentation file follows for reference
+purposes.
+
+
+How to enable CMI 8330 (SOUNDPRO) soundchip on Linux
+------------------------------------------
+Stefan Laudat
+
+[Note: The CMI 8338 is unrelated and is supported by cmpci.o]
+
+
+ In order to use CMI8330 under Linux you just have to use a proper isapnp.conf, a good isapnp and a little bit of patience. I use isapnp 1.17, but
+you may get a better one I guess at http://www.roestock.demon.co.uk/isapnptools/.
+
+ Of course you will have to compile kernel sound support as module, as shown below:
+
+CONFIG_SOUND=m
+CONFIG_SOUND_OSS=m
+CONFIG_SOUND_SB=m
+CONFIG_SOUND_ADLIB=m
+CONFIG_SOUND_MPU401=m
+# Mikro$chaft sound system (kinda useful here ;))
+CONFIG_SOUND_MSS=m
+
+ The /etc/isapnp.conf file will be:
+
+
+
+
+(READPORT 0x0203)
+(ISOLATE PRESERVE)
+(IDENTIFY *)
+(VERBOSITY 2)
+(CONFLICT (IO FATAL)(IRQ FATAL)(DMA FATAL)(MEM FATAL)) # or WARNING
+(VERIFYLD N)
+
+
+# WSS
+
+(CONFIGURE CMI0001/16777472 (LD 0
+(IO 0 (SIZE 8) (BASE 0x0530))
+(IO 1 (SIZE 8) (BASE 0x0388))
+(INT 0 (IRQ 7 (MODE +E)))
+(DMA 0 (CHANNEL 0))
+(NAME "CMI0001/16777472[0]{CMI8330/C3D Audio Adapter}")
+(ACT Y)
+))
+
+# MPU
+
+(CONFIGURE CMI0001/16777472 (LD 1
+(IO 0 (SIZE 2) (BASE 0x0330))
+(INT 0 (IRQ 11 (MODE +E)))
+(NAME "CMI0001/16777472[1]{CMI8330/C3D Audio Adapter}")
+(ACT Y)
+))
+
+# Joystick
+
+(CONFIGURE CMI0001/16777472 (LD 2
+(IO 0 (SIZE 8) (BASE 0x0200))
+(NAME "CMI0001/16777472[2]{CMI8330/C3D Audio Adapter}")
+(ACT Y)
+))
+
+# SoundBlaster
+
+(CONFIGURE CMI0001/16777472 (LD 3
+(IO 0 (SIZE 16) (BASE 0x0220))
+(INT 0 (IRQ 5 (MODE +E)))
+(DMA 0 (CHANNEL 1))
+(DMA 1 (CHANNEL 5))
+(NAME "CMI0001/16777472[3]{CMI8330/C3D Audio Adapter}")
+(ACT Y)
+))
+
+
+(WAITFORKEY)
+
+
+
+ The module sequence is trivial:
+
+/sbin/insmod soundcore
+/sbin/insmod sound
+/sbin/insmod uart401
+# insert this first
+/sbin/insmod ad1848 io=0x530 irq=7 dma=0 soundpro=1
+# The sb module is an alternative to the ad1848 (Microsoft Sound System)
+# Anyhow, this is full duplex and has MIDI
+/sbin/insmod sb io=0x220 dma=1 dma16=5 irq=5 mpu_io=0x330
+
+
+
+Alma Chao suggests the following /etc/modules.conf:
+
+alias sound ad1848
+alias synth0 opl3
+options ad1848 io=0x530 irq=7 dma=0 soundpro=1
+options opl3 io=0x388
+
+
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/CMI8338 linux-2.5.6/Documentation/sound/oss/CMI8338
--- linux-2.5.6-pre3/Documentation/sound/oss/CMI8338 Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/CMI8338 Thu Mar 7 18:24:41 2002
@@ -0,0 +1,85 @@
+Audio driver for CM8338/CM8738 chips by Chen-Li Tien
+
+
+HARDWARE SUPPORTED
+================================================================================
+C-Media CMI8338
+C-Media CMI8738
+On-board C-Media chips
+
+
+STEPS TO BUILD DRIVER
+================================================================================
+
+ 1. Backup the Config.in and Makefile in the sound driver directory
+ (/usr/src/linux/driver/sound).
+ The Configure.help provide help when you config driver in step
+ 4, please backup the original one (/usr/src/linux/Document) and
+ copy this file.
+ The cmpci is document for the driver in detail, please copy it
+ to /usr/src/linux/Document/sound so you can refer it. Backup if
+ there is already one.
+
+ 2. Extract the tar file by 'tar xvzf cmpci-xx.tar.gz' in the above
+ directory.
+
+ 3. Change directory to /usr/src/linux
+
+ 4. Config cm8338 driver by 'make menuconfig', 'make config' or
+ 'make xconfig' command.
+
+ 5. Please select Sound Card (CONFIG_SOUND=m) support and CMPCI
+ driver (CONFIG_SOUND_CMPCI=m) as modules. Resident mode not tested.
+ For driver option, please refer 'DRIVER PARAMETER'
+
+ 6. Compile the kernel if necessary.
+
+ 7. Compile the modules by 'make modules'.
+
+ 8. Install the modules by 'make modules_install'
+
+
+INSTALL DRIVER
+================================================================================
+
+ 1. Before first time to run the driver, create module dependency by
+ 'depmod -a'
+
+ 2. To install the driver manually, enter 'modprobe cmpci'.
+
+ 3. Driver installation for various distributions:
+
+ a. Slackware 4.0
+ Add the 'modprobe cmpci' command in your /etc/rc.d/rc.modules
+ file.so you can start the driver automatically each time booting.
+
+ b. Caldera OpenLinux 2.2
+ Use LISA to load the cmpci module.
+
+ c. RedHat 6.0 and S.u.S.E. 6.1
+ Add following command in /etc/conf.modules:
+
+ alias sound cmpci
+
+ also visit http://www.cmedia.com.tw for installation instruction.
+
+DRIVER PARAMETER
+================================================================================
+
+ Some functions for the cm8738 can be configured in Kernel Configuration
+ or modules parameters. Set these parameters to 1 to enable.
+
+ mpuio: I/O ports base for MPU-401, 0 if disabled.
+ fmio: I/O ports base for OPL-3, 0 if disabled.
+ spdif_inverse:Inverse the S/PDIF-in signal, this depends on your
+ CD-ROM or DVD-ROM.
+ spdif_loop: Enable S/PDIF loop, this route S/PDIF-in to S/PDIF-out
+ directly.
+ speakers: Number of speakers used.
+ use_line_as_rear:Enable this if you want to use line-in as
+ rear-out.
+ use_line_as_bass:Enable this if you want to use line-in as
+ bass-out.
+ joystick: Enable joystick. You will need to install Linux joystick
+ driver.
+
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/CS4232 linux-2.5.6/Documentation/sound/oss/CS4232
--- linux-2.5.6-pre3/Documentation/sound/oss/CS4232 Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/CS4232 Thu Mar 7 18:24:41 2002
@@ -0,0 +1,23 @@
+To configure the Crystal CS423x sound chip and activate its DSP functions,
+modules may be loaded in this order:
+
+ modprobe sound
+ insmod ad1848
+ insmod uart401
+ insmod cs4232 io=* irq=* dma=* dma2=*
+
+This is the meaning of the parameters:
+
+ io--I/O address of the Windows Sound System (normally 0x534)
+ irq--IRQ of this device
+ dma and dma2--DMA channels (DMA2 may be 0)
+
+On some cards, the board attempts to do non-PnP setup, and fails. If you
+have problems, use Linux' PnP facilities.
+
+To get MIDI facilities add
+
+ insmod opl3 io=*
+
+where "io" is the I/O address of the OPL3 synthesizer. This will be shown
+in /proc/sys/pnp and is normally 0x388.
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/ChangeLog.awe linux-2.5.6/Documentation/sound/oss/ChangeLog.awe
--- linux-2.5.6-pre3/Documentation/sound/oss/ChangeLog.awe Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/ChangeLog.awe Thu Mar 7 18:24:41 2002
@@ -0,0 +1,230 @@
+ver.0.4.3p4
+ - Bug fix for invalid memory detection when initialized twice
+ - Add sample sharing function - works together with awesfx-0.4.3p3
+ - Add AWE_PROBE_DATA for probing sample id
+
+ver.0.4.3p3
+ - Replace memset to MEMSET (for FreeBSD)
+ - Add PAN_EXCHANGE switch
+
+ver.0.4.3p2
+ - MIDI emulation device is added
+ - Controls volume and filter targets
+ - Include chorus/reverb/equalizer values in MISC_MODE
+
+ver.0.4.3p1
+ - Change the volume calculation method
+ - Support for Tom Lees' PnP driver (v0.3)
+
+ver.0.4.2d
+ - Support for OSS/Free 3.8 on 2.0 kernels.
+ - Support for Linux PnP driver
+ - Support for module (for recent 2.1 kernels and RH5.0)
+ - Support for FreeBSD-3.0 system
+
+ver.0.4.2c
+ - Add a mode to enable drum channel toggle via bank number
+ change.
+
+ver.0.4.2b
+ - Clear voice position after note on
+ - Change nrvoices according to the current playing mode
+
+ver.0.4.2a
+ - Fix a bug in pitch calculation with scale parameter
+ - Change default chorus & reverb modes
+
+ver.0.4.2
+ - Use indirect voice allocation mode; used as default mode
+ - Add preset mapping
+ - Free buffers when resetting samples
+ - Set default preset/bank/drumset as variable
+ - Fix a bug in exclusive note-off
+ - Add channel reset control macro
+ - Change modwheel sensitivity as variable
+ - Add lock option in open_patch
+ - Add channel priority mode macro, and disable it as default
+ - Add unset effect macro
+ - Add user defined chorus/reverb modes
+ - Do not initialize effect parameters when allocating voices
+ - Accept realtime filter-Q parameter change
+ - Check value range of set/add effects
+ - Change drum flags automatically when receiving bank #128
+
+ver.0.4.1 development versions
+
+ver.0.4.0c
+ - Fix kernel oops when setting AWE_FX_ATTEN
+
+ver.0.4.0b
+ - Do not kill_note in start_note when velocity is zero
+
+ver.0.4.0a
+ - Fix a bug in channel pressure effects
+
+ver.0.4.0
+ - Support dynamic buffer allocation
+ - Add functions to open/close/unload a patch
+ - Change from pointer to integer index in voice/sample lists
+ - Support for Linux/Alpha-AXP
+ - Fix for FreeBSD
+ - Add sostenuto control
+ - Add midi channel priority
+ - Fix a bug in all notes off control
+ - Use AWE_DEFAULT_MEMSIZE always if defined
+ - Fix a bug in awe_reset causes seg fault when no DRAM onboard
+ - Use awe_mem_start variable instead of constant
+
+ver.0.3.3c
+ - Fix IOCTL_TO_USER for OSS-3.8 (on Linux-2.1.25)
+ - Fix i/o macros for mixer controls
+
+ver.0.3.3b
+ - Fix version number in awe_version.h
+ - Fix a small bug in noteoff/release all
+
+ver.0.3.3a
+ - Fix all notes/sounds off
+ - Add layer effect control
+ - Add misc mode controls; realtime pan, version number, etc.
+ - Move gus bank control in misc mode control
+ - Modify awe_operations for OSS3.8b5
+ - Fix installation script
+
+ver.0.3.3
+ - Add bass/treble control in Emu8000 chip
+ - Add mixer device
+ - Fix sustain on to value 127
+
+ver.0.3.2
+ - Refuse linux-2.0.0 at installation
+ - Move awe_voice.h to /usr/include/linux
+
+ver.0.3.1b (not released)
+ - Rewrite chorus/reverb mode change functions
+ - Rewrite awe_detect & awe_check_dram routines
+
+ver.0.3.1a
+ - Fix a bug to reset voice counter in awe_reset
+ - Fix voice balance on GUS mode
+ - Make symlink on /usr/include/asm in install script
+
+ver.0.3.1
+ - Remove zero size arrays from awe_voice.h
+ - Fix init_fm routine
+ - Remove all samples except primary samples in REMOVE_LAST_SAMPLES
+
+ver.0.3.0a
+ - Add AWE_NOTEOFF_ALL control
+ - Remove AWE_INIT_ATTEN control
+
+ver.0.3.0
+ - Fix decay time table
+ - Add exclusive sounds mode
+ - Add capability to get current status
+
+ver.0.2.99e
+ - Add #ifdef for all sounds/notes off controls.
+ - Fix bugs on searching the default drumset/preset.
+ - Fix usslite patch to modify the default Config.in.
+
+ver.0.2.99d
+ - Fix bugs of attack/hold parameters
+ - Fix attack & decay time table
+
+ver.0.2.99c
+ - Change volume control messages (main & expression volume)
+ to accesspt normal MIDI parameters in channel mode.
+ - Use channel mode in SEQ2 controls.
+
+ver.0.2.99b
+ - #ifdef patch manager functions (for OSS-3.7)
+
+ver.0.2.99a
+ - Fix sustain bug
+
+ver.0.2.99 (0.3 beta)
+ - Support multiple instruments
+
+ver.0.2.0c
+ - Add copyright notice
+ - FreeBSD 2.2-ALPHA integration
+
+ver.0.2.0b
+ - Remove buffered reading appended in v0.2.0a
+ - Remove SMAxW register check on writing
+ - Support Linux 2.1.x kernel
+ - Rewrite installation script
+
+ver.0.2.0a
+ - Define SEQUENCER_C for tuning.h for FreeBSD system
+ - Improvement of sample loading speed
+ - Fix installation script
+ - Add PnP driver functions for ISA PnP driver support
+
+ver.0.2.0
+ - Includes FreeBSD port
+ - Can load GUS compatible patches
+ - Change values of hardware control parameters for compatibility
+ with GUS driver
+ - Accept 8bit or unsigned wave data
+ - Accept no blank loop data
+ - Add sample mode flags in sample_info
+
+ver.0.1.6
+ - Add voice effects control
+ - Fix awe_voice.h for word alignment
+
+ver.0.1.5c
+ - Fix FM(OPL) playback problem
+
+ver.0.1.5b
+ - Fix pitch calculation for fixed midi key
+
+ver.0.1.5a
+ - Fix bugs in removing samples from linked list.
+
+ver.0.1.5
+ - Add checksum verification for sample uploading
+ (not compatible from older sample_info structure)
+ - Fix sample offset pointers to (actual value - 1)
+ - Add sequencer command to initialize awe32
+
+ver.0.1.4c
+ - Fix card detection and memory check function to avoid system crash
+ at booting
+
+ver.0.1.4b
+ - Add release sustain mode
+ - Initialize FM each time after loading samples
+
+ver.0.1.4a
+ - Fix AWE card detection code
+ - Correct FM initialize position
+ - Add non-releasing mode on voice info
+
+ver.0.1.4
+ - Add AWE card and DRAM detection codes
+ - Add FM initialization code
+ - Modify volume control
+ - Remove linear volume mode
+ - Change memory management; not using malloc dynamically
+ - Add remove-samples command
+ - Use internal id implicitly at loading samples
+
+ver.0.1.3
+ - Fix a bug on patch uploading to RAM
+
+ver.0.1.2
+ - Divide to separated packages
+ - Fix disagreed macro conditions
+ - Fix unresolved function bugs
+ - Integrate VoxWare and USS-Lite driver source (awe_voice.c)
+ and remove awe_card.c
+
+ver.0.1.1
+ - Fix wrong sample numbers in sbktext
+ - Fix txt2sfx bug
+ - Fix pan parameter calculation
+ - Append USS-Lite/Linux2.0 driver
+
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/ChangeLog.multisound linux-2.5.6/Documentation/sound/oss/ChangeLog.multisound
--- linux-2.5.6-pre3/Documentation/sound/oss/ChangeLog.multisound Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/ChangeLog.multisound Thu Mar 7 18:24:41 2002
@@ -0,0 +1,213 @@
+1998-12-04 Andrew T. Veliath
+
+ * Update version to 0.8.2.2
+
+ * Add msndreset program to shell archive.
+
+1998-11-11 Andrew T. Veliath
+
+ * msnd_pinnacle.c (mixer_ioctl): Add a mixer ioctl for
+ SOUND_MIXER_PRIVATE1 which does a full reset on the card.
+ (mixer_set): Move line in recording source to input monitor, aux
+ input level added, some mixer fixes.
+
+1998-09-10 Andrew Veliath
+
+ * Update version to 0.8.2
+
+ * Add SNDCTL_DSP_GETOSPACE and SNDCTL_DSP_GETISPACE ioctls.
+
+1998-09-09 Andrew Veliath
+
+ * Update version to 0.8.1
+
+ * msnd_pinnacle.c: Fix resetting of default audio parameters. Turn
+ flush code from dsp_halt into dsp_write_flush, and use that for
+ SNDCTL_DSP_SYNC.
+
+1998-09-07 Andrew Veliath
+
+ * Update version to 0.8.0
+
+ * Provide separate signal parameters for play and record.
+
+ * Cleanups to locking and interrupt handling, change default
+ fifosize to 128kB.
+
+ * Update version to 0.7.15
+
+ * Interprocess full-duplex support (ie `cat /dev/dsp > /dev/dsp').
+
+ * More mutex sections for read and write fifos (read + write locks
+ added).
+
+1998-09-05 Andrew Veliath
+
+ * msnd_pinnacle.c: (chk_send_dsp_cmd) Do full DSP reset upon DSP
+ timeout (when not in interrupt; maintains mixer settings). Fixes
+ to flushing and IRQ ref counting. Rewrote queuing for smoother
+ playback and fixed initial playback cutoff problem.
+
+1998-09-03 Andrew Veliath
+
+ * Replaced packed structure accesses with standard C equivalents.
+
+1998-09-01 Andrew Veliath
+
+ * msnd_pinnacle.c: Add non-PnP configuration to driver code, which
+ will facilitate compiled-in operation.
+
+1998-08-29 Andrew Veliath
+
+ * Update version to 0.7.6
+
+ * msnd_pinnacle.c (dsp_ioctl): Add DSP_GETFMTS, change SAMPLESIZE
+ to DSP_SETFMT.
+
+ * Update version to 0.7.5
+
+ * Create pinnaclecfg.c and turn MultiSound doc into a shell
+ archive with pinnaclecfg.c included. pinnaclecfg.c can
+ now fully configure the card in non-PnP mode, including the
+ joystick and IDE controller. Also add an isapnp conf
+ example.
+
+ * Reduce DSP reset timeout from 20000 to 100
+
+1998-08-06 Andrew Veliath
+
+ * Update version to 0.7.2
+
+ * After A/D calibration, do an explicit set to the line input,
+ rather than using set_recsrc
+
+1998-07-20 Andrew Veliath
+
+ * Update version to 0.7.1
+
+ * Add more OSS ioctls
+
+1998-07-19 Andrew Veliath
+
+ * Update doc file
+
+ * Bring back DIGITAL1 with digital parameter to msnd_pinnacle.c
+ and CONFIG_MSNDPIN_DIGITAL. I'm not sure this actually works,
+ since I find audio playback goes into a very speeded mode of
+ operation, however it might be due to a lack of a digital
+ source, which I don't have to test.
+
+1998-07-18 Andrew Veliath
+
+ * Update version to 0.7.0
+
+ * Can now compile with Alan Cox' 2.0.34-modular-sound patch (so
+ now it requires >= 2.1.106 or 2.0.34-ms) (note for 2.0.34-ms it
+ is in the Experimental section)
+
+ * More modularization, consolidation, also some MIDI hooks
+ installed for future MIDI modules
+
+ * Write flush
+
+ * Change default speed, channels, bit size to OSS/Free defaults
+
+1998-06-02 Andrew Veliath
+
+ * Update version to 0.5b
+
+ * Fix version detection
+
+ * Remove underflow and overflow resets (delay was too long)
+
+ * Replace spinlocked bitops with atomic bit ops
+
+1998-05-27 Andrew Veliath
+
+ * Update version to 0.5a
+
+ * Better recovery from underflow or overflow conditions
+
+ * Fix a deadlock condition with one thread reading and the other
+ writing
+
+1998-05-26 Andrew Veliath
+
+ * Update version to 0.5
+
+ * Separate reset queue functions for play and record
+
+ * Add delays in dsp_halt
+
+1998-05-24 Andrew Veliath
+
+ * Add a check for Linux >= 2.1.95
+
+ * Remove DIGITAL1 input until I figure out how to make it work
+
+ * Add HAVE_DSPCODEH which when not defined will load firmware from
+ files using mod_firmware_load, then release memory after they
+ are uploaded (requires reorganized OSS).
+
+1998-05-22 Andrew Veliath
+
+ * Update version to 0.4c
+
+ * Hopefully fix the mixer volume problem
+
+1998-05-19 Andrew Veliath
+
+ * Add __initfuncs and __initdatas to reduce resident code size
+
+ * Move bunch of code around, remove some protos
+
+ * Integrate preliminary changes for Alan Cox's OSS reorganization
+ for non-OSS drivers to coexist with OSS devices on the same
+ major. To compile standalone, must now define STANDALONE.
+
+1998-05-16 Andrew Veliath
+
+ * Update version to 0.4b
+
+ * Integrated older card support into a unified driver, tested on a
+ MultiSound Classic c/o Kendrick Vargas.
+
+1998-05-15 Andrew Veliath
+
+ * Update version to 0.4
+
+ * Fix read/write return values
+
+1998-05-13 Andrew Veliath
+
+ * Update version to 0.3
+
+ * Stop play gracefully
+
+ * Add busy flag
+
+ * Add major and calibrate_signal module parameters
+
+ * Add ADC calibration
+
+ * Add some OSS compatibility ioctls
+
+ * Add mixer record selection
+
+ * Add O_NONBLOCK support, separate read/write wait queues
+
+ * Add sample bit size ioctl, expanded sample rate ioctl
+
+ * Playback suspension now resumes
+
+ * Use signal_pending after interruptible_sleep_on
+
+ * Add recording, change ints to bit flags
+
+1998-05-11 Andrew Veliath
+
+ * Update version to 0.2
+
+ * Add preliminary playback support
+
+ * Use new Turtle Beach DSP code
\ No newline at end of file
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/ESS linux-2.5.6/Documentation/sound/oss/ESS
--- linux-2.5.6-pre3/Documentation/sound/oss/ESS Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/ESS Thu Mar 7 18:24:41 2002
@@ -0,0 +1,34 @@
+Documentation for the ESS AudioDrive chips
+
+In 2.4 kernels the SoundBlaster driver not only tries to detect an ESS chip, it
+tries to detect the type of ESS chip too. The correct detection of the chip
+doesn't always succeed however, so unless you use the kernel isapnp facilities
+(and you chip is pnp capable) the default behaviour is 2.0 behaviour which
+means: only detect ES688 and ES1688.
+
+All ESS chips now have a recording level setting. This is a need-to-have for
+people who want to use their ESS for recording sound.
+
+Every chip that's detected as a later-than-es1688 chip has a 6 bits logarithmic
+master volume control.
+
+Every chip that's detected as a ES1887 now has Full Duplex support. Made a
+little testprogram that shows that is works, haven't seen a real program that
+needs this however.
+
+For ESS chips an additional parameter "esstype" can be specified. This controls
+the (auto) detection of the ESS chips. It can have 3 kinds of values:
+
+-1 Act like 2.0 kernels: only detect ES688 or ES1688.
+0 Try to auto-detect the chip (may fail for ES1688)
+688 The chip will be treated as ES688
+1688 ,, ,, ,, ,, ,, ,, ES1688
+1868 ,, ,, ,, ,, ,, ,, ES1868
+1869 ,, ,, ,, ,, ,, ,, ES1869
+1788 ,, ,, ,, ,, ,, ,, ES1788
+1887 ,, ,, ,, ,, ,, ,, ES1887
+1888 ,, ,, ,, ,, ,, ,, ES1888
+
+Because Full Duplex is supported for ES1887 you can specify a second DMA
+channel by specifying module parameter dma16. It can be one of: 0, 1, 3 or 5.
+
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/ESS1868 linux-2.5.6/Documentation/sound/oss/ESS1868
--- linux-2.5.6-pre3/Documentation/sound/oss/ESS1868 Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/ESS1868 Thu Mar 7 18:24:41 2002
@@ -0,0 +1,55 @@
+Documentation for the ESS1868F AudioDrive PnP sound card
+
+The ESS1868 sound card is a PnP ESS1688-compatible 16-bit sound card.
+
+It should be automatically detected by the Linux Kernel isapnp support when you
+load the sb.o module. Otherwise you should take care of:
+
+ * The ESS1868 does not allow use of a 16-bit DMA, thus DMA 0, 1, 2, and 3
+ may only be used.
+
+ * isapnptools version 1.14 does work with ESS1868. Earlier versions might
+ not.
+
+ * Sound support MUST be compiled as MODULES, not statically linked
+ into the kernel.
+
+
+NOTE: this is only needed when not using the kernel isapnp support!
+
+For configuring the sound card's I/O addresses, IRQ and DMA, here is a
+sample copy of the isapnp.conf directives regarding the ESS1868:
+
+(CONFIGURE ESS1868/-1 (LD 1
+(IO 0 (BASE 0x0220))
+(IO 1 (BASE 0x0388))
+(IO 2 (BASE 0x0330))
+(DMA 0 (CHANNEL 1))
+(INT 0 (IRQ 5 (MODE +E)))
+(ACT Y)
+))
+
+(for a full working isapnp.conf file, remember the
+(ISOLATE)
+(IDENTIFY *)
+at the beginning and the
+(WAITFORKEY)
+at the end.)
+
+In this setup, the main card I/O is 0x0220, FM synthesizer is 0x0388, and
+the MPU-401 MIDI port is located at 0x0330. IRQ is IRQ 5, DMA is channel 1.
+
+After configuring the sound card via isapnp, to use the card you must load
+the sound modules with the proper I/O information. Here is my setup:
+
+# ESS1868F AudioDrive initialization
+
+/sbin/modprobe sound
+/sbin/insmod uart401
+/sbin/insmod sb io=0x220 irq=5 dma=1 dma16=-1
+/sbin/insmod mpu401 io=0x330
+/sbin/insmod opl3 io=0x388
+/sbin/insmod v_midi
+
+opl3 is the FM synthesizer
+/sbin/insmod opl3 io=0x388
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/INSTALL.awe linux-2.5.6/Documentation/sound/oss/INSTALL.awe
--- linux-2.5.6-pre3/Documentation/sound/oss/INSTALL.awe Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/INSTALL.awe Thu Mar 7 18:24:41 2002
@@ -0,0 +1,134 @@
+================================================================
+ INSTALLATION OF AWE32 SOUND DRIVER FOR LINUX
+ Takashi Iwai
+================================================================
+
+----------------------------------------------------------------
+* Attention to SB-PnP Card Users
+
+If you're using PnP cards, the initialization of PnP is required
+before loading this driver. You have now three options:
+ 1. Use isapnptools.
+ 2. Use in-kernel isapnp support.
+ 3. Initialize PnP on DOS/Windows, then boot linux by loadlin.
+In this document, only the case 1 case is treated.
+
+----------------------------------------------------------------
+* Installation on Red Hat 5.0 Sound Driver
+
+Please use install-rh.sh under RedHat5.0 directory.
+DO NOT USE install.sh below.
+See INSTALL.RH for more details.
+
+----------------------------------------------------------------
+* Installation/Update by Shell Script
+
+ 1. Become root
+
+ % su
+
+ 2. If you have never configured the kernel tree yet, run make config
+ once (to make dependencies and symlinks).
+
+ # cd /usr/src/linux
+ # make xconfig
+
+ 3. Run install.sh script
+
+ # sh ./install.sh
+
+ 4. Configure your kernel
+
+ (for Linux 2.[01].x user)
+ # cd /usr/src/linux
+ # make xconfig (or make menuconfig)
+
+ (for Linux 1.2.x user)
+ # cd /usr/src/linux
+ # make config
+
+ Answer YES to both "lowlevel drivers" and "AWE32 wave synth" items
+ in Sound menu. ("lowlevel drivers" will appear only in 2.x
+ kernel.)
+
+ 5. Make your kernel (and modules), and install them as usual.
+
+ 5a. make kernel image
+ # make zImage
+
+ 5b. make modules and install them
+ # make modules && make modules_install
+
+ 5c. If you're using lilo, copy the kernel image and run lilo.
+ Otherwise, copy the kernel image to suitable directory or
+ media for your system.
+
+ 6. Reboot the kernel if necessary.
+ - If you updated only the modules, you don't have to reboot
+ the system. Just remove the old sound modules here.
+ in
+ # rmmod sound.o (linux-2.0 or OSS/Free)
+ # rmmod awe_wave.o (linux-2.1)
+
+ 7. If your AWE card is a PnP and not initialized yet, you'll have to
+ do it by isapnp tools. Otherwise, skip to 8.
+
+ This section described only a brief explanation. For more
+ details, please see the AWE64-Mini-HOWTO or isapnp tools FAQ.
+
+ 7a. If you have no isapnp.conf file, generate it by pnpdump.
+ Otherwise, skip to 7d.
+ # pnpdump > /etc/isapnp.conf
+
+ 7b. Edit isapnp.conf file. Comment out the appropriate
+ lines containing desirable I/O ports, DMA and IRQs.
+ Don't forget to enable (ACT Y) line.
+
+ 7c. Add two i/o ports (0xA20 and 0xE20) in WaveTable part.
+ ex)
+ (CONFIGURE CTL0048/58128 (LD 2
+ # ANSI string -->WaveTable<--
+ (IO 0 (BASE 0x0620))
+ (IO 1 (BASE 0x0A20))
+ (IO 2 (BASE 0x0E20))
+ (ACT Y)
+ ))
+
+ 7d. Load the config file.
+ CAUTION: This will reset all PnP cards!
+
+ # isapnp /etc/isapnp.conf
+
+ 8. Load the sound module (if you configured it as a module):
+
+ for 2.0 kernel or OSS/Free monolithic module:
+
+ # modprobe sound.o
+
+ for 2.1 kernel:
+
+ # modprobe sound
+ # insmod uart401
+ # insmod sb io=0x220 irq=5 dma=1 dma16=5 mpu_io=0x330
+ (These values depend on your settings.)
+ # insmod awe_wave
+ (Be sure to load awe_wave after sb!)
+
+ See /usr/src/linux/Documentation/sound/AWE32 for
+ more details.
+
+ 9. (only for obsolete systems) If you don't have /dev/sequencer
+ device file, make it according to Readme.linux file on
+ /usr/src/linux/drivers/sound. (Run a shell script included in
+ that file). <-- This file no longer exists in the recent kernels!
+
+ 10. OK, load your own soundfont file, and enjoy MIDI!
+
+ % sfxload synthgm.sbk
+ % drvmidi foo.mid
+
+ 11. For more advanced use (eg. dynamic loading, virtual bank and
+ etc.), please read the awedrv FAQ or the instructions in awesfx
+ and awemidi packages.
+
+Good luck!
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/Introduction linux-2.5.6/Documentation/sound/oss/Introduction
--- linux-2.5.6-pre3/Documentation/sound/oss/Introduction Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/Introduction Thu Mar 7 18:24:41 2002
@@ -0,0 +1,461 @@
+Introduction Notes on Modular Sound Drivers and Soundcore
+Wade Hampton
+2/14/2001
+
+Purpose:
+========
+This document provides some general notes on the modular
+sound drivers and their configuration, along with the
+support modules sound.o and soundcore.o.
+
+Note, some of this probably should be added to the Sound-HOWTO!
+
+Note, soundlow.o was present with 2.2 kernels but is not
+required for 2.4.x kernels. References have been removed
+to this.
+
+
+Copying:
+========
+none
+
+
+History:
+========
+0.1.0 11/20/1998 First version, draft
+1.0.0 11/1998 Alan Cox changes, incorporation in 2.2.0
+ as /usr/src/linux/Documentation/sound/Introduction
+1.1.0 6/30/1999 Second version, added notes on making the drivers,
+ added info on multiple sound cards of similar types,]
+ added more diagnostics info, added info about esd.
+ added info on OSS and ALSA.
+1.1.1 19991031 Added notes on sound-slot- and sound-service.
+ (Alan Cox)
+1.1.2 20000920 Modified for Kernel 2.4 (Christoph Hellwig)
+1.1.3 20010214 Minor notes and corrections (Wade Hampton)
+ Added examples of sound-slot-0, etc.
+
+
+Modular Sound Drivers:
+======================
+
+Thanks to the GREAT work by Alan Cox (alan@lxorguk.ukuu.org.uk),
+
+[And Oleg Drokin, Thomas Sailer, Andrew Veliath and more than a few
+ others - not to mention Hannu's original code being designed well
+ enough to cope with that kind of chopping up](Alan)
+
+the standard Linux kernels support a modular sound driver. From
+Alan's comments in linux/drivers/sound/README.FIRST:
+
+ The modular sound driver patches were funded by Red Hat Software
+ (www.redhat.com). The sound driver here is thus a modified version of
+ Hannu's code. Please bear that in mind when considering the appropriate
+ forums for bug reporting.
+
+The modular sound drivers may be loaded via insmod or modprobe.
+To support all the various sound modules, there are two general
+support modules that must be loaded first:
+
+ soundcore.o: Top level handler for the sound system, provides
+ a set of functions for registration of devices
+ by type.
+
+ sound.o: Common sound functions required by all modules.
+
+For the specific sound modules (e.g., sb.o for the Soundblaster),
+read the documentation on that module to determine what options
+are available, for example IRQ, address, DMA.
+
+Warning, the options for different cards sometime use different names
+for the same or a similar feature (dma1= versus dma16=). As a last
+resort, inspect the code (search for MODULE_PARM).
+
+Notes:
+
+1. There is a new OpenSource sound driver called ALSA which is
+ currently under development: http://www.alsa-project.org/
+ The ALSA drivers support some newer hardware that may not
+ be supported by this sound driver and also provide some
+ additional features.
+
+2. The commercial OSS driver may be obtained from the site:
+ http://www/opensound.com. This may be used for cards that
+ are unsupported by the kernel driver, or may be used
+ by other operating systems.
+
+3. The enlightenment sound daemon may be used for playing
+ multiple sounds at the same time via a single card, eliminating
+ some of the requirements for multiple sound card systems. For
+ more information, see: http://www.tux.org/~ricdude/EsounD.html
+ The "esd" program may be used with the real-player and mpeg
+ players like mpg123 and x11amp. The newer real-player
+ and some games even include built-in support for ESD!
+
+
+Building the Modules:
+=====================
+
+This document does not provide full details on building the
+kernel, etc. The notes below apply only to making the kernel
+sound modules. If this conflicts with the kernel's README,
+the README takes precedence.
+
+1. To make the kernel sound modules, cd to your /usr/src/linux
+ directory (typically) and type make config, make menuconfig,
+ or make xconfig (to start the command line, dialog, or x-based
+ configuration tool).
+
+2. Select the Sound option and a dialog will be displayed.
+
+3. Select M (module) for "Sound card support".
+
+4. Select your sound driver(s) as a module. For ProAudio, Sound
+ Blaster, etc., select M (module) for OSS sound modules.
+ [thanks to Marvin Stodolsky ]A
+
+5. Make the kernel (e.g., make dep ; make bzImage), and install
+ the kernel.
+
+6. Make the modules and install them (make modules; make modules_install).
+
+Note, for 2.4.x kernels, make sure you have the newer modutils
+loaded or modules will not be loaded properly. 2.4.x changed the
+layout of /lib/modules/2.4.x and requires an updated modutils.
+
+
+Plug and Play (PnP:
+===================
+
+If the sound card is an ISA PnP card, isapnp may be used
+to configure the card. See the file isapnp.txt in the
+directory one level up (e.g., /usr/src/linux/Documentation).
+
+Also the 2.4.x kernels provide PnP capabilities, see the
+file NEWS in this directory.
+
+PCI sound cards are highly recommended, as they are far
+easier to configure and from what I have read, they use
+less resources and are more CPU efficient.
+
+
+INSMOD:
+=======
+
+If loading via insmod, the common modules must be loaded in the
+order below BEFORE loading the other sound modules. The card-specific
+modules may then be loaded (most require parameters). For example,
+I use the following via a shell script to load my SoundBlaster:
+
+SB_BASE=0x240
+SB_IRQ=9
+SB_DMA=3
+SB_DMA2=5
+SB_MPU=0x300
+#
+echo Starting sound
+/sbin/insmod soundcore
+/sbin/insmod sound
+#
+echo Starting sound blaster....
+/sbin/insmod uart401
+/sbin/insmod sb io=$SB_BASE irq=$SB_IRQ dma=$SB_DMA dma16=$SB_DMA2 mpu_io=$SB_MP
+
+When using sound as a module, I typically put these commands
+in a file such as /root/soundon.sh.
+
+
+MODPROBE:
+=========
+
+If loading via modprobe, these common files are automatically loaded
+when requested by modprobe. For example, my /etc/modules.conf contains:
+
+alias sound sb
+options sb io=0x240 irq=9 dma=3 dma16=5 mpu_io=0x300
+
+All you need to do to load the module is:
+
+ /sbin/modprobe sb
+
+
+Sound Status:
+=============
+
+The status of sound may be read/checked by:
+ cat (anyfile).au >/dev/audio
+
+[WWH: This may not work properly for SoundBlaster PCI 128 cards
+such as the es1370/1 (see the es1370/1 files in this directory)
+as they do not automatically support uLaw on /dev/audio.]
+
+The status of the modules and which modules depend on
+which other modules may be checked by:
+ /sbin/lsmod
+
+/sbin/lsmod should show something like the following:
+ sb 26280 0
+ uart401 5640 0 [sb]
+ sound 57112 0 [sb uart401]
+ soundcore 1968 8 [sb sound]
+
+
+Removing Sound:
+===============
+
+Sound may be removed by using /sbin/rmmod in the reverse order
+in which you load the modules. Note, if a program has a sound device
+open (e.g., xmixer), that module (and the modules on which it
+depends) may not be unloaded.
+
+For example, I use the following to remove my Soundblaster (rmmod
+in the reverse order in which I loaded the modules):
+
+/sbin/rmmod sb
+/sbin/rmmod uart401
+/sbin/rmmod sound
+/sbin/rmmod soundcore
+
+When using sound as a module, I typically put these commands
+in a script such as /root/soundoff.sh.
+
+
+Removing Sound for use with OSS:
+================================
+
+If you get really stuck or have a card that the kernel modules
+will not support, you can get a commercial sound driver from
+http://www.opensound.com. Before loading the commercial sound
+driver, you should do the following:
+
+1. remove sound modules (detailed above)
+2. remove the sound modules from /etc/modules.conf
+3. move the sound modules from /lib/modules//misc
+ (for example, I make a /lib/modules//misc/tmp
+ directory and copy the sound module files to that
+ directory).
+
+
+Multiple Sound Cards:
+=====================
+
+The sound drivers will support multiple sound cards and there
+are some great applications like multitrack that support them.
+Typically, you need two sound cards of different types. Note, this
+uses more precious interrupts and DMA channels and sometimes
+can be a configuration nightmare. I have heard reports of 3-4
+sound cards (typically I only use 2). You can sometimes use
+multiple PCI sound cards of the same type.
+
+On my machine I have two sound cards (cs4232 and Soundblaster Vibra
+16). By loading sound as modules, I can control which is the first
+sound device (/dev/dsp, /dev/audio, /dev/mixer) and which is
+the second. Normally, the cs4232 (Dell sound on the motherboard)
+would be the first sound device, but I prefer the Soundblaster.
+All you have to do is to load the one you want as /dev/dsp
+first (in my case "sb") and then load the other one
+(in my case "cs4232").
+
+If you have two cards of the same type that are jumpered
+cards or different PnP revisions, you may load the same
+module twice. For example, I have a SoundBlaster vibra 16
+and an older SoundBlaster 16 (jumpers). To load the module
+twice, you need to do the following:
+
+1. Copy the sound modules to a new name. For example
+ sb.o could be copied (or symlinked) to sb1.o for the
+ second SoundBlaster.
+
+2. Make a second entry in /etc/modules.conf, for example,
+ sound1 or sb1. This second entry should refer to the
+ new module names for example sb1, and should include
+ the I/O, etc. for the second sound card.
+
+3. Update your soundon.sh script, etc.
+
+Warning: I have never been able to get two PnP sound cards of the
+same type to load at the same time. I have tried this several times
+with the Soundblaster Vibra 16 cards. OSS has indicated that this
+is a PnP problem.... If anyone has any luck doing this, please
+send me an E-MAIL. PCI sound cards should not have this problem.a
+Since this was originally release, I have received a couple of
+mails from people who have accomplished this!
+
+NOTE: In Linux 2.4 the Sound Blaster driver (and only this one yet)
+supports multiple cards with one module by default.
+Read the file 'Soundblaster' in this directory for details.
+
+
+Sound Problems:
+===============
+
+First RTFM (including the troubleshooting section
+in the Sound-HOWTO).
+
+1) If you are having problems loading the modules (for
+ example, if you get device conflict errors) try the
+ following:
+
+ A) If you have Win95 or NT on the same computer,
+ write down what addresses, IRQ, and DMA channels
+ those were using for the same hardware. You probably
+ can use these addresses, IRQs, and DMA channels.
+ You should really do this BEFORE attempting to get
+ sound working!
+
+ B) Check (cat) /proc/interrupts, /proc/ioports,
+ and /proc/dma. Are you trying to use an address,
+ IRQ or DMA port that another device is using?
+
+ C) Check (cat) /proc/isapnp
+
+ D) Inspect your /var/log/messages file. Often that will
+ indicate what IRQ or IO port could not be obtained.
+
+ E) Try another port or IRQ. Note this may involve
+ using the PnP tools to move the sound card to
+ another location. Sometimes this is the only way
+ and it is more or less trial and error.
+
+2) If you get motor-boating (the same sound or part of a
+ sound clip repeated), you probably have either an IRQ
+ or DMA conflict. Move the card to another IRQ or DMA
+ port. This has happened to me when playing long files
+ when I had an IRQ conflict.
+
+3. If you get dropouts or pauses when playing high sample
+ rate files such as using mpg123 or x11amp/xmms, you may
+ have too slow of a CPU and may have to use the options to
+ play the files at 1/2 speed. For example, you may use
+ the -2 or -4 option on mpg123. You may also get this
+ when trying to play mpeg files stored on a CD-ROM
+ (my Toshiba T8000 PII/366 sometimes has this problem).
+
+4. If you get "cannot access device" errors, your /dev/dsp
+ files, etc. may be set to owner root, mode 600. You
+ may have to use the command:
+ chmod 666 /dev/dsp /dev/mixer /dev/audio
+
+5. If you get "device busy" errors, another program has the
+ sound device open. For example, if using the Enlightenment
+ sound daemon "esd", the "esd" program has the sound device.
+ If using "esd", please RTFM the docs on ESD. For example,
+ esddsp may be used to play files via a non-esd
+ aware program.
+
+6) Ask for help on the sound list or send E-MAIL to the
+ sound driver author/maintainer.
+
+7) Turn on debug in drivers/sound/sound_config.h (DEB, DDB, MDB).
+
+8) If the system reports insufficient DMA memory then you may want to
+ load sound with the "dmabufs=1" option. Or in /etc/conf.modules add
+
+ preinstall sound dmabufs=1
+
+ This makes the sound system allocate its buffers and hang onto them.
+
+ You may also set persistent DMA when building a 2.4.x kernel.
+
+
+Configuring Sound:
+==================
+
+There are several ways of configuring your sound:
+
+1) On the kernel command line (when using the sound driver(s)
+ compiled in the kernel). Check the driver source and
+ documentation for details.
+
+2) On the command line when using insmod or in a bash script
+ using command line calls to load sound.
+
+3) In /etc/modules.conf when using modprobe.
+
+4) Via Red Hat's GPL'd /usr/sbin/sndconfig program (text based).
+
+5) Via the OSS soundconf program (with the commercial version
+ of the OSS driver.
+
+6) By just loading the module and let isapnp do everything relevant
+ for you. This works only with a few drivers yet and - of course -
+ only with isapnp hardware.
+
+And I am sure, several other ways.
+
+Anyone want to write a linuxconf module for configuring sound?
+
+
+Module Loading:
+===============
+
+When a sound card is first referenced and sound is modular, the sound system
+will ask for the sound devices to be loaded. Initially it requests that
+the driver for the sound system is loaded. It then will ask for
+sound-slot-0, where 0 is the first sound card. (sound-slot-1 the second and
+so on). Thus you can do
+
+alias sound-slot-0 sb
+
+To load a soundblaster at this point. If the slot loading does not provide
+the desired device - for example a soundblaster does not directly provide
+a midi synth in all cases then it will request "sound-service-0-n" where n
+is
+
+ 0 Mixer
+
+ 2 MIDI
+
+ 3, 4 DSP audio
+
+
+For example, I use the following to load my Soundblaster PCI 128
+(ES 1371) card first, followed by my SoundBlaster Vibra 16 card,
+then by my TV card:
+
+# Load the Soundblaster PCI 128 as /dev/dsp, /dev/dsp1, /dev/mixer
+alias sound-slot-0 es1371
+
+# Load the Soundblaster Vibra 16 as /dev/dsp2, /dev/mixer1
+alias sound-slot-1 sb
+options sb io=0x240 irq=5 dma=1 dma16=5 mpu_io=0x330
+
+# Load the BTTV (TV card) as /dev/mixer2
+alias sound-slot-2 bttv
+alias sound-service-2-0 tvmixer
+
+pre-install bttv modprobe tuner ; modprobe tvmixer
+pre-install tvmixer modprobe msp3400; modprobe tvaudio
+options tuner debug=0 type=8
+options bttv card=0 radio=0 pll=0
+
+
+For More Information (RTFM):
+============================
+1) Information on kernel modules: linux/Documentation/modules.txt
+ and manual pages for insmod and modprobe.
+
+2) Information on PnP, RTFM manual pages for isapnp.
+
+3) Sound-HOWTO and Sound-Playing-HOWTO.
+
+4) OSS's WWW site at http://www.opensound.com.
+
+5) All the files in linux/Documentation/sound.
+
+6) The comments and code in linux/drivers/sound.
+
+7) The sndconfig and rhsound documentation from Red Hat.
+
+8) The Linux-sound mailing list: sound-list@redhat.com.
+
+9) Enlightenment documentation (for info on esd)
+ http://www.tux.org/~ricdude/EsounD.html.
+
+10) ALSA home page: http://www.alsa-project.org/
+
+
+Contact Information:
+====================
+Wade Hampton: (whampton@staffnet.com)
+
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/MAD16 linux-2.5.6/Documentation/sound/oss/MAD16
--- linux-2.5.6-pre3/Documentation/sound/oss/MAD16 Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/MAD16 Thu Mar 7 18:24:41 2002
@@ -0,0 +1,55 @@
+(This recipe has been edited to update the configuration symbols.)
+
+From: Shaw Carruthers
+
+I have been using mad16 sound for some time now with no problems, current
+kernel 2.1.89
+
+lsmod shows:
+
+mad16 5176 0
+sb 22044 0 [mad16]
+uart401 5576 0 [mad16 sb]
+ad1848 14176 1 [mad16]
+sound 61928 0 [mad16 sb uart401 ad1848]
+
+.config has:
+
+CONFIG_SOUND=m
+CONFIG_SOUND_ADLIB=m
+CONFIG_SOUND_MAD16=m
+CONFIG_SOUND_YM3812=m
+
+modules.conf has:
+
+alias char-major-14 mad16
+options sb mad16=1
+options mad16 io=0x530 irq=7 dma=0 dma16=1 && /usr/local/bin/aumix -w 15 -p 20 -m 0 -1 0 -2 0 -3 0 -i 0
+
+
+To get the built in mixer to work this needs to be:
+
+options adlib_card io=0x388 # FM synthesizer
+options sb mad16=1
+options mad16 io=0x530 irq=7 dma=0 dma16=1 mpu_io=816 mpu_irq=5 && /usr/local/bin/aumix -w 15 -p 20 -m 0 -1 0 -2 0 -3 0 -i 0
+
+The addition of the "mpu_io=816 mpu_irq=5" to the mad16 options line is
+
+------------------------------------------------------------------------
+The mad16 module in addition supports the following options:
+
+option: meaning: default:
+joystick=0,1 disabled, enabled disabled
+cdtype=0x00,0x02,0x04, disabled, Sony CDU31A, disabled
+ 0x06,0x08,0x0a Mitsumi, Panasonic,
+ Secondary IDE, Primary IDE
+cdport=0x340,0x320, 0x340
+ 0x330,0x360
+cdirq=0,3,5,7,9,10,11 disabled, IRQ3, ... disabled
+cddma=0,5,6,7 disabled, DMA5, ... DMA5 for Mitsumi or IDE
+cddma=0,1,2,3 disabled, DMA1, ... DMA3 for Sony or Panasonic
+opl4=0,1 OPL3, OPL4 OPL3
+
+for more details see linux/drivers/sound/mad16.c
+
+Rui Sousa
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/Maestro linux-2.5.6/Documentation/sound/oss/Maestro
--- linux-2.5.6-pre3/Documentation/sound/oss/Maestro Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/Maestro Thu Mar 7 18:24:41 2002
@@ -0,0 +1,123 @@
+ An OSS/Lite Driver for the ESS Maestro family of sound cards
+
+ Zach Brown, December 1999
+
+Driver Status and Availability
+------------------------------
+
+The most recent version of this driver will hopefully always be available at
+ http://www.zabbo.net/maestro/
+
+I will try and maintain the most recent stable version of the driver
+in both the stable and development kernel lines.
+
+ESS Maestro Chip Family
+-----------------------
+
+There are 3 main variants of the ESS Maestro PCI sound chip. The first
+is the Maestro 1. It was originally produced by Platform Tech as the
+'AGOGO'. It can be recognized by Platform Tech's PCI ID 0x1285 with
+0x0100 as the device ID. It was put on some sound boards and a few laptops.
+ESS bought the design and cleaned it up as the Maestro 2. This starts
+their marking with the ESS vendor ID 0x125D and the 'year' device IDs.
+The Maestro 2 claims 0x1968 while the Maestro 2e has 0x1978.
+
+The various families of Maestro are mostly identical as far as this
+driver is concerned. It doesn't touch the DSP parts that differ (though
+it could for FM synthesis).
+
+Driver OSS Behavior
+--------------------
+
+This OSS driver exports /dev/mixer and /dev/dsp to applications, which
+mostly adhere to the OSS spec. This driver doesn't register itself
+with /dev/sndstat, so don't expect information to appear there.
+
+The /dev/dsp device exported behaves almost as expected. Playback is
+supported in all the various lovely formats. 8/16bit stereo/mono from
+8khz to 48khz, and mmap()ing for playback behaves. Capture/recording
+is limited due to oddities with the Maestro hardware. One can only
+record in 16bit stereo. For recording the maestro uses non interleaved
+stereo buffers so that mmap()ing the incoming data does not result in
+a ring buffer of LRLR data. mmap()ing of the read buffers is therefore
+disallowed until this can be cleaned up.
+
+/dev/mixer is an interface to the AC'97 codec on the Maestro. It is
+worth noting that there are a variety of AC'97s that can be wired to
+the Maestro. Which is used is entirely up to the hardware implementor.
+This should only be visible to the user by the presence, or lack, of
+'Bass' and 'Treble' sliders in the mixer. Not all AC'97s have them.
+
+The driver doesn't support MIDI or FM playback at the moment. Typically
+the Maestro is wired to an MPU MIDI chip, but some hardware implementations
+don't. We need to assemble a white list of hardware implementations that
+have MIDI wired properly before we can claim to support it safely.
+
+Compiling and Installing
+------------------------
+
+With the drivers inclusion into the kernel, compiling and installing
+is the same as most OSS/Lite modular sound drivers. Compilation
+of the driver is enabled through the CONFIG_SOUND_MAESTRO variable
+in the config system.
+
+It may be modular or statically linked. If it is modular it should be
+installed with the rest of the modules for the kernel on the system.
+Typically this will be in /lib/modules/ somewhere. 'alias sound maestro'
+should also be added to your module configs (typically /etc/conf.modules)
+if you're using modular OSS/Lite sound and want to default to using a
+maestro chip.
+
+As this is a PCI device, the module does not need to be informed of
+any IO or IRQ resources it should use, it devines these from the
+system. Sometimes, on sucky PCs, the BIOS fails to allocated resources
+for the maestro. This will result in a message like:
+ maestro: PCI subsystem reports IRQ 0, this might not be correct.
+from the kernel. Should this happen the sound chip most likely will
+not operate correctly. To solve this one has to dig through their BIOS
+(typically entered by hitting a hot key at boot time) and figure out
+what magic needs to happen so that the BIOS will reward the maestro with
+an IRQ. This operation is incredibly system specific, so you're on your
+own. Sometimes the magic lies in 'PNP Capable Operating System' settings.
+
+There are very few options to the driver. One is 'debug' which will
+tell the driver to print minimal debugging information as it runs. This
+can be collected with 'dmesg' or through the klogd daemon.
+
+The other, more interesting option, is 'dsps_order'. Typically at
+install time the driver will only register one available /dev/dsp device
+for its use. The 'dsps_order' module parameter allows for more devices
+to be allocated, as a power of two. Up to 4 devices can be registered
+( dsps_order=2 ). These devices act as fully distinct units and use
+separate channels in the maestro.
+
+Power Management
+----------------
+
+As of version 0.14, this driver has a minimal understanding of PCI
+Power Management. If it finds a valid power management capability
+on the PCI device it will attempt to use the power management
+functions of the maestro. It will only do this on Maestro 2Es and
+only on machines that are known to function well. You can
+force the use of power management by setting the 'use_pm' module
+option to 1, or can disable it entirely by setting it to 0.
+
+When using power management, the driver does a few things
+differently. It will keep the chip in a lower power mode
+when the module is inserted but /dev/dsp is not open. This
+allows the mixer to function but turns off the clocks
+on other parts of the chip. When /dev/dsp is opened the chip
+is brought into full power mode, and brought back down
+when it is closed. It also powers down the chip entirely
+when the module is removed or the machine is shutdown. This
+can have nonobvious consequences. CD audio may not work
+after a power managing driver is removed. Also, software that
+doesn't understand power management may not be able to talk
+to the powered down chip until the machine goes through a hard
+reboot to bring it back.
+
+.. more details ..
+------------------
+
+drivers/sound/maestro.c contains comments that hopefully explain
+the maestro implementation.
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/Maestro3 linux-2.5.6/Documentation/sound/oss/Maestro3
--- linux-2.5.6-pre3/Documentation/sound/oss/Maestro3 Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/Maestro3 Thu Mar 7 18:24:41 2002
@@ -0,0 +1,84 @@
+ An OSS/Lite Driver for the ESS Maestro3 family of sound chips
+
+ Zach Brown, January 2001
+
+Driver Status and Availability
+------------------------------
+
+The most recent version of this driver will hopefully always be available at
+ http://www.zabbo.net/maestro3/
+
+I will try and maintain the most recent stable version of the driver
+in both the stable and development kernel lines.
+
+Historically I've sucked pretty hard at actually doing that, however.
+
+ESS Maestro3 Chip Family
+-----------------------
+
+The 'Maestro3' is much like the Maestro2 chip. The noted improvement
+is the removal of the silicon in the '2' that did PCM mixing. All that
+work is now done through a custom DSP called the ASSP, the Asynchronus
+Specific Signal Processor.
+
+The 'Allegro' is a baby version of the Maestro3. I'm not entirely clear
+on the extent of the differences, but the driver supports them both :)
+
+The 'Allegro' shows up as PCI ID 0x1988 and the Maestro3 as 0x1998,
+both under ESS's vendor ID of 0x125D. The Maestro3 can also show up as
+0x199a when hardware strapping is used.
+
+The chip can also act as a multi function device. The modem IDs follow
+the audio multimedia device IDs. (so the modem part of an Allegro shows
+up as 0x1989)
+
+Driver OSS Behavior
+--------------------
+
+This OSS driver exports /dev/mixer and /dev/dsp to applications, which
+mostly adhere to the OSS spec. This driver doesn't register itself
+with /dev/sndstat, so don't expect information to appear there.
+
+The /dev/dsp device exported behaves as expected. Playback is
+supported in all the various lovely formats. 8/16bit stereo/mono from
+8khz to 48khz, with both read()/write(), and mmap().
+
+/dev/mixer is an interface to the AC'97 codec on the Maestro3. It is
+worth noting that there are a variety of AC'97s that can be wired to
+the Maestro3. Which is used is entirely up to the hardware implementor.
+This should only be visible to the user by the presence, or lack, of
+'Bass' and 'Treble' sliders in the mixer. Not all AC'97s have them.
+The Allegro has an onchip AC'97.
+
+The driver doesn't support MIDI or FM playback at the moment.
+
+Compiling and Installing
+------------------------
+
+With the drivers inclusion into the kernel, compiling and installing
+is the same as most OSS/Lite modular sound drivers. Compilation
+of the driver is enabled through the CONFIG_SOUND_MAESTRO3 variable
+in the config system.
+
+It may be modular or statically linked. If it is modular it should be
+installed with the rest of the modules for the kernel on the system.
+Typically this will be in /lib/modules/ somewhere. 'alias sound-slot-0
+maestro3' should also be added to your module configs (typically
+/etc/modules.conf) if you're using modular OSS/Lite sound and want to
+default to using a maestro3 chip.
+
+There are very few options to the driver. One is 'debug' which will
+tell the driver to print minimal debugging information as it runs. This
+can be collected with 'dmesg' or through the klogd daemon.
+
+The other is 'external_amp', which tells the driver to attempt to enable
+an external amplifier. This defaults to '1', you can tell the driver
+not to bother enabling such an amplifier by setting it to '0'.
+
+Power Management
+----------------
+
+This driver has a minimal understanding of PCI Power Management. It will
+try and power down the chip when the system is suspended, and power
+it up with it is resumed. It will also try and power down the chip
+when the machine is shut down.
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/MultiSound linux-2.5.6/Documentation/sound/oss/MultiSound
--- linux-2.5.6-pre3/Documentation/sound/oss/MultiSound Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/MultiSound Thu Mar 7 18:24:41 2002
@@ -0,0 +1,1137 @@
+#! /bin/sh
+#
+# Turtle Beach MultiSound Driver Notes
+# -- Andrew Veliath
+#
+# Last update: September 10, 1998
+# Corresponding msnd driver: 0.8.3
+#
+# ** This file is a README (top part) and shell archive (bottom part).
+# The corresponding archived utility sources can be unpacked by
+# running `sh MultiSound' (the utilities are only needed for the
+# Pinnacle and Fiji cards). **
+#
+#
+# -=-=- Getting Firmware -=-=-
+# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+# See the section `Obtaining and Creating Firmware Files' in this
+# document for instructions on obtaining the necessary firmware
+# files.
+#
+#
+# Supported Features
+# ~~~~~~~~~~~~~~~~~~
+#
+# Currently, full-duplex digital audio (/dev/dsp only, /dev/audio is
+# not currently available) and mixer functionality (/dev/mixer) are
+# supported (memory mapped digital audio is not yet supported).
+# Digital transfers and monitoring can be done as well if you have
+# the digital daughterboard (see the section on using the S/PDIF port
+# for more information).
+#
+# Support for the Turtle Beach MultiSound Hurricane architecture is
+# composed of the following modules (these can also operate compiled
+# into the kernel):
+#
+# msnd - MultiSound base (requires soundcore)
+#
+# msnd_classic - Base audio/mixer support for Classic, Monetery and
+# Tahiti cards
+#
+# msnd_pinnacle - Base audio/mixer support for Pinnacle and Fiji cards
+#
+#
+# Important Notes - Read Before Using
+# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+# The firmware files are not included (may change in future). You
+# must obtain these images from Turtle Beach (they are included in
+# the MultiSound Development Kits), and place them in /etc/sound for
+# example, and give the full paths in the Linux configuration. If
+# you are compiling in support for the MultiSound driver rather than
+# using it as a module, these firmware files must be accessible
+# during kernel compilation.
+#
+# Please note these files must be binary files, not assembler. See
+# the section later in this document for instructions to obtain these
+# files.
+#
+#
+# Configuring Card Resources
+# ~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+# ** This section is very important, as your card may not work at all
+# or your machine may crash if you do not do this correctly. **
+#
+# * Classic/Monterey/Tahiti
+#
+# These cards are configured through the driver msnd_classic. You must
+# know the io port, then the driver will select the irq and memory resources
+# on the card. It is up to you to know if these are free locations or now,
+# a conflict can lock the machine up.
+#
+# * Pinnacle/Fiji
+#
+# The Pinnacle and Fiji cards have an extra config port, either
+# 0x250, 0x260 or 0x270. This port can be disabled to have the card
+# configured strictly through PnP, however you lose the ability to
+# access the IDE controller and joystick devices on this card when
+# using PnP. The included pinnaclecfg program in this shell archive
+# can be used to configure the card in non-PnP mode, and in PnP mode
+# you can use isapnptools. These are described briefly here.
+#
+# pinnaclecfg is not required; you can use the msnd_pinnacle module
+# to fully configure the card as well. However, pinnaclecfg can be
+# used to change the resource values of a particular device after the
+# msnd_pinnacle module has been loaded. If you are compiling the
+# driver into the kernel, you must set these values during compile
+# time, however other peripheral resource values can be changed with
+# the pinnaclecfg program after the kernel is loaded.
+#
+#
+# *** PnP mode
+#
+# Use pnpdump to obtain a sample configuration if you can; I was able
+# to obtain one with the command `pnpdump 1 0x203' -- this may vary
+# for you (running pnpdump by itself did not work for me). Then,
+# edit this file and use isapnp to uncomment and set the card values.
+# Use these values when inserting the msnd_pinnacle module. Using
+# this method, you can set the resources for the DSP and the Kurzweil
+# synth (Pinnacle). Since Linux does not directly support PnP
+# devices, you may have difficulty when using the card in PnP mode
+# when it the driver is compiled into the kernel. Using non-PnP mode
+# is preferable in this case.
+#
+# Here is an example mypinnacle.conf for isapnp that sets the card to
+# io base 0x210, irq 5 and mem 0xd8000, and also sets the Kurzweil
+# synth to 0x330 and irq 9 (may need editing for your system):
+#
+# (READPORT 0x0203)
+# (CSN 2)
+# (IDENTIFY *)
+#
+# # DSP
+# (CONFIGURE BVJ0440/-1 (LD 0
+# (INT 0 (IRQ 5 (MODE +E))) (IO 0 (BASE 0x0210)) (MEM 0 (BASE 0x0d8000))
+# (ACT Y)))
+#
+# # Kurzweil Synth (Pinnacle Only)
+# (CONFIGURE BVJ0440/-1 (LD 1
+# (IO 0 (BASE 0x0330)) (INT 0 (IRQ 9 (MODE +E)))
+# (ACT Y)))
+#
+# (WAITFORKEY)
+#
+#
+# *** Non-PnP mode
+#
+# The second way is by running the card in non-PnP mode. This
+# actually has some advantages in that you can access some other
+# devices on the card, such as the joystick and IDE controller. To
+# configure the card, unpack this shell archive and build the
+# pinnaclecfg program. Using this program, you can assign the
+# resource values to the card's devices, or disable the devices. As
+# an alternative to using pinnaclecfg, you can specify many of the
+# configuration values when loading the msnd_pinnacle module (or
+# during kernel configuration when compiling the driver into the
+# kernel).
+#
+# If you specify cfg=0x250 for the msnd_pinnacle module, it
+# automatically configure the card to the given io, irq and memory
+# values using that config port (the config port is jumper selectable
+# on the card to 0x250, 0x260 or 0x270).
+#
+# See the `msnd_pinnacle Additional Options' section below for more
+# information on these parameters (also, if you compile the driver
+# directly into the kernel, these extra parameters can be useful
+# here).
+#
+#
+# ** It is very easy to cause problems in your machine if you choose a
+# resource value which is incorrect. **
+#
+#
+# Examples
+# ~~~~~~~~
+#
+# * MultiSound Classic/Monterey/Tahiti:
+#
+# modprobe soundcore
+# insmod msnd
+# insmod msnd_classic io=0x290 irq=7 mem=0xd0000
+#
+# * MultiSound Pinnacle in PnP mode:
+#
+# modprobe soundcore
+# insmod msnd
+# isapnp mypinnacle.conf
+# insmod msnd_pinnacle io=0x210 irq=5 mem=0xd8000 <-- match mypinnacle.conf values
+#
+# * MultiSound Pinnacle in non-PnP mode (replace 0x250 with your configuration port,
+# one of 0x250, 0x260 or 0x270):
+#
+# insmod soundcore
+# insmod msnd
+# insmod msnd_pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000
+#
+# * To use the MPU-compatible Kurzweil synth on the Pinnacle in PnP
+# mode, add the following (assumes you did `isapnp mypinnacle.conf'):
+#
+# insmod sound
+# insmod mpu401 io=0x330 irq=9 <-- match mypinnacle.conf values
+#
+# * To use the MPU-compatible Kurzweil synth on the Pinnacle in non-PnP
+# mode, add the following. Note how we first configure the peripheral's
+# resources, _then_ install a Linux driver for it:
+#
+# insmod sound
+# pinnaclecfg 0x250 mpu 0x330 9
+# insmod mpu401 io=0x330 irq=9
+#
+# -- OR you can use the following sequence without pinnaclecfg in non-PnP mode:
+#
+# insmod soundcore
+# insmod msnd
+# insmod msnd_pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000 mpu_io=0x330 mpu_irq=9
+# insmod sound
+# insmod mpu401 io=0x330 irq=9
+#
+# * To setup the joystick port on the Pinnacle in non-PnP mode (though
+# you have to find the actual Linux joystick driver elsewhere), you
+# can use pinnaclecfg:
+#
+# pinnaclecfg 0x250 joystick 0x200
+#
+# -- OR you can configure this using msnd_pinnacle with the following:
+#
+# insmod soundcore
+# insmod msnd
+# insmod msnd_pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000 joystick_io=0x200
+#
+#
+# msnd_classic, msnd_pinnacle Required Options
+# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+# If the following options are not given, the module will not load.
+# Examine the kernel message log for informative error messages.
+# WARNING--probing isn't supported so try to make sure you have the
+# correct shared memory area, otherwise you may experience problems.
+#
+# io I/O base of DSP, e.g. io=0x210
+# irq IRQ number, e.g. irq=5
+# mem Shared memory area, e.g. mem=0xd8000
+#
+#
+# msnd_classic, msnd_pinnacle Additional Options
+# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+# fifosize The digital audio FIFOs, in kilobytes. If not
+# specified, the default will be used. Increasing
+# this value will reduce the chance of a FIFO
+# underflow at the expense of increasing overall
+# latency. For example, fifosize=512 will
+# allocate 512kB read and write FIFOs (1MB total).
+# While this may reduce dropouts, a heavy machine
+# load will undoubtedly starve the FIFO of data
+# and you will eventually get dropouts. One
+# option is to alter the scheduling priority of
+# the playback process, using `nice' or some form
+# of POSIX soft real-time scheduling.
+#
+# calibrate_signal Setting this to one calibrates the ADCs to the
+# signal, zero calibrates to the card (defaults
+# to zero).
+#
+#
+# msnd_pinnacle Additional Options
+# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+# digital Specify digital=1 to enable the S/PDIF input
+# if you have the digital daughterboard
+# adapter. This will enable access to the
+# DIGITAL1 input for the soundcard in the mixer.
+# Some mixer programs might have trouble setting
+# the DIGITAL1 source as an input. If you have
+# trouble, you can try the setdigital.c program
+# at the bottom of this document.
+#
+# cfg Non-PnP configuration port for the Pinnacle
+# and Fiji (typically 0x250, 0x260 or 0x270,
+# depending on the jumper configuration). If
+# this option is omitted, then it is assumed
+# that the card is in PnP mode, and that the
+# specified DSP resource values are already
+# configured with PnP (i.e. it won't attempt to
+# do any sort of configuration).
+#
+# When the Pinnacle is in non-PnP mode, you can use the following
+# options to configure particular devices. If a full specification
+# for a device is not given, then the device is not configured. Note
+# that you still must use a Linux driver for any of these devices
+# once their resources are setup (such as the Linux joystick driver,
+# or the MPU401 driver from OSS for the Kurzweil synth).
+#
+# mpu_io I/O port of MPU (on-board Kurzweil synth)
+# mpu_irq IRQ of MPU (on-board Kurzweil synth)
+# ide_io0 First I/O port of IDE controller
+# ide_io1 Second I/O port of IDE controller
+# ide_irq IRQ IDE controller
+# joystick_io I/O port of joystick
+#
+#
+# Obtaining and Creating Firmware Files
+# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+# For the Classic/Tahiti/Monterey
+# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+# Download to /tmp and unzip the following file from Turtle Beach:
+#
+# ftp://ftp.voyetra.com/pub/tbs/msndcl/msndvkit.zip
+#
+# When unzipped, unzip the file named MsndFiles.zip. Then copy the
+# following firmware files to /etc/sound (note the file renaming):
+#
+# cp DSPCODE/MSNDINIT.BIN /etc/sound/msndinit.bin
+# cp DSPCODE/MSNDPERM.REB /etc/sound/msndperm.bin
+#
+# When configuring the Linux kernel, specify /etc/sound/msndinit.bin and
+# /etc/sound/msndperm.bin for the two firmware files (Linux kernel
+# versions older than 2.2 do not ask for firmware paths, and are
+# hardcoded to /etc/sound).
+#
+# If you are compiling the driver into the kernel, these files must
+# be accessible during compilation, but will not be needed later.
+# The files must remain, however, if the driver is used as a module.
+#
+#
+# For the Pinnacle/Fiji
+# ~~~~~~~~~~~~~~~~~~~~~
+#
+# Download to /tmp and unzip the following file from Turtle Beach (be
+# sure to use the entire URL; some have had trouble navigating to the
+# URL):
+#
+# ftp://ftp.voyetra.com/pub/tbs/pinn/pnddk100.zip
+#
+# Unpack this shell archive, and run make in the created directory
+# (you need a C compiler and flex to build the utilities). This
+# should give you the executables conv, pinnaclecfg and setdigital.
+# conv is only used temporarily here to create the firmware files,
+# while pinnaclecfg is used to configure the Pinnacle or Fiji card in
+# non-PnP mode, and setdigital can be used to set the S/PDIF input on
+# the mixer (pinnaclecfg and setdigital should be copied to a
+# convenient place, possibly run during system initialization).
+#
+# To generating the firmware files with the `conv' program, we create
+# the binary firmware files by doing the following conversion
+# (assuming the archive unpacked into a directory named PINNDDK):
+#
+# ./conv < PINNDDK/dspcode/pndspini.asm > /etc/sound/pndspini.bin
+# ./conv < PINNDDK/dspcode/pndsperm.asm > /etc/sound/pndsperm.bin
+#
+# The conv (and conv.l) program is not needed after conversion and can
+# be safely deleted. Then, when configuring the Linux kernel, specify
+# /etc/sound/pndspini.bin and /etc/sound/pndsperm.bin for the two
+# firmware files (Linux kernel versions older than 2.2 do not ask for
+# firmware paths, and are hardcoded to /etc/sound).
+#
+# If you are compiling the driver into the kernel, these files must
+# be accessible during compilation, but will not be needed later.
+# The files must remain, however, if the driver is used as a module.
+#
+#
+# Using Digital I/O with the S/PDIF Port
+# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+# If you have a Pinnacle or Fiji with the digital daughterboard and
+# want to set it as the input source, you can use this program if you
+# have trouble trying to do it with a mixer program (be sure to
+# insert the module with the digital=1 option, or say Y to the option
+# during compiled-in kernel operation). Upon selection of the S/PDIF
+# port, you should be able monitor and record from it.
+#
+# There is something to note about using the S/PDIF port. Digital
+# timing is taken from the digital signal, so if a signal is not
+# connected to the port and it is selected as recording input, you
+# will find PCM playback to be distorted in playback rate. Also,
+# attempting to record at a sampling rate other than the DAT rate may
+# be problematic (i.e. trying to record at 8000Hz when the DAT signal
+# is 44100Hz). If you have a problem with this, set the recording
+# input to analog if you need to record at a rate other than that of
+# the DAT rate.
+#
+#
+# -- Shell archive attached below, just run `sh MultiSound' to extract.
+# Contains Pinnacle/Fiji utilities to convert firmware, configure
+# in non-PnP mode, and select the DIGITAL1 input for the mixer.
+#
+#
+#!/bin/sh
+# This is a shell archive (produced by GNU sharutils 4.2).
+# To extract the files from this archive, save it to some FILE, remove
+# everything before the `!/bin/sh' line above, then type `sh FILE'.
+#
+# Made on 1998-12-04 10:07 EST by .
+# Source directory was `/home/andrewtv/programming/pinnacle/pinnacle'.
+#
+# Existing files will *not* be overwritten unless `-c' is specified.
+#
+# This shar contains:
+# length mode name
+# ------ ---------- ------------------------------------------
+# 2046 -rw-rw-r-- MultiSound.d/setdigital.c
+# 10235 -rw-rw-r-- MultiSound.d/pinnaclecfg.c
+# 106 -rw-rw-r-- MultiSound.d/Makefile
+# 141 -rw-rw-r-- MultiSound.d/conv.l
+# 1472 -rw-rw-r-- MultiSound.d/msndreset.c
+#
+save_IFS="${IFS}"
+IFS="${IFS}:"
+gettext_dir=FAILED
+locale_dir=FAILED
+first_param="$1"
+for dir in $PATH
+do
+ if test "$gettext_dir" = FAILED && test -f $dir/gettext \
+ && ($dir/gettext --version >/dev/null 2>&1)
+ then
+ set `$dir/gettext --version 2>&1`
+ if test "$3" = GNU
+ then
+ gettext_dir=$dir
+ fi
+ fi
+ if test "$locale_dir" = FAILED && test -f $dir/shar \
+ && ($dir/shar --print-text-domain-dir >/dev/null 2>&1)
+ then
+ locale_dir=`$dir/shar --print-text-domain-dir`
+ fi
+done
+IFS="$save_IFS"
+if test "$locale_dir" = FAILED || test "$gettext_dir" = FAILED
+then
+ echo=echo
+else
+ TEXTDOMAINDIR=$locale_dir
+ export TEXTDOMAINDIR
+ TEXTDOMAIN=sharutils
+ export TEXTDOMAIN
+ echo="$gettext_dir/gettext -s"
+fi
+touch -am 1231235999 $$.touch >/dev/null 2>&1
+if test ! -f 1231235999 && test -f $$.touch; then
+ shar_touch=touch
+else
+ shar_touch=:
+ echo
+ $echo 'WARNING: not restoring timestamps. Consider getting and'
+ $echo "installing GNU \`touch', distributed in GNU File Utilities..."
+ echo
+fi
+rm -f 1231235999 $$.touch
+#
+if mkdir _sh01426; then
+ $echo 'x -' 'creating lock directory'
+else
+ $echo 'failed to create lock directory'
+ exit 1
+fi
+# ============= MultiSound.d/setdigital.c ==============
+if test ! -d 'MultiSound.d'; then
+ $echo 'x -' 'creating directory' 'MultiSound.d'
+ mkdir 'MultiSound.d'
+fi
+if test -f 'MultiSound.d/setdigital.c' && test "$first_param" != -c; then
+ $echo 'x -' SKIPPING 'MultiSound.d/setdigital.c' '(file already exists)'
+else
+ $echo 'x -' extracting 'MultiSound.d/setdigital.c' '(text)'
+ sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/setdigital.c' &&
+/*********************************************************************
+X *
+X * setdigital.c - sets the DIGITAL1 input for a mixer
+X *
+X * Copyright (C) 1998 Andrew Veliath
+X *
+X * This program is free software; you can redistribute it and/or modify
+X * it under the terms of the GNU General Public License as published by
+X * the Free Software Foundation; either version 2 of the License, or
+X * (at your option) any later version.
+X *
+X * This program is distributed in the hope that it will be useful,
+X * but WITHOUT ANY WARRANTY; without even the implied warranty of
+X * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+X * GNU General Public License for more details.
+X *
+X * You should have received a copy of the GNU General Public License
+X * along with this program; if not, write to the Free Software
+X * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+X *
+X ********************************************************************/
+X
+#include
+#include
+#include
+#include
+#include
+#include
+#include
+X
+int main(int argc, char *argv[])
+{
+X int fd;
+X unsigned long recmask, recsrc;
+X
+X if (argc != 2) {
+X fprintf(stderr, "usage: setdigital \n");
+X exit(1);
+X }
+X
+X if ((fd = open(argv[1], O_RDWR)) < 0) {
+X perror(argv[1]);
+X exit(1);
+X }
+X
+X if (ioctl(fd, SOUND_MIXER_READ_RECMASK, &recmask) < 0) {
+X fprintf(stderr, "error: ioctl read recording mask failed\n");
+X perror("ioctl");
+X close(fd);
+X exit(1);
+X }
+X
+X if (!(recmask & SOUND_MASK_DIGITAL1)) {
+X fprintf(stderr, "error: cannot find DIGITAL1 device in mixer\n");
+X close(fd);
+X exit(1);
+X }
+X
+X if (ioctl(fd, SOUND_MIXER_READ_RECSRC, &recsrc) < 0) {
+X fprintf(stderr, "error: ioctl read recording source failed\n");
+X perror("ioctl");
+X close(fd);
+X exit(1);
+X }
+X
+X recsrc |= SOUND_MASK_DIGITAL1;
+X
+X if (ioctl(fd, SOUND_MIXER_WRITE_RECSRC, &recsrc) < 0) {
+X fprintf(stderr, "error: ioctl write recording source failed\n");
+X perror("ioctl");
+X close(fd);
+X exit(1);
+X }
+X
+X close(fd);
+X
+X return 0;
+}
+SHAR_EOF
+ $shar_touch -am 1204092598 'MultiSound.d/setdigital.c' &&
+ chmod 0664 'MultiSound.d/setdigital.c' ||
+ $echo 'restore of' 'MultiSound.d/setdigital.c' 'failed'
+ if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \
+ && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then
+ md5sum -c << SHAR_EOF >/dev/null 2>&1 \
+ || $echo 'MultiSound.d/setdigital.c:' 'MD5 check failed'
+e87217fc3e71288102ba41fd81f71ec4 MultiSound.d/setdigital.c
+SHAR_EOF
+ else
+ shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/setdigital.c'`"
+ test 2046 -eq "$shar_count" ||
+ $echo 'MultiSound.d/setdigital.c:' 'original size' '2046,' 'current size' "$shar_count!"
+ fi
+fi
+# ============= MultiSound.d/pinnaclecfg.c ==============
+if test -f 'MultiSound.d/pinnaclecfg.c' && test "$first_param" != -c; then
+ $echo 'x -' SKIPPING 'MultiSound.d/pinnaclecfg.c' '(file already exists)'
+else
+ $echo 'x -' extracting 'MultiSound.d/pinnaclecfg.c' '(text)'
+ sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/pinnaclecfg.c' &&
+/*********************************************************************
+X *
+X * pinnaclecfg.c - Pinnacle/Fiji Device Configuration Program
+X *
+X * This is for NON-PnP mode only. For PnP mode, use isapnptools.
+X *
+X * This is Linux-specific, and must be run with root permissions.
+X *
+X * Part of the Turtle Beach MultiSound Sound Card Driver for Linux
+X *
+X * Copyright (C) 1998 Andrew Veliath
+X *
+X * This program is free software; you can redistribute it and/or modify
+X * it under the terms of the GNU General Public License as published by
+X * the Free Software Foundation; either version 2 of the License, or
+X * (at your option) any later version.
+X *
+X * This program is distributed in the hope that it will be useful,
+X * but WITHOUT ANY WARRANTY; without even the implied warranty of
+X * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+X * GNU General Public License for more details.
+X *
+X * You should have received a copy of the GNU General Public License
+X * along with this program; if not, write to the Free Software
+X * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+X *
+X ********************************************************************/
+X
+#include
+#include
+#include
+#include
+#include
+#include
+#include
+X
+#define IREG_LOGDEVICE 0x07
+#define IREG_ACTIVATE 0x30
+#define LD_ACTIVATE 0x01
+#define LD_DISACTIVATE 0x00
+#define IREG_EECONTROL 0x3F
+#define IREG_MEMBASEHI 0x40
+#define IREG_MEMBASELO 0x41
+#define IREG_MEMCONTROL 0x42
+#define IREG_MEMRANGEHI 0x43
+#define IREG_MEMRANGELO 0x44
+#define MEMTYPE_8BIT 0x00
+#define MEMTYPE_16BIT 0x02
+#define MEMTYPE_RANGE 0x00
+#define MEMTYPE_HIADDR 0x01
+#define IREG_IO0_BASEHI 0x60
+#define IREG_IO0_BASELO 0x61
+#define IREG_IO1_BASEHI 0x62
+#define IREG_IO1_BASELO 0x63
+#define IREG_IRQ_NUMBER 0x70
+#define IREG_IRQ_TYPE 0x71
+#define IRQTYPE_HIGH 0x02
+#define IRQTYPE_LOW 0x00
+#define IRQTYPE_LEVEL 0x01
+#define IRQTYPE_EDGE 0x00
+X
+#define HIBYTE(w) ((BYTE)(((WORD)(w) >> 8) & 0xFF))
+#define LOBYTE(w) ((BYTE)(w))
+#define MAKEWORD(low,hi) ((WORD)(((BYTE)(low))|(((WORD)((BYTE)(hi)))<<8)))
+X
+typedef __u8 BYTE;
+typedef __u16 USHORT;
+typedef __u16 WORD;
+X
+static int config_port = -1;
+X
+static int msnd_write_cfg(int cfg, int reg, int value)
+{
+X outb(reg, cfg);
+X outb(value, cfg + 1);
+X if (value != inb(cfg + 1)) {
+X fprintf(stderr, "error: msnd_write_cfg: I/O error\n");
+X return -EIO;
+X }
+X return 0;
+}
+X
+static int msnd_read_cfg(int cfg, int reg)
+{
+X outb(reg, cfg);
+X return inb(cfg + 1);
+}
+X
+static int msnd_write_cfg_io0(int cfg, int num, WORD io)
+{
+X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X return -EIO;
+X if (msnd_write_cfg(cfg, IREG_IO0_BASEHI, HIBYTE(io)))
+X return -EIO;
+X if (msnd_write_cfg(cfg, IREG_IO0_BASELO, LOBYTE(io)))
+X return -EIO;
+X return 0;
+}
+X
+static int msnd_read_cfg_io0(int cfg, int num, WORD *io)
+{
+X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X return -EIO;
+X
+X *io = MAKEWORD(msnd_read_cfg(cfg, IREG_IO0_BASELO),
+X msnd_read_cfg(cfg, IREG_IO0_BASEHI));
+X
+X return 0;
+}
+X
+static int msnd_write_cfg_io1(int cfg, int num, WORD io)
+{
+X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X return -EIO;
+X if (msnd_write_cfg(cfg, IREG_IO1_BASEHI, HIBYTE(io)))
+X return -EIO;
+X if (msnd_write_cfg(cfg, IREG_IO1_BASELO, LOBYTE(io)))
+X return -EIO;
+X return 0;
+}
+X
+static int msnd_read_cfg_io1(int cfg, int num, WORD *io)
+{
+X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X return -EIO;
+X
+X *io = MAKEWORD(msnd_read_cfg(cfg, IREG_IO1_BASELO),
+X msnd_read_cfg(cfg, IREG_IO1_BASEHI));
+X
+X return 0;
+}
+X
+static int msnd_write_cfg_irq(int cfg, int num, WORD irq)
+{
+X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X return -EIO;
+X if (msnd_write_cfg(cfg, IREG_IRQ_NUMBER, LOBYTE(irq)))
+X return -EIO;
+X if (msnd_write_cfg(cfg, IREG_IRQ_TYPE, IRQTYPE_EDGE))
+X return -EIO;
+X return 0;
+}
+X
+static int msnd_read_cfg_irq(int cfg, int num, WORD *irq)
+{
+X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X return -EIO;
+X
+X *irq = msnd_read_cfg(cfg, IREG_IRQ_NUMBER);
+X
+X return 0;
+}
+X
+static int msnd_write_cfg_mem(int cfg, int num, int mem)
+{
+X WORD wmem;
+X
+X mem >>= 8;
+X mem &= 0xfff;
+X wmem = (WORD)mem;
+X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X return -EIO;
+X if (msnd_write_cfg(cfg, IREG_MEMBASEHI, HIBYTE(wmem)))
+X return -EIO;
+X if (msnd_write_cfg(cfg, IREG_MEMBASELO, LOBYTE(wmem)))
+X return -EIO;
+X if (wmem && msnd_write_cfg(cfg, IREG_MEMCONTROL, (MEMTYPE_HIADDR | MEMTYPE_16BIT)))
+X return -EIO;
+X return 0;
+}
+X
+static int msnd_read_cfg_mem(int cfg, int num, int *mem)
+{
+X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X return -EIO;
+X
+X *mem = MAKEWORD(msnd_read_cfg(cfg, IREG_MEMBASELO),
+X msnd_read_cfg(cfg, IREG_MEMBASEHI));
+X *mem <<= 8;
+X
+X return 0;
+}
+X
+static int msnd_activate_logical(int cfg, int num)
+{
+X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X return -EIO;
+X if (msnd_write_cfg(cfg, IREG_ACTIVATE, LD_ACTIVATE))
+X return -EIO;
+X return 0;
+}
+X
+static int msnd_write_cfg_logical(int cfg, int num, WORD io0, WORD io1, WORD irq, int mem)
+{
+X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X return -EIO;
+X if (msnd_write_cfg_io0(cfg, num, io0))
+X return -EIO;
+X if (msnd_write_cfg_io1(cfg, num, io1))
+X return -EIO;
+X if (msnd_write_cfg_irq(cfg, num, irq))
+X return -EIO;
+X if (msnd_write_cfg_mem(cfg, num, mem))
+X return -EIO;
+X if (msnd_activate_logical(cfg, num))
+X return -EIO;
+X return 0;
+}
+X
+static int msnd_read_cfg_logical(int cfg, int num, WORD *io0, WORD *io1, WORD *irq, int *mem)
+{
+X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X return -EIO;
+X if (msnd_read_cfg_io0(cfg, num, io0))
+X return -EIO;
+X if (msnd_read_cfg_io1(cfg, num, io1))
+X return -EIO;
+X if (msnd_read_cfg_irq(cfg, num, irq))
+X return -EIO;
+X if (msnd_read_cfg_mem(cfg, num, mem))
+X return -EIO;
+X return 0;
+}
+X
+static void usage(void)
+{
+X fprintf(stderr,
+X "\n"
+X "pinnaclecfg 1.0\n"
+X "\n"
+X "usage: pinnaclecfg [device config]\n"
+X "\n"
+X "This is for use with the card in NON-PnP mode only.\n"
+X "\n"
+X "Available devices (not all available for Fiji):\n"
+X "\n"
+X " Device Description\n"
+X " -------------------------------------------------------------------\n"
+X " reset Reset all devices (i.e. disable)\n"
+X " show Display current device configurations\n"
+X "\n"
+X " dsp Audio device\n"
+X " mpu Internal Kurzweil synth\n"
+X " ide On-board IDE controller\n"
+X " joystick Joystick port\n"
+X "\n");
+X exit(1);
+}
+X
+static int cfg_reset(void)
+{
+X int i;
+X
+X for (i = 0; i < 4; ++i)
+X msnd_write_cfg_logical(config_port, i, 0, 0, 0, 0);
+X
+X return 0;
+}
+X
+static int cfg_show(void)
+{
+X int i;
+X int count = 0;
+X
+X for (i = 0; i < 4; ++i) {
+X WORD io0, io1, irq;
+X int mem;
+X msnd_read_cfg_logical(config_port, i, &io0, &io1, &irq, &mem);
+X switch (i) {
+X case 0:
+X if (io0 || irq || mem) {
+X printf("dsp 0x%x %d 0x%x\n", io0, irq, mem);
+X ++count;
+X }
+X break;
+X case 1:
+X if (io0 || irq) {
+X printf("mpu 0x%x %d\n", io0, irq);
+X ++count;
+X }
+X break;
+X case 2:
+X if (io0 || io1 || irq) {
+X printf("ide 0x%x 0x%x %d\n", io0, io1, irq);
+X ++count;
+X }
+X break;
+X case 3:
+X if (io0) {
+X printf("joystick 0x%x\n", io0);
+X ++count;
+X }
+X break;
+X }
+X }
+X
+X if (count == 0)
+X fprintf(stderr, "no devices configured\n");
+X
+X return 0;
+}
+X
+static int cfg_dsp(int argc, char *argv[])
+{
+X int io, irq, mem;
+X
+X if (argc < 3 ||
+X sscanf(argv[0], "0x%x", &io) != 1 ||
+X sscanf(argv[1], "%d", &irq) != 1 ||
+X sscanf(argv[2], "0x%x", &mem) != 1)
+X usage();
+X
+X if (!(io == 0x290 ||
+X io == 0x260 ||
+X io == 0x250 ||
+X io == 0x240 ||
+X io == 0x230 ||
+X io == 0x220 ||
+X io == 0x210 ||
+X io == 0x3e0)) {
+X fprintf(stderr, "error: io must be one of "
+X "210, 220, 230, 240, 250, 260, 290, or 3E0\n");
+X usage();
+X }
+X
+X if (!(irq == 5 ||
+X irq == 7 ||
+X irq == 9 ||
+X irq == 10 ||
+X irq == 11 ||
+X irq == 12)) {
+X fprintf(stderr, "error: irq must be one of "
+X "5, 7, 9, 10, 11 or 12\n");
+X usage();
+X }
+X
+X if (!(mem == 0xb0000 ||
+X mem == 0xc8000 ||
+X mem == 0xd0000 ||
+X mem == 0xd8000 ||
+X mem == 0xe0000 ||
+X mem == 0xe8000)) {
+X fprintf(stderr, "error: mem must be one of "
+X "0xb0000, 0xc8000, 0xd0000, 0xd8000, 0xe0000 or 0xe8000\n");
+X usage();
+X }
+X
+X return msnd_write_cfg_logical(config_port, 0, io, 0, irq, mem);
+}
+X
+static int cfg_mpu(int argc, char *argv[])
+{
+X int io, irq;
+X
+X if (argc < 2 ||
+X sscanf(argv[0], "0x%x", &io) != 1 ||
+X sscanf(argv[1], "%d", &irq) != 1)
+X usage();
+X
+X return msnd_write_cfg_logical(config_port, 1, io, 0, irq, 0);
+}
+X
+static int cfg_ide(int argc, char *argv[])
+{
+X int io0, io1, irq;
+X
+X if (argc < 3 ||
+X sscanf(argv[0], "0x%x", &io0) != 1 ||
+X sscanf(argv[0], "0x%x", &io1) != 1 ||
+X sscanf(argv[1], "%d", &irq) != 1)
+X usage();
+X
+X return msnd_write_cfg_logical(config_port, 2, io0, io1, irq, 0);
+}
+X
+static int cfg_joystick(int argc, char *argv[])
+{
+X int io;
+X
+X if (argc < 1 ||
+X sscanf(argv[0], "0x%x", &io) != 1)
+X usage();
+X
+X return msnd_write_cfg_logical(config_port, 3, io, 0, 0, 0);
+}
+X
+int main(int argc, char *argv[])
+{
+X char *device;
+X int rv = 0;
+X
+X --argc; ++argv;
+X
+X if (argc < 2)
+X usage();
+X
+X sscanf(argv[0], "0x%x", &config_port);
+X if (config_port != 0x250 && config_port != 0x260 && config_port != 0x270) {
+X fprintf(stderr, "error: must be 0x250, 0x260 or 0x270\n");
+X exit(1);
+X }
+X if (ioperm(config_port, 2, 1)) {
+X perror("ioperm");
+X fprintf(stderr, "note: pinnaclecfg must be run as root\n");
+X exit(1);
+X }
+X device = argv[1];
+X
+X argc -= 2; argv += 2;
+X
+X if (strcmp(device, "reset") == 0)
+X rv = cfg_reset();
+X else if (strcmp(device, "show") == 0)
+X rv = cfg_show();
+X else if (strcmp(device, "dsp") == 0)
+X rv = cfg_dsp(argc, argv);
+X else if (strcmp(device, "mpu") == 0)
+X rv = cfg_mpu(argc, argv);
+X else if (strcmp(device, "ide") == 0)
+X rv = cfg_ide(argc, argv);
+X else if (strcmp(device, "joystick") == 0)
+X rv = cfg_joystick(argc, argv);
+X else {
+X fprintf(stderr, "error: unknown device %s\n", device);
+X usage();
+X }
+X
+X if (rv)
+X fprintf(stderr, "error: device configuration failed\n");
+X
+X return 0;
+}
+SHAR_EOF
+ $shar_touch -am 1204092598 'MultiSound.d/pinnaclecfg.c' &&
+ chmod 0664 'MultiSound.d/pinnaclecfg.c' ||
+ $echo 'restore of' 'MultiSound.d/pinnaclecfg.c' 'failed'
+ if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \
+ && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then
+ md5sum -c << SHAR_EOF >/dev/null 2>&1 \
+ || $echo 'MultiSound.d/pinnaclecfg.c:' 'MD5 check failed'
+366bdf27f0db767a3c7921d0a6db20fe MultiSound.d/pinnaclecfg.c
+SHAR_EOF
+ else
+ shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/pinnaclecfg.c'`"
+ test 10235 -eq "$shar_count" ||
+ $echo 'MultiSound.d/pinnaclecfg.c:' 'original size' '10235,' 'current size' "$shar_count!"
+ fi
+fi
+# ============= MultiSound.d/Makefile ==============
+if test -f 'MultiSound.d/Makefile' && test "$first_param" != -c; then
+ $echo 'x -' SKIPPING 'MultiSound.d/Makefile' '(file already exists)'
+else
+ $echo 'x -' extracting 'MultiSound.d/Makefile' '(text)'
+ sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/Makefile' &&
+CC = gcc
+CFLAGS = -O
+PROGS = setdigital msndreset pinnaclecfg conv
+X
+all: $(PROGS)
+X
+clean:
+X rm -f $(PROGS)
+SHAR_EOF
+ $shar_touch -am 1204092398 'MultiSound.d/Makefile' &&
+ chmod 0664 'MultiSound.d/Makefile' ||
+ $echo 'restore of' 'MultiSound.d/Makefile' 'failed'
+ if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \
+ && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then
+ md5sum -c << SHAR_EOF >/dev/null 2>&1 \
+ || $echo 'MultiSound.d/Makefile:' 'MD5 check failed'
+76ca8bb44e3882edcf79c97df6c81845 MultiSound.d/Makefile
+SHAR_EOF
+ else
+ shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/Makefile'`"
+ test 106 -eq "$shar_count" ||
+ $echo 'MultiSound.d/Makefile:' 'original size' '106,' 'current size' "$shar_count!"
+ fi
+fi
+# ============= MultiSound.d/conv.l ==============
+if test -f 'MultiSound.d/conv.l' && test "$first_param" != -c; then
+ $echo 'x -' SKIPPING 'MultiSound.d/conv.l' '(file already exists)'
+else
+ $echo 'x -' extracting 'MultiSound.d/conv.l' '(text)'
+ sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/conv.l' &&
+%%
+[ \n\t,\r]
+\;.*
+DB
+[0-9A-Fa-f]+H { int n; sscanf(yytext, "%xH", &n); printf("%c", n); }
+%%
+int yywrap() { return 1; }
+main() { yylex(); }
+SHAR_EOF
+ $shar_touch -am 0828231798 'MultiSound.d/conv.l' &&
+ chmod 0664 'MultiSound.d/conv.l' ||
+ $echo 'restore of' 'MultiSound.d/conv.l' 'failed'
+ if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \
+ && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then
+ md5sum -c << SHAR_EOF >/dev/null 2>&1 \
+ || $echo 'MultiSound.d/conv.l:' 'MD5 check failed'
+d2411fc32cd71a00dcdc1f009e858dd2 MultiSound.d/conv.l
+SHAR_EOF
+ else
+ shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/conv.l'`"
+ test 141 -eq "$shar_count" ||
+ $echo 'MultiSound.d/conv.l:' 'original size' '141,' 'current size' "$shar_count!"
+ fi
+fi
+# ============= MultiSound.d/msndreset.c ==============
+if test -f 'MultiSound.d/msndreset.c' && test "$first_param" != -c; then
+ $echo 'x -' SKIPPING 'MultiSound.d/msndreset.c' '(file already exists)'
+else
+ $echo 'x -' extracting 'MultiSound.d/msndreset.c' '(text)'
+ sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/msndreset.c' &&
+/*********************************************************************
+X *
+X * msndreset.c - resets the MultiSound card
+X *
+X * Copyright (C) 1998 Andrew Veliath
+X *
+X * This program is free software; you can redistribute it and/or modify
+X * it under the terms of the GNU General Public License as published by
+X * the Free Software Foundation; either version 2 of the License, or
+X * (at your option) any later version.
+X *
+X * This program is distributed in the hope that it will be useful,
+X * but WITHOUT ANY WARRANTY; without even the implied warranty of
+X * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+X * GNU General Public License for more details.
+X *
+X * You should have received a copy of the GNU General Public License
+X * along with this program; if not, write to the Free Software
+X * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+X *
+X ********************************************************************/
+X
+#include
+#include
+#include
+#include
+#include
+#include
+#include
+X
+int main(int argc, char *argv[])
+{
+X int fd;
+X
+X if (argc != 2) {
+X fprintf(stderr, "usage: msndreset \n");
+X exit(1);
+X }
+X
+X if ((fd = open(argv[1], O_RDWR)) < 0) {
+X perror(argv[1]);
+X exit(1);
+X }
+X
+X if (ioctl(fd, SOUND_MIXER_PRIVATE1, 0) < 0) {
+X fprintf(stderr, "error: msnd ioctl reset failed\n");
+X perror("ioctl");
+X close(fd);
+X exit(1);
+X }
+X
+X close(fd);
+X
+X return 0;
+}
+SHAR_EOF
+ $shar_touch -am 1204100698 'MultiSound.d/msndreset.c' &&
+ chmod 0664 'MultiSound.d/msndreset.c' ||
+ $echo 'restore of' 'MultiSound.d/msndreset.c' 'failed'
+ if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \
+ && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then
+ md5sum -c << SHAR_EOF >/dev/null 2>&1 \
+ || $echo 'MultiSound.d/msndreset.c:' 'MD5 check failed'
+c52f876521084e8eb25e12e01dcccb8a MultiSound.d/msndreset.c
+SHAR_EOF
+ else
+ shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/msndreset.c'`"
+ test 1472 -eq "$shar_count" ||
+ $echo 'MultiSound.d/msndreset.c:' 'original size' '1472,' 'current size' "$shar_count!"
+ fi
+fi
+rm -fr _sh01426
+exit 0
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/NEWS linux-2.5.6/Documentation/sound/oss/NEWS
--- linux-2.5.6-pre3/Documentation/sound/oss/NEWS Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/NEWS Thu Mar 7 18:24:41 2002
@@ -0,0 +1,42 @@
+Linux 2.4 Sound Changes
+2000-September-25
+Christoph Hellwig,
+
+
+
+=== isapnp support
+
+The Linux 2.4 Kernel does have reliable in-kernel isapnp support.
+Some drivers (sb.o, ad1816.o awe_wave.o) do now support automatically
+detecting and configuring isapnp devices.
+If you have a not yet supported isapnp soundcard, mail me the content
+of '/proc/isapnp' on your system and some information about your card
+and its driver(s) so I can try to get isapnp working for it.
+
+
+
+=== soundcard resources on kernel commandline
+
+Before Linux 2.4 you had to specify the resources for sounddrivers
+statically linked into the kernel at compile time
+(in make config/menuconfig/xconfig). In Linux 2.4 the ressources are
+now specified at the boot-time kernel commandline (e.g. the lilo
+'append=' line or everything that's after the kernel name in grub).
+Read the Configure.help entry for your card for the parameters.
+
+
+=== softoss is gone
+
+In Linux 2.4 the softoss in-kernel software synthesizer is no more aviable.
+Use a user space software synthesizer like timidity instead.
+
+
+
+=== /dev/sndstat and /proc/sound are gone
+
+In older Linux versions those files exported some information about the
+OSS/Free configuration to userspace. In Linux 2.3 they were removed because
+they did not support the growing number of pci soundcards and there were
+some general problems with this interface.
+
+
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/NM256 linux-2.5.6/Documentation/sound/oss/NM256
--- linux-2.5.6-pre3/Documentation/sound/oss/NM256 Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/NM256 Thu Mar 7 18:24:41 2002
@@ -0,0 +1,280 @@
+=======================================================
+Documentation for the NeoMagic 256AV/256ZX sound driver
+=======================================================
+
+You're looking at version 1.1 of the driver. (Woohoo!) It has been
+successfully tested against the following laptop models:
+
+ Sony Z505S/Z505SX/Z505DX/Z505RX
+ Sony F150, F160, F180, F250, F270, F280, PCG-F26
+ Dell Latitude CPi, CPt (various submodels)
+
+There are a few caveats, which is why you should read the entirety of
+this document first.
+
+This driver was developed without any support or assistance from
+NeoMagic. There is no warranty, expressed, implied, or otherwise. It
+is free software in the public domain; feel free to use it, sell it,
+give it to your best friends, even claim that you wrote it (but why?!)
+but don't go whining to me, NeoMagic, Sony, Dell, or anyone else
+when it blows up your computer.
+
+Version 1.1 contains a change to try and detect non-AC97 versions of
+the hardware, and not install itself appropriately. It should also
+reinitialize the hardware on an APM resume event, assuming that APM
+was configured into your kernel.
+
+============
+Installation
+============
+
+Enable the sound drivers, the OSS sound drivers, and then the NM256
+driver. The NM256 driver *must* be configured as a module (it won't
+give you any other choice).
+
+Next, do the usual "make modules" and "make modules_install".
+Finally, insmod the soundcore, sound and nm256 modules.
+
+When the nm256 driver module is loaded, you should see a couple of
+confirmation messages in the kernel logfile indicating that it found
+the device (the device does *not* use any I/O ports or DMA channels).
+Now try playing a wav file, futz with the CD-ROM if you have one, etc.
+
+The NM256 is entirely a PCI-based device, and all the necessary
+information is automatically obtained from the card. It can only be
+configured as a module in a vain attempt to prevent people from
+hurting themselves. It works correctly if it shares an IRQ with
+another device (it normally shares IRQ 9 with the builtin eepro100
+ethernet on the Sony Z505 laptops).
+
+It does not run the card in any sort of compatibility mode. It will
+not work on laptops that have the SB16-compatible, AD1848-compatible
+or CS4232-compatible codec/mixer; you will want to use the appropriate
+compatible OSS driver with these chipsets. I cannot provide any
+assistance with machines using the SB16, AD1848 or CS4232 compatible
+versions. (The driver now attempts to detect the mixer version, and
+will refuse to load if it believes the hardware is not
+AC97-compatible.)
+
+The sound support is very basic, but it does include simultaneous
+playback and record capability. The mixer support is also quite
+simple, although this is in keeping with the rather limited
+functionality of the chipset.
+
+There is no hardware synthesizer available, as the Losedows OPL-3 and
+MIDI support is done via hardware emulation.
+
+Only three recording devices are available on the Sony: the
+microphone, the CD-ROM input, and the volume device (which corresponds
+to the stereo output). (Other devices may be available on other
+models of laptops.) The Z505 series does not have a builtin CD-ROM,
+so of course the CD-ROM input doesn't work. It does work on laptops
+with a builtin CD-ROM drive.
+
+The mixer device does not appear to have any tone controls, at least
+on the Z505 series. The mixer module checks for tone controls in the
+AC97 mixer, and will enable them if they are available.
+
+==============
+Known problems
+==============
+
+ * There are known problems with PCMCIA cards and the eepro100 ethernet
+ driver on the Z505S/Z505SX/Z505DX. Keep reading.
+
+ * There are also potential problems with using a virtual X display, and
+ also problems loading the module after the X server has been started.
+ Keep reading.
+
+ * The volume control isn't anywhere near linear. Sorry. This will be
+ fixed eventually, when I get sufficiently annoyed with it. (I doubt
+ it will ever be fixed now, since I've never gotten sufficiently
+ annoyed with it and nobody else seems to care.)
+
+ * There are reports that the CD-ROM volume is very low. Since I do not
+ have a CD-ROM equipped laptop, I cannot test this (it's kinda hard to
+ do remotely).
+
+ * Only 8 fixed-rate speeds are supported. This is mainly a chipset
+ limitation. It may be possible to support other speeds in the future.
+
+ * There is no support for the telephone mixer/codec. There is support
+ for a phonein/phoneout device in the mixer driver; whether or not
+ it does anything is anyone's guess. (Reports on this would be
+ appreciated. You'll have to figure out how to get the phone to
+ go off-hook before it'll work, tho.)
+
+ * This driver was not written with any cooperation or support from
+ NeoMagic. If you have any questions about this, see their website
+ for their official stance on supporting open source drivers.
+
+============
+Video memory
+============
+
+The NeoMagic sound engine uses a portion of the display memory to hold
+the sound buffer. (Crazy, eh?) The NeoMagic video BIOS sets up a
+special pointer at the top of video RAM to indicate where the top of
+the audio buffer should be placed.
+
+At the present time XFree86 is apparently not aware of this. It will
+thus write over either the pointer or the sound buffer with abandon.
+(Accelerated-X seems to do a better job here.)
+
+This implies a few things:
+
+ * Sometimes the NM256 driver has to guess at where the buffer
+ should be placed, especially if the module is loaded after the
+ X server is started. It's usually correct, but it will consistently
+ fail on the Sony F250.
+
+ * Virtual screens greater than 1024x768x16 under XFree86 are
+ problematic on laptops with only 2.5MB of screen RAM. This
+ includes all of the 256AV-equipped laptops. (Virtual displays
+ may or may not work on the 256ZX, which has at least 4MB of
+ video RAM.)
+
+If you start having problems with random noise being output either
+constantly (this is the usual symptom on the F250), or when windows
+are moved around (this is the usual symptom when using a virtual
+screen), the best fix is to
+
+ * Don't use a virtual frame buffer.
+ * Make sure you load the NM256 module before the X server is
+ started.
+
+On the F250, it is possible to force the driver to load properly even
+after the XFree86 server is started by doing:
+
+ insmod nm256 buffertop=0x25a800
+
+This forces the audio buffers to the correct offset in screen RAM.
+
+One user has reported a similar problem on the Sony F270, although
+others apparently aren't seeing any problems. His suggested command
+is
+
+ insmod nm256 buffertop=0x272800
+
+=================
+Official WWW site
+=================
+
+The official site for the NM256 driver is:
+
+ http://www.uglx.org/sony.html
+
+You should always be able to get the latest version of the driver there,
+and the driver will be supported for the foreseeable future.
+
+==============
+Z505RX and IDE
+==============
+
+There appears to be a problem with the IDE chipset on the Z505RX; one
+of the symptoms is that sound playback periodically hangs (when the
+disk is accessed). The user reporting the problem also reported that
+enabling all of the IDE chipset workarounds in the kernel solved the
+problem, tho obviously only one of them should be needed--if someone
+can give me more details I would appreciate it.
+
+==============================
+Z505S/Z505SX on-board Ethernet
+==============================
+
+If you're using the on-board Ethernet Pro/100 ethernet support on the Z505
+series, I strongly encourage you to download the latest eepro100 driver from
+Donald Becker's site:
+
+ ftp://cesdis.gsfc.nasa.gov/pub/linux/drivers/test/eepro100.c
+
+There was a reported problem on the Z505SX that if the ethernet
+interface is disabled and reenabled while the sound driver is loaded,
+the machine would lock up. I have included a workaround that is
+working satisfactorily. However, you may occasionally see a message
+about "Releasing interrupts, over 1000 bad interrupts" which indicates
+that the workaround is doing its job.
+
+==================================
+PCMCIA and the Z505S/Z505SX/Z505DX
+==================================
+
+There is also a known problem with the Sony Z505S and Z505SX hanging
+if a PCMCIA card is inserted while the ethernet driver is loaded, or
+in some cases if the laptop is suspended. This is caused by tons of
+spurious IRQ 9s, probably generated from the PCMCIA or ACPI bridges.
+
+There is currently no fix for the problem that works in every case.
+The only known workarounds are to disable the ethernet interface
+before inserting or removing a PCMCIA card, or with some cards
+disabling the PCMCIA card before ejecting it will also help the
+problem with the laptop hanging when the card is ejected.
+
+One user has reported that setting the tcic's cs_irq to some value
+other than 9 (like 11) fixed the problem. This doesn't work on my
+Z505S, however--changing the value causes the cardmgr to stop seeing
+card insertions and removals, cards don't seem to work correctly, and
+I still get hangs if a card is inserted when the kernel is booted.
+
+Using the latest ethernet driver and pcmcia package allows me to
+insert an Adaptec 1480A SlimScsi card without the laptop hanging,
+although I still have to shut down the card before ejecting or
+powering down the laptop. However, similar experiments with a DE-660
+ethernet card still result in hangs when the card is inserted. I am
+beginning to think that the interrupts are CardBus-related, since the
+Adaptec card is a CardBus card, and the DE-660 is not; however, I
+don't have any other CardBus cards to test with.
+
+======
+Thanks
+======
+
+First, I want to thank everyone (except NeoMagic of course) for their
+generous support and encouragement. I'd like to list everyone's name
+here that replied during the development phase, but the list is
+amazingly long.
+
+I will be rather unfair and single out a few people, however:
+
+ Justin Maurer, for being the first random net.person to try it,
+ and for letting me login to his Z505SX to get it working there
+
+ Edi Weitz for trying out several different versions, and giving
+ me a lot of useful feedback
+
+ Greg Rumple for letting me login remotely to get the driver
+ functional on the 256ZX, for his assistance on tracking
+ down all sorts of random stuff, and for trying out Accel-X
+
+ Zach Brown, for the initial AC97 mixer interface design
+
+ Jeff Garzik, for various helpful suggestions on the AC97
+ interface
+
+ "Mr. Bumpy" for feedback on the Z505RX
+
+ Bill Nottingham, for generous assistance in getting the mixer ID
+ code working
+
+=================
+Previous versions
+=================
+
+Versions prior to 0.3 (aka `noname') had problems with weird artifacts
+in the output and failed to set the recording rate properly. These
+problems have long since been fixed.
+
+Versions prior to 0.5 had problems with clicks in the output when
+anything other than 16-bit stereo sound was being played, and also had
+periodic clicks when recording.
+
+Version 0.7 first incorporated support for the NM256ZX chipset, which
+is found on some Dell Latitude laptops (the CPt, and apparently
+some CPi models as well). It also included the generic AC97
+mixer module.
+
+Version 0.75 renamed all the functions and files with slightly more
+generic names.
+
+Note that previous versions of this document claimed that recording was
+8-bit only; it actually has been working for 16-bits all along.
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/OPL3 linux-2.5.6/Documentation/sound/oss/OPL3
--- linux-2.5.6-pre3/Documentation/sound/oss/OPL3 Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/OPL3 Thu Mar 7 18:24:41 2002
@@ -0,0 +1,6 @@
+A pure OPL3 card is nice and easy to configure. Simply do
+
+insmod opl3 io=0x388
+
+Change the I/O address in the very unlikely case this card is differently
+configured
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/OPL3-SA linux-2.5.6/Documentation/sound/oss/OPL3-SA
--- linux-2.5.6-pre3/Documentation/sound/oss/OPL3-SA Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/OPL3-SA Thu Mar 7 18:24:41 2002
@@ -0,0 +1,52 @@
+OPL3-SA1 sound driver (opl3sa.o)
+
+---
+Note: This howto only describes how to setup the OPL3-SA1 chip; this info
+does not apply to the SA2, SA3, or SA4.
+---
+
+The Yamaha OPL3-SA1 sound chip is usually found built into motherboards, and
+it's a decent little chip offering a WSS mode, a SB Pro emulation mode, MPU401
+and OPL3 FM Synth capabilities.
+
+You can enable inclusion of the driver via CONFIG_SOUND_OPL3SA1=m, or
+CONFIG_SOUND_OPL3SA1=y through 'make config/xconfig/menuconfig'.
+
+You'll need to know all of the relevant info (irq, dma, and io port) for the
+chip's WSS mode, since that is the mode the kernel sound driver uses, and of
+course you'll also need to know about where the MPU401 and OPL3 ports and
+IRQs are if you want to use those.
+
+Here's the skinny on how to load it as a module:
+
+ modprobe opl3sa io=0x530 irq=11 dma=0 dma2=1 mpu_io=0x330 mpu_irq=5
+
+Module options in detail:
+
+ io: This is the WSS's port base.
+ irq: This is the WSS's IRQ.
+ dma: This is the WSS's DMA line. In my BIOS setup screen this was
+ listed as "WSS Play DMA"
+ dma2: This is the WSS's secondary DMA line. My BIOS calls it the
+ "WSS capture DMA"
+
+ mpu_io: This is the MPU401's port base.
+ mpu_irq: This is the MPU401's IRQ.
+
+If you'd like to use the OPL3 FM Synthesizer, make sure you enable
+CONFIG_YM3812 (in 'make config'). That'll build the opl3.o module.
+
+Then a simple 'insmod opl3 io=0x388', and you now have FM Synth.
+
+You can also use the SoftOSS software synthesizer instead of the builtin OPL3.
+Here's how:
+
+Say 'y' or 'm' to "SoftOSS software wave table engine" in make config.
+
+If you said yes, the software synth is available once you boot your new
+kernel.
+
+If you chose to build it as a module, just insmod the resulting softoss2.o
+
+Questions? Comments?
+
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/OPL3-SA2 linux-2.5.6/Documentation/sound/oss/OPL3-SA2
--- linux-2.5.6-pre3/Documentation/sound/oss/OPL3-SA2 Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/OPL3-SA2 Thu Mar 7 18:24:41 2002
@@ -0,0 +1,210 @@
+Documentation for the OPL3-SA2, SA3, and SAx driver (opl3sa2.o)
+---------------------------------------------------------------
+
+Scott Murray, scott@spiteful.org
+January 7, 2001
+
+NOTE: All trade-marked terms mentioned below are properties of their
+ respective owners.
+
+
+Supported Devices
+-----------------
+
+This driver is for PnP soundcards based on the following Yamaha audio
+controller chipsets:
+
+YMF711 aka OPL3-SA2
+YMF715 and YMF719 aka OPL3-SA3
+
+Up until recently (December 2000), I'd thought the 719 to be a
+different chipset, the OPL3-SAx. After an email exhange with
+Yamaha, however, it turns out that the 719 is just a re-badged
+715, and the chipsets are identical. The chipset detection code
+has been updated to reflect this.
+
+Anyways, all of these chipsets implement the following devices:
+
+OPL3 FM synthesizer
+Soundblaster Pro
+Microsoft/Windows Sound System
+MPU401 MIDI interface
+
+Note that this driver uses the MSS device, and to my knowledge these
+chipsets enforce an either/or situation with the Soundblaster Pro
+device and the MSS device. Since the MSS device has better
+capabilities, I have implemented the driver to use it.
+
+
+Mixer Channels
+--------------
+
+Older versions of this driver (pre-December 2000) had two mixers,
+an OPL3-SA2 or SA3 mixer and a MSS mixer. The OPL3-SA[23] mixer
+device contained a superset of mixer channels consisting of its own
+channels and all of the MSS mixer channels. To simplify the driver
+considerably, and to partition functionality better, the OPL3-SA[23]
+mixer device now contains has its own specific mixer channels. They
+are:
+
+Volume - Hardware master volume control
+Bass - SA3 only, now supports left and right channels
+Treble - SA3 only, now supports left and right channels
+Microphone - Hardware microphone input volume control
+Digital1 - Yamaha 3D enhancement "Wide" mixer
+
+All other mixer channels (e.g. "PCM", "CD", etc.) now have to be
+controlled via the "MS Sound System (CS4231)" mixer. To facilitate
+this, the mixer device creation order has been switched so that
+the MSS mixer is created first. This allows accessing the majority
+of the useful mixer channels even via single mixer-aware tools
+such as "aumix".
+
+
+Plug 'n Play
+------------
+
+In previous kernels (2.2.x), some configuration was required to
+get the driver to talk to the card. Being the new millennium and
+all, the 2.4.x kernels now support auto-configuration if ISA PnP
+support is configured in. Theoretically, the driver even supports
+having more than one card in this case.
+
+With the addition of PnP support to the driver, two new parameters
+have been added to control it:
+
+isapnp - set to 0 to disable ISA PnP card detection
+
+multiple - set to 0 to disable multiple PnP card detection
+
+
+Optional Parameters
+-------------------
+
+Recent (December 2000) additions to the driver (based on a patch
+provided by Peter Englmaier) are two new parameters:
+
+ymode - Set Yamaha 3D enhancement mode:
+ 0 = Desktop/Normal 5-12 cm speakers
+ 1 = Notebook PC (1) 3 cm speakers
+ 2 = Notebook PC (2) 1.5 cm speakers
+ 3 = Hi-Fi 16-38 cm speakers
+
+loopback - Set A/D input source. Useful for echo cancellation:
+ 0 = Mic Right channel (default)
+ 1 = Mono output loopback
+
+The ymode parameter has been tested and does work. The loopback
+parameter, however, is untested. Any feedback on its usefulness
+would be appreciated.
+
+
+Manual Configuration
+--------------------
+
+If for some reason you decide not to compile ISA PnP support into
+your kernel, or disabled the driver's usage of it by setting the
+isapnp parameter as discussed above, then you will need to do some
+manual configuration. There are two ways of doing this. The most
+common is to use the isapnptools package to initialize the card, and
+use the kernel module form of the sound subsystem and sound drivers.
+Alternatively, some BIOS's allow manual configuration of installed
+PnP devices in a BIOS menu, which should allow using the non-modular
+sound drivers, i.e. built into the kernel.
+
+I personally use isapnp and modules, and do not have access to a PnP
+BIOS machine to test. If you have such a beast, configuring the
+driver to be built into the kernel should just work (thanks to work
+done by David Luyer ). You will still need
+to specify settings, which can be done by adding:
+
+opl3sa2=,,,,,
+
+to the kernel command line. For example:
+
+opl3sa2=0x370,5,0,1,0x530,0x330
+
+If you are instead using the isapnp tools (as most people have been
+before Linux 2.4.x), follow the directions in their documentation to
+produce a configuration file. Here is the relevant excerpt I used to
+use for my SA3 card from my isapnp.conf:
+
+(CONFIGURE YMH0800/-1 (LD 0
+
+# NOTE: IO 0 is for the unused SoundBlaster part of the chipset.
+(IO 0 (BASE 0x0220))
+(IO 1 (BASE 0x0530))
+(IO 2 (BASE 0x0388))
+(IO 3 (BASE 0x0330))
+(IO 4 (BASE 0x0370))
+(INT 0 (IRQ 5 (MODE +E)))
+(DMA 0 (CHANNEL 0))
+(DMA 1 (CHANNEL 1))
+
+Here, note that:
+
+Port Acceptable Range Purpose
+---- ---------------- -------
+IO 0 0x0220 - 0x0280 SB base address, unused.
+IO 1 0x0530 - 0x0F48 MSS base address
+IO 2 0x0388 - 0x03F8 OPL3 base address
+IO 3 0x0300 - 0x0334 MPU base address
+IO 4 0x0100 - 0x0FFE card's own base address for its control I/O ports
+
+The IRQ and DMA values can be any that are considered acceptable for a
+MSS. Assuming you've got isapnp all happy, then you should be able to
+do something like the following (which matches up with the isapnp
+configuration above):
+
+modprobe mpu401
+modprobe ad1848
+modprobe opl3sa2 io=0x370 mss_io=0x530 mpu_io=0x330 irq=5 dma=0 dma2=1
+modprobe opl3 io=0x388
+
+See the section "Automatic Module Loading" below for how to set up
+/etc/modules.conf to automate this.
+
+An important thing to remember that the opl3sa2 module's io argument is
+for it's own control port, which handles the card's master mixer for
+volume (on all cards), and bass and treble (on SA3 cards).
+
+
+Troubleshooting
+---------------
+
+If all goes well and you see no error messages, you should be able to
+start using the sound capabilities of your system. If you get an
+error message while trying to insert the opl3sa2 module, then make
+sure that the values of the various arguments match what you specified
+in your isapnp configuration file, and that there is no conflict with
+another device for an I/O port or interrupt. Checking the contents of
+/proc/ioports and /proc/interrupts can be useful to see if you're
+butting heads with another device.
+
+If you still cannot get the module to load, look at the contents of
+your system log file, usually /var/log/messages. If you see the
+message "opl3sa2: Unknown Yamaha audio controller version", then you
+have a different chipset version than I've encountered so far. Look
+for all messages in the log file that start with "opl3sa2: " and see
+if they provide any clues. If you do not see the chipset version
+message, and none of the other messages present in the system log are
+helpful, email me some details and I'll try my best to help.
+
+
+Automatic Module Loading
+------------------------
+
+Lastly, if you're using modules and want to set up automatic module
+loading with kmod, the kernel module loader, here is the section I
+currently use in my modules.conf file:
+
+# Sound
+alias sound-slot-0 opl3sa2
+options opl3sa2 io=0x370 mss_io=0x530 mpu_io=0x330 irq=7 dma=0 dma2=3
+options opl3 io=0x388
+
+That's all it currently takes to get an OPL3-SA3 card working on my
+system. Once again, if you have any other problems, email me at the
+address listed above.
+
+Scott
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/Opti linux-2.5.6/Documentation/sound/oss/Opti
--- linux-2.5.6-pre3/Documentation/sound/oss/Opti Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/Opti Thu Mar 7 18:24:41 2002
@@ -0,0 +1,222 @@
+Support for the OPTi 82C931 chip
+--------------------------------
+Note: parts of this README file apply also to other
+cards that use the mad16 driver.
+
+Some items in this README file are based on features
+added to the sound driver after Linux-2.1.91 was out.
+By the time of writing this I do not know which official
+kernel release will include these features.
+Please do not report inconsistencies on older Linux
+kernels.
+
+The OPTi 82C931 is supported in its non-PnP mode.
+Usually you do not need to set jumpers, etc. The sound driver
+will check the card status and if it is required it will
+force the card into a mode in which it can be programmed.
+
+If you have another OS installed on your computer it is recommended
+that Linux and the other OS use the same resources.
+
+Also, it is recommended that resources specified in /etc/modules.conf
+and resources specified in /etc/isapnp.conf agree.
+
+Compiling the sound driver
+--------------------------
+I highly recommend that you build a modularized sound driver.
+This document does not cover a sound-driver which is built in
+the kernel.
+
+Sound card support should be enabled as a module (chose m).
+Answer 'm' for these items:
+ Generic OPL2/OPL3 FM synthesizer support (CONFIG_SOUND_ADLIB)
+ Microsoft Sound System support (CONFIG_SOUND_MSS)
+ Support for OPTi MAD16 and/or Mozart based cards (CONFIG_SOUND_MAD16)
+ FM synthesizer (YM3812/OPL-3) support (CONFIG_SOUND_YM3812)
+
+The configuration menu may ask for addresses, IRQ lines or DMA
+channels. If the card is used as a module the module loading
+options will override these values.
+
+For the OPTi 931 you can answer 'n' to:
+ Support MIDI in older MAD16 based cards (requires SB) (CONFIG_SOUND_MAD16_OLDCARD)
+If you do need MIDI support in a Mozart or C928 based card you
+need to answer 'm' to the above question. In that case you will
+also need to answer 'm' to:
+ '100% Sound Blaster compatibles (SB16/32/64, ESS, Jazz16) support' (CONFIG_SOUND_SB)
+
+Go on and compile your kernel and modules. Install the modules. Run depmod -a.
+
+Using isapnptools
+-----------------
+In most systems with a PnP BIOS you do not need to use isapnp. The
+initialization provided by the BIOS is sufficient for the driver
+to pick up the card and continue initialization.
+
+If that fails, or if you have other PnP cards, you need to use isapnp
+to initialize the card.
+This was tested with isapnptools-1.11 but I recommend that you use
+isapnptools-1.13 (or newer). Run pnpdump to dump the information
+about your PnP cards. Then edit the resulting file and select
+the options of your choice. This file is normally installed as
+/etc/isapnp.conf.
+
+The driver has one limitation with respect to I/O port resources:
+IO3 base must be 0x0E0C. Although isapnp allows other ports, this
+address is hard-coded into the driver.
+
+Using kmod and autoloading the sound driver
+-------------------------------------------
+Comment: as of linux-2.1.90 kmod is replacing kerneld.
+The config file '/etc/modules.conf' is used as before.
+
+This is the sound part of my /etc/modules.conf file.
+Following that I will explain each line.
+
+alias mixer0 mad16
+alias audio0 mad16
+alias midi0 mad16
+alias synth0 opl3
+options sb mad16=1
+options mad16 irq=10 dma=0 dma16=1 io=0x530 joystick=1 cdtype=0
+options opl3 io=0x388
+post-install mad16 /sbin/ad1848_mixer_reroute 14 8 15 3 16 6
+
+If you have an MPU daughtercard or onboard MPU you will want to add to the
+"options mad16" line - eg
+
+options mad16 irq=5 dma=0 dma16=3 io=0x530 mpu_io=0x330 mpu_irq=9
+
+To set the I/O and IRQ of the MPU.
+
+
+Explain:
+
+alias mixer0 mad16
+alias audio0 mad16
+alias midi0 mad16
+alias synth0 opl3
+
+When any sound device is opened the kernel requests auto-loading
+of char-major-14. There is a built-in alias that translates this
+request to loading the main sound module.
+
+The sound module in its turn will request loading of a sub-driver
+for mixer, audio, midi or synthesizer device. The first 3 are
+supported by the mad16 driver. The synth device is supported
+by the opl3 driver.
+
+There is currently no way to autoload the sound device driver
+if more than one card is installed.
+
+options sb mad16=1
+
+This is left for historical reasons. If you enable the
+config option 'Support MIDI in older MAD16 based cards (requires SB)'
+or if you use an older mad16 driver it will force loading of the
+SoundBlaster driver. This option tells the SB driver not to look
+for a SB card but to wait for the mad16 driver.
+
+options mad16 irq=10 dma=0 dma16=1 io=0x530 joystick=1 cdtype=0
+options opl3 io=0x388
+
+post-install mad16 /sbin/ad1848_mixer_reroute 14 8 15 3 16 6
+
+This sets resources and options for the mad16 and opl3 drivers.
+I use two DMA channels (only one is required) to enable full duplex.
+joystick=1 enables the joystick port. cdtype=0 disables the cd port.
+You can also set mpu_io and mpu_irq in the mad16 options for the
+uart401 driver.
+
+This tells modprobe to run /sbin/ad1848_mixer_reroute after
+mad16 is successfully loaded and initialized. The source
+for ad1848_mixer_reroute is appended to the end of this readme
+file. It is impossible for the sound driver to know the actual
+connections to the mixer. The 3 inputs intended for cd, synth
+and line-in are mapped to the generic inputs line1, line2 and
+line3. This program reroutes these mixer channels to their
+right names (note the right mapping depends on the actual sound
+card that you use).
+The numeric parameters mean:
+ 14=line1 8=cd - reroute line1 to the CD input.
+ 15=line2 3=synth - reroute line2 to the synthesizer input.
+ 16=line3 6=line - reroute line3 to the line input.
+For reference on other input names look at the file
+/usr/include/linux/soundcard.h.
+
+Using a joystick
+-----------------
+You must enable a joystick in the mad16 options. (also
+in /etc/isapnp.conf if you use it).
+Tested with regular analog joysticks.
+
+A CDROM drive connected to the sound card
+-----------------------------------------
+The 82C931 chip has support only for secondary ATAPI cdrom.
+(cdtype=8). Loading the mad16 driver resets the C931 chip
+and if a cdrom was already mounted it may cause a complete
+system hang. Do not use the sound card if you have an alternative.
+If you do use the sound card it is important that you load
+the mad16 driver (use "modprobe mad16" to prevent auto-unloading)
+before the cdrom is accessed the first time.
+
+Using the sound driver built-in to the kernel may help here, but...
+Most new systems have a PnP BIOS and also two IDE controllers.
+The IDE controller on the sound card may be needed only on older
+systems (which have only one IDE controller) but these systems
+also do not have a PnP BIOS - requiring isapnptools and a modularized
+driver.
+
+Known problems
+--------------
+1. See the section on "A CDROM drive connected to the sound card".
+
+2. On my system the codec cannot capture companded sound samples.
+ (eg., recording from /dev/audio). When any companded capture is
+ requested I get stereo-16 bit samples instead. Playback of
+ companded samples works well. Apparently this problem is not common
+ to all C931 based cards. I do not know how to identify cards that
+ have this problem.
+
+Source for ad1848_mixer_reroute.c
+---------------------------------
+#include
+#include
+#include
+
+static char *mixer_names[SOUND_MIXER_NRDEVICES] =
+ SOUND_DEVICE_LABELS;
+
+int
+main(int argc, char **argv) {
+ int val, from, to;
+ int i, fd;
+
+ fd = open("/dev/mixer", O_RDWR);
+ if(fd < 0) {
+ perror("/dev/mixer");
+ return 1;
+ }
+
+ for(i = 2; i < argc; i += 2) {
+ from = atoi(argv[i-1]);
+ to = atoi(argv[i]);
+
+ if(to == SOUND_MIXER_NONE)
+ fprintf(stderr, "%s: turning off mixer %s\n",
+ argv[0], mixer_names[to]);
+ else
+ fprintf(stderr, "%s: rerouting mixer %s to %s\n",
+ argv[0], mixer_names[from], mixer_names[to]);
+
+ val = from << 8 | to;
+
+ if(ioctl(fd, SOUND_MIXER_PRIVATE2, &val)) {
+ perror("AD1848 mixer reroute");
+ return 1;
+ }
+ }
+
+ return 0;
+}
+
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/PAS16 linux-2.5.6/Documentation/sound/oss/PAS16
--- linux-2.5.6-pre3/Documentation/sound/oss/PAS16 Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/PAS16 Thu Mar 7 18:24:41 2002
@@ -0,0 +1,163 @@
+Pro Audio Spectrum 16 for 2.3.99 and later
+=========================================
+by Thomas Molina (tmolina@home.com)
+last modified 3 Mar 2001
+Acknowledgement to Axel Boldt (boldt@math.ucsb.edu) for stuff taken
+from Configure.help, Riccardo Facchetti for stuff from README.OSS,
+and others whose names I could not find.
+
+This documentation is relevant for the PAS16 driver (pas2_card.c and
+friends) under kernel version 2.3.99 and later. If you are
+unfamiliar with configuring sound under Linux, please read the
+Sound-HOWTO, linux/Documentation/sound/Introduction and other
+relevant docs first.
+
+The following information is relevant information from README.OSS
+and legacy docs for the Pro Audio Spectrum 16 (PAS16):
+==================================================================
+
+The pas2_card.c driver supports the following cards --
+Pro Audio Spectrum 16 (PAS16) and compatibles:
+ Pro Audio Spectrum 16
+ Pro Audio Studio 16
+ Logitech Sound Man 16
+ NOTE! The original Pro Audio Spectrum as well as the PAS+ are not
+ and will not be supported by the driver.
+
+The sound driver configuration dialog
+-------------------------------------
+
+Sound configuration starts by making some yes/no questions. Be careful
+when answering to these questions since answering y to a question may
+prevent some later ones from being asked. For example don't answer y to
+the question about (PAS16) if you don't really have a PAS16. Sound
+configuration may also be made modular by answering m to configuration
+options presented.
+
+Note also that all questions may not be asked. The configuration program
+may disable some questions depending on the earlier choices. It may also
+select some options automatically as well.
+
+ "ProAudioSpectrum 16 support",
+ - Answer 'y'_ONLY_ if you have a Pro Audio Spectrum _16_,
+ Pro Audio Studio 16 or Logitech SoundMan 16 (be sure that
+ you read the above list correctly). Don't answer 'y' if you
+ have some other card made by Media Vision or Logitech since they
+ are not PAS16 compatible.
+ NOTE! Since 3.5-beta10 you need to enable SB support (next question)
+ if you want to use the SB emulation of PAS16. It's also possible to
+ the emulation if you want to use a true SB card together with PAS16
+ (there is another question about this that is asked later).
+
+ "Generic OPL2/OPL3 FM synthesizer support",
+ - Answer 'y' if your card has a FM chip made by Yamaha (OPL2/OPL3/OPL4).
+ The PAS16 has an OPL3-compatible FM chip.
+
+With PAS16 you can use two audio device files at the same time. /dev/dsp (and
+/dev/audio) is connected to the 8/16 bit native codec and the /dev/dsp1 (and
+/dev/audio1) is connected to the SB emulation (8 bit mono only).
+
+
+The new stuff for 2.3.99 and later
+============================================================================
+The following configuration options from linux/Documentation/Configure.help
+are relevant to configuring the PAS16:
+
+Sound card support
+CONFIG_SOUND
+ If you have a sound card in your computer, i.e. if it can say more
+ than an occasional beep, say Y. Be sure to have all the information
+ about your sound card and its configuration down (I/O port,
+ interrupt and DMA channel), because you will be asked for it.
+
+ You want to read the Sound-HOWTO, available from
+ http://www.linuxdoc.org/docs.html#howto . General information
+ about the modular sound system is contained in the files
+ Documentation/sound/Introduction. The file
+ Documentation/sound/README.OSS contains some slightly outdated but
+ still useful information as well.
+
+OSS sound modules
+CONFIG_SOUND_OSS
+ OSS is the Open Sound System suite of sound card drivers. They make
+ sound programming easier since they provide a common API. Say Y or M
+ here (the module will be called sound.o) if you haven't found a
+ driver for your sound card above, then pick your driver from the
+ list below.
+
+Persistent DMA buffers
+CONFIG_SOUND_DMAP
+ Linux can often have problems allocating DMA buffers for ISA sound
+ cards on machines with more than 16MB of RAM. This is because ISA
+ DMA buffers must exist below the 16MB boundary and it is quite
+ possible that a large enough free block in this region cannot be
+ found after the machine has been running for a while. If you say Y
+ here the DMA buffers (64Kb) will be allocated at boot time and kept
+ until the shutdown. This option is only useful if you said Y to
+ "OSS sound modules", above. If you said M to "OSS sound modules"
+ then you can get the persistent DMA buffer functionality by passing
+ the command-line argument "dmabuf=1" to the sound.o module.
+
+ Say y here for PAS16.
+
+ProAudioSpectrum 16 support
+CONFIG_SOUND_PAS
+ Answer Y only if you have a Pro Audio Spectrum 16, ProAudio Studio
+ 16 or Logitech SoundMan 16 sound card. Don't answer Y if you have
+ some other card made by Media Vision or Logitech since they are not
+ PAS16 compatible. It is not necessary to enable the separate
+ Sound Blaster support; it is included in the PAS driver.
+
+ If you compile the driver into the kernel, you have to add
+ "pas2=,,,,,,,
+ to the kernel command line.
+
+FM Synthesizer (YM3812/OPL-3) support
+CONFIG_SOUND_YM3812
+ Answer Y if your card has a FM chip made by Yamaha (OPL2/OPL3/OPL4).
+ Answering Y is usually a safe and recommended choice, however some
+ cards may have software (TSR) FM emulation. Enabling FM support with
+ these cards may cause trouble (I don't currently know of any such
+ cards, however).
+ Please read the file Documentation/sound/OPL3 if your card has an
+ OPL3 chip.
+ If you compile the driver into the kernel, you have to add
+ "opl3=" to the kernel command line.
+
+ If you compile your drivers into the kernel, you MUST configure
+ OPL3 support as a module for PAS16 support to work properly.
+ You can then get OPL3 functionality by issuing the command:
+ insmod opl3
+ In addition, you must either add the following line to
+ /etc/modules.conf:
+ options opl3 io=0x388
+ or else add the following line to /etc/lilo.conf:
+ opl3=0x388
+
+
+EXAMPLES
+===================================================================
+To use the PAS16 in my computer I have enabled the following sound
+configuration options:
+
+CONFIG_SOUND=y
+CONFIG_SOUND_OSS=y
+CONFIG_SOUND_TRACEINIT=y
+CONFIG_SOUND_DMAP=y
+CONFIG_SOUND_PAS=y
+CONFIG_SOUND_SB=n
+CONFIG_SOUND_YM3812=m
+
+I have also included the following append line in /etc/lilo.conf:
+append="pas2=0x388,10,3,-1,0x220,5,1,-1 sb=0x220,5,1,-1 opl3=0x388"
+
+The io address of 0x388 is default configuration on the PAS16. The
+irq of 10 and dma of 3 may not match your installation. The above
+configuration enables PAS16, 8-bit Soundblaster and OPL3
+functionality. If Soundblaster functionality is not desired, the
+following line would be appropriate:
+append="pas2=0x388,10,3,-1,0,-1,-1,-1 opl3=0x388"
+
+If sound is built totally modular, the above options may be
+specified in /etc/modules.conf for pas2.o, sb.o and opl3.o
+respectively.
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/PSS linux-2.5.6/Documentation/sound/oss/PSS
--- linux-2.5.6-pre3/Documentation/sound/oss/PSS Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/PSS Thu Mar 7 18:24:41 2002
@@ -0,0 +1,41 @@
+The PSS cards and other ECHO based cards provide an onboard DSP with
+downloadable programs and also has an AD1848 "Microsoft Sound System"
+device. The PSS driver enables MSS and MPU401 modes of the card. SB
+is not enabled since it doesn't work concurrently with MSS.
+
+If you build this driver as a module then the driver takes the following
+parameters
+
+pss_io. The I/O base the PSS card is configured at (normally 0x220
+ or 0x240)
+
+mss_io The base address of the Microsoft Sound System interface.
+ This is normally 0x530, but may be 0x604 or other addresses.
+
+mss_irq The interrupt assigned to the Microsoft Sound System
+ emulation. IRQ's 3,5,7,9,10,11 and 12 are available. If you
+ get IRQ errors be sure to check the interrupt is set to
+ "ISA/Legacy" in the BIOS on modern machines.
+
+mss_dma The DMA channel used by the Microsoft Sound System.
+ This can be 0, 1, or 3. DMA 0 is not available on older
+ machines and will cause a crash on them.
+
+mpu_io The MPU emulation base address. This sets the base of the
+ synthesizer. It is typically 0x330 but can be altered.
+
+mpu_irq The interrupt to use for the synthesizer. It must differ
+ from the IRQ used by the Microsoft Sound System port.
+
+
+The mpu_io/mpu_irq fields are optional. If they are not specified the
+synthesizer parts are not configured.
+
+When the module is loaded it looks for a file called
+/etc/sound/pss_synth. This is the firmware file from the DOS install disks.
+This fil holds a general MIDI emulation. The file expected is called
+genmidi.ld on newer DOS driver install disks and synth.ld on older ones.
+
+You can also load alternative DSP algorithms into the card if you wish. One
+alternative driver can be found at http://www.mpg123.de/
+
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/PSS-updates linux-2.5.6/Documentation/sound/oss/PSS-updates
--- linux-2.5.6-pre3/Documentation/sound/oss/PSS-updates Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/PSS-updates Thu Mar 7 18:24:41 2002
@@ -0,0 +1,88 @@
+ This file contains notes for users of PSS sound cards who wish to use the
+newly added features of the newest version of this driver.
+
+ The major enhancements present in this new revision of this driver is the
+addition of two new module parameters that allow you to take full advantage of
+all the features present on your PSS sound card. These features include the
+ability to enable both the builtin CDROM and joystick ports.
+
+pss_enable_joystick
+
+ This parameter is basically a flag. A 0 will leave the joystick port
+disabled, while a non-zero value would enable the joystick port. The default
+setting is pss_enable_joystick=0 as this keeps this driver fully compatable
+with systems that were using previous versions of this driver. If you wish to
+enable the joystick port you will have to add pss_enable_joystick=1 as an
+argument to the driver. To actually use the joystick port you will then have
+to load the joystick driver itself. Just remember to load the joystick driver
+AFTER the pss sound driver.
+
+pss_cdrom_port
+
+ This parameter takes a port address as its parameter. Any available port
+address can be specified to enable the CDROM port, except for 0x0 and -1 as
+these values would leave the port disabled. Like the joystick port, the cdrom
+port will require that an appropiate CDROM driver be loaded before you can make
+use of the newly enabled CDROM port. Like the joystick port option above,
+remember to load the CDROM driver AFTER the pss sound driver. While it may
+differ on some PSS sound cards, all the PSS sound cards that I have seen have a
+builtin Wearnes CDROM port. If this is the case with your PSS sound card you
+should load aztcd with the appropiate port option that matches the port you
+assigned to the CDROM port when you loaded your pss sound driver. (ex.
+modprobe pss pss_cdrom_port=0x340 && modprobe aztcd aztcd=0x340) The default
+setting of this parameter leaves the CDROM port disabled to maintain full
+compatability with systems using previous versions of this driver.
+
+ Other options have also been added for the added convenience and utility
+of the user. These options are only available if this driver is loaded as a
+module.
+
+pss_no_sound
+
+ This module parameter is a flag that can be used to tell the driver to
+just configure non-sound components. 0 configures all components, a non-0
+value will only attept to configure the CDROM and joystick ports. This
+parameter can be used by a user who only wished to use the builtin joystick
+and/or CDROM port(s) of his PSS sound card. If this driver is loaded with this
+parameter and with the paramter below set to true then a user can safely unload
+this driver with the following command "rmmod pss && rmmod ad1848 && rmmod
+mpu401 && rmmod sound && rmmod soundcore" and retain the full functionality of
+his CDROM and/or joystick port(s) while gaining back the memory previously used
+by the sound drivers. This default setting of this parameter is 0 to retain
+full behavioral compatability with previous versions of this driver.
+
+pss_keep_settings
+
+ This parameter can be used to specify whether you want the driver to reset
+all emulations whenever its unloaded. This can be useful for those who are
+sharing resources (io ports, IRQ's, DMA's) between different ISA cards. This
+flag can also be useful in that future versions of this driver may reset all
+emulations by default on the driver's unloading (as it probably should), so
+specifying it now will ensure that all future versions of this driver will
+continue to work as expected. The default value of this parameter is 1 to
+retain full behavioral compatability with previous versions of this driver.
+
+pss_firmware
+
+ This parameter can be used to specify the file containing the firmware
+code so that a user could tell the driver where that file is located instead
+of having to put it in a predefined location with a predefined name. The
+default setting of this parameter is "/etc/sound/pss_synth" as this was the
+path and filename the hardcoded value in the previous versions of this driver.
+
+Examples:
+
+# Normal PSS sound card system, loading of drivers.
+# Should be specified in an rc file (ex. Slackware uses /etc/rc.d/rc.modules).
+
+/sbin/modprobe pss pss_io=0x220 mpu_io=0x338 mpu_irq=9 mss_io=0x530 mss_irq=10 mss_dma=1 pss_cdrom_port=0x340 pss_enable_joystick=1
+/sbin/modprobe aztcd aztcd=0x340
+/sbin/modprobe joystick
+
+# System using the PSS sound card just for its CDROM and joystick ports.
+# Should be specified in an rc file (ex. Slackware uses /etc/rc.d/rc.modules).
+
+/sbin/modprobe pss pss_io=0x220 pss_cdrom_port=0x340 pss_enable_joystick=1 pss_no_sound=1
+/sbin/rmmod pss && /sbin/rmmod ad1848 && /sbin/rmmod mpu401 && /sbin/rmmod sound && /sbin/rmmod soundcore # This line not needed, but saves memory.
+/sbin/modprobe aztcd aztcd=0x340
+/sbin/modprobe joystick
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/README.OSS linux-2.5.6/Documentation/sound/oss/README.OSS
--- linux-2.5.6-pre3/Documentation/sound/oss/README.OSS Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/README.OSS Thu Mar 7 18:24:41 2002
@@ -0,0 +1,1456 @@
+Introduction
+------------
+
+This file is a collection of all the old Readme files distributed with
+OSS/Lite by Hannu Savolainen. Since the new Linux sound driver is founded
+on it I think these information may still be interesting for users that
+have to configure their sound system.
+
+Be warned: Alan Cox is the current maintainer of the Linux sound driver so if
+you have problems with it, please contact him or the current device-specific
+driver maintainer (e.g. for aedsp16 specific problems contact me). If you have
+patches, contributions or suggestions send them to Alan: I'm sure they are
+welcome.
+
+In this document you will find a lot of references about OSS/Lite or ossfree:
+they are gone forever. Keeping this in mind and with a grain of salt this
+document can be still interesting and very helpful.
+
+[ File edited 17.01.1999 - Riccardo Facchetti ]
+[ Edited miroSOUND section 19.04.2001 - Robert Siemer ]
+
+OSS/Free version 3.8 release notes
+----------------------------------
+
+Please read the SOUND-HOWTO (available from sunsite.unc.edu and other Linux FTP
+sites). It gives instructions about using sound with Linux. It's bit out of
+date but still very useful. Information about bug fixes and such things
+is available from the web page (see above).
+
+Please check http://www.opensound.com/pguide for more info about programming
+with OSS API.
+
+ ====================================================
+- THIS VERSION ____REQUIRES____ Linux 2.1.57 OR LATER.
+ ====================================================
+
+Packages "snd-util-3.8.tar.gz" and "snd-data-0.1.tar.Z"
+contain useful utilities to be used with this driver.
+See http://www.opensound.com/ossfree/getting.html for
+download instructions.
+
+If you are looking for the installation instructions, please
+look forward into this document.
+
+Supported sound cards
+---------------------
+
+See below.
+
+Contributors
+------------
+
+This driver contains code by several contributors. In addition several other
+persons have given useful suggestions. The following is a list of major
+contributors. (I could have forgotten some names.)
+
+ Craig Metz 1/2 of the PAS16 Mixer and PCM support
+ Rob Hooft Volume computation algorithm for the FM synth.
+ Mika Liljeberg uLaw encoding and decoding routines
+ Jeff Tranter Linux SOUND HOWTO document
+ Greg Lee Volume computation algorithm for the GUS and
+ lots of valuable suggestions.
+ Andy Warner ISC port
+ Jim Lowe,
+ Amancio Hasty Jr FreeBSD/NetBSD port
+ Anders Baekgaard Bug hunting and valuable suggestions.
+ Joerg Schubert SB16 DSP support (initial version).
+ Andrew Robinson Improvements to the GUS driver
+ Megens SA MIDI recording for SB and SB Pro (initial version).
+ Mikael Nordqvist Linear volume support for GUS and
+ nonblocking /dev/sequencer.
+ Ian Hartas SVR4.2 port
+ Markus Aroharju and
+ Risto Kankkunen Major contributions to the mixer support
+ of GUS v3.7.
+ Hunyue Yau Mixer support for SG NX Pro.
+ Marc Hoffman PSS support (initial version).
+ Rainer Vranken Initialization for Jazz16 (initial version).
+ Peter Trattler Initial version of loadable module support for Linux.
+ JRA Gibson 16 bit mode for Jazz16 (initial version)
+ Davor Jadrijevic MAD16 support (initial version)
+ Gregor Hoffleit Mozart support (initial version)
+ Riccardo Facchetti Audio Excel DSP 16 (aedsp16) support
+ James Hightower Spotting a tiny but important bug in CS423x support.
+ Denis Sablic OPTi 82C924 specific enhancements (non PnP mode)
+ Tim MacKenzie Full duplex support for OPTi 82C930.
+
+ Please look at lowlevel/README for more contributors.
+
+There are probably many other names missing. If you have sent me some
+patches and your name is not in the above list, please inform me.
+
+Sending your contributions or patches
+-------------------------------------
+
+First of all it's highly recommended to contact me before sending anything
+or before even starting to do any work. Tell me what you suggest to be
+changed or what you have planned to do. Also ensure you are using the
+very latest (development) version of OSS/Free since the change may already be
+implemented there. In general it's a major waste of time to try to improve a
+several months old version. Information about the latest version can be found
+from http://www.opensound.com/ossfree. In general there is no point in
+sending me patches relative to production kernels.
+
+Sponsors etc.
+-------------
+
+The following companies have greatly helped development of this driver
+in form of a free copy of their product:
+
+Novell, Inc. UnixWare personal edition + SDK
+The Santa Cruz Operation, Inc. A SCO OpenServer + SDK
+Ensoniq Corp, a SoundScape card and extensive amount of assistance
+MediaTrix Peripherals Inc, a AudioTrix Pro card + SDK
+Acer, Inc. a pair of AcerMagic S23 cards.
+
+In addition the following companies have provided me sufficient amount
+of technical information at least some of their products (free or $$$):
+
+Advanced Gravis Computer Technology Ltd.
+Media Vision Inc.
+Analog Devices Inc.
+Logitech Inc.
+Aztech Labs Inc.
+Crystal Semiconductor Corporation,
+Integrated Circuit Systems Inc.
+OAK Technology
+OPTi
+Turtle Beach
+miro
+Ad Lib Inc. ($$)
+Music Quest Inc. ($$)
+Creative Labs ($$$)
+
+If you have some problems
+=========================
+
+Read the sound HOWTO (sunsite.unc.edu:/pub/Linux/docs/...?).
+Also look at the home page (http://www.opensound.com/ossfree). It may
+contain info about some recent bug fixes.
+
+It's likely that you have some problems when trying to use the sound driver
+first time. Sound cards don't have standard configuration so there are no
+good default configuration to use. Please try to use same I/O, DMA and IRQ
+values for the sound card than with DOS.
+
+If you get an error message when trying to use the driver, please look
+at /var/adm/messages for more verbose error message.
+
+
+The following errors are likely with /dev/dsp and /dev/audio.
+
+ - "No such device or address".
+ This error indicates that there are no suitable hardware for the
+ device file or the sound driver has been compiled without support for
+ this particular device. For example /dev/audio and /dev/dsp will not
+ work if "digitized voice support" was not enabled during "make config".
+
+ - "Device or resource busy". Probably the IRQ (or DMA) channel
+ required by the sound card is in use by some other device/driver.
+
+ - "I/O error". Almost certainly (99%) it's an IRQ or DMA conflict.
+ Look at the kernel messages in /var/adm/notice for more info.
+
+ - "Invalid argument". The application is calling ioctl()
+ with impossible parameters. Check that the application is
+ for sound driver version 2.X or later.
+
+Linux installation
+==================
+
+IMPORTANT! Read this if you are installing a separately
+ distributed version of this driver.
+
+ Check that your kernel version works with this
+ release of the driver (see Readme). Also verify
+ that your current kernel version doesn't have more
+ recent sound driver version than this one. IT'S HIGHLY
+ RECOMMENDED THAT YOU USE THE SOUND DRIVER VERSION THAT
+ IS DISTRIBUTED WITH KERNEL SOURCES.
+
+- When installing separately distributed sound driver you should first
+ read the above notice. Then try to find proper directory where and how
+ to install the driver sources. You should not try to install a separately
+ distributed driver version if you are not able to find the proper way
+ yourself (in this case use the version that is distributed with kernel
+ sources). Remove old version of linux/drivers/sound directory before
+ installing new files.
+
+- To build the device files you need to run the enclosed shell script
+ (see below). You need to do this only when installing sound driver
+ first time or when upgrading to much recent version than the earlier
+ one.
+
+- Configure and compile Linux as normally (remember to include the
+ sound support during "make config"). Please refer to kernel documentation
+ for instructions about configuring and compiling kernel. File Readme.cards
+ contains card specific instructions for configuring this driver for
+ use with various sound cards.
+
+Boot time configuration (using lilo and insmod)
+-----------------------------------------------
+
+This information has been removed. Too many users didn't believe
+that it's really not necessary to use this method. Please look at
+Readme of sound driver version 3.0.1 if you still want to use this method.
+
+Problems
+--------
+
+Common error messages:
+
+- /dev/???????: No such file or directory.
+Run the script at the end of this file.
+
+- /dev/???????: No such device.
+You are not running kernel which contains the sound driver. When using
+modularized sound driver this error means that the sound driver is not
+loaded.
+
+- /dev/????: No such device or address.
+Sound driver didn't detect suitable card when initializing. Please look at
+Readme.cards for info about configuring the driver with your card. Also
+check for possible boot (insmod) time error messages in /var/adm/messages.
+
+- Other messages or problems
+Please check http://www.opensound.com/ossfree for more info.
+
+Configuring version 3.8 (for Linux) with some common sound cards
+================================================================
+
+This document describes configuring sound cards with the freeware version of
+Open Sound Systems (OSS/Free). Information about the commercial version
+(OSS/Linux) and its configuration is available from
+http://www.opensound.com/linux.html. Information presented here is
+not valid for OSS/Linux.
+
+If you are unsure about how to configure OSS/Free
+you can download the free evaluation version of OSS/Linux from the above
+address. There is a chance that it can autodetect your sound card. In this case
+you can use the information included in soundon.log when configuring OSS/Free.
+
+
+IMPORTANT! This document covers only cards that were "known" when
+ this driver version was released. Please look at
+ http://www.opensound.com/ossfree for info about
+ cards introduced recently.
+
+ When configuring the sound driver, you should carefully
+ check each sound configuration option (particularly
+ "Support for /dev/dsp and /dev/audio"). The default values
+ offered by these programs are not necessarily valid.
+
+
+THE BIGGEST MISTAKES YOU CAN MAKE
+=================================
+
+1. Assuming that the card is Sound Blaster compatible when it's not.
+--------------------------------------------------------------------
+
+The number one mistake is to assume that your card is compatible with
+Sound Blaster. Only the cards made by Creative Technology or which have
+one or more chips labeled by Creative are SB compatible. In addition there
+are few sound chipsets which are SB compatible in Linux such as ESS1688 or
+Jazz16. Note that SB compatibility in DOS/Windows does _NOT_ mean anything
+in Linux.
+
+IF YOU REALLY ARE 150% SURE YOU HAVE A SOUND BLASTER YOU CAN SKIP THE REST OF
+THIS CHAPTER.
+
+For most other "supposed to be SB compatible" cards you have to use other
+than SB drivers (see below). It is possible to get most sound cards to work
+in SB mode but in general it's a complete waste of time. There are several
+problems which you will encounter by using SB mode with cards that are not
+truly SB compatible:
+
+- The SB emulation is at most SB Pro (DSP version 3.x) which means that
+you get only 8 bit audio (there is always an another ("native") mode which
+gives the 16 bit capability). The 8 bit only operation is the reason why
+many users claim that sound quality in Linux is much worse than in DOS.
+In addition some applications require 16 bit mode and they produce just
+noise with a 8 bit only device.
+- The card may work only in some cases but refuse to work most of the
+time. The SB compatible mode always requires special initialization which is
+done by the DOS/Windows drivers. This kind of cards work in Linux after
+you have warm booted it after DOS but they don't work after cold boot
+(power on or reset).
+- You get the famous "DMA timed out" messages. Usually all SB clones have
+software selectable IRQ and DMA settings. If the (power on default) values
+currently used by the card don't match configuration of the driver you will
+get the above error message whenever you try to record or play. There are
+few other reasons to the DMA timeout message but using the SB mode seems
+to be the most common cause.
+
+2. Trying to use a PnP (Plug & Play) card just like an ordinary sound card
+--------------------------------------------------------------------------
+
+Plug & Play is a protocol defined by Intel and Microsoft. It lets operating
+systems to easily identify and reconfigure I/O ports, IRQs and DMAs of ISA
+cards. The problem with PnP cards is that the standard Linux doesn't currently
+(versions 2.1.x and earlier) don't support PnP. This means that you will have
+to use some special tricks (see later) to get a PnP card alive. Many PnP cards
+work after they have been initialized but this is not always the case.
+
+There are sometimes both PnP and non-PnP versions of the same sound card.
+The non-PnP version is the original model which usually has been discontinued
+more than an year ago. The PnP version has the same name but with "PnP"
+appended to it (sometimes not). This causes major confusion since the non-PnP
+model works with Linux but the PnP one doesn't.
+
+You should carefully check if "Plug & Play" or "PnP" is mentioned in the name
+of the card or in the documentation or package that came with the card.
+Everything described in the rest of this document is not necessarily valid for
+PnP models of sound cards even you have managed to wake up the card properly.
+Many PnP cards are simply too different from their non-PnP ancestors which are
+covered by this document.
+
+
+Cards that are not (fully) supported by this driver
+===================================================
+
+See http://www.opensound.com/ossfree for information about sound cards
+to be supported in future.
+
+
+How to use sound without recompiling kernel and/or sound driver
+===============================================================
+
+There is a commercial sound driver which comes in precompiled form and doesn't
+require recompiling of the kernel. See http://www.4Front-tech.com/oss.html for
+more info.
+
+
+Configuring PnP cards
+=====================
+
+New versions of most sound cards use the so-called ISA PnP protocol for
+soft configuring their I/O, IRQ, DMA and shared memory resources.
+Currently at least cards made by Creative Technology (SB32 and SB32AWE
+PnP), Gravis (GUS PnP and GUS PnP Pro), Ensoniq (Soundscape PnP) and
+Aztech (some Sound Galaxy models) use PnP technology. The CS4232/4236 audio
+chip by Crystal Semiconductor (Intel Atlantis, HP Pavilion and many other
+motherboards) is also based on PnP technology but there is a "native" driver
+available for it (see information about CS4232 later in this document).
+
+PnP sound cards (as well as most other PnP ISA cards) are not supported
+by this version of the driver . Proper
+support for them should be released during 97 once the kernel level
+PnP support is available.
+
+There is a method to get most of the PnP cards to work. The basic method
+is the following:
+
+1) Boot DOS so the card's DOS drivers have a chance to initialize it.
+2) _Cold_ boot to Linux by using "loadlin.exe". Hitting ctrl-alt-del
+works with older machines but causes a hard reset of all cards on recent
+(Pentium) machines.
+3) If you have the sound driver in Linux configured properly, the card should
+work now. "Proper" means that I/O, IRQ and DMA settings are the same as in
+DOS. The hard part is to find which settings were used. See the documentation of
+your card for more info.
+
+Windows 95 could work as well as DOS but running loadlin may be difficult.
+Probably you should "shut down" your machine to MS-DOS mode before running it.
+
+Some machines have a BIOS utility for setting PnP resources. This is a good
+way to configure some cards. In this case you don't need to boot DOS/Win95
+before starting Linux.
+
+Another way to initialize PnP cards without DOS/Win95 is a Linux based
+PnP isolation tool. When writing this there is a pre alpha test version
+of such a tool available from ftp://ftp.demon.co.uk/pub/unix/linux/utils. The
+file is called isapnptools-*. Please note that this tool is just a temporary
+solution which may be incompatible with future kernel versions having proper
+support for PnP cards. There are bugs in setting DMA channels in earlier
+versions of isapnptools so at least version 1.6 is required with sound cards.
+
+Yet another way to use PnP cards is to use (commercial) OSS/Linux drivers. See
+http://www.opensound.com/linux.html for more info. This is probably the way you
+should do it if you don't want to spend time recompiling the kernel and
+required tools.
+
+
+Read this before trying to configure the driver
+===============================================
+
+There are currently many cards that work with this driver. Some of the cards
+have native support while others work since they emulate some other
+card (usually SB, MSS/WSS and/or MPU401). The following cards have native
+support in the driver. Detailed instructions for configuring these cards
+will be given later in this document.
+
+Pro Audio Spectrum 16 (PAS16) and compatibles:
+ Pro Audio Spectrum 16
+ Pro Audio Studio 16
+ Logitech Sound Man 16
+ NOTE! The original Pro Audio Spectrum as well as the PAS+ are not
+ and will not be supported by the driver.
+
+Media Vision Jazz16 based cards
+ Pro Sonic 16
+ Logitech SoundMan Wave
+ (Other Jazz based cards should work but I don't have any reports
+ about them).
+
+Sound Blasters
+ SB 1.0 to 2.0
+ SB Pro
+ SB 16
+ SB32/64/AWE
+ Configure SB32/64/AWE just like SB16. See lowlevel/README.awe
+ for information about using the wave table synth.
+ NOTE! AWE63/Gold and 16/32/AWE "PnP" cards need to be activated
+ using isapnptools before they work with OSS/Free.
+ SB16 compatible cards by other manufacturers than Creative.
+ You have been fooled since there are _no_ SB16 compatible
+ cards on the market (as of May 1997). It's likely that your card
+ is compatible just with SB Pro but there is also a non-SB-
+ compatible 16 bit mode. Usually it's MSS/WSS but it could also
+ be a proprietary one like MV Jazz16 or ESS ES688. OPTi
+ MAD16 chips are very common in so called "SB 16 bit cards"
+ (try with the MAD16 driver).
+
+ ======================================================================
+ "Supposed to be SB compatible" cards.
+ Forget the SB compatibility and check for other alternatives
+ first. The only cards that work with the SB driver in
+ Linux have been made by Creative Technology (there is at least
+ one chip on the card with "CREATIVE" printed on it). The
+ only other SB compatible chips are ESS and Jazz16 chips
+ (maybe ALSxxx chips too but they probably don't work).
+ Most other "16 bit SB compatible" cards such as "OPTi/MAD16" or
+ "Crystal" are _NOT_ SB compatible in Linux.
+
+ Practically all sound cards have some kind of SB emulation mode
+ in addition to their native (16 bit) mode. In most cases this
+ (8 bit only) SB compatible mode doesn't work with Linux. If
+ you get it working it may cause problems with games and
+ applications which require 16 bit audio. Some 16 bit only
+ applications don't check if the card actually supports 16 bits.
+ They just dump 16 bit data to a 8 bit card which produces just
+ noise.
+
+ In most cases the 16 bit native mode is supported by Linux.
+ Use the SB mode with "clones" only if you don't find anything
+ better from the rest of this doc.
+ ======================================================================
+
+Gravis Ultrasound (GUS)
+ GUS
+ GUS + the 16 bit option
+ GUS MAX
+ GUS ACE (No MIDI port and audio recording)
+ GUS PnP (with RAM)
+
+MPU-401 and compatibles
+ The driver works both with the full (intelligent mode) MPU-401
+ cards (such as MPU IPC-T and MQX-32M) and with the UART only
+ dumb MIDI ports. MPU-401 is currently the most common MIDI
+ interface. Most sound cards are compatible with it. However,
+ don't enable MPU401 mode blindly. Many cards with native support
+ in the driver have their own MPU401 driver. Enabling the standard one
+ will cause a conflict with these cards. So check if your card is
+ in the list of supported cards before enabling MPU401.
+
+Windows Sound System (MSS/WSS)
+ Even when Microsoft has discontinued their own Sound System card
+ they managed to make it a standard. MSS compatible cards are based on
+ a codec chip which is easily available from at least two manufacturers
+ (AD1848 by Analog Devices and CS4231/CS4248 by Crystal Semiconductor).
+ Currently most sound cards are based on one of the MSS compatible codec
+ chips. The CS4231 is used in the high quality cards such as GUS MAX,
+ MediaTrix AudioTrix Pro and TB Tropez (GUS MAX is not MSS compatible).
+
+ Having a AD1848, CS4248 or CS4231 codec chip on the card is a good
+ sign. Even if the card is not MSS compatible, it could be easy to write
+ support for it. Note also that most MSS compatible cards
+ require special boot time initialization which may not be present
+ in the driver. Also, some MSS compatible cards have native support.
+ Enabling the MSS support with these cards is likely to
+ cause a conflict. So check if your card is listed in this file before
+ enabling the MSS support.
+
+Yamaha FM synthesizers (OPL2, OPL3 (not OPL3-SA) and OPL4)
+ Most sound cards have a FM synthesizer chip. The OPL2 is a 2
+ operator chip used in the original AdLib card. Currently it's used
+ only in the cheapest (8 bit mono) cards. The OPL3 is a 4 operator
+ FM chip which provides better sound quality and/or more available
+ voices than the OPL2. The OPL4 is a new chip that has an OPL3 and
+ a wave table synthesizer packed onto the same chip. The driver supports
+ just the OPL3 mode directly. Most cards with an OPL4 (like
+ SM Wave and AudioTrix Pro) support the OPL4 mode using MPU401
+ emulation. Writing a native OPL4 support is difficult
+ since Yamaha doesn't give information about their sample ROM chip.
+
+ Enable the generic OPL2/OPL3 FM synthesizer support if your
+ card has a FM chip made by Yamaha. Don't enable it if your card
+ has a software (TRS) based FM emulator.
+
+ ----------------------------------------------------------------
+ NOTE! OPL3-SA is different chip than the ordinary OPL3. In addition
+ to the FM synth this chip has also digital audio (WSS) and
+ MIDI (MPU401) capabilities. Support for OPL3-SA is described below.
+ ----------------------------------------------------------------
+
+Yamaha OPL3-SA1
+
+ Yamaha OPL3-SA1 (YMF701) is an audio controller chip used on some
+ (Intel) motherboards and on cheap sound cards. It should not be
+ confused with the original OPL3 chip (YMF278) which is entirely
+ different chip. OPL3-SA1 has support for MSS, MPU401 and SB Pro
+ (not used in OSS/Free) in addition to the OPL3 FM synth.
+
+ There are also chips called OPL3-SA2, OPL3-SA3, ..., OPL3SA-N. They
+ are PnP chips and will not work with the OPL3-SA1 driver. You should
+ use the standard MSS, MPU401 and OPL3 options with these chips and to
+ activate the card using isapnptools.
+
+4Front Technologies SoftOSS
+
+ SoftOSS is a software based wave table emulation which works with
+ any 16 bit stereo sound card. Due to its nature a fast CPU is
+ required (P133 is minimum). Although SoftOSS does _not_ use MMX
+ instructions it has proven out that recent processors (which appear
+ to have MMX) perform significantly better with SoftOSS than earlier
+ ones. For example a P166MMX beats a PPro200. SoftOSS should not be used
+ on 486 or 386 machines.
+
+ The amount of CPU load caused by SoftOSS can be controlled by
+ selecting the CONFIG_SOFTOSS_RATE and CONFIG_SOFTOSS_VOICES
+ parameters properly (they will be prompted by make config). It's
+ recommended to set CONFIG_SOFTOSS_VOICES to 32. If you have a
+ P166MMX or faster (PPro200 is not faster) you can set
+ CONFIG_SOFTOSS_RATE to 44100 (kHz). However with slower systems it
+ recommended to use sampling rates around 22050 or even 16000 kHz.
+ Selecting too high values for these parameters may hang your
+ system when playing MIDI files with hight degree of polyphony
+ (number of concurrently playing notes). It's also possible to
+ decrease CONFIG_SOFTOSS_VOICES. This makes it possible to use
+ higher sampling rates. However using fewer voices decreases
+ playback quality more than decreasing the sampling rate.
+
+ SoftOSS keeps the samples loaded on the system's RAM so much RAM is
+ required. SoftOSS should never be used on machines with less than 16 MB
+ of RAM since this is potentially dangerous (you may accidentally run out
+ of memory which probably crashes the machine).
+
+ SoftOSS implements the wave table API originally designed for GUS. For
+ this reason all applications designed for GUS should work (at least
+ after minor modifications). For example gmod/xgmod and playmidi -g are
+ known to work.
+
+ To work SoftOSS will require GUS compatible
+ patch files to be installed on the system (in /dos/ultrasnd/midi). You
+ can use the public domain MIDIA patchset available from several ftp
+ sites.
+
+ *********************************************************************
+ IMPORTANT NOTICE! The original patch set distributed with the Gravis
+ Ultrasound card is not in public domain (even though it's available from
+ some FTP sites). You should contact Voice Crystal (www.voicecrystal.com)
+ if you like to use these patches with SoftOSS included in OSS/Free.
+ *********************************************************************
+
+PSS based cards (AD1848 + ADSP-2115 + Echo ESC614 ASIC)
+ Analog Devices and Echo Speech have together defined a sound card
+ architecture based on the above chips. The DSP chip is used
+ for emulation of SB Pro, FM and General MIDI/MT32.
+
+ There are several cards based on this architecture. The most known
+ ones are Orchid SW32 and Cardinal DSP16.
+
+ The driver supports downloading DSP algorithms to these cards.
+
+ NOTE! You will have to use the "old" config script when configuring
+ PSS cards.
+
+MediaTrix AudioTrix Pro
+ The ATP card is built around a CS4231 codec and an OPL4 synthesizer
+ chips. The OPL4 mode is supported by a microcontroller running a
+ General MIDI emulator. There is also a SB 1.5 compatible playback mode.
+
+Ensoniq SoundScape and compatibles
+ Ensoniq has designed a sound card architecture based on the
+ OTTO synthesizer chip used in their professional MIDI synthesizers.
+ Several companies (including Ensoniq, Reveal and Spea) are selling
+ cards based on this architecture.
+
+ NOTE! The SoundScape PnP is not supported by OSS/Free. Ensoniq VIVO and
+ VIVO90 cards are not compatible with Soundscapes so the Soundscape
+ driver will not work with them. You may want to use OSS/Linux with these
+ cards.
+
+OPTi MAD16 and Mozart based cards
+ The Mozart (OAK OTI-601), MAD16 (OPTi 82C928), MAD16 Pro (OPTi 82C929),
+ OPTi 82C924/82C925 (in _non_ PnP mode) and OPTi 82C930 interface
+ chips are used in many different sound cards, including some
+ cards by Reveal miro and Turtle Beach (Tropez). The purpose of these
+ chips is to connect other audio components to the PC bus. The
+ interface chip performs address decoding for the other chips.
+ NOTE! Tropez Plus is not MAD16 but CS4232 based.
+ NOTE! MAD16 PnP cards (82C924, 82C925, 82C931) are not MAD16 compatible
+ in the PnP mode. You will have to use them in MSS mode after having
+ initialized them using isapnptools or DOS. 82C931 probably requires
+ initialization using DOS/Windows (running isapnptools is not enough).
+ It's possible to use 82C931 with OSS/Free by jumpering it to non-PnP
+ mode (provided that the card has a jumper for this). In non-PnP mode
+ 82C931 is compatible with 82C930 and should work with the MAD16 driver
+ (without need to use isapnptools or DOS to initialize it). All OPTi
+ chips are supported by OSS/Linux (both in PnP and non-PnP modes).
+
+Audio Excel DSP16
+ Support for this card was written by Riccardo Faccetti
+ (riccardo@cdc8g5.cdc.polimi.it). The AEDSP16 driver included in
+ the lowlevel/ directory. To use it you should enable the
+ "Additional low level drivers" option.
+
+Crystal CS4232 and CS4236 based cards such as AcerMagic S23, TB Tropez _Plus_ and
+ many PC motherboards (Compaq, HP, Intel, ...)
+ CS4232 is a PnP multimedia chip which contains a CS3231A codec,
+ SB and MPU401 emulations. There is support for OPL3 too.
+ Unfortunately the MPU401 mode doesn't work (I don't know how to
+ initialize it). CS4236 is an enhanced (compatible) version of CS4232.
+ NOTE! Don't ever try to use isapnptools with CS4232 since this will just
+ freeze your machine (due to chip bugs). If you have problems in getting
+ CS4232 working you could try initializing it with DOS (CS4232C.EXE) and
+ then booting Linux using loadlin. CS4232C.EXE loads a secret firmware
+ patch which is not documented by Crystal.
+
+Turtle Beach Maui and Tropez "classic"
+ This driver version supports sample, patch and program loading commands
+ described in the Maui/Tropez User's manual.
+ There is now full initialization support too. The audio side of
+ the Tropez is based on the MAD16 chip (see above).
+ NOTE! Tropez Plus is different card than Tropez "classic" and will not
+ work fully in Linux. You can get audio features working by configuring
+ the card as a CS4232 based card (above).
+
+
+Jumpers and software configuration
+==================================
+
+Some of the earliest sound cards were jumper configurable. You have to
+configure the driver use I/O, IRQ and DMA settings
+that match the jumpers. Just few 8 bit cards are fully jumper
+configurable (SB 1.x/2.x, SB Pro and clones).
+Some cards made by Aztech have an EEPROM which contains the
+config info. These cards behave much like hardware jumpered cards.
+
+Most cards have jumper for the base I/O address but other parameters
+are software configurable. Sometimes there are few other jumpers too.
+
+Latest cards are fully software configurable or they are PnP ISA
+compatible. There are no jumpers on the board.
+
+The driver handles software configurable cards automatically. Just configure
+the driver to use I/O, IRQ and DMA settings which are known to work.
+You could usually use the same values than with DOS and/or Windows.
+Using different settings is possible but not recommended since it may cause
+some trouble (for example when warm booting from an OS to another or
+when installing new hardware to the machine).
+
+Sound driver sets the soft configurable parameters of the card automatically
+during boot. Usually you don't need to run any extra initialization
+programs when booting Linux but there are some exceptions. See the
+card-specific instructions below for more info.
+
+The drawback of software configuration is that the driver needs to know
+how the card must be initialized. It cannot initialize unknown cards
+even if they are otherwise compatible with some other cards (like SB,
+MPU401 or Windows Sound System).
+
+
+What if your card was not listed above?
+=======================================
+
+The first thing to do is to look at the major IC chips on the card.
+Many of the latest sound cards are based on some standard chips. If you
+are lucky, all of them could be supported by the driver. The most common ones
+are the OPTi MAD16, Mozart, SoundScape (Ensoniq) and the PSS architectures
+listed above. Also look at the end of this file for list of unsupported
+cards and the ones which could be supported later.
+
+The last resort is to send _exact_ name and model information of the card
+to me together with a list of the major IC chips (manufactured, model) to
+me. I could then try to check if your card looks like something familiar.
+
+There are many more cards in the world than listed above. The first thing to
+do with these cards is to check if they emulate some other card or interface
+such as SB, MSS and/or MPU401. In this case there is a chance to get the
+card to work by booting DOS before starting Linux (boot DOS, hit ctrl-alt-del
+and boot Linux without hard resetting the machine). In this method the
+DOS based driver initializes the hardware to use known I/O, IRQ and DMA
+settings. If sound driver is configured to use the same settings, everything
+should work OK.
+
+
+Configuring sound driver (with Linux)
+=====================================
+
+The sound driver is currently distributed as part of the Linux kernel. The
+files are in /usr/src/linux/drivers/sound/.
+
+****************************************************************************
+* ALWAYS USE THE SOUND DRIVER VERSION WHICH IS DISTRIBUTED WITH *
+* THE KERNEL SOURCE PACKAGE YOU ARE USING. SOME ALPHA AND BETA TEST *
+* VERSIONS CAN BE INSTALLED FROM A SEPARATELY DISTRIBUTED PACKAGE *
+* BUT CHECK THAT THE PACKAGE IS NOT MUCH OLDER (OR NEWER) THAN THE *
+* KERNEL YOU ARE USING. IT'S POSSIBLE THAT THE KERNEL/DRIVER *
+* INTERFACE CHANGES BETWEEN KERNEL RELEASES WHICH MAY CAUSE SOME *
+* INCOMPATIBILITY PROBLEMS. *
+* *
+* IN CASE YOU INSTALL A SEPARATELY DISTRIBUTED SOUND DRIVER VERSION, *
+* BE SURE TO REMOVE OR RENAME THE OLD SOUND DRIVER DIRECTORY BEFORE *
+* INSTALLING THE NEW ONE. LEAVING OLD FILES TO THE SOUND DRIVER *
+* DIRECTORY _WILL_ CAUSE PROBLEMS WHEN THE DRIVER IS USED OR *
+* COMPILED. *
+****************************************************************************
+
+To configure the driver, run "make config" in the kernel source directory
+(/usr/src/linux). Answer "y" or "m" to the question about Sound card support
+(after the questions about mouse, CD-ROM, ftape, etc. support). Questions
+about options for sound will then be asked.
+
+After configuring the kernel and sound driver, run "make dep" and compile
+the kernel following instructions in the kernel README.
+
+The sound driver configuration dialog
+-------------------------------------
+
+Sound configuration starts by making some yes/no questions. Be careful
+when answering to these questions since answering y to a question may
+prevent some later ones from being asked. For example don't answer y to
+the first question (PAS16) if you don't really have a PAS16. Don't enable
+more cards than you really need since they just consume memory. Also
+some drivers (like MPU401) may conflict with your SCSI controller and
+prevent kernel from booting. If you card was in the list of supported
+cards (above), please look at the card specific config instructions
+(later in this file) before starting to configure. Some cards must be
+configured in way which is not obvious.
+
+So here is the beginning of the config dialog. Answer 'y' or 'n' to these
+questions. The default answer is shown so that (y/n) means 'y' by default and
+(n/y) means 'n'. To use the default value, just hit ENTER. But be careful
+since using the default _doesn't_ guarantee anything.
+
+Note also that all questions may not be asked. The configuration program
+may disable some questions depending on the earlier choices. It may also
+select some options automatically as well.
+
+ "ProAudioSpectrum 16 support",
+ - Answer 'y'_ONLY_ if you have a Pro Audio Spectrum _16_,
+ Pro Audio Studio 16 or Logitech SoundMan 16 (be sure that
+ you read the above list correctly). Don't answer 'y' if you
+ have some other card made by Media Vision or Logitech since they
+ are not PAS16 compatible.
+ NOTE! Since 3.5-beta10 you need to enable SB support (next question)
+ if you want to use the SB emulation of PAS16. It's also possible to
+ the emulation if you want to use a true SB card together with PAS16
+ (there is another question about this that is asked later).
+ "Sound Blaster support",
+ - Answer 'y' if you have an original SB card made by Creative Labs
+ or a full 100% hardware compatible clone (like Thunderboard or
+ SM Games). If your card was in the list of supported cards (above),
+ please look at the card specific instructions later in this file
+ before answering this question. For an unknown card you may answer
+ 'y' if the card claims to be SB compatible.
+ Enable this option also with PAS16 (changed since v3.5-beta9).
+
+ Don't enable SB if you have a MAD16 or Mozart compatible card.
+
+ "Generic OPL2/OPL3 FM synthesizer support",
+ - Answer 'y' if your card has a FM chip made by Yamaha (OPL2/OPL3/OPL4).
+ Answering 'y' is usually a safe and recommended choice. However some
+ cards may have software (TSR) FM emulation. Enabling FM support
+ with these cards may cause trouble. However I don't currently know
+ such cards.
+ "Gravis Ultrasound support",
+ - Answer 'y' if you have GUS or GUS MAX. Answer 'n' if you don't
+ have GUS since the GUS driver consumes much memory.
+ Currently I don't have experiences with the GUS ACE so I don't
+ know what to answer with it.
+ "MPU-401 support (NOT for SB16)",
+ - Be careful with this question. The MPU401 interface is supported
+ by almost any sound card today. However some natively supported cards
+ have their own driver for MPU401. Enabling the MPU401 option with
+ these cards will cause a conflict. Also enabling MPU401 on a system
+ that doesn't really have a MPU401 could cause some trouble. If your
+ card was in the list of supported cards (above), please look at
+ the card specific instructions later in this file.
+
+ In MOST cases this MPU401 driver should only be used with "true"
+ MIDI-only MPU401 professional cards. In most other cases there
+ is another way to get the MPU401 compatible interface of a
+ sound card to work.
+ Support for the MPU401 compatible MIDI port of SB16, ESS1688
+ and MV Jazz16 cards is included in the SB driver. Use it instead
+ of this separate MPU401 driver with these cards. As well
+ Soundscape, PSS and Maui drivers include their own MPU401
+ options.
+
+ It's safe to answer 'y' if you have a true MPU401 MIDI interface
+ card.
+ "6850 UART Midi support",
+ - It's safe to answer 'n' to this question in all cases. The 6850
+ UART interface is so rarely used.
+ "PSS (ECHO-ADI2111) support",
+ - Answer 'y' only if you have Orchid SW32, Cardinal DSP16 or some
+ other card based on the PSS chipset (AD1848 codec + ADSP-2115
+ DSP chip + Echo ESC614 ASIC CHIP).
+ "16 bit sampling option of GUS (_NOT_ GUS MAX)",
+ - Answer 'y' if you have installed the 16 bit sampling daughtercard
+ to your GUS. Answer 'n' if you have GUS MAX. Enabling this option
+ disables GUS MAX support.
+ "GUS MAX support",
+ - Answer 'y' only if you have a GUS MAX.
+ "Microsoft Sound System support",
+ - Again think carefully before answering 'y' to this question. It's
+ safe to answer 'y' in case you have the original Windows Sound
+ System card made by Microsoft or Aztech SG 16 Pro (or NX16 Pro).
+ Also you may answer 'y' in case your card was not listed earlier
+ in this file. For cards having native support in the driver, consult
+ the card specific instructions later in this file. Some drivers
+ have their own MSS support and enabling this option will cause a
+ conflict.
+ Note! The MSS driver permits configuring two DMA channels. This is a
+ "nonstandard" feature and works only with very few cards (if any).
+ In most cases the second DMA channel should be disabled or set to
+ the same channel than the first one. Trying to configure two separate
+ channels with cards that don't support this feature will prevent
+ audio (at least recording) from working.
+ "Ensoniq Soundscape support",
+ - Answer 'y' if you have a sound card based on the Ensoniq SoundScape
+ chipset. Such cards are being manufactured at least by Ensoniq,
+ Spea and Reveal (note that Reveal makes other cards also). The oldest
+ cards made by Spea don't work properly with Linux.
+ Soundscape PnP as well as Ensoniq VIVO work only with the commercial
+ OSS/Linux version.
+ "MediaTrix AudioTrix Pro support",
+ - Answer 'y' if you have the AudioTrix Pro.
+ "Support for MAD16 and/or Mozart based cards",
+ - Answer y if your card has a Mozart (OAK OTI-601) or MAD16
+ (OPTi 82C928, 82C929, 82C924/82C925 or 82C930) audio interface chip.
+ These chips are
+ currently quite common so it's possible that many no-name cards
+ have one of them. In addition the MAD16 chip is used in some
+ cards made by known manufacturers such as Turtle Beach (Tropez),
+ Reveal (some models) and Diamond (some recent models).
+ Note OPTi 82C924 and 82C925 are MAD16 compatible only in non PnP
+ mode (jumper selectable on many cards).
+ "Support for TB Maui"
+ - This enables TB Maui specific initialization. Works with TB Maui
+ and TB Tropez (may not work with Tropez Plus).
+
+
+Then the configuration program asks some y/n questions about the higher
+level services. It's recommended to answer 'y' to each of these questions.
+Answer 'n' only if you know you will not need the option.
+
+ "MIDI interface support",
+ - Answering 'n' disables /dev/midi## devices and access to any
+ MIDI ports using /dev/sequencer and /dev/music. This option
+ also affects any MPU401 and/or General MIDI compatible devices.
+ "FM synthesizer (YM3812/OPL-3) support",
+ - Answer 'y' here.
+ "/dev/sequencer support",
+ - Answering 'n' disables /dev/sequencer and /dev/music.
+
+Entering the I/O, IRQ and DMA config parameters
+-----------------------------------------------
+
+After the above questions the configuration program prompts for the
+card specific configuration information. Usually just a set of
+I/O address, IRQ and DMA numbers are asked. With some cards the program
+asks for some files to be used during initialization of the card. For example
+many cards have a DSP chip or microprocessor which must be initialized by
+downloading a program (microcode) file to the card.
+
+Instructions for answering these questions are given in the next section.
+
+
+Card specific information
+=========================
+
+This section gives additional instructions about configuring some cards.
+Please refer manual of your card for valid I/O, IRQ and DMA numbers. Using
+the same settings with DOS/Windows and Linux is recommended. Using
+different values could cause some problems when switching between
+different operating systems.
+
+Sound Blasters (the original ones by Creative)
+---------------------------------------------
+
+NOTE! Check if you have a PnP Sound Blaster (cards sold after summer 1995
+ are almost certainly PnP ones). With PnP cards you should use isapnptools
+ to activate them (see above).
+
+It's possible to configure these cards to use different I/O, IRQ and
+DMA settings. Since the possible/default settings have changed between various
+models, you have to consult manual of your card for the proper ones. It's
+a good idea to use the same values than with DOS/Windows. With SB and SB Pro
+it's the only choice. SB16 has software selectable IRQ and DMA channels but
+using different values with DOS and Linux is likely to cause troubles. The
+DOS driver is not able to reset the card properly after warm boot from Linux
+if Linux has used different IRQ or DMA values.
+
+The original (steam) Sound Blaster (versions 1.x and 2.x) use always
+DMA1. There is no way to change it.
+
+The SB16 needs two DMA channels. A 8 bit one (1 or 3) is required for
+8 bit operation and a 16 bit one (5, 6 or 7) for the 16 bit mode. In theory
+it's possible to use just one (8 bit) DMA channel by answering the 8 bit
+one when the configuration program asks for the 16 bit one. This may work
+in some systems but is likely to cause terrible noise on some other systems.
+
+It's possible to use two SB16/32/64 at the same time. To do this you should
+first configure OSS/Free for one card. Then edit local.h manually and define
+SB2_BASE, SB2_IRQ, SB2_DMA and SB2_DMA2 for the second one. You can't get
+the OPL3, MIDI and EMU8000 devices of the second card to work. If you are
+going to use two PnP Sound Blasters, ensure that they are of different model
+and have different PnP IDs. There is no way to get two cards with the same
+card ID and serial number to work. The easiest way to check this is trying
+if isapnptools can see both cards or just one.
+
+NOTE! Don't enable the SM Games option (asked by the configuration program)
+ if you are not 101% sure that your card is a Logitech Soundman Games
+ (not a SM Wave or SM16).
+
+SB Clones
+---------
+
+First of all: There are no SB16 clones. There are SB Pro clones with a
+16 bit mode which is not SB16 compatible. The most likely alternative is that
+the 16 bit mode means MSS/WSS.
+
+There are just a few fully 100% hardware SB or SB Pro compatible cards.
+I know just Thunderboard and SM Games. Other cards require some kind of
+hardware initialization before they become SB compatible. Check if your card
+was listed in the beginning of this file. In this case you should follow
+instructions for your card later in this file.
+
+For other not fully SB clones you may try initialization using DOS in
+the following way:
+
+ - Boot DOS so that the card specific driver gets run.
+ - Hit ctrl-alt-del (or use loadlin) to boot Linux. Don't
+ switch off power or press the reset button.
+ - If you use the same I/O, IRQ and DMA settings in Linux, the
+ card should work.
+
+If your card is both SB and MSS compatible, I recommend using the MSS mode.
+Most cards of this kind are not able to work in the SB and the MSS mode
+simultaneously. Using the MSS mode provides 16 bit recording and playback.
+
+ProAudioSpectrum 16 and compatibles
+-----------------------------------
+
+PAS16 has a SB emulation chip which can be used together with the native
+(16 bit) mode of the card. To enable this emulation you should configure
+the driver to have SB support too (this has been changed since version
+3.5-beta9 of this driver).
+
+With current driver versions it's also possible to use PAS16 together with
+another SB compatible card. In this case you should configure SB support
+for the other card and to disable the SB emulation of PAS16 (there is a
+separate questions about this).
+
+With PAS16 you can use two audio device files at the same time. /dev/dsp (and
+/dev/audio) is connected to the 8/16 bit native codec and the /dev/dsp1 (and
+/dev/audio1) is connected to the SB emulation (8 bit mono only).
+
+Gravis Ultrasound
+-----------------
+
+There are many different revisions of the Ultrasound card (GUS). The
+earliest ones (pre 3.7) don't have a hardware mixer. With these cards
+the driver uses a software emulation for synth and pcm playbacks. It's
+also possible to switch some of the inputs (line in, mic) off by setting
+mixer volume of the channel level below 10%. For recording you have
+to select the channel as a recording source and to use volume above 10%.
+
+GUS 3.7 has a hardware mixer.
+
+GUS MAX and the 16 bit sampling daughtercard have a CS4231 codec chip which
+also contains a mixer.
+
+Configuring GUS is simple. Just enable the GUS support and GUS MAX or
+the 16 bit daughtercard if you have them. Note that enabling the daughter
+card disables GUS MAX driver.
+
+NOTE for owners of the 16 bit daughtercard: By default the daughtercard
+uses /dev/dsp (and /dev/audio). Command "ln -sf /dev/dsp1 /dev/dsp"
+selects the daughter card as the default device.
+
+With just the standard GUS enabled the configuration program prompts
+for the I/O, IRQ and DMA numbers for the card. Use the same values than
+with DOS.
+
+With the daughter card option enabled you will be prompted for the I/O,
+IRQ and DMA numbers for the daughter card. You have to use different I/O
+and DMA values than for the standard GUS. The daughter card permits
+simultaneous recording and playback. Use /dev/dsp (the daughtercard) for
+recording and /dev/dsp1 (GUS GF1) for playback.
+
+GUS MAX uses the same I/O address and IRQ settings than the original GUS
+(GUS MAX = GUS + a CS4231 codec). In addition an extra DMA channel may be used.
+Using two DMA channels permits simultaneous playback using two devices
+(dev/dsp0 and /dev/dsp1). The second DMA channel is required for
+full duplex audio.
+To enable the second DMA channels, give a valid DMA channel when the config
+program asks for the GUS MAX DMA (entering -1 disables the second DMA).
+Using 16 bit DMA channels (5,6 or 7) is recommended.
+
+If you have problems in recording with GUS MAX, you could try to use
+just one 8 bit DMA channel. Recording will not work with one DMA
+channel if it's a 16 bit one.
+
+Microphone input of GUS MAX is connected to mixer in little bit nonstandard
+way. There is actually two microphone volume controls. Normal "mic" controls
+only recording level. Mixer control "speaker" is used to control volume of
+microphone signal connected directly to line/speaker out. So just decrease
+volume of "speaker" if you have problems with microphone feedback.
+
+GUS ACE works too but any attempt to record or to use the MIDI port
+will fail.
+
+GUS PnP (with RAM) is partially supported but it needs to be initialized using
+DOS or isapnptools before starting the driver.
+
+MPU401 and Windows Sound System
+-------------------------------
+
+Again. Don't enable these options in case your card is listed
+somewhere else in this file.
+
+Configuring these cards is obvious (or it should be). With MSS
+you should probably enable the OPL3 synth also since
+most MSS compatible cards have it. However check that this is true
+before enabling OPL3.
+
+Sound driver supports more than one MPU401 compatible cards at the same time
+but the config program asks config info for just the first of them.
+Adding the second or third MPU interfaces must be done manually by
+editing sound/local.h (after running the config program). Add defines for
+MPU2_BASE & MPU2_IRQ (and MPU3_BASE & MPU3_IRQ) to the file.
+
+CAUTION!
+
+The default I/O base of Adaptec AHA-1542 SCSI controller is 0x330 which
+is also the default of the MPU401 driver. Don't configure the sound driver to
+use 0x330 as the MPU401 base if you have a AHA1542. The kernel will not boot
+if you make this mistake.
+
+PSS
+---
+
+Even the PSS cards are compatible with SB, MSS and MPU401, you must not
+enable these options when configuring the driver. The configuration
+program handles these options itself. (You may use the SB, MPU and MSS options
+together with PSS if you have another card on the system).
+
+The PSS driver enables MSS and MPU401 modes of the card. SB is not enabled
+since it doesn't work concurrently with MSS. The driver loads also a
+DSP algorithm which is used to for the general MIDI emulation. The
+algorithm file (.ld) is read by the config program and written to a
+file included when the pss.c is compiled. For this reason the config
+program asks if you want to download the file. Use the genmidi.ld file
+distributed with the DOS/Windows drivers of the card (don't use the mt32.ld).
+With some cards the file is called 'synth.ld'. You must have access to
+the file when configuring the driver. The easiest way is to mount the DOS
+partition containing the file with Linux.
+
+It's possible to load your own DSP algorithms and run them with the card.
+Look at the directory pss_test of snd-util-3.0.tar.gz for more info.
+
+AudioTrix Pro
+-------------
+
+You have to enable the OPL3 and SB (not SB Pro or SB16) drivers in addition
+to the native AudioTrix driver. Don't enable MSS or MPU drivers.
+
+Configuring ATP is little bit tricky since it uses so many I/O, IRQ and
+DMA numbers. Using the same values than with DOS/Win is a good idea. Don't
+attempt to use the same IRQ or DMA channels twice.
+
+The SB mode of ATP is implemented so the ATP driver just enables SB
+in the proper address. The SB driver handles the rest. You have to configure
+both the SB driver and the SB mode of ATP to use the same IRQ, DMA and I/O
+settings.
+
+Also the ATP has a microcontroller for the General MIDI emulation (OPL4).
+For this reason the driver asks for the name of a file containing the
+microcode (TRXPRO.HEX). This file is usually located in the directory
+where the DOS drivers were installed. You must have access to this file
+when configuring the driver.
+
+If you have the effects daughtercard, it must be initialized by running
+the setfx program of snd-util-3.0.tar.gz package. This step is not required
+when using the (future) binary distribution version of the driver.
+
+Ensoniq SoundScape
+------------------
+
+NOTE! The new PnP SoundScape is not supported yet. Soundscape compatible
+ cards made by Reveal don't work with Linux. They use older revision
+ of the Soundscape chipset which is not fully compatible with
+ newer cards made by Ensoniq.
+
+The SoundScape driver handles initialization of MSS and MPU supports
+itself so you don't need to enable other drivers than SoundScape
+(enable also the /dev/dsp, /dev/sequencer and MIDI supports).
+
+!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
+!!!!! !!!!
+!!!!! NOTE! Before version 3.5-beta6 there WERE two sets of audio !!!!
+!!!!! device files (/dev/dsp0 and /dev/dsp1). The first one WAS !!!!
+!!!!! used only for card initialization and the second for audio !!!!
+!!!!! purposes. It WAS required to change /dev/dsp (a symlink) to !!!!
+!!!!! point to /dev/dsp1. !!!!
+!!!!! !!!!
+!!!!! This is not required with OSS versions 3.5-beta6 and later !!!!
+!!!!! since there is now just one audio device file. Please !!!!
+!!!!! change /dev/dsp to point back to /dev/dsp0 if you are !!!!
+!!!!! upgrading from an earlier driver version using !!!!
+!!!!! (cd /dev;rm dsp;ln -s dsp0 dsp). !!!!
+!!!!! !!!!
+!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
+
+The configuration program asks one DMA channel and two interrupts. One IRQ
+and one DMA is used by the MSS codec. The second IRQ is required for the
+MPU401 mode (you have to use different IRQs for both purposes).
+There were earlier two DMA channels for SoundScape but the current driver
+version requires just one.
+
+The SoundScape card has a Motorola microcontroller which must initialized
+_after_ boot (the driver doesn't initialize it during boot).
+The initialization is done by running the 'ssinit' program which is
+distributed in the snd-util-3.0.tar.gz package. You have to edit two
+defines in the ssinit.c and then compile the program. You may run ssinit
+manually (after each boot) or add it to /etc/rc.d/rc.local.
+
+The ssinit program needs the microcode file that comes with the DOS/Windows
+driver of the card. You will need to use version 1.30.00 or later
+of the microcode file (sndscape.co0 or sndscape.co1 depending on
+your card model). THE OLD sndscape.cod WILL NOT WORK. IT WILL HANG YOUR
+MACHINE. The only way to get the new microcode file is to download
+and install the DOS/Windows driver from ftp://ftp.ensoniq.com/pub.
+
+Then you have to select the proper microcode file to use: soundscape.co0
+is the right one for most cards and sndscape.co1 is for few (older) cards
+made by Reveal and/or Spea. The driver has capability to detect the card
+version during boot. Look at the boot log messages in /var/adm/messages
+and locate the sound driver initialization message for the SoundScape
+card. If the driver displays string , you have
+an old card and you will need to use sndscape.co1. For other cards use
+soundscape.co0. New Soundscape revisions such as Elite and PnP use
+code files with higher numbers (.co2, .co3, etc.).
+
+NOTE! Ensoniq Soundscape VIVO is not compatible with other Soundscape cards.
+ Currently it's possible to use it in Linux only with OSS/Linux
+ drivers.
+
+Check /var/adm/messages after running ssinit. The driver prints
+the board version after downloading the microcode file. That version
+number must match the number in the name of the microcode file (extension).
+
+Running ssinit with a wrong version of the sndscape.co? file is not
+dangerous as long as you don't try to use a file called sndscape.cod.
+If you have initialized the card using a wrong microcode file (sounds
+are terrible), just modify ssinit.c to use another microcode file and try
+again. It's possible to use an earlier version of sndscape.co[01] but it
+may sound weird.
+
+MAD16 (Pro) and Mozart
+----------------------
+
+You need to enable just the MAD16 /Mozart support when configuring
+the driver. _Don't_ enable SB, MPU401 or MSS. However you will need the
+/dev/audio, /dev/sequencer and MIDI supports.
+
+Mozart and OPTi 82C928 (the original MAD16) chips don't support
+MPU401 mode so enter just 0 when the configuration program asks the
+MPU/MIDI I/O base. The MAD16 Pro (OPTi 82C929) and 82C930 chips have MPU401
+mode.
+
+TB Tropez is based on the 82C929 chip. It has two MIDI ports.
+The one connected to the MAD16 chip is the second one (there is a second
+MIDI connector/pins somewhere??). If you have not connected the second MIDI
+port, just disable the MIDI port of MAD16. The 'Maui' compatible synth of
+Tropez is jumper configurable and not connected to the MAD16 chip (the
+Maui driver can be used with it).
+
+Some MAD16 based cards may cause feedback, whistle or terrible noise if the
+line3 mixer channel is turned too high. This happens at least with Shuttle
+Sound System. Current driver versions set volume of line3 low enough so
+this should not be a problem.
+
+If you have a MAD16 card which have an OPL4 (FM + Wave table) synthesizer
+chip (_not_ an OPL3), you have to append a line containing #define MAD16_OPL4
+to the file linux/drivers/sound/local.h (after running make config).
+
+MAD16 cards having a CS4231 codec support full duplex mode. This mode
+can be enabled by configuring the card to use two DMA channels. Possible
+DMA channel pairs are: 0&1, 1&0 and 3&0.
+
+NOTE! Cards having an OPTi 82C924/82C925 chip work with OSS/Free only in
+non-PnP mode (usually jumper selectable). The PnP mode is supported only
+by OSS/Linux.
+
+MV Jazz (ProSonic)
+------------------
+
+The Jazz16 driver is just a hack made to the SB Pro driver. However it works
+fairly well. You have to enable SB, SB Pro (_not_ SB16) and MPU401 supports
+when configuring the driver. The configuration program asks later if you
+want support for MV Jazz16 based cards (after asking SB base address). Answer
+'y' here and the driver asks the second (16 bit) DMA channel.
+
+The Jazz16 driver uses the MPU401 driver in a way which will cause
+problems if you have another MPU401 compatible card. In this case you must
+give address of the Jazz16 based MPU401 interface when the config
+program prompts for the MPU401 information. Then look at the MPU401
+specific section for instructions about configuring more than one MPU401 cards.
+
+Logitech Soundman Wave
+----------------------
+
+Read the above MV Jazz specific instructions first.
+
+The Logitech SoundMan Wave (don't confuse this with the SM16 or SM Games) is
+a MV Jazz based card which has an additional OPL4 based wave table
+synthesizer. The OPL4 chip is handled by an on board microcontroller
+which must be initialized during boot. The config program asks if
+you have a SM Wave immediately after asking the second DMA channel of jazz16.
+If you answer 'y', the config program will ask name of the file containing
+code to be loaded to the microcontroller. The file is usually called
+MIDI0001.BIN and it's located in the DOS/Windows driver directory. The file
+may also be called as TSUNAMI.BIN or something else (older cards?).
+
+The OPL4 synth will be inaccessible without loading the microcontroller code.
+
+Also remember to enable SB MPU401 support if you want to use the OPL4 mode.
+(Don't enable the 'normal' MPU401 device as with some earlier driver
+versions (pre 3.5-alpha8)).
+
+NOTE! Don't answer 'y' when the driver asks about SM Games support
+ (the next question after the MIDI0001.BIN name). However
+ answering 'y' doesn't cause damage your computer so don't panic.
+
+Sound Galaxies
+--------------
+
+There are many different Sound Galaxy cards made by Aztech. The 8 bit
+ones are fully SB or SB Pro compatible and there should be no problems
+with them.
+
+The older 16 bit cards (SG Pro16, SG NX Pro16, Nova and Lyra) have
+an EEPROM chip for storing the configuration data. There is a microcontroller
+which initializes the card to match the EEPROM settings when the machine
+is powered on. These cards actually behave just like they have jumpers
+for all of the settings. Configure driver for MSS, MPU, SB/SB Pro and OPL3
+supports with these cards.
+
+There are some new Sound Galaxies in the market. I have no experience with
+them so read the card's manual carefully.
+
+ESS ES1688 and ES688 'AudioDrive' based cards
+---------------------------------------------
+
+Support for these two ESS chips is embedded in the SB driver.
+Configure these cards just like SB. Enable the 'SB MPU401 MIDI port'
+if you want to use MIDI features of ES1688. ES688 doesn't have MPU mode
+so you don't need to enable it (the driver uses normal SB MIDI automatically
+with ES688).
+
+NOTE! ESS cards are not compatible with MSS/WSS so don't worry if MSS support
+of OSS doesn't work with it.
+
+There are some ES1688/688 based sound cards and (particularly) motherboards
+which use software configurable I/O port relocation feature of the chip.
+This ESS proprietary feature is supported only by OSS/Linux.
+
+There are ES1688 based cards which use different interrupt pin assignment than
+recommended by ESS (5, 7, 9/2 and 10). In this case all IRQs don't work.
+At least a card called (Pearl?) Hypersound 16 supports IRQ 15 but it doesn't
+work.
+
+ES1868 is a PnP chip which is (supposed to be) compatible with ESS1688
+probably works with OSS/Free after initialization using isapnptools.
+
+Reveal cards
+------------
+
+There are several different cards made/marketed by Reveal. Some of them
+are compatible with SoundScape and some use the MAD16 chip. You may have
+to look at the card and try to identify its origin.
+
+Diamond
+-------
+
+The oldest (Sierra Aria based) sound cards made by Diamond are not supported
+(they may work if the card is initialized using DOS). The recent (LX?)
+models are based on the MAD16 chip which is supported by the driver.
+
+Audio Excel DSP16
+-----------------
+
+Support for this card is currently not functional. A new driver for it
+should be available later this year.
+
+PCMCIA cards
+------------
+
+Sorry, can't help. Some cards may work and some don't.
+
+TI TM4000M notebooks
+--------------------
+
+These computers have a built in sound support based on the Jazz chipset.
+Look at the instructions for MV Jazz (above). It's also important to note
+that there is something wrong with the mouse port and sound at least on
+some TM models. Don't enable the "C&T 82C710 mouse port support" when
+configuring Linux. Having it enabled is likely to cause mysterious problems
+and kernel failures when sound is used.
+
+miroSOUND
+---------
+
+The miroSOUND PCM1-pro, PCM12 and PCM20 radio has been used
+successfully. These cards are based on the MAD16, OPL4, and CS4231A chips
+and everything said in the section about MAD16 cards applies here,
+too. The only major difference between the PCMxx and other MAD16 cards
+is that instead of the mixer in the CS4231 codec a separate mixer
+controlled by an on-board 80C32 microcontroller is used. Control of
+the mixer takes place via the ACI (miro's audio control interface)
+protocol that is implemented in a separate lowlevel driver. Make sure
+you compile this ACI driver together with the normal MAD16 support
+when you use a miroSOUND PCMxx card. The ACI mixer is controlled by
+/dev/mixer and the CS4231 mixer by /dev/mixer1 (depends on load
+time). Only in special cases you want to change something regularly on
+the CS4231 mixer.
+
+The miroSOUND PCM12 and PCM20 radio is capable of full duplex
+operation (simultaneous PCM replay and recording), which allows you to
+implement nice real-time signal processing audio effect software and
+network telephones. The ACI mixer has to be switched into the "solo"
+mode for duplex operation in order to avoid feedback caused by the
+mixer (input hears output signal). You can de-/activate this mode
+through toggleing the record button for the wave controller with an
+OSS-mixer.
+
+The PCM20 contains a radio tuner, which is also controlled by
+ACI. This radio tuner is supported by the ACI driver together with the
+miropcm20.o module. Also the 7-band equalizer is integrated
+(limited by the OSS-design). Developement has started and maybe
+finished for the RDS decoder on this card, too. You will be able to
+read RadioText, the Programme Service name, Programme TYpe and
+others. Even the v4l radio module benefits from it with a refined
+strength value. See aci.[ch] and miropcm20*.[ch] for more details.
+
+The following configuration parameters have worked fine for the PCM12
+in Markus Kuhn's system, many other configurations might work, too:
+CONFIG_MAD16_BASE=0x530, CONFIG_MAD16_IRQ=11, CONFIG_MAD16_DMA=3,
+CONFIG_MAD16_DMA2=0, CONFIG_MAD16_MPU_BASE=0x330, CONFIG_MAD16_MPU_IRQ=10,
+DSP_BUFFSIZE=65536, SELECTED_SOUND_OPTIONS=0x00281000.
+
+Bas van der Linden is using his PCM1-pro with a configuration that
+differs in: CONFIG_MAD16_IRQ=7, CONFIG_MAD16_DMA=1, CONFIG_MAD16_MPU_IRQ=9
+
+Compaq Deskpro XL
+-----------------
+
+The builtin sound hardware of Compaq Deskpro XL is now supported.
+You need to configure the driver with MSS and OPL3 supports enabled.
+In addition you need to manually edit linux/drivers/sound/local.h and
+to add a line containing "#define DESKPROXL" if you used
+make menuconfig/xconfig.
+
+Others?
+-------
+
+Since there are so many different sound cards, it's likely that I have
+forgotten to mention many of them. Please inform me if you know yet another
+card which works with Linux, please inform me (or is anybody else
+willing to maintain a database of supported cards (just like in XF86)?).
+
+Cards not supported yet
+=======================
+
+Please check the version of sound driver you are using before
+complaining that your card is not supported. It's possible you are
+using a driver version which was released months before your card was
+introduced.
+
+First of all, there is an easy way to make most sound cards work with Linux.
+Just use the DOS based driver to initialize the card to a known state, then use
+loadlin.exe to boot Linux. If Linux is configured to use the same I/O, IRQ and
+DMA numbers as DOS, the card could work.
+(ctrl-alt-del can be used in place of loadlin.exe but it doesn't work with
+new motherboards). This method works also with all/most PnP sound cards.
+
+Don't get fooled with SB compatibility. Most cards are compatible with
+SB but that may require a TSR which is not possible with Linux. If
+the card is compatible with MSS, it's a better choice. Some cards
+don't work in the SB and MSS modes at the same time.
+
+Then there are cards which are no longer manufactured and/or which
+are relatively rarely used (such as the 8 bit ProAudioSpectrum
+models). It's extremely unlikely that such cards ever get supported.
+Adding support for a new card requires much work and increases time
+required in maintaining the driver (some changes need to be done
+to all low level drivers and be tested too, maybe with multiple
+operating systems). For this reason I have made a decision to not support
+obsolete cards. It's possible that someone else makes a separately
+distributed driver (diffs) for the card.
+
+Writing a driver for a new card is not possible if there are no
+programming information available about the card. If you don't
+find your new card from this file, look from the home page
+(http://www.opensound.com/ossfree). Then please contact
+manufacturer of the card and ask if they have (or are willing to)
+released technical details of the card. Do this before contacting me. I
+can only answer 'no' if there are no programming information available.
+
+I have made decision to not accept code based on reverse engineering
+to the driver. There are three main reasons: First I don't want to break
+relationships to sound card manufacturers. The second reason is that
+maintaining and supporting a driver without any specs will be a pain.
+The third reason is that companies have freedom to refuse selling their
+products to other than Windows users.
+
+Some companies don't give low level technical information about their
+products to public or at least their require signing a NDA. It's not
+possible to implement a freeware driver for them. However it's possible
+that support for such cards become available in the commercial version
+of this driver (see http://www.4Front-tech.com/oss.html for more info).
+
+There are some common audio chipsets that are not supported yet. For example
+Sierra Aria and IBM Mwave. It's possible that these architectures
+get some support in future but I can't make any promises. Just look
+at the home page (http://www.opensound.com/ossfree/new_cards.html)
+for latest info.
+
+Information about unsupported sound cards and chipsets is welcome as well
+as free copies of sound cards, SDKs and operating systems.
+
+If you have any corrections and/or comments, please contact me.
+
+Hannu Savolainen
+hannu@opensound.com
+
+Personal home page: http://www.compusonic.fi/~hannu
+home page of OSS/Free: http://www.opensound.com/ossfree
+
+home page of commercial OSS
+(Open Sound System) drivers: http://www.opensound.com/oss.html
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/README.awe linux-2.5.6/Documentation/sound/oss/README.awe
--- linux-2.5.6-pre3/Documentation/sound/oss/README.awe Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/README.awe Thu Mar 7 18:24:41 2002
@@ -0,0 +1,218 @@
+================================================================
+ AWE32 Sound Driver for Linux / FreeBSD
+ version 0.4.3; Nov. 1, 1998
+
+ Takashi Iwai
+================================================================
+
+* GENERAL NOTES
+
+This is a sound driver extension for SoundBlaster AWE32 and other
+compatible cards (AWE32-PnP, SB32, SB32-PnP, AWE64 & etc) to enable
+the wave synth operations. The driver is provided for Linux 1.2.x
+and 2.[012].x kernels, as well as FreeBSD, on Intel x86 and DEC
+Alpha systems.
+
+This driver was written by Takashi Iwai ,
+and provided "as is". The original source (awedrv-0.4.3.tar.gz) and
+binary packages are available on the following URL:
+ http://bahamut.mm.t.u-tokyo.ac.jp/~iwai/awedrv/
+Note that since the author is apart from this web site, the update is
+not frequent now.
+
+
+* NOTE TO LINUX USERS
+
+To enable this driver on linux-2.[01].x kernels, you need turn on
+"AWE32 synth" options in sound menu when configure your linux kernel
+and modules. The precise installation procedure is described in the
+AWE64-Mini-HOWTO and linux-kernel/Documetation/sound/AWE32.
+
+If you're using PnP cards, the card must be initialized before loading
+the sound driver. There're several options to do this:
+ - Initialize the card via ISA PnP tools, and load the sound module.
+ - Initialize the card on DOS, and load linux by loadlin.exe
+ - Use PnP kernel driver (for Linux-2.x.x)
+The detailed instruction for the solution using isapnp tools is found
+in many documents like above. A brief instruction is also included in
+the installation document of this package.
+For PnP driver project, please refer to the following URL:
+ http://www-jcr.lmh.ox.ac.uk/~pnp/
+
+
+* USING THE DRIVER
+
+The awedrv has several different playing modes to realize easy channel
+allocation for MIDI songs. To hear the exact sound quality, you need
+to obtain the extended sequencer program, drvmidi or playmidi-2.5.
+
+For playing MIDI files, you *MUST* load the soundfont file on the
+driver previously by sfxload utility. Otherwise you'll here no sounds
+at all! All the utilities and driver source packages are found in the
+above URL. The sfxload program is included in the package
+awesfx-0.4.3.tgz. Binary packages are available there, too. See the
+instruction in each package for installation.
+
+Loading a soundfont file is very simple. Just execute the command
+
+ % sfxload synthgm.sbk
+
+Then, sfxload transfers the file "synthgm.sbk" to the driver.
+Both SF1 and SF2 formats are accepted.
+
+Now you can hear midi musics by a midi player.
+
+ % drvmidi foo.mid
+
+If you run MIDI player after MOD player, you need to load soundfont
+files again, since MOD player programs clear the previous loaded
+samples by their own data.
+
+If you have only 512kb on the sound card, I recommend to use dynamic
+sample loading via -L option of drvmidi. 2MB GM/GS soundfont file is
+available in most midi files.
+
+ % sfxload synthgm
+ % drvmidi -L 2mbgmgs foo.mid
+
+This makes a big difference (believe me)! For more details, please
+refer to the FAQ list which is available on the URL above.
+
+The current chorus, reverb and equalizer status can be changed by
+aweset utility program (included in awesfx package). Note that
+some awedrv-native programs (like drvmidi and xmp) will change the
+current settings by themselves. The aweset program is effective
+only for other programs like playmidi.
+
+Enjoy.
+
+
+* COMPILE FLAGS
+
+Compile conditions are defined in awe_config.h.
+
+[Compatibility Conditions]
+The following flags are defined automatically when using installation
+shell script.
+
+- AWE_MODULE_SUPPORT
+ indicates your Linux kernel supports module for each sound card
+ (in recent 2.1 or 2.2 kernels and unofficial patched 2.0 kernels
+ as distributed in the RH5.0 package).
+ This flag is automatically set when you're using 2.1.x kernels.
+ You can pass the base address and memory size via the following
+ module options,
+ io = base I/O port address (eg. 0x620)
+ memsize = DRAM size in kilobytes (eg. 512)
+ As default, AWE driver probes these values automatically.
+
+
+[Hardware Conditions]
+You DON'T have to define the following two values.
+Define them only when the driver couldn't detect the card properly.
+
+- AWE_DEFAULT_BASE_ADDR (default: not defined)
+ specifies the base port address of your AWE32 card.
+ 0 means to autodetect the address.
+
+- AWE_DEFAULT_MEM_SIZE (default: not defined)
+ specifies the memory size of your AWE32 card in kilobytes.
+ -1 means to autodetect its size.
+
+
+[Sample Table Size]
+From ver.0.4.0, sample tables are allocated dynamically (except
+Linux-1.2.x system), so you need NOT to touch these parameters.
+Linux-1.2.x users may need to increase these values to appropriate size
+if the sound card is equipped with more DRAM.
+
+- AWE_MAX_SF_LISTS, AWE_MAX_SAMPLES, AWE_MAX_INFOS
+
+
+[Other Conditions]
+
+- AWE_ALWAYS_INIT_FM (default: not defined)
+ indicates the AWE driver always initialize FM passthrough even
+ without DRAM on board. Emu8000 chip has a restriction for playing
+ samples on DRAM that at least two channels must be occupied as
+ passthrough channels.
+
+- AWE_DEBUG_ON (default: defined)
+ turns on debugging messages if defined.
+
+- AWE_HAS_GUS_COMPATIBILITY (default: defined)
+ Enables GUS compatibility mode if defined, reading GUS patches and
+ GUS control commands. Define this option to use GMOD or other
+ GUS module players.
+
+- CONFIG_AWE32_MIDIEMU (default: defined)
+ Adds a MIDI emulation device by Emu8000 wavetable. The emulation
+ device can be accessed as an external MIDI, and sends the MIDI
+ control codes directly. XG and GS sysex/NRPN are accepted.
+ No MIDI input is supported.
+
+- CONFIG_AWE32_MIXER (default: not defined)
+ Adds a mixer device for AWE32 bass/treble equalizer control.
+ You can access this device using /dev/mixer?? (usually mixer01).
+
+- AWE_USE_NEW_VOLUME_CALC (default: defined)
+ Use the new method to calculate the volume change as compatible
+ with DOS/Win drivers. This option can be toggled via aweset
+ program, or drvmidi player.
+
+- AWE_CHECK_VTARGET (default: defined)
+ Check the current volume target value when searching for an
+ empty channel to allocate a new voice. This is experimentally
+ implemented in this version. (probably, this option doesn't
+ affect the sound quality severely...)
+
+- AWE_ALLOW_SAMPLE_SHARING (default: defined)
+ Allow sample sharing for differently loaded patches.
+ This function is available only together with awesfx-0.4.3p3.
+ Note that this is still an experimental option.
+
+- DEF_FM_CHORUS_DEPTH (default: 0x10)
+ The default strength to be sent to the chorus effect engine.
+ From 0 to 0xff. Larger numbers may often cause weird sounds.
+
+- DEF_FM_REVERB_DEPTH (default: 0x10)
+ The default strength to be sent to the reverb effect engine.
+ From 0 to 0xff. Larger numbers may often cause weird sounds.
+
+
+* ACKNOWLEDGMENTS
+
+Thanks to Witold Jachimczyk (witek@xfactor.wpi.edu) for much advice
+on programming of AWE32. Much code is brought from his AWE32-native
+MOD player, ALMP.
+The port of awedrv to FreeBSD is done by Randall Hopper
+(rhh@ct.picker.com).
+The new volume calculation routine was derived from Mark Weaver's
+ADIP compatible routines.
+I also thank linux-awe-ml members for their efforts
+to reboot their system many times :-)
+
+
+* TODO'S
+
+- Complete DOS/Win compatibility
+- DSP-like output
+
+
+* COPYRIGHT
+
+Copyright (C) 1996-1998 Takashi Iwai
+
+This program is free software; you can redistribute it and/or modify
+it under the terms of the GNU General Public License as published by
+the Free Software Foundation; either version 2 of the License, or
+(at your option) any later version.
+
+This program is distributed in the hope that it will be useful,
+but WITHOUT ANY WARRANTY; without even the implied warranty of
+MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+GNU General Public License for more details.
+
+You should have received a copy of the GNU General Public License
+along with this program; if not, write to the Free Software
+Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/README.modules linux-2.5.6/Documentation/sound/oss/README.modules
--- linux-2.5.6-pre3/Documentation/sound/oss/README.modules Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/README.modules Thu Mar 7 18:24:41 2002
@@ -0,0 +1,105 @@
+Building a modular sound driver
+================================
+
+ The following information is current as of linux-2.1.85. Check the other
+readme files, especially README.OSS, for information not specific to
+making sound modular.
+
+ First, configure your kernel. This is an idea of what you should be
+setting in the sound section:
+
+ Sound card support
+
+ 100% Sound Blaster compatibles (SB16/32/64, ESS, Jazz16) support
+
+ I have SoundBlaster. Select your card from the list.
+
+ Generic OPL2/OPL3 FM synthesizer support
+ FM synthesizer (YM3812/OPL-3) support
+
+ If you don't set these, you will probably find you can play .wav files
+but not .midi. As the help for them says, set them unless you know your
+card does not use one of these chips for FM support.
+
+ Once you are configured, make zlilo, modules, modules_install; reboot.
+Note that it is no longer necessary or possible to configure sound in the
+drivers/sound dir. Now one simply configures and makes one's kernel and
+modules in the usual way.
+
+ Then, add to your /etc/modules.conf something like:
+
+alias char-major-14 sb
+post-install sb /sbin/modprobe "-k" "adlib_card"
+options sb io=0x220 irq=7 dma=1 dma16=5 mpu_io=0x330
+options adlib_card io=0x388 # FM synthesizer
+
+ Alternatively, if you have compiled in kernel level ISAPnP support:
+
+alias char-major-14 sb
+post-install sb /sbin/modprobe "-k" "adlib_card"
+options adlib_card io=0x388
+
+ The effect of this is that the sound driver and all necessary bits and
+pieces autoload on demand, assuming you use kerneld (a sound choice) and
+autoclean when not in use. Also, options for the device drivers are
+set. They will not work without them. Change as appropriate for your card.
+If you are not yet using the very cool kerneld, you will have to "modprobe
+-k sb" yourself to get things going. Eventually things may be fixed so
+that this kludgery is not necessary; for the time being, it seems to work
+well.
+
+ Replace 'sb' with the driver for your card, and give it the right
+options. To find the filename of the driver, look in
+/lib/modules//misc. Mine looks like:
+
+adlib_card.o # This is the generic OPLx driver
+opl3.o # The OPL3 driver
+sb.o # <>
+sound.o # The sound driver
+uart401.o # Used by sb, maybe other cards
+
+ Whichever card you have, try feeding it the options that would be the
+default if you were making the driver wired, not as modules. You can look
+at the init_module() code for the card to see what args are expected.
+
+ Note that at present there is no way to configure the io, irq and other
+parameters for the modular drivers as one does for the wired drivers.. One
+needs to pass the modules the necessary parameters as arguments, either
+with /etc/modules.conf or with command-line args to modprobe, e.g.
+
+modprobe -k sb io=0x220 irq=7 dma=1 dma16=5 mpu_io=0x330
+modprobe -k adlib_card io=0x388
+
+ recommend using /etc/modules.conf.
+
+Persistent DMA Buffers:
+
+The sound modules normally allocate DMA buffers during open() and
+deallocate them during close(). Linux can often have problems allocating
+DMA buffers for ISA cards on machines with more than 16MB RAM. This is
+because ISA DMA buffers must exist below the 16MB boundary and it is quite
+possible that we can't find a large enough free block in this region after
+the machine has been running for any amount of time. The way to avoid this
+problem is to allocate the DMA buffers during module load and deallocate
+them when the module is unloaded. For this to be effective we need to load
+the sound modules right after the kernel boots, either manually or by an
+init script, and keep them around until we shut down. This is a little
+wasteful of RAM, but it guarantees that sound always works.
+
+To make the sound driver use persistent DMA buffers we need to pass the
+sound.o module a "dmabuf=1" command-line argument. This is normally done
+in /etc/modules.conf like so:
+
+options sound dmabuf=1
+
+If you have 16MB or less RAM or a PCI sound card, this is wasteful and
+unnecessary. It is possible that machine with 16MB or less RAM will find
+this option useful, but if your machine is so memory-starved that it
+cannot find a 64K block free, you will be wasting even more RAM by keeping
+the sound modules loaded and the DMA buffers allocated when they are not
+needed. The proper solution is to upgrade your RAM. But you do also have
+this improper solution as well. Use it wisely.
+
+ I'm afraid I know nothing about anything but my setup, being more of a
+text-mode guy anyway. If you have options for other cards or other helpful
+hints, send them to me, Jim Bray, jb@as220.org, http://as220.org/jb.
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/README.ymfsb linux-2.5.6/Documentation/sound/oss/README.ymfsb
--- linux-2.5.6-pre3/Documentation/sound/oss/README.ymfsb Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/README.ymfsb Thu Mar 7 18:24:41 2002
@@ -0,0 +1,107 @@
+Legacy audio driver for YMF7xx PCI cards.
+
+
+FIRST OF ALL
+============
+
+ This code references YAMAHA's sample codes and data sheets.
+ I respect and thank for all people they made open the informations
+ about YMF7xx cards.
+
+ And this codes heavily based on Jeff Garzik 's
+ old VIA 82Cxxx driver (via82cxxx.c). I also respect him.
+
+
+DISCLIMER
+=========
+
+ This driver is currently at early ALPHA stage. It may cause serious
+ damage to your computer when used.
+ PLEASE USE IT AT YOUR OWN RISK.
+
+
+ABOUT THIS DRIVER
+=================
+
+ This code enables you to use your YMF724[A-F], YMF740[A-C], YMF744, YMF754
+ cards. When enabled, your card acts as "SoundBlaster Pro" compatible card.
+ It can only play 22.05kHz / 8bit / Stereo samples, control external MIDI
+ port.
+ If you want to use your card as recent "16-bit" card, you should use
+ Alsa or OSS/Linux driver. Of course you can write native PCI driver for
+ your cards :)
+
+
+USAGE
+=====
+
+ # modprobe ymfsb (options)
+
+
+OPTIONS FOR MODULE
+==================
+
+ io : SB base address (0x220, 0x240, 0x260, 0x280)
+ synth_io : OPL3 base address (0x388, 0x398, 0x3a0, 0x3a8)
+ dma : DMA number (0,1,3)
+ master_volume: AC'97 PCM out Vol (0-100)
+ spdif_out : SPDIF-out flag (0:disable 1:enable)
+
+ These options will change in future...
+
+
+FREQUENCY
+=========
+
+ When playing sounds via this driver, you will hear its pitch is slightly
+ lower than original sounds. Since this driver recognizes your card acts
+ with 21.739kHz sample rates rather than 22.050kHz (I think it must be
+ hardware restriction). So many players become tone deafness.
+ To prevent this, you should express some options to your sound player
+ that specify correct sample frequency. For example, to play your MP3 file
+ correctly with mpg123, specify the frequency like following:
+
+ % mpg123 -r 21739 foo.mp3
+
+
+SPDIF OUT
+=========
+
+ With installing modules with option 'spdif_out=1', you can enjoy your
+ sounds from SPDIF-out of your card (if it had).
+ Its Fs is fixed to 48kHz (It never means the sample frequency become
+ up to 48kHz. All sounds via SPDIF-out also 22kHz samples). So your
+ digital-in capable components has to be able to handle 48kHz Fs.
+
+
+COPYING
+=======
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of the GNU General Public License as published by
+ the Free Software Foundation; either version 2, or (at your option)
+ any later version.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+
+
+TODO
+====
+ * support for multiple cards
+ (set the different SB_IO,MPU_IO,OPL_IO for each cards)
+
+ * support for OPL (dmfm) : There will be no requirements... :-<
+
+
+AUTHOR
+======
+
+ Daisuke Nagano
+
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/SoundPro linux-2.5.6/Documentation/sound/oss/SoundPro
--- linux-2.5.6-pre3/Documentation/sound/oss/SoundPro Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/SoundPro Thu Mar 7 18:24:41 2002
@@ -0,0 +1,105 @@
+Documentation for the SoundPro CMI8330 extensions in the WSS driver (ad1848.o)
+------------------------------------------------------------------------------
+
+( Be sure to read Documentation/sound/CMI8330 too )
+
+Ion Badulescu, ionut@cs.columbia.edu
+February 24, 1999
+
+(derived from the OPL3-SA2 documentation by Scott Murray)
+
+The SoundPro CMI8330 (ISA) is a chip usually found on some Taiwanese
+motherboards. The official name in the documentation is CMI8330, SoundPro
+is the nickname and the big inscription on the chip itself.
+
+The chip emulates a WSS as well as a SB16, but it has certain differences
+in the mixer section which require separate support. It also emulates an
+MPU401 and an OPL3 synthesizer, so you probably want to enable support
+for these, too.
+
+The chip identifies itself as an AD1848, but its mixer is significantly
+more advanced than the original AD1848 one. If your system works with
+either WSS or SB16 and you are having problems with some mixer controls
+(no CD audio, no line-in, etc), you might want to give this driver a try.
+Detection should work, but it hasn't been widely tested, so it might still
+mis-identify the chip. You can still force soundpro=1 in the modprobe
+parameters for ad1848. Please let me know if it happens to you, so I can
+adjust the detection routine.
+
+The chip is capable of doing full-duplex, but since the driver sees it as an
+AD1848, it cannot take advantage of this. Moreover, the full-duplex mode is
+not achievable through the WSS interface, b/c it needs a dma16 line which is
+assigned only to the SB16 subdevice (with isapnp). Windows documentation
+says the user must use WSS Playback and SB16 Recording for full-duplex, so
+it might be possible to do the same thing under Linux. You can try loading
+up both ad1848 and sb then use one for playback and the other for
+recording. I don't know if this works, b/c I haven't tested it. Anyway, if
+you try it, be very careful: the SB16 mixer *mostly* works, but certain
+settings can have unexpected effects. Use the WSS mixer for best results.
+
+There is also a PCI SoundPro chip. I have not seen this chip, so I have
+no idea if the driver will work with it. I suspect it won't.
+
+As with PnP cards, some configuration is required. There are two ways
+of doing this. The most common is to use the isapnptools package to
+initialize the card, and use the kernel module form of the sound
+subsystem and sound drivers. Alternatively, some BIOS's allow manual
+configuration of installed PnP devices in a BIOS menu, which should
+allow using the non-modular sound drivers, i.e. built into the kernel.
+Since in this latter case you cannot use module parameters, you will
+have to enable support for the SoundPro at compile time.
+
+The IRQ and DMA values can be any that are considered acceptable for a
+WSS. Assuming you've got isapnp all happy, then you should be able to
+do something like the following (which *must* match the isapnp/BIOS
+configuration):
+
+modprobe ad1848 io=0x530 irq=11 dma=0 soundpro=1
+-and maybe-
+modprobe sb io=0x220 irq=5 dma=1 dma16=5
+
+-then-
+modprobe mpu401 io=0x330 irq=9
+modprobe opl3 io=0x388
+
+If all goes well and you see no error messages, you should be able to
+start using the sound capabilities of your system. If you get an
+error message while trying to insert the module(s), then make
+sure that the values of the various arguments match what you specified
+in your isapnp configuration file, and that there is no conflict with
+another device for an I/O port or interrupt. Checking the contents of
+/proc/ioports and /proc/interrupts can be useful to see if you're
+butting heads with another device.
+
+If you do not see the chipset version message, and none of the other
+messages present in the system log are helpful, try adding 'debug=1'
+to the ad1848 parameters, email me the syslog results and I'll do
+my best to help.
+
+Lastly, if you're using modules and want to set up automatic module
+loading with kmod, the kernel module loader, here is the section I
+currently use in my conf.modules file:
+
+# Sound
+post-install sound modprobe -k ad1848; modprobe -k mpu401; modprobe -k opl3
+options ad1848 io=0x530 irq=11 dma=0
+options sb io=0x220 irq=5 dma=1 dma16=5
+options mpu401 io=0x330 irq=9
+options opl3 io=0x388
+
+The above ensures that ad1848 will be loaded whenever the sound system
+is being used.
+
+Good luck.
+
+Ion
+
+NOT REALLY TESTED:
+- recording
+- recording device selection
+- full-duplex
+
+TODO:
+- implement mixer support for surround, loud, digital CD switches.
+- come up with a scheme which allows recording volumes for each subdevice.
+This is a major OSS API change.
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/Soundblaster linux-2.5.6/Documentation/sound/oss/Soundblaster
--- linux-2.5.6-pre3/Documentation/sound/oss/Soundblaster Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/Soundblaster Thu Mar 7 18:24:41 2002
@@ -0,0 +1,53 @@
+modprobe sound
+insmod uart401
+insmod sb ...
+
+This loads the driver for the Sound Blaster and assorted clones. Cards that
+are covered by other drivers should not be using this driver.
+
+The Sound Blaster module takes the following arguments
+
+io I/O address of the Sound Blaster chip (0x220,0x240,0x260,0x280)
+irq IRQ of the Sound Blaster chip (5,7,9,10)
+dma 8-bit DMA channel for the Sound Blaster (0,1,3)
+dma16 16-bit DMA channel for SB16 and equivalent cards (5,6,7)
+mpu_io I/O for MPU chip if present (0x300,0x330)
+
+sm_games=1 Set if you have a Logitech soundman games
+acer=1 Set this to detect cards in some ACER notebooks
+mwave_bug=1 Set if you are trying to use this driver with mwave (see on)
+type Use this to specify a specific card type
+
+The following arguments are taken if ISAPnP support is compiled in
+
+isapnp=0 Set this to disable ISAPnP detection (use io=0xXXX etc. above)
+multiple=0 Set to disable detection of multiple Soundblaster cards.
+ Consider it a bug if this option is needed, and send in a
+ report.
+pnplegacy=1 Set this to be able to use a PnP card(s) along with a single
+ non-PnP (legacy) card. Above options for io, irq, etc. are
+ needed, and will apply only to the legacy card.
+reverse=1 Reverses the order of the search in the PnP table.
+uart401=1 Set to enable detection of mpu devices on some clones.
+isapnpjump=n Jumps to slot n in the driver's PnP table. Use the source,
+ Luke.
+
+You may well want to load the opl3 driver for synth music on most SB and
+clone SB devices
+
+insmod opl3 io=0x388
+
+Using Mwave
+
+To make this driver work with Mwave you must set mwave_bug. You also need
+to warm boot from DOS/Windows with the required firmware loaded under this
+OS. IBM are being difficult about documenting how to load this firmware.
+
+Avance Logic ALS007
+
+This card is supported; see the separate file ALS007 for full details.
+
+Avance Logic ALS100
+
+This card is supported; setup should be as for a standard Sound Blaster 16.
+The driver will identify the audio device as a "Sound Blaster 16 (ALS-100)".
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/Tropez+ linux-2.5.6/Documentation/sound/oss/Tropez+
--- linux-2.5.6-pre3/Documentation/sound/oss/Tropez+ Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/Tropez+ Thu Mar 7 18:24:41 2002
@@ -0,0 +1,26 @@
+From: Paul Barton-Davis
+
+Here is the configuration I use with a Tropez+ and my modular
+driver:
+
+ alias char-major-14 wavefront
+ alias synth0 wavefront
+ alias mixer0 cs4232
+ alias audio0 cs4232
+ pre-install wavefront modprobe "-k" "cs4232"
+ post-install wavefront modprobe "-k" "opl3"
+ options wavefront io=0x200 irq=9
+ options cs4232 synthirq=9 synthio=0x200 io=0x530 irq=5 dma=1 dma2=0
+ options opl3 io=0x388
+
+Things to note:
+
+ the wavefront options "io" and "irq" ***MUST*** match the "synthio"
+ and "synthirq" cs4232 options.
+
+ you can do without the opl3 module if you don't
+ want to use the OPL/[34] synth on the soundcard
+
+ the opl3 io parameter is conventionally not adjustable.
+
+Please see drivers/sound/README.wavefront for more details.
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/VIA-chipset linux-2.5.6/Documentation/sound/oss/VIA-chipset
--- linux-2.5.6-pre3/Documentation/sound/oss/VIA-chipset Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/VIA-chipset Thu Mar 7 18:24:41 2002
@@ -0,0 +1,43 @@
+Running sound cards on VIA chipsets
+
+o There are problems with VIA chipsets and sound cards that appear to
+ lock the hardware solidly. Test programs under DOS have verified the
+ problem exists on at least some (but apparently not all) VIA boards
+
+o VIA have so far failed to bother to answer support mail on the subject
+ so if you are a VIA engineer feeling aggrieved as you read this
+ document go chase your own people. If there is a workaround please
+ let us know so we can implement it.
+
+
+Certain patterns of ISA DMA access used for most PC sound cards cause the
+VIA chipsets to lock up. From the collected reports this appears to cover a
+wide range of boards. Some also lock up with sound cards under Win* as well.
+
+Linux implements a workaround providing your chipset is PCI and you compiled
+with PCI Quirks enabled. If so you will see a message
+ "Activating ISA DMA bug workarounds"
+
+during booting. If you have a VIA PCI chipset that hangs when you use the
+sound and is not generating this message even with PCI quirks enabled
+please report the information to the linux-kernel list (see REPORTING-BUGS).
+
+If you are one of the tiny number of unfortunates with a 486 ISA/VLB VIA
+chipset board you need to do the following to build a special kernel for
+your board
+
+ edit linux/include/asm-i386/dma.h
+
+change
+
+#define isa_dma_bridge_buggy (0)
+
+to
+
+#define isa_dma_bridge_buggy (1)
+
+and rebuild a kernel without PCI quirk support.
+
+
+Other than this particular glitch the VIA [M]VP* chipsets appear to work
+perfectly with Linux.
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/VIBRA16 linux-2.5.6/Documentation/sound/oss/VIBRA16
--- linux-2.5.6-pre3/Documentation/sound/oss/VIBRA16 Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/VIBRA16 Thu Mar 7 18:24:41 2002
@@ -0,0 +1,80 @@
+Sound Blaster 16X Vibra addendum
+--------------------------------
+by Marius Ilioaea
+ Stefan Laudat
+
+Sat Mar 6 23:55:27 EET 1999
+
+ Hello again,
+
+ Playing with a SB Vibra 16x soundcard we found it very difficult
+to setup because the kernel reported a lot of DMA errors and wouldn't
+simply play any sound.
+ A good starting point is that the vibra16x chip full-duplex facility
+is neither still exploited by the sb driver found in the linux kernel
+(tried it with a 2.2.2-ac7), nor in the commercial OSS package (it reports
+it as half-duplex soundcard). Oh, I almost forgot, the RedHat sndconfig
+failed detecting it ;)
+ So, the big problem still remains, because the sb module wants a
+8-bit and a 16-bit dma, which we could not allocate for vibra... it supports
+only two 8-bit dma channels, the second one will be passed to the module
+as a 16 bit channel, the kernel will yield about that but everything will
+be okay, trust us.
+ The only inconvenient you may find is that you will have
+some sound playing jitters if you have HDD dma support enabled - but this
+will happen with almost all soundcards...
+
+ A fully working isapnp.conf is just here:
+
+
+
+(READPORT 0x0203)
+(ISOLATE PRESERVE)
+(IDENTIFY *)
+(VERBOSITY 2)
+(CONFLICT (IO FATAL)(IRQ FATAL)(DMA FATAL)(MEM FATAL)) # or WARNING
+# SB 16 and OPL3 devices
+(CONFIGURE CTL00f0/-1 (LD 0
+(INT 0 (IRQ 5 (MODE +E)))
+(DMA 0 (CHANNEL 1))
+(DMA 1 (CHANNEL 3))
+(IO 0 (SIZE 16) (BASE 0x0220))
+(IO 2 (SIZE 4) (BASE 0x0388))
+(NAME "CTL00f0/-1[0]{Audio }")
+(ACT Y)
+))
+
+# Joystick device - only if you need it :-/
+
+(CONFIGURE CTL00f0/-1 (LD 1
+(IO 0 (SIZE 1) (BASE 0x0200))
+(NAME "CTL00f0/-1[1]{Game }")
+(ACT Y)
+))
+(WAITFORKEY)
+
+
+
+ So, after a good kernel modules compilation and a 'depmod -a kernel_ver'
+you may want to:
+
+modprobe sb io=0x220 irq=5 dma=1 dma16=3
+
+ Or, take the hard way:
+
+insmod souncore
+insmod sound
+insmod uart401
+insmod sb io=0x220 irq=5 dma=1 dma16=3
+# do you need MIDI?
+insmod opl3=0x388
+
+ Just in case, the kernel sound support should be:
+
+CONFIG_SOUND=m
+CONFIG_SOUND_OSS=m
+CONFIG_SOUND_SB=m
+
+ Enjoy your new noisy Linux box! ;)
+
+
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/WaveArtist linux-2.5.6/Documentation/sound/oss/WaveArtist
--- linux-2.5.6-pre3/Documentation/sound/oss/WaveArtist Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/WaveArtist Thu Mar 7 18:24:41 2002
@@ -0,0 +1,170 @@
+
+ (the following is from the armlinux CVS)
+
+ WaveArtist mixer and volume levels can be accessed via these commands:
+
+ nn30 read registers nn, where nn = 00 - 09 for mixer settings
+ 0a - 13 for channel volumes
+ mm31 write the volume setting in pairs, where mm = (nn - 10) / 2
+ rr32 write the mixer settings in pairs, where rr = nn/2
+ xx33 reset all settings to default
+ 0y34 select mono source, y=0 = left, y=1 = right
+
+ bits
+ nn 15 14 13 12 11 10 9 8 7 6 5 4 3 2 1 0
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 00 | 0 | 0 0 1 1 | left line mixer gain | left aux1 mixer gain |lmute|
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 01 | 0 | 0 1 0 1 | left aux2 mixer gain | right 2 left mic gain |mmute|
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 02 | 0 | 0 1 1 1 | left mic mixer gain | left mic | left mixer gain |dith |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 03 | 0 | 1 0 0 1 | left mixer input select |lrfg | left ADC gain |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 04 | 0 | 1 0 1 1 | right line mixer gain | right aux1 mixer gain |rmute|
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 05 | 0 | 1 1 0 1 | right aux2 mixer gain | left 2 right mic gain |test |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 06 | 0 | 1 1 1 1 | right mic mixer gain | right mic |right mixer gain |rbyps|
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 07 | 1 | 0 0 0 1 | right mixer select |rrfg | right ADC gain |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 08 | 1 | 0 0 1 1 | mono mixer gain |right ADC mux sel|left ADC mux sel |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 09 | 1 | 0 1 0 1 |loopb|left linout|loop|ADCch|TxFch|OffCD|test |loopb|loopb|osamp|
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 0a | 0 | left PCM channel volume |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 0b | 0 | right PCM channel volume |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 0c | 0 | left FM channel volume |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 0d | 0 | right FM channel volume |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 0e | 0 | left wavetable channel volume |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 0f | 0 | right wavetable channel volume |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 10 | 0 | left PCM expansion channel volume |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 11 | 0 | right PCM expansion channel volume |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 12 | 0 | left FM expansion channel volume |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 13 | 0 | right FM expansion channel volume |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+
+ lmute: left mute
+ mmute: mono mute
+ dith: dithds
+ lrfg:
+ rmute: right mute
+ rbyps: right bypass
+ rrfg:
+ ADCch:
+ TxFch:
+ OffCD:
+ osamp:
+
+ And the following diagram is derived from the description in the CVS archive:
+
+ MIC L (mouthpiece)
+ +------+
+ -->PreAmp>-\
+ +--^---+ |
+ | |
+ r2b4-5 | +--------+
+ /----*-------------------------------->5 |
+ | | |
+ | /----------------------------------->4 |
+ | | | |
+ | | /--------------------------------->3 1of5 | +---+
+ | | | | mux >-->AMP>--> ADC L
+ | | | /------------------------------->2 | +-^-+
+ | | | | | | |
+ Line | | | | +----+ +------+ +---+ /---->1 | r3b3-0
+ ------------*->mute>--> Gain >--> | | | |
+ L | | | +----+ +------+ | | | *->0 |
+ | | | | | | +---^----+
+ Aux2 | | | +----+ +------+ | | | |
+ ----------*--->mute>--> Gain >--> M | | r8b0-2
+ L | | +----+ +------+ | | |
+ | | | | \------\
+ Aux1 | | +----+ +------+ | | |
+ --------*----->mute>--> Gain >--> I | |
+ L | +----+ +------+ | | |
+ | | | |
+ | +----+ +------+ | | +---+ |
+ *------->mute>--> Gain >--> X >-->AMP>--*
+ | +----+ +------+ | | +-^-+ |
+ | | | | |
+ | +----+ +------+ | | r2b1-3 |
+ | /----->mute>--> Gain >--> E | |
+ | | +----+ +------+ | | |
+ | | | | |
+ | | +----+ +------+ | | |
+ | | /--->mute>--> Gain >--> R | |
+ | | | +----+ +------+ | | |
+ | | | | | | r9b8-9
+ | | | +----+ +------+ | | | |
+ | | | /->mute>--> Gain >--> | | +---v---+
+ | | | | +----+ +------+ +---+ /-*->0 |
+ DAC | | | | | | |
+ ------------*----------------------------------->? | +----+
+ L | | | | | Mux >-->mute>--> L output
+ | | | | /->? | +--^-+
+ | | | | | | | |
+ | | | /--------->? | r0b0
+ | | | | | | +-------+
+ | | | | | |
+ Mono | | | | | | +-------+
+ ----------* | \---> | +----+
+ | | | | | | Mix >-->mute>--> Mono output
+ | | | | *-> | +--^-+
+ | | | | | +-------+ |
+ | | | | | r1b0
+ DAC | | | | | +-------+
+ ------------*-------------------------*--------->1 | +----+
+ R | | | | | | Mux >-->mute>--> R output
+ | | | | +----+ +------+ +---+ *->0 | +--^-+
+ | | | \->mute>--> Gain >--> | | +---^---+ |
+ | | | +----+ +------+ | | | | r5b0
+ | | | | | | r6b0
+ | | | +----+ +------+ | | |
+ | | \--->mute>--> Gain >--> M | |
+ | | +----+ +------+ | | |
+ | | | | |
+ | | +----+ +------+ | | |
+ | *----->mute>--> Gain >--> I | |
+ | | +----+ +------+ | | |
+ | | | | |
+ | | +----+ +------+ | | +---+ |
+ \------->mute>--> Gain >--> X >-->AMP>--*
+ | +----+ +------+ | | +-^-+ |
+ /--/ | | | |
+ Aux1 | +----+ +------+ | | r6b1-3 |
+ -------*------>mute>--> Gain >--> E | |
+ R | | +----+ +------+ | | |
+ | | | | |
+ Aux2 | | +----+ +------+ | | /------/
+ ---------*---->mute>--> Gain >--> R | |
+ R | | | +----+ +------+ | | |
+ | | | | | | +--------+
+ Line | | | +----+ +------+ | | | *->0 |
+ -----------*-->mute>--> Gain >--> | | | |
+ R | | | | +----+ +------+ +---+ \---->1 |
+ | | | | | |
+ | | | \-------------------------------->2 | +---+
+ | | | | Mux >-->AMP>--> ADC R
+ | | \---------------------------------->3 | +-^-+
+ | | | | |
+ | \------------------------------------>4 | r7b3-0
+ | | |
+ \-----*-------------------------------->5 |
+ | +---^----+
+ r6b4-5 | |
+ | | r8b3-5
+ +--v---+ |
+ -->PreAmp>-/
+ +------+
+ MIC R (electret mic)
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/Wavefront linux-2.5.6/Documentation/sound/oss/Wavefront
--- linux-2.5.6-pre3/Documentation/sound/oss/Wavefront Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/Wavefront Thu Mar 7 18:24:41 2002
@@ -0,0 +1,341 @@
+ An OSS/Free Driver for WaveFront soundcards
+ (Turtle Beach Maui, Tropez, Tropez Plus)
+
+ Paul Barton-Davis, July 1998
+
+ VERSION 0.2.5
+
+Driver Status
+-------------
+
+Requires: Kernel 2.1.106 or later (the driver is included with kernels
+2.1.109 and above)
+
+As of 7/22/1998, this driver is currently in *BETA* state. This means
+that it compiles and runs, and that I use it on my system (Linux
+2.1.106) with some reasonably demanding applications and uses. I
+believe the code is approaching an initial "finished" state that
+provides bug-free support for the Tropez Plus.
+
+Please note that to date, the driver has ONLY been tested on a Tropez
+Plus. I would very much like to hear (and help out) people with Tropez
+and Maui cards, since I think the driver can support those cards as
+well.
+
+Finally, the driver has not been tested (or even compiled) as a static
+(non-modular) part of the kernel. Alan Cox's good work in modularizing
+OSS/Free for Linux makes this rather unnecessary.
+
+Some Questions
+--------------
+
+**********************************************************************
+0) What does this driver do that the maui driver did not ?
+**********************************************************************
+
+* can fully initialize a WaveFront card from cold boot - no DOS
+ utilities needed
+* working patch/sample/program loading and unloading (the maui
+ driver didn't document how to make this work, and assumed
+ user-level preparation of the patch data for writing
+ to the board. ick.)
+* full user-level access to all WaveFront commands
+* for the Tropez Plus, (primitive) control of the YSS225 FX processor
+* Virtual MIDI mode supported - 2 MIDI devices accessible via the
+ WaveFront's MPU401/UART emulation. One
+ accesses the WaveFront synth, the other accesses the
+ external MIDI connector. Full MIDI read/write semantics
+ for both devices.
+* OSS-compliant /dev/sequencer interface for the WaveFront synth,
+ including native and GUS-format patch downloading.
+* semi-intelligent patch management (prototypical at this point)
+
+**********************************************************************
+1) What to do about MIDI interfaces ?
+**********************************************************************
+
+The Tropez Plus (and perhaps other WF cards) can in theory support up
+to 2 physical MIDI interfaces. One of these is connected to the
+ICS2115 chip (the WaveFront synth itself) and is controlled by
+MPU/UART-401 emulation code running as part of the WaveFront OS. The
+other is controlled by the CS4232 chip present on the board. However,
+physical access to the CS4232 connector is difficult, and it is
+unlikely (though not impossible) that you will want to use it.
+
+An older version of this driver introduced an additional kernel config
+variable which controlled whether or not the CS4232 MIDI interface was
+configured. Because of Alan Cox's work on modularizing the sound
+drivers, and now backporting them to 2.0.34 kernels, there seems to be
+little reason to support "static" configuration variables, and so this
+has been abandoned in favor of *only* module parameters. Specifying
+"mpuio" and "mpuirq" for the cs4232 parameter will result in the
+CS4232 MIDI interface being configured; leaving them unspecified will
+leave it unconfigured (and thus unusable).
+
+BTW, I have heard from one Tropez+ user that the CS4232 interface is
+more reliable than the ICS2115 one. I have had no problems with the
+latter, and I don't have the right cable to test the former one
+out. Reports welcome.
+
+**********************************************************************
+2) Why does line XXX of the code look like this .... ?
+**********************************************************************
+
+Either because its not finished yet, or because you're a better coder
+than I am, or because you don't understand some aspect of how the card
+or the code works.
+
+I absolutely welcome comments, criticisms and suggestions about the
+design and implementation of the driver.
+
+**********************************************************************
+3) What files are included ?
+**********************************************************************
+
+ drivers/sound/README.wavefront -- this file
+
+ drivers/sound/wavefront.patch -- patches for the 2.1.106 sound drivers
+ needed to make the rest of this work
+ DO NOT USE IF YOU'VE APPLIED THEM
+ BEFORE, OR HAVE 2.1.109 OR ABOVE
+
+ drivers/sound/wavfront.c -- the driver
+ drivers/sound/ys225.h -- data declarations for FX config
+ drivers/sound/ys225.c -- data definitions for FX config
+ drivers/sound/wf_midi.c -- the "uart401" driver
+ to support virtual MIDI mode.
+ include/wavefront.h -- the header file
+ Documentation/sound/Tropez+ -- short docs on configuration
+
+**********************************************************************
+4) How do I compile/install/use it ?
+**********************************************************************
+
+PART ONE: install the source code into your sound driver directory
+
+ cd
+ tar -zxvf
+
+PART TWO: apply the patches
+
+ DO THIS ONLY IF YOU HAVE A KERNEL VERSION BELOW 2.1.109
+ AND HAVE NOT ALREADY INSTALLED THE PATCH(ES).
+
+ cd drivers/sound
+ patch < wavefront.patch
+
+PART THREE: configure your kernel
+
+ cd
+ make xconfig (or whichever config option you use)
+
+ - choose YES for Sound Support
+ - choose MODULE (M) for OSS Sound Modules
+ - choose MODULE(M) to YM3812/OPL3 support
+ - choose MODULE(M) for WaveFront support
+ - choose MODULE(M) for CS4232 support
+
+ - choose "N" for everything else (unless you have other
+ soundcards you want support for)
+
+
+ make dep
+ make boot
+ .
+ .
+ .
+
+ make modules
+ .
+ .
+ .
+ make modules_install
+
+Here's my autoconf.h SOUND section:
+
+/*
+ * Sound
+ */
+#define CONFIG_SOUND 1
+#undef CONFIG_SOUND_OSS
+#define CONFIG_SOUND_OSS_MODULE 1
+#undef CONFIG_SOUND_PAS
+#undef CONFIG_SOUND_SB
+#undef CONFIG_SOUND_ADLIB
+#undef CONFIG_SOUND_GUS
+#undef CONFIG_SOUND_MPU401
+#undef CONFIG_SOUND_PSS
+#undef CONFIG_SOUND_MSS
+#undef CONFIG_SOUND_SSCAPE
+#undef CONFIG_SOUND_TRIX
+#undef CONFIG_SOUND_MAD16
+#undef CONFIG_SOUND_WAVEFRONT
+#define CONFIG_SOUND_WAVEFRONT_MODULE 1
+#undef CONFIG_SOUND_CS4232
+#define CONFIG_SOUND_CS4232_MODULE 1
+#undef CONFIG_SOUND_MAUI
+#undef CONFIG_SOUND_SGALAXY
+#undef CONFIG_SOUND_OPL3SA1
+#undef CONFIG_SOUND_SOFTOSS
+#undef CONFIG_SOUND_YM3812
+#define CONFIG_SOUND_YM3812_MODULE 1
+#undef CONFIG_SOUND_VMIDI
+#undef CONFIG_SOUND_UART6850
+/*
+ * Additional low level sound drivers
+ */
+#undef CONFIG_LOWLEVEL_SOUND
+
+************************************************************
+6) How do I configure my card ?
+************************************************************
+
+You need to edit /etc/modules.conf. Here's mine (edited to show the
+relevant details):
+
+ # Sound system
+ alias char-major-14 wavefront
+ alias synth0 wavefront
+ alias mixer0 cs4232
+ alias audio0 cs4232
+ pre-install wavefront modprobe "-k" "cs4232"
+ post-install wavefront modprobe "-k" "opl3"
+ options wavefront io=0x200 irq=9
+ options cs4232 synthirq=9 synthio=0x200 io=0x530 irq=5 dma=1 dma2=0
+ options opl3 io=0x388
+
+Things to note:
+
+ the wavefront options "io" and "irq" ***MUST*** match the "synthio"
+ and "synthirq" cs4232 options.
+
+ you can do without the opl3 module if you don't
+ want to use the OPL/[34] FM synth on the soundcard
+
+ the opl3 io parameter is conventionally not adjustable.
+ In theory, any not-in-use IO port address would work, but
+ just use 0x388 and stick with the crowd.
+
+**********************************************************************
+7) What about firmware ?
+**********************************************************************
+
+Turtle Beach have not given me permission to distribute their firmware
+for the ICS2115. However, if you have a WaveFront card, then you
+almost certainly have the firmware, and if not, its freely available
+on their website, at:
+
+ http://www.tbeach.com/tbs/downloads/scardsdown.htm#tropezplus
+
+The file is called WFOS2001.MOT (for the Tropez+).
+
+This driver, however, doesn't use the pure firmware as distributed,
+but instead relies on a somewhat processed form of it. You can
+generate this very easily. Following an idea from Andrew Veliath's
+Pinnacle driver, the following flex program will generate the
+processed version:
+
+---- cut here -------------------------
+%option main
+%%
+^S[28].*\r$ printf ("%c%.*s", yyleng-1,yyleng-1,yytext);
+<> { fputc ('\0', stdout); return; }
+\n {}
+. {}
+---- cut here -------------------------
+
+To use it, put the above in file (say, ws.l) compile it like this:
+
+ shell> flex -ows.c ws.l
+ shell> cc -o ws ws.c
+
+and then use it like this:
+
+ ws < my-copy-of-the-oswf.mot-file > /etc/sound/wavefront.os
+
+If you put it somewhere else, you'll always have to use the wf_ospath
+module parameter (see below) or alter the source code.
+
+**********************************************************************
+7) How do I get it working ?
+**********************************************************************
+
+Optionally, you can reboot with the "new" kernel (even though the only
+changes have really been made to a module).
+
+Then, as root do:
+
+ modprobe wavefront
+
+You should get something like this in /var/log/messages:
+
+ WaveFront: firmware 1.20 already loaded.
+
+or
+
+ WaveFront: no response to firmware probe, assume raw.
+
+then:
+
+ WaveFront: waiting for memory configuration ...
+ WaveFront: hardware version 1.64
+ WaveFront: available DRAM 8191k
+ WaveFront: 332 samples used (266 real, 13 aliases, 53 multi), 180 empty
+ WaveFront: 128 programs slots in use
+ WaveFront: 256 patch slots filled, 142 in use
+
+The whole process takes about 16 seconds, the longest waits being
+after reporting the hardware version (during the firmware download),
+and after reporting program status (during patch status inquiry). Its
+shorter (about 10 secs) if the firmware is already loaded (i.e. only
+warm reboots since the last firmware load).
+
+The "available DRAM" line will vary depending on how much added RAM
+your card has. Mine has 8MB.
+
+To check basically functionality, use play(1) or splay(1) to send a
+.WAV or other audio file through the audio portion. Then use playmidi
+to play a General MIDI file. Try the "-D 0" to hear the
+difference between sending MIDI to the WaveFront and using the OPL/3,
+which is the default (I think ...). If you have an external synth(s)
+hooked to the soundcard, you can use "-e" to route to the
+external synth(s) (in theory, -D 1 should work as well, but I think
+there is a bug in playmidi which prevents this from doing what it
+should).
+
+**********************************************************************
+8) What are the module parameters ?
+**********************************************************************
+
+Its best to read wavefront.c for this, but here is a summary:
+
+integers:
+ wf_raw - if set, ignore apparent presence of firmware
+ loaded onto the ICS2115, reset the whole
+ board, and initialize it from scratch. (default = 0)
+
+ fx_raw - if set, always initialize the YSS225 processor
+ on the Tropez plus. (default = 1)
+
+ < The next 4 are basically for kernel hackers to allow
+ tweaking the driver for testing purposes. >
+
+ wait_usecs - loop timer used when waiting for
+ status conditions on the board.
+ The default is 150.
+
+ debug_default - debugging flags. See sound/wavefront.h
+ for WF_DEBUG_* values. Default is zero.
+ Setting this allows you to debug the
+ driver during module installation.
+strings:
+ ospath - path to get to the pre-processed OS firmware.
+ (default: /etc/sound/wavefront.os)
+
+**********************************************************************
+9) Who should I contact if I have problems?
+**********************************************************************
+
+Just me: Paul Barton-Davis
+
+
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/btaudio linux-2.5.6/Documentation/sound/oss/btaudio
--- linux-2.5.6-pre3/Documentation/sound/oss/btaudio Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/btaudio Thu Mar 7 18:24:41 2002
@@ -0,0 +1,92 @@
+
+Intro
+=====
+
+people start bugging me about this with questions, looks like I
+should write up some documentation for this beast. That way I
+don't have to answer that much mails I hope. Yes, I'm lazy...
+
+
+You might have noticed that the bt878 grabber cards have actually
+_two_ PCI functions:
+
+$ lspci
+[ ... ]
+00:0a.0 Multimedia video controller: Brooktree Corporation Bt878 (rev 02)
+00:0a.1 Multimedia controller: Brooktree Corporation Bt878 (rev 02)
+[ ... ]
+
+The first does video, it is backward compatible to the bt848. The second
+does audio. btaudio is a driver for the second function. It's a sound
+driver which can be used for recording sound (and _only_ recording, no
+playback). As most TV cards come with a short cable which can be plugged
+into your sound card's line-in you probably don't need this driver if all
+you want to do is just watching TV...
+
+
+Driver Status
+=============
+
+Still somewhat experimental. The driver should work stable, i.e. it
+should'nt crash your box. It might not work as expected, have bugs,
+not being fully OSS API compilant, ...
+
+Latest versions are available from http://bytesex.org/bttv/, the
+driver is in the bttv tarball. Kernel patches might be available too,
+have a look at http://bytesex.org/bttv/listing.html.
+
+The chip knows two different modes. btaudio registers two dsp
+devices, one for each mode. They can not be used at the same time.
+
+
+Digital audio mode
+==================
+
+The chip gives you 16 bit stereo sound. The sample rate depends on
+the external source which feeds the bt878 with digital sound via I2S
+interface. There is a insmod option (rate) to tell the driver which
+sample rate the hardware uses (32000 is the default).
+
+One possible source for digital sound is the msp34xx audio processor
+chip which provides digital sound via I2S with 32 kHz sample rate. My
+Hauppauge board works this way.
+
+The Osprey-200 reportly gives you digital sound with 44100 Hz sample
+rate. It is also possible that you get no sound at all.
+
+
+analog mode (A/D)
+=================
+
+You can tell the driver to use this mode with the insmod option "analog=1".
+The chip has three analog inputs. Consequently you'll get a mixer device
+to control these.
+
+The analog mode supports mono only. Both 8 + 16 bit. Both are _signed_
+int, which is uncommon for the 8 bit case. Sample rate range is 119 kHz
+to 448 kHz. Yes, the number of digits is correct. The driver supports
+downsampling by powers of two, so you can ask for more usual sample rates
+like 44 kHz too.
+
+With my Hauppauge I get noisy sound on the second input (mapped to line2
+by the mixer device). Others get a useable signal on line1.
+
+
+some examples
+=============
+
+* read audio data from btaudio (dsp2), send to es1730 (dsp,dsp1):
+ $ sox -w -r 32000 -t ossdsp /dev/dsp2 -t ossdsp /dev/dsp
+
+* read audio data from btaudio, send to esound daemon (which might be
+ running on another host):
+ $ sox -c 2 -w -r 32000 -t ossdsp /dev/dsp2 -t sw - | esdcat -r 32000
+ $ sox -c 1 -w -r 32000 -t ossdsp /dev/dsp2 -t sw - | esdcat -m -r 32000
+
+
+Have fun,
+
+ Gerd
+
+--
+Gerd Knorr
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/cs46xx linux-2.5.6/Documentation/sound/oss/cs46xx
--- linux-2.5.6-pre3/Documentation/sound/oss/cs46xx Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/cs46xx Thu Mar 7 18:24:41 2002
@@ -0,0 +1,138 @@
+
+Documentation for the Cirrus Logic/Crystal SoundFusion cs46xx/cs4280 audio
+controller chips (2001/05/11)
+
+The cs46xx audio driver supports the DSP line of Cirrus controllers.
+Specifically, the cs4610, cs4612, cs4614, cs4622, cs4624, cs4630 and the cs4280
+products. This driver uses the generic ac97_codec driver for AC97 codec
+support.
+
+
+Features:
+
+Full Duplex Playback/Capture supported from 8k-48k.
+16Bit Signed LE & 8Bit Unsigned, with Mono or Stereo supported.
+
+APM/PM - 2.2.x PM is enabled and functional. APM can also
+be enabled for 2.4.x by modifying the CS46XX_ACPI_SUPPORT macro
+definition.
+
+DMA playback buffer size is configurable from 16k (defaultorder=2) up to 2Meg
+(defaultorder=11). DMA capture buffer size is fixed at a single 4k page as
+two 2k fragments.
+
+MMAP seems to work well with QuakeIII, and test XMMS plugin.
+
+Myth2 works, but the polling logic is not fully correct, but is functional.
+
+The 2.4.4-ac6 gameport code in the cs461x joystick driver has been tested
+with a Microsoft Sidewinder joystick (cs461x.o and sidewinder.o). This
+audio driver must be loaded prior to the joystick driver to enable the
+DSP task image supporting the joystick device.
+
+
+Limitations:
+
+SPDIF is currently not supported.
+
+Primary codec support only. No secondary codec support is implemented.
+
+
+
+NOTES:
+
+Hercules Game Theatre XP - the EGPIO2 pin controls the external Amp,
+and has been tested.
+Module parameter hercules_egpio_disable set to 1, will force a 0 to EGPIODR
+to disable the external amplifier.
+
+VTB Santa Cruz - the GPIO7/GPIO8 on the Secondary Codec control
+the external amplifier for the "back" speakers, since we do not
+support the secondary codec then this external amp is not
+turned on. The primary codec external amplifier is supported but
+note that the AC97 EAPD bit is inverted logic (amp_voyetra()).
+
+DMA buffer size - there are issues with many of the Linux applications
+concerning the optimal buffer size. Several applications request a
+certain fragment size and number and then do not verify that the driver
+has the ability to support the requested configuration.
+SNDCTL_DSP_SETFRAGMENT ioctl is used to request a fragment size and
+number of fragments. Some applications exit if an error is returned
+on this particular ioctl. Therefore, in alignment with the other OSS audio
+drivers, no error is returned when a SETFRAGs IOCTL is received, but the
+values passed from the app are not used in any buffer calculation
+(ossfragshift/ossmaxfrags are not used).
+Use the "defaultorder=N" module parameter to change the buffer size if
+you have an application that requires a specific number of fragments
+or a specific buffer size (see below).
+
+Debug Interface
+---------------
+There is an ioctl debug interface to allow runtime modification of the
+debug print levels. This debug interface code can be disabled from the
+compilation process with commenting the following define:
+#define CSDEBUG_INTERFACE 1
+There is also a debug print methodolgy to select printf statements from
+different areas of the driver. A debug print level is also used to allow
+additional printfs to be active. Comment out the following line in the
+driver to disable compilation of the CS_DBGOUT print statements:
+#define CSDEBUG 1
+
+Please see the defintions for cs_debuglevel and cs_debugmask for additional
+information on the debug levels and sections.
+
+There is also a csdbg executable to allow runtime manipulation of these
+parameters. for a copy email: twoller@crystal.cirrus.com
+
+
+
+MODULE_PARMS definitions
+------------------------
+MODULE_PARM(defaultorder, "i");
+defaultorder=N
+where N is a value from 1 to 12
+The buffer order determines the size of the dma buffer for the driver.
+under Linux, a smaller buffer allows more responsiveness from many of the
+applications (e.g. games). A larger buffer allows some of the apps (esound)
+to not underrun the dma buffer as easily. As default, use 32k (order=3)
+rather than 64k as some of the games work more responsively.
+(2^N) * PAGE_SIZE = allocated buffer size
+
+MODULE_PARM(cs_debuglevel, "i");
+MODULE_PARM(cs_debugmask, "i");
+cs_debuglevel=N
+cs_debugmask=0xMMMMMMMM
+where N is a value from 0 (no debug printfs), to 9 (maximum)
+0xMMMMMMMM is a debug mask corresponding to the CS_xxx bits (see driver source).
+
+MODULE_PARM(hercules_egpio_disable, "i");
+hercules_egpio_disable=N
+where N is a 0 (enable egpio), or a 1 (disable egpio support)
+
+MODULE_PARM(initdelay, "i");
+initdelay=N
+This value is used to determine the millescond delay during the initialization
+code prior to powering up the PLL. On laptops this value can be used to
+assist with errors on resume, mostly with IBM laptops. Basically, if the
+system is booted under battery power then the mdelay()/udelay() functions fail to
+properly delay the required time. Also, if the system is booted under AC power
+and then the power removed, the mdelay()/udelay() functions will not delay properly.
+
+MODULE_PARM(powerdown, "i");
+powerdown=N
+where N is 0 (disable any powerdown of the internal blocks) or 1 (enable powerdown)
+
+
+MODULE_PARM(external_amp, "i");
+external_amp=1
+if N is set to 1, then force enabling the EAPD support in the primary AC97 codec.
+override the detection logic and force the external amp bit in the AC97 0x26 register
+to be reset (0). EAPD should be 0 for powerup, and 1 for powerdown. The VTB Santa Cruz
+card has inverted logic, so there is a special function for these cards.
+
+MODULE_PARM(thinkpad, "i");
+thinkpad=1
+if N is set to 1, then force enabling the clkrun functionality.
+Currently, when the part is being used, then clkrun is disabled for the entire system,
+but re-enabled when the driver is released or there is no outstanding open count.
+
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/es1370 linux-2.5.6/Documentation/sound/oss/es1370
--- linux-2.5.6-pre3/Documentation/sound/oss/es1370 Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/es1370 Thu Mar 7 18:24:41 2002
@@ -0,0 +1,70 @@
+/proc/sound, /dev/sndstat
+-------------------------
+
+/proc/sound and /dev/sndstat is not supported by the
+driver. To find out whether the driver succeeded loading,
+check the kernel log (dmesg).
+
+
+ALaw/uLaw sample formats
+------------------------
+
+This driver does not support the ALaw/uLaw sample formats.
+ALaw is the default mode when opening a sound device
+using OSS/Free. The reason for the lack of support is
+that the hardware does not support these formats, and adding
+conversion routines to the kernel would lead to very ugly
+code in the presence of the mmap interface to the driver.
+And since xquake uses mmap, mmap is considered important :-)
+and no sane application uses ALaw/uLaw these days anyway.
+In short, playing a Sun .au file as follows:
+
+cat my_file.au > /dev/dsp
+
+does not work. Instead, you may use the play script from
+Chris Bagwell's sox-12.14 package (available from the URL
+below) to play many different audio file formats.
+The script automatically determines the audio format
+and does do audio conversions if necessary.
+http://home.sprynet.com/sprynet/cbagwell/projects.html
+
+
+Blocking vs. nonblocking IO
+---------------------------
+
+Unlike OSS/Free this driver honours the O_NONBLOCK file flag
+not only during open, but also during read and write.
+This is an effort to make the sound driver interface more
+regular. Timidity has problems with this; a patch
+is available from http://www.ife.ee.ethz.ch/~sailer/linux/pciaudio.html.
+(Timidity patched will also run on OSS/Free).
+
+
+MIDI UART
+---------
+
+The driver supports a simple MIDI UART interface, with
+no ioctl's supported.
+
+
+MIDI synthesizer
+----------------
+
+This soundcard does not have any hardware MIDI synthesizer;
+MIDI synthesis has to be done in software. To allow this
+the driver/soundcard supports two PCM (/dev/dsp) interfaces.
+The second one goes to the mixer "synth" setting and supports
+only a limited set of sampling rates (44100, 22050, 11025, 5512).
+By setting lineout to 1 on the driver command line
+(eg. insmod es1370 lineout=1) it is even possible on some
+cards to convert the LINEIN jack into a second LINEOUT jack, thus
+making it possible to output four independent audio channels!
+
+There is a freely available software package that allows
+MIDI file playback on this soundcard called Timidity.
+See http://www.cgs.fi/~tt/timidity/.
+
+
+
+Thomas Sailer
+t.sailer@alumni.ethz.ch
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/es1371 linux-2.5.6/Documentation/sound/oss/es1371
--- linux-2.5.6-pre3/Documentation/sound/oss/es1371 Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/es1371 Thu Mar 7 18:24:41 2002
@@ -0,0 +1,64 @@
+/proc/sound, /dev/sndstat
+-------------------------
+
+/proc/sound and /dev/sndstat is not supported by the
+driver. To find out whether the driver succeeded loading,
+check the kernel log (dmesg).
+
+
+ALaw/uLaw sample formats
+------------------------
+
+This driver does not support the ALaw/uLaw sample formats.
+ALaw is the default mode when opening a sound device
+using OSS/Free. The reason for the lack of support is
+that the hardware does not support these formats, and adding
+conversion routines to the kernel would lead to very ugly
+code in the presence of the mmap interface to the driver.
+And since xquake uses mmap, mmap is considered important :-)
+and no sane application uses ALaw/uLaw these days anyway.
+In short, playing a Sun .au file as follows:
+
+cat my_file.au > /dev/dsp
+
+does not work. Instead, you may use the play script from
+Chris Bagwell's sox-12.14 package (available from the URL
+below) to play many different audio file formats.
+The script automatically determines the audio format
+and does do audio conversions if necessary.
+http://home.sprynet.com/sprynet/cbagwell/projects.html
+
+
+Blocking vs. nonblocking IO
+---------------------------
+
+Unlike OSS/Free this driver honours the O_NONBLOCK file flag
+not only during open, but also during read and write.
+This is an effort to make the sound driver interface more
+regular. Timidity has problems with this; a patch
+is available from http://www.ife.ee.ethz.ch/~sailer/linux/pciaudio.html.
+(Timidity patched will also run on OSS/Free).
+
+
+MIDI UART
+---------
+
+The driver supports a simple MIDI UART interface, with
+no ioctl's supported.
+
+
+MIDI synthesizer
+----------------
+
+This soundcard does not have any hardware MIDI synthesizer;
+MIDI synthesis has to be done in software. To allow this
+the driver/soundcard supports two PCM (/dev/dsp) interfaces.
+
+There is a freely available software package that allows
+MIDI file playback on this soundcard called Timidity.
+See http://www.cgs.fi/~tt/timidity/.
+
+
+
+Thomas Sailer
+t.sailer@alumni.ethz.ch
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/mwave linux-2.5.6/Documentation/sound/oss/mwave
--- linux-2.5.6-pre3/Documentation/sound/oss/mwave Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/mwave Thu Mar 7 18:24:41 2002
@@ -0,0 +1,185 @@
+ How to try to survive an IBM Mwave under Linux SB drivers
+
+
++ IBM have now released documentation of sorts and Torsten is busy
+ trying to make the Mwave work. This is not however a trivial task.
+
+----------------------------------------------------------------------------
+
+OK, first thing - the IRQ problem IS a problem, whether the test is bypassed or
+not. It is NOT a Linux problem, but an MWAVE problem that is fixed with the
+latest MWAVE patches. So, in other words, don't bypass the test for MWAVES!
+
+I have Windows 95 on /dev/hda1, swap on /dev/hda2, and Red Hat 5 on /dev/hda3.
+
+The steps, then:
+
+ Boot to Linux.
+ Mount Windows 95 file system (assume mount point = /dos95).
+ mkdir /dos95/linux
+ mkdir /dos95/linux/boot
+ mkdir /dos95/linux/boot/parms
+
+ Copy the kernel, any initrd image, and loadlin to /dos95/linux/boot/.
+
+ Reboot to Windows 95.
+
+ Edit C:/msdos.sys and add or change the following:
+
+ Logo=0
+ BootGUI=0
+
+ Note that msdos.sys is a text file but it needs to be made 'unhidden',
+ readable and writable before it can be edited. This can be done with
+ DOS' "attrib" command.
+
+ Edit config.sys to have multiple config menus. I have one for windows 95 and
+ five for Linux, like this:
+------------
+[menu]
+menuitem=W95, Windows 95
+menuitem=LINTP, Linux - ThinkPad
+menuitem=LINTP3, Linux - ThinkPad Console
+menuitem=LINDOC, Linux - Docked
+menuitem=LINDOC3, Linux - Docked Console
+menuitem=LIN1, Linux - Single User Mode
+REM menudefault=W95,10
+
+[W95]
+
+[LINTP]
+
+[LINDOC]
+
+[LINTP3]
+
+[LINDOC3]
+
+[LIN1]
+
+[COMMON]
+FILES=30
+REM Please read README.TXT in C:\MWW subdirectory before changing the DOS= statement.
+DOS=HIGH,UMB
+DEVICE=C:\MWW\MANAGER\MWD50430.EXE
+SHELL=c:\command.com /e:2048
+-------------------
+
+The important things are the SHELL and DEVICE statements.
+
+ Then change autoexec.bat. Basically everything in there originally should be
+ done ONLY when Windows 95 is booted. Then you add new things specifically
+ for Linux. Mine is as follows
+
+---------------
+@ECHO OFF
+if "%CONFIG%" == "W95" goto W95
+
+REM
+REM Linux stuff
+REM
+SET MWPATH=C:\MWW\DLL;C:\MWW\MWGAMES;C:\MWW\DSP
+SET BLASTER=A220 I5 D1
+SET MWROOT=C:\MWW
+SET LIBPATH=C:\MWW\DLL
+SET PATH=C:\WINDOWS;C:\MWW\DLL;
+CALL MWAVE START NOSHOW
+c:\linux\boot\loadlin.exe @c:\linux\boot\parms\%CONFIG%.par
+
+:W95
+REM
+REM Windows 95 stuff
+REM
+c:\toolkit\guard
+SET MSINPUT=C:\MSINPUT
+SET MWPATH=C:\MWW\DLL;C:\MWW\MWGAMES;C:\MWW\DSP
+REM The following is used by DOS games to recognize Sound Blaster hardware.
+REM If hardware settings are changed, please change this line as well.
+REM See the Mwave README file for instructions.
+SET BLASTER=A220 I5 D1
+SET MWROOT=C:\MWW
+SET LIBPATH=C:\MWW\DLL
+SET PATH=C:\WINDOWS;C:\WINDOWS\COMMAND;E:\ORAWIN95\BIN;f:\msdev\bin;e:\v30\bin.dbg;v:\devt\v30\bin;c:\JavaSDK\Bin;C:\MWW\DLL;
+SET INCLUDE=f:\MSDEV\INCLUDE;F:\MSDEV\MFC\INCLUDE
+SET LIB=F:\MSDEV\LIB;F:\MSDEV\MFC\LIB
+win
+
+------------------------
+
+Now build a file in c:\linux\boot\parms for each Linux config that you have.
+
+For example, my LINDOC3 config is for a docked Thinkpad at runlevel 3 with no
+initrd image, and has a parameter file named LINDOC3.PAR in c:\linux\boot\parms:
+
+-----------------------
+# LOADLIN @param_file image=other_image root=/dev/other
+#
+# Linux Console in docking station
+#
+c:\linux\boot\zImage.krn # First value must be filename of Linux kernel.
+root=/dev/hda3 # device which gets mounted as root FS
+ro # Other kernel arguments go here.
+apm=off
+doc=yes
+3
+-----------------------
+
+The doc=yes parameter is an environment variable used by my init scripts, not
+a kernel argument.
+
+However, the apm=off parameter IS a kernel argument! APM, at least in my setup,
+causes the kernel to crash when loaded via loadlin (but NOT when loaded via
+LILO). The APM stuff COULD be forced out of the kernel via the kernel compile
+options. Instead, I got an unofficial patch to the APM drivers that allows them
+to be dynamically deactivated via kernel arguments. Whatever you chose to
+document, APM, it seems, MUST be off for setups like mine.
+
+Now make sure C:\MWW\MWCONFIG.REF looks like this:
+
+----------------------
+[NativeDOS]
+Default=SB1.5
+SBInputSource=CD
+SYNTH=FM
+QSound=OFF
+Reverb=OFF
+Chorus=OFF
+ReverbDepth=5
+ChorusDepth=5
+SBInputVolume=5
+SBMainVolume=10
+SBWaveVolume=10
+SBSynthVolume=10
+WaveTableVolume=10
+AudioPowerDriver=ON
+
+[FastCFG]
+Show=No
+HideOption=Off
+-----------------------------
+
+OR the Default= line COULD be
+
+Default=SBPRO
+
+Reboot to Windows 95 and choose Linux. When booted, use sndconfig to configure
+the sound modules and voilà - ThinkPad sound with Linux.
+
+Now the gotchas - you can either have CD sound OR Mixers but not both. That's a
+problem with the SB1.5 (CD sound) or SBPRO (Mixers) settings. No one knows why
+this is!
+
+For some reason MPEG3 files, when played through mpg123, sound like they
+are playing at 1/8th speed - not very useful! If you have ANY insight
+on why this second thing might be happening, I would be grateful.
+
+===========================================================
+ _/ _/_/_/_/
+ _/_/ _/_/ _/
+ _/ _/_/ _/_/_/_/ Martin John Bartlett
+ _/ _/ _/ _/ (martin@nitram.demon.co.uk)
+_/ _/_/_/_/
+ _/
+_/ _/
+ _/_/
+===========================================================
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/solo1 linux-2.5.6/Documentation/sound/oss/solo1
--- linux-2.5.6-pre3/Documentation/sound/oss/solo1 Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/solo1 Thu Mar 7 18:24:41 2002
@@ -0,0 +1,70 @@
+Recording
+---------
+
+Recording does not work on the author's card, but there
+is at least one report of it working on later silicon.
+The chip behaves differently than described in the data sheet,
+likely due to a chip bug. Working around this would require
+the help of ESS (for example by publishing an errata sheet),
+but ESS has not done so so far.
+
+Also, the chip only supports 24 bit addresses for recording,
+which means it cannot work on some Alpha mainboards.
+
+
+/proc/sound, /dev/sndstat
+-------------------------
+
+/proc/sound and /dev/sndstat is not supported by the
+driver. To find out whether the driver succeeded loading,
+check the kernel log (dmesg).
+
+
+ALaw/uLaw sample formats
+------------------------
+
+This driver does not support the ALaw/uLaw sample formats.
+ALaw is the default mode when opening a sound device
+using OSS/Free. The reason for the lack of support is
+that the hardware does not support these formats, and adding
+conversion routines to the kernel would lead to very ugly
+code in the presence of the mmap interface to the driver.
+And since xquake uses mmap, mmap is considered important :-)
+and no sane application uses ALaw/uLaw these days anyway.
+In short, playing a Sun .au file as follows:
+
+cat my_file.au > /dev/dsp
+
+does not work. Instead, you may use the play script from
+Chris Bagwell's sox-12.14 package (or later, available from the URL
+below) to play many different audio file formats.
+The script automatically determines the audio format
+and does do audio conversions if necessary.
+http://home.sprynet.com/sprynet/cbagwell/projects.html
+
+
+Blocking vs. nonblocking IO
+---------------------------
+
+Unlike OSS/Free this driver honours the O_NONBLOCK file flag
+not only during open, but also during read and write.
+This is an effort to make the sound driver interface more
+regular. Timidity has problems with this; a patch
+is available from http://www.ife.ee.ethz.ch/~sailer/linux/pciaudio.html.
+(Timidity patched will also run on OSS/Free).
+
+
+MIDI UART
+---------
+
+The driver supports a simple MIDI UART interface, with
+no ioctl's supported.
+
+
+MIDI synthesizer
+----------------
+
+The card has an OPL compatible FM synthesizer.
+
+Thomas Sailer
+t.sailer@alumni.ethz.ch
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/sonicvibes linux-2.5.6/Documentation/sound/oss/sonicvibes
--- linux-2.5.6-pre3/Documentation/sound/oss/sonicvibes Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/sonicvibes Thu Mar 7 18:24:41 2002
@@ -0,0 +1,81 @@
+/proc/sound, /dev/sndstat
+-------------------------
+
+/proc/sound and /dev/sndstat is not supported by the
+driver. To find out whether the driver succeeded loading,
+check the kernel log (dmesg).
+
+
+ALaw/uLaw sample formats
+------------------------
+
+This driver does not support the ALaw/uLaw sample formats.
+ALaw is the default mode when opening a sound device
+using OSS/Free. The reason for the lack of support is
+that the hardware does not support these formats, and adding
+conversion routines to the kernel would lead to very ugly
+code in the presence of the mmap interface to the driver.
+And since xquake uses mmap, mmap is considered important :-)
+and no sane application uses ALaw/uLaw these days anyway.
+In short, playing a Sun .au file as follows:
+
+cat my_file.au > /dev/dsp
+
+does not work. Instead, you may use the play script from
+Chris Bagwell's sox-12.14 package (available from the URL
+below) to play many different audio file formats.
+The script automatically determines the audio format
+and does do audio conversions if necessary.
+http://home.sprynet.com/sprynet/cbagwell/projects.html
+
+
+Blocking vs. nonblocking IO
+---------------------------
+
+Unlike OSS/Free this driver honours the O_NONBLOCK file flag
+not only during open, but also during read and write.
+This is an effort to make the sound driver interface more
+regular. Timidity has problems with this; a patch
+is available from http://www.ife.ee.ethz.ch/~sailer/linux/pciaudio.html.
+(Timidity patched will also run on OSS/Free).
+
+
+MIDI UART
+---------
+
+The driver supports a simple MIDI UART interface, with
+no ioctl's supported.
+
+
+MIDI synthesizer
+----------------
+
+The card both has an OPL compatible FM synthesizer as well as
+a wavetable synthesizer.
+
+I haven't managed so far to get the OPL synth running.
+
+Using the wavetable synthesizer requires allocating
+1-4MB of physically contiguous memory, which isn't possible
+currently on Linux without ugly hacks like the bigphysarea
+patch. Therefore, the driver doesn't support wavetable
+synthesis.
+
+
+No support from S3
+------------------
+
+I do not get any support from S3. Therefore, the driver
+still has many problems. For example, although the manual
+states that the chip should be able to access the sample
+buffer anywhere in 32bit address space, I haven't managed to
+get it working with buffers above 16M. Therefore, the card
+has the same disadvantages as ISA soundcards.
+
+Given that the card is also very noisy, and if you haven't
+already bought it, you should strongly opt for one of the
+comparatively priced Ensoniq products.
+
+
+Thomas Sailer
+t.sailer@alumni.ethz.ch
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/ultrasound linux-2.5.6/Documentation/sound/oss/ultrasound
--- linux-2.5.6-pre3/Documentation/sound/oss/ultrasound Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/ultrasound Thu Mar 7 18:24:41 2002
@@ -0,0 +1,30 @@
+modprobe sound
+insmod ad1848
+insmod gus io=* irq=* dma=* ...
+
+This loads the driver for the Gravis Ultrasound family of sound cards.
+
+The gus module takes the following arguments
+
+io I/O address of the Ultrasound card (eg. io=0x220)
+irq IRQ of the Sound Blaster card
+dma DMA channel for the Sound Blaster
+dma16 2nd DMA channel, only needed for full duplex operation
+type 1 for PnP card
+gus16 1 for using 16 bit sampling daughter board
+no_wave_dma Set to disable DMA usage for wavetable (see note)
+db16 ???
+
+
+no_wave_dma option
+
+This option defaults to a value of 0, which allows the Ultrasound wavetable
+DSP to use DMA for for playback and downloading samples. This is the same
+as the old behaviour. If set to 1, no DMA is needed for downloading samples,
+and allows owners of a GUS MAX to make use of simultaneous digital audio
+(/dev/dsp), MIDI, and wavetable playback.
+
+
+If you have problems in recording with GUS MAX, you could try to use
+just one 8 bit DMA channel. Recording will not work with one DMA
+channel if it's a 16 bit one.
diff -urN linux-2.5.6-pre3/Documentation/sound/oss/vwsnd linux-2.5.6/Documentation/sound/oss/vwsnd
--- linux-2.5.6-pre3/Documentation/sound/oss/vwsnd Wed Dec 31 16:00:00 1969
+++ linux-2.5.6/Documentation/sound/oss/vwsnd Thu Mar 7 18:24:41 2002
@@ -0,0 +1,293 @@
+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, 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(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);
+ }
+
diff -urN linux-2.5.6-pre3/Documentation/sound/solo1 linux-2.5.6/Documentation/sound/solo1
--- linux-2.5.6-pre3/Documentation/sound/solo1 Tue Feb 19 18:11:05 2002
+++ linux-2.5.6/Documentation/sound/solo1 Wed Dec 31 16:00:00 1969
@@ -1,70 +0,0 @@
-Recording
----------
-
-Recording does not work on the author's card, but there
-is at least one report of it working on later silicon.
-The chip behaves differently than described in the data sheet,
-likely due to a chip bug. Working around this would require
-the help of ESS (for example by publishing an errata sheet),
-but ESS has not done so so far.
-
-Also, the chip only supports 24 bit addresses for recording,
-which means it cannot work on some Alpha mainboards.
-
-
-/proc/sound, /dev/sndstat
--------------------------
-
-/proc/sound and /dev/sndstat is not supported by the
-driver. To find out whether the driver succeeded loading,
-check the kernel log (dmesg).
-
-
-ALaw/uLaw sample formats
-------------------------
-
-This driver does not support the ALaw/uLaw sample formats.
-ALaw is the default mode when opening a sound device
-using OSS/Free. The reason for the lack of support is
-that the hardware does not support these formats, and adding
-conversion routines to the kernel would lead to very ugly
-code in the presence of the mmap interface to the driver.
-And since xquake uses mmap, mmap is considered important :-)
-and no sane application uses ALaw/uLaw these days anyway.
-In short, playing a Sun .au file as follows:
-
-cat my_file.au > /dev/dsp
-
-does not work. Instead, you may use the play script from
-Chris Bagwell's sox-12.14 package (or later, available from the URL
-below) to play many different audio file formats.
-The script automatically determines the audio format
-and does do audio conversions if necessary.
-http://home.sprynet.com/sprynet/cbagwell/projects.html
-
-
-Blocking vs. nonblocking IO
----------------------------
-
-Unlike OSS/Free this driver honours the O_NONBLOCK file flag
-not only during open, but also during read and write.
-This is an effort to make the sound driver interface more
-regular. Timidity has problems with this; a patch
-is available from http://www.ife.ee.ethz.ch/~sailer/linux/pciaudio.html.
-(Timidity patched will also run on OSS/Free).
-
-
-MIDI UART
----------
-
-The driver supports a simple MIDI UART interface, with
-no ioctl's supported.
-
-
-MIDI synthesizer
-----------------
-
-The card has an OPL compatible FM synthesizer.
-
-Thomas Sailer
-t.sailer@alumni.ethz.ch
diff -urN linux-2.5.6-pre3/Documentation/sound/sonicvibes linux-2.5.6/Documentation/sound/sonicvibes
--- linux-2.5.6-pre3/Documentation/sound/sonicvibes Tue Feb 19 18:10:59 2002
+++ linux-2.5.6/Documentation/sound/sonicvibes Wed Dec 31 16:00:00 1969
@@ -1,81 +0,0 @@
-/proc/sound, /dev/sndstat
--------------------------
-
-/proc/sound and /dev/sndstat is not supported by the
-driver. To find out whether the driver succeeded loading,
-check the kernel log (dmesg).
-
-
-ALaw/uLaw sample formats
-------------------------
-
-This driver does not support the ALaw/uLaw sample formats.
-ALaw is the default mode when opening a sound device
-using OSS/Free. The reason for the lack of support is
-that the hardware does not support these formats, and adding
-conversion routines to the kernel would lead to very ugly
-code in the presence of the mmap interface to the driver.
-And since xquake uses mmap, mmap is considered important :-)
-and no sane application uses ALaw/uLaw these days anyway.
-In short, playing a Sun .au file as follows:
-
-cat my_file.au > /dev/dsp
-
-does not work. Instead, you may use the play script from
-Chris Bagwell's sox-12.14 package (available from the URL
-below) to play many different audio file formats.
-The script automatically determines the audio format
-and does do audio conversions if necessary.
-http://home.sprynet.com/sprynet/cbagwell/projects.html
-
-
-Blocking vs. nonblocking IO
----------------------------
-
-Unlike OSS/Free this driver honours the O_NONBLOCK file flag
-not only during open, but also during read and write.
-This is an effort to make the sound driver interface more
-regular. Timidity has problems with this; a patch
-is available from http://www.ife.ee.ethz.ch/~sailer/linux/pciaudio.html.
-(Timidity patched will also run on OSS/Free).
-
-
-MIDI UART
----------
-
-The driver supports a simple MIDI UART interface, with
-no ioctl's supported.
-
-
-MIDI synthesizer
-----------------
-
-The card both has an OPL compatible FM synthesizer as well as
-a wavetable synthesizer.
-
-I haven't managed so far to get the OPL synth running.
-
-Using the wavetable synthesizer requires allocating
-1-4MB of physically contiguous memory, which isn't possible
-currently on Linux without ugly hacks like the bigphysarea
-patch. Therefore, the driver doesn't support wavetable
-synthesis.
-
-
-No support from S3
-------------------
-
-I do not get any support from S3. Therefore, the driver
-still has many problems. For example, although the manual
-states that the chip should be able to access the sample
-buffer anywhere in 32bit address space, I haven't managed to
-get it working with buffers above 16M. Therefore, the card
-has the same disadvantages as ISA soundcards.
-
-Given that the card is also very noisy, and if you haven't
-already bought it, you should strongly opt for one of the
-comparatively priced Ensoniq products.
-
-
-Thomas Sailer
-t.sailer@alumni.ethz.ch
diff -urN linux-2.5.6-pre3/Documentation/sound/ultrasound linux-2.5.6/Documentation/sound/ultrasound
--- linux-2.5.6-pre3/Documentation/sound/ultrasound Tue Feb 19 18:10:54 2002
+++ linux-2.5.6/Documentation/sound/ultrasound Wed Dec 31 16:00:00 1969
@@ -1,30 +0,0 @@
-modprobe sound
-insmod ad1848
-insmod gus io=* irq=* dma=* ...
-
-This loads the driver for the Gravis Ultrasound family of sound cards.
-
-The gus module takes the following arguments
-
-io I/O address of the Ultrasound card (eg. io=0x220)
-irq IRQ of the Sound Blaster card
-dma DMA channel for the Sound Blaster
-dma16 2nd DMA channel, only needed for full duplex operation
-type 1 for PnP card
-gus16 1 for using 16 bit sampling daughter board
-no_wave_dma Set to disable DMA usage for wavetable (see note)
-db16 ???
-
-
-no_wave_dma option
-
-This option defaults to a value of 0, which allows the Ultrasound wavetable
-DSP to use DMA for for playback and downloading samples. This is the same
-as the old behaviour. If set to 1, no DMA is needed for downloading samples,
-and allows owners of a GUS MAX to make use of simultaneous digital audio
-(/dev/dsp), MIDI, and wavetable playback.
-
-
-If you have problems in recording with GUS MAX, you could try to use
-just one 8 bit DMA channel. Recording will not work with one DMA
-channel if it's a 16 bit one.
diff -urN linux-2.5.6-pre3/Documentation/sound/vwsnd linux-2.5.6/Documentation/sound/vwsnd
--- linux-2.5.6-pre3/Documentation/sound/vwsnd Tue Feb 19 18:10:56 2002
+++ linux-2.5.6/Documentation/sound/vwsnd Wed Dec 31 16:00:00 1969
@@ -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 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, 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(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);
- }
-
diff -urN linux-2.5.6-pre3/MAINTAINERS linux-2.5.6/MAINTAINERS
--- linux-2.5.6-pre3/MAINTAINERS Thu Mar 7 18:24:24 2002
+++ linux-2.5.6/MAINTAINERS Thu Mar 7 18:24:42 2002
@@ -1722,12 +1722,12 @@
S: Maintained
USB SERIAL KEYSPAN DRIVER
-P: Hugh Blemings
-M: hugh@misc.nu
+P: Greg Kroah-Hartman
+M: greg@kroah.com
L: linux-usb-users@lists.sourceforge.net
L: linux-usb-devel@lists.sourceforge.net
+W: http://www.kroah.com/linux/
S: Maintained
-W: http://misc.nu/hugh/keyspan/
USB SUBSYSTEM
P: Greg Kroah-Hartman
diff -urN linux-2.5.6-pre3/Makefile linux-2.5.6/Makefile
--- linux-2.5.6-pre3/Makefile Thu Mar 7 18:24:24 2002
+++ linux-2.5.6/Makefile Thu Mar 7 18:24:42 2002
@@ -1,7 +1,7 @@
VERSION = 2
PATCHLEVEL = 5
SUBLEVEL = 6
-EXTRAVERSION =-pre3
+EXTRAVERSION =
KERNELRELEASE=$(VERSION).$(PATCHLEVEL).$(SUBLEVEL)$(EXTRAVERSION)
@@ -162,6 +162,7 @@
DRIVERS-$(CONFIG_PCMCIA) += drivers/pcmcia/pcmcia.o
DRIVERS-$(CONFIG_NET_PCMCIA) += drivers/net/pcmcia/pcmcia_net.o
DRIVERS-$(CONFIG_NET_WIRELESS) += drivers/net/wireless/wireless_net.o
+DRIVERS-$(CONFIG_NET_TULIP) += drivers/net/tulip/tulip_net.o
DRIVERS-$(CONFIG_PCMCIA_CHRDEV) += drivers/char/pcmcia/pcmcia_char.o
DRIVERS-$(CONFIG_DIO) += drivers/dio/dio.a
DRIVERS-$(CONFIG_SBUS) += drivers/sbus/sbus_all.o
diff -urN linux-2.5.6-pre3/arch/alpha/Makefile linux-2.5.6/arch/alpha/Makefile
--- linux-2.5.6-pre3/arch/alpha/Makefile Thu Mar 7 18:24:24 2002
+++ linux-2.5.6/arch/alpha/Makefile Thu Mar 7 18:24:42 2002
@@ -45,22 +45,22 @@
CFLAGS := $(CFLAGS) -mcpu=pca56
mcpu_done := y
endif
- ifeq ($(mcpu_done)$(CONFIG_ALPHA_PYXIS),ny)
- CFLAGS := $(CFLAGS) -mcpu=ev56
- mcpu_done := y
- endif
- ifeq ($(mcpu_done)$(CONFIG_ALPHA_POLARIS),ny)
- ifeq ($(have_mcpu_pca56),y)
- CFLAGS := $(CFLAGS) -mcpu=pca56
- else
- CFLAGS := $(CFLAGS) -mcpu=ev56
- endif
+ ifeq ($(mcpu_done)$(CONFIG_ALPHA_POLARIS)$(have_mcpu_pca56),nyy)
+ CFLAGS := $(CFLAGS) -mcpu=pca56
mcpu_done := y
endif
ifeq ($(mcpu_done)$(CONFIG_ALPHA_EV4),ny)
CFLAGS := $(CFLAGS) -mcpu=ev4
mcpu_done := y
endif
+ ifeq ($(mcpu_done)$(CONFIG_ALPHA_EV56),ny)
+ CFLAGS := $(CFLAGS) -mcpu=ev56
+ mcpu_done := y
+ endif
+ ifeq ($(mcpu_done)$(CONFIG_ALPHA_EV5),ny)
+ CFLAGS := $(CFLAGS) -mcpu=ev5
+ mcpu_done := y
+ endif
ifeq ($(mcpu_done)$(CONFIG_ALPHA_EV67)$(have_mcpu_ev67),nyy)
CFLAGS := $(CFLAGS) -mcpu=ev67
mcpu_done := y
diff -urN linux-2.5.6-pre3/arch/alpha/config.in linux-2.5.6/arch/alpha/config.in
--- linux-2.5.6-pre3/arch/alpha/config.in Tue Feb 19 18:10:57 2002
+++ linux-2.5.6/arch/alpha/config.in Thu Mar 7 18:24:42 2002
@@ -88,19 +88,43 @@
then
define_bool CONFIG_ALPHA_EB64P y
fi
-if [ "$CONFIG_ALPHA_EB164" = "y" -o "$CONFIG_ALPHA_PC164" = "y" \
- -o "$CONFIG_ALPHA_ALCOR" = "y" -o "$CONFIG_ALPHA_TAKARA" = "y" ]
+if [ "$CONFIG_ALPHA_ALCOR" = "y" ]
then
define_bool CONFIG_ALPHA_EV5 y
define_bool CONFIG_ALPHA_CIA y
+ bool 'EV56 CPU (speed >= 366MHz)?' CONFIG_ALPHA_EV56
fi
-if [ "$CONFIG_ALPHA_MIKASA" = "y" -o "$CONFIG_ALPHA_NORITAKE" = "y" ]
+if [ "$CONFIG_ALPHA_EB164" = "y" ]
+then
+ define_bool CONFIG_ALPHA_EV5 y
+ define_bool CONFIG_ALPHA_CIA y
+fi
+if [ "$CONFIG_ALPHA_PC164" = "y" -o "$CONFIG_ALPHA_TAKARA" = "y" ]
+then
+ define_bool CONFIG_ALPHA_EV5 y
+ define_bool CONFIG_ALPHA_EV56 y
+ define_bool CONFIG_ALPHA_CIA y
+fi
+if [ "$CONFIG_ALPHA_MIKASA" = "y" ]
+then
+ bool 'EV5 CPU daughtercard (model 5/xxx)?' CONFIG_ALPHA_PRIMO
+ if [ "$CONFIG_ALPHA_PRIMO" = "y" ]
+ then
+ define_bool CONFIG_ALPHA_EV5 y
+ define_bool CONFIG_ALPHA_CIA y
+ else
+ define_bool CONFIG_ALPHA_EV4 y
+ define_bool CONFIG_ALPHA_APECS y
+ fi
+fi
+if [ "$CONFIG_ALPHA_NORITAKE" = "y" ]
then
bool 'EV5 CPU daughtercard (model 5/xxx)?' CONFIG_ALPHA_PRIMO
if [ "$CONFIG_ALPHA_PRIMO" = "y" ]
then
define_bool CONFIG_ALPHA_EV5 y
define_bool CONFIG_ALPHA_CIA y
+ bool 'EV56 CPU (speed >= 333MHz)?' CONFIG_ALPHA_EV56
else
define_bool CONFIG_ALPHA_EV4 y
define_bool CONFIG_ALPHA_APECS y
@@ -121,6 +145,7 @@
-o "$CONFIG_ALPHA_SX164" = "y" -o "$CONFIG_ALPHA_RUFFIAN" = "y" ]
then
define_bool CONFIG_ALPHA_EV5 y
+ define_bool CONFIG_ALPHA_EV56 y
define_bool CONFIG_ALPHA_CIA y
define_bool CONFIG_ALPHA_PYXIS y
fi
@@ -145,10 +170,12 @@
then
define_bool CONFIG_ALPHA_EV5 y
define_bool CONFIG_ALPHA_MCPCIA y
+ bool 'EV56 CPU (speed >= 400MHz)?' CONFIG_ALPHA_EV56
fi
if [ "$CONFIG_ALPHA_RX164" = "y" ]
then
define_bool CONFIG_ALPHA_EV5 y
+ define_bool CONFIG_ALPHA_EV56 y
define_bool CONFIG_ALPHA_POLARIS y
fi
if [ "$CONFIG_ALPHA_JENSEN" = "y" ]
diff -urN linux-2.5.6-pre3/arch/alpha/kernel/core_irongate.c linux-2.5.6/arch/alpha/kernel/core_irongate.c
--- linux-2.5.6-pre3/arch/alpha/kernel/core_irongate.c Tue Feb 19 18:10:56 2002
+++ linux-2.5.6/arch/alpha/kernel/core_irongate.c Thu Mar 7 18:24:42 2002
@@ -455,7 +455,7 @@
if (address >= end)
BUG();
do {
- pte_t * pte = pte_alloc(&init_mm, pmd, address);
+ pte_t * pte = pte_alloc_kernel(&init_mm, pmd, address);
if (!pte)
return -ENOMEM;
irongate_remap_area_pte(pte, address, end - address,
diff -urN linux-2.5.6-pre3/arch/alpha/kernel/process.c linux-2.5.6/arch/alpha/kernel/process.c
--- linux-2.5.6-pre3/arch/alpha/kernel/process.c Thu Mar 7 18:24:24 2002
+++ linux-2.5.6/arch/alpha/kernel/process.c Thu Mar 7 18:24:42 2002
@@ -64,7 +64,6 @@
while (!need_resched())
barrier();
schedule();
- check_pgt_cache();
}
}
diff -urN linux-2.5.6-pre3/arch/alpha/mm/init.c linux-2.5.6/arch/alpha/mm/init.c
--- linux-2.5.6-pre3/arch/alpha/mm/init.c Tue Feb 19 18:11:00 2002
+++ linux-2.5.6/arch/alpha/mm/init.c Thu Mar 7 18:24:42 2002
@@ -42,12 +42,8 @@
static struct pcb_struct original_pcb;
-#ifndef CONFIG_SMP
-struct pgtable_cache_struct quicklists;
-#endif
-
pgd_t *
-get_pgd_slow(void)
+pgd_alloc(struct mm_struct *mm)
{
pgd_t *ret, *init;
@@ -69,28 +65,26 @@
return ret;
}
-int do_check_pgt_cache(int low, int high)
+pte_t *
+pte_alloc_one_kernel(struct mm_struct *mm, unsigned long address)
{
- int freed = 0;
- if(pgtable_cache_size > high) {
- do {
- if(pgd_quicklist) {
- free_pgd_slow(get_pgd_fast());
- freed++;
- }
- if(pmd_quicklist) {
- pmd_free_slow(pmd_alloc_one_fast(NULL, 0));
- freed++;
- }
- if(pte_quicklist) {
- pte_free_slow(pte_alloc_one_fast(NULL, 0));
- freed++;
- }
- } while(pgtable_cache_size > low);
+ pte_t *pte;
+ long timeout = 10;
+
+ retry:
+ pte = (pte_t *) __get_free_page(GFP_KERNEL);
+ if (pte)
+ clear_page(pte);
+ else if (--timeout >= 0) {
+ current->state = TASK_UNINTERRUPTIBLE;
+ schedule_timeout(HZ);
+ goto retry;
}
- return freed;
+
+ return pte;
}
+
/*
* BAD_PAGE is the page that is used for page faults when linux
* is out-of-memory. Older versions of linux just did a
@@ -145,7 +139,6 @@
printk("%ld reserved pages\n",reserved);
printk("%ld pages shared\n",shared);
printk("%ld pages swap cached\n",cached);
- printk("%ld pages in page table cache\n",pgtable_cache_size);
show_buffers();
}
#endif
@@ -260,7 +253,7 @@
unsigned long paddr = crb->map[i].pa;
crb->map[i].va = vaddr;
for (j = 0; j < crb->map[i].count; ++j) {
- set_pte(pte_offset(pmd, vaddr),
+ set_pte(pte_offset_kernel(pmd, vaddr),
mk_pte_phys(paddr, PAGE_KERNEL));
paddr += PAGE_SIZE;
vaddr += PAGE_SIZE;
diff -urN linux-2.5.6-pre3/arch/i386/defconfig linux-2.5.6/arch/i386/defconfig
--- linux-2.5.6-pre3/arch/i386/defconfig Thu Mar 7 18:24:24 2002
+++ linux-2.5.6/arch/i386/defconfig Thu Mar 7 18:24:42 2002
@@ -435,12 +435,9 @@
# CONFIG_AC3200 is not set
# CONFIG_APRICOT is not set
# CONFIG_CS89x0 is not set
-# CONFIG_DE2104X is not set
-# CONFIG_TULIP is not set
-# CONFIG_DE4X5 is not set
# CONFIG_DGRS is not set
-# CONFIG_DM9102 is not set
-CONFIG_EEPRO100=y
+# CONFIG_EEPRO100 is not set
+CONFIG_E100=y
# CONFIG_LNE390 is not set
# CONFIG_FEALNX is not set
# CONFIG_NATSEMI is not set
@@ -459,7 +456,6 @@
# CONFIG_TLAN is not set
# CONFIG_VIA_RHINE is not set
# CONFIG_VIA_RHINE_MMIO is not set
-# CONFIG_WINBOND_840 is not set
# CONFIG_NET_POCKET is not set
#
@@ -473,6 +469,7 @@
# CONFIG_HAMACHI is not set
# CONFIG_YELLOWFIN is not set
# CONFIG_SK98LIN is not set
+# CONFIG_TIGON3 is not set
# CONFIG_FDDI is not set
# CONFIG_PLIP is not set
# CONFIG_PPP is not set
@@ -495,6 +492,11 @@
# CONFIG_WAN is not set
#
+# "Tulip" family network device support
+#
+# CONFIG_NET_TULIP is not set
+
+#
# PCMCIA network device support
#
CONFIG_NET_PCMCIA=y
@@ -508,8 +510,6 @@
# CONFIG_PCMCIA_AXNET is not set
# CONFIG_ARCNET_COM20020_CS is not set
# CONFIG_PCMCIA_IBMTR is not set
-# CONFIG_PCMCIA_XIRCOM is not set
-# CONFIG_PCMCIA_XIRTULIP is not set
CONFIG_NET_PCMCIA_RADIO=y
CONFIG_PCMCIA_RAYCS=y
# CONFIG_AIRONET4500_CS is not set
diff -urN linux-2.5.6-pre3/arch/i386/kernel/Makefile linux-2.5.6/arch/i386/kernel/Makefile
--- linux-2.5.6-pre3/arch/i386/kernel/Makefile Tue Feb 19 18:10:57 2002
+++ linux-2.5.6/arch/i386/kernel/Makefile Thu Mar 7 18:24:42 2002
@@ -14,7 +14,7 @@
O_TARGET := kernel.o
-export-objs := mca.o mtrr.o msr.o cpuid.o microcode.o i386_ksyms.o
+export-objs := mca.o mtrr.o msr.o cpuid.o microcode.o i386_ksyms.o time.o
obj-y := process.o semaphore.o signal.o entry.o traps.o irq.o vm86.o \
ptrace.o i8259.o ioport.o ldt.o setup.o time.o sys_i386.o \
diff -urN linux-2.5.6-pre3/arch/i386/kernel/apm.c linux-2.5.6/arch/i386/kernel/apm.c
--- linux-2.5.6-pre3/arch/i386/kernel/apm.c Tue Feb 19 18:10:54 2002
+++ linux-2.5.6/arch/i386/kernel/apm.c Thu Mar 7 18:24:42 2002
@@ -812,7 +812,7 @@
t1 = IDLE_LEAKY_MAX;
- while (need_resched()) {
+ while (!need_resched()) {
if (use_apm_idle) {
unsigned int t;
diff -urN linux-2.5.6-pre3/arch/i386/kernel/mpparse.c linux-2.5.6/arch/i386/kernel/mpparse.c
--- linux-2.5.6-pre3/arch/i386/kernel/mpparse.c Tue Feb 19 18:10:59 2002
+++ linux-2.5.6/arch/i386/kernel/mpparse.c Thu Mar 7 18:24:42 2002
@@ -37,6 +37,8 @@
int apic_version [MAX_APICS];
int mp_bus_id_to_type [MAX_MP_BUSSES];
int mp_bus_id_to_node [MAX_MP_BUSSES];
+int mp_bus_id_to_local [MAX_MP_BUSSES];
+int quad_local_to_mp_bus_id [NR_CPUS/4][4];
int mp_bus_id_to_pci_bus [MAX_MP_BUSSES] = { [0 ... MAX_MP_BUSSES-1] = -1 };
int mp_current_pci_id;
@@ -241,13 +243,17 @@
static void __init MP_bus_info (struct mpc_config_bus *m)
{
char str[7];
+ int quad;
memcpy(str, m->mpc_bustype, 6);
str[6] = 0;
if (clustered_apic_mode) {
- mp_bus_id_to_node[m->mpc_busid] = translation_table[mpc_record]->trans_quad;
- printk("Bus #%d is %s (node %d)\n", m->mpc_busid, str, mp_bus_id_to_node[m->mpc_busid]);
+ quad = translation_table[mpc_record]->trans_quad;
+ mp_bus_id_to_node[m->mpc_busid] = quad;
+ mp_bus_id_to_local[m->mpc_busid] = translation_table[mpc_record]->trans_local;
+ quad_local_to_mp_bus_id[quad][translation_table[mpc_record]->trans_local] = m->mpc_busid;
+ printk("Bus #%d is %s (node %d)\n", m->mpc_busid, str, quad);
} else {
Dprintk("Bus #%d is %s\n", m->mpc_busid, str);
}
@@ -324,13 +330,14 @@
static void __init MP_translation_info (struct mpc_config_translation *m)
{
- printk("Translation: record %d, type %d, quad %d, global %d, local %d\n", mpc_record, m->trans_type,
- m->trans_quad, m->trans_global, m->trans_local);
+ printk("Translation: record %d, type %d, quad %d, global %d, local %d\n", mpc_record, m->trans_type, m->trans_quad, m->trans_global, m->trans_local);
if (mpc_record >= MAX_MPC_ENTRY)
printk("MAX_MPC_ENTRY exceeded!\n");
else
translation_table[mpc_record] = m; /* stash this for later */
+ if (m->trans_quad+1 > numnodes)
+ numnodes = m->trans_quad+1;
}
/*
@@ -495,10 +502,6 @@
}
++mpc_record;
}
- if (clustered_apic_mode && nr_ioapics > 2) {
- /* don't initialise IO apics on secondary quads */
- nr_ioapics = 2;
- }
if (!num_processors)
printk(KERN_ERR "SMP mptable: no processors registered!\n");
return num_processors;
diff -urN linux-2.5.6-pre3/arch/i386/kernel/pci-pc.c linux-2.5.6/arch/i386/kernel/pci-pc.c
--- linux-2.5.6-pre3/arch/i386/kernel/pci-pc.c Tue Feb 19 18:11:01 2002
+++ linux-2.5.6/arch/i386/kernel/pci-pc.c Thu Mar 7 18:24:42 2002
@@ -14,6 +14,7 @@
#include