diff options
| author | Linus Torvalds <torvalds@ppc970.osdl.org> | 2005-04-16 15:20:36 -0700 |
|---|---|---|
| committer | Linus Torvalds <torvalds@ppc970.osdl.org> | 2005-04-16 15:20:36 -0700 |
| commit | 1da177e4c3f41524e886b7f1b8a0c1fc7321cac2 (patch) | |
| tree | 0bba044c4ce775e45a88a51686b5d9f90697ea9d /Documentation/sound/oss | |
| download | talos-obmc-linux-1da177e4c3f41524e886b7f1b8a0c1fc7321cac2.tar.gz talos-obmc-linux-1da177e4c3f41524e886b7f1b8a0c1fc7321cac2.zip | |
Linux-2.6.12-rc2v2.6.12-rc2
Initial git repository build. I'm not bothering with the full history,
even though we have it. We can create a separate "historical" git
archive of that later if we want to, and in the meantime it's about
3.2GB when imported into git - space that would just make the early
git days unnecessarily complicated, when we don't have a lot of good
infrastructure for it.
Let it rip!
Diffstat (limited to 'Documentation/sound/oss')
45 files changed, 8275 insertions, 0 deletions
diff --git a/Documentation/sound/oss/AD1816 b/Documentation/sound/oss/AD1816 new file mode 100644 index 000000000000..14bd8f25d523 --- /dev/null +++ b/Documentation/sound/oss/AD1816 @@ -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 <tek@rbg.informatik.tu-darmstadt.de> +Christoph Hellwig <hch@infradead.org> + Last modified: 2000/09/20 diff --git a/Documentation/sound/oss/ALS b/Documentation/sound/oss/ALS new file mode 100644 index 000000000000..d01ffbfd5808 --- /dev/null +++ b/Documentation/sound/oss/ALS @@ -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 --git a/Documentation/sound/oss/AWE32 b/Documentation/sound/oss/AWE32 new file mode 100644 index 000000000000..cb179bfeb522 --- /dev/null +++ b/Documentation/sound/oss/AWE32 @@ -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/modprobe.conf, and insert the following lines at the end of the + file: + + alias sound-slot-0 sb + alias sound-service-0-1 awe_wave + install awe_wave /sbin/modprobe --first-time -i 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 --git a/Documentation/sound/oss/AudioExcelDSP16 b/Documentation/sound/oss/AudioExcelDSP16 new file mode 100644 index 000000000000..c0f08922993b --- /dev/null +++ b/Documentation/sound/oss/AudioExcelDSP16 @@ -0,0 +1,101 @@ +Driver +------ + +Informations about Audio Excel DSP 16 driver can be found in the source +file 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/modprobe.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 modprobe.conf +file: + +install ad1848 /sbin/modprobe aedsp16 && /sbin/modprobe -i ad1848 +install opl3 /sbin/modprobe aedsp16 && /sbin/modprobe -i opl3 + +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 +<ctrl>-<alt>-<del> 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) <ctrl>-<alt>-<del> 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 <sjg95@unixfe.rl.ac.uk> +> 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 --git a/Documentation/sound/oss/CMI8330 b/Documentation/sound/oss/CMI8330 new file mode 100644 index 000000000000..9c439f1a6dba --- /dev/null +++ b/Documentation/sound/oss/CMI8330 @@ -0,0 +1,153 @@ +Documentation for CMI 8330 (SoundPRO) +------------------------------------- +Alessandro Zummo <azummo@ita.flashnet.it> + +( Be sure to read Documentation/sound/oss/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 <azummo@ita.flashnet.it> +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 <Stefan.Laudat@asit.ro> + +[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: + +<snip below> + + +(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) + +<end of snip> + + 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 <elysian@ethereal.torsion.org> suggests the following /etc/modprobe.conf: + +alias sound ad1848 +alias synth0 opl3 +options ad1848 io=0x530 irq=7 dma=0 soundpro=1 +options opl3 io=0x388 + + diff --git a/Documentation/sound/oss/CMI8338 b/Documentation/sound/oss/CMI8338 new file mode 100644 index 000000000000..387d058c3f95 --- /dev/null +++ b/Documentation/sound/oss/CMI8338 @@ -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 --git a/Documentation/sound/oss/CS4232 b/Documentation/sound/oss/CS4232 new file mode 100644 index 000000000000..7d6af7a5c1c2 --- /dev/null +++ b/Documentation/sound/oss/CS4232 @@ -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 --git a/Documentation/sound/oss/ESS b/Documentation/sound/oss/ESS new file mode 100644 index 000000000000..bba93b4d2def --- /dev/null +++ b/Documentation/sound/oss/ESS @@ -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 --git a/Documentation/sound/oss/ESS1868 b/Documentation/sound/oss/ESS1868 new file mode 100644 index 000000000000..55e922f21bc0 --- /dev/null +++ b/Documentation/sound/oss/ESS1868 @@ -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 --git a/Documentation/sound/oss/INSTALL.awe b/Documentation/sound/oss/INSTALL.awe new file mode 100644 index 000000000000..310f42ca1e83 --- /dev/null +++ b/Documentation/sound/oss/INSTALL.awe @@ -0,0 +1,134 @@ +================================================================ + INSTALLATION OF AWE32 SOUND DRIVER FOR LINUX + Takashi Iwai <iwai@ww.uni-erlangen.de> +================================================================ + +---------------------------------------------------------------- +* 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 Documentation/sound/oss/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 --git a/Documentation/sound/oss/Introduction b/Documentation/sound/oss/Introduction new file mode 100644 index 000000000000..15d4fb975ac0 --- /dev/null +++ b/Documentation/sound/oss/Introduction @@ -0,0 +1,459 @@ +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 Documentation/sound/oss/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 <stodolsk@erols.com>]A + +5. Make the kernel (e.g., make bzImage), and install the kernel. + +6. Make the modules and install them (make modules; make modules_install). + +Note, for 2.5.x kernels, make sure you have the newer module-init-tools +installed or modules will not be loaded properly. 2.5.x requires an +updated module-init-tools. + + +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/modprobe.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/modprobe.conf +3. move the sound modules from /lib/modules/<kernel>/misc + (for example, I make a /lib/modules/<kernel>/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/modprobe.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 <program> 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/modprobe.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: 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 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 --git a/Documentation/sound/oss/MAD16 b/Documentation/sound/oss/MAD16 new file mode 100644 index 000000000000..865dbd848742 --- /dev/null +++ b/Documentation/sound/oss/MAD16 @@ -0,0 +1,56 @@ +(This recipe has been edited to update the configuration symbols, + and change over to modprobe.conf for 2.6) + +From: Shaw Carruthers <shaw@shawc.demon.co.uk> + +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 + +modprobe.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 --git a/Documentation/sound/oss/Maestro b/Documentation/sound/oss/Maestro new file mode 100644 index 000000000000..4a80eb3f8e00 --- /dev/null +++ b/Documentation/sound/oss/Maestro @@ -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 --git a/Documentation/sound/oss/Maestro3 b/Documentation/sound/oss/Maestro3 new file mode 100644 index 000000000000..a113718e8034 --- /dev/null +++ b/Documentation/sound/oss/Maestro3 @@ -0,0 +1,92 @@ + 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/modprobe.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. + +One 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'. + +And the last is 'gpio_pin', which tells the driver which GPIO pin number +the external amp uses (0-15), The Allegro uses 8 by default, all others 1. +If everything loads correctly and seems to be working but you get no sound, +try tweaking this value. + +Systems known to need a different value + Panasonic ToughBook CF-72: gpio_pin=13 + +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 --git a/Documentation/sound/oss/MultiSound b/Documentation/sound/oss/MultiSound new file mode 100644 index 000000000000..e4a18bb7f73a --- /dev/null +++ b/Documentation/sound/oss/MultiSound @@ -0,0 +1,1137 @@ +#! /bin/sh +# +# Turtle Beach MultiSound Driver Notes +# -- Andrew Veliath <andrewtv@usa.net> +# +# 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 <andrewtv@ztransform.velsoft.com>. +# 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 <stdio.h> +#include <unistd.h> +#include <fcntl.h> +#include <sys/types.h> +#include <sys/stat.h> +#include <sys/ioctl.h> +#include <sys/soundcard.h> +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 <mixer device>\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 <stdio.h> +#include <stdlib.h> +#include <string.h> +#include <errno.h> +#include <unistd.h> +#include <asm/io.h> +#include <asm/types.h> +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 <config port> [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 <io> <irq> <mem> Audio device\n" +X " mpu <io> <irq> Internal Kurzweil synth\n" +X " ide <io0> <io1> <irq> On-board IDE controller\n" +X " joystick <io> 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: <config port> 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 <stdio.h> +#include <unistd.h> +#include <fcntl.h> +#include <sys/types.h> +#include <sys/stat.h> +#include <sys/ioctl.h> +#include <sys/soundcard.h> +X +int main(int argc, char *argv[]) +{ +X int fd; +X +X if (argc != 2) { +X fprintf(stderr, "usage: msndreset <mixer device>\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 --git a/Documentation/sound/oss/NEWS b/Documentation/sound/oss/NEWS new file mode 100644 index 000000000000..a81e0ef72ae9 --- /dev/null +++ b/Documentation/sound/oss/NEWS @@ -0,0 +1,42 @@ +Linux 2.4 Sound Changes +2000-September-25 +Christoph Hellwig, <hch@infradead.org> + + + +=== 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 resources 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 --git a/Documentation/sound/oss/NM256 b/Documentation/sound/oss/NM256 new file mode 100644 index 000000000000..b503217488b3 --- /dev/null +++ b/Documentation/sound/oss/NM256 @@ -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 --git a/Documentation/sound/oss/OPL3 b/Documentation/sound/oss/OPL3 new file mode 100644 index 000000000000..2468ff827688 --- /dev/null +++ b/Documentation/sound/oss/OPL3 @@ -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 --git a/Documentation/sound/oss/OPL3-SA b/Documentation/sound/oss/OPL3-SA new file mode 100644 index 000000000000..66a91835d918 --- /dev/null +++ b/Documentation/sound/oss/OPL3-SA @@ -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_SOUND_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? +<stiker@northlink.com> diff --git a/Documentation/sound/oss/OPL3-SA2 b/Documentation/sound/oss/OPL3-SA2 new file mode 100644 index 000000000000..d8b6d2bbada6 --- /dev/null +++ b/Documentation/sound/oss/OPL3-SA2 @@ -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 <luyer@ucs.uwa.edu.au>). You will still need +to specify settings, which can be done by adding: + +opl3sa2=<io>,<irq>,<dma>,<dma2>,<mssio>,<mpuio> + +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/modprobe.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 modprobe.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 --git a/Documentation/sound/oss/Opti b/Documentation/sound/oss/Opti new file mode 100644 index 000000000000..c15af3c07d46 --- /dev/null +++ b/Documentation/sound/oss/Opti @@ -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/modprobe.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/modprobe.conf' is used as before. + +This is the sound part of my /etc/modprobe.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 +install mad16 /sbin/modprobe -i 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 <stdio.h> +#include <fcntl.h> +#include <linux/soundcard.h> + +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 --git a/Documentation/sound/oss/PAS16 b/Documentation/sound/oss/PAS16 new file mode 100644 index 000000000000..951b3dce51b4 --- /dev/null +++ b/Documentation/sound/oss/PAS16 @@ -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, Documentation/sound/oss/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 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.tldp.org/docs.html#howto . General information + about the modular sound system is contained in the files + Documentation/sound/oss/Introduction. The file + Documentation/sound/oss/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=<io>,<irq>,<dma>,<dma2>,<sbio>,<sbirq>,<sbdma>,<sbdma2> + 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/oss/OPL3 if your card has an + OPL3 chip. + If you compile the driver into the kernel, you have to add + "opl3=<io>" 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/modprobe.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/modprobe.conf for pas2, sb and opl3 +respectively. diff --git a/Documentation/sound/oss/PSS b/Documentation/sound/oss/PSS new file mode 100644 index 000000000000..187b9525e1f6 --- /dev/null +++ b/Documentation/sound/oss/PSS @@ -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 --git a/Documentation/sound/oss/PSS-updates b/Documentation/sound/oss/PSS-updates new file mode 100644 index 000000000000..c84dd7597e64 --- /dev/null +++ b/Documentation/sound/oss/PSS-updates @@ -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 compatible +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 appropriate 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 appropriate 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 +compatibility 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 parameter 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 compatibility 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 compatibility 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 --git a/Documentation/sound/oss/README.OSS b/Documentation/sound/oss/README.OSS new file mode 100644 index 000000000000..fd42b05b2f55 --- /dev/null +++ b/Documentation/sound/oss/README.OSS @@ -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 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 <Ensoniq Soundscape (old)>, 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 --git a/Documentation/sound/oss/README.awe b/Documentation/sound/oss/README.awe new file mode 100644 index 000000000000..80054cd8fcde --- /dev/null +++ b/Documentation/sound/oss/README.awe @@ -0,0 +1,218 @@ +================================================================ + AWE32 Sound Driver for Linux / FreeBSD + version 0.4.3; Nov. 1, 1998 + + Takashi Iwai <iwai@ww.uni-erlangen.de> +================================================================ + +* 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 <iwai@ww.uni-erlangen.de>, +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 --git a/Documentation/sound/oss/README.modules b/Documentation/sound/oss/README.modules new file mode 100644 index 000000000000..e691d74e1e5e --- /dev/null +++ b/Documentation/sound/oss/README.modules @@ -0,0 +1,106 @@ +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: + +<M> Sound card support + +<M> 100% Sound Blaster compatibles (SB16/32/64, ESS, Jazz16) support + + I have SoundBlaster. Select your card from the list. + +<M> Generic OPL2/OPL3 FM synthesizer support +<M> 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/modprobe.conf something like: + +alias char-major-14-* sb +install sb /sbin/modprobe -i sb && /sbin/modprobe 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/<kernel-version>/misc. Mine looks like: + +adlib_card.o # This is the generic OPLx driver +opl3.o # The OPL3 driver +sb.o # <<The SoundBlaster driver. Yours may differ.>> +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 function referred to by module_init() 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/modprobe.conf or with command-line args to modprobe, e.g. + +modprobe sb io=0x220 irq=7 dma=1 dma16=5 mpu_io=0x330 +modprobe adlib_card io=0x388 + + recommend using /etc/modprobe.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/modprobe.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 --git a/Documentation/sound/oss/README.ymfsb b/Documentation/sound/oss/README.ymfsb new file mode 100644 index 000000000000..af8a7d3a4e8e --- /dev/null +++ b/Documentation/sound/oss/README.ymfsb @@ -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 <jgarzik@pobox.com>'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 <breeze.nagano@nifty.ne.jp> + diff --git a/Documentation/sound/oss/SoundPro b/Documentation/sound/oss/SoundPro new file mode 100644 index 000000000000..9d4db1f29d3c --- /dev/null +++ b/Documentation/sound/oss/SoundPro @@ -0,0 +1,105 @@ +Documentation for the SoundPro CMI8330 extensions in the WSS driver (ad1848.o) +------------------------------------------------------------------------------ + +( Be sure to read Documentation/sound/oss/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 --git a/Documentation/sound/oss/Soundblaster b/Documentation/sound/oss/Soundblaster new file mode 100644 index 000000000000..b288d464ba8b --- /dev/null +++ b/Documentation/sound/oss/Soundblaster @@ -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 --git a/Documentation/sound/oss/Tropez+ b/Documentation/sound/oss/Tropez+ new file mode 100644 index 000000000000..b93a6b734fc0 --- /dev/null +++ b/Documentation/sound/oss/Tropez+ @@ -0,0 +1,26 @@ +From: Paul Barton-Davis <pbd@op.net> + +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 --git a/Documentation/sound/oss/VIA-chipset b/Documentation/sound/oss/VIA-chipset new file mode 100644 index 000000000000..37865234e54d --- /dev/null +++ b/Documentation/sound/oss/VIA-chipset @@ -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 --git a/Documentation/sound/oss/VIBRA16 b/Documentation/sound/oss/VIBRA16 new file mode 100644 index 000000000000..68a5a46beb88 --- /dev/null +++ b/Documentation/sound/oss/VIBRA16 @@ -0,0 +1,80 @@ +Sound Blaster 16X Vibra addendum +-------------------------------- +by Marius Ilioaea <mariusi@protv.ro> + Stefan Laudat <stefan@asit.ro> + +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: + +<snip 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) + +<end of snipping> + + 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: + +modprobe soundcore +modprobe sound +modprobe uart401 +modprobe sb io=0x220 irq=5 dma=1 dma16=3 +# do you need MIDI? +modprobe 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 --git a/Documentation/sound/oss/WaveArtist b/Documentation/sound/oss/WaveArtist new file mode 100644 index 000000000000..f4f3407cd818 --- /dev/null +++ b/Documentation/sound/oss/WaveArtist @@ -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 --git a/Documentation/sound/oss/Wavefront b/Documentation/sound/oss/Wavefront new file mode 100644 index 000000000000..16f57ea43052 --- /dev/null +++ b/Documentation/sound/oss/Wavefront @@ -0,0 +1,339 @@ + 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 it's 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/oss/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 <top-of-your-2.1.106-code-base-e.g.-/usr/src/linux> + tar -zxvf <where-you-put/wavefront.tar.gz> + +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 <top of your kernel tree> + 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 boot + . + . + . + <whatever you normally do for a kernel install> + 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/modprobe.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 + install wavefront /sbin/modprobe cs4232 && /sbin/modprobe -i wavefront && /sbin/modprobe 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); +<<EOF>> { 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 <pbd@op.net> + + diff --git a/Documentation/sound/oss/btaudio b/Documentation/sound/oss/btaudio new file mode 100644 index 000000000000..1a693e69d44b --- /dev/null +++ b/Documentation/sound/oss/btaudio @@ -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 <kraxel@bytesex.org> diff --git a/Documentation/sound/oss/cs46xx b/Documentation/sound/oss/cs46xx new file mode 100644 index 000000000000..88d6cf8b39f3 --- /dev/null +++ b/Documentation/sound/oss/cs46xx @@ -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 definitions 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 --git a/Documentation/sound/oss/es1370 b/Documentation/sound/oss/es1370 new file mode 100644 index 000000000000..7b38b1a096a3 --- /dev/null +++ b/Documentation/sound/oss/es1370 @@ -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 --git a/Documentation/sound/oss/es1371 b/Documentation/sound/oss/es1371 new file mode 100644 index 000000000000..c3151266771c --- /dev/null +++ b/Documentation/sound/oss/es1371 @@ -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 --git a/Documentation/sound/oss/mwave b/Documentation/sound/oss/mwave new file mode 100644 index 000000000000..858334bb46b0 --- /dev/null +++ b/Documentation/sound/oss/mwave @@ -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 voilŕ - 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 --git a/Documentation/sound/oss/rme96xx b/Documentation/sound/oss/rme96xx new file mode 100644 index 000000000000..87d7b7b65fa1 --- /dev/null +++ b/Documentation/sound/oss/rme96xx @@ -0,0 +1,767 @@ +Beta release of the rme96xx (driver for RME 96XX cards like the +"Hammerfall" and the "Hammerfall light") + +Important: The driver module has to be installed on a freshly rebooted system, +otherwise the driver might not be able to acquire its buffers. + +features: + + - OSS programming interface (i.e. runs with standard OSS soundsoftware) + - OSS/Multichannel interface (OSS multichannel is done by just aquiring + more than 2 channels). The driver does not use more than one device + ( yet .. this feature may be implemented later ) + - more than one RME card supported + +The driver uses a specific multichannel interface, which I will document +when the driver gets stable. (take a look at the defines in rme96xx.h, +which adds blocked multichannel formats i.e instead of +lrlrlrlr --> llllrrrr etc. + +Use the "rmectrl" programm to look at the status of the card .. +or use xrmectrl, a GUI interface for the ctrl program. + +What you can do with the rmectrl program is to set the stereo device for +OSS emulation (e.g. if you use SPDIF out). + +You do: + +./ctrl offset 24 24 + +which makes the stereo device use channels 25 and 26. + +Guenter Geiger <geiger@epy.co.at> + +copy the first part of the attached source code into rmectrl.c +and the second part into xrmectrl (or get the program from +http://gige.xdv.org/pages/soft/pages/rme) + +to compile: gcc -o rmectrl rmectrl.c +------------------------------ snip ------------------------------------ + +#include <stdio.h> +#include <sys/types.h> +#include <sys/stat.h> +#include <sys/ioctl.h> +#include <fcntl.h> +#include <linux/soundcard.h> +#include <math.h> +#include <unistd.h> +#include <stdlib.h> +#include "rme96xx.h" + +/* + remctrl.c + (C) 2000 Guenter Geiger <geiger@debian.org> + HP20020201 - Heiko Purnhagen <purnhage@tnt.uni-hannover.de> +*/ + +/* # define DEVICE_NAME "/dev/mixer" */ +# define DEVICE_NAME "/dev/mixer1" + + +void usage(void) +{ + fprintf(stderr,"usage: rmectrl [/dev/mixer<n>] [command [options]]\n\n"); + fprintf(stderr,"where command is one of:\n"); + fprintf(stderr," help show this help\n"); + fprintf(stderr," status show status bits\n"); + fprintf(stderr," control show control bits\n"); + fprintf(stderr," mix show mixer/offset status\n"); + fprintf(stderr," master <n> set sync master\n"); + fprintf(stderr," pro <n> set spdif out pro\n"); + fprintf(stderr," emphasis <n> set spdif out emphasis\n"); + fprintf(stderr," dolby <n> set spdif out no audio\n"); + fprintf(stderr," optout <n> set spdif out optical\n"); + fprintf(stderr," wordclock <n> set sync wordclock\n"); + fprintf(stderr," spdifin <n> set spdif in (0=optical,1=coax,2=intern)\n"); + fprintf(stderr," syncref <n> set sync source (0=ADAT1,1=ADAT2,2=ADAT3,3=SPDIF)\n"); + fprintf(stderr," adat1cd <n> set ADAT1 on internal CD\n"); + fprintf(stderr," offset <devnr> <in> <out> set dev (0..3) offset (0..25)\n"); + exit(-1); +} + + +int main(int argc, char* argv[]) +{ + int cards; + int ret; + int i; + double ft; + int fd, fdwr; + int param,orig; + rme_status_t stat; + rme_ctrl_t ctrl; + char *device; + int argidx; + + if (argc < 2) + usage(); + + if (*argv[1]=='/') { + device = argv[1]; + argidx = 2; + } + else { + device = DEVICE_NAME; + argidx = 1; + } + + fprintf(stdout,"mixer device %s\n",device); + if ((fd = open(device,O_RDONLY)) < 0) { + fprintf(stdout,"opening device failed\n"); + exit(-1); + } + + if ((fdwr = open(device,O_WRONLY)) < 0) { + fprintf(stdout,"opening device failed\n"); + exit(-1); + } + + if (argc < argidx+1) + usage(); + + if (!strcmp(argv[argidx],"help")) + usage(); + if (!strcmp(argv[argidx],"-h")) + usage(); + if (!strcmp(argv[argidx],"--help")) + usage(); + + if (!strcmp(argv[argidx],"status")) { + ioctl(fd,SOUND_MIXER_PRIVATE2,&stat); + fprintf(stdout,"stat.irq %d\n",stat.irq); + fprintf(stdout,"stat.lockmask %d\n",stat.lockmask); + fprintf(stdout,"stat.sr48 %d\n",stat.sr48); + fprintf(stdout,"stat.wclock %d\n",stat.wclock); + fprintf(stdout,"stat.bufpoint %d\n",stat.bufpoint); + fprintf(stdout,"stat.syncmask %d\n",stat.syncmask); + fprintf(stdout,"stat.doublespeed %d\n",stat.doublespeed); + fprintf(stdout,"stat.tc_busy %d\n",stat.tc_busy); + fprintf(stdout,"stat.tc_out %d\n",stat.tc_out); + fprintf(stdout,"stat.crystalrate %d (0=64k 3=96k 4=88.2k 5=48k 6=44.1k 7=32k)\n",stat.crystalrate); + fprintf(stdout,"stat.spdif_error %d\n",stat.spdif_error); + fprintf(stdout,"stat.bufid %d\n",stat.bufid); + fprintf(stdout,"stat.tc_valid %d\n",stat.tc_valid); + exit (0); + } + + if (!strcmp(argv[argidx],"control")) { + ioctl(fd,SOUND_MIXER_PRIVATE3,&ctrl); + fprintf(stdout,"ctrl.start %d\n",ctrl.start); + fprintf(stdout,"ctrl.latency %d (0=64 .. 7=8192)\n",ctrl.latency); + fprintf(stdout,"ctrl.master %d\n",ctrl.master); + fprintf(stdout,"ctrl.ie %d\n",ctrl.ie); + fprintf(stdout,"ctrl.sr48 %d\n",ctrl.sr48); + fprintf(stdout,"ctrl.spare %d\n",ctrl.spare); + fprintf(stdout,"ctrl.doublespeed %d\n",ctrl.doublespeed); + fprintf(stdout,"ctrl.pro %d\n",ctrl.pro); + fprintf(stdout,"ctrl.emphasis %d\n",ctrl.emphasis); + fprintf(stdout,"ctrl.dolby %d\n",ctrl.dolby); + fprintf(stdout,"ctrl.opt_out %d\n",ctrl.opt_out); + fprintf(stdout,"ctrl.wordclock %d\n",ctrl.wordclock); + fprintf(stdout,"ctrl.spdif_in %d (0=optical,1=coax,2=intern)\n",ctrl.spdif_in); + fprintf(stdout,"ctrl.sync_ref %d (0=ADAT1,1=ADAT2,2=ADAT3,3=SPDIF)\n",ctrl.sync_ref); + fprintf(stdout,"ctrl.spdif_reset %d\n",ctrl.spdif_reset); + fprintf(stdout,"ctrl.spdif_select %d\n",ctrl.spdif_select); + fprintf(stdout,"ctrl.spdif_clock %d\n",ctrl.spdif_clock); + fprintf(stdout,"ctrl.spdif_write %d\n",ctrl.spdif_write); + fprintf(stdout,"ctrl.adat1_cd %d\n",ctrl.adat1_cd); + exit (0); + } + + if (!strcmp(argv[argidx],"mix")) { + rme_mixer mix; + int i; + + for (i=0; i<4; i++) { + mix.devnr = i; + ioctl(fd,SOUND_MIXER_PRIVATE1,&mix); + if (mix.devnr == i) { + fprintf(stdout,"devnr %d\n",mix.devnr); + fprintf(stdout,"mix.i_offset %2d (0-25)\n",mix.i_offset); + fprintf(stdout,"mix.o_offset %2d (0-25)\n",mix.o_offset); + } + } + exit (0); + } + +/* the control flags */ + + if (argc < argidx+2) + usage(); + + if (!strcmp(argv[argidx],"master")) { + int val = atoi(argv[argidx+1]); + ioctl(fd,SOUND_MIXER_PRIVATE3,&ctrl); + printf("master = %d\n",val); + ctrl.master = val; + ioctl(fdwr,SOUND_MIXER_PRIVATE3,&ctrl); + exit (0); + } + + if (!strcmp(argv[argidx],"pro")) { + int val = atoi(argv[argidx+1]); + ioctl(fd,SOUND_MIXER_PRIVATE3,&ctrl); + printf("pro = %d\n",val); + ctrl.pro = val; + ioctl(fdwr,SOUND_MIXER_PRIVATE3,&ctrl); + exit (0); + } + + if (!strcmp(argv[argidx],"emphasis")) { + int val = atoi(argv[argidx+1]); + ioctl(fd,SOUND_MIXER_PRIVATE3,&ctrl); + printf("emphasis = %d\n",val); + ctrl.emphasis = val; + ioctl(fdwr,SOUND_MIXER_PRIVATE3,&ctrl); + exit (0); + } + + if (!strcmp(argv[argidx],"dolby")) { + int val = atoi(argv[argidx+1]); + ioctl(fd,SOUND_MIXER_PRIVATE3,&ctrl); + printf("dolby = %d\n",val); + ctrl.dolby = val; + ioctl(fdwr,SOUND_MIXER_PRIVATE3,&ctrl); + exit (0); + } + + if (!strcmp(argv[argidx],"optout")) { + int val = atoi(argv[argidx+1]); + ioctl(fd,SOUND_MIXER_PRIVATE3,&ctrl); + printf("optout = %d\n",val); + ctrl.opt_out = val; + ioctl(fdwr,SOUND_MIXER_PRIVATE3,&ctrl); + exit (0); + } + + if (!strcmp(argv[argidx],"wordclock")) { + int val = atoi(argv[argidx+1]); + ioctl(fd,SOUND_MIXER_PRIVATE3,&ctrl); + printf("wordclock = %d\n",val); + ctrl.wordclock = val; + ioctl(fdwr,SOUND_MIXER_PRIVATE3,&ctrl); + exit (0); + } + + if (!strcmp(argv[argidx],"spdifin")) { + int val = atoi(argv[argidx+1]); + ioctl(fd,SOUND_MIXER_PRIVATE3,&ctrl); + printf("spdifin = %d\n",val); + ctrl.spdif_in = val; + ioctl(fdwr,SOUND_MIXER_PRIVATE3,&ctrl); + exit (0); + } + + if (!strcmp(argv[argidx],"syncref")) { + int val = atoi(argv[argidx+1]); + ioctl(fd,SOUND_MIXER_PRIVATE3,&ctrl); + printf("syncref = %d\n",val); + ctrl.sync_ref = val; + ioctl(fdwr,SOUND_MIXER_PRIVATE3,&ctrl); + exit (0); + } + + if (!strcmp(argv[argidx],"adat1cd")) { + int val = atoi(argv[argidx+1]); + ioctl(fd,SOUND_MIXER_PRIVATE3,&ctrl); + printf("adat1cd = %d\n",val); + ctrl.adat1_cd = val; + ioctl(fdwr,SOUND_MIXER_PRIVATE3,&ctrl); + exit (0); + } + +/* setting offset */ + + if (argc < argidx+4) + usage(); + + if (!strcmp(argv[argidx],"offset")) { + rme_mixer mix; + + mix.devnr = atoi(argv[argidx+1]); + + mix.i_offset = atoi(argv[argidx+2]); + mix.o_offset = atoi(argv[argidx+3]); + ioctl(fdwr,SOUND_MIXER_PRIVATE1,&mix); + fprintf(stdout,"devnr %d\n",mix.devnr); + fprintf(stdout,"mix.i_offset to %d\n",mix.i_offset); + fprintf(stdout,"mix.o_offset to %d\n",mix.o_offset); + exit (0); + } + + usage(); + exit (0); /* to avoid warning */ +} + + +---------------------------- <snip> -------------------------------- +#!/usr/bin/wish + +# xrmectrl +# (C) 2000 Guenter Geiger <geiger@debian.org> +# HP20020201 - Heiko Purnhagen <purnhage@tnt.uni-hannover.de> + +#set defaults "-relief ridged" +set CTRLPROG "./rmectrl" +if {$argc} { + set CTRLPROG "$CTRLPROG $argv" +} +puts "CTRLPROG $CTRLPROG" + +frame .butts +button .butts.exit -text "Exit" -command "exit" -relief ridge +#button .butts.state -text "State" -command "get_all" + +pack .butts.exit -side left +pack .butts -side bottom + + +# +# STATUS +# + +frame .status + +# Sampling Rate + +frame .status.sr +label .status.sr.text -text "Sampling Rate" -justify left +radiobutton .status.sr.441 -selectcolor red -text "44.1 kHz" -width 10 -anchor nw -variable srate -value 44100 -font times +radiobutton .status.sr.480 -selectcolor red -text "48 kHz" -width 10 -anchor nw -variable srate -value 48000 -font times +radiobutton .status.sr.882 -selectcolor red -text "88.2 kHz" -width 10 -anchor nw -variable srate -value 88200 -font times +radiobutton .status.sr.960 -selectcolor red -text "96 kHz" -width 10 -anchor nw -variable srate -value 96000 -font times + +pack .status.sr.text .status.sr.441 .status.sr.480 .status.sr.882 .status.sr.960 -side top -padx 3 + +# Lock + +frame .status.lock +label .status.lock.text -text "Lock" -justify left +checkbutton .status.lock.adat1 -selectcolor red -text "ADAT1" -anchor nw -width 10 -variable adatlock1 -font times +checkbutton .status.lock.adat2 -selectcolor red -text "ADAT2" -anchor nw -width 10 -variable adatlock2 -font times +checkbutton .status.lock.adat3 -selectcolor red -text "ADAT3" -anchor nw -width 10 -variable adatlock3 -font times + +pack .status.lock.text .status.lock.adat1 .status.lock.adat2 .status.lock.adat3 -side top -padx 3 + +# Sync + +frame .status.sync +label .status.sync.text -text "Sync" -justify left +checkbutton .status.sync.adat1 -selectcolor red -text "ADAT1" -anchor nw -width 10 -variable adatsync1 -font times +checkbutton .status.sync.adat2 -selectcolor red -text "ADAT2" -anchor nw -width 10 -variable adatsync2 -font times +checkbutton .status.sync.adat3 -selectcolor red -text "ADAT3" -anchor nw -width 10 -variable adatsync3 -font times + +pack .status.sync.text .status.sync.adat1 .status.sync.adat2 .status.sync.adat3 -side top -padx 3 + +# Timecode + +frame .status.tc +label .status.tc.text -text "Timecode" -justify left +checkbutton .status.tc.busy -selectcolor red -text "busy" -anchor nw -width 10 -variable tcbusy -font times +checkbutton .status.tc.out -selectcolor red -text "out" -anchor nw -width 10 -variable tcout -font times +checkbutton .status.tc.valid -selectcolor red -text "valid" -anchor nw -width 10 -variable tcvalid -font times + +pack .status.tc.text .status.tc.busy .status.tc.out .status.tc.valid -side top -padx 3 + +# SPDIF In + +frame .status.spdif +label .status.spdif.text -text "SPDIF In" -justify left +label .status.spdif.sr -text "--.- kHz" -anchor n -width 10 -font times +checkbutton .status.spdif.error -selectcolor red -text "Input Lock" -anchor nw -width 10 -variable spdiferr -font times + +pack .status.spdif.text .status.spdif.sr .status.spdif.error -side top -padx 3 + +pack .status.sr .status.lock .status.sync .status.tc .status.spdif -side left -fill x -anchor n -expand 1 + + +# +# CONTROL +# + +proc setprof {} { + global CTRLPROG + global spprof + exec $CTRLPROG pro $spprof +} + +proc setemph {} { + global CTRLPROG + global spemph + exec $CTRLPROG emphasis $spemph +} + +proc setnoaud {} { + global CTRLPROG + global spnoaud + exec $CTRLPROG dolby $spnoaud +} + +proc setoptical {} { + global CTRLPROG + global spoptical + exec $CTRLPROG optout $spoptical +} + +proc setspdifin {} { + global CTRLPROG + global spdifin + exec $CTRLPROG spdifin [expr $spdifin - 1] +} + +proc setsyncsource {} { + global CTRLPROG + global syncsource + exec $CTRLPROG syncref [expr $syncsource -1] +} + + +proc setmaster {} { + global CTRLPROG + global master + exec $CTRLPROG master $master +} + +proc setwordclock {} { + global CTRLPROG + global wordclock + exec $CTRLPROG wordclock $wordclock +} + +proc setadat1cd {} { + global CTRLPROG + global adat1cd + exec $CTRLPROG adat1cd $adat1cd +} + + +frame .control + +# SPDIF In & SPDIF Out + + +frame .control.spdif + +frame .control.spdif.in +label .control.spdif.in.text -text "SPDIF In" -justify left +radiobutton .control.spdif.in.input1 -text "Optical" -anchor nw -width 13 -variable spdifin -value 1 -command setspdifin -selectcolor blue -font times +radiobutton .control.spdif.in.input2 -text "Coaxial" -anchor nw -width 13 -variable spdifin -value 2 -command setspdifin -selectcolor blue -font times +radiobutton .control.spdif.in.input3 -text "Intern " -anchor nw -width 13 -variable spdifin -command setspdifin -value 3 -selectcolor blue -font times + +checkbutton .control.spdif.in.adat1cd -text "ADAT1 Intern" -anchor nw -width 13 -variable adat1cd -command setadat1cd -selectcolor blue -font times + +pack .control.spdif.in.text .control.spdif.in.input1 .control.spdif.in.input2 .control.spdif.in.input3 .control.spdif.in.adat1cd + +label .control.spdif.space + +frame .control.spdif.out +label .control.spdif.out.text -text "SPDIF Out" -justify left +checkbutton .control.spdif.out.pro -text "Professional" -anchor nw -width 13 -variable spprof -command setprof -selectcolor blue -font times +checkbutton .control.spdif.out.emphasis -text "Emphasis" -anchor nw -width 13 -variable spemph -command setemph -selectcolor blue -font times +checkbutton .control.spdif.out.dolby -text "NoAudio" -anchor nw -width 13 -variable spnoaud -command setnoaud -selectcolor blue -font times +checkbutton .control.spdif.out.optout -text "Optical Out" -anchor nw -width 13 -variable spoptical -command setoptical -selectcolor blue -font times + +pack .control.spdif.out.optout .control.spdif.out.dolby .control.spdif.out.emphasis .control.spdif.out.pro .control.spdif.out.text -side bottom + +pack .control.spdif.in .control.spdif.space .control.spdif.out -side top -fill y -padx 3 -expand 1 + +# Sync Mode & Sync Source + +frame .control.sync +frame .control.sync.mode +label .control.sync.mode.text -text "Sync Mode" -justify left +checkbutton .control.sync.mode.master -text "Master" -anchor nw -width 13 -variable master -command setmaster -selectcolor blue -font times +checkbutton .control.sync.mode.wc -text "Wordclock" -anchor nw -width 13 -variable wordclock -command setwordclock -selectcolor blue -font times + +pack .control.sync.mode.text .control.sync.mode.master .control.sync.mode.wc + +label .control.sync.space + +frame .control.sync.src +label .control.sync.src.text -text "Sync Source" -justify left +radiobutton .control.sync.src.input1 -text "ADAT1" -anchor nw -width 13 -variable syncsource -value 1 -command setsyncsource -selectcolor blue -font times +radiobutton .control.sync.src.input2 -text "ADAT2" -anchor nw -width 13 -variable syncsource -value 2 -command setsyncsource -selectcolor blue -font times +radiobutton .control.sync.src.input3 -text "ADAT3" -anchor nw -width 13 -variable syncsource -command setsyncsource -value 3 -selectcolor blue -font times +radiobutton .control.sync.src.input4 -text "SPDIF" -anchor nw -width 13 -variable syncsource -command setsyncsource -value 4 -selectcolor blue -font times + +pack .control.sync.src.input4 .control.sync.src.input3 .control.sync.src.input2 .control.sync.src.input1 .control.sync.src.text -side bottom + +pack .control.sync.mode .control.sync.space .control.sync.src -side top -fill y -padx 3 -expand 1 + +label .control.space -text "" -width 10 + +# Buffer Size + +frame .control.buf +label .control.buf.text -text "Buffer Size (Latency)" -justify left +radiobutton .control.buf.b1 -selectcolor red -text "64 (1.5 ms)" -width 13 -anchor nw -variable ssrate -value 1 -font times +radiobutton .control.buf.b2 -selectcolor red -text "128 (3 ms)" -width 13 -anchor nw -variable ssrate -value 2 -font times +radiobutton .control.buf.b3 -selectcolor red -text "256 (6 ms)" -width 13 -anchor nw -variable ssrate -value 3 -font times +radiobutton .control.buf.b4 -selectcolor red -text "512 (12 ms)" -width 13 -anchor nw -variable ssrate -value 4 -font times +radiobutton .control.buf.b5 -selectcolor red -text "1024 (23 ms)" -width 13 -anchor nw -variable ssrate -value 5 -font times +radiobutton .control.buf.b6 -selectcolor red -text "2048 (46 ms)" -width 13 -anchor nw -variable ssrate -value 6 -font times +radiobutton .control.buf.b7 -selectcolor red -text "4096 (93 ms)" -width 13 -anchor nw -variable ssrate -value 7 -font times +radiobutton .control.buf.b8 -selectcolor red -text "8192 (186 ms)" -width 13 -anchor nw -variable ssrate -value 8 -font times + +pack .control.buf.text .control.buf.b1 .control.buf.b2 .control.buf.b3 .control.buf.b4 .control.buf.b5 .control.buf.b6 .control.buf.b7 .control.buf.b8 -side top -padx 3 + +# Offset + +frame .control.offset + +frame .control.offset.in +label .control.offset.in.text -text "Offset In" -justify left +label .control.offset.in.off0 -text "dev\#0: -" -anchor nw -width 10 -font times +label .control.offset.in.off1 -text "dev\#1: -" -anchor nw -width 10 -font times +label .control.offset.in.off2 -text "dev\#2: -" -anchor nw -width 10 -font times +label .control.offset.in.off3 -text "dev\#3: -" -anchor nw -width 10 -font times + +pack .control.offset.in.text .control.offset.in.off0 .control.offset.in.off1 .control.offset.in.off2 .control.offset.in.off3 + +label .control.offset.space + +frame .control.offset.out +label .control.offset.out.text -text "Offset Out" -justify left +label .control.offset.out.off0 -text "dev\#0: -" -anchor nw -width 10 -font times +label .control.offset.out.off1 -text "dev\#1: -" -anchor nw -width 10 -font times +label .control.offset.out.off2 -text "dev\#2: -" -anchor nw -width 10 -font times +label .control.offset.out.off3 -text "dev\#3: -" -anchor nw -width 10 -font times + +pack .control.offset.out.off3 .control.offset.out.off2 .control.offset.out.off1 .control.offset.out.off0 .control.offset.out.text -side bottom + +pack .control.offset.in .control.offset.space .control.offset.out -side top -fill y -padx 3 -expand 1 + + +pack .control.spdif .control.sync .control.space .control.buf .control.offset -side left -fill both -anchor n -expand 1 + + +label .statustext -text Status -justify center -relief ridge +label .controltext -text Control -justify center -relief ridge + +label .statusspace +label .controlspace + +pack .statustext .status .statusspace .controltext .control .controlspace -side top -anchor nw -fill both -expand 1 + + +proc get_bit {output sstr} { + set idx1 [string last [concat $sstr 1] $output] + set idx1 [expr $idx1 != -1] + return $idx1 +} + +proc get_val {output sstr} { + set val [string wordend $output [string last $sstr $output]] + set val [string range $output $val [expr $val+1]] + return $val +} + +proc get_val2 {output sstr} { + set val [string wordend $output [string first $sstr $output]] + set val [string range $output $val [expr $val+2]] + return $val +} + +proc get_control {} { + global spprof + global spemph + global spnoaud + global spoptical + global spdifin + global ssrate + global master + global wordclock + global syncsource + global CTRLPROG + + set f [open "| $CTRLPROG control" r+] + set ooo [read $f 1000] + close $f +# puts $ooo + + set spprof [ get_bit $ooo "pro"] + set spemph [ get_bit $ooo "emphasis"] + set spnoaud [ get_bit $ooo "dolby"] + set spoptical [ get_bit $ooo "opt_out"] + set spdifin [ expr [ get_val $ooo "spdif_in"] + 1] + set ssrate [ expr [ get_val $ooo "latency"] + 1] + set master [ expr [ get_val $ooo "master"]] + set wordclock [ expr [ get_val $ooo "wordclock"]] + set syncsource [ expr [ get_val $ooo "sync_ref"] + 1] +} + +proc get_status {} { + global srate + global ctrlcom + + global adatlock1 + global adatlock2 + global adatlock3 + + global adatsync1 + global adatsync2 + global adatsync3 + + global tcbusy + global tcout + global tcvalid + + global spdiferr + global crystal + global .status.spdif.text + global CTRLPROG + + + set f [open "| $CTRLPROG status" r+] + set ooo [read $f 1000] + close $f +# puts $ooo + +# samplerate + + set idx1 [string last "sr48 1" $ooo] + set idx2 [string last "doublespeed 1" $ooo] + if {$idx1 >= 0} { + set fact1 48000 + } else { + set fact1 44100 + } + + if {$idx2 >= 0} { + set fact2 2 + } else { + set fact2 1 + } + set srate [expr $fact1 * $fact2] +# ADAT lock + + set val [get_val $ooo lockmask] + set adatlock1 0 + set adatlock2 0 + set adatlock3 0 + if {[expr $val & 1]} { + set adatlock3 1 + } + if {[expr $val & 2]} { + set adatlock2 1 + } + if {[expr $val & 4]} { + set adatlock1 1 + } + +# ADAT sync + set val [get_val $ooo syncmask] + set adatsync1 0 + set adatsync2 0 + set adatsync3 0 + + if {[expr $val & 1]} { + set adatsync3 1 + } + if {[expr $val & 2]} { + set adatsync2 1 + } + if {[expr $val & 4]} { + set adatsync1 1 + } + +# TC busy + + set tcbusy [get_bit $ooo "busy"] + set tcout [get_bit $ooo "out"] + set tcvalid [get_bit $ooo "valid"] + set spdiferr [expr [get_bit $ooo "spdif_error"] == 0] + +# 000=64kHz, 100=88.2kHz, 011=96kHz +# 111=32kHz, 110=44.1kHz, 101=48kHz + + set val [get_val $ooo crystalrate] + + set crystal "--.- kHz" + if {$val == 0} { + set crystal "64 kHz" + } + if {$val == 4} { + set crystal "88.2 kHz" + } + if {$val == 3} { + set crystal "96 kHz" + } + if {$val == 7} { + set crystal "32 kHz" + } + if {$val == 6} { + set crystal "44.1 kHz" + } + if {$val == 5} { + set crystal "48 kHz" + } + .status.spdif.sr configure -text $crystal +} + +proc get_offset {} { + global inoffset + global outoffset + global CTRLPROG + + set f [open "| $CTRLPROG mix" r+] + set ooo [read $f 1000] + close $f +# puts $ooo + + if { [string match "*devnr*" $ooo] } { + set ooo [string range $ooo [string wordend $ooo [string first devnr $ooo]] end] + set val [get_val2 $ooo i_offset] + .control.offset.in.off0 configure -text "dev\#0: $val" + set val [get_val2 $ooo o_offset] + .control.offset.out.off0 configure -text "dev\#0: $val" + } else { + .control.offset.in.off0 configure -text "dev\#0: -" + .control.offset.out.off0 configure -text "dev\#0: -" + } + if { [string match "*devnr*" $ooo] } { + set ooo [string range $ooo [string wordend $ooo [string first devnr $ooo]] end] + set val [get_val2 $ooo i_offset] + .control.offset.in.off1 configure -text "dev\#1: $val" + set val [get_val2 $ooo o_offset] + .control.offset.out.off1 configure -text "dev\#1: $val" + } else { + .control.offset.in.off1 configure -text "dev\#1: -" + .control.offset.out.off1 configure -text "dev\#1: -" + } + if { [string match "*devnr*" $ooo] } { + set ooo [string range $ooo [string wordend $ooo [string first devnr $ooo]] end] + set val [get_val2 $ooo i_offset] + .control.offset.in.off2 configure -text "dev\#2: $val" + set val [get_val2 $ooo o_offset] + .control.offset.out.off2 configure -text "dev\#2: $val" + } else { + .control.offset.in.off2 configure -text "dev\#2: -" + .control.offset.out.off2 configure -text "dev\#2: -" + } + if { [string match "*devnr*" $ooo] } { + set ooo [string range $ooo [string wordend $ooo [string first devnr $ooo]] end] + set val [get_val2 $ooo i_offset] + .control.offset.in.off3 configure -text "dev\#3: $val" + set val [get_val2 $ooo o_offset] + .control.offset.out.off3 configure -text "dev\#3: $val" + } else { + .control.offset.in.off3 configure -text "dev\#3: -" + .control.offset.out.off3 configure -text "dev\#3: -" + } +} + + +proc get_all {} { +get_status +get_control +get_offset +} + +# main +while {1} { + after 200 + get_all + update +} diff --git a/Documentation/sound/oss/solo1 b/Documentation/sound/oss/solo1 new file mode 100644 index 000000000000..6f53d407d027 --- /dev/null +++ b/Documentation/sound/oss/solo1 @@ -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 --git a/Documentation/sound/oss/sonicvibes b/Documentation/sound/oss/sonicvibes new file mode 100644 index 000000000000..84dee2e0b37d --- /dev/null +++ b/Documentation/sound/oss/sonicvibes @@ -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 --git a/Documentation/sound/oss/ultrasound b/Documentation/sound/oss/ultrasound new file mode 100644 index 000000000000..32cd50478b36 --- /dev/null +++ b/Documentation/sound/oss/ultrasound @@ -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 --git a/Documentation/sound/oss/vwsnd b/Documentation/sound/oss/vwsnd new file mode 100644 index 000000000000..a6ea0a1df9e4 --- /dev/null +++ b/Documentation/sound/oss/vwsnd @@ -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, <sys/soundcard.h> defines constants SOUND_MIXER_READ_xxx and +SOUND_MIXER_WRITE_xxx which work just as well. + +SOUND_MIXER_CAPS Read-only + +This is a mask of optional driver capabilities that are implemented. +This driver's only capability is SOUND_CAP_EXCL_INPUT, which means +that only one recording source can be active at a time. + +SOUND_MIXER_DEVMASK Read-only + +This is a mask of the sound channels. This driver's channels are PCM, +LINE, MIC, CD, and RECLEV. + +SOUND_MIXER_STEREODEVS Read-only + +This is a mask of which sound channels are capable of stereo. All +channels are capable of stereo. (But see caveat on MIC input in I/O +CONNECTIONS section above). + +SOUND_MIXER_OUTMASK Read-only + +This is a mask of channels that route inputs through to outputs. +Those are LINE, MIC, and CD. + +SOUND_MIXER_RECMASK Read-only + +This is a mask of channels that can be recording sources. Those are +PCM, LINE, MIC, CD. + +SOUND_MIXER_PCM Default: 0x5757 (0 dB) + +This is the gain control for PCM output. The left and right channel +gain are controlled independently. This gain control has 64 levels, +which range from -82.5 dB to +12.0 dB in 1.5 dB steps. Those 64 +levels are mapped onto 100 levels at the ioctl, see below. + +SOUND_MIXER_LINE Default: 0x4a4a (0 dB) + +This is the gain control for mixing the Line In source into the +outputs. The left and right channel gain are controlled +independently. This gain control has 32 levels, which range from +-34.5 dB to +12.0 dB in 1.5 dB steps. Those 32 levels are mapped onto +100 levels at the ioctl, see below. + +SOUND_MIXER_MIC Default: 0x4a4a (0 dB) + +This is the gain control for mixing the MIC source into the outputs. +The left and right channel gain are controlled independently. This +gain control has 32 levels, which range from -34.5 dB to +12.0 dB in +1.5 dB steps. Those 32 levels are mapped onto 100 levels at the +ioctl, see below. + +SOUND_MIXER_CD Default: 0x4a4a (0 dB) + +This is the gain control for mixing the CD audio source into the +outputs. The left and right channel gain are controlled +independently. This gain control has 32 levels, which range from +-34.5 dB to +12.0 dB in 1.5 dB steps. Those 32 levels are mapped onto +100 levels at the ioctl, see below. + +SOUND_MIXER_RECLEV Default: 0 (0 dB) + +This is the gain control for PCM input (RECording LEVel). The left +and right channel gain are controlled independently. This gain +control has 16 levels, which range from 0 dB to +22.5 dB in 1.5 dB +steps. Those 16 levels are mapped onto 100 levels at the ioctl, see +below. + +SOUND_MIXER_RECSRC Default: SOUND_MASK_LINE + +This is a mask of currently selected PCM input sources (RECording +SouRCes). Because the AD1843 can only have a single recording source +at a time, only one bit at a time can be set in this mask. The +allowable values are SOUND_MASK_PCM, SOUND_MASK_LINE, SOUND_MASK_MIC, +or SOUND_MASK_CD. Selecting SOUND_MASK_PCM sets up internal +resampling which is useful for loopback testing and for hardware +sample rate conversion. But software sample rate conversion is +probably faster, so I don't know how useful that is. + +SOUND_MIXER_OUTSRC DEFAULT: SOUND_MASK_LINE|SOUND_MASK_MIC|SOUND_MASK_CD + +This is a mask of sources that are currently passed through to the +outputs. Those sources whose bits are not set are muted. + +============================================================================== +GAIN CONTROL + +There are five gain controls listed above. Each has 16, 32, or 64 +steps. Each control has 1.5 dB of gain per step. Each control is +stereo. + +The OSS defines the argument to a channel gain ioctl as having two +components, left and right, each of which ranges from 0 to 100. The +two components are packed into the same word, with the left side gain +in the least significant byte, and the right side gain in the second +least significant byte. In C, we would say this. + + #include <assert.h> + + ... + + assert(leftgain >= 0 && leftgain <= 100); + assert(rightgain >= 0 && rightgain <= 100); + arg = leftgain | rightgain << 8; + +So each OSS gain control has 101 steps. But the hardware has 16, 32, +or 64 steps. The hardware steps are spread across the 101 OSS steps +nearly evenly. The conversion formulas are like this, given N equals +16, 32, or 64. + + int round = N/2 - 1; + OSS_gain_steps = (hw_gain_steps * 100 + round) / (N - 1); + hw_gain_steps = (OSS_gain_steps * (N - 1) + round) / 100; + +Here is a snippet of C code that will return the left and right gain +of any channel in dB. Pass it one of the predefined gain_desc_t +structures to access any of the five channels' gains. + + typedef struct gain_desc { + float min_gain; + float gain_step; + int nbits; + int chan; + } gain_desc_t; + + const gain_desc_t gain_pcm = { -82.5, 1.5, 6, SOUND_MIXER_PCM }; + const gain_desc_t gain_line = { -34.5, 1.5, 5, SOUND_MIXER_LINE }; + const gain_desc_t gain_mic = { -34.5, 1.5, 5, SOUND_MIXER_MIC }; + const gain_desc_t gain_cd = { -34.5, 1.5, 5, SOUND_MIXER_CD }; + const gain_desc_t gain_reclev = { 0.0, 1.5, 4, SOUND_MIXER_RECLEV }; + + int get_gain_dB(int fd, const gain_desc_t *gp, + float *left, float *right) + { + int word; + int lg, rg; + int mask = (1 << gp->nbits) - 1; + + if (ioctl(fd, MIXER_READ(gp->chan), &word) != 0) + return -1; /* fail */ + lg = word & 0xFF; + rg = word >> 8 & 0xFF; + lg = (lg * mask + mask / 2) / 100; + rg = (rg * mask + mask / 2) / 100; + *left = gp->min_gain + gp->gain_step * lg; + *right = gp->min_gain + gp->gain_step * rg; + return 0; + } + +And here is the corresponding routine to set a channel's gain in dB. + + int set_gain_dB(int fd, const gain_desc_t *gp, float left, float right) + { + float max_gain = + gp->min_gain + (1 << gp->nbits) * gp->gain_step; + float round = gp->gain_step / 2; + int mask = (1 << gp->nbits) - 1; + int word; + int lg, rg; + + if (left < gp->min_gain || right < gp->min_gain) + return EINVAL; + lg = (left - gp->min_gain + round) / gp->gain_step; + rg = (right - gp->min_gain + round) / gp->gain_step; + if (lg >= (1 << gp->nbits) || rg >= (1 << gp->nbits)) + return EINVAL; + lg = (100 * lg + mask / 2) / mask; + rg = (100 * rg + mask / 2) / mask; + word = lg | rg << 8; + + return ioctl(fd, MIXER_WRITE(gp->chan), &word); + } + |
