diff options
Diffstat (limited to 'Documentation/sound/oss')
| -rw-r--r-- | Documentation/sound/oss/AD1816 | 84 | ||||
| -rw-r--r-- | Documentation/sound/oss/ALS | 4 | ||||
| -rw-r--r-- | Documentation/sound/oss/AudioExcelDSP16 | 16 | ||||
| -rw-r--r-- | Documentation/sound/oss/CMI8330 | 5 | ||||
| -rw-r--r-- | Documentation/sound/oss/CS4232 | 23 | ||||
| -rw-r--r-- | Documentation/sound/oss/Introduction | 12 | ||||
| -rw-r--r-- | Documentation/sound/oss/NM256 | 280 | ||||
| -rw-r--r-- | Documentation/sound/oss/OPL3-SA2 | 210 | ||||
| -rw-r--r-- | Documentation/sound/oss/Opti | 8 | ||||
| -rw-r--r-- | Documentation/sound/oss/PAS16 | 7 | ||||
| -rw-r--r-- | Documentation/sound/oss/README.OSS | 7 | ||||
| -rw-r--r-- | Documentation/sound/oss/README.modules | 10 | ||||
| -rw-r--r-- | Documentation/sound/oss/README.ymfsb | 2 | ||||
| -rw-r--r-- | Documentation/sound/oss/VIA-chipset | 43 | ||||
| -rw-r--r-- | Documentation/sound/oss/cs46xx | 138 | ||||
| -rw-r--r-- | Documentation/sound/oss/es1371 | 64 | ||||
| -rw-r--r-- | Documentation/sound/oss/mwave | 2 | ||||
| -rw-r--r-- | Documentation/sound/oss/oss-parameters.txt | 51 | ||||
| -rw-r--r-- | Documentation/sound/oss/vwsnd | 293 |
19 files changed, 84 insertions, 1175 deletions
diff --git a/Documentation/sound/oss/AD1816 b/Documentation/sound/oss/AD1816 deleted file mode 100644 index 14bd8f25d52..00000000000 --- a/Documentation/sound/oss/AD1816 +++ /dev/null @@ -1,84 +0,0 @@ -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 index d01ffbfd580..bf10bed4574 100644 --- a/Documentation/sound/oss/ALS +++ b/Documentation/sound/oss/ALS @@ -57,10 +57,10 @@ The resulting sound driver will provide the following capabilities: DSP/PCM/audio out (L&R), FM (L&R) and Mic in (mono). Jonathan Woithe -jwoithe@physics.adelaide.edu.au +jwoithe@just42.net 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 +Modified 2000-11-19 by Jonathan Woithe, jwoithe@just42.net - updated information for kernel 2.4.x. diff --git a/Documentation/sound/oss/AudioExcelDSP16 b/Documentation/sound/oss/AudioExcelDSP16 index c0f08922993..ea8549faede 100644 --- a/Documentation/sound/oss/AudioExcelDSP16 +++ b/Documentation/sound/oss/AudioExcelDSP16 @@ -1,10 +1,10 @@ Driver ------ -Informations about Audio Excel DSP 16 driver can be found in the source +Information 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. +information. Configuration ------------- @@ -41,7 +41,7 @@ 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: +A configuration file in /etc/modprobe.d/ directory will have lines like this: options opl3 io=0x388 options ad1848 io=0x530 irq=11 dma=3 @@ -51,11 +51,11 @@ 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: +the sound card. Installation dependencies must be written in configuration +files under /etc/modprobe.d/ directory: -install ad1848 /sbin/modprobe aedsp16 && /sbin/modprobe -i ad1848 -install opl3 /sbin/modprobe aedsp16 && /sbin/modprobe -i opl3 +softdep ad1848 pre: aedsp16 +softdep opl3 pre: aedsp16 Then you must load the sound modules stack in this order: sound -> aedsp16 -> [ ad1848, opl3 ] @@ -68,7 +68,7 @@ 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. +information 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. diff --git a/Documentation/sound/oss/CMI8330 b/Documentation/sound/oss/CMI8330 index 9c439f1a6db..8a5fd1611c6 100644 --- a/Documentation/sound/oss/CMI8330 +++ b/Documentation/sound/oss/CMI8330 @@ -143,11 +143,10 @@ CONFIG_SOUND_MSS=m -Alma Chao <elysian@ethereal.torsion.org> suggests the following /etc/modprobe.conf: +Alma Chao <elysian@ethereal.torsion.org> suggests the following in +a /etc/modprobe.d/*conf file: 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/CS4232 b/Documentation/sound/oss/CS4232 deleted file mode 100644 index 7d6af7a5c1c..00000000000 --- a/Documentation/sound/oss/CS4232 +++ /dev/null @@ -1,23 +0,0 @@ -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/Introduction b/Documentation/sound/oss/Introduction index f04ba6bb739..42da2d8fa37 100644 --- a/Documentation/sound/oss/Introduction +++ b/Documentation/sound/oss/Introduction @@ -80,7 +80,7 @@ Notes: additional features. 2. The commercial OSS driver may be obtained from the site: - http://www/opensound.com. This may be used for cards that + 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. @@ -167,8 +167,8 @@ 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: +If loading via modprobe, these common files are automatically loaded when +requested by modprobe. For example, my /etc/modprobe.d/oss.conf contains: alias sound sb options sb io=0x240 irq=9 dma=3 dma16=5 mpu_io=0x300 @@ -228,7 +228,7 @@ 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 +2. remove the sound modules from /etc/modprobe.d/*.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 @@ -265,7 +265,7 @@ twice, you need to do the following: 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, +2. Make a second entry in /etc/modprobe.d/*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. @@ -369,7 +369,7 @@ There are several ways of configuring your sound: 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. +3) In /etc/modprobe.d/*conf when using modprobe. 4) Via Red Hat's GPL'd /usr/sbin/sndconfig program (text based). diff --git a/Documentation/sound/oss/NM256 b/Documentation/sound/oss/NM256 deleted file mode 100644 index b503217488b..00000000000 --- a/Documentation/sound/oss/NM256 +++ /dev/null @@ -1,280 +0,0 @@ -======================================================= -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-SA2 b/Documentation/sound/oss/OPL3-SA2 deleted file mode 100644 index d8b6d2bbada..00000000000 --- a/Documentation/sound/oss/OPL3-SA2 +++ /dev/null @@ -1,210 +0,0 @@ -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 index c15af3c07d4..4cd5d9ab358 100644 --- a/Documentation/sound/oss/Opti +++ b/Documentation/sound/oss/Opti @@ -18,7 +18,7 @@ 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 +Also, it is recommended that resources specified in /etc/modprobe.d/*.conf and resources specified in /etc/isapnp.conf agree. Compiling the sound driver @@ -67,11 +67,7 @@ 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. +Config files in '/etc/modprobe.d/' are used as below: alias mixer0 mad16 alias audio0 mad16 diff --git a/Documentation/sound/oss/PAS16 b/Documentation/sound/oss/PAS16 index 951b3dce51b..5c27229eec8 100644 --- a/Documentation/sound/oss/PAS16 +++ b/Documentation/sound/oss/PAS16 @@ -60,8 +60,7 @@ With PAS16 you can use two audio device files at the same time. /dev/dsp (and The new stuff for 2.3.99 and later ============================================================================ -The following configuration options from Documentation/Configure.help -are relevant to configuring the PAS16: +The following configuration options are relevant to configuring the PAS16: Sound card support CONFIG_SOUND @@ -129,7 +128,7 @@ CONFIG_SOUND_YM3812 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: + /etc/modprobe.d/*.conf: options opl3 io=0x388 or else add the following line to /etc/lilo.conf: opl3=0x388 @@ -159,5 +158,5 @@ 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 +specified in /etc/modprobe.d/*.conf for pas2, sb and opl3 respectively. diff --git a/Documentation/sound/oss/README.OSS b/Documentation/sound/oss/README.OSS index fd42b05b2f5..4be259428a1 100644 --- a/Documentation/sound/oss/README.OSS +++ b/Documentation/sound/oss/README.OSS @@ -36,7 +36,7 @@ with OSS API. 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 +See http://www.opensound.com/ossfree/ for download instructions. If you are looking for the installation instructions, please @@ -1352,7 +1352,7 @@ 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 +(limited by the OSS-design). Development 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 @@ -1438,7 +1438,7 @@ 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) +at the home page (http://www.opensound.com/ossfree/) for latest info. Information about unsupported sound cards and chipsets is welcome as well @@ -1449,7 +1449,6 @@ 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 diff --git a/Documentation/sound/oss/README.modules b/Documentation/sound/oss/README.modules index e691d74e1e5..cdc039421a4 100644 --- a/Documentation/sound/oss/README.modules +++ b/Documentation/sound/oss/README.modules @@ -26,7 +26,7 @@ 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: + Then, add to your /etc/modprobe.d/oss.conf something like: alias char-major-14-* sb install sb /sbin/modprobe -i sb && /sbin/modprobe adlib_card @@ -36,7 +36,7 @@ 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" +softdep sb post: adlib_card options adlib_card io=0x388 The effect of this is that the sound driver and all necessary bits and @@ -66,12 +66,12 @@ 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. +with /etc/modprobe.d/*.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. + recommend using /etc/modprobe.d/*.conf. Persistent DMA Buffers: @@ -89,7 +89,7 @@ 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: +in /etc/modprobe.d/*.conf files like so: options sound dmabuf=1 diff --git a/Documentation/sound/oss/README.ymfsb b/Documentation/sound/oss/README.ymfsb index af8a7d3a4e8..b6b77906b58 100644 --- a/Documentation/sound/oss/README.ymfsb +++ b/Documentation/sound/oss/README.ymfsb @@ -5,7 +5,7 @@ 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 + I respect and thank for all people they made open the information about YMF7xx cards. And this codes heavily based on Jeff Garzik <jgarzik@pobox.com>'s diff --git a/Documentation/sound/oss/VIA-chipset b/Documentation/sound/oss/VIA-chipset deleted file mode 100644 index 37865234e54..00000000000 --- a/Documentation/sound/oss/VIA-chipset +++ /dev/null @@ -1,43 +0,0 @@ -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/cs46xx b/Documentation/sound/oss/cs46xx deleted file mode 100644 index b5443270986..00000000000 --- a/Documentation/sound/oss/cs46xx +++ /dev/null @@ -1,138 +0,0 @@ - -Documentation for the Cirrus Logic/Crystal SoundFusion cs46xx/cs4280 audio -controller chips (2001/05/11) - -The cs46xx audio driver supports the DSP line of Cirrus controllers. -Specifically, the cs4610, cs4612, cs4614, cs4622, cs4624, cs4630 and the cs4280 -products. This driver uses the generic ac97_codec driver for AC97 codec -support. - - -Features: - -Full Duplex Playback/Capture supported from 8k-48k. -16Bit Signed LE & 8Bit Unsigned, with Mono or Stereo supported. - -APM/PM - 2.2.x PM is enabled and functional. APM can also -be enabled for 2.4.x by modifying the CS46XX_ACPI_SUPPORT macro -definition. - -DMA playback buffer size is configurable from 16k (defaultorder=2) up to 2Meg -(defaultorder=11). DMA capture buffer size is fixed at a single 4k page as -two 2k fragments. - -MMAP seems to work well with QuakeIII, and test XMMS plugin. - -Myth2 works, but the polling logic is not fully correct, but is functional. - -The 2.4.4-ac6 gameport code in the cs461x joystick driver has been tested -with a Microsoft Sidewinder joystick (cs461x.o and sidewinder.o). This -audio driver must be loaded prior to the joystick driver to enable the -DSP task image supporting the joystick device. - - -Limitations: - -SPDIF is currently not supported. - -Primary codec support only. No secondary codec support is implemented. - - - -NOTES: - -Hercules Game Theatre XP - the EGPIO2 pin controls the external Amp, -and has been tested. -Module parameter hercules_egpio_disable set to 1, will force a 0 to EGPIODR -to disable the external amplifier. - -VTB Santa Cruz - the GPIO7/GPIO8 on the Secondary Codec control -the external amplifier for the "back" speakers, since we do not -support the secondary codec then this external amp is not -turned on. The primary codec external amplifier is supported but -note that the AC97 EAPD bit is inverted logic (amp_voyetra()). - -DMA buffer size - there are issues with many of the Linux applications -concerning the optimal buffer size. Several applications request a -certain fragment size and number and then do not verify that the driver -has the ability to support the requested configuration. -SNDCTL_DSP_SETFRAGMENT ioctl is used to request a fragment size and -number of fragments. Some applications exit if an error is returned -on this particular ioctl. Therefore, in alignment with the other OSS audio -drivers, no error is returned when a SETFRAGs IOCTL is received, but the -values passed from the app are not used in any buffer calculation -(ossfragshift/ossmaxfrags are not used). -Use the "defaultorder=N" module parameter to change the buffer size if -you have an application that requires a specific number of fragments -or a specific buffer size (see below). - -Debug Interface ---------------- -There is an ioctl debug interface to allow runtime modification of the -debug print levels. This debug interface code can be disabled from the -compilation process with commenting the following define: -#define CSDEBUG_INTERFACE 1 -There is also a debug print methodolgy to select printf statements from -different areas of the driver. A debug print level is also used to allow -additional printfs to be active. Comment out the following line in the -driver to disable compilation of the CS_DBGOUT print statements: -#define CSDEBUG 1 - -Please see the 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_param(defaultorder, ulong, 0); -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_param(cs_debuglevel, ulong, 0644); -module_param(cs_debugmask, ulong, 0644); -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_param(hercules_egpio_disable, ulong, 0); -hercules_egpio_disable=N -where N is a 0 (enable egpio), or a 1 (disable egpio support) - -module_param(initdelay, ulong, 0); -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_param(powerdown, ulong, 0); -powerdown=N -where N is 0 (disable any powerdown of the internal blocks) or 1 (enable powerdown) - - -module_param(external_amp, bool, 0); -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_param(thinkpad, bool, 0); -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/es1371 b/Documentation/sound/oss/es1371 deleted file mode 100644 index c3151266771..00000000000 --- a/Documentation/sound/oss/es1371 +++ /dev/null @@ -1,64 +0,0 @@ -/proc/sound, /dev/sndstat -------------------------- - -/proc/sound and /dev/sndstat is not supported by the -driver. To find out whether the driver succeeded loading, -check the kernel log (dmesg). - - -ALaw/uLaw sample formats ------------------------- - -This driver does not support the ALaw/uLaw sample formats. -ALaw is the default mode when opening a sound device -using OSS/Free. The reason for the lack of support is -that the hardware does not support these formats, and adding -conversion routines to the kernel would lead to very ugly -code in the presence of the mmap interface to the driver. -And since xquake uses mmap, mmap is considered important :-) -and no sane application uses ALaw/uLaw these days anyway. -In short, playing a Sun .au file as follows: - -cat my_file.au > /dev/dsp - -does not work. Instead, you may use the play script from -Chris Bagwell's sox-12.14 package (available from the URL -below) to play many different audio file formats. -The script automatically determines the audio format -and does do audio conversions if necessary. -http://home.sprynet.com/sprynet/cbagwell/projects.html - - -Blocking vs. nonblocking IO ---------------------------- - -Unlike OSS/Free this driver honours the O_NONBLOCK file flag -not only during open, but also during read and write. -This is an effort to make the sound driver interface more -regular. Timidity has problems with this; a patch -is available from http://www.ife.ee.ethz.ch/~sailer/linux/pciaudio.html. -(Timidity patched will also run on OSS/Free). - - -MIDI UART ---------- - -The driver supports a simple MIDI UART interface, with -no ioctl's supported. - - -MIDI synthesizer ----------------- - -This soundcard does not have any hardware MIDI synthesizer; -MIDI synthesis has to be done in software. To allow this -the driver/soundcard supports two PCM (/dev/dsp) interfaces. - -There is a freely available software package that allows -MIDI file playback on this soundcard called Timidity. -See http://www.cgs.fi/~tt/timidity/. - - - -Thomas Sailer -t.sailer@alumni.ethz.ch diff --git a/Documentation/sound/oss/mwave b/Documentation/sound/oss/mwave index 858334bb46b..5fbcb160927 100644 --- a/Documentation/sound/oss/mwave +++ b/Documentation/sound/oss/mwave @@ -163,7 +163,7 @@ 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 voilą - ThinkPad sound with Linux. +the sound modules and voilĆ - 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 diff --git a/Documentation/sound/oss/oss-parameters.txt b/Documentation/sound/oss/oss-parameters.txt new file mode 100644 index 00000000000..3ab391e7c29 --- /dev/null +++ b/Documentation/sound/oss/oss-parameters.txt @@ -0,0 +1,51 @@ + OSS Kernel Parameters + ~~~~~~~~~~~~~~~~~~~~~ + +See Documentation/kernel-parameters.txt for general information on +specifying module parameters. + +This document may not be entirely up to date and comprehensive. The command +"modinfo -p ${modulename}" shows a current list of all parameters of a loadable +module. Loadable modules, after being loaded into the running kernel, also +reveal their parameters in /sys/module/${modulename}/parameters/. Some of these +parameters may be changed at runtime by the command +"echo -n ${value} > /sys/module/${modulename}/parameters/${parm}". + + + ad1848= [HW,OSS] + Format: <io>,<irq>,<dma>,<dma2>,<type> + + aedsp16= [HW,OSS] Audio Excel DSP 16 + Format: <io>,<irq>,<dma>,<mss_io>,<mpu_io>,<mpu_irq> + See also header of sound/oss/aedsp16.c. + + dmasound= [HW,OSS] Sound subsystem buffers + + mpu401= [HW,OSS] + Format: <io>,<irq> + + opl3= [HW,OSS] + Format: <io> + + pas2= [HW,OSS] Format: + <io>,<irq>,<dma>,<dma16>,<sb_io>,<sb_irq>,<sb_dma>,<sb_dma16> + + pss= [HW,OSS] Personal Sound System (ECHO ESC614) + Format: + <io>,<mss_io>,<mss_irq>,<mss_dma>,<mpu_io>,<mpu_irq> + + sscape= [HW,OSS] + Format: <io>,<irq>,<dma>,<mpu_io>,<mpu_irq> + + trix= [HW,OSS] MediaTrix AudioTrix Pro + Format: + <io>,<irq>,<dma>,<dma2>,<sb_io>,<sb_irq>,<sb_dma>,<mpu_io>,<mpu_irq> + + uart401= [HW,OSS] + Format: <io>,<irq> + + uart6850= [HW,OSS] + Format: <io>,<irq> + + waveartist= [HW,OSS] + Format: <io>,<irq>,<dma>,<dma2> diff --git a/Documentation/sound/oss/vwsnd b/Documentation/sound/oss/vwsnd deleted file mode 100644 index 4c6cbdb3c54..00000000000 --- a/Documentation/sound/oss/vwsnd +++ /dev/null @@ -1,293 +0,0 @@ -vwsnd - Sound driver for the Silicon Graphics 320 and 540 Visual -Workstations' onboard audio. - -Copyright 1999 Silicon Graphics, Inc. All rights reserved. - - -At the time of this writing, March 1999, there are two models of -Visual Workstation, the 320 and the 540. This document only describes -those models. Future Visual Workstation models may have different -sound capabilities, and this driver will probably not work on those -boxes. - -The Visual Workstation has an Analog Devices AD1843 "SoundComm" audio -codec chip. The AD1843 is accessed through the Cobalt I/O ASIC, also -known as Lithium. This driver programs both 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); - } - |