diff options
Diffstat (limited to 'Documentation/sound')
| -rw-r--r-- | Documentation/sound/alsa/ALSA-Configuration.txt | 75 | ||||
| -rw-r--r-- | Documentation/sound/alsa/Audiophile-Usb.txt | 242 | ||||
| -rw-r--r-- | Documentation/sound/alsa/OSS-Emulation.txt | 15 | ||||
| -rw-r--r-- | Documentation/sound/oss/AD1816 | 84 | ||||
| -rw-r--r-- | Documentation/sound/oss/NM256 | 280 | ||||
| -rw-r--r-- | Documentation/sound/oss/OPL3-SA2 | 210 | ||||
| -rw-r--r-- | Documentation/sound/oss/VIA-chipset | 43 | ||||
| -rw-r--r-- | Documentation/sound/oss/cs46xx | 138 |
8 files changed, 246 insertions, 841 deletions
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 355ff0a2bb7..241e26c4ff9 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -467,7 +467,12 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. above explicitly. The power-management is supported. - + + Module snd-cs5530 + _________________ + + Module for Cyrix/NatSemi Geode 5530 chip. + Module snd-cs5535audio ---------------------- @@ -759,6 +764,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. model - force the model name position_fix - Fix DMA pointer (0 = auto, 1 = none, 2 = POSBUF, 3 = FIFO size) + probe_mask - Bitmask to probe codecs (default = -1, meaning all slots) single_cmd - Use single immediate commands to communicate with codecs (for debugging only) enable_msi - Enable Message Signaled Interrupt (MSI) (default = off) @@ -803,6 +809,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. hp-3013 HP machines (3013-variant) fujitsu Fujitsu S7020 acer Acer TravelMate + will Will laptops (PB V7900) + replacer Replacer 672V basic fixed pin assignment (old default model) auto auto-config reading BIOS (default) @@ -811,16 +819,31 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. hp-bpc HP xw4400/6400/8400/9400 laptops hp-bpc-d7000 HP BPC D7000 benq Benq ED8 + benq-t31 Benq T31 hippo Hippo (ATI) with jack detection, Sony UX-90s hippo_1 Hippo (Benq) with jack detection + sony-assamd Sony ASSAMD basic fixed pin assignment w/o SPDIF auto auto-config reading BIOS (default) + ALC268 + 3stack 3-stack model + auto auto-config reading BIOS (default) + + ALC662 + 3stack-dig 3-stack (2-channel) with SPDIF + 3stack-6ch 3-stack (6-channel) + 3stack-6ch-dig 3-stack (6-channel) with SPDIF + 6stack-dig 6-stack with SPDIF + lenovo-101e Lenovo laptop + auto auto-config reading BIOS (default) + ALC882/885 3stack-dig 3-jack with SPDIF I/O 6stack-dig 6-jack digital with SPDIF I/O arima Arima W820Di1 macpro MacPro support + imac24 iMac 24'' with jack detection w2jc ASUS W2JC auto auto-config reading BIOS (default) @@ -832,9 +855,15 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 6stack-dig-demo 6-jack digital for Intel demo board acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc) medion Medion Laptops + medion-md2 Medion MD2 targa-dig Targa/MSI targa-2ch-dig Targs/MSI with 2-channel laptop-eapd 3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE) + lenovo-101e Lenovo 101E + lenovo-nb0763 Lenovo NB0763 + lenovo-ms7195-dig Lenovo MS7195 + 6stack-hp HP machines with 6stack (Nettle boards) + 3stack-hp HP machines with 3stack (Lucknow, Samba boards) auto auto-config reading BIOS (default) ALC861/660 @@ -853,7 +882,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 3stack-dig 3-jack with SPDIF OUT 6stack-dig 6-jack with SPDIF OUT 3stack-660 3-jack (for ALC660VD) + 3stack-660-digout 3-jack with SPDIF OUT (for ALC660VD) lenovo Lenovo 3000 C200 + dallas Dallas laptops auto auto-config reading BIOS (default) CMI9880 @@ -864,12 +895,26 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. allout 5-jack in back, 2-jack in front, SPDIF out auto auto-config reading BIOS (default) + AD1882 + 3stack 3-stack mode (default) + 6stack 6-stack mode + + AD1884 + N/A + AD1981 basic 3-jack (default) hp HP nx6320 thinkpad Lenovo Thinkpad T60/X60/Z60 toshiba Toshiba U205 + AD1983 + N/A + + AD1984 + basic default configuration + thinkpad Lenovo Thinkpad T61/X61 + AD1986A 6stack 6-jack, separate surrounds (default) 3stack 3-stack, shared surrounds @@ -907,11 +952,18 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. ref Reference board 3stack D945 3stack 5stack D945 5stack + SPDIF - macmini Intel Mac Mini - macbook Intel Mac Book - macbook-pro-v1 Intel Mac Book Pro 1st generation - macbook-pro Intel Mac Book Pro 2nd generation - imac-intel Intel iMac + dell Dell XPS M1210 + intel-mac-v1 Intel Mac Type 1 + intel-mac-v2 Intel Mac Type 2 + intel-mac-v3 Intel Mac Type 3 + intel-mac-v4 Intel Mac Type 4 + intel-mac-v5 Intel Mac Type 5 + macmini Intel Mac Mini (equivalent with type 3) + macbook Intel Mac Book (eq. type 5) + macbook-pro-v1 Intel Mac Book Pro 1st generation (eq. type 3) + macbook-pro Intel Mac Book Pro 2nd generation (eq. type 3) + imac-intel Intel iMac (eq. type 2) + imac-intel-20 Intel iMac (newer version) (eq. type 3) STAC9202/9250/9251 ref Reference board, base config @@ -956,6 +1008,17 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. from the irq. Remember this is a last resort, and should be avoided as much as possible... + MORE NOTES ON "azx_get_response timeout" PROBLEMS: + On some hardwares, you may need to add a proper probe_mask option + to avoid the "azx_get_response timeout" problem above, instead. + This occurs when the access to non-existing or non-working codec slot + (likely a modem one) causes a stall of the communication via HD-audio + bus. You can see which codec slots are probed by enabling + CONFIG_SND_DEBUG_DETECT, or simply from the file name of the codec + proc files. Then limit the slots to probe by probe_mask option. + For example, probe_mask=1 means to probe only the first slot, and + probe_mask=4 means only the third slot. + The power-management is supported. Module snd-hdsp diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt index e40cce83327..2ad5e6306c4 100644 --- a/Documentation/sound/alsa/Audiophile-Usb.txt +++ b/Documentation/sound/alsa/Audiophile-Usb.txt @@ -1,4 +1,4 @@ - Guide to using M-Audio Audiophile USB with ALSA and Jack v1.3 + Guide to using M-Audio Audiophile USB with ALSA and Jack v1.5 ======================================================== Thibault Le Meur <Thibault.LeMeur@supelec.fr> @@ -6,8 +6,19 @@ This document is a guide to using the M-Audio Audiophile USB (tm) device with ALSA and JACK. +History +======= +* v1.4 - Thibault Le Meur (2007-07-11) + - Added Low Endianness nature of 16bits-modes + found by Hakan Lennestal <Hakan.Lennestal@brfsodrahamn.se> + - Modifying document structure +* v1.5 - Thibault Le Meur (2007-07-12) + - Added AC3/DTS passthru info + + 1 - Audiophile USB Specs and correct usage ========================================== + This part is a reminder of important facts about the functions and limitations of the device. @@ -25,18 +36,18 @@ The device has 4 audio interfaces, and 2 MIDI ports: The internal DAC/ADC has the following characteristics: * sample depth of 16 or 24 bits * sample rate from 8kHz to 96kHz -* Two ports can't use different sample depths at the same time. Moreover, the -Audiophile USB documentation gives the following Warning: "Please exit any -audio application running before switching between bit depths" +* Two interfaces can't use different sample depths at the same time. +Moreover, the Audiophile USB documentation gives the following Warning: +"Please exit any audio application running before switching between bit depths" Due to the USB 1.1 bandwidth limitation, a limited number of interfaces can be activated at the same time depending on the audio mode selected: - * 16-bit/48kHz ==> 4 channels in/4 channels out + * 16-bit/48kHz ==> 4 channels in + 4 channels out - Ai+Ao+Di+Do - * 24-bit/48kHz ==> 4 channels in/2 channels out, - or 2 channels in/4 channels out + * 24-bit/48kHz ==> 4 channels in + 2 channels out, + or 2 channels in + 4 channels out - Ai+Ao+Do or Ai+Di+Ao or Ai+Di+Do or Di+Ao+Do - * 24-bit/96kHz ==> 2 channels in, or 2 channels out (half duplex only) + * 24-bit/96kHz ==> 2 channels in _or_ 2 channels out (half duplex only) - Ai or Ao or Di or Do Important facts about the Digital interface: @@ -52,44 +63,56 @@ source is connected synchronization error (for instance sound played at an odd sample rate) -2 - Audiophile USB support in ALSA -================================== +2 - Audiophile USB MIDI support in ALSA +======================================= -2.1 - MIDI ports ----------------- -The Audiophile USB MIDI ports will be automatically supported once the +The Audiophile USB MIDI ports will be automatically supported once the following modules have been loaded: * snd-usb-audio * snd-seq-midi No additional setting is required. -2.2 - Audio ports ------------------ + +3 - Audiophile USB Audio support in ALSA +======================================== Audio functions of the Audiophile USB device are handled by the snd-usb-audio module. This module can work in a default mode (without any device-specific parameter), or in an "advanced" mode with the device-specific parameter called "device_setup". -2.2.1 - Default Alsa driver mode - -The default behavior of the snd-usb-audio driver is to parse the device -capabilities at startup and enable all functions inside the device (including -all ports at any supported sample rates and sample depths). This approach -has the advantage to let the driver easily switch from sample rates/depths -automatically according to the need of the application claiming the device. - -In this case the Audiophile ports are mapped to alsa pcm devices in the -following way (I suppose the device's index is 1): +3.1 - Default Alsa driver mode +------------------------------ + +The default behavior of the snd-usb-audio driver is to list the device +capabilities at startup and activate the required mode when required +by the applications: for instance if the user is recording in a +24bit-depth-mode and immediately after wants to switch to a 16bit-depth mode, +the snd-usb-audio module will reconfigure the device on the fly. + +This approach has the advantage to let the driver automatically switch from sample +rates/depths automatically according to the user's needs. However, those who +are using the device under windows know that this is not how the device is meant to +work: under windows applications must be closed before using the m-audio control +panel to switch the device working mode. Thus as we'll see in next section, this +Default Alsa driver mode can lead to device misconfigurations. + +Let's get back to the Default Alsa driver mode for now. In this case the +Audiophile interfaces are mapped to alsa pcm devices in the following +way (I suppose the device's index is 1): * hw:1,0 is Ao in playback and Di in capture * hw:1,1 is Do in playback and Ai in capture * hw:1,2 is Do in AC3/DTS passthrough mode -You must note as well that the device uses Big Endian byte encoding so that -supported audio format are S16_BE for 16-bit depth modes and S24_3BE for -24-bits depth mode. One exception is the hw:1,2 port which is Little Endian -compliant and thus uses S16_LE. +In this mode, the device uses Big Endian byte-encoding so that +supported audio format are S16_BE for 16-bit depth modes and S24_3BE for +24-bits depth mode. + +One exception is the hw:1,2 port which was reported to be Little Endian +compliant (supposedly supporting S16_LE) but processes in fact only S16_BE streams. +This has been fixed in kernel 2.6.23 and above and now the hw:1,2 interface +is reported to be big endian in this default driver mode. Examples: * playing a S24_3BE encoded raw file to the Ao port @@ -98,22 +121,26 @@ Examples: % arecord -D hw:1,1 -c2 -t raw -r48000 -fS24_3BE test.raw * playing a S16_BE encoded raw file to the Do port % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test.raw + * playing an ac3 sample file to the Do port + % aplay -D hw:1,2 --channels=6 ac3_S16_BE_encoded_file.raw -If you're happy with the default Alsa driver setup and don't experience any +If you're happy with the default Alsa driver mode and don't experience any issue with this mode, then you can skip the following chapter. -2.2.2 - Advanced module setup +3.2 - Advanced module setup +--------------------------- Due to the hardware constraints described above, the device initialization made by the Alsa driver in default mode may result in a corrupted state of the device. For instance, a particularly annoying issue is that the sound captured -from the Ai port sounds distorted (as if boosted with an excessive high volume -gain). +from the Ai interface sounds distorted (as if boosted with an excessive high +volume gain). For people having this problem, the snd-usb-audio module has a new module -parameter called "device_setup". +parameter called "device_setup" (this parameter was introduced in kernel +release 2.6.17) -2.2.2.1 - Initializing the working mode of the Audiophile USB +3.2.1 - Initializing the working mode of the Audiophile USB As far as the Audiophile USB device is concerned, this value let the user specify: @@ -121,33 +148,57 @@ specify: * the sample rate * whether the Di port is used or not -Here is a list of supported device_setup values for this device: - * device_setup=0x00 (or omitted) - - Alsa driver default mode - - maintains backward compatibility with setups that do not use this - parameter by not introducing any change - - results sometimes in corrupted sound as described earlier +When initialized with "device_setup=0x00", the snd-usb-audio module has +the same behaviour as when the parameter is omitted (see paragraph "Default +Alsa driver mode" above) + +Others modes are described in the following subsections. + +3.2.1.1 - 16-bit modes + +The two supported modes are: + * device_setup=0x01 - 16bits 48kHz mode with Di disabled - Ai,Ao,Do can be used at the same time - hw:1,0 is not available in capture mode - hw:1,2 is not available + * device_setup=0x11 - 16bits 48kHz mode with Di enabled - Ai,Ao,Di,Do can be used at the same time - hw:1,0 is available in capture mode - hw:1,2 is not available + +In this modes the device operates only at 16bits-modes. Before kernel 2.6.23, +the devices where reported to be Big-Endian when in fact they were Little-Endian +so that playing a file was a matter of using: + % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test_S16_LE.raw +where "test_S16_LE.raw" was in fact a little-endian sample file. + +Thanks to Hakan Lennestal (who discovered the Little-Endiannes of the device in +these modes) a fix has been committed (expected in kernel 2.6.23) and +Alsa now reports Little-Endian interfaces. Thus playing a file now is as simple as +using: + % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_LE test_S16_LE.raw + +3.2.1.2 - 24-bit modes + +The three supported modes are: + * device_setup=0x09 - 24bits 48kHz mode with Di disabled - Ai,Ao,Do can be used at the same time - hw:1,0 is not available in capture mode - hw:1,2 is not available + * device_setup=0x19 - 24bits 48kHz mode with Di enabled - 3 ports from {Ai,Ao,Di,Do} can be used at the same time - hw:1,0 is available in capture mode and an active digital source must be connected to Di - hw:1,2 is not available + * device_setup=0x0D or 0x10 - 24bits 96kHz mode - Di is enabled by default for this mode but does not need to be connected @@ -155,34 +206,64 @@ Here is a list of supported device_setup values for this device: - Only 1 port from {Ai,Ao,Di,Do} can be used at the same time - hw:1,0 is available in captured mode - hw:1,2 is not available + +In these modes the device is only Big-Endian compliant (see "Default Alsa driver +mode" above for an aplay command example) + +3.2.1.3 - AC3 w/ DTS passthru mode + +Thanks to Hakan Lennestal, I now have a report saying that this mode works. + * device_setup=0x03 - 16bits 48kHz mode with only the Do port enabled - - AC3 with DTS passthru (not tested) + - AC3 with DTS passthru - Caution with this setup the Do port is mapped to the pcm device hw:1,0 -2.2.2.2 - Setting and switching configurations with the device_setup parameter +The command line used to playback the AC3/DTS encoded .wav-files in this mode: + % aplay -D hw:1,0 --channels=6 ac3_S16_LE_encoded_file.raw + +3.2.2 - How to use the device_setup parameter +---------------------------------------------- The parameter can be given: + * By manually probing the device (as root): # modprobe -r snd-usb-audio # modprobe snd-usb-audio index=1 device_setup=0x09 + * Or while configuring the modules options in your modules configuration file - For Fedora distributions, edit the /etc/modprobe.conf file: alias snd-card-1 snd-usb-audio options snd-usb-audio index=1 device_setup=0x09 -IMPORTANT NOTE WHEN SWITCHING CONFIGURATION: -------------------------------------------- - * You may need to _first_ initialize the module with the correct device_setup - parameter and _only_after_ turn on the Audiophile USB device - * This is especially true when switching the sample depth: +CAUTION when initializaing the device +------------------------------------- + + * Correct initialization on the device requires that device_setup is given to + the module BEFORE the device is turned on. So, if you use the "manual probing" + method described above, take care to power-on the device AFTER this initialization. + + * Failing to respect this will lead in a misconfiguration of the device. In this case + turn off the device, unproble the snd-usb-audio module, then probe it again with + correct device_setup parameter and then (and only then) turn on the device again. + + * If you've correctly initialized the device in a valid mode and then want to switch + to another mode (possibly with another sample-depth), please use also the following + procedure: - first turn off the device - de-register the snd-usb-audio module (modprobe -r) - change the device_setup parameter by changing the device_setup option in /etc/modprobe.conf - turn on the device + * A workaround for this last issue has been applied to kernel 2.6.23, but it may not + be enough to ensure the 'stability' of the device initialization. -2.2.2.3 - Audiophile USB's device_setup structure +3.2.3 - Technical details for hackers +------------------------------------- +This section is for hackers, wanting to understand details about the device +internals and how Alsa supports it. + +3.2.3.1 - Audiophile USB's device_setup structure If you want to understand the device_setup magic numbers for the Audiophile USB, you need some very basic understanding of binary computation. However, @@ -228,12 +309,12 @@ Caution: - choosing b2 will prepare all interfaces for 24bits/96kHz but you'll only be able to use one at the same time -2.2.3 - USB implementation details for this device +3.2.3.2 - USB implementation details for this device You may safely skip this section if you're not interested in driver -development. +hacking. -This section describes some internal aspects of the device and summarize the +This section describes some internal aspects of the device and summarizes the data I got by usb-snooping the windows and Linux drivers. The M-Audio Audiophile USB has 7 USB Interfaces: @@ -293,43 +374,45 @@ parse_audio_endpoints function uses a quirk called "audiophile_skip_setting_quirk" in order to prevent AltSettings not corresponding to device_setup from being registered in the driver. -3 - Audiophile USB and Jack support +4 - Audiophile USB and Jack support =================================== This section deals with support of the Audiophile USB device in Jack. -The main issue regarding this support is that the device is Big Endian -compliant. -3.1 - Using the plug alsa plugin --------------------------------- +There are 2 main potential issues when using Jackd with the device: +* support for Big-Endian devices in 24-bit modes +* support for 4-in / 4-out channels + +4.1 - Direct support in Jackd +----------------------------- -Jack doesn't directly support big endian devices. Thus, one way to have support -for this device with Alsa is to use the Alsa "plug" converter. +Jack supports big endian devices only in recent versions (thanks to +Andreas Steinmetz for his first big-endian patch). I can't remember +extacly when this support was released into jackd, let's just say that +with jackd version 0.103.0 it's almost ok (just a small bug is affecting +16bits Big-Endian devices, but since you've read carefully the above +paragraphs, you're now using kernel >= 2.6.23 and your 16bits devices +are now Little Endians ;-) ). + +You can run jackd with the following command for playback with Ao and +record with Ai: + % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1 + +4.2 - Using Alsa plughw +----------------------- +If you don't have a recent Jackd installed, you can downgrade to using +the Alsa "plug" converter. For instance here is one way to run Jack with 2 playback channels on Ao and 2 capture channels from Ai: % jackd -R -dalsa -dplughw:1 -r48000 -p256 -n2 -D -Cplughw:1,1 - However you may see the following warning message: "You appear to be using the ALSA software "plug" layer, probably a result of using the "default" ALSA device. This is less efficient than it could be. Consider using a hardware device instead rather than using the plug layer." -3.2 - Patching alsa to use direct pcm device --------------------------------------------- -A patch for Jack by Andreas Steinmetz adds support for Big Endian devices. -However it has not been included in the CVS tree. - -You can find it at the following URL: -http://sourceforge.net/tracker/index.php?func=detail&aid=1289682&group_id=39687& -atid=425939 - -After having applied the patch you can run jackd with the following command -line: - % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1 - -3.2 - Getting 2 input and/or output interfaces in Jack +4.3 - Getting 2 input and/or output interfaces in Jack ------------------------------------------------------ As you can see, starting the Jack server this way will only enable 1 stereo @@ -339,6 +422,7 @@ This is due to the following restrictions: * Jack can only open one capture device and one playback device at a time * The Audiophile USB is seen as 2 (or three) Alsa devices: hw:1,0, hw:1,1 (and optionally hw:1,2) + If you want to get Ai+Di and/or Ao+Do support with Jack, you would need to combine the Alsa devices into one logical "complex" device. @@ -348,13 +432,11 @@ It is related to another device (ice1712) but can be adapted to suit the Audiophile USB. Enabling multiple Audiophile USB interfaces for Jackd will certainly require: -* patching Jack with the previously mentioned "Big Endian" patch -* patching Jackd with the MMAP_COMPLEX patch (see the ice1712 page) -* patching the alsa-lib/src/pcm/pcm_multi.c file (see the ice1712 page) +* Making sure your Jackd version has the MMAP_COMPLEX patch (see the ice1712 page) +* (maybe) patching the alsa-lib/src/pcm/pcm_multi.c file (see the ice1712 page) * define a multi device (combination of hw:1,0 and hw:1,1) in your .asoundrc file * start jackd with this device -I had no success in testing this for now, but this may be due to my OS -configuration. If you have any success with this kind of setup, please -drop me an email. +I had no success in testing this for now, if you have any success with this kind +of setup, please drop me an email. diff --git a/Documentation/sound/alsa/OSS-Emulation.txt b/Documentation/sound/alsa/OSS-Emulation.txt index ec2a02541d5..bfa0c9aacb4 100644 --- a/Documentation/sound/alsa/OSS-Emulation.txt +++ b/Documentation/sound/alsa/OSS-Emulation.txt @@ -278,6 +278,21 @@ current mixer configuration by reading and writing the whole file image. +Duplex Streams +============== + +Note that when attempting to use a single device file for playback and +capture, the OSS API provides no way to set the format, sample rate or +number of channels different in each direction. Thus + io_handle = open("device", O_RDWR) +will only function correctly if the values are the same in each direction. + +To use different values in the two directions, use both + input_handle = open("device", O_RDONLY) + output_handle = open("device", O_WRONLY) +and set the values for the corresponding handle. + + Unsupported Features ==================== 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/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/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. - |