Merge branch 'master'

5 files changed, 414 insertions, 16 deletions

diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 36b511c7cade..1def6049784c 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -513,6 +513,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     This module supports multiple cards and autoprobe.
     
+    The power-management is supported.
+
   Module snd-ens1371
   ------------------
 
@@ -526,6 +528,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     This module supports multiple cards and autoprobe.
     
+    The power-management is supported.
+
   Module snd-es968
   ----------------
 
@@ -671,6 +675,8 @@ 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)
+    single_cmd  - Use single immediate commands to communicate with
+                codecs (for debugging only)
 
     This module supports one card and autoprobe.
 
@@ -694,13 +700,34 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
           asus          3-jack
           uniwill       3-jack
           F1734         2-jack
+          lg            LG laptop (m1 express dual)
           test          for testing/debugging purpose, almost all controls can be
                         adjusted.  Appearing only when compiled with
                         $CONFIG_SND_DEBUG=y
+          auto          auto-config reading BIOS (default)
 
         ALC260
           hp            HP machines
           fujitsu       Fujitsu S7020
+          acer          Acer TravelMate
+          basic         fixed pin assignment (old default model)
+          auto          auto-config reading BIOS (default)
+
+        ALC262
+          fujitsu       Fujitsu Laptop
+          basic         fixed pin assignment w/o SPDIF
+          auto          auto-config reading BIOS (default)
+
+        ALC882/883/885
+          3stack-dig    3-jack with SPDIF I/O
+          6stck-dig     6-jack digital with SPDIF I/O
+          auto          auto-config reading BIOS (default)
+
+        ALC861
+          3stack        3-jack
+          3stack-dig    3-jack with SPDIF I/O
+          6stack-dig    6-jack with SPDIF I/O
+          auto          auto-config reading BIOS (default)
 
         CMI9880
           minimal       3-jack in back
@@ -710,6 +737,28 @@ 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)
 
+        AD1981
+          basic         3-jack (default)
+          hp            HP nx6320
+
+        AD1986A
+          6stack        6-jack, separate surrounds (default)
+          3stack        3-stack, shared surrounds
+          laptop        2-channel only (FSC V2060, Samsung M50)
+          laptop-eapd   2-channel with EAPD (Samsung R65, ASUS A6J)
+
+        AD1988
+          6stack        6-jack
+          6stack-dig    ditto with SPDIF
+          3stack        3-jack
+          3stack-dig    ditto with SPDIF
+          laptop        3-jack with hp-jack automute
+          laptop-dig    ditto with SPDIF
+          auto          auto-confgi reading BIOS (default)
+
+        STAC7661(?)
+          vaio          Setup for VAIO FE550G/SZ110
+
     If the default configuration doesn't work and one of the above
     matches with your device, report it together with the PCI
     subsystem ID (output of "lspci -nv") to ALSA BTS or alsa-devel
@@ -723,6 +772,17 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
             (Usually SD_LPLIB register is more accurate than the
             position buffer.)
 
+    NB: If you get many "azx_get_response timeout" messages at
+    loading, it's likely a problem of interrupts (e.g. ACPI irq
+    routing).  Try to boot with options like "pci=noacpi".  Also, you
+    can try "single_cmd=1" module option.  This will switch the
+    communication method between HDA controller and codecs to the
+    single immediate commands instead of CORB/RIRB.  Basically, the
+    single command mode is provided only for BIOS, and you won't get
+    unsolicited events, too.  But, at least, this works independently
+    from the irq.  Remember this is a last resort, and should be
+    avoided as much as possible...
+    
     The power-management is supported.
 
   Module snd-hdsp
@@ -802,6 +862,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
   ------------------
 
     Module for Envy24HT (VT/ICE1724), Envy24PT (VT1720) based PCI sound cards.
+                        * MidiMan M Audio Revolution 5.1
                         * MidiMan M Audio Revolution 7.1
                         * AMP Ltd AUDIO2000
                         * TerraTec Aureon 5.1 Sky
@@ -810,6 +871,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
                         * TerraTec Phase 22
                         * TerraTec Phase 28
                         * AudioTrak Prodigy 7.1
+                        * AudioTrak Prodigy 7.1LT
                         * AudioTrak Prodigy 192
                         * Pontis MS300
                         * Albatron K8X800 Pro II 
@@ -820,9 +882,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
                         * Shuttle SN25P
 
     model       - Use the given board model, one of the following:
-                  revo71, amp2000, prodigy71, prodigy192, aureon51,
-                  aureon71, universe, k8x800, phase22, phase28, ms300,
-                  av710
+                  revo51, revo71, amp2000, prodigy71, prodigy71lt,
+                  prodigy192, aureon51, aureon71, universe,
+                  k8x800, phase22, phase28, ms300, av710
 
     This module supports multiple cards and autoprobe.
 
@@ -1353,6 +1415,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     vid             - Vendor ID for the device (optional)
     pid             - Product ID for the device (optional)
+    device_setup    - Device specific magic number (optional)
+                    - Influence depends on the device
+                    - Default: 0x0000 
 
     This module supports multiple devices, autoprobe and hotplugging.
 
diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt
new file mode 100644
index 000000000000..4692c8e77dc1
--- /dev/null
+++ b/Documentation/sound/alsa/Audiophile-Usb.txt
@@ -0,0 +1,333 @@
+        Guide to using M-Audio Audiophile USB with ALSA and Jack        v1.2
+        ========================================================
+
+            Thibault Le Meur <Thibault.LeMeur@supelec.fr>
+
+This document is a guide to using the M-Audio Audiophile USB (tm) device with 
+ALSA and JACK.
+
+1 - Audiophile USB Specs and correct usage
+==========================================
+This part is a reminder of important facts about the functions and limitations 
+of the device.
+
+The device has 4 audio interfaces, and 2 MIDI ports:
+ * Analog Stereo Input (Ai)
+   - This port supports 2 pairs of line-level audio inputs (1/4" TS and RCA) 
+   - When the 1/4" TS (jack) connectors are connected, the RCA connectors
+     are disabled
+ * Analog Stereo Output (Ao)
+ * Digital Stereo Input (Di)
+ * Digital Stereo Output (Do)
+ * Midi In (Mi)
+ * Midi Out (Mo)
+
+The internal DAC/ADC has the following caracteristics:
+* 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"
+
+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
+   - Ai+Ao+Di+Do
+ * 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)
+   - Ai or Ao or Di or Do
+
+Important facts about the Digital interface:
+--------------------------------------------
+ * The Do port additionnaly supports surround-encoded AC-3 and DTS passthrough, 
+though I haven't tested it under linux
+   - Note that in this setup only the Do interface can be enabled
+ * Apart from recording an audio digital stream, enabling the Di port is a way 
+to synchronize the device to an external sample clock
+   - As a consequence, the Di port must be enable only if an active Digital 
+source is connected
+   - Enabling Di when no digital source is connected can result in a 
+synchronization error (for instance sound played at an odd sample rate)
+
+
+2 - Audiophile USB support in ALSA
+==================================
+
+2.1 - MIDI ports
+----------------
+The Audiophile USB MIDI ports will be automatically supported once the 
+following modules have been loaded:
+ * snd-usb-audio
+ * snd-seq
+ * snd-seq-midi
+
+No additionnal setting is required.
+
+2.2 - Audio ports
+-----------------
+
+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 behaviour 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 sample rates and any sample depths supported). 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):
+ * 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.
+
+Examples:
+ * playing a S24_3BE encoded raw file to the Ao port
+   % aplay -D hw:1,0 -c2 -t raw -r48000 -fS24_3BE test.raw
+ * recording a  S24_3BE encoded raw file from the Ai port
+   % 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
+
+If you're happy with the default Alsa driver setup and don't experience any 
+issue with this mode, then you can skip the following chapter.
+
+2.2.2 - 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).
+
+For people having this problem, the snd-usb-audio module has a new module 
+parameter called "device_setup".
+
+2.2.2.1 - Initializing the working mode of the Audiohile USB
+
+As far as the Audiohile USB device is concerned, this value let the user 
+specify:
+ * the sample depth
+ * 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 decribed earlier
+ * 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
+ * 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 
+     to an active source
+   - 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
+ * device_setup=0x03
+   - 16bits 48kHz mode with only the Do port enabled 
+   - AC3 with DTS passthru (not tested)
+   - 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 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_ intialize 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:
+   - first trun off the device
+   - de-register the snd-usb-audio module
+   - change the device_setup parameter (by either manually reprobing the module 
+     or changing modprobe.conf)
+   - turn on the device
+
+2.2.2.3 - 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, 
+this is not required to use the parameter and you may skip thi section.
+
+The device_setup is one byte long and its structure is the following:
+
+       +---+---+---+---+---+---+---+---+
+       | b7| b6| b5| b4| b3| b2| b1| b0|
+       +---+---+---+---+---+---+---+---+
+       | 0 | 0 | 0 | Di|24B|96K|DTS|SET|
+       +---+---+---+---+---+---+---+---+
+
+Where:
+ * b0 is the "SET" bit
+   - it MUST be set if device_setup is initialized 
+ * b1 is the "DTS" bit
+   - it is set only for Digital output with DTS/AC3
+   - this setup is not tested
+ * b2 is the Rate selection flag
+   - When set to "1" the rate range is 48.1-96kHz
+   - Otherwise the sample rate range is 8-48kHz
+ * b3 is the bit depth selection flag
+   - When set to "1" samples are 24bits long
+   - Otherwise they are 16bits long
+   - Note that b2 implies b3 as the 96kHz mode is only supported for 24 bits 
+     samples
+ * b4 is the Digital input flag
+   - When set to "1" the device assumes that an active digital source is 
+     connected 
+   - You shouldn't enable Di if no source is seen on the port (this leads to 
+     synchronization issues)
+   - b4 is implied by b2 (since only one port is enabled at a time no synch 
+     error can occur) 
+ * b5 to b7 are reserved for future uses, and must be set to "0"
+   - might become Ao, Do, Ai, for b7, b6, b4 respectively
+
+Caution:
+ * there is no check on the value you will give to device_setup
+   - for instance choosing 0x05 (16bits 96kHz) will fail back to 0x09 since 
+     b2 implies b3. But _there_will_be_no_warning_ in /var/log/messages
+ * Hardware constraints due to the USB bus limitation aren't checked
+   - 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
+
+You may safely skip this section if you're not interrested in driver 
+development.
+
+This section describes some internals aspect of the device and summarize the 
+data I got by usb-snooping the windows and linux drivers.
+
+The M-Audio Audiophile USB has 7 USB Interfaces:
+a "USB interface":
+ * USB Interface nb.0
+ * USB Interface nb.1
+   - Audio Control function
+ * USB Interface nb.2
+   - Analog Output
+ * USB Interface nb.3
+   - Digital Output
+ * USB Interface nb.4
+   - Analog Input
+ * USB Interface nb.5
+   - Digital Input
+ * USB Interface nb.6
+   - MIDI interface compliant with the MIDIMAN quirk 
+
+Each interface has 5 altsettings (AltSet 1,2,3,4,5) except:
+ * Interface 3 (Digital Out) has an extra Alset nb.6 
+ * Interface 5 (Digital In) does not have Alset nb.3 and 5 
+
+Here is a short description of the AltSettings capabilities:
+ * AltSettings 1 corresponds to
+  - 24-bit depth, 48.1-96kHz sample mode
+  - Adaptive playback (Ao and Do), Synch capture (Ai), or Asynch capture (Di)
+ * AltSettings 2 corresponds to
+  - 24-bit depth, 8-48kHz sample mode
+  - Asynch capture and playback  (Ao,Ai,Do,Di)
+ * AltSettings 3 corresponds to
+  - 24-bit depth, 8-48kHz sample mode
+  - Synch capture (Ai) and Adaptive playback (Ao,Do)
+ * AltSettings 4 corresponds to
+  - 16-bit depth, 8-48kHz sample mode
+  - Asynch capture and playback  (Ao,Ai,Do,Di)
+ * AltSettings 5 corresponds to
+  - 16-bit depth, 8-48kHz sample mode
+  - Synch capture (Ai) and Adaptive playback (Ao,Do)
+ * AltSettings 6 corresponds to
+  - 16-bit depth, 8-48kHz sample mode
+  - Synch playback (Do), audio format type III IEC1937_AC-3
+
+In order to ensure a correct intialization of the device, the driver 
+_must_know_ how the device will be used:
+ * if DTS is choosen, only Interface 2 with AltSet nb.6 must be
+   registered
+ * if 96KHz only AltSets nb.1 of each interface must be selected
+ * if samples are using 24bits/48KHz then AltSet 2 must me used if
+   Digital input is connected, and only AltSet nb.3 if Digital input
+   is not connected
+ * if samples are using 16bits/48KHz then AltSet 4 must me used if
+   Digital input is connected, and only AltSet nb.5 if Digital input
+   is not connected
+
+When device_setup is given as a parameter to the snd-usb-audio module, the 
+parse_audio_enpoint 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
+===================================
+
+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
+--------------------------------
+
+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.
+
+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
+
diff --git a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
index 4251085d38d3..6feef9e82b63 100644
--- a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
+++ b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
@@ -1834,7 +1834,7 @@
           mychip_set_sample_format(chip, runtime->format);
           mychip_set_sample_rate(chip, runtime->rate);
           mychip_set_channels(chip, runtime->channels);
-          mychip_set_dma_setup(chip, runtime->dma_area,
+          mychip_set_dma_setup(chip, runtime->dma_addr,
                                chip->buffer_size,
                                chip->period_size);
           return 0;
@@ -2836,7 +2836,7 @@ struct _snd_pcm_runtime {
 
         <para>
         Note that this callback became non-atomic since the recent version.
-        You can use schedule-related fucntions safely in this callback now.
+        You can use schedule-related functions safely in this callback now.
         </para>
 
         <para>
@@ -3388,7 +3388,7 @@ struct _snd_pcm_runtime {
           .name = "PCM Playback Switch",
           .index = 0,
           .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
-          .private_values = 0xffff,
+          .private_value = 0xffff,
           .info = my_control_info,
           .get = my_control_get,
           .put = my_control_put
@@ -3449,7 +3449,7 @@ struct _snd_pcm_runtime {
       </para>
 
       <para>
-        The <structfield>private_values</structfield> field contains
+        The <structfield>private_value</structfield> field contains
       an arbitrary long integer value for this record. When using
       generic <structfield>info</structfield>,
       <structfield>get</structfield> and
diff --git a/Documentation/sound/oss/Introduction b/Documentation/sound/oss/Introduction
index 15d4fb975ac0..f04ba6bb7395 100644
--- a/Documentation/sound/oss/Introduction
+++ b/Documentation/sound/oss/Introduction
@@ -69,7 +69,7 @@ are available, for example IRQ, address, DMA.
 
 Warning, the options for different cards sometime use different names 
 for the same or a similar feature (dma1= versus dma16=).  As a last 
-resort, inspect the code (search for MODULE_PARM).
+resort, inspect the code (search for module_param).
 
 Notes:
 
diff --git a/Documentation/sound/oss/cs46xx b/Documentation/sound/oss/cs46xx
index 88d6cf8b39f3..b54432709863 100644
--- a/Documentation/sound/oss/cs46xx
+++ b/Documentation/sound/oss/cs46xx
@@ -88,7 +88,7 @@ parameters.  for a copy email: twoller@crystal.cirrus.com
 
 MODULE_PARMS definitions
 ------------------------
-MODULE_PARM(defaultorder, "i");
+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.
@@ -98,18 +98,18 @@ to not underrun the dma buffer as easily.  As default, use 32k (order=3)
 rather than 64k as some of the games work more responsively.
 (2^N) * PAGE_SIZE = allocated buffer size
 
-MODULE_PARM(cs_debuglevel, "i");
-MODULE_PARM(cs_debugmask, "i");
+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_PARM(hercules_egpio_disable, "i");
+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_PARM(initdelay, "i");
+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
@@ -118,19 +118,19 @@ system is booted under battery power then the mdelay()/udelay() functions fail t
 properly delay the required time.  Also, if the system is booted under AC power
 and then the power removed, the mdelay()/udelay() functions will not delay properly.
  
-MODULE_PARM(powerdown, "i");
+module_param(powerdown, ulong, 0);
 powerdown=N
 where N is 0 (disable any powerdown of the internal blocks) or 1 (enable powerdown)
 
 
-MODULE_PARM(external_amp, "i");
+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_PARM(thinkpad, "i");
+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,