Merge suse.cz:/home/perex/bk/linux-sound/linux-2.5

into suse.cz:/home/perex/bk/linux-sound/linux-sound

5 files changed, 442 insertions, 185 deletions

diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index addec50998e6b8..10c5bed26b1c01 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -416,7 +416,8 @@ Module parameters
 	* Creative Card w/Digital out + Digital I/O 2	[0x0fc3/0x1f0f]
 	* Creative Card w/Digital CD in + Digital I/O 2	[0x0fcf/0x1f0f]
         * Creative Card 5.1/w Digital out + LiveDrive	[0x3fc3/0x1fff]
-        * Creative Card all ins and outs		[0x3fff/0x1fff]
+	* Creative Card 5.1 (c) 2003			[0x3fc3/0x7cff]
+        * Creative Card all ins and outs		[0x3fff/0x7fff]
     
   Module snd-ens1370
   ------------------
@@ -609,6 +610,10 @@ Module parameters
                         * Hoontech SoundTrack DSP 24 Media 7.1
                         * Digigram VX442
 
+    model       - Use the given board model, one of the following:
+		  delta1010, dio2496, delta66, delta44, audiophile, delta410,
+		  delta1010lt, vx442, ewx2496, ews88mt, ews88mt_new, ews88d,
+		  dmx6fire, dsp24, dsp24_71, ez8
     omni	- Omni I/O support for MidiMan M-Audio Delta44/66
     cs8427_timeout - reset timeout for the CS8427 chip (S/PDIF transciever)
                      in msec resolution, default value is 500 (0.5 sec)
@@ -625,6 +630,9 @@ Module parameters
 			* AMP Ltd AUDIO2000
 			* TerraTec Aureon Sky-5.1, Space-7.1
 
+    model       - Use the given board model, one of the following:
+		  revo71, amp2000, prodigy71, aureon51, aureon71
+
     Module supports up to 8 cards and autoprobe.
 
   Module snd-intel8x0
@@ -787,6 +795,8 @@ Module parameters
 
     Module supports autoprobe and multiple chips (max 8).
 
+    The power-management is supported.
+
     Note: on some notebooks the buffer address cannot be detected
     automatically, or causes hang-up during initialization.
     In such a case, specify the buffer top address explicity via
@@ -796,9 +806,24 @@ Module parameters
       Sony F270: buffer_top=0x272800
     The driver supports only ac97 codec.  It's possible to force
     to initialize/use ac97 although it's not detected.  In such a
-    case, use force_ac97=1 option.
-
-    The power-management is supported.
+    case, use force_ac97=1 option - but *NO* guarantee whether it
+    works!
+
+    Note: The NM256 chip can be linked internally with non-AC97
+    codecs.  This driver supports only the AC97 codec, and won't work
+    with machines with other (most likely CS423x or OPL3SAx) chips,
+    even though the device is detected in lspci.  In such a case, try
+    other drivers, e.g. snd-cs4232 or snd-opl3sa2.  Some has ISA-PnP
+    but some doesn't have ISA PnP.  You'll need to speicfy isapnp=0
+    and proper hardware parameters in the case without ISA PnP.
+
+    Note: This driver is really crappy.  It's a porting from the
+    OSS driver, which is a result of black-magic reverse engineering.
+    The detection of codec will fail if the driver is loaded *after*
+    X-server as described above.  You might be able to force to load
+    the module, but it may result in hang-up.   Hence, make sure that
+    you load this module *before* X if you encounter this kind of
+    problem.
 
   Module snd-opl3sa2
   ------------------
diff --git a/Documentation/sound/alsa/Audigy-mixer.txt b/Documentation/sound/alsa/Audigy-mixer.txt
new file mode 100644
index 00000000000000..5132fd95e07435
--- /dev/null
+++ b/Documentation/sound/alsa/Audigy-mixer.txt
@@ -0,0 +1,345 @@
+
+		Sound Blaster Audigy mixer / default DSP code
+		===========================================
+
+This is based on SB-Live-mixer.txt.
+
+The EMU10K2 chips have a DSP part which can be programmed to support 
+various ways of sample processing, which is described here.
+(This acticle does not deal with the overall functionality of the 
+EMU10K2 chips. See the manuals section for further details.)
+
+The ALSA driver programs this portion of chip by default code
+(can be altered later) which offers the following functionality:
+
+
+1) Digital mixer controls
+-------------------------
+
+These controls are built using the DSP instructions. They offer extended
+functionality. Only the default build-in code in the ALSA driver is described
+here. Note that the controls work as attenuators: the maximum value is the 
+neutral position leaving the signal unchanged. Note that if the  same destination 
+is mentioned in multiple controls, the signal is accumulated and can be wrapped 
+(set to maximal or minimal value without checking of overflow).
+
+
+Explanation of used abbreviations:
+
+DAC    - digital to analog converter
+ADC    - analog to digital converter
+I2S    - one-way three wire serial bus for digital sound by Philips Semiconductors
+         (this standard is used for connecting standalone DAC and ADC converters)
+LFE    - low frequency effects (subwoofer signal)
+AC97   - a chip containing an analog mixer, DAC and ADC converters
+IEC958 - S/PDIF
+FX-bus - the EMU10K2 chip has an effect bus containing 64 accumulators.
+         Each of the synthesizer voices can feed its output to these accumulators
+         and the DSP microcontroller can operate with the resulting sum.
+
+name='PCM Front Playback Volume',index=0
+
+This control is used to attenuate samples for left and right front PCM FX-bus
+accumulators. ALSA uses accumulators 8 and 9 for left and right front PCM 
+samples for 5.1 playback. The result samples are forwarded to the front DAC PCM 
+slots of the Philips DAC.
+
+name='PCM Surround Playback Volume',index=0
+
+This control is used to attenuate samples for left and right surround PCM FX-bus
+accumulators. ALSA uses accumulators 2 and 3 for left and right surround PCM 
+samples for 5.1 playback. The result samples are forwarded to the surround DAC PCM 
+slots of the Philips DAC.
+
+name='PCM Center Playback Volume',index=0
+
+This control is used to attenuate samples for center PCM FX-bus accumulator.
+ALSA uses accumulator 6 for center PCM sample for 5.1 playback. The result sample
+is forwarded to the center DAC PCM slot of the Philips DAC.
+
+name='PCM LFE Playback Volume',index=0
+
+This control is used to attenuate sample for LFE PCM FX-bus accumulator. 
+ALSA uses accumulator 7 for LFE PCM sample for 5.1 playback. The result sample 
+is forwarded to the LFE DAC PCM slot of the Philips DAC.
+
+name='PCM Playback Volume',index=0
+
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples for
+stereo playback. The result samples are forwarded to the front DAC PCM slots 
+of the Philips DAC.
+
+name='PCM Capture Volume',index=0
+
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulator. ALSA uses accumulators 0 and 1 for left and right PCM.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Music Playback Volume',index=0
+
+This control is used to attenuate samples for left and right MIDI FX-bus
+accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples.
+The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
+
+name='Music Capture Volume',index=0
+
+These controls are used to attenuate samples for left and right MIDI FX-bus
+accumulator. ALSA uses accumulators 4 and 5 for left and right PCM.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Mic Playback Volume',index=0
+
+This control is used to attenuate samples for left and right Mic input.
+For Mic input is used AC97 codec. The result samples are forwarded to 
+the front DAC PCM slots of the Philips DAC. Samples are forwarded to Mic
+capture FIFO (device 1 - 16bit/8KHz mono) too without volume control.
+
+name='Mic Capture Volume',index=0
+
+This control is used to attenuate samples for left and right Mic input.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Audigy CD Playback Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 TTL
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the front DAC PCM slots of the Philips DAC.
+
+name='Audigy CD Capture Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 TTL
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the ADC capture FIFO (thus to the standard capture PCM device).
+
+name='IEC958 Optical Playback Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 optical
+digital input. The result samples are forwarded to the front DAC PCM slots
+of the Philips DAC.
+
+name='IEC958 Optical Capture Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 optical
+digital inputs. The result samples are forwarded to the ADC capture FIFO
+(thus to the standard capture PCM device).
+
+name='Line2 Playback Volume',index=0
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the AudigyDrive). The result samples are forwarded to the front
+DAC PCM slots of the Philips DAC.
+
+name='Line2 Capture Volume',index=1
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the AudigyDrive). The result samples are forwarded to the ADC
+capture FIFO (thus to the standard capture PCM device).
+
+name='Analog Mix Playback Volume',index=0
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs from Philips ADC. The result samples are forwarded to the front
+DAC PCM slots of the Philips DAC. This contains mix from analog sources
+like CD, Line In, Aux, ....
+
+name='Analog Mix Capture Volume',index=1
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs Philips ADC. The result samples are forwarded to the ADC
+capture FIFO (thus to the standard capture PCM device).
+
+name='Aux2 Playback Volume',index=0
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the AudigyDrive). The result samples are forwarded to the front
+DAC PCM slots of the Philips DAC.
+
+name='Aux2 Capture Volume',index=1
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the AudigyDrive). The result samples are forwarded to the ADC
+capture FIFO (thus to the standard capture PCM device).
+
+name='Front Playback Volume',index=0
+
+All stereo signals are mixed together and mirrored to surround, center and LFE.
+This control is used to attenuate samples for left and right front speakers of
+this mix.
+
+name='Surround Playback Volume',index=0
+
+All stereo signals are mixed together and mirrored to surround, center and LFE.
+This control is used to attenuate samples for left and right surround speakers of
+this mix.
+
+name='Center Playback Volume',index=0
+
+All stereo signals are mixed together and mirrored to surround, center and LFE.
+This control is used to attenuate sample for center speaker of this mix.
+
+name='LFE Playback Volume',index=0
+
+All stereo signals are mixed together and mirrored to surround, center and LFE.
+This control is used to attenuate sample for LFE speaker of this mix.
+
+name='Tone Control - Switch',index=0
+
+This control turns the tone control on or off. The samples for front, rear
+and center / LFE outputs are affected.
+
+name='Tone Control - Bass',index=0
+
+This control sets the bass intensity. There is no neutral value!!
+When the tone control code is activated, the samples are always modified.
+The closest value to pure signal is 20.
+
+name='Tone Control - Treble',index=0
+
+This control sets the treble intensity. There is no neutral value!!
+When the tone control code is activated, the samples are always modified.
+The closest value to pure signal is 20.
+
+name='Master Playback Volume',index=0
+
+This control is used to attenuate samples for front, surround, center and 
+LFE outputs.
+
+name='IEC958 Optical Raw Playback Switch',index=0
+
+If this switch is on, then the samples for the IEC958 (S/PDIF) digital
+output are taken only from the raw FX8010 PCM, otherwise standard front
+PCM samples are taken.
+
+
+2) PCM stream related controls
+------------------------------
+
+name='EMU10K1 PCM Volume',index 0-31
+
+Channel volume attenuation in range 0-0xffff. The maximum value (no
+attenuation) is default. The channel mapping for three values is
+as follows:
+
+	0 - mono, default 0xffff (no attenuation)
+	1 - left, default 0xffff (no attenuation)
+	2 - right, default 0xffff (no attenuation)
+
+name='EMU10K1 PCM Send Routing',index 0-31
+
+This control specifies the destination - FX-bus accumulators. There 24
+values with this mapping:
+
+	 0 -  mono, A destination (FX-bus 0-63), default 0
+	 1 -  mono, B destination (FX-bus 0-63), default 1
+	 2 -  mono, C destination (FX-bus 0-63), default 2
+	 3 -  mono, D destination (FX-bus 0-63), default 3
+	 4 -  mono, E destination (FX-bus 0-63), default 0
+	 5 -  mono, F destination (FX-bus 0-63), default 0
+	 6 -  mono, G destination (FX-bus 0-63), default 0
+	 7 -  mono, H destination (FX-bus 0-63), default 0
+	 8 -  left, A destination (FX-bus 0-63), default 0
+	 9 -  left, B destination (FX-bus 0-63), default 1
+	10 -  left, C destination (FX-bus 0-63), default 2
+	11 -  left, D destination (FX-bus 0-63), default 3
+	12 -  left, E destination (FX-bus 0-63), default 0
+	13 -  left, F destination (FX-bus 0-63), default 0
+	14 -  left, G destination (FX-bus 0-63), default 0
+	15 -  left, H destination (FX-bus 0-63), default 0
+	16 - right, A destination (FX-bus 0-63), default 0
+	17 - right, B destination (FX-bus 0-63), default 1
+	18 - right, C destination (FX-bus 0-63), default 2
+	19 - right, D destination (FX-bus 0-63), default 3
+	20 - right, E destination (FX-bus 0-63), default 0
+	21 - right, F destination (FX-bus 0-63), default 0
+	22 - right, G destination (FX-bus 0-63), default 0
+	23 - right, H destination (FX-bus 0-63), default 0
+
+Don't forget that it's illegal to assign a channel to the same FX-bus accumulator 
+more than once (it means 0=0 && 1=0 is an invalid combination).
+ 
+name='EMU10K1 PCM Send Volume',index 0-31
+
+It specifies the attenuation (amount) for given destination in range 0-255.
+The channel mapping is following:
+
+	 0 -  mono, A destination attn, default 255 (no attenuation)
+	 1 -  mono, B destination attn, default 255 (no attenuation)
+	 2 -  mono, C destination attn, default 0 (mute)
+	 3 -  mono, D destination attn, default 0 (mute)
+	 4 -  mono, E destination attn, default 0 (mute)
+	 5 -  mono, F destination attn, default 0 (mute)
+	 6 -  mono, G destination attn, default 0 (mute)
+	 7 -  mono, H destination attn, default 0 (mute)
+	 8 -  left, A destination attn, default 255 (no attenuation)
+	 9 -  left, B destination attn, default 0 (mute)
+	10 -  left, C destination attn, default 0 (mute)
+	11 -  left, D destination attn, default 0 (mute)
+	12 -  left, E destination attn, default 0 (mute)
+	13 -  left, F destination attn, default 0 (mute)
+	14 -  left, G destination attn, default 0 (mute)
+	15 -  left, H destination attn, default 0 (mute)
+	16 - right, A destination attn, default 0 (mute)
+	17 - right, B destination attn, default 255 (no attenuation)
+	18 - right, C destination attn, default 0 (mute)
+	19 - right, D destination attn, default 0 (mute)
+	20 - right, E destination attn, default 0 (mute)
+	21 - right, F destination attn, default 0 (mute)
+	22 - right, G destination attn, default 0 (mute)
+	23 - right, H destination attn, default 0 (mute)
+
+
+
+4) MANUALS/PATENTS:
+-------------------
+
+ftp://opensource.creative.com/pub/doc
+-------------------------------------
+
+        Files:
+        LM4545.pdf      AC97 Codec
+
+        m2049.pdf       The EMU10K1 Digital Audio Processor
+
+        hog63.ps        FX8010 - A DSP Chip Architecture for Audio Effects
+
+
+WIPO Patents
+------------
+        Patent numbers:
+        WO 9901813 (A1) Audio Effects Processor with multiple asynchronous (Jan. 14, 1999)
+                        streams
+
+        WO 9901814 (A1) Processor with Instruction Set for Audio Effects (Jan. 14, 1999)
+
+        WO 9901953 (A1) Audio Effects Processor having Decoupled Instruction
+                        Execution and Audio Data Sequencing (Jan. 14, 1999)
+
+
+US Patents (http://www.uspto.gov/)
+----------------------------------
+
+        US 5925841      Digital Sampling Instrument employing cache memory (Jul. 20, 1999)
+
+        US 5928342      Audio Effects Processor integrated on a single chip (Jul. 27, 1999)
+                        with a multiport memory onto which multiple asynchronous
+                        digital sound samples can be concurrently loaded
+
+        US 5930158      Processor with Instruction Set for Audio Effects (Jul. 27, 1999)
+
+        US 6032235      Memory initialization circuit (Tram) (Feb. 29, 2000)
+
+        US 6138207      Interpolation looping of audio samples in cache connected to    (Oct. 24, 2000)
+                        system bus with prioritization and modification of bus transfers
+                        in accordance with loop ends and minimum block sizes
+
+        US 6151670      Method for conserving memory storage using a (Nov. 21, 2000)
+                        pool of  short term memory registers
+
+        US 6195715      Interrupt control for multiple programs communicating with      (Feb. 27, 2001)
+                        a common interrupt by associating programs to GP registers,
+                        defining interrupt register, polling GP registers, and invoking
+                        callback routine associated with defined interrupt register
diff --git a/Documentation/sound/alsa/CMIPCI.txt b/Documentation/sound/alsa/CMIPCI.txt
index f7e5ff7ba34142..4a7df771b806c7 100644
--- a/Documentation/sound/alsa/CMIPCI.txt
+++ b/Documentation/sound/alsa/CMIPCI.txt
@@ -180,8 +180,8 @@ Similarly the following switches are off: "IEC958 Mix Analog" and
 device automatically to the previous state.
 
 On the model 033, AC3 is implemented by the software conversion in
-the driver.  This prevents the mmap support.  If you need mmap
-support, pass the "soft_ac3=0" module option.  This doesn't matter
+the alsa-lib.  If you need to bypass the software conversion of IEC958
+subframes, pass the "soft_ac3=0" module option.  This doesn't matter
 on the newer models.
 
 
diff --git a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
index 63a46fd2b52737..934a49a595fbea 100644
--- a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
+++ b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
@@ -511,7 +511,7 @@
           }
 
           // (7)
-          pci_set_drvdata(pci, chip);
+          pci_set_drvdata(pci, card);
           dev++;
           return 0;
   }
@@ -519,10 +519,7 @@
   // destructor -- see "Destructor" sub-section
   static void __devexit snd_mychip_remove(struct pci_dev *pci)
   {
-          mychip_t *chip = snd_magic_cast(mychip_t,
-                                     pci_get_drvdata(pci), return);
-          if (chip)
-                  snd_card_free(chip->card);
+          snd_card_free(pci_get_drvdata(pci));
           pci_set_drvdata(pci, NULL);
   }
 ]]>
@@ -691,21 +688,16 @@
           <informalexample>
             <programlisting>
 <![CDATA[
-        pci_set_drvdata(pci, chip);
+        pci_set_drvdata(pci, card);
         dev++;
         return 0;
 ]]>
             </programlisting>
           </informalexample>
 
-          In the above, the chip record is stored. This pointer is
+          In the above, the card record is stored. This pointer is
         referred in the remove callback and power-management
         callbacks, too. 
-	If the card doesn't support the suspend/resume, you can store
-        the card pointer instead of the chip pointer, so that
-        <function>snd_card_free</function> can be called directly
-        without cast in the remove callback.  But anyway, be sure
-        which pointer is used.
         </para>
       </section>
     </section>
@@ -726,21 +718,15 @@
 <![CDATA[
   static void __devexit snd_mychip_remove(struct pci_dev *pci)
   {
-          mychip_t *chip = snd_magic_cast(mychip_t,
-                                    pci_get_drvdata(pci), return);
-          if (chip)
-                  snd_card_free(chip->card);
+          snd_card_free(pci_get_drvdata(pci));
           pci_set_drvdata(pci, NULL);
   }
 ]]>
           </programlisting>
         </informalexample>
 
-        The above code assumes that the chip is allocated
-	with snd_magic stuff and
-      has the field to hold the card pointer (see <link
-      linkend="card-management"><citetitle>the next
-      section</citetitle></link>). 
+        The above code assumes that the card pointer is set to the PCI
+	driver data.
       </para>
     </section>
 
@@ -1355,16 +1341,7 @@
   // initialization of the module
   static int __init alsa_card_mychip_init(void)
   {
-          int err;
-
-          if ((err = pci_module_init(&driver)) < 0) {
-  #ifdef MODULE
-                  printk(KERN_ERR "My chip soundcard not found "
-                                  "or device busy\n");
-  #endif
-                  return err;
-          }
-          return 0;
+          return pci_module_init(&driver);
   }
 
   // clean up the module
@@ -1781,16 +1758,7 @@
 <![CDATA[
   static int __init alsa_card_mychip_init(void)
   {
-          int err;
-
-          if ((err = pci_module_init(&driver)) < 0) {
-  #ifdef MODULE
-                  printk(KERN_ERR "My chip soundcard not found"
-                                  " or device busy\n");
-  #endif
-                  return err;
-          }
-          return 0;
+          return pci_module_init(&driver);
   }
 
   static void __exit alsa_card_mychip_exit(void)
@@ -5254,47 +5222,23 @@ struct _snd_pcm_runtime {
     </para>
 
     <para>
-      Basic jobs of suspend/resume are done in
-      <structfield>suspend</structfield> and
-      <structfield>resume</structfield> callbacks of
-      <structname>pci_driver</structname> struct. Unfortunately, the
-      API of these callbacks was changed at the middle time of Linux
-      2.4.x, if you want to keep the support for older kernels, you
-      have to write two different callbacks. The example below is the
-      skeleton callbacks which just call the real suspend and resume
-      functions. 
+      ALSA provides the common power-management layer. Each card driver
+      needs to have only low-level suspend and resume callbacks.
 
       <informalexample>
         <programlisting>
 <![CDATA[
-  #ifndef PCI_OLD_SUSPEND
-  static int snd_my_suspend(struct pci_dev *dev, u32 state)
+  #ifdef CONFIG_PM
+  static int snd_my_suspend(snd_card_t *card, unsigned int state)
   {
-          mychip_t *chip = snd_magic_cast(mychip_t,
-                             pci_get_drvdata(dev), return -ENXIO);
-          mychip_suspend(chip);
+          .... // do things for suspsend
           return 0;
   }
-  static int snd_my_resume(struct pci_dev *dev)
+  static int snd_my_resume(snd_card_t *card, unsigned int state)
   {
-          mychip_t *chip = snd_magic_cast(mychip_t,
-                             pci_get_drvdata(dev), return -ENXIO);
-          mychip_resume(chip);
+          .... // do things for suspsend
           return 0;
   }
-  #else
-  static void snd_my_suspend(struct pci_dev *dev)
-  {
-          mychip_t *chip = snd_magic_cast(mychip_t,
-                             pci_get_drvdata(dev), return);
-          mychip_suspend(chip);
-  }
-  static void snd_mychip_resume(struct pci_dev *dev)
-  {
-          mychip_t *chip = snd_magic_cast(mychip_t,
-                             pci_get_drvdata(dev), return);
-          mychip_resume(chip);
-  }
   #endif
 ]]>
         </programlisting>
@@ -5302,17 +5246,10 @@ struct _snd_pcm_runtime {
     </para>
 
     <para>
-      For keeping the readability of 2.6 source code, it's recommended to
-      separate the above ifdef condition as the patch file in alsa-driver
-      directory.
-      See <filename>alsa-driver/pci/ali5451.c</filename> for example.
-   </para>
-
-    <para>
       The scheme of the real suspend job is as following.
 
       <orderedlist>
-        <listitem><para>Check whether the power-state is already D3hot. If yes, skip the job.</para></listitem>
+        <listitem><para>Retrieve the chip data from pm_private_data field.</para></listitem>
         <listitem><para>Call <function>snd_pcm_suspend_all()</function> to suspend the running PCM streams.</para></listitem>
         <listitem><para>Save the register values if necessary.</para></listitem>
         <listitem><para>Stop the hardware if necessary.</para></listitem>
@@ -5326,12 +5263,11 @@ struct _snd_pcm_runtime {
       <informalexample>
         <programlisting>
 <![CDATA[
-  static void mychip_suspend(mychip_t *chip)
+  static int mychip_suspend(snd_card_t *card, unsigned int state)
   {
-          snd_card_t *card = chip->card;
           // (1)
-          if (card->power_state == SNDRV_CTL_POWER_D3hot)
-                  return;
+          mychip_t *chip = snd_magic_cast(mychip_t, card->pm_private_data,
+                                          return -ENXIO);
           // (2)
           snd_pcm_suspend_all(chip->pcm);
           // (3)
@@ -5340,6 +5276,7 @@ struct _snd_pcm_runtime {
           snd_mychip_stop_hardware(chip);
           // (5)
           snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+          return 0;
   }
 ]]>
         </programlisting>
@@ -5350,8 +5287,7 @@ struct _snd_pcm_runtime {
     The scheme of the real resume job is as following.
 
     <orderedlist>
-    <listitem><para>Check whether the power-state is already D0.
-    If yes, skip the job.</para></listitem>
+    <listitem><para>Retrieve the chip data from pm_private_data field.</para></listitem>
     <listitem><para>Enable the pci device again by calling
     <function>pci_enable_device()</function>.</para></listitem>
     <listitem><para>Re-initialize the chip.</para></listitem>
@@ -5372,10 +5308,9 @@ struct _snd_pcm_runtime {
 <![CDATA[
   static void mychip_resume(mychip_t *chip)
   {
-          snd_card_t *card = chip->card;
           // (1)
-          if (card->power_state == SNDRV_CTL_POWER_D0)
-                  return;
+          mychip_t *chip = snd_magic_cast(mychip_t, card->pm_private_data,
+                                          return -ENXIO);
           // (2)
           pci_enable_device(chip->pci);
           // (3)
@@ -5388,38 +5323,6 @@ struct _snd_pcm_runtime {
           snd_mychip_restart_chip(chip);
           // (7)
           snd_power_change_state(card, SNDRV_CTL_POWER_D0);
-  }
-]]>
-        </programlisting>
-      </informalexample>
-    </para>
-
-    <para>
-      In addition to the callbacks above, you should define a callback
-      for the changes via the ALSA control interface. It's defined
-      like below: 
-
-      <informalexample>
-        <programlisting>
-<![CDATA[
-  static int snd_mychip_set_power_state(snd_card_t *card,
-                                        unsigned int power_state)
-  {
-          mychip_t *chip = snd_magic_cast(mychip_t,
-                   card->power_state_private_data, return -ENXIO);
-          switch (power_state) {
-          case SNDRV_CTL_POWER_D0:
-          case SNDRV_CTL_POWER_D1:
-          case SNDRV_CTL_POWER_D2:
-                  mychip_resume(chip);
-                  break;
-          case SNDRV_CTL_POWER_D3hot:
-          case SNDRV_CTL_POWER_D3cold:
-                  mychip_suspend(chip);
-                  break;
-          default:
-                  return -EINVAL;
-          }
           return 0;
   }
 ]]>
@@ -5439,41 +5342,42 @@ struct _snd_pcm_runtime {
   {
           ....
           snd_card_t *card;
+          mychip_t *chip;
           ....
-  #ifdef CONFIG_PM
-          card->set_power_state = snd_mychip_set_power_state;
-          card->power_state_private_data = chip;
-  #endif
+          snd_card_set_pm_callback(card, snd_my_suspend, snd_my_resume, chip);
           ....
   }
 ]]>
         </programlisting>
       </informalexample>
+
+    Here you don't have to put ifdef CONFIG_PM around, since it's already
+    checked in the header and expanded to empty if not needed.
     </para>
 
     <para>
       If you need a space for saving the registers, you'll need to
-    allocate the buffer for it here, too, since you cannot call
-    <function>kmalloc()</function> with
-    <constant>GFP_KERNEL</constant> flag or
-    <function>vmalloc()</function> in the suspend callback.
+    allocate the buffer for it here, too, since it would be fatal
+    if you cannot allocate a memory in the suspend phase.
     The allocated buffer should be released in the corresponding
     destructor.
     </para>
 
     <para>
       And next, set suspend/resume callbacks to the pci_driver,
+      This can be done by passing a macro SND_PCI_PM_CALLBACKS
+      in the pci_driver struct.  This macro is expanded to the correct
+      (global) callbacks if CONFIG_PM is set.
 
       <informalexample>
         <programlisting>
 <![CDATA[
   static struct pci_driver driver = {
           .name = "My Chip",
-          ....
-  #ifdef CONFIG_PM
-          .suspend = snd_mychip_suspend,
-          .resume = snd_mychip_resume,
-  #endif
+          .id_table = snd_my_ids,
+          .probe = snd_my_probe,
+          .remove = __devexit_p(snd_my_remove),
+          SND_PCI_PM_CALLBACKS
   };
 ]]>
         </programlisting>
@@ -5521,7 +5425,8 @@ struct _snd_pcm_runtime {
 
     <para>
       The module parameters must be declared with the standard
-    <function>MODULE_PARM()</function> and
+    <function>module_param()()</function>,
+    <function>module_param_array()()</function> and
     <function>MODULE_PARM_DESC()</function> macros. The ALSA provides
     an additional macro, <function>MODULE_PARM_SYNTAX()</function>,
     for describing its syntax. The strings will be written to
@@ -5545,18 +5450,22 @@ struct _snd_pcm_runtime {
 <![CDATA[
   #define CARD_NAME "My Chip"
 
-  MODULE_PARM(index, "1-" __MODULE_STRING(SNDRV_CARDS) "i");
+  static int boot_devs;
+  module_param_array(index, int, boot_devs, 0444);
   MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard.");
   MODULE_PARM_SYNTAX(index, SNDRV_INDEX_DESC);
-  MODULE_PARM(id, "1-" __MODULE_STRING(SNDRV_CARDS) "s");
+  module_param_array(id, charp, boot_devs, 0444);
   MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard.");
   MODULE_PARM_SYNTAX(id, SNDRV_ID_DESC);
-  MODULE_PARM(enable, "1-" __MODULE_STRING(SNDRV_CARDS) "i");
+  module_param_array(enable, bool, boot_devs, 0444);
   MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard.");
   MODULE_PARM_SYNTAX(enable, SNDRV_ENABLE_DESC);
 ]]>
         </programlisting>
       </informalexample>
+
+    Here boot_devs is passed but simply ignored since we don't care
+    the number of parsed parameters.
     </para>
 
     <para>
@@ -5577,39 +5486,6 @@ struct _snd_pcm_runtime {
       </informalexample>
     </para>
 
-    <para>
-      For building the driver into kernel, you should define the
-      <function>setup()</function> function in addition, too. 
-      ALSA provides <function>get_id()</function> function to retrieve
-      a string argument from the kernel boot parameters.
-
-      <informalexample>
-        <programlisting>
-<![CDATA[
-  #ifndef MODULE
-
-  /* format is: snd-mychip=enable,index,id */
-
-  static int __init alsa_card_mychip_setup(char *str)
-  {
-          static unsigned __initdata nr_dev = 0;
-
-          if (nr_dev >= SNDRV_CARDS)
-                  return 0;
-          (void)(get_option(&str,&enable[nr_dev]) == 2 &&
-                 get_option(&str,&index[nr_dev]) == 2 &&
-                 get_id(&str,&id[nr_dev]) == 2);
-          nr_dev++;
-          return 1;
-  }
-
-  __setup("snd-mychip=", alsa_card_mychip_setup);
-
-  #endif /* ifndef MODULE */
-]]>
-        </programlisting>
-      </informalexample>
-    </para>
   </chapter>
 
 
@@ -5631,10 +5507,14 @@ struct _snd_pcm_runtime {
 	Suppose that you'll create a new PCI driver for the card
 	<quote>xyz</quote>.  The card module name would be
 	snd-xyz.  The new driver is usually put into alsa-driver
-	tree.  Then the driver is evaluated, audited and tested
+	tree, <filename>alsa-driver/pci</filename> directory in
+	the case of PCI cards.
+	Then the driver is evaluated, audited and tested
 	by developers and users.  After a certain time, the driver
-	will go to alsa-kernel tree and eventually integrated into
-	Linux 2.6 tree.
+	will go to alsa-kernel tree (to the corresponding directory,
+	such as <filename>alsa-kernel/pci</filename>) and eventually
+	integrated into Linux 2.6 tree (the directory would be
+	<filename>linux/sound/pci</filename>).
 	</para>
 
 	<para>
@@ -5661,7 +5541,7 @@ struct _snd_pcm_runtime {
         <programlisting>
 <![CDATA[
   snd-xyz-objs := xyz.o
-  extra-obj-$(CONFIG_SND_XYZ) += snd-xyz.o
+  obj-$(CONFIG_SND_XYZ) += snd-xyz.o
 ]]>
         </programlisting>
       </informalexample>
@@ -5678,8 +5558,8 @@ struct _snd_pcm_runtime {
       <informalexample>
         <programlisting>
 <![CDATA[
-  config SND_BT87X
-          tristate "Foobar XYX"
+  config SND_XYZ
+          tristate "Foobar XYZ"
           depends on SND
           select SND_PCM
           help
@@ -5730,7 +5610,8 @@ struct _snd_pcm_runtime {
 	<orderedlist>
 	<listitem>
 	<para>
-	Add a new directory (xyz) to extra-subdir-y list in alsa-driver/pci/Makefile
+	Add a new directory (<filename>xyz</filename>) in
+	<filename>alsa-driver/pci/Makefile</filename> like below
 
       <informalexample>
         <programlisting>
@@ -5744,7 +5625,7 @@ struct _snd_pcm_runtime {
 
 	<listitem>
 	<para>
-	Under the directory xyz, create a Makefile
+	Under the directory <filename>xyz</filename>, create a Makefile
 
       <example>
 	<title>Sample Makefile for a driver xyz</title>
diff --git a/Documentation/sound/alsa/Procfile.txt b/Documentation/sound/alsa/Procfile.txt
index d686ea3c39eb59..25c5d648aef6c2 100644
--- a/Documentation/sound/alsa/Procfile.txt
+++ b/Documentation/sound/alsa/Procfile.txt
@@ -131,6 +131,12 @@ card*/codec97#*/ac97#?-?
 card*/codec97#0/ac97#?-?+regs
 	Shows the AC97 register dump.  Useful for debugging.
 
+	When CONFIG_SND_DEBUG is enabled, you can write to this file for
+	changing an AC97 register directly.  Pass two hex numbers.
+	For example,
+
+	# echo 02 9f1f > /proc/asound/card0/codec97#0/ac97#0-0+regs
+
 
 Sequencer Information
 ---------------------