Merge branch 'linus' of master.kernel.org:/pub/scm/linux/kernel/git/perex/alsa

* 'linus' of master.kernel.org:/pub/scm/linux/kernel/git/perex/alsa: (264 commits)
  [ALSA] version 1.0.15
  [ALSA] Fix thinko in cs4231 mce down check
  [ALSA] sun-cs4231: improved waiting after MCE down
  [ALSA] sun-cs4231: use cs4231-regs.h
  [ALSA] This simplifies and fixes waiting loops of the mce_down()
  [ALSA] This patch adds support for a wavetable chip on
  [ALSA] This patch removes open_mutex from the ad1848-lib as
  [ALSA] fix bootup crash in snd_gus_interrupt()
  [ALSA] hda-codec - Fix SKU ID function for realtek codecs
  [ALSA] Support  ASUS P701 eeepc [0x1043 0x82a1] support
  [ALSA] hda-codec - Add array terminator for dmic in STAC codec
  [ALSA] hdsp - Fix zero division
  [ALSA] usb-audio - Fix double comment
  [ALSA] hda-codec - Fix STAC922x volume knob control
  [ALSA] Changed Jaroslav Kysela's e-mail from perex@suse.cz to perex@perex.cz
  [ALSA] hda-codec - Fix for Fujitsu Lifebook C1410
  [ALSA] mpu-401: remove MPU401_INFO_UART_ONLY flag
  [ALSA] mpu-401: do not require an ACK byte for the ENTER_UART command
  [ALSA] via82xx - Add DXS quirk for Shuttle AK31v2
  [ALSA] hda-codec - Fix input_mux numbers for vaio stac92xx
  ...

CREDITS

@@ -1933,7 +1933,7 @@ M: seasons@makosteszta.sote.hu
 D: Original author of software suspend
 
 N: Jaroslav Kysela
-E: perex@suse.cz
+E: perex@perex.cz
 W: http://www.perex.cz
 D: Original Author and Maintainer for HP 10/100 Mbit Network Adapters
 D: ISA PnP

Documentation/sound/alsa/ALSA-Configuration.txt

@@ -365,13 +365,14 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
   Module snd-cmipci
   -----------------
 
-    Module for C-Media CMI8338 and 8738 PCI sound cards.
+    Module for C-Media CMI8338/8738/8768/8770 PCI sound cards.
 
-    mpu_port	- 0x300,0x310,0x320,0x330 = legacy port,
+    mpu_port    - port address of MIDI interface (8338 only):
-		  1 = integrated PCI port,
+		  0x300,0x310,0x320,0x330 = legacy port,
 		  0 = disable (default)
-    fm_port     - 0x388 = legacy port,
+    fm_port     - port address of OPL-3 FM synthesizer (8x38 only):
-		  1 = integrated PCI port (default),
+		  0x388 = legacy port,
+		  1 = integrated PCI port (default on 8738),
 		  0 = disable
     soft_ac3    - Software-conversion of raw SPDIF packets (model 033 only)
                   (default = 1)
@@ -768,6 +769,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     single_cmd  - Use single immediate commands to communicate with
 		codecs (for debugging only)
     enable_msi	- Enable Message Signaled Interrupt (MSI) (default = off)
+    power_save	- Automatic power-saving timtout (in second, 0 =
+		disable)
+    power_save_controller - Reset HD-audio controller in power-saving mode
+		(default = on)
 
     This module supports one card and autoprobe.
 
@@ -828,6 +833,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
 	ALC268
 	  3stack	3-stack model
+	  toshiba	Toshiba A205
+	  acer		Acer laptops
 	  auto		auto-config reading BIOS (default)
 
 	ALC662
@@ -842,7 +849,11 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  3stack-dig	3-jack with SPDIF I/O
 	  6stack-dig	6-jack digital with SPDIF I/O
 	  arima		Arima W820Di1
+	  targa		Targa T8, MSI-1049 T8
+	  asus-a7j	ASUS A7J
+	  asus-a7m	ASUS A7M
 	  macpro	MacPro support
+	  mbp3		Macbook Pro rev3
 	  imac24	iMac 24'' with jack detection
 	  w2jc		ASUS W2JC
 	  auto		auto-config reading BIOS (default)
@@ -854,6 +865,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  3stack-6ch-dig 3-jack 6-channel with SPDIF I/O
 	  6stack-dig-demo  6-jack digital for Intel demo board
 	  acer		Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc)
+	  acer-aspire	Acer Aspire 9810
 	  medion	Medion Laptops
 	  medion-md2	Medion MD2
 	  targa-dig	Targa/MSI
@@ -862,6 +874,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  lenovo-101e	Lenovo 101E
 	  lenovo-nb0763	Lenovo NB0763
 	  lenovo-ms7195-dig Lenovo MS7195
+	  haier-w66	Haier W66
 	  6stack-hp	HP machines with 6stack (Nettle boards)
 	  3stack-hp	HP machines with 3stack (Lucknow, Samba boards)
 	  auto		auto-config reading BIOS (default)
@@ -885,6 +898,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  3stack-660-digout 3-jack with SPDIF OUT (for ALC660VD)
 	  lenovo	Lenovo 3000 C200
 	  dallas	Dallas laptops
+	  hp		HP TX1000
 	  auto		auto-config reading BIOS (default)
 
 	CMI9880
@@ -920,6 +934,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  3stack	3-stack, shared surrounds
 	  laptop	2-channel only (FSC V2060, Samsung M50)
 	  laptop-eapd	2-channel with EAPD (Samsung R65, ASUS A6J)
+	  laptop-automute 2-channel with EAPD and HP-automute (Lenovo N100)
 	  ultra		2-channel with EAPD (Samsung Ultra tablet PC)
 
 	AD1988
@@ -945,14 +960,30 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 			can be adjusted.  Appearing only when compiled with
 			$CONFIG_SND_DEBUG=y
 
-	STAC9200/9205/9254
+	STAC9200
 	  ref		Reference board
+	  dell-d21	Dell (unknown)
+	  dell-d22	Dell (unknown)
+	  dell-d23	Dell (unknown)
+	  dell-m21	Dell Inspiron 630m, Dell Inspiron 640m
+	  dell-m22	Dell Latitude D620, Dell Latitude D820
+	  dell-m23	Dell XPS M1710, Dell Precision M90
+	  dell-m24	Dell Latitude 120L
+	  dell-m25	Dell Inspiron E1505n
+	  dell-m26	Dell Inspiron 1501
+	  dell-m27	Dell Inspiron E1705/9400
+	  gateway	Gateway laptops with EAPD control
+
+	STAC9205/9254
+	  ref		Reference board
+	  dell-m42	Dell (unknown)
+	  dell-m43	Dell Precision
+	  dell-m44	Dell Inspiron
 
 	STAC9220/9221
 	  ref		Reference board
 	  3stack	D945 3stack
 	  5stack	D945 5stack + SPDIF
-	  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
@@ -964,6 +995,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  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)
+	  dell-d81	Dell (unknown)
+	  dell-d82	Dell (unknown)
+	  dell-m81	Dell (unknown)
+	  dell-m82	Dell XPS M1210
 
 	STAC9202/9250/9251
 	  ref		Reference board, base config
@@ -975,6 +1010,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  ref		Reference board
 	  3stack	D965 3stack
 	  5stack	D965 5stack + SPDIF
+	  dell-3stack	Dell Dimension E520
 
 	STAC9872
 	  vaio		Setup for VAIO FE550G/SZ110
@@ -989,6 +1025,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     subsystem ID (output of "lspci -nv") to ALSA BTS or alsa-devel
     ML (see the section "Links and Addresses").
 
+    power_save and power_save_controller options are for power-saving
+    mode.  See powersave.txt for details.
+
     Note 2: If you get click noises on output, try the module option
 	    position_fix=1 or 2.  position_fix=1 will use the SD_LPIB
 	    register value without FIFO size correction as the current
@@ -1349,7 +1388,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     port	- port number or -1 (disable)
     irq		- IRQ number or -1 (disable)
     pnp		- PnP detection - 0 = disable, 1 = enable (default)
-    uart_enter	- Issue UART_ENTER command at open - bool, default = on
 
     This module supports multiple devices and PnP.
     
@@ -1630,6 +1668,21 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     
     The power-management is supported.
 
+  Module snd-sc6000
+  -----------------
+
+    Module for Gallant SC-6000 soundcard.
+
+    port	- Port # (0x220 or 0x240)
+    mss_port	- MSS Port # (0x530 or 0xe80)
+    irq		- IRQ # (5,7,9,10,11)
+    mpu_irq	- MPU-401 IRQ # (5,7,9,10) ,0 - no MPU-401 irq
+    dma		- DMA # (1,3,0)
+
+    This module supports multiple cards.
+
+    This card is also known as Audio Excel DSP 16 or Zoltrix AV302.
+
   Module snd-sgalaxy
   ------------------
 
@@ -1650,9 +1703,11 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     Module for ENSONIQ SoundScape PnP cards.
 
     port	- Port # (PnP setup)
+    wss_port	- WSS Port # (PnP setup)
     irq		- IRQ # (PnP setup)
     mpu_irq	- MPU-401 IRQ # (PnP setup)
     dma		- DMA # (PnP setup)
+    dma2	- 2nd DMA # (PnP setup, -1 to disable)
 
     This module supports multiple cards.  ISA PnP must be enabled.
     You need sscape_ctl tool in alsa-tools package for loading
@@ -1697,8 +1752,52 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     dma2            - DMA2 # for CS4232 PCM interface.
     isapnp          - ISA PnP detection - 0 = disable, 1 = enable (default)
 
+    The below are options for wavefront_synth features:
+    wf_raw	    - Assume that we need to boot the OS (default:no)
+	If yes, then during driver loading, the state of the board is
+	ignored, and we reset the board and load the firmware anyway.
+    fx_raw	    - Assume that the FX process needs help (default:yes)
+	If false, we'll leave the FX processor in whatever state it is
+	when the driver is loaded.  The default is to download the
+	microprogram and associated coefficients to set it up for
+	"default" operation, whatever that means.
+    debug_default   - Debug parameters for card initialization
+    wait_usecs	    - How long to wait without sleeping, usecs
+		      (default:150)
+	This magic number seems to give pretty optimal throughput
+	based on my limited experimentation. 
+	If you want to play around with it and find a better value, be
+	my guest. Remember, the idea is to get a number that causes us
+	to just busy wait for as many WaveFront commands as possible,
+	without coming up with a number so large that we hog the whole
+	CPU. 
+	Specifically, with this number, out of about 134,000 status
+	waits, only about 250 result in a sleep. 
+    sleep_interval  - How long to sleep when waiting for reply
+		      (default: 100)
+    sleep_tries	    - How many times to try sleeping during a wait
+		      (default: 50)
+    ospath	    - Pathname to processed ICS2115 OS firmware
+		      (default:wavefront.os)
+	The path name of the ISC2115 OS firmware.  In the recent
+	version, it's handled via firmware loader framework, so it
+	must be installed in the proper path, typically,
+	/lib/firmware.
+    reset_time	    - How long to wait for a reset to take effect
+		      (default:2)
+    ramcheck_time   - How many seconds to wait for the RAM test
+		      (default:20)
+    osrun_time	    - How many seconds to wait for the ICS2115 OS
+		      (default:10)
+
     This module supports multiple cards and ISA PnP.
 
+    Note: the firmware file "wavefront.os" was located in the earlier
+          version in /etc.  Now it's loaded via firmware loader, and
+	  must be in the proper firmware path, such as /lib/firmware.
+	  Copy (or symlink) the file appropriately if you get an error
+	  regarding firmware downloading after upgrading the kernel.
+
   Module snd-sonicvibes
   ---------------------
 

Documentation/sound/alsa/CMIPCI.txt

@@ -1,5 +1,5 @@
-              Brief Notes on C-Media 8738/8338 Driver
+         Brief Notes on C-Media 8338/8738/8768/8770 Driver
-              =======================================
+         =================================================
 
                    Takashi Iwai <tiwai@suse.de>
 
@@ -209,10 +209,13 @@ In addition to the standard SB mixer, CM8x38 provides more functions.
 MIDI CONTROLLER
 ---------------
 
-The MPU401-UART interface is disabled as default.  You need to set
+With CMI8338 chips, the MPU401-UART interface is disabled as default.
-module option "mpu_port" with a valid I/O port address to enable the
+You need to set the module option "mpu_port" to a valid I/O port address
-MIDI support.  The valid I/O ports are 0x300, 0x310, 0x320 and 0x330.
+to enable MIDI support.  Valid I/O ports are 0x300, 0x310, 0x320 and
-Choose the value which doesn't conflict with other cards.
+0x330.  Choose a value that doesn't conflict with other cards.
+
+With CMI8738 and newer chips, the MIDI interface is enabled by default
+and the driver automatically chooses a port address.
 
 There is _no_ hardware wavetable function on this chip (except for
 OPL3 synth below).
@@ -230,6 +233,8 @@ Set "fm_port" module option for more cards.
 The output quality of FM OPL/3 is, however, very weird.
 I don't know why..
 
+CMI8768 and newer chips do not have the FM synth.
+
 
 Joystick and Modem
 ------------------

Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl

@@ -18,8 +18,8 @@
       </affiliation>
      </author>
 
-     <date>November 17, 2005</date>
+     <date>September 10, 2007</date>
-     <edition>0.3.6</edition>
+     <edition>0.3.7</edition>
 
     <abstract>
       <para>
@@ -405,8 +405,9 @@
   /* definition of the chip-specific record */
   struct mychip {
           struct snd_card *card;
-          // rest of implementation will be in the section
+          /* rest of implementation will be in the section
-          // "PCI Resource Managements"
+           * "PCI Resource Managements"
+           */
   };
 
   /* chip-specific destructor
@@ -414,7 +415,7 @@
    */
   static int snd_mychip_free(struct mychip *chip)
   {
-          .... // will be implemented later...
+          .... /* will be implemented later... */
   }
 
   /* component-destructor
@@ -440,8 +441,9 @@
 
           *rchip = NULL;
 
-          // check PCI availability here
+          /* check PCI availability here
-          // (see "PCI Resource Managements")
+           * (see "PCI Resource Managements")
+           */
           ....
 
           /* allocate a chip-specific data with zero filled */
@@ -451,12 +453,13 @@
 
           chip->card = card;
 
-          // rest of initialization here; will be implemented
+          /* rest of initialization here; will be implemented
-          // later, see "PCI Resource Managements"
+           * later, see "PCI Resource Managements"
+           */
           ....
 
-          if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL,
+          err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
-                                    chip, &ops)) < 0) {
+          if (err < 0) {
                   snd_mychip_free(chip);
                   return err;
           }
@@ -490,7 +493,8 @@
                   return -ENOMEM;
 
           /* (3) */
-          if ((err = snd_mychip_create(card, pci, &chip)) < 0) {
+          err = snd_mychip_create(card, pci, &chip);
+          if (err < 0) {
                   snd_card_free(card);
                   return err;
           }
@@ -502,10 +506,11 @@
                   card->shortname, chip->ioport, chip->irq);
 
           /* (5) */
-          .... // implemented later
+          .... /* implemented later */
 
           /* (6) */
-          if ((err = snd_card_register(card)) < 0) {
+          err = snd_card_register(card);
+          if (err < 0) {
                   snd_card_free(card);
                   return err;
           }
@@ -605,7 +610,8 @@
 <![CDATA[
   struct mychip *chip;
   ....
-  if ((err = snd_mychip_create(card, pci, &chip)) < 0) {
+  err = snd_mychip_create(card, pci, &chip);
+  if (err < 0) {
           snd_card_free(card);
           return err;
   }
@@ -666,7 +672,8 @@
           <informalexample>
             <programlisting>
 <![CDATA[
-  if ((err = snd_card_register(card)) < 0) {
+  err = snd_card_register(card);
+  if (err < 0) {
           snd_card_free(card);
           return err;
   }
@@ -1091,7 +1098,7 @@
   static int snd_mychip_free(struct mychip *chip)
   {
           /* disable hardware here if any */
-          .... // (not implemented in this document)
+          .... /* (not implemented in this document) */
 
           /* release the irq */
           if (chip->irq >= 0)
@@ -1119,7 +1126,8 @@
           *rchip = NULL;
 
           /* initialize the PCI entry */
-          if ((err = pci_enable_device(pci)) < 0)
+          err = pci_enable_device(pci);
+          if (err < 0)
                   return err;
           /* check PCI availability (28bit DMA) */
           if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 ||
@@ -1141,7 +1149,8 @@
           chip->irq = -1;
 
           /* (1) PCI resource allocation */
-          if ((err = pci_request_regions(pci, "My Chip")) < 0) {
+          err = pci_request_regions(pci, "My Chip");
+          if (err < 0) {
                   kfree(chip);
                   pci_disable_device(pci);
                   return err;
@@ -1156,10 +1165,10 @@
           chip->irq = pci->irq;
 
           /* (2) initialization of the chip hardware */
-          .... //   (not implemented in this document)
+          .... /*   (not implemented in this document) */
 
-          if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL,
+          err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
-                                    chip, &ops)) < 0) {
+          if (err < 0) {
                   snd_mychip_free(chip);
                   return err;
           }
@@ -1233,7 +1242,8 @@
         <informalexample>
           <programlisting>
 <![CDATA[
-  if ((err = pci_enable_device(pci)) < 0)
+  err = pci_enable_device(pci);
+  if (err < 0)
           return err;
   if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 ||
       pci_set_consistent_dma_mask(pci, DMA_28BIT_MASK) < 0) {
@@ -1294,7 +1304,8 @@
         <informalexample>
           <programlisting>
 <![CDATA[
-  if ((err = pci_request_regions(pci, "My Chip")) < 0) { 
+  err = pci_request_regions(pci, "My Chip");
+  if (err < 0) { 
           kfree(chip);
           pci_disable_device(pci);
           return err;
@@ -1322,7 +1333,7 @@
           <programlisting>
 <![CDATA[
   if (request_irq(pci->irq, snd_mychip_interrupt,
-                  IRQF_DISABLED|IRQF_SHARED, "My Chip", chip)) {
+                  IRQF_SHARED, "My Chip", chip)) {
           printk(KERN_ERR "cannot grab irq %d\n", pci->irq);
           snd_mychip_free(chip);
           return -EBUSY;
@@ -1773,7 +1784,8 @@
           struct snd_pcm_runtime *runtime = substream->runtime;
 
           runtime->hw = snd_mychip_playback_hw;
-          // more hardware-initialization will be done here
+          /* more hardware-initialization will be done here */
+          ....
           return 0;
   }
 
@@ -1781,7 +1793,8 @@
   static int snd_mychip_playback_close(struct snd_pcm_substream *substream)
   {
           struct mychip *chip = snd_pcm_substream_chip(substream);
-          // the hardware-specific codes will be here
+          /* the hardware-specific codes will be here */
+          ....
           return 0;
 
   }
@@ -1793,7 +1806,8 @@
           struct snd_pcm_runtime *runtime = substream->runtime;
 
           runtime->hw = snd_mychip_capture_hw;
-          // more hardware-initialization will be done here
+          /* more hardware-initialization will be done here */
+          ....
           return 0;
   }
 
@@ -1801,7 +1815,8 @@
   static int snd_mychip_capture_close(struct snd_pcm_substream *substream)
   {
           struct mychip *chip = snd_pcm_substream_chip(substream);
-          // the hardware-specific codes will be here
+          /* the hardware-specific codes will be here */
+          ....
           return 0;
 
   }
@@ -1844,10 +1859,12 @@
   {
           switch (cmd) {
           case SNDRV_PCM_TRIGGER_START:
-                  // do something to start the PCM engine
+                  /* do something to start the PCM engine */
+                  ....
                   break;
           case SNDRV_PCM_TRIGGER_STOP:
-                  // do something to stop the PCM engine
+                  /* do something to stop the PCM engine */
+                  ....
                   break;
           default:
                   return -EINVAL;
@@ -1900,8 +1917,8 @@
           struct snd_pcm *pcm;
           int err;
 
-          if ((err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1,
+          err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, &pcm);
-                                 &pcm)) < 0) 
+          if (err < 0) 
                   return err;
           pcm->private_data = chip;
           strcpy(pcm->name, "My Chip");
@@ -1939,8 +1956,8 @@
           struct snd_pcm *pcm;
           int err;
 
-          if ((err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1,
+          err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, &pcm);
-                                 &pcm)) < 0) 
+          if (err < 0) 
                   return err;
           pcm->private_data = chip;
           strcpy(pcm->name, "My Chip");
@@ -2097,7 +2114,7 @@
           struct mychip *chip = snd_pcm_chip(pcm);
           /* free your own data */
           kfree(chip->my_private_pcm_data);
-          // do what you like else
+          /* do what you like else */
           ....
   }
 
@@ -2884,10 +2901,10 @@ struct _snd_pcm_runtime {
 <![CDATA[
   switch (cmd) {
   case SNDRV_PCM_TRIGGER_START:
-          // do something to start the PCM engine
+          /* do something to start the PCM engine */
           break;
   case SNDRV_PCM_TRIGGER_STOP:
-          // do something to stop the PCM engine
+          /* do something to stop the PCM engine */
           break;
   default:
           return -EINVAL;
@@ -3071,7 +3088,7 @@ struct _snd_pcm_runtime {
                   spin_unlock(&chip->lock);
                   snd_pcm_period_elapsed(chip->substream);
                   spin_lock(&chip->lock);
-                  // acknowledge the interrupt if necessary
+                  /* acknowledge the interrupt if necessary */
           }
           ....
           spin_unlock(&chip->lock);
@@ -3134,7 +3151,7 @@ struct _snd_pcm_runtime {
                           snd_pcm_period_elapsed(substream);
                           spin_lock(&chip->lock);
                   }
-                  // acknowledge the interrupt if necessary
+                  /* acknowledge the interrupt if necessary */
           }
           ....
           spin_unlock(&chip->lock);
@@ -3455,6 +3472,13 @@ struct _snd_pcm_runtime {
       (casted to unsigned long) of some record to this field, too. 
       </para>
 
+      <para>
+      The <structfield>tlv</structfield> field can be used to provide
+      metadata about the control; see the
+      <link linkend="control-interface-tlv">
+      <citetitle>Metadata</citetitle></link> subsection.
+      </para>
+
       <para>
         The other three are
 	<link linkend="control-interface-callbacks"><citetitle>
@@ -3604,7 +3628,7 @@ struct _snd_pcm_runtime {
 	    <title>Example of info callback</title>
             <programlisting>
 <![CDATA[
-  static int snd_myctl_info(struct snd_kcontrol *kcontrol,
+  static int snd_myctl_mono_info(struct snd_kcontrol *kcontrol,
                           struct snd_ctl_elem_info *uinfo)
   {
           uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
@@ -3639,7 +3663,7 @@ struct _snd_pcm_runtime {
           <informalexample>
             <programlisting>
 <![CDATA[
-  static int snd_myctl_info(struct snd_kcontrol *kcontrol,
+  static int snd_myctl_enum_info(struct snd_kcontrol *kcontrol,
                           struct snd_ctl_elem_info *uinfo)
   {
           static char *texts[4] = {
@@ -3658,6 +3682,16 @@ struct _snd_pcm_runtime {
             </programlisting>
           </informalexample>
         </para>
+
+        <para>
+	  Some common info callbacks are prepared for easy use:
+	<function>snd_ctl_boolean_mono_info()</function> and
+	<function>snd_ctl_boolean_stereo_info()</function>.
+	Obviously, the former is an info callback for a mono channel
+	boolean item, just like <function>snd_myctl_mono_info</function>
+	above, and the latter is for a stereo channel boolean item.
+	</para>
+
       </section>
 
       <section id="control-interface-callbacks-get">
@@ -3794,7 +3828,8 @@ struct _snd_pcm_runtime {
         <informalexample>
           <programlisting>
 <![CDATA[
-  if ((err = snd_ctl_add(card, snd_ctl_new1(&my_control, chip))) < 0)
+  err = snd_ctl_add(card, snd_ctl_new1(&my_control, chip));
+  if (err < 0)
           return err;
 ]]>
           </programlisting>
@@ -3843,6 +3878,56 @@ struct _snd_pcm_runtime {
       </para>
     </section>
 
+    <section id="control-interface-tlv">
+      <title>Metadata</title>
+      <para>
+      To provide information about the dB values of a mixer control, use
+      on of the <constant>DECLARE_TLV_xxx</constant> macros from
+      <filename>&lt;sound/tlv.h&gt;</filename> to define a variable
+      containing this information, set the<structfield>tlv.p
+      </structfield> field to point to this variable, and include the
+      <constant>SNDRV_CTL_ELEM_ACCESS_TLV_READ</constant> flag in the
+      <structfield>access</structfield> field; like this:
+      <informalexample>
+        <programlisting>
+<![CDATA[
+  static DECLARE_TLV_DB_SCALE(db_scale_my_control, -4050, 150, 0);
+
+  static struct snd_kcontrol_new my_control __devinitdata = {
+          ...
+          .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+                    SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+          ...
+          .tlv.p = db_scale_my_control,
+  };
+]]>
+        </programlisting>
+      </informalexample>
+      </para>
+
+      <para>
+      The <function>DECLARE_TLV_DB_SCALE</function> macro defines
+      information about a mixer control where each step in the control's
+      value changes the dB value by a constant dB amount.
+      The first parameter is the name of the variable to be defined.
+      The second parameter is the minimum value, in units of 0.01 dB.
+      The third parameter is the step size, in units of 0.01 dB.
+      Set the fourth parameter to 1 if the minimum value actually mutes
+      the control.
+      </para>
+
+      <para>
+      The <function>DECLARE_TLV_DB_LINEAR</function> macro defines
+      information about a mixer control where the control's value affects
+      the output linearly.
+      The first parameter is the name of the variable to be defined.
+      The second parameter is the minimum value, in units of 0.01 dB.
+      The third parameter is the maximum value, in units of 0.01 dB.
+      If the minimum value mutes the control, set the second parameter to
+      <constant>TLV_DB_GAIN_MUTE</constant>.
+      </para>
+    </section>
+
   </chapter>
 
 
@@ -3880,7 +3965,7 @@ struct _snd_pcm_runtime {
   {
           struct mychip *chip = ac97->private_data;
           ....
-          // read a register value here from the codec
+          /* read a register value here from the codec */
           return the_register_value;
   }
 
@@ -3889,7 +3974,7 @@ struct _snd_pcm_runtime {
   {
           struct mychip *chip = ac97->private_data;
           ....
-          // write the given register value to the codec
+          /* write the given register value to the codec */
   }
 
   static int snd_mychip_ac97(struct mychip *chip)
@@ -3902,7 +3987,8 @@ struct _snd_pcm_runtime {
                   .read = snd_mychip_ac97_read,
           };
 
-          if ((err = snd_ac97_bus(chip->card, 0, &ops, NULL, &bus)) < 0)
+          err = snd_ac97_bus(chip->card, 0, &ops, NULL, &bus);
+          if (err < 0)
                   return err;
           memset(&ac97, 0, sizeof(ac97));
           ac97.private_data = chip;
@@ -4447,10 +4533,10 @@ struct _snd_pcm_runtime {
         <informalexample>
           <programlisting>
 <![CDATA[
-  struct list_head *list;
   struct snd_rawmidi_substream *substream;
-  list_for_each(list, &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams) {
+  list_for_each_entry(substream,
-          substream = list_entry(list, struct snd_rawmidi_substream, list);
+                      &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams,
+                      list {
           sprintf(substream->name, "My MIDI Port %d", substream->number + 1);
   }
   /* same for SNDRV_RAWMIDI_STREAM_INPUT */

Documentation/sound/alsa/OSS-Emulation.txt

@@ -303,10 +303,3 @@ ICE1712 supports only the unconventional format, interleaved
 the buffer as the conventional (mono or 2-channels, 8 or 16bit) format
 on OSS.
 
-USB devices
------------
-Some USB devices support only 24bit format packed in 3bytes.  This
-format is not supported by OSS and no conversion is provided by kernel
-OSS emulation.  You can use the user-space OSS emulation via libaoss
-instead.
-

Documentation/sound/alsa/hda_codec.txt

@@ -49,6 +49,9 @@ struct hda_bus_ops {
 		       unsigned int verb, unsigned int parm);
 	unsigned int (*get_response)(struct hda_codec *codec);
 	void (*private_free)(struct hda_bus *);
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+	void (*pm_notify)(struct hda_codec *codec);
+#endif
 };
 
 The command callback is called when the codec module needs to send a
@@ -56,9 +59,16 @@ VERB to the controller.  It's always a single command.
 The get_response callback is called when the codec requires the answer
 for the last command.  These two callbacks are mandatory and have to
 be given.
-The last, private_free callback, is optional.  It's called in the
+The third, private_free callback, is optional.  It's called in the
 destructor to release any necessary data in the lowlevel driver.
 
+The pm_notify callback is available only with
+CONFIG_SND_HDA_POWER_SAVE kconfig.  It's called when the codec needs
+to power up or may power down.  The controller should check the all
+belonging codecs on the bus whether they are actually powered off
+(check codec->power_on), and optionally the driver may power down the
+contoller side, too.
+
 The bus instance is created via snd_hda_bus_new().  You need to pass
 the card instance, the template, and the pointer to store the
 resultant bus instance.
@@ -86,10 +96,8 @@ resultant codec instance (can be NULL if not needed).
 The codec is stored in a linked list of bus instance.  You can follow
 the codec list like:
 
-	struct list_head *p;
 	struct hda_codec *codec;
-	list_for_each(p, &bus->codec_list) {
+	list_for_each_entry(codec, &bus->codec_list, list) {
-		codec = list_entry(p, struct hda_codec, list);
 		...
 	}
 
@@ -100,10 +108,15 @@ initialization sequence is called when the controls are built later.
 Codec Access
 ============
 
-To access codec, use snd_codec_read() and snd_codec_write().
+To access codec, use snd_hda_codec_read() and snd_hda_codec_write().
 snd_hda_param_read() is for reading parameters.
 For writing a sequence of verbs, use snd_hda_sequence_write().
 
+There are variants of cached read/write, snd_hda_codec_write_cache(),
+snd_hda_sequence_write_cache().  These are used for recording the
+register states for the power-mangement resume.  When no PM is needed,
+these are equivalent with non-cached version.
+
 To retrieve the number of sub nodes connected to the given node, use
 snd_hda_get_sub_nodes().  The connection list can be obtained via
 snd_hda_get_connections() call.
@@ -239,6 +252,10 @@ set the codec->patch_ops field.  This is defined as below:
 		int (*suspend)(struct hda_codec *codec, pm_message_t state);
 		int (*resume)(struct hda_codec *codec);
 	#endif
+	#ifdef CONFIG_SND_HDA_POWER_SAVE
+		int (*check_power_status)(struct hda_codec *codec,
+					  hda_nid_t nid);
+	#endif
 	};
 
 The build_controls callback is called from snd_hda_build_controls().
@@ -251,6 +268,18 @@ The unsol_event callback is called when an unsolicited event is
 received.
 
 The suspend and resume callbacks are for power management.
+They can be NULL if no special sequence is required.  When the resume
+callback is NULL, the driver calls the init callback and resumes the
+registers from the cache.  If other handling is needed, you'd need to
+write your own resume callback.  There, the amp values can be resumed
+via
+	void snd_hda_codec_resume_amp(struct hda_codec *codec);
+and the other codec registers via
+	void snd_hda_codec_resume_cache(struct hda_codec *codec);
+
+The check_power_status callback is called when the amp value of the
+given widget NID is changed.  The codec code can turn on/off the power
+appropriately from this information.
 
 Each entry can be NULL if not necessary to be called.
 
@@ -267,8 +296,7 @@ Digital I/O
 ===========
 
 Call snd_hda_create_spdif_out_ctls() from the patch to create controls
-related with SPDIF out.  In the patch resume callback, call
+related with SPDIF out.
-snd_hda_resume_spdif().
 
 
 Helper Functions
@@ -284,12 +312,7 @@ as a module parameter, and PCI subsystem IDs.  If the matching entry
 is found, it returns the config field value.
 
 snd_hda_add_new_ctls() can be used to create and add control entries.
-Pass the zero-terminated array of struct snd_kcontrol_new.  The same array
+Pass the zero-terminated array of struct snd_kcontrol_new
-can be passed to snd_hda_resume_ctls() for resume.
-Note that this will call control->put callback of these entries.  So,
-put callback should check codec->in_resume and force to restore the
-given value if it's non-zero even if the value is identical with the
-cached value.
 
 Macros HDA_CODEC_VOLUME(), HDA_CODEC_MUTE() and their variables can be
 used for the entry of struct snd_kcontrol_new.

Documentation/sound/alsa/powersave.txt

@@ -0,0 +1,41 @@
+Notes on Power-Saving Mode
+==========================
+
+AC97 and HD-audio drivers have the automatic power-saving mode.
+This feature is enabled via Kconfig CONFIG_SND_AC97_POWER_SAVE
+and CONFIG_SND_HDA_POWER_SAVE options, respectively.
+
+With the automatic power-saving, the driver turns off the codec power
+appropriately when no operation is required.  When no applications use
+the device and/or no analog loopback is set, the power disablement is
+done fully or partially.  It'll save a certain power consumption, thus
+good for laptops (even for desktops).
+
+The time-out for automatic power-off can be specified via power_save
+module option of snd-ac97-codec and snd-hda-intel modules.  Specify
+the time-out value in seconds.  0 means to disable the automatic
+power-saving.  The default value of timeout is given via
+CONFIG_SND_AC97_POWER_SAVE_DEFAULT and
+CONFIG_SND_HDA_POWER_SAVE_DEFAULT Kconfig options.  Setting this to 1
+(the minimum value) isn't recommended because many applications try to
+reopen the device frequently.  10 would be a good choice for normal
+operations.
+
+The power_save option is exported as writable.  This means you can
+adjust the value via sysfs on the fly.  For example, to turn on the
+automatic power-save mode with 10 seconds, write to
+/sys/modules/snd_ac97_codec/parameters/power_save (usually as root):
+
+	# echo 10 > /sys/modules/snd_ac97_codec/parameters/power_save
+
+
+Note that you might hear click noise/pop when changing the power
+state.  Also, it often takes certain time to wake up from the
+power-down to the active state.  These are often hardly to fix, so
+don't report extra bug reports unless you have a fix patch ;-)
+
+For HD-audio interface, there is another module option,
+power_save_controller.  This enables/disables the power-save mode of
+the controller side.  Setting this on may reduce a bit more power
+consumption, but might result in longer wake-up time and click noise.
+Try to turn it off when you experience such a thing too often.

MAINTAINERS

@@ -1769,7 +1769,7 @@ S:	Maintained
 
 HP100:	Driver for HP 10/100 Mbit/s Voice Grade Network Adapter Series
 P:	Jaroslav Kysela
-M:	perex@suse.cz
+M:	perex@perex.cz
 S:	Maintained
 
 HPET:	High Precision Event Timers driver (hpet.c)
@@ -2132,7 +2132,7 @@ S:	Maintained
 
 ISAPNP
 P:	Jaroslav Kysela
-M:	perex@suse.cz
+M:	perex@perex.cz
 S:	Maintained
 
 ISDN SUBSYSTEM
@@ -3523,7 +3523,7 @@ S:	Maintained
 
 SOUND
 P:	Jaroslav Kysela
-M:	perex@suse.cz
+M:	perex@perex.cz
 L:	alsa-devel@alsa-project.org (subscribers-only)
 S:	Maintained
 

drivers/media/video/cx88/cx88-alsa.c

@@ -7,7 +7,7 @@
  *    (c) 2005,2006 Ricardo Cerqueira <v4l@cerqueira.org>
  *    (c) 2005 Mauro Carvalho Chehab <mchehab@infradead.org>
  *    Based on a dummy cx88 module by Gerd Knorr <kraxel@bytesex.org>
- *    Based on dummy.c by Jaroslav Kysela <perex@suse.cz>
+ *    Based on dummy.c by Jaroslav Kysela <perex@perex.cz>
  *
  *  This program is free software; you can redistribute it and/or modify
  *  it under the terms of the GNU General Public License as published by

drivers/net/hp100.c

@@ -8,7 +8,7 @@
 ** Extended for new busmaster capable chipsets by
 ** Siegfried "Frieder" Loeffler (dg1sek) <floeff@mathematik.uni-stuttgart.de>
 **
-** Maintained by: Jaroslav Kysela <perex@suse.cz>
+** Maintained by: Jaroslav Kysela <perex@perex.cz>
 **
 ** This driver has only been tested with
 ** -- HP J2585B 10/100 Mbit/s PCI Busmaster
@@ -2951,7 +2951,7 @@ static struct pci_driver hp100_pci_driver = {
  */
 
 MODULE_LICENSE("GPL");
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>, "
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, "
               "Siegfried \"Frieder\" Loeffler (dg1sek) <floeff@mathematik.uni-stuttgart.de>");
 MODULE_DESCRIPTION("HP CASCADE Architecture Driver for 100VG-AnyLan Network Adapters");
 

drivers/pnp/interface.c

@@ -1,7 +1,7 @@
 /*
  * interface.c - contains everything related to the user interface
  *
- * Some code, especially possible resource dumping is based on isapnp_proc.c (c) Jaroslav Kysela <perex@suse.cz>
+ * Some code, especially possible resource dumping is based on isapnp_proc.c (c) Jaroslav Kysela <perex@perex.cz>
  * Copyright 2002 Adam Belay <ambx1@neo.rr.com>
  */
 

drivers/pnp/isapnp/core.c

@@ -1,6 +1,6 @@
 /*
  *  ISA Plug & Play support
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify
@@ -53,7 +53,7 @@ static int isapnp_rdp;		/* Read Data Port */
 static int isapnp_reset = 1;	/* reset all PnP cards (deactivate) */
 static int isapnp_verbose = 1;	/* verbose mode */
 
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("Generic ISA Plug & Play support");
 module_param(isapnp_disable, int, 0);
 MODULE_PARM_DESC(isapnp_disable, "ISA Plug & Play disable");

drivers/pnp/isapnp/proc.c

@@ -1,6 +1,6 @@
 /*
  *  ISA Plug & Play support
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *   This program is free software; you can redistribute it and/or modify
  *   it under the terms of the GNU General Public License as published by

drivers/pnp/manager.c

@@ -1,7 +1,7 @@
 /*
  * manager.c - Resource Management, Conflict Resolution, Activation and Disabling of Devices
  *
- * based on isapnp.c resource management (c) Jaroslav Kysela <perex@suse.cz>
+ * based on isapnp.c resource management (c) Jaroslav Kysela <perex@perex.cz>
  * Copyright 2003 Adam Belay <ambx1@neo.rr.com>
  */
 

drivers/pnp/resource.c

@@ -1,7 +1,7 @@
 /*
  * resource.c - Contains functions for registering and analyzing resource information
  *
- * based on isapnp.c resource management (c) Jaroslav Kysela <perex@suse.cz>
+ * based on isapnp.c resource management (c) Jaroslav Kysela <perex@perex.cz>
  * Copyright 2003 Adam Belay <ambx1@neo.rr.com>
  */
 

include/linux/i2c-id.h

@@ -120,6 +120,7 @@
 #define I2C_DRIVERID_WM8753	91	/* Wolfson WM8753 audio codec */
 #define I2C_DRIVERID_LM4857 	92 	/* LM4857 Audio Amplifier */
 #define I2C_DRIVERID_VP27SMPX	93	/* Panasonic VP27s tuner internal MPX */
+#define I2C_DRIVERID_CS4270	94	/* Cirrus Logic 4270 audio codec */
 
 #define I2C_DRIVERID_I2CDEV	900
 #define I2C_DRIVERID_ARP        902    /* SMBus ARP Client              */

include/linux/spi/at73c213.h

@@ -0,0 +1,25 @@
+/*
+ * Board-specific data used to set up AT73c213 audio DAC driver.
+ */
+
+#ifndef __LINUX_SPI_AT73C213_H
+#define __LINUX_SPI_AT73C213_H
+
+/**
+ * at73c213_board_info - how the external DAC is wired to the device.
+ *
+ * @ssc_id: SSC platform_driver id the DAC shall use to stream the audio.
+ * @dac_clk: the external clock used to provide master clock to the DAC.
+ * @shortname: a short discription for the DAC, seen by userspace tools.
+ *
+ * This struct contains the configuration of the hardware connection to the
+ * external DAC. The DAC needs a master clock and a I2S audio stream. It also
+ * provides a name which is used to identify it in userspace tools.
+ */
+struct at73c213_board_info {
+	int		ssc_id;
+	struct clk	*dac_clk;
+	char		shortname[32];
+};
+
+#endif /* __LINUX_SPI_AT73C213_H */

include/sound/ac97_codec.h

@@ -2,7 +2,7 @@
 #define __SOUND_AC97_CODEC_H
 
 /*
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *  Universal interface for Audio Codec '97
  *
  *  For more details look to AC '97 component specification revision 2.1
@@ -345,9 +345,9 @@
 #define AC97_ALC650_GPIO_STATUS		0x78
 #define AC97_ALC650_CLOCK		0x7a
 
-/* specific - Yamaha YMF753 */
+/* specific - Yamaha YMF7x3 */
-#define AC97_YMF753_DIT_CTRL2	0x66	/* DIT Control 2 */
+#define AC97_YMF7X3_DIT_CTRL	0x66	/* DIT Control (YMF743) / 2 (YMF753) */
-#define AC97_YMF753_3D_MODE_SEL	0x68	/* 3D Mode Select */
+#define AC97_YMF7X3_3D_MODE_SEL	0x68	/* 3D Mode Select */
 
 /* specific - C-Media */
 #define AC97_CM9738_VENDOR_CTRL	0x5a

include/sound/ad1848.h

@@ -2,7 +2,7 @@
 #define __SOUND_AD1848_H
 
 /*
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *  Definitions for AD1847/AD1848/CS4248 chips
  *
  *
@@ -27,7 +27,7 @@
 
 /* IO ports */
 
-#define AD1848P( codec, x ) ( (chip) -> port + c_d_c_AD1848##x )
+#define AD1848P( chip, x ) ( (chip) -> port + c_d_c_AD1848##x )
 
 #define c_d_c_AD1848REGSEL	0
 #define c_d_c_AD1848REG		1
@@ -154,7 +154,6 @@ struct snd_ad1848 {
 #endif
 
 	spinlock_t reg_lock;
-	struct mutex open_mutex;
 };
 
 /* exported functions */

include/sound/ainstr_gf1.h

@@ -2,7 +2,7 @@
  *  Advanced Linux Sound Architecture
  *
  *  GF1 (GUS) Patch Instrument Format
- *  Copyright (c) 1994-99 by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) 1994-99 by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify

include/sound/ainstr_iw.h

@@ -2,7 +2,7 @@
  *  Advanced Linux Sound Architecture
  *
  *  InterWave FFFF Instrument Format
- *  Copyright (c) 1994-99 by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) 1994-99 by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify

include/sound/ainstr_simple.h

@@ -2,7 +2,7 @@
  *  Advanced Linux Sound Architecture
  *
  *  Simple (MOD player) Instrument Format
- *  Copyright (c) 1994-99 by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) 1994-99 by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify

include/sound/ak4114.h

@@ -3,7 +3,7 @@
 
 /*
  *  Routines for Asahi Kasei AK4114
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>,
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
  *
  *
  *   This program is free software; you can redistribute it and/or modify

include/sound/ak4117.h

@@ -3,7 +3,7 @@
 
 /*
  *  Routines for Asahi Kasei AK4117
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>,
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
  *
  *
  *   This program is free software; you can redistribute it and/or modify

include/sound/ak4531_codec.h

@@ -2,7 +2,7 @@
 #define __SOUND_AK4531_CODEC_H
 
 /*
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *  Universal interface for Audio Codec '97
  *
  *  For more details look to AC '97 component specification revision 2.1

include/sound/ak4xxx-adda.h

@@ -5,7 +5,7 @@
  *   ALSA driver for AK4524 / AK4528 / AK4529 / AK4355 / AK4381
  *   AD and DA converters
  *
- *	Copyright (c) 2000 Jaroslav Kysela <perex@suse.cz>
+ *	Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
  *
  *   This program is free software; you can redistribute it and/or modify
  *   it under the terms of the GNU General Public License as published by

include/sound/asequencer.h

@@ -1,7 +1,7 @@
 /*
  *  Main header file for the ALSA sequencer
  *  Copyright (c) 1998-1999 by Frank van de Pol <fvdpol@coil.demon.nl>
- *            (c) 1998-1999 by Jaroslav Kysela <perex@suse.cz>
+ *            (c) 1998-1999 by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify

include/sound/asound.h

@@ -1,6 +1,6 @@
 /*
  *  Advanced Linux Sound Architecture - ALSA - Driver
- *  Copyright (c) 1994-2003 by Jaroslav Kysela <perex@suse.cz>,
+ *  Copyright (c) 1994-2003 by Jaroslav Kysela <perex@perex.cz>,
  *                             Abramo Bagnara <abramo@alsa-project.org>
  *
  *
@@ -92,6 +92,7 @@ enum {
 	SNDRV_HWDEP_IFACE_USX2Y_PCM,	/* Tascam US122, US224 & US428 rawusb pcm */
 	SNDRV_HWDEP_IFACE_PCXHR,	/* Digigram PCXHR */
 	SNDRV_HWDEP_IFACE_SB_RC,	/* SB Extigy/Audigy2NX remote control */
+	SNDRV_HWDEP_IFACE_HDA,		/* HD-audio */
 
 	/* Don't forget to change the following: */
 	SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_SB_RC

include/sound/asound_fm.h

@@ -5,7 +5,7 @@
  *  Advanced Linux Sound Architecture - ALSA
  *
  *  Interface file between ALSA driver & user space
- *  Copyright (c) 1994-98 by Jaroslav Kysela <perex@suse.cz>,
+ *  Copyright (c) 1994-98 by Jaroslav Kysela <perex@perex.cz>,
  *                           4Front Technologies
  *
  *  Direct FM control

include/sound/asoundef.h

@@ -3,7 +3,7 @@
 
 /*
  *  Advanced Linux Sound Architecture - ALSA - Driver
- *  Copyright (c) 1994-2000 by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) 1994-2000 by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify

include/sound/control.h

@@ -3,7 +3,7 @@
 
 /*
  *  Header file for control interface
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify
@@ -161,4 +161,12 @@ static inline struct snd_ctl_elem_id *snd_ctl_build_ioff(struct snd_ctl_elem_id
 	return dst_id;
 }
 
+/*
+ * Frequently used control callbacks
+ */
+int snd_ctl_boolean_mono_info(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_info *uinfo);
+int snd_ctl_boolean_stereo_info(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_info *uinfo);
+
 #endif	/* __SOUND_CONTROL_H */

include/sound/core.h

@@ -3,7 +3,7 @@
 
 /*
  *  Main header file for the ALSA driver
- *  Copyright (c) 1994-2001 by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) 1994-2001 by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify

include/sound/cs4231-regs.h

@@ -0,0 +1,180 @@
+#ifndef __SOUND_CS4231_REGS_H
+#define __SOUND_CS4231_REGS_H
+
+/*
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *  Definitions for CS4231 & InterWave chips & compatible chips registers
+ *
+ *
+ *   This program is free software; you can redistribute it and/or modify
+ *   it under the terms of the GNU General Public License as published by
+ *   the Free Software Foundation; either version 2 of the License, or
+ *   (at your option) any later version.
+ *
+ *   This program is distributed in the hope that it will be useful,
+ *   but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *   GNU General Public License for more details.
+ *
+ *   You should have received a copy of the GNU General Public License
+ *   along with this program; if not, write to the Free Software
+ *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ */
+
+/* IO ports */
+
+#define CS4231P(x)		(c_d_c_CS4231##x)
+
+#define c_d_c_CS4231REGSEL	0
+#define c_d_c_CS4231REG		1
+#define c_d_c_CS4231STATUS	2
+#define c_d_c_CS4231PIO		3
+
+/* codec registers */
+
+#define CS4231_LEFT_INPUT	0x00	/* left input control */
+#define CS4231_RIGHT_INPUT	0x01	/* right input control */
+#define CS4231_AUX1_LEFT_INPUT	0x02	/* left AUX1 input control */
+#define CS4231_AUX1_RIGHT_INPUT	0x03	/* right AUX1 input control */
+#define CS4231_AUX2_LEFT_INPUT	0x04	/* left AUX2 input control */
+#define CS4231_AUX2_RIGHT_INPUT	0x05	/* right AUX2 input control */
+#define CS4231_LEFT_OUTPUT	0x06	/* left output control register */
+#define CS4231_RIGHT_OUTPUT	0x07	/* right output control register */
+#define CS4231_PLAYBK_FORMAT	0x08	/* clock and data format - playback - bits 7-0 MCE */
+#define CS4231_IFACE_CTRL	0x09	/* interface control - bits 7-2 MCE */
+#define CS4231_PIN_CTRL		0x0a	/* pin control */
+#define CS4231_TEST_INIT	0x0b	/* test and initialization */
+#define CS4231_MISC_INFO	0x0c	/* miscellaneaous information */
+#define CS4231_LOOPBACK		0x0d	/* loopback control */
+#define CS4231_PLY_UPR_CNT	0x0e	/* playback upper base count */
+#define CS4231_PLY_LWR_CNT	0x0f	/* playback lower base count */
+#define CS4231_ALT_FEATURE_1	0x10	/* alternate #1 feature enable */
+#define AD1845_AF1_MIC_LEFT	0x10	/* alternate #1 feature + MIC left */
+#define CS4231_ALT_FEATURE_2	0x11	/* alternate #2 feature enable */
+#define AD1845_AF2_MIC_RIGHT	0x11	/* alternate #2 feature + MIC right */
+#define CS4231_LEFT_LINE_IN	0x12	/* left line input control */
+#define CS4231_RIGHT_LINE_IN	0x13	/* right line input control */
+#define CS4231_TIMER_LOW	0x14	/* timer low byte */
+#define CS4231_TIMER_HIGH	0x15	/* timer high byte */
+#define CS4231_LEFT_MIC_INPUT	0x16	/* left MIC input control register (InterWave only) */
+#define AD1845_UPR_FREQ_SEL	0x16	/* upper byte of frequency select */
+#define CS4231_RIGHT_MIC_INPUT	0x17	/* right MIC input control register (InterWave only) */
+#define AD1845_LWR_FREQ_SEL	0x17	/* lower byte of frequency select */
+#define CS4236_EXT_REG		0x17	/* extended register access */
+#define CS4231_IRQ_STATUS	0x18	/* irq status register */
+#define CS4231_LINE_LEFT_OUTPUT	0x19	/* left line output control register (InterWave only) */
+#define CS4231_VERSION		0x19	/* CS4231(A) - version values */
+#define CS4231_MONO_CTRL	0x1a	/* mono input/output control */
+#define CS4231_LINE_RIGHT_OUTPUT 0x1b	/* right line output control register (InterWave only) */
+#define AD1845_PWR_DOWN		0x1b	/* power down control */
+#define CS4235_LEFT_MASTER	0x1b	/* left master output control */
+#define CS4231_REC_FORMAT	0x1c	/* clock and data format - record - bits 7-0 MCE */
+#define CS4231_PLY_VAR_FREQ	0x1d	/* playback variable frequency */
+#define AD1845_CLOCK		0x1d	/* crystal clock select and total power down */
+#define CS4235_RIGHT_MASTER	0x1d	/* right master output control */
+#define CS4231_REC_UPR_CNT	0x1e	/* record upper count */
+#define CS4231_REC_LWR_CNT	0x1f	/* record lower count */
+
+/* definitions for codec register select port - CODECP( REGSEL ) */
+
+#define CS4231_INIT		0x80	/* CODEC is initializing */
+#define CS4231_MCE		0x40	/* mode change enable */
+#define CS4231_TRD		0x20	/* transfer request disable */
+
+/* definitions for codec status register - CODECP( STATUS ) */
+
+#define CS4231_GLOBALIRQ	0x01	/* IRQ is active */
+
+/* definitions for codec irq status */
+
+#define CS4231_PLAYBACK_IRQ	0x10
+#define CS4231_RECORD_IRQ	0x20
+#define CS4231_TIMER_IRQ	0x40
+#define CS4231_ALL_IRQS		0x70
+#define CS4231_REC_UNDERRUN	0x08
+#define CS4231_REC_OVERRUN	0x04
+#define CS4231_PLY_OVERRUN	0x02
+#define CS4231_PLY_UNDERRUN	0x01
+
+/* definitions for CS4231_LEFT_INPUT and CS4231_RIGHT_INPUT registers */
+
+#define CS4231_ENABLE_MIC_GAIN	0x20
+
+#define CS4231_MIXS_LINE	0x00
+#define CS4231_MIXS_AUX1	0x40
+#define CS4231_MIXS_MIC		0x80
+#define CS4231_MIXS_ALL		0xc0
+
+/* definitions for clock and data format register - CS4231_PLAYBK_FORMAT */
+
+#define CS4231_LINEAR_8		0x00	/* 8-bit unsigned data */
+#define CS4231_ALAW_8		0x60	/* 8-bit A-law companded */
+#define CS4231_ULAW_8		0x20	/* 8-bit U-law companded */
+#define CS4231_LINEAR_16	0x40	/* 16-bit twos complement data - little endian */
+#define CS4231_LINEAR_16_BIG	0xc0	/* 16-bit twos complement data - big endian */
+#define CS4231_ADPCM_16		0xa0	/* 16-bit ADPCM */
+#define CS4231_STEREO		0x10	/* stereo mode */
+/* bits 3-1 define frequency divisor */
+#define CS4231_XTAL1		0x00	/* 24.576 crystal */
+#define CS4231_XTAL2		0x01	/* 16.9344 crystal */
+
+/* definitions for interface control register - CS4231_IFACE_CTRL */
+
+#define CS4231_RECORD_PIO	0x80	/* record PIO enable */
+#define CS4231_PLAYBACK_PIO	0x40	/* playback PIO enable */
+#define CS4231_CALIB_MODE	0x18	/* calibration mode bits */
+#define CS4231_AUTOCALIB	0x08	/* auto calibrate */
+#define CS4231_SINGLE_DMA	0x04	/* use single DMA channel */
+#define CS4231_RECORD_ENABLE	0x02	/* record enable */
+#define CS4231_PLAYBACK_ENABLE	0x01	/* playback enable */
+
+/* definitions for pin control register - CS4231_PIN_CTRL */
+
+#define CS4231_IRQ_ENABLE	0x02	/* enable IRQ */
+#define CS4231_XCTL1		0x40	/* external control #1 */
+#define CS4231_XCTL0		0x80	/* external control #0 */
+
+/* definitions for test and init register - CS4231_TEST_INIT */
+
+#define CS4231_CALIB_IN_PROGRESS 0x20	/* auto calibrate in progress */
+#define CS4231_DMA_REQUEST	0x10	/* DMA request in progress */
+
+/* definitions for misc control register - CS4231_MISC_INFO */
+
+#define CS4231_MODE2		0x40	/* MODE 2 */
+#define CS4231_IW_MODE3		0x6c	/* MODE 3 - InterWave enhanced mode */
+#define CS4231_4236_MODE3	0xe0	/* MODE 3 - CS4236+ enhanced mode */
+
+/* definitions for alternate feature 1 register - CS4231_ALT_FEATURE_1 */
+
+#define	CS4231_DACZ		0x01	/* zero DAC when underrun */
+#define CS4231_TIMER_ENABLE	0x40	/* codec timer enable */
+#define CS4231_OLB		0x80	/* output level bit */
+
+/* definitions for Extended Registers - CS4236+ */
+
+#define CS4236_REG(i23val)	(((i23val << 2) & 0x10) | ((i23val >> 4) & 0x0f))
+#define CS4236_I23VAL(reg)	((((reg)&0xf) << 4) | (((reg)&0x10) >> 2) | 0x8)
+
+#define CS4236_LEFT_LINE	0x08	/* left LINE alternate volume */
+#define CS4236_RIGHT_LINE	0x18	/* right LINE alternate volume */
+#define CS4236_LEFT_MIC		0x28	/* left MIC volume */
+#define CS4236_RIGHT_MIC	0x38	/* right MIC volume */
+#define CS4236_LEFT_MIX_CTRL	0x48	/* synthesis and left input mixer control */
+#define CS4236_RIGHT_MIX_CTRL	0x58	/* right input mixer control */
+#define CS4236_LEFT_FM		0x68	/* left FM volume */
+#define CS4236_RIGHT_FM		0x78	/* right FM volume */
+#define CS4236_LEFT_DSP		0x88	/* left DSP serial port volume */
+#define CS4236_RIGHT_DSP	0x98	/* right DSP serial port volume */
+#define CS4236_RIGHT_LOOPBACK	0xa8	/* right loopback monitor volume */
+#define CS4236_DAC_MUTE		0xb8	/* DAC mute and IFSE enable */
+#define CS4236_ADC_RATE		0xc8	/* indenpendent ADC sample frequency */
+#define CS4236_DAC_RATE		0xd8	/* indenpendent DAC sample frequency */
+#define CS4236_LEFT_MASTER	0xe8	/* left master digital audio volume */
+#define CS4236_RIGHT_MASTER	0xf8	/* right master digital audio volume */
+#define CS4236_LEFT_WAVE	0x0c	/* left wavetable serial port volume */
+#define CS4236_RIGHT_WAVE	0x1c	/* right wavetable serial port volume */
+#define CS4236_VERSION		0x9c	/* chip version and ID */
+
+#endif /* __SOUND_CS4231_REGS_H */

include/sound/cs4231.h

@@ -2,7 +2,7 @@
 #define __SOUND_CS4231_H
 
 /*
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *  Definitions for CS4231 & InterWave chips & compatible chips
  *
  *
@@ -26,160 +26,7 @@
 #include "pcm.h"
 #include "timer.h"
 
-/* IO ports */
+#include "cs4231-regs.h"
-
-#define CS4231P(x)		(c_d_c_CS4231##x)
-
-#define c_d_c_CS4231REGSEL	0
-#define c_d_c_CS4231REG		1
-#define c_d_c_CS4231STATUS	2
-#define c_d_c_CS4231PIO		3
-
-/* codec registers */
-
-#define CS4231_LEFT_INPUT	0x00	/* left input control */
-#define CS4231_RIGHT_INPUT	0x01	/* right input control */
-#define CS4231_AUX1_LEFT_INPUT	0x02	/* left AUX1 input control */
-#define CS4231_AUX1_RIGHT_INPUT	0x03	/* right AUX1 input control */
-#define CS4231_AUX2_LEFT_INPUT	0x04	/* left AUX2 input control */
-#define CS4231_AUX2_RIGHT_INPUT	0x05	/* right AUX2 input control */
-#define CS4231_LEFT_OUTPUT	0x06	/* left output control register */
-#define CS4231_RIGHT_OUTPUT	0x07	/* right output control register */
-#define CS4231_PLAYBK_FORMAT	0x08	/* clock and data format - playback - bits 7-0 MCE */
-#define CS4231_IFACE_CTRL	0x09	/* interface control - bits 7-2 MCE */
-#define CS4231_PIN_CTRL		0x0a	/* pin control */
-#define CS4231_TEST_INIT	0x0b	/* test and initialization */
-#define CS4231_MISC_INFO	0x0c	/* miscellaneaous information */
-#define CS4231_LOOPBACK		0x0d	/* loopback control */
-#define CS4231_PLY_UPR_CNT	0x0e	/* playback upper base count */
-#define CS4231_PLY_LWR_CNT	0x0f	/* playback lower base count */
-#define CS4231_ALT_FEATURE_1	0x10	/* alternate #1 feature enable */
-#define AD1845_AF1_MIC_LEFT	0x10	/* alternate #1 feature + MIC left */
-#define CS4231_ALT_FEATURE_2	0x11	/* alternate #2 feature enable */
-#define AD1845_AF2_MIC_RIGHT	0x11	/* alternate #2 feature + MIC right */
-#define CS4231_LEFT_LINE_IN	0x12	/* left line input control */
-#define CS4231_RIGHT_LINE_IN	0x13	/* right line input control */
-#define CS4231_TIMER_LOW	0x14	/* timer low byte */
-#define CS4231_TIMER_HIGH	0x15	/* timer high byte */
-#define CS4231_LEFT_MIC_INPUT	0x16	/* left MIC input control register (InterWave only) */
-#define AD1845_UPR_FREQ_SEL	0x16	/* upper byte of frequency select */
-#define CS4231_RIGHT_MIC_INPUT	0x17	/* right MIC input control register (InterWave only) */
-#define AD1845_LWR_FREQ_SEL	0x17	/* lower byte of frequency select */
-#define CS4236_EXT_REG		0x17	/* extended register access */
-#define CS4231_IRQ_STATUS	0x18	/* irq status register */
-#define CS4231_LINE_LEFT_OUTPUT	0x19	/* left line output control register (InterWave only) */
-#define CS4231_VERSION		0x19	/* CS4231(A) - version values */
-#define CS4231_MONO_CTRL	0x1a	/* mono input/output control */
-#define CS4231_LINE_RIGHT_OUTPUT 0x1b	/* right line output control register (InterWave only) */
-#define AD1845_PWR_DOWN		0x1b	/* power down control */
-#define CS4235_LEFT_MASTER	0x1b	/* left master output control */
-#define CS4231_REC_FORMAT	0x1c	/* clock and data format - record - bits 7-0 MCE */
-#define CS4231_PLY_VAR_FREQ	0x1d	/* playback variable frequency */
-#define AD1845_CLOCK		0x1d	/* crystal clock select and total power down */
-#define CS4235_RIGHT_MASTER	0x1d	/* right master output control */
-#define CS4231_REC_UPR_CNT	0x1e	/* record upper count */
-#define CS4231_REC_LWR_CNT	0x1f	/* record lower count */
-
-/* definitions for codec register select port - CODECP( REGSEL ) */
-
-#define CS4231_INIT		0x80	/* CODEC is initializing */
-#define CS4231_MCE		0x40	/* mode change enable */
-#define CS4231_TRD		0x20	/* transfer request disable */
-
-/* definitions for codec status register - CODECP( STATUS ) */
-
-#define CS4231_GLOBALIRQ	0x01	/* IRQ is active */
-
-/* definitions for codec irq status */
-
-#define CS4231_PLAYBACK_IRQ	0x10
-#define CS4231_RECORD_IRQ	0x20
-#define CS4231_TIMER_IRQ	0x40
-#define CS4231_ALL_IRQS		0x70
-#define CS4231_REC_UNDERRUN	0x08
-#define CS4231_REC_OVERRUN	0x04
-#define CS4231_PLY_OVERRUN	0x02
-#define CS4231_PLY_UNDERRUN	0x01
-
-/* definitions for CS4231_LEFT_INPUT and CS4231_RIGHT_INPUT registers */
-
-#define CS4231_ENABLE_MIC_GAIN	0x20
-
-#define CS4231_MIXS_LINE	0x00
-#define CS4231_MIXS_AUX1	0x40
-#define CS4231_MIXS_MIC		0x80
-#define CS4231_MIXS_ALL		0xc0
-
-/* definitions for clock and data format register - CS4231_PLAYBK_FORMAT */
-
-#define CS4231_LINEAR_8		0x00	/* 8-bit unsigned data */
-#define CS4231_ALAW_8		0x60	/* 8-bit A-law companded */
-#define CS4231_ULAW_8		0x20	/* 8-bit U-law companded */
-#define CS4231_LINEAR_16	0x40	/* 16-bit twos complement data - little endian */
-#define CS4231_LINEAR_16_BIG	0xc0	/* 16-bit twos complement data - big endian */
-#define CS4231_ADPCM_16		0xa0	/* 16-bit ADPCM */
-#define CS4231_STEREO		0x10	/* stereo mode */
-/* bits 3-1 define frequency divisor */
-#define CS4231_XTAL1		0x00	/* 24.576 crystal */
-#define CS4231_XTAL2		0x01	/* 16.9344 crystal */
-
-/* definitions for interface control register - CS4231_IFACE_CTRL */
-
-#define CS4231_RECORD_PIO	0x80	/* record PIO enable */
-#define CS4231_PLAYBACK_PIO	0x40	/* playback PIO enable */
-#define CS4231_CALIB_MODE	0x18	/* calibration mode bits */
-#define CS4231_AUTOCALIB	0x08	/* auto calibrate */
-#define CS4231_SINGLE_DMA	0x04	/* use single DMA channel */
-#define CS4231_RECORD_ENABLE	0x02	/* record enable */
-#define CS4231_PLAYBACK_ENABLE	0x01	/* playback enable */
-
-/* definitions for pin control register - CS4231_PIN_CTRL */
-
-#define CS4231_IRQ_ENABLE	0x02	/* enable IRQ */
-#define CS4231_XCTL1		0x40	/* external control #1 */
-#define CS4231_XCTL0		0x80	/* external control #0 */
-
-/* definitions for test and init register - CS4231_TEST_INIT */
-
-#define CS4231_CALIB_IN_PROGRESS 0x20	/* auto calibrate in progress */
-#define CS4231_DMA_REQUEST	0x10	/* DMA request in progress */
-
-/* definitions for misc control register - CS4231_MISC_INFO */
-
-#define CS4231_MODE2		0x40	/* MODE 2 */
-#define CS4231_IW_MODE3		0x6c	/* MODE 3 - InterWave enhanced mode */
-#define CS4231_4236_MODE3	0xe0	/* MODE 3 - CS4236+ enhanced mode */
-
-/* definitions for alternate feature 1 register - CS4231_ALT_FEATURE_1 */
-
-#define	CS4231_DACZ		0x01	/* zero DAC when underrun */
-#define CS4231_TIMER_ENABLE	0x40	/* codec timer enable */
-#define CS4231_OLB		0x80	/* output level bit */
-
-/* definitions for Extended Registers - CS4236+ */
-
-#define CS4236_REG(i23val)	(((i23val << 2) & 0x10) | ((i23val >> 4) & 0x0f))
-#define CS4236_I23VAL(reg)	((((reg)&0xf) << 4) | (((reg)&0x10) >> 2) | 0x8)
-
-#define CS4236_LEFT_LINE	0x08	/* left LINE alternate volume */
-#define CS4236_RIGHT_LINE	0x18	/* right LINE alternate volume */
-#define CS4236_LEFT_MIC		0x28	/* left MIC volume */
-#define CS4236_RIGHT_MIC	0x38	/* right MIC volume */
-#define CS4236_LEFT_MIX_CTRL	0x48	/* synthesis and left input mixer control */
-#define CS4236_RIGHT_MIX_CTRL	0x58	/* right input mixer control */
-#define CS4236_LEFT_FM		0x68	/* left FM volume */
-#define CS4236_RIGHT_FM		0x78	/* right FM volume */
-#define CS4236_LEFT_DSP		0x88	/* left DSP serial port volume */
-#define CS4236_RIGHT_DSP	0x98	/* right DSP serial port volume */
-#define CS4236_RIGHT_LOOPBACK	0xa8	/* right loopback monitor volume */
-#define CS4236_DAC_MUTE		0xb8	/* DAC mute and IFSE enable */
-#define CS4236_ADC_RATE		0xc8	/* indenpendent ADC sample frequency */
-#define CS4236_DAC_RATE		0xd8	/* indenpendent DAC sample frequency */
-#define CS4236_LEFT_MASTER	0xe8	/* left master digital audio volume */
-#define CS4236_RIGHT_MASTER	0xf8	/* right master digital audio volume */
-#define CS4236_LEFT_WAVE	0x0c	/* left wavetable serial port volume */
-#define CS4236_RIGHT_WAVE	0x1c	/* right wavetable serial port volume */
-#define CS4236_VERSION		0x9c	/* chip version and ID */
 
 /* defines for codec.mode */
 
@@ -210,7 +57,7 @@
 #define CS4231_HW_CS4239	0x0404	/* CS4239 - Crystal Clear (tm) stereo enhancement */
 /* compatible, but clones */
 #define CS4231_HW_INTERWAVE     0x1000	/* InterWave chip */
-#define CS4231_HW_OPL3SA2       0x1001	/* OPL3-SA2 chip */
+#define CS4231_HW_OPL3SA2       0x1101	/* OPL3-SA2 chip, similar to cs4231 */
 
 /* defines for codec.hwshare */
 #define CS4231_HWSHARE_IRQ	(1<<0)

include/sound/cs46xx.h

@@ -2,7 +2,7 @@
 #define __SOUND_CS46XX_H
 
 /*
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>,
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
  *		     Cirrus Logic, Inc.
  *  Definitions for Cirrus Logic CS46xx chips
  *

include/sound/cs46xx_dsp_scb_types.h

@@ -1,6 +1,6 @@
 /*
  *  The driver for the Cirrus Logic's Sound Fusion CS46XX based soundcards
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify

include/sound/cs46xx_dsp_spos.h

@@ -1,6 +1,6 @@
 /*
  *  The driver for the Cirrus Logic's Sound Fusion CS46XX based soundcards
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify

include/sound/cs46xx_dsp_task_types.h

@@ -1,6 +1,6 @@
 /*
  *  The driver for the Cirrus Logic's Sound Fusion CS46XX based soundcards
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify

include/sound/cs8403.h

@@ -3,7 +3,7 @@
 
 /*
  *  Routines for Cirrus Logic CS8403/CS8404A IEC958 (S/PDIF) Transmitter
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>,
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
  *		     Takashi Iwai <tiwai@suse.de>
  *
  *

include/sound/cs8427.h

@@ -3,7 +3,7 @@
 
 /*
  *  Routines for Cirrus Logic CS8427
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>,
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
  *
  *
  *   This program is free software; you can redistribute it and/or modify

include/sound/driver.h

@@ -3,7 +3,7 @@
 
 /*
  *  Main header file for the ALSA driver
- *  Copyright (c) 1994-2000 by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) 1994-2000 by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify

include/sound/emu10k1.h

@@ -2,7 +2,7 @@
 #define __SOUND_EMU10K1_H
 
 /*
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>,
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
  *		     Creative Labs, Inc.
  *  Definitions for EMU10K1 (SB Live!) chips
  *
@@ -1408,8 +1408,6 @@ struct snd_emu10k1_fx8010 {
 	struct snd_emu10k1_fx8010_irq *irq_handlers;
 };
 
-#define emu10k1_gpr_ctl(n) list_entry(n, struct snd_emu10k1_fx8010_ctl, list)
-
 struct snd_emu10k1_midi {
 	struct snd_emu10k1 *emu;
 	struct snd_rawmidi *rmidi;
@@ -1456,6 +1454,9 @@ struct snd_emu1010 {
 	unsigned int adc_pads; /* bit mask */
 	unsigned int dac_pads; /* bit mask */
 	unsigned int internal_clock; /* 44100 or 48000 */
+	unsigned int optical_in; /* 0:SPDIF, 1:ADAT */
+	unsigned int optical_out; /* 0:SPDIF, 1:ADAT */
+	struct task_struct *firmware_thread;
 };
 
 struct snd_emu10k1 {
@@ -1599,9 +1600,9 @@ unsigned int snd_emu10k1_ptr20_read(struct snd_emu10k1 * emu, unsigned int reg,
 void snd_emu10k1_ptr20_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned int chn, unsigned int data);
 int snd_emu10k1_spi_write(struct snd_emu10k1 * emu, unsigned int data);
 int snd_emu10k1_i2c_write(struct snd_emu10k1 *emu, u32 reg, u32 value);
-int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, int reg, int value);
+int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, u32 reg, u32 value);
-int snd_emu1010_fpga_read(struct snd_emu10k1 * emu, int reg, int *value);
+int snd_emu1010_fpga_read(struct snd_emu10k1 * emu, u32 reg, u32 *value);
-int snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 * emu, int dst, int src);
+int snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 * emu, u32 dst, u32 src);
 unsigned int snd_emu10k1_efx_read(struct snd_emu10k1 *emu, unsigned int pc);
 void snd_emu10k1_intr_enable(struct snd_emu10k1 *emu, unsigned int intrenb);
 void snd_emu10k1_intr_disable(struct snd_emu10k1 *emu, unsigned int intrenb);
@@ -1746,6 +1747,8 @@ int snd_emu10k1_fx8010_unregister_irq_handler(struct snd_emu10k1 *emu,
 #define A_FXBUS2(x)	(0x80 + (x))	/* x = 0x00 - 0x1f extra outs used for EFX capture -> A_FXWC2 */
 #define A_EMU32OUTH(x)	(0xa0 + (x))	/* x = 0x00 - 0x0f "EMU32_OUT_10 - _1F" - ??? */
 #define A_EMU32OUTL(x)	(0xb0 + (x))	/* x = 0x00 - 0x0f "EMU32_OUT_1 - _F" - ??? */
+#define A3_EMU32IN(x)	(0x160 + (x))	/* x = 0x00 - 0x3f "EMU32_IN_00 - _3F" - Only when .device = 0x0008 */
+#define A3_EMU32OUT(x)	(0x1E0 + (x))	/* x = 0x00 - 0x0f "EMU32_OUT_00 - _3F" - Only when .device = 0x0008 */
 #define A_GPR(x)	(A_FXGPREGBASE + (x))
 
 /* cc_reg constants */

include/sound/es1688.h

@@ -3,7 +3,7 @@
 
 /*
  *  Header file for ES488/ES1688
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify

include/sound/gus.h

@@ -3,7 +3,7 @@
 
 /*
  *  Global structures used for GUS part of ALSA driver
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify

include/sound/hda_hwdep.h

@@ -0,0 +1,44 @@
+/*
+ * HWDEP Interface for HD-audio codec
+ *
+ * Copyright (c) 2007 Takashi Iwai <tiwai@suse.de>
+ *
+ *  This driver is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; either version 2 of the License, or
+ *  (at your option) any later version.
+ *
+ *  This driver is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *  GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License
+ *  along with this program; if not, write to the Free Software
+ *  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ */
+
+#ifndef __SOUND_HDA_HWDEP_H
+#define __SOUND_HDA_HWDEP_H
+
+#define HDA_HWDEP_VERSION	((1 << 16) | (0 << 8) | (0 << 0)) /* 1.0.0 */
+
+/* verb */
+#define HDA_REG_NID_SHIFT	24
+#define HDA_REG_VERB_SHIFT	8
+#define HDA_REG_VAL_SHIFT	0
+#define HDA_VERB(nid,verb,param)	((nid)<<24 | (verb)<<8 | (param))
+
+struct hda_verb_ioctl {
+	u32 verb;	/* HDA_VERB() */
+	u32 res;	/* response */
+};
+
+/*
+ * ioctls
+ */
+#define HDA_IOCTL_PVERSION		_IOR('H', 0x10, int)
+#define HDA_IOCTL_VERB_WRITE		_IOWR('H', 0x11, struct hda_verb_ioctl)
+#define HDA_IOCTL_GET_WCAP		_IOWR('H', 0x12, struct hda_verb_ioctl)
+
+#endif

include/sound/hdspm.h

@@ -1,4 +1,4 @@
-#ifndef __SOUND_HDSPM_H		/* -*- linux-c -*- */
+#ifndef __SOUND_HDSPM_H
 #define __SOUND_HDSPM_H
 /*
  *   Copyright (C) 2003 Winfried Ritsch (IEM)
@@ -61,7 +61,8 @@ struct hdspm_peak_rms_ioctl {
 };
 
 /* use indirect access due to the limit of ioctl bit size */
-#define SNDRV_HDSPM_IOCTL_GET_PEAK_RMS _IOR('H', 0x40, struct hdspm_peak_rms_ioctl)
+#define SNDRV_HDSPM_IOCTL_GET_PEAK_RMS \
+	_IOR('H', 0x40, struct hdspm_peak_rms_ioctl)
 
 /* ------------ CONFIG block IOCTL ---------------------- */
 
@@ -79,7 +80,8 @@ struct hdspm_config_info {
 	unsigned int analog_out;
 };
 
-#define SNDRV_HDSPM_IOCTL_GET_CONFIG_INFO _IOR('H', 0x41, struct hdspm_config_info)
+#define SNDRV_HDSPM_IOCTL_GET_CONFIG_INFO \
+	_IOR('H', 0x41, struct hdspm_config_info)
 
 
 /* get Soundcard Version */
@@ -93,10 +95,14 @@ struct hdspm_version {
 
 /* ------------- get Matrix Mixer IOCTL --------------- */
 
-/* MADI mixer: 64inputs+64playback in 64outputs = 8192 => *4Byte = 32768 Bytes */
+/* MADI mixer: 64inputs+64playback in 64outputs = 8192 => *4Byte =
+ * 32768 Bytes
+ */
 
 /* organisation is 64 channelfader in a continous memory block */
-/* equivalent to hardware definition, maybe for future feature of mmap of them */
+/* equivalent to hardware definition, maybe for future feature of mmap of
+ * them
+ */
 /* each of 64 outputs has 64 infader and 64 outfader: 
    Ins to Outs mixer[out].in[in], Outstreams to Outs mixer[out].pb[pb] */
 

include/sound/hwdep.h

@@ -3,7 +3,7 @@
 
 /*
  *  Hardware dependent layer 
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify

include/sound/info.h

@@ -3,7 +3,7 @@
 
 /*
  *  Header file for info interface
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify

include/sound/initval.h

@@ -3,7 +3,7 @@
 
 /*
  *  Init values for soundcard modules
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *   This program is free software; you can redistribute it and/or modify
  *   it under the terms of the GNU General Public License as published by

include/sound/memalloc.h

@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *                   Takashi Iwai <tiwai@suse.de>
  * 
  *  Generic memory allocators

include/sound/mixer_oss.h

@@ -3,7 +3,7 @@
 
 /*
  *  OSS MIXER API
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify

include/sound/mpu401.h

@@ -3,7 +3,7 @@
 
 /*
  *  Header file for MPU-401 and compatible cards
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify
@@ -50,7 +50,6 @@
 #define MPU401_INFO_INTEGRATED	(1 << 2)	/* integrated h/w port */
 #define MPU401_INFO_MMIO	(1 << 3)	/* MMIO access */
 #define MPU401_INFO_TX_IRQ	(1 << 4)	/* independent TX irq */
-#define MPU401_INFO_UART_ONLY	(1 << 5)	/* No ENTER_UART cmd needed */
 
 #define MPU401_MODE_BIT_INPUT		0
 #define MPU401_MODE_BIT_OUTPUT		1

include/sound/opl3.h

@@ -4,7 +4,7 @@
 /*
  * Definitions of the OPL-3 registers.
  *
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>,
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
  *                  Hannu Savolainen 1993-1996
  *
  *

include/sound/pcm-indirect.h

@@ -2,7 +2,7 @@
  * Helper functions for indirect PCM data transfer
  *
  *  Copyright (c) by Takashi Iwai <tiwai@suse.de>
- *                   Jaroslav Kysela <perex@suse.cz>
+ *                   Jaroslav Kysela <perex@perex.cz>
  *
  *   This program is free software; you can redistribute it and/or modify
  *   it under the terms of the GNU General Public License as published by

include/sound/pcm.h

@@ -3,7 +3,7 @@
 
 /*
  *  Digital Audio (PCM) abstract layer
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *                   Abramo Bagnara <abramo@alsa-project.org>
  *
  *
@@ -301,8 +301,8 @@ struct snd_pcm_runtime {
 	union snd_pcm_sync_id sync;	/* hardware synchronization ID */
 
 	/* -- mmap -- */
-	volatile struct snd_pcm_mmap_status *status;
+	struct snd_pcm_mmap_status *status;
-	volatile struct snd_pcm_mmap_control *control;
+	struct snd_pcm_mmap_control *control;
 
 	/* -- locking / scheduling -- */
 	wait_queue_head_t sleep;
@@ -791,13 +791,13 @@ static inline struct snd_interval *hw_param_interval(struct snd_pcm_hw_params *p
 static inline const struct snd_mask *hw_param_mask_c(const struct snd_pcm_hw_params *params,
 					     snd_pcm_hw_param_t var)
 {
-	return (const struct snd_mask *)hw_param_mask((struct snd_pcm_hw_params*) params, var);
+	return &params->masks[var - SNDRV_PCM_HW_PARAM_FIRST_MASK];
 }
 
 static inline const struct snd_interval *hw_param_interval_c(const struct snd_pcm_hw_params *params,
 						     snd_pcm_hw_param_t var)
 {
-	return (const struct snd_interval *)hw_param_interval((struct snd_pcm_hw_params*) params, var);
+	return &params->intervals[var - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL];
 }
 
 #define params_access(p) snd_mask_min(hw_param_mask((p), SNDRV_PCM_HW_PARAM_ACCESS))
@@ -922,7 +922,10 @@ snd_pcm_sframes_t snd_pcm_lib_writev(struct snd_pcm_substream *substream,
 snd_pcm_sframes_t snd_pcm_lib_readv(struct snd_pcm_substream *substream,
 				    void __user **bufs, snd_pcm_uframes_t frames);
 
+extern const struct snd_pcm_hw_constraint_list snd_pcm_known_rates;
+
 int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime);
+unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate);
 
 static inline void snd_pcm_set_runtime_buffer(struct snd_pcm_substream *substream,
 					      struct snd_dma_buffer *bufp)

include/sound/pcm_oss.h

@@ -3,7 +3,7 @@
 
 /*
  *  Digital Audio (PCM) - OSS compatibility abstract layer
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify

include/sound/rawmidi.h

@@ -3,7 +3,7 @@
 
 /*
  *  Abstract layer for MIDI v1.0 stream
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify

include/sound/sb.h

@@ -3,7 +3,7 @@
 
 /*
  *  Header file for SoundBlaster cards
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify

include/sound/seq_instr.h

@@ -3,7 +3,7 @@
 
 /*
  *  Main kernel header file for the ALSA sequencer
- *  Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify

include/sound/seq_midi_event.h

@@ -5,7 +5,7 @@
  *  MIDI byte <-> sequencer event coder
  *
  *  Copyright (C) 1998,99 Takashi Iwai <tiwai@suse.de>,
- *                        Jaroslav Kysela <perex@suse.cz>
+ *                        Jaroslav Kysela <perex@perex.cz>
  *
  *   This program is free software; you can redistribute it and/or modify
  *   it under the terms of the GNU General Public License as published by

include/sound/seq_virmidi.h

@@ -4,7 +4,7 @@
 /*
  *  Virtual Raw MIDI client on Sequencer
  *  Copyright (c) 2000 by Takashi Iwai <tiwai@suse.de>,
- *                        Jaroslav Kysela <perex@suse.cz>
+ *                        Jaroslav Kysela <perex@perex.cz>
  *
  *   This program is free software; you can redistribute it and/or modify
  *   it under the terms of the GNU General Public License as published by

include/sound/soc.h

@@ -201,8 +201,7 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_info *uinfo);
 int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_info *uinfo);
-int snd_soc_info_bool_ext(struct snd_kcontrol *kcontrol,
+#define snd_soc_info_bool_ext		snd_ctl_boolean_mono
-	struct snd_ctl_elem_info *uinfo);
 int snd_soc_get_volsw(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol);
 int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,

include/sound/tea575x-tuner.h

@@ -4,7 +4,7 @@
 /*
  *   ALSA driver for TEA5757/5759 Philips AM/FM tuner chips
  *
- *	Copyright (c) 2004 Jaroslav Kysela <perex@suse.cz>
+ *	Copyright (c) 2004 Jaroslav Kysela <perex@perex.cz>
  *
  *   This program is free software; you can redistribute it and/or modify
  *   it under the terms of the GNU General Public License as published by

include/sound/timer.h

@@ -3,7 +3,7 @@
 
 /*
  *  Timer abstract layer
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>,
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
  *		     Abramo Bagnara <abramo@alsa-project.org>
  *
  *

include/sound/tlv.h

@@ -3,7 +3,7 @@
 
 /*
  *  Advanced Linux Sound Architecture - ALSA - Driver
- *  Copyright (c) 2006 by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) 2006 by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify

include/sound/version.h

@@ -1,3 +1,3 @@
 /* include/version.h.  Generated by alsa/ksync script.  */
-#define CONFIG_SND_VERSION "1.0.14"
+#define CONFIG_SND_VERSION "1.0.15"
-#define CONFIG_SND_DATE " (Fri Jul 20 09:12:58 2007 UTC)"
+#define CONFIG_SND_DATE " (Tue Oct 16 14:57:44 2007 UTC)"

include/sound/ymfpci.h

@@ -2,7 +2,7 @@
 #define __SOUND_YMFPCI_H
 
 /*
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *  Definitions for Yahama YMF724/740/744/754 chips
  *
  *

sound/Kconfig

@@ -63,6 +63,10 @@ source "sound/aoa/Kconfig"
 
 source "sound/arm/Kconfig"
 
+if SPI
+source "sound/spi/Kconfig"
+endif
+
 source "sound/mips/Kconfig"
 
 source "sound/sh/Kconfig"

sound/Makefile

@@ -5,7 +5,8 @@ obj-$(CONFIG_SOUND) += soundcore.o
 obj-$(CONFIG_SOUND_PRIME) += sound_firmware.o
 obj-$(CONFIG_SOUND_PRIME) += oss/
 obj-$(CONFIG_DMASOUND) += oss/
-obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ soc/
+obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ \
+	sparc/ spi/ parisc/ pcmcia/ mips/ soc/
 obj-$(CONFIG_SND_AOA) += aoa/
 
 # This one must be compilable even if sound is configured out

sound/aoa/codecs/snd-aoa-codec-onyx.c

@@ -297,15 +297,7 @@ static struct snd_kcontrol_new capture_source_control = {
 	.put = onyx_snd_capture_source_put,
 };
 
-static int onyx_snd_mute_info(struct snd_kcontrol *kcontrol,
+#define onyx_snd_mute_info	snd_ctl_boolean_stereo_info
-	struct snd_ctl_elem_info *uinfo)
-{
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
-	uinfo->count = 2;
-	uinfo->value.integer.min = 0;
-	uinfo->value.integer.max = 1;
-	return 0;
-}
 
 static int onyx_snd_mute_get(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
@@ -359,15 +351,7 @@ static struct snd_kcontrol_new mute_control = {
 };
 
 
-static int onyx_snd_single_bit_info(struct snd_kcontrol *kcontrol,
+#define onyx_snd_single_bit_info	snd_ctl_boolean_mono_info
-	struct snd_ctl_elem_info *uinfo)
-{
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
-	uinfo->count = 1;
-	uinfo->value.integer.min = 0;
-	uinfo->value.integer.max = 1;
-	return 0;
-}
 
 #define FLAG_POLARITY_INVERT	1
 #define FLAG_SPDIFLOCK		2

sound/aoa/codecs/snd-aoa-codec-tas.c

@@ -272,15 +272,7 @@ static struct snd_kcontrol_new volume_control = {
 	.put = tas_snd_vol_put,
 };
 
-static int tas_snd_mute_info(struct snd_kcontrol *kcontrol,
+#define tas_snd_mute_info	snd_ctl_boolean_stereo_info
-	struct snd_ctl_elem_info *uinfo)
-{
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
-	uinfo->count = 2;
-	uinfo->value.integer.min = 0;
-	uinfo->value.integer.max = 1;
-	return 0;
-}
 
 static int tas_snd_mute_get(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
@@ -431,15 +423,7 @@ static struct snd_kcontrol_new drc_range_control = {
 	.put = tas_snd_drc_range_put,
 };
 
-static int tas_snd_drc_switch_info(struct snd_kcontrol *kcontrol,
+#define tas_snd_drc_switch_info		snd_ctl_boolean_mono_info
-	struct snd_ctl_elem_info *uinfo)
-{
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
-	uinfo->count = 1;
-	uinfo->value.integer.min = 0;
-	uinfo->value.integer.max = 1;
-	return 0;
-}
 
 static int tas_snd_drc_switch_get(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
@@ -743,6 +727,7 @@ static int tas_switch_clock(struct codec_info_item *cii, enum clock_switch clock
 	return 0;
 }
 
+#ifdef CONFIG_PM
 /* we are controlled via i2c and assume that is always up
  * If that wasn't the case, we'd have to suspend once
  * our i2c device is suspended, and then take note of that! */
@@ -768,7 +753,6 @@ static int tas_resume(struct tas *tas)
 	return 0;
 }
 
-#ifdef CONFIG_PM
 static int _tas_suspend(struct codec_info_item *cii, pm_message_t state)
 {
 	return tas_suspend(cii->codec_data);
@@ -778,7 +762,10 @@ static int _tas_resume(struct codec_info_item *cii)
 {
 	return tas_resume(cii->codec_data);
 }
-#endif
+#else /* CONFIG_PM */
+#define _tas_suspend	NULL
+#define _tas_resume	NULL
+#endif /* CONFIG_PM */
 
 static struct codec_info tas_codec_info = {
 	.transfers = tas_transfers,
@@ -791,10 +778,8 @@ static struct codec_info tas_codec_info = {
 	.owner = THIS_MODULE,
 	.usable = tas_usable,
 	.switch_clock = tas_switch_clock,
-#ifdef CONFIG_PM
 	.suspend = _tas_suspend,
 	.resume = _tas_resume,
-#endif
 };
 
 static int tas_init_codec(struct aoa_codec *codec)

sound/aoa/fabrics/snd-aoa-fabric-layout.c

@@ -582,15 +582,7 @@ static int layouts_list_items;
  * make the fabric handle all the card stuff, etc... */
 static struct layout_dev *layout_device;
 
-static int control_info(struct snd_kcontrol *kcontrol,
+#define control_info	snd_ctl_boolean_mono_info
-			struct snd_ctl_elem_info *uinfo)
-{
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
-	uinfo->count = 1;
-	uinfo->value.integer.min = 0;
-	uinfo->value.integer.max = 1;
-	return 0;
-}
 
 #define AMP_CONTROL(n, description)					\
 static int n##_control_get(struct snd_kcontrol *kcontrol,		\

sound/arm/sa11xx-uda1341.c

@@ -79,12 +79,6 @@
 #include <asm/mach-types.h>
 #include <asm/dma.h>
 
-#ifdef CONFIG_H3600_HAL
-#include <asm/semaphore.h>
-#include <asm/uaccess.h>
-#include <asm/arch/h3600_hal.h>
-#endif
-
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/initval.h>
@@ -100,9 +94,6 @@
  * We use DMA stuff from 2.4.18-rmk3-hh24 here to be able to compile this
  * module for Familiar 0.6.1
  */
-#ifdef CONFIG_H3600_HAL
-#define HH_VERSION 1
-#endif
 
 /* {{{ Type definitions */
 
@@ -238,11 +229,8 @@ static void sa11xx_uda1341_set_samplerate(struct sa11xx_uda1341 *sa11xx_uda1341,
 		rate = 8000;
 
 	/* Set the external clock generator */
-#ifdef CONFIG_H3600_HAL
+	
-	h3600_audio_clock(rate);
-#else	
 	sa11xx_uda1341_set_audio_clock(rate);
-#endif
 
 	/* Select the clock divisor */
 	switch (rate) {
@@ -307,13 +295,10 @@ static void sa11xx_uda1341_audio_init(struct sa11xx_uda1341 *sa11xx_uda1341)
 	local_irq_restore(flags);
 
 	/* Enable the audio power */
-#ifdef CONFIG_H3600_HAL
+
-	h3600_audio_power(AUDIO_RATE_DEFAULT);
-#else
 	clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
 	set_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
 	set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
-#endif
  
 	/* Wait for the UDA1341 to wake up */
 	mdelay(1); //FIXME - was removed by Perex - Why?
@@ -331,21 +316,13 @@ static void sa11xx_uda1341_audio_init(struct sa11xx_uda1341 *sa11xx_uda1341)
 	/* make the left and right channels unswapped (flip the WS latch) */
 	Ser4SSDR = 0;
 
-#ifdef CONFIG_H3600_HAL
+	clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
-	h3600_audio_mute(0);
-#else	
-	clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);        
-#endif     
 }
 
 static void sa11xx_uda1341_audio_shutdown(struct sa11xx_uda1341 *sa11xx_uda1341)
 {
 	/* mute on */
-#ifdef CONFIG_H3600_HAL
-	h3600_audio_mute(1);
-#else	
 	set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
-#endif
 	
 	/* disable the audio power and all signals leading to the audio chip */
 	l3_close(sa11xx_uda1341->uda1341);
@@ -354,13 +331,9 @@ static void sa11xx_uda1341_audio_shutdown(struct sa11xx_uda1341 *sa11xx_uda1341)
 
 	/* power off and mute off */
 	/* FIXME - is muting off necesary??? */
-#ifdef CONFIG_H3600_HAL
+
-	h3600_audio_power(0);
-	h3600_audio_mute(0);
-#else	
 	clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
 	clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
-#endif	
 }
 
 /* }}} */

sound/core/Makefile

@@ -1,20 +1,17 @@
 #
 # Makefile for ALSA
-# Copyright (c) 1999,2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 1999,2001 by Jaroslav Kysela <perex@perex.cz>
 #
 
-snd-objs     := sound.o init.o memory.o info.o control.o misc.o device.o
+snd-y     := sound.o init.o memory.o info.o control.o misc.o device.o
-ifeq ($(CONFIG_ISA_DMA_API),y)
+snd-$(CONFIG_ISA_DMA_API) += isadma.o
-snd-objs     += isadma.o
+snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o info_oss.o
-endif
-ifeq ($(CONFIG_SND_OSSEMUL),y)
-snd-objs     += sound_oss.o info_oss.o
-endif
 
 snd-pcm-objs := pcm.o pcm_native.o pcm_lib.o pcm_timer.o pcm_misc.o \
 		pcm_memory.o
 
-snd-page-alloc-objs := memalloc.o sgbuf.o
+snd-page-alloc-y := memalloc.o
+snd-page-alloc-$(CONFIG_HAS_DMA) += sgbuf.o
 
 snd-rawmidi-objs  := rawmidi.o
 snd-timer-objs    := timer.o

sound/core/control.c

@@ -1,6 +1,6 @@
 /*
  *  Routines for driver control interface
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify
@@ -716,8 +716,6 @@ int snd_ctl_elem_read(struct snd_card *card, struct snd_ctl_elem_value *control)
 	return result;
 }
 
-EXPORT_SYMBOL(snd_ctl_elem_read);
-
 static int snd_ctl_elem_read_user(struct snd_card *card,
 				  struct snd_ctl_elem_value __user *_control)
 {
@@ -781,8 +779,6 @@ int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file,
 	return result;
 }
 
-EXPORT_SYMBOL(snd_ctl_elem_write);
-
 static int snd_ctl_elem_write_user(struct snd_ctl_file *file,
 				   struct snd_ctl_elem_value __user *_control)
 {
@@ -1486,3 +1482,30 @@ int snd_ctl_create(struct snd_card *card)
 	snd_assert(card != NULL, return -ENXIO);
 	return snd_device_new(card, SNDRV_DEV_CONTROL, card, &ops);
 }
+
+/*
+ * Frequently used control callbacks
+ */
+int snd_ctl_boolean_mono_info(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+	uinfo->count = 1;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = 1;
+	return 0;
+}
+
+EXPORT_SYMBOL(snd_ctl_boolean_mono_info);
+
+int snd_ctl_boolean_stereo_info(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+	uinfo->count = 2;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = 1;
+	return 0;
+}
+
+EXPORT_SYMBOL(snd_ctl_boolean_stereo_info);

sound/core/device.c

@@ -1,6 +1,6 @@
 /*
  *  Device management routines
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify

sound/core/hwdep.c

@@ -1,6 +1,6 @@
 /*
  *  Hardware dependent layer
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify
@@ -31,7 +31,7 @@
 #include <sound/hwdep.h>
 #include <sound/info.h>
 
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("Hardware dependent layer");
 MODULE_LICENSE("GPL");
 

sound/core/info.c

@@ -1,6 +1,6 @@
 /*
  *  Information interface for ALSA driver
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify

sound/core/info_oss.c

@@ -1,6 +1,6 @@
 /*
  *  Information interface for ALSA driver
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify

sound/core/init.c

@@ -1,6 +1,6 @@
 /*
  *  Initialization routines
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify

sound/core/isadma.c

@@ -1,6 +1,6 @@
 /*
  *  ISA DMA support functions
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify

sound/core/memalloc.c

@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *                   Takashi Iwai <tiwai@suse.de>
  * 
  *  Generic memory allocators
@@ -38,7 +38,7 @@
 #endif
 
 
-MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>, Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>, Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("Memory allocator for ALSA system.");
 MODULE_LICENSE("GPL");
 
@@ -206,6 +206,7 @@ void snd_free_pages(void *ptr, size_t size)
  *
  */
 
+#ifdef CONFIG_HAS_DMA
 /* allocate the coherent DMA pages */
 static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *dma)
 {
@@ -239,6 +240,7 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr,
 	dec_snd_pages(pg);
 	dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma);
 }
+#endif /* CONFIG_HAS_DMA */
 
 #ifdef CONFIG_SBUS
 
@@ -312,12 +314,14 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size,
 		dmab->area = snd_malloc_sbus_pages(device, size, &dmab->addr);
 		break;
 #endif
+#ifdef CONFIG_HAS_DMA
 	case SNDRV_DMA_TYPE_DEV:
 		dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr);
 		break;
 	case SNDRV_DMA_TYPE_DEV_SG:
 		snd_malloc_sgbuf_pages(device, size, dmab, NULL);
 		break;
+#endif
 	default:
 		printk(KERN_ERR "snd-malloc: invalid device type %d\n", type);
 		dmab->area = NULL;
@@ -383,12 +387,14 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab)
 		snd_free_sbus_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr);
 		break;
 #endif
+#ifdef CONFIG_HAS_DMA
 	case SNDRV_DMA_TYPE_DEV:
 		snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr);
 		break;
 	case SNDRV_DMA_TYPE_DEV_SG:
 		snd_free_sgbuf_pages(dmab);
 		break;
+#endif
 	default:
 		printk(KERN_ERR "snd-malloc: invalid device type %d\n", dmab->dev.type);
 	}

sound/core/memory.c

@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  * 
  *  Misc memory accessors
  *

sound/core/misc.c

@@ -1,6 +1,6 @@
 /*
  *  Misc and compatibility things
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify

sound/core/oss/Makefile

@@ -1,12 +1,13 @@
 #
 # Makefile for ALSA
-# Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
 #
 
 snd-mixer-oss-objs := mixer_oss.o
 
-snd-pcm-oss-objs := pcm_oss.o pcm_plugin.o \
+snd-pcm-oss-y := pcm_oss.o
-		    io.o copy.o linear.o mulaw.o route.o rate.o
+snd-pcm-oss-$(CONFIG_SND_PCM_OSS_PLUGINS) += pcm_plugin.o \
+	io.o copy.o linear.o mulaw.o route.o rate.o
 
 obj-$(CONFIG_SND_MIXER_OSS) += snd-mixer-oss.o
 obj-$(CONFIG_SND_PCM_OSS) += snd-pcm-oss.o

sound/core/oss/copy.c

@@ -20,9 +20,6 @@
  */
 
 #include <sound/driver.h>
-
-#ifdef CONFIG_SND_PCM_OSS_PLUGINS
-
 #include <linux/time.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -88,5 +85,3 @@ int snd_pcm_plugin_build_copy(struct snd_pcm_substream *plug,
 	*r_plugin = plugin;
 	return 0;
 }
-
-#endif

sound/core/oss/io.c

@@ -1,6 +1,6 @@
 /*
  *  PCM I/O Plug-In Interface
- *  Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This library is free software; you can redistribute it and/or modify
@@ -20,9 +20,6 @@
  */
   
 #include <sound/driver.h>
-
-#ifdef CONFIG_SND_PCM_OSS_PLUGINS
-
 #include <linux/time.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -135,5 +132,3 @@ int snd_pcm_plugin_build_io(struct snd_pcm_substream *plug,
 	*r_plugin = plugin;
 	return 0;
 }
-
-#endif

sound/core/oss/linear.c

@@ -1,6 +1,6 @@
 /*
  *  Linear conversion Plug-In
- *  Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz>,
+ *  Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>,
  *			  Abramo Bagnara <abramo@alsa-project.org>
  *
  *
@@ -21,9 +21,6 @@
  */
 
 #include <sound/driver.h>
-
-#ifdef CONFIG_SND_PCM_OSS_PLUGINS
-
 #include <linux/time.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -34,19 +31,34 @@
  */
  
 struct linear_priv {
-	int conv;
+	int cvt_endian;		/* need endian conversion? */
+	unsigned int src_ofs;	/* byte offset in source format */
+	unsigned int dst_ofs;	/* byte soffset in destination format */
+	unsigned int copy_ofs;	/* byte offset in temporary u32 data */
+	unsigned int dst_bytes;		/* byte size of destination format */
+	unsigned int copy_bytes;	/* bytes to copy per conversion */
+	unsigned int flip; /* MSB flip for signeness, done after endian conv */
 };
 
+static inline void do_convert(struct linear_priv *data,
+			      unsigned char *dst, unsigned char *src)
+{
+	unsigned int tmp = 0;
+	unsigned char *p = (unsigned char *)&tmp;
+
+	memcpy(p + data->copy_ofs, src + data->src_ofs, data->copy_bytes);
+	if (data->cvt_endian)
+		tmp = swab32(tmp);
+	tmp ^= data->flip;
+	memcpy(dst, p + data->dst_ofs, data->dst_bytes);
+}
+
 static void convert(struct snd_pcm_plugin *plugin,
 		    const struct snd_pcm_plugin_channel *src_channels,
 		    struct snd_pcm_plugin_channel *dst_channels,
 		    snd_pcm_uframes_t frames)
 {
-#define CONV_LABELS
-#include "plugin_ops.h"
-#undef CONV_LABELS
 	struct linear_priv *data = (struct linear_priv *)plugin->extra_data;
-	void *conv = conv_labels[data->conv];
 	int channel;
 	int nchannels = plugin->src_format.channels;
 	for (channel = 0; channel < nchannels; ++channel) {
@@ -67,11 +79,7 @@ static void convert(struct snd_pcm_plugin *plugin,
 		dst_step = dst_channels[channel].area.step / 8;
 		frames1 = frames;
 		while (frames1-- > 0) {
-			goto *conv;
+			do_convert(data, dst, src);
-#define CONV_END after
-#include "plugin_ops.h"
-#undef CONV_END
-		after:
 			src += src_step;
 			dst += dst_step;
 		}
@@ -106,29 +114,36 @@ static snd_pcm_sframes_t linear_transfer(struct snd_pcm_plugin *plugin,
 	return frames;
 }
 
-static int conv_index(int src_format, int dst_format)
+static void init_data(struct linear_priv *data, int src_format, int dst_format)
 {
-	int src_endian, dst_endian, sign, src_width, dst_width;
+	int src_le, dst_le, src_bytes, dst_bytes;
 
-	sign = (snd_pcm_format_signed(src_format) !=
+	src_bytes = snd_pcm_format_width(src_format) / 8;
-		snd_pcm_format_signed(dst_format));
+	dst_bytes = snd_pcm_format_width(dst_format) / 8;
-#ifdef SNDRV_LITTLE_ENDIAN
+	src_le = snd_pcm_format_little_endian(src_format) > 0;
-	src_endian = snd_pcm_format_big_endian(src_format);
+	dst_le = snd_pcm_format_little_endian(dst_format) > 0;
-	dst_endian = snd_pcm_format_big_endian(dst_format);
-#else
-	src_endian = snd_pcm_format_little_endian(src_format);
-	dst_endian = snd_pcm_format_little_endian(dst_format);
-#endif
 
-	if (src_endian < 0)
+	data->dst_bytes = dst_bytes;
-		src_endian = 0;
+	data->cvt_endian = src_le != dst_le;
-	if (dst_endian < 0)
+	data->copy_bytes = src_bytes < dst_bytes ? src_bytes : dst_bytes;
-		dst_endian = 0;
+	if (src_le) {
-
+		data->copy_ofs = 4 - data->copy_bytes;
-	src_width = snd_pcm_format_width(src_format) / 8 - 1;
+		data->src_ofs = src_bytes - data->copy_bytes;
-	dst_width = snd_pcm_format_width(dst_format) / 8 - 1;
+	} else
-
+		data->src_ofs = snd_pcm_format_physical_width(src_format) / 8 -
-	return src_width * 32 + src_endian * 16 + sign * 8 + dst_width * 2 + dst_endian;
+			src_bytes;
+	if (dst_le)
+		data->dst_ofs = 4 - data->dst_bytes;
+	else
+		data->dst_ofs = snd_pcm_format_physical_width(dst_format) / 8 -
+			dst_bytes;
+	if (snd_pcm_format_signed(src_format) !=
+	    snd_pcm_format_signed(dst_format)) {
+		if (dst_le)
+			data->flip = cpu_to_le32(0x80000000);
+		else
+			data->flip = cpu_to_be32(0x80000000);
+	}
 }
 
 int snd_pcm_plugin_build_linear(struct snd_pcm_substream *plug,
@@ -154,10 +169,8 @@ int snd_pcm_plugin_build_linear(struct snd_pcm_substream *plug,
 	if (err < 0)
 		return err;
 	data = (struct linear_priv *)plugin->extra_data;
-	data->conv = conv_index(src_format->format, dst_format->format);
+	init_data(data, src_format->format, dst_format->format);
 	plugin->transfer = linear_transfer;
 	*r_plugin = plugin;
 	return 0;
 }
-
-#endif

sound/core/oss/mixer_oss.c

@@ -1,6 +1,6 @@
 /*
  *  OSS emulation layer for the mixer interface
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify
@@ -33,7 +33,7 @@
 
 #define OSS_ALSAEMULVER         _SIOR ('M', 249, int)
 
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("Mixer OSS emulation for ALSA.");
 MODULE_LICENSE("GPL");
 MODULE_ALIAS_SNDRV_MINOR(SNDRV_MINOR_OSS_MIXER);

sound/core/oss/mulaw.c

@@ -1,6 +1,6 @@
 /*
  *  Mu-Law conversion Plug-In Interface
- *  Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
  *                        Uros Bizjak <uros@kss-loka.si>
  *
  *  Based on reference implementation by Sun Microsystems, Inc.
@@ -22,9 +22,6 @@
  */
   
 #include <sound/driver.h>
-
-#ifdef CONFIG_SND_PCM_OSS_PLUGINS
-
 #include <linux/time.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -149,19 +146,32 @@ typedef void (*mulaw_f)(struct snd_pcm_plugin *plugin,
 
 struct mulaw_priv {
 	mulaw_f func;
-	int conv;
+	int cvt_endian;			/* need endian conversion? */
+	unsigned int native_ofs;	/* byte offset in native format */
+	unsigned int copy_ofs;		/* byte offset in s16 format */
+	unsigned int native_bytes;	/* byte size of the native format */
+	unsigned int copy_bytes;	/* bytes to copy per conversion */
+	u16 flip; /* MSB flip for signedness, done after endian conversion */
 };
 
+static inline void cvt_s16_to_native(struct mulaw_priv *data,
+				     unsigned char *dst, u16 sample)
+{
+	sample ^= data->flip;
+	if (data->cvt_endian)
+		sample = swab16(sample);
+	if (data->native_bytes > data->copy_bytes)
+		memset(dst, 0, data->native_bytes);
+	memcpy(dst + data->native_ofs, (char *)&sample + data->copy_ofs,
+	       data->copy_bytes);
+}
+
 static void mulaw_decode(struct snd_pcm_plugin *plugin,
 			const struct snd_pcm_plugin_channel *src_channels,
 			struct snd_pcm_plugin_channel *dst_channels,
 			snd_pcm_uframes_t frames)
 {
-#define PUT_S16_LABELS
-#include "plugin_ops.h"
-#undef PUT_S16_LABELS
 	struct mulaw_priv *data = (struct mulaw_priv *)plugin->extra_data;
-	void *put = put_s16_labels[data->conv];
 	int channel;
 	int nchannels = plugin->src_format.channels;
 	for (channel = 0; channel < nchannels; ++channel) {
@@ -183,30 +193,33 @@ static void mulaw_decode(struct snd_pcm_plugin *plugin,
 		frames1 = frames;
 		while (frames1-- > 0) {
 			signed short sample = ulaw2linear(*src);
-			goto *put;
+			cvt_s16_to_native(data, dst, sample);
-#define PUT_S16_END after
-#include "plugin_ops.h"
-#undef PUT_S16_END
-		after:
 			src += src_step;
 			dst += dst_step;
 		}
 	}
 }
 
+static inline signed short cvt_native_to_s16(struct mulaw_priv *data,
+					     unsigned char *src)
+{
+	u16 sample = 0;
+	memcpy((char *)&sample + data->copy_ofs, src + data->native_ofs,
+	       data->copy_bytes);
+	if (data->cvt_endian)
+		sample = swab16(sample);
+	sample ^= data->flip;
+	return (signed short)sample;
+}
+
 static void mulaw_encode(struct snd_pcm_plugin *plugin,
 			const struct snd_pcm_plugin_channel *src_channels,
 			struct snd_pcm_plugin_channel *dst_channels,
 			snd_pcm_uframes_t frames)
 {
-#define GET_S16_LABELS
-#include "plugin_ops.h"
-#undef GET_S16_LABELS
 	struct mulaw_priv *data = (struct mulaw_priv *)plugin->extra_data;
-	void *get = get_s16_labels[data->conv];
 	int channel;
 	int nchannels = plugin->src_format.channels;
-	signed short sample = 0;
 	for (channel = 0; channel < nchannels; ++channel) {
 		char *src;
 		char *dst;
@@ -225,11 +238,7 @@ static void mulaw_encode(struct snd_pcm_plugin *plugin,
 		dst_step = dst_channels[channel].area.step / 8;
 		frames1 = frames;
 		while (frames1-- > 0) {
-			goto *get;
+			signed short sample = cvt_native_to_s16(data, src);
-#define GET_S16_END after
-#include "plugin_ops.h"
-#undef GET_S16_END
-		after:
 			*dst = linear2ulaw(sample);
 			src += src_step;
 			dst += dst_step;
@@ -265,23 +274,25 @@ static snd_pcm_sframes_t mulaw_transfer(struct snd_pcm_plugin *plugin,
 	return frames;
 }
 
-static int getput_index(int format)
+static void init_data(struct mulaw_priv *data, int format)
 {
-	int sign, width, endian;
-	sign = !snd_pcm_format_signed(format);
-	width = snd_pcm_format_width(format) / 8 - 1;
-	if (width < 0 || width > 3) {
-		snd_printk(KERN_ERR "snd-pcm-oss: invalid format %d\n", format);
-		width = 0;
-	}
 #ifdef SNDRV_LITTLE_ENDIAN
-	endian = snd_pcm_format_big_endian(format);
+	data->cvt_endian = snd_pcm_format_big_endian(format) > 0;
 #else
-	endian = snd_pcm_format_little_endian(format);
+	data->cvt_endian = snd_pcm_format_little_endian(format) > 0;
 #endif
-	if (endian < 0)
+	if (!snd_pcm_format_signed(format))
-		endian = 0;
+		data->flip = 0x8000;
-	return width * 4 + endian * 2 + sign;
+	data->native_bytes = snd_pcm_format_physical_width(format) / 8;
+	data->copy_bytes = data->native_bytes < 2 ? 1 : 2;
+	if (snd_pcm_format_little_endian(format)) {
+		data->native_ofs = data->native_bytes - data->copy_bytes;
+		data->copy_ofs = 2 - data->copy_bytes;
+	} else {
+		/* S24 in 4bytes need an 1 byte offset */
+		data->native_ofs = data->native_bytes -
+			snd_pcm_format_width(format) / 8;
+	}
 }
 
 int snd_pcm_plugin_build_mulaw(struct snd_pcm_substream *plug,
@@ -322,11 +333,8 @@ int snd_pcm_plugin_build_mulaw(struct snd_pcm_substream *plug,
 		return err;
 	data = (struct mulaw_priv *)plugin->extra_data;
 	data->func = func;
-	data->conv = getput_index(format->format);
+	init_data(data, format->format);
-	snd_assert(data->conv >= 0 && data->conv < 4*2*2, return -EINVAL);
 	plugin->transfer = mulaw_transfer;
 	*r_plugin = plugin;
 	return 0;
 }
-
-#endif

sound/core/oss/pcm_oss.c

@@ -1,6 +1,6 @@
 /*
  *  Digital Audio (PCM) abstract layer / OSS compatible
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify
@@ -48,7 +48,7 @@ static int dsp_map[SNDRV_CARDS];
 static int adsp_map[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = 1};
 static int nonblock_open = 1;
 
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>, Abramo Bagnara <abramo@alsa-project.org>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, Abramo Bagnara <abramo@alsa-project.org>");
 MODULE_DESCRIPTION("PCM OSS emulation for ALSA.");
 MODULE_LICENSE("GPL");
 module_param_array(dsp_map, int, NULL, 0444);
@@ -633,6 +633,22 @@ static long snd_pcm_alsa_frames(struct snd_pcm_substream *substream, long bytes)
 	return bytes_to_frames(runtime, (buffer_size * bytes) / runtime->oss.buffer_bytes);
 }
 
+/* define extended formats in the recent OSS versions (if any) */
+/* linear formats */
+#define AFMT_S32_LE      0x00001000
+#define AFMT_S32_BE      0x00002000
+#define AFMT_S24_LE      0x00008000
+#define AFMT_S24_BE      0x00010000
+#define AFMT_S24_PACKED  0x00040000
+
+/* other supported formats */
+#define AFMT_FLOAT       0x00004000
+#define AFMT_SPDIF_RAW   0x00020000
+
+/* unsupported formats */
+#define AFMT_AC3         0x00000400
+#define AFMT_VORBIS      0x00000800
+
 static int snd_pcm_oss_format_from(int format)
 {
 	switch (format) {
@@ -646,6 +662,13 @@ static int snd_pcm_oss_format_from(int format)
 	case AFMT_U16_LE:	return SNDRV_PCM_FORMAT_U16_LE;
 	case AFMT_U16_BE:	return SNDRV_PCM_FORMAT_U16_BE;
 	case AFMT_MPEG:		return SNDRV_PCM_FORMAT_MPEG;
+	case AFMT_S32_LE:	return SNDRV_PCM_FORMAT_S32_LE;
+	case AFMT_S32_BE:	return SNDRV_PCM_FORMAT_S32_BE;
+	case AFMT_S24_LE:	return SNDRV_PCM_FORMAT_S24_LE;
+	case AFMT_S24_BE:	return SNDRV_PCM_FORMAT_S24_BE;
+	case AFMT_S24_PACKED:	return SNDRV_PCM_FORMAT_S24_3LE;
+	case AFMT_FLOAT:	return SNDRV_PCM_FORMAT_FLOAT;
+	case AFMT_SPDIF_RAW:	return SNDRV_PCM_FORMAT_IEC958_SUBFRAME;
 	default:		return SNDRV_PCM_FORMAT_U8;
 	}
 }
@@ -663,6 +686,13 @@ static int snd_pcm_oss_format_to(int format)
 	case SNDRV_PCM_FORMAT_U16_LE:	return AFMT_U16_LE;
 	case SNDRV_PCM_FORMAT_U16_BE:	return AFMT_U16_BE;
 	case SNDRV_PCM_FORMAT_MPEG:		return AFMT_MPEG;
+	case SNDRV_PCM_FORMAT_S32_LE:	return AFMT_S32_LE;
+	case SNDRV_PCM_FORMAT_S32_BE:	return AFMT_S32_BE;
+	case SNDRV_PCM_FORMAT_S24_LE:	return AFMT_S24_LE;
+	case SNDRV_PCM_FORMAT_S24_BE:	return AFMT_S24_BE;
+	case SNDRV_PCM_FORMAT_S24_3LE:	return AFMT_S24_PACKED;
+	case SNDRV_PCM_FORMAT_FLOAT:	return AFMT_FLOAT;
+	case SNDRV_PCM_FORMAT_IEC958_SUBFRAME: return AFMT_SPDIF_RAW;
 	default:			return -EINVAL;
 	}
 }
@@ -1725,7 +1755,10 @@ static int snd_pcm_oss_get_formats(struct snd_pcm_oss_file *pcm_oss_file)
 		return AFMT_MU_LAW | AFMT_U8 |
 		       AFMT_S16_LE | AFMT_S16_BE |
 		       AFMT_S8 | AFMT_U16_LE |
-		       AFMT_U16_BE;
+		       AFMT_U16_BE |
+			AFMT_S32_LE | AFMT_S32_BE |
+			AFMT_S24_LE | AFMT_S24_LE |
+			AFMT_S24_PACKED;
 	params = kmalloc(sizeof(*params), GFP_KERNEL);
 	if (!params)
 		return -ENOMEM;

sound/core/oss/pcm_plugin.c

@@ -1,6 +1,6 @@
 /*
  *  PCM Plug-In shared (kernel/library) code
- *  Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
  *  Copyright (c) 2000 by Abramo Bagnara <abramo@alsa-project.org>
  *
  *
@@ -25,9 +25,6 @@
 #endif
 
 #include <sound/driver.h>
-
-#ifdef CONFIG_SND_PCM_OSS_PLUGINS
-
 #include <linux/slab.h>
 #include <linux/time.h>
 #include <linux/vmalloc.h>
@@ -267,6 +264,8 @@ static int snd_pcm_plug_formats(struct snd_mask *mask, int format)
 		       SNDRV_PCM_FMTBIT_U16_BE | SNDRV_PCM_FMTBIT_S16_BE |
 		       SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_S24_LE |
 		       SNDRV_PCM_FMTBIT_U24_BE | SNDRV_PCM_FMTBIT_S24_BE |
+		       SNDRV_PCM_FMTBIT_U24_3LE | SNDRV_PCM_FMTBIT_S24_3LE |
+		       SNDRV_PCM_FMTBIT_U24_3BE | SNDRV_PCM_FMTBIT_S24_3BE |
 		       SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_S32_LE |
 		       SNDRV_PCM_FMTBIT_U32_BE | SNDRV_PCM_FMTBIT_S32_BE);
 	snd_mask_set(&formats, SNDRV_PCM_FORMAT_MU_LAW);
@@ -283,6 +282,10 @@ static int preferred_formats[] = {
 	SNDRV_PCM_FORMAT_S16_BE,
 	SNDRV_PCM_FORMAT_U16_LE,
 	SNDRV_PCM_FORMAT_U16_BE,
+	SNDRV_PCM_FORMAT_S24_3LE,
+	SNDRV_PCM_FORMAT_S24_3BE,
+	SNDRV_PCM_FORMAT_U24_3LE,
+	SNDRV_PCM_FORMAT_U24_3BE,
 	SNDRV_PCM_FORMAT_S24_LE,
 	SNDRV_PCM_FORMAT_S24_BE,
 	SNDRV_PCM_FORMAT_U24_LE,
@@ -297,41 +300,37 @@ static int preferred_formats[] = {
 
 int snd_pcm_plug_slave_format(int format, struct snd_mask *format_mask)
 {
+	int i;
+
 	if (snd_mask_test(format_mask, format))
 		return format;
 	if (! snd_pcm_plug_formats(format_mask, format))
 		return -EINVAL;
 	if (snd_pcm_format_linear(format)) {
-		int width = snd_pcm_format_width(format);
+		unsigned int width = snd_pcm_format_width(format);
-		int unsignd = snd_pcm_format_unsigned(format);
+		int unsignd = snd_pcm_format_unsigned(format) > 0;
-		int big = snd_pcm_format_big_endian(format);
+		int big = snd_pcm_format_big_endian(format) > 0;
-		int format1;
+		unsigned int badness, best = -1;
-		int wid, width1=width;
+		int best_format = -1;
-		int dwidth1 = 8;
+		for (i = 0; i < ARRAY_SIZE(preferred_formats); i++) {
-		for (wid = 0; wid < 4; ++wid) {
+			int f = preferred_formats[i];
-			int end, big1 = big;
+			unsigned int w;
-			for (end = 0; end < 2; ++end) {
+			if (!snd_mask_test(format_mask, f))
-				int sgn, unsignd1 = unsignd;
+				continue;
-				for (sgn = 0; sgn < 2; ++sgn) {
+			w = snd_pcm_format_width(f);
-					format1 = snd_pcm_build_linear_format(width1, unsignd1, big1);
+			if (w >= width)
-					if (format1 >= 0 &&
+				badness = w - width;
-					    snd_mask_test(format_mask, format1))
+			else
-						goto _found;
+				badness = width - w + 32;
-					unsignd1 = !unsignd1;
+			badness += snd_pcm_format_unsigned(f) != unsignd;
-				}
+			badness += snd_pcm_format_big_endian(f) != big;
-				big1 = !big1;
+			if (badness < best) {
+				best_format = f;
+				best = badness;
 			}
-			if (width1 == 32) {
-				dwidth1 = -dwidth1;
-				width1 = width;
-			}
-			width1 += dwidth1;
 		}
-		return -EINVAL;
+		return best_format >= 0 ? best_format : -EINVAL;
-	_found:
-		return format1;
 	} else {
-		unsigned int i;
 		switch (format) {
 		case SNDRV_PCM_FORMAT_MU_LAW:
 			for (i = 0; i < ARRAY_SIZE(preferred_formats); ++i) {
@@ -740,5 +739,3 @@ int snd_pcm_area_copy(const struct snd_pcm_channel_area *src_area, size_t src_of
 	}
 	return 0;
 }
-
-#endif

sound/core/oss/pcm_plugin.h

@@ -3,7 +3,7 @@
 
 /*
  *  Digital Audio (Plugin interface) abstract layer
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify

sound/core/oss/plugin_ops.h

@@ -1,370 +0,0 @@
-/*
- *  Plugin sample operators with fast switch
- *  Copyright (c) 2000 by Jaroslav Kysela <perex@suse.cz>
- *
- *
- *   This library is free software; you can redistribute it and/or modify
- *   it under the terms of the GNU Library General Public License as
- *   published by the Free Software Foundation; either version 2 of
- *   the License, or (at your option) any later version.
- *
- *   This program is distributed in the hope that it will be useful,
- *   but WITHOUT ANY WARRANTY; without even the implied warranty of
- *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- *   GNU Library General Public License for more details.
- *
- *   You should have received a copy of the GNU Library General Public
- *   License along with this library; if not, write to the Free Software
- *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
- *
- */
-
-
-#define as_u8(ptr) (*(u_int8_t*)(ptr))
-#define as_u16(ptr) (*(u_int16_t*)(ptr))
-#define as_u32(ptr) (*(u_int32_t*)(ptr))
-#define as_u64(ptr) (*(u_int64_t*)(ptr))
-#define as_s8(ptr) (*(int8_t*)(ptr))
-#define as_s16(ptr) (*(int16_t*)(ptr))
-#define as_s32(ptr) (*(int32_t*)(ptr))
-#define as_s64(ptr) (*(int64_t*)(ptr))
-
-#ifdef COPY_LABELS
-static void *copy_labels[4] = {
-	&&copy_8,
-	&&copy_16,
-	&&copy_32,
-	&&copy_64
-};
-#endif
-
-#ifdef COPY_END
-while(0) {
-copy_8: as_s8(dst) = as_s8(src); goto COPY_END;
-copy_16: as_s16(dst) = as_s16(src); goto COPY_END;
-copy_32: as_s32(dst) = as_s32(src); goto COPY_END;
-copy_64: as_s64(dst) = as_s64(src); goto COPY_END;
-}
-#endif
-
-#ifdef CONV_LABELS
-/* src_wid src_endswap sign_toggle dst_wid dst_endswap */
-static void *conv_labels[4 * 2 * 2 * 4 * 2] = {
-	&&conv_xxx1_xxx1,	 /*  8h ->  8h */
-	&&conv_xxx1_xxx1,	 /*  8h ->  8s */
-	&&conv_xxx1_xx10,	 /*  8h -> 16h */
-	&&conv_xxx1_xx01,	 /*  8h -> 16s */
-	&&conv_xxx1_x100,	 /*  8h -> 24h */
-	&&conv_xxx1_001x,	 /*  8h -> 24s */
-	&&conv_xxx1_1000,	 /*  8h -> 32h */
-	&&conv_xxx1_0001,	 /*  8h -> 32s */
-	&&conv_xxx1_xxx9,	 /*  8h ^>  8h */
-	&&conv_xxx1_xxx9,	 /*  8h ^>  8s */
-	&&conv_xxx1_xx90,	 /*  8h ^> 16h */
-	&&conv_xxx1_xx09,	 /*  8h ^> 16s */
-	&&conv_xxx1_x900,	 /*  8h ^> 24h */
-	&&conv_xxx1_009x,	 /*  8h ^> 24s */
-	&&conv_xxx1_9000,	 /*  8h ^> 32h */
-	&&conv_xxx1_0009,	 /*  8h ^> 32s */
-	&&conv_xxx1_xxx1,	 /*  8s ->  8h */
-	&&conv_xxx1_xxx1,	 /*  8s ->  8s */
-	&&conv_xxx1_xx10,	 /*  8s -> 16h */
-	&&conv_xxx1_xx01,	 /*  8s -> 16s */
-	&&conv_xxx1_x100,	 /*  8s -> 24h */
-	&&conv_xxx1_001x,	 /*  8s -> 24s */
-	&&conv_xxx1_1000,	 /*  8s -> 32h */
-	&&conv_xxx1_0001,	 /*  8s -> 32s */
-	&&conv_xxx1_xxx9,	 /*  8s ^>  8h */
-	&&conv_xxx1_xxx9,	 /*  8s ^>  8s */
-	&&conv_xxx1_xx90,	 /*  8s ^> 16h */
-	&&conv_xxx1_xx09,	 /*  8s ^> 16s */
-	&&conv_xxx1_x900,	 /*  8s ^> 24h */
-	&&conv_xxx1_009x,	 /*  8s ^> 24s */
-	&&conv_xxx1_9000,	 /*  8s ^> 32h */
-	&&conv_xxx1_0009,	 /*  8s ^> 32s */
-	&&conv_xx12_xxx1,	 /* 16h ->  8h */
-	&&conv_xx12_xxx1,	 /* 16h ->  8s */
-	&&conv_xx12_xx12,	 /* 16h -> 16h */
-	&&conv_xx12_xx21,	 /* 16h -> 16s */
-	&&conv_xx12_x120,	 /* 16h -> 24h */
-	&&conv_xx12_021x,	 /* 16h -> 24s */
-	&&conv_xx12_1200,	 /* 16h -> 32h */
-	&&conv_xx12_0021,	 /* 16h -> 32s */
-	&&conv_xx12_xxx9,	 /* 16h ^>  8h */
-	&&conv_xx12_xxx9,	 /* 16h ^>  8s */
-	&&conv_xx12_xx92,	 /* 16h ^> 16h */
-	&&conv_xx12_xx29,	 /* 16h ^> 16s */
-	&&conv_xx12_x920,	 /* 16h ^> 24h */
-	&&conv_xx12_029x,	 /* 16h ^> 24s */
-	&&conv_xx12_9200,	 /* 16h ^> 32h */
-	&&conv_xx12_0029,	 /* 16h ^> 32s */
-	&&conv_xx12_xxx2,	 /* 16s ->  8h */
-	&&conv_xx12_xxx2,	 /* 16s ->  8s */
-	&&conv_xx12_xx21,	 /* 16s -> 16h */
-	&&conv_xx12_xx12,	 /* 16s -> 16s */
-	&&conv_xx12_x210,	 /* 16s -> 24h */
-	&&conv_xx12_012x,	 /* 16s -> 24s */
-	&&conv_xx12_2100,	 /* 16s -> 32h */
-	&&conv_xx12_0012,	 /* 16s -> 32s */
-	&&conv_xx12_xxxA,	 /* 16s ^>  8h */
-	&&conv_xx12_xxxA,	 /* 16s ^>  8s */
-	&&conv_xx12_xxA1,	 /* 16s ^> 16h */
-	&&conv_xx12_xx1A,	 /* 16s ^> 16s */
-	&&conv_xx12_xA10,	 /* 16s ^> 24h */
-	&&conv_xx12_01Ax,	 /* 16s ^> 24s */
-	&&conv_xx12_A100,	 /* 16s ^> 32h */
-	&&conv_xx12_001A,	 /* 16s ^> 32s */
-	&&conv_x123_xxx1,	 /* 24h ->  8h */
-	&&conv_x123_xxx1,	 /* 24h ->  8s */
-	&&conv_x123_xx12,	 /* 24h -> 16h */
-	&&conv_x123_xx21,	 /* 24h -> 16s */
-	&&conv_x123_x123,	 /* 24h -> 24h */
-	&&conv_x123_321x,	 /* 24h -> 24s */
-	&&conv_x123_1230,	 /* 24h -> 32h */
-	&&conv_x123_0321,	 /* 24h -> 32s */
-	&&conv_x123_xxx9,	 /* 24h ^>  8h */
-	&&conv_x123_xxx9,	 /* 24h ^>  8s */
-	&&conv_x123_xx92,	 /* 24h ^> 16h */
-	&&conv_x123_xx29,	 /* 24h ^> 16s */
-	&&conv_x123_x923,	 /* 24h ^> 24h */
-	&&conv_x123_329x,	 /* 24h ^> 24s */
-	&&conv_x123_9230,	 /* 24h ^> 32h */
-	&&conv_x123_0329,	 /* 24h ^> 32s */
-	&&conv_123x_xxx3,	 /* 24s ->  8h */
-	&&conv_123x_xxx3,	 /* 24s ->  8s */
-	&&conv_123x_xx32,	 /* 24s -> 16h */
-	&&conv_123x_xx23,	 /* 24s -> 16s */
-	&&conv_123x_x321,	 /* 24s -> 24h */
-	&&conv_123x_123x,	 /* 24s -> 24s */
-	&&conv_123x_3210,	 /* 24s -> 32h */
-	&&conv_123x_0123,	 /* 24s -> 32s */
-	&&conv_123x_xxxB,	 /* 24s ^>  8h */
-	&&conv_123x_xxxB,	 /* 24s ^>  8s */
-	&&conv_123x_xxB2,	 /* 24s ^> 16h */
-	&&conv_123x_xx2B,	 /* 24s ^> 16s */
-	&&conv_123x_xB21,	 /* 24s ^> 24h */
-	&&conv_123x_12Bx,	 /* 24s ^> 24s */
-	&&conv_123x_B210,	 /* 24s ^> 32h */
-	&&conv_123x_012B,	 /* 24s ^> 32s */
-	&&conv_1234_xxx1,	 /* 32h ->  8h */
-	&&conv_1234_xxx1,	 /* 32h ->  8s */
-	&&conv_1234_xx12,	 /* 32h -> 16h */
-	&&conv_1234_xx21,	 /* 32h -> 16s */
-	&&conv_1234_x123,	 /* 32h -> 24h */
-	&&conv_1234_321x,	 /* 32h -> 24s */
-	&&conv_1234_1234,	 /* 32h -> 32h */
-	&&conv_1234_4321,	 /* 32h -> 32s */
-	&&conv_1234_xxx9,	 /* 32h ^>  8h */
-	&&conv_1234_xxx9,	 /* 32h ^>  8s */
-	&&conv_1234_xx92,	 /* 32h ^> 16h */
-	&&conv_1234_xx29,	 /* 32h ^> 16s */
-	&&conv_1234_x923,	 /* 32h ^> 24h */
-	&&conv_1234_329x,	 /* 32h ^> 24s */
-	&&conv_1234_9234,	 /* 32h ^> 32h */
-	&&conv_1234_4329,	 /* 32h ^> 32s */
-	&&conv_1234_xxx4,	 /* 32s ->  8h */
-	&&conv_1234_xxx4,	 /* 32s ->  8s */
-	&&conv_1234_xx43,	 /* 32s -> 16h */
-	&&conv_1234_xx34,	 /* 32s -> 16s */
-	&&conv_1234_x432,	 /* 32s -> 24h */
-	&&conv_1234_234x,	 /* 32s -> 24s */
-	&&conv_1234_4321,	 /* 32s -> 32h */
-	&&conv_1234_1234,	 /* 32s -> 32s */
-	&&conv_1234_xxxC,	 /* 32s ^>  8h */
-	&&conv_1234_xxxC,	 /* 32s ^>  8s */
-	&&conv_1234_xxC3,	 /* 32s ^> 16h */
-	&&conv_1234_xx3C,	 /* 32s ^> 16s */
-	&&conv_1234_xC32,	 /* 32s ^> 24h */
-	&&conv_1234_23Cx,	 /* 32s ^> 24s */
-	&&conv_1234_C321,	 /* 32s ^> 32h */
-	&&conv_1234_123C,	 /* 32s ^> 32s */
-};
-#endif
-
-#ifdef CONV_END
-while(0) {
-conv_xxx1_xxx1: as_u8(dst) = as_u8(src); goto CONV_END;
-conv_xxx1_xx10: as_u16(dst) = (u_int16_t)as_u8(src) << 8; goto CONV_END;
-conv_xxx1_xx01: as_u16(dst) = (u_int16_t)as_u8(src); goto CONV_END;
-conv_xxx1_x100: as_u32(dst) = (u_int32_t)as_u8(src) << 16; goto CONV_END;
-conv_xxx1_001x: as_u32(dst) = (u_int32_t)as_u8(src) << 8; goto CONV_END;
-conv_xxx1_1000: as_u32(dst) = (u_int32_t)as_u8(src) << 24; goto CONV_END;
-conv_xxx1_0001: as_u32(dst) = (u_int32_t)as_u8(src); goto CONV_END;
-conv_xxx1_xxx9: as_u8(dst) = as_u8(src) ^ 0x80; goto CONV_END;
-conv_xxx1_xx90: as_u16(dst) = (u_int16_t)(as_u8(src) ^ 0x80) << 8; goto CONV_END;
-conv_xxx1_xx09: as_u16(dst) = (u_int16_t)(as_u8(src) ^ 0x80); goto CONV_END;
-conv_xxx1_x900: as_u32(dst) = (u_int32_t)(as_u8(src) ^ 0x80) << 16; goto CONV_END;
-conv_xxx1_009x: as_u32(dst) = (u_int32_t)(as_u8(src) ^ 0x80) << 8; goto CONV_END;
-conv_xxx1_9000: as_u32(dst) = (u_int32_t)(as_u8(src) ^ 0x80) << 24; goto CONV_END;
-conv_xxx1_0009: as_u32(dst) = (u_int32_t)(as_u8(src) ^ 0x80); goto CONV_END;
-conv_xx12_xxx1: as_u8(dst) = as_u16(src) >> 8; goto CONV_END;
-conv_xx12_xx12: as_u16(dst) = as_u16(src); goto CONV_END;
-conv_xx12_xx21: as_u16(dst) = swab16(as_u16(src)); goto CONV_END;
-conv_xx12_x120: as_u32(dst) = (u_int32_t)as_u16(src) << 8; goto CONV_END;
-conv_xx12_021x: as_u32(dst) = (u_int32_t)swab16(as_u16(src)) << 8; goto CONV_END;
-conv_xx12_1200: as_u32(dst) = (u_int32_t)as_u16(src) << 16; goto CONV_END;
-conv_xx12_0021: as_u32(dst) = (u_int32_t)swab16(as_u16(src)); goto CONV_END;
-conv_xx12_xxx9: as_u8(dst) = (as_u16(src) >> 8) ^ 0x80; goto CONV_END;
-conv_xx12_xx92: as_u16(dst) = as_u16(src) ^ 0x8000; goto CONV_END;
-conv_xx12_xx29: as_u16(dst) = swab16(as_u16(src)) ^ 0x80; goto CONV_END;
-conv_xx12_x920: as_u32(dst) = (u_int32_t)(as_u16(src) ^ 0x8000) << 8; goto CONV_END;
-conv_xx12_029x: as_u32(dst) = (u_int32_t)(swab16(as_u16(src)) ^ 0x80) << 8; goto CONV_END;
-conv_xx12_9200: as_u32(dst) = (u_int32_t)(as_u16(src) ^ 0x8000) << 16; goto CONV_END;
-conv_xx12_0029: as_u32(dst) = (u_int32_t)(swab16(as_u16(src)) ^ 0x80); goto CONV_END;
-conv_xx12_xxx2: as_u8(dst) = as_u16(src) & 0xff; goto CONV_END;
-conv_xx12_x210: as_u32(dst) = (u_int32_t)swab16(as_u16(src)) << 8; goto CONV_END;
-conv_xx12_012x: as_u32(dst) = (u_int32_t)as_u16(src) << 8; goto CONV_END;
-conv_xx12_2100: as_u32(dst) = (u_int32_t)swab16(as_u16(src)) << 16; goto CONV_END;
-conv_xx12_0012: as_u32(dst) = (u_int32_t)as_u16(src); goto CONV_END; 
-conv_xx12_xxxA: as_u8(dst) = (as_u16(src) ^ 0x80) & 0xff; goto CONV_END;
-conv_xx12_xxA1: as_u16(dst) = swab16(as_u16(src) ^ 0x80); goto CONV_END;
-conv_xx12_xx1A: as_u16(dst) = as_u16(src) ^ 0x80; goto CONV_END;
-conv_xx12_xA10: as_u32(dst) = (u_int32_t)swab16(as_u16(src) ^ 0x80) << 8; goto CONV_END;
-conv_xx12_01Ax: as_u32(dst) = (u_int32_t)(as_u16(src) ^ 0x80) << 8; goto CONV_END;
-conv_xx12_A100: as_u32(dst) = (u_int32_t)swab16(as_u16(src) ^ 0x80) << 16; goto CONV_END;
-conv_xx12_001A: as_u32(dst) = (u_int32_t)(as_u16(src) ^ 0x80); goto CONV_END;
-conv_x123_xxx1: as_u8(dst) = as_u32(src) >> 16; goto CONV_END;
-conv_x123_xx12: as_u16(dst) = as_u32(src) >> 8; goto CONV_END;
-conv_x123_xx21: as_u16(dst) = swab16(as_u32(src) >> 8); goto CONV_END;
-conv_x123_x123: as_u32(dst) = as_u32(src); goto CONV_END;
-conv_x123_321x: as_u32(dst) = swab32(as_u32(src)); goto CONV_END;
-conv_x123_1230: as_u32(dst) = as_u32(src) << 8; goto CONV_END;
-conv_x123_0321: as_u32(dst) = swab32(as_u32(src)) >> 8; goto CONV_END;
-conv_x123_xxx9: as_u8(dst) = (as_u32(src) >> 16) ^ 0x80; goto CONV_END;
-conv_x123_xx92: as_u16(dst) = (as_u32(src) >> 8) ^ 0x8000; goto CONV_END;
-conv_x123_xx29: as_u16(dst) = swab16(as_u32(src) >> 8) ^ 0x80; goto CONV_END;
-conv_x123_x923: as_u32(dst) = as_u32(src) ^ 0x800000; goto CONV_END;
-conv_x123_329x: as_u32(dst) = swab32(as_u32(src)) ^ 0x8000; goto CONV_END;
-conv_x123_9230: as_u32(dst) = (as_u32(src) ^ 0x800000) << 8; goto CONV_END;
-conv_x123_0329: as_u32(dst) = (swab32(as_u32(src)) >> 8) ^ 0x80; goto CONV_END;
-conv_123x_xxx3: as_u8(dst) = (as_u32(src) >> 8) & 0xff; goto CONV_END;
-conv_123x_xx32: as_u16(dst) = swab16(as_u32(src) >> 8); goto CONV_END;
-conv_123x_xx23: as_u16(dst) = (as_u32(src) >> 8) & 0xffff; goto CONV_END;
-conv_123x_x321: as_u32(dst) = swab32(as_u32(src)); goto CONV_END;
-conv_123x_123x: as_u32(dst) = as_u32(src); goto CONV_END;
-conv_123x_3210: as_u32(dst) = swab32(as_u32(src)) << 8; goto CONV_END;
-conv_123x_0123: as_u32(dst) = as_u32(src) >> 8; goto CONV_END;
-conv_123x_xxxB: as_u8(dst) = ((as_u32(src) >> 8) & 0xff) ^ 0x80; goto CONV_END;
-conv_123x_xxB2: as_u16(dst) = swab16((as_u32(src) >> 8) ^ 0x80); goto CONV_END;
-conv_123x_xx2B: as_u16(dst) = ((as_u32(src) >> 8) & 0xffff) ^ 0x80; goto CONV_END;
-conv_123x_xB21: as_u32(dst) = swab32(as_u32(src)) ^ 0x800000; goto CONV_END;
-conv_123x_12Bx: as_u32(dst) = as_u32(src) ^ 0x8000; goto CONV_END;
-conv_123x_B210: as_u32(dst) = swab32(as_u32(src) ^ 0x8000) << 8; goto CONV_END;
-conv_123x_012B: as_u32(dst) = (as_u32(src) >> 8) ^ 0x80; goto CONV_END;
-conv_1234_xxx1: as_u8(dst) = as_u32(src) >> 24; goto CONV_END;
-conv_1234_xx12: as_u16(dst) = as_u32(src) >> 16; goto CONV_END;
-conv_1234_xx21: as_u16(dst) = swab16(as_u32(src) >> 16); goto CONV_END;
-conv_1234_x123: as_u32(dst) = as_u32(src) >> 8; goto CONV_END;
-conv_1234_321x: as_u32(dst) = swab32(as_u32(src)) << 8; goto CONV_END;
-conv_1234_1234: as_u32(dst) = as_u32(src); goto CONV_END;
-conv_1234_4321: as_u32(dst) = swab32(as_u32(src)); goto CONV_END;
-conv_1234_xxx9: as_u8(dst) = (as_u32(src) >> 24) ^ 0x80; goto CONV_END;
-conv_1234_xx92: as_u16(dst) = (as_u32(src) >> 16) ^ 0x8000; goto CONV_END;
-conv_1234_xx29: as_u16(dst) = swab16(as_u32(src) >> 16) ^ 0x80; goto CONV_END;
-conv_1234_x923: as_u32(dst) = (as_u32(src) >> 8) ^ 0x800000; goto CONV_END;
-conv_1234_329x: as_u32(dst) = (swab32(as_u32(src)) ^ 0x80) << 8; goto CONV_END;
-conv_1234_9234: as_u32(dst) = as_u32(src) ^ 0x80000000; goto CONV_END;
-conv_1234_4329: as_u32(dst) = swab32(as_u32(src)) ^ 0x80; goto CONV_END;
-conv_1234_xxx4: as_u8(dst) = as_u32(src) & 0xff; goto CONV_END;
-conv_1234_xx43: as_u16(dst) = swab16(as_u32(src)); goto CONV_END;
-conv_1234_xx34: as_u16(dst) = as_u32(src) & 0xffff; goto CONV_END;
-conv_1234_x432: as_u32(dst) = swab32(as_u32(src)) >> 8; goto CONV_END;
-conv_1234_234x: as_u32(dst) = as_u32(src) << 8; goto CONV_END;
-conv_1234_xxxC: as_u8(dst) = (as_u32(src) & 0xff) ^ 0x80; goto CONV_END;
-conv_1234_xxC3: as_u16(dst) = swab16(as_u32(src) ^ 0x80); goto CONV_END;
-conv_1234_xx3C: as_u16(dst) = (as_u32(src) & 0xffff) ^ 0x80; goto CONV_END;
-conv_1234_xC32: as_u32(dst) = (swab32(as_u32(src)) >> 8) ^ 0x800000; goto CONV_END;
-conv_1234_23Cx: as_u32(dst) = (as_u32(src) ^ 0x80) << 8; goto CONV_END;
-conv_1234_C321: as_u32(dst) = swab32(as_u32(src) ^ 0x80); goto CONV_END;
-conv_1234_123C: as_u32(dst) = as_u32(src) ^ 0x80; goto CONV_END;
-}
-#endif
-
-#ifdef GET_S16_LABELS
-/* src_wid src_endswap unsigned */
-static void *get_s16_labels[4 * 2 * 2] = {
-	&&get_s16_xxx1_xx10,	 /*  8h -> 16h */
-	&&get_s16_xxx1_xx90,	 /*  8h ^> 16h */
-	&&get_s16_xxx1_xx10,	 /*  8s -> 16h */
-	&&get_s16_xxx1_xx90,	 /*  8s ^> 16h */
-	&&get_s16_xx12_xx12,	 /* 16h -> 16h */
-	&&get_s16_xx12_xx92,	 /* 16h ^> 16h */
-	&&get_s16_xx12_xx21,	 /* 16s -> 16h */
-	&&get_s16_xx12_xxA1,	 /* 16s ^> 16h */
-	&&get_s16_x123_xx12,	 /* 24h -> 16h */
-	&&get_s16_x123_xx92,	 /* 24h ^> 16h */
-	&&get_s16_123x_xx32,	 /* 24s -> 16h */
-	&&get_s16_123x_xxB2,	 /* 24s ^> 16h */
-	&&get_s16_1234_xx12,	 /* 32h -> 16h */
-	&&get_s16_1234_xx92,	 /* 32h ^> 16h */
-	&&get_s16_1234_xx43,	 /* 32s -> 16h */
-	&&get_s16_1234_xxC3,	 /* 32s ^> 16h */
-};
-#endif
-
-#ifdef GET_S16_END
-while(0) {
-get_s16_xxx1_xx10: sample = (u_int16_t)as_u8(src) << 8; goto GET_S16_END;
-get_s16_xxx1_xx90: sample = (u_int16_t)(as_u8(src) ^ 0x80) << 8; goto GET_S16_END;
-get_s16_xx12_xx12: sample = as_u16(src); goto GET_S16_END;
-get_s16_xx12_xx92: sample = as_u16(src) ^ 0x8000; goto GET_S16_END;
-get_s16_xx12_xx21: sample = swab16(as_u16(src)); goto GET_S16_END;
-get_s16_xx12_xxA1: sample = swab16(as_u16(src) ^ 0x80); goto GET_S16_END;
-get_s16_x123_xx12: sample = as_u32(src) >> 8; goto GET_S16_END;
-get_s16_x123_xx92: sample = (as_u32(src) >> 8) ^ 0x8000; goto GET_S16_END;
-get_s16_123x_xx32: sample = swab16(as_u32(src) >> 8); goto GET_S16_END;
-get_s16_123x_xxB2: sample = swab16((as_u32(src) >> 8) ^ 0x8000); goto GET_S16_END;
-get_s16_1234_xx12: sample = as_u32(src) >> 16; goto GET_S16_END;
-get_s16_1234_xx92: sample = (as_u32(src) >> 16) ^ 0x8000; goto GET_S16_END;
-get_s16_1234_xx43: sample = swab16(as_u32(src)); goto GET_S16_END;
-get_s16_1234_xxC3: sample = swab16(as_u32(src) ^ 0x80); goto GET_S16_END;
-}
-#endif
-
-#ifdef PUT_S16_LABELS
-/* dst_wid dst_endswap unsigned */
-static void *put_s16_labels[4 * 2 * 2] = {
-	&&put_s16_xx12_xxx1,	 /* 16h ->  8h */
-	&&put_s16_xx12_xxx9,	 /* 16h ^>  8h */
-	&&put_s16_xx12_xxx1,	 /* 16h ->  8s */
-	&&put_s16_xx12_xxx9,	 /* 16h ^>  8s */
-	&&put_s16_xx12_xx12,	 /* 16h -> 16h */
-	&&put_s16_xx12_xx92,	 /* 16h ^> 16h */
-	&&put_s16_xx12_xx21,	 /* 16h -> 16s */
-	&&put_s16_xx12_xx29,	 /* 16h ^> 16s */
-	&&put_s16_xx12_x120,	 /* 16h -> 24h */
-	&&put_s16_xx12_x920,	 /* 16h ^> 24h */
-	&&put_s16_xx12_021x,	 /* 16h -> 24s */
-	&&put_s16_xx12_029x,	 /* 16h ^> 24s */
-	&&put_s16_xx12_1200,	 /* 16h -> 32h */
-	&&put_s16_xx12_9200,	 /* 16h ^> 32h */
-	&&put_s16_xx12_0021,	 /* 16h -> 32s */
-	&&put_s16_xx12_0029,	 /* 16h ^> 32s */
-};
-#endif
-
-#ifdef PUT_S16_END
-while (0) {
-put_s16_xx12_xxx1: as_u8(dst) = sample >> 8; goto PUT_S16_END;
-put_s16_xx12_xxx9: as_u8(dst) = (sample >> 8) ^ 0x80; goto PUT_S16_END;
-put_s16_xx12_xx12: as_u16(dst) = sample; goto PUT_S16_END;
-put_s16_xx12_xx92: as_u16(dst) = sample ^ 0x8000; goto PUT_S16_END;
-put_s16_xx12_xx21: as_u16(dst) = swab16(sample); goto PUT_S16_END;
-put_s16_xx12_xx29: as_u16(dst) = swab16(sample) ^ 0x80; goto PUT_S16_END;
-put_s16_xx12_x120: as_u32(dst) = (u_int32_t)sample << 8; goto PUT_S16_END;
-put_s16_xx12_x920: as_u32(dst) = (u_int32_t)(sample ^ 0x8000) << 8; goto PUT_S16_END;
-put_s16_xx12_021x: as_u32(dst) = (u_int32_t)swab16(sample) << 8; goto PUT_S16_END;
-put_s16_xx12_029x: as_u32(dst) = (u_int32_t)(swab16(sample) ^ 0x80) << 8; goto PUT_S16_END;
-put_s16_xx12_1200: as_u32(dst) = (u_int32_t)sample << 16; goto PUT_S16_END;
-put_s16_xx12_9200: as_u32(dst) = (u_int32_t)(sample ^ 0x8000) << 16; goto PUT_S16_END;
-put_s16_xx12_0021: as_u32(dst) = (u_int32_t)swab16(sample); goto PUT_S16_END;
-put_s16_xx12_0029: as_u32(dst) = (u_int32_t)swab16(sample) ^ 0x80; goto PUT_S16_END;
-}
-#endif
-
-#undef as_u8
-#undef as_u16
-#undef as_u32
-#undef as_s8
-#undef as_s16
-#undef as_s32

sound/core/oss/rate.c

@@ -1,6 +1,6 @@
 /*
  *  Rate conversion Plug-In
- *  Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This library is free software; you can redistribute it and/or modify
@@ -20,9 +20,6 @@
  */
   
 #include <sound/driver.h>
-
-#ifdef CONFIG_SND_PCM_OSS_PLUGINS
-
 #include <linux/time.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -340,5 +337,3 @@ int snd_pcm_plugin_build_rate(struct snd_pcm_substream *plug,
 	*r_plugin = plugin;
 	return 0;
 }
-
-#endif

sound/core/oss/route.c

@@ -20,9 +20,6 @@
  */
 
 #include <sound/driver.h>
-
-#ifdef CONFIG_SND_PCM_OSS_PLUGINS
-
 #include <linux/slab.h>
 #include <linux/time.h>
 #include <sound/core.h>
@@ -108,5 +105,3 @@ int snd_pcm_plugin_build_route(struct snd_pcm_substream *plug,
 	*r_plugin = plugin;
 	return 0;
 }
-
-#endif

sound/core/pcm.c

@@ -1,6 +1,6 @@
 /*
  *  Digital Audio (PCM) abstract layer
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify
@@ -30,7 +30,7 @@
 #include <sound/control.h>
 #include <sound/info.h>
 
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>, Abramo Bagnara <abramo@alsa-project.org>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, Abramo Bagnara <abramo@alsa-project.org>");
 MODULE_DESCRIPTION("Midlevel PCM code for ALSA.");
 MODULE_LICENSE("GPL");
 

sound/core/pcm_lib.c

@@ -1,6 +1,6 @@
 /*
  *  Digital Audio (PCM) abstract layer
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *                   Abramo Bagnara <abramo@alsa-project.org>
  *
  *

sound/core/pcm_memory.c

@@ -1,6 +1,6 @@
 /*
  *  Digital Audio (PCM) abstract layer
- *  Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This program is free software; you can redistribute it and/or modify

sound/core/pcm_misc.c

@@ -1,6 +1,6 @@
 /*
  *  PCM Interface - misc routines
- *  Copyright (c) 1998 by Jaroslav Kysela <perex@suse.cz>
+ *  Copyright (c) 1998 by Jaroslav Kysela <perex@perex.cz>
  *
  *
  *   This library is free software; you can redistribute it and/or modify
@@ -422,38 +422,6 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int
 
 EXPORT_SYMBOL(snd_pcm_format_set_silence);
 
-/* [width][unsigned][bigendian] */
-static int linear_formats[4][2][2] = {
-	{{ SNDRV_PCM_FORMAT_S8, SNDRV_PCM_FORMAT_S8},
-	 { SNDRV_PCM_FORMAT_U8, SNDRV_PCM_FORMAT_U8}},
-	{{SNDRV_PCM_FORMAT_S16_LE, SNDRV_PCM_FORMAT_S16_BE},
-	 {SNDRV_PCM_FORMAT_U16_LE, SNDRV_PCM_FORMAT_U16_BE}},
-	{{SNDRV_PCM_FORMAT_S24_LE, SNDRV_PCM_FORMAT_S24_BE},
-	 {SNDRV_PCM_FORMAT_U24_LE, SNDRV_PCM_FORMAT_U24_BE}},
-	{{SNDRV_PCM_FORMAT_S32_LE, SNDRV_PCM_FORMAT_S32_BE},
-	 {SNDRV_PCM_FORMAT_U32_LE, SNDRV_PCM_FORMAT_U32_BE}}
-};
-
-/**
- * snd_pcm_build_linear_format - return the suitable linear format for the given condition
- * @width: the bit-width
- * @unsignd: 1 if unsigned, 0 if signed.
- * @big_endian: 1 if big-endian, 0 if little-endian
- *
- * Returns the suitable linear format for the given condition.
- */
-snd_pcm_format_t snd_pcm_build_linear_format(int width, int unsignd, int big_endian)
-{
-	if (width & 7)
-		return SND_PCM_FORMAT_UNKNOWN;
-	width = (width / 8) - 1;
-	if (width < 0 || width >= 4)
-		return SND_PCM_FORMAT_UNKNOWN;
-	return linear_formats[width][!!unsignd][!!big_endian];
-}
-
-EXPORT_SYMBOL(snd_pcm_build_linear_format);
-
 /**
  * snd_pcm_limit_hw_rates - determine rate_min/rate_max fields
  * @runtime: the runtime instance
@@ -465,21 +433,16 @@ EXPORT_SYMBOL(snd_pcm_build_linear_format);
  */
 int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime)
 {
-	static unsigned rates[] = {
-		/* ATTENTION: these values depend on the definition in pcm.h! */
-		5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000,
-		64000, 88200, 96000, 176400, 192000
-	};
 	int i;
-	for (i = 0; i < (int)ARRAY_SIZE(rates); i++) {
+	for (i = 0; i < (int)snd_pcm_known_rates.count; i++) {
 		if (runtime->hw.rates & (1 << i)) {
-			runtime->hw.rate_min = rates[i];
+			runtime->hw.rate_min = snd_pcm_known_rates.list[i];
 			break;
 		}
 	}
-	for (i = (int)ARRAY_SIZE(rates) - 1; i >= 0; i--) {
+	for (i = (int)snd_pcm_known_rates.count - 1; i >= 0; i--) {
 		if (runtime->hw.rates & (1 << i)) {
-			runtime->hw.rate_max = rates[i];
+			runtime->hw.rate_max = snd_pcm_known_rates.list[i];
 			break;
 		}
 	}
@@ -487,3 +450,21 @@ int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime)
 }
 
 EXPORT_SYMBOL(snd_pcm_limit_hw_rates);
+
+/**
+ * snd_pcm_rate_to_rate_bit - converts sample rate to SNDRV_PCM_RATE_xxx bit
+ * @rate: the sample rate to convert
+ *
+ * Returns the SNDRV_PCM_RATE_xxx flag that corresponds to the given rate, or
+ * SNDRV_PCM_RATE_KNOT for an unknown rate.
+ */
+unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate)
+{
+	unsigned int i;
+
+	for (i = 0; i < snd_pcm_known_rates.count; i++)
+		if (snd_pcm_known_rates.list[i] == rate)
+			return 1u << i;
+	return SNDRV_PCM_RATE_KNOT;
+}
+EXPORT_SYMBOL(snd_pcm_rate_to_rate_bit);