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: (102 commits)
  [ALSA] version 1.0.14
  [ALSA] remove duplicate Logitech Quickcam USB ID in usbquirks.h
  [ALSA] hda-codec - Fix input with STAC92xx
  [ALSA] hda-intel: support for iMac 24'' released on 09/2006
  [ALSA] hda-codec - Add quirk for Asus P5LD2
  [ALSA] snd-ca0106: Add support for X-Fi Extreme Audio.
  [ALSA] snd-emu10k1:Enable E-Mu 1616m notebook firmware loading.
  [ALSA] snd-emu10k1: Initial support for E-Mu 1616 and 1616m.
  [ALSA] cs46xx - Fix PM resume
  [ALSA] hda: Enable SPDIF in/out on some stac9205 boards
  [ALSA] timer: check for incorrect device state in non-debug compiles, too
  [ALSA] snd-aoa-codec-onyx: fix typo
  [ALSA] hda-codec - Add quirks for HP dx2200/dx2250
  [ALSA] hda-codec - Rename HP model-specific quirks
  [ALSA] hda-codec - Add quirk for HP Samba
  [ALSA] hda-codec - Add LG LW20 line-in capture source
  [ALSA] usb-audio - Fix AC3 with M-Audio Audiophile USB
  [ALSA] hda: stac9202 mixer fix
  [ALSA] Make s3c24xx_i2s_set_clkdiv() change the correct bits
  [ALSA] hda-codec - Add LG LW20 si3054 modem id
  ...

CREDITS

@@ -2212,13 +2212,13 @@ S: 2300 Copenhagen S
 S: Denmark
 
 N: Claudio S. Matsuoka
-E: claudio@conectiva.com
-E: claudio@helllabs.org
+E: cmatsuoka@gmail.com
+E: claudio@mandriva.com
 W: http://helllabs.org/~claudio
-D: V4L, OV511 driver hacks
+D: V4L, OV511 and HDA-codec hacks
 S: Conectiva S.A.
-S: R. Tocantins 89
-S: 80050-430  Curitiba PR
+S: Souza Naves 1250
+S: 80050-040  Curitiba PR
 S: Brazil
 
 N: Heinz Mauelshagen

Documentation/sound/alsa/ALSA-Configuration.txt

@@ -467,7 +467,12 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     above explicitly.
 
     The power-management is supported.
-    
+
+  Module snd-cs5530
+  _________________
+
+    Module for Cyrix/NatSemi Geode 5530 chip. 
+  
   Module snd-cs5535audio
   ----------------------
 
@@ -759,6 +764,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     model	- force the model name
     position_fix - Fix DMA pointer (0 = auto, 1 = none, 2 = POSBUF, 3 = FIFO size)
+    probe_mask  - Bitmask to probe codecs (default = -1, meaning all slots)
     single_cmd  - Use single immediate commands to communicate with
 		codecs (for debugging only)
     enable_msi	- Enable Message Signaled Interrupt (MSI) (default = off)
@@ -803,6 +809,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  hp-3013	HP machines (3013-variant)
 	  fujitsu	Fujitsu S7020
 	  acer		Acer TravelMate
+	  will		Will laptops (PB V7900)
+	  replacer	Replacer 672V
 	  basic		fixed pin assignment (old default model)
 	  auto		auto-config reading BIOS (default)
 
@@ -811,16 +819,31 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  hp-bpc	HP xw4400/6400/8400/9400 laptops
 	  hp-bpc-d7000	HP BPC D7000
 	  benq		Benq ED8
+	  benq-t31	Benq T31
 	  hippo		Hippo (ATI) with jack detection, Sony UX-90s
 	  hippo_1	Hippo (Benq) with jack detection
+	  sony-assamd	Sony ASSAMD
 	  basic		fixed pin assignment w/o SPDIF
 	  auto		auto-config reading BIOS (default)
 
+	ALC268
+	  3stack	3-stack model
+	  auto		auto-config reading BIOS (default)
+
+	ALC662
+	  3stack-dig	3-stack (2-channel) with SPDIF
+	  3stack-6ch	 3-stack (6-channel)
+	  3stack-6ch-dig 3-stack (6-channel) with SPDIF
+	  6stack-dig	 6-stack with SPDIF
+	  lenovo-101e	 Lenovo laptop
+	  auto		auto-config reading BIOS (default)
+
 	ALC882/885
 	  3stack-dig	3-jack with SPDIF I/O
 	  6stack-dig	6-jack digital with SPDIF I/O
 	  arima		Arima W820Di1
 	  macpro	MacPro support
+	  imac24	iMac 24'' with jack detection
 	  w2jc		ASUS W2JC
 	  auto		auto-config reading BIOS (default)
 
@@ -832,9 +855,15 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  6stack-dig-demo  6-jack digital for Intel demo board
 	  acer		Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc)
 	  medion	Medion Laptops
+	  medion-md2	Medion MD2
 	  targa-dig	Targa/MSI
 	  targa-2ch-dig	Targs/MSI with 2-channel
 	  laptop-eapd   3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE)
+	  lenovo-101e	Lenovo 101E
+	  lenovo-nb0763	Lenovo NB0763
+	  lenovo-ms7195-dig Lenovo MS7195
+	  6stack-hp	HP machines with 6stack (Nettle boards)
+	  3stack-hp	HP machines with 3stack (Lucknow, Samba boards)
 	  auto		auto-config reading BIOS (default)
 
 	ALC861/660
@@ -853,7 +882,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  3stack-dig	3-jack with SPDIF OUT
 	  6stack-dig	6-jack with SPDIF OUT
 	  3stack-660	3-jack (for ALC660VD)
+	  3stack-660-digout 3-jack with SPDIF OUT (for ALC660VD)
 	  lenovo	Lenovo 3000 C200
+	  dallas	Dallas laptops
 	  auto		auto-config reading BIOS (default)
 
 	CMI9880
@@ -864,12 +895,26 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  allout	5-jack in back, 2-jack in front, SPDIF out
 	  auto		auto-config reading BIOS (default)
 
+	AD1882
+	  3stack	3-stack mode (default)
+	  6stack	6-stack mode
+
+	AD1884
+	  N/A
+
 	AD1981
 	  basic		3-jack (default)
 	  hp		HP nx6320
 	  thinkpad	Lenovo Thinkpad T60/X60/Z60
 	  toshiba	Toshiba U205
 
+	AD1983
+	  N/A
+
+	AD1984
+	  basic		default configuration
+	  thinkpad	Lenovo Thinkpad T61/X61
+
 	AD1986A
 	  6stack	6-jack, separate surrounds (default)
 	  3stack	3-stack, shared surrounds
@@ -907,11 +952,18 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  ref		Reference board
 	  3stack	D945 3stack
 	  5stack	D945 5stack + SPDIF
-	  macmini	Intel Mac Mini
-	  macbook	Intel Mac Book
-	  macbook-pro-v1 Intel Mac Book Pro 1st generation
-	  macbook-pro	Intel Mac Book Pro 2nd generation
-	  imac-intel	Intel iMac
+	  dell		Dell XPS M1210
+	  intel-mac-v1	Intel Mac Type 1
+	  intel-mac-v2	Intel Mac Type 2
+	  intel-mac-v3	Intel Mac Type 3
+	  intel-mac-v4	Intel Mac Type 4
+	  intel-mac-v5	Intel Mac Type 5
+	  macmini	Intel Mac Mini (equivalent with type 3)
+	  macbook	Intel Mac Book (eq. type 5)
+	  macbook-pro-v1 Intel Mac Book Pro 1st generation (eq. type 3)
+	  macbook-pro	Intel Mac Book Pro 2nd generation (eq. type 3)
+	  imac-intel	Intel iMac (eq. type 2)
+	  imac-intel-20	Intel iMac (newer version) (eq. type 3)
 
 	STAC9202/9250/9251
 	  ref		Reference board, base config
@@ -956,6 +1008,17 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     from the irq.  Remember this is a last resort, and should be
     avoided as much as possible...
     
+    MORE NOTES ON "azx_get_response timeout" PROBLEMS:
+    On some hardwares, you may need to add a proper probe_mask option
+    to avoid the "azx_get_response timeout" problem above, instead.
+    This occurs when the access to non-existing or non-working codec slot
+    (likely a modem one) causes a stall of the communication via HD-audio
+    bus.  You can see which codec slots are probed by enabling
+    CONFIG_SND_DEBUG_DETECT, or simply from the file name of the codec
+    proc files.  Then limit the slots to probe by probe_mask option.
+    For example, probe_mask=1 means to probe only the first slot, and
+    probe_mask=4 means only the third slot.
+
     The power-management is supported.
 
   Module snd-hdsp

Documentation/sound/alsa/Audiophile-Usb.txt

-	Guide to using M-Audio Audiophile USB with ALSA and Jack	v1.3
+	Guide to using M-Audio Audiophile USB with ALSA and Jack	v1.5
 	========================================================
 
 	    Thibault Le Meur <Thibault.LeMeur@supelec.fr>
@@ -6,8 +6,19 @@
 This document is a guide to using the M-Audio Audiophile USB (tm) device with 
 ALSA and JACK.
 
+History
+=======
+* v1.4 - Thibault Le Meur (2007-07-11)
+ - Added Low Endianness nature of 16bits-modes
+   found by Hakan Lennestal <Hakan.Lennestal@brfsodrahamn.se>
+ - Modifying document structure
+* v1.5 - Thibault Le Meur (2007-07-12)
+ - Added AC3/DTS passthru info
+
+
 1 - Audiophile USB Specs and correct usage
 ==========================================
+
 This part is a reminder of important facts about the functions and limitations 
 of the device.
 
@@ -25,18 +36,18 @@ The device has 4 audio interfaces, and 2 MIDI ports:
 The internal DAC/ADC has the following characteristics:
 * sample depth of 16 or 24 bits
 * sample rate from 8kHz to 96kHz
-* Two ports can't use different sample depths at the same time. Moreover, the 
-Audiophile USB documentation gives the following Warning: "Please exit any 
-audio application running before switching between bit depths"
+* Two interfaces can't use different sample depths at the same time.
+Moreover, the Audiophile USB documentation gives the following Warning:
+"Please exit any audio application running before switching between bit depths"
 
 Due to the USB 1.1 bandwidth limitation, a limited number of interfaces can be 
 activated at the same time depending on the audio mode selected:
- * 16-bit/48kHz ==> 4 channels in/4 channels out
+ * 16-bit/48kHz ==> 4 channels in + 4 channels out
    - Ai+Ao+Di+Do
- * 24-bit/48kHz ==> 4 channels in/2 channels out, 
-                    or 2 channels in/4 channels out
+ * 24-bit/48kHz ==> 4 channels in + 2 channels out, 
+                    or 2 channels in + 4 channels out
    - Ai+Ao+Do or Ai+Di+Ao or Ai+Di+Do or Di+Ao+Do
- * 24-bit/96kHz ==> 2 channels in, or 2 channels out (half duplex only)
+ * 24-bit/96kHz ==> 2 channels in _or_ 2 channels out (half duplex only)
    - Ai or Ao or Di or Do
 
 Important facts about the Digital interface:
@@ -52,44 +63,56 @@ source is connected
 synchronization error (for instance sound played at an odd sample rate)
 
 
-2 - Audiophile USB support in ALSA
-==================================
+2 - Audiophile USB MIDI support in ALSA
+=======================================
 
-2.1 - MIDI ports
-----------------
-The Audiophile USB MIDI ports will be automatically supported once the 
+The Audiophile USB MIDI ports will be automatically supported once the
 following modules have been loaded:
  * snd-usb-audio
  * snd-seq-midi
 
 No additional setting is required.
 
-2.2 - Audio ports
------------------
+
+3 - Audiophile USB Audio support in ALSA
+========================================
 
 Audio functions of the Audiophile USB device are handled by the snd-usb-audio 
 module. This module can work in a default mode (without any device-specific 
 parameter), or in an "advanced" mode with the device-specific parameter called 
 "device_setup".
 
-2.2.1 - Default Alsa driver mode
-
-The default behavior of the snd-usb-audio driver is to parse the device 
-capabilities at startup and enable all functions inside the device (including 
-all ports at any supported sample rates and sample depths). This approach 
-has the advantage to let the driver easily switch from sample rates/depths 
-automatically according to the need of the application claiming the device.
-
-In this case the Audiophile ports are mapped to alsa pcm devices in the 
-following way (I suppose the device's index is 1):
+3.1 - Default Alsa driver mode
+------------------------------
+
+The default behavior of the snd-usb-audio driver is to list the device 
+capabilities at startup and activate the required mode when required 
+by the applications: for instance if the user is recording in a 
+24bit-depth-mode and immediately after wants to switch to a 16bit-depth mode,
+the snd-usb-audio module will reconfigure the device on the fly.
+
+This approach has the advantage to let the driver automatically switch from sample 
+rates/depths automatically according to the user's needs. However, those who 
+are using the device under windows know that this is not how the device is meant to
+work: under windows applications must be closed before using the m-audio control
+panel to switch the device working mode. Thus as we'll see in next section, this 
+Default Alsa driver mode can lead to device misconfigurations.
+
+Let's get back to the Default Alsa driver mode for now.  In this case the 
+Audiophile interfaces are mapped to alsa pcm devices in the following 
+way (I suppose the device's index is 1):
  * hw:1,0 is Ao in playback and Di in capture
  * hw:1,1 is Do in playback and Ai in capture
  * hw:1,2 is Do in AC3/DTS passthrough mode
 
-You must note as well that the device uses Big Endian byte encoding so that 
-supported audio format are S16_BE  for 16-bit depth modes and S24_3BE for 
-24-bits depth mode. One exception is the hw:1,2 port which is Little Endian 
-compliant and thus uses S16_LE.
+In this mode, the device uses Big Endian byte-encoding so that 
+supported audio format are S16_BE for 16-bit depth modes and S24_3BE for 
+24-bits depth mode.
+
+One exception is the hw:1,2 port which was reported to be Little Endian 
+compliant (supposedly supporting S16_LE) but processes in fact only S16_BE streams.
+This has been fixed in kernel 2.6.23 and above and now the hw:1,2 interface 
+is reported to be big endian in this default driver mode.
 
 Examples:
  * playing a S24_3BE encoded raw file to the Ao port
@@ -98,22 +121,26 @@ Examples:
    % arecord -D hw:1,1 -c2  -t raw -r48000 -fS24_3BE test.raw
  * playing a S16_BE encoded raw file to the Do port
    % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test.raw
+ * playing an ac3 sample file to the Do port
+   % aplay -D hw:1,2 --channels=6 ac3_S16_BE_encoded_file.raw
 
-If you're happy with the default Alsa driver setup and don't experience any 
+If you're happy with the default Alsa driver mode and don't experience any 
 issue with this mode, then you can skip the following chapter.
 
-2.2.2 - Advanced module setup
+3.2 - Advanced module setup
+---------------------------
 
 Due to the hardware constraints described above, the device initialization made 
 by the Alsa driver in default mode may result in a corrupted state of the 
 device. For instance, a particularly annoying issue is that the sound captured 
-from the Ai port sounds distorted (as if boosted with an excessive high volume 
-gain).
+from the Ai interface sounds distorted (as if boosted with an excessive high
+volume gain).
 
 For people having this problem, the snd-usb-audio module has a new module 
-parameter called "device_setup".
+parameter called "device_setup" (this parameter was introduced in kernel
+release 2.6.17)
 
-2.2.2.1 - Initializing the working mode of the Audiophile USB
+3.2.1 - Initializing the working mode of the Audiophile USB
 
 As far as the Audiophile USB device is concerned, this value let the user 
 specify:
@@ -121,33 +148,57 @@ specify:
  * the sample rate
  * whether the Di port is used or not 
 
-Here is a list of supported device_setup values for this device:
- * device_setup=0x00 (or omitted)
-   - Alsa driver default mode
-   - maintains backward compatibility with setups that do not use this 
-     parameter by not introducing any change
-   - results sometimes in corrupted sound as described earlier
+When initialized with "device_setup=0x00", the snd-usb-audio module has
+the same behaviour as when the parameter is omitted (see paragraph "Default 
+Alsa driver mode" above)
+
+Others modes are described in the following subsections.
+
+3.2.1.1 - 16-bit modes
+
+The two supported modes are:
+
  * device_setup=0x01
    - 16bits 48kHz mode with Di disabled
    - Ai,Ao,Do can be used at the same time
    - hw:1,0 is not available in capture mode
    - hw:1,2 is not available
+
  * device_setup=0x11
    - 16bits 48kHz mode with Di enabled
    - Ai,Ao,Di,Do can be used at the same time
    - hw:1,0 is available in capture mode
    - hw:1,2 is not available
+
+In this modes the device operates only at 16bits-modes. Before kernel 2.6.23,
+the devices where reported to be Big-Endian when in fact they were Little-Endian
+so that playing a file was a matter of using:
+   % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test_S16_LE.raw
+where "test_S16_LE.raw" was in fact a little-endian sample file.
+
+Thanks to Hakan Lennestal (who discovered the Little-Endiannes of the device in
+these modes) a fix has been committed (expected in kernel 2.6.23) and
+Alsa now reports Little-Endian interfaces. Thus playing a file now is as simple as
+using:
+   % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_LE test_S16_LE.raw
+
+3.2.1.2 - 24-bit modes
+
+The three supported modes are:
+
  * device_setup=0x09
    - 24bits 48kHz mode with Di disabled
    - Ai,Ao,Do can be used at the same time
    - hw:1,0 is not available in capture mode
    - hw:1,2 is not available
+
  * device_setup=0x19
    - 24bits 48kHz mode with Di enabled
    - 3 ports from {Ai,Ao,Di,Do} can be used at the same time
    - hw:1,0 is available in capture mode and an active digital source must be 
      connected to Di
    - hw:1,2 is not available
+
  * device_setup=0x0D or 0x10
    - 24bits 96kHz mode
    - Di is enabled by default for this mode but does not need to be connected 
@@ -155,34 +206,64 @@ Here is a list of supported device_setup values for this device:
    - Only 1 port from {Ai,Ao,Di,Do} can be used at the same time
    - hw:1,0 is available in captured mode
    - hw:1,2 is not available
+
+In these modes the device is only Big-Endian compliant (see "Default Alsa driver 
+mode" above for an aplay command example)
+
+3.2.1.3 - AC3 w/ DTS passthru mode
+
+Thanks to Hakan Lennestal, I now have a report saying that this mode works.
+
  * device_setup=0x03
    - 16bits 48kHz mode with only the Do port enabled 
-   - AC3 with DTS passthru (not tested)
+   - AC3 with DTS passthru
    - Caution with this setup the Do port is mapped to the pcm device hw:1,0
 
-2.2.2.2 - Setting and switching configurations with the device_setup parameter
+The command line used to playback the AC3/DTS encoded .wav-files in this mode:
+   % aplay -D hw:1,0 --channels=6 ac3_S16_LE_encoded_file.raw
+
+3.2.2 - How to use the device_setup parameter
+----------------------------------------------
 
 The parameter can be given:
+
  * By manually probing the device (as root):
    # modprobe -r snd-usb-audio
    # modprobe snd-usb-audio index=1 device_setup=0x09
+
  * Or while configuring the modules options in your modules configuration file
    - For Fedora distributions, edit the /etc/modprobe.conf file:
        alias snd-card-1 snd-usb-audio
        options snd-usb-audio index=1 device_setup=0x09
 
-IMPORTANT NOTE WHEN SWITCHING CONFIGURATION:
--------------------------------------------
- * You may need to _first_ initialize the module with the correct device_setup 
-   parameter and _only_after_ turn on the Audiophile USB device
- * This is especially true when switching the sample depth:
+CAUTION when initializaing the device
+-------------------------------------
+
+ * Correct initialization on the device requires that device_setup is given to
+   the module BEFORE the device is turned on. So, if you use the "manual probing"
+   method described above, take care to power-on the device AFTER this initialization.
+
+ * Failing to respect this will lead in a misconfiguration of the device. In this case
+   turn off the device, unproble the snd-usb-audio module, then probe it again with 
+   correct device_setup parameter and then (and only then) turn on the device again.
+
+ * If you've correctly initialized the device in a valid mode and then want to switch
+   to  another mode (possibly with another sample-depth), please use also the following 
+   procedure:
    - first turn off the device
    - de-register the snd-usb-audio module (modprobe -r)
    - change the device_setup parameter by changing the device_setup
      option in /etc/modprobe.conf 
    - turn on the device
+ * A workaround for this last issue has been applied to kernel 2.6.23, but it may not
+   be enough to ensure the 'stability' of the device initialization.
 
-2.2.2.3 - Audiophile USB's device_setup structure
+3.2.3 - Technical details for hackers
+-------------------------------------
+This section is for hackers, wanting to understand details about the device
+internals and how Alsa supports it.
+
+3.2.3.1 - Audiophile USB's device_setup structure
 
 If you want to understand the device_setup magic numbers for the Audiophile 
 USB, you need some very basic understanding of binary computation. However, 
@@ -228,12 +309,12 @@ Caution:
    - choosing b2 will prepare all interfaces for 24bits/96kHz but you'll
      only be able to use one at the same time
 
-2.2.3 -  USB implementation details for this device
+3.2.3.2 -  USB implementation details for this device
 
 You may safely skip this section if you're not interested in driver 
-development.
+hacking.
 
-This section describes some internal aspects of the device and summarize the 
+This section describes some internal aspects of the device and summarizes the 
 data I got by usb-snooping the windows and Linux drivers.
 
 The M-Audio Audiophile USB has 7 USB Interfaces:
@@ -293,43 +374,45 @@ parse_audio_endpoints function uses a quirk called
 "audiophile_skip_setting_quirk" in order to prevent AltSettings not 
 corresponding to device_setup from being registered in the driver.
 
-3 - Audiophile USB and Jack support
+4 - Audiophile USB and Jack support
 ===================================
 
 This section deals with support of the Audiophile USB device in Jack.
-The main issue regarding this support is that the device is Big Endian 
-compliant.
 
-3.1 - Using the plug alsa plugin
---------------------------------
+There are 2 main potential issues when using Jackd with the device:
+* support for Big-Endian devices in 24-bit modes
+* support for 4-in / 4-out channels
+
+4.1 - Direct support in Jackd
+-----------------------------
 
-Jack doesn't directly support big endian devices. Thus, one way to have support 
-for this device with Alsa is to use the Alsa "plug" converter.
+Jack supports big endian devices only in recent versions (thanks to
+Andreas Steinmetz for his first big-endian patch). I can't remember 
+extacly when this support was released into jackd, let's just say that 
+with jackd version 0.103.0 it's almost ok (just a small bug is affecting 
+16bits Big-Endian devices, but since you've read  carefully the above 
+paragraphs, you're now using kernel >= 2.6.23 and your 16bits devices 
+are now Little Endians ;-) ).
+
+You can run jackd with the following command for playback with Ao and
+record with Ai:
+  % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1
+
+4.2 - Using Alsa plughw
+-----------------------
+If you don't have a recent Jackd installed, you can downgrade to using
+the Alsa "plug" converter.
 
 For instance here is one way to run Jack with 2 playback channels on Ao and 2 
 capture channels from Ai:
   % jackd -R -dalsa -dplughw:1 -r48000 -p256 -n2 -D -Cplughw:1,1
 
-
 However you may see the following warning message:
 "You appear to be using the ALSA software "plug" layer, probably a result of 
 using the "default" ALSA device. This is less efficient than it could be. 
 Consider using a hardware device instead rather than using the plug layer."
 
-3.2 - Patching alsa to use direct pcm device
---------------------------------------------
-A patch for Jack by Andreas Steinmetz adds support for Big Endian devices. 
-However it has not been included in the CVS tree.
-
-You can find it at the following URL:
-http://sourceforge.net/tracker/index.php?func=detail&aid=1289682&group_id=39687&
-atid=425939
-
-After having applied the patch you can run jackd with the following command 
-line:
-  % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1
-
-3.2 - Getting 2 input and/or output interfaces in Jack
+4.3 - Getting 2 input and/or output interfaces in Jack
 ------------------------------------------------------
 
 As you can see, starting the Jack server this way will only enable 1 stereo
@@ -339,6 +422,7 @@ This is due to the following restrictions:
 * Jack can only open one capture device and one playback device at a time
 * The Audiophile USB is seen as 2 (or three) Alsa devices: hw:1,0, hw:1,1
   (and optionally hw:1,2)
+
 If you want to get Ai+Di and/or Ao+Do support with Jack, you would need to
 combine the Alsa devices into one logical "complex" device.
 
@@ -348,13 +432,11 @@ It is related to another device (ice1712) but can be adapted to suit
 the Audiophile USB.
 
 Enabling multiple Audiophile USB interfaces for Jackd will certainly require:
-* patching Jack with the previously mentioned "Big Endian" patch
-* patching Jackd with the MMAP_COMPLEX patch (see the ice1712 page)
-* patching the alsa-lib/src/pcm/pcm_multi.c file (see the ice1712 page)
+* Making sure your Jackd version has the MMAP_COMPLEX patch (see the ice1712 page)
+* (maybe) patching the alsa-lib/src/pcm/pcm_multi.c file (see the ice1712 page)
 * define a multi device (combination of hw:1,0 and hw:1,1) in your .asoundrc
   file 
 * start jackd with this device
 
-I had no success in testing this for now, but this may be due to my OS
-configuration. If you have any success with this kind of setup, please
-drop me an email.
+I had no success in testing this for now, if you have any success with this kind 
+of setup, please drop me an email.

Documentation/sound/alsa/OSS-Emulation.txt

@@ -278,6 +278,21 @@ current mixer configuration by reading and writing the whole file
 image.
 
 
+Duplex Streams
+==============
+
+Note that when attempting to use a single device file for playback and
+capture, the OSS API provides no way to set the format, sample rate or
+number of channels different in each direction.  Thus
+	io_handle = open("device", O_RDWR)
+will only function correctly if the values are the same in each direction.
+
+To use different values in the two directions, use both
+	input_handle = open("device", O_RDONLY)
+	output_handle = open("device", O_WRONLY)
+and set the values for the corresponding handle.
+
+
 Unsupported Features
 ====================
 

include/linux/i2c-id.h

@@ -115,9 +115,10 @@
 #define I2C_DRIVERID_KS0127	86	/* Samsung ks0127 video decoder */
 #define I2C_DRIVERID_TLV320AIC23B 87	/* TI TLV320AIC23B audio codec  */
 #define I2C_DRIVERID_ISL1208	88	/* Intersil ISL1208 RTC		*/
-#define I2C_DRIVERID_WM8731		89	/* Wolfson WM8731 audio codec */
-#define I2C_DRIVERID_WM8750		90	/* Wolfson WM8750 audio codec */
-#define I2C_DRIVERID_WM8753		91	/* Wolfson WM8753 audio codec */
+#define I2C_DRIVERID_WM8731	89	/* Wolfson WM8731 audio codec */
+#define I2C_DRIVERID_WM8750	90	/* Wolfson WM8750 audio codec */
+#define I2C_DRIVERID_WM8753	91	/* Wolfson WM8753 audio codec */
+#define I2C_DRIVERID_LM4857 	92 	/* LM4857 Audio Amplifier */
 
 #define I2C_DRIVERID_I2CDEV	900
 #define I2C_DRIVERID_ARP        902    /* SMBus ARP Client              */

include/sound/ak4xxx-adda.h

@@ -43,6 +43,7 @@ struct snd_ak4xxx_ops {
 struct snd_akm4xxx_dac_channel {
 	char *name;		/* mixer volume name */
 	unsigned int num_channels;
+	char *switch_name;		/* mixer switch*/
 };
 
 /* ADC labels and channels */

include/sound/cs46xx.h

@@ -1723,6 +1723,10 @@ struct snd_cs46xx {
 	struct snd_cs46xx_pcm *playback_pcm;
 	unsigned int play_ctl;
 #endif
+
+#ifdef CONFIG_PM
+	u32 *saved_regs;
+#endif
 };
 
 int snd_cs46xx_create(struct snd_card *card,

include/sound/cs46xx_dsp_spos.h

@@ -107,6 +107,7 @@ struct dsp_scb_descriptor {
 	char scb_name[DSP_MAX_SCB_NAME];
 	u32 address;
 	int index;
+	u32 *data;
 
 	struct dsp_scb_descriptor * sub_list_ptr;
 	struct dsp_scb_descriptor * next_scb_ptr;
@@ -127,6 +128,7 @@ struct dsp_task_descriptor {
 	int size;
 	u32 address;
 	int index;
+	u32 *data;
 };
 
 struct dsp_pcm_channel_descriptor {

include/sound/emu10k1.h

@@ -1120,6 +1120,16 @@
 /************************************************************************************************/
 /* EMU1010m HANA Destinations									*/
 /************************************************************************************************/
+/* 32-bit destinations of signal in the Hana FPGA. Destinations are either
+ * physical outputs of Hana, or outputs going to Alice2 (audigy) for capture
+ * - 16 x EMU_DST_ALICE2_EMU32_X.
+ */
+/* EMU32 = 32-bit serial channel between Alice2 (audigy) and Hana (FPGA) */
+/* EMU_DST_ALICE2_EMU32_X - data channels from Hana to Alice2 used for capture.
+ * Which data is fed into a EMU_DST_ALICE2_EMU32_X channel in Hana depends on
+ * setup of mixer control for each destination - see emumixer.c -
+ * snd_emu1010_output_enum_ctls[], snd_emu1010_input_enum_ctls[]
+ */
 #define EMU_DST_ALICE2_EMU32_0	0x000f	/* 16 EMU32 channels to Alice2 +0 to +0xf */
 #define EMU_DST_ALICE2_EMU32_1	0x0000	/* 16 EMU32 channels to Alice2 +0 to +0xf */
 #define EMU_DST_ALICE2_EMU32_2	0x0001	/* 16 EMU32 channels to Alice2 +0 to +0xf */
@@ -1199,6 +1209,12 @@
 /************************************************************************************************/
 /* EMU1010m HANA Sources									*/
 /************************************************************************************************/
+/* 32-bit sources of signal in the Hana FPGA. The sources are routed to
+ * destinations using mixer control for each destination - see emumixer.c
+ * Sources are either physical inputs of FPGA,
+ * or outputs from Alice (audigy) - 16 x EMU_SRC_ALICE_EMU32A +
+ * 16 x EMU_SRC_ALICE_EMU32B
+ */
 #define EMU_SRC_SILENCE		0x0000	/* Silence */
 #define EMU_SRC_DOCK_MIC_A1	0x0100	/* Audio Dock Mic A, 1st or 48kHz only */
 #define EMU_SRC_DOCK_MIC_A2	0x0101	/* Audio Dock Mic A, 2nd or 96kHz */

include/sound/sb.h

@@ -38,6 +38,7 @@ enum sb_hw_type {
 	SB_HW_ALS100,		/* Avance Logic ALS100 chip */
 	SB_HW_ALS4000,		/* Avance Logic ALS4000 chip */
 	SB_HW_DT019X,		/* Diamond Tech. DT-019X / Avance Logic ALS-007 */
+	SB_HW_CS5530,		/* Cyrix/NatSemi 5530 VSA1 */
 };
 
 #define SB_OPEN_PCM			0x01

include/sound/version.h

 /* include/version.h.  Generated by alsa/ksync script.  */
 #define CONFIG_SND_VERSION "1.0.14"
-#define CONFIG_SND_DATE " (Thu May 31 09:03:25 2007 UTC)"
+#define CONFIG_SND_DATE " (Fri Jul 20 09:12:58 2007 UTC)"

include/sound/wavefront_fx.h

-#ifndef __SOUND_WAVEFRONT_FX_H
-#define __SOUND_WAVEFRONT_FX_H
-
-extern int  snd_wavefront_fx_detect (snd_wavefront_t *);
-extern void snd_wavefront_fx_ioctl  (snd_synth_t *sdev, 
-				     unsigned int cmd, 
-				     unsigned long arg);
-
-#endif  __SOUND_WAVEFRONT_FX_H

sound/Kconfig

@@ -65,6 +65,8 @@ source "sound/arm/Kconfig"
 
 source "sound/mips/Kconfig"
 
+source "sound/sh/Kconfig"
+
 # the following will depend on the order of config.
 # here assuming USB is defined before ALSA
 source "sound/usb/Kconfig"

sound/Makefile

@@ -5,7 +5,7 @@ obj-$(CONFIG_SOUND) += soundcore.o
 obj-$(CONFIG_SOUND_PRIME) += sound_firmware.o
 obj-$(CONFIG_SOUND_PRIME) += oss/
 obj-$(CONFIG_DMASOUND) += oss/
-obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ soc/
+obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ sparc/ 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

@@ -661,7 +661,7 @@ static struct transfer_info onyx_transfers[] = {
 		.tag = 2,
 	},
 #ifdef SNDRV_PCM_FMTBIT_COMPRESSED_16BE
-Once alsa gets supports for this kind of thing we can add it...
+	/* Once alsa gets supports for this kind of thing we can add it... */
 	{
 		/* digital compressed output */
 		.formats =  SNDRV_PCM_FMTBIT_COMPRESSED_16BE,
@@ -713,7 +713,7 @@ static int onyx_prepare(struct codec_info_item *cii,
 	if (substream->runtime->format == SNDRV_PCM_FMTBIT_COMPRESSED_16BE) {
 		/* mute and lock analog output */
 		onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &v);
-		if (onyx_write_register(onyx
+		if (onyx_write_register(onyx,
 					ONYX_REG_DAC_CONTROL,
 					v | ONYX_MUTE_RIGHT | ONYX_MUTE_LEFT))
 			goto out_unlock;

sound/core/pcm_native.c

@@ -1487,7 +1487,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream)
 
 	snd_pcm_stream_lock_irq(substream);
 	/* resume pause */
-	if (runtime->status->state == SNDRV_PCM_STATE_PAUSED)
+	if (substream->runtime->status->state == SNDRV_PCM_STATE_PAUSED)
 		snd_pcm_pause(substream, 0);
 
 	/* pre-start/stop - all running streams are changed to DRAINING state */

sound/core/seq/seq_instr.c

@@ -109,7 +109,7 @@ void snd_seq_instr_list_free(struct snd_seq_kinstr_list **list_ptr)
 			spin_lock_irqsave(&list->lock, flags);
 			while (instr->use) {
 				spin_unlock_irqrestore(&list->lock, flags);
-				schedule_timeout_interruptible(1);
+				schedule_timeout(1);
 				spin_lock_irqsave(&list->lock, flags);
 			}				
 			spin_unlock_irqrestore(&list->lock, flags);
@@ -199,7 +199,7 @@ int snd_seq_instr_list_free_cond(struct snd_seq_kinstr_list *list,
 			instr = flist;
 			flist = instr->next;
 			while (instr->use)
-				schedule_timeout_interruptible(1);
+				schedule_timeout(1);
 			if (snd_seq_instr_free(instr, atomic)<0)
 				snd_printk(KERN_WARNING "instrument free problem\n");
 			instr = next;
@@ -555,7 +555,7 @@ static int instr_free(struct snd_seq_kinstr_ops *ops,
 					   SNDRV_SEQ_INSTR_NOTIFY_REMOVE);
 		while (instr->use) {
 			spin_unlock_irqrestore(&list->lock, flags);
-			schedule_timeout_interruptible(1);
+			schedule_timeout(1);
 			spin_lock_irqsave(&list->lock, flags);
 		}				
 		spin_unlock_irqrestore(&list->lock, flags);

sound/core/timer.c

@@ -1549,9 +1549,11 @@ static int snd_timer_user_info(struct file *file,
 	int err = 0;
 
 	tu = file->private_data;
-	snd_assert(tu->timeri != NULL, return -ENXIO);
+	if (!tu->timeri)
+		return -EBADFD;
 	t = tu->timeri->timer;
-	snd_assert(t != NULL, return -ENXIO);
+	if (!t)
+		return -EBADFD;
 
 	info = kzalloc(sizeof(*info), GFP_KERNEL);
 	if (! info)
@@ -1579,9 +1581,11 @@ static int snd_timer_user_params(struct file *file,
 	int err;
 
 	tu = file->private_data;
-	snd_assert(tu->timeri != NULL, return -ENXIO);
+	if (!tu->timeri)
+		return -EBADFD;
 	t = tu->timeri->timer;
-	snd_assert(t != NULL, return -ENXIO);
+	if (!t)
+		return -EBADFD;
 	if (copy_from_user(&params, _params, sizeof(params)))
 		return -EFAULT;
 	if (!(t->hw.flags & SNDRV_TIMER_HW_SLAVE) && params.ticks < 1) {
@@ -1675,7 +1679,8 @@ static int snd_timer_user_status(struct file *file,
 	struct snd_timer_status status;
 
 	tu = file->private_data;
-	snd_assert(tu->timeri != NULL, return -ENXIO);
+	if (!tu->timeri)
+		return -EBADFD;
 	memset(&status, 0, sizeof(status));
 	status.tstamp = tu->tstamp;
 	status.resolution = snd_timer_resolution(tu->timeri);
@@ -1695,7 +1700,8 @@ static int snd_timer_user_start(struct file *file)
 	struct snd_timer_user *tu;
 
 	tu = file->private_data;
-	snd_assert(tu->timeri != NULL, return -ENXIO);
+	if (!tu->timeri)
+		return -EBADFD;
 	snd_timer_stop(tu->timeri);
 	tu->timeri->lost = 0;
 	tu->last_resolution = 0;
@@ -1708,7 +1714,8 @@ static int snd_timer_user_stop(struct file *file)
 	struct snd_timer_user *tu;
 
 	tu = file->private_data;
-	snd_assert(tu->timeri != NULL, return -ENXIO);
+	if (!tu->timeri)
+		return -EBADFD;
 	return (err = snd_timer_stop(tu->timeri)) < 0 ? err : 0;
 }
 
@@ -1718,7 +1725,8 @@ static int snd_timer_user_continue(struct file *file)
 	struct snd_timer_user *tu;
 
 	tu = file->private_data;
-	snd_assert(tu->timeri != NULL, return -ENXIO);
+	if (!tu->timeri)
+		return -EBADFD;
 	tu->timeri->lost = 0;
 	return (err = snd_timer_continue(tu->timeri)) < 0 ? err : 0;
 }
@@ -1729,7 +1737,8 @@ static int snd_timer_user_pause(struct file *file)
 	struct snd_timer_user *tu;
 
 	tu = file->private_data;
-	snd_assert(tu->timeri != NULL, return -ENXIO);
+	if (!tu->timeri)
+		return -EBADFD;
 	return (err = snd_timer_pause(tu->timeri)) < 0 ? err : 0;
 }
 

sound/drivers/dummy.c

@@ -659,7 +659,7 @@ static struct platform_driver snd_dummy_driver = {
 	},
 };
 
-static void __init_or_module snd_dummy_unregister_all(void)
+static void snd_dummy_unregister_all(void)
 {
 	int i;
 

sound/drivers/mpu401/mpu401.c

@@ -228,7 +228,7 @@ static struct pnp_driver snd_mpu401_pnp_driver = {
 static struct pnp_driver snd_mpu401_pnp_driver;
 #endif
 
-static void __init_or_module snd_mpu401_unregister_all(void)
+static void snd_mpu401_unregister_all(void)
 {
 	int i;