---

yaml --- r: 47305 b: refs/heads/master c: 2fd592e h: refs/heads/master i: 47303: 553ff69 v: v3
git-mirror · Feb 9, 2007 · f4859ac · f4859ac
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diff --git a/[refs] b/[refs]
@@ -1,2 +1,2 @@
 ---
-refs/heads/master: 0a9b0db19262dbb09f3a34195e68cafd5dc3fa10
+refs/heads/master: 2fd592e45b9c89d69e126f172d0f991e2af955e5
diff --git a/trunk/Documentation/crypto/api-intro.txt b/trunk/Documentation/crypto/api-intro.txt
@@ -193,6 +193,7 @@ Original developers of the crypto algorithms:
   Kartikey Mahendra Bhatt (CAST6)
   Jon Oberheide (ARC4)
   Jouni Malinen (Michael MIC)
+  NTT(Nippon Telegraph and Telephone Corporation) (Camellia)
 
 SHA1 algorithm contributors:
   Jean-Francois Dive
@@ -246,6 +247,9 @@ Tiger algorithm contributors:
 VIA PadLock contributors:
   Michal Ludvig
 
+Camellia algorithm contributors:
+  NTT(Nippon Telegraph and Telephone Corporation) (Camellia)
+
 Generic scatterwalk code by Adam J. Richter <adam@yggdrasil.com>
 
 Please send any credits updates or corrections to:

diff --git a/trunk/Documentation/feature-removal-schedule.txt b/trunk/Documentation/feature-removal-schedule.txt
@@ -186,18 +186,6 @@ Who:	Greg Kroah-Hartman <gregkh@suse.de>
 
 ---------------------------
 
-What:	find_trylock_page
-When:	January 2007
-Why:	The interface no longer has any callers left in the kernel. It
-	is an odd interface (compared with other find_*_page functions), in
-	that it does not take a refcount to the page, only the page lock.
-	It should be replaced with find_get_page or find_lock_page if possible.
-	This feature removal can be reevaluated if users of the interface
-	cannot cleanly use something else.
-Who:	Nick Piggin <npiggin@suse.de>
-
----------------------------
-
 What:	Interrupt only SA_* flags
 When:	Januar 2007
 Why:	The interrupt related SA_* flags are replaced by IRQF_* to move them

diff --git a/trunk/Documentation/sound/alsa/ALSA-Configuration.txt b/trunk/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -242,6 +242,12 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     ac97_clock		- AC'97 clock (default = 48000)
     ac97_quirk		- AC'97 workaround for strange hardware
 			  See "AC97 Quirk Option" section below.
+    ac97_codec		- Workaround to specify which AC'97 codec 
+			  instead of probing.  If this works for you
+			  file a bug with your `lspci -vn` output.
+			  -2  -- Force probing.
+			  -1  -- Default behavior.
+			  0-2 -- Use the specified codec.
     spdif_aclink	- S/PDIF transfer over AC-link (default = 1)
 
     This module supports one card and autoprobe.
@@ -779,6 +785,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  asus-dig	ASUS with SPDIF out
 	  asus-dig2	ASUS with SPDIF out (using GPIO2)
 	  uniwill	3-jack
+	  fujitsu	Fujitsu Laptops (Pi1536)
 	  F1734		2-jack
 	  lg		LG laptop (m1 express dual)
 	  lg-lw		LG LW20/LW25 laptop
@@ -800,14 +807,18 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	ALC262
 	  fujitsu	Fujitsu Laptop
 	  hp-bpc	HP xw4400/6400/8400/9400 laptops
+	  hp-bpc-d7000	HP BPC D7000
 	  benq		Benq ED8
+	  hippo		Hippo (ATI) with jack detection, Sony UX-90s
+	  hippo_1	Hippo (Benq) with jack detection
 	  basic		fixed pin assignment w/o SPDIF
 	  auto		auto-config reading BIOS (default)
 
 	ALC882/885
 	  3stack-dig	3-jack with SPDIF I/O
-	  6stck-dig	6-jack digital with SPDIF I/O
+	  6stack-dig	6-jack digital with SPDIF I/O
 	  arima		Arima W820Di1
+	  macpro	MacPro support
 	  auto		auto-config reading BIOS (default)
 
 	ALC883/888
@@ -817,6 +828,10 @@ 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)
+	  medion	Medion Laptops
+	  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)
 	  auto		auto-config reading BIOS (default)
 
 	ALC861/660
@@ -825,6 +840,16 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  6stack-dig	6-jack with SPDIF I/O
 	  3stack-660	3-jack (for ALC660)
 	  uniwill-m31	Uniwill M31 laptop
+	  toshiba	Toshiba laptop support
+	  asus		Asus laptop support
+	  asus-laptop	ASUS F2/F3 laptops
+	  auto		auto-config reading BIOS (default)
+
+	ALC861VD/660VD
+	  3stack	3-jack
+	  3stack-dig	3-jack with SPDIF OUT
+	  6stack-dig	6-jack with SPDIF OUT
+	  3stack-660	3-jack (for ALC660VD)
 	  auto		auto-config reading BIOS (default)
 
 	CMI9880
@@ -845,6 +870,7 @@ 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)
+	  ultra		2-channel with EAPD (Samsung Ultra tablet PC)
 
 	AD1988
 	  6stack	6-jack
@@ -854,12 +880,31 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  laptop	3-jack with hp-jack automute
 	  laptop-dig	ditto with SPDIF
 	  auto		auto-config reading BIOS (default)
+
+	Conexant 5045
+	  laptop	Laptop config 
+	  test		for testing/debugging purpose, almost all controls
+			can be adjusted.  Appearing only when compiled with
+			$CONFIG_SND_DEBUG=y
+
+	Conexant 5047
+	  laptop	Basic Laptop config 
+	  laptop-hp	Laptop config for some HP models (subdevice 30A5)
+	  laptop-eapd	Laptop config with EAPD support
+	  test		for testing/debugging purpose, almost all controls
+			can be adjusted.  Appearing only when compiled with
+			$CONFIG_SND_DEBUG=y
 
 	STAC9200/9205/9220/9221/9254
 	  ref		Reference board
 	  3stack	D945 3stack
 	  5stack	D945 5stack + SPDIF
 
+	STAC9202/9250/9251
+	  ref		Reference board, base config
+	  m2-2		Some Gateway MX series laptops
+	  m6		Some Gateway NX series laptops
+
 	STAC9227/9228/9229/927x
 	  ref		Reference board
 	  3stack	D965 3stack
@@ -974,6 +1019,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     Module for Envy24HT (VT/ICE1724), Envy24PT (VT1720) based PCI sound cards.
 			* MidiMan M Audio Revolution 5.1
 			* MidiMan M Audio Revolution 7.1
+			* MidiMan M Audio Audiophile 192
 			* AMP Ltd AUDIO2000
 			* TerraTec Aureon 5.1 Sky
 			* TerraTec Aureon 7.1 Space
@@ -993,7 +1039,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     model       - Use the given board model, one of the following:
 		  revo51, revo71, amp2000, prodigy71, prodigy71lt,
-		  prodigy192, aureon51, aureon71, universe,
+		  prodigy192, aureon51, aureon71, universe, ap192,
 		  k8x800, phase22, phase28, ms300, av710
 
     This module supports multiple cards and autoprobe.
@@ -1049,6 +1095,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     buggy_semaphore - Enable workaround for hardwares with buggy
 		    semaphores (e.g. on some ASUS laptops)
 		    (default off)
+    spdif_aclink  - Use S/PDIF over AC-link instead of direct connection
+		    from the controller chip
+		    (0 = off, 1 = on, -1 = default)
 
     This module supports one chip and autoprobe.
 
@@ -1371,6 +1420,13 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     This module supports multiple cards.
 
+  Module snd-portman2x4
+  ---------------------
+
+    Module for Midiman Portman 2x4 parallel port MIDI interface
+
+    This module supports multiple cards.
+
   Module snd-powermac (on ppc only)
   ---------------------------------
 

diff --git a/trunk/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl b/trunk/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl
@@ -36,7 +36,7 @@
   </bookinfo>
 
   <chapter><title>Management of Cards and Devices</title>
-     <sect1><title>Card Managment</title>
+     <sect1><title>Card Management</title>
 !Esound/core/init.c
      </sect1>
      <sect1><title>Device Components</title>
@@ -59,7 +59,7 @@
      <sect1><title>PCM Format Helpers</title>
 !Esound/core/pcm_misc.c
      </sect1>
-     <sect1><title>PCM Memory Managment</title>
+     <sect1><title>PCM Memory Management</title>
 !Esound/core/pcm_memory.c
      </sect1>
   </chapter>

diff --git a/trunk/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl b/trunk/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
@@ -1360,8 +1360,7 @@
         <informalexample>
           <programlisting>
 <![CDATA[
-  static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id,
-                                          struct pt_regs *regs)
+  static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id)
   {
           struct mychip *chip = dev_id;
           ....
@@ -2127,7 +2126,7 @@
 	accessible via <constant>substream-&gt;runtime</constant>.
 	This runtime pointer holds the various information; it holds
 	the copy of hw_params and sw_params configurations, the buffer
-	pointers, mmap records, spinlocks, etc.  Almost everyhing you
+	pointers, mmap records, spinlocks, etc.  Almost everything you
 	need for controlling the PCM can be found there.
 	</para>
 
@@ -2340,7 +2339,7 @@ struct _snd_pcm_runtime {
 
 	<para>
 	  When the PCM substreams can be synchronized (typically,
-	synchorinized start/stop of a playback and a capture streams),
+	synchronized start/stop of a playback and a capture streams),
 	you can give <constant>SNDRV_PCM_INFO_SYNC_START</constant>,
 	too.  In this case, you'll need to check the linked-list of
 	PCM substreams in the trigger callback.  This will be
@@ -3062,8 +3061,7 @@ struct _snd_pcm_runtime {
 	    <title>Interrupt Handler Case #1</title>
             <programlisting>
 <![CDATA[
-  static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id,
-                                          struct pt_regs *regs)
+  static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id)
   {
           struct mychip *chip = dev_id;
           spin_lock(&chip->lock);
@@ -3106,8 +3104,7 @@ struct _snd_pcm_runtime {
 	    <title>Interrupt Handler Case #2</title>
             <programlisting>
 <![CDATA[
-  static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id,
-                                          struct pt_regs *regs)
+  static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id)
   {
           struct mychip *chip = dev_id;
           spin_lock(&chip->lock);
@@ -3247,7 +3244,7 @@ struct _snd_pcm_runtime {
         You can even define your own constraint rules.
         For example, let's suppose my_chip can manage a substream of 1 channel
         if and only if the format is S16_LE, otherwise it supports any format
-        specified in the <structname>snd_pcm_hardware</structname> stucture (or in any
+        specified in the <structname>snd_pcm_hardware</structname> structure (or in any
         other constraint_list). You can build a rule like this:
 
         <example>
@@ -3690,16 +3687,6 @@ struct _snd_pcm_runtime {
           </example>
         </para>
 
-        <para>
-          Here, the chip instance is retrieved via
-        <function>snd_kcontrol_chip()</function> macro.  This macro
-        just accesses to kcontrol-&gt;private_data. The
-        kcontrol-&gt;private_data field is 
-        given as the argument of <function>snd_ctl_new()</function>
-        (see the later subsection
-        <link linkend="control-interface-constructor"><citetitle>Constructor</citetitle></link>).
-        </para>
-
         <para>
 	The <structfield>value</structfield> field is depending on
         the type of control as well as on info callback.  For example,
@@ -3780,7 +3767,7 @@ struct _snd_pcm_runtime {
         <para>
 	Like <structfield>get</structfield> callback,
 	when the control has more than one elements,
-	all elemehts must be evaluated in this callback, too.
+	all elements must be evaluated in this callback, too.
         </para>
       </section>
 
@@ -5541,12 +5528,12 @@ struct _snd_pcm_runtime {
   #ifdef CONFIG_PM
   static int snd_my_suspend(struct pci_dev *pci, pm_message_t state)
   {
-          .... /* do things for suspsend */
+          .... /* do things for suspend */
           return 0;
   }
   static int snd_my_resume(struct pci_dev *pci)
   {
-          .... /* do things for suspsend */
+          .... /* do things for suspend */
           return 0;
   }
   #endif
@@ -6111,7 +6098,7 @@ struct _snd_pcm_runtime {
 <!-- ****************************************************** -->
 <!-- Acknowledgments  -->
 <!-- ****************************************************** -->
-  <chapter id="acknowledments">
+  <chapter id="acknowledgments">
     <title>Acknowledgments</title>
     <para>
       I would like to thank Phil Kerr for his help for improvement and

diff --git a/trunk/Documentation/sound/alsa/hda_codec.txt b/trunk/Documentation/sound/alsa/hda_codec.txt
@@ -277,11 +277,11 @@ Helper Functions
 snd_hda_get_codec_name() stores the codec name on the given string.
 
 snd_hda_check_board_config() can be used to obtain the configuration
-information matching with the device.  Define the table with struct
-hda_board_config entries (zero-terminated), and pass it to the
-function.  The function checks the modelname given as a module
-parameter, and PCI subsystem IDs.  If the matching entry is found, it
-returns the config field value.
+information matching with the device.  Define the model string table
+and the table with struct snd_pci_quirk entries (zero-terminated),
+and pass it to the function.  The function checks the modelname given
+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

diff --git a/trunk/Documentation/sound/alsa/soc/DAI.txt b/trunk/Documentation/sound/alsa/soc/DAI.txt
@@ -0,0 +1,56 @@
+ASoC currently supports the three main Digital Audio Interfaces (DAI) found on
+SoC controllers and portable audio CODECS today, namely AC97, I2S and PCM.
+
+
+AC97
+====
+
+  AC97 is a five wire interface commonly found on many PC sound cards. It is
+now also popular in many portable devices. This DAI has a reset line and time
+multiplexes its data on its SDATA_OUT (playback) and SDATA_IN (capture) lines.
+The bit clock (BCLK) is always driven by the CODEC (usually 12.288MHz) and the
+frame (FRAME) (usually 48kHz) is always driven by the controller. Each AC97
+frame is 21uS long and is divided into 13 time slots.
+
+The AC97 specification can be found at :-
+http://www.intel.com/design/chipsets/audio/ac97_r23.pdf
+
+
+I2S
+===
+
+ I2S is a common 4 wire DAI used in HiFi, STB and portable devices. The Tx and
+Rx lines are used for audio transmision, whilst the bit clock (BCLK) and
+left/right clock (LRC) synchronise the link. I2S is flexible in that either the
+controller or CODEC can drive (master) the BCLK and LRC clock lines. Bit clock
+usually varies depending on the sample rate and the master system clock
+(SYSCLK). LRCLK is the same as the sample rate. A few devices support separate
+ADC and DAC LRCLK's, this allows for similtanious capture and playback at
+different sample rates.
+
+I2S has several different operating modes:-
+
+ o I2S - MSB is transmitted on the falling edge of the first BCLK after LRC
+         transition.
+
+ o Left Justified - MSB is transmitted on transition of LRC.
+
+ o Right Justified - MSB is transmitted sample size BCLK's before LRC
+                     transition.
+
+PCM
+===
+
+PCM is another 4 wire interface, very similar to I2S, that can support a more
+flexible protocol. It has bit clock (BCLK) and sync (SYNC) lines that are used
+to synchronise the link whilst the Tx and Rx lines are used to transmit and
+receive the audio data. Bit clock usually varies depending on sample rate
+whilst sync runs at the sample rate. PCM also supports Time Division
+Multiplexing (TDM) in that several devices can use the bus similtaniuosly (This
+is sometimes referred to as network mode).
+
+Common PCM operating modes:-
+
+ o Mode A - MSB is transmitted on falling edge of first BCLK after FRAME/SYNC.
+
+ o Mode B - MSB is transmitted on rising edge of FRAME/SYNC.