---

yaml
---
r: 353472
b: refs/heads/master
c: 83048ea
h: refs/heads/master
v: v3

[refs]

@@ -1,2 +1,2 @@
 ---
-refs/heads/master: 699ba4546cfe3f4eee73aff4f307bf16362f9232
+refs/heads/master: 83048ea7b1d11f5e560dea53873fb51d860202eb

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

@@ -871,9 +871,8 @@
       <para>
       This function itself doesn't allocate the data space. The data
       must be allocated manually beforehand, and its pointer is passed
-      as the argument. This pointer is used as the
-      (<parameter>chip</parameter> identifier in the above example)
-      for the instance. 
+      as the argument. This pointer (<parameter>chip</parameter> in the
+      above example) is used as the identifier for the instance.
       </para>
 
       <para>
@@ -2304,7 +2303,7 @@ struct _snd_pcm_runtime {
         <constant>SNDRV_PCM_INFO_XXX</constant>. Here, at least, you
         have to specify whether the mmap is supported and which
         interleaved format is supported.
-        When the is supported, add the
+        When the hardware supports mmap, add the
         <constant>SNDRV_PCM_INFO_MMAP</constant> flag here. When the
         hardware supports the interleaved or the non-interleaved
         formats, <constant>SNDRV_PCM_INFO_INTERLEAVED</constant> or
@@ -2898,7 +2897,7 @@ struct _snd_pcm_runtime {
 
         <para>
           When the pcm supports the pause operation (given in the info
-        field of the hardware table), the <constant>PAUSE_PUSE</constant>
+        field of the hardware table), the <constant>PAUSE_PUSH</constant>
         and <constant>PAUSE_RELEASE</constant> commands must be
         handled here, too. The former is the command to pause the pcm,
         and the latter to restart the pcm again. 
@@ -3085,7 +3084,7 @@ struct _snd_pcm_runtime {
       <section id="pcm-interface-interrupt-handler-timer">
         <title>High frequency timer interrupts</title>
         <para>
-	This happense when the hardware doesn't generate interrupts
+	This happens when the hardware doesn't generate interrupts
         at the period boundary but issues timer interrupts at a fixed
         timer rate (e.g. es1968 or ymfpci drivers). 
         In this case, you need to check the current hardware
@@ -3251,18 +3250,19 @@ struct _snd_pcm_runtime {
 	  <title>Example of Hardware Constraints for Channels</title>
 	  <programlisting>
 <![CDATA[
-  static int hw_rule_format_by_channels(struct snd_pcm_hw_params *params,
+  static int hw_rule_channels_by_format(struct snd_pcm_hw_params *params,
                                         struct snd_pcm_hw_rule *rule)
   {
           struct snd_interval *c = hw_param_interval(params,
-                SNDRV_PCM_HW_PARAM_CHANNELS);
+                        SNDRV_PCM_HW_PARAM_CHANNELS);
           struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
-          struct snd_mask fmt;
+          struct snd_interval ch;
 
-          snd_mask_any(&fmt);    /* Init the struct */
-          if (c->min < 2) {
-                  fmt.bits[0] &= SNDRV_PCM_FMTBIT_S16_LE;
-                  return snd_mask_refine(f, &fmt);
+          snd_interval_any(&ch);
+          if (f->bits[0] == SNDRV_PCM_FMTBIT_S16_LE) {
+                  ch.min = ch.max = 1;
+                  ch.integer = 1;
+                  return snd_interval_refine(c, &ch);
           }
           return 0;
   }
@@ -3278,35 +3278,35 @@ struct _snd_pcm_runtime {
 	 <programlisting>
 <![CDATA[
   snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
-                      hw_rule_channels_by_format, 0, SNDRV_PCM_HW_PARAM_FORMAT,
-                      -1);
+                      hw_rule_channels_by_format, NULL,
+                      SNDRV_PCM_HW_PARAM_FORMAT, -1);
 ]]>
           </programlisting>
         </informalexample>
       </para>
 
       <para>
-        The rule function is called when an application sets the number of
-        channels. But an application can set the format before the number of
-        channels. Thus you also need to define the inverse rule:
+        The rule function is called when an application sets the PCM
+	format, and it refines the number of channels accordingly.
+        But an application may set the number of channels before
+	setting the format. Thus you also need to define the inverse rule:
 
        <example>
-	 <title>Example of Hardware Constraints for Channels</title>
+	 <title>Example of Hardware Constraints for Formats</title>
 	 <programlisting>
 <![CDATA[
-  static int hw_rule_channels_by_format(struct snd_pcm_hw_params *params,
+  static int hw_rule_format_by_channels(struct snd_pcm_hw_params *params,
                                         struct snd_pcm_hw_rule *rule)
   {
           struct snd_interval *c = hw_param_interval(params,
-                        SNDRV_PCM_HW_PARAM_CHANNELS);
+                SNDRV_PCM_HW_PARAM_CHANNELS);
           struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
-          struct snd_interval ch;
+          struct snd_mask fmt;
 
-          snd_interval_any(&ch);
-          if (f->bits[0] == SNDRV_PCM_FMTBIT_S16_LE) {
-                  ch.min = ch.max = 1;
-                  ch.integer = 1;
-                  return snd_interval_refine(c, &ch);
+          snd_mask_any(&fmt);    /* Init the struct */
+          if (c->min < 2) {
+                  fmt.bits[0] &= SNDRV_PCM_FMTBIT_S16_LE;
+                  return snd_mask_refine(f, &fmt);
           }
           return 0;
   }
@@ -3321,8 +3321,8 @@ struct _snd_pcm_runtime {
 	 <programlisting>
 <![CDATA[
   snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT,
-                      hw_rule_format_by_channels, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
-                      -1);
+                      hw_rule_format_by_channels, NULL,
+                      SNDRV_PCM_HW_PARAM_CHANNELS, -1);
 ]]>
           </programlisting>
         </informalexample>

trunk/Documentation/sound/alsa/ALSA-Configuration.txt

@@ -890,8 +890,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     enable_msi	- Enable Message Signaled Interrupt (MSI) (default = off)
     power_save	- Automatic power-saving timeout (in second, 0 =
 		disable)
-    power_save_controller - Reset HD-audio controller in power-saving mode
-		(default = on)
+    power_save_controller - Support runtime D3 of HD-audio controller
+		(-1 = on for supported chip (default), false = off,
+		 true = force to on even for unsupported hardware)
     align_buffer_size - Force rounding of buffer/period sizes to multiples
     		      of 128 bytes. This is more efficient in terms of memory
 		      access but isn't required by the HDA spec and prevents

trunk/Documentation/sound/alsa/HD-Audio.txt

@@ -176,14 +176,14 @@ support the automatic probing (yet as of 2.6.28).  And, BIOS is often,
 yes, pretty often broken.  It sets up wrong values and screws up the
 driver.
 
-The preset model is provided basically to overcome such a situation.
-When the matching preset model is found in the white-list, the driver
-assumes the static configuration of that preset and builds the mixer
-elements and PCM streams based on the static information.  Thus, if
-you have a newer machine with a slightly different PCI SSID from the
-existing one, you may have a good chance to re-use the same model.
-You can pass the `model` option to specify the preset model instead of
-PCI SSID look-up.
+The preset model (or recently called as "fix-up") is provided
+basically to overcome such a situation.  When the matching preset
+model is found in the white-list, the driver assumes the static
+configuration of that preset with the correct pin setup, etc.
+Thus, if you have a newer machine with a slightly different PCI SSID
+(or codec SSID) from the existing one, you may have a good chance to
+re-use the same model.  You can pass the `model` option to specify the
+preset model instead of PCI (and codec-) SSID look-up.
 
 What `model` option values are available depends on the codec chip.
 Check your codec chip from the codec proc file (see "Codec Proc-File"
@@ -199,17 +199,12 @@ non-working HD-audio hardware is to check HD-audio codec and several
 different `model` option values.  If you have any luck, some of them
 might suit with your device well.
 
-Some codecs such as ALC880 have a special model option `model=test`.
-This configures the driver to provide as many mixer controls as
-possible for every single pin feature except for the unsolicited
-events (and maybe some other specials).  Adjust each mixer element and
-try the I/O in the way of trial-and-error until figuring out the whole
-I/O pin mappings.
+There are a few special model option values:
+- when 'nofixup' is passed, the device-specific fixups in the codec
+  parser are skipped.
+- when `generic` is passed, the codec-specific parser is skipped and
+  only the generic parser is used.
 
-Note that `model=generic` has a special meaning.  It means to use the
-generic parser regardless of the codec.  Usually the codec-specific
-parser is much better than the generic parser (as now).  Thus this
-option is more about the debugging purpose.
 
 Speaker and Headphone Output
 ~~~~~~~~~~~~~~~~~~~~~~~~~~~~
@@ -387,9 +382,8 @@ init_verbs::
   (separated with a space).
 hints::
   Shows / stores hint strings for codec parsers for any use.
-  Its format is `key = value`.  For example, passing `hp_detect = yes`
-  to IDT/STAC codec parser will result in the disablement of the
-  headphone detection.
+  Its format is `key = value`.  For example, passing `jack_detect = no`
+  will disable the jack detection of the machine completely.
 init_pin_configs::
   Shows the initial pin default config values set by BIOS.
 driver_pin_configs::
@@ -421,6 +415,61 @@ re-configure based on that state, run like below:
 ------------------------------------------------------------------------
 
 
+Hint Strings
+~~~~~~~~~~~~
+The codec parser have several switches and adjustment knobs for
+matching better with the actual codec or device behavior.  Many of
+them can be adjusted dynamically via "hints" strings as mentioned in
+the section above.  For example, by passing `jack_detect = no` string
+via sysfs or a patch file, you can disable the jack detection, thus
+the codec parser will skip the features like auto-mute or mic
+auto-switch.  As a boolean value, either `yes`, `no`, `true`, `false`,
+`1` or `0` can be passed.
+
+The generic parser supports the following hints:
+
+- jack_detect (bool): specify whether the jack detection is available
+  at all on this machine; default true
+- inv_jack_detect (bool): indicates that the jack detection logic is
+  inverted
+- trigger_sense (bool): indicates that the jack detection needs the
+  explicit call of AC_VERB_SET_PIN_SENSE verb
+- inv_eapd (bool): indicates that the EAPD is implemented in the
+  inverted logic
+- pcm_format_first (bool): sets the PCM format before the stream tag
+  and channel ID
+- sticky_stream (bool): keep the PCM format, stream tag and ID as long
+  as possible; default true
+- spdif_status_reset (bool): reset the SPDIF status bits at each time
+  the SPDIF stream is set up
+-  pin_amp_workaround (bool): the output pin may have multiple amp
+  values
+- single_adc_amp (bool): ADCs can have only single input amps
+- auto_mute (bool): enable/disable the headphone auto-mute feature;
+  default true
+- auto_mic (bool): enable/disable the mic auto-switch feature; default
+  true
+- line_in_auto_switch (bool): enable/disable the line-in auto-switch
+  feature; default false
+- need_dac_fix (bool): limits the DACs depending on the channel count
+- primary_hp (bool): probe headphone jacks as the primary outputs;
+  default true
+- multi_cap_vol (bool): provide multiple capture volumes
+- inv_dmic_split (bool): provide split internal mic volume/switch for
+  phase-inverted digital mics
+- indep_hp (bool): provide the independent headphone PCM stream and
+  the corresponding mixer control, if available
+- add_stereo_mix_input (bool): add the stereo mix (analog-loopback
+  mix) to the input mux if available
+- add_out_jack_modes (bool): add "xxx Jack Mode" enum controls to each
+  output jack for allowing to change the headphone amp capability
+- add_in_jack_modes (bool): add "xxx Jack Mode" enum controls to each
+  input jack for allowing to change the mic bias vref
+- power_down_unused (bool): power down the unused widgets
+- mixer_nid (int): specifies the widget NID of the analog-loopback
+  mixer
+
+
 Early Patching
 ~~~~~~~~~~~~~~
 When CONFIG_SND_HDA_PATCH_LOADER=y is set, you can pass a "patch" as a
@@ -445,7 +494,7 @@ A patch file is a plain text file which looks like below:
   0x20 0x400 0xff
 
   [hint]
-  hp_detect = yes
+  jack_detect = no
 ------------------------------------------------------------------------
 
 The file needs to have a line `[codec]`.  The next line should contain
@@ -531,6 +580,13 @@ cable is unplugged.  Thus, if you hear noises, suspect first the
 power-saving.  See /sys/module/snd_hda_intel/parameters/power_save to
 check the current value.  If it's non-zero, the feature is turned on.
 
+The recent kernel supports the runtime PM for the HD-audio controller
+chip, too.  It means that the HD-audio controller is also powered up /
+down dynamically.  The feature is enabled only for certain controller
+chips like Intel LynxPoint.  You can enable/disable this feature
+forcibly by setting `power_save_controller` option, which is also
+available at /sys/module/snd_hda_intel/parameters directory.
+
 
 Tracepoints
 ~~~~~~~~~~~
@@ -587,8 +643,9 @@ The latest development codes for HD-audio are found on sound git tree:
 - git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git
 
 The master branch or for-next branches can be used as the main
-development branches in general while the HD-audio specific patches
-are committed in topic/hda branch.
+development branches in general while the development for the current
+and next kernels are found in for-linus and for-next branches,
+respectively.
 
 If you are using the latest Linus tree, it'd be better to pull the
 above GIT tree onto it.  If you are using the older kernels, an easy
@@ -699,7 +756,11 @@ won't be always updated.  For example, the volume values are usually
 cached in the driver, and thus changing the widget amp value directly
 via hda-verb won't change the mixer value.
 
-The hda-verb program is found in the ftp directory:
+The hda-verb program is included now in alsa-tools:
+
+- git://git.alsa-project.org/alsa-tools.git
+
+Also, the old stand-alone package is found in the ftp directory:
 
 - ftp://ftp.suse.com/pub/people/tiwai/misc/
 
@@ -777,3 +838,18 @@ A git repository is available:
 
 See README file in the tarball for more details about hda-emu
 program.
+
+
+hda-jack-retask
+~~~~~~~~~~~~~~~
+hda-jack-retask is a user-friendly GUI program to manipulate the
+HD-audio pin control for jack retasking.  If you have a problem about
+the jack assignment, try this program and check whether you can get
+useful results.  Once when you figure out the proper pin assignment,
+it can be fixed either in the driver code statically or via passing a
+firmware patch file (see "Early Patching" section).
+
+The program is included in alsa-tools now:
+
+- git://git.alsa-project.org/alsa-tools.git
+

trunk/include/sound/core.h

@@ -394,8 +394,11 @@ void __snd_printk(unsigned int level, const char *file, int line,
 
 #else /* !CONFIG_SND_DEBUG */
 
-#define snd_printd(fmt, args...)	do { } while (0)
-#define _snd_printd(level, fmt, args...) do { } while (0)
+__printf(1, 2)
+static inline void snd_printd(const char *format, ...) {}
+__printf(2, 3)
+static inline void _snd_printd(int level, const char *format, ...) {}
+
 #define snd_BUG()			do { } while (0)
 static inline int __snd_bug_on(int cond)
 {
@@ -416,7 +419,8 @@ static inline int __snd_bug_on(int cond)
 #define snd_printdd(format, args...) \
 	__snd_printk(2, __FILE__, __LINE__, format, ##args)
 #else
-#define snd_printdd(format, args...)	do { } while (0)
+__printf(1, 2)
+static inline void snd_printdd(const char *format, ...) {}
 #endif
 
 
@@ -454,13 +458,15 @@ struct snd_pci_quirk {
 #define SND_PCI_QUIRK_MASK(vend, mask, dev, xname, val)			\
 	{_SND_PCI_QUIRK_ID_MASK(vend, mask, dev),			\
 			.value = (val), .name = (xname)}
+#define snd_pci_quirk_name(q)	((q)->name)
 #else
 #define SND_PCI_QUIRK(vend,dev,xname,val) \
 	{_SND_PCI_QUIRK_ID(vend, dev), .value = (val)}
 #define SND_PCI_QUIRK_MASK(vend, mask, dev, xname, val)			\
 	{_SND_PCI_QUIRK_ID_MASK(vend, mask, dev), .value = (val)}
 #define SND_PCI_QUIRK_VENDOR(vend, xname, val)			\
 	{_SND_PCI_QUIRK_ID_MASK(vend, 0, 0), .value = (val)}
+#define snd_pci_quirk_name(q)	""
 #endif
 
 const struct snd_pci_quirk *

trunk/include/sound/memalloc.h

@@ -37,7 +37,7 @@ struct snd_dma_device {
 #ifndef snd_dma_pci_data
 #define snd_dma_pci_data(pci)	(&(pci)->dev)
 #define snd_dma_isa_data()	NULL
-#define snd_dma_continuous_data(x)	((struct device *)(unsigned long)(x))
+#define snd_dma_continuous_data(x)	((struct device *)(__force unsigned long)(x))
 #endif