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-rw-r--r--Documentation/sound/alsa/ALSA-Configuration.txt5
-rw-r--r--Documentation/sound/alsa/HD-Audio-Models.txt2
-rw-r--r--Documentation/sound/alsa/HD-Audio.txt126
-rw-r--r--Documentation/sound/alsa/compress_offload.txt46
4 files changed, 151 insertions, 28 deletions
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index b9cfd339a6fa..ce6581c8ca26 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/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
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index 16dfe57f1731..bb8b0dc532b8 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -53,7 +53,7 @@ ALC882/883/885/888/889
acer-aspire-8930g Acer Aspire 8330G/6935G
acer-aspire Acer Aspire others
inv-dmic Inverted internal mic workaround
- no-primary-hp VAIO Z workaround (for fixed speaker DAC)
+ no-primary-hp VAIO Z/VGC-LN51JGB workaround (for fixed speaker DAC)
ALC861/660
==========
diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt
index 7813c06a5c71..d4faa63ff352 100644
--- a/Documentation/sound/alsa/HD-Audio.txt
+++ b/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
+
diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt
index 90e9b3a11abc..0bcc55155911 100644
--- a/Documentation/sound/alsa/compress_offload.txt
+++ b/Documentation/sound/alsa/compress_offload.txt
@@ -145,6 +145,52 @@ Modifications include:
- Addition of encoding options when required (derived from OpenMAX IL)
- Addition of rateControlSupported (missing in OpenMAX AL)
+Gapless Playback
+================
+When playing thru an album, the decoders have the ability to skip the encoder
+delay and padding and directly move from one track content to another. The end
+user can perceive this as gapless playback as we dont have silence while
+switching from one track to another
+
+Also, there might be low-intensity noises due to encoding. Perfect gapless is
+difficult to reach with all types of compressed data, but works fine with most
+music content. The decoder needs to know the encoder delay and encoder padding.
+So we need to pass this to DSP. This metadata is extracted from ID3/MP4 headers
+and are not present by default in the bitstream, hence the need for a new
+interface to pass this information to the DSP. Also DSP and userspace needs to
+switch from one track to another and start using data for second track.
+
+The main additions are:
+
+- set_metadata
+This routine sets the encoder delay and encoder padding. This can be used by
+decoder to strip the silence. This needs to be set before the data in the track
+is written.
+
+- set_next_track
+This routine tells DSP that metadata and write operation sent after this would
+correspond to subsequent track
+
+- partial drain
+This is called when end of file is reached. The userspace can inform DSP that
+EOF is reached and now DSP can start skipping padding delay. Also next write
+data would belong to next track
+
+Sequence flow for gapless would be:
+- Open
+- Get caps / codec caps
+- Set params
+- Set metadata of the first track
+- Fill data of the first track
+- Trigger start
+- User-space finished sending all,
+- Indicaite next track data by sending set_next_track
+- Set metadata of the next track
+- then call partial_drain to flush most of buffer in DSP
+- Fill data of the next track
+- DSP switches to second track
+(note: order for partial_drain and write for next track can be reversed as well)
+
Not supported:
- Support for VoIP/circuit-switched calls is not the target of this