diff options
Diffstat (limited to 'Documentation/sound/alsa')
| -rw-r--r-- | Documentation/sound/alsa/ALSA-Configuration.txt | 5 | ||||
| -rw-r--r-- | Documentation/sound/alsa/HD-Audio-Models.txt | 2 | ||||
| -rw-r--r-- | Documentation/sound/alsa/HD-Audio.txt | 126 | ||||
| -rw-r--r-- | Documentation/sound/alsa/compress_offload.txt | 46 |
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 |