diff options
Diffstat (limited to 'Documentation/sound')
| -rw-r--r-- | Documentation/sound/alsa/ALSA-Configuration.txt | 38 | ||||
| -rw-r--r-- | Documentation/sound/alsa/HD-Audio-Models.txt | 1 | ||||
| -rw-r--r-- | Documentation/sound/alsa/HD-Audio.txt | 2 | ||||
| -rw-r--r-- | Documentation/sound/alsa/Procfile.txt | 39 | ||||
| -rw-r--r-- | Documentation/sound/alsa/README.maya44 | 163 | ||||
| -rw-r--r-- | Documentation/sound/alsa/hda_codec.txt | 2 | ||||
| -rw-r--r-- | Documentation/sound/alsa/soc/dapm.txt | 1 |
7 files changed, 227 insertions, 19 deletions
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 414700b996a..f9d11140af9 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -460,6 +460,25 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. The power-management is supported. + Module snd-ctxfi + ---------------- + + Module for Creative Sound Blaster X-Fi boards (20k1 / 20k2 chips) + * Creative Sound Blaster X-Fi Titanium Fatal1ty Champion Series + * Creative Sound Blaster X-Fi Titanium Fatal1ty Professional Series + * Creative Sound Blaster X-Fi Titanium Professional Audio + * Creative Sound Blaster X-Fi Titanium + * Creative Sound Blaster X-Fi Elite Pro + * Creative Sound Blaster X-Fi Platinum + * Creative Sound Blaster X-Fi Fatal1ty + * Creative Sound Blaster X-Fi XtremeGamer + * Creative Sound Blaster X-Fi XtremeMusic + + reference_rate - reference sample rate, 44100 or 48000 (default) + multiple - multiple to ref. sample rate, 1 or 2 (default) + + This module supports multiple cards. + Module snd-darla20 ------------------ @@ -758,7 +777,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. single_cmd - Use single immediate commands to communicate with codecs (for debugging only) enable_msi - Enable Message Signaled Interrupt (MSI) (default = off) - power_save - Automatic power-saving timtout (in second, 0 = + power_save - Automatic power-saving timeout (in second, 0 = disable) power_save_controller - Reset HD-audio controller in power-saving mode (default = on) @@ -929,6 +948,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. * Onkyo SE-90PCI * Onkyo SE-200PCI * ESI Juli@ + * ESI Maya44 * Hercules Fortissimo IV * EGO-SYS WaveTerminal 192M @@ -937,7 +957,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. prodigy71xt, prodigy71hifi, prodigyhd2, prodigy192, juli, aureon51, aureon71, universe, ap192, k8x800, phase22, phase28, ms300, av710, se200pci, se90pci, - fortissimo4, sn25p, WT192M + fortissimo4, sn25p, WT192M, maya44 This module supports multiple cards and autoprobe. @@ -1097,6 +1117,13 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. This module supports multiple cards. The driver requires the firmware loader support on kernel. + Module snd-lx6464es + ------------------- + + Module for Digigram LX6464ES boards + + This module supports multiple cards. + Module snd-maestro3 ------------------- @@ -1547,13 +1574,15 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module snd-sc6000 ----------------- - Module for Gallant SC-6000 soundcard. + Module for Gallant SC-6000 soundcard and later models: SC-6600 + and SC-7000. port - Port # (0x220 or 0x240) mss_port - MSS Port # (0x530 or 0xe80) irq - IRQ # (5,7,9,10,11) mpu_irq - MPU-401 IRQ # (5,7,9,10) ,0 - no MPU-401 irq dma - DMA # (1,3,0) + joystick - Enable gameport - 0 = disable (default), 1 = enable This module supports multiple cards. @@ -1863,7 +1892,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. ------------------- Module for sound cards based on the Asus AV100/AV200 chips, - i.e., Xonar D1, DX, D2, D2X, HDAV1.3 (Deluxe), and Essence STX. + i.e., Xonar D1, DX, D2, D2X, HDAV1.3 (Deluxe), Essence ST + (Deluxe) and Essence STX. This module supports autoprobe and multiple cards. diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 0d8d23581c4..939a3dd5814 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -240,6 +240,7 @@ AD1986A laptop-automute 2-channel with EAPD and HP-automute (Lenovo N100) ultra 2-channel with EAPD (Samsung Ultra tablet PC) samsung 2-channel with EAPD (Samsung R65) + samsung-p50 2-channel with HP-automute (Samsung P50) AD1988/AD1988B/AD1989A/AD1989B ============================== diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index 55aab116823..0b5b480708f 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -16,7 +16,7 @@ methods for the HD-audio hardware. The HD-audio component consists of two parts: the controller chip and the codec chips on the HD-audio bus. Linux provides a single driver for all controllers, snd-hda-intel. Although the driver name contains -a word of a well-known harware vendor, it's not specific to it but for +a word of a well-known hardware vendor, it's not specific to it but for all controller chips by other companies. Since the HD-audio controllers are supposed to be compatible, the single snd-hda-driver should work in most cases. But, not surprisingly, there are known diff --git a/Documentation/sound/alsa/Procfile.txt b/Documentation/sound/alsa/Procfile.txt index bba2dbb79d8..381908d8ca4 100644 --- a/Documentation/sound/alsa/Procfile.txt +++ b/Documentation/sound/alsa/Procfile.txt @@ -88,21 +88,34 @@ card*/pcm*/info substreams, etc. card*/pcm*/xrun_debug - This file appears when CONFIG_SND_DEBUG=y. - This shows the status of xrun (= buffer overrun/xrun) debug of - ALSA PCM middle layer, as an integer from 0 to 2. The value - can be changed by writing to this file, such as - - # cat 2 > /proc/asound/card0/pcm0p/xrun_debug - - When this value is greater than 0, the driver will show the - messages to kernel log when an xrun is detected. The debug - message is shown also when the invalid H/W pointer is detected - at the update of periods (usually called from the interrupt + This file appears when CONFIG_SND_DEBUG=y and + CONFIG_PCM_XRUN_DEBUG=y. + This shows the status of xrun (= buffer overrun/xrun) and + invalid PCM position debug/check of ALSA PCM middle layer. + It takes an integer value, can be changed by writing to this + file, such as + + # cat 5 > /proc/asound/card0/pcm0p/xrun_debug + + The value consists of the following bit flags: + bit 0 = Enable XRUN/jiffies debug messages + bit 1 = Show stack trace at XRUN / jiffies check + bit 2 = Enable additional jiffies check + + When the bit 0 is set, the driver will show the messages to + kernel log when an xrun is detected. The debug message is + shown also when the invalid H/W pointer is detected at the + update of periods (usually called from the interrupt handler). - When this value is greater than 1, the driver will show the - stack trace additionally. This may help the debugging. + When the bit 1 is set, the driver will show the stack trace + additionally. This may help the debugging. + + Since 2.6.30, this option can enable the hwptr check using + jiffies. This detects spontaneous invalid pointer callback + values, but can be lead to too much corrections for a (mostly + buggy) hardware that doesn't give smooth pointer updates. + This feature is enabled via the bit 2. card*/pcm*/sub*/info The general information of this PCM sub-stream. diff --git a/Documentation/sound/alsa/README.maya44 b/Documentation/sound/alsa/README.maya44 new file mode 100644 index 00000000000..0e41576fa13 --- /dev/null +++ b/Documentation/sound/alsa/README.maya44 @@ -0,0 +1,163 @@ +NOTE: The following is the original document of Rainer's patch that the +current maya44 code based on. Some contents might be obsoleted, but I +keep here as reference -- tiwai + +---------------------------------------------------------------- + +STATE OF DEVELOPMENT: + +This driver is being developed on the initiative of Piotr Makowski (oponek@gmail.com) and financed by Lars Bergmann. +Development is carried out by Rainer Zimmermann (mail@lightshed.de). + +ESI provided a sample Maya44 card for the development work. + +However, unfortunately it has turned out difficult to get detailed programming information, so I (Rainer Zimmermann) had to find out some card-specific information by experiment and conjecture. Some information (in particular, several GPIO bits) is still missing. + +This is the first testing version of the Maya44 driver released to the alsa-devel mailing list (Feb 5, 2008). + + +The following functions work, as tested by Rainer Zimmermann and Piotr Makowski: + +- playback and capture at all sampling rates +- input/output level +- crossmixing +- line/mic switch +- phantom power switch +- analogue monitor a.k.a bypass + + +The following functions *should* work, but are not fully tested: + +- Channel 3+4 analogue - S/PDIF input switching +- S/PDIF output +- all inputs/outputs on the M/IO/DIO extension card +- internal/external clock selection + + +*In particular, we would appreciate testing of these functions by anyone who has access to an M/IO/DIO extension card.* + + +Things that do not seem to work: + +- The level meters ("multi track") in 'alsamixer' do not seem to react to signals in (if this is a bug, it would probably be in the existing ICE1724 code). + +- Ardour 2.1 seems to work only via JACK, not using ALSA directly or via OSS. This still needs to be tracked down. + + +DRIVER DETAILS: + +the following files were added: + +pci/ice1724/maya44.c - Maya44 specific code +pci/ice1724/maya44.h +pci/ice1724/ice1724.patch +pci/ice1724/ice1724.h.patch - PROPOSED patch to ice1724.h (see SAMPLING RATES) +i2c/other/wm8776.c - low-level access routines for Wolfson WM8776 codecs +include/wm8776.h + + +Note that the wm8776.c code is meant to be card-independent and does not actually register the codec with the ALSA infrastructure. +This is done in maya44.c, mainly because some of the WM8776 controls are used in Maya44-specific ways, and should be named appropriately. + + +the following files were created in pci/ice1724, simply #including the corresponding file from the alsa-kernel tree: + +wtm.h +vt1720_mobo.h +revo.h +prodigy192.h +pontis.h +phase.h +maya44.h +juli.h +aureon.h +amp.h +envy24ht.h +se.h +prodigy_hifi.h + + +*I hope this is the correct way to do things.* + + +SAMPLING RATES: + +The Maya44 card (or more exactly, the Wolfson WM8776 codecs) allow a maximum sampling rate of 192 kHz for playback and 92 kHz for capture. + +As the ICE1724 chip only allows one global sampling rate, this is handled as follows: + +* setting the sampling rate on any open PCM device on the maya44 card will always set the *global* sampling rate for all playback and capture channels. + +* In the current state of the driver, setting rates of up to 192 kHz is permitted even for capture devices. + +*AVOID CAPTURING AT RATES ABOVE 96kHz*, even though it may appear to work. The codec cannot actually capture at such rates, meaning poor quality. + + +I propose some additional code for limiting the sampling rate when setting on a capture pcm device. However because of the global sampling rate, this logic would be somewhat problematic. + +The proposed code (currently deactivated) is in ice1712.h.patch, ice1724.c and maya44.c (in pci/ice1712). + + +SOUND DEVICES: + +PCM devices correspond to inputs/outputs as follows (assuming Maya44 is card #0): + +hw:0,0 input - stereo, analog input 1+2 +hw:0,0 output - stereo, analog output 1+2 +hw:0,1 input - stereo, analog input 3+4 OR S/PDIF input +hw:0,1 output - stereo, analog output 3+4 (and SPDIF out) + + +NAMING OF MIXER CONTROLS: + +(for more information about the signal flow, please refer to the block diagram on p.24 of the ESI Maya44 manual, or in the ESI windows software). + + +PCM: (digital) output level for channel 1+2 +PCM 1: same for channel 3+4 + +Mic Phantom+48V: switch for +48V phantom power for electrostatic microphones on input 1/2. + Make sure this is not turned on while any other source is connected to input 1/2. + It might damage the source and/or the maya44 card. + +Mic/Line input: if switch is is on, input jack 1/2 is microphone input (mono), otherwise line input (stereo). + +Bypass: analogue bypass from ADC input to output for channel 1+2. Same as "Monitor" in the windows driver. +Bypass 1: same for channel 3+4. + +Crossmix: cross-mixer from channels 1+2 to channels 3+4 +Crossmix 1: cross-mixer from channels 3+4 to channels 1+2 + +IEC958 Output: switch for S/PDIF output. + This is not supported by the ESI windows driver. + S/PDIF should output the same signal as channel 3+4. [untested!] + + +Digitial output selectors: + + These switches allow a direct digital routing from the ADCs to the DACs. + Each switch determines where the digital input data to one of the DACs comes from. + They are not supported by the ESI windows driver. + For normal operation, they should all be set to "PCM out". + +H/W: Output source channel 1 +H/W 1: Output source channel 2 +H/W 2: Output source channel 3 +H/W 3: Output source channel 4 + +H/W 4 ... H/W 9: unknown function, left in to enable testing. + Possibly some of these control S/PDIF output(s). + If these turn out to be unused, they will go away in later driver versions. + +Selectable values for each of the digital output selectors are: + "PCM out" -> DAC output of the corresponding channel (default setting) + "Input 1"... + "Input 4" -> direct routing from ADC output of the selected input channel + + +-------- + +Feb 14, 2008 +Rainer Zimmermann +mail@lightshed.de + diff --git a/Documentation/sound/alsa/hda_codec.txt b/Documentation/sound/alsa/hda_codec.txt index 34e87ec1379..de8efbc7e4b 100644 --- a/Documentation/sound/alsa/hda_codec.txt +++ b/Documentation/sound/alsa/hda_codec.txt @@ -114,7 +114,7 @@ For writing a sequence of verbs, use snd_hda_sequence_write(). There are variants of cached read/write, snd_hda_codec_write_cache(), snd_hda_sequence_write_cache(). These are used for recording the -register states for the power-mangement resume. When no PM is needed, +register states for the power-management resume. When no PM is needed, these are equivalent with non-cached version. To retrieve the number of sub nodes connected to the given node, use diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt index 9e6763264a2..9ac842be9b4 100644 --- a/Documentation/sound/alsa/soc/dapm.txt +++ b/Documentation/sound/alsa/soc/dapm.txt @@ -62,6 +62,7 @@ Audio DAPM widgets fall into a number of types:- o Mic - Mic (and optional Jack) o Line - Line Input/Output (and optional Jack) o Speaker - Speaker + o Supply - Power or clock supply widget used by other widgets. o Pre - Special PRE widget (exec before all others) o Post - Special POST widget (exec after all others) |