diff options
Diffstat (limited to 'Documentation/sound')
| -rw-r--r-- | Documentation/sound/alsa/ALSA-Configuration.txt | 71 | ||||
| -rw-r--r-- | Documentation/sound/alsa/Audiophile-Usb.txt | 333 | ||||
| -rw-r--r-- | Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl | 6 |
3 files changed, 404 insertions, 6 deletions
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 36b511c7cad..1def6049784 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -513,6 +513,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. This module supports multiple cards and autoprobe. + The power-management is supported. + Module snd-ens1371 ------------------ @@ -526,6 +528,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. This module supports multiple cards and autoprobe. + The power-management is supported. + Module snd-es968 ---------------- @@ -671,6 +675,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. model - force the model name position_fix - Fix DMA pointer (0 = auto, 1 = none, 2 = POSBUF, 3 = FIFO size) + single_cmd - Use single immediate commands to communicate with + codecs (for debugging only) This module supports one card and autoprobe. @@ -694,13 +700,34 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. asus 3-jack uniwill 3-jack F1734 2-jack + lg LG laptop (m1 express dual) test for testing/debugging purpose, almost all controls can be adjusted. Appearing only when compiled with $CONFIG_SND_DEBUG=y + auto auto-config reading BIOS (default) ALC260 hp HP machines fujitsu Fujitsu S7020 + acer Acer TravelMate + basic fixed pin assignment (old default model) + auto auto-config reading BIOS (default) + + ALC262 + fujitsu Fujitsu Laptop + basic fixed pin assignment w/o SPDIF + auto auto-config reading BIOS (default) + + ALC882/883/885 + 3stack-dig 3-jack with SPDIF I/O + 6stck-dig 6-jack digital with SPDIF I/O + auto auto-config reading BIOS (default) + + ALC861 + 3stack 3-jack + 3stack-dig 3-jack with SPDIF I/O + 6stack-dig 6-jack with SPDIF I/O + auto auto-config reading BIOS (default) CMI9880 minimal 3-jack in back @@ -710,6 +737,28 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. allout 5-jack in back, 2-jack in front, SPDIF out auto auto-config reading BIOS (default) + AD1981 + basic 3-jack (default) + hp HP nx6320 + + AD1986A + 6stack 6-jack, separate surrounds (default) + 3stack 3-stack, shared surrounds + laptop 2-channel only (FSC V2060, Samsung M50) + laptop-eapd 2-channel with EAPD (Samsung R65, ASUS A6J) + + AD1988 + 6stack 6-jack + 6stack-dig ditto with SPDIF + 3stack 3-jack + 3stack-dig ditto with SPDIF + laptop 3-jack with hp-jack automute + laptop-dig ditto with SPDIF + auto auto-confgi reading BIOS (default) + + STAC7661(?) + vaio Setup for VAIO FE550G/SZ110 + If the default configuration doesn't work and one of the above matches with your device, report it together with the PCI subsystem ID (output of "lspci -nv") to ALSA BTS or alsa-devel @@ -723,6 +772,17 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. (Usually SD_LPLIB register is more accurate than the position buffer.) + NB: If you get many "azx_get_response timeout" messages at + loading, it's likely a problem of interrupts (e.g. ACPI irq + routing). Try to boot with options like "pci=noacpi". Also, you + can try "single_cmd=1" module option. This will switch the + communication method between HDA controller and codecs to the + single immediate commands instead of CORB/RIRB. Basically, the + single command mode is provided only for BIOS, and you won't get + unsolicited events, too. But, at least, this works independently + from the irq. Remember this is a last resort, and should be + avoided as much as possible... + The power-management is supported. Module snd-hdsp @@ -802,6 +862,7 @@ 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 * AMP Ltd AUDIO2000 * TerraTec Aureon 5.1 Sky @@ -810,6 +871,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. * TerraTec Phase 22 * TerraTec Phase 28 * AudioTrak Prodigy 7.1 + * AudioTrak Prodigy 7.1LT * AudioTrak Prodigy 192 * Pontis MS300 * Albatron K8X800 Pro II @@ -820,9 +882,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. * Shuttle SN25P model - Use the given board model, one of the following: - revo71, amp2000, prodigy71, prodigy192, aureon51, - aureon71, universe, k8x800, phase22, phase28, ms300, - av710 + revo51, revo71, amp2000, prodigy71, prodigy71lt, + prodigy192, aureon51, aureon71, universe, + k8x800, phase22, phase28, ms300, av710 This module supports multiple cards and autoprobe. @@ -1353,6 +1415,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. vid - Vendor ID for the device (optional) pid - Product ID for the device (optional) + device_setup - Device specific magic number (optional) + - Influence depends on the device + - Default: 0x0000 This module supports multiple devices, autoprobe and hotplugging. diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt new file mode 100644 index 00000000000..4692c8e77dc --- /dev/null +++ b/Documentation/sound/alsa/Audiophile-Usb.txt @@ -0,0 +1,333 @@ + Guide to using M-Audio Audiophile USB with ALSA and Jack v1.2 + ======================================================== + + Thibault Le Meur <Thibault.LeMeur@supelec.fr> + +This document is a guide to using the M-Audio Audiophile USB (tm) device with +ALSA and JACK. + +1 - Audiophile USB Specs and correct usage +========================================== +This part is a reminder of important facts about the functions and limitations +of the device. + +The device has 4 audio interfaces, and 2 MIDI ports: + * Analog Stereo Input (Ai) + - This port supports 2 pairs of line-level audio inputs (1/4" TS and RCA) + - When the 1/4" TS (jack) connectors are connected, the RCA connectors + are disabled + * Analog Stereo Output (Ao) + * Digital Stereo Input (Di) + * Digital Stereo Output (Do) + * Midi In (Mi) + * Midi Out (Mo) + +The internal DAC/ADC has the following caracteristics: +* sample depth of 16 or 24 bits +* sample rate from 8kHz to 96kHz +* Two ports can't use different sample depths at the same time.Moreover, the +Audiophile USB documentation gives the following Warning: "Please exit any +audio application running before switching between bit depths" + +Due to the USB 1.1 bandwidth limitation, a limited number of interfaces can be +activated at the same time depending on the audio mode selected: + * 16-bit/48kHz ==> 4 channels in/ 4 channels out + - Ai+Ao+Di+Do + * 24-bit/48kHz ==> 4 channels in/2 channels out, + or 2 channels in/4 channels out + - Ai+Ao+Do or Ai+Di+Ao or Ai+Di+Do or Di+Ao+Do + * 24-bit/96kHz ==> 2 channels in, or 2 channels out (half duplex only) + - Ai or Ao or Di or Do + +Important facts about the Digital interface: +-------------------------------------------- + * The Do port additionnaly supports surround-encoded AC-3 and DTS passthrough, +though I haven't tested it under linux + - Note that in this setup only the Do interface can be enabled + * Apart from recording an audio digital stream, enabling the Di port is a way +to synchronize the device to an external sample clock + - As a consequence, the Di port must be enable only if an active Digital +source is connected + - Enabling Di when no digital source is connected can result in a +synchronization error (for instance sound played at an odd sample rate) + + +2 - Audiophile USB support in ALSA +================================== + +2.1 - MIDI ports +---------------- +The Audiophile USB MIDI ports will be automatically supported once the +following modules have been loaded: + * snd-usb-audio + * snd-seq + * snd-seq-midi + +No additionnal setting is required. + +2.2 - Audio ports +----------------- + +Audio functions of the Audiophile USB device are handled by the snd-usb-audio +module. This module can work in a default mode (without any device-specific +parameter), or in an advanced mode with the device-specific parameter called +"device_setup". + +2.2.1 - Default Alsa driver mode + +The default behaviour of the snd-usb-audio driver is to parse the device +capabilities at startup and enable all functions inside the device (including +all ports at any sample rates and any sample depths supported). This approach +has the advantage to let the driver easily switch from sample rates/depths +automatically according to the need of the application claiming the device. + +In this case the Audiophile ports are mapped to alsa pcm devices in the +following way (I suppose the device's index is 1): + * hw:1,0 is Ao in playback and Di in capture + * hw:1,1 is Do in playback and Ai in capture + * hw:1,2 is Do in AC3/DTS passthrough mode + +You must note as well that the device uses Big Endian byte encoding so that +supported audio format are S16_BE for 16-bit depth modes and S24_3BE for +24-bits depth mode. One exception is the hw:1,2 port which is Little Endian +compliant and thus uses S16_LE. + +Examples: + * playing a S24_3BE encoded raw file to the Ao port + % aplay -D hw:1,0 -c2 -t raw -r48000 -fS24_3BE test.raw + * recording a S24_3BE encoded raw file from the Ai port + % arecord -D hw:1,1 -c2 -t raw -r48000 -fS24_3BE test.raw + * playing a S16_BE encoded raw file to the Do port + % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test.raw + +If you're happy with the default Alsa driver setup and don't experience any +issue with this mode, then you can skip the following chapter. + +2.2.2 - Advanced module setup + +Due to the hardware constraints described above, the device initialization made +by the Alsa driver in default mode may result in a corrupted state of the +device. For instance, a particularly annoying issue is that the sound captured +from the Ai port sounds distorted (as if boosted with an excessive high volume +gain). + +For people having this problem, the snd-usb-audio module has a new module +parameter called "device_setup". + +2.2.2.1 - Initializing the working mode of the Audiohile USB + +As far as the Audiohile USB device is concerned, this value let the user +specify: + * the sample depth + * the sample rate + * whether the Di port is used or not + +Here is a list of supported device_setup values for this device: + * device_setup=0x00 (or omitted) + - Alsa driver default mode + - maintains backward compatibility with setups that do not use this + parameter by not introducing any change + - results sometimes in corrupted sound as decribed earlier + * device_setup=0x01 + - 16bits 48kHz mode with Di disabled + - Ai,Ao,Do can be used at the same time + - hw:1,0 is not available in capture mode + - hw:1,2 is not available + * device_setup=0x11 + - 16bits 48kHz mode with Di enabled + - Ai,Ao,Di,Do can be used at the same time + - hw:1,0 is available in capture mode + - hw:1,2 is not available + * device_setup=0x09 + - 24bits 48kHz mode with Di disabled + - Ai,Ao,Do can be used at the same time + - hw:1,0 is not available in capture mode + - hw:1,2 is not available + * device_setup=0x19 + - 24bits 48kHz mode with Di enabled + - 3 ports from {Ai,Ao,Di,Do} can be used at the same time + - hw:1,0 is available in capture mode and an active digital source must be + connected to Di + - hw:1,2 is not available + * device_setup=0x0D or 0x10 + - 24bits 96kHz mode + - Di is enabled by default for this mode but does not need to be connected + to an active source + - Only 1 port from {Ai,Ao,Di,Do} can be used at the same time + - hw:1,0 is available in captured mode + - hw:1,2 is not available + * device_setup=0x03 + - 16bits 48kHz mode with only the Do port enabled + - AC3 with DTS passthru (not tested) + - Caution with this setup the Do port is mapped to the pcm device hw:1,0 + +2.2.2.2 - Setting and switching configurations with the device_setup parameter + +The parameter can be given: + * By manually probing the device (as root): + # modprobe -r snd-usb-audio + # modprobe snd-usb-audio index=1 device_setup=0x09 + * Or while configuring the modules options in your modules configuration file + - For Fedora distributions, edit the /etc/modprobe.conf file: + alias snd-card-1 snd-usb-audio + options snd-usb-audio index=1 device_setup=0x09 + +IMPORTANT NOTE WHEN SWITCHING CONFIGURATION: +------------------------------------------- + * You may need to _first_ intialize the module with the correct device_setup + parameter and _only_after_ turn on the Audiophile USB device + * This is especially true when switching the sample depth: + - first trun off the device + - de-register the snd-usb-audio module + - change the device_setup parameter (by either manually reprobing the module + or changing modprobe.conf) + - turn on the device + +2.2.2.3 - Audiophile USB's device_setup structure + +If you want to understand the device_setup magic numbers for the Audiophile +USB, you need some very basic understanding of binary computation. However, +this is not required to use the parameter and you may skip thi section. + +The device_setup is one byte long and its structure is the following: + + +---+---+---+---+---+---+---+---+ + | b7| b6| b5| b4| b3| b2| b1| b0| + +---+---+---+---+---+---+---+---+ + | 0 | 0 | 0 | Di|24B|96K|DTS|SET| + +---+---+---+---+---+---+---+---+ + +Where: + * b0 is the "SET" bit + - it MUST be set if device_setup is initialized + * b1 is the "DTS" bit + - it is set only for Digital output with DTS/AC3 + - this setup is not tested + * b2 is the Rate selection flag + - When set to "1" the rate range is 48.1-96kHz + - Otherwise the sample rate range is 8-48kHz + * b3 is the bit depth selection flag + - When set to "1" samples are 24bits long + - Otherwise they are 16bits long + - Note that b2 implies b3 as the 96kHz mode is only supported for 24 bits + samples + * b4 is the Digital input flag + - When set to "1" the device assumes that an active digital source is + connected + - You shouldn't enable Di if no source is seen on the port (this leads to + synchronization issues) + - b4 is implied by b2 (since only one port is enabled at a time no synch + error can occur) + * b5 to b7 are reserved for future uses, and must be set to "0" + - might become Ao, Do, Ai, for b7, b6, b4 respectively + +Caution: + * there is no check on the value you will give to device_setup + - for instance choosing 0x05 (16bits 96kHz) will fail back to 0x09 since + b2 implies b3. But _there_will_be_no_warning_ in /var/log/messages + * Hardware constraints due to the USB bus limitation aren't checked + - choosing b2 will prepare all interfaces for 24bits/96kHz but you'll + only be able to use one at the same time + +2.2.3 - USB implementation details for this device + +You may safely skip this section if you're not interrested in driver +development. + +This section describes some internals aspect of the device and summarize the +data I got by usb-snooping the windows and linux drivers. + +The M-Audio Audiophile USB has 7 USB Interfaces: +a "USB interface": + * USB Interface nb.0 + * USB Interface nb.1 + - Audio Control function + * USB Interface nb.2 + - Analog Output + * USB Interface nb.3 + - Digital Output + * USB Interface nb.4 + - Analog Input + * USB Interface nb.5 + - Digital Input + * USB Interface nb.6 + - MIDI interface compliant with the MIDIMAN quirk + +Each interface has 5 altsettings (AltSet 1,2,3,4,5) except: + * Interface 3 (Digital Out) has an extra Alset nb.6 + * Interface 5 (Digital In) does not have Alset nb.3 and 5 + +Here is a short description of the AltSettings capabilities: + * AltSettings 1 corresponds to + - 24-bit depth, 48.1-96kHz sample mode + - Adaptive playback (Ao and Do), Synch capture (Ai), or Asynch capture (Di) + * AltSettings 2 corresponds to + - 24-bit depth, 8-48kHz sample mode + - Asynch capture and playback (Ao,Ai,Do,Di) + * AltSettings 3 corresponds to + - 24-bit depth, 8-48kHz sample mode + - Synch capture (Ai) and Adaptive playback (Ao,Do) + * AltSettings 4 corresponds to + - 16-bit depth, 8-48kHz sample mode + - Asynch capture and playback (Ao,Ai,Do,Di) + * AltSettings 5 corresponds to + - 16-bit depth, 8-48kHz sample mode + - Synch capture (Ai) and Adaptive playback (Ao,Do) + * AltSettings 6 corresponds to + - 16-bit depth, 8-48kHz sample mode + - Synch playback (Do), audio format type III IEC1937_AC-3 + +In order to ensure a correct intialization of the device, the driver +_must_know_ how the device will be used: + * if DTS is choosen, only Interface 2 with AltSet nb.6 must be + registered + * if 96KHz only AltSets nb.1 of each interface must be selected + * if samples are using 24bits/48KHz then AltSet 2 must me used if + Digital input is connected, and only AltSet nb.3 if Digital input + is not connected + * if samples are using 16bits/48KHz then AltSet 4 must me used if + Digital input is connected, and only AltSet nb.5 if Digital input + is not connected + +When device_setup is given as a parameter to the snd-usb-audio module, the +parse_audio_enpoint function uses a quirk called +"audiophile_skip_setting_quirk" in order to prevent AltSettings not +corresponding to device_setup from being registered in the driver. + +3 - Audiophile USB and Jack support +=================================== + +This section deals with support of the Audiophile USB device in Jack. +The main issue regarding this support is that the device is Big Endian +compliant. + +3.1 - Using the plug alsa plugin +-------------------------------- + +Jack doesn't directly support big endian devices. Thus, one way to have support +for this device with Alsa is to use the Alsa "plug" converter. + +For instance here is one way to run Jack with 2 playback channels on Ao and 2 +capture channels from Ai: + % jackd -R -dalsa -dplughw:1 -r48000 -p256 -n2 -D -Cplughw:1,1 + + +However you may see the following warning message: +"You appear to be using the ALSA software "plug" layer, probably a result of +using the "default" ALSA device. This is less efficient than it could be. +Consider using a hardware device instead rather than using the plug layer." + + +3.2 - Patching alsa to use direct pcm device +------------------------------------------- +A patch for Jack by Andreas Steinmetz adds support for Big Endian devices. +However it has not been included in the CVS tree. + +You can find it at the following URL: +http://sourceforge.net/tracker/index.php?func=detail&aid=1289682&group_id=39687& +atid=425939 + +After having applied the patch you can run jackd with the following command +line: + % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1 + diff --git a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl index 4251085d38d..6dc9d9f622c 100644 --- a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl +++ b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl @@ -1834,7 +1834,7 @@ mychip_set_sample_format(chip, runtime->format); mychip_set_sample_rate(chip, runtime->rate); mychip_set_channels(chip, runtime->channels); - mychip_set_dma_setup(chip, runtime->dma_area, + mychip_set_dma_setup(chip, runtime->dma_addr, chip->buffer_size, chip->period_size); return 0; @@ -3388,7 +3388,7 @@ struct _snd_pcm_runtime { .name = "PCM Playback Switch", .index = 0, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, - .private_values = 0xffff, + .private_value = 0xffff, .info = my_control_info, .get = my_control_get, .put = my_control_put @@ -3449,7 +3449,7 @@ struct _snd_pcm_runtime { </para> <para> - The <structfield>private_values</structfield> field contains + The <structfield>private_value</structfield> field contains an arbitrary long integer value for this record. When using generic <structfield>info</structfield>, <structfield>get</structfield> and |