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Diffstat (limited to 'Documentation/sound/cards')
| -rw-r--r-- | Documentation/sound/cards/audigy-mixer.rst | 368 | ||||
| -rw-r--r-- | Documentation/sound/cards/audiophile-usb.rst | 550 | ||||
| -rw-r--r-- | Documentation/sound/cards/bt87x.rst | 83 | ||||
| -rw-r--r-- | Documentation/sound/cards/cmipci.rst | 272 | ||||
| -rw-r--r-- | Documentation/sound/cards/emu10k1-jack.rst | 78 | ||||
| -rw-r--r-- | Documentation/sound/cards/hdspm.rst | 379 | ||||
| -rw-r--r-- | Documentation/sound/cards/img-spdif-in.rst | 53 | ||||
| -rw-r--r-- | Documentation/sound/cards/index.rst | 19 | ||||
| -rw-r--r-- | Documentation/sound/cards/joystick.rst | 91 | ||||
| -rw-r--r-- | Documentation/sound/cards/maya44.rst | 186 | ||||
| -rw-r--r-- | Documentation/sound/cards/mixart.rst | 110 | ||||
| -rw-r--r-- | Documentation/sound/cards/sb-live-mixer.rst | 373 | ||||
| -rw-r--r-- | Documentation/sound/cards/serial-u16550.rst | 93 | ||||
| -rw-r--r-- | Documentation/sound/cards/via82xx-mixer.rst | 8 |
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diff --git a/Documentation/sound/cards/audigy-mixer.rst b/Documentation/sound/cards/audigy-mixer.rst new file mode 100644 index 000000000000..86213234435f --- /dev/null +++ b/Documentation/sound/cards/audigy-mixer.rst @@ -0,0 +1,368 @@ +============================================= +Sound Blaster Audigy mixer / default DSP code +============================================= + +This is based on sb-live-mixer.rst. + +The EMU10K2 chips have a DSP part which can be programmed to support +various ways of sample processing, which is described here. +(This article does not deal with the overall functionality of the +EMU10K2 chips. See the manuals section for further details.) + +The ALSA driver programs this portion of chip by default code +(can be altered later) which offers the following functionality: + + +Digital mixer controls +====================== + +These controls are built using the DSP instructions. They offer extended +functionality. Only the default build-in code in the ALSA driver is described +here. Note that the controls work as attenuators: the maximum value is the +neutral position leaving the signal unchanged. Note that if the same destination +is mentioned in multiple controls, the signal is accumulated and can be wrapped +(set to maximal or minimal value without checking of overflow). + + +Explanation of used abbreviations: + +DAC + digital to analog converter +ADC + analog to digital converter +I2S + one-way three wire serial bus for digital sound by Philips Semiconductors + (this standard is used for connecting standalone DAC and ADC converters) +LFE + low frequency effects (subwoofer signal) +AC97 + a chip containing an analog mixer, DAC and ADC converters +IEC958 + S/PDIF +FX-bus + the EMU10K2 chip has an effect bus containing 64 accumulators. + Each of the synthesizer voices can feed its output to these accumulators + and the DSP microcontroller can operate with the resulting sum. + +name='PCM Front Playback Volume',index=0 +---------------------------------------- +This control is used to attenuate samples for left and right front PCM FX-bus +accumulators. ALSA uses accumulators 8 and 9 for left and right front PCM +samples for 5.1 playback. The result samples are forwarded to the front DAC PCM +slots of the Philips DAC. + +name='PCM Surround Playback Volume',index=0 +------------------------------------------- +This control is used to attenuate samples for left and right surround PCM FX-bus +accumulators. ALSA uses accumulators 2 and 3 for left and right surround PCM +samples for 5.1 playback. The result samples are forwarded to the surround DAC PCM +slots of the Philips DAC. + +name='PCM Center Playback Volume',index=0 +----------------------------------------- +This control is used to attenuate samples for center PCM FX-bus accumulator. +ALSA uses accumulator 6 for center PCM sample for 5.1 playback. The result sample +is forwarded to the center DAC PCM slot of the Philips DAC. + +name='PCM LFE Playback Volume',index=0 +-------------------------------------- +This control is used to attenuate sample for LFE PCM FX-bus accumulator. +ALSA uses accumulator 7 for LFE PCM sample for 5.1 playback. The result sample +is forwarded to the LFE DAC PCM slot of the Philips DAC. + +name='PCM Playback Volume',index=0 +---------------------------------- +This control is used to attenuate samples for left and right PCM FX-bus +accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples for +stereo playback. The result samples are forwarded to the front DAC PCM slots +of the Philips DAC. + +name='PCM Capture Volume',index=0 +--------------------------------- +This control is used to attenuate samples for left and right PCM FX-bus +accumulator. ALSA uses accumulators 0 and 1 for left and right PCM. +The result is forwarded to the ADC capture FIFO (thus to the standard capture +PCM device). + +name='Music Playback Volume',index=0 +------------------------------------ +This control is used to attenuate samples for left and right MIDI FX-bus +accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples. +The result samples are forwarded to the front DAC PCM slots of the AC97 codec. + +name='Music Capture Volume',index=0 +----------------------------------- +These controls are used to attenuate samples for left and right MIDI FX-bus +accumulator. ALSA uses accumulators 4 and 5 for left and right PCM. +The result is forwarded to the ADC capture FIFO (thus to the standard capture +PCM device). + +name='Mic Playback Volume',index=0 +---------------------------------- +This control is used to attenuate samples for left and right Mic input. +For Mic input is used AC97 codec. The result samples are forwarded to +the front DAC PCM slots of the Philips DAC. Samples are forwarded to Mic +capture FIFO (device 1 - 16bit/8KHz mono) too without volume control. + +name='Mic Capture Volume',index=0 +--------------------------------- +This control is used to attenuate samples for left and right Mic input. +The result is forwarded to the ADC capture FIFO (thus to the standard capture +PCM device). + +name='Audigy CD Playback Volume',index=0 +---------------------------------------- +This control is used to attenuate samples from left and right IEC958 TTL +digital inputs (usually used by a CDROM drive). The result samples are +forwarded to the front DAC PCM slots of the Philips DAC. + +name='Audigy CD Capture Volume',index=0 +--------------------------------------- +This control is used to attenuate samples from left and right IEC958 TTL +digital inputs (usually used by a CDROM drive). The result samples are +forwarded to the ADC capture FIFO (thus to the standard capture PCM device). + +name='IEC958 Optical Playback Volume',index=0 +--------------------------------------------- +This control is used to attenuate samples from left and right IEC958 optical +digital input. The result samples are forwarded to the front DAC PCM slots +of the Philips DAC. + +name='IEC958 Optical Capture Volume',index=0 +-------------------------------------------- +This control is used to attenuate samples from left and right IEC958 optical +digital inputs. The result samples are forwarded to the ADC capture FIFO +(thus to the standard capture PCM device). + +name='Line2 Playback Volume',index=0 +------------------------------------ +This control is used to attenuate samples from left and right I2S ADC +inputs (on the AudigyDrive). The result samples are forwarded to the front +DAC PCM slots of the Philips DAC. + +name='Line2 Capture Volume',index=1 +----------------------------------- +This control is used to attenuate samples from left and right I2S ADC +inputs (on the AudigyDrive). The result samples are forwarded to the ADC +capture FIFO (thus to the standard capture PCM device). + +name='Analog Mix Playback Volume',index=0 +----------------------------------------- +This control is used to attenuate samples from left and right I2S ADC +inputs from Philips ADC. The result samples are forwarded to the front +DAC PCM slots of the Philips DAC. This contains mix from analog sources +like CD, Line In, Aux, .... + +name='Analog Mix Capture Volume',index=1 +---------------------------------------- +This control is used to attenuate samples from left and right I2S ADC +inputs Philips ADC. The result samples are forwarded to the ADC +capture FIFO (thus to the standard capture PCM device). + +name='Aux2 Playback Volume',index=0 +----------------------------------- +This control is used to attenuate samples from left and right I2S ADC +inputs (on the AudigyDrive). The result samples are forwarded to the front +DAC PCM slots of the Philips DAC. + +name='Aux2 Capture Volume',index=1 +---------------------------------- +This control is used to attenuate samples from left and right I2S ADC +inputs (on the AudigyDrive). The result samples are forwarded to the ADC +capture FIFO (thus to the standard capture PCM device). + +name='Front Playback Volume',index=0 +------------------------------------ +All stereo signals are mixed together and mirrored to surround, center and LFE. +This control is used to attenuate samples for left and right front speakers of +this mix. + +name='Surround Playback Volume',index=0 +--------------------------------------- +All stereo signals are mixed together and mirrored to surround, center and LFE. +This control is used to attenuate samples for left and right surround speakers of +this mix. + +name='Center Playback Volume',index=0 +------------------------------------- +All stereo signals are mixed together and mirrored to surround, center and LFE. +This control is used to attenuate sample for center speaker of this mix. + +name='LFE Playback Volume',index=0 +---------------------------------- +All stereo signals are mixed together and mirrored to surround, center and LFE. +This control is used to attenuate sample for LFE speaker of this mix. + +name='Tone Control - Switch',index=0 +------------------------------------ +This control turns the tone control on or off. The samples for front, rear +and center / LFE outputs are affected. + +name='Tone Control - Bass',index=0 +---------------------------------- +This control sets the bass intensity. There is no neutral value!! +When the tone control code is activated, the samples are always modified. +The closest value to pure signal is 20. + +name='Tone Control - Treble',index=0 +------------------------------------ +This control sets the treble intensity. There is no neutral value!! +When the tone control code is activated, the samples are always modified. +The closest value to pure signal is 20. + +name='Master Playback Volume',index=0 +------------------------------------- +This control is used to attenuate samples for front, surround, center and +LFE outputs. + +name='IEC958 Optical Raw Playback Switch',index=0 +------------------------------------------------- +If this switch is on, then the samples for the IEC958 (S/PDIF) digital +output are taken only from the raw FX8010 PCM, otherwise standard front +PCM samples are taken. + + +PCM stream related controls +=========================== + +name='EMU10K1 PCM Volume',index 0-31 +------------------------------------ +Channel volume attenuation in range 0-0xffff. The maximum value (no +attenuation) is default. The channel mapping for three values is +as follows: + +* 0 - mono, default 0xffff (no attenuation) +* 1 - left, default 0xffff (no attenuation) +* 2 - right, default 0xffff (no attenuation) + +name='EMU10K1 PCM Send Routing',index 0-31 +------------------------------------------ +This control specifies the destination - FX-bus accumulators. There 24 +values with this mapping: + +* 0 - mono, A destination (FX-bus 0-63), default 0 +* 1 - mono, B destination (FX-bus 0-63), default 1 +* 2 - mono, C destination (FX-bus 0-63), default 2 +* 3 - mono, D destination (FX-bus 0-63), default 3 +* 4 - mono, E destination (FX-bus 0-63), default 0 +* 5 - mono, F destination (FX-bus 0-63), default 0 +* 6 - mono, G destination (FX-bus 0-63), default 0 +* 7 - mono, H destination (FX-bus 0-63), default 0 +* 8 - left, A destination (FX-bus 0-63), default 0 +* 9 - left, B destination (FX-bus 0-63), default 1 +* 10 - left, C destination (FX-bus 0-63), default 2 +* 11 - left, D destination (FX-bus 0-63), default 3 +* 12 - left, E destination (FX-bus 0-63), default 0 +* 13 - left, F destination (FX-bus 0-63), default 0 +* 14 - left, G destination (FX-bus 0-63), default 0 +* 15 - left, H destination (FX-bus 0-63), default 0 +* 16 - right, A destination (FX-bus 0-63), default 0 +* 17 - right, B destination (FX-bus 0-63), default 1 +* 18 - right, C destination (FX-bus 0-63), default 2 +* 19 - right, D destination (FX-bus 0-63), default 3 +* 20 - right, E destination (FX-bus 0-63), default 0 +* 21 - right, F destination (FX-bus 0-63), default 0 +* 22 - right, G destination (FX-bus 0-63), default 0 +* 23 - right, H destination (FX-bus 0-63), default 0 + +Don't forget that it's illegal to assign a channel to the same FX-bus accumulator +more than once (it means 0=0 && 1=0 is an invalid combination). + +name='EMU10K1 PCM Send Volume',index 0-31 +----------------------------------------- +It specifies the attenuation (amount) for given destination in range 0-255. +The channel mapping is following: + +* 0 - mono, A destination attn, default 255 (no attenuation) +* 1 - mono, B destination attn, default 255 (no attenuation) +* 2 - mono, C destination attn, default 0 (mute) +* 3 - mono, D destination attn, default 0 (mute) +* 4 - mono, E destination attn, default 0 (mute) +* 5 - mono, F destination attn, default 0 (mute) +* 6 - mono, G destination attn, default 0 (mute) +* 7 - mono, H destination attn, default 0 (mute) +* 8 - left, A destination attn, default 255 (no attenuation) +* 9 - left, B destination attn, default 0 (mute) +* 10 - left, C destination attn, default 0 (mute) +* 11 - left, D destination attn, default 0 (mute) +* 12 - left, E destination attn, default 0 (mute) +* 13 - left, F destination attn, default 0 (mute) +* 14 - left, G destination attn, default 0 (mute) +* 15 - left, H destination attn, default 0 (mute) +* 16 - right, A destination attn, default 0 (mute) +* 17 - right, B destination attn, default 255 (no attenuation) +* 18 - right, C destination attn, default 0 (mute) +* 19 - right, D destination attn, default 0 (mute) +* 20 - right, E destination attn, default 0 (mute) +* 21 - right, F destination attn, default 0 (mute) +* 22 - right, G destination attn, default 0 (mute) +* 23 - right, H destination attn, default 0 (mute) + + + +MANUALS/PATENTS +=============== + +ftp://opensource.creative.com/pub/doc +------------------------------------- + +LM4545.pdf + AC97 Codec + +m2049.pdf + The EMU10K1 Digital Audio Processor + +hog63.ps + FX8010 - A DSP Chip Architecture for Audio Effects + + +WIPO Patents +------------ + +WO 9901813 (A1) + Audio Effects Processor with multiple asynchronous streams + (Jan. 14, 1999) + +WO 9901814 (A1) + Processor with Instruction Set for Audio Effects (Jan. 14, 1999) + +WO 9901953 (A1) + Audio Effects Processor having Decoupled Instruction + Execution and Audio Data Sequencing (Jan. 14, 1999) + + +US Patents (http://www.uspto.gov/) +---------------------------------- + +US 5925841 + Digital Sampling Instrument employing cache memory (Jul. 20, 1999) + +US 5928342 + Audio Effects Processor integrated on a single chip + with a multiport memory onto which multiple asynchronous + digital sound samples can be concurrently loaded + (Jul. 27, 1999) + +US 5930158 + Processor with Instruction Set for Audio Effects (Jul. 27, 1999) + +US 6032235 + Memory initialization circuit (Tram) (Feb. 29, 2000) + +US 6138207 + Interpolation looping of audio samples in cache connected to + system bus with prioritization and modification of bus transfers + in accordance with loop ends and minimum block sizes + (Oct. 24, 2000) + +US 6151670 + Method for conserving memory storage using a + pool of short term memory registers + (Nov. 21, 2000) + +US 6195715 + Interrupt control for multiple programs communicating with + a common interrupt by associating programs to GP registers, + defining interrupt register, polling GP registers, and invoking + callback routine associated with defined interrupt register + (Feb. 27, 2001) diff --git a/Documentation/sound/cards/audiophile-usb.rst b/Documentation/sound/cards/audiophile-usb.rst new file mode 100644 index 000000000000..a7bb5648331f --- /dev/null +++ b/Documentation/sound/cards/audiophile-usb.rst @@ -0,0 +1,550 @@ +======================================================== +Guide to using M-Audio Audiophile USB with ALSA and Jack +======================================================== + +v1.5 + +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. + +History +======= + +* v1.4 - Thibault Le Meur (2007-07-11) + + - Added Low Endianness nature of 16bits-modes + found by Hakan Lennestal <Hakan.Lennestal@brfsodrahamn.se> + - Modifying document structure + +* v1.5 - Thibault Le Meur (2007-07-12) + - Added AC3/DTS passthru info + + +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 characteristics: + +* sample depth of 16 or 24 bits +* sample rate from 8kHz to 96kHz +* Two interfaces 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 additionally 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) + + +Audiophile USB MIDI support in ALSA +=================================== + +The Audiophile USB MIDI ports will be automatically supported once the +following modules have been loaded: + + * snd-usb-audio + * snd-seq-midi + +No additional setting is required. + + +Audiophile USB Audio support in ALSA +==================================== + +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``. + +Default Alsa driver mode +------------------------ + +The default behavior of the snd-usb-audio driver is to list the device +capabilities at startup and activate the required mode when required +by the applications: for instance if the user is recording in a +24bit-depth-mode and immediately after wants to switch to a 16bit-depth mode, +the snd-usb-audio module will reconfigure the device on the fly. + +This approach has the advantage to let the driver automatically switch from sample +rates/depths automatically according to the user's needs. However, those who +are using the device under windows know that this is not how the device is meant to +work: under windows applications must be closed before using the m-audio control +panel to switch the device working mode. Thus as we'll see in next section, this +Default Alsa driver mode can lead to device misconfigurations. + +Let's get back to the Default Alsa driver mode for now. In this case the +Audiophile interfaces 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 + +In this mode, 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 was reported to be Little Endian +compliant (supposedly supporting S16_LE) but processes in fact only S16_BE streams. +This has been fixed in kernel 2.6.23 and above and now the hw:1,2 interface +is reported to be big endian in this default driver mode. + +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 + + * playing an ac3 sample file to the Do port:: + + % aplay -D hw:1,2 --channels=6 ac3_S16_BE_encoded_file.raw + +If you're happy with the default Alsa driver mode and don't experience any +issue with this mode, then you can skip the following chapter. + +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 interface 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`` (this parameter was introduced in kernel +release 2.6.17) + +Initializing the working mode of the Audiophile USB +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +As far as the Audiophile USB device is concerned, this value let the user +specify: + + * the sample depth + * the sample rate + * whether the Di port is used or not + +When initialized with ``device_setup=0x00``, the snd-usb-audio module has +the same behaviour as when the parameter is omitted (see paragraph "Default +Alsa driver mode" above) + +Others modes are described in the following subsections. + +16-bit modes +~~~~~~~~~~~~ + +The two supported modes are: + + * ``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 + +In this modes the device operates only at 16bits-modes. Before kernel 2.6.23, +the devices where reported to be Big-Endian when in fact they were Little-Endian +so that playing a file was a matter of using: +:: + + % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test_S16_LE.raw + +where "test_S16_LE.raw" was in fact a little-endian sample file. + +Thanks to Hakan Lennestal (who discovered the Little-Endiannes of the device in +these modes) a fix has been committed (expected in kernel 2.6.23) and +Alsa now reports Little-Endian interfaces. Thus playing a file now is as simple as +using: +:: + + % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_LE test_S16_LE.raw + + +24-bit modes +~~~~~~~~~~~~ + +The three supported modes are: + + * ``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 + +In these modes the device is only Big-Endian compliant (see "Default Alsa driver +mode" above for an aplay command example) + +AC3 w/ DTS passthru mode +~~~~~~~~~~~~~~~~~~~~~~~~ + +Thanks to Hakan Lennestal, I now have a report saying that this mode works. + + * ``device_setup=0x03`` + + - 16bits 48kHz mode with only the Do port enabled + - AC3 with DTS passthru + - Caution with this setup the Do port is mapped to the pcm device hw:1,0 + +The command line used to playback the AC3/DTS encoded .wav-files in this mode: +:: + + % aplay -D hw:1,0 --channels=6 ac3_S16_LE_encoded_file.raw + +How to use 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 + (typically a .conf file in /etc/modprobe.d/ directory::: + + alias snd-card-1 snd-usb-audio + options snd-usb-audio index=1 device_setup=0x09 + +CAUTION when initializing the device +------------------------------------- + + * Correct initialization on the device requires that device_setup is given to + the module BEFORE the device is turned on. So, if you use the "manual probing" + method described above, take care to power-on the device AFTER this initialization. + + * Failing to respect this will lead to a misconfiguration of the device. In this case + turn off the device, unprobe the snd-usb-audio module, then probe it again with + correct device_setup parameter and then (and only then) turn on the device again. + + * If you've correctly initialized the device in a valid mode and then want to switch + to another mode (possibly with another sample-depth), please use also the following + procedure: + + - first turn off the device + - de-register the snd-usb-audio module (modprobe -r) + - change the device_setup parameter by changing the device_setup + option in ``/etc/modprobe.d/*.conf`` + - turn on the device + + * A workaround for this last issue has been applied to kernel 2.6.23, but it may not + be enough to ensure the 'stability' of the device initialization. + +Technical details for hackers +----------------------------- + +This section is for hackers, wanting to understand details about the device +internals and how Alsa supports it. + +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 this 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 + +USB implementation details for this device +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +You may safely skip this section if you're not interested in driver +hacking. + +This section describes some internal aspects of the device and summarizes 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 initialization of the device, the driver +*must* *know* how the device will be used: + + * if DTS is chosen, 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_endpoints 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. + +Audiophile USB and Jack support +=============================== + +This section deals with support of the Audiophile USB device in Jack. + +There are 2 main potential issues when using Jackd with the device: + +* support for Big-Endian devices in 24-bit modes +* support for 4-in / 4-out channels + +Direct support in Jackd +----------------------- + +Jack supports big endian devices only in recent versions (thanks to +Andreas Steinmetz for his first big-endian patch). I can't remember +exactly when this support was released into jackd, let's just say that +with jackd version 0.103.0 it's almost ok (just a small bug is affecting +16bits Big-Endian devices, but since you've read carefully the above +paragraphs, you're now using kernel >= 2.6.23 and your 16bits devices +are now Little Endians ;-) ). + +You can run jackd with the following command for playback with Ao and +record with Ai: +:: + + % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1 + +Using Alsa plughw +----------------- + +If you don't have a recent Jackd installed, you can downgrade to using +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. + +Getting 2 input and/or output interfaces in Jack +------------------------------------------------ + +As you can see, starting the Jack server this way will only enable 1 stereo +input (Di or Ai) and 1 stereo output (Ao or Do). + +This is due to the following restrictions: + +* Jack can only open one capture device and one playback device at a time +* The Audiophile USB is seen as 2 (or three) Alsa devices: hw:1,0, hw:1,1 + (and optionally hw:1,2) + +If you want to get Ai+Di and/or Ao+Do support with Jack, you would need to +combine the Alsa devices into one logical "complex" device. + +If you want to give it a try, I recommend reading the information from +this page: http://www.sound-man.co.uk/linuxaudio/ice1712multi.html +It is related to another device (ice1712) but can be adapted to suit +the Audiophile USB. + +Enabling multiple Audiophile USB interfaces for Jackd will certainly require: + +* Making sure your Jackd version has the MMAP_COMPLEX patch (see the ice1712 page) +* (maybe) patching the alsa-lib/src/pcm/pcm_multi.c file (see the ice1712 page) +* define a multi device (combination of hw:1,0 and hw:1,1) in your .asoundrc + file +* start jackd with this device + +I had no success in testing this for now, if you have any success with this kind +of setup, please drop me an email. diff --git a/Documentation/sound/cards/bt87x.rst b/Documentation/sound/cards/bt87x.rst new file mode 100644 index 000000000000..912732d3ef9e --- /dev/null +++ b/Documentation/sound/cards/bt87x.rst @@ -0,0 +1,83 @@ +================= +ALSA BT87x Driver +================= + +Intro +===== + +You might have noticed that the bt878 grabber cards have actually +*two* PCI functions: +:: + + $ lspci + [ ... ] + 00:0a.0 Multimedia video controller: Brooktree Corporation Bt878 (rev 02) + 00:0a.1 Multimedia controller: Brooktree Corporation Bt878 (rev 02) + [ ... ] + +The first does video, it is backward compatible to the bt848. The second +does audio. snd-bt87x is a driver for the second function. It's a sound +driver which can be used for recording sound (and *only* recording, no +playback). As most TV cards come with a short cable which can be plugged +into your sound card's line-in you probably don't need this driver if all +you want to do is just watching TV... + +Some cards do not bother to connect anything to the audio input pins of +the chip, and some other cards use the audio function to transport MPEG +video data, so it's quite possible that audio recording may not work +with your card. + + +Driver Status +============= + +The driver is now stable. However, it doesn't know about many TV cards, +and it refuses to load for cards it doesn't know. + +If the driver complains ("Unknown TV card found, the audio driver will +not load"), you can specify the ``load_all=1`` option to force the driver to +try to use the audio capture function of your card. If the frequency of +recorded data is not right, try to specify the ``digital_rate`` option with +other values than the default 32000 (often it's 44100 or 64000). + +If you have an unknown card, please mail the ID and board name to +<alsa-devel@alsa-project.org>, regardless of whether audio capture works +or not, so that future versions of this driver know about your card. + + +Audio modes +=========== + +The chip knows two different modes (digital/analog). snd-bt87x +registers two PCM devices, one for each mode. They cannot be used at +the same time. + + +Digital audio mode +================== + +The first device (hw:X,0) gives you 16 bit stereo sound. The sample +rate depends on the external source which feeds the Bt87x with digital +sound via I2S interface. + + +Analog audio mode (A/D) +======================= + +The second device (hw:X,1) gives you 8 or 16 bit mono sound. Supported +sample rates are between 119466 and 448000 Hz (yes, these numbers are +that high). If you've set the CONFIG_SND_BT87X_OVERCLOCK option, the +maximum sample rate is 1792000 Hz, but audio data becomes unusable +beyond 896000 Hz on my card. + +The chip has three analog inputs. Consequently you'll get a mixer +device to control these. + + +Have fun, + + Clemens + + +Written by Clemens Ladisch <clemens@ladisch.de> +big parts copied from btaudio.txt by Gerd Knorr <kraxel@bytesex.org> diff --git a/Documentation/sound/cards/cmipci.rst b/Documentation/sound/cards/cmipci.rst new file mode 100644 index 000000000000..9ea1de6ec4ce --- /dev/null +++ b/Documentation/sound/cards/cmipci.rst @@ -0,0 +1,272 @@ +================================================= +Brief Notes on C-Media 8338/8738/8768/8770 Driver +================================================= + +Takashi Iwai <tiwai@suse.de> + + +Front/Rear Multi-channel Playback +--------------------------------- + +CM8x38 chip can use ADC as the second DAC so that two different stereo +channels can be used for front/rear playbacks. Since there are two +DACs, both streams are handled independently unlike the 4/6ch multi- +channel playbacks in the section below. + +As default, ALSA driver assigns the first PCM device (i.e. hw:0,0 for +card#0) for front and 4/6ch playbacks, while the second PCM device +(hw:0,1) is assigned to the second DAC for rear playback. + +There are slight differences between the two DACs: + +- The first DAC supports U8 and S16LE formats, while the second DAC + supports only S16LE. +- The second DAC supports only two channel stereo. + +Please note that the CM8x38 DAC doesn't support continuous playback +rate but only fixed rates: 5512, 8000, 11025, 16000, 22050, 32000, +44100 and 48000 Hz. + +The rear output can be heard only when "Four Channel Mode" switch is +disabled. Otherwise no signal will be routed to the rear speakers. +As default it's turned on. + +.. WARNING:: + When "Four Channel Mode" switch is off, the output from rear speakers + will be FULL VOLUME regardless of Master and PCM volumes [#]_. + This might damage your audio equipment. Please disconnect speakers + before your turn off this switch. + + +.. [#] + Well.. I once got the output with correct volume (i.e. same with the + front one) and was so excited. It was even with "Four Channel" bit + on and "double DAC" mode. Actually I could hear separate 4 channels + from front and rear speakers! But.. after reboot, all was gone. + It's a very pity that I didn't save the register dump at that + time.. Maybe there is an unknown register to achieve this... + +If your card has an extra output jack for the rear output, the rear +playback should be routed there as default. If not, there is a +control switch in the driver "Line-In As Rear", which you can change +via alsamixer or somewhat else. When this switch is on, line-in jack +is used as rear output. + +There are two more controls regarding to the rear output. +The "Exchange DAC" switch is used to exchange front and rear playback +routes, i.e. the 2nd DAC is output from front output. + + +4/6 Multi-Channel Playback +-------------------------- + +The recent CM8738 chips support for the 4/6 multi-channel playback +function. This is useful especially for AC3 decoding. + +When the multi-channel is supported, the driver name has a suffix +"-MC" such like "CMI8738-MC6". You can check this name from +/proc/asound/cards. + +When the 4/6-ch output is enabled, the second DAC accepts up to 6 (or +4) channels. While the dual DAC supports two different rates or +formats, the 4/6-ch playback supports only the same condition for all +channels. Since the multi-channel playback mode uses both DACs, you +cannot operate with full-duplex. + +The 4.0 and 5.1 modes are defined as the pcm "surround40" and "surround51" +in alsa-lib. For example, you can play a WAV file with 6 channels like +:: + + % aplay -Dsurround51 sixchannels.wav + +For programming the 4/6 channel playback, you need to specify the PCM +channels as you like and set the format S16LE. For example, for playback +with 4 channels, +:: + + snd_pcm_hw_params_set_access(pcm, hw, SND_PCM_ACCESS_RW_INTERLEAVED); + // or mmap if you like + snd_pcm_hw_params_set_format(pcm, hw, SND_PCM_FORMAT_S16_LE); + snd_pcm_hw_params_set_channels(pcm, hw, 4); + +and use the interleaved 4 channel data. + +There are some control switches affecting to the speaker connections: + +Line-In Mode + an enum control to change the behavior of line-in + jack. Either "Line-In", "Rear Output" or "Bass Output" can + be selected. The last item is available only with model 039 + or newer. + When "Rear Output" is chosen, the surround channels 3 and 4 + are output to line-in jack. +Mic-In Mode + an enum control to change the behavior of mic-in + jack. Either "Mic-In" or "Center/LFE Output" can be + selected. + When "Center/LFE Output" is chosen, the center and bass + channels (channels 5 and 6) are output to mic-in jack. + +Digital I/O +----------- + +The CM8x38 provides the excellent SPDIF capability with very cheap +price (yes, that's the reason I bought the card :) + +The SPDIF playback and capture are done via the third PCM device +(hw:0,2). Usually this is assigned to the PCM device "spdif". +The available rates are 44100 and 48000 Hz. +For playback with aplay, you can run like below: +:: + + % aplay -Dhw:0,2 foo.wav + +or + +:: + + % aplay -Dspdif foo.wav + +24bit format is also supported experimentally. + +The playback and capture over SPDIF use normal DAC and ADC, +respectively, so you cannot playback both analog and digital streams +simultaneously. + +To enable SPDIF output, you need to turn on "IEC958 Output Switch" +control via mixer or alsactl ("IEC958" is the official name of +so-called S/PDIF). Then you'll see the red light on from the card so +you know that's working obviously :) +The SPDIF input is always enabled, so you can hear SPDIF input data +from line-out with "IEC958 In Monitor" switch at any time (see +below). + +You can play via SPDIF even with the first device (hw:0,0), +but SPDIF is enabled only when the proper format (S16LE), sample rate +(441100 or 48000) and channels (2) are used. Otherwise it's turned +off. (Also don't forget to turn on "IEC958 Output Switch", too.) + + +Additionally there are relevant control switches: + +IEC958 Mix Analog + Mix analog PCM playback and FM-OPL/3 streams and + output through SPDIF. This switch appears only on old chip + models (CM8738 033 and 037). + + Note: without this control you can output PCM to SPDIF. + This is "mixing" of streams, so e.g. it's not for AC3 output + (see the next section). + +IEC958 In Select + Select SPDIF input, the internal CD-in (false) + and the external input (true). + +IEC958 Loop + SPDIF input data is loop back into SPDIF + output (aka bypass) + +IEC958 Copyright + Set the copyright bit. + +IEC958 5V + Select 0.5V (coax) or 5V (optical) interface. + On some cards this doesn't work and you need to change the + configuration with hardware dip-switch. + +IEC958 In Monitor + SPDIF input is routed to DAC. + +IEC958 In Phase Inverse + Set SPDIF input format as inverse. + [FIXME: this doesn't work on all chips..] + +IEC958 In Valid + Set input validity flag detection. + +Note: When "PCM Playback Switch" is on, you'll hear the digital output +stream through analog line-out. + + +The AC3 (RAW DIGITAL) OUTPUT +---------------------------- + +The driver supports raw digital (typically AC3) i/o over SPDIF. This +can be toggled via IEC958 playback control, but usually you need to +access it via alsa-lib. See alsa-lib documents for more details. + +On the raw digital mode, the "PCM Playback Switch" is automatically +turned off so that non-audio data is heard from the analog line-out. +Similarly the following switches are off: "IEC958 Mix Analog" and +"IEC958 Loop". The switches are resumed after closing the SPDIF PCM +device automatically to the previous state. + +On the model 033, AC3 is implemented by the software conversion in +the alsa-lib. If you need to bypass the software conversion of IEC958 +subframes, pass the "soft_ac3=0" module option. This doesn't matter +on the newer models. + + +ANALOG MIXER INTERFACE +---------------------- + +The mixer interface on CM8x38 is similar to SB16. +There are Master, PCM, Synth, CD, Line, Mic and PC Speaker playback +volumes. Synth, CD, Line and Mic have playback and capture switches, +too, as well as SB16. + +In addition to the standard SB mixer, CM8x38 provides more functions. +- PCM playback switch +- PCM capture switch (to capture the data sent to DAC) +- Mic Boost switch +- Mic capture volume +- Aux playback volume/switch and capture switch +- 3D control switch + + +MIDI CONTROLLER +--------------- + +With CMI8338 chips, the MPU401-UART interface is disabled as default. +You need to set the module option "mpu_port" to a valid I/O port address +to enable MIDI support. Valid I/O ports are 0x300, 0x310, 0x320 and +0x330. Choose a value that doesn't conflict with other cards. + +With CMI8738 and newer chips, the MIDI interface is enabled by default +and the driver automatically chooses a port address. + +There is *no* hardware wavetable function on this chip (except for +OPL3 synth below). +What's said as MIDI synth on Windows is a software synthesizer +emulation. On Linux use TiMidity or other softsynth program for +playing MIDI music. + + +FM OPL/3 Synth +-------------- + +The FM OPL/3 is also enabled as default only for the first card. +Set "fm_port" module option for more cards. + +The output quality of FM OPL/3 is, however, very weird. +I don't know why.. + +CMI8768 and newer chips do not have the FM synth. + + +Joystick and Modem +------------------ + +The legacy joystick is supported. To enable the joystick support, pass +joystick_port=1 module option. The value 1 means the auto-detection. +If the auto-detection fails, try to pass the exact I/O address. + +The modem is enabled dynamically via a card control switch "Modem". + + +Debugging Information +--------------------- + +The registers are shown in /proc/asound/cardX/cmipci. If you have any +problem (especially unexpected behavior of mixer), please attach the +output of this proc file together with the bug report. diff --git a/Documentation/sound/cards/emu10k1-jack.rst b/Documentation/sound/cards/emu10k1-jack.rst new file mode 100644 index 000000000000..6597f1ea83f0 --- /dev/null +++ b/Documentation/sound/cards/emu10k1-jack.rst @@ -0,0 +1,78 @@ +================================================================= +Low latency, multichannel audio with JACK and the emu10k1/emu10k2 +================================================================= + +This document is a guide to using the emu10k1 based devices with JACK for low +latency, multichannel recording functionality. All of my recent work to allow +Linux users to use the full capabilities of their hardware has been inspired +by the kX Project. Without their work I never would have discovered the true +power of this hardware. + + http://www.kxproject.com + - Lee Revell, 2005.03.30 + + +Until recently, emu10k1 users on Linux did not have access to the same low +latency, multichannel features offered by the "kX ASIO" feature of their +Windows driver. As of ALSA 1.0.9 this is no more! + +For those unfamiliar with kX ASIO, this consists of 16 capture and 16 playback +channels. With a post 2.6.9 Linux kernel, latencies down to 64 (1.33 ms) or +even 32 (0.66ms) frames should work well. + +The configuration is slightly more involved than on Windows, as you have to +select the correct device for JACK to use. Actually, for qjackctl users it's +fairly self explanatory - select Duplex, then for capture and playback select +the multichannel devices, set the in and out channels to 16, and the sample +rate to 48000Hz. The command line looks like this: +:: + + /usr/local/bin/jackd -R -dalsa -r48000 -p64 -n2 -D -Chw:0,2 -Phw:0,3 -S + +This will give you 16 input ports and 16 output ports. + +The 16 output ports map onto the 16 FX buses (or the first 16 of 64, for the +Audigy). The mapping from FX bus to physical output is described in +sb-live-mixer.rst (or audigy-mixer.rst). + +The 16 input ports are connected to the 16 physical inputs. Contrary to +popular belief, all emu10k1 cards are multichannel cards. Which of these +input channels have physical inputs connected to them depends on the card +model. Trial and error is highly recommended; the pinout diagrams +for the card have been reverse engineered by some enterprising kX users and are +available on the internet. Meterbridge is helpful here, and the kX forums are +packed with useful information. + +Each input port will either correspond to a digital (SPDIF) input, an analog +input, or nothing. The one exception is the SBLive! 5.1. On these devices, +the second and third input ports are wired to the center/LFE output. You will +still see 16 capture channels, but only 14 are available for recording inputs. + +This chart, borrowed from kxfxlib/da_asio51.cpp, describes the mapping of JACK +ports to FXBUS2 (multitrack recording input) and EXTOUT (physical output) +channels. + +JACK (& ASIO) mappings on 10k1 5.1 SBLive cards: + +============== ======== ============ +JACK Epilog FXBUS2(nr) +============== ======== ============ +capture_1 asio14 FXBUS2(0xe) +capture_2 asio15 FXBUS2(0xf) +capture_3 asio0 FXBUS2(0x0) +~capture_4 Center EXTOUT(0x11) // mapped to by Center +~capture_5 LFE EXTOUT(0x12) // mapped to by LFE +capture_6 asio3 FXBUS2(0x3) +capture_7 asio4 FXBUS2(0x4) +capture_8 asio5 FXBUS2(0x5) +capture_9 asio6 FXBUS2(0x6) +capture_10 asio7 FXBUS2(0x7) +capture_11 asio8 FXBUS2(0x8) +capture_12 asio9 FXBUS2(0x9) +capture_13 asio10 FXBUS2(0xa) +capture_14 asio11 FXBUS2(0xb) +capture_15 asio12 FXBUS2(0xc) +capture_16 asio13 FXBUS2(0xd) +============== ======== ============ + +TODO: describe use of ld10k1/qlo10k1 in conjunction with JACK diff --git a/Documentation/sound/cards/hdspm.rst b/Documentation/sound/cards/hdspm.rst new file mode 100644 index 000000000000..5373e51ed076 --- /dev/null +++ b/Documentation/sound/cards/hdspm.rst @@ -0,0 +1,379 @@ +======================================= +Software Interface ALSA-DSP MADI Driver +======================================= + +(translated from German, so no good English ;-), + +2004 - winfried ritsch + + +Full functionality has been added to the driver. Since some of +the Controls and startup-options are ALSA-Standard and only the +special Controls are described and discussed below. + + +Hardware functionality +====================== + +Audio transmission +------------------ + +* number of channels -- depends on transmission mode + + The number of channels chosen is from 1..Nmax. The reason to + use for a lower number of channels is only resource allocation, + since unused DMA channels are disabled and less memory is + allocated. So also the throughput of the PCI system can be + scaled. (Only important for low performance boards). + +* Single Speed -- 1..64 channels + +.. note:: + (Note: Choosing the 56channel mode for transmission or as + receiver, only 56 are transmitted/received over the MADI, but + all 64 channels are available for the mixer, so channel count + for the driver) + +* Double Speed -- 1..32 channels + +.. note:: + Note: Choosing the 56-channel mode for + transmission/receive-mode , only 28 are transmitted/received + over the MADI, but all 32 channels are available for the mixer, + so channel count for the driver + + +* Quad Speed -- 1..16 channels + +.. note:: + Choosing the 56-channel mode for + transmission/receive-mode , only 14 are transmitted/received + over the MADI, but all 16 channels are available for the mixer, + so channel count for the driver + +* Format -- signed 32 Bit Little Endian (SNDRV_PCM_FMTBIT_S32_LE) + +* Sample Rates -- + + Single Speed -- 32000, 44100, 48000 + + Double Speed -- 64000, 88200, 96000 (untested) + + Quad Speed -- 128000, 176400, 192000 (untested) + +* access-mode -- MMAP (memory mapped), Not interleaved (PCM_NON-INTERLEAVED) + +* buffer-sizes -- 64,128,256,512,1024,2048,8192 Samples + +* fragments -- 2 + +* Hardware-pointer -- 2 Modi + + + The Card supports the readout of the actual Buffer-pointer, + where DMA reads/writes. Since of the bulk mode of PCI it is only + 64 Byte accurate. SO it is not really usable for the + ALSA-mid-level functions (here the buffer-ID gives a better + result), but if MMAP is used by the application. Therefore it + can be configured at load-time with the parameter + precise-pointer. + + +.. hint:: + (Hint: Experimenting I found that the pointer is maximum 64 to + large never to small. So if you subtract 64 you always have a + safe pointer for writing, which is used on this mode inside + ALSA. In theory now you can get now a latency as low as 16 + Samples, which is a quarter of the interrupt possibilities.) + + * Precise Pointer -- off + interrupt used for pointer-calculation + + * Precise Pointer -- on + hardware pointer used. + +Controller +---------- + +Since DSP-MADI-Mixer has 8152 Fader, it does not make sense to +use the standard mixer-controls, since this would break most of +(especially graphic) ALSA-Mixer GUIs. So Mixer control has be +provided by a 2-dimensional controller using the +hwdep-interface. + +Also all 128+256 Peak and RMS-Meter can be accessed via the +hwdep-interface. Since it could be a performance problem always +copying and converting Peak and RMS-Levels even if you just need +one, I decided to export the hardware structure, so that of +needed some driver-guru can implement a memory-mapping of mixer +or peak-meters over ioctl, or also to do only copying and no +conversion. A test-application shows the usage of the controller. + +* Latency Controls --- not implemented !!! + +.. note:: + Note: Within the windows-driver the latency is accessible of a + control-panel, but buffer-sizes are controlled with ALSA from + hwparams-calls and should not be changed in run-state, I did not + implement it here. + + +* System Clock -- suspended !!!! + + * Name -- "System Clock Mode" + + * Access -- Read Write + + * Values -- "Master" "Slave" + +.. note:: + !!!! This is a hardware-function but is in conflict with the + Clock-source controller, which is a kind of ALSA-standard. I + makes sense to set the card to a special mode (master at some + frequency or slave), since even not using an Audio-application + a studio should have working synchronisations setup. So use + Clock-source-controller instead !!!! + +* Clock Source + + * Name -- "Sample Clock Source" + + * Access -- Read Write + + * Values -- "AutoSync", "Internal 32.0 kHz", "Internal 44.1 kHz", + "Internal 48.0 kHz", "Internal 64.0 kHz", "Internal 88.2 kHz", + "Internal 96.0 kHz" + + Choose between Master at a specific Frequency and so also the + Speed-mode or Slave (Autosync). Also see "Preferred Sync Ref" + +.. warning:: + !!!! This is no pure hardware function but was implemented by + ALSA by some ALSA-drivers before, so I use it also. !!! + + +* Preferred Sync Ref + + * Name -- "Preferred Sync Reference" + + * Access -- Read Write + + * Values -- "Word" "MADI" + + + Within the Auto-sync-Mode the preferred Sync Source can be + chosen. If it is not available another is used if possible. + +.. note:: + Note: Since MADI has a much higher bit-rate than word-clock, the + card should synchronise better in MADI Mode. But since the + RME-PLL is very good, there are almost no problems with + word-clock too. I never found a difference. + + +* TX 64 channel + + * Name -- "TX 64 channels mode" + + * Access -- Read Write + + * Values -- 0 1 + + Using 64-channel-modus (1) or 56-channel-modus for + MADI-transmission (0). + + +.. note:: + Note: This control is for output only. Input-mode is detected + automatically from hardware sending MADI. + + +* Clear TMS + + * Name -- "Clear Track Marker" + + * Access -- Read Write + + * Values -- 0 1 + + + Don't use to lower 5 Audio-bits on AES as additional Bits. + + +* Safe Mode oder Auto Input + + * Name -- "Safe Mode" + + * Access -- Read Write + + * Values -- 0 1 (default on) + + If on (1), then if either the optical or coaxial connection + has a failure, there is a takeover to the working one, with no + sample failure. Its only useful if you use the second as a + backup connection. + +* Input + + * Name -- "Input Select" + + * Access -- Read Write + + * Values -- optical coaxial + + + Choosing the Input, optical or coaxial. If Safe-mode is active, + this is the preferred Input. + +Mixer +----- + +* Mixer + + * Name -- "Mixer" + + * Access -- Read Write + + * Values - <channel-number 0-127> <Value 0-65535> + + + Here as a first value the channel-index is taken to get/set the + corresponding mixer channel, where 0-63 are the input to output + fader and 64-127 the playback to outputs fader. Value 0 + is channel muted 0 and 32768 an amplification of 1. + +* Chn 1-64 + + fast mixer for the ALSA-mixer utils. The diagonal of the + mixer-matrix is implemented from playback to output. + + +* Line Out + + * Name -- "Line Out" + + * Access -- Read Write + + * Values -- 0 1 + + Switching on and off the analog out, which has nothing to do + with mixing or routing. the analog outs reflects channel 63,64. + + +Information (only read access) +------------------------------ + +* Sample Rate + + * Name -- "System Sample Rate" + + * Access -- Read-only + + getting the sample rate. + + +* External Rate measured + + * Name -- "External Rate" + + * Access -- Read only + + + Should be "Autosync Rate", but Name used is + ALSA-Scheme. External Sample frequency liked used on Autosync is + reported. + + +* MADI Sync Status + + * Name -- "MADI Sync Lock Status" + + * Access -- Read + + * Values -- 0,1,2 + + MADI-Input is 0=Unlocked, 1=Locked, or 2=Synced. + + +* Word Clock Sync Status + + * Name -- "Word Clock Lock Status" + + * Access -- Read + + * Values -- 0,1,2 + + Word Clock Input is 0=Unlocked, 1=Locked, or 2=Synced. + +* AutoSync + + * Name -- "AutoSync Reference" + + * Access -- Read + + * Values -- "WordClock", "MADI", "None" + + Sync-Reference is either "WordClock", "MADI" or none. + +* RX 64ch --- noch nicht implementiert + + MADI-Receiver is in 64 channel mode oder 56 channel mode. + + +* AB_inp --- not tested + + Used input for Auto-Input. + + +* actual Buffer Position --- not implemented + + !!! this is a ALSA internal function, so no control is used !!! + + + +Calling Parameter +================= + +* index int array (min = 1, max = 8) + + Index value for RME HDSPM interface. card-index within ALSA + + note: ALSA-standard + +* id string array (min = 1, max = 8) + + ID string for RME HDSPM interface. + + note: ALSA-standard + +* enable int array (min = 1, max = 8) + + Enable/disable specific HDSPM sound-cards. + + note: ALSA-standard + +* precise_ptr int array (min = 1, max = 8) + + Enable precise pointer, or disable. + +.. note:: + note: Use only when the application supports this (which is a special case). + +* line_outs_monitor int array (min = 1, max = 8) + + Send playback streams to analog outs by default. + +.. note:: + note: each playback channel is mixed to the same numbered output + channel (routed). This is against the ALSA-convention, where all + channels have to be muted on after loading the driver, but was + used before on other cards, so i historically use it again) + + + +* enable_monitor int array (min = 1, max = 8) + + Enable Analog Out on Channel 63/64 by default. + +.. note :: + note: here the analog output is enabled (but not routed). diff --git a/Documentation/sound/cards/img-spdif-in.rst b/Documentation/sound/cards/img-spdif-in.rst new file mode 100644 index 000000000000..7df9f5ae2609 --- /dev/null +++ b/Documentation/sound/cards/img-spdif-in.rst @@ -0,0 +1,53 @@ +================================================ +Imagination Technologies SPDIF Input Controllers +================================================ + +The Imagination Technologies SPDIF Input controller contains the following +controls: + +* name='IEC958 Capture Mask',index=0 + +This control returns a mask that shows which of the IEC958 status bits +can be read using the 'IEC958 Capture Default' control. + +* name='IEC958 Capture Default',index=0 + +This control returns the status bits contained within the SPDIF stream that +is being received. The 'IEC958 Capture Mask' shows which bits can be read +from this control. + +* name='SPDIF In Multi Frequency Acquire',index=0 +* name='SPDIF In Multi Frequency Acquire',index=1 +* name='SPDIF In Multi Frequency Acquire',index=2 +* name='SPDIF In Multi Frequency Acquire',index=3 + +This control is used to attempt acquisition of up to four different sample +rates. The active rate can be obtained by reading the 'SPDIF In Lock Frequency' +control. + +When the value of this control is set to {0,0,0,0}, the rate given to hw_params +will determine the single rate the block will capture. Else, the rate given to +hw_params will be ignored, and the block will attempt capture for each of the +four sample rates set here. + +If less than four rates are required, the same rate can be specified more than +once + +* name='SPDIF In Lock Frequency',index=0 + +This control returns the active capture rate, or 0 if a lock has not been +acquired + +* name='SPDIF In Lock TRK',index=0 + +This control is used to modify the locking/jitter rejection characteristics +of the block. Larger values increase the locking range, but reduce jitter +rejection. + +* name='SPDIF In Lock Acquire Threshold',index=0 + +This control is used to change the threshold at which a lock is acquired. + +* name='SPDIF In Lock Release Threshold',index=0 + +This control is used to change the threshold at which a lock is released. diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst new file mode 100644 index 000000000000..c016f8c3b88b --- /dev/null +++ b/Documentation/sound/cards/index.rst @@ -0,0 +1,19 @@ +Card-Specific Information +========================= + +.. toctree:: + :maxdepth: 2 + + joystick + cmipci + sb-live-mixer + audigy-mixer + emu10k1-jack + via82xx-mixer + audiophile-usb + mixart + bt87x + maya44 + hdspm + serial-u16550 + img-spdif-in diff --git a/Documentation/sound/cards/joystick.rst b/Documentation/sound/cards/joystick.rst new file mode 100644 index 000000000000..a6e468c81d02 --- /dev/null +++ b/Documentation/sound/cards/joystick.rst @@ -0,0 +1,91 @@ +======================================= +Analog Joystick Support on ALSA Drivers +======================================= + +Oct. 14, 2003 + +Takashi Iwai <tiwai@suse.de> + +General +------- + +First of all, you need to enable GAMEPORT support on Linux kernel for +using a joystick with the ALSA driver. For the details of gameport +support, refer to Documentation/input/joystick.txt. + +The joystick support of ALSA drivers is different between ISA and PCI +cards. In the case of ISA (PnP) cards, it's usually handled by the +independent module (ns558). Meanwhile, the ALSA PCI drivers have the +built-in gameport support. Hence, when the ALSA PCI driver is built +in the kernel, CONFIG_GAMEPORT must be 'y', too. Otherwise, the +gameport support on that card will be (silently) disabled. + +Some adapter modules probe the physical connection of the device at +the load time. It'd be safer to plug in the joystick device before +loading the module. + + +PCI Cards +--------- + +For PCI cards, the joystick is enabled when the appropriate module +option is specified. Some drivers don't need options, and the +joystick support is always enabled. In the former ALSA version, there +was a dynamic control API for the joystick activation. It was +changed, however, to the static module options because of the system +stability and the resource management. + +The following PCI drivers support the joystick natively. + +============== ============= ============================================ +Driver Module Option Available Values +============== ============= ============================================ +als4000 joystick_port 0 = disable (default), 1 = auto-detect, + manual: any address (e.g. 0x200) +au88x0 N/A N/A +azf3328 joystick 0 = disable, 1 = enable, -1 = auto (default) +ens1370 joystick 0 = disable (default), 1 = enable +ens1371 joystick_port 0 = disable (default), 1 = auto-detect, + manual: 0x200, 0x208, 0x210, 0x218 +cmipci joystick_port 0 = disable (default), 1 = auto-detect, + manual: any address (e.g. 0x200) +cs4281 N/A N/A +cs46xx N/A N/A +es1938 N/A N/A +es1968 joystick 0 = disable (default), 1 = enable +sonicvibes N/A N/A +trident N/A N/A +via82xx [#f1]_ joystick 0 = disable (default), 1 = enable +ymfpci joystick_port 0 = disable (default), 1 = auto-detect, + manual: 0x201, 0x202, 0x204, 0x205 [#f2]_ +============== ============= ============================================ + +.. [#f1] VIA686A/B only +.. [#f2] With YMF744/754 chips, the port address can be chosen arbitrarily + +The following drivers don't support gameport natively, but there are +additional modules. Load the corresponding module to add the gameport +support. + +======= ================= +Driver Additional Module +======= ================= +emu10k1 emu10k1-gp +fm801 fm801-gp +======= ================= + +Note: the "pcigame" and "cs461x" modules are for the OSS drivers only. +These ALSA drivers (cs46xx, trident and au88x0) have the +built-in gameport support. + +As mentioned above, ALSA PCI drivers have the built-in gameport +support, so you don't have to load ns558 module. Just load "joydev" +and the appropriate adapter module (e.g. "analog"). + + +ISA Cards +--------- + +ALSA ISA drivers don't have the built-in gameport support. +Instead, you need to load "ns558" module in addition to "joydev" and +the adapter module (e.g. "analog"). diff --git a/Documentation/sound/cards/maya44.rst b/Documentation/sound/cards/maya44.rst new file mode 100644 index 000000000000..bf09a584b443 --- /dev/null +++ b/Documentation/sound/cards/maya44.rst @@ -0,0 +1,186 @@ +================================= +Notes on Maya44 USB Audio Support +================================= + +.. 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 + +Feb 14, 2008 + +Rainer Zimmermann <mail@lightshed.de> + +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 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 + diff --git a/Documentation/sound/cards/mixart.rst b/Documentation/sound/cards/mixart.rst new file mode 100644 index 000000000000..48aba98b088f --- /dev/null +++ b/Documentation/sound/cards/mixart.rst @@ -0,0 +1,110 @@ +============================================================== +Alsa driver for Digigram miXart8 and miXart8AES/EBU soundcards +============================================================== + +Digigram <alsa@digigram.com> + + +GENERAL +======= + +The miXart8 is a multichannel audio processing and mixing soundcard +that has 4 stereo audio inputs and 4 stereo audio outputs. +The miXart8AES/EBU is the same with a add-on card that offers further +4 digital stereo audio inputs and outputs. +Furthermore the add-on card offers external clock synchronisation +(AES/EBU, Word Clock, Time Code and Video Synchro) + +The mainboard has a PowerPC that offers onboard mpeg encoding and +decoding, samplerate conversions and various effects. + +The driver don't work properly at all until the certain firmwares +are loaded, i.e. no PCM nor mixer devices will appear. +Use the mixartloader that can be found in the alsa-tools package. + + +VERSION 0.1.0 +============= + +One miXart8 board will be represented as 4 alsa cards, each with 1 +stereo analog capture 'pcm0c' and 1 stereo analog playback 'pcm0p' device. +With a miXart8AES/EBU there is in addition 1 stereo digital input +'pcm1c' and 1 stereo digital output 'pcm1p' per card. + +Formats +------- +U8, S16_LE, S16_BE, S24_3LE, S24_3BE, FLOAT_LE, FLOAT_BE +Sample rates : 8000 - 48000 Hz continuously + +Playback +-------- +For instance the playback devices are configured to have max. 4 +substreams performing hardware mixing. This could be changed to a +maximum of 24 substreams if wished. +Mono files will be played on the left and right channel. Each channel +can be muted for each stream to use 8 analog/digital outputs separately. + +Capture +------- +There is one substream per capture device. For instance only stereo +formats are supported. + +Mixer +----- +<Master> and <Master Capture> + analog volume control of playback and capture PCM. +<PCM 0-3> and <PCM Capture> + digital volume control of each analog substream. +<AES 0-3> and <AES Capture> + digital volume control of each AES/EBU substream. +<Monitoring> + Loopback from 'pcm0c' to 'pcm0p' with digital volume + and mute control. + +Rem : for best audio quality try to keep a 0 attenuation on the PCM +and AES volume controls which is set by 219 in the range from 0 to 255 +(about 86% with alsamixer) + + +NOT YET IMPLEMENTED +=================== + +- external clock support (AES/EBU, Word Clock, Time Code, Video Sync) +- MPEG audio formats +- mono record +- on-board effects and samplerate conversions +- linked streams + + +FIRMWARE +======== + +[As of 2.6.11, the firmware can be loaded automatically with hotplug + when CONFIG_FW_LOADER is set. The mixartloader is necessary only + for older versions or when you build the driver into kernel.] + +For loading the firmware automatically after the module is loaded, use a +install command. For example, add the following entry to +/etc/modprobe.d/mixart.conf for miXart driver: +:: + + install snd-mixart /sbin/modprobe --first-time -i snd-mixart && \ + /usr/bin/mixartloader + + +(for 2.2/2.4 kernels, add "post-install snd-mixart /usr/bin/vxloader" to +/etc/modules.conf, instead.) + +The firmware binaries are installed on /usr/share/alsa/firmware +(or /usr/local/share/alsa/firmware, depending to the prefix option of +configure). There will be a miXart.conf file, which define the dsp image +files. + +The firmware files are copyright by Digigram SA + + +COPYRIGHT +========= + +Copyright (c) 2003 Digigram SA <alsa@digigram.com> +Distributable under GPL. diff --git a/Documentation/sound/cards/sb-live-mixer.rst b/Documentation/sound/cards/sb-live-mixer.rst new file mode 100644 index 000000000000..bcb62fc99bbb --- /dev/null +++ b/Documentation/sound/cards/sb-live-mixer.rst @@ -0,0 +1,373 @@ +=========================================== +Sound Blaster Live mixer / default DSP code +=========================================== + + +The EMU10K1 chips have a DSP part which can be programmed to support +various ways of sample processing, which is described here. +(This article does not deal with the overall functionality of the +EMU10K1 chips. See the manuals section for further details.) + +The ALSA driver programs this portion of chip by default code +(can be altered later) which offers the following functionality: + + +IEC958 (S/PDIF) raw PCM +======================= + +This PCM device (it's the 4th PCM device (index 3!) and first subdevice +(index 0) for a given card) allows to forward 48kHz, stereo, 16-bit +little endian streams without any modifications to the digital output +(coaxial or optical). The universal interface allows the creation of up +to 8 raw PCM devices operating at 48kHz, 16-bit little endian. It would +be easy to add support for multichannel devices to the current code, +but the conversion routines exist only for stereo (2-channel streams) +at the time. + +Look to tram_poke routines in lowlevel/emu10k1/emufx.c for more details. + + +Digital mixer controls +====================== + +These controls are built using the DSP instructions. They offer extended +functionality. Only the default build-in code in the ALSA driver is described +here. Note that the controls work as attenuators: the maximum value is the +neutral position leaving the signal unchanged. Note that if the same destination +is mentioned in multiple controls, the signal is accumulated and can be wrapped +(set to maximal or minimal value without checking of overflow). + + +Explanation of used abbreviations: + +DAC + digital to analog converter +ADC + analog to digital converter +I2S + one-way three wire serial bus for digital sound by Philips Semiconductors + (this standard is used for connecting standalone DAC and ADC converters) +LFE + low frequency effects (subwoofer signal) +AC97 + a chip containing an analog mixer, DAC and ADC converters +IEC958 + S/PDIF +FX-bus + the EMU10K1 chip has an effect bus containing 16 accumulators. + Each of the synthesizer voices can feed its output to these accumulators + and the DSP microcontroller can operate with the resulting sum. + + +``name='Wave Playback Volume',index=0`` +--------------------------------------- +This control is used to attenuate samples for left and right PCM FX-bus +accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples. +The result samples are forwarded to the front DAC PCM slots of the AC97 codec. + +``name='Wave Surround Playback Volume',index=0`` +------------------------------------------------ +This control is used to attenuate samples for left and right PCM FX-bus +accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples. +The result samples are forwarded to the rear I2S DACs. These DACs operates +separately (they are not inside the AC97 codec). + +``name='Wave Center Playback Volume',index=0`` +---------------------------------------------- +This control is used to attenuate samples for left and right PCM FX-bus +accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples. +The result is mixed to mono signal (single channel) and forwarded to +the ??rear?? right DAC PCM slot of the AC97 codec. + +``name='Wave LFE Playback Volume',index=0`` +------------------------------------------- +This control is used to attenuate samples for left and right PCM FX-bus +accumulators. ALSA uses accumulators 0 and 1 for left and right PCM. +The result is mixed to mono signal (single channel) and forwarded to +the ??rear?? left DAC PCM slot of the AC97 codec. + +``name='Wave Capture Volume',index=0``, ``name='Wave Capture Switch',index=0`` +------------------------------------------------------------------------------ +These controls are used to attenuate samples for left and right PCM FX-bus +accumulator. ALSA uses accumulators 0 and 1 for left and right PCM. +The result is forwarded to the ADC capture FIFO (thus to the standard capture +PCM device). + +``name='Synth Playback Volume',index=0`` +---------------------------------------- +This control is used to attenuate samples for left and right MIDI FX-bus +accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples. +The result samples are forwarded to the front DAC PCM slots of the AC97 codec. + +``name='Synth Capture Volume',index=0``, ``name='Synth Capture Switch',index=0`` +-------------------------------------------------------------------------------- +These controls are used to attenuate samples for left and right MIDI FX-bus +accumulator. ALSA uses accumulators 4 and 5 for left and right PCM. +The result is forwarded to the ADC capture FIFO (thus to the standard capture +PCM device). + +``name='Surround Playback Volume',index=0`` +------------------------------------------- +This control is used to attenuate samples for left and right rear PCM FX-bus +accumulators. ALSA uses accumulators 2 and 3 for left and right rear PCM samples. +The result samples are forwarded to the rear I2S DACs. These DACs operate +separately (they are not inside the AC97 codec). + +``name='Surround Capture Volume',index=0``, ``name='Surround Capture Switch',index=0`` +-------------------------------------------------------------------------------------- +These controls are used to attenuate samples for left and right rear PCM FX-bus +accumulators. ALSA uses accumulators 2 and 3 for left and right rear PCM samples. +The result is forwarded to the ADC capture FIFO (thus to the standard capture +PCM device). + +``name='Center Playback Volume',index=0`` +----------------------------------------- +This control is used to attenuate sample for center PCM FX-bus accumulator. +ALSA uses accumulator 6 for center PCM sample. The result sample is forwarded +to the ??rear?? right DAC PCM slot of the AC97 codec. + +``name='LFE Playback Volume',index=0`` +-------------------------------------- +This control is used to attenuate sample for center PCM FX-bus accumulator. +ALSA uses accumulator 6 for center PCM sample. The result sample is forwarded +to the ??rear?? left DAC PCM slot of the AC97 codec. + +``name='AC97 Playback Volume',index=0`` +--------------------------------------- +This control is used to attenuate samples for left and right front ADC PCM slots +of the AC97 codec. The result samples are forwarded to the front DAC PCM +slots of the AC97 codec. + +.. note:: + This control should be zero for the standard operations, otherwise + a digital loopback is activated. + + +``name='AC97 Capture Volume',index=0`` +-------------------------------------- +This control is used to attenuate samples for left and right front ADC PCM slots +of the AC97 codec. The result is forwarded to the ADC capture FIFO (thus to +the standard capture PCM device). + +.. note:: + This control should be 100 (maximal value), otherwise no analog + inputs of the AC97 codec can be captured (recorded). + +``name='IEC958 TTL Playback Volume',index=0`` +--------------------------------------------- +This control is used to attenuate samples from left and right IEC958 TTL +digital inputs (usually used by a CDROM drive). The result samples are +forwarded to the front DAC PCM slots of the AC97 codec. + +``name='IEC958 TTL Capture Volume',index=0`` +-------------------------------------------- +This control is used to attenuate samples from left and right IEC958 TTL +digital inputs (usually used by a CDROM drive). The result samples are +forwarded to the ADC capture FIFO (thus to the standard capture PCM device). + +``name='Zoom Video Playback Volume',index=0`` +--------------------------------------------- +This control is used to attenuate samples from left and right zoom video +digital inputs (usually used by a CDROM drive). The result samples are +forwarded to the front DAC PCM slots of the AC97 codec. + +``name='Zoom Video Capture Volume',index=0`` +-------------------------------------------- +This control is used to attenuate samples from left and right zoom video +digital inputs (usually used by a CDROM drive). The result samples are +forwarded to the ADC capture FIFO (thus to the standard capture PCM device). + +``name='IEC958 LiveDrive Playback Volume',index=0`` +--------------------------------------------------- +This control is used to attenuate samples from left and right IEC958 optical +digital input. The result samples are forwarded to the front DAC PCM slots +of the AC97 codec. + +``name='IEC958 LiveDrive Capture Volume',index=0`` +-------------------------------------------------- +This control is used to attenuate samples from left and right IEC958 optical +digital inputs. The result samples are forwarded to the ADC capture FIFO +(thus to the standard capture PCM device). + +``name='IEC958 Coaxial Playback Volume',index=0`` +------------------------------------------------- +This control is used to attenuate samples from left and right IEC958 coaxial +digital inputs. The result samples are forwarded to the front DAC PCM slots +of the AC97 codec. + +``name='IEC958 Coaxial Capture Volume',index=0`` +------------------------------------------------ +This control is used to attenuate samples from left and right IEC958 coaxial +digital inputs. The result samples are forwarded to the ADC capture FIFO +(thus to the standard capture PCM device). + +``name='Line LiveDrive Playback Volume',index=0``, ``name='Line LiveDrive Playback Volume',index=1`` +---------------------------------------------------------------------------------------------------- +This control is used to attenuate samples from left and right I2S ADC +inputs (on the LiveDrive). The result samples are forwarded to the front +DAC PCM slots of the AC97 codec. + +``name='Line LiveDrive Capture Volume',index=1``, ``name='Line LiveDrive Capture Volume',index=1`` +-------------------------------------------------------------------------------------------------- +This control is used to attenuate samples from left and right I2S ADC +inputs (on the LiveDrive). The result samples are forwarded to the ADC +capture FIFO (thus to the standard capture PCM device). + +``name='Tone Control - Switch',index=0`` +---------------------------------------- +This control turns the tone control on or off. The samples for front, rear +and center / LFE outputs are affected. + +``name='Tone Control - Bass',index=0`` +-------------------------------------- +This control sets the bass intensity. There is no neutral value!! +When the tone control code is activated, the samples are always modified. +The closest value to pure signal is 20. + +``name='Tone Control - Treble',index=0`` +---------------------------------------- +This control sets the treble intensity. There is no neutral value!! +When the tone control code is activated, the samples are always modified. +The closest value to pure signal is 20. + +``name='IEC958 Optical Raw Playback Switch',index=0`` +----------------------------------------------------- +If this switch is on, then the samples for the IEC958 (S/PDIF) digital +output are taken only from the raw FX8010 PCM, otherwise standard front +PCM samples are taken. + +``name='Headphone Playback Volume',index=1`` +-------------------------------------------- +This control attenuates the samples for the headphone output. + +``name='Headphone Center Playback Switch',index=1`` +--------------------------------------------------- +If this switch is on, then the sample for the center PCM is put to the +left headphone output (useful for SB Live cards without separate center/LFE +output). + +``name='Headphone LFE Playback Switch',index=1`` +------------------------------------------------ +If this switch is on, then the sample for the center PCM is put to the +right headphone output (useful for SB Live cards without separate center/LFE +output). + + +PCM stream related controls +=========================== + +``name='EMU10K1 PCM Volume',index 0-31`` +---------------------------------------- +Channel volume attenuation in range 0-0xffff. The maximum value (no +attenuation) is default. The channel mapping for three values is +as follows: + +* 0 - mono, default 0xffff (no attenuation) +* 1 - left, default 0xffff (no attenuation) +* 2 - right, default 0xffff (no attenuation) + +``name='EMU10K1 PCM Send Routing',index 0-31`` +---------------------------------------------- +This control specifies the destination - FX-bus accumulators. There are +twelve values with this mapping: + +* 0 - mono, A destination (FX-bus 0-15), default 0 +* 1 - mono, B destination (FX-bus 0-15), default 1 +* 2 - mono, C destination (FX-bus 0-15), default 2 +* 3 - mono, D destination (FX-bus 0-15), default 3 +* 4 - left, A destination (FX-bus 0-15), default 0 +* 5 - left, B destination (FX-bus 0-15), default 1 +* 6 - left, C destination (FX-bus 0-15), default 2 +* 7 - left, D destination (FX-bus 0-15), default 3 +* 8 - right, A destination (FX-bus 0-15), default 0 +* 9 - right, B destination (FX-bus 0-15), default 1 +* 10 - right, C destination (FX-bus 0-15), default 2 +* 11 - right, D destination (FX-bus 0-15), default 3 + +Don't forget that it's illegal to assign a channel to the same FX-bus accumulator +more than once (it means 0=0 && 1=0 is an invalid combination). + +``name='EMU10K1 PCM Send Volume',index 0-31`` +--------------------------------------------- +It specifies the attenuation (amount) for given destination in range 0-255. +The channel mapping is following: + +* 0 - mono, A destination attn, default 255 (no attenuation) +* 1 - mono, B destination attn, default 255 (no attenuation) +* 2 - mono, C destination attn, default 0 (mute) +* 3 - mono, D destination attn, default 0 (mute) +* 4 - left, A destination attn, default 255 (no attenuation) +* 5 - left, B destination attn, default 0 (mute) +* 6 - left, C destination attn, default 0 (mute) +* 7 - left, D destination attn, default 0 (mute) +* 8 - right, A destination attn, default 0 (mute) +* 9 - right, B destination attn, default 255 (no attenuation) +* 10 - right, C destination attn, default 0 (mute) +* 11 - right, D destination attn, default 0 (mute) + + + +MANUALS/PATENTS +=============== + +ftp://opensource.creative.com/pub/doc +------------------------------------- + +LM4545.pdf + AC97 Codec +m2049.pdf + The EMU10K1 Digital Audio Processor +hog63.ps + FX8010 - A DSP Chip Architecture for Audio Effects + + +WIPO Patents +------------ + +WO 9901813 (A1) + Audio Effects Processor with multiple asynchronous streams + (Jan. 14, 1999) + +WO 9901814 (A1) + Processor with Instruction Set for Audio Effects (Jan. 14, 1999) + +WO 9901953 (A1) + Audio Effects Processor having Decoupled Instruction + Execution and Audio Data Sequencing (Jan. 14, 1999) + + +US Patents (http://www.uspto.gov/) +---------------------------------- + +US 5925841 + Digital Sampling Instrument employing cache memory (Jul. 20, 1999) + +US 5928342 + Audio Effects Processor integrated on a single chip + with a multiport memory onto which multiple asynchronous + digital sound samples can be concurrently loaded + (Jul. 27, 1999) + +US 5930158 + Processor with Instruction Set for Audio Effects (Jul. 27, 1999) + +US 6032235 + Memory initialization circuit (Tram) (Feb. 29, 2000) + +US 6138207 + Interpolation looping of audio samples in cache connected to + system bus with prioritization and modification of bus transfers + in accordance with loop ends and minimum block sizes + (Oct. 24, 2000) + +US 6151670 + Method for conserving memory storage using a + pool of short term memory registers + (Nov. 21, 2000) + +US 6195715 + Interrupt control for multiple programs communicating with + a common interrupt by associating programs to GP registers, + defining interrupt register, polling GP registers, and invoking + callback routine associated with defined interrupt register + (Feb. 27, 2001) diff --git a/Documentation/sound/cards/serial-u16550.rst b/Documentation/sound/cards/serial-u16550.rst new file mode 100644 index 000000000000..197aeacea3da --- /dev/null +++ b/Documentation/sound/cards/serial-u16550.rst @@ -0,0 +1,93 @@ +=================================== +Serial UART 16450/16550 MIDI driver +=================================== + +The adaptor module parameter allows you to select either: + +* 0 - Roland Soundcanvas support (default) +* 1 - Midiator MS-124T support (1) +* 2 - Midiator MS-124W S/A mode (2) +* 3 - MS-124W M/B mode support (3) +* 4 - Generic device with multiple input support (4) + +For the Midiator MS-124W, you must set the physical M-S and A-B +switches on the Midiator to match the driver mode you select. + +In Roland Soundcanvas mode, multiple ALSA raw MIDI substreams are supported +(midiCnD0-midiCnD15). Whenever you write to a different substream, the driver +sends the nonstandard MIDI command sequence F5 NN, where NN is the substream +number plus 1. Roland modules use this command to switch between different +"parts", so this feature lets you treat each part as a distinct raw MIDI +substream. The driver provides no way to send F5 00 (no selection) or to not +send the F5 NN command sequence at all; perhaps it ought to. + +Usage example for simple serial converter: +:: + + /sbin/setserial /dev/ttyS0 uart none + /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 speed=115200 + +Usage example for Roland SoundCanvas with 4 MIDI ports: +:: + + /sbin/setserial /dev/ttyS0 uart none + /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 outs=4 + +In MS-124T mode, one raw MIDI substream is supported (midiCnD0); the outs +module parameter is automatically set to 1. The driver sends the same data to +all four MIDI Out connectors. Set the A-B switch and the speed module +parameter to match (A=19200, B=9600). + +Usage example for MS-124T, with A-B switch in A position: +:: + + /sbin/setserial /dev/ttyS0 uart none + /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=1 \ + speed=19200 + +In MS-124W S/A mode, one raw MIDI substream is supported (midiCnD0); +the outs module parameter is automatically set to 1. The driver sends +the same data to all four MIDI Out connectors at full MIDI speed. + +Usage example for S/A mode: +:: + + /sbin/setserial /dev/ttyS0 uart none + /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=2 + +In MS-124W M/B mode, the driver supports 16 ALSA raw MIDI substreams; +the outs module parameter is automatically set to 16. The substream +number gives a bitmask of which MIDI Out connectors the data should be +sent to, with midiCnD1 sending to Out 1, midiCnD2 to Out 2, midiCnD4 to +Out 3, and midiCnD8 to Out 4. Thus midiCnD15 sends the data to all 4 ports. +As a special case, midiCnD0 also sends to all ports, since it is not useful +to send the data to no ports. M/B mode has extra overhead to select the MIDI +Out for each byte, so the aggregate data rate across all four MIDI Outs is +at most one byte every 520 us, as compared with the full MIDI data rate of +one byte every 320 us per port. + +Usage example for M/B mode: +:: + + /sbin/setserial /dev/ttyS0 uart none + /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=3 + +The MS-124W hardware's M/A mode is currently not supported. This mode allows +the MIDI Outs to act independently at double the aggregate throughput of M/B, +but does not allow sending the same byte simultaneously to multiple MIDI Outs. +The M/A protocol requires the driver to twiddle the modem control lines under +timing constraints, so it would be a bit more complicated to implement than +the other modes. + +Midiator models other than MS-124W and MS-124T are currently not supported. +Note that the suffix letter is significant; the MS-124 and MS-124B are not +compatible, nor are the other known models MS-101, MS-101B, MS-103, and MS-114. +I do have documentation (tim.mann@compaq.com) that partially covers these models, +but no units to experiment with. The MS-124W support is tested with a real unit. +The MS-124T support is untested, but should work. + +The Generic driver supports multiple input and output substreams over a single +serial port. Similar to Roland Soundcanvas mode, F5 NN is used to select the +appropriate input or output stream (depending on the data direction). +Additionally, the CTS signal is used to regulate the data flow. The number of +inputs is specified by the ins parameter. diff --git a/Documentation/sound/cards/via82xx-mixer.rst b/Documentation/sound/cards/via82xx-mixer.rst new file mode 100644 index 000000000000..6ee993d4535b --- /dev/null +++ b/Documentation/sound/cards/via82xx-mixer.rst @@ -0,0 +1,8 @@ +============= +VIA82xx mixer +============= + +On many VIA82xx boards, the ``Input Source Select`` mixer control does not work. +Setting it to ``Input2`` on such boards will cause recording to hang, or fail +with EIO (input/output error) via OSS emulation. This control should be left +at ``Input1`` for such cards. |