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linux-7.2.3/Documentation/sound/cards/audigy-mixer.rst
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=============================================
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 built-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 clipped
(set to maximal or minimal value without checking for 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 D/A and A/D converters)
LFE
low frequency effects (used as subwoofer signal)
AC97
a chip containing an analog mixer, D/A and A/D 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 from 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 speakers.
name='PCM Surround Playback Volume',index=0
-------------------------------------------
This control is used to attenuate samples from 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 (rear)
speakers.
name='PCM Side Playback Volume',index=0
---------------------------------------
This control is used to attenuate samples from left and right side PCM FX-bus
accumulators. ALSA uses accumulators 14 and 15 for left and right side PCM
samples for 7.1 playback. The result samples are forwarded to the side speakers.
name='PCM Center Playback Volume',index=0
-----------------------------------------
This control is used to attenuate samples from center PCM FX-bus accumulator.
ALSA uses accumulator 6 for center PCM samples for 5.1 playback. The result
samples are forwarded to the center speaker.
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 samples for 5.1 playback. The result
samples are forwarded to the subwoofer.
name='PCM Playback Volume',index=0
----------------------------------
This control is used to attenuate samples from 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 speakers.
name='PCM Capture Volume',index=0
---------------------------------
This control is used to attenuate samples from left and right PCM FX-bus
accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples for
stereo playback. The result is forwarded to the standard capture PCM device.
name='Music Playback Volume',index=0
------------------------------------
This control is used to attenuate samples from 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 virtual stereo mixer.
name='Music Capture Volume',index=0
-----------------------------------
These controls are used to attenuate samples from left and right MIDI FX-bus
accumulator. ALSA uses accumulators 4 and 5 for left and right MIDI samples.
The result is forwarded to the standard capture PCM device.
name='Mic Playback Volume',index=0
----------------------------------
This control is used to attenuate samples from left and right Mic input of
the AC97 codec. The result samples are forwarded to the virtual stereo mixer.
name='Mic Capture Volume',index=0
---------------------------------
This control is used to attenuate samples from left and right Mic input of
the AC97 codec. The result is forwarded to the standard capture PCM device.
The original samples are also forwarded to the Mic capture PCM device (device 1;
16bit/8KHz mono) without volume control.
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 virtual stereo mixer.
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 is forwarded
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 virtual stereo mixer.
name='IEC958 Optical Capture Volume',index=0
--------------------------------------------
This control is used to attenuate samples from left and right IEC958 optical
digital inputs. The result is forwarded 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 virtual
stereo mixer.
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 is forwarded 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 virtual
stereo mixer. 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 is forwarded 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 virtual
stereo mixer.
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 is forwarded to the standard capture
PCM device.
name='Front Playback Volume',index=0
------------------------------------
This control is used to attenuate samples from the virtual stereo mixer.
The result samples are forwarded to the front speakers.
name='Surround Playback Volume',index=0
---------------------------------------
This control is used to attenuate samples from the virtual stereo mixer.
The result samples are forwarded to the surround (rear) speakers.
name='Side Playback Volume',index=0
-----------------------------------
This control is used to attenuate samples from the virtual stereo mixer.
The result samples are forwarded to the side speakers.
name='Center Playback Volume',index=0
-------------------------------------
This control is used to attenuate samples from the virtual stereo mixer.
The result samples are forwarded to the center speaker.
name='LFE Playback Volume',index=0
----------------------------------
This control is used to attenuate samples from the virtual stereo mixer.
The result samples are forwarded to the subwoofer.
name='Tone Control - Switch',index=0
------------------------------------
This control turns the tone control on or off. The samples forwarded to
the speaker 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 forwarded to the speaker 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 iec958 ALSA PCM device (which uses
accumulators 20 and 21 for left and right PCM by default).
PCM stream related controls
===========================
name='EMU10K1 PCM Volume',index 0-31
------------------------------------
Channel volume attenuation in range 0-0x1fffd. The middle 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 24
values in 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 4
* 5 - mono, F destination (FX-bus 0-63), default 5
* 6 - mono, G destination (FX-bus 0-63), default 6
* 7 - mono, H destination (FX-bus 0-63), default 7
* 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 4
* 13 - left, F destination (FX-bus 0-63), default 5
* 14 - left, G destination (FX-bus 0-63), default 6
* 15 - left, H destination (FX-bus 0-63), default 7
* 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 4
* 21 - right, F destination (FX-bus 0-63), default 5
* 22 - right, G destination (FX-bus 0-63), default 6
* 23 - right, H destination (FX-bus 0-63), default 7
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
===============
See sb-live-mixer.rst.
linux-7.2.3/Documentation/sound/cards/audiophile-usb.rst
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========================================================
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.
linux-7.2.3/Documentation/sound/cards/bt87x.rst
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=================
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>
linux-7.2.3/Documentation/sound/cards/cmipci.rst
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=================================================
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.
linux-7.2.3/Documentation/sound/cards/emu-mixer.rst
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==================================================
E-MU Digital Audio System mixer / default DSP code
==================================================
This document covers the E-MU 0404/1010/1212/1616/1820 PCI/PCI-e/CardBus
cards.
These cards use regular EMU10K2 (SoundBlaster Audigy) chips, but with an
alternative front-end geared towards semi-professional studio recording.
This document is based on audigy-mixer.rst.
Hardware compatibility
======================
The EMU10K2 chips have a very short capture FIFO, which makes recording
unreliable if the card's PCI bus requests are not handled with the
appropriate priority.
This is the case on more modern motherboards, where the PCI bus is only a
secondary peripheral, rather than the actual arbiter of device access.
In particular, I got recording glitches during simultaneous playback on an
Intel DP55 board (memory controller in the CPU), but had success with an
Intel DP45 board (memory controller in the north bridge).
The PCI Express variants of these cards (which have a PCI bridge on board,
but are otherwise identical) may be less problematic.
Driver capabilities
===================
This driver supports only 16-bit 44.1/48 kHz operation. The multi-channel
device (see emu10k1-jack.rst) additionally supports 24-bit capture.
A patchset to enhance the driver is available from `a GitHub repository
<https://github.com/ossilator/linux/tree/ossis-emu10k1>`_.
Its multi-channel device supports 24-bit for both playback and capture,
and also supports full 88.2/96/176.4/192 kHz operation.
It is not going to be upstreamed due to a fundamental disagreement about
what constitutes a good user experience.
Digital mixer controls
======================
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 clipped
(set to maximal or minimal value without checking for overflow).
Explanation of used abbreviations:
DAC
digital to analog converter
ADC
analog to digital converter
LFE
low frequency effects (used as subwoofer signal)
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='Clock Source',index=0
---------------------------
This control allows switching the word clock between internally generated
44.1 or 48 kHz, or a number of external sources.
Note: the sources for the 1616 CardBus card are unclear. Please report your
findings.
name='Clock Fallback',index=0
-----------------------------
This control determines the internal clock which the card switches to when
the selected external clock source is/becomes invalid.
name='DAC1 0202 14dB PAD',index=0, etc.
---------------------------------------
Output attenuation controls. Not available on 0404 cards.
name='ADC1 14dB PAD 0202',index=0, etc.
---------------------------------------
Input attenuation controls. Not available on 0404 cards.
name='Optical Output Mode',index=0
----------------------------------
Switches the TOSLINK output port between S/PDIF and ADAT.
Not available on 0404 cards (fixed to S/PDIF).
name='Optical Input Mode',index=0
---------------------------------
Switches the TOSLINK input port between S/PDIF and ADAT.
Not available on 0404 cards (fixed to S/PDIF).
name='PCM Front Playback Volume',index=0
----------------------------------------
This control is used to attenuate samples from 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 DSP 0 & 1
playback channels.
name='PCM Surround Playback Volume',index=0
-------------------------------------------
This control is used to attenuate samples from 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 DSP 2 & 3
playback channels.
name='PCM Side Playback Volume',index=0
---------------------------------------
This control is used to attenuate samples from left and right side PCM FX-bus
accumulators. ALSA uses accumulators 14 and 15 for left and right side PCM
samples for 7.1 playback. The result samples are forwarded to the DSP 6 & 7
playback channels.
name='PCM Center Playback Volume',index=0
-----------------------------------------
This control is used to attenuate samples from the center PCM FX-bus accumulator.
ALSA uses accumulator 6 for center PCM samples for 5.1 playback. The result samples
are forwarded to the DSP 4 playback channel.
name='PCM LFE Playback Volume',index=0
--------------------------------------
This control is used to attenuate samples from the LFE PCM FX-bus accumulator.
ALSA uses accumulator 7 for LFE PCM samples for 5.1 playback. The result samples
are forwarded to the DSP 5 playback channel.
name='PCM Playback Volume',index=0
----------------------------------
This control is used to attenuate samples from 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 virtual stereo mixer.
name='PCM Capture Volume',index=0
---------------------------------
This control is used to attenuate samples from left and right PCM FX-bus
accumulators. ALSA uses accumulators 0 and 1 for left and right PCM.
The result is forwarded to the standard capture PCM device.
name='Music Playback Volume',index=0
------------------------------------
This control is used to attenuate samples from 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 virtual stereo mixer.
name='Music Capture Volume',index=0
-----------------------------------
These controls are used to attenuate samples from left and right MIDI FX-bus
accumulator. ALSA uses accumulators 4 and 5 for left and right MIDI samples.
The result is forwarded to the standard capture PCM device.
name='Front Playback Volume',index=0
------------------------------------
This control is used to attenuate samples from the virtual stereo mixer.
The result samples are forwarded to the DSP 0 & 1 playback channels.
name='Surround Playback Volume',index=0
---------------------------------------
This control is used to attenuate samples from the virtual stereo mixer.
The result samples are forwarded to the DSP 2 & 3 playback channels.
name='Side Playback Volume',index=0
-----------------------------------
This control is used to attenuate samples from the virtual stereo mixer.
The result samples are forwarded to the DSP 6 & 7 playback channels.
name='Center Playback Volume',index=0
-------------------------------------
This control is used to attenuate samples from the virtual stereo mixer.
The result samples are forwarded to the DSP 4 playback channel.
name='LFE Playback Volume',index=0
----------------------------------
This control is used to attenuate samples from the virtual stereo mixer.
The result samples are forwarded to the DSP 5 playback channel.
name='Tone Control - Switch',index=0
------------------------------------
This control turns the tone control on or off. The samples forwarded to
the DSP playback channels 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 all DSP playback channels.
name='EMU Capture Volume',index=0
----------------------------------
This control is used to attenuate samples from the DSP 0 & 1 capture channels.
The result is forwarded to the standard capture PCM device.
name='DAC Left',index=0, etc.
-----------------------------
Select the source for the given physical audio output. These may be physical
inputs, playback channels (DSP xx, specified as a decimal number), or silence.
name='DSP x',index=0
--------------------
Select the source for the given capture channel (specified as a hexadecimal
digit). Same options as for the physical audio outputs.
PCM stream related controls
===========================
These controls are described in audigy-mixer.rst.
MANUALS/PATENTS
===============
See sb-live-mixer.rst.
linux-7.2.3/Documentation/sound/cards/emu10k1-jack.rst
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=================================================================
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
linux-7.2.3/Documentation/sound/cards/hdspm.rst
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=======================================
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).
linux-7.2.3/Documentation/sound/cards/img-spdif-in.rst
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================================================
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.
linux-7.2.3/Documentation/sound/cards/index.rst
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Card-Specific Information
=========================
.. toctree::
:maxdepth: 2
joystick
cmipci
sb-live-mixer
audigy-mixer
emu-mixer
emu10k1-jack
via82xx-mixer
audiophile-usb
mixart
bt87x
maya44
hdspm
serial-u16550
img-spdif-in
pcmtest
linux-7.2.3/Documentation/sound/cards/joystick.rst
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=======================================
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/joydev/joystick.rst.
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").
linux-7.2.3/Documentation/sound/cards/maya44.rst
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=================================
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!]
Digital 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
linux-7.2.3/Documentation/sound/cards/mixart.rst
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==============================================================
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.
linux-7.2.3/Documentation/sound/cards/multisound.sh
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linux-7.2.3/Documentation/sound/cards/pcmtest.rst
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.. SPDX-License-Identifier: GPL-2.0
The Virtual PCM Test Driver
===========================
The Virtual PCM Test Driver emulates a generic PCM device, and can be used for
testing/fuzzing of the userspace ALSA applications, as well as for testing/fuzzing of
the PCM middle layer. Additionally, it can be used for simulating hard to reproduce
problems with PCM devices.
What can this driver do?
~~~~~~~~~~~~~~~~~~~~~~~~
At this moment the driver can do the following things:
* Simulate both capture and playback processes
* Generate random or pattern-based capturing data
* Inject delays into the playback and capturing processes
* Inject errors during the PCM callbacks
It supports up to 8 substreams and 4 channels. Also it supports both interleaved and
non-interleaved access modes.
Also, this driver can check the playback stream for containing the predefined pattern,
which is used in the corresponding selftest (alsa/pcmtest-test.sh) to check the PCM middle
layer data transferring functionality. Additionally, this driver redefines the default
RESET ioctl, and the selftest covers this PCM API functionality as well.
Configuration
-------------
The driver has several parameters besides the common ALSA module parameters:
* fill_mode (bool) - Buffer fill mode (see below)
* inject_delay (int)
* inject_hwpars_err (bool)
* inject_prepare_err (bool)
* inject_trigger_err (bool)
Capture Data Generation
-----------------------
The driver has two modes of data generation: the first (0 in the fill_mode parameter)
means random data generation, the second (1 in the fill_mode) - pattern-based
data generation. Let's look at the second mode.
First of all, you may want to specify the pattern for data generation. You can do it
by writing the pattern to the debugfs file. There are pattern buffer debugfs entries
for each channel, as well as entries which contain the pattern buffer length.
* /sys/kernel/debug/pcmtest/fill_pattern[0-3]
* /sys/kernel/debug/pcmtest/fill_pattern[0-3]_len
To set the pattern for the channel 0 you can execute the following command:
.. code-block:: bash
echo -n mycoolpattern > /sys/kernel/debug/pcmtest/fill_pattern0
Then, after every capture action performed on the 'pcmtest' device the buffer for the
channel 0 will contain 'mycoolpatternmycoolpatternmycoolpatternmy...'.
The pattern itself can be up to 4096 bytes long.
Delay injection
---------------
The driver has 'inject_delay' parameter, which has very self-descriptive name and
can be used for time delay/speedup simulations. The parameter has integer type, and
it means the delay added between module's internal timer ticks.
If the 'inject_delay' value is positive, the buffer will be filled slower, if it is
negative - faster. You can try it yourself by starting a recording in any
audiorecording application (like Audacity) and selecting the 'pcmtest' device as a
source.
This parameter can be also used for generating a huge amount of sound data in a very
short period of time (with the negative 'inject_delay' value).
Errors injection
----------------
This module can be used for injecting errors into the PCM communication process. This
action can help you to figure out how the userspace ALSA program behaves under unusual
circumstances.
For example, you can make all 'hw_params' PCM callback calls return EBUSY error by
writing '1' to the 'inject_hwpars_err' module parameter:
.. code-block:: bash
echo 1 > /sys/module/snd_pcmtest/parameters/inject_hwpars_err
Errors can be injected into the following PCM callbacks:
* hw_params (EBUSY)
* prepare (EINVAL)
* trigger (EINVAL)
Playback test
-------------
This driver can be also used for the playback functionality testing - every time you
write the playback data to the 'pcmtest' PCM device and close it, the driver checks the
buffer for containing the looped pattern (which is specified in the fill_pattern
debugfs file for each channel). If the playback buffer content represents the looped
pattern, 'pc_test' debugfs entry is set into '1'. Otherwise, the driver sets it to '0'.
ioctl redefinition test
-----------------------
The driver redefines the 'reset' ioctl, which is default for all PCM devices. To test
this functionality, we can trigger the reset ioctl and check the 'ioctl_test' debugfs
entry:
.. code-block:: bash
cat /sys/kernel/debug/pcmtest/ioctl_test
If the ioctl is triggered successfully, this file will contain '1', and '0' otherwise.
linux-7.2.3/Documentation/sound/cards/sb-live-mixer.rst
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===========================================
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 3rd PCM device (index 2!) 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 built-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 clipped
(set to maximal or minimal value without checking for 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 D/A and A/D converters)
LFE
low frequency effects (used as subwoofer signal)
AC97
a chip containing an analog mixer, D/A and A/D 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 from 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 from 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 from 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 from 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 from 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 from 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 from left and right MIDI FX-bus
accumulator. ALSA uses accumulators 4 and 5 for left and right MIDI samples.
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 from 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 from 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 from 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 from 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 0-0x1fffd. The middle 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
-------------------------------------
Note that the site is defunct, but the documents are available
from various other locations.
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 (https://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)
linux-7.2.3/Documentation/sound/cards/serial-u16550.rst
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===================================
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.
linux-7.2.3/Documentation/sound/cards/via82xx-mixer.rst
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=============
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.