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linux-7.2.3/Documentation/sound/alsa-configuration.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.
linux-7.2.3/Documentation/sound/codecs/cs35l56.rst
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.. SPDX-License-Identifier: GPL-2.0-only
========================================================================
Audio drivers for Cirrus Logic CS35L54/56/57/63 Boosted Smart Amplifiers
========================================================================
:Copyright: 2025 Cirrus Logic, Inc. and
Cirrus Logic International Semiconductor Ltd.
Contact: patches@opensource.cirrus.com
Summary
=======
The high-level summary of this document is:
**If you have a laptop that uses CS35L54/56/57/63 amplifiers but audio is not
working, DO NOT ATTEMPT TO USE FIRMWARE AND SETTINGS FROM ANOTHER LAPTOP,
EVEN IF THAT LAPTOP SEEMS SIMILAR.**
The CS35L54/56/57/63 amplifiers must be correctly configured for the power
supply voltage, speaker impedance, maximum speaker voltage/current, and
other external hardware connections.
The amplifiers feature advanced boost technology that increases the voltage
used to drive the speakers, while proprietary speaker protection algorithms
allow these boosted amplifiers to push the limits of the speakers without
causing damage. These **must** be configured correctly.
Supported Cirrus Logic amplifiers
---------------------------------
The cs35l56 drivers support:
* CS35L54
* CS35L56
* CS35L57
* CS35L63
There are two drivers in the kernel
*For systems using SoundWire*: sound/soc/codecs/cs35l56.c and associated files
*For systems using HDA*: sound/hda/codecs/side-codecs/cs35l56_hda.c
Firmware
========
The amplifier is controlled and managed by firmware running on the internal
DSP. Firmware files are essential to enable the full capabilities of the
amplifier.
Firmware is distributed in the linux-firmware repository:
https://gitlab.com/kernel-firmware/linux-firmware.git
On most SoundWire systems the amplifier has a default minimum capability to
produce audio. However this will be
* at low volume, to protect the speakers, since the speaker specifications
and power supply voltages are unknown.
* a mono mix of left and right channels.
On some SoundWire systems that have both CS42L43 and CS35L56/57 the CS35L56/57
receive their audio from the CS42L43 instead of directly from the host
SoundWire interface. These systems can be identified by the CS42L43 showing
in dmesg as a SoundWire device, but the CS35L56/57 as SPI. On these systems
the firmware is *mandatory* to enable receiving the audio from the CS42L43.
On HDA systems the firmware is *mandatory* to enable HDA bridge mode. There
will not be any audio from the amplifiers without firmware.
Cirrus Logic firmware files
---------------------------
Each amplifier requires two firmware files. One file has a .wmfw suffix, the
other has a .bin suffix.
The firmware is customized by the OEM to match the hardware of each laptop,
and the firmware is specific to that laptop. Because of this, there are many
firmware files in linux-firmware for these amplifiers. Firmware files are
**not interchangeable between laptops**.
Cirrus Logic submits files for known laptops to the upstream linux-firmware
repository. Providing Cirrus Logic is aware of a particular laptop and has
permission from the manufacturer to publish the firmware, it will be pushed
to linux-firmware. You may need to upgrade to a newer release of
linux-firmware to obtain the firmware for your laptop.
**Important:** the Makefile for linux-firmware creates symlinks that are listed
in the WHENCE file. These symlinks are required for the CS35L56 driver to be
able to load the firmware.
How do I know which firmware file I should have?
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
All firmware file names are qualified with a unique "system ID". On normal
x86 PCs with PCI audio this is the Vendor Subsystem ID (SSID) of the host
PCI audio interface.
The SSID can be viewed using the lspci tool::
lspci -v -nn | grep -A2 -i audio
0000:00:1f.3 Audio device [0403]: Intel Corporation Meteor Lake-P HD Audio Controller [8086:7e28]
Subsystem: Dell Meteor Lake-P HD Audio Controller [1028:0c63]
In this example the SSID is 10280c63.
The format of the firmware file names is:
SoundWire:
cs35lxx-b0-dsp1-misc-SSID[-spkidX]-l?u?
SoundWire CS35L56 Rev B0 firmware released before kernel version 6.16:
cs35lxx-b0-dsp1-misc-SSID[-spkidX]-ampN
Non-SoundWire (HDA and I2S):
cs35lxx-b0-dsp1-misc-SSID[-spkidX]-ampN
Where:
* cs35lxx-b0 is the amplifier model and silicon revision. This information
is logged by the driver during initialization.
* SSID is the 8-digit hexadecimal SSID value.
* l?u? is the physical address on the SoundWire bus of the amp this
file applies to.
* ampN is the amplifier number (for example amp1). This is the same as
the prefix on the ALSA control names except that it is always lower-case
in the file name.
* spkidX is an optional part, used for laptops that have firmware
configurations for different makes and models of internal speakers.
Early firmware for CS35L56 Rev B0 used the ALSA prefix (ampN) as the
filename qualifier. Support for the l?u? qualifier was added in kernel 6.16.
Sound Open Firmware and ALSA topology files
-------------------------------------------
All SoundWire systems will require a Sound Open Firmware (SOF) for the
host CPU audio DSP, together with an ALSA topology file (.tplg).
The SOF firmware will usually be provided by the manufacturer of the host
CPU (i.e. Intel or AMD). The .tplg file is normally part of the SOF firmware
release.
SOF binary builds are available from: https://github.com/thesofproject/sof-bin/releases
The main SOF source is here: https://github.com/thesofproject
ALSA-ucm configurations
-----------------------
Typically an appropriate ALSA-ucm configuration file is needed for
use-case managers and audio servers such as PipeWire.
Configuration files are available from the alsa-ucm-conf repository:
https://git.alsa-project.org/?p=alsa-ucm-conf.git
Kernel log messages
===================
SoundWire
---------
A successful initialization will look like this (this will be repeated for
each amplifier)::
[ 7.568374] cs35l56 sdw:0:0:01fa:3556:01:0: supply VDD_P not found, using dummy regulator
[ 7.605208] cs35l56 sdw:0:0:01fa:3556:01:0: supply VDD_IO not found, using dummy regulator
[ 7.605313] cs35l56 sdw:0:0:01fa:3556:01:0: supply VDD_A not found, using dummy regulator
[ 7.939279] cs35l56 sdw:0:0:01fa:3556:01:0: Cirrus Logic CS35L56 Rev B0 OTP3 fw:3.4.4 (patched=0)
[ 7.947844] cs35l56 sdw:0:0:01fa:3556:01:0: Slave 4 state check1: UNATTACHED, status was 1
[ 8.740280] cs35l56 sdw:0:0:01fa:3556:01:0: supply VDD_B not found, using dummy regulator
[ 8.740552] cs35l56 sdw:0:0:01fa:3556:01:0: supply VDD_AMP not found, using dummy regulator
[ 9.242164] cs35l56 sdw:0:0:01fa:3556:01:0: DSP1: cirrus/cs35l56-b0-dsp1-misc-xxxxxxxx.wmfw: format 3 timestamp 0x66b2b872
[ 9.242173] cs35l56 sdw:0:0:01fa:3556:01:0: DSP1: cirrus/cs35l56-b0-dsp1-misc-xxxxxxxx.wmfw: Tue 05 Dec 2023 21:37:21 GMT Standard Time
[ 9.991709] cs35l56 sdw:0:0:01fa:3556:01:0: DSP1: Firmware: 1a00d6 vendor: 0x2 v3.11.23, 41 algorithms
[10.039098] cs35l56 sdw:0:0:01fa:3556:01:0: DSP1: cirrus/cs35l56-b0-dsp1-misc-xxxxxxxx-amp1.bin: v3.11.23
[10.879235] cs35l56 sdw:0:0:01fa:3556:01:0: Slave 4 state check1: UNATTACHED, status was 1
[11.401536] cs35l56 sdw:0:0:01fa:3556:01:0: Calibration applied
HDA
---
A successful initialization will look like this (this will be repeated for
each amplifier)::
[ 6.306475] cs35l56-hda i2c-CSC3556:00-cs35l56-hda.0: Cirrus Logic CS35L56 Rev B0 OTP3 fw:3.4.4 (patched=0)
[ 6.613892] cs35l56-hda i2c-CSC3556:00-cs35l56-hda.0: DSP system name: 'xxxxxxxx', amp name: 'AMP1'
[ 8.266660] snd_hda_codec_cs8409 ehdaudio0D0: bound i2c-CSC3556:00-cs35l56-hda.0 (ops cs35l56_hda_comp_ops [snd_hda_scodec_cs35l56])
[ 8.287525] cs35l56-hda i2c-CSC3556:00-cs35l56-hda.0: DSP1: cirrus/cs35l56-b0-dsp1-misc-xxxxxxxx.wmfw: format 3 timestamp 0x66b2b872
[ 8.287528] cs35l56-hda i2c-CSC3556:00-cs35l56-hda.0: DSP1: cirrus/cs35l56-b0-dsp1-misc-xxxxxxxx.wmfw: Tue 05 Dec 2023 21:37:21 GMT Standard Time
[ 9.984335] cs35l56-hda i2c-CSC3556:00-cs35l56-hda.0: DSP1: Firmware: 1a00d6 vendor: 0x2 v3.11.23, 41 algorithms
[10.085797] cs35l56-hda i2c-CSC3556:00-cs35l56-hda.0: DSP1: cirrus/cs35l56-b0-dsp1-misc-xxxxxxxx-amp1.bin: v3.11.23
[10.655237] cs35l56-hda i2c-CSC3556:00-cs35l56-hda.0: Calibration applied
Important messages
~~~~~~~~~~~~~~~~~~
Cirrus Logic CS35L56 Rev B0 OTP3 fw:3.4.4 (patched=0)
Shows that the driver has been able to read device ID registers from the
amplifier.
* The actual amplifier type and silicon revision (CS35L56 B0 in this
example) is shown, as read from the amplifier identification registers.
* (patched=0) is normal, and indicates that the amplifier has been hard
reset and is running default ROM firmware.
* (patched=1) means that something has previously downloaded firmware
to the amplifier and the driver does not have control of the RESET
signal to be able to replace this preloaded firmware. This is normal
for systems where the BIOS downloads firmware to the amplifiers
before OS boot.
This status can also be seen if the cs35l56 kernel module is unloaded
and reloaded on a system where the driver does not have control of
RESET. SoundWire systems typically do not give the driver control of
RESET and only a BIOS (re)boot can reset the amplifiers.
DSP1: cirrus/cs35l56-b0-dsp1-misc-xxxxxxxx.wmfw
Shows that a .wmfw firmware file was found and downloaded.
DSP1: cirrus/cs35l56-b0-dsp1-misc-xxxxxxxx-amp1.bin
Shows that a .bin firmware file was found and downloaded.
Calibration applied
Factory calibration data in EFI was written to the amplifier.
Error messages
==============
This section explains some of the error messages that the driver can log.
Algorithm coefficient version %d.%d.%d but expected %d.%d.%d
The version of the .bin file content does not match the loaded firmware.
Caused by mismatched .wmfw and .bin file, or .bin file was found but
.wmfw was not.
No %s for algorithm %x
The version of the .bin file content does not match the loaded firmware.
Caused by mismatched .wmfw and .bin file, or .bin file was found but
.wmfw was not.
.bin file required but not found
HDA driver did not find a .bin file that matches this hardware.
Calibration disabled due to missing firmware controls
Driver was not able to write EFI calibration data to firmware registers.
This typically means that either:
* The driver did not find a suitable wmfw for this hardware, or
* The amplifier has already been patched with firmware by something
previously, and the driver does not have control of a hard RESET line
to be able to reset the amplifier and download the firmware files it
found. This situation is indicated by the device identification
string in the kernel log shows "(patched=1)"
Failed to write calibration
Same meaning and cause as "Calibration disabled due to missing firmware
controls"
Failed to read calibration data from EFI
Factory calibration data in EFI is missing, empty or corrupt.
This is most likely to be cause by accidentally deleting the file from
the EFI filesystem.
No calibration for silicon ID
The factory calibration data in EFI does not match this hardware.
The most likely cause is that an amplifier has been replaced on the
motherboard without going through manufacturer calibration process to
generate calibration data for the new amplifier.
Did not find any buses for CSCxxxx
Only on HDA systems. The HDA codec driver found an ACPI entry for
Cirrus Logic companion amps, but could not enumerate the ACPI entries for
the I2C/SPI buses. The most likely cause of this is that:
* The relevant bus driver (I2C or SPI) is not part of the kernel.
* The HDA codec driver was built-in to the kernel but the I2C/SPI
bus driver is a module and so the HDA codec driver cannot call the
bus driver functions.
init_completion timed out
The SoundWire bus controller (host end) did not enumerate the amplifier.
In other words, the ACPI says there is an amplifier but for some reason
it was not detected on the bus.
No AF01 node
Indicates an error in ACPI. A SoundWire system should have a Device()
node named "AF01" but it was not found.
Failed to get spk-id-gpios
ACPI says that the driver should request a GPIO but the driver was not
able to get that GPIO. The most likely cause is that the kernel does not
include the correct GPIO or PINCTRL driver for this system.
Failed to read spk-id
ACPI says that the driver should request a GPIO but the driver was not
able to read that GPIO.
Unexpected spk-id element count
AF01 contains more speaker ID GPIO entries than the driver supports
Overtemp error
Amplifier overheat protection was triggered and the amplifier shut down
to protect itself.
Amp short error
Amplifier detected a short-circuit on the speaker output pins and shut
down for protection. This would normally indicate a damaged speaker.
Hibernate wake failed
The driver tried to wake the amplifier from its power-saving state but
did not see the expected responses from the amplifier. This can be caused
by using firmware that does not match the hardware.
linux-7.2.3/Documentation/sound/codecs/index.rst
+10
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@@ -0,0 +1,10 @@
.. SPDX-License-Identifier: GPL-2.0
Codec-Specific Information
==========================
.. toctree::
:maxdepth: 2
cs35l56
tas675x
linux-7.2.3/Documentation/sound/codecs/tas675x.rst
+821
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@@ -0,0 +1,821 @@
.. SPDX-License-Identifier: GPL-2.0
======================================
TAS675x Codec Mixer Controls
======================================
This document describes the ALSA mixer controls for the TAS675x
4-channel amplifier driver.
For device tree bindings, see:
Documentation/devicetree/bindings/sound/ti,tas67524.yaml
DSP Signal Path Mode
====================
DSP Signal Path Mode
--------------------
:Description: Signal processing mode selection.
:Type: Enumerated
:Default: Normal
:Options: Normal, LLP, FFLP
:Register: 0x32 bits [1:0]
Normal
Full DSP with all features available.
LLP (Low Latency Path)
Bypasses DSP processing. DSP protection features (Thermal Foldback,
PVDD Foldback, Clip Detect) and Real-Time Load Diagnostics unavailable.
FFLP (Full Feature Low Latency Path)
Reduced latency. Real-Time Load Diagnostics unavailable.
The following controls are unavailable in LLP mode:
``Thermal Foldback Switch``, ``PVDD Foldback Switch``,
``DC Blocker Bypass Switch``, ``Clip Detect Switch``, ``Audio SDOUT Switch``.
The following controls require Normal mode (unavailable in FFLP and LLP):
``CHx RTLDG Switch``, ``RTLDG Clip Mask Switch``, ``ISENSE Calibration Switch``,
``RTLDG Open Load Threshold``, ``RTLDG Short Load Threshold``,
``CHx RTLDG Impedance``.
Volume Controls
===============
Analog Playback Volume
----------------------
:Description: Analog output gain for all channels (CH1/CH2 and CH3/CH4 pairs).
:Type: Volume (TLV)
:Default: 0 dB
:Range: -15.5 dB to 0 dB (0.5 dB steps)
:Register: 0x4A (CH1/CH2), 0x4B (CH3/CH4)
Analog Gain Ramp Step
---------------------
:Description: Anti-pop ramp step duration for analog gain transitions.
:Type: Enumerated
:Default: 15us
:Options: 15us, 60us, 200us, 400us
:Register: 0x4E bits [3:2]
CHx Digital Playback Volume
---------------------------
:Description: Per-channel digital volume control (x = 1, 2, 3, 4).
:Type: Volume (TLV)
:Default: 0 dB
:Range: -103 dB to 0 dB (0.5 dB steps)
:Bounds: 0x30 (min/mute) to 0xFF (max)
:Register: 0x40 (CH1), 0x41 (CH2), 0x42 (CH3), 0x43 (CH4)
Volume Ramp Down Rate
---------------------
:Description: Update frequency during mute transition.
:Type: Enumerated
:Default: 16 FS
:Options: 4 FS, 16 FS, 32 FS, Instant
:Register: 0x44 bits [7:6]
Volume Ramp Down Step
---------------------
:Description: dB change per update during mute.
:Type: Enumerated
:Default: 0.5dB
:Options: 4dB, 2dB, 1dB, 0.5dB
:Register: 0x44 bits [5:4]
Volume Ramp Up Rate
-------------------
:Description: Update frequency during unmute transition.
:Type: Enumerated
:Default: 16 FS
:Options: 4 FS, 16 FS, 32 FS, Instant
:Register: 0x44 bits [3:2]
Volume Ramp Up Step
-------------------
:Description: dB change per update during unmute.
:Type: Enumerated
:Default: 0.5dB
:Options: 4dB, 2dB, 1dB, 0.5dB
:Register: 0x44 bits [1:0]
CH1/2 Volume Combine
--------------------
:Description: Links digital volume controls for CH1 and CH2.
:Type: Enumerated
:Default: Independent
:Options: Independent, CH2 follows CH1, CH1 follows CH2
:Register: 0x46 bits [1:0]
CH3/4 Volume Combine
--------------------
:Description: Links digital volume controls for CH3 and CH4.
:Type: Enumerated
:Default: Independent
:Options: Independent, CH4 follows CH3, CH3 follows CH4
:Register: 0x46 bits [3:2]
Auto Mute & Silence Detection
==============================
CHx Auto Mute Switch
--------------------
:Description: Enables automatic muting on zero-signal detection (x = 1, 2, 3, 4).
:Type: Boolean Switch
:Default: Disabled (0)
:Register: 0x47 bit [0] (CH1), bit [1] (CH2), bit [2] (CH3), bit [3] (CH4)
Auto Mute Combine Switch
------------------------
:Description: Coordinated muting behaviour across all channels.
:Type: Boolean Switch
:Default: Disabled (0)
:Behavior: Disabled: channels mute independently when their signal is zero.
Enabled: all channels mute together only when all detect zero
signal; unmute when any channel has non-zero signal.
:Register: 0x47 bit [4]
CHx Auto Mute Time
------------------
:Description: Duration of zero signal before muting triggers (x = 1, 2, 3, 4).
:Type: Enumerated
:Default: 11.5ms
:Options: 11.5ms, 53ms, 106.5ms, 266.5ms, 535ms, 1065ms, 2665ms, 5330ms
:Register: 0x48 bits [7:4] (CH1), bits [3:0] (CH2),
0x49 bits [7:4] (CH3), bits [3:0] (CH4)
:Note: Values are at 96 kHz. At 48 kHz, times are doubled.
Clock & EMI Management
======================
Spread Spectrum Mode
--------------------
:Description: Frequency dithering mode to reduce peak EMI.
:Type: Enumerated
:Default: Disabled
:Options: Disabled, Triangle, Random, Triangle and Random
:Register: 0x61 bits [1:0]
SS Triangle Range
-----------------
:Description: Frequency deviation range for Triangle spread spectrum.
:Type: Enumerated
:Default: 6.5%
:Options: 6.5%, 13.5%, 5%, 10%
:Register: 0x62 bits [1:0]
:Note: Applies only when Spread Spectrum Mode includes Triangle.
SS Random Range
---------------
:Description: Frequency deviation range for Random spread spectrum.
:Type: Enumerated
:Default: 0.83%
:Options: 0.83%, 2.50%, 5.83%, 12.50%, 25.83%
:Register: 0x62 bits [6:4]
:Note: Applies only when Spread Spectrum Mode includes Random.
SS Random Dwell Range
---------------------
:Description: Dwell time range for Random spread spectrum (FSS = spread
spectrum modulation frequency).
:Type: Enumerated
:Default: 1/FSS to 2/FSS
:Options: 1/FSS to 2/FSS, 1/FSS to 4/FSS, 1/FSS to 8/FSS, 1/FSS to 15/FSS
:Register: 0x62 bits [3:2]
:Note: Applies only when Spread Spectrum Mode includes Random.
SS Triangle Dwell Min
---------------------
:Description: Minimum dwell time at Triangle spread spectrum frequency extremes.
:Type: Integer
:Default: 0
:Range: 0 to 15 (0 = feature disabled)
:Register: 0x66 bits [7:4]
:Note: Counts in FSS clock cycles. The modulator holds the extreme
frequency for at least this many FSS cycles before reversing.
When Dwell Min equals Dwell Max, the dwell feature is inactive.
For FSS values at each PWM frequency refer to the "Spread
Spectrum" section of the TRM.
SS Triangle Dwell Max
---------------------
:Description: Maximum dwell time at Triangle spread spectrum frequency extremes.
:Type: Integer
:Default: 0
:Range: 0 to 15 (0 = feature disabled)
:Register: 0x66 bits [3:0]
:Note: Counts in FSS clock cycles. Must be >= Dwell Min. When Dwell Max
equals Dwell Min, the dwell feature is inactive.
Hardware Protection
===================
OTSD Auto Recovery Switch
--------------------------
:Description: Enables automatic recovery from over-temperature shutdown.
:Type: Boolean Switch
:Default: Disabled (0)
:Register: 0x8F bit [1]
:Note: When disabled, manual fault clearing is required after OTSD events.
Overcurrent Limit Level
-----------------------
:Description: Current-limit trip point sensitivity.
:Type: Enumerated
:Default: Level 4
:Options: Level 4, Level 3, Level 2, Level 1
:Register: 0x55 bits [1:0]
:Note: Level 4 is the least sensitive (highest trip current); Level 1 is
the most sensitive. The exact ILIM values depend on operating
conditions (PVDD voltage, switching frequency, and temperature).
Refer to the Electrical Characteristics table and the
"Overcurrent Limit (Cycle-By-Cycle)" section of the TRM.
CHx OTW Threshold
-----------------
:Description: Over-temperature warning threshold per channel (x = 1, 2, 3, 4).
:Type: Enumerated
:Default: >95C
:Options: Disabled, >95C, >110C, >125C, >135C, >145C, >155C, >165C
:Register: 0xE2 bits [6:4] (CH1), bits [2:0] (CH2),
0xE3 bits [6:4] (CH3), bits [2:0] (CH4)
Temperature and Voltage Monitoring
===================================
PVDD Sense
----------
:Description: Supply voltage sense register.
:Type: Integer (read-only)
:Range: 0 to 255
:Conversion: value × 0.19 V
:Register: 0x74
Global Temperature
------------------
:Description: Global die temperature sense register.
:Type: Integer (read-only)
:Range: 0 to 255
:Conversion: value × 2.19 K; subtract 273.15 for °C
:Register: 0x75
CHx Temperature Range
---------------------
:Description: Per-channel coarse temperature range indicator (x = 1, 2, 3, 4).
:Type: Integer (read-only)
:Range: 0 to 7
:Mapping: 0 = <95 °C, 1 = 95–110 °C, 2 = 110–125 °C, 3 = 125–135 °C,
4 = 135–145 °C, 5 = 145–155 °C, 6 = 155–165 °C, 7 = >165 °C
:Register: 0xBB bits [2:0] (CH1), bits [5:3] (CH2),
0xBC bits [2:0] (CH3), bits [5:3] (CH4)
Load Diagnostics
================
The TAS675x provides three load diagnostic modes:
DC Load Diagnostics (DC LDG)
Measures DC resistance to detect S2G (short-to-ground), S2P
(short-to-power), OL (open load), and SL (shorted load) faults.
AC Load Diagnostics (AC LDG)
Measures complex AC impedance at a configurable frequency. Detects
capacitive loads and tweeter configurations.
Real-Time Load Diagnostics (RTLDG)
Monitors impedance continuously during playback using a pilot tone.
Normal DSP mode only, at 48 kHz or 96 kHz.
Fast Boot Mode
--------------
By default the device runs DC load diagnostics at initialization before
accepting audio. Setting ``ti,fast-boot`` in the device tree bypasses this
initial diagnostic run for faster startup. Automatic diagnostics after
fault recovery remain enabled.
DC Load Diagnostics
-------------------
The ``CHx DC LDG Report`` 4-bit fault field uses the following encoding:
====== =========== ===================================================
Bit Fault Description
====== =========== ===================================================
[3] S2G Short-to-Ground
[2] S2P Short-to-Power
[1] OL Open Load
[0] SL Shorted Load
====== =========== ===================================================
DC LDG Trigger
~~~~~~~~~~~~~~
:Description: Triggers manual DC load diagnostics on all channels.
:Type: Boolean (write-only)
:Note: Returns -EBUSY if any DAI stream (playback or capture) is active.
The driver manages all channel state transitions. Blocks until
diagnostics complete or time out (300 ms).
DC LDG Auto Diagnostics Switch
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
:Description: Enables automatic DC diagnostics after fault recovery.
:Type: Boolean Switch
:Default: Enabled (1)
:Register: 0xB0 bit [0]
:Note: Active-low, when enabled, affected channels re-run diagnostics after
fault recovery and retry approximately every 750 ms until resolved.
CHx LO LDG Switch
~~~~~~~~~~~~~~~~~
:Description: Enables line output load detection per channel (x = 1, 2, 3, 4).
:Type: Boolean Switch
:Default: Disabled (0)
:Register: 0xB1 bit [3] (CH1), bit [2] (CH2), bit [1] (CH3), bit [0] (CH4)
:Note: When enabled and DC diagnostics report OL, the device tests for
a high-impedance line output load.
DC LDG SLOL Ramp Time
~~~~~~~~~~~~~~~~~~~~~
:Description: Voltage ramp time for shorted-load and open-load detection.
:Type: Enumerated
:Default: 15 ms
:Options: 15 ms, 30 ms, 10 ms, 20 ms
:Register: 0xB2 bits [7:6]
DC LDG SLOL Settling Time
~~~~~~~~~~~~~~~~~~~~~~~~~~
:Description: Settling time for shorted-load and open-load detection.
:Type: Enumerated
:Default: 10 ms
:Options: 10 ms, 5 ms, 20 ms, 15 ms
:Register: 0xB2 bits [5:4]
DC LDG S2PG Ramp Time
~~~~~~~~~~~~~~~~~~~~~
:Description: Voltage ramp time for short-to-power and short-to-ground detection.
:Type: Enumerated
:Default: 5 ms
:Options: 5 ms, 2.5 ms, 10 ms, 15 ms
:Register: 0xB2 bits [3:2]
DC LDG S2PG Settling Time
~~~~~~~~~~~~~~~~~~~~~~~~~~
:Description: Settling time for short-to-power and short-to-ground detection.
:Type: Enumerated
:Default: 10 ms
:Options: 10 ms, 5 ms, 20 ms, 30 ms
:Register: 0xB2 bits [1:0]
CHx DC LDG SL Threshold
~~~~~~~~~~~~~~~~~~~~~~~~
:Description: Shorted-load detection threshold per channel (x = 1, 2, 3, 4).
:Type: Enumerated
:Default: 1 Ohm
:Options: 0.5 Ohm, 1 Ohm, 1.5 Ohm, 2 Ohm, 2.5 Ohm,
3 Ohm, 3.5 Ohm, 4 Ohm, 4.5 Ohm, 5 Ohm
:Register: 0xB3 bits [7:4] (CH1), bits [3:0] (CH2),
0xB4 bits [7:4] (CH3), bits [3:0] (CH4)
DC LDG Result
~~~~~~~~~~~~~
:Description: Overall DC diagnostic result register.
:Type: Integer (read-only)
:Range: 0x00 to 0xFF
:Register: 0xC2
:Bit Encoding:
======== =====================================================
Bits Description
======== =====================================================
[7:4] Line output detection result, one bit per channel
[3:0] DC diagnostic pass/fail per channel (1=pass, 0=fail)
======== =====================================================
CHx DC LDG Report
~~~~~~~~~~~~~~~~~
:Description: DC diagnostic fault status per channel (x = 1, 2, 3, 4).
:Type: Integer (read-only)
:Range: 0x0 to 0xF
:Register: 0xC0 bits [7:4] (CH1), bits [3:0] (CH2),
0xC1 bits [7:4] (CH3), bits [3:0] (CH4)
:Note: See fault bit encoding table at the start of this section.
CHx LO LDG Report
~~~~~~~~~~~~~~~~~
:Description: Line output load detection result per channel (x = 1, 2, 3, 4).
:Type: Boolean (read-only)
:Values: 0 = not detected, 1 = line output load detected
:Register: 0xC2 bit [7] (CH1), bit [6] (CH2), bit [5] (CH3), bit [4] (CH4)
CHx DC Resistance
~~~~~~~~~~~~~~~~~
:Description: Measured DC load resistance per channel (x = 1, 2, 3, 4).
:Type: Float (read-only, displayed in ohms)
:Resolution: 0.1 ohm per code (10-bit value)
:Range: 0.0 to 102.3 ohms
:Register: 0xD9 bits [7:6]/[5:4]/[3:2]/[1:0] (MSB for CH1–CH4),
0xDA (CH1 LSB), 0xDB (CH2 LSB), 0xDC (CH3 LSB), 0xDD (CH4 LSB)
AC Load Diagnostics
-------------------
AC LDG Trigger
~~~~~~~~~~~~~~
:Description: Triggers AC impedance measurement on all channels.
:Type: Boolean (write-only)
:Note: Returns -EBUSY if any DAI stream (playback or capture) is active.
The driver transitions all channels to SLEEP state before starting
the measurement. Blocks until diagnostics complete or time out.
AC LDG Gain
~~~~~~~~~~~
:Description: Measurement resolution for AC diagnostics.
:Type: Boolean Switch
:Default: 1 (Gain 8)
:Values: 0 = 0.8 ohm/code (Gain 1), 1 = 0.1 ohm/code (Gain 8)
:Register: 0xB5 bit [4]
:Note: Gain 8 recommended for load impedances below 8 ohms.
AC LDG Test Frequency
~~~~~~~~~~~~~~~~~~~~~
:Description: Test signal frequency for AC impedance measurement.
:Type: Integer
:Default: 200 (0xC8 = 18.75 kHz)
:Range: 0x01 to 0xFF (0x00 reserved)
:Formula: Frequency = 93.75 Hz × register value
:Register: 0xB8
CHx AC LDG Real / CHx AC LDG Imag
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
:Description: Real and imaginary AC impedance components per channel
(x = 1, 2, 3, 4).
:Type: Integer (read-only)
:Range: 0x00 to 0xFF (8-bit signed)
:Register: 0xC3 (CH1 Real), 0xC4 (CH1 Imag), 0xC5 (CH2 Real), 0xC6 (CH2 Imag),
0xC7 (CH3 Real), 0xC8 (CH3 Imag), 0xC9 (CH4 Real), 0xCA (CH4 Imag)
:Note: Scale set by AC LDG Gain.
Speaker Protection & Detection
-------------------------------
Tweeter Detection Switch
~~~~~~~~~~~~~~~~~~~~~~~~
:Description: Enables tweeter detection using the AC impedance magnitude comparator.
:Type: Boolean Switch
:Default: Enabled (1)
:Register: 0xB6 bit [0]
:Note: The underlying register bit is TWEETER DETECT DISABLE (active-low).
Control value 1 = detection enabled (register bit 0), 0 = disabled
(register bit 1).
Tweeter Detect Threshold
~~~~~~~~~~~~~~~~~~~~~~~~~
:Description: Magnitude threshold for tweeter detection.
:Type: Integer
:Default: 0
:Range: 0x00 to 0xFF
:Resolution: 0.8 ohm/code (AC LDG Gain=0) or 0.1 ohm/code (AC LDG Gain=1)
:Register: 0xB7
CHx Tweeter Detect Report
~~~~~~~~~~~~~~~~~~~~~~~~~~
:Description: Tweeter detection result per channel (x = 1, 2, 3, 4).
:Type: Boolean (read-only)
:Values: 0 = no tweeter detected, 1 = tweeter detected
:Register: 0xCB bit [3] (CH1), bit [2] (CH2), bit [1] (CH3), bit [0] (CH4)
DSP Protection Features
=======================
These controls are unavailable in LLP mode.
Thermal Foldback Switch
-----------------------
:Description: Enables dynamic gain reduction based on die temperature.
:Type: Boolean Switch
:Default: Disabled (0)
:Register: 0x3A bit [0]
PVDD Foldback Switch
--------------------
:Description: Enables automatic gain limiting when supply voltage drops
(Automatic Gain Limiter).
:Type: Boolean Switch
:Default: Disabled (0)
:Register: 0x3A bit [4]
DC Blocker Bypass Switch
------------------------
:Description: Bypasses the DC-blocking high-pass filter.
:Type: Boolean Switch
:Default: Not bypassed (0)
:Register: 0x39 bit [0]
Clip Detect Switch
------------------
:Description: Enables DSP-based clip detection (Pseudo-Analog Clip Detect).
:Type: Boolean Switch
:Default: Disabled (0)
:Register: 0x93 bit [6]
Audio SDOUT Switch
------------------
:Description: Routes post-processed audio to the SDOUT pin instead of
Vpredict data.
:Type: Boolean Switch
:Default: Disabled (0)
:Register: 0x3A bit [5]
:Note: When enabled, replaces Vpredict data on SDOUT with post-processed
SDIN. All SDOUT configurations that apply to Vpredict also apply
to SDIN-to-SDOUT transmission.
Real-Time Load Diagnostics
===========================
These controls require Normal DSP mode at 48 kHz or 96 kHz. They are
unavailable at 192 kHz and in FFLP and LLP modes.
CHx RTLDG Switch
----------------
:Description: Enables real-time impedance monitoring during playback
(x = 1, 2, 3, 4).
:Type: Boolean Switch
:Default: Disabled (0)
:Register: 0x37 bit [3] (CH1), bit [2] (CH2), bit [1] (CH3), bit [0] (CH4)
RTLDG Clip Mask Switch
----------------------
:Description: Suppresses impedance updates during clipping events.
:Type: Boolean Switch
:Default: Enabled (1)
:Register: 0x37 bit [4]
ISENSE Calibration Switch
--------------------------
:Description: Enables current sense calibration for accurate impedance
measurements.
:Type: Boolean Switch
:Default: Disabled (0)
:Register: 0x5B bit [3]
RTLDG Open Load Threshold
--------------------------
:Description: DSP coefficient for open load fault detection threshold.
:Type: DSP coefficient (extended control)
:Register: DSP Book 0x8C, page 0x22, 0x98
RTLDG Short Load Threshold
---------------------------
:Description: DSP coefficient for shorted load fault detection threshold.
:Type: DSP coefficient (extended control)
:Register: DSP Book 0x8C, page 0x22, 0x9C
CHx RTLDG Impedance
-------------------
:Description: Real-time load impedance per channel (x = 1, 2, 3, 4).
:Type: Float (read-only, displayed in ohms)
:Register: 0xD1–0xD2 (CH1), 0xD3–0xD4 (CH2), 0xD5–0xD6 (CH3), 0xD7–0xD8 (CH4)
:Note: Valid only during PLAY state with RTLDG enabled at 48 or
96 kHz. Holds stale data in SLEEP, MUTE, or Hi-Z states.
Fault Monitoring
================
The driver monitors faults via IRQ or periodic polling (fallback when
no ``interrupts`` DT property is present). Detected faults are logged
and cleared so affected channels can recover.
Kernel log messages use the format::
tas675x <addr>: <Name> Latched: 0x<value>
Critical faults are logged at ``CRIT`` level; warnings at ``WARN``.
The following fault registers are monitored:
Critical Faults
---------------
These faults place affected channels into the FAULT state. The driver
issues fault clear (register 0x01 bit 3) to allow recovery.
Overtemperature Shutdown (0x87)
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
======== ==========================================
Bits Description
======== ==========================================
[4] Global OTSD
[3:0] CH1-CH4 OTSD respectively
======== ==========================================
Fires when die temperature exceeds the OTSD threshold. If auto-recovery
is enabled (``OTSD Auto Recovery Switch``), channels return to their
previous state after cooling. Otherwise fault clear is required.
Overcurrent / DC Fault (0x8E)
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
======== ==========================================
Bits Description
======== ==========================================
[7:4] Overcurrent Shutdown, CH1-CH4 respectively
[3:0] DC Fault, CH1-CH4 respectively
======== ==========================================
Overcurrent shutdown occurs when output current reaches the shutdown
threshold (e.g. output short to GND). DC fault fires when DC offset
exceeds the DCFAULT threshold.
Real-Time Load Diagnostic Fault (0x8B)
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
======== ==========================================
Bits Description
======== ==========================================
[7:4] Shorted Load faults, CH1-CH4 respectively
[3:0] Open Load faults, CH1-CH4 respectively
======== ==========================================
Reported during playback when RTLDG detects impedance outside the
configured open/short thresholds. Requires RTLDG and current sense to
be enabled, and a sampling rate of 48 kHz or 96 kHz.
CBC Fault/Warning (0x8D)
~~~~~~~~~~~~~~~~~~~~~~~~
======== ==========================================
Bits Description
======== ==========================================
[7:4] Load current warning, CH1-CH4 respectively
[3:0] Load current fault, CH1-CH4 respectively
======== ==========================================
Cycle-by-cycle current limiting events. A warning fires when the
current limit is active for at least 25% of a 21.3 ms window. A fault
fires if the warning persists for 170.4 ms continuously.
Warnings
--------
These conditions are logged but do not place channels into FAULT state.
Corresponding register bits are cleared when read. Nonetheless, the driver
still clears all events as warnings also latch the (mapped) FAULT pin.
Power Fault (0x86)
~~~~~~~~~~~~~~~~~~
======== ==========================================
Bits Description
======== ==========================================
[7] DVDD Power-On-Reset
[4] DVDD Undervoltage
[3] PVDD Overvoltage
[2] VBAT Overvoltage
[1] PVDD Undervoltage
[0] VBAT Undervoltage
======== ==========================================
Supply-related events. DVDD POR at bit 7 is expected after every
power-up and can be ignored.
CP / OUTM Soft Short Fault (0x7D)
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
======== ==========================================
Bits Description
======== ==========================================
[5:2] OUTM soft short, CH4-CH1 respectively
[1] Charge Pump UVLO (latched)
[0] Charge Pump UVLO (unlatched/live status)
======== ==========================================
Per-channel out_minus soft short faults and charge pump undervoltage lockout.
Clock Fault (0x8A)
~~~~~~~~~~~~~~~~~~
======== ==========================================
Bits Description
======== ==========================================
[0] Clock error
======== ==========================================
Fires on clock halt or unsupported SCLK/FSYNC ratio. Only logged when
a stream is active (suppressed during idle to avoid noise from normal
stream stop/start transitions). See `Clock Fault Behaviour`_ below.
Overtemperature Warning (0x88)
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
======== ==========================================
Bits Description
======== ==========================================
[4] Global OTW
[3:0] CH1-CH4 OTW respectively
======== ==========================================
Fires when temperature crosses the OTW threshold. The device continues
to operate. Only logged on change to avoid log spam.
Clip Warning (0x89)
~~~~~~~~~~~~~~~~~~~
======== ==========================================
Bits Description
======== ==========================================
[3:0] CH1-CH4 clip detected respectively
======== ==========================================
Fires when the audio signal magnitude exceeds the clip detection
threshold. Requires Clip Detect to be enabled. Only logged when a
stream is active.
Driver Known Limitations
========================
Clock Fault Behaviour
---------------------
On Stream Stop
~~~~~~~~~~~~~~
Every time a playback stream stops the FAULT pin briefly asserts. The
ASoC PCM trigger stop sequence calls DAIs in reverse order, so the CPU
DAI stops SCLK before the codec can transition to sleep. The device
detects the clock halt and latches ``CLK_FAULT_LATCHED``, which asserts
the FAULT pin. The driver clears the latch in the ``mute_stream``
callback that follows, so the FAULT pin flicker lasts only a few
milliseconds. Audio output is not affected and no kernel log message
is produced.
On Rapid Rate Switching
~~~~~~~~~~~~~~~~~~~~~~~
When streams are started in rapid succession, an intermittent
``Clock Fault Latched: 0x01`` message may appear in the kernel log.
A (conservative) ~0.5 second gap between sessions eliminates this.
References
==========
- TAS675x Technical Reference Manual: SLOU589A
- Device Tree Bindings: Documentation/devicetree/bindings/sound/ti,tas67524.yaml
- ALSA Control Name Conventions: Documentation/sound/designs/control-names.rst
linux-7.2.3/Documentation/sound/designs/channel-mapping-api.rst
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============================
ALSA PCM channel-mapping API
============================
Takashi Iwai <tiwai@suse.de>
General
=======
The channel mapping API allows user to query the possible channel maps
and the current channel map, also optionally to modify the channel map
of the current stream.
A channel map is an array of position for each PCM channel.
Typically, a stereo PCM stream has a channel map of
``{ front_left, front_right }``
while a 4.0 surround PCM stream has a channel map of
``{ front left, front right, rear left, rear right }.``
The problem, so far, was that we had no standard channel map
explicitly, and applications had no way to know which channel
corresponds to which (speaker) position. Thus, applications applied
wrong channels for 5.1 outputs, and you hear suddenly strange sound
from rear. Or, some devices secretly assume that center/LFE is the
third/fourth channels while others that C/LFE as 5th/6th channels.
Also, some devices such as HDMI are configurable for different speaker
positions even with the same number of total channels. However, there
was no way to specify this because of lack of channel map
specification. These are the main motivations for the new channel
mapping API.
Design
======
Actually, "the channel mapping API" doesn't introduce anything new in
the kernel/user-space ABI perspective. It uses only the existing
control element features.
As a ground design, each PCM substream may contain a control element
providing the channel mapping information and configuration. This
element is specified by:
* iface = SNDRV_CTL_ELEM_IFACE_PCM
* name = "Playback Channel Map" or "Capture Channel Map"
* device = the same device number for the assigned PCM substream
* index = the same index number for the assigned PCM substream
Note the name is different depending on the PCM substream direction.
Each control element provides at least the TLV read operation and the
read operation. Optionally, the write operation can be provided to
allow user to change the channel map dynamically.
TLV
---
The TLV operation gives the list of available channel
maps. A list item of a channel map is usually a TLV of
``type data-bytes ch0 ch1 ch2...``
where type is the TLV type value, the second argument is the total
bytes (not the numbers) of channel values, and the rest are the
position value for each channel.
As a TLV type, either ``SNDRV_CTL_TLVT_CHMAP_FIXED``,
``SNDRV_CTL_TLV_CHMAP_VAR`` or ``SNDRV_CTL_TLVT_CHMAP_PAIRED`` can be used.
The ``_FIXED`` type is for a channel map with the fixed channel position
while the latter two are for flexible channel positions. ``_VAR`` type is
for a channel map where all channels are freely swappable and ``_PAIRED``
type is where pair-wise channels are swappable. For example, when you
have {FL/FR/RL/RR} channel map, ``_PAIRED`` type would allow you to swap
only {RL/RR/FL/FR} while ``_VAR`` type would allow even swapping FL and
RR.
These new TLV types are defined in ``sound/tlv.h``.
The available channel position values are defined in ``sound/asound.h``,
here is a cut:
::
/* channel positions */
enum {
SNDRV_CHMAP_UNKNOWN = 0,
SNDRV_CHMAP_NA, /* N/A, silent */
SNDRV_CHMAP_MONO, /* mono stream */
/* this follows the alsa-lib mixer channel value + 3 */
SNDRV_CHMAP_FL, /* front left */
SNDRV_CHMAP_FR, /* front right */
SNDRV_CHMAP_RL, /* rear left */
SNDRV_CHMAP_RR, /* rear right */
SNDRV_CHMAP_FC, /* front center */
SNDRV_CHMAP_LFE, /* LFE */
SNDRV_CHMAP_SL, /* side left */
SNDRV_CHMAP_SR, /* side right */
SNDRV_CHMAP_RC, /* rear center */
/* new definitions */
SNDRV_CHMAP_FLC, /* front left center */
SNDRV_CHMAP_FRC, /* front right center */
SNDRV_CHMAP_RLC, /* rear left center */
SNDRV_CHMAP_RRC, /* rear right center */
SNDRV_CHMAP_FLW, /* front left wide */
SNDRV_CHMAP_FRW, /* front right wide */
SNDRV_CHMAP_FLH, /* front left high */
SNDRV_CHMAP_FCH, /* front center high */
SNDRV_CHMAP_FRH, /* front right high */
SNDRV_CHMAP_TC, /* top center */
SNDRV_CHMAP_TFL, /* top front left */
SNDRV_CHMAP_TFR, /* top front right */
SNDRV_CHMAP_TFC, /* top front center */
SNDRV_CHMAP_TRL, /* top rear left */
SNDRV_CHMAP_TRR, /* top rear right */
SNDRV_CHMAP_TRC, /* top rear center */
SNDRV_CHMAP_LAST = SNDRV_CHMAP_TRC,
};
When a PCM stream can provide more than one channel map, you can
provide multiple channel maps in a TLV container type. The TLV data
to be returned will contain such as:
::
SNDRV_CTL_TLVT_CONTAINER 96
SNDRV_CTL_TLVT_CHMAP_FIXED 4 SNDRV_CHMAP_FC
SNDRV_CTL_TLVT_CHMAP_FIXED 8 SNDRV_CHMAP_FL SNDRV_CHMAP_FR
SNDRV_CTL_TLVT_CHMAP_FIXED 16 NDRV_CHMAP_FL SNDRV_CHMAP_FR \
SNDRV_CHMAP_RL SNDRV_CHMAP_RR
The channel position is provided in LSB 16bits. The upper bits are
used for bit flags.
::
#define SNDRV_CHMAP_POSITION_MASK 0xffff
#define SNDRV_CHMAP_PHASE_INVERSE (0x01 << 16)
#define SNDRV_CHMAP_DRIVER_SPEC (0x02 << 16)
``SNDRV_CHMAP_PHASE_INVERSE`` indicates the channel is phase inverted,
(thus summing left and right channels would result in almost silence).
Some digital mic devices have this.
When ``SNDRV_CHMAP_DRIVER_SPEC`` is set, all the channel position values
don't follow the standard definition above but driver-specific.
Read Operation
--------------
The control read operation is for providing the current channel map of
the given stream. The control element returns an integer array
containing the position of each channel.
When this is performed before the number of the channel is specified
(i.e. hw_params is set), it should return all channels set to
``UNKNOWN``.
Write Operation
---------------
The control write operation is optional, and only for devices that can
change the channel configuration on the fly, such as HDMI. User needs
to pass an integer value containing the valid channel positions for
all channels of the assigned PCM substream.
This operation is allowed only at PCM PREPARED state. When called in
other states, it shall return an error.
linux-7.2.3/Documentation/sound/designs/compress-accel.rst
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==================================
ALSA Co-processor Acceleration API
==================================
Jaroslav Kysela <perex@perex.cz>
Overview
========
There is a requirement to expose the audio hardware that accelerates various
tasks for user space such as sample rate converters, compressed
stream decoders, etc.
This is description for the API extension for the compress ALSA API which
is able to handle "tasks" that are not bound to real-time operations
and allows for the serialization of operations.
Requirements
============
The main requirements are:
- serialization of multiple tasks for user space to allow multiple
operations without user space intervention
- separate buffers (input + output) for each operation
- expose buffers using mmap to user space
- signal user space when the task is finished (standard poll mechanism)
Design
======
A new direction SND_COMPRESS_ACCEL is introduced to identify
the passthrough API.
The API extension shares device enumeration and parameters handling from
the main compressed API. All other realtime streaming ioctls are deactivated
and a new set of task related ioctls are introduced. The standard
read/write/mmap I/O operations are not supported in the passthrough device.
Device ("stream") state handling is reduced to OPEN/SETUP. All other
states are not available for the passthrough mode.
Data I/O mechanism is using standard dma-buf interface with all advantages
like mmap, standard I/O, buffer sharing etc. One buffer is used for the
input data and second (separate) buffer is used for the output data. Each task
have separate I/O buffers.
For the buffering parameters, the fragments means a limit of allocated tasks
for given device. The fragment_size limits the input buffer size for the given
device. The output buffer size is determined by the driver (may be different
from the input buffer size).
State Machine
=============
The passthrough audio stream state machine is described below::
+----------+
| |
| OPEN |
| |
+----------+
|
|
| compr_set_params()
|
v
all passthrough task ops +----------+
+------------------------------------| |
| | SETUP |
| |
| +----------+
| |
+------------------------------------------+
Passthrough operations (ioctls)
===============================
All operations are protected using stream->device->lock (mutex).
CREATE
------
Creates a set of input/output buffers. The input buffer size is
fragment_size. Allocates unique seqno.
The hardware drivers allocate internal 'struct dma_buf' for both input and
output buffers (using 'dma_buf_export()' function). The anonymous
file descriptors for those buffers are passed to user space.
FREE
----
Free a set of input/output buffers. If a task is active, the stop
operation is executed before. If seqno is zero, operation is executed for all
tasks.
START
-----
Starts (queues) a task. There are two cases of the task start - right after
the task is created. In this case, origin_seqno must be zero.
The second case is for reusing of already finished task. The origin_seqno
must identify the task to be reused. In both cases, a new seqno value
is allocated and returned to user space.
The prerequisite is that application filled input dma buffer with
new source data and set input_size to pass the real data size to the driver.
The order of data processing is preserved (first started job must be
finished at first).
If the multiple tasks require a state handling (e.g. resampling operation),
the user space may set SND_COMPRESS_TFLG_NEW_STREAM flag to mark the
start of the new stream data. It is useful to keep the allocated buffers
for the new operation rather using open/close mechanism.
STOP
----
Stop (dequeues) a task. If seqno is zero, operation is executed for all
tasks.
STATUS
------
Obtain the task status (active, finished). Also, the driver will set
the real output data size (valid area in the output buffer).
Credits
=======
- Shengjiu Wang <shengjiu.wang@gmail.com>
- Takashi Iwai <tiwai@suse.de>
- Vinod Koul <vkoul@kernel.org>
linux-7.2.3/Documentation/sound/designs/compress-offload.rst
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=========================
ALSA Compress-Offload API
=========================
Pierre-Louis.Bossart <pierre-louis.bossart@linux.intel.com>
Vinod Koul <vinod.koul@linux.intel.com>
Overview
========
Since its early days, the ALSA API was defined with PCM support or
constant bitrates payloads such as IEC61937 in mind. Arguments and
returned values in frames are the norm, making it a challenge to
extend the existing API to compressed data streams.
In recent years, audio digital signal processors (DSP) were integrated
in system-on-chip designs, and DSPs are also integrated in audio
codecs. Processing compressed data on such DSPs results in a dramatic
reduction of power consumption compared to host-based
processing. Support for such hardware has not been very good in Linux,
mostly because of a lack of a generic API available in the mainline
kernel.
Rather than requiring a compatibility break with an API change of the
ALSA PCM interface, a new 'Compressed Data' API is introduced to
provide a control and data-streaming interface for audio DSPs.
The design of this API was inspired by the 2-year experience with the
Intel Moorestown SOC, with many corrections required to upstream the
API in the mainline kernel instead of the staging tree and make it
usable by others.
Requirements
============
The main requirements are:
- separation between byte counts and time. Compressed formats may have
a header per file, per frame, or no header at all. The payload size
may vary from frame-to-frame. As a result, it is not possible to
estimate reliably the duration of audio buffers when handling
compressed data. Dedicated mechanisms are required to allow for
reliable audio-video synchronization, which requires precise
reporting of the number of samples rendered at any given time.
- Handling of multiple formats. PCM data only requires a specification
of the sampling rate, number of channels and bits per sample. In
contrast, compressed data comes in a variety of formats. Audio DSPs
may also provide support for a limited number of audio encoders and
decoders embedded in firmware, or may support more choices through
dynamic download of libraries.
- Focus on main formats. This API provides support for the most
popular formats used for audio and video capture and playback. It is
likely that as audio compression technology advances, new formats
will be added.
- Handling of multiple configurations. Even for a given format like
AAC, some implementations may support AAC multichannel but HE-AAC
stereo. Likewise WMA10 level M3 may require too much memory and cpu
cycles. The new API needs to provide a generic way of listing these
formats.
- Rendering/Grabbing only. This API does not provide any means of
hardware acceleration, where PCM samples are provided back to
user-space for additional processing. This API focuses instead on
streaming compressed data to a DSP, with the assumption that the
decoded samples are routed to a physical output or logical back-end.
- Complexity hiding. Existing user-space multimedia frameworks all
have existing enums/structures for each compressed format. This new
API assumes the existence of a platform-specific compatibility layer
to expose, translate and make use of the capabilities of the audio
DSP, eg. Android HAL or PulseAudio sinks. By construction, regular
applications are not supposed to make use of this API.
Design
======
The new API shares a number of concepts with the PCM API for flow
control. Start, pause, resume, drain and stop commands have the same
semantics no matter what the content is.
The concept of memory ring buffer divided in a set of fragments is
borrowed from the ALSA PCM API. However, only sizes in bytes can be
specified.
Seeks/trick modes are assumed to be handled by the host.
The notion of rewinds/forwards is not supported. Data committed to the
ring buffer cannot be invalidated, except when dropping all buffers.
The Compressed Data API does not make any assumptions on how the data
is transmitted to the audio DSP. DMA transfers from main memory to an
embedded audio cluster or to a SPI interface for external DSPs are
possible. As in the ALSA PCM case, a core set of routines is exposed;
each driver implementer will have to write support for a set of
mandatory routines and possibly make use of optional ones.
The main additions are
get_caps
This routine returns the list of audio formats supported. Querying the
codecs on a capture stream will return encoders, decoders will be
listed for playback streams.
get_codec_caps
For each codec, this routine returns a list of
capabilities. The intent is to make sure all the capabilities
correspond to valid settings, and to minimize the risks of
configuration failures. For example, for a complex codec such as AAC,
the number of channels supported may depend on a specific profile. If
the capabilities were exposed with a single descriptor, it may happen
that a specific combination of profiles/channels/formats may not be
supported. Likewise, embedded DSPs have limited memory and cpu cycles,
it is likely that some implementations make the list of capabilities
dynamic and dependent on existing workloads. In addition to codec
settings, this routine returns the minimum buffer size handled by the
implementation. This information can be a function of the DMA buffer
sizes, the number of bytes required to synchronize, etc, and can be
used by userspace to define how much needs to be written in the ring
buffer before playback can start.
set_params
This routine sets the configuration chosen for a specific codec. The
most important field in the parameters is the codec type; in most
cases decoders will ignore other fields, while encoders will strictly
comply to the settings
get_params
This routines returns the actual settings used by the DSP. Changes to
the settings should remain the exception.
get_timestamp
The timestamp becomes a multiple field structure. It lists the number
of bytes transferred, the number of samples processed and the number
of samples rendered/grabbed. All these values can be used to determine
the average bitrate, figure out if the ring buffer needs to be
refilled or the delay due to decoding/encoding/io on the DSP.
Note that the list of codecs/profiles/modes was derived from the
OpenMAX AL specification instead of reinventing the wheel.
Modifications include:
- Addition of FLAC and IEC formats
- Merge of encoder/decoder capabilities
- Profiles/modes listed as bitmasks to make descriptors more compact
- Addition of set_params for decoders (missing in OpenMAX AL)
- Addition of AMR/AMR-WB encoding modes (missing in OpenMAX AL)
- Addition of format information for WMA
- Addition of encoding options when required (derived from OpenMAX IL)
- Addition of rateControlSupported (missing in OpenMAX AL)
State Machine
=============
The compressed audio stream state machine is described below ::
+----------+
| |
| OPEN |
| |
+----------+
|
|
| compr_set_params()
|
v
compr_free() +----------+
+------------------------------------| |
| | SETUP |
| +-------------------------| |<-------------------------+
| | compr_write() +----------+ |
| | ^ |
| | | compr_drain_notify() |
| | | or |
| | | compr_stop() |
| | | |
| | +----------+ |
| | | | |
| | | DRAIN | |
| | | | |
| | +----------+ |
| | ^ |
| | | |
| | | compr_drain() |
| | | |
| v | |
| +----------+ +----------+ |
| | | compr_start() | | compr_stop() |
| | PREPARE |------------------->| RUNNING |--------------------------+
| | | | | |
| +----------+ +----------+ |
| | | ^ |
| |compr_free() | | |
| | compr_pause() | | compr_resume() |
| | | | |
| v v | |
| +----------+ +----------+ |
| | | | | compr_stop() |
+--->| FREE | | PAUSE |---------------------------+
| | | |
+----------+ +----------+
Gapless Playback
================
When playing thru an album, the decoders have the ability to skip the encoder
delay and padding and directly move from one track content to another. The end
user can perceive this as gapless playback as we don't have silence while
switching from one track to another
Also, there might be low-intensity noises due to encoding. Perfect gapless is
difficult to reach with all types of compressed data, but works fine with most
music content. The decoder needs to know the encoder delay and encoder padding.
So we need to pass this to DSP. This metadata is extracted from ID3/MP4 headers
and are not present by default in the bitstream, hence the need for a new
interface to pass this information to the DSP. Also DSP and userspace needs to
switch from one track to another and start using data for second track.
The main additions are:
set_metadata
This routine sets the encoder delay and encoder padding. This can be used by
decoder to strip the silence. This needs to be set before the data in the track
is written.
set_next_track
This routine tells DSP that metadata and write operation sent after this would
correspond to subsequent track
partial drain
This is called when end of file is reached. The userspace can inform DSP that
EOF is reached and now DSP can start skipping padding delay. Also next write
data would belong to next track
Sequence flow for gapless would be:
- Open
- Get caps / codec caps
- Set params
- Set metadata of the first track
- Fill data of the first track
- Trigger start
- User-space finished sending all,
- Indicate next track data by sending set_next_track
- Set metadata of the next track
- then call partial_drain to flush most of buffer in DSP
- Fill data of the next track
- DSP switches to second track
(note: order for partial_drain and write for next track can be reversed as well)
Gapless Playback SM
===================
For Gapless, we move from running state to partial drain and back, along
with setting of meta_data and signalling for next track ::
+----------+
compr_drain_notify() | |
+------------------------>| RUNNING |
| | |
| +----------+
| |
| |
| | compr_next_track()
| |
| V
| +----------+
| compr_set_params() | |
| +-----------|NEXT_TRACK|
| | | |
| | +--+-------+
| | | |
| +--------------+ |
| |
| | compr_partial_drain()
| |
| V
| +----------+
| | |
+------------------------ | PARTIAL_ |
| DRAIN |
+----------+
Not supported
=============
- Support for VoIP/circuit-switched calls is not the target of this
API. Support for dynamic bit-rate changes would require a tight
coupling between the DSP and the host stack, limiting power savings.
- Packet-loss concealment is not supported. This would require an
additional interface to let the decoder synthesize data when frames
are lost during transmission. This may be added in the future.
- Volume control/routing is not handled by this API. Devices exposing a
compressed data interface will be considered as regular ALSA devices;
volume changes and routing information will be provided with regular
ALSA kcontrols.
- Embedded audio effects. Such effects should be enabled in the same
manner, no matter if the input was PCM or compressed.
- multichannel IEC encoding. Unclear if this is required.
- Encoding/decoding acceleration is not supported as mentioned
above. It is possible to route the output of a decoder to a capture
stream, or even implement transcoding capabilities. This routing
would be enabled with ALSA kcontrols.
- Audio policy/resource management. This API does not provide any
hooks to query the utilization of the audio DSP, nor any preemption
mechanisms.
- No notion of underrun/overrun. Since the bytes written are compressed
in nature and data written/read doesn't translate directly to
rendered output in time, this does not deal with underrun/overrun and
maybe dealt in user-library
Credits
=======
- Mark Brown and Liam Girdwood for discussions on the need for this API
- Harsha Priya for her work on intel_sst compressed API
- Rakesh Ughreja for valuable feedback
- Sing Nallasellan, Sikkandar Madar and Prasanna Samaga for
demonstrating and quantifying the benefits of audio offload on a
real platform.
linux-7.2.3/Documentation/sound/designs/control-names.rst
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===========================
Standard ALSA Control Names
===========================
This document describes standard names of mixer controls.
Standard Syntax
---------------
Syntax: [LOCATION] SOURCE [CHANNEL] [DIRECTION] FUNCTION
DIRECTION
~~~~~~~~~
================ ===============
<nothing> both directions
Playback one direction
Capture one direction
Bypass Playback one direction
Bypass Capture one direction
================ ===============
FUNCTION
~~~~~~~~
======== =================================
Switch on/off switch
Volume amplifier
Route route control, hardware specific
======== =================================
CHANNEL
~~~~~~~
============ ==================================================
<nothing> channel independent, or applies to all channels
Front front left/right channels
Surround rear left/right in 4.0/5.1 surround
CLFE C/LFE channels
Center center channel
LFE LFE channel
Side side left/right for 7.1 surround
============ ==================================================
LOCATION (Physical location of source)
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
============ =====================
Front front position
Rear rear position
Dock on docking station
Internal internal
============ =====================
SOURCE
~~~~~~
=================== =================================================
Master
Master Mono
Hardware Master
Speaker internal speaker
Bass Speaker internal LFE speaker
Headphone
Line Out
Beep beep generator
Phone
Phone Input
Phone Output
Synth
FM
Mic
Headset Mic mic part of combined headset jack - 4-pin
headphone + mic
Headphone Mic mic part of either/or - 3-pin headphone or mic
Line input only, use "Line Out" for output
CD
Video
Zoom Video
Aux
PCM
PCM Pan
Loopback
Analog Loopback D/A -> A/D loopback
Digital Loopback playback -> capture loopback -
without analog path
Mono
Mono Output
Multi
ADC
Wave
Music
I2S
IEC958
HDMI
SPDIF output only
SPDIF In
Digital In
HDMI/DP either HDMI or DisplayPort
=================== =================================================
Exceptions (deprecated)
-----------------------
===================================== =======================
[Analogue|Digital] Capture Source
[Analogue|Digital] Capture Switch aka input gain switch
[Analogue|Digital] Capture Volume aka input gain volume
[Analogue|Digital] Playback Switch aka output gain switch
[Analogue|Digital] Playback Volume aka output gain volume
Tone Control - Switch
Tone Control - Bass
Tone Control - Treble
3D Control - Switch
3D Control - Center
3D Control - Depth
3D Control - Wide
3D Control - Space
3D Control - Level
Mic Boost [(?dB)]
===================================== =======================
PCM interface
-------------
=================== ========================================
Sample Clock Source { "Word", "Internal", "AutoSync" }
Clock Sync Status { "Lock", "Sync", "No Lock" }
External Rate external capture rate
Capture Rate capture rate taken from external source
=================== ========================================
IEC958 (S/PDIF) interface
-------------------------
============================================ ======================================
IEC958 [...] [Playback|Capture] Switch turn on/off the IEC958 interface
IEC958 [...] [Playback|Capture] Volume digital volume control
IEC958 [...] [Playback|Capture] Default default or global value - read/write
IEC958 [...] [Playback|Capture] Mask consumer and professional mask
IEC958 [...] [Playback|Capture] Con Mask consumer mask
IEC958 [...] [Playback|Capture] Pro Mask professional mask
IEC958 [...] [Playback|Capture] PCM Stream the settings assigned to a PCM stream
IEC958 Q-subcode [Playback|Capture] Default Q-subcode bits
IEC958 Preamble [Playback|Capture] Default burst preamble words (4*16bits)
============================================ ======================================
linux-7.2.3/Documentation/sound/designs/index.rst
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Designs and Implementations
===========================
.. toctree::
:maxdepth: 2
control-names
channel-mapping-api
compress-accel
compress-offload
timestamping
jack-controls
tracepoints
procfile
powersave
oss-emulation
seq-oss
jack-injection
midi-2.0
linux-7.2.3/Documentation/sound/designs/jack-controls.rst
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==================
ALSA Jack Controls
==================
Why we need Jack kcontrols
==========================
ALSA uses kcontrols to export audio controls(switch, volume, Mux, ...)
to user space. This means userspace applications like pulseaudio can
switch off headphones and switch on speakers when no headphones are
plugged in.
The old ALSA jack code only created input devices for each registered
jack. These jack input devices are not readable by userspace devices
that run as non root.
The new jack code creates embedded jack kcontrols for each jack that
can be read by any process.
This can be combined with UCM to allow userspace to route audio more
intelligently based on jack insertion or removal events.
Jack Kcontrol Internals
=======================
Each jack will have a kcontrol list, so that we can create a kcontrol
and attach it to the jack, at jack creation stage. We can also add a
kcontrol to an existing jack, at anytime when required.
Those kcontrols will be freed automatically when the Jack is freed.
How to use jack kcontrols
=========================
In order to keep compatibility, snd_jack_new() has been modified by
adding two params:
initial_kctl
if true, create a kcontrol and add it to the jack list.
phantom_jack
Don't create a input device for phantom jacks.
HDA jacks can set phantom_jack to true in order to create a phantom
jack and set initial_kctl to true to create an initial kcontrol with
the correct id.
ASoC jacks should set initial_kctl as false. The pin name will be
assigned as the jack kcontrol name.
linux-7.2.3/Documentation/sound/designs/jack-injection.rst
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============================
ALSA Jack Software Injection
============================
Simple Introduction On Jack Injection
=====================================
Here jack injection means users could inject plugin or plugout events
to the audio jacks through debugfs interface, it is helpful to
validate ALSA userspace changes. For example, we change the audio
profile switching code in the pulseaudio, and we want to verify if the
change works as expected and if the change introduce the regression,
in this case, we could inject plugin or plugout events to an audio
jack or to some audio jacks, we don't need to physically access the
machine and plug/unplug physical devices to the audio jack.
In this design, an audio jack doesn't equal to a physical audio jack.
Sometimes a physical audio jack contains multi functions, and the
ALSA driver creates multi ``jack_kctl`` for a ``snd_jack``, here the
``snd_jack`` represents a physical audio jack and the ``jack_kctl``
represents a function, for example a physical jack has two functions:
headphone and mic_in, the ALSA ASoC driver will build 2 ``jack_kctl``
for this jack. The jack injection is implemented based on the
``jack_kctl`` instead of ``snd_jack``.
To inject events to audio jacks, we need to enable the jack injection
via ``sw_inject_enable`` first, once it is enabled, this jack will not
change the state by hardware events anymore, we could inject plugin or
plugout events via ``jackin_inject`` and check the jack state via
``status``, after we finish our test, we need to disable the jack
injection via ``sw_inject_enable`` too, once it is disabled, the jack
state will be restored according to the last reported hardware events
and will change by future hardware events.
The Layout of Jack Injection Interface
======================================
If users enable the SND_JACK_INJECTION_DEBUG in the kernel, the audio
jack injection interface will be created as below:
::
$debugfs_mount_dir/sound
|-- card0
|-- |-- HDMI_DP_pcm_10_Jack
|-- |-- |-- jackin_inject
|-- |-- |-- kctl_id
|-- |-- |-- mask_bits
|-- |-- |-- status
|-- |-- |-- sw_inject_enable
|-- |-- |-- type
...
|-- |-- HDMI_DP_pcm_9_Jack
|-- |-- jackin_inject
|-- |-- kctl_id
|-- |-- mask_bits
|-- |-- status
|-- |-- sw_inject_enable
|-- |-- type
|-- card1
|-- HDMI_DP_pcm_5_Jack
|-- |-- jackin_inject
|-- |-- kctl_id
|-- |-- mask_bits
|-- |-- status
|-- |-- sw_inject_enable
|-- |-- type
...
|-- Headphone_Jack
|-- |-- jackin_inject
|-- |-- kctl_id
|-- |-- mask_bits
|-- |-- status
|-- |-- sw_inject_enable
|-- |-- type
|-- Headset_Mic_Jack
|-- jackin_inject
|-- kctl_id
|-- mask_bits
|-- status
|-- sw_inject_enable
|-- type
The Explanation Of The Nodes
======================================
kctl_id
read-only, get jack_kctl->kctl's id
::
sound/card1/Headphone_Jack# cat kctl_id
Headphone Jack
mask_bits
read-only, get jack_kctl's supported events mask_bits
::
sound/card1/Headphone_Jack# cat mask_bits
0x0001 HEADPHONE(0x0001)
status
read-only, get jack_kctl's current status
- headphone unplugged:
::
sound/card1/Headphone_Jack# cat status
Unplugged
- headphone plugged:
::
sound/card1/Headphone_Jack# cat status
Plugged
type
read-only, get snd_jack's supported events from type (all supported events on the physical audio jack)
::
sound/card1/Headphone_Jack# cat type
0x7803 HEADPHONE(0x0001) MICROPHONE(0x0002) BTN_3(0x0800) BTN_2(0x1000) BTN_1(0x2000) BTN_0(0x4000)
sw_inject_enable
read-write, enable or disable injection
- injection disabled:
::
sound/card1/Headphone_Jack# cat sw_inject_enable
Jack: Headphone Jack Inject Enabled: 0
- injection enabled:
::
sound/card1/Headphone_Jack# cat sw_inject_enable
Jack: Headphone Jack Inject Enabled: 1
- to enable jack injection:
::
sound/card1/Headphone_Jack# echo 1 > sw_inject_enable
- to disable jack injection:
::
sound/card1/Headphone_Jack# echo 0 > sw_inject_enable
jackin_inject
write-only, inject plugin or plugout
- to inject plugin:
::
sound/card1/Headphone_Jack# echo 1 > jackin_inject
- to inject plugout:
::
sound/card1/Headphone_Jack# echo 0 > jackin_inject
linux-7.2.3/Documentation/sound/designs/midi-2.0.rst
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=================
MIDI 2.0 on Linux
=================
General
=======
MIDI 2.0 is an extended protocol for providing higher resolutions and
more fine controls over the legacy MIDI 1.0. The fundamental changes
introduced for supporting MIDI 2.0 are:
- Support of Universal MIDI Packet (UMP)
- Support of MIDI 2.0 protocol messages
- Transparent conversions between UMP and legacy MIDI 1.0 byte stream
- MIDI-CI for property and profile configurations
UMP is a new container format to hold all MIDI protocol 1.0 and MIDI
2.0 protocol messages. Unlike the former byte stream, it's 32bit
aligned, and each message can be put in a single packet. UMP can send
the events up to 16 "UMP Groups", where each UMP Group contain up to
16 MIDI channels.
MIDI 2.0 protocol is an extended protocol to achieve the higher
resolution and more controls over the old MIDI 1.0 protocol.
MIDI-CI is a high-level protocol that can talk with the MIDI device
for the flexible profiles and configurations. It's represented in the
form of special SysEx.
For Linux implementations, the kernel supports the UMP transport and
the encoding/decoding of MIDI protocols on UMP, while MIDI-CI is
supported in user-space over the standard SysEx.
As of this writing, only USB MIDI device supports the UMP and Linux
2.0 natively. The UMP support itself is pretty generic, hence it
could be used by other transport layers, although it could be
implemented differently (e.g. as a ALSA sequencer client), too.
The access to UMP devices are provided in two ways: the access via
rawmidi device and the access via ALSA sequencer API.
ALSA sequencer API was extended to allow the payload of UMP packets.
It's allowed to connect freely between MIDI 1.0 and MIDI 2.0 sequencer
clients, and the events are converted transparently.
Kernel Configuration
====================
The following new configs are added for supporting MIDI 2.0:
`CONFIG_SND_UMP`, `CONFIG_SND_UMP_LEGACY_RAWMIDI`,
`CONFIG_SND_SEQ_UMP`, `CONFIG_SND_SEQ_UMP_CLIENT`, and
`CONFIG_SND_USB_AUDIO_MIDI_V2`. The first visible one is
`CONFIG_SND_USB_AUDIO_MIDI_V2`, and when you choose it (to set `=y`),
the core support for UMP (`CONFIG_SND_UMP`) and the sequencer binding
(`CONFIG_SND_SEQ_UMP_CLIENT`) will be automatically selected.
Additionally, `CONFIG_SND_UMP_LEGACY_RAWMIDI=y` will enable the
support for the legacy raw MIDI device for UMP Endpoints.
Rawmidi Device with USB MIDI 2.0
================================
When a device supports MIDI 2.0, the USB-audio driver probes and uses
the MIDI 2.0 interface (that is found always at the altset 1) as
default instead of the MIDI 1.0 interface (at altset 0). You can
switch back to the binding with the old MIDI 1.0 interface by passing
`midi2_enable=0` option to snd-usb-audio driver module, too.
The USB audio driver tries to query the UMP Endpoint and UMP Function
Block information that are provided since UMP v1.1, and builds up the
topology based on those information. When the device is older and
doesn't respond to the new UMP inquiries, the driver falls back and
builds the topology based on Group Terminal Block (GTB) information
from the USB descriptor. Some device might be screwed up by the
unexpected UMP command; in such a case, pass `midi2_ump_probe=0`
option to snd-usb-audio driver for skipping the UMP v1.1 inquiries.
When the MIDI 2.0 device is probed, the kernel creates a rawmidi
device for each UMP Endpoint of the device. Its device name is
`/dev/snd/umpC*D*` and different from the standard rawmidi device name
`/dev/snd/midiC*D*` for MIDI 1.0, in order to avoid confusing the
legacy applications accessing mistakenly to UMP devices.
You can read and write UMP packet data directly from/to this UMP
rawmidi device. For example, reading via `hexdump` like below will
show the incoming UMP packets of the card 0 device 0 in the hex
format::
% hexdump -C /dev/snd/umpC0D0
00000000 01 07 b0 20 00 07 b0 20 64 3c 90 20 64 3c 80 20 |... ... d<. d<. |
Unlike the MIDI 1.0 byte stream, UMP is a 32bit packet, and the size
for reading or writing the device is also aligned to 32bit (which is 4
bytes).
The 32-bit words in the UMP packet payload are always in CPU native
endianness. Transport drivers are responsible to convert UMP words
from / to system endianness to required transport endianness / byte
order.
When `CONFIG_SND_UMP_LEGACY_RAWMIDI` is set, the driver creates
another standard raw MIDI device additionally as `/dev/snd/midiC*D*`.
This contains 16 substreams, and each substream corresponds to a
(0-based) UMP Group. Legacy applications can access to the specified
group via each substream in MIDI 1.0 byte stream format. With the
ALSA rawmidi API, you can open the arbitrary substream, while just
opening `/dev/snd/midiC*D*` will end up with opening the first
substream.
Each UMP Endpoint can provide the additional information, constructed
from the information inquired via UMP 1.1 Stream messages or USB MIDI
2.0 descriptors. And a UMP Endpoint may contain one or more UMP
Blocks, where UMP Block is an abstraction introduced in the ALSA UMP
implementations to represent the associations among UMP Groups. UMP
Block corresponds to Function Block in UMP 1.1 specification. When
UMP 1.1 Function Block information isn't available, it's filled
partially from Group Terminal Block (GTB) as defined in USB MIDI 2.0
specifications.
The information of UMP Endpoints and UMP Blocks are found in the proc
file `/proc/asound/card*/midi*`. For example::
% cat /proc/asound/card1/midi0
ProtoZOA MIDI
Type: UMP
EP Name: ProtoZOA
EP Product ID: ABCD12345678
UMP Version: 0x0000
Protocol Caps: 0x00000100
Protocol: 0x00000100
Num Blocks: 3
Block 0 (ProtoZOA Main)
Direction: bidirection
Active: Yes
Groups: 1-1
Is MIDI1: No
Block 1 (ProtoZOA Ext IN)
Direction: output
Active: Yes
Groups: 2-2
Is MIDI1: Yes (Low Speed)
....
Note that `Groups` field shown in the proc file above indicates the
1-based UMP Group numbers (from-to).
Those additional UMP Endpoint and UMP Block information can be
obtained via the new ioctls `SNDRV_UMP_IOCTL_ENDPOINT_INFO` and
`SNDRV_UMP_IOCTL_BLOCK_INFO`, respectively.
The rawmidi name and the UMP Endpoint name are usually identical, and
in the case of USB MIDI, it's taken from `iInterface` of the
corresponding USB MIDI interface descriptor. If it's not provided,
it's copied from `iProduct` of the USB device descriptor as a
fallback.
The Endpoint Product ID is a string field and supposed to be unique.
It's copied from `iSerialNumber` of the device for USB MIDI.
The protocol capabilities and the actual protocol bits are defined in
`asound.h`.
ALSA Sequencer with USB MIDI 2.0
================================
In addition to the rawmidi interfaces, ALSA sequencer interface
supports the new UMP MIDI 2.0 device, too. Now, each ALSA sequencer
client may set its MIDI version (0, 1 or 2) to declare itself being
either the legacy, UMP MIDI 1.0 or UMP MIDI 2.0 device, respectively.
The first, legacy client is the one that sends/receives the old
sequencer event as was. Meanwhile, UMP MIDI 1.0 and 2.0 clients send
and receive in the extended event record for UMP. The MIDI version is
seen in the new `midi_version` field of `snd_seq_client_info`.
A UMP packet can be sent/received in a sequencer event embedded by
specifying the new event flag bit `SNDRV_SEQ_EVENT_UMP`. When this
flag is set, the event has 16 byte (128 bit) data payload for holding
the UMP packet. Without the `SNDRV_SEQ_EVENT_UMP` bit flag, the event
is treated as a legacy event as it was (with max 12 byte data
payload).
With `SNDRV_SEQ_EVENT_UMP` flag set, the type field of a UMP sequencer
event is ignored (but it should be set to 0 as default).
The type of each client can be seen in `/proc/asound/seq/clients`.
For example::
% cat /proc/asound/seq/clients
Client info
cur clients : 3
....
Client 14 : "Midi Through" [Kernel Legacy]
Port 0 : "Midi Through Port-0" (RWe-)
Client 20 : "ProtoZOA" [Kernel UMP MIDI1]
UMP Endpoint: ProtoZOA
UMP Block 0: ProtoZOA Main [Active]
Groups: 1-1
UMP Block 1: ProtoZOA Ext IN [Active]
Groups: 2-2
UMP Block 2: ProtoZOA Ext OUT [Active]
Groups: 3-3
Port 0 : "MIDI 2.0" (RWeX) [In/Out]
Port 1 : "ProtoZOA Main" (RWeX) [In/Out]
Port 2 : "ProtoZOA Ext IN" (-We-) [Out]
Port 3 : "ProtoZOA Ext OUT" (R-e-) [In]
Here you can find two types of kernel clients, "Legacy" for client 14,
and "UMP MIDI1" for client 20, which is a USB MIDI 2.0 device.
A USB MIDI 2.0 client gives always the port 0 as "MIDI 2.0" and the
rest ports from 1 for each UMP Group (e.g. port 1 for Group 1).
In this example, the device has three active groups (Main, Ext IN and
Ext OUT), and those are exposed as sequencer ports from 1 to 3.
The "MIDI 2.0" port is for a UMP Endpoint, and its difference from
other UMP Group ports is that UMP Endpoint port sends the events from
the all ports on the device ("catch-all"), while each UMP Group port
sends only the events from the given UMP Group.
Also, UMP groupless messages (such as the UMP message type 0x0f) are
sent only to the UMP Endpoint port.
Note that, although each UMP sequencer client usually creates 16
ports, those ports that don't belong to any UMP Blocks (or belonging
to inactive UMP Blocks) are marked as inactive, and they don't appear
in the proc outputs. In the example above, the sequencer ports from 4
to 16 are present but not shown there.
The proc file above shows the UMP Block information, too. The same
entry (but with more detailed information) is found in the rawmidi
proc output.
When clients are connected between different MIDI versions, the events
are translated automatically depending on the client's version, not
only between the legacy and the UMP MIDI 1.0/2.0 types, but also
between UMP MIDI 1.0 and 2.0 types, too. For example, running
`aseqdump` program on the ProtoZOA Main port in the legacy mode will
give you the output like::
% aseqdump -p 20:1
Waiting for data. Press Ctrl+C to end.
Source Event Ch Data
20:1 Note on 0, note 60, velocity 100
20:1 Note off 0, note 60, velocity 100
20:1 Control change 0, controller 11, value 4
When you run `aseqdump` in MIDI 2.0 mode, it'll receive the high
precision data like::
% aseqdump -u 2 -p 20:1
Waiting for data. Press Ctrl+C to end.
Source Event Ch Data
20:1 Note on 0, note 60, velocity 0xc924, attr type = 0, data = 0x0
20:1 Note off 0, note 60, velocity 0xc924, attr type = 0, data = 0x0
20:1 Control change 0, controller 11, value 0x2000000
while the data is automatically converted by ALSA sequencer core.
Rawmidi API Extensions
======================
* The additional UMP Endpoint information can be obtained via the new
ioctl `SNDRV_UMP_IOCTL_ENDPOINT_INFO`. It contains the associated
card and device numbers, the bit flags, the protocols, the number of
UMP Blocks, the name string of the endpoint, etc.
The protocols are specified in two field, the protocol capabilities
and the current protocol. Both contain the bit flags specifying the
MIDI protocol version (`SNDRV_UMP_EP_INFO_PROTO_MIDI1` or
`SNDRV_UMP_EP_INFO_PROTO_MIDI2`) in the upper byte and the jitter
reduction timestamp (`SNDRV_UMP_EP_INFO_PROTO_JRTS_TX` and
`SNDRV_UMP_EP_INFO_PROTO_JRTS_RX`) in the lower byte.
A UMP Endpoint may contain up to 32 UMP Blocks, and the number of
the currently assigned blocks are shown in the Endpoint information.
* Each UMP Block information can be obtained via another new ioctl
`SNDRV_UMP_IOCTL_BLOCK_INFO`. The block ID number (0-based) has to
be passed for the block to query. The received data contains the
associated the direction of the block, the first associated group ID
(0-based) and the number of groups, the name string of the block,
etc.
The direction is either `SNDRV_UMP_DIR_INPUT`,
`SNDRV_UMP_DIR_OUTPUT` or `SNDRV_UMP_DIR_BIDIRECTION`.
* For the device supports UMP v1.1, the UMP MIDI protocol can be
switched via "Stream Configuration Request" message (UMP type 0x0f,
status 0x05). When UMP core receives such a message, it updates the
UMP EP info and the corresponding sequencer clients as well.
* The legacy rawmidi device number is found in the new `tied_device`
field of the rawmidi info.
On the other hand, the UMP rawmidi device number is found in
`tied_device` field of the legacy rawmidi info, too.
* Each substream of the legacy rawmidi may be enabled / disabled
dynamically depending on the UMP FB state.
When the selected substream is inactive, it's indicated by the bit
0x10 (`SNDRV_RAWMIDI_INFO_STREAM_INACTIVE`) in the `flags` field of
the legacy rawmidi info.
Control API Extensions
======================
* The new ioctl `SNDRV_CTL_IOCTL_UMP_NEXT_DEVICE` is introduced for
querying the next UMP rawmidi device, while the existing ioctl
`SNDRV_CTL_IOCTL_RAWMIDI_NEXT_DEVICE` queries only the legacy
rawmidi devices.
For setting the subdevice (substream number) to be opened, use the
ioctl `SNDRV_CTL_IOCTL_RAWMIDI_PREFER_SUBDEVICE` like the normal
rawmidi.
* Two new ioctls `SNDRV_CTL_IOCTL_UMP_ENDPOINT_INFO` and
`SNDRV_CTL_IOCTL_UMP_BLOCK_INFO` provide the UMP Endpoint and UMP
Block information of the specified UMP device via ALSA control API
without opening the actual (UMP) rawmidi device.
The `card` field is ignored upon inquiry, always tied with the card
of the control interface.
Sequencer API Extensions
========================
* `midi_version` field is added to `snd_seq_client_info` to indicate
the current MIDI version (either 0, 1 or 2) of each client.
When `midi_version` is 1 or 2, the alignment of read from a UMP
sequencer client is also changed from the former 28 bytes to 32
bytes for the extended payload. The alignment size for the write
isn't changed, but each event size may differ depending on the new
bit flag below.
* `SNDRV_SEQ_EVENT_UMP` flag bit is added for each sequencer event
flags. When this bit flag is set, the sequencer event is extended
to have a larger payload of 16 bytes instead of the legacy 12
bytes, and the event contains the UMP packet in the payload.
* The new sequencer port type bit (`SNDRV_SEQ_PORT_TYPE_MIDI_UMP`)
indicates the port being UMP-capable.
* The sequencer ports have new capability bits to indicate the
inactive ports (`SNDRV_SEQ_PORT_CAP_INACTIVE`) and the UMP Endpoint
port (`SNDRV_SEQ_PORT_CAP_UMP_ENDPOINT`).
* The event conversion of ALSA sequencer clients can be suppressed the
new filter bit `SNDRV_SEQ_FILTER_NO_CONVERT` set to the client info.
For example, the kernel pass-through client (`snd-seq-dummy`) sets
this flag internally.
* The port information gained the new field `direction` to indicate
the direction of the port (either `SNDRV_SEQ_PORT_DIR_INPUT`,
`SNDRV_SEQ_PORT_DIR_OUTPUT` or `SNDRV_SEQ_PORT_DIR_BIDIRECTION`).
* Another additional field for the port information is `ump_group`
which specifies the associated UMP Group Number (1-based).
When it's non-zero, the UMP group field in the UMP packet updated
upon delivery to the specified group (corrected to be 0-based).
Each sequencer port is supposed to set this field if it's a port to
specific to a certain UMP group.
* Each client may set the additional event filter for UMP Groups in
`group_filter` bitmap. The filter consists of bitmap from 1-based
Group numbers. For example, when the bit 1 is set, messages from
Group 1 (i.e. the very first group) are filtered and not delivered.
The bit 0 is used for filtering UMP groupless messages.
* Two new ioctls are added for UMP-capable clients:
`SNDRV_SEQ_IOCTL_GET_CLIENT_UMP_INFO` and
`SNDRV_SEQ_IOCTL_SET_CLIENT_UMP_INFO`. They are used to get and set
either `snd_ump_endpoint_info` or `snd_ump_block_info` data
associated with the sequencer client. The USB MIDI driver provides
those information from the underlying UMP rawmidi, while a
user-space client may provide its own data via `*_SET` ioctl.
For an Endpoint data, pass 0 to the `type` field, while for a Block
data, pass the block number + 1 to the `type` field.
Setting the data for a kernel client shall result in an error.
* With UMP 1.1, Function Block information may be changed
dynamically. When the update of Function Block is received from the
device, ALSA sequencer core changes the corresponding sequencer port
name and attributes accordingly, and notifies the changes via the
announcement to the ALSA sequencer system port, similarly like the
normal port change notification.
* There are two extended event types for notifying the UMP Endpoint and
Function Block changes via the system announcement port:
type 68 (`SNDRV_SEQ_EVENT_UMP_EP_CHANGE`) and type 69
(`SNDRV_SEQ_EVENT_UMP_BLOCK_CHANGE`). They take the new type,
`snd_seq_ev_ump_notify` in the payload, indicating the client number
and the FB number that are changed.
MIDI2 USB Gadget Function Driver
================================
The latest kernel contains the support for USB MIDI 2.0 gadget
function driver, which can be used for prototyping and debugging MIDI
2.0 features.
`CONFIG_USB_GADGET`, `CONFIG_USB_CONFIGFS` and
`CONFIG_USB_CONFIGFS_F_MIDI2` need to be enabled for the MIDI2 gadget
driver.
In addition, for using a gadget driver, you need a working UDC driver.
In the example below, we use `dummy_hcd` driver (enabled via
`CONFIG_USB_DUMMY_HCD`) that is available on PC and VM for debugging
purpose. There are other UDC drivers depending on the platform, and
those can be used for a real device, instead, too.
At first, on a system to run the gadget, load `libcomposite` module::
% modprobe libcomposite
and you'll have `usb_gadget` subdirectory under configfs space
(typically `/sys/kernel/config` on modern OS). Then create a gadget
instance and add configurations there, for example::
% cd /sys/kernel/config
% mkdir usb_gadget/g1
% cd usb_gadget/g1
% mkdir configs/c.1
% mkdir functions/midi2.usb0
% echo 0x0004 > idProduct
% echo 0x17b3 > idVendor
% mkdir strings/0x409
% echo "ACME Enterprises" > strings/0x409/manufacturer
% echo "ACMESynth" > strings/0x409/product
% echo "ABCD12345" > strings/0x409/serialnumber
% mkdir configs/c.1/strings/0x409
% echo "Monosynth" > configs/c.1/strings/0x409/configuration
% echo 120 > configs/c.1/MaxPower
At this point, there must be a subdirectory `ep.0`, and that is the
configuration for a UMP Endpoint. You can fill the Endpoint
information like::
% echo "ACMESynth" > functions/midi2.usb0/iface_name
% echo "ACMESynth" > functions/midi2.usb0/ep.0/ep_name
% echo "ABCD12345" > functions/midi2.usb0/ep.0/product_id
% echo 0x0123 > functions/midi2.usb0/ep.0/family
% echo 0x4567 > functions/midi2.usb0/ep.0/model
% echo 0x123456 > functions/midi2.usb0/ep.0/manufacturer
% echo 0x12345678 > functions/midi2.usb0/ep.0/sw_revision
The default MIDI protocol can be set either 1 or 2::
% echo 2 > functions/midi2.usb0/ep.0/protocol
And, you can find a subdirectory `block.0` under this Endpoint
subdirectory. This defines the Function Block information::
% echo "Monosynth" > functions/midi2.usb0/ep.0/block.0/name
% echo 0 > functions/midi2.usb0/ep.0/block.0/first_group
% echo 1 > functions/midi2.usb0/ep.0/block.0/num_groups
Finally, link the configuration and enable it::
% ln -s functions/midi2.usb0 configs/c.1
% echo dummy_udc.0 > UDC
where `dummy_udc.0` is an example case and it differs depending on the
system. You can find the UDC instances in `/sys/class/udc` and pass
the found name instead::
% ls /sys/class/udc
dummy_udc.0
Now, the MIDI 2.0 gadget device is enabled, and the gadget host
creates a new sound card instance containing a UMP rawmidi device by
`f_midi2` driver::
% cat /proc/asound/cards
....
1 [Gadget ]: f_midi2 - MIDI 2.0 Gadget
MIDI 2.0 Gadget
And on the connected host, a similar card should appear, too, but with
the card and device names given in the configfs above::
% cat /proc/asound/cards
....
2 [ACMESynth ]: USB-Audio - ACMESynth
ACME Enterprises ACMESynth at usb-dummy_hcd.0-1, high speed
You can play a MIDI file on the gadget side::
% aplaymidi -p 20:1 to_host.mid
and this will appear as an input from a MIDI device on the connected
host::
% aseqdump -p 20:0 -u 2
Vice versa, a playback on the connected host will work as an input on
the gadget, too.
Each Function Block may have different direction and UI-hint,
specified via `direction` and `ui_hint` attributes.
Passing `1` is for input-only, `2` for out-only and `3` for
bidirectional (the default value). For example::
% echo 2 > functions/midi2.usb0/ep.0/block.0/direction
% echo 2 > functions/midi2.usb0/ep.0/block.0/ui_hint
When you need more than one Function Blocks, you can create
subdirectories `block.1`, `block.2`, etc dynamically, and configure
them in the configuration procedure above before linking.
For example, to create a second Function Block for a keyboard::
% mkdir functions/midi2.usb0/ep.0/block.1
% echo "Keyboard" > functions/midi2.usb0/ep.0/block.1/name
% echo 1 > functions/midi2.usb0/ep.0/block.1/first_group
% echo 1 > functions/midi2.usb0/ep.0/block.1/num_groups
% echo 1 > functions/midi2.usb0/ep.0/block.1/direction
% echo 1 > functions/midi2.usb0/ep.0/block.1/ui_hint
The `block.*` subdirectories can be removed dynamically, too (except
for `block.0` which is persistent).
For assigning a Function Block for MIDI 1.0 I/O, set up in `is_midi1`
attribute. 1 is for MIDI 1.0, and 2 is for MIDI 1.0 with low speed
connection::
% echo 2 > functions/midi2.usb0/ep.0/block.1/is_midi1
For disabling the processing of UMP Stream messages in the gadget
driver, pass `0` to `process_ump` attribute in the top-level config::
% echo 0 > functions/midi2.usb0/process_ump
The MIDI 1.0 interface at altset 0 is supported by the gadget driver,
too. When MIDI 1.0 interface is selected by the connected host, the
UMP I/O on the gadget is translated from/to USB MIDI 1.0 packets
accordingly while the gadget driver keeps communicating with the
user-space over UMP rawmidi.
MIDI 1.0 ports are set up from the config in each Function Block.
For example::
% echo 0 > functions/midi2.usb0/ep.0/block.0/midi1_first_group
% echo 1 > functions/midi2.usb0/ep.0/block.0/midi1_num_groups
The configuration above will enable the Group 1 (the index 0) for MIDI
1.0 interface. Note that those groups must be in the groups defined
for the Function Block itself.
The gadget driver supports more than one UMP Endpoints, too.
Similarly like the Function Blocks, you can create a new subdirectory
`ep.1` (but under the card top-level config) to enable a new Endpoint::
% mkdir functions/midi2.usb0/ep.1
and create a new Function Block there. For example, to create 4
Groups for the Function Block of this new Endpoint::
% mkdir functions/midi2.usb0/ep.1/block.0
% echo 4 > functions/midi2.usb0/ep.1/block.0/num_groups
Now, you'll have 4 rawmidi devices in total: the first two are UMP
rawmidi devices for Endpoint 0 and Endpoint 1, and other two for the
legacy MIDI 1.0 rawmidi devices corresponding to both EP 0 and EP 1.
The current altsetting on the gadget can be informed via a control
element "Operation Mode" with `RAWMIDI` iface. e.g. you can read it
via `amixer` program running on the gadget host like::
% amixer -c1 cget iface=RAWMIDI,name='Operation Mode'
; type=INTEGER,access=r--v----,values=1,min=0,max=2,step=0
: values=2
The value (shown in the second returned line with `: values=`)
indicates 1 for MIDI 1.0 (altset 0), 2 for MIDI 2.0 (altset 1) and 0
for unset.
As of now, the configurations can't be changed after binding.
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=============================
Notes on Kernel OSS-Emulation
=============================
Jan. 22, 2004 Takashi Iwai <tiwai@suse.de>
Modules
=======
ALSA provides a powerful OSS emulation on the kernel.
The OSS emulation for PCM, mixer and sequencer devices is implemented
as add-on kernel modules, snd-pcm-oss, snd-mixer-oss and snd-seq-oss.
When you need to access the OSS PCM, mixer or sequencer devices, the
corresponding module has to be loaded.
These modules are loaded automatically when the corresponding service
is called. The alias is defined ``sound-service-x-y``, where x and y are
the card number and the minor unit number. Usually you don't have to
define these aliases by yourself.
Only necessary step for auto-loading of OSS modules is to define the
card alias in ``/etc/modprobe.d/alsa.conf``, such as::
alias sound-slot-0 snd-emu10k1
As the second card, define ``sound-slot-1`` as well.
Note that you can't use the aliased name as the target name (i.e.
``alias sound-slot-0 snd-card-0`` doesn't work any more like the old
modutils).
The currently available OSS configuration is shown in
/proc/asound/oss/sndstat. This shows in the same syntax of
/dev/sndstat, which is available on the commercial OSS driver.
On ALSA, you can symlink /dev/sndstat to this proc file.
Please note that the devices listed in this proc file appear only
after the corresponding OSS-emulation module is loaded. Don't worry
even if "NOT ENABLED IN CONFIG" is shown in it.
Device Mapping
==============
ALSA supports the following OSS device files:
::
PCM:
/dev/dspX
/dev/adspX
Mixer:
/dev/mixerX
MIDI:
/dev/midi0X
/dev/amidi0X
Sequencer:
/dev/sequencer
/dev/sequencer2 (aka /dev/music)
where X is the card number from 0 to 7.
(NOTE: Some distributions have the device files like /dev/midi0 and
/dev/midi1. They are NOT for OSS but for tclmidi, which is
a totally different thing.)
Unlike the real OSS, ALSA cannot use the device files more than the
assigned ones. For example, the first card cannot use /dev/dsp1 or
/dev/dsp2, but only /dev/dsp0 and /dev/adsp0.
As seen above, PCM and MIDI may have two devices. Usually, the first
PCM device (``hw:0,0`` in ALSA) is mapped to /dev/dsp and the secondary
device (``hw:0,1``) to /dev/adsp (if available). For MIDI, /dev/midi and
/dev/amidi, respectively.
You can change this device mapping via the module options of
snd-pcm-oss and snd-rawmidi. In the case of PCM, the following
options are available for snd-pcm-oss:
dsp_map
PCM device number assigned to /dev/dspX
(default = 0)
adsp_map
PCM device number assigned to /dev/adspX
(default = 1)
For example, to map the third PCM device (``hw:0,2``) to /dev/adsp0,
define like this:
::
options snd-pcm-oss adsp_map=2
The options take arrays. For configuring the second card, specify
two entries separated by comma. For example, to map the third PCM
device on the second card to /dev/adsp1, define like below:
::
options snd-pcm-oss adsp_map=0,2
To change the mapping of MIDI devices, the following options are
available for snd-rawmidi:
midi_map
MIDI device number assigned to /dev/midi0X
(default = 0)
amidi_map
MIDI device number assigned to /dev/amidi0X
(default = 1)
For example, to assign the third MIDI device on the first card to
/dev/midi00, define as follows:
::
options snd-rawmidi midi_map=2
PCM Mode
========
As default, ALSA emulates the OSS PCM with so-called plugin layer,
i.e. tries to convert the sample format, rate or channels
automatically when the card doesn't support it natively.
This will lead to some problems for some applications like quake or
wine, especially if they use the card only in the MMAP mode.
In such a case, you can change the behavior of PCM per application by
writing a command to the proc file. There is a proc file for each PCM
stream, ``/proc/asound/cardX/pcmY[cp]/oss``, where X is the card number
(zero-based), Y the PCM device number (zero-based), and ``p`` is for
playback and ``c`` for capture, respectively. Note that this proc file
exists only after snd-pcm-oss module is loaded.
The command sequence has the following syntax:
::
app_name fragments fragment_size [options]
``app_name`` is the name of application with (higher priority) or without
path.
``fragments`` specifies the number of fragments or zero if no specific
number is given.
``fragment_size`` is the size of fragment in bytes or zero if not given.
``options`` is the optional parameters. The following options are
available:
disable
the application tries to open a pcm device for
this channel but does not want to use it.
direct
don't use plugins
block
force block open mode
non-block
force non-block open mode
partial-frag
write also partial fragments (affects playback only)
no-silence
do not fill silence ahead to avoid clicks
The ``disable`` option is useful when one stream direction (playback or
capture) is not handled correctly by the application although the
hardware itself does support both directions.
The ``direct`` option is used, as mentioned above, to bypass the automatic
conversion and useful for MMAP-applications.
For example, to playback the first PCM device without plugins for
quake, send a command via echo like the following:
::
% echo "quake 0 0 direct" > /proc/asound/card0/pcm0p/oss
While quake wants only playback, you may append the second command
to notify driver that only this direction is about to be allocated:
::
% echo "quake 0 0 disable" > /proc/asound/card0/pcm0c/oss
The permission of proc files depend on the module options of snd.
As default it's set as root, so you'll likely need to be superuser for
sending the command above.
The block and non-block options are used to change the behavior of
opening the device file.
As default, ALSA behaves as original OSS drivers, i.e. does not block
the file when it's busy. The -EBUSY error is returned in this case.
This blocking behavior can be changed globally via nonblock_open
module option of snd-pcm-oss. For using the blocking mode as default
for OSS devices, define like the following:
::
options snd-pcm-oss nonblock_open=0
The ``partial-frag`` and ``no-silence`` commands have been added recently.
Both commands are for optimization use only. The former command
specifies to invoke the write transfer only when the whole fragment is
filled. The latter stops writing the silence data ahead
automatically. Both are disabled as default.
You can check the currently defined configuration by reading the proc
file. The read image can be sent to the proc file again, hence you
can save the current configuration
::
% cat /proc/asound/card0/pcm0p/oss > /somewhere/oss-cfg
and restore it like
::
% cat /somewhere/oss-cfg > /proc/asound/card0/pcm0p/oss
Also, for clearing all the current configuration, send ``erase`` command
as below:
::
% echo "erase" > /proc/asound/card0/pcm0p/oss
Mixer Elements
==============
Since ALSA has completely different mixer interface, the emulation of
OSS mixer is relatively complicated. ALSA builds up a mixer element
from several different ALSA (mixer) controls based on the name
string. For example, the volume element SOUND_MIXER_PCM is composed
from "PCM Playback Volume" and "PCM Playback Switch" controls for the
playback direction and from "PCM Capture Volume" and "PCM Capture
Switch" for the capture directory (if exists). When the PCM volume of
OSS is changed, all the volume and switch controls above are adjusted
automatically.
As default, ALSA uses the following control for OSS volumes:
==================== ===================== =====
OSS volume ALSA control Index
==================== ===================== =====
SOUND_MIXER_VOLUME Master 0
SOUND_MIXER_BASS Tone Control - Bass 0
SOUND_MIXER_TREBLE Tone Control - Treble 0
SOUND_MIXER_SYNTH Synth 0
SOUND_MIXER_PCM PCM 0
SOUND_MIXER_SPEAKER PC Speaker 0
SOUND_MIXER_LINE Line 0
SOUND_MIXER_MIC Mic 0
SOUND_MIXER_CD CD 0
SOUND_MIXER_IMIX Monitor Mix 0
SOUND_MIXER_ALTPCM PCM 1
SOUND_MIXER_RECLEV (not assigned)
SOUND_MIXER_IGAIN Capture 0
SOUND_MIXER_OGAIN Playback 0
SOUND_MIXER_LINE1 Aux 0
SOUND_MIXER_LINE2 Aux 1
SOUND_MIXER_LINE3 Aux 2
SOUND_MIXER_DIGITAL1 Digital 0
SOUND_MIXER_DIGITAL2 Digital 1
SOUND_MIXER_DIGITAL3 Digital 2
SOUND_MIXER_PHONEIN Phone 0
SOUND_MIXER_PHONEOUT Phone 1
SOUND_MIXER_VIDEO Video 0
SOUND_MIXER_RADIO Radio 0
SOUND_MIXER_MONITOR Monitor 0
==================== ===================== =====
The second column is the base-string of the corresponding ALSA
control. In fact, the controls with ``XXX [Playback|Capture]
[Volume|Switch]`` will be checked in addition.
The current assignment of these mixer elements is listed in the proc
file, /proc/asound/cardX/oss_mixer, which will be like the following
::
VOLUME "Master" 0
BASS "" 0
TREBLE "" 0
SYNTH "" 0
PCM "PCM" 0
...
where the first column is the OSS volume element, the second column
the base-string of the corresponding ALSA control, and the third the
control index. When the string is empty, it means that the
corresponding OSS control is not available.
For changing the assignment, you can write the configuration to this
proc file. For example, to map "Wave Playback" to the PCM volume,
send the command like the following:
::
% echo 'VOLUME "Wave Playback" 0' > /proc/asound/card0/oss_mixer
The command is exactly as same as listed in the proc file. You can
change one or more elements, one volume per line. In the last
example, both "Wave Playback Volume" and "Wave Playback Switch" will
be affected when PCM volume is changed.
Like the case of PCM proc file, the permission of proc files depend on
the module options of snd. you'll likely need to be superuser for
sending the command above.
As well as in the case of PCM proc file, you can save and restore the
current mixer configuration by reading and writing the whole file
image.
Duplex Streams
==============
Note that when attempting to use a single device file for playback and
capture, the OSS API provides no way to set the format, sample rate or
number of channels different in each direction. Thus
::
io_handle = open("device", O_RDWR)
will only function correctly if the values are the same in each direction.
To use different values in the two directions, use both
::
input_handle = open("device", O_RDONLY)
output_handle = open("device", O_WRONLY)
and set the values for the corresponding handle.
Unsupported Features
====================
MMAP on ICE1712 driver
----------------------
ICE1712 supports only the unconventional format, interleaved
10-channels 24bit (packed in 32bit) format. Therefore you cannot mmap
the buffer as the conventional (mono or 2-channels, 8 or 16bit) format
on OSS.
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==========================
Notes on Power-Saving Mode
==========================
AC97 and HD-audio drivers have the automatic power-saving mode.
This feature is enabled via Kconfig ``CONFIG_SND_AC97_POWER_SAVE``
and ``CONFIG_SND_HDA_POWER_SAVE`` options, respectively.
With the automatic power-saving, the driver turns off the codec power
appropriately when no operation is required. When no applications use
the device and/or no analog loopback is set, the power disablement is
done fully or partially. It'll save a certain power consumption, thus
good for laptops (even for desktops).
The time-out for automatic power-off can be specified via ``power_save``
module option of snd-ac97-codec and snd-hda-intel modules. Specify
the time-out value in seconds. 0 means to disable the automatic
power-saving. The default value of timeout is given via
``CONFIG_SND_AC97_POWER_SAVE_DEFAULT`` and
``CONFIG_SND_HDA_POWER_SAVE_DEFAULT`` Kconfig options. Setting this to 1
(the minimum value) isn't recommended because many applications try to
reopen the device frequently. 10 would be a good choice for normal
operations.
The ``power_save`` option is exported as writable. This means you can
adjust the value via sysfs on the fly. For example, to turn on the
automatic power-save mode with 10 seconds, write to
``/sys/module/snd_ac97_codec/parameters/power_save`` (usually as root):
::
# echo 10 > /sys/module/snd_ac97_codec/parameters/power_save
Note that you might hear click noise/pop when changing the power
state. Also, it often takes certain time to wake up from the
power-down to the active state. These are often hard to fix, so
don't report extra bug reports unless you have a fix patch ;-)
For HD-audio interface, there is another module option,
power_save_controller. This enables/disables the power-save mode of
the controller side. Setting this on may reduce a bit more power
consumption, but might result in longer wake-up time and click noise.
Try to turn it off when you experience such a thing too often.
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==========================
Proc Files of ALSA Drivers
==========================
Takashi Iwai <tiwai@suse.de>
General
=======
ALSA has its own proc tree, /proc/asound. Many useful information are
found in this tree. When you encounter a problem and need debugging,
check the files listed in the following sections.
Each card has its subtree cardX, where X is from 0 to 7. The
card-specific files are stored in the ``card*`` subdirectories.
Global Information
==================
cards
Shows the list of currently configured ALSA drivers,
index, the id string, short and long descriptions.
version
Shows the version string and compile date.
modules
Lists the module of each card
devices
Lists the ALSA native device mappings.
meminfo
Shows the status of allocated pages via ALSA drivers.
Appears only when ``CONFIG_SND_DEBUG=y``.
hwdep
Lists the currently available hwdep devices in format of
``<card>-<device>: <name>``
pcm
Lists the currently available PCM devices in format of
``<card>-<device>: <id>: <name> : <sub-streams>``
timer
Lists the currently available timer devices
oss/devices
Lists the OSS device mappings.
oss/sndstat
Provides the output compatible with /dev/sndstat.
You can symlink this to /dev/sndstat.
Card Specific Files
===================
The card-specific files are found in ``/proc/asound/card*`` directories.
Some drivers (e.g. cmipci) have their own proc entries for the
register dump, etc (e.g. ``/proc/asound/card*/cmipci`` shows the register
dump). These files would be really helpful for debugging.
When PCM devices are available on this card, you can see directories
like pcm0p or pcm1c. They hold the PCM information for each PCM
stream. The number after ``pcm`` is the PCM device number from 0, and
the last ``p`` or ``c`` means playback or capture direction. The files in
this subtree is described later.
The status of MIDI I/O is found in ``midi*`` files. It shows the device
name and the received/transmitted bytes through the MIDI device.
When the card is equipped with AC97 codecs, there are ``codec97#*``
subdirectories (described later).
When the OSS mixer emulation is enabled (and the module is loaded),
oss_mixer file appears here, too. This shows the current mapping of
OSS mixer elements to the ALSA control elements. You can change the
mapping by writing to this device. Read OSS-Emulation.txt for
details.
PCM Proc Files
==============
``card*/pcm*/info``
The general information of this PCM device: card #, device #,
substreams, etc.
``card*/pcm*/xrun_debug``
This file appears when ``CONFIG_SND_DEBUG=y`` and
``CONFIG_SND_PCM_XRUN_DEBUG=y``.
This shows the status of xrun (= buffer overrun/xrun) and
invalid PCM position debug/check of ALSA PCM middle layer.
It takes an integer value, can be changed by writing to this
file, such as::
# echo 5 > /proc/asound/card0/pcm0p/xrun_debug
The value consists of the following bit flags:
* bit 0 = Enable XRUN/jiffies debug messages
* bit 1 = Show stack trace at XRUN / jiffies check
* bit 2 = Enable additional jiffies check
When the bit 0 is set, the driver will show the messages to
kernel log when an xrun is detected. The debug message is
shown also when the invalid H/W pointer is detected at the
update of periods (usually called from the interrupt
handler).
When the bit 1 is set, the driver will show the stack trace
additionally. This may help the debugging.
Since 2.6.30, this option can enable the hwptr check using
jiffies. This detects spontaneous invalid pointer callback
values, but can be lead to too much corrections for a (mostly
buggy) hardware that doesn't give smooth pointer updates.
This feature is enabled via the bit 2.
``card*/pcm*/sub*/info``
The general information of this PCM sub-stream.
``card*/pcm*/sub*/status``
The current status of this PCM sub-stream, elapsed time,
H/W position, etc.
``card*/pcm*/sub*/hw_params``
The hardware parameters set for this sub-stream.
``card*/pcm*/sub*/sw_params``
The soft parameters set for this sub-stream.
``card*/pcm*/sub*/prealloc``
The buffer pre-allocation information.
``card*/pcm*/sub*/xrun_injection``
Triggers an XRUN to the running stream when any value is
written to this proc file. Used for fault injection.
This entry is write-only.
AC97 Codec Information
======================
``card*/codec97#*/ac97#?-?``
Shows the general information of this AC97 codec chip, such as
name, capabilities, set up.
``card*/codec97#0/ac97#?-?+regs``
Shows the AC97 register dump. Useful for debugging.
When CONFIG_SND_DEBUG is enabled, you can write to this file for
changing an AC97 register directly. Pass two hex numbers.
For example,
::
# echo 02 9f1f > /proc/asound/card0/codec97#0/ac97#0-0+regs
USB Audio Streams
=================
``card*/stream*``
Shows the assignment and the current status of each audio stream
of the given card. This information is very useful for debugging.
HD-Audio Codecs
===============
``card*/codec#*``
Shows the general codec information and the attribute of each
widget node.
``card*/eld#*``
Available for HDMI or DisplayPort interfaces.
Shows ELD(EDID Like Data) info retrieved from the attached HDMI sink,
and describes its audio capabilities and configurations.
Some ELD fields may be modified by doing ``echo name hex_value > eld#*``.
Only do this if you are sure the HDMI sink provided value is wrong.
And if that makes your HDMI audio work, please report to us so that we
can fix it in future kernel releases.
Sequencer Information
=====================
seq/drivers
Lists the currently available ALSA sequencer drivers.
seq/clients
Shows the list of currently available sequencer clients and
ports. The connection status and the running status are shown
in this file, too.
seq/queues
Lists the currently allocated/running sequencer queues.
seq/timer
Lists the currently allocated/running sequencer timers.
seq/oss
Lists the OSS-compatible sequencer stuffs.
Help For Debugging?
===================
When the problem is related with PCM, first try to turn on xrun_debug
mode. This will give you the kernel messages when and where xrun
happened.
If it's really a bug, report it with the following information:
- the name of the driver/card, show in ``/proc/asound/cards``
- the register dump, if available (e.g. ``card*/cmipci``)
when it's a PCM problem,
- set-up of PCM, shown in hw_parms, sw_params, and status in the PCM
sub-stream directory
when it's a mixer problem,
- AC97 proc files, ``codec97#*/*`` files
for USB audio/midi,
- output of ``lsusb -v``
- ``stream*`` files in card directory
The ALSA bug-tracking system is found at:
https://bugtrack.alsa-project.org/alsa-bug/
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===============================
OSS Sequencer Emulation on ALSA
===============================
Copyright (c) 1998,1999 by Takashi Iwai
ver.0.1.8; Nov. 16, 1999
Description
===========
This directory contains the OSS sequencer emulation driver on ALSA. Note
that this program is still in the development state.
What this does - it provides the emulation of the OSS sequencer, access
via ``/dev/sequencer`` and ``/dev/music`` devices.
The most of applications using OSS can run if the appropriate ALSA
sequencer is prepared.
The following features are emulated by this driver:
* Normal sequencer and MIDI events:
They are converted to the ALSA sequencer events, and sent to the
corresponding port.
* Timer events:
The timer is not selectable by ioctl. The control rate is fixed to
100 regardless of HZ. That is, even on Alpha system, a tick is always
1/100 second. The base rate and tempo can be changed in ``/dev/music``.
* Patch loading:
It purely depends on the synth drivers whether it's supported since
the patch loading is realized by callback to the synth driver.
* I/O controls:
Most of controls are accepted. Some controls
are dependent on the synth driver, as well as even on original OSS.
Furthermore, you can find the following advanced features:
* Better queue mechanism:
The events are queued before processing them.
* Multiple applications:
You can run two or more applications simultaneously (even for OSS
sequencer)!
However, each MIDI device is exclusive - that is, if a MIDI device
is opened once by some application, other applications can't use
it. No such a restriction in synth devices.
* Real-time event processing:
The events can be processed in real time without using out of bound
ioctl. To switch to real-time mode, send ABSTIME 0 event. The followed
events will be processed in real-time without queued. To switch off the
real-time mode, send RELTIME 0 event.
* ``/proc`` interface:
The status of applications and devices can be shown via
``/proc/asound/seq/oss`` at any time. In the later version,
configuration will be changed via ``/proc`` interface, too.
Installation
============
Run configure script with both sequencer support (``--with-sequencer=yes``)
and OSS emulation (``--with-oss=yes``) options. A module ``snd-seq-oss.o``
will be created. If the synth module of your sound card supports for OSS
emulation (so far, only Emu8000 driver), this module will be loaded
automatically.
Otherwise, you need to load this module manually.
At beginning, this module probes all the MIDI ports which have been
already connected to the sequencer. Once after that, the creation and deletion
of ports are watched by announcement mechanism of ALSA sequencer.
The available synth and MIDI devices can be found in proc interface.
Run ``cat /proc/asound/seq/oss``, and check the devices. For example,
if you use an AWE64 card, you'll see like the following:
::
OSS sequencer emulation version 0.1.8
ALSA client number 63
ALSA receiver port 0
Number of applications: 0
Number of synth devices: 1
synth 0: [EMU8000]
type 0x1 : subtype 0x20 : voices 32
capabilities : ioctl enabled / load_patch enabled
Number of MIDI devices: 3
midi 0: [Emu8000 Port-0] ALSA port 65:0
capability write / opened none
midi 1: [Emu8000 Port-1] ALSA port 65:1
capability write / opened none
midi 2: [0: MPU-401 (UART)] ALSA port 64:0
capability read/write / opened none
Note that the device number may be different from the information of
``/proc/asound/oss-devices`` or ones of the original OSS driver.
Use the device number listed in ``/proc/asound/seq/oss``
to play via OSS sequencer emulation.
Using Synthesizer Devices
=========================
Run your favorite program. I've tested playmidi-2.4, awemidi-0.4.3, gmod-3.1
and xmp-1.1.5. You can load samples via ``/dev/sequencer`` like sfxload,
too.
If the lowlevel driver supports multiple access to synth devices (like
Emu8000 driver), two or more applications are allowed to run at the same
time.
Using MIDI Devices
==================
So far, only MIDI output was tested. MIDI input was not checked at all,
but hopefully it will work. Use the device number listed in
``/proc/asound/seq/oss``.
Be aware that these numbers are mostly different from the list in
``/proc/asound/oss-devices``.
Module Options
==============
The following module options are available:
maxqlen
specifies the maximum read/write queue length. This queue is private
for OSS sequencer, so that it is independent from the queue length of ALSA
sequencer. Default value is 1024.
seq_oss_debug
specifies the debug level and accepts zero (= no debug message) or
positive integer. Default value is 0.
Queue Mechanism
===============
OSS sequencer emulation uses an ALSA priority queue. The
events from ``/dev/sequencer`` are processed and put onto the queue
specified by module option.
All the events from ``/dev/sequencer`` are parsed at beginning.
The timing events are also parsed at this moment, so that the events may
be processed in real-time. Sending an event ABSTIME 0 switches the operation
mode to real-time mode, and sending an event RELTIME 0 switches it off.
In the real-time mode, all events are dispatched immediately.
The queued events are dispatched to the corresponding ALSA sequencer
ports after scheduled time by ALSA sequencer dispatcher.
If the write-queue is full, the application sleeps until a certain amount
(as default one half) becomes empty in blocking mode. The synchronization
to write timing was implemented, too.
The input from MIDI devices or echo-back events are stored on read FIFO
queue. If application reads ``/dev/sequencer`` in blocking mode, the
process will be awaked.
Interface to Synthesizer Device
===============================
Registration
------------
To register an OSS synthesizer device, use snd_seq_oss_synth_register()
function:
::
int snd_seq_oss_synth_register(char *name, int type, int subtype, int nvoices,
snd_seq_oss_callback_t *oper, void *private_data)
The arguments ``name``, ``type``, ``subtype`` and ``nvoices``
are used for making the appropriate synth_info structure for ioctl. The
return value is an index number of this device. This index must be remembered
for unregister. If registration is failed, -errno will be returned.
To release this device, call snd_seq_oss_synth_unregister() function:
::
int snd_seq_oss_synth_unregister(int index)
where the ``index`` is the index number returned by register function.
Callbacks
---------
OSS synthesizer devices have capability for sample downloading and ioctls
like sample reset. In OSS emulation, these special features are realized
by using callbacks. The registration argument oper is used to specify these
callbacks. The following callback functions must be defined:
::
snd_seq_oss_callback_t:
int (*open)(snd_seq_oss_arg_t *p, void *closure);
int (*close)(snd_seq_oss_arg_t *p);
int (*ioctl)(snd_seq_oss_arg_t *p, unsigned int cmd, unsigned long arg);
int (*load_patch)(snd_seq_oss_arg_t *p, int format, const char *buf, int offs, int count);
int (*reset)(snd_seq_oss_arg_t *p);
Except for ``open`` and ``close`` callbacks, they are allowed to be NULL.
Each callback function takes the argument type ``snd_seq_oss_arg_t`` as the
first argument.
::
struct snd_seq_oss_arg_t {
int app_index;
int file_mode;
int seq_mode;
snd_seq_addr_t addr;
void *private_data;
int event_passing;
};
The first three fields, ``app_index``, ``file_mode`` and ``seq_mode``
are initialized by OSS sequencer. The ``app_index`` is the application
index which is unique to each application opening OSS sequencer. The
``file_mode`` is bit-flags indicating the file operation mode. See
``seq_oss.h`` for its meaning. The ``seq_mode`` is sequencer operation
mode. In the current version, only ``SND_OSSSEQ_MODE_SYNTH`` is used.
The next two fields, ``addr`` and ``private_data``, must be
filled by the synth driver at open callback. The ``addr`` contains
the address of ALSA sequencer port which is assigned to this device. If
the driver allocates memory for ``private_data``, it must be released
in close callback by itself.
The last field, ``event_passing``, indicates how to translate note-on
/ off events. In ``PROCESS_EVENTS`` mode, the note 255 is regarded
as velocity change, and key pressure event is passed to the port. In
``PASS_EVENTS`` mode, all note on/off events are passed to the port
without modified. ``PROCESS_KEYPRESS`` mode checks the note above 128
and regards it as key pressure event (mainly for Emu8000 driver).
Open Callback
-------------
The ``open`` is called at each time this device is opened by an application
using OSS sequencer. This must not be NULL. Typically, the open callback
does the following procedure:
#. Allocate private data record.
#. Create an ALSA sequencer port.
#. Set the new port address on ``arg->addr``.
#. Set the private data record pointer on ``arg->private_data``.
Note that the type bit-flags in port_info of this synth port must NOT contain
``TYPE_MIDI_GENERIC``
bit. Instead, ``TYPE_SPECIFIC`` should be used. Also, ``CAP_SUBSCRIPTION``
bit should NOT be included, too. This is necessary to tell it from other
normal MIDI devices. If the open procedure succeeded, return zero. Otherwise,
return -errno.
Ioctl Callback
--------------
The ``ioctl`` callback is called when the sequencer receives device-specific
ioctls. The following two ioctls should be processed by this callback:
IOCTL_SEQ_RESET_SAMPLES
reset all samples on memory -- return 0
IOCTL_SYNTH_MEMAVL
return the available memory size
FM_4OP_ENABLE
can be ignored usually
The other ioctls are processed inside the sequencer without passing to
the lowlevel driver.
Load_Patch Callback
-------------------
The ``load_patch`` callback is used for sample-downloading. This callback
must read the data on user-space and transfer to each device. Return 0
if succeeded, and -errno if failed. The format argument is the patch key
in patch_info record. The buf is user-space pointer where patch_info record
is stored. The offs can be ignored. The count is total data size of this
sample data.
Close Callback
--------------
The ``close`` callback is called when this device is closed by the
application. If any private data was allocated in open callback, it must
be released in the close callback. The deletion of ALSA port should be
done here, too. This callback must not be NULL.
Reset Callback
--------------
The ``reset`` callback is called when sequencer device is reset or
closed by applications. The callback should turn off the sounds on the
relevant port immediately, and initialize the status of the port. If this
callback is undefined, OSS seq sends a ``HEARTBEAT`` event to the
port.
Events
======
Most of the events are processed by sequencer and translated to the adequate
ALSA sequencer events, so that each synth device can receive by input_event
callback of ALSA sequencer port. The following ALSA events should be
implemented by the driver:
============= ===================
ALSA event Original OSS events
============= ===================
NOTEON SEQ_NOTEON, MIDI_NOTEON
NOTE SEQ_NOTEOFF, MIDI_NOTEOFF
KEYPRESS MIDI_KEY_PRESSURE
CHANPRESS SEQ_AFTERTOUCH, MIDI_CHN_PRESSURE
PGMCHANGE SEQ_PGMCHANGE, MIDI_PGM_CHANGE
PITCHBEND SEQ_CONTROLLER(CTRL_PITCH_BENDER),
MIDI_PITCH_BEND
CONTROLLER MIDI_CTL_CHANGE,
SEQ_BALANCE (with CTL_PAN)
CONTROL14 SEQ_CONTROLLER
REGPARAM SEQ_CONTROLLER(CTRL_PITCH_BENDER_RANGE)
SYSEX SEQ_SYSEX
============= ===================
The most of these behavior can be realized by MIDI emulation driver
included in the Emu8000 lowlevel driver. In the future release, this module
will be independent.
Some OSS events (``SEQ_PRIVATE`` and ``SEQ_VOLUME`` events) are passed as event
type SND_SEQ_OSS_PRIVATE. The OSS sequencer passes these event 8 byte
packets without any modification. The lowlevel driver should process these
events appropriately.
Interface to MIDI Device
========================
Since the OSS emulation probes the creation and deletion of ALSA MIDI
sequencer ports automatically by receiving announcement from ALSA
sequencer, the MIDI devices don't need to be registered explicitly
like synth devices.
However, the MIDI port_info registered to ALSA sequencer must include
a group name ``SND_SEQ_GROUP_DEVICE`` and a capability-bit
``CAP_READ`` or ``CAP_WRITE``. Also, subscription capabilities,
``CAP_SUBS_READ`` or ``CAP_SUBS_WRITE``, must be defined, too. If
these conditions are not satisfied, the port is not registered as OSS
sequencer MIDI device.
The events via MIDI devices are parsed in OSS sequencer and converted
to the corresponding ALSA sequencer events. The input from MIDI sequencer
is also converted to MIDI byte events by OSS sequencer. This works just
a reverse way of seq_midi module.
Known Problems / TODO's
=======================
* Patch loading via ALSA instrument layer is not implemented yet.
linux-7.2.3/Documentation/sound/designs/timestamping.rst
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=====================
ALSA PCM Timestamping
=====================
The ALSA API can provide two different system timestamps:
- Trigger_tstamp is the system time snapshot taken when the .trigger
callback is invoked. This snapshot is taken by the ALSA core in the
general case, but specific hardware may have synchronization
capabilities or conversely may only be able to provide a correct
estimate with a delay. In the latter two cases, the low-level driver
is responsible for updating the trigger_tstamp at the most appropriate
and precise moment. Applications should not rely solely on the first
trigger_tstamp but update their internal calculations if the driver
provides a refined estimate with a delay.
- tstamp is the current system timestamp updated during the last
event or application query.
The difference (tstamp - trigger_tstamp) defines the elapsed time.
The ALSA API provides two basic pieces of information, avail
and delay, which combined with the trigger and current system
timestamps allow for applications to keep track of the 'fullness' of
the ring buffer and the amount of queued samples.
The use of these different pointers and time information depends on
the application needs:
- ``avail`` reports how much can be written in the ring buffer
- ``delay`` reports the time it will take to hear a new sample after all
queued samples have been played out.
When timestamps are enabled, the avail/delay information is reported
along with a snapshot of system time. Applications can select from
``CLOCK_REALTIME`` (NTP corrections including going backwards),
``CLOCK_MONOTONIC`` (NTP corrections but never going backwards),
``CLOCK_MONOTIC_RAW`` (without NTP corrections) and change the mode
dynamically with sw_params
The ALSA API also provide an audio_tstamp which reflects the passage
of time as measured by different components of audio hardware. In
ascii-art, this could be represented as follows (for the playback
case):
::
--------------------------------------------------------------> time
^ ^ ^ ^ ^
| | | | |
analog link dma app FullBuffer
time time time time time
| | | | |
|< codec delay >|<--hw delay-->|<queued samples>|<---avail->|
|<----------------- delay---------------------->| |
|<----ring buffer length---->|
The analog time is taken at the last stage of the playback, as close
as possible to the actual transducer
The link time is taken at the output of the SoC/chipset as the samples
are pushed on a link. The link time can be directly measured if
supported in hardware by sample counters or wallclocks (e.g. with
HDAudio 24MHz or PTP clock for networked solutions) or indirectly
estimated (e.g. with the frame counter in USB).
The DMA time is measured using counters - typically the least reliable
of all measurements due to the bursty nature of DMA transfers.
The app time corresponds to the time tracked by an application after
writing in the ring buffer.
The application can query the hardware capabilities, define which
audio time it wants reported by selecting the relevant settings in
audio_tstamp_config fields, thus get an estimate of the timestamp
accuracy. It can also request the delay-to-analog be included in the
measurement. Direct access to the link time is very interesting on
platforms that provide an embedded DSP; measuring directly the link
time with dedicated hardware, possibly synchronized with system time,
removes the need to keep track of internal DSP processing times and
latency.
In case the application requests an audio tstamp that is not supported
in hardware/low-level driver, the type is overridden as DEFAULT and the
timestamp will report the DMA time based on the hw_pointer value.
For backwards compatibility with previous implementations that did not
provide timestamp selection, with a zero-valued COMPAT timestamp type
the results will default to the HDAudio wall clock for playback
streams and to the DMA time (hw_ptr) in all other cases.
The audio timestamp accuracy can be returned to user-space, so that
appropriate decisions are made:
- for dma time (default), the granularity of the transfers can be
inferred from the steps between updates and in turn provide
information on how much the application pointer can be rewound
safely.
- the link time can be used to track long-term drifts between audio
and system time using the (tstamp-trigger_tstamp)/audio_tstamp
ratio, the precision helps define how much smoothing/low-pass
filtering is required. The link time can be either reset on startup
or reported as is (the latter being useful to compare progress of
different streams - but may require the wallclock to be always
running and not wrap-around during idle periods). If supported in
hardware, the absolute link time could also be used to define a
precise start time (patches WIP)
- including the delay in the audio timestamp may
counter-intuitively not increase the precision of timestamps, e.g. if a
codec includes variable-latency DSP processing or a chain of
hardware components the delay is typically not known with precision.
The accuracy is reported in nanosecond units (using an unsigned 32-bit
word), which gives a max precision of 4.29s, more than enough for
audio applications...
Due to the varied nature of timestamping needs, even for a single
application, the audio_tstamp_config can be changed dynamically. In
the ``STATUS`` ioctl, the parameters are read-only and do not allow for
any application selection. To work around this limitation without
impacting legacy applications, a new ``STATUS_EXT`` ioctl is introduced
with read/write parameters. ALSA-lib will be modified to make use of
``STATUS_EXT`` and effectively deprecate ``STATUS``.
The ALSA API only allows for a single audio timestamp to be reported
at a time. This is a conscious design decision, reading the audio
timestamps from hardware registers or from IPC takes time, the more
timestamps are read the more imprecise the combined measurements
are. To avoid any interpretation issues, a single (system, audio)
timestamp is reported. Applications that need different timestamps
will be required to issue multiple queries and perform an
interpolation of the results
In some hardware-specific configuration, the system timestamp is
latched by a low-level audio subsystem, and the information provided
back to the driver. Due to potential delays in the communication with
the hardware, there is a risk of misalignment with the avail and delay
information. To make sure applications are not confused, a
driver_timestamp field is added in the snd_pcm_status structure; this
timestamp shows when the information is put together by the driver
before returning from the ``STATUS`` and ``STATUS_EXT`` ioctl. in most cases
this driver_timestamp will be identical to the regular system tstamp.
Examples of timestamping with HDAudio:
1. DMA timestamp, no compensation for DMA+analog delay
::
$ ./audio_time -p --ts_type=1
playback: systime: 341121338 nsec, audio time 342000000 nsec, systime delta -878662
playback: systime: 426236663 nsec, audio time 427187500 nsec, systime delta -950837
playback: systime: 597080580 nsec, audio time 598000000 nsec, systime delta -919420
playback: systime: 682059782 nsec, audio time 683020833 nsec, systime delta -961051
playback: systime: 852896415 nsec, audio time 853854166 nsec, systime delta -957751
playback: systime: 937903344 nsec, audio time 938854166 nsec, systime delta -950822
2. DMA timestamp, compensation for DMA+analog delay
::
$ ./audio_time -p --ts_type=1 -d
playback: systime: 341053347 nsec, audio time 341062500 nsec, systime delta -9153
playback: systime: 426072447 nsec, audio time 426062500 nsec, systime delta 9947
playback: systime: 596899518 nsec, audio time 596895833 nsec, systime delta 3685
playback: systime: 681915317 nsec, audio time 681916666 nsec, systime delta -1349
playback: systime: 852741306 nsec, audio time 852750000 nsec, systime delta -8694
3. link timestamp, compensation for DMA+analog delay
::
$ ./audio_time -p --ts_type=2 -d
playback: systime: 341060004 nsec, audio time 341062791 nsec, systime delta -2787
playback: systime: 426242074 nsec, audio time 426244875 nsec, systime delta -2801
playback: systime: 597080992 nsec, audio time 597084583 nsec, systime delta -3591
playback: systime: 682084512 nsec, audio time 682088291 nsec, systime delta -3779
playback: systime: 852936229 nsec, audio time 852940916 nsec, systime delta -4687
playback: systime: 938107562 nsec, audio time 938112708 nsec, systime delta -5146
Example 1 shows that the timestamp at the DMA level is close to 1ms
ahead of the actual playback time (as a side time this sort of
measurement can help define rewind safeguards). Compensating for the
DMA-link delay in example 2 helps remove the hardware buffering but
the information is still very jittery, with up to one sample of
error. In example 3 where the timestamps are measured with the link
wallclock, the timestamps show a monotonic behavior and a lower
dispersion.
Example 3 and 4 are with USB audio class. Example 3 shows a high
offset between audio time and system time due to buffering. Example 4
shows how compensating for the delay exposes a 1ms accuracy (due to
the use of the frame counter by the driver)
Example 3: DMA timestamp, no compensation for delay, delta of ~5ms
::
$ ./audio_time -p -Dhw:1 -t1
playback: systime: 120174019 nsec, audio time 125000000 nsec, systime delta -4825981
playback: systime: 245041136 nsec, audio time 250000000 nsec, systime delta -4958864
playback: systime: 370106088 nsec, audio time 375000000 nsec, systime delta -4893912
playback: systime: 495040065 nsec, audio time 500000000 nsec, systime delta -4959935
playback: systime: 620038179 nsec, audio time 625000000 nsec, systime delta -4961821
playback: systime: 745087741 nsec, audio time 750000000 nsec, systime delta -4912259
playback: systime: 870037336 nsec, audio time 875000000 nsec, systime delta -4962664
Example 4: DMA timestamp, compensation for delay, delay of ~1ms
::
$ ./audio_time -p -Dhw:1 -t1 -d
playback: systime: 120190520 nsec, audio time 120000000 nsec, systime delta 190520
playback: systime: 245036740 nsec, audio time 244000000 nsec, systime delta 1036740
playback: systime: 370034081 nsec, audio time 369000000 nsec, systime delta 1034081
playback: systime: 495159907 nsec, audio time 494000000 nsec, systime delta 1159907
playback: systime: 620098824 nsec, audio time 619000000 nsec, systime delta 1098824
playback: systime: 745031847 nsec, audio time 744000000 nsec, systime delta 1031847
linux-7.2.3/Documentation/sound/designs/tracepoints.rst
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===================
Tracepoints in ALSA
===================
2017/07/02
Takasahi Sakamoto
Tracepoints in ALSA PCM core
============================
ALSA PCM core registers ``snd_pcm`` subsystem to kernel tracepoint system.
This subsystem includes two categories of tracepoints; for state of PCM buffer
and for processing of PCM hardware parameters. These tracepoints are available
when corresponding kernel configurations are enabled. When ``CONFIG_SND_DEBUG``
is enabled, the latter tracepoints are available. When additional
``SND_PCM_XRUN_DEBUG`` is enabled too, the former trace points are enabled.
Tracepoints for state of PCM buffer
------------------------------------
This category includes four tracepoints; ``hwptr``, ``applptr``, ``xrun`` and
``hw_ptr_error``.
Tracepoints for processing of PCM hardware parameters
-----------------------------------------------------
This category includes two tracepoints; ``hw_mask_param`` and
``hw_interval_param``.
In a design of ALSA PCM core, data transmission is abstracted as PCM substream.
Applications manage PCM substream to maintain data transmission for PCM frames.
Before starting the data transmission, applications need to configure PCM
substream. In this procedure, PCM hardware parameters are decided by
interaction between applications and ALSA PCM core. Once decided, runtime of
the PCM substream keeps the parameters.
The parameters are described in struct snd_pcm_hw_params. This
structure includes several types of parameters. Applications set preferable
value to these parameters, then execute ioctl(2) with SNDRV_PCM_IOCTL_HW_REFINE
or SNDRV_PCM_IOCTL_HW_PARAMS. The former is used just for refining available
set of parameters. The latter is used for an actual decision of the parameters.
The struct snd_pcm_hw_params structure has below members:
``flags``
Configurable. ALSA PCM core and some drivers handle this flag to select
convenient parameters or change their behaviour.
``masks``
Configurable. This type of parameter is described in
struct snd_mask and represent mask values. As of PCM protocol
v2.0.13, three types are defined.
- SNDRV_PCM_HW_PARAM_ACCESS
- SNDRV_PCM_HW_PARAM_FORMAT
- SNDRV_PCM_HW_PARAM_SUBFORMAT
``intervals``
Configurable. This type of parameter is described in
struct snd_interval and represent values with a range. As of
PCM protocol v2.0.13, twelve types are defined.
- SNDRV_PCM_HW_PARAM_SAMPLE_BITS
- SNDRV_PCM_HW_PARAM_FRAME_BITS
- SNDRV_PCM_HW_PARAM_CHANNELS
- SNDRV_PCM_HW_PARAM_RATE
- SNDRV_PCM_HW_PARAM_PERIOD_TIME
- SNDRV_PCM_HW_PARAM_PERIOD_SIZE
- SNDRV_PCM_HW_PARAM_PERIOD_BYTES
- SNDRV_PCM_HW_PARAM_PERIODS
- SNDRV_PCM_HW_PARAM_BUFFER_TIME
- SNDRV_PCM_HW_PARAM_BUFFER_SIZE
- SNDRV_PCM_HW_PARAM_BUFFER_BYTES
- SNDRV_PCM_HW_PARAM_TICK_TIME
``rmask``
Configurable. This is evaluated at ioctl(2) with
SNDRV_PCM_IOCTL_HW_REFINE only. Applications can select which
mask/interval parameter can be changed by ALSA PCM core. For
SNDRV_PCM_IOCTL_HW_PARAMS, this mask is ignored and all of parameters
are going to be changed.
``cmask``
Read-only. After returning from ioctl(2), buffer in user space for
struct snd_pcm_hw_params includes result of each operation.
This mask represents which mask/interval parameter is actually changed.
``info``
Read-only. This represents hardware/driver capabilities as bit flags
with SNDRV_PCM_INFO_XXX. Typically, applications execute ioctl(2) with
SNDRV_PCM_IOCTL_HW_REFINE to retrieve this flag, then decide candidates
of parameters and execute ioctl(2) with SNDRV_PCM_IOCTL_HW_PARAMS to
configure PCM substream.
``msbits``
Read-only. This value represents available bit width in MSB side of
a PCM sample. When a parameter of SNDRV_PCM_HW_PARAM_SAMPLE_BITS was
decided as a fixed number, this value is also calculated according to
it. Else, zero. But this behaviour depends on implementations in driver
side.
``rate_num``
Read-only. This value represents numerator of sampling rate in fraction
notation. Basically, when a parameter of SNDRV_PCM_HW_PARAM_RATE was
decided as a single value, this value is also calculated according to
it. Else, zero. But this behaviour depends on implementations in driver
side.
``rate_den``
Read-only. This value represents denominator of sampling rate in
fraction notation. Basically, when a parameter of
SNDRV_PCM_HW_PARAM_RATE was decided as a single value, this value is
also calculated according to it. Else, zero. But this behaviour depends
on implementations in driver side.
``fifo_size``
Read-only. This value represents the size of FIFO in serial sound
interface of hardware. Basically, each driver can assigns a proper
value to this parameter but some drivers intentionally set zero with
a care of hardware design or data transmission protocol.
ALSA PCM core handles buffer of struct snd_pcm_hw_params when
applications execute ioctl(2) with SNDRV_PCM_HW_REFINE or SNDRV_PCM_HW_PARAMS.
Parameters in the buffer are changed according to
struct snd_pcm_hardware and rules of constraints in the runtime. The
structure describes capabilities of handled hardware. The rules describes
dependencies on which a parameter is decided according to several parameters.
A rule has a callback function, and drivers can register arbitrary functions
to compute the target parameter. ALSA PCM core registers some rules to the
runtime as a default.
Each driver can join in the interaction as long as it prepared for two stuffs
in a callback of struct snd_pcm_ops.open.
1. In the callback, drivers are expected to change a member of
struct snd_pcm_hardware type in the runtime, according to
capacities of corresponding hardware.
2. In the same callback, drivers are also expected to register additional rules
of constraints into the runtime when several parameters have dependencies
due to hardware design.
The driver can refers to result of the interaction in a callback of
struct snd_pcm_ops.hw_params, however it should not change the
content.
Tracepoints in this category are designed to trace changes of the
mask/interval parameters. When ALSA PCM core changes them, ``hw_mask_param`` or
``hw_interval_param`` event is probed according to type of the changed parameter.
ALSA PCM core also has a pretty print format for each of the tracepoints. Below
is an example for ``hw_mask_param``.
::
hw_mask_param: pcmC0D0p 001/023 FORMAT 00000000000000000000001000000044 00000000000000000000001000000044
Below is an example for ``hw_interval_param``.
::
hw_interval_param: pcmC0D0p 000/023 BUFFER_SIZE 0 0 [0 4294967295] 0 1 [0 4294967295]
The first three fields are common. They represent name of ALSA PCM character
device, rules of constraint and name of the changed parameter, in order. The
field for rules of constraint consists of two sub-fields; index of applied rule
and total number of rules added to the runtime. As an exception, the index 000
means that the parameter is changed by ALSA PCM core, regardless of the rules.
The rest of field represent state of the parameter before/after changing. These
fields are different according to type of the parameter. For parameters of mask
type, the fields represent hexadecimal dump of content of the parameter. For
parameters of interval type, the fields represent values of each member of
``empty``, ``integer``, ``openmin``, ``min``, ``max``, ``openmax`` in
struct snd_interval in this order.
Tracepoints in drivers
======================
Some drivers have tracepoints for developers' convenience. For them, please
refer to each documentation or implementation.
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======================================
HD-Audio Codec-Specific Mixer Controls
======================================
This file explains the codec-specific mixer controls.
Realtek codecs
--------------
Channel Mode
This is an enum control to change the surround-channel setup,
appears only when the surround channels are available.
It gives the number of channels to be used, "2ch", "4ch", "6ch",
and "8ch". According to the configuration, this also controls the
jack-retasking of multi-I/O jacks.
Auto-Mute Mode
This is an enum control to change the auto-mute behavior of the
headphone and line-out jacks. If built-in speakers and headphone
and/or line-out jacks are available on a machine, this controls
appears.
When there are only either headphones or line-out jacks, it gives
"Disabled" and "Enabled" state. When enabled, the speaker is muted
automatically when a jack is plugged.
When both headphone and line-out jacks are present, it gives
"Disabled", "Speaker Only" and "Line-Out+Speaker". When
speaker-only is chosen, plugging into a headphone or a line-out jack
mutes the speakers, but not line-outs. When line-out+speaker is
selected, plugging to a headphone jack mutes both speakers and
line-outs.
IDT/Sigmatel codecs
-------------------
Analog Loopback
This control enables/disables the analog-loopback circuit. This
appears only when "loopback" is set to true in a codec hint
(see HD-Audio.txt). Note that on some codecs the analog-loopback
and the normal PCM playback are exclusive, i.e. when this is on, you
won't hear any PCM stream.
Swap Center/LFE
Swaps the center and LFE channel order. Normally, the left
corresponds to the center and the right to the LFE. When this is
ON, the left to the LFE and the right to the center.
Headphone as Line Out
When this control is ON, treat the headphone jacks as line-out
jacks. That is, the headphone won't auto-mute the other line-outs,
and no HP-amp is set to the pins.
Mic Jack Mode, Line Jack Mode, etc
These enum controls the direction and the bias of the input jack
pins. Depending on the jack type, it can set as "Mic In" and "Line
In", for determining the input bias, or it can be set to "Line Out"
when the pin is a multi-I/O jack for surround channels.
VIA codecs
----------
Smart 5.1
An enum control to re-task the multi-I/O jacks for surround outputs.
When it's ON, the corresponding input jacks (usually a line-in and a
mic-in) are switched as the surround and the CLFE output jacks.
Independent HP
When this enum control is enabled, the headphone output is routed
from an individual stream (the third PCM such as hw:0,2) instead of
the primary stream. In the case the headphone DAC is shared with a
side or a CLFE-channel DAC, the DAC is switched to the headphone
automatically.
Loopback Mixing
An enum control to determine whether the analog-loopback route is
enabled or not. When it's enabled, the analog-loopback is mixed to
the front-channel. Also, the same route is used for the headphone
and speaker outputs. As a side-effect, when this mode is set, the
individual volume controls will be no longer available for
headphones and speakers because there is only one DAC connected to a
mixer widget.
Dynamic Power-Control
This control determines whether the dynamic power-control per jack
detection is enabled or not. When enabled, the widgets power state
(D0/D3) are changed dynamically depending on the jack plugging
state for saving power consumptions. However, if your system
doesn't provide a proper jack-detection, this won't work; in such a
case, turn this control OFF.
Jack Detect
This control is provided only for VT1708 codec which gives no proper
unsolicited event per jack plug. When this is on, the driver polls
the jack detection so that the headphone auto-mute can work, while
turning this off would reduce the power consumption.
Conexant codecs
---------------
Auto-Mute Mode
See Realtek codecs.
Analog codecs
--------------
Channel Mode
This is an enum control to change the surround-channel setup,
appears only when the surround channels are available.
It gives the number of channels to be used, "2ch", "4ch" and "6ch".
According to the configuration, this also controls the
jack-retasking of multi-I/O jacks.
Independent HP
When this enum control is enabled, the headphone output is routed
from an individual stream (the third PCM such as hw:0,2) instead of
the primary stream.
linux-7.2.3/Documentation/sound/hd-audio/dp-mst.rst
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=======================
HD-Audio DP-MST Support
=======================
To support DP MST audio, HD Audio hdmi codec driver introduces virtual pin
and dynamic pcm assignment.
Virtual pin is an extension of per_pin. The most difference of DP MST
from legacy is that DP MST introduces device entry. Each pin can contain
several device entries. Each device entry behaves as a pin.
As each pin may contain several device entries and each codec may contain
several pins, if we use one pcm per per_pin, there will be many PCMs.
The new solution is to create a few PCMs and to dynamically bind pcm to
per_pin. Driver uses spec->dyn_pcm_assign flag to indicate whether to use
the new solution.
PCM
===
To be added
Pin Initialization
==================
Each pin may have several device entries (virtual pins). On Intel platform,
the device entries number is dynamically changed. If DP MST hub is connected,
it is in DP MST mode, and the device entries number is 3. Otherwise, the
device entries number is 1.
To simplify the implementation, all the device entries will be initialized
when bootup no matter whether it is in DP MST mode or not.
Connection list
===============
DP MST reuses connection list code. The code can be reused because
device entries on the same pin have the same connection list.
This means DP MST gets the device entry connection list without the
device entry setting.
Jack
====
Presume:
- MST must be dyn_pcm_assign, and it is acomp (for Intel scenario);
- NON-MST may or may not be dyn_pcm_assign, it can be acomp or !acomp;
So there are the following scenarios:
a. MST (&& dyn_pcm_assign && acomp)
b. NON-MST && dyn_pcm_assign && acomp
c. NON-MST && !dyn_pcm_assign && !acomp
Below discussion will ignore MST and NON-MST difference as it doesn't
impact on jack handling too much.
Driver uses struct hdmi_pcm pcm[] array in hdmi_spec and snd_jack is
a member of hdmi_pcm. Each pin has one struct hdmi_pcm * pcm pointer.
For !dyn_pcm_assign, per_pin->pcm will assigned to spec->pcm[n] statically.
For dyn_pcm_assign, per_pin->pcm will assigned to spec->pcm[n]
when monitor is hotplugged.
Build Jack
----------
- dyn_pcm_assign
Will not use hda_jack but use snd_jack in spec->pcm_rec[pcm_idx].jack directly.
- !dyn_pcm_assign
Use hda_jack and assign spec->pcm_rec[pcm_idx].jack = jack->jack statically.
Unsolicited Event Enabling
--------------------------
Enable unsolicited event if !acomp.
Monitor Hotplug Event Handling
------------------------------
- acomp
pin_eld_notify() -> check_presence_and_report() -> hdmi_present_sense() ->
sync_eld_via_acomp().
Use directly snd_jack_report() on spec->pcm_rec[pcm_idx].jack for
both dyn_pcm_assign and !dyn_pcm_assign
- !acomp
hdmi_unsol_event() -> hdmi_intrinsic_event() -> check_presence_and_report() ->
hdmi_present_sense() -> hdmi_prepsent_sense_via_verbs()
Use directly snd_jack_report() on spec->pcm_rec[pcm_idx].jack for dyn_pcm_assign.
Use hda_jack mechanism to handle jack events.
Others to be added later
========================
linux-7.2.3/Documentation/sound/hd-audio/index.rst
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HD-Audio
========
.. toctree::
:maxdepth: 2
notes
models
controls
dp-mst
realtek-pc-beep
intel-multi-link
linux-7.2.3/Documentation/sound/hd-audio/intel-multi-link.rst
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.. SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause)
.. include:: <isonum.txt>
================================================
HDAudio multi-link extensions on Intel platforms
================================================
:Copyright: |copy| 2023 Intel Corporation
This file documents the 'multi-link structure' introduced in 2015 with
the Skylake processor and recently extended in newer Intel platforms
HDaudio existing link mapping (2015 addition in SkyLake)
========================================================
External HDAudio codecs are handled with link #0, while iDISP codec
for HDMI/DisplayPort is handled with link #1.
The only change to the 2015 definitions is the declaration of the
LCAP.ALT=0x0 - since the ALT bit was previously reserved, this is a
backwards-compatible change.
LCTL.SPA and LCTL.CPA are automatically set when exiting reset. They
are only used in existing drivers when the SCF value needs to be
corrected.
Basic structure for HDaudio codecs
----------------------------------
::
+-----------+
| ML cap #0 |
+-----------+
| ML cap #1 |---+
+-----------+ |
|
+--> 0x0 +---------------+ LCAP
| ALT=0 |
+---------------+
| S192 |
+---------------+
| S96 |
+---------------+
| S48 |
+---------------+
| S24 |
+---------------+
| S12 |
+---------------+
| S6 |
+---------------+
0x4 +---------------+ LCTL
| INTSTS |
+---------------+
| CPA |
+---------------+
| SPA |
+---------------+
| SCF |
+---------------+
0x8 +---------------+ LOSIDV
| L1OSIVD15 |
+---------------+
| L1OSIDV.. |
+---------------+
| L1OSIDV1 |
+---------------+
0xC +---------------+ LSDIID
| SDIID14 |
+---------------+
| SDIID... |
+---------------+
| SDIID0 |
+---------------+
SoundWire HDaudio extended link mapping
=======================================
A SoundWire extended link is identified when LCAP.ALT=1 and
LEPTR.ID=0.
DMA control uses the existing LOSIDV register.
Changes include additional descriptions for enumeration that were not
present in earlier generations.
- multi-link synchronization: capabilities in LCAP.LSS and control in LSYNC
- number of sublinks (manager IP) in LCAP.LSCOUNT
- power management moved from SHIM to LCTL.SPA bits
- hand-over to the DSP for access to multi-link registers, SHIM/IP with LCTL.OFLEN
- mapping of SoundWire codecs to SDI ID bits
- move of SHIM and Cadence registers to different offsets, with no
change in functionality. The LEPTR.PTR value is an offset from the
ML address, with a default value of 0x30000.
Extended structure for SoundWire (assuming 4 Manager IP)
--------------------------------------------------------
::
+-----------+
| ML cap #0 |
+-----------+
| ML cap #1 |
+-----------+
| ML cap #2 |---+
+-----------+ |
|
+--> 0x0 +---------------+ LCAP
| ALT=1 |
+---------------+
| INTC |
+---------------+
| OFLS |
+---------------+
| LSS |
+---------------+
| SLCOUNT=4 |-----------+
+---------------+ |
|
0x4 +---------------+ LCTL |
| INTSTS | |
+---------------+ |
| CPA (x bits) | |
+---------------+ |
| SPA (x bits) | |
+---------------+ for each sublink x
| INTEN | |
+---------------+ |
| OFLEN | |
+---------------+ |
|
0x8 +---------------+ LOSIDV |
| L1OSIVD15 | |
+---------------+ |
| L1OSIDV.. | |
+---------------+ |
| L1OSIDV1 | +---+----------------------------------------------------------+
+---------------+ | |
v |
0xC + 0x2 * x +---------------+ LSDIIDx +---> 0x30000 +-----------------+ 0x00030000 |
| SDIID14 | | | SoundWire SHIM | |
+---------------+ | | generic | |
| SDIID... | | +-----------------+ 0x00030100 |
+---------------+ | | SoundWire IP | |
| SDIID0 | | +-----------------+ 0x00036000 |
+---------------+ | | SoundWire SHIM | |
| | vendor-specific | |
0x1C +---------------+ LSYNC | +-----------------+ |
| CMDSYNC | | v
+---------------+ | +-----------------+ 0x00030000 + 0x8000 * x
| SYNCGO | | | SoundWire SHIM |
+---------------+ | | generic |
| SYNCPU | | +-----------------+ 0x00030100 + 0x8000 * x
+---------------+ | | SoundWire IP |
| SYNPRD | | +-----------------+ 0x00036000 + 0x8000 * x
+---------------+ | | SoundWire SHIM |
| | vendor-specific |
0x20 +---------------+ LEPTR | +-----------------+
| ID = 0 | |
+---------------+ |
| VER | |
+---------------+ |
| PTR |------------+
+---------------+
DMIC HDaudio extended link mapping
==================================
A DMIC extended link is identified when LCAP.ALT=1 and
LEPTR.ID=0xC1 are set.
DMA control uses the existing LOSIDV register
Changes include additional descriptions for enumeration that were not
present in earlier generations.
- multi-link synchronization: capabilities in LCAP.LSS and control in LSYNC
- power management with LCTL.SPA bits
- hand-over to the DSP for access to multi-link registers, SHIM/IP with LCTL.OFLEN
- move of DMIC registers to different offsets, with no change in
functionality. The LEPTR.PTR value is an offset from the ML
address, with a default value of 0x10000.
Extended structure for DMIC
---------------------------
::
+-----------+
| ML cap #0 |
+-----------+
| ML cap #1 |
+-----------+
| ML cap #2 |---+
+-----------+ |
|
+--> 0x0 +---------------+ LCAP
| ALT=1 |
+---------------+
| INTC |
+---------------+
| OFLS |
+---------------+
| SLCOUNT=1 |
+---------------+
0x4 +---------------+ LCTL
| INTSTS |
+---------------+
| CPA |
+---------------+
| SPA |
+---------------+
| INTEN |
+---------------+
| OFLEN |
+---------------+ +---> 0x10000 +-----------------+ 0x00010000
| | DMIC SHIM |
0x8 +---------------+ LOSIDV | | generic |
| L1OSIVD15 | | +-----------------+ 0x00010100
+---------------+ | | DMIC IP |
| L1OSIDV.. | | +-----------------+ 0x00016000
+---------------+ | | DMIC SHIM |
| L1OSIDV1 | | | vendor-specific |
+---------------+ | +-----------------+
|
0x20 +---------------+ LEPTR |
| ID = 0xC1 | |
+---------------+ |
| VER | |
+---------------+ |
| PTR |-----------+
+---------------+
SSP HDaudio extended link mapping
=================================
A DMIC extended link is identified when LCAP.ALT=1 and
LEPTR.ID=0xC0 are set.
DMA control uses the existing LOSIDV register
Changes include additional descriptions for enumeration and control that were not
present in earlier generations:
- number of sublinks (SSP IP instances) in LCAP.LSCOUNT
- power management moved from SHIM to LCTL.SPA bits
- hand-over to the DSP for access to multi-link registers, SHIM/IP
with LCTL.OFLEN
- move of SHIM and SSP IP registers to different offsets, with no
change in functionality. The LEPTR.PTR value is an offset from the ML
address, with a default value of 0x28000.
Extended structure for SSP (assuming 3 instances of the IP)
-----------------------------------------------------------
::
+-----------+
| ML cap #0 |
+-----------+
| ML cap #1 |
+-----------+
| ML cap #2 |---+
+-----------+ |
|
+--> 0x0 +---------------+ LCAP
| ALT=1 |
+---------------+
| INTC |
+---------------+
| OFLS |
+---------------+
| SLCOUNT=3 |-------------------------for each sublink x -------------------------+
+---------------+ |
|
0x4 +---------------+ LCTL |
| INTSTS | |
+---------------+ |
| CPA (x bits) | |
+---------------+ |
| SPA (x bits) | |
+---------------+ |
| INTEN | |
+---------------+ |
| OFLEN | |
+---------------+ +---> 0x28000 +-----------------+ 0x00028000 |
| | SSP SHIM | |
0x8 +---------------+ LOSIDV | | generic | |
| L1OSIVD15 | | +-----------------+ 0x00028100 |
+---------------+ | | SSP IP | |
| L1OSIDV.. | | +-----------------+ 0x00028C00 |
+---------------+ | | SSP SHIM | |
| L1OSIDV1 | | | vendor-specific | |
+---------------+ | +-----------------+ |
| v
0x20 +---------------+ LEPTR | +-----------------+ 0x00028000 + 0x1000 * x
| ID = 0xC0 | | | SSP SHIM |
+---------------+ | | generic |
| VER | | +-----------------+ 0x00028100 + 0x1000 * x
+---------------+ | | SSP IP |
| PTR |-----------+ +-----------------+ 0x00028C00 + 0x1000 * x
+---------------+ | SSP SHIM |
| vendor-specific |
+-----------------+
linux-7.2.3/Documentation/sound/hd-audio/models.rst
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==============================
HD-Audio Codec-Specific Models
==============================
ALC880
======
3stack
3-jack in back and a headphone out
3stack-digout
3-jack in back, a HP out and a SPDIF out
5stack
5-jack in back, 2-jack in front
5stack-digout
5-jack in back, 2-jack in front, a SPDIF out
6stack
6-jack in back, 2-jack in front
6stack-digout
6-jack with a SPDIF out
6stack-automute
6-jack with headphone jack detection
ALC260
======
gpio1
Enable GPIO1
coef
Enable EAPD via COEF table
fujitsu
Quirk for FSC S7020
fujitsu-jwse
Quirk for FSC S7020 with jack modes and HP mic support
ALC262
======
inv-dmic
Inverted internal mic workaround
fsc-h270
Fixups for Fujitsu-Siemens Celsius H270
fsc-s7110
Fixups for Fujitsu-Siemens Lifebook S7110
hp-z200
Fixups for HP Z200
tyan
Fixups for Tyan Thunder n6650W
lenovo-3000
Fixups for Lenovo 3000
benq
Fixups for Benq ED8
benq-t31
Fixups for Benq T31
bayleybay
Fixups for Intel BayleyBay
ALC267/268
==========
inv-dmic
Inverted internal mic workaround
hp-eapd
Disable HP EAPD on NID 0x15
spdif
Enable SPDIF output on NID 0x1e
ALC22x/23x/25x/269/27x/28x/29x (and vendor-specific ALC3xxx models)
===================================================================
laptop-amic
Laptops with analog-mic input
laptop-dmic
Laptops with digital-mic input
alc269-dmic
Enable ALC269(VA) digital mic workaround
alc271-dmic
Enable ALC271X digital mic workaround
inv-dmic
Inverted internal mic workaround
headset-mic
Indicates a combined headset (headphone+mic) jack
headset-mode
More comprehensive headset support for ALC269 & co
headset-mode-no-hp-mic
Headset mode support without headphone mic
lenovo-dock
Enables docking station I/O for some Lenovos
hp-gpio-led
GPIO LED support on HP laptops
hp-dock-gpio-mic1-led
HP dock with mic LED support
dell-headset-multi
Headset jack, which can also be used as mic-in
dell-headset-dock
Headset jack (without mic-in), and also dock I/O
dell-headset3
Headset jack (without mic-in), and also dock I/O, variant 3
dell-headset4
Headset jack (without mic-in), and also dock I/O, variant 4
alc283-dac-wcaps
Fixups for Chromebook with ALC283
alc283-sense-combo
Combo jack sensing on ALC283
tpt440-dock
Pin configs for Lenovo Thinkpad Dock support
tpt440
Lenovo Thinkpad T440s setup
tpt460
Lenovo Thinkpad T460/560 setup
tpt470-dock
Lenovo Thinkpad T470 dock setup
dual-codecs
Lenovo laptops with dual codecs
alc700-ref
Intel reference board with ALC700 codec
vaio
Pin fixups for Sony VAIO laptops
dell-m101z
COEF setup for Dell M101z
asus-g73jw
Subwoofer pin fixup for ASUS G73JW
lenovo-eapd
Inversed EAPD setup for Lenovo laptops
sony-hweq
H/W EQ COEF setup for Sony laptops
pcm44k
Fixed PCM 44kHz constraints (for buggy devices)
lifebook
Dock pin fixups for Fujitsu Lifebook
lifebook-extmic
Headset mic fixup for Fujitsu Lifebook
lifebook-hp-pin
Headphone pin fixup for Fujitsu Lifebook
lifebook-u7x7
Lifebook U7x7 fixups
alc269vb-amic
ALC269VB analog mic pin fixups
alc269vb-dmic
ALC269VB digital mic pin fixups
hp-mute-led-mic1
Mute LED via Mic1 pin on HP
hp-mute-led-mic2
Mute LED via Mic2 pin on HP
hp-mute-led-mic3
Mute LED via Mic3 pin on HP
hp-gpio-mic1
GPIO + Mic1 pin LED on HP
hp-line1-mic1
Mute LED via Line1 + Mic1 pins on HP
noshutup
Skip shutup callback
sony-nomic
Headset mic fixup for Sony laptops
aspire-headset-mic
Headset pin fixup for Acer Aspire
asus-x101
ASUS X101 fixups
acer-ao7xx
Acer AO7xx fixups
acer-aspire-e1
Acer Aspire E1 fixups
acer-ac700
Acer AC700 fixups
limit-mic-boost
Limit internal mic boost on Lenovo machines
asus-zenbook
ASUS Zenbook fixups
asus-zenbook-ux31a
ASUS Zenbook UX31A fixups
ordissimo
Ordissimo EVE2 (or Malata PC-B1303) fixups
asus-tx300
ASUS TX300 fixups
alc283-int-mic
ALC283 COEF setup for Lenovo machines
mono-speakers
Subwoofer and headset fixupes for Dell Inspiron
alc290-subwoofer
Subwoofer fixups for Dell Vostro
thinkpad
Binding with thinkpad_acpi driver for Lenovo machines
dmic-thinkpad
thinkpad_acpi binding + digital mic support
alc255-acer
ALC255 fixups on Acer machines
alc255-asus
ALC255 fixups on ASUS machines
alc255-dell1
ALC255 fixups on Dell machines
alc255-dell2
ALC255 fixups on Dell machines, variant 2
alc293-dell1
ALC293 fixups on Dell machines
alc283-headset
Headset pin fixups on ALC283
aspire-v5
Acer Aspire V5 fixups
hp-gpio4
GPIO and Mic1 pin mute LED fixups for HP
hp-gpio-led
GPIO mute LEDs on HP
hp-gpio2-hotkey
GPIO mute LED with hot key handling on HP
hp-dock-pins
GPIO mute LEDs and dock support on HP
hp-dock-gpio-mic
GPIO, Mic mute LED and dock support on HP
hp-9480m
HP 9480m fixups
alc288-dell1
ALC288 fixups on Dell machines
alc288-dell-xps13
ALC288 fixups on Dell XPS13
dell-e7x
Dell E7x fixups
alc293-dell
ALC293 fixups on Dell machines
alc298-dell1
ALC298 fixups on Dell machines
alc298-dell-aio
ALC298 fixups on Dell AIO machines
alc275-dell-xps
ALC275 fixups on Dell XPS models
lenovo-spk-noise
Workaround for speaker noise on Lenovo machines
lenovo-hotkey
Hot-key support via Mic2 pin on Lenovo machines
dell-spk-noise
Workaround for speaker noise on Dell machines
alc255-dell1
ALC255 fixups on Dell machines
alc295-disable-dac3
Disable DAC3 routing on ALC295
alc280-hp-headset
HP Elitebook fixups
alc221-hp-mic
Front mic pin fixup on HP machines
alc298-spk-volume
Speaker pin routing workaround on ALC298
dell-inspiron-7559
Dell Inspiron 7559 fixups
ativ-book
Samsung Ativ book 8 fixups
alc221-hp-mic
ALC221 headset fixups on HP machines
alc256-asus-mic
ALC256 fixups on ASUS machines
alc256-asus-aio
ALC256 fixups on ASUS AIO machines
alc233-eapd
ALC233 fixups on ASUS machines
alc294-lenovo-mic
ALC294 Mic pin fixup for Lenovo AIO machines
alc225-wyse
Dell Wyse fixups
alc274-dell-aio
ALC274 fixups on Dell AIO machines
alc255-dummy-lineout
Dell Precision 3930 fixups
alc255-dell-headset
Dell Precision 3630 fixups
alc295-hp-x360
HP Spectre X360 fixups
alc-sense-combo
Headset button support for Chrome platform
huawei-mbx-stereo
Enable initialization verbs for Huawei MBX stereo speakers;
might be risky, try this at your own risk
alc298-samsung-headphone
Samsung laptops with ALC298
alc256-samsung-headphone
Samsung laptops with ALC256
ALC66x/67x/892
==============
aspire
Subwoofer pin fixup for Aspire laptops
ideapad
Subwoofer pin fixup for Ideapad laptops
mario
Chromebook mario model fixup
hp-rp5800
Headphone pin fixup for HP RP5800
asus-mode1
ASUS
asus-mode2
ASUS
asus-mode3
ASUS
asus-mode4
ASUS
asus-mode5
ASUS
asus-mode6
ASUS
asus-mode7
ASUS
asus-mode8
ASUS
zotac-z68
Front HP fixup for Zotac Z68
inv-dmic
Inverted internal mic workaround
alc662-headset-multi
Dell headset jack, which can also be used as mic-in (ALC662)
dell-headset-multi
Headset jack, which can also be used as mic-in
alc662-headset
Headset mode support on ALC662
alc668-headset
Headset mode support on ALC668
bass16
Bass speaker fixup on pin 0x16
bass1a
Bass speaker fixup on pin 0x1a
automute
Auto-mute fixups for ALC668
dell-xps13
Dell XPS13 fixups
asus-nx50
ASUS Nx50 fixups
asus-nx51
ASUS Nx51 fixups
asus-g751
ASUS G751 fixups
alc891-headset
Headset mode support on ALC891
alc891-headset-multi
Dell headset jack, which can also be used as mic-in (ALC891)
acer-veriton
Acer Veriton speaker pin fixup
asrock-mobo
Fix invalid 0x15 / 0x16 pins
usi-headset
Headset support on USI machines
dual-codecs
Lenovo laptops with dual codecs
alc285-hp-amp-init
HP laptops which require speaker amplifier initialization (ALC285)
ALC680
======
N/A
ALC88x/898/1150/1220
====================
abit-aw9d
Pin fixups for Abit AW9D-MAX
lenovo-y530
Pin fixups for Lenovo Y530
acer-aspire-7736
Fixup for Acer Aspire 7736
asus-w90v
Pin fixup for ASUS W90V
cd
Enable audio CD pin NID 0x1c
no-front-hp
Disable front HP pin NID 0x1b
vaio-tt
Pin fixup for VAIO TT
eee1601
COEF setups for ASUS Eee 1601
alc882-eapd
Change EAPD COEF mode on ALC882
alc883-eapd
Change EAPD COEF mode on ALC883
gpio1
Enable GPIO1
gpio2
Enable GPIO2
gpio3
Enable GPIO3
alc889-coef
Setup ALC889 COEF
asus-w2jc
Fixups for ASUS W2JC
acer-aspire-4930g
Acer Aspire 4930G/5930G/6530G/6930G/7730G
acer-aspire-8930g
Acer Aspire 8330G/6935G
acer-aspire
Acer Aspire others
macpro-gpio
GPIO setup for Mac Pro
dac-route
Workaround for DAC routing on Acer Aspire
mbp-vref
Vref setup for Macbook Pro
imac91-vref
Vref setup for iMac 9,1
mba11-vref
Vref setup for MacBook Air 1,1
mba21-vref
Vref setup for MacBook Air 2,1
mp11-vref
Vref setup for Mac Pro 1,1
mp41-vref
Vref setup for Mac Pro 4,1
inv-dmic
Inverted internal mic workaround
no-primary-hp
VAIO Z/VGC-LN51JGB workaround (for fixed speaker DAC)
asus-bass
Bass speaker setup for ASUS ET2700
dual-codecs
ALC1220 dual codecs for Gaming mobos
clevo-p950
Fixups for Clevo P950
ALC861/660
==========
N/A
ALC861VD/660VD
==============
N/A
CMI9880
=======
minimal
3-jack in back
min_fp
3-jack in back, 2-jack in front
full
6-jack in back, 2-jack in front
full_dig
6-jack in back, 2-jack in front, SPDIF I/O
allout
5-jack in back, 2-jack in front, SPDIF out
auto
auto-config reading BIOS (default)
AD1882 / AD1882A
================
3stack
3-stack mode
3stack-automute
3-stack with automute front HP (default)
6stack
6-stack mode
AD1884A / AD1883 / AD1984A / AD1984B
====================================
desktop 3-stack desktop (default)
laptop laptop with HP jack sensing
mobile mobile devices with HP jack sensing
thinkpad Lenovo Thinkpad X300
touchsmart HP Touchsmart
AD1884
======
N/A
AD1981
======
basic 3-jack (default)
hp HP nx6320
thinkpad Lenovo Thinkpad T60/X60/Z60
toshiba Toshiba U205
AD1983
======
N/A
AD1984
======
basic default configuration
thinkpad Lenovo Thinkpad T61/X61
dell_desktop Dell T3400
AD1986A
=======
3stack
3-stack, shared surrounds
laptop
2-channel only (FSC V2060, Samsung M50)
laptop-imic
2-channel with built-in mic
eapd
Turn on EAPD constantly
AD1988/AD1988B/AD1989A/AD1989B
==============================
6stack
6-jack
6stack-dig
ditto with SPDIF
3stack
3-jack
3stack-dig
ditto with SPDIF
laptop
3-jack with hp-jack automute
laptop-dig
ditto with SPDIF
auto
auto-config reading BIOS (default)
Conexant 5045
=============
cap-mix-amp
Fix max input level on mixer widget
toshiba-p105
Toshiba P105 quirk
hp-530
HP 530 quirk
Conexant 5047
=============
cap-mix-amp
Fix max input level on mixer widget
Conexant 5051
=============
lenovo-x200
Lenovo X200 quirk
Conexant 5066
=============
stereo-dmic
Workaround for inverted stereo digital mic
gpio1
Enable GPIO1 pin
headphone-mic-pin
Enable headphone mic NID 0x18 without detection
tp410
Thinkpad T400 & co quirks
thinkpad
Thinkpad mute/mic LED quirk
lemote-a1004
Lemote A1004 quirk
lemote-a1205
Lemote A1205 quirk
olpc-xo
OLPC XO quirk
mute-led-eapd
Mute LED control via EAPD
hp-dock
HP dock support
mute-led-gpio
Mute LED control via GPIO
hp-mic-fix
Fix for headset mic pin on HP boxes
STAC9200
========
ref
Reference board
oqo
OQO Model 2
dell-d21
Dell (unknown)
dell-d22
Dell (unknown)
dell-d23
Dell (unknown)
dell-m21
Dell Inspiron 630m, Dell Inspiron 640m
dell-m22
Dell Latitude D620, Dell Latitude D820
dell-m23
Dell XPS M1710, Dell Precision M90
dell-m24
Dell Latitude 120L
dell-m25
Dell Inspiron E1505n
dell-m26
Dell Inspiron 1501
dell-m27
Dell Inspiron E1705/9400
gateway-m4
Gateway laptops with EAPD control
gateway-m4-2
Gateway laptops with EAPD control
panasonic
Panasonic CF-74
auto
BIOS setup (default)
STAC9205/9254
=============
ref
Reference board
dell-m42
Dell (unknown)
dell-m43
Dell Precision
dell-m44
Dell Inspiron
eapd
Keep EAPD on (e.g. Gateway T1616)
auto
BIOS setup (default)
STAC9220/9221
=============
ref
Reference board
3stack
D945 3stack
5stack
D945 5stack + SPDIF
intel-mac-v1
Intel Mac Type 1
intel-mac-v2
Intel Mac Type 2
intel-mac-v3
Intel Mac Type 3
intel-mac-v4
Intel Mac Type 4
intel-mac-v5
Intel Mac Type 5
intel-mac-auto
Intel Mac (detect type according to subsystem id)
macmini
Intel Mac Mini (equivalent with type 3)
macbook
Intel Mac Book (eq. type 5)
macbook-pro-v1
Intel Mac Book Pro 1st generation (eq. type 3)
macbook-pro
Intel Mac Book Pro 2nd generation (eq. type 3)
imac-intel
Intel iMac (eq. type 2)
imac-intel-20
Intel iMac (newer version) (eq. type 3)
ecs202
ECS/PC chips
dell-d81
Dell (unknown)
dell-d82
Dell (unknown)
dell-m81
Dell (unknown)
dell-m82
Dell XPS M1210
auto
BIOS setup (default)
STAC9202/9250/9251
==================
ref
Reference board, base config
m1
Some Gateway MX series laptops (NX560XL)
m1-2
Some Gateway MX series laptops (MX6453)
m2
Some Gateway MX series laptops (M255)
m2-2
Some Gateway MX series laptops
m3
Some Gateway MX series laptops
m5
Some Gateway MX series laptops (MP6954)
m6
Some Gateway NX series laptops
auto
BIOS setup (default)
STAC9227/9228/9229/927x
=======================
ref
Reference board
ref-no-jd
Reference board without HP/Mic jack detection
3stack
D965 3stack
5stack
D965 5stack + SPDIF
5stack-no-fp
D965 5stack without front panel
dell-3stack
Dell Dimension E520
dell-bios
Fixes with Dell BIOS setup
dell-bios-amic
Fixes with Dell BIOS setup including analog mic
volknob
Fixes with volume-knob widget 0x24
auto
BIOS setup (default)
STAC92HD71B*
============
ref
Reference board
dell-m4-1
Dell desktops
dell-m4-2
Dell desktops
dell-m4-3
Dell desktops
hp-m4
HP mini 1000
hp-dv5
HP dv series
hp-hdx
HP HDX series
hp-dv4-1222nr
HP dv4-1222nr (with LED support)
auto
BIOS setup (default)
STAC92HD73*
===========
ref
Reference board
no-jd
BIOS setup but without jack-detection
intel
Intel D*45* mobos
dell-m6-amic
Dell desktops/laptops with analog mics
dell-m6-dmic
Dell desktops/laptops with digital mics
dell-m6
Dell desktops/laptops with both type of mics
dell-eq
Dell desktops/laptops
alienware
Alienware M17x
asus-mobo
Pin configs for ASUS mobo with 5.1/SPDIF out
auto
BIOS setup (default)
STAC92HD83*
===========
ref
Reference board
mic-ref
Reference board with power management for ports
dell-s14
Dell laptop
dell-vostro-3500
Dell Vostro 3500 laptop
hp-dv7-4000
HP dv-7 4000
hp_cNB11_intquad
HP CNB models with 4 speakers
hp-zephyr
HP Zephyr
hp-led
HP with broken BIOS for mute LED
hp-inv-led
HP with broken BIOS for inverted mute LED
hp-mic-led
HP with mic-mute LED
headset-jack
Dell Latitude with a 4-pin headset jack
hp-envy-bass
Pin fixup for HP Envy bass speaker (NID 0x0f)
hp-envy-ts-bass
Pin fixup for HP Envy TS bass speaker (NID 0x10)
hp-bnb13-eq
Hardware equalizer setup for HP laptops
hp-envy-ts-bass
HP Envy TS bass support
auto
BIOS setup (default)
STAC92HD95
==========
hp-led
LED support for HP laptops
hp-bass
Bass HPF setup for HP Spectre 13
STAC9872
========
vaio
VAIO laptop without SPDIF
auto
BIOS setup (default)
Cirrus Logic CS4206/4207
========================
mbp53
MacBook Pro 5,3
mbp55
MacBook Pro 5,5
imac27
IMac 27 Inch
imac27_122
iMac 12,2
apple
Generic Apple quirk
mbp101
MacBookPro 10,1
mbp81
MacBookPro 8,1
mba42
MacBookAir 4,2
auto
BIOS setup (default)
Cirrus Logic CS4208
===================
mba6
MacBook Air 6,1 and 6,2
gpio0
Enable GPIO 0 amp
mbp11
MacBookPro 11,2
macmini
MacMini 7,1
auto
BIOS setup (default)
VIA VT17xx/VT18xx/VT20xx
========================
auto
BIOS setup (default)
linux-7.2.3/Documentation/sound/hd-audio/notes.rst
+892
View File
@@ -0,0 +1,892 @@
=============================
More Notes on HD-Audio Driver
=============================
Takashi Iwai <tiwai@suse.de>
General
=======
HD-audio is the new standard on-board audio component on modern PCs
after AC97. Although Linux has been supporting HD-audio since long
time ago, there are often problems with new machines. A part of the
problem is broken BIOS, and the rest is the driver implementation.
This document explains the brief trouble-shooting and debugging
methods for the HD-audio hardware.
The HD-audio component consists of two parts: the controller chip and
the codec chips on the HD-audio bus. Linux provides a single driver
for all controllers, snd-hda-intel. Although the driver name contains
a word of a well-known hardware vendor, it's not specific to it but for
all controller chips by other companies. Since the HD-audio
controllers are supposed to be compatible, the single snd-hda-driver
should work in most cases. But, not surprisingly, there are known
bugs and issues specific to each controller type. The snd-hda-intel
driver has a bunch of workarounds for these as described below.
A controller may have multiple codecs. Usually you have one audio
codec and optionally one modem codec. In theory, there might be
multiple audio codecs, e.g. for analog and digital outputs, and the
driver might not work properly because of conflict of mixer elements.
This should be fixed in future if such hardware really exists.
The snd-hda-intel driver has several different codec parsers depending
on the codec. It has a generic parser as a fallback, but this
functionality is fairly limited until now. Instead of the generic
parser, usually the codec-specific parser (coded in patch_*.c) is used
for the codec-specific implementations. The details about the
codec-specific problems are explained in the later sections.
If you are interested in the deep debugging of HD-audio, read the
HD-audio specification at first. The specification is found on
Intel's web page, for example:
* https://www.intel.com/content/www/us/en/standards/high-definition-audio-specification.html
HD-Audio Controller
===================
DMA-Position Problem
--------------------
The most common problem of the controller is the inaccurate DMA
pointer reporting. The DMA pointer for playback and capture can be
read in two ways, either via a LPIB register or via a position-buffer
map. As default the driver tries to read from the io-mapped
position-buffer, and falls back to LPIB if the position-buffer appears
dead. However, this detection isn't perfect on some devices. In such
a case, you can change the default method via ``position_fix`` option.
``position_fix=1`` means to use LPIB method explicitly.
``position_fix=2`` means to use the position-buffer.
``position_fix=3`` means to use a combination of both methods, needed
for some VIA controllers. The capture stream position is corrected
by comparing both LPIB and position-buffer values.
``position_fix=4`` is another combination available for all controllers,
and uses LPIB for the playback and the position-buffer for the capture
streams.
``position_fix=5`` is specific to Intel platforms, so far, for Skylake
and onward. It applies the delay calculation for the precise position
reporting.
``position_fix=6`` is to correct the position with the fixed FIFO
size, mainly targeted for the recent AMD controllers.
0 is the default value for all other
controllers, the automatic check and fallback to LPIB as described in
the above. If you get a problem of repeated sounds, this option might
help.
In addition to that, every controller is known to be broken regarding
the wake-up timing. It wakes up a few samples before actually
processing the data on the buffer. This caused a lot of problems, for
example, with ALSA dmix or JACK. Since 2.6.27 kernel, the driver puts
an artificial delay to the wake up timing. This delay is controlled
via ``bdl_pos_adj`` option.
When ``bdl_pos_adj`` is a negative value (as default), it's assigned to
an appropriate value depending on the controller chip. For Intel
chips, it'd be 1 while it'd be 32 for others. Usually this works.
Only in case it doesn't work and you get warning messages, you should
change this parameter to other values.
Codec-Probing Problem
---------------------
A less often but a more severe problem is the codec probing. When
BIOS reports the available codec slots wrongly, the driver gets
confused and tries to access the non-existing codec slot. This often
results in the total screw-up, and destructs the further communication
with the codec chips. The symptom appears usually as error messages
like:
::
hda_intel: azx_get_response timeout, switching to polling mode:
last cmd=0x12345678
hda_intel: azx_get_response timeout, switching to single_cmd mode:
last cmd=0x12345678
The first line is a warning, and this is usually relatively harmless.
It means that the codec response isn't notified via an IRQ. The
driver uses explicit polling method to read the response. It gives
very slight CPU overhead, but you'd unlikely notice it.
The second line is, however, a fatal error. If this happens, usually
it means that something is really wrong. Most likely you are
accessing a non-existing codec slot.
Thus, if the second error message appears, try to narrow the probed
codec slots via ``probe_mask`` option. It's a bitmask, and each bit
corresponds to the codec slot. For example, to probe only the first
slot, pass ``probe_mask=1``. For the first and the third slots, pass
``probe_mask=5`` (where 5 = 1 | 4), and so on.
Since 2.6.29 kernel, the driver has a more robust probing method, so
this error might happen rarely, though.
On a machine with a broken BIOS, sometimes you need to force the
driver to probe the codec slots the hardware doesn't report for use.
In such a case, turn the bit 8 (0x100) of ``probe_mask`` option on.
Then the rest 8 bits are passed as the codec slots to probe
unconditionally. For example, ``probe_mask=0x103`` will force to probe
the codec slots 0 and 1 no matter what the hardware reports.
Interrupt Handling
------------------
HD-audio driver uses MSI as default (if available) since 2.6.33
kernel as MSI works better on some machines, and in general, it's
better for performance. However, Nvidia controllers showed bad
regressions with MSI (especially in a combination with AMD chipset),
thus we disabled MSI for them.
There seem also still other devices that don't work with MSI. If you
see a regression wrt the sound quality (stuttering, etc) or a lock-up
in the recent kernel, try to pass ``enable_msi=0`` option to disable
MSI. If it works, you can add the known bad device to the blacklist
defined in hda_intel.c. In such a case, please report and give the
patch back to the upstream developer.
HD-Audio Codec
==============
Model Option
------------
The most common problem regarding the HD-audio driver is the
unsupported codec features or the mismatched device configuration.
Most of codec-specific code has several preset models, either to
override the BIOS setup or to provide more comprehensive features.
The driver checks PCI SSID and looks through the static configuration
table until any matching entry is found. If you have a new machine,
you may see a message like below:
::
hda_codec: ALC880: BIOS auto-probing.
Meanwhile, in the earlier versions, you would see a message like:
::
hda_codec: Unknown model for ALC880, trying auto-probe from BIOS...
Even if you see such a message, DON'T PANIC. Take a deep breath and
keep your towel. First of all, it's an informational message, no
warning, no error. This means that the PCI SSID of your device isn't
listed in the known preset model (white-)list. But, this doesn't mean
that the driver is broken. Many codec-drivers provide the automatic
configuration mechanism based on the BIOS setup.
The HD-audio codec has usually "pin" widgets, and BIOS sets the default
configuration of each pin, which indicates the location, the
connection type, the jack color, etc. The HD-audio driver can guess
the right connection judging from these default configuration values.
However -- some codec-support codes, such as patch_analog.c, don't
support the automatic probing (yet as of 2.6.28). And, BIOS is often,
yes, pretty often broken. It sets up wrong values and screws up the
driver.
The preset model (or recently called as "fix-up") is provided
basically to overcome such a situation. When the matching preset
model is found in the white-list, the driver assumes the static
configuration of that preset with the correct pin setup, etc.
Thus, if you have a newer machine with a slightly different PCI SSID
(or codec SSID) from the existing one, you may have a good chance to
reuse the same model. You can pass the ``model`` option to specify the
preset model instead of PCI (and codec-) SSID look-up.
What ``model`` option values are available depends on the codec chip.
Check your codec chip from the codec proc file (see "Codec Proc-File"
section below). It will show the vendor/product name of your codec
chip. Then, see Documentation/sound/hd-audio/models.rst file,
the section of HD-audio driver. You can find a list of codecs
and ``model`` options belonging to each codec. For example, for Realtek
ALC262 codec chip, pass ``model=ultra`` for devices that are compatible
with Samsung Q1 Ultra.
Thus, the first thing you can do for any brand-new, unsupported and
non-working HD-audio hardware is to check HD-audio codec and several
different ``model`` option values. If you have any luck, some of them
might suit with your device well.
There are a few special model option values:
* when 'nofixup' is passed, the device-specific fixups in the codec
parser are skipped.
* when ``generic`` is passed, the codec-specific parser is skipped and
only the generic parser is used.
A new style for the model option that was introduced since 5.15 kernel
is to pass the PCI or codec SSID in the form of ``model=XXXX:YYYY``
where XXXX and YYYY are the sub-vendor and sub-device IDs in hex
numbers, respectively. This is a kind of aliasing to another device;
when this form is given, the driver will refer to that SSID as a
reference to the quirk table. It'd be useful especially when the
target quirk isn't listed in the model table. For example, passing
model=103c:8862 will apply the quirk for HP ProBook 445 G8 (which
isn't found in the model table as of writing) as long as the device is
handled equivalently by the same driver.
Speaker and Headphone Output
----------------------------
One of the most frequent (and obvious) bugs with HD-audio is the
silent output from either or both of a built-in speaker and a
headphone jack. In general, you should try a headphone output at
first. A speaker output often requires more additional controls like
the external amplifier bits. Thus a headphone output has a slightly
better chance.
Before making a bug report, double-check whether the mixer is set up
correctly. The recent version of snd-hda-intel driver provides mostly
"Master" volume control as well as "Front" volume (where Front
indicates the front-channels). In addition, there can be individual
"Headphone" and "Speaker" controls.
Ditto for the speaker output. There can be "External Amplifier"
switch on some codecs. Turn on this if present.
Another related problem is the automatic mute of speaker output by
headphone plugging. This feature is implemented in most cases, but
not on every preset model or codec-support code.
In anyway, try a different model option if you have such a problem.
Some other models may match better and give you more matching
functionality. If none of the available models works, send a bug
report. See the bug report section for details.
If you are masochistic enough to debug the driver problem, note the
following:
* The speaker (and the headphone, too) output often requires the
external amplifier. This can be set usually via EAPD verb or a
certain GPIO. If the codec pin supports EAPD, you have a better
chance via SET_EAPD_BTL verb (0x70c). On others, GPIO pin (mostly
it's either GPIO0 or GPIO1) may turn on/off EAPD.
* Some Realtek codecs require special vendor-specific coefficients to
turn on the amplifier. See patch_realtek.c.
* IDT codecs may have extra power-enable/disable controls on each
analog pin. See patch_sigmatel.c.
* Very rare but some devices don't accept the pin-detection verb until
triggered. Issuing GET_PIN_SENSE verb (0xf09) may result in the
codec-communication stall. Some examples are found in
patch_realtek.c.
Capture Problems
----------------
The capture problems are often because of missing setups of mixers.
Thus, before submitting a bug report, make sure that you set up the
mixer correctly. For example, both "Capture Volume" and "Capture
Switch" have to be set properly in addition to the right "Capture
Source" or "Input Source" selection. Some devices have "Mic Boost"
volume or switch.
When the PCM device is opened via "default" PCM (without pulse-audio
plugin), you'll likely have "Digital Capture Volume" control as well.
This is provided for the extra gain/attenuation of the signal in
software, especially for the inputs without the hardware volume
control such as digital microphones. Unless really needed, this
should be set to exactly 50%, corresponding to 0dB -- neither extra
gain nor attenuation. When you use "hw" PCM, i.e., a raw access PCM,
this control will have no influence, though.
It's known that some codecs / devices have fairly bad analog circuits,
and the recorded sound contains a certain DC-offset. This is no bug
of the driver.
Most of modern laptops have no analog CD-input connection. Thus, the
recording from CD input won't work in many cases although the driver
provides it as the capture source. Use CDDA instead.
The automatic switching of the built-in and external mic per plugging
is implemented on some codec models but not on every model. Partly
because of my laziness but mostly lack of testers. Feel free to
submit the improvement patch to the author.
Direct Debugging
----------------
If no model option gives you a better result, and you are a tough guy
to fight against evil, try debugging via hitting the raw HD-audio
codec verbs to the device. Some tools are available: hda-emu and
hda-analyzer. The detailed description is found in the sections
below. You'd need to enable hwdep for using these tools. See "Kernel
Configuration" section.
Other Issues
============
Kernel Configuration
--------------------
In general, I recommend you to enable the sound debug option,
``CONFIG_SND_DEBUG=y``, no matter whether you are debugging or not.
Don't forget to turn on the appropriate ``CONFIG_SND_HDA_CODEC_*``
options. Note that each of them corresponds to the codec chip, not
the controller chip. Thus, even if lspci shows the Nvidia controller,
you may need to choose the option for other vendors. If you are
unsure, just select all yes.
``CONFIG_SND_HDA_HWDEP`` is a useful option for debugging the driver.
When this is enabled, the driver creates hardware-dependent devices
(one per each codec), and you have a raw access to the device via
these device files. For example, ``hwC0D2`` will be created for the
codec slot #2 of the first card (#0). For debug-tools such as
hda-verb and hda-analyzer, the hwdep device has to be enabled.
Thus, it'd be better to turn this on always.
``CONFIG_SND_HDA_RECONFIG`` is a new option, and this depends on the
hwdep option above. When enabled, you'll have some sysfs files under
the corresponding hwdep directory. See "HD-audio reconfiguration"
section below.
``CONFIG_SND_HDA_POWER_SAVE`` option enables the power-saving feature.
See "Power-saving" section below.
Codec Proc-File
---------------
The codec proc-file is a treasure-chest for debugging HD-audio.
It shows most of useful information of each codec widget.
The proc file is located in /proc/asound/card*/codec#*, one file per
each codec slot. You can know the codec vendor, product id and
names, the type of each widget, capabilities and so on.
This file, however, doesn't show the jack sensing state, so far. This
is because the jack-sensing might be depending on the trigger state.
This file will be picked up by the debug tools, and also it can be fed
to the emulator as the primary codec information. See the debug tools
section below.
This proc file can be also used to check whether the generic parser is
used. When the generic parser is used, the vendor/product ID name
will appear as "Realtek ID 0262", instead of "Realtek ALC262".
HD-Audio Reconfiguration
------------------------
This is an experimental feature to allow you re-configure the HD-audio
codec dynamically without reloading the driver. The following sysfs
files are available under each codec-hwdep device directory (e.g.
/sys/class/sound/hwC0D0):
vendor_id
Shows the 32bit codec vendor-id hex number. You can change the
vendor-id value by writing to this file.
subsystem_id
Shows the 32bit codec subsystem-id hex number. You can change the
subsystem-id value by writing to this file.
revision_id
Shows the 32bit codec revision-id hex number. You can change the
revision-id value by writing to this file.
afg
Shows the AFG ID. This is read-only.
mfg
Shows the MFG ID. This is read-only.
name
Shows the codec name string. Can be changed by writing to this
file.
modelname
Shows the currently set ``model`` option. Can be changed by writing
to this file.
init_verbs
The extra verbs to execute at initialization. You can add a verb by
writing to this file. Pass three numbers: nid, verb and parameter
(separated with a space).
hints
Shows / stores hint strings for codec parsers for any use.
Its format is ``key = value``. For example, passing ``jack_detect = no``
will disable the jack detection of the machine completely.
init_pin_configs
Shows the initial pin default config values set by BIOS.
driver_pin_configs
Shows the pin default values set by the codec parser explicitly.
This doesn't show all pin values but only the changed values by
the parser. That is, if the parser doesn't change the pin default
config values by itself, this will contain nothing.
user_pin_configs
Shows the pin default config values to override the BIOS setup.
Writing this (with two numbers, NID and value) appends the new
value. The given will be used instead of the initial BIOS value at
the next reconfiguration time. Note that this config will override
even the driver pin configs, too.
reconfig
Triggers the codec re-configuration. When any value is written to
this file, the driver re-initialize and parses the codec tree
again. All the changes done by the sysfs entries above are taken
into account.
clear
Resets the codec, removes the mixer elements and PCM stuff of the
specified codec, and clear all init verbs and hints.
For example, when you want to change the pin default configuration
value of the pin widget 0x14 to 0x9993013f, and let the driver
re-configure based on that state, run like below:
::
# echo 0x14 0x9993013f > /sys/class/sound/hwC0D0/user_pin_configs
# echo 1 > /sys/class/sound/hwC0D0/reconfig
Hint Strings
------------
The codec parser have several switches and adjustment knobs for
matching better with the actual codec or device behavior. Many of
them can be adjusted dynamically via "hints" strings as mentioned in
the section above. For example, by passing ``jack_detect = no`` string
via sysfs or a patch file, you can disable the jack detection, thus
the codec parser will skip the features like auto-mute or mic
auto-switch. As a boolean value, either ``yes``, ``no``, ``true``, ``false``,
``1`` or ``0`` can be passed.
The generic parser supports the following hints:
jack_detect (bool)
specify whether the jack detection is available at all on this
machine; default true
inv_jack_detect (bool)
indicates that the jack detection logic is inverted
trigger_sense (bool)
indicates that the jack detection needs the explicit call of
AC_VERB_SET_PIN_SENSE verb
inv_eapd (bool)
indicates that the EAPD is implemented in the inverted logic
pcm_format_first (bool)
sets the PCM format before the stream tag and channel ID
sticky_stream (bool)
keep the PCM format, stream tag and ID as long as possible;
default true
spdif_status_reset (bool)
reset the SPDIF status bits at each time the SPDIF stream is set
up
pin_amp_workaround (bool)
the output pin may have multiple amp values
single_adc_amp (bool)
ADCs can have only single input amps
auto_mute (bool)
enable/disable the headphone auto-mute feature; default true
auto_mic (bool)
enable/disable the mic auto-switch feature; default true
line_in_auto_switch (bool)
enable/disable the line-in auto-switch feature; default false
need_dac_fix (bool)
limits the DACs depending on the channel count
primary_hp (bool)
probe headphone jacks as the primary outputs; default true
multi_io (bool)
try probing multi-I/O config (e.g. shared line-in/surround,
mic/clfe jacks)
multi_cap_vol (bool)
provide multiple capture volumes
inv_dmic_split (bool)
provide split internal mic volume/switch for phase-inverted
digital mics
indep_hp (bool)
provide the independent headphone PCM stream and the corresponding
mixer control, if available
add_stereo_mix_input (bool)
add the stereo mix (analog-loopback mix) to the input mux if
available
add_jack_modes (bool)
add "xxx Jack Mode" enum controls to each I/O jack for allowing to
change the headphone amp and mic bias VREF capabilities
power_save_node (bool)
advanced power management for each widget, controlling the power
state (D0/D3) of each widget node depending on the actual pin and
stream states
power_down_unused (bool)
power down the unused widgets, a subset of power_save_node, and
will be dropped in future
add_hp_mic (bool)
add the headphone to capture source if possible
hp_mic_detect (bool)
enable/disable the hp/mic shared input for a single built-in mic
case; default true
vmaster (bool)
enable/disable the virtual Master control; default true
mixer_nid (int)
specifies the widget NID of the analog-loopback mixer
Early Patching
--------------
When ``CONFIG_SND_HDA_PATCH_LOADER=y`` is set, you can pass a "patch"
as a firmware file for modifying the HD-audio setup before
initializing the codec. This can work basically like the
reconfiguration via sysfs in the above, but it does it before the
first codec configuration.
A patch file is a plain text file which looks like below:
::
[codec]
0x12345678 0xabcd1234 2
[model]
auto
[pincfg]
0x12 0x411111f0
[verb]
0x20 0x500 0x03
0x20 0x400 0xff
[hint]
jack_detect = no
The file needs to have a line ``[codec]``. The next line should contain
three numbers indicating the codec vendor-id (0x12345678 in the
example), the codec subsystem-id (0xabcd1234) and the address (2) of
the codec. The rest patch entries are applied to this specified codec
until another codec entry is given. Passing 0 or a negative number to
the first or the second value will make the check of the corresponding
field be skipped. It'll be useful for really broken devices that don't
initialize SSID properly.
The ``[model]`` line allows to change the model name of the each codec.
In the example above, it will be changed to model=auto.
Note that this overrides the module option.
After the ``[pincfg]`` line, the contents are parsed as the initial
default pin-configurations just like ``user_pin_configs`` sysfs above.
The values can be shown in user_pin_configs sysfs file, too.
Similarly, the lines after ``[verb]`` are parsed as ``init_verbs``
sysfs entries, and the lines after ``[hint]`` are parsed as ``hints``
sysfs entries, respectively.
Another example to override the codec vendor id from 0x12345678 to
0xdeadbeef is like below:
::
[codec]
0x12345678 0xabcd1234 2
[vendor_id]
0xdeadbeef
In the similar way, you can override the codec subsystem_id via
``[subsystem_id]``, the revision id via ``[revision_id]`` line.
Also, the codec chip name can be rewritten via ``[chip_name]`` line.
::
[codec]
0x12345678 0xabcd1234 2
[subsystem_id]
0xffff1111
[revision_id]
0x10
[chip_name]
My-own NEWS-0002
The hd-audio driver reads the file via request_firmware(). Thus,
a patch file has to be located on the appropriate firmware path,
typically, /lib/firmware. For example, when you pass the option
``patch=hda-init.fw``, the file /lib/firmware/hda-init.fw must be
present.
The patch module option is specific to each card instance, and you
need to give one file name for each instance, separated by commas.
For example, if you have two cards, one for an on-board analog and one
for an HDMI video board, you may pass patch option like below:
::
options snd-hda-intel patch=on-board-patch,hdmi-patch
Power-Saving
------------
The power-saving is a kind of auto-suspend of the device. When the
device is inactive for a certain time, the device is automatically
turned off to save the power. The time to go down is specified via
``power_save`` module option, and this option can be changed dynamically
via sysfs.
The power-saving won't work when the analog loopback is enabled on
some codecs. Make sure that you mute all unneeded signal routes when
you want the power-saving.
The power-saving feature might cause audible click noises at each
power-down/up depending on the device. Some of them might be
solvable, but some are hard, I'm afraid. Some distros such as
openSUSE enables the power-saving feature automatically when the power
cable is unplugged. Thus, if you hear noises, suspect first the
power-saving. See /sys/module/snd_hda_intel/parameters/power_save to
check the current value. If it's non-zero, the feature is turned on.
The recent kernel supports the runtime PM for the HD-audio controller
chip, too. It means that the HD-audio controller is also powered up /
down dynamically. The feature is enabled only for certain controller
chips like Intel LynxPoint. You can enable/disable this feature
forcibly by setting ``power_save_controller`` option, which is also
available at /sys/module/snd_hda_intel/parameters directory.
Tracepoints
-----------
The hd-audio driver gives a few basic tracepoints.
``hda:hda_send_cmd`` traces each CORB write while ``hda:hda_get_response``
traces the response from RIRB (only when read from the codec driver).
``hda:hda_bus_reset`` traces the bus-reset due to fatal error, etc,
``hda:hda_unsol_event`` traces the unsolicited events, and
``hda:hda_power_down`` and ``hda:hda_power_up`` trace the power down/up
via power-saving behavior.
Enabling all tracepoints can be done like
::
# echo 1 > /sys/kernel/tracing/events/hda/enable
then after some commands, you can traces from
/sys/kernel/tracing/trace file. For example, when you want to
trace what codec command is sent, enable the tracepoint like:
::
# cat /sys/kernel/tracing/trace
# tracer: nop
#
# TASK-PID CPU# TIMESTAMP FUNCTION
# | | | | |
<...>-7807 [002] 105147.774889: hda_send_cmd: [0:0] val=e3a019
<...>-7807 [002] 105147.774893: hda_send_cmd: [0:0] val=e39019
<...>-7807 [002] 105147.999542: hda_send_cmd: [0:0] val=e3a01a
<...>-7807 [002] 105147.999543: hda_send_cmd: [0:0] val=e3901a
<...>-26764 [001] 349222.837143: hda_send_cmd: [0:0] val=e3a019
<...>-26764 [001] 349222.837148: hda_send_cmd: [0:0] val=e39019
<...>-26764 [001] 349223.058539: hda_send_cmd: [0:0] val=e3a01a
<...>-26764 [001] 349223.058541: hda_send_cmd: [0:0] val=e3901a
Here ``[0:0]`` indicates the card number and the codec address, and
``val`` shows the value sent to the codec, respectively. The value is
a packed value, and you can decode it via hda-decode-verb program
included in hda-emu package below. For example, the value e3a019 is
to set the left output-amp value to 25.
::
% hda-decode-verb 0xe3a019
raw value = 0x00e3a019
cid = 0, nid = 0x0e, verb = 0x3a0, parm = 0x19
raw value: verb = 0x3a0, parm = 0x19
verbname = set_amp_gain_mute
amp raw val = 0xa019
output, left, idx=0, mute=0, val=25
Development Tree
----------------
The latest development codes for HD-audio are found on sound git tree:
* git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git
The master branch or for-next branches can be used as the main
development branches in general while the development for the current
and next kernels are found in for-linus and for-next branches,
respectively.
Sending a Bug Report
--------------------
If any model or module options don't work for your device, it's time
to send a bug report to the developers. Give the following in your
bug report:
* Hardware vendor, product and model names
* Kernel version (and ALSA-driver version if you built externally)
* ``alsa-info.sh`` output; run with ``--no-upload`` option. See the
section below about alsa-info
If it's a regression, at best, send alsa-info outputs of both working
and non-working kernels. This is really helpful because we can
compare the codec registers directly.
Send a bug report either the following:
kernel-bugzilla
https://bugzilla.kernel.org/
alsa-devel ML
alsa-devel@alsa-project.org
Debug Tools
===========
This section describes some tools available for debugging HD-audio
problems.
alsa-info
---------
The script ``alsa-info.sh`` is a very useful tool to gather the audio
device information. It's included in alsa-utils package. The latest
version can be found on git repository:
* git://git.alsa-project.org/alsa-utils.git
The script can be fetched directly from the following URL, too:
* https://www.alsa-project.org/alsa-info.sh
Run this script as root, and it will gather the important information
such as the module lists, module parameters, proc file contents
including the codec proc files, mixer outputs and the control
elements. As default, it will store the information onto a web server
on alsa-project.org. But, if you send a bug report, it'd be better to
run with ``--no-upload`` option, and attach the generated file.
There are some other useful options. See ``--help`` option output for
details.
When a probe error occurs or when the driver obviously assigns a
mismatched model, it'd be helpful to load the driver with
``probe_only=1`` option (at best after the cold reboot) and run
alsa-info at this state. With this option, the driver won't configure
the mixer and PCM but just tries to probe the codec slot. After
probing, the proc file is available, so you can get the raw codec
information before modified by the driver. Of course, the driver
isn't usable with ``probe_only=1``. But you can continue the
configuration via hwdep sysfs file if hda-reconfig option is enabled.
Using ``probe_only`` mask 2 skips the reset of HDA codecs (use
``probe_only=3`` as module option). The hwdep interface can be used
to determine the BIOS codec initialization.
hda-verb
--------
hda-verb is a tiny program that allows you to access the HD-audio
codec directly. You can execute a raw HD-audio codec verb with this.
This program accesses the hwdep device, thus you need to enable the
kernel config ``CONFIG_SND_HDA_HWDEP=y`` beforehand.
The hda-verb program takes four arguments: the hwdep device file, the
widget NID, the verb and the parameter. When you access to the codec
on the slot 2 of the card 0, pass /dev/snd/hwC0D2 to the first
argument, typically. (However, the real path name depends on the
system.)
The second parameter is the widget number-id to access. The third
parameter can be either a hex/digit number or a string corresponding
to a verb. Similarly, the last parameter is the value to write, or
can be a string for the parameter type.
::
% hda-verb /dev/snd/hwC0D0 0x12 0x701 2
nid = 0x12, verb = 0x701, param = 0x2
value = 0x0
% hda-verb /dev/snd/hwC0D0 0x0 PARAMETERS VENDOR_ID
nid = 0x0, verb = 0xf00, param = 0x0
value = 0x10ec0262
% hda-verb /dev/snd/hwC0D0 2 set_a 0xb080
nid = 0x2, verb = 0x300, param = 0xb080
value = 0x0
Although you can issue any verbs with this program, the driver state
won't be always updated. For example, the volume values are usually
cached in the driver, and thus changing the widget amp value directly
via hda-verb won't change the mixer value.
The hda-verb program is included now in alsa-tools:
* git://git.alsa-project.org/alsa-tools.git
Also, the old stand-alone package is found in the ftp directory:
* ftp://ftp.suse.com/pub/people/tiwai/misc/
Also a git repository is available:
* git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-verb.git
See README file in the tarball for more details about hda-verb
program.
hda-analyzer
------------
hda-analyzer provides a graphical interface to access the raw HD-audio
control, based on pyGTK2 binding. It's a more powerful version of
hda-verb. The program gives you an easy-to-use GUI stuff for showing
the widget information and adjusting the amp values, as well as the
proc-compatible output.
The hda-analyzer:
* https://git.alsa-project.org/?p=alsa.git;a=tree;f=hda-analyzer
is a part of alsa.git repository in alsa-project.org:
* git://git.alsa-project.org/alsa.git
Codecgraph
----------
Codecgraph is a utility program to generate a graph and visualizes the
codec-node connection of a codec chip. It's especially useful when
you analyze or debug a codec without a proper datasheet. The program
parses the given codec proc file and converts to SVG via graphiz
program.
The tarball and GIT trees are found in the web page at:
* http://helllabs.org/codecgraph/
hda-emu
-------
hda-emu is an HD-audio emulator. The main purpose of this program is
to debug an HD-audio codec without the real hardware. Thus, it
doesn't emulate the behavior with the real audio I/O, but it just
dumps the codec register changes and the ALSA-driver internal changes
at probing and operating the HD-audio driver.
The program requires a codec proc-file to simulate. Get a proc file
for the target codec beforehand, or pick up an example codec from the
codec proc collections in the tarball. Then, run the program with the
proc file, and the hda-emu program will start parsing the codec file
and simulates the HD-audio driver:
::
% hda-emu codecs/stac9200-dell-d820-laptop
# Parsing..
hda_codec: Unknown model for STAC9200, using BIOS defaults
hda_codec: pin nid 08 bios pin config 40c003fa
....
The program gives you only a very dumb command-line interface. You
can get a proc-file dump at the current state, get a list of control
(mixer) elements, set/get the control element value, simulate the PCM
operation, the jack plugging simulation, etc.
The program is found in the git repository below:
* git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-emu.git
See README file in the repository for more details about hda-emu
program.
hda-jack-retask
---------------
hda-jack-retask is a user-friendly GUI program to manipulate the
HD-audio pin control for jack retasking. If you have a problem about
the jack assignment, try this program and check whether you can get
useful results. Once when you figure out the proper pin assignment,
it can be fixed either in the driver code statically or via passing a
firmware patch file (see "Early Patching" section).
The program is included in alsa-tools now:
* git://git.alsa-project.org/alsa-tools.git
linux-7.2.3/Documentation/sound/hd-audio/realtek-pc-beep.rst
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===============================
Realtek PC Beep Hidden Register
===============================
This file documents the "PC Beep Hidden Register", which is present in certain
Realtek HDA codecs and controls a muxer and pair of passthrough mixers that can
route audio between pins but aren't themselves exposed as HDA widgets. As far
as I can tell, these hidden routes are designed to allow flexible PC Beep output
for codecs that don't have mixer widgets in their output paths. Why it's easier
to hide a mixer behind an undocumented vendor register than to just expose it
as a widget, I have no idea.
Register Description
====================
The register is accessed via processing coefficient 0x36 on NID 20h. Bits not
identified below have no discernible effect on my machine, a Dell XPS 13 9350::
MSB LSB
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |h|S|L| | B |R| | Known bits
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
|0|0|1|1| 0x7 |0|0x0|1| 0x7 | Reset value
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
1Ah input select (B): 2 bits
When zero, expose the PC Beep line (from the internal beep generator, when
enabled with the Set Beep Generation verb on NID 01h, or else from the
external PCBEEP pin) on the 1Ah pin node. When nonzero, expose the headphone
jack (or possibly Line In on some machines) input instead. If PC Beep is
selected, the 1Ah boost control has no effect.
Amplify 1Ah loopback, left (L): 1 bit
Amplify the left channel of 1Ah before mixing it into outputs as specified
by h and S bits. Does not affect the level of 1Ah exposed to other widgets.
Amplify 1Ah loopback, right (R): 1 bit
Amplify the right channel of 1Ah before mixing it into outputs as specified
by h and S bits. Does not affect the level of 1Ah exposed to other widgets.
Loopback 1Ah to 21h [active low] (h): 1 bit
When zero, mix 1Ah (possibly with amplification, depending on L and R bits)
into 21h (headphone jack on my machine). Mixed signal respects the mute
setting on 21h.
Loopback 1Ah to 14h (S): 1 bit
When one, mix 1Ah (possibly with amplification, depending on L and R bits)
into 14h (internal speaker on my machine). Mixed signal **ignores** the mute
setting on 14h and is present whenever 14h is configured as an output.
Path diagrams
=============
1Ah input selection (DIV is the PC Beep divider set on NID 01h)::
<Beep generator> <PCBEEP pin> <Headphone jack>
| | |
+--DIV--+--!DIV--+ {1Ah boost control}
| |
+--(b == 0)--+--(b != 0)--+
|
>1Ah (Beep/Headphone Mic/Line In)<
Loopback of 1Ah to 21h/14h::
<1Ah (Beep/Headphone Mic/Line In)>
|
{amplify if L/R}
|
+-----!h-----+-----S-----+
| |
{21h mute control} |
| |
>21h (Headphone)< >14h (Internal Speaker)<
Background
==========
All Realtek HDA codecs have a vendor-defined widget with node ID 20h which
provides access to a bank of registers that control various codec functions.
Registers are read and written via the standard HDA processing coefficient
verbs (Set/Get Coefficient Index, Set/Get Processing Coefficient). The node is
named "Realtek Vendor Registers" in public datasheets' verb listings and,
apart from that, is entirely undocumented.
This particular register, exposed at coefficient 0x36 and named in commits from
Realtek, is of note: unlike most registers, which seem to control detailed
amplifier parameters not in scope of the HDA specification, it controls audio
routing which could just as easily have been defined using standard HDA mixer
and selector widgets.
Specifically, it selects between two sources for the input pin widget with Node
ID (NID) 1Ah: the widget's signal can come either from an audio jack (on my
laptop, a Dell XPS 13 9350, it's the headphone jack, but comments in Realtek
commits indicate that it might be a Line In on some machines) or from the PC
Beep line (which is itself multiplexed between the codec's internal beep
generator and external PCBEEP pin, depending on if the beep generator is
enabled via verbs on NID 01h). Additionally, it can mix (with optional
amplification) that signal onto the 21h and/or 14h output pins.
The register's reset value is 0x3717, corresponding to PC Beep on 1Ah that is
then amplified and mixed into both the headphones and the speakers. Not only
does this violate the HDA specification, which says that "[a vendor defined
beep input pin] connection may be maintained *only* while the Link reset
(**RST#**) is asserted", it means that we cannot ignore the register if we care
about the input that 1Ah would otherwise expose or if the PCBEEP trace is
poorly shielded and picks up chassis noise (both of which are the case on my
machine).
Unfortunately, there are lots of ways to get this register configuration wrong.
Linux, it seems, has gone through most of them. For one, the register resets
after S3 suspend: judging by existing code, this isn't the case for all vendor
registers, and it's led to some fixes that improve behavior on cold boot but
don't last after suspend. Other fixes have successfully switched the 1Ah input
away from PC Beep but have failed to disable both loopback paths. On my
machine, this means that the headphone input is amplified and looped back to
the headphone output, which uses the exact same pins! As you might expect, this
causes terrible headphone noise, the character of which is controlled by the
1Ah boost control. (If you've seen instructions online to fix XPS 13 headphone
noise by changing "Headphone Mic Boost" in ALSA, now you know why.)
The information here has been obtained through black-box reverse engineering of
the ALC256 codec's behavior and is not guaranteed to be correct. It likely
also applies for the ALC255, ALC257, ALC235, and ALC236, since those codecs
seem to be close relatives of the ALC256. (They all share one initialization
function.) Additionally, other codecs like the ALC225 and ALC285 also have this
register, judging by existing fixups in ``patch_realtek.c``, but specific
data (e.g. node IDs, bit positions, pin mappings) for those codecs may differ
from what I've described here.
linux-7.2.3/Documentation/sound/index.rst
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.. SPDX-License-Identifier: GPL-2.0
=============================
Sound Subsystem Documentation
=============================
.. toctree::
:maxdepth: 2
kernel-api/index
designs/index
soc/index
alsa-configuration
hd-audio/index
cards/index
codecs/index
utimers
linux-7.2.3/Documentation/sound/kernel-api/alsa-driver-api.rst
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===================
The ALSA Driver API
===================
Management of Cards and Devices
===============================
Card Management
---------------
.. kernel-doc:: sound/core/init.c
Device Components
-----------------
.. kernel-doc:: sound/core/device.c
Module requests and Device File Entries
---------------------------------------
.. kernel-doc:: sound/core/sound.c
Memory Management Helpers
-------------------------
.. kernel-doc:: sound/core/memory.c
.. kernel-doc:: sound/core/memalloc.c
PCM API
=======
PCM Core
--------
.. kernel-doc:: sound/core/pcm.c
.. kernel-doc:: sound/core/pcm_lib.c
.. kernel-doc:: sound/core/pcm_native.c
.. kernel-doc:: include/sound/pcm.h
PCM Format Helpers
------------------
.. kernel-doc:: sound/core/pcm_misc.c
PCM Memory Management
---------------------
.. kernel-doc:: sound/core/pcm_memory.c
PCM DMA Engine API
------------------
.. kernel-doc:: sound/core/pcm_dmaengine.c
.. kernel-doc:: include/sound/dmaengine_pcm.h
Control/Mixer API
=================
General Control Interface
-------------------------
.. kernel-doc:: sound/core/control.c
AC97 Codec API
--------------
.. kernel-doc:: sound/pci/ac97/ac97_codec.c
.. kernel-doc:: sound/pci/ac97/ac97_pcm.c
Virtual Master Control API
--------------------------
.. kernel-doc:: sound/core/vmaster.c
.. kernel-doc:: include/sound/control.h
MIDI API
========
Raw MIDI API
------------
.. kernel-doc:: sound/core/rawmidi.c
MPU401-UART API
---------------
.. kernel-doc:: sound/drivers/mpu401/mpu401_uart.c
Proc Info API
=============
Proc Info Interface
-------------------
.. kernel-doc:: sound/core/info.c
Compress Offload
================
Compress Offload API
--------------------
.. kernel-doc:: sound/core/compress_offload.c
.. kernel-doc:: include/uapi/sound/compress_offload.h
.. kernel-doc:: include/uapi/sound/compress_params.h
.. kernel-doc:: include/sound/compress_driver.h
ASoC
====
ASoC Core API
-------------
.. kernel-doc:: include/sound/soc.h
.. kernel-doc:: sound/soc/soc-core.c
.. kernel-doc:: sound/soc/soc-devres.c
.. kernel-doc:: sound/soc/soc-component.c
.. kernel-doc:: sound/soc/soc-pcm.c
.. kernel-doc:: sound/soc/soc-ops.c
.. kernel-doc:: sound/soc/soc-compress.c
ASoC DAPM API
-------------
.. kernel-doc:: sound/soc/soc-dapm.c
ASoC DMA Engine API
-------------------
.. kernel-doc:: sound/soc/soc-generic-dmaengine-pcm.c
Miscellaneous Functions
=======================
Hardware-Dependent Devices API
------------------------------
.. kernel-doc:: sound/core/hwdep.c
Jack Abstraction Layer API
--------------------------
.. kernel-doc:: include/sound/jack.h
.. kernel-doc:: sound/core/jack.c
.. kernel-doc:: sound/soc/soc-jack.c
ISA DMA Helpers
---------------
.. kernel-doc:: sound/core/isadma.c
Other Helper Macros
-------------------
.. kernel-doc:: include/sound/core.h
.. kernel-doc:: sound/sound_core.c
linux-7.2.3/Documentation/sound/kernel-api/index.rst
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ALSA Kernel API Documentation
=============================
.. toctree::
:maxdepth: 2
alsa-driver-api
writing-an-alsa-driver
linux-7.2.3/Documentation/sound/kernel-api/writing-an-alsa-driver.rst
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linux-7.2.3/Documentation/sound/soc/clocking.rst
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==============
Audio Clocking
==============
This text describes the audio clocking terms in ASoC and digital audio in
general. Note: Audio clocking can be complex!
Master Clock
------------
Every audio subsystem is driven by a master clock (sometimes referred to as MCLK
or SYSCLK). This audio master clock can be derived from a number of sources
(e.g. crystal, PLL, CPU clock) and is responsible for producing the correct
audio playback and capture sample rates.
Some master clocks (e.g. PLLs and CPU based clocks) are configurable in that
their speed can be altered by software (depending on the system use and to save
power). Other master clocks are fixed at a set frequency (i.e. crystals).
DAI Clocks
----------
The Digital Audio Interface is usually driven by a Bit Clock (often referred to
as BCLK). This clock is used to drive the digital audio data across the link
between the codec and CPU.
The DAI also has a frame clock to signal the start of each audio frame. This
clock is sometimes referred to as LRC (left right clock) or FRAME. This clock
runs at exactly the sample rate (LRC = Rate).
Bit Clock can be generated as follows:-
- BCLK = MCLK / x, or
- BCLK = LRC * x, or
- BCLK = LRC * Channels * Word Size
This relationship depends on the codec or SoC CPU in particular. In general
it is best to configure BCLK to the lowest possible speed (depending on your
rate, number of channels and word size) to save on power.
It is also desirable to use the codec (if possible) to drive (or master) the
audio clocks as it usually gives more accurate sample rates than the CPU.
ASoC provided clock APIs
------------------------
.. kernel-doc:: sound/soc/soc-dai.c
:identifiers: snd_soc_dai_set_sysclk
.. kernel-doc:: sound/soc/soc-dai.c
:identifiers: snd_soc_dai_set_clkdiv
.. kernel-doc:: sound/soc/soc-dai.c
:identifiers: snd_soc_dai_set_pll
.. kernel-doc:: sound/soc/soc-dai.c
:identifiers: snd_soc_dai_set_bclk_ratio
linux-7.2.3/Documentation/sound/soc/codec-to-codec.rst
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==============================================
Creating codec to codec dai link for ALSA dapm
==============================================
Mostly the flow of audio is always from CPU to codec so your system
will look as below:
::
--------- ---------
| | dai | |
CPU -------> codec
| | | |
--------- ---------
In case your system looks as below:
::
---------
| |
codec-2
| |
---------
|
dai-2
|
---------- ---------
| | dai-1 | |
CPU -------> codec-1
| | | |
---------- ---------
|
dai-3
|
---------
| |
codec-3
| |
---------
Suppose codec-2 is a bluetooth chip and codec-3 is connected to
a speaker and you have a below scenario:
codec-2 will receive the audio data and the user wants to play that
audio through codec-3 without involving the CPU.This
aforementioned case is the ideal case when codec to codec
connection should be used.
Your dai_link should appear as below in your machine
file:
::
/*
* this pcm stream only supports 24 bit, 2 channel and
* 48k sampling rate.
*/
static const struct snd_soc_pcm_stream dsp_codec_params = {
.formats = SNDRV_PCM_FMTBIT_S24_LE,
.rate_min = 48000,
.rate_max = 48000,
.channels_min = 2,
.channels_max = 2,
};
{
.name = "CPU-DSP",
.stream_name = "CPU-DSP",
.cpu_dai_name = "samsung-i2s.0",
.codec_name = "codec-2,
.codec_dai_name = "codec-2-dai_name",
.platform_name = "samsung-i2s.0",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBP_CFP,
.ignore_suspend = 1,
.c2c_params = &dsp_codec_params,
.num_c2c_params = 1,
},
{
.name = "DSP-CODEC",
.stream_name = "DSP-CODEC",
.cpu_dai_name = "wm0010-sdi2",
.codec_name = "codec-3,
.codec_dai_name = "codec-3-dai_name",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBP_CFP,
.ignore_suspend = 1,
.c2c_params = &dsp_codec_params,
.num_c2c_params = 1,
},
Above code snippet is motivated from sound/soc/samsung/speyside.c.
Note the "c2c_params" callback which lets the dapm know that this
dai_link is a codec to codec connection.
In dapm core a route is created between cpu_dai playback widget
and codec_dai capture widget for playback path and vice-versa is
true for capture path. In order for this aforementioned route to get
triggered, DAPM needs to find a valid endpoint which could be either
a sink or source widget corresponding to playback and capture path
respectively.
In order to trigger this dai_link widget, a thin codec driver for
the speaker amp can be created as demonstrated in wm8727.c file, it
sets appropriate constraints for the device even if it needs no control.
Make sure to name your corresponding cpu and codec playback and capture
dai names ending with "Playback" and "Capture" respectively as dapm core
will link and power those dais based on the name.
A dai_link in a "simple-audio-card" will automatically be detected as
codec to codec when all DAIs on the link belong to codec components.
The dai_link will be initialized with the subset of stream parameters
(channels, format, sample rate) supported by all DAIs on the link. Since
there is no way to provide these parameters in the device tree, this is
mostly useful for communication with simple fixed-function codecs, such
as a Bluetooth controller or cellular modem.
linux-7.2.3/Documentation/sound/soc/codec.rst
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=======================
ASoC Codec Class Driver
=======================
The codec class driver is generic and hardware independent code that configures
the codec, FM, MODEM, BT or external DSP to provide audio capture and playback.
It should contain no code that is specific to the target platform or machine.
All platform and machine specific code should be added to the platform and
machine drivers respectively.
Each codec class driver *must* provide the following features:-
1. Codec DAI and PCM configuration
2. Codec control IO - using RegMap API
3. Mixers and audio controls
4. Codec audio operations
5. DAPM description.
6. DAPM event handler.
Optionally, codec drivers can also provide:-
7. DAC Digital mute control.
Its probably best to use this guide in conjunction with the existing codec
driver code in sound/soc/codecs/
ASoC Codec driver breakdown
===========================
Codec DAI and PCM configuration
-------------------------------
Each codec driver must have a struct snd_soc_dai_driver to define its DAI and
PCM capabilities and operations. This struct is exported so that it can be
registered with the core by your machine driver.
e.g.
::
static struct snd_soc_dai_ops wm8731_dai_ops = {
.prepare = wm8731_pcm_prepare,
.hw_params = wm8731_hw_params,
.shutdown = wm8731_shutdown,
.mute_stream = wm8731_mute,
.set_sysclk = wm8731_set_dai_sysclk,
.set_fmt = wm8731_set_dai_fmt,
};
struct snd_soc_dai_driver wm8731_dai = {
.name = "wm8731-hifi",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 2,
.rates = WM8731_RATES,
.formats = WM8731_FORMATS,},
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM8731_RATES,
.formats = WM8731_FORMATS,},
.ops = &wm8731_dai_ops,
.symmetric_rate = 1,
};
Codec control IO
----------------
The codec can usually be controlled via an I2C or SPI style interface
(AC97 combines control with data in the DAI). The codec driver should use the
Regmap API for all codec IO. Please see include/linux/regmap.h and existing
codec drivers for example regmap usage.
Mixers and audio controls
-------------------------
All the codec mixers and audio controls can be defined using the convenience
macros defined in soc.h.
::
#define SOC_SINGLE(xname, reg, shift, mask, invert)
Defines a single control as follows:-
::
xname = Control name e.g. "Playback Volume"
reg = codec register
shift = control bit(s) offset in register
mask = control bit size(s) e.g. mask of 7 = 3 bits
invert = the control is inverted
Other macros include:-
::
#define SOC_DOUBLE(xname, reg, shift_left, shift_right, mask, invert)
A stereo control
::
#define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, mask, invert)
A stereo control spanning 2 registers
::
#define SOC_ENUM_SINGLE(xreg, xshift, xmask, xtexts)
Defines an single enumerated control as follows:-
::
xreg = register
xshift = control bit(s) offset in register
xmask = control bit(s) size
xtexts = pointer to array of strings that describe each setting
#define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts)
Defines a stereo enumerated control
Codec Audio Operations
----------------------
The codec driver also supports the following ALSA PCM operations:-
::
/* SoC audio ops */
struct snd_soc_ops {
int (*startup)(struct snd_pcm_substream *);
void (*shutdown)(struct snd_pcm_substream *);
int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *);
int (*hw_free)(struct snd_pcm_substream *);
int (*prepare)(struct snd_pcm_substream *);
};
Please refer to the :doc:`ALSA driver PCM documentation
<../kernel-api/writing-an-alsa-driver>` for details.
DAPM description
----------------
The Dynamic Audio Power Management description describes the codec power
components and their relationships and registers to the ASoC core.
Please read dapm.rst for details of building the description.
Please also see the examples in other codec drivers.
DAPM event handler
------------------
This function is a callback that handles codec domain PM calls and system
domain PM calls (e.g. suspend and resume). It is used to put the codec
to sleep when not in use.
Power states:-
::
SNDRV_CTL_POWER_D0: /* full On */
/* vref/mid, clk and osc on, active */
SNDRV_CTL_POWER_D1: /* partial On */
SNDRV_CTL_POWER_D2: /* partial On */
SNDRV_CTL_POWER_D3hot: /* Off, with power */
/* everything off except vref/vmid, inactive */
SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */
Codec DAC digital mute control
------------------------------
Most codecs have a digital mute before the DACs that can be used to
minimise any system noise. The mute stops any digital data from
entering the DAC.
A callback can be created that is called by the core for each codec DAI
when the mute is applied or freed.
i.e.
::
static int wm8974_mute(struct snd_soc_dai *dai, int mute, int direction)
{
struct snd_soc_component *component = dai->component;
u16 mute_reg = snd_soc_component_read(component, WM8974_DAC) & 0xffbf;
if (mute)
snd_soc_component_write(component, WM8974_DAC, mute_reg | 0x40);
else
snd_soc_component_write(component, WM8974_DAC, mute_reg);
return 0;
}
linux-7.2.3/Documentation/sound/soc/dai.rst
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==================================
ASoC Digital Audio Interface (DAI)
==================================
ASoC currently supports the three main Digital Audio Interfaces (DAI) found on
SoC controllers and portable audio CODECs today, namely AC97, I2S and PCM.
AC97
====
AC97 is a five wire interface commonly found on many PC sound cards. It is
now also popular in many portable devices. This DAI has a RESET line and time
multiplexes its data on its SDATA_OUT (playback) and SDATA_IN (capture) lines.
The bit clock (BCLK) is always driven by the CODEC (usually 12.288MHz) and the
frame (FRAME) (usually 48kHz) is always driven by the controller. Each AC97
frame is 21uS long and is divided into 13 time slots.
The AC97 specification can be found at :
https://www.intel.com/p/en_US/business/design
I2S
===
I2S is a common 4 wire DAI used in HiFi, STB and portable devices. The Tx and
Rx lines are used for audio transmission, while the bit clock (BCLK) and
left/right clock (LRC) synchronise the link. I2S is flexible in that either the
controller or CODEC can drive (master) the BCLK and LRC clock lines. Bit clock
usually varies depending on the sample rate and the master system clock
(SYSCLK). LRCLK is the same as the sample rate. A few devices support separate
ADC and DAC LRCLKs, this allows for simultaneous capture and playback at
different sample rates.
I2S has several different operating modes:-
I2S
MSB is transmitted on the falling edge of the first BCLK after LRC
transition.
Left Justified
MSB is transmitted on transition of LRC.
Right Justified
MSB is transmitted sample size BCLKs before LRC transition.
PCM
===
PCM is another 4 wire interface, very similar to I2S, which can support a more
flexible protocol. It has bit clock (BCLK) and sync (SYNC) lines that are used
to synchronise the link while the Tx and Rx lines are used to transmit and
receive the audio data. Bit clock usually varies depending on sample rate
while sync runs at the sample rate. PCM also supports Time Division
Multiplexing (TDM) in that several devices can use the bus simultaneously (this
is sometimes referred to as network mode).
Common PCM operating modes:-
Mode A
MSB is transmitted on falling edge of first BCLK after FRAME/SYNC.
Mode B
MSB is transmitted on rising edge of FRAME/SYNC.
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linux-7.2.3/Documentation/sound/soc/dapm.rst
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===================================================
Dynamic Audio Power Management for Portable Devices
===================================================
Description
===========
Dynamic Audio Power Management (DAPM) is designed to allow portable
Linux devices to use the minimum amount of power within the audio
subsystem at all times. It is independent of other kernel power
management frameworks and, as such, can easily co-exist with them.
DAPM is also completely transparent to all user space applications as
all power switching is done within the ASoC core. No code changes or
recompiling are required for user space applications. DAPM makes power
switching decisions based upon any audio stream (capture/playback)
activity and audio mixer settings within the device.
DAPM is based on two basic elements, called widgets and routes:
* a **widget** is every part of the audio hardware that can be enabled by
software when in use and disabled to save power when not in use
* a **route** is an interconnection between widgets that exists when sound
can flow from one widget to the other
All DAPM power switching decisions are made automatically by consulting an
audio routing graph. This graph is specific to each sound card and spans
the whole sound card, so some DAPM routes connect two widgets belonging to
different components (e.g. the LINE OUT pin of a CODEC and the input pin of
an amplifier).
The graph for the STM32MP1-DK1 sound card is shown in picture:
.. kernel-figure:: dapm-graph.svg
:alt: Example DAPM graph
:align: center
You can also generate compatible graph for your sound card using
`tools/sound/dapm-graph` utility.
DAPM power domains
==================
There are 4 power domains within DAPM:
Codec bias domain
VREF, VMID (core codec and audio power)
Usually controlled at codec probe/remove and suspend/resume, although
can be set at stream time if power is not needed for sidetone, etc.
Platform/Machine domain
physically connected inputs and outputs
Is platform/machine and user action specific, is configured by the
machine driver and responds to asynchronous events e.g when HP
are inserted
Path domain
audio subsystem signal paths
Automatically set when mixer and mux settings are changed by the user.
e.g. alsamixer, amixer.
Stream domain
DACs and ADCs.
Enabled and disabled when stream playback/capture is started and
stopped respectively. e.g. aplay, arecord.
DAPM Widgets
============
Audio DAPM widgets fall into a number of types:
Mixer
Mixes several analog signals into a single analog signal.
Mux
An analog switch that outputs only one of many inputs.
PGA
A programmable gain amplifier or attenuation widget.
ADC
Analog to Digital Converter
DAC
Digital to Analog Converter
Switch
An analog switch
Input
A codec input pin
Output
A codec output pin
Headphone
Headphone (and optional Jack)
Mic
Mic (and optional Jack)
Line
Line Input/Output (and optional Jack)
Speaker
Speaker
Supply
Power or clock supply widget used by other widgets.
Regulator
External regulator that supplies power to audio components.
Clock
External clock that supplies clock to audio components.
AIF IN
Audio Interface Input (with TDM slot mask).
AIF OUT
Audio Interface Output (with TDM slot mask).
Siggen
Signal Generator.
DAI IN
Digital Audio Interface Input.
DAI OUT
Digital Audio Interface Output.
DAI Link
DAI Link between two DAI structures
Pre
Special PRE widget (exec before all others)
Post
Special POST widget (exec after all others)
Buffer
Inter widget audio data buffer within a DSP.
Scheduler
DSP internal scheduler that schedules component/pipeline processing
work.
Effect
Widget that performs an audio processing effect.
SRC
Sample Rate Converter within DSP or CODEC
ASRC
Asynchronous Sample Rate Converter within DSP or CODEC
Encoder
Widget that encodes audio data from one format (usually PCM) to another
usually more compressed format.
Decoder
Widget that decodes audio data from a compressed format to an
uncompressed format like PCM.
(Widgets are defined in include/sound/soc-dapm.h)
Widgets can be added to the sound card by any of the component driver types.
There are convenience macros defined in soc-dapm.h that can be used to quickly
build a list of widgets of the codecs and machines DAPM widgets.
Most widgets have a name, register, shift and invert. Some widgets have extra
parameters for stream name and kcontrols.
Stream Domain Widgets
---------------------
Stream Widgets relate to the stream power domain and only consist of ADCs
(analog to digital converters), DACs (digital to analog converters),
AIF IN and AIF OUT.
Stream widgets have the following format:
::
SND_SOC_DAPM_DAC(name, stream name, reg, shift, invert),
SND_SOC_DAPM_AIF_IN(name, stream, slot, reg, shift, invert)
NOTE: the stream name must match the corresponding stream name in your codec
snd_soc_dai_driver.
e.g. stream widgets for HiFi playback and capture
::
SND_SOC_DAPM_DAC("HiFi DAC", "HiFi Playback", REG, 3, 1),
SND_SOC_DAPM_ADC("HiFi ADC", "HiFi Capture", REG, 2, 1),
e.g. stream widgets for AIF
::
SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0),
Path Domain Widgets
-------------------
Path domain widgets have a ability to control or affect the audio signal or
audio paths within the audio subsystem. They have the following form:
::
SND_SOC_DAPM_PGA(name, reg, shift, invert, controls, num_controls)
Any widget kcontrols can be set using the controls and num_controls members.
e.g. Mixer widget (the kcontrols are declared first)
::
/* Output Mixer */
static const snd_kcontrol_new_t wm8731_output_mixer_controls[] = {
SOC_DAPM_SINGLE("Line Bypass Switch", WM8731_APANA, 3, 1, 0),
SOC_DAPM_SINGLE("Mic Sidetone Switch", WM8731_APANA, 5, 1, 0),
SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0),
};
SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, wm8731_output_mixer_controls,
ARRAY_SIZE(wm8731_output_mixer_controls)),
If you don't want the mixer elements prefixed with the name of the mixer widget,
you can use SND_SOC_DAPM_MIXER_NAMED_CTL instead. the parameters are the same
as for SND_SOC_DAPM_MIXER.
Machine domain Widgets
----------------------
Machine widgets are different from codec widgets in that they don't have a
codec register bit associated with them. A machine widget is assigned to each
machine audio component (non codec or DSP) that can be independently
powered. e.g.
* Speaker Amp
* Microphone Bias
* Jack connectors
A machine widget can have an optional call back.
e.g. Jack connector widget for an external Mic that enables Mic Bias
when the Mic is inserted::
static int spitz_mic_bias(struct snd_soc_dapm_widget* w, int event)
{
gpio_set_value(SPITZ_GPIO_MIC_BIAS, SND_SOC_DAPM_EVENT_ON(event));
return 0;
}
SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias),
Codec (BIAS) Domain
-------------------
The codec bias power domain has no widgets and is handled by the codec DAPM
event handler. This handler is called when the codec powerstate is changed wrt
to any stream event or by kernel PM events.
Virtual Widgets
---------------
Sometimes widgets exist in the codec or machine audio graph that don't have any
corresponding soft power control. In this case it is necessary to create
a virtual widget - a widget with no control bits e.g.
::
SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
This can be used to merge two signal paths together in software.
Registering DAPM controls
=========================
In many cases the DAPM widgets are implemented statically in a ``static
const struct snd_soc_dapm_widget`` array in a codec driver, and simply
declared via the ``dapm_widgets`` and ``num_dapm_widgets`` fields of the
``struct snd_soc_component_driver``.
Similarly, routes connecting them are implemented statically in a ``static
const struct snd_soc_dapm_route`` array and declared via the
``dapm_routes`` and ``num_dapm_routes`` fields of the same struct.
With the above declared, the driver registration will take care of
populating them::
static const struct snd_soc_dapm_widget wm2000_dapm_widgets[] = {
SND_SOC_DAPM_OUTPUT("SPKN"),
SND_SOC_DAPM_OUTPUT("SPKP"),
...
};
/* Target, Path, Source */
static const struct snd_soc_dapm_route wm2000_audio_map[] = {
{ "SPKN", NULL, "ANC Engine" },
{ "SPKP", NULL, "ANC Engine" },
...
};
static const struct snd_soc_component_driver soc_component_dev_wm2000 = {
...
.dapm_widgets = wm2000_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(wm2000_dapm_widgets),
.dapm_routes = wm2000_audio_map,
.num_dapm_routes = ARRAY_SIZE(wm2000_audio_map),
...
};
In more complex cases the list of DAPM widgets and/or routes can be only
known at probe time. This happens for example when a driver supports
different models having a different set of features. In those cases
separate widgets and routes arrays implementing the case-specific features
can be registered programmatically by calling snd_soc_dapm_new_controls()
and snd_soc_dapm_add_routes().
Codec/DSP Widget Interconnections
=================================
Widgets are connected to each other within the codec, platform and machine by
audio paths (called interconnections). Each interconnection must be defined in
order to create a graph of all audio paths between widgets.
This is easiest with a diagram of the codec or DSP (and schematic of the machine
audio system), as it requires joining widgets together via their audio signal
paths.
For example the WM8731 output mixer (wm8731.c) has 3 inputs (sources):
1. Line Bypass Input
2. DAC (HiFi playback)
3. Mic Sidetone Input
Each input in this example has a kcontrol associated with it (defined in
the example above) and is connected to the output mixer via its kcontrol
name. We can now connect the destination widget (wrt audio signal) with its
source widgets. ::
/* output mixer */
{"Output Mixer", "Line Bypass Switch", "Line Input"},
{"Output Mixer", "HiFi Playback Switch", "DAC"},
{"Output Mixer", "Mic Sidetone Switch", "Mic Bias"},
So we have:
* Destination Widget <=== Path Name <=== Source Widget, or
* Sink, Path, Source, or
* ``Output Mixer`` is connected to the ``DAC`` via the ``HiFi Playback Switch``.
When there is no path name connecting widgets (e.g. a direct connection) we
pass NULL for the path name.
Interconnections are created with a call to::
snd_soc_dapm_connect_input(codec, sink, path, source);
Finally, snd_soc_dapm_new_widgets() must be called after all widgets and
interconnections have been registered with the core. This causes the core to
scan the codec and machine so that the internal DAPM state matches the
physical state of the machine.
Machine Widget Interconnections
-------------------------------
Machine widget interconnections are created in the same way as codec ones and
directly connect the codec pins to machine level widgets.
e.g. connects the speaker out codec pins to the internal speaker.
::
/* ext speaker connected to codec pins LOUT2, ROUT2 */
{"Ext Spk", NULL , "ROUT2"},
{"Ext Spk", NULL , "LOUT2"},
This allows the DAPM to power on and off pins that are connected (and in use)
and pins that are NC respectively.
Endpoint Widgets
================
An endpoint is a start or end point (widget) of an audio signal within the
machine and includes the codec. e.g.
* Headphone Jack
* Internal Speaker
* Internal Mic
* Mic Jack
* Codec Pins
Endpoints are added to the DAPM graph so that their usage can be determined in
order to save power. e.g. NC codecs pins will be switched OFF, unconnected
jacks can also be switched OFF.
DAPM Widget Events
==================
Widgets needing to implement a more complex behaviour than what DAPM can do
can set a custom "event handler" by setting a function pointer. An example
is a power supply needing to enable a GPIO::
static int sof_es8316_speaker_power_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
if (SND_SOC_DAPM_EVENT_ON(event))
gpiod_set_value_cansleep(gpio_pa, true);
else
gpiod_set_value_cansleep(gpio_pa, false);
return 0;
}
static const struct snd_soc_dapm_widget st_widgets[] = {
...
SND_SOC_DAPM_SUPPLY("Speaker Power", SND_SOC_NOPM, 0, 0,
sof_es8316_speaker_power_event,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
};
See soc-dapm.h for all other widgets that support events.
Event types
-----------
The following event types are supported by event widgets::
/* dapm event types */
#define SND_SOC_DAPM_PRE_PMU 0x1 /* before widget power up */
#define SND_SOC_DAPM_POST_PMU 0x2 /* after widget power up */
#define SND_SOC_DAPM_PRE_PMD 0x4 /* before widget power down */
#define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */
#define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */
#define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */
#define SND_SOC_DAPM_WILL_PMU 0x40 /* called at start of sequence */
#define SND_SOC_DAPM_WILL_PMD 0x80 /* called at start of sequence */
#define SND_SOC_DAPM_PRE_POST_PMD (SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD)
#define SND_SOC_DAPM_PRE_POST_PMU (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU)
linux-7.2.3/Documentation/sound/soc/dpcm.rst
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===========
Dynamic PCM
===========
Description
===========
Dynamic PCM allows an ALSA PCM device to digitally route its PCM audio to
various digital endpoints during the PCM stream runtime. e.g. PCM0 can route
digital audio to I2S DAI0, I2S DAI1 or PDM DAI2. This is useful for on SoC DSP
drivers that expose several ALSA PCMs and can route to multiple DAIs.
The DPCM runtime routing is determined by the ALSA mixer settings in the same
way as the analog signal is routed in an ASoC codec driver. DPCM uses a DAPM
graph representing the DSP internal audio paths and uses the mixer settings to
determine the path used by each ALSA PCM.
DPCM re-uses all the existing component codec, platform and DAI drivers without
any modifications.
Phone Audio System with SoC based DSP
-------------------------------------
Consider the following phone audio subsystem. This will be used in this
document for all examples :-
::
| Front End PCMs | SoC DSP | Back End DAIs | Audio devices |
*************
PCM0 <------------> * * <----DAI0-----> Codec Headset
* *
PCM1 <------------> * * <----DAI1-----> Codec Speakers
* DSP *
PCM2 <------------> * * <----DAI2-----> MODEM
* *
PCM3 <------------> * * <----DAI3-----> BT
* *
* * <----DAI4-----> DMIC
* *
* * <----DAI5-----> FM
*************
This diagram shows a simple smart phone audio subsystem. It supports Bluetooth,
FM digital radio, Speakers, Headset Jack, digital microphones and cellular
modem. This sound card exposes 4 DSP front end (FE) ALSA PCM devices and
supports 6 back end (BE) DAIs. Each FE PCM can digitally route audio data to any
of the BE DAIs. The FE PCM devices can also route audio to more than 1 BE DAI.
Example - DPCM Switching playback from DAI0 to DAI1
---------------------------------------------------
Audio is being played to the Headset. After a while the user removes the headset
and audio continues playing on the speakers.
Playback on PCM0 to Headset would look like :-
::
*************
PCM0 <============> * * <====DAI0=====> Codec Headset
* *
PCM1 <------------> * * <----DAI1-----> Codec Speakers
* DSP *
PCM2 <------------> * * <----DAI2-----> MODEM
* *
PCM3 <------------> * * <----DAI3-----> BT
* *
* * <----DAI4-----> DMIC
* *
* * <----DAI5-----> FM
*************
The headset is removed from the jack by user so the speakers must now be used :-
::
*************
PCM0 <============> * * <----DAI0-----> Codec Headset
* *
PCM1 <------------> * * <====DAI1=====> Codec Speakers
* DSP *
PCM2 <------------> * * <----DAI2-----> MODEM
* *
PCM3 <------------> * * <----DAI3-----> BT
* *
* * <----DAI4-----> DMIC
* *
* * <----DAI5-----> FM
*************
The audio driver processes this as follows :-
1. Machine driver receives Jack removal event.
2. Machine driver OR audio HAL disables the Headset path.
3. DPCM runs the PCM trigger(stop), hw_free(), shutdown() operations on DAI0
for headset since the path is now disabled.
4. Machine driver or audio HAL enables the speaker path.
5. DPCM runs the PCM ops for startup(), hw_params(), prepare() and
trigger(start) for DAI1 Speakers since the path is enabled.
In this example, the machine driver or userspace audio HAL can alter the routing
and then DPCM will take care of managing the DAI PCM operations to either bring
the link up or down. Audio playback does not stop during this transition.
DPCM machine driver
===================
The DPCM enabled ASoC machine driver is similar to normal machine drivers
except that we also have to :-
1. Define the FE and BE DAI links.
2. Define any FE/BE PCM operations.
3. Define widget graph connections.
FE and BE DAI links
-------------------
::
| Front End PCMs | SoC DSP | Back End DAIs | Audio devices |
*************
PCM0 <------------> * * <----DAI0-----> Codec Headset
* *
PCM1 <------------> * * <----DAI1-----> Codec Speakers
* DSP *
PCM2 <------------> * * <----DAI2-----> MODEM
* *
PCM3 <------------> * * <----DAI3-----> BT
* *
* * <----DAI4-----> DMIC
* *
* * <----DAI5-----> FM
*************
For the example above we have to define 4 FE DAI links and 6 BE DAI links. The
FE DAI links are defined as follows :-
::
SND_SOC_DAILINK_DEFS(pcm0,
DAILINK_COMP_ARRAY(COMP_CPU("System Pin")),
DAILINK_COMP_ARRAY(COMP_DUMMY()),
DAILINK_COMP_ARRAY(COMP_PLATFORM("dsp-audio")));
static struct snd_soc_dai_link machine_dais[] = {
{
.name = "PCM0 System",
.stream_name = "System Playback",
SND_SOC_DAILINK_REG(pcm0),
.dynamic = 1,
.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
},
.....< other FE and BE DAI links here >
};
This FE DAI link is pretty similar to a regular DAI link except that we also
set the DAI link to a DPCM FE with the ``dynamic = 1``.
There is also an option to specify the ordering of the trigger call for
each FE. This allows the ASoC core to trigger the DSP before or after the other
components (as some DSPs have strong requirements for the ordering DAI/DSP
start and stop sequences).
The FE DAI above sets the codec and code DAIs to dummy devices since the BE is
dynamic and will change depending on runtime config.
The BE DAIs are configured as follows :-
::
SND_SOC_DAILINK_DEFS(headset,
DAILINK_COMP_ARRAY(COMP_CPU("ssp-dai.0")),
DAILINK_COMP_ARRAY(COMP_CODEC("rt5640.0-001c", "rt5640-aif1")));
static struct snd_soc_dai_link machine_dais[] = {
.....< FE DAI links here >
{
.name = "Codec Headset",
SND_SOC_DAILINK_REG(headset),
.no_pcm = 1,
.ignore_suspend = 1,
.ignore_pmdown_time = 1,
.be_hw_params_fixup = hswult_ssp0_fixup,
.ops = &haswell_ops,
},
.....< other BE DAI links here >
};
This BE DAI link connects DAI0 to the codec (in this case RT5460 AIF1). It sets
the ``no_pcm`` flag to mark it has a BE.
The BE has also flags set for ignoring suspend and PM down time. This allows
the BE to work in a hostless mode where the host CPU is not transferring data
like a BT phone call :-
::
*************
PCM0 <------------> * * <----DAI0-----> Codec Headset
* *
PCM1 <------------> * * <----DAI1-----> Codec Speakers
* DSP *
PCM2 <------------> * * <====DAI2=====> MODEM
* *
PCM3 <------------> * * <====DAI3=====> BT
* *
* * <----DAI4-----> DMIC
* *
* * <----DAI5-----> FM
*************
This allows the host CPU to sleep while the DSP, MODEM DAI and the BT DAI are
still in operation.
A BE DAI link can also set the codec to a dummy device if the codec is a device
that is managed externally.
Likewise a BE DAI can also set a dummy cpu DAI if the CPU DAI is managed by the
DSP firmware.
FE/BE PCM operations
--------------------
The BE above also exports some PCM operations and a ``fixup`` callback. The fixup
callback is used by the machine driver to (re)configure the DAI based upon the
FE hw params. i.e. the DSP may perform SRC or ASRC from the FE to BE.
e.g. DSP converts all FE hw params to run at fixed rate of 48k, 16bit, stereo for
DAI0. This means all FE hw_params have to be fixed in the machine driver for
DAI0 so that the DAI is running at desired configuration regardless of the FE
configuration.
::
static int dai0_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
{
struct snd_interval *rate = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_RATE);
struct snd_interval *channels = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
/* The DSP will convert the FE rate to 48k, stereo */
rate->min = rate->max = 48000;
channels->min = channels->max = 2;
/* set DAI0 to 16 bit */
params_set_format(params, SNDRV_PCM_FORMAT_S16_LE);
return 0;
}
The other PCM operation are the same as for regular DAI links. Use as necessary.
Widget graph connections
------------------------
The BE DAI links will normally be connected to the graph at initialisation time
by the ASoC DAPM core. However, if the BE codec or BE DAI is a dummy then this
has to be set explicitly in the driver :-
::
/* BE for codec Headset - DAI0 is dummy and managed by DSP FW */
{"DAI0 CODEC IN", NULL, "AIF1 Capture"},
{"AIF1 Playback", NULL, "DAI0 CODEC OUT"},
Writing a DPCM DSP driver
=========================
The DPCM DSP driver looks much like a standard platform class ASoC driver
combined with elements from a codec class driver. A DSP platform driver must
implement :-
1. Front End PCM DAIs - i.e. struct snd_soc_dai_driver.
2. DAPM graph showing DSP audio routing from FE DAIs to BEs.
3. DAPM widgets from DSP graph.
4. Mixers for gains, routing, etc.
5. DMA configuration.
6. BE AIF widgets.
Items 6 is important for routing the audio outside of the DSP. AIF need to be
defined for each BE and each stream direction. e.g for BE DAI0 above we would
have :-
::
SND_SOC_DAPM_AIF_IN("DAI0 RX", NULL, 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("DAI0 TX", NULL, 0, SND_SOC_NOPM, 0, 0),
The BE AIF are used to connect the DSP graph to the graphs for the other
component drivers (e.g. codec graph).
Hostless PCM streams
====================
A hostless PCM stream is a stream that is not routed through the host CPU. An
example of this would be a phone call from handset to modem.
::
*************
PCM0 <------------> * * <----DAI0-----> Codec Headset
* *
PCM1 <------------> * * <====DAI1=====> Codec Speakers/Mic
* DSP *
PCM2 <------------> * * <====DAI2=====> MODEM
* *
PCM3 <------------> * * <----DAI3-----> BT
* *
* * <----DAI4-----> DMIC
* *
* * <----DAI5-----> FM
*************
In this case the PCM data is routed via the DSP. The host CPU in this use case
is only used for control and can sleep during the runtime of the stream.
The host can control the hostless link either by :-
1. Configuring the link as a CODEC <-> CODEC style link. In this case the link
is enabled or disabled by the state of the DAPM graph. This usually means
there is a mixer control that can be used to connect or disconnect the path
between both DAIs.
2. Hostless FE. This FE has a virtual connection to the BE DAI links on the DAPM
graph. Control is then carried out by the FE as regular PCM operations.
This method gives more control over the DAI links, but requires much more
userspace code to control the link. Its recommended to use CODEC<->CODEC
unless your HW needs more fine grained sequencing of the PCM ops.
CODEC <-> CODEC link
--------------------
This DAI link is enabled when DAPM detects a valid path within the DAPM graph.
The machine driver sets some additional parameters to the DAI link i.e.
::
static const struct snd_soc_pcm_stream dai_params = {
.formats = SNDRV_PCM_FMTBIT_S32_LE,
.rate_min = 8000,
.rate_max = 8000,
.channels_min = 2,
.channels_max = 2,
};
static struct snd_soc_dai_link dais[] = {
< ... more DAI links above ... >
{
.name = "MODEM",
.stream_name = "MODEM",
.cpu_dai_name = "dai2",
.codec_dai_name = "modem-aif1",
.codec_name = "modem",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBP_CFP,
.c2c_params = &dai_params,
.num_c2c_params = 1,
}
< ... more DAI links here ... >
These parameters are used to configure the DAI hw_params() when DAPM detects a
valid path and then calls the PCM operations to start the link. DAPM will also
call the appropriate PCM operations to disable the DAI when the path is no
longer valid.
Hostless FE
-----------
The DAI link(s) are enabled by a FE that does not read or write any PCM data.
This means creating a new FE that is connected with a virtual path to both
DAI links. The DAI links will be started when the FE PCM is started and stopped
when the FE PCM is stopped. Note that the FE PCM cannot read or write data in
this configuration.
linux-7.2.3/Documentation/sound/soc/index.rst
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==============
ALSA SoC Layer
==============
The documentation is split into the following sections:-
.. toctree::
:maxdepth: 2
overview
codec
dai
dapm
platform
machine
pops-clicks
clocking
jack
dpcm
codec-to-codec
usb
linux-7.2.3/Documentation/sound/soc/jack.rst
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===================
ASoC jack detection
===================
ALSA has a standard API for representing physical jacks to user space,
the kernel side of which can be seen in include/sound/jack.h. ASoC
provides a version of this API adding two additional features:
- It allows more than one jack detection method to work together on one
user visible jack. In embedded systems it is common for multiple
to be present on a single jack but handled by separate bits of
hardware.
- Integration with DAPM, allowing DAPM endpoints to be updated
automatically based on the detected jack status (eg, turning off the
headphone outputs if no headphones are present).
This is done by splitting the jacks up into three things working
together: the jack itself represented by a struct snd_soc_jack, sets of
snd_soc_jack_pins representing DAPM endpoints to update and blocks of
code providing jack reporting mechanisms.
For example, a system may have a stereo headset jack with two reporting
mechanisms, one for the headphone and one for the microphone. Some
systems won't be able to use their speaker output while a headphone is
connected and so will want to make sure to update both speaker and
headphone when the headphone jack status changes.
The jack - struct snd_soc_jack
==============================
This represents a physical jack on the system and is what is visible to
user space. The jack itself is completely passive, it is set up by the
machine driver and updated by jack detection methods.
Jacks are created by the machine driver calling snd_soc_jack_new().
snd_soc_jack_pin
================
These represent a DAPM pin to update depending on some of the status
bits supported by the jack. Each snd_soc_jack has zero or more of these
which are updated automatically. They are created by the machine driver
and associated with the jack using snd_soc_jack_add_pins(). The status
of the endpoint may configured to be the opposite of the jack status if
required (eg, enabling a built in microphone if a microphone is not
connected via a jack).
Jack detection methods
======================
Actual jack detection is done by code which is able to monitor some
input to the system and update a jack by calling snd_soc_jack_report(),
specifying a subset of bits to update. The jack detection code should
be set up by the machine driver, taking configuration for the jack to
update and the set of things to report when the jack is connected.
Often this is done based on the status of a GPIO - a handler for this is
provided by the snd_soc_jack_add_gpio() function. Other methods are
also available, for example integrated into CODECs. One example of
CODEC integrated jack detection can be see in the WM8350 driver.
Each jack may have multiple reporting mechanisms, though it will need at
least one to be useful.
Machine drivers
===============
These are all hooked together by the machine driver depending on the
system hardware. The machine driver will set up the snd_soc_jack and
the list of pins to update then set up one or more jack detection
mechanisms to update that jack based on their current status.
linux-7.2.3/Documentation/sound/soc/machine.rst
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===================
ASoC Machine Driver
===================
The ASoC machine (or board) driver is the code that glues together all the
component drivers (e.g. codecs, platforms and DAIs). It also describes the
relationships between each component which include audio paths, GPIOs,
interrupts, clocking, jacks and voltage regulators.
The machine driver can contain codec and platform specific code. It registers
the audio subsystem with the kernel as a platform device and is represented by
the following struct:-
::
/* SoC machine */
struct snd_soc_card {
char *name;
...
int (*probe)(struct platform_device *pdev);
int (*remove)(struct platform_device *pdev);
/* the pre and post PM functions are used to do any PM work before and
* after the codec and DAIs do any PM work. */
int (*suspend_pre)(struct platform_device *pdev, pm_message_t state);
int (*suspend_post)(struct platform_device *pdev, pm_message_t state);
int (*resume_pre)(struct platform_device *pdev);
int (*resume_post)(struct platform_device *pdev);
...
/* CPU <--> Codec DAI links */
struct snd_soc_dai_link *dai_link;
int num_links;
...
};
probe()/remove()
----------------
probe/remove are optional. Do any machine specific probe here.
suspend()/resume()
------------------
The machine driver has pre and post versions of suspend and resume to take care
of any machine audio tasks that have to be done before or after the codec, DAIs
and DMA is suspended and resumed. Optional.
Machine DAI Configuration
-------------------------
The machine DAI configuration glues all the codec and CPU DAIs together. It can
also be used to set up the DAI system clock and for any machine related DAI
initialisation e.g. the machine audio map can be connected to the codec audio
map, unconnected codec pins can be set as such.
struct snd_soc_dai_link is used to set up each DAI in your machine. e.g.
::
/* corgi digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link corgi_dai = {
.name = "WM8731",
.stream_name = "WM8731",
.cpu_dai_name = "pxa-is2-dai",
.codec_dai_name = "wm8731-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm8713-codec.0-001a",
.init = corgi_wm8731_init,
.ops = &corgi_ops,
};
In the above struct, dai’s are registered using names but you can pass
either dai name or device tree node but not both. Also, names used here
for cpu/codec/platform dais should be globally unique.
Additionally below example macro can be used to register cpu, codec and
platform dai::
SND_SOC_DAILINK_DEFS(wm2200_cpu_dsp,
DAILINK_COMP_ARRAY(COMP_CPU("samsung-i2s.0")),
DAILINK_COMP_ARRAY(COMP_CODEC("spi0.0", "wm0010-sdi1")),
DAILINK_COMP_ARRAY(COMP_PLATFORM("samsung-i2s.0")));
struct snd_soc_card then sets up the machine with its DAIs. e.g.
::
/* corgi audio machine driver */
static struct snd_soc_card snd_soc_corgi = {
.name = "Corgi",
.dai_link = &corgi_dai,
.num_links = 1,
};
Following this, ``devm_snd_soc_register_card`` can be used to register
the sound card. During the registration, the individual components
such as the codec, CPU, and platform are probed. If all these components
are successfully probed, the sound card gets registered.
Machine Power Map
-----------------
The machine driver can optionally extend the codec power map and to become an
audio power map of the audio subsystem. This allows for automatic power up/down
of speaker/HP amplifiers, etc. Codec pins can be connected to the machines jack
sockets in the machine init function.
Machine Controls
----------------
Machine specific audio mixer controls can be added in the DAI init function.
Clocking Controls
-----------------
As previously noted, clock configuration is handled within the machine driver.
For details on the clock APIs that the machine driver can utilize for
setup, please refer to Documentation/sound/soc/clocking.rst. However, the
callback needs to be registered by the CPU/Codec/Platform drivers to configure
the clocks that is needed for the corresponding device operation.
linux-7.2.3/Documentation/sound/soc/overview.rst
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=======================
ALSA SoC Layer Overview
=======================
The overall project goal of the ALSA System on Chip (ASoC) layer is to
provide better ALSA support for embedded system-on-chip processors (e.g.
pxa2xx, au1x00, iMX, etc) and portable audio codecs. Prior to the ASoC
subsystem there was some support in the kernel for SoC audio, however it
had some limitations:-
* Codec drivers were often tightly coupled to the underlying SoC
CPU. This is not ideal and leads to code duplication - for example,
Linux had different wm8731 drivers for 4 different SoC platforms.
* There was no standard method to signal user initiated audio events (e.g.
Headphone/Mic insertion, Headphone/Mic detection after an insertion
event). These are quite common events on portable devices and often require
machine specific code to re-route audio, enable amps, etc., after such an
event.
* Drivers tended to power up the entire codec when playing (or
recording) audio. This is fine for a PC, but tends to waste a lot of
power on portable devices. There was also no support for saving
power via changing codec oversampling rates, bias currents, etc.
ASoC Design
===========
The ASoC layer is designed to address these issues and provide the following
features :-
* Codec independence. Allows reuse of codec drivers on other platforms
and machines.
* Easy I2S/PCM audio interface setup between codec and SoC. Each SoC
interface and codec registers its audio interface capabilities with the
core and are subsequently matched and configured when the application
hardware parameters are known.
* Dynamic Audio Power Management (DAPM). DAPM automatically sets the codec to
its minimum power state at all times. This includes powering up/down
internal power blocks depending on the internal codec audio routing and any
active streams.
* Pop and click reduction. Pops and clicks can be reduced by powering the
codec up/down in the correct sequence (including using digital mute). ASoC
signals the codec when to change power states.
* Machine specific controls: Allow machines to add controls to the sound card
(e.g. volume control for speaker amplifier).
To achieve all this, ASoC basically splits an embedded audio system into
multiple re-usable component drivers :-
* Codec class drivers: The codec class driver is platform independent and
contains audio controls, audio interface capabilities, codec DAPM
definition and codec IO functions. This class extends to BT, FM and MODEM
ICs if required. Codec class drivers should be generic code that can run
on any architecture and machine.
* Platform class drivers: The platform class driver includes the audio DMA
engine driver, digital audio interface (DAI) drivers (e.g. I2S, AC97, PCM)
and any audio DSP drivers for that platform.
* Machine class driver: The machine driver class acts as the glue that
describes and binds the other component drivers together to form an ALSA
"sound card device". It handles any machine specific controls and
machine level audio events (e.g. turning on an amp at start of playback).
linux-7.2.3/Documentation/sound/soc/platform.rst
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====================
ASoC Platform Driver
====================
An ASoC platform driver class can be divided into audio DMA drivers, SoC DAI
drivers and DSP drivers. The platform drivers only target the SoC CPU and must
have no board specific code.
Audio DMA
=========
The platform DMA driver optionally supports the following ALSA operations:-
::
/* SoC audio ops */
struct snd_soc_ops {
int (*startup)(struct snd_pcm_substream *);
void (*shutdown)(struct snd_pcm_substream *);
int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *);
int (*hw_free)(struct snd_pcm_substream *);
int (*prepare)(struct snd_pcm_substream *);
int (*trigger)(struct snd_pcm_substream *, int);
};
The platform driver exports its DMA functionality via struct
snd_soc_component_driver:-
::
struct snd_soc_component_driver {
const char *name;
...
int (*probe)(struct snd_soc_component *);
void (*remove)(struct snd_soc_component *);
int (*suspend)(struct snd_soc_component *);
int (*resume)(struct snd_soc_component *);
/* pcm creation and destruction */
int (*pcm_new)(struct snd_soc_pcm_runtime *);
void (*pcm_free)(struct snd_pcm *);
...
const struct snd_pcm_ops *ops;
const struct snd_compr_ops *compr_ops;
...
};
Please refer to the :doc:`ALSA driver documentation
<../kernel-api/writing-an-alsa-driver>` for details of audio DMA.
An example DMA driver is soc/pxa/pxa2xx-pcm.c
SoC DAI Drivers
===============
Each SoC DAI driver must provide the following features:-
1. Digital audio interface (DAI) description
2. Digital audio interface configuration
3. PCM's description
4. SYSCLK configuration
5. Suspend and resume (optional)
Please see codec.rst for a description of items 1 - 4.
SoC DSP Drivers
===============
Each SoC DSP driver usually supplies the following features :-
1. DAPM graph
2. Mixer controls
3. DMA IO to/from DSP buffers (if applicable)
4. Definition of DSP front end (FE) PCM devices.
Please see dpcm.rst for a description of item 4.
linux-7.2.3/Documentation/sound/soc/pops-clicks.rst
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=====================
Audio Pops and Clicks
=====================
Pops and clicks are unwanted audio artifacts caused by the powering up and down
of components within the audio subsystem. This is noticeable on PCs when an
audio module is either loaded or unloaded (at module load time the sound card is
powered up and causes a popping noise on the speakers).
Pops and clicks can be more frequent on portable systems with DAPM. This is
because the components within the subsystem are being dynamically powered
depending on the audio usage and this can subsequently cause a small pop or
click every time a component power state is changed.
Minimising Playback Pops and Clicks
===================================
Playback pops in portable audio subsystems cannot be completely eliminated
currently, however future audio codec hardware will have better pop and click
suppression. Pops can be reduced within playback by powering the audio
components in a specific order. This order is different for startup and
shutdown and follows some basic rules:-
::
Startup Order :- DAC --> Mixers --> Output PGA --> Digital Unmute
Shutdown Order :- Digital Mute --> Output PGA --> Mixers --> DAC
This assumes that the codec PCM output path from the DAC is via a mixer and then
a PGA (programmable gain amplifier) before being output to the speakers.
Minimising Capture Pops and Clicks
==================================
Capture artifacts are somewhat easier to get rid as we can delay activating the
ADC until all the pops have occurred. This follows similar power rules to
playback in that components are powered in a sequence depending upon stream
startup or shutdown.
::
Startup Order - Input PGA --> Mixers --> ADC
Shutdown Order - ADC --> Mixers --> Input PGA
Zipper Noise
============
An unwanted zipper noise can occur within the audio playback or capture stream
when a volume control is changed near its maximum gain value. The zipper noise
is heard when the gain increase or decrease changes the mean audio signal
amplitude too quickly. It can be minimised by enabling the zero cross setting
for each volume control. The ZC forces the gain change to occur when the signal
crosses the zero amplitude line.
linux-7.2.3/Documentation/sound/soc/usb.rst
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================
ASoC USB support
================
Overview
========
In order to leverage the existing USB sound device support in ALSA, the
ASoC USB APIs are introduced to allow the subsystems to exchange
configuration information.
One potential use case would be to support USB audio offloading, which is
an implementation that allows for an alternate power-optimized path in the audio
subsystem to handle the transfer of audio data over the USB bus. This would
let the main processor to stay in lower power modes for longer duration. The
following is an example design of how the ASoC and ALSA pieces can be connected
together to achieve this:
::
USB | ASoC
| _________________________
| | ASoC Platform card |
| |_________________________|
| | |
| ___V____ ____V____
| |ASoC BE | |ASoC FE |
| |DAI LNK | |DAI LNK |
| |________| |_________|
| ^ ^ ^
| | |________|
| ___V____ |
| |SoC-USB | |
________ ________ | | |
|USB SND |<--->|USBSND |<------------>|________| |
|(card.c)| |offld |<---------- |
|________| |________|___ | | |
^ ^ | | | ____________V_________
| | | | | |IPC |
__ V_______________V_____ | | | |______________________|
|USB SND (endpoint.c) | | | | ^
|_________________________| | | | |
^ | | | ___________V___________
| | | |->|audio DSP |
___________V_____________ | | |_______________________|
|XHCI HCD |<- |
|_________________________| |
SoC USB driver
==============
Structures
----------
``struct snd_soc_usb``
- ``list``: list head for SND SoC struct list
- ``component``: reference to ASoC component
- ``connection_status_cb``: callback to notify connection events
- ``update_offload_route_info``: callback to fetch selected USB sound card/PCM
device
- ``priv_data``: driver data
The snd_soc_usb structure can be referenced using the ASoC platform card
device, or a USB device (udev->dev). This is created by the ASoC BE DAI
link, and the USB sound entity will be able to pass information to the
ASoC BE DAI link using this structure.
``struct snd_soc_usb_device``
- ``card_idx``: sound card index associated with USB sound device
- ``chip_idx``: USB sound chip array index
- ``cpcm_idx``: capture pcm device indexes associated with the USB sound device
- ``ppcm_idx``: playback pcm device indexes associated with the USB sound device
- ``num_playback``: number of playback streams
- ``num_capture``: number of capture streams
- ``list``: list head for the USB sound device list
The struct snd_soc_usb_device is created by the USB sound offload driver.
This will carry basic parameters/limitations that will be used to
determine the possible offloading paths for this USB audio device.
Functions
---------
.. code-block:: rst
int snd_soc_usb_find_supported_format(int card_idx,
struct snd_pcm_hw_params *params, int direction)
..
- ``card_idx``: the index into the USB sound chip array.
- ``params``: Requested PCM parameters from the USB DPCM BE DAI link
- ``direction``: capture or playback
**snd_soc_usb_find_supported_format()** ensures that the requested audio profile
being requested by the external DSP is supported by the USB device.
Returns 0 on success, and -EOPNOTSUPP on failure.
.. code-block:: rst
int snd_soc_usb_connect(struct device *usbdev, struct snd_soc_usb_device *sdev)
..
- ``usbdev``: the usb device that was discovered
- ``sdev``: capabilities of the device
**snd_soc_usb_connect()** notifies the ASoC USB DPCM BE DAI link of a USB
audio device detection. This can be utilized in the BE DAI
driver to keep track of available USB audio devices. This is intended
to be called by the USB offload driver residing in USB SND.
Returns 0 on success, negative error code on failure.
.. code-block:: rst
int snd_soc_usb_disconnect(struct device *usbdev, struct snd_soc_usb_device *sdev)
..
- ``usbdev``: the usb device that was removed
- ``sdev``: capabilities to free
**snd_soc_usb_disconnect()** notifies the ASoC USB DPCM BE DAI link of a USB
audio device removal. This is intended to be called by the USB offload
driver that resides in USB SND.
.. code-block:: rst
void *snd_soc_usb_find_priv_data(struct device *usbdev)
..
- ``usbdev``: the usb device to reference to find private data
**snd_soc_usb_find_priv_data()** fetches the private data saved to the SoC USB
device.
Returns pointer to priv_data on success, NULL on failure.
.. code-block:: rst
int snd_soc_usb_setup_offload_jack(struct snd_soc_component *component,
struct snd_soc_jack *jack)
..
- ``component``: ASoC component to add the jack
- ``jack``: jack component to populate
**snd_soc_usb_setup_offload_jack()** is a helper to add a sound jack control to
the platform sound card. This will allow for consistent naming to be used on
designs that support USB audio offloading. Additionally, this will enable the
jack to notify of changes.
Returns 0 on success, negative otherwise.
.. code-block:: rst
int snd_soc_usb_update_offload_route(struct device *dev, int card, int pcm,
int direction, enum snd_soc_usb_kctl path,
long *route)
..
- ``dev``: USB device to look up offload path mapping
- ``card``: USB sound card index
- ``pcm``: USB sound PCM device index
- ``direction``: direction to fetch offload routing information
- ``path``: kcontrol selector - pcm device or card index
- ``route``: mapping of sound card and pcm indexes for the offload path. This is
an array of two integers that will carry the card and pcm device indexes
in that specific order. This can be used as the array for the kcontrol
output.
**snd_soc_usb_update_offload_route()** calls a registered callback to the USB BE DAI
link to fetch the information about the mapped ASoC devices for executing USB audio
offload for the device. ``route`` may be a pointer to a kcontrol value output array,
which carries values when the kcontrol is read.
Returns 0 on success, negative otherwise.
.. code-block:: rst
struct snd_soc_usb *snd_soc_usb_allocate_port(struct snd_soc_component *component,
void *data);
..
- ``component``: DPCM BE DAI link component
- ``data``: private data
**snd_soc_usb_allocate_port()** allocates a SoC USB device and populates standard
parameters that is used for further operations.
Returns a pointer to struct soc_usb on success, negative on error.
.. code-block:: rst
void snd_soc_usb_free_port(struct snd_soc_usb *usb);
..
- ``usb``: SoC USB device to free
**snd_soc_usb_free_port()** frees a SoC USB device.
.. code-block:: rst
void snd_soc_usb_add_port(struct snd_soc_usb *usb);
..
- ``usb``: SoC USB device to add
**snd_soc_usb_add_port()** add an allocated SoC USB device to the SOC USB framework.
Once added, this device can be referenced by further operations.
.. code-block:: rst
void snd_soc_usb_remove_port(struct snd_soc_usb *usb);
..
- ``usb``: SoC USB device to remove
**snd_soc_usb_remove_port()** removes a SoC USB device from the SoC USB framework.
After removing a device, any SOC USB operations would not be able to reference the
device removed.
How to Register to SoC USB
--------------------------
The ASoC DPCM USB BE DAI link is the entity responsible for allocating and
registering the SoC USB device on the component bind. Likewise, it will
also be responsible for freeing the allocated resources. An example can
be shown below:
.. code-block:: rst
static int q6usb_component_probe(struct snd_soc_component *component)
{
...
data->usb = snd_soc_usb_allocate_port(component, 1, &data->priv);
if (!data->usb)
return -ENOMEM;
usb->connection_status_cb = q6usb_alsa_connection_cb;
ret = snd_soc_usb_add_port(usb);
if (ret < 0) {
dev_err(component->dev, "failed to add usb port\n");
goto free_usb;
}
...
}
static void q6usb_component_remove(struct snd_soc_component *component)
{
...
snd_soc_usb_remove_port(data->usb);
snd_soc_usb_free_port(data->usb);
}
static const struct snd_soc_component_driver q6usb_dai_component = {
.probe = q6usb_component_probe,
.remove = q6usb_component_remove,
.name = "q6usb-dai-component",
...
};
..
BE DAI links can pass along vendor specific information as part of the
call to allocate the SoC USB device. This will allow any BE DAI link
parameters or settings to be accessed by the USB offload driver that
resides in USB SND.
USB Audio Device Connection Flow
--------------------------------
USB devices can be hotplugged into the USB ports at any point in time.
The BE DAI link should be aware of the current state of the physical USB
port, i.e. if there are any USB devices with audio interface(s) connected.
connection_status_cb() can be used to notify the BE DAI link of any change.
This is called whenever there is a USB SND interface bind or remove event,
using snd_soc_usb_connect() or snd_soc_usb_disconnect():
.. code-block:: rst
static void qc_usb_audio_offload_probe(struct snd_usb_audio *chip)
{
...
snd_soc_usb_connect(usb_get_usb_backend(udev), sdev);
...
}
static void qc_usb_audio_offload_disconnect(struct snd_usb_audio *chip)
{
...
snd_soc_usb_disconnect(usb_get_usb_backend(chip->dev), dev->sdev);
...
}
..
In order to account for conditions where driver or device existence is
not guaranteed, USB SND exposes snd_usb_rediscover_devices() to resend the
connect events for any identified USB audio interfaces. Consider the
the following situation:
**usb_audio_probe()**
| --> USB audio streams allocated and saved to usb_chip[]
| --> Propagate connect event to USB offload driver in USB SND
| --> **snd_soc_usb_connect()** exits as USB BE DAI link is not ready
BE DAI link component probe
| --> DAI link is probed and SoC USB port is allocated
| --> The USB audio device connect event is missed
To ensure connection events are not missed, **snd_usb_rediscover_devices()**
is executed when the SoC USB device is registered. Now, when the BE DAI
link component probe occurs, the following highlights the sequence:
BE DAI link component probe
| --> DAI link is probed and SoC USB port is allocated
| --> SoC USB device added, and **snd_usb_rediscover_devices()** runs
**snd_usb_rediscover_devices()**
| --> Traverses through usb_chip[] and for non-NULL entries issue
| **connection_status_cb()**
In the case where the USB offload driver is unbound, while USB SND is ready,
the **snd_usb_rediscover_devices()** is called during module init. This allows
for the offloading path to also be enabled with the following flow:
**usb_audio_probe()**
| --> USB audio streams allocated and saved to usb_chip[]
| --> Propagate connect event to USB offload driver in USB SND
| --> USB offload driver **NOT** ready!
BE DAI link component probe
| --> DAI link is probed and SoC USB port is allocated
| --> No USB connect event due to missing USB offload driver
USB offload driver probe
| --> **qc_usb_audio_offload_init()**
| --> Calls **snd_usb_rediscover_devices()** to notify of devices
USB Offload Related Kcontrols
=============================
Details
-------
A set of kcontrols can be utilized by applications to help select the proper sound
devices to enable USB audio offloading. SoC USB exposes the get_offload_dev()
callback that designs can use to ensure that the proper indices are returned to the
application.
Implementation
--------------
**Example:**
**Sound Cards**:
::
0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D
SM8250-MTP-WCD9380-WSA8810-VA-DMIC
1 [Seri ]: USB-Audio - Plantronics Blackwire 3225 Seri
Plantronics Plantronics Blackwire
3225 Seri at usb-xhci-hcd.1.auto-1.1,
full sp
2 [C320M ]: USB-Audio - Plantronics C320-M
Plantronics Plantronics C320-M at usb-xhci-hcd.1.auto-1.2, full speed
**PCM Devices**:
::
card 0: SM8250MTPWCD938 [SM8250-MTP-WCD9380-WSA8810-VA-D], device 0: MultiMedia1 (*) []
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: SM8250MTPWCD938 [SM8250-MTP-WCD9380-WSA8810-VA-D], device 1: MultiMedia2 (*) []
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: Seri [Plantronics Blackwire 3225 Seri], device 0: USB Audio [USB Audio]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 2: C320M [Plantronics C320-M], device 0: USB Audio [USB Audio]
Subdevices: 1/1
Subdevice #0: subdevice #0
**USB Sound Card** - card#1:
::
USB Offload Playback Card Route PCM#0 -1 (range -1->32)
USB Offload Playback PCM Route PCM#0 -1 (range -1->255)
**USB Sound Card** - card#2:
::
USB Offload Playback Card Route PCM#0 0 (range -1->32)
USB Offload Playback PCM Route PCM#0 1 (range -1->255)
The above example shows a scenario where the system has one ASoC platform card
(card#0) and two USB sound devices connected (card#1 and card#2). When reading
the available kcontrols for each USB audio device, the following kcontrols lists
the mapped offload card and pcm device indexes for the specific USB device:
``USB Offload Playback Card Route PCM#*``
``USB Offload Playback PCM Route PCM#*``
The kcontrol is indexed, because a USB audio device could potentially have
several PCM devices. The above kcontrols are defined as:
- ``USB Offload Playback Card Route PCM#`` **(R)**: Returns the ASoC platform sound
card index for a mapped offload path. The output **"0"** (card index) signifies
that there is an available offload path for the USB SND device through card#0.
If **"-1"** is seen, then no offload path is available for the USB SND device.
This kcontrol exists for each USB audio device that exists in the system, and
its expected to derive the current status of offload based on the output value
for the kcontrol along with the PCM route kcontrol.
- ``USB Offload Playback PCM Route PCM#`` **(R)**: Returns the ASoC platform sound
PCM device index for a mapped offload path. The output **"1"** (PCM device index)
signifies that there is an available offload path for the USB SND device through
PCM device#0. If **"-1"** is seen, then no offload path is available for the USB\
SND device. This kcontrol exists for each USB audio device that exists in the
system, and its expected to derive the current status of offload based on the
output value for this kcontrol, in addition to the card route kcontrol.
USB Offload Playback Route Kcontrol
-----------------------------------
In order to allow for vendor specific implementations on audio offloading device
selection, the SoC USB layer exposes the following:
.. code-block:: rst
int (*update_offload_route_info)(struct snd_soc_component *component,
int card, int pcm, int direction,
enum snd_soc_usb_kctl path,
long *route)
..
These are specific for the **USB Offload Playback Card Route PCM#** and **USB
Offload PCM Route PCM#** kcontrols.
When users issue get calls to the kcontrol, the registered SoC USB callbacks will
execute the registered function calls to the DPCM BE DAI link.
**Callback Registration:**
.. code-block:: rst
static int q6usb_component_probe(struct snd_soc_component *component)
{
...
usb = snd_soc_usb_allocate_port(component, 1, &data->priv);
if (IS_ERR(usb))
return -ENOMEM;
usb->connection_status_cb = q6usb_alsa_connection_cb;
usb->update_offload_route_info = q6usb_get_offload_dev;
ret = snd_soc_usb_add_port(usb);
..
Existing USB Sound Kcontrol
---------------------------
With the introduction of USB offload support, the above USB offload kcontrol
will be added to the pre existing list of kcontrols identified by the USB sound
framework. These kcontrols are still the main controls that are used to
modify characteristics pertaining to the USB audio device.
::
Number of controls: 9
ctl type num name value
0 INT 2 Capture Channel Map 0, 0 (range 0->36)
1 INT 2 Playback Channel Map 0, 0 (range 0->36)
2 BOOL 1 Headset Capture Switch On
3 INT 1 Headset Capture Volume 10 (range 0->13)
4 BOOL 1 Sidetone Playback Switch On
5 INT 1 Sidetone Playback Volume 4096 (range 0->8192)
6 BOOL 1 Headset Playback Switch On
7 INT 2 Headset Playback Volume 20, 20 (range 0->24)
8 INT 1 USB Offload Playback Card Route PCM#0 0 (range -1->32)
9 INT 1 USB Offload Playback PCM Route PCM#0 1 (range -1->255)
Since USB audio device controls are handled over the USB control endpoint, use the
existing mechanisms present in the USB mixer to set parameters, such as volume.
linux-7.2.3/Documentation/sound/utimers.rst
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.. SPDX-License-Identifier: GPL-2.0
=======================
Userspace-driven timers
=======================
:Author: Ivan Orlov <ivan.orlov0322@gmail.com>
Preface
=======
This document describes the userspace-driven timers: virtual ALSA timers
which could be created and controlled by userspace applications using
IOCTL calls. Such timers could be useful when synchronizing audio
stream with timer sources which we don't have ALSA timers exported for
(e.g. PTP clocks), and when synchronizing the audio stream going through
two virtual sound devices using ``snd-aloop`` (for instance, when
we have a network application sending frames to one snd-aloop device,
and another sound application listening on the other end of snd-aloop).
Enabling userspace-driven timers
================================
The userspace-driven timers could be enabled in the kernel using the
``CONFIG_SND_UTIMER`` configuration option. It depends on the
``CONFIG_SND_TIMER`` option, so it also should be enabled.
Userspace-driven timers API
===========================
Userspace application can create a userspace-driven ALSA timer by
executing the ``SNDRV_TIMER_IOCTL_CREATE`` ioctl call on the
``/dev/snd/timer`` device file descriptor. The ``snd_timer_uinfo``
structure should be passed as an ioctl argument:
::
struct snd_timer_uinfo {
__u64 resolution;
int fd;
unsigned int id;
unsigned char reserved[16];
}
The ``resolution`` field sets the desired resolution in nanoseconds for
the virtual timer. ``resolution`` field simply provides an information
about the virtual timer, but does not affect the timing itself. ``id``
field gets overwritten by the ioctl, and the identifier you get in this
field after the call can be used as a timer subdevice number when
passing the timer to ``snd-aloop`` kernel module or other userspace
applications. There could be up to 128 userspace-driven timers in the
system at one moment of time, thus the id value ranges from 0 to 127.
Besides from overwriting the ``snd_timer_uinfo`` struct, ioctl stores
a timer file descriptor, which can be used to trigger the timer, in the
``fd`` field of the ``snd_timer_uinfo`` struct. Allocation of a file
descriptor for the timer guarantees that the timer can only be triggered
by the process which created it. The timer then can be triggered with
``SNDRV_TIMER_IOCTL_TRIGGER`` ioctl call on the timer file descriptor.
So, the example code for creating and triggering the timer would be:
::
static struct snd_timer_uinfo utimer_info = {
/* Timer is going to tick (presumably) every 1000000 ns */
.resolution = 1000000ULL,
.id = -1,
};
int timer_device_fd = open("/dev/snd/timer", O_RDWR | O_CLOEXEC);
if (ioctl(timer_device_fd, SNDRV_TIMER_IOCTL_CREATE, &utimer_info)) {
perror("Failed to create the timer");
return -1;
}
...
/*
* Now we want to trigger the timer. Callbacks of all of the
* timer instances binded to this timer will be executed after
* this call.
*/
ioctl(utimer_info.fd, SNDRV_TIMER_IOCTL_TRIGGER, NULL);
...
/* Now, destroy the timer */
close(timer_info.fd);
More detailed example of creating and ticking the timer could be found
in the utimer ALSA selftest.
Userspace-driven timers and snd-aloop
-------------------------------------
Userspace-driven timers could be easily used with ``snd-aloop`` module
when synchronizing two sound applications on both ends of the virtual
sound loopback. For instance, if one of the applications receives sound
frames from network and sends them to snd-aloop pcm device, and another
application listens for frames on the other snd-aloop pcm device, it
makes sense that the ALSA middle layer should initiate a data
transaction when the new period of data is received through network, but
not when the certain amount of jiffies elapses. Userspace-driven ALSA
timers could be used to achieve this.
To use userspace-driven ALSA timer as a timer source of snd-aloop, pass
the following string as the snd-aloop ``timer_source`` parameter:
::
# modprobe snd-aloop timer_source="-1.4.<utimer_id>"
Where ``utimer_id`` is the id of the timer you created with
``SNDRV_TIMER_IOCTL_CREATE``, and ``4`` is the number of
userspace-driven timers device (``SNDRV_TIMER_GLOBAL_UDRIVEN``).
``resolution`` for the userspace-driven ALSA timer used with snd-aloop
should be calculated as ``1000000000ULL / frame_rate * period_size`` as
the timer is going to tick every time a new period of frames is ready.
After that, each time you trigger the timer with
``SNDRV_TIMER_IOCTL_TRIGGER`` the new period of data will be transferred
from one snd-aloop device to another.