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Fixed MTP to work with TWRP

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Sound Blaster Audigy mixer / default DSP code
===========================================
This is based on SB-Live-mixer.txt.
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:
1) Digital mixer controls
-------------------------
These controls are built using the DSP instructions. They offer extended
functionality. Only the default build-in code in the ALSA driver is described
here. Note that the controls work as attenuators: the maximum value is the
neutral position leaving the signal unchanged. Note that if the same destination
is mentioned in multiple controls, the signal is accumulated and can be wrapped
(set to maximal or minimal value without checking of overflow).
Explanation of used abbreviations:
DAC - digital to analog converter
ADC - analog to digital converter
I2S - one-way three wire serial bus for digital sound by Philips Semiconductors
(this standard is used for connecting standalone DAC and ADC converters)
LFE - low frequency effects (subwoofer signal)
AC97 - a chip containing an analog mixer, DAC and ADC converters
IEC958 - S/PDIF
FX-bus - the EMU10K2 chip has an effect bus containing 64 accumulators.
Each of the synthesizer voices can feed its output to these accumulators
and the DSP microcontroller can operate with the resulting sum.
name='PCM Front Playback Volume',index=0
This control is used to attenuate samples for left and right front PCM FX-bus
accumulators. ALSA uses accumulators 8 and 9 for left and right front PCM
samples for 5.1 playback. The result samples are forwarded to the front DAC PCM
slots of the Philips DAC.
name='PCM Surround Playback Volume',index=0
This control is used to attenuate samples for left and right surround PCM FX-bus
accumulators. ALSA uses accumulators 2 and 3 for left and right surround PCM
samples for 5.1 playback. The result samples are forwarded to the surround DAC PCM
slots of the Philips DAC.
name='PCM Center Playback Volume',index=0
This control is used to attenuate samples for center PCM FX-bus accumulator.
ALSA uses accumulator 6 for center PCM sample for 5.1 playback. The result sample
is forwarded to the center DAC PCM slot of the Philips DAC.
name='PCM LFE Playback Volume',index=0
This control is used to attenuate sample for LFE PCM FX-bus accumulator.
ALSA uses accumulator 7 for LFE PCM sample for 5.1 playback. The result sample
is forwarded to the LFE DAC PCM slot of the Philips DAC.
name='PCM Playback Volume',index=0
This control is used to attenuate samples for left and right PCM FX-bus
accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples for
stereo playback. The result samples are forwarded to the front DAC PCM slots
of the Philips DAC.
name='PCM Capture Volume',index=0
This control is used to attenuate samples for left and right PCM FX-bus
accumulator. ALSA uses accumulators 0 and 1 for left and right PCM.
The result is forwarded to the ADC capture FIFO (thus to the standard capture
PCM device).
name='Music Playback Volume',index=0
This control is used to attenuate samples for left and right MIDI FX-bus
accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples.
The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
name='Music Capture Volume',index=0
These controls are used to attenuate samples for left and right MIDI FX-bus
accumulator. ALSA uses accumulators 4 and 5 for left and right PCM.
The result is forwarded to the ADC capture FIFO (thus to the standard capture
PCM device).
name='Mic Playback Volume',index=0
This control is used to attenuate samples for left and right Mic input.
For Mic input is used AC97 codec. The result samples are forwarded to
the front DAC PCM slots of the Philips DAC. Samples are forwarded to Mic
capture FIFO (device 1 - 16bit/8KHz mono) too without volume control.
name='Mic Capture Volume',index=0
This control is used to attenuate samples for left and right Mic input.
The result is forwarded to the ADC capture FIFO (thus to the standard capture
PCM device).
name='Audigy CD Playback Volume',index=0
This control is used to attenuate samples from left and right IEC958 TTL
digital inputs (usually used by a CDROM drive). The result samples are
forwarded to the front DAC PCM slots of the Philips DAC.
name='Audigy CD Capture Volume',index=0
This control is used to attenuate samples from left and right IEC958 TTL
digital inputs (usually used by a CDROM drive). The result samples are
forwarded to the ADC capture FIFO (thus to the standard capture PCM device).
name='IEC958 Optical Playback Volume',index=0
This control is used to attenuate samples from left and right IEC958 optical
digital input. The result samples are forwarded to the front DAC PCM slots
of the Philips DAC.
name='IEC958 Optical Capture Volume',index=0
This control is used to attenuate samples from left and right IEC958 optical
digital inputs. The result samples are forwarded to the ADC capture FIFO
(thus to the standard capture PCM device).
name='Line2 Playback Volume',index=0
This control is used to attenuate samples from left and right I2S ADC
inputs (on the AudigyDrive). The result samples are forwarded to the front
DAC PCM slots of the Philips DAC.
name='Line2 Capture Volume',index=1
This control is used to attenuate samples from left and right I2S ADC
inputs (on the AudigyDrive). The result samples are forwarded to the ADC
capture FIFO (thus to the standard capture PCM device).
name='Analog Mix Playback Volume',index=0
This control is used to attenuate samples from left and right I2S ADC
inputs from Philips ADC. The result samples are forwarded to the front
DAC PCM slots of the Philips DAC. This contains mix from analog sources
like CD, Line In, Aux, ....
name='Analog Mix Capture Volume',index=1
This control is used to attenuate samples from left and right I2S ADC
inputs Philips ADC. The result samples are forwarded to the ADC
capture FIFO (thus to the standard capture PCM device).
name='Aux2 Playback Volume',index=0
This control is used to attenuate samples from left and right I2S ADC
inputs (on the AudigyDrive). The result samples are forwarded to the front
DAC PCM slots of the Philips DAC.
name='Aux2 Capture Volume',index=1
This control is used to attenuate samples from left and right I2S ADC
inputs (on the AudigyDrive). The result samples are forwarded to the ADC
capture FIFO (thus to the standard capture PCM device).
name='Front Playback Volume',index=0
All stereo signals are mixed together and mirrored to surround, center and LFE.
This control is used to attenuate samples for left and right front speakers of
this mix.
name='Surround Playback Volume',index=0
All stereo signals are mixed together and mirrored to surround, center and LFE.
This control is used to attenuate samples for left and right surround speakers of
this mix.
name='Center Playback Volume',index=0
All stereo signals are mixed together and mirrored to surround, center and LFE.
This control is used to attenuate sample for center speaker of this mix.
name='LFE Playback Volume',index=0
All stereo signals are mixed together and mirrored to surround, center and LFE.
This control is used to attenuate sample for LFE speaker of this mix.
name='Tone Control - Switch',index=0
This control turns the tone control on or off. The samples for front, rear
and center / LFE outputs are affected.
name='Tone Control - Bass',index=0
This control sets the bass intensity. There is no neutral value!!
When the tone control code is activated, the samples are always modified.
The closest value to pure signal is 20.
name='Tone Control - Treble',index=0
This control sets the treble intensity. There is no neutral value!!
When the tone control code is activated, the samples are always modified.
The closest value to pure signal is 20.
name='Master Playback Volume',index=0
This control is used to attenuate samples for front, surround, center and
LFE outputs.
name='IEC958 Optical Raw Playback Switch',index=0
If this switch is on, then the samples for the IEC958 (S/PDIF) digital
output are taken only from the raw FX8010 PCM, otherwise standard front
PCM samples are taken.
2) PCM stream related controls
------------------------------
name='EMU10K1 PCM Volume',index 0-31
Channel volume attenuation in range 0-0xffff. The maximum value (no
attenuation) is default. The channel mapping for three values is
as follows:
0 - mono, default 0xffff (no attenuation)
1 - left, default 0xffff (no attenuation)
2 - right, default 0xffff (no attenuation)
name='EMU10K1 PCM Send Routing',index 0-31
This control specifies the destination - FX-bus accumulators. There 24
values with this mapping:
0 - mono, A destination (FX-bus 0-63), default 0
1 - mono, B destination (FX-bus 0-63), default 1
2 - mono, C destination (FX-bus 0-63), default 2
3 - mono, D destination (FX-bus 0-63), default 3
4 - mono, E destination (FX-bus 0-63), default 0
5 - mono, F destination (FX-bus 0-63), default 0
6 - mono, G destination (FX-bus 0-63), default 0
7 - mono, H destination (FX-bus 0-63), default 0
8 - left, A destination (FX-bus 0-63), default 0
9 - left, B destination (FX-bus 0-63), default 1
10 - left, C destination (FX-bus 0-63), default 2
11 - left, D destination (FX-bus 0-63), default 3
12 - left, E destination (FX-bus 0-63), default 0
13 - left, F destination (FX-bus 0-63), default 0
14 - left, G destination (FX-bus 0-63), default 0
15 - left, H destination (FX-bus 0-63), default 0
16 - right, A destination (FX-bus 0-63), default 0
17 - right, B destination (FX-bus 0-63), default 1
18 - right, C destination (FX-bus 0-63), default 2
19 - right, D destination (FX-bus 0-63), default 3
20 - right, E destination (FX-bus 0-63), default 0
21 - right, F destination (FX-bus 0-63), default 0
22 - right, G destination (FX-bus 0-63), default 0
23 - right, H destination (FX-bus 0-63), default 0
Don't forget that it's illegal to assign a channel to the same FX-bus accumulator
more than once (it means 0=0 && 1=0 is an invalid combination).
name='EMU10K1 PCM Send Volume',index 0-31
It specifies the attenuation (amount) for given destination in range 0-255.
The channel mapping is following:
0 - mono, A destination attn, default 255 (no attenuation)
1 - mono, B destination attn, default 255 (no attenuation)
2 - mono, C destination attn, default 0 (mute)
3 - mono, D destination attn, default 0 (mute)
4 - mono, E destination attn, default 0 (mute)
5 - mono, F destination attn, default 0 (mute)
6 - mono, G destination attn, default 0 (mute)
7 - mono, H destination attn, default 0 (mute)
8 - left, A destination attn, default 255 (no attenuation)
9 - left, B destination attn, default 0 (mute)
10 - left, C destination attn, default 0 (mute)
11 - left, D destination attn, default 0 (mute)
12 - left, E destination attn, default 0 (mute)
13 - left, F destination attn, default 0 (mute)
14 - left, G destination attn, default 0 (mute)
15 - left, H destination attn, default 0 (mute)
16 - right, A destination attn, default 0 (mute)
17 - right, B destination attn, default 255 (no attenuation)
18 - right, C destination attn, default 0 (mute)
19 - right, D destination attn, default 0 (mute)
20 - right, E destination attn, default 0 (mute)
21 - right, F destination attn, default 0 (mute)
22 - right, G destination attn, default 0 (mute)
23 - right, H destination attn, default 0 (mute)
4) MANUALS/PATENTS:
-------------------
ftp://opensource.creative.com/pub/doc
-------------------------------------
Files:
LM4545.pdf AC97 Codec
m2049.pdf The EMU10K1 Digital Audio Processor
hog63.ps FX8010 - A DSP Chip Architecture for Audio Effects
WIPO Patents
------------
Patent numbers:
WO 9901813 (A1) Audio Effects Processor with multiple asynchronous (Jan. 14, 1999)
streams
WO 9901814 (A1) Processor with Instruction Set for Audio Effects (Jan. 14, 1999)
WO 9901953 (A1) Audio Effects Processor having Decoupled Instruction
Execution and Audio Data Sequencing (Jan. 14, 1999)
US Patents (http://www.uspto.gov/)
----------------------------------
US 5925841 Digital Sampling Instrument employing cache memory (Jul. 20, 1999)
US 5928342 Audio Effects Processor integrated on a single chip (Jul. 27, 1999)
with a multiport memory onto which multiple asynchronous
digital sound samples can be concurrently loaded
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 (Oct. 24, 2000)
system bus with prioritization and modification of bus transfers
in accordance with loop ends and minimum block sizes
US 6151670 Method for conserving memory storage using a (Nov. 21, 2000)
pool of short term memory registers
US 6195715 Interrupt control for multiple programs communicating with (Feb. 27, 2001)
a common interrupt by associating programs to GP registers,
defining interrupt register, polling GP registers, and invoking
callback routine associated with defined interrupt register

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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
1 - Audiophile USB Specs and correct usage
==========================================
This part is a reminder of important facts about the functions and limitations
of the device.
The device has 4 audio interfaces, and 2 MIDI ports:
* Analog Stereo Input (Ai)
- This port supports 2 pairs of line-level audio inputs (1/4" TS and RCA)
- When the 1/4" TS (jack) connectors are connected, the RCA connectors
are disabled
* Analog Stereo Output (Ao)
* Digital Stereo Input (Di)
* Digital Stereo Output (Do)
* Midi In (Mi)
* Midi Out (Mo)
The internal DAC/ADC has the following 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)
2 - 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.
3 - 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".
3.1 - 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.
3.2 - Advanced module setup
---------------------------
Due to the hardware constraints described above, the device initialization made
by the Alsa driver in default mode may result in a corrupted state of the
device. For instance, a particularly annoying issue is that the sound captured
from the Ai 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)
3.2.1 - 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.
3.2.1.1 - 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
3.2.1.2 - 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)
3.2.1.3 - 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
3.2.2 - 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.
3.2.3 - Technical details for hackers
-------------------------------------
This section is for hackers, wanting to understand details about the device
internals and how Alsa supports it.
3.2.3.1 - 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
3.2.3.2 - 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.
4 - 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
4.1 - 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
4.2 - 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."
4.3 - 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.

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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>

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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.
*** WARNING ***
[ 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.

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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.

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This document describes standard names of mixer controls.
Syntax: SOURCE [DIRECTION] FUNCTION
DIRECTION:
<nothing> (both directions)
Playback
Capture
Bypass Playback
Bypass Capture
FUNCTION:
Switch (on/off switch)
Volume
Route (route control, hardware specific)
SOURCE:
Master
Master Mono
Hardware Master
Speaker (internal speaker)
Headphone
Beep (beep generator)
Phone
Phone Input
Phone Output
Synth
FM
Mic
Line
CD
Video
Zoom Video
Aux
PCM
PCM Front
PCM Rear
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
Exceptions:
[Digital] Capture Source
[Digital] Capture Switch (aka input gain switch)
[Digital] Capture Volume (aka input gain volume)
[Digital] Playback Switch (aka output gain switch)
[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) */

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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 Reatek 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.

318
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Model name Description
---------- -----------
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
ALC260
======
N/A
ALC262
======
inv-dmic Inverted internal mic workaround
ALC267/268
==========
inv-dmic Inverted internal mic workaround
ALC269/270/275/276/28x/29x
======
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
lenovo-dock Enables docking station I/O for some Lenovos
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
ALC66x/67x/892
==============
mario Chromebook mario model fixup
asus-mode1 ASUS
asus-mode2 ASUS
asus-mode3 ASUS
asus-mode4 ASUS
asus-mode5 ASUS
asus-mode6 ASUS
asus-mode7 ASUS
asus-mode8 ASUS
inv-dmic Inverted internal mic workaround
dell-headset-multi Headset jack, which can also be used as mic-in
ALC680
======
N/A
ALC88x/898/1150
======================
acer-aspire-4930g Acer Aspire 4930G/5930G/6530G/6930G/7730G
acer-aspire-8930g Acer Aspire 8330G/6935G
acer-aspire Acer Aspire others
inv-dmic Inverted internal mic workaround
no-primary-hp VAIO Z/VGC-LN51JGB workaround (for fixed speaker DAC)
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
=======
6stack 6-jack, separate surrounds (default)
3stack 3-stack, shared surrounds
laptop 2-channel only (FSC V2060, Samsung M50)
laptop-eapd 2-channel with EAPD (ASUS A6J)
laptop-automute 2-channel with EAPD and HP-automute (Lenovo N100)
ultra 2-channel with EAPD (Samsung Ultra tablet PC)
samsung 2-channel with EAPD (Samsung R65)
samsung-p50 2-channel with HP-automute (Samsung P50)
AD1988/AD1988B/AD1989A/AD1989B
==============================
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
=============
laptop-hpsense Laptop with HP sense (old model laptop)
laptop-micsense Laptop with Mic sense (old model fujitsu)
laptop-hpmicsense Laptop with HP and Mic senses
benq Benq R55E
laptop-hp530 HP 530 laptop
test for testing/debugging purpose, almost all controls
can be adjusted. Appearing only when compiled with
$CONFIG_SND_DEBUG=y
Conexant 5047
=============
laptop Basic Laptop config
laptop-hp Laptop config for some HP models (subdevice 30A5)
laptop-eapd Laptop config with EAPD support
test for testing/debugging purpose, almost all controls
can be adjusted. Appearing only when compiled with
$CONFIG_SND_DEBUG=y
Conexant 5051
=============
laptop Basic Laptop config (default)
hp HP Spartan laptop
hp-dv6736 HP dv6736
hp-f700 HP Compaq Presario F700
ideapad Lenovo IdeaPad laptop
toshiba Toshiba Satellite M300
Conexant 5066
=============
laptop Basic Laptop config (default)
hp-laptop HP laptops, e g G60
asus Asus K52JU, Lenovo G560
dell-laptop Dell laptops
dell-vostro Dell Vostro
olpc-xo-1_5 OLPC XO 1.5
ideapad Lenovo IdeaPad U150
thinkpad Lenovo Thinkpad
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 DG45* 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
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
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
========================
mbp55 MacBook Pro 5,5
imac27 IMac 27 Inch
auto BIOS setup (default)
Cirrus Logic CS4208
===================
mba6 MacBook Air 6,1 and 6,2
gpio0 Enable GPIO 0 amp
auto BIOS setup (default)
VIA VT17xx/VT18xx/VT20xx
========================
auto BIOS setup (default)

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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:
- http://www.intel.com/standards/hdaudio/
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.
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
re-use 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/alsa/HD-Audio-Models.txt 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.
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.
This enables snd_printd() macro and others, and you'll get additional
kernel messages at probing.
In addition, you can enable `CONFIG_SND_DEBUG_VERBOSE=y`. But this
will give you far more messages. Thus turn this on only when you are
sure to want it.
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_down_unused (bool): power down the unused widgets
- 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
- 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/debug/tracing/events/hda/enable
------------------------------------------------------------------------
then after some commands, you can traces from
/sys/kernel/debug/tracing/trace file. For example, when you want to
trace what codec command is sent, enable the tracepoint like:
------------------------------------------------------------------------
# cat /sys/kernel/debug/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.
If you are using the latest Linus tree, it'd be better to pull the
above GIT tree onto it. If you are using the older kernels, an easy
way to try the latest ALSA code is to build from the snapshot
tarball. There are daily tarballs and the latest snapshot tarball.
All can be built just like normal alsa-driver release packages, that
is, installed via the usual spells: configure, make and make
install(-modules). See INSTALL in the package. The snapshot tarballs
are found at:
- ftp://ftp.suse.com/pub/people/tiwai/snapshot/
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 followings:
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. You can fetch the latest version from:
- http://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:
- http://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 package is found in:
- ftp://ftp.suse.com/pub/people/tiwai/misc/
A git repository is available:
- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-emu.git
See README file in the tarball 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

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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/joystick.txt.
The joystick support of ALSA drivers is different between ISA and PCI
cards. In the case of ISA (PnP) cards, it's usually handled by the
independent module (ns558). Meanwhile, the ALSA PCI drivers have the
built-in gameport support. Hence, when the ALSA PCI driver is built
in the kernel, CONFIG_GAMEPORT must be 'y', too. Otherwise, the
gameport support on that card will be (silently) disabled.
Some adapter modules probe the physical connection of the device at
the load time. It'd be safer to plug in the joystick device before
loading the module.
PCI Cards
---------
For PCI cards, the joystick is enabled when the appropriate module
option is specified. Some drivers don't need options, and the
joystick support is always enabled. In the former ALSA version, there
was a dynamic control API for the joystick activation. It was
changed, however, to the static module options because of the system
stability and the resource management.
The following PCI drivers support the joystick natively.
Driver Module Option Available Values
---------------------------------------------------------------------------
als4000 joystick_port 0 = disable (default), 1 = auto-detect,
manual: any address (e.g. 0x200)
au88x0 N/A N/A
azf3328 joystick 0 = disable, 1 = enable, -1 = auto (default)
ens1370 joystick 0 = disable (default), 1 = enable
ens1371 joystick_port 0 = disable (default), 1 = auto-detect,
manual: 0x200, 0x208, 0x210, 0x218
cmipci joystick_port 0 = disable (default), 1 = auto-detect,
manual: any address (e.g. 0x200)
cs4281 N/A N/A
cs46xx N/A N/A
es1938 N/A N/A
es1968 joystick 0 = disable (default), 1 = enable
sonicvibes N/A N/A
trident N/A N/A
via82xx(*1) joystick 0 = disable (default), 1 = enable
ymfpci joystick_port 0 = disable (default), 1 = auto-detect,
manual: 0x201, 0x202, 0x204, 0x205(*2)
---------------------------------------------------------------------------
*1) VIA686A/B only
*2) 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").

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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.

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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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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_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
bit 3 = Log hwptr update at each period interrupt
bit 4 = Log hwptr update at each snd_pcm_update_hw_ptr()
bit 5 = Show last 10 positions on error
bit 6 = Do above only once
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.
Bits 3 and 4 are for logging the hwptr records. Note that
these will give flood of kernel messages.
When bit 5 is set, the driver logs the last 10 xrun errors and
the proc file shows each jiffies, position, period_size,
buffer_size, old_hw_ptr, and hw_ptr_base values.
When bit 6 is set, the full xrun log is shown only once.
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.
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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NOTE: The following is the original document of Rainer's patch that the
current maya44 code based on. Some contents might be obsoleted, but I
keep here as reference -- tiwai
----------------------------------------------------------------
STATE OF DEVELOPMENT:
This driver is being developed on the initiative of Piotr Makowski (oponek@gmail.com) and financed by Lars Bergmann.
Development is carried out by Rainer Zimmermann (mail@lightshed.de).
ESI provided a sample Maya44 card for the development work.
However, unfortunately it has turned out difficult to get detailed programming information, so I (Rainer Zimmermann) had to find out some card-specific information by experiment and conjecture. Some information (in particular, several GPIO bits) is still missing.
This is the first testing version of the Maya44 driver released to the alsa-devel mailing list (Feb 5, 2008).
The following functions work, as tested by Rainer Zimmermann and Piotr Makowski:
- playback and capture at all sampling rates
- input/output level
- crossmixing
- line/mic switch
- phantom power switch
- analogue monitor a.k.a bypass
The following functions *should* work, but are not fully tested:
- Channel 3+4 analogue - S/PDIF input switching
- S/PDIF output
- all inputs/outputs on the M/IO/DIO extension card
- internal/external clock selection
*In particular, we would appreciate testing of these functions by anyone who has access to an M/IO/DIO extension card.*
Things that do not seem to work:
- The level meters ("multi track") in 'alsamixer' do not seem to react to signals in (if this is a bug, it would probably be in the existing ICE1724 code).
- Ardour 2.1 seems to work only via JACK, not using ALSA directly or via OSS. This still needs to be tracked down.
DRIVER DETAILS:
the following files were added:
pci/ice1724/maya44.c - Maya44 specific code
pci/ice1724/maya44.h
pci/ice1724/ice1724.patch
pci/ice1724/ice1724.h.patch - PROPOSED patch to ice1724.h (see SAMPLING RATES)
i2c/other/wm8776.c - low-level access routines for Wolfson WM8776 codecs
include/wm8776.h
Note that the wm8776.c code is meant to be card-independent and does not actually register the codec with the ALSA infrastructure.
This is done in maya44.c, mainly because some of the WM8776 controls are used in Maya44-specific ways, and should be named appropriately.
the following files were created in pci/ice1724, simply #including the corresponding file from the alsa-kernel tree:
wtm.h
vt1720_mobo.h
revo.h
prodigy192.h
pontis.h
phase.h
maya44.h
juli.h
aureon.h
amp.h
envy24ht.h
se.h
prodigy_hifi.h
*I hope this is the correct way to do things.*
SAMPLING RATES:
The Maya44 card (or more exactly, the Wolfson WM8776 codecs) allow a maximum sampling rate of 192 kHz for playback and 92 kHz for capture.
As the ICE1724 chip only allows one global sampling rate, this is handled as follows:
* setting the sampling rate on any open PCM device on the maya44 card will always set the *global* sampling rate for all playback and capture channels.
* In the current state of the driver, setting rates of up to 192 kHz is permitted even for capture devices.
*AVOID CAPTURING AT RATES ABOVE 96kHz*, even though it may appear to work. The codec cannot actually capture at such rates, meaning poor quality.
I propose some additional code for limiting the sampling rate when setting on a capture pcm device. However because of the global sampling rate, this logic would be somewhat problematic.
The proposed code (currently deactivated) is in ice1712.h.patch, ice1724.c and maya44.c (in pci/ice1712).
SOUND DEVICES:
PCM devices correspond to inputs/outputs as follows (assuming Maya44 is card #0):
hw:0,0 input - stereo, analog input 1+2
hw:0,0 output - stereo, analog output 1+2
hw:0,1 input - stereo, analog input 3+4 OR S/PDIF input
hw:0,1 output - stereo, analog output 3+4 (and SPDIF out)
NAMING OF MIXER CONTROLS:
(for more information about the signal flow, please refer to the block diagram on p.24 of the ESI Maya44 manual, or in the ESI windows software).
PCM: (digital) output level for channel 1+2
PCM 1: same for channel 3+4
Mic Phantom+48V: switch for +48V phantom power for electrostatic microphones on input 1/2.
Make sure this is not turned on while any other source is connected to input 1/2.
It might damage the source and/or the maya44 card.
Mic/Line input: if switch is on, input jack 1/2 is microphone input (mono), otherwise line input (stereo).
Bypass: analogue bypass from ADC input to output for channel 1+2. Same as "Monitor" in the windows driver.
Bypass 1: same for channel 3+4.
Crossmix: cross-mixer from channels 1+2 to channels 3+4
Crossmix 1: cross-mixer from channels 3+4 to channels 1+2
IEC958 Output: switch for S/PDIF output.
This is not supported by the ESI windows driver.
S/PDIF should output the same signal as channel 3+4. [untested!]
Digitial output selectors:
These switches allow a direct digital routing from the ADCs to the DACs.
Each switch determines where the digital input data to one of the DACs comes from.
They are not supported by the ESI windows driver.
For normal operation, they should all be set to "PCM out".
H/W: Output source channel 1
H/W 1: Output source channel 2
H/W 2: Output source channel 3
H/W 3: Output source channel 4
H/W 4 ... H/W 9: unknown function, left in to enable testing.
Possibly some of these control S/PDIF output(s).
If these turn out to be unused, they will go away in later driver versions.
Selectable values for each of the digital output selectors are:
"PCM out" -> DAC output of the corresponding channel (default setting)
"Input 1"...
"Input 4" -> direct routing from ADC output of the selected input channel
--------
Feb 14, 2008
Rainer Zimmermann
mail@lightshed.de

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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:
1) IEC958 (S/PDIF) raw PCM
--------------------------
This PCM device (it's the 4th PCM device (index 3!) and first subdevice
(index 0) for a given card) allows to forward 48kHz, stereo, 16-bit
little endian streams without any modifications to the digital output
(coaxial or optical). The universal interface allows the creation of up
to 8 raw PCM devices operating at 48kHz, 16-bit little endian. It would
be easy to add support for multichannel devices to the current code,
but the conversion routines exist only for stereo (2-channel streams)
at the time.
Look to tram_poke routines in lowlevel/emu10k1/emufx.c for more details.
2) Digital mixer controls
-------------------------
These controls are built using the DSP instructions. They offer extended
functionality. Only the default build-in code in the ALSA driver is described
here. Note that the controls work as attenuators: the maximum value is the
neutral position leaving the signal unchanged. Note that if the same destination
is mentioned in multiple controls, the signal is accumulated and can be wrapped
(set to maximal or minimal value without checking of overflow).
Explanation of used abbreviations:
DAC - digital to analog converter
ADC - analog to digital converter
I2S - one-way three wire serial bus for digital sound by Philips Semiconductors
(this standard is used for connecting standalone DAC and ADC converters)
LFE - low frequency effects (subwoofer signal)
AC97 - a chip containing an analog mixer, DAC and ADC converters
IEC958 - S/PDIF
FX-bus - the EMU10K1 chip has an effect bus containing 16 accumulators.
Each of the synthesizer voices can feed its output to these accumulators
and the DSP microcontroller can operate with the resulting sum.
name='Wave Playback Volume',index=0
This control is used to attenuate samples for left and right PCM FX-bus
accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples.
The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
name='Wave Surround Playback Volume',index=0
This control is used to attenuate samples for left and right PCM FX-bus
accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples.
The result samples are forwarded to the rear I2S DACs. These DACs operates
separately (they are not inside the AC97 codec).
name='Wave Center Playback Volume',index=0
This control is used to attenuate samples for left and right PCM FX-bus
accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples.
The result is mixed to mono signal (single channel) and forwarded to
the ??rear?? right DAC PCM slot of the AC97 codec.
name='Wave LFE Playback Volume',index=0
This control is used to attenuate samples for left and right PCM FX-bus
accumulators. ALSA uses accumulators 0 and 1 for left and right PCM.
The result is mixed to mono signal (single channel) and forwarded to
the ??rear?? left DAC PCM slot of the AC97 codec.
name='Wave Capture Volume',index=0
name='Wave Capture Switch',index=0
These controls are used to attenuate samples for left and right PCM FX-bus
accumulator. ALSA uses accumulators 0 and 1 for left and right PCM.
The result is forwarded to the ADC capture FIFO (thus to the standard capture
PCM device).
name='Synth Playback Volume',index=0
This control is used to attenuate samples for left and right MIDI FX-bus
accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples.
The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
name='Synth Capture Volume',index=0
name='Synth Capture Switch',index=0
These controls are used to attenuate samples for left and right MIDI FX-bus
accumulator. ALSA uses accumulators 4 and 5 for left and right PCM.
The result is forwarded to the ADC capture FIFO (thus to the standard capture
PCM device).
name='Surround Playback Volume',index=0
This control is used to attenuate samples for left and right rear PCM FX-bus
accumulators. ALSA uses accumulators 2 and 3 for left and right rear PCM samples.
The result samples are forwarded to the rear I2S DACs. These DACs operate
separately (they are not inside the AC97 codec).
name='Surround Capture Volume',index=0
name='Surround Capture Switch',index=0
These controls are used to attenuate samples for left and right rear PCM FX-bus
accumulators. ALSA uses accumulators 2 and 3 for left and right rear PCM samples.
The result is forwarded to the ADC capture FIFO (thus to the standard capture
PCM device).
name='Center Playback Volume',index=0
This control is used to attenuate sample for center PCM FX-bus accumulator.
ALSA uses accumulator 6 for center PCM sample. The result sample is forwarded
to the ??rear?? right DAC PCM slot of the AC97 codec.
name='LFE Playback Volume',index=0
This control is used to attenuate sample for center PCM FX-bus accumulator.
ALSA uses accumulator 6 for center PCM sample. The result sample is forwarded
to the ??rear?? left DAC PCM slot of the AC97 codec.
name='AC97 Playback Volume',index=0
This control is used to attenuate samples for left and right front ADC PCM slots
of the AC97 codec. The result samples are forwarded to the front DAC PCM
slots of the AC97 codec.
********************************************************************************
*** Note: This control should be zero for the standard operations, otherwise ***
*** a digital loopback is activated. ***
********************************************************************************
name='AC97 Capture Volume',index=0
This control is used to attenuate samples for left and right front ADC PCM slots
of the AC97 codec. The result is forwarded to the ADC capture FIFO (thus to
the standard capture PCM device).
********************************************************************************
*** Note: This control should be 100 (maximal value), otherwise no analog ***
*** inputs of the AC97 codec can be captured (recorded). ***
********************************************************************************
name='IEC958 TTL Playback Volume',index=0
This control is used to attenuate samples from left and right IEC958 TTL
digital inputs (usually used by a CDROM drive). The result samples are
forwarded to the front DAC PCM slots of the AC97 codec.
name='IEC958 TTL Capture Volume',index=0
This control is used to attenuate samples from left and right IEC958 TTL
digital inputs (usually used by a CDROM drive). The result samples are
forwarded to the ADC capture FIFO (thus to the standard capture PCM device).
name='Zoom Video Playback Volume',index=0
This control is used to attenuate samples from left and right zoom video
digital inputs (usually used by a CDROM drive). The result samples are
forwarded to the front DAC PCM slots of the AC97 codec.
name='Zoom Video Capture Volume',index=0
This control is used to attenuate samples from left and right zoom video
digital inputs (usually used by a CDROM drive). The result samples are
forwarded to the ADC capture FIFO (thus to the standard capture PCM device).
name='IEC958 LiveDrive Playback Volume',index=0
This control is used to attenuate samples from left and right IEC958 optical
digital input. The result samples are forwarded to the front DAC PCM slots
of the AC97 codec.
name='IEC958 LiveDrive Capture Volume',index=0
This control is used to attenuate samples from left and right IEC958 optical
digital inputs. The result samples are forwarded to the ADC capture FIFO
(thus to the standard capture PCM device).
name='IEC958 Coaxial Playback Volume',index=0
This control is used to attenuate samples from left and right IEC958 coaxial
digital inputs. The result samples are forwarded to the front DAC PCM slots
of the AC97 codec.
name='IEC958 Coaxial Capture Volume',index=0
This control is used to attenuate samples from left and right IEC958 coaxial
digital inputs. The result samples are forwarded to the ADC capture FIFO
(thus to the standard capture PCM device).
name='Line LiveDrive Playback Volume',index=0
name='Line LiveDrive Playback Volume',index=1
This control is used to attenuate samples from left and right I2S ADC
inputs (on the LiveDrive). The result samples are forwarded to the front
DAC PCM slots of the AC97 codec.
name='Line LiveDrive Capture Volume',index=1
name='Line LiveDrive Capture Volume',index=1
This control is used to attenuate samples from left and right I2S ADC
inputs (on the LiveDrive). The result samples are forwarded to the ADC
capture FIFO (thus to the standard capture PCM device).
name='Tone Control - Switch',index=0
This control turns the tone control on or off. The samples for front, rear
and center / LFE outputs are affected.
name='Tone Control - Bass',index=0
This control sets the bass intensity. There is no neutral value!!
When the tone control code is activated, the samples are always modified.
The closest value to pure signal is 20.
name='Tone Control - Treble',index=0
This control sets the treble intensity. There is no neutral value!!
When the tone control code is activated, the samples are always modified.
The closest value to pure signal is 20.
name='IEC958 Optical Raw Playback Switch',index=0
If this switch is on, then the samples for the IEC958 (S/PDIF) digital
output are taken only from the raw FX8010 PCM, otherwise standard front
PCM samples are taken.
name='Headphone Playback Volume',index=1
This control attenuates the samples for the headphone output.
name='Headphone Center Playback Switch',index=1
If this switch is on, then the sample for the center PCM is put to the
left headphone output (useful for SB Live cards without separate center/LFE
output).
name='Headphone LFE Playback Switch',index=1
If this switch is on, then the sample for the center PCM is put to the
right headphone output (useful for SB Live cards without separate center/LFE
output).
3) PCM stream related controls
------------------------------
name='EMU10K1 PCM Volume',index 0-31
Channel volume attenuation in range 0-0xffff. The maximum value (no
attenuation) is default. The channel mapping for three values is
as follows:
0 - mono, default 0xffff (no attenuation)
1 - left, default 0xffff (no attenuation)
2 - right, default 0xffff (no attenuation)
name='EMU10K1 PCM Send Routing',index 0-31
This control specifies the destination - FX-bus accumulators. There are
twelve values with this mapping:
0 - mono, A destination (FX-bus 0-15), default 0
1 - mono, B destination (FX-bus 0-15), default 1
2 - mono, C destination (FX-bus 0-15), default 2
3 - mono, D destination (FX-bus 0-15), default 3
4 - left, A destination (FX-bus 0-15), default 0
5 - left, B destination (FX-bus 0-15), default 1
6 - left, C destination (FX-bus 0-15), default 2
7 - left, D destination (FX-bus 0-15), default 3
8 - right, A destination (FX-bus 0-15), default 0
9 - right, B destination (FX-bus 0-15), default 1
10 - right, C destination (FX-bus 0-15), default 2
11 - right, D destination (FX-bus 0-15), default 3
Don't forget that it's illegal to assign a channel to the same FX-bus accumulator
more than once (it means 0=0 && 1=0 is an invalid combination).
name='EMU10K1 PCM Send Volume',index 0-31
It specifies the attenuation (amount) for given destination in range 0-255.
The channel mapping is following:
0 - mono, A destination attn, default 255 (no attenuation)
1 - mono, B destination attn, default 255 (no attenuation)
2 - mono, C destination attn, default 0 (mute)
3 - mono, D destination attn, default 0 (mute)
4 - left, A destination attn, default 255 (no attenuation)
5 - left, B destination attn, default 0 (mute)
6 - left, C destination attn, default 0 (mute)
7 - left, D destination attn, default 0 (mute)
8 - right, A destination attn, default 0 (mute)
9 - right, B destination attn, default 255 (no attenuation)
10 - right, C destination attn, default 0 (mute)
11 - right, D destination attn, default 0 (mute)
4) MANUALS/PATENTS:
-------------------
ftp://opensource.creative.com/pub/doc
-------------------------------------
Files:
LM4545.pdf AC97 Codec
m2049.pdf The EMU10K1 Digital Audio Processor
hog63.ps FX8010 - A DSP Chip Architecture for Audio Effects
WIPO Patents
------------
Patent numbers:
WO 9901813 (A1) Audio Effects Processor with multiple asynchronous (Jan. 14, 1999)
streams
WO 9901814 (A1) Processor with Instruction Set for Audio Effects (Jan. 14, 1999)
WO 9901953 (A1) Audio Effects Processor having Decoupled Instruction
Execution and Audio Data Sequencing (Jan. 14, 1999)
US Patents (http://www.uspto.gov/)
----------------------------------
US 5925841 Digital Sampling Instrument employing cache memory (Jul. 20, 1999)
US 5928342 Audio Effects Processor integrated on a single chip (Jul. 27, 1999)
with a multiport memory onto which multiple asynchronous
digital sound samples can be concurrently loaded
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 (Oct. 24, 2000)
system bus with prioritization and modification of bus transfers
in accordance with loop ends and minimum block sizes
US 6151670 Method for conserving memory storage using a (Nov. 21, 2000)
pool of short term memory registers
US 6195715 Interrupt control for multiple programs communicating with (Feb. 27, 2001)
a common interrupt by associating programs to GP registers,
defining interrupt register, polling GP registers, and invoking
callback routine associated with defined interrupt register

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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.

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ALSA Kernel Parameters
~~~~~~~~~~~~~~~~~~~~~~
See Documentation/kernel-parameters.txt for general information on
specifying module parameters.
This document may not be entirely up to date and comprehensive. The command
"modinfo -p ${modulename}" shows a current list of all parameters of a loadable
module. Loadable modules, after being loaded into the running kernel, also
reveal their parameters in /sys/module/${modulename}/parameters/. Some of these
parameters may be changed at runtime by the command
"echo -n ${value} > /sys/module/${modulename}/parameters/${parm}".
snd-ad1816a= [HW,ALSA]
snd-ad1848= [HW,ALSA]
snd-ali5451= [HW,ALSA]
snd-als100= [HW,ALSA]
snd-als4000= [HW,ALSA]
snd-azt2320= [HW,ALSA]
snd-cmi8330= [HW,ALSA]
snd-cmipci= [HW,ALSA]
snd-cs4231= [HW,ALSA]
snd-cs4232= [HW,ALSA]
snd-cs4236= [HW,ALSA]
snd-cs4281= [HW,ALSA]
snd-cs46xx= [HW,ALSA]
snd-dt019x= [HW,ALSA]
snd-dummy= [HW,ALSA]
snd-emu10k1= [HW,ALSA]
snd-ens1370= [HW,ALSA]
snd-ens1371= [HW,ALSA]
snd-es968= [HW,ALSA]
snd-es1688= [HW,ALSA]
snd-es18xx= [HW,ALSA]
snd-es1938= [HW,ALSA]
snd-es1968= [HW,ALSA]
snd-fm801= [HW,ALSA]
snd-gusclassic= [HW,ALSA]
snd-gusextreme= [HW,ALSA]
snd-gusmax= [HW,ALSA]
snd-hdsp= [HW,ALSA]
snd-ice1712= [HW,ALSA]
snd-intel8x0= [HW,ALSA]
snd-interwave= [HW,ALSA]
snd-interwave-stb=
[HW,ALSA]
snd-korg1212= [HW,ALSA]
snd-maestro3= [HW,ALSA]
snd-mpu401= [HW,ALSA]
snd-mtpav= [HW,ALSA]
snd-nm256= [HW,ALSA]
snd-opl3sa2= [HW,ALSA]
snd-opti92x-ad1848=
[HW,ALSA]
snd-opti92x-cs4231=
[HW,ALSA]
snd-opti93x= [HW,ALSA]
snd-pmac= [HW,ALSA]
snd-rme32= [HW,ALSA]
snd-rme96= [HW,ALSA]
snd-rme9652= [HW,ALSA]
snd-sb8= [HW,ALSA]
snd-sb16= [HW,ALSA]
snd-sbawe= [HW,ALSA]
snd-serial= [HW,ALSA]
snd-sgalaxy= [HW,ALSA]
snd-sonicvibes= [HW,ALSA]
snd-sun-amd7930=
[HW,ALSA]
snd-sun-cs4231= [HW,ALSA]
snd-trident= [HW,ALSA]
snd-usb-audio= [HW,ALSA,USB]
snd-via82xx= [HW,ALSA]
snd-virmidi= [HW,ALSA]
snd-wavefront= [HW,ALSA]
snd-ymfpci= [HW,ALSA]

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compress_offload.txt
=====================
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)
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 dont 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,
- Indicaite 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)
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.

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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
Low latency, multichannel audio with JACK and the emu10k1/emu10k2
-----------------------------------------------------------------
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.txt (or Audigy-mixer.txt).
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

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Notes on Universal Interface for Intel High Definition Audio Codec
------------------------------------------------------------------
Takashi Iwai <tiwai@suse.de>
[Still a draft version]
General
=======
The snd-hda-codec module supports the generic access function for the
High Definition (HD) audio codecs. It's designed to be independent
from the controller code like ac97 codec module. The real accessors
from/to the controller must be implemented in the lowlevel driver.
The structure of this module is similar with ac97_codec module.
Each codec chip belongs to a bus class which communicates with the
controller.
Initialization of Bus Instance
==============================
The card driver has to create struct hda_bus at first. The template
struct should be filled and passed to the constructor:
struct hda_bus_template {
void *private_data;
struct pci_dev *pci;
const char *modelname;
struct hda_bus_ops ops;
};
The card driver can set and use the private_data field to retrieve its
own data in callback functions. The pci field is used when the patch
needs to check the PCI subsystem IDs, so on. For non-PCI system, it
doesn't have to be set, of course.
The modelname field specifies the board's specific configuration. The
string is passed to the codec parser, and it depends on the parser how
the string is used.
These fields, private_data, pci and modelname are all optional.
The ops field contains the callback functions as the following:
struct hda_bus_ops {
int (*command)(struct hda_codec *codec, hda_nid_t nid, int direct,
unsigned int verb, unsigned int parm);
unsigned int (*get_response)(struct hda_codec *codec);
void (*private_free)(struct hda_bus *);
#ifdef CONFIG_SND_HDA_POWER_SAVE
void (*pm_notify)(struct hda_codec *codec);
#endif
};
The command callback is called when the codec module needs to send a
VERB to the controller. It's always a single command.
The get_response callback is called when the codec requires the answer
for the last command. These two callbacks are mandatory and have to
be given.
The third, private_free callback, is optional. It's called in the
destructor to release any necessary data in the lowlevel driver.
The pm_notify callback is available only with
CONFIG_SND_HDA_POWER_SAVE kconfig. It's called when the codec needs
to power up or may power down. The controller should check the all
belonging codecs on the bus whether they are actually powered off
(check codec->power_on), and optionally the driver may power down the
controller side, too.
The bus instance is created via snd_hda_bus_new(). You need to pass
the card instance, the template, and the pointer to store the
resultant bus instance.
int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp,
struct hda_bus **busp);
It returns zero if successful. A negative return value means any
error during creation.
Creation of Codec Instance
==========================
Each codec chip on the board is then created on the BUS instance.
To create a codec instance, call snd_hda_codec_new().
int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
struct hda_codec **codecp);
The first argument is the BUS instance, the second argument is the
address of the codec, and the last one is the pointer to store the
resultant codec instance (can be NULL if not needed).
The codec is stored in a linked list of bus instance. You can follow
the codec list like:
struct hda_codec *codec;
list_for_each_entry(codec, &bus->codec_list, list) {
...
}
The codec isn't initialized at this stage properly. The
initialization sequence is called when the controls are built later.
Codec Access
============
To access codec, use snd_hda_codec_read() and snd_hda_codec_write().
snd_hda_param_read() is for reading parameters.
For writing a sequence of verbs, use snd_hda_sequence_write().
There are variants of cached read/write, snd_hda_codec_write_cache(),
snd_hda_sequence_write_cache(). These are used for recording the
register states for the power-management resume. When no PM is needed,
these are equivalent with non-cached version.
To retrieve the number of sub nodes connected to the given node, use
snd_hda_get_sub_nodes(). The connection list can be obtained via
snd_hda_get_connections() call.
When an unsolicited event happens, pass the event via
snd_hda_queue_unsol_event() so that the codec routines will process it
later.
(Mixer) Controls
================
To create mixer controls of all codecs, call
snd_hda_build_controls(). It then builds the mixers and does
initialization stuff on each codec.
PCM Stuff
=========
snd_hda_build_pcms() gives the necessary information to create PCM
streams. When it's called, each codec belonging to the bus stores
codec->num_pcms and codec->pcm_info fields. The num_pcms indicates
the number of elements in pcm_info array. The card driver is supposed
to traverse the codec linked list, read the pcm information in
pcm_info array, and build pcm instances according to them.
The pcm_info array contains the following record:
/* PCM information for each substream */
struct hda_pcm_stream {
unsigned int substreams; /* number of substreams, 0 = not exist */
unsigned int channels_min; /* min. number of channels */
unsigned int channels_max; /* max. number of channels */
hda_nid_t nid; /* default NID to query rates/formats/bps, or set up */
u32 rates; /* supported rates */
u64 formats; /* supported formats (SNDRV_PCM_FMTBIT_) */
unsigned int maxbps; /* supported max. bit per sample */
struct hda_pcm_ops ops;
};
/* for PCM creation */
struct hda_pcm {
char *name;
struct hda_pcm_stream stream[2];
};
The name can be passed to snd_pcm_new(). The stream field contains
the information for playback (SNDRV_PCM_STREAM_PLAYBACK = 0) and
capture (SNDRV_PCM_STREAM_CAPTURE = 1) directions. The card driver
should pass substreams to snd_pcm_new() for the number of substreams
to create.
The channels_min, channels_max, rates and formats should be copied to
runtime->hw record. They and maxbps fields are used also to compute
the format value for the HDA codec and controller. Call
snd_hda_calc_stream_format() to get the format value.
The ops field contains the following callback functions:
struct hda_pcm_ops {
int (*open)(struct hda_pcm_stream *info, struct hda_codec *codec,
struct snd_pcm_substream *substream);
int (*close)(struct hda_pcm_stream *info, struct hda_codec *codec,
struct snd_pcm_substream *substream);
int (*prepare)(struct hda_pcm_stream *info, struct hda_codec *codec,
unsigned int stream_tag, unsigned int format,
struct snd_pcm_substream *substream);
int (*cleanup)(struct hda_pcm_stream *info, struct hda_codec *codec,
struct snd_pcm_substream *substream);
};
All are non-NULL, so you can call them safely without NULL check.
The open callback should be called in PCM open after runtime->hw is
set up. It may override some setting and constraints additionally.
Similarly, the close callback should be called in the PCM close.
The prepare callback should be called in PCM prepare. This will set
up the codec chip properly for the operation. The cleanup should be
called in hw_free to clean up the configuration.
The caller should check the return value, at least for open and
prepare callbacks. When a negative value is returned, some error
occurred.
Proc Files
==========
Each codec dumps the widget node information in
/proc/asound/card*/codec#* file. This information would be really
helpful for debugging. Please provide its contents together with the
bug report.
Power Management
================
It's simple:
Call snd_hda_suspend() in the PM suspend callback.
Call snd_hda_resume() in the PM resume callback.
Codec Preset (Patch)
====================
To set up and handle the codec functionality fully, each codec may
have a codec preset (patch). It's defined in struct hda_codec_preset:
struct hda_codec_preset {
unsigned int id;
unsigned int mask;
unsigned int subs;
unsigned int subs_mask;
unsigned int rev;
const char *name;
int (*patch)(struct hda_codec *codec);
};
When the codec id and codec subsystem id match with the given id and
subs fields bitwise (with bitmask mask and subs_mask), the callback
patch is called. The patch callback should initialize the codec and
set the codec->patch_ops field. This is defined as below:
struct hda_codec_ops {
int (*build_controls)(struct hda_codec *codec);
int (*build_pcms)(struct hda_codec *codec);
int (*init)(struct hda_codec *codec);
void (*free)(struct hda_codec *codec);
void (*unsol_event)(struct hda_codec *codec, unsigned int res);
#ifdef CONFIG_PM
int (*suspend)(struct hda_codec *codec, pm_message_t state);
int (*resume)(struct hda_codec *codec);
#endif
#ifdef CONFIG_SND_HDA_POWER_SAVE
int (*check_power_status)(struct hda_codec *codec,
hda_nid_t nid);
#endif
};
The build_controls callback is called from snd_hda_build_controls().
Similarly, the build_pcms callback is called from
snd_hda_build_pcms(). The init callback is called after
build_controls to initialize the hardware.
The free callback is called as a destructor.
The unsol_event callback is called when an unsolicited event is
received.
The suspend and resume callbacks are for power management.
They can be NULL if no special sequence is required. When the resume
callback is NULL, the driver calls the init callback and resumes the
registers from the cache. If other handling is needed, you'd need to
write your own resume callback. There, the amp values can be resumed
via
void snd_hda_codec_resume_amp(struct hda_codec *codec);
and the other codec registers via
void snd_hda_codec_resume_cache(struct hda_codec *codec);
The check_power_status callback is called when the amp value of the
given widget NID is changed. The codec code can turn on/off the power
appropriately from this information.
Each entry can be NULL if not necessary to be called.
Generic Parser
==============
When the device doesn't match with any given presets, the widgets are
parsed via th generic parser (hda_generic.c). Its support is
limited: no multi-channel support, for example.
Digital I/O
===========
Call snd_hda_create_spdif_out_ctls() from the patch to create controls
related with SPDIF out.
Helper Functions
================
snd_hda_get_codec_name() stores the codec name on the given string.
snd_hda_check_board_config() can be used to obtain the configuration
information matching with the device. Define the model string table
and the table with struct snd_pci_quirk entries (zero-terminated),
and pass it to the function. The function checks the modelname given
as a module parameter, and PCI subsystem IDs. If the matching entry
is found, it returns the config field value.
snd_hda_add_new_ctls() can be used to create and add control entries.
Pass the zero-terminated array of struct snd_kcontrol_new
Macros HDA_CODEC_VOLUME(), HDA_CODEC_MUTE() and their variables can be
used for the entry of struct snd_kcontrol_new.
The input MUX helper callbacks for such a control are provided, too:
snd_hda_input_mux_info() and snd_hda_input_mux_put(). See
patch_realtek.c for example.

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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: 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: 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: 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: 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"
!!!! 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"
!!!! 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: 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: 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: 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: 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: here the analog output is enabled (but not routed).

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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/modules/snd_ac97_codec/parameters/power_save (usually as root):
# echo 10 > /sys/modules/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 hardly 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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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML>
<HEAD>
<TITLE>OSS Sequencer Emulation on ALSA</TITLE>
</HEAD>
<BODY>
<CENTER>
<H1>
<HR WIDTH="100%"></H1></CENTER>
<CENTER>
<H1>
OSS Sequencer Emulation on ALSA</H1></CENTER>
<HR WIDTH="100%">
<P>Copyright (c) 1998,1999 by Takashi Iwai
<TT><A HREF="mailto:iwai@ww.uni-erlangen.de">&lt;iwai@ww.uni-erlangen.de></A></TT>
<P>ver.0.1.8; Nov. 16, 1999
<H2>
<HR WIDTH="100%"></H2>
<H2>
1. Description</H2>
This directory contains the OSS sequencer emulation driver on ALSA. Note
that this program is still in the development state.
<P>What this does - it provides the emulation of the OSS sequencer, access
via
<TT>/dev/sequencer</TT> and <TT>/dev/music</TT> devices.
The most of applications using OSS can run if the appropriate ALSA
sequencer is prepared.
<P>The following features are emulated by this driver:
<UL>
<LI>
Normal sequencer and MIDI events:</LI>
<BR>They are converted to the ALSA sequencer events, and sent to the corresponding
port.
<LI>
Timer events:</LI>
<BR>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 <TT>/dev/music</TT>.
<LI>
Patch loading:</LI>
<BR>It purely depends on the synth drivers whether it's supported since
the patch loading is realized by callback to the synth driver.
<LI>
I/O controls:</LI>
<BR>Most of controls are accepted. Some controls
are dependent on the synth driver, as well as even on original OSS.</UL>
Furthermore, you can find the following advanced features:
<UL>
<LI>
Better queue mechanism:</LI>
<BR>The events are queued before processing them.
<LI>
Multiple applications:</LI>
<BR>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.
<LI>
Real-time event processing:</LI>
<BR>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.
<LI>
<TT>/proc</TT> interface:</LI>
<BR>The status of applications and devices can be shown via <TT>/proc/asound/seq/oss</TT>
at any time. In the later version, configuration will be changed via <TT>/proc</TT>
interface, too.</UL>
<H2>
2. Installation</H2>
Run configure script with both sequencer support (<TT>--with-sequencer=yes</TT>)
and OSS emulation (<TT>--with-oss=yes</TT>) options. A module <TT>snd-seq-oss.o</TT>
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.
<P>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.
<P>The available synth and MIDI devices can be found in proc interface.
Run "<TT>cat /proc/asound/seq/oss</TT>", and check the devices. For example,
if you use an AWE64 card, you'll see like the following:
<PRE>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; OSS sequencer emulation version 0.1.8
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ALSA client number 63
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ALSA receiver port 0
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Number of applications: 0
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Number of synth devices: 1
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; synth 0: [EMU8000]
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; type 0x1 : subtype 0x20 : voices 32
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; capabilties : ioctl enabled / load_patch enabled
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Number of MIDI devices: 3
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; midi 0: [Emu8000 Port-0] ALSA port 65:0
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; capability write / opened none
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; midi 1: [Emu8000 Port-1] ALSA port 65:1
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; capability write / opened none
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; midi 2: [0: MPU-401 (UART)] ALSA port 64:0
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; capability read/write / opened none</PRE>
Note that the device number may be different from the information of
<TT>/proc/asound/oss-devices</TT>
or ones of the original OSS driver. Use the device number listed in <TT>/proc/asound/seq/oss</TT>
to play via OSS sequencer emulation.
<H2>
3. Using Synthesizer Devices</H2>
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 <TT>/dev/sequencer</TT> like sfxload,
too.
<P>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.
<H2>
4. Using MIDI Devices</H2>
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 <TT>/proc/asound/seq/oss</TT>.
Be aware that these numbers are mostly different from the list in
<TT>/proc/asound/oss-devices</TT>.
<H2>
5. Module Options</H2>
The following module options are available:
<UL>
<LI>
<TT>maxqlen</TT></LI>
<BR>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.
<LI>
<TT>seq_oss_debug</TT></LI>
<BR>specifies the debug level and accepts zero (= no debug message) or
positive integer. Default value is 0.</UL>
<H2>
6. Queue Mechanism</H2>
OSS sequencer emulation uses an ALSA priority queue. The
events from <TT>/dev/sequencer</TT> are processed and put onto the queue
specified by module option.
<P>All the events from <TT>/dev/sequencer</TT> 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.
<P>The queued events are dispatched to the corresponding ALSA sequencer
ports after scheduled time by ALSA sequencer dispatcher.
<P>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.
<P>The input from MIDI devices or echo-back events are stored on read FIFO
queue. If application reads <TT>/dev/sequencer</TT> in blocking mode, the
process will be awaked.
<H2>
7. Interface to Synthesizer Device</H2>
<H3>
7.1. Registration</H3>
To register an OSS synthesizer device, use <TT>snd_seq_oss_synth_register</TT>
function.
<PRE>int snd_seq_oss_synth_register(char *name, int type, int subtype, int nvoices,
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; snd_seq_oss_callback_t *oper, void *private_data)</PRE>
The arguments <TT>name</TT>, <TT>type</TT>, <TT>subtype</TT> and
<TT>nvoices</TT>
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.
<P>To release this device, call <TT>snd_seq_oss_synth_unregister function</TT>:
<PRE>int snd_seq_oss_synth_unregister(int index),</PRE>
where the <TT>index</TT> is the index number returned by register function.
<H3>
7.2. Callbacks</H3>
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:
<PRE>snd_seq_oss_callback_t:
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int (*open)(snd_seq_oss_arg_t *p, void *closure);
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int (*close)(snd_seq_oss_arg_t *p);
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int (*ioctl)(snd_seq_oss_arg_t *p, unsigned int cmd, unsigned long arg);
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int (*load_patch)(snd_seq_oss_arg_t *p, int format, const char *buf, int offs, int count);
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int (*reset)(snd_seq_oss_arg_t *p);
Except for <TT>open</TT> and <TT>close</TT> callbacks, they are allowed
to be NULL.
<P>Each callback function takes the argument type snd_seq_oss_arg_t as the
first argument.
<PRE>struct snd_seq_oss_arg_t {
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int app_index;
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int file_mode;
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int seq_mode;
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; snd_seq_addr_t addr;
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; void *private_data;
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int event_passing;
};</PRE>
The first three fields, <TT>app_index</TT>, <TT>file_mode</TT> and
<TT>seq_mode</TT>
are initialized by OSS sequencer. The <TT>app_index</TT> is the application
index which is unique to each application opening OSS sequencer. The
<TT>file_mode</TT>
is bit-flags indicating the file operation mode. See
<TT>seq_oss.h</TT>
for its meaning. The <TT>seq_mode</TT> is sequencer operation mode. In
the current version, only <TT>SND_OSSSEQ_MODE_SYNTH</TT> is used.
<P>The next two fields, <TT>addr</TT> and <TT>private_data</TT>, must be
filled by the synth driver at open callback. The <TT>addr</TT> contains
the address of ALSA sequencer port which is assigned to this device. If
the driver allocates memory for <TT>private_data</TT>, it must be released
in close callback by itself.
<P>The last field, <TT>event_passing</TT>, indicates how to translate note-on
/ off events. In <TT>PROCESS_EVENTS</TT> mode, the note 255 is regarded
as velocity change, and key pressure event is passed to the port. In <TT>PASS_EVENTS</TT>
mode, all note on/off events are passed to the port without modified. <TT>PROCESS_KEYPRESS</TT>
mode checks the note above 128 and regards it as key pressure event (mainly
for Emu8000 driver).
<H4>
7.2.1. Open Callback</H4>
The <TT>open</TT> 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:
<OL>
<LI>
Allocate private data record.</LI>
<LI>
Create an ALSA sequencer port.</LI>
<LI>
Set the new port address on arg->addr.</LI>
<LI>
Set the private data record pointer on arg->private_data.</LI>
</OL>
Note that the type bit-flags in port_info of this synth port must NOT contain
<TT>TYPE_MIDI_GENERIC</TT>
bit. Instead, <TT>TYPE_SPECIFIC</TT> should be used. Also, <TT>CAP_SUBSCRIPTION</TT>
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.
<H4>
7.2.2 Ioctl Callback</H4>
The <TT>ioctl</TT> callback is called when the sequencer receives device-specific
ioctls. The following two ioctls should be processed by this callback:
<UL>
<LI>
<TT>IOCTL_SEQ_RESET_SAMPLES</TT></LI>
<BR>reset all samples on memory -- return 0
<LI>
<TT>IOCTL_SYNTH_MEMAVL</TT></LI>
<BR>return the available memory size
<LI>
<TT>FM_4OP_ENABLE</TT></LI>
<BR>can be ignored usually</UL>
The other ioctls are processed inside the sequencer without passing to
the lowlevel driver.
<H4>
7.2.3 Load_Patch Callback</H4>
The <TT>load_patch</TT> 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.
<H4>
7.2.4 Close Callback</H4>
The <TT>close</TT> 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.
<H4>
7.2.5 Reset Callback</H4>
The <TT>reset</TT> 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 <TT>HEARTBEAT</TT> event to the
port.
<H3>
7.3 Events</H3>
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:
<BR>&nbsp;
<TABLE BORDER WIDTH="75%" NOSAVE >
<TR NOSAVE>
<TD NOSAVE><B>ALSA event</B></TD>
<TD><B>Original OSS events</B></TD>
</TR>
<TR>
<TD>NOTEON</TD>
<TD>SEQ_NOTEON
<BR>MIDI_NOTEON</TD>
</TR>
<TR>
<TD>NOTE</TD>
<TD>SEQ_NOTEOFF
<BR>MIDI_NOTEOFF</TD>
</TR>
<TR NOSAVE>
<TD NOSAVE>KEYPRESS</TD>
<TD>MIDI_KEY_PRESSURE</TD>
</TR>
<TR NOSAVE>
<TD>CHANPRESS</TD>
<TD NOSAVE>SEQ_AFTERTOUCH
<BR>MIDI_CHN_PRESSURE</TD>
</TR>
<TR NOSAVE>
<TD NOSAVE>PGMCHANGE</TD>
<TD NOSAVE>SEQ_PGMCHANGE
<BR>MIDI_PGM_CHANGE</TD>
</TR>
<TR>
<TD>PITCHBEND</TD>
<TD>SEQ_CONTROLLER(CTRL_PITCH_BENDER)
<BR>MIDI_PITCH_BEND</TD>
</TR>
<TR>
<TD>CONTROLLER</TD>
<TD>MIDI_CTL_CHANGE
<BR>SEQ_BALANCE (with CTL_PAN)</TD>
</TR>
<TR>
<TD>CONTROL14</TD>
<TD>SEQ_CONTROLLER</TD>
</TR>
<TR>
<TD>REGPARAM</TD>
<TD>SEQ_CONTROLLER(CTRL_PITCH_BENDER_RANGE)</TD>
</TR>
<TR>
<TD>SYSEX</TD>
<TD>SEQ_SYSEX</TD>
</TR>
</TABLE>
<P>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.
<P>Some OSS events (<TT>SEQ_PRIVATE</TT> and <TT>SEQ_VOLUME</TT> 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.
<H2>
8. Interface to MIDI Device</H2>
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 <TT>SND_SEQ_GROUP_DEVICE</TT> and a capability-bit <TT>CAP_READ</TT> or
<TT>CAP_WRITE</TT>. Also, subscription capabilities, <TT>CAP_SUBS_READ</TT> or <TT>CAP_SUBS_WRITE</TT>,
must be defined, too. If these conditions are not satisfied, the port is not
registered as OSS sequencer MIDI device.
<P>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.
<H2>
9. Known Problems / TODO's</H2>
<UL>
<LI>
Patch loading via ALSA instrument layer is not implemented yet.</LI>
</UL>
</BODY>
</HTML>

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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.

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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 :-
http://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, whilst 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:-
o I2S - MSB is transmitted on the falling edge of the first BCLK after LRC
transition.
o Left Justified - MSB is transmitted on transition of LRC.
o 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 whilst the Tx and Rx lines are used to transmit and
receive the audio data. Bit clock usually varies depending on sample rate
whilst 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:-
o Mode A - MSB is transmitted on falling edge of first BCLK after FRAME/SYNC.
o Mode B - MSB is transmitted on rising edge of FRAME/SYNC.

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Dynamic PCM
===========
1. 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 patch 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(), prepapre() 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.
1 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 :-
static struct snd_soc_dai_link machine_dais[] = {
{
.name = "PCM0 System",
.stream_name = "System Playback",
.cpu_dai_name = "System Pin",
.platform_name = "dsp-audio",
.codec_name = "snd-soc-dummy",
.codec_dai_name = "snd-soc-dummy-dai",
.dynamic = 1,
.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
.dpcm_playback = 1,
},
.....< 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". The supported FE stream
directions should also be set with the "dpcm_playback" and "dpcm_capture"
flags. 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 :-
static struct snd_soc_dai_link machine_dais[] = {
.....< FE DAI links here >
{
.name = "Codec Headset",
.cpu_dai_name = "ssp-dai.0",
.platform_name = "snd-soc-dummy",
.no_pcm = 1,
.codec_name = "rt5640.0-001c",
.codec_dai_name = "rt5640-aif1",
.ignore_suspend = 1,
.ignore_pmdown_time = 1,
.be_hw_params_fixup = hswult_ssp0_fixup,
.ops = &haswell_ops,
.dpcm_playback = 1,
.dpcm_capture = 1,
},
.....< 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 and sets flags for supported stream
directions using "dpcm_playback" and "dpcm_capture" above.
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 whilst 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 code 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.
2 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 covert the FE rate to 48k, stereo */
rate->min = rate->max = 48000;
channels->min = channels->max = 2;
/* set DAI0 to 16 bit */
snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
SNDRV_PCM_HW_PARAM_FIRST_MASK],
SNDRV_PCM_FORMAT_S16_LE);
return 0;
}
The other PCM operation are the same as for regular DAI links. Use as necessary.
3 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_CBM_CFM,
.params = &dai_params,
}
< ... 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.

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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.

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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
===========================
1 - 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,
.digital_mute = 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_rates = 1,
};
2 - 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.
3 - 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
4 - 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 ALSA driver PCM documentation for details.
http://www.alsa-project.org/~iwai/writing-an-alsa-driver/
5 - 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.txt for details of building the description.
Please also see the examples in other codec drivers.
6 - 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 */
7 - 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)
{
struct snd_soc_codec *codec = dai->codec;
u16 mute_reg = snd_soc_read(codec, WM8974_DAC) & 0xffbf;
if (mute)
snd_soc_write(codec, WM8974_DAC, mute_reg | 0x40);
else
snd_soc_write(codec, WM8974_DAC, mute_reg);
return 0;
}

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Dynamic Audio Power Management for Portable Devices
===================================================
1. 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 PM and as
such, can easily co-exist with the other PM systems.
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 spans the whole machine. It covers power control within the entire
audio subsystem, this includes internal codec power blocks and machine
level power systems.
There are 4 power domains within DAPM
1. 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.
2. 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
3. Path domain - audio subsystem signal paths
Automatically set when mixer and mux settings are changed by the user.
e.g. alsamixer, amixer.
4. Stream domain - DACs and ADCs.
Enabled and disabled when stream playback/capture is started and
stopped respectively. e.g. aplay, arecord.
All DAPM power switching decisions are made automatically by consulting an audio
routing map of the whole machine. This map is specific to each machine and
consists of the interconnections between every audio component (including
internal codec components). All audio components that effect power are called
widgets hereafter.
2. DAPM Widgets
===============
Audio DAPM widgets fall into a number of types:-
o Mixer - Mixes several analog signals into a single analog signal.
o Mux - An analog switch that outputs only one of many inputs.
o PGA - A programmable gain amplifier or attenuation widget.
o ADC - Analog to Digital Converter
o DAC - Digital to Analog Converter
o Switch - An analog switch
o Input - A codec input pin
o Output - A codec output pin
o Headphone - Headphone (and optional Jack)
o Mic - Mic (and optional Jack)
o Line - Line Input/Output (and optional Jack)
o Speaker - Speaker
o Supply - Power or clock supply widget used by other widgets.
o Regulator - External regulator that supplies power to audio components.
o Clock - External clock that supplies clock to audio components.
o AIF IN - Audio Interface Input (with TDM slot mask).
o AIF OUT - Audio Interface Output (with TDM slot mask).
o Siggen - Signal Generator.
o DAI IN - Digital Audio Interface Input.
o DAI OUT - Digital Audio Interface Output.
o DAI Link - DAI Link between two DAI structures */
o Pre - Special PRE widget (exec before all others)
o Post - Special POST widget (exec after all others)
(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.
2.1 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_codec_dai.
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),
2.2 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 dont 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.
2.3 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.
o Speaker Amp
o Microphone Bias
o 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),
2.4 Codec (BIAS) Domain
-----------------------
The codec bias power domain has no widgets and is handled by the codecs DAPM
event handler. This handler is called when the codec powerstate is changed wrt
to any stream event or by kernel PM events.
2.5 Virtual Widgets
-------------------
Sometimes widgets exist in the codec or machine audio map 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_DAPM_NOPM, 0, 0, NULL, 0),
This can be used to merge to signal paths together in software.
After all the widgets have been defined, they can then be added to the DAPM
subsystem individually with a call to snd_soc_dapm_new_control().
3. 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 map 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.
e.g., from the WM8731 output mixer (wm8731.c)
The WM8731 output mixer 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 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(codec) 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.
3.1 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.
4 Endpoint Widgets
===================
An endpoint is a start or end point (widget) of an audio signal within the
machine and includes the codec. e.g.
o Headphone Jack
o Internal Speaker
o Internal Mic
o Mic Jack
o 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.
5 DAPM Widget Events
====================
Some widgets can register their interest with the DAPM core in PM events.
e.g. A Speaker with an amplifier registers a widget so the amplifier can be
powered only when the spk is in use.
/* turn speaker amplifier on/off depending on use */
static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event)
{
gpio_set_value(CORGI_GPIO_APM_ON, SND_SOC_DAPM_EVENT_ON(event));
return 0;
}
/* corgi machine dapm widgets */
static const struct snd_soc_dapm_widget wm8731_dapm_widgets =
SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event);
Please see soc-dapm.h for all other widgets that support events.
5.1 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 */

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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.

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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 componnent 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,
};
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,
};
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.

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ALSA SoC Layer
==============
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
decribes 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).
Documentation
=============
The documentation is spilt into the following sections:-
overview.txt: This file.
codec.txt: Codec driver internals.
DAI.txt: Description of Digital Audio Interface standards and how to configure
a DAI within your codec and CPU DAI drivers.
dapm.txt: Dynamic Audio Power Management
platform.txt: Platform audio DMA and DAI.
machine.txt: Machine driver internals.
pop_clicks.txt: How to minimise audio artifacts.
clocking.txt: ASoC clocking for best power performance.
jack.txt: ASoC jack detection.
DPCM.txt: Dynamic PCM - Describes DPCM with DSP examples.

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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_platform_driver:-
struct snd_soc_platform_driver {
char *name;
int (*probe)(struct platform_device *pdev);
int (*remove)(struct platform_device *pdev);
int (*suspend)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai);
int (*resume)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai);
/* pcm creation and destruction */
int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *, struct snd_pcm *);
void (*pcm_free)(struct snd_pcm *);
/*
* For platform caused delay reporting.
* Optional.
*/
snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
struct snd_soc_dai *);
/* platform stream ops */
struct snd_pcm_ops *pcm_ops;
};
Please refer to the ALSA driver documentation for details of audio DMA.
http://www.alsa-project.org/~iwai/writing-an-alsa-driver/
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.txt 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.txt for a description of item 4.

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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.