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Merge remote-tracking branch 'sound/topic/restize-docs' into sound 

Bring in the sphinxification of the sound documentation.
This commit is contained in:
Jonathan Corbet 2016-11-18 16:19:28 -07:00
parent 917fef6f7e c6ab9e57e8
commit 726d661fea
56 changed files with 10264 additions and 11242 deletions
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3
Documentation/DocBook/Makefile
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@ -12,8 +12,7 @@ DOCBOOKS := z8530book.xml \
kernel-api.xml filesystems.xml lsm.xml kgdb.xml \
gadget.xml libata.xml mtdnand.xml librs.xml rapidio.xml \
genericirq.xml s390-drivers.xml uio-howto.xml scsi.xml \
debugobjects.xml sh.xml regulator.xml \
alsa-driver-api.xml writing-an-alsa-driver.xml \
80211.xml debugobjects.xml sh.xml regulator.xml \
tracepoint.xml w1.xml \
writing_musb_glue_layer.xml crypto-API.xml iio.xml

142
Documentation/DocBook/alsa-driver-api.tmpl
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@ -1,142 +0,0 @@
<?xml version="1.0" encoding="UTF-8"?>
<!DOCTYPE book PUBLIC "-//OASIS//DTD DocBook XML V4.1.2//EN"
"http://www.oasis-open.org/docbook/xml/4.1.2/docbookx.dtd" []>
<!-- ****************************************************** -->
<!-- Header -->
<!-- ****************************************************** -->
<book id="ALSA-Driver-API">
<bookinfo>
<title>The ALSA Driver API</title>
<legalnotice>
<para>
This document is free; you can redistribute it and/or modify it
under the terms of the GNU General Public License as published by
the Free Software Foundation; either version 2 of the License, or
(at your option) any later version.
</para>
<para>
This document is distributed in the hope that it will be useful,
but <emphasis>WITHOUT ANY WARRANTY</emphasis>; without even the
implied warranty of <emphasis>MERCHANTABILITY or FITNESS FOR A
PARTICULAR PURPOSE</emphasis>. See the GNU General Public License
for more details.
</para>
<para>
You should have received a copy of the GNU General Public
License along with this program; if not, write to the Free
Software Foundation, Inc., 59 Temple Place, Suite 330, Boston,
MA 02111-1307 USA
</para>
</legalnotice>
</bookinfo>
<toc></toc>
<chapter><title>Management of Cards and Devices</title>
<sect1><title>Card Management</title>
!Esound/core/init.c
</sect1>
<sect1><title>Device Components</title>
!Esound/core/device.c
</sect1>
<sect1><title>Module requests and Device File Entries</title>
!Esound/core/sound.c
</sect1>
<sect1><title>Memory Management Helpers</title>
!Esound/core/memory.c
!Esound/core/memalloc.c
</sect1>
</chapter>
<chapter><title>PCM API</title>
<sect1><title>PCM Core</title>
!Esound/core/pcm.c
!Esound/core/pcm_lib.c
!Esound/core/pcm_native.c
!Iinclude/sound/pcm.h
</sect1>
<sect1><title>PCM Format Helpers</title>
!Esound/core/pcm_misc.c
</sect1>
<sect1><title>PCM Memory Management</title>
!Esound/core/pcm_memory.c
</sect1>
<sect1><title>PCM DMA Engine API</title>
!Esound/core/pcm_dmaengine.c
!Iinclude/sound/dmaengine_pcm.h
</sect1>
</chapter>
<chapter><title>Control/Mixer API</title>
<sect1><title>General Control Interface</title>
!Esound/core/control.c
</sect1>
<sect1><title>AC97 Codec API</title>
!Esound/pci/ac97/ac97_codec.c
!Esound/pci/ac97/ac97_pcm.c
</sect1>
<sect1><title>Virtual Master Control API</title>
!Esound/core/vmaster.c
!Iinclude/sound/control.h
</sect1>
</chapter>
<chapter><title>MIDI API</title>
<sect1><title>Raw MIDI API</title>
!Esound/core/rawmidi.c
</sect1>
<sect1><title>MPU401-UART API</title>
!Esound/drivers/mpu401/mpu401_uart.c
</sect1>
</chapter>
<chapter><title>Proc Info API</title>
<sect1><title>Proc Info Interface</title>
!Esound/core/info.c
</sect1>
</chapter>
<chapter><title>Compress Offload</title>
<sect1><title>Compress Offload API</title>
!Esound/core/compress_offload.c
!Iinclude/uapi/sound/compress_offload.h
!Iinclude/uapi/sound/compress_params.h
!Iinclude/sound/compress_driver.h
</sect1>
</chapter>
<chapter><title>ASoC</title>
<sect1><title>ASoC Core API</title>
!Iinclude/sound/soc.h
!Esound/soc/soc-core.c
<!-- !Esound/soc/soc-cache.c no docbook comments here -->
!Esound/soc/soc-devres.c
!Esound/soc/soc-io.c
!Esound/soc/soc-pcm.c
!Esound/soc/soc-ops.c
!Esound/soc/soc-compress.c
</sect1>
<sect1><title>ASoC DAPM API</title>
!Esound/soc/soc-dapm.c
</sect1>
<sect1><title>ASoC DMA Engine API</title>
!Esound/soc/soc-generic-dmaengine-pcm.c
</sect1>
</chapter>
<chapter><title>Miscellaneous Functions</title>
<sect1><title>Hardware-Dependent Devices API</title>
!Esound/core/hwdep.c
</sect1>
<sect1><title>Jack Abstraction Layer API</title>
!Iinclude/sound/jack.h
!Esound/core/jack.c
!Esound/soc/soc-jack.c
</sect1>
<sect1><title>ISA DMA Helpers</title>
!Esound/core/isadma.c
</sect1>
<sect1><title>Other Helper Macros</title>
!Iinclude/sound/core.h
</sect1>
</chapter>
</book>

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1
Documentation/index.rst
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@ -58,6 +58,7 @@ needed).
gpu/index
80211/index
security/index
sound/index
Korean translations
-------------------

2683
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Documentation/sound/alsa/ALSA-Configuration.txt
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107
Documentation/sound/alsa/ControlNames.txt
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@ -1,107 +0,0 @@
This document describes standard names of mixer controls.
Syntax: [LOCATION] SOURCE [CHANNEL] [DIRECTION] FUNCTION
DIRECTION:
<nothing> (both directions)
Playback
Capture
Bypass Playback
Bypass Capture
FUNCTION:
Switch (on/off switch)
Volume
Route (route control, hardware specific)
CHANNEL:
<nothing> (channel independent, or applies to all channels)
Front
Surround (rear left/right in 4.0/5.1 surround)
CLFE
Center
LFE
Side (side left/right for 7.1 surround)
LOCATION: (physical location of source)
Front
Rear
Dock (docking station)
Internal
SOURCE:
Master
Master Mono
Hardware Master
Speaker (internal speaker)
Bass Speaker (internal LFE speaker)
Headphone
Line Out
Beep (beep generator)
Phone
Phone Input
Phone Output
Synth
FM
Mic
Headset Mic (mic part of combined headset jack - 4-pin headphone + mic)
Headphone Mic (mic part of either/or - 3-pin headphone or mic)
Line (input only, use "Line Out" for output)
CD
Video
Zoom Video
Aux
PCM
PCM Pan
Loopback
Analog Loopback (D/A -> A/D loopback)
Digital Loopback (playback -> capture loopback - without analog path)
Mono
Mono Output
Multi
ADC
Wave
Music
I2S
IEC958
HDMI
SPDIF (output only)
SPDIF In
Digital In
HDMI/DP (either HDMI or DisplayPort)
Exceptions (deprecated):
[Analogue|Digital] Capture Source
[Analogue|Digital] Capture Switch (aka input gain switch)
[Analogue|Digital] Capture Volume (aka input gain volume)
[Analogue|Digital] Playback Switch (aka output gain switch)
[Analogue|Digital] Playback Volume (aka output gain volume)
Tone Control - Switch
Tone Control - Bass
Tone Control - Treble
3D Control - Switch
3D Control - Center
3D Control - Depth
3D Control - Wide
3D Control - Space
3D Control - Level
Mic Boost [(?dB)]
PCM interface:
Sample Clock Source { "Word", "Internal", "AutoSync" }
Clock Sync Status { "Lock", "Sync", "No Lock" }
External Rate /* external capture rate */
Capture Rate /* capture rate taken from external source */
IEC958 (S/PDIF) interface:
IEC958 [...] [Playback|Capture] Switch /* turn on/off the IEC958 interface */
IEC958 [...] [Playback|Capture] Volume /* digital volume control */
IEC958 [...] [Playback|Capture] Default /* default or global value - read/write */
IEC958 [...] [Playback|Capture] Mask /* consumer and professional mask */
IEC958 [...] [Playback|Capture] Con Mask /* consumer mask */
IEC958 [...] [Playback|Capture] Pro Mask /* professional mask */
IEC958 [...] [Playback|Capture] PCM Stream /* the settings assigned to a PCM stream */
IEC958 Q-subcode [Playback|Capture] Default /* Q-subcode bits */
IEC958 Preamble [Playback|Capture] Default /* burst preamble words (4*16bits) */

324
Documentation/sound/alsa/HD-Audio-Models.txt
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@ -1,324 +0,0 @@
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
======
gpio1 Enable GPIO1
coef Enable EAPD via COEF table
fujitsu Quirk for FSC S7020
fujitsu-jwse Quirk for FSC S7020 with jack modes and HP mic support
ALC262
======
inv-dmic Inverted internal mic workaround
ALC267/268
==========
inv-dmic Inverted internal mic workaround
hp-eapd Disable HP EAPD on NID 0x15
ALC22x/23x/25x/269/27x/28x/29x (and vendor-specific ALC3xxx models)
======
laptop-amic Laptops with analog-mic input
laptop-dmic Laptops with digital-mic input
alc269-dmic Enable ALC269(VA) digital mic workaround
alc271-dmic Enable ALC271X digital mic workaround
inv-dmic Inverted internal mic workaround
headset-mic Indicates a combined headset (headphone+mic) jack
headset-mode More comprehensive headset support for ALC269 & co
headset-mode-no-hp-mic Headset mode support without headphone mic
lenovo-dock Enables docking station I/O for some Lenovos
hp-gpio-led GPIO LED support on HP laptops
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
alc283-dac-wcaps Fixups for Chromebook with ALC283
alc283-sense-combo Combo jack sensing on ALC283
tpt440-dock Pin configs for Lenovo Thinkpad Dock support
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
=======
3stack 3-stack, shared surrounds
laptop 2-channel only (FSC V2060, Samsung M50)
laptop-imic 2-channel with built-in mic
eapd Turn on EAPD constantly
AD1988/AD1988B/AD1989A/AD1989B
==============================
6stack 6-jack
6stack-dig ditto with SPDIF
3stack 3-jack
3stack-dig ditto with SPDIF
laptop 3-jack with hp-jack automute
laptop-dig ditto with SPDIF
auto auto-config reading BIOS (default)
Conexant 5045
=============
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)

8
Documentation/sound/alsa/VIA82xx-mixer.txt
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@ -1,8 +0,0 @@
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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Documentation/sound/alsa/alsa-parameters.txt
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ALSA Kernel Parameters
~~~~~~~~~~~~~~~~~~~~~~
See Documentation/admin-guide/kernel-parameters.rst 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]

409
Documentation/sound/alsa/seq_oss.html
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@ -1,409 +0,0 @@
<!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>

279
Documentation/sound/alsa/Audigy-mixer.txt → Documentation/sound/cards/audigy-mixer.rst
View File

@ -1,8 +1,8 @@
=============================================
Sound Blaster Audigy mixer / default DSP code
=============================================
Sound Blaster Audigy mixer / default DSP code
===========================================
This is based on SB-Live-mixer.txt.
This is based on sb-live-mixer.rst.
The EMU10K2 chips have a DSP part which can be programmed to support
various ways of sample processing, which is described here.
@ -13,8 +13,8 @@ The ALSA driver programs this portion of chip by default code
(can be altered later) which offers the following functionality:
1) Digital mixer controls
-------------------------
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
@ -26,320 +26,343 @@ is mentioned in multiple controls, the signal is accumulated and can be wrapped
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.
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
------------------------------
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)
* 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
* 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)
* 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:
-------------------
MANUALS/PATENTS
===============
ftp://opensource.creative.com/pub/doc
-------------------------------------
Files:
LM4545.pdf AC97 Codec
LM4545.pdf
AC97 Codec
m2049.pdf The EMU10K1 Digital Audio Processor
m2049.pdf
The EMU10K1 Digital Audio Processor
hog63.ps FX8010 - A DSP Chip Architecture for Audio Effects
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 9901813 (A1)
Audio Effects Processor with multiple asynchronous streams
(Jan. 14, 1999)
WO 9901953 (A1) Audio Effects Processor having Decoupled Instruction
Execution and Audio Data Sequencing (Jan. 14, 1999)
WO 9901814 (A1)
Processor with Instruction Set for Audio Effects (Jan. 14, 1999)
WO 9901953 (A1)
Audio Effects Processor having Decoupled Instruction
Execution and Audio Data Sequencing (Jan. 14, 1999)
US Patents (http://www.uspto.gov/)
----------------------------------
US 5925841 Digital Sampling Instrument employing cache memory (Jul. 20, 1999)
US 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 5928342
Audio Effects Processor integrated on a single chip
with a multiport memory onto which multiple asynchronous
digital sound samples can be concurrently loaded
(Jul. 27, 1999)
US 5930158 Processor with Instruction Set for Audio Effects (Jul. 27, 1999)
US 5930158
Processor with Instruction Set for Audio Effects (Jul. 27, 1999)
US 6032235 Memory initialization circuit (Tram) (Feb. 29, 2000)
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 6138207
Interpolation looping of audio samples in cache connected to
system bus with prioritization and modification of bus transfers
in accordance with loop ends and minimum block sizes
(Oct. 24, 2000)
US 6151670 Method for conserving memory storage using a (Nov. 21, 2000)
pool of short term memory registers
US 6151670
Method for conserving memory storage using a
pool of short term memory registers
(Nov. 21, 2000)
US 6195715 Interrupt control for multiple programs communicating with (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
US 6195715
Interrupt control for multiple programs communicating with
a common interrupt by associating programs to GP registers,
defining interrupt register, polling GP registers, and invoking
callback routine associated with defined interrupt register
(Feb. 27, 2001)

258
Documentation/sound/alsa/Audiophile-Usb.txt → Documentation/sound/cards/audiophile-usb.rst
View File

@ -1,32 +1,41 @@
Guide to using M-Audio Audiophile USB with ALSA and Jack v1.5
========================================================
========================================================
Guide to using M-Audio Audiophile USB with ALSA and Jack
========================================================
Thibault Le Meur <Thibault.LeMeur@supelec.fr>
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
- 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
- Added AC3/DTS passthru info
1 - Audiophile USB Specs and correct usage
==========================================
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)
@ -34,56 +43,69 @@ The device has 4 audio interfaces, and 2 MIDI ports:
* 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"
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
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
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
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
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)
synchronization error (for instance sound played at an odd sample rate)
2 - Audiophile USB MIDI support in ALSA
=======================================
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
========================================
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".
``device_setup``.
3.1 - Default Alsa driver mode
------------------------------
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
@ -101,6 +123,7 @@ 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
@ -115,20 +138,28 @@ 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
* 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
* 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
* 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
* 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
---------------------------
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
@ -137,34 +168,39 @@ 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
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
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
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
16-bit modes
~~~~~~~~~~~~
The two supported modes are:
* device_setup=0x01
* ``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
* ``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
@ -173,33 +209,43 @@ The two supported modes are:
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
24-bit modes
~~~~~~~~~~~~
The three supported modes are:
* device_setup=0x09
* ``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
* ``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
* ``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
@ -210,29 +256,35 @@ The three supported modes are:
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
AC3 w/ DTS passthru mode
~~~~~~~~~~~~~~~~~~~~~~~~
Thanks to Hakan Lennestal, I now have a report saying that this mode works.
* device_setup=0x03
* ``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
----------------------------------------------
How to use the ``device_setup`` parameter
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
The parameter can be given:
* By manually probing the device (as root):
* 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:
(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
@ -250,26 +302,31 @@ CAUTION when initializing the device
* 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
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
-------------------------------------
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
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|
@ -278,38 +335,55 @@ The device_setup is one byte long and its structure is the following:
+---+---+---+---+---+---+---+---+
Where:
* b0 is the "SET" bit
* b0 is the ``SET`` bit
- it MUST be set if device_setup is initialized
* b1 is the "DTS" bit
* 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
- 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
- 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
- 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"
* 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
USB implementation details for this device
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
You may safely skip this section if you're not interested in driver
hacking.
@ -319,46 +393,72 @@ 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
* 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
* AltSettings 2 corresponds to
- 24-bit depth, 8-48kHz sample mode
- Asynch capture and playback (Ao,Ai,Do,Di)
* AltSettings 3 corresponds to
* AltSettings 3 corresponds to
- 24-bit depth, 8-48kHz sample mode
- Synch capture (Ai) and Adaptive playback (Ao,Do)
* AltSettings 4 corresponds to
* AltSettings 4 corresponds to
- 16-bit depth, 8-48kHz sample mode
- Asynch capture and playback (Ao,Ai,Do,Di)
* AltSettings 5 corresponds to
* AltSettings 5 corresponds to
- 16-bit depth, 8-48kHz sample mode
- Synch capture (Ai) and Adaptive playback (Ao,Do)
* AltSettings 6 corresponds to
* 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:
*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
@ -371,20 +471,21 @@ _must_know_ how the device will be used:
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
``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
===================================
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
-----------------------------
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
@ -396,29 +497,35 @@ 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
-----------------------
Using Alsa plughw
-----------------
If you don't have a recent Jackd installed, you can downgrade to using
the Alsa "plug" converter.
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."
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
------------------------------------------------------
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)
@ -432,6 +539,7 @@ 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

23
Documentation/sound/alsa/Bt87x.txt → Documentation/sound/cards/bt87x.rst
View File

@ -1,18 +1,23 @@
=================
ALSA BT87x Driver
=================
Intro
=====
You might have noticed that the bt878 grabber cards have actually
_two_ PCI functions:
*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)
[ ... ]
$ 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
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...
@ -30,9 +35,9 @@ 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
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
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

62
Documentation/sound/alsa/CMIPCI.txt → Documentation/sound/cards/cmipci.rst
View File

@ -1,7 +1,8 @@
Brief Notes on C-Media 8338/8738/8768/8770 Driver
=================================================
=================================================
Brief Notes on C-Media 8338/8738/8768/8770 Driver
=================================================
Takashi Iwai <tiwai@suse.de>
Takashi Iwai <tiwai@suse.de>
Front/Rear Multi-channel Playback
@ -30,19 +31,20 @@ 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 ***
.. WARNING::
When "Four Channel Mode" switch is off, the output from rear speakers
will be FULL VOLUME regardless of Master and PCM volumes [#]_.
This might damage your audio equipment. Please disconnect speakers
before your turn off this switch.
[ Well.. I once got the output with correct volume (i.e. same with the
.. [#]
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... ]
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
@ -73,12 +75,14 @@ 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
@ -89,13 +93,15 @@ 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
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
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
@ -111,11 +117,14 @@ 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.
@ -140,31 +149,40 @@ 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
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)
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
IEC958 Loop
SPDIF input data is loop back into SPDIF
output (aka bypass)
"IEC958 Copyright" - Set the copyright bit.
IEC958 Copyright
Set the copyright bit.
"IEC958 5V" - Select 0.5V (coax) or 5V (optical) interface.
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 Monitor
SPDIF input is routed to DAC.
"IEC958 In Phase Inverse" - Set SPDIF input format as inverse.
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.
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.
@ -217,7 +235,7 @@ to enable MIDI support. Valid I/O ports are 0x300, 0x310, 0x320 and
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
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

20
Documentation/sound/alsa/emu10k1-jack.txt → Documentation/sound/cards/emu10k1-jack.rst
View File

@ -1,3 +1,7 @@
=================================================================
Low latency, multichannel audio with JACK and the emu10k1/emu10k2
=================================================================
This document is a guide to using the emu10k1 based devices with JACK for low
latency, multichannel recording functionality. All of my recent work to allow
Linux users to use the full capabilities of their hardware has been inspired
@ -7,8 +11,6 @@ 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
@ -23,14 +25,15 @@ 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
/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).
sb-live-mixer.rst (or audigy-mixer.rst).
The 16 input ports are connected to the 16 physical inputs. Contrary to
popular belief, all emu10k1 cards are multichannel cards. Which of these
@ -49,10 +52,11 @@ 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 (& 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)
@ -69,6 +73,6 @@ 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

253
Documentation/sound/alsa/hdspm.txt → Documentation/sound/cards/hdspm.rst
View File

@ -1,21 +1,24 @@
=======================================
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.
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:
Hardware functionality
======================
Audio transmission:
Audio transmission
------------------
number of channels -- depends on transmission mode
* 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,
@ -23,31 +26,34 @@ Software Interface ALSA-DSP MADI Driver
allocated. So also the throughput of the PCI system can be
scaled. (Only important for low performance boards).
Single Speed -- 1..64 channels
* Single Speed -- 1..64 channels
.. note::
(Note: Choosing the 56channel mode for transmission or as
receiver, only 56 are transmitted/received over the MADI, but
all 64 channels are available for the mixer, so channel count
for the driver)
Double Speed -- 1..32 channels
* Double Speed -- 1..32 channels
.. note::
Note: Choosing the 56-channel mode for
transmission/receive-mode , only 28 are transmitted/received
over the MADI, but all 32 channels are available for the mixer,
so channel count for the driver
Quad Speed -- 1..16 channels
* Quad Speed -- 1..16 channels
Note: Choosing the 56-channel mode for
.. 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)
* Format -- signed 32 Bit Little Endian (SNDRV_PCM_FMTBIT_S32_LE)
Sample Rates --
* Sample Rates --
Single Speed -- 32000, 44100, 48000
@ -55,14 +61,13 @@ Software Interface ALSA-DSP MADI Driver
Quad Speed -- 128000, 176400, 192000 (untested)
access-mode -- MMAP (memory mapped), Not interleaved
(PCM_NON-INTERLEAVED)
* access-mode -- MMAP (memory mapped), Not interleaved (PCM_NON-INTERLEAVED)
buffer-sizes -- 64,128,256,512,1024,2048,8192 Samples
* buffer-sizes -- 64,128,256,512,1024,2048,8192 Samples
fragments -- 2
* fragments -- 2
Hardware-pointer -- 2 Modi
* Hardware-pointer -- 2 Modi
The Card supports the readout of the actual Buffer-pointer,
@ -74,53 +79,54 @@ Software Interface ALSA-DSP MADI Driver
precise-pointer.
.. hint::
(Hint: Experimenting I found that the pointer is maximum 64 to
large never to small. So if you subtract 64 you always have a
safe pointer for writing, which is used on this mode inside
ALSA. In theory now you can get now a latency as low as 16
Samples, which is a quarter of the interrupt possibilities.)
Precise Pointer -- off
* Precise Pointer -- off
interrupt used for pointer-calculation
Precise Pointer -- on
* Precise Pointer -- on
hardware pointer used.
Controller:
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.
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 !!!
Also all 128+256 Peak and RMS-Meter can be accessed via the
hwdep-interface. Since it could be a performance problem always
copying and converting Peak and RMS-Levels even if you just need
one, I decided to export the hardware structure, so that of
needed some driver-guru can implement a memory-mapping of mixer
or peak-meters over ioctl, or also to do only copying and no
conversion. A test-application shows the usage of the controller.
* Latency Controls --- not implemented !!!
.. note::
Note: Within the windows-driver the latency is accessible of a
control-panel, but buffer-sizes are controlled with ALSA from
hwparams-calls and should not be changed in run-state, I did not
implement it here.
System Clock -- suspended !!!!
* System Clock -- suspended !!!!
Name -- "System Clock Mode"
Access -- Read Write
Values -- "Master" "Slave"
* Name -- "System Clock Mode"
* Access -- Read Write
* Values -- "Master" "Slave"
.. note::
!!!! This is a hardware-function but is in conflict with the
Clock-source controller, which is a kind of ALSA-standard. I
makes sense to set the card to a special mode (master at some
@ -128,106 +134,107 @@ Software Interface ALSA-DSP MADI Driver
a studio should have working synchronisations setup. So use
Clock-source-controller instead !!!!
Clock Source
* Clock Source
Name -- "Sample Clock Source"
* Name -- "Sample Clock Source"
Access -- Read Write
* 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"
* Values -- "AutoSync", "Internal 32.0 kHz", "Internal 44.1 kHz",
"Internal 48.0 kHz", "Internal 64.0 kHz", "Internal 88.2 kHz",
"Internal 96.0 kHz"
Choose between Master at a specific Frequency and so also the
Speed-mode or Slave (Autosync). Also see "Preferred Sync Ref"
.. warning::
!!!! This is no pure hardware function but was implemented by
ALSA by some ALSA-drivers before, so I use it also. !!!
Preferred Sync Ref
* Preferred Sync Ref
Name -- "Preferred Sync Reference"
* Name -- "Preferred Sync Reference"
Access -- Read Write
* Access -- Read Write
Values -- "Word" "MADI"
* Values -- "Word" "MADI"
Within the Auto-sync-Mode the preferred Sync Source can be
chosen. If it is not available another is used if possible.
.. note::
Note: Since MADI has a much higher bit-rate than word-clock, the
card should synchronise better in MADI Mode. But since the
RME-PLL is very good, there are almost no problems with
word-clock too. I never found a difference.
TX 64 channel ---
* TX 64 channel
Name -- "TX 64 channels mode"
* Name -- "TX 64 channels mode"
Access -- Read Write
* Access -- Read Write
Values -- 0 1
* Values -- 0 1
Using 64-channel-modus (1) or 56-channel-modus for
MADI-transmission (0).
.. note::
Note: This control is for output only. Input-mode is detected
automatically from hardware sending MADI.
Clear TMS ---
* Clear TMS
Name -- "Clear Track Marker"
* Name -- "Clear Track Marker"
Access -- Read Write
* Access -- Read Write
Values -- 0 1
* Values -- 0 1
Don't use to lower 5 Audio-bits on AES as additional Bits.
Safe Mode oder Auto Input ---
* Safe Mode oder Auto Input
Name -- "Safe Mode"
* Name -- "Safe Mode"
Access -- Read Write
* Access -- Read Write
Values -- 0 1
(default on)
* 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 ---
* Input
Name -- "Input Select"
* Name -- "Input Select"
Access -- Read Write
* Access -- Read Write
Values -- optical coaxial
* Values -- optical coaxial
Choosing the Input, optical or coaxial. If Safe-mode is active,
this is the preferred Input.
-------------- Mixer ----------------------
Mixer
-----
Mixer
* Mixer
Name -- "Mixer"
* Name -- "Mixer"
Access -- Read Write
* Access -- Read Write
Values - <channel-number 0-127> <Value 0-65535>
* Values - <channel-number 0-127> <Value 0-65535>
Here as a first value the channel-index is taken to get/set the
@ -235,40 +242,41 @@ Software Interface ALSA-DSP MADI Driver
fader and 64-127 the playback to outputs fader. Value 0
is channel muted 0 and 32768 an amplification of 1.
Chn 1-64
* Chn 1-64
fast mixer for the ALSA-mixer utils. The diagonal of the
mixer-matrix is implemented from playback to output.
Line Out
* Line Out
Name -- "Line Out"
* Name -- "Line Out"
Access -- Read Write
* Access -- Read Write
Values -- 0 1
* 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):
Information (only read access)
------------------------------
Sample Rate
* Sample Rate
Name -- "System Sample Rate"
* Name -- "System Sample Rate"
Access -- Read-only
* Access -- Read-only
getting the sample rate.
External Rate measured
* External Rate measured
Name -- "External Rate"
* Name -- "External Rate"
Access -- Read only
* Access -- Read only
Should be "Autosync Rate", but Name used is
@ -276,79 +284,86 @@ Software Interface ALSA-DSP MADI Driver
reported.
MADI Sync Status
* MADI Sync Status
Name -- "MADI Sync Lock Status"
* Name -- "MADI Sync Lock Status"
Access -- Read
* Access -- Read
Values -- 0,1,2
* Values -- 0,1,2
MADI-Input is 0=Unlocked, 1=Locked, or 2=Synced.
Word Clock Sync Status
* Word Clock Sync Status
Name -- "Word Clock Lock Status"
* Name -- "Word Clock Lock Status"
Access -- Read
* Access -- Read
Values -- 0,1,2
* Values -- 0,1,2
Word Clock Input is 0=Unlocked, 1=Locked, or 2=Synced.
AutoSync
* AutoSync
Name -- "AutoSync Reference"
* Name -- "AutoSync Reference"
Access -- Read
* Access -- Read
Values -- "WordClock", "MADI", "None"
* Values -- "WordClock", "MADI", "None"
Sync-Reference is either "WordClock", "MADI" or none.
RX 64ch --- noch nicht implementiert
* RX 64ch --- noch nicht implementiert
MADI-Receiver is in 64 channel mode oder 56 channel mode.
AB_inp --- not tested
* AB_inp --- not tested
Used input for Auto-Input.
actual Buffer Position --- not implemented
* actual Buffer Position --- not implemented
!!! this is a ALSA internal function, so no control is used !!!
Calling Parameter:
Calling Parameter
=================
index int array (min = 1, max = 8),
"Index value for RME HDSPM interface." card-index within ALSA
* 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."
* 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."
* 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."
* precise_ptr int array (min = 1, max = 8)
Enable precise pointer, or disable.
.. note::
note: Use only when the application supports this (which is a special case).
line_outs_monitor int array (min = 1, max = 8),
"Send playback streams to analog outs by default."
* line_outs_monitor int array (min = 1, max = 8)
Send playback streams to analog outs by default.
.. note::
note: each playback channel is mixed to the same numbered output
channel (routed). This is against the ALSA-convention, where all
channels have to be muted on after loading the driver, but was
@ -356,7 +371,9 @@ Calling Parameter:
enable_monitor int array (min = 1, max = 8),
"Enable Analog Out on Channel 63/64 by default."
* enable_monitor int array (min = 1, max = 8)
Enable Analog Out on Channel 63/64 by default.
.. note ::
note: here the analog output is enabled (but not routed).

24
Documentation/sound/alsa/img,spdif-in.txt → Documentation/sound/cards/img-spdif-in.rst
View File

@ -1,21 +1,25 @@
================================================
Imagination Technologies SPDIF Input Controllers
================================================
The Imagination Technologies SPDIF Input controller contains the following
controls:
name='IEC958 Capture Mask',index=0
* name='IEC958 Capture Mask',index=0
This control returns a mask that shows which of the IEC958 status bits
can be read using the 'IEC958 Capture Default' control.
name='IEC958 Capture Default',index=0
* name='IEC958 Capture Default',index=0
This control returns the status bits contained within the SPDIF stream that
is being received. The 'IEC958 Capture Mask' shows which bits can be read
from this control.
name='SPDIF In Multi Frequency Acquire',index=0
name='SPDIF In Multi Frequency Acquire',index=1
name='SPDIF In Multi Frequency Acquire',index=2
name='SPDIF In Multi Frequency Acquire',index=3
* name='SPDIF In Multi Frequency Acquire',index=0
* name='SPDIF In Multi Frequency Acquire',index=1
* name='SPDIF In Multi Frequency Acquire',index=2
* name='SPDIF In Multi Frequency Acquire',index=3
This control is used to attempt acquisition of up to four different sample
rates. The active rate can be obtained by reading the 'SPDIF In Lock Frequency'
@ -29,21 +33,21 @@ four sample rates set here.
If less than four rates are required, the same rate can be specified more than
once
name='SPDIF In Lock Frequency',index=0
* name='SPDIF In Lock Frequency',index=0
This control returns the active capture rate, or 0 if a lock has not been
acquired
name='SPDIF In Lock TRK',index=0
* name='SPDIF In Lock TRK',index=0
This control is used to modify the locking/jitter rejection characteristics
of the block. Larger values increase the locking range, but reduce jitter
rejection.
name='SPDIF In Lock Acquire Threshold',index=0
* name='SPDIF In Lock Acquire Threshold',index=0
This control is used to change the threshold at which a lock is acquired.
name='SPDIF In Lock Release Threshold',index=0
* name='SPDIF In Lock Release Threshold',index=0
This control is used to change the threshold at which a lock is released.

19
Documentation/sound/cards/index.rst Normal file
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@ -0,0 +1,19 @@
Card-Specific Information
=========================
.. toctree::
:maxdepth: 2
joystick
cmipci
sb-live-mixer
audigy-mixer
emu10k1-jack
via82xx-mixer
audiophile-usb
mixart
bt87x
maya44
hdspm
serial-u16550
img-spdif-in

69
Documentation/sound/alsa/Joystick.txt → Documentation/sound/cards/joystick.rst
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@ -1,7 +1,10 @@
=======================================
Analog Joystick Support on ALSA Drivers
=======================================
Oct. 14, 2003
Takashi Iwai <tiwai@suse.de>
Oct. 14, 2003
Takashi Iwai <tiwai@suse.de>
General
-------
@ -34,44 +37,46 @@ 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)
---------------------------------------------------------------------------
============== ============= ============================================
Driver Module Option Available Values
============== ============= ============================================
als4000 joystick_port 0 = disable (default), 1 = auto-detect,
manual: any address (e.g. 0x200)
au88x0 N/A N/A
azf3328 joystick 0 = disable, 1 = enable, -1 = auto (default)
ens1370 joystick 0 = disable (default), 1 = enable
ens1371 joystick_port 0 = disable (default), 1 = auto-detect,
manual: 0x200, 0x208, 0x210, 0x218
cmipci joystick_port 0 = disable (default), 1 = auto-detect,
manual: any address (e.g. 0x200)
cs4281 N/A N/A
cs46xx N/A N/A
es1938 N/A N/A
es1968 joystick 0 = disable (default), 1 = enable
sonicvibes N/A N/A
trident N/A N/A
via82xx [#f1]_ joystick 0 = disable (default), 1 = enable
ymfpci joystick_port 0 = disable (default), 1 = auto-detect,
manual: 0x201, 0x202, 0x204, 0x205 [#f2]_
============== ============= ============================================
*1) VIA686A/B only
*2) With YMF744/754 chips, the port address can be chosen arbitrarily
.. [#f1] VIA686A/B only
.. [#f2] With YMF744/754 chips, the port address can be chosen arbitrarily
The following drivers don't support gameport natively, but there are
additional modules. Load the corresponding module to add the gameport
support.
Driver Additional Module
-----------------------------
emu10k1 emu10k1-gp
fm801 fm801-gp
-----------------------------
======= =================
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.
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"

137
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@ -1,10 +1,18 @@
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
=================================
Notes on Maya44 USB Audio Support
=================================
----------------------------------------------------------------
.. note::
The following is the original document of Rainer's patch that the
current maya44 code based on. Some contents might be obsoleted, but I
keep here as reference -- tiwai
Feb 14, 2008
Rainer Zimmermann <mail@lightshed.de>
STATE OF DEVELOPMENT:
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).
@ -44,16 +52,17 @@ Things that do not seem to work:
- 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:
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
* 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.
@ -62,25 +71,26 @@ This is done in maya44.c, mainly because some of the WM8776 controls are used in
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
* 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:
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.
@ -98,66 +108,79 @@ I propose some additional code for limiting the sampling rate when setting on a
The proposed code (currently deactivated) is in ice1712.h.patch, ice1724.c and maya44.c (in pci/ice1712).
SOUND DEVICES:
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)
* 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:
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
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.
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).
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.
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
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.
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:
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
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.
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
PCM out
DAC output of the corresponding channel (default setting)
Input 1 ... Input 4
direct routing from ADC output of the selected input channel

26
Documentation/sound/alsa/MIXART.txt → Documentation/sound/cards/mixart.rst
View File

@ -1,5 +1,8 @@
Alsa driver for Digigram miXart8 and miXart8AES/EBU soundcards
Digigram <alsa@digigram.com>
==============================================================
Alsa driver for Digigram miXart8 and miXart8AES/EBU soundcards
==============================================================
Digigram <alsa@digigram.com>
GENERAL
@ -48,11 +51,15 @@ 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.
<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
@ -79,11 +86,14 @@ FIRMWARE
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.)
/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

321
Documentation/sound/alsa/SB-Live-mixer.txt → Documentation/sound/cards/sb-live-mixer.rst
View File

@ -1,6 +1,6 @@
Sound Blaster Live mixer / default DSP code
===========================================
===========================================
Sound Blaster Live mixer / default DSP code
===========================================
The EMU10K1 chips have a DSP part which can be programmed to support
@ -12,8 +12,8 @@ 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
--------------------------
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
@ -27,8 +27,8 @@ at the time.
Look to tram_poke routines in lowlevel/emu10k1/emufx.c for more details.
2) Digital mixer controls
-------------------------
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
@ -40,317 +40,334 @@ is mentioned in multiple controls, the signal is accumulated and can be wrapped
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.
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
``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
``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
``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
``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
``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
``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
``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
``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
``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
``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
``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
``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
.. 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
.. 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
``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
``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
``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
``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
``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
``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
``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
``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
``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
``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
``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
``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
``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
``name='Headphone Playback Volume',index=1``
--------------------------------------------
This control attenuates the samples for the headphone output.
name='Headphone Center Playback Switch',index=1
``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
``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
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
* 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
* 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
``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)
* 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:
-------------------
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
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 9901813 (A1)
Audio Effects Processor with multiple asynchronous streams
(Jan. 14, 1999)
WO 9901953 (A1) Audio Effects Processor having Decoupled Instruction
Execution and Audio Data Sequencing (Jan. 14, 1999)
WO 9901814 (A1)
Processor with Instruction Set for Audio Effects (Jan. 14, 1999)
WO 9901953 (A1)
Audio Effects Processor having Decoupled Instruction
Execution and Audio Data Sequencing (Jan. 14, 1999)
US Patents (http://www.uspto.gov/)
----------------------------------
US 5925841 Digital Sampling Instrument employing cache memory (Jul. 20, 1999)
US 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 5928342
Audio Effects Processor integrated on a single chip
with a multiport memory onto which multiple asynchronous
digital sound samples can be concurrently loaded
(Jul. 27, 1999)
US 5930158 Processor with Instruction Set for Audio Effects (Jul. 27, 1999)
US 5930158
Processor with Instruction Set for Audio Effects (Jul. 27, 1999)
US 6032235 Memory initialization circuit (Tram) (Feb. 29, 2000)
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 6138207
Interpolation looping of audio samples in cache connected to
system bus with prioritization and modification of bus transfers
in accordance with loop ends and minimum block sizes
(Oct. 24, 2000)
US 6151670 Method for conserving memory storage using a (Nov. 21, 2000)
pool of short term memory registers
US 6151670
Method for conserving memory storage using a
pool of short term memory registers
(Nov. 21, 2000)
US 6195715 Interrupt control for multiple programs communicating with (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
US 6195715
Interrupt control for multiple programs communicating with
a common interrupt by associating programs to GP registers,
defining interrupt register, polling GP registers, and invoking
callback routine associated with defined interrupt register
(Feb. 27, 2001)

21
Documentation/sound/alsa/serial-u16550.txt → Documentation/sound/cards/serial-u16550.rst
View File

@ -1,14 +1,14 @@
Serial UART 16450/16550 MIDI driver
===================================
===================================
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)
* 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.
@ -22,11 +22,13 @@ 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
@ -37,6 +39,7 @@ 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 \
@ -47,6 +50,7 @@ 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
@ -63,6 +67,7 @@ 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

8
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@ -0,0 +1,8 @@
=============
VIA82xx mixer
=============
On many VIA82xx boards, the ``Input Source Select`` mixer control does not work.
Setting it to ``Input2`` on such boards will cause recording to hang, or fail
with EIO (input/output error) via OSS emulation. This control should be left
at ``Input1`` for such cards.

77
Documentation/sound/alsa/Channel-Mapping-API.txt → Documentation/sound/designs/channel-mapping-api.rst
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@ -1,9 +1,11 @@
============================
ALSA PCM channel-mapping API
============================
Takashi Iwai <tiwai@suse.de>
GENERAL
-------
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
@ -11,9 +13,9 @@ 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 }
``{ front_left, front_right }``
while a 4.0 surround PCM stream has a channel map of
{ front left, front right, rear left, rear right }.
``{ 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
@ -29,8 +31,8 @@ specification. These are the main motivations for the new channel
mapping API.
DESIGN
------
Design
======
Actually, "the channel mapping API" doesn't introduce anything new in
the kernel/user-space ABI perspective. It uses only the existing
@ -39,10 +41,11 @@ 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
* 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.
@ -50,32 +53,35 @@ 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
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...
``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
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
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.
These new TLV types are defined in ``sound/tlv.h``.
The available channel position values are defined in sound/asound.h,
The available channel position values are defined in ``sound/asound.h``,
here is a cut:
/* channel positions */
enum {
::
/* channel positions */
enum {
SNDRV_CHMAP_UNKNOWN = 0,
SNDRV_CHMAP_NA, /* N/A, silent */
SNDRV_CHMAP_MONO, /* mono stream */
@ -107,11 +113,13 @@ enum {
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
@ -120,19 +128,21 @@ to be returned will contain such as:
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)
#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,
``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
When ``SNDRV_CHMAP_DRIVER_SPEC`` is set, all the channel position values
don't follow the standard definition above but driver-specific.
* READ OPERATION
Read Operation
--------------
The control read operation is for providing the current channel map of
the given stream. The control element returns an integer array
@ -140,9 +150,10 @@ 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.
``UNKNOWN``.
* WRITE OPERATION
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

119
Documentation/sound/alsa/compress_offload.txt → Documentation/sound/designs/compress-offload.rst
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@ -1,10 +1,14 @@
compress_offload.txt
=====================
Pierre-Louis.Bossart <pierre-louis.bossart@linux.intel.com>
Vinod Koul <vinod.koul@linux.intel.com>
=========================
ALSA Compress-Offload API
=========================
Pierre-Louis.Bossart <pierre-louis.bossart@linux.intel.com>
Vinod Koul <vinod.koul@linux.intel.com>
Overview
========
Since its early days, the ALSA API was defined with PCM support or
constant bitrates payloads such as IEC61937 in mind. Arguments and
returned values in frames are the norm, making it a challenge to
@ -27,8 +31,9 @@ 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
Requirements
============
The main requirements are:
- separation between byte counts and time. Compressed formats may have
@ -63,7 +68,7 @@ The main requirements are:
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
- 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
@ -72,7 +77,7 @@ The main requirements are:
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.
@ -95,43 +100,44 @@ 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_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.
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
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_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.
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.
@ -145,6 +151,7 @@ Modifications include:
- 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
@ -162,19 +169,19 @@ 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_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
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
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
@ -189,10 +196,12 @@ Sequence flow for gapless would be:
- 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:
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.
@ -225,7 +234,9 @@ Not supported:
rendered output in time, this does not deal with underrun/overrun and
maybe dealt in user-library
Credits:
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

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@ -0,0 +1,142 @@
===========================
Standard ALSA Control Names
===========================
This document describes standard names of mixer controls.
Standard Syntax
---------------
Syntax: [LOCATION] SOURCE [CHANNEL] [DIRECTION] FUNCTION
DIRECTION
~~~~~~~~~
================ ===============
<nothing> both directions
Playback one direction
Capture one direction
Bypass Playback one direction
Bypass Capture one direction
================ ===============
FUNCTION
~~~~~~~~
======== =================================
Switch on/off switch
Volume amplifier
Route route control, hardware specific
======== =================================
CHANNEL
~~~~~~~
============ ==================================================
<nothing> channel independent, or applies to all channels
Front front left/right channels
Surround rear left/right in 4.0/5.1 surround
CLFE C/LFE channels
Center center cannel
LFE LFE channel
Side side left/right for 7.1 surround
============ ==================================================
LOCATION (Physical location of source)
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
============ =====================
Front front position
Rear rear position
Dock on docking station
Internal internal
============ =====================
SOURCE
~~~~~~
=================== =================================================
Master
Master Mono
Hardware Master
Speaker internal speaker
Bass Speaker internal LFE speaker
Headphone
Line Out
Beep beep generator
Phone
Phone Input
Phone Output
Synth
FM
Mic
Headset Mic mic part of combined headset jack - 4-pin
headphone + mic
Headphone Mic mic part of either/or - 3-pin headphone or mic
Line input only, use "Line Out" for output
CD
Video
Zoom Video
Aux
PCM
PCM Pan
Loopback
Analog Loopback D/A -> A/D loopback
Digital Loopback playback -> capture loopback -
without analog path
Mono
Mono Output
Multi
ADC
Wave
Music
I2S
IEC958
HDMI
SPDIF output only
SPDIF In
Digital In
HDMI/DP either HDMI or DisplayPort
=================== =================================================
Exceptions (deprecated)
-----------------------
===================================== =======================
[Analogue|Digital] Capture Source
[Analogue|Digital] Capture Switch aka input gain switch
[Analogue|Digital] Capture Volume aka input gain volume
[Analogue|Digital] Playback Switch aka output gain switch
[Analogue|Digital] Playback Volume aka output gain volume
Tone Control - Switch
Tone Control - Bass
Tone Control - Treble
3D Control - Switch
3D Control - Center
3D Control - Depth
3D Control - Wide
3D Control - Space
3D Control - Level
Mic Boost [(?dB)]
===================================== =======================
PCM interface
-------------
=================== ========================================
Sample Clock Source { "Word", "Internal", "AutoSync" }
Clock Sync Status { "Lock", "Sync", "No Lock" }
External Rate external capture rate
Capture Rate capture rate taken from external source
=================== ========================================
IEC958 (S/PDIF) interface
-------------------------
============================================ ======================================
IEC958 [...] [Playback|Capture] Switch turn on/off the IEC958 interface
IEC958 [...] [Playback|Capture] Volume digital volume control
IEC958 [...] [Playback|Capture] Default default or global value - read/write
IEC958 [...] [Playback|Capture] Mask consumer and professional mask
IEC958 [...] [Playback|Capture] Con Mask consumer mask
IEC958 [...] [Playback|Capture] Pro Mask professional mask
IEC958 [...] [Playback|Capture] PCM Stream the settings assigned to a PCM stream
IEC958 Q-subcode [Playback|Capture] Default Q-subcode bits
IEC958 Preamble [Playback|Capture] Default burst preamble words (4*16bits)
============================================ ======================================

15
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@ -0,0 +1,15 @@
Designs and Implementations
===========================
.. toctree::
:maxdepth: 2
control-names
channel-mapping-api
compress-offload
timestamping
jack-controls
procfile
powersave
oss-emulation
seq-oss

13
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@ -1,3 +1,7 @@
==================
ALSA Jack Controls
==================
Why we need Jack kcontrols
==========================
@ -29,11 +33,12 @@ How to use jack kcontrols
=========================
In order to keep compatibility, snd_jack_new() has been modified by
adding two params :-
adding two params:
- @initial_kctl: if true, create a kcontrol and add it to the jack
list.
- @phantom_jack: Don't create a input device for phantom jacks.
initial_kctl
if true, create a kcontrol and add it to the jack list.
phantom_jack
Don't create a input device for phantom jacks.
HDA jacks can set phantom_jack to true in order to create a phantom
jack and set initial_kctl to true to create an initial kcontrol with

167
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@ -1,7 +1,8 @@
NOTES ON KERNEL OSS-EMULATION
=============================
=============================
Notes on Kernel OSS-Emulation
=============================
Jan. 22, 2004 Takashi Iwai <tiwai@suse.de>
Jan. 22, 2004 Takashi Iwai <tiwai@suse.de>
Modules
@ -14,18 +15,18 @@ 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
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
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.
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
``alias sound-slot-0 snd-card-0`` doesn't work any more like the old
modutils).
The currently available OSS configuration is shown in
@ -42,6 +43,7 @@ Device Mapping
==============
ALSA supports the following OSS device files:
::
PCM:
/dev/dspX
@ -61,48 +63,55 @@ ALSA supports the following OSS device files:
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.)
/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
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)
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,
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)
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
@ -118,43 +127,52 @@ 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
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
``app_name`` is the name of application with (higher priority) or without
path.
fragments specifies the number of fragments or zero if no specific
``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
``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
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
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
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
@ -171,10 +189,11 @@ 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.
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
@ -183,15 +202,18 @@ 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
Also, for clearing all the current configuration, send ``erase`` command
as below:
::
% echo "erase" > /proc/asound/card0/pcm0p/oss
@ -211,40 +233,43 @@ 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
==================== ===================== =====
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.
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
@ -261,6 +286,7 @@ 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
@ -284,12 +310,18 @@ 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.
@ -302,4 +334,3 @@ 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.

16
Documentation/sound/alsa/powersave.txt → Documentation/sound/designs/powersave.rst
View File

@ -1,9 +1,10 @@
==========================
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.
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
@ -11,20 +12,21 @@ 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
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
``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
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):
``/sys/modules/snd_ac97_codec/parameters/power_save`` (usually as root):
::
# echo 10 > /sys/modules/snd_ac97_codec/parameters/power_save

106
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View File

@ -1,20 +1,22 @@
Proc Files of ALSA Drivers
==========================
Takashi Iwai <tiwai@suse.de>
==========================
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.
card-specific files are stored in the ``card*`` subdirectories.
Global Information
------------------
==================
cards
Shows the list of currently configured ALSA drivers,
@ -31,15 +33,15 @@ devices
meminfo
Shows the status of allocated pages via ALSA drivers.
Appears only when CONFIG_SND_DEBUG=y.
Appears only when ``CONFIG_SND_DEBUG=y``.
hwdep
Lists the currently available hwdep devices in format of
<card>-<device>: <name>
``<card>-<device>: <name>``
pcm
Lists the currently available PCM devices in format of
<card>-<device>: <id>: <name> : <sub-streams>
``<card>-<device>: <id>: <name> : <sub-streams>``
timer
Lists the currently available timer devices
@ -54,23 +56,23 @@ oss/sndstat
Card Specific Files
-------------------
===================
The card-specific files are found in /proc/asound/card* directories.
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
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
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
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#*
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),
@ -81,26 +83,27 @@ details.
PCM Proc Files
--------------
==============
card*/pcm*/info
``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.
``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
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 0 = Enable XRUN/jiffies debug messages
* bit 1 = Show stack trace at XRUN / jiffies check
* bit 2 = Enable additional jiffies check
When the bit 0 is set, the driver will show the messages to
kernel log when an xrun is detected. The debug message is
@ -117,72 +120,74 @@ card*/pcm*/xrun_debug
buggy) hardware that doesn't give smooth pointer updates.
This feature is enabled via the bit 2.
card*/pcm*/sub*/info
``card*/pcm*/sub*/info``
The general information of this PCM sub-stream.
card*/pcm*/sub*/status
``card*/pcm*/sub*/status``
The current status of this PCM sub-stream, elapsed time,
H/W position, etc.
card*/pcm*/sub*/hw_params
``card*/pcm*/sub*/hw_params``
The hardware parameters set for this sub-stream.
card*/pcm*/sub*/sw_params
``card*/pcm*/sub*/sw_params``
The soft parameters set for this sub-stream.
card*/pcm*/sub*/prealloc
``card*/pcm*/sub*/prealloc``
The buffer pre-allocation information.
card*/pcm*/sub*/xrun_injection
``card*/pcm*/sub*/xrun_injection``
Triggers an XRUN to the running stream when any value is
written to this proc file. Used for fault injection.
This entry is write-only.
AC97 Codec Information
----------------------
======================
card*/codec97#*/ac97#?-?
``card*/codec97#*/ac97#?-?``
Shows the general information of this AC97 codec chip, such as
name, capabilities, set up.
card*/codec97#0/ac97#?-?+regs
``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*
``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#*
``card*/codec#*``
Shows the general codec information and the attribute of each
widget node.
card*/eld#*
``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#*`.
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.
@ -203,7 +208,7 @@ seq/oss
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
@ -211,24 +216,23 @@ 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)
- 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
- 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
- AC97 proc files, ``codec97#*/*`` files
for USB audio/midi,
- output of lsusb -v
- stream* files in card directory
- output of ``lsusb -v``
- ``stream*`` files in card directory
The ALSA bug-tracking system is found at:
https://bugtrack.alsa-project.org/alsa-bug/
https://bugtrack.alsa-project.org/alsa-bug/

371
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@ -0,0 +1,371 @@
===============================
OSS Sequencer Emulation on ALSA
===============================
Copyright (c) 1998,1999 by Takashi Iwai
ver.0.1.8; Nov. 16, 1999
Description
===========
This directory contains the OSS sequencer emulation driver on ALSA. Note
that this program is still in the development state.
What this does - it provides the emulation of the OSS sequencer, access
via ``/dev/sequencer`` and ``/dev/music`` devices.
The most of applications using OSS can run if the appropriate ALSA
sequencer is prepared.
The following features are emulated by this driver:
* Normal sequencer and MIDI events:
They are converted to the ALSA sequencer events, and sent to the
corresponding port.
* Timer events:
The timer is not selectable by ioctl. The control rate is fixed to
100 regardless of HZ. That is, even on Alpha system, a tick is always
1/100 second. The base rate and tempo can be changed in ``/dev/music``.
* Patch loading:
It purely depends on the synth drivers whether it's supported since
the patch loading is realized by callback to the synth driver.
* I/O controls:
Most of controls are accepted. Some controls
are dependent on the synth driver, as well as even on original OSS.
Furthermore, you can find the following advanced features:
* Better queue mechanism:
The events are queued before processing them.
* Multiple applications:
You can run two or more applications simultaneously (even for OSS
sequencer)!
However, each MIDI device is exclusive - that is, if a MIDI device
is opened once by some application, other applications can't use
it. No such a restriction in synth devices.
* Real-time event processing:
The events can be processed in real time without using out of bound
ioctl. To switch to real-time mode, send ABSTIME 0 event. The followed
events will be processed in real-time without queued. To switch off the
real-time mode, send RELTIME 0 event.
* ``/proc`` interface:
The status of applications and devices can be shown via
``/proc/asound/seq/oss`` at any time. In the later version,
configuration will be changed via ``/proc`` interface, too.
Installation
============
Run configure script with both sequencer support (``--with-sequencer=yes``)
and OSS emulation (``--with-oss=yes``) options. A module ``snd-seq-oss.o``
will be created. If the synth module of your sound card supports for OSS
emulation (so far, only Emu8000 driver), this module will be loaded
automatically.
Otherwise, you need to load this module manually.
At beginning, this module probes all the MIDI ports which have been
already connected to the sequencer. Once after that, the creation and deletion
of ports are watched by announcement mechanism of ALSA sequencer.
The available synth and MIDI devices can be found in proc interface.
Run ``cat /proc/asound/seq/oss``, and check the devices. For example,
if you use an AWE64 card, you'll see like the following:
::
OSS sequencer emulation version 0.1.8
ALSA client number 63
ALSA receiver port 0
Number of applications: 0
Number of synth devices: 1
synth 0: [EMU8000]
type 0x1 : subtype 0x20 : voices 32
capabilties : ioctl enabled / load_patch enabled
Number of MIDI devices: 3
midi 0: [Emu8000 Port-0] ALSA port 65:0
capability write / opened none
midi 1: [Emu8000 Port-1] ALSA port 65:1
capability write / opened none
midi 2: [0: MPU-401 (UART)] ALSA port 64:0
capability read/write / opened none
Note that the device number may be different from the information of
``/proc/asound/oss-devices`` or ones of the original OSS driver.
Use the device number listed in ``/proc/asound/seq/oss``
to play via OSS sequencer emulation.
Using Synthesizer Devices
=========================
Run your favorite program. I've tested playmidi-2.4, awemidi-0.4.3, gmod-3.1
and xmp-1.1.5. You can load samples via ``/dev/sequencer`` like sfxload,
too.
If the lowlevel driver supports multiple access to synth devices (like
Emu8000 driver), two or more applications are allowed to run at the same
time.
Using MIDI Devices
==================
So far, only MIDI output was tested. MIDI input was not checked at all,
but hopefully it will work. Use the device number listed in
``/proc/asound/seq/oss``.
Be aware that these numbers are mostly different from the list in
``/proc/asound/oss-devices``.
Module Options
==============
The following module options are available:
maxqlen
specifies the maximum read/write queue length. This queue is private
for OSS sequencer, so that it is independent from the queue length of ALSA
sequencer. Default value is 1024.
seq_oss_debug
specifies the debug level and accepts zero (= no debug message) or
positive integer. Default value is 0.
Queue Mechanism
===============
OSS sequencer emulation uses an ALSA priority queue. The
events from ``/dev/sequencer`` are processed and put onto the queue
specified by module option.
All the events from ``/dev/sequencer`` are parsed at beginning.
The timing events are also parsed at this moment, so that the events may
be processed in real-time. Sending an event ABSTIME 0 switches the operation
mode to real-time mode, and sending an event RELTIME 0 switches it off.
In the real-time mode, all events are dispatched immediately.
The queued events are dispatched to the corresponding ALSA sequencer
ports after scheduled time by ALSA sequencer dispatcher.
If the write-queue is full, the application sleeps until a certain amount
(as default one half) becomes empty in blocking mode. The synchronization
to write timing was implemented, too.
The input from MIDI devices or echo-back events are stored on read FIFO
queue. If application reads ``/dev/sequencer`` in blocking mode, the
process will be awaked.
Interface to Synthesizer Device
===============================
Registration
------------
To register an OSS synthesizer device, use snd_seq_oss_synth_register()
function:
::
int snd_seq_oss_synth_register(char *name, int type, int subtype, int nvoices,
snd_seq_oss_callback_t *oper, void *private_data)
The arguments ``name``, ``type``, ``subtype`` and ``nvoices``
are used for making the appropriate synth_info structure for ioctl. The
return value is an index number of this device. This index must be remembered
for unregister. If registration is failed, -errno will be returned.
To release this device, call snd_seq_oss_synth_unregister() function:
::
int snd_seq_oss_synth_unregister(int index)
where the ``index`` is the index number returned by register function.
Callbacks
---------
OSS synthesizer devices have capability for sample downloading and ioctls
like sample reset. In OSS emulation, these special features are realized
by using callbacks. The registration argument oper is used to specify these
callbacks. The following callback functions must be defined:
::
snd_seq_oss_callback_t:
int (*open)(snd_seq_oss_arg_t *p, void *closure);
int (*close)(snd_seq_oss_arg_t *p);
int (*ioctl)(snd_seq_oss_arg_t *p, unsigned int cmd, unsigned long arg);
int (*load_patch)(snd_seq_oss_arg_t *p, int format, const char *buf, int offs, int count);
int (*reset)(snd_seq_oss_arg_t *p);
Except for ``open`` and ``close`` callbacks, they are allowed to be NULL.
Each callback function takes the argument type ``snd_seq_oss_arg_t`` as the
first argument.
::
struct snd_seq_oss_arg_t {
int app_index;
int file_mode;
int seq_mode;
snd_seq_addr_t addr;
void *private_data;
int event_passing;
};
The first three fields, ``app_index``, ``file_mode`` and ``seq_mode``
are initialized by OSS sequencer. The ``app_index`` is the application
index which is unique to each application opening OSS sequencer. The
``file_mode`` is bit-flags indicating the file operation mode. See
``seq_oss.h`` for its meaning. The ``seq_mode`` is sequencer operation
mode. In the current version, only ``SND_OSSSEQ_MODE_SYNTH`` is used.
The next two fields, ``addr`` and ``private_data``, must be
filled by the synth driver at open callback. The ``addr`` contains
the address of ALSA sequencer port which is assigned to this device. If
the driver allocates memory for ``private_data``, it must be released
in close callback by itself.
The last field, ``event_passing``, indicates how to translate note-on
/ off events. In ``PROCESS_EVENTS`` mode, the note 255 is regarded
as velocity change, and key pressure event is passed to the port. In
``PASS_EVENTS`` mode, all note on/off events are passed to the port
without modified. ``PROCESS_KEYPRESS`` mode checks the note above 128
and regards it as key pressure event (mainly for Emu8000 driver).
Open Callback
-------------
The ``open`` is called at each time this device is opened by an application
using OSS sequencer. This must not be NULL. Typically, the open callback
does the following procedure:
#. Allocate private data record.
#. Create an ALSA sequencer port.
#. Set the new port address on ``arg->addr``.
#. Set the private data record pointer on ``arg->private_data``.
Note that the type bit-flags in port_info of this synth port must NOT contain
``TYPE_MIDI_GENERIC``
bit. Instead, ``TYPE_SPECIFIC`` should be used. Also, ``CAP_SUBSCRIPTION``
bit should NOT be included, too. This is necessary to tell it from other
normal MIDI devices. If the open procedure succeeded, return zero. Otherwise,
return -errno.
Ioctl Callback
--------------
The ``ioctl`` callback is called when the sequencer receives device-specific
ioctls. The following two ioctls should be processed by this callback:
IOCTL_SEQ_RESET_SAMPLES
reset all samples on memory -- return 0
IOCTL_SYNTH_MEMAVL
return the available memory size
FM_4OP_ENABLE
can be ignored usually
The other ioctls are processed inside the sequencer without passing to
the lowlevel driver.
Load_Patch Callback
-------------------
The ``load_patch`` callback is used for sample-downloading. This callback
must read the data on user-space and transfer to each device. Return 0
if succeeded, and -errno if failed. The format argument is the patch key
in patch_info record. The buf is user-space pointer where patch_info record
is stored. The offs can be ignored. The count is total data size of this
sample data.
Close Callback
--------------
The ``close`` callback is called when this device is closed by the
application. If any private data was allocated in open callback, it must
be released in the close callback. The deletion of ALSA port should be
done here, too. This callback must not be NULL.
Reset Callback
--------------
The ``reset`` callback is called when sequencer device is reset or
closed by applications. The callback should turn off the sounds on the
relevant port immediately, and initialize the status of the port. If this
callback is undefined, OSS seq sends a ``HEARTBEAT`` event to the
port.
Events
======
Most of the events are processed by sequencer and translated to the adequate
ALSA sequencer events, so that each synth device can receive by input_event
callback of ALSA sequencer port. The following ALSA events should be
implemented by the driver:
============= ===================
ALSA event Original OSS events
============= ===================
NOTEON SEQ_NOTEON, MIDI_NOTEON
NOTE SEQ_NOTEOFF, MIDI_NOTEOFF
KEYPRESS MIDI_KEY_PRESSURE
CHANPRESS SEQ_AFTERTOUCH, MIDI_CHN_PRESSURE
PGMCHANGE SEQ_PGMCHANGE, MIDI_PGM_CHANGE
PITCHBEND SEQ_CONTROLLER(CTRL_PITCH_BENDER),
MIDI_PITCH_BEND
CONTROLLER MIDI_CTL_CHANGE,
SEQ_BALANCE (with CTL_PAN)
CONTROL14 SEQ_CONTROLLER
REGPARAM SEQ_CONTROLLER(CTRL_PITCH_BENDER_RANGE)
SYSEX SEQ_SYSEX
============= ===================
The most of these behavior can be realized by MIDI emulation driver
included in the Emu8000 lowlevel driver. In the future release, this module
will be independent.
Some OSS events (``SEQ_PRIVATE`` and ``SEQ_VOLUME`` events) are passed as event
type SND_SEQ_OSS_PRIVATE. The OSS sequencer passes these event 8 byte
packets without any modification. The lowlevel driver should process these
events appropriately.
Interface to MIDI Device
========================
Since the OSS emulation probes the creation and deletion of ALSA MIDI
sequencer ports automatically by receiving announcement from ALSA
sequencer, the MIDI devices don't need to be registered explicitly
like synth devices.
However, the MIDI port_info registered to ALSA sequencer must include
a group name ``SND_SEQ_GROUP_DEVICE`` and a capability-bit
``CAP_READ`` or ``CAP_WRITE``. Also, subscription capabilities,
``CAP_SUBS_READ`` or ``CAP_SUBS_WRITE``, must be defined, too. If
these conditions are not satisfied, the port is not registered as OSS
sequencer MIDI device.
The events via MIDI devices are parsed in OSS sequencer and converted
to the corresponding ALSA sequencer events. The input from MIDI sequencer
is also converted to MIDI byte events by OSS sequencer. This works just
a reverse way of seq_midi module.
Known Problems / TODO's
=======================
* Patch loading via ALSA instrument layer is not implemented yet.

143
Documentation/sound/alsa/timestamping.txt → Documentation/sound/designs/timestamping.rst
View File

@ -1,18 +1,22 @@
=====================
ALSA PCM Timestamping
=====================
The ALSA API can provide two different system timestamps:
- Trigger_tstamp is the system time snapshot taken when the .trigger
callback is invoked. This snapshot is taken by the ALSA core in the
general case, but specific hardware may have synchronization
capabilities or conversely may only be able to provide a correct
estimate with a delay. In the latter two cases, the low-level driver
is responsible for updating the trigger_tstamp at the most appropriate
and precise moment. Applications should not rely solely on the first
trigger_tstamp but update their internal calculations if the driver
provides a refined estimate with a delay.
callback is invoked. This snapshot is taken by the ALSA core in the
general case, but specific hardware may have synchronization
capabilities or conversely may only be able to provide a correct
estimate with a delay. In the latter two cases, the low-level driver
is responsible for updating the trigger_tstamp at the most appropriate
and precise moment. Applications should not rely solely on the first
trigger_tstamp but update their internal calculations if the driver
provides a refined estimate with a delay.
- tstamp is the current system timestamp updated during the last
event or application query.
The difference (tstamp - trigger_tstamp) defines the elapsed time.
event or application query.
The difference (tstamp - trigger_tstamp) defines the elapsed time.
The ALSA API provides two basic pieces of information, avail
and delay, which combined with the trigger and current system
@ -22,15 +26,15 @@ the ring buffer and the amount of queued samples.
The use of these different pointers and time information depends on
the application needs:
- 'avail' reports how much can be written in the ring buffer
- 'delay' reports the time it will take to hear a new sample after all
queued samples have been played out.
- ``avail`` reports how much can be written in the ring buffer
- ``delay`` reports the time it will take to hear a new sample after all
queued samples have been played out.
When timestamps are enabled, the avail/delay information is reported
along with a snapshot of system time. Applications can select from
CLOCK_REALTIME (NTP corrections including going backwards),
CLOCK_MONOTONIC (NTP corrections but never going backwards),
CLOCK_MONOTIC_RAW (without NTP corrections) and change the mode
``CLOCK_REALTIME`` (NTP corrections including going backwards),
``CLOCK_MONOTONIC`` (NTP corrections but never going backwards),
``CLOCK_MONOTIC_RAW`` (without NTP corrections) and change the mode
dynamically with sw_params
@ -38,17 +42,18 @@ The ALSA API also provide an audio_tstamp which reflects the passage
of time as measured by different components of audio hardware. In
ascii-art, this could be represented as follows (for the playback
case):
::
--------------------------------------------------------------> time
^ ^ ^ ^ ^
| | | | |
analog link dma app FullBuffer
time time time time time
| | | | |
|< codec delay >|<--hw delay-->|<queued samples>|<---avail->|
|<----------------- delay---------------------->| |
|<----ring buffer length---->|
--------------------------------------------------------------> time
^ ^ ^ ^ ^
| | | | |
analog link dma app FullBuffer
time time time time time
| | | | |
|< codec delay >|<--hw delay-->|<queued samples>|<---avail->|
|<----------------- delay---------------------->| |
|<----ring buffer length---->|
The analog time is taken at the last stage of the playback, as close
as possible to the actual transducer
@ -113,11 +118,11 @@ audio applications...
Due to the varied nature of timestamping needs, even for a single
application, the audio_tstamp_config can be changed dynamically. In
the STATUS ioctl, the parameters are read-only and do not allow for
the ``STATUS`` ioctl, the parameters are read-only and do not allow for
any application selection. To work around this limitation without
impacting legacy applications, a new STATUS_EXT ioctl is introduced
impacting legacy applications, a new ``STATUS_EXT`` ioctl is introduced
with read/write parameters. ALSA-lib will be modified to make use of
STATUS_EXT and effectively deprecate STATUS.
``STATUS_EXT`` and effectively deprecate ``STATUS``.
The ALSA API only allows for a single audio timestamp to be reported
at a time. This is a conscious design decision, reading the audio
@ -135,36 +140,42 @@ the hardware, there is a risk of misalignment with the avail and delay
information. To make sure applications are not confused, a
driver_timestamp field is added in the snd_pcm_status structure; this
timestamp shows when the information is put together by the driver
before returning from the STATUS and STATUS_EXT ioctl. in most cases
before returning from the ``STATUS`` and ``STATUS_EXT`` ioctl. in most cases
this driver_timestamp will be identical to the regular system tstamp.
Examples of typestamping with HDaudio:
1. DMA timestamp, no compensation for DMA+analog delay
$ ./audio_time -p --ts_type=1
playback: systime: 341121338 nsec, audio time 342000000 nsec, systime delta -878662
playback: systime: 426236663 nsec, audio time 427187500 nsec, systime delta -950837
playback: systime: 597080580 nsec, audio time 598000000 nsec, systime delta -919420
playback: systime: 682059782 nsec, audio time 683020833 nsec, systime delta -961051
playback: systime: 852896415 nsec, audio time 853854166 nsec, systime delta -957751
playback: systime: 937903344 nsec, audio time 938854166 nsec, systime delta -950822
::
$ ./audio_time -p --ts_type=1
playback: systime: 341121338 nsec, audio time 342000000 nsec, systime delta -878662
playback: systime: 426236663 nsec, audio time 427187500 nsec, systime delta -950837
playback: systime: 597080580 nsec, audio time 598000000 nsec, systime delta -919420
playback: systime: 682059782 nsec, audio time 683020833 nsec, systime delta -961051
playback: systime: 852896415 nsec, audio time 853854166 nsec, systime delta -957751
playback: systime: 937903344 nsec, audio time 938854166 nsec, systime delta -950822
2. DMA timestamp, compensation for DMA+analog delay
$ ./audio_time -p --ts_type=1 -d
playback: systime: 341053347 nsec, audio time 341062500 nsec, systime delta -9153
playback: systime: 426072447 nsec, audio time 426062500 nsec, systime delta 9947
playback: systime: 596899518 nsec, audio time 596895833 nsec, systime delta 3685
playback: systime: 681915317 nsec, audio time 681916666 nsec, systime delta -1349
playback: systime: 852741306 nsec, audio time 852750000 nsec, systime delta -8694
::
$ ./audio_time -p --ts_type=1 -d
playback: systime: 341053347 nsec, audio time 341062500 nsec, systime delta -9153
playback: systime: 426072447 nsec, audio time 426062500 nsec, systime delta 9947
playback: systime: 596899518 nsec, audio time 596895833 nsec, systime delta 3685
playback: systime: 681915317 nsec, audio time 681916666 nsec, systime delta -1349
playback: systime: 852741306 nsec, audio time 852750000 nsec, systime delta -8694
3. link timestamp, compensation for DMA+analog delay
$ ./audio_time -p --ts_type=2 -d
playback: systime: 341060004 nsec, audio time 341062791 nsec, systime delta -2787
playback: systime: 426242074 nsec, audio time 426244875 nsec, systime delta -2801
playback: systime: 597080992 nsec, audio time 597084583 nsec, systime delta -3591
playback: systime: 682084512 nsec, audio time 682088291 nsec, systime delta -3779
playback: systime: 852936229 nsec, audio time 852940916 nsec, systime delta -4687
playback: systime: 938107562 nsec, audio time 938112708 nsec, systime delta -5146
::
$ ./audio_time -p --ts_type=2 -d
playback: systime: 341060004 nsec, audio time 341062791 nsec, systime delta -2787
playback: systime: 426242074 nsec, audio time 426244875 nsec, systime delta -2801
playback: systime: 597080992 nsec, audio time 597084583 nsec, systime delta -3591
playback: systime: 682084512 nsec, audio time 682088291 nsec, systime delta -3779
playback: systime: 852936229 nsec, audio time 852940916 nsec, systime delta -4687
playback: systime: 938107562 nsec, audio time 938112708 nsec, systime delta -5146
Example 1 shows that the timestamp at the DMA level is close to 1ms
ahead of the actual playback time (as a side time this sort of
@ -181,20 +192,24 @@ shows how compensating for the delay exposes a 1ms accuracy (due to
the use of the frame counter by the driver)
Example 3: DMA timestamp, no compensation for delay, delta of ~5ms
$ ./audio_time -p -Dhw:1 -t1
playback: systime: 120174019 nsec, audio time 125000000 nsec, systime delta -4825981
playback: systime: 245041136 nsec, audio time 250000000 nsec, systime delta -4958864
playback: systime: 370106088 nsec, audio time 375000000 nsec, systime delta -4893912
playback: systime: 495040065 nsec, audio time 500000000 nsec, systime delta -4959935
playback: systime: 620038179 nsec, audio time 625000000 nsec, systime delta -4961821
playback: systime: 745087741 nsec, audio time 750000000 nsec, systime delta -4912259
playback: systime: 870037336 nsec, audio time 875000000 nsec, systime delta -4962664
::
$ ./audio_time -p -Dhw:1 -t1
playback: systime: 120174019 nsec, audio time 125000000 nsec, systime delta -4825981
playback: systime: 245041136 nsec, audio time 250000000 nsec, systime delta -4958864
playback: systime: 370106088 nsec, audio time 375000000 nsec, systime delta -4893912
playback: systime: 495040065 nsec, audio time 500000000 nsec, systime delta -4959935
playback: systime: 620038179 nsec, audio time 625000000 nsec, systime delta -4961821
playback: systime: 745087741 nsec, audio time 750000000 nsec, systime delta -4912259
playback: systime: 870037336 nsec, audio time 875000000 nsec, systime delta -4962664
Example 4: DMA timestamp, compensation for delay, delay of ~1ms
$ ./audio_time -p -Dhw:1 -t1 -d
playback: systime: 120190520 nsec, audio time 120000000 nsec, systime delta 190520
playback: systime: 245036740 nsec, audio time 244000000 nsec, systime delta 1036740
playback: systime: 370034081 nsec, audio time 369000000 nsec, systime delta 1034081
playback: systime: 495159907 nsec, audio time 494000000 nsec, systime delta 1159907
playback: systime: 620098824 nsec, audio time 619000000 nsec, systime delta 1098824
playback: systime: 745031847 nsec, audio time 744000000 nsec, systime delta 1031847
::
$ ./audio_time -p -Dhw:1 -t1 -d
playback: systime: 120190520 nsec, audio time 120000000 nsec, systime delta 190520
playback: systime: 245036740 nsec, audio time 244000000 nsec, systime delta 1036740
playback: systime: 370034081 nsec, audio time 369000000 nsec, systime delta 1034081
playback: systime: 495159907 nsec, audio time 494000000 nsec, systime delta 1159907
playback: systime: 620098824 nsec, audio time 619000000 nsec, systime delta 1098824
playback: systime: 745031847 nsec, audio time 744000000 nsec, systime delta 1031847

33
Documentation/sound/alsa/HD-Audio-Controls.txt → Documentation/sound/hd-audio/controls.rst
View File

@ -1,16 +1,21 @@
======================================
HD-Audio Codec-Specific Mixer Controls
======================================
This file explains the codec-specific mixer controls.
Realtek codecs
--------------
* Channel Mode
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
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
@ -30,24 +35,24 @@ Realtek codecs
IDT/Sigmatel codecs
-------------------
* Analog Loopback
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
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
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
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"
@ -57,19 +62,19 @@ IDT/Sigmatel codecs
VIA codecs
----------
* Smart 5.1
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
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
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
@ -78,7 +83,7 @@ VIA codecs
headphones and speakers because there is only one DAC connected to a
mixer widget.
* Dynamic Power-Control
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
@ -86,7 +91,7 @@ VIA codecs
doesn't provide a proper jack-detection, this won't work; in such a
case, turn this control OFF.
* Jack Detect
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
@ -96,21 +101,21 @@ VIA codecs
Conexant codecs
---------------
* Auto-Mute Mode
Auto-Mute Mode
See Reatek codecs.
Analog codecs
--------------
* Channel Mode
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
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.

30
Documentation/sound/alsa/HD-Audio-DP-MST-audio.txt → Documentation/sound/hd-audio/dp-mst.rst
View File

@ -1,3 +1,7 @@
=======================
HD-Audio DP-MST Support
=======================
To support DP MST audio, HD Audio hdmi codec driver introduces virtual pin
and dynamic pcm assignment.
@ -44,10 +48,12 @@ Build Jack
----------
- dyn_pcm_assign
Will not use hda_jack but use snd_jack in spec->pcm_rec[pcm_idx].jack directly.
Will not use hda_jack but use snd_jack in spec->pcm_rec[pcm_idx].jack directly.
- !dyn_pcm_assign
Use hda_jack and assign spec->pcm_rec[pcm_idx].jack = jack->jack statically.
Use hda_jack and assign spec->pcm_rec[pcm_idx].jack = jack->jack statically.
Unsolicited Event Enabling
@ -58,16 +64,20 @@ Enable unsolicited event if !acomp.
Monitor Hotplug Event Handling
------------------------------
- acomp
pin_eld_notify() -> check_presence_and_report() -> hdmi_present_sense() ->
sync_eld_via_acomp().
Use directly snd_jack_report() on spec->pcm_rec[pcm_idx].jack for
both dyn_pcm_assign and !dyn_pcm_assign
pin_eld_notify() -> check_presence_and_report() -> hdmi_present_sense() ->
sync_eld_via_acomp().
Use directly snd_jack_report() on spec->pcm_rec[pcm_idx].jack for
both dyn_pcm_assign and !dyn_pcm_assign
- !acomp
Hdmi_unsol_event() -> hdmi_intrinsic_event() -> check_presence_and_report() ->
hdmi_present_sense() -> hdmi_prepsent_sense_via_verbs()
Use directly snd_jack_report() on spec->pcm_rec[pcm_idx].jack for dyn_pcm_assign.
Use hda_jack mechanism to handle jack events.
hdmi_unsol_event() -> hdmi_intrinsic_event() -> check_presence_and_report() ->
hdmi_present_sense() -> hdmi_prepsent_sense_via_verbs()
Use directly snd_jack_report() on spec->pcm_rec[pcm_idx].jack for dyn_pcm_assign.
Use hda_jack mechanism to handle jack events.
Others to be added later

10
Documentation/sound/hd-audio/index.rst Normal file
View File

@ -0,0 +1,10 @@
HD-Audio
========
.. toctree::
:maxdepth: 2
notes
models
controls
dp-mst

518
Documentation/sound/hd-audio/models.rst Normal file
View File

@ -0,0 +1,518 @@
==============================
HD-Audio Codec-Specific Models
==============================
ALC880
======
3stack
3-jack in back and a headphone out
3stack-digout
3-jack in back, a HP out and a SPDIF out
5stack
5-jack in back, 2-jack in front
5stack-digout
5-jack in back, 2-jack in front, a SPDIF out
6stack
6-jack in back, 2-jack in front
6stack-digout
6-jack with a SPDIF out
ALC260
======
gpio1
Enable GPIO1
coef
Enable EAPD via COEF table
fujitsu
Quirk for FSC S7020
fujitsu-jwse
Quirk for FSC S7020 with jack modes and HP mic support
ALC262
======
inv-dmic
Inverted internal mic workaround
ALC267/268
==========
inv-dmic
Inverted internal mic workaround
hp-eapd
Disable HP EAPD on NID 0x15
ALC22x/23x/25x/269/27x/28x/29x (and vendor-specific ALC3xxx models)
===================================================================
laptop-amic
Laptops with analog-mic input
laptop-dmic
Laptops with digital-mic input
alc269-dmic
Enable ALC269(VA) digital mic workaround
alc271-dmic
Enable ALC271X digital mic workaround
inv-dmic
Inverted internal mic workaround
headset-mic
Indicates a combined headset (headphone+mic) jack
headset-mode
More comprehensive headset support for ALC269 & co
headset-mode-no-hp-mic
Headset mode support without headphone mic
lenovo-dock
Enables docking station I/O for some Lenovos
hp-gpio-led
GPIO LED support on HP laptops
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
alc283-dac-wcaps
Fixups for Chromebook with ALC283
alc283-sense-combo
Combo jack sensing on ALC283
tpt440-dock
Pin configs for Lenovo Thinkpad Dock support
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
=======
3stack
3-stack, shared surrounds
laptop
2-channel only (FSC V2060, Samsung M50)
laptop-imic
2-channel with built-in mic
eapd
Turn on EAPD constantly
AD1988/AD1988B/AD1989A/AD1989B
==============================
6stack
6-jack
6stack-dig
ditto with SPDIF
3stack
3-jack
3stack-dig
ditto with SPDIF
laptop
3-jack with hp-jack automute
laptop-dig
ditto with SPDIF
auto
auto-config reading BIOS (default)
Conexant 5045
=============
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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@ -0,0 +1,20 @@
===================================
Linux Sound Subsystem Documentation
===================================
.. toctree::
:maxdepth: 2
kernel-api/index
designs/index
soc/index
alsa-configuration
hd-audio/index
cards/index
.. only:: subproject
Indices
=======
* :ref:`genindex`

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@ -0,0 +1,134 @@
===================
The ALSA Driver API
===================
Management of Cards and Devices
===============================
Card Management
---------------
.. kernel-doc:: sound/core/init.c
Device Components
-----------------
.. kernel-doc:: sound/core/device.c
Module requests and Device File Entries
---------------------------------------
.. kernel-doc:: sound/core/sound.c
Memory Management Helpers
-------------------------
.. kernel-doc:: sound/core/memory.c
.. kernel-doc:: sound/core/memalloc.c
PCM API
=======
PCM Core
--------
.. kernel-doc:: sound/core/pcm.c
.. kernel-doc:: sound/core/pcm_lib.c
.. kernel-doc:: sound/core/pcm_native.c
.. kernel-doc:: include/sound/pcm.h
PCM Format Helpers
------------------
.. kernel-doc:: sound/core/pcm_misc.c
PCM Memory Management
---------------------
.. kernel-doc:: sound/core/pcm_memory.c
PCM DMA Engine API
------------------
.. kernel-doc:: sound/core/pcm_dmaengine.c
.. kernel-doc:: include/sound/dmaengine_pcm.h
Control/Mixer API
=================
General Control Interface
-------------------------
.. kernel-doc:: sound/core/control.c
AC97 Codec API
--------------
.. kernel-doc:: sound/pci/ac97/ac97_codec.c
.. kernel-doc:: sound/pci/ac97/ac97_pcm.c
Virtual Master Control API
--------------------------
.. kernel-doc:: sound/core/vmaster.c
.. kernel-doc:: include/sound/control.h
MIDI API
========
Raw MIDI API
------------
.. kernel-doc:: sound/core/rawmidi.c
MPU401-UART API
---------------
.. kernel-doc:: sound/drivers/mpu401/mpu401_uart.c
Proc Info API
=============
Proc Info Interface
-------------------
.. kernel-doc:: sound/core/info.c
Compress Offload
================
Compress Offload API
--------------------
.. kernel-doc:: sound/core/compress_offload.c
.. kernel-doc:: include/uapi/sound/compress_offload.h
.. kernel-doc:: include/uapi/sound/compress_params.h
.. kernel-doc:: include/sound/compress_driver.h
ASoC
====
ASoC Core API
-------------
.. kernel-doc:: include/sound/soc.h
.. kernel-doc:: sound/soc/soc-core.c
.. kernel-doc:: sound/soc/soc-devres.c
.. kernel-doc:: sound/soc/soc-io.c
.. kernel-doc:: sound/soc/soc-pcm.c
.. kernel-doc:: sound/soc/soc-ops.c
.. kernel-doc:: sound/soc/soc-compress.c
ASoC DAPM API
-------------
.. kernel-doc:: sound/soc/soc-dapm.c
ASoC DMA Engine API
-------------------
.. kernel-doc:: sound/soc/soc-generic-dmaengine-pcm.c
Miscellaneous Functions
=======================
Hardware-Dependent Devices API
------------------------------
.. kernel-doc:: sound/core/hwdep.c
Jack Abstraction Layer API
--------------------------
.. kernel-doc:: include/sound/jack.h
.. kernel-doc:: sound/core/jack.c
.. kernel-doc:: sound/soc/soc-jack.c
ISA DMA Helpers
---------------
.. kernel-doc:: sound/core/isadma.c
Other Helper Macros
-------------------
.. kernel-doc:: include/sound/core.h

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@ -0,0 +1,8 @@
ALSA Kernel API Documentation
=============================
.. toctree::
:maxdepth: 2
alsa-driver-api
writing-an-alsa-driver

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13
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@ -1,3 +1,4 @@
==============
Audio Clocking
==============
@ -30,15 +31,9 @@ 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
- 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

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@ -0,0 +1,108 @@
==============================================
Creating codec to codec dai link for ALSA dapm
==============================================
Mostly the flow of audio is always from CPU to codec so your system
will look as below:
::
--------- ---------
| | dai | |
CPU -------> codec
| | | |
--------- ---------
In case your system looks as below:
::
---------
| |
codec-2
| |
---------
|
dai-2
|
---------- ---------
| | dai-1 | |
CPU -------> codec-1
| | | |
---------- ---------
|
dai-3
|
---------
| |
codec-3
| |
---------
Suppose codec-2 is a bluetooth chip and codec-3 is connected to
a speaker and you have a below scenario:
codec-2 will receive the audio data and the user wants to play that
audio through codec-3 without involving the CPU.This
aforementioned case is the ideal case when codec to codec
connection should be used.
Your dai_link should appear as below in your machine
file:
::
/*
* this pcm stream only supports 24 bit, 2 channel and
* 48k sampling rate.
*/
static const struct snd_soc_pcm_stream dsp_codec_params = {
.formats = SNDRV_PCM_FMTBIT_S24_LE,
.rate_min = 48000,
.rate_max = 48000,
.channels_min = 2,
.channels_max = 2,
};
{
.name = "CPU-DSP",
.stream_name = "CPU-DSP",
.cpu_dai_name = "samsung-i2s.0",
.codec_name = "codec-2,
.codec_dai_name = "codec-2-dai_name",
.platform_name = "samsung-i2s.0",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBM_CFM,
.ignore_suspend = 1,
.params = &dsp_codec_params,
},
{
.name = "DSP-CODEC",
.stream_name = "DSP-CODEC",
.cpu_dai_name = "wm0010-sdi2",
.codec_name = "codec-3,
.codec_dai_name = "codec-3-dai_name",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBM_CFM,
.ignore_suspend = 1,
.params = &dsp_codec_params,
},
Above code snippet is motivated from sound/soc/samsung/speyside.c.
Note the "params" callback which lets the dapm know that this
dai_link is a codec to codec connection.
In dapm core a route is created between cpu_dai playback widget
and codec_dai capture widget for playback path and vice-versa is
true for capture path. In order for this aforementioned route to get
triggered, DAPM needs to find a valid endpoint which could be either
a sink or source widget corresponding to playback and capture path
respectively.
In order to trigger this dai_link widget, a thin codec driver for
the speaker amp can be created as demonstrated in wm8727.c file, it
sets appropriate constraints for the device even if it needs no control.
Make sure to name your corresponding cpu and codec playback and capture
dai names ending with "Playback" and "Capture" respectively as dapm core
will link and power those dais based on the name.
Note that in current device tree there is no way to mark a dai_link
as codec to codec. However, it may change in future.

75
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@ -1,3 +1,4 @@
=======================
ASoC Codec Class Driver
=======================
@ -9,16 +10,16 @@ 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.
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.
7. DAC Digital mute control.
Its probably best to use this guide in conjunction with the existing codec
driver code in sound/soc/codecs/
@ -26,24 +27,25 @@ driver code in sound/soc/codecs/
ASoC Codec driver breakdown
===========================
1 - Codec DAI and PCM configuration
-----------------------------------
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 = {
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 = {
};
struct snd_soc_dai_driver wm8731_dai = {
.name = "wm8731-hifi",
.playback = {
.stream_name = "Playback",
@ -59,25 +61,27 @@ struct snd_soc_dai_driver wm8731_dai = {
.formats = WM8731_FORMATS,},
.ops = &wm8731_dai_ops,
.symmetric_rates = 1,
};
};
2 - Codec control IO
--------------------
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
-----------------------------
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
@ -86,18 +90,22 @@ Defines a single control as follows:-
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
@ -109,25 +117,26 @@ Defines an single enumerated control as follows:-
Defines a stereo enumerated control
4 - Codec Audio Operations
--------------------------
Codec Audio Operations
----------------------
The codec driver also supports the following ALSA PCM operations:-
::
/* SoC audio ops */
struct snd_soc_ops {
/* 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.
---------------------
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.
@ -135,13 +144,14 @@ Please read dapm.txt for details of building the description.
Please also see the examples in other codec drivers.
6 - DAPM event handler
----------------------
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 */
@ -155,8 +165,8 @@ Power states:-
SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */
7 - Codec DAC digital mute control
----------------------------------
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.
@ -165,9 +175,10 @@ 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)
{
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;
@ -176,4 +187,4 @@ static int wm8974_mute(struct snd_soc_dai *dai, int mute)
else
snd_soc_write(codec, WM8974_DAC, mute_reg);
return 0;
}
}

28
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@ -1,3 +1,7 @@
==================================
ASoC Digital Audio Interface (DAI)
==================================
ASoC currently supports the three main Digital Audio Interfaces (DAI) found on
SoC controllers and portable audio CODECs today, namely AC97, I2S and PCM.
@ -5,21 +9,21 @@ 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
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 :-
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
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
@ -30,13 +34,15 @@ 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.
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.
Left Justified
MSB is transmitted on transition of LRC.
o Right Justified - MSB is transmitted sample size BCLKs before LRC
transition.
Right Justified
MSB is transmitted sample size BCLKs before LRC transition.
PCM
===
@ -51,6 +57,8 @@ 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.
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.
Mode B
MSB is transmitted on rising edge of FRAME/SYNC.

247
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@ -1,8 +1,9 @@
===================================================
Dynamic Audio Power Management for Portable Devices
===================================================
1. Description
==============
Description
===========
Dynamic Audio Power Management (DAPM) is designed to allow portable
Linux devices to use the minimum amount of power within the audio
@ -21,20 +22,28 @@ level power systems.
There are 4 power domains within DAPM
1. Codec bias domain - VREF, VMID (core codec and audio power)
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
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
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.
Stream domain
DACs and ADCs.
Enabled and disabled when stream playback/capture is started and
stopped respectively. e.g. aplay, arecord.
@ -45,34 +54,57 @@ internal codec components). All audio components that effect power are called
widgets hereafter.
2. DAPM Widgets
===============
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)
Mixer
Mixes several analog signals into a single analog signal.
Mux
An analog switch that outputs only one of many inputs.
PGA
A programmable gain amplifier or attenuation widget.
ADC
Analog to Digital Converter
DAC
Digital to Analog Converter
Switch
An analog switch
Input
A codec input pin
Output
A codec output pin
Headphone
Headphone (and optional Jack)
Mic
Mic (and optional Jack)
Line
Line Input/Output (and optional Jack)
Speaker
Speaker
Supply
Power or clock supply widget used by other widgets.
Regulator
External regulator that supplies power to audio components.
Clock
External clock that supplies clock to audio components.
AIF IN
Audio Interface Input (with TDM slot mask).
AIF OUT
Audio Interface Output (with TDM slot mask).
Siggen
Signal Generator.
DAI IN
Digital Audio Interface Input.
DAI OUT
Digital Audio Interface Output.
DAI Link
DAI Link between two DAI structures
Pre
Special PRE widget (exec before all others)
Post
Special POST widget (exec after all others)
(Widgets are defined in include/sound/soc-dapm.h)
@ -84,52 +116,57 @@ 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 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)
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),
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),
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
-------------------
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)
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),
};
/* 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,
SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, wm8731_output_mixer_controls,
ARRAY_SIZE(wm8731_output_mixer_controls)),
If you don't want the mixer elements prefixed with the name of the mixer widget,
@ -137,48 +174,49 @@ 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 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
* Speaker Amp
* Microphone Bias
* Jack connectors
A machine widget can have an optional call back.
e.g. Jack connector widget for an external Mic that enables Mic Bias
when the Mic is inserted:-
when the Mic is inserted:-::
static int spitz_mic_bias(struct snd_soc_dapm_widget* w, int event)
{
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),
SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias),
2.4 Codec (BIAS) Domain
-----------------------
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
-------------------
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),
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.
@ -186,8 +224,8 @@ 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
====================================
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
@ -201,13 +239,14 @@ 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
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"},
@ -216,22 +255,17 @@ connect the destination widget (wrt audio signal) with its source widgets.
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".
* 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);
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
@ -239,12 +273,13 @@ 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
-------------------------------
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"},
@ -254,52 +289,54 @@ 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
===================
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
* Headphone Jack
* Internal Speaker
* Internal Mic
* Mic Jack
* Codec Pins
Endpoints are added to the DAPM graph so that their usage can be determined in
order to save power. e.g. NC codecs pins will be switched OFF, unconnected
jacks can also be switched OFF.
5 DAPM Widget Events
====================
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)
{
/* 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 =
/* 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
---------------
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 */
/* 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 */

270
Documentation/sound/alsa/soc/DPCM.txt → Documentation/sound/soc/dpcm.rst
View File

@ -1,8 +1,9 @@
===========
Dynamic PCM
===========
1. Description
==============
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
@ -23,22 +24,23 @@ 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
*************
| 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
@ -55,50 +57,52 @@ 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
*************
*************
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
*************
*************
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.
1. Machine driver receives Jack removal event.
2) Machine driver OR audio HAL disables the Headset path.
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.
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.
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.
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
@ -112,36 +116,38 @@ 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.
1. Define the FE and BE DAI links.
2) Define any FE/BE PCM operations.
2. Define any FE/BE PCM operations.
3) Define widget graph connections.
3. Define widget graph connections.
1 FE and BE DAI links
---------------------
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
*************
| 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[] = {
static struct snd_soc_dai_link machine_dais[] = {
{
.name = "PCM0 System",
.stream_name = "System Playback",
@ -154,11 +160,11 @@ static struct snd_soc_dai_link machine_dais[] = {
.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"
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
@ -168,8 +174,9 @@ 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[] = {
static struct snd_soc_dai_link machine_dais[] = {
.....< FE DAI links here >
{
.name = "Codec Headset",
@ -186,29 +193,30 @@ static struct snd_soc_dai_link machine_dais[] = {
.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 ``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
*************
*************
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.
@ -220,10 +228,10 @@ 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
----------------------
FE/BE PCM operations
--------------------
The BE above also exports some PCM operations and a "fixup" callback. The fixup
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.
@ -231,10 +239,11 @@ e.g. DSP converts all FE hw params to run at fixed rate of 48k, 16bit, stereo fo
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,
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,
@ -249,21 +258,22 @@ static int dai0_fixup(struct snd_soc_pcm_runtime *rtd,
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
--------------------------
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"},
/* 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
@ -273,24 +283,25 @@ 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.
1. Front End PCM DAIs - i.e. struct snd_soc_dai_driver.
2) DAPM graph showing DSP audio routing from FE DAIs to BEs.
2. DAPM graph showing DSP audio routing from FE DAIs to BEs.
3) DAPM widgets from DSP graph.
3. DAPM widgets from DSP graph.
4) Mixers for gains, routing, etc.
4. Mixers for gains, routing, etc.
5) DMA configuration.
5. DMA configuration.
6) BE AIF widgets.
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),
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).
@ -301,33 +312,33 @@ 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
*************
*************
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
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
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
@ -339,16 +350,17 @@ 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 = {
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[] = {
static struct snd_soc_dai_link dais[] = {
< ... more DAI links above ... >
{
.name = "MODEM",

20
Documentation/sound/soc/index.rst Normal file
View File

@ -0,0 +1,20 @@
==============
ALSA SoC Layer
==============
The documentation is spilt into the following sections:-
.. toctree::
:maxdepth: 2
overview
codec
dai
dapm
platform
machine
pops-clicks
clocking
jack
dpcm
codec-to-codec

1
Documentation/sound/alsa/soc/jack.txt → Documentation/sound/soc/jack.rst
View File

@ -1,3 +1,4 @@
===================
ASoC jack detection
===================

22
Documentation/sound/alsa/soc/machine.txt → Documentation/sound/soc/machine.rst
View File

@ -1,3 +1,4 @@
===================
ASoC Machine Driver
===================
@ -9,9 +10,10 @@ 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 {
/* SoC machine */
struct snd_soc_card {
char *name;
...
@ -33,7 +35,7 @@ struct snd_soc_card {
int num_links;
...
};
};
probe()/remove()
----------------
@ -55,9 +57,10 @@ 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 = {
/* 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",
@ -66,16 +69,17 @@ static struct snd_soc_dai_link corgi_dai = {
.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 = {
/* corgi audio machine driver */
static struct snd_soc_card snd_soc_corgi = {
.name = "Corgi",
.dai_link = &corgi_dai,
.num_links = 1,
};
};
Machine Power Map

32
Documentation/sound/alsa/soc/overview.txt → Documentation/sound/soc/overview.rst
View File

@ -1,5 +1,6 @@
ALSA SoC Layer
==============
=======================
ALSA SoC Layer Overview
=======================
The overall project goal of the ALSA System on Chip (ASoC) layer is to
provide better ALSA support for embedded system-on-chip processors (e.g.
@ -66,30 +67,3 @@ multiple re-usable component drivers :-
describes and binds the other component drivers together to form an ALSA
"sound card device". It handles any machine specific controls and
machine level audio events (e.g. turning on an amp at start of playback).
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.

31
Documentation/sound/alsa/soc/platform.txt → Documentation/sound/soc/platform.rst
View File

@ -1,3 +1,4 @@
====================
ASoC Platform Driver
====================
@ -9,21 +10,23 @@ Audio DMA
=========
The platform DMA driver optionally supports the following ALSA operations:-
::
/* SoC audio ops */
struct snd_soc_ops {
/* 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 {
struct snd_soc_platform_driver {
char *name;
int (*probe)(struct platform_device *pdev);
@ -44,7 +47,7 @@ struct snd_soc_platform_driver {
/* 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/
@ -57,11 +60,11 @@ 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)
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.
@ -71,9 +74,9 @@ 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.
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.

15
Documentation/sound/alsa/soc/pops_clicks.txt → Documentation/sound/soc/pops-clicks.rst
View File

@ -1,3 +1,4 @@
=====================
Audio Pops and Clicks
=====================
@ -20,10 +21,11 @@ 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
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.
@ -36,10 +38,11 @@ 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
Startup Order - Input PGA --> Mixers --> ADC
Shutdown Order - ADC --> Mixers --> Input PGA
Zipper Noise