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提交 a04dae9e 编写于 作者: T Takashi Iwai
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Merge branch 'topic/restize-docs' into for-next

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Documentation/DocBook/Makefile
浏览文件 @ a04dae9e
...@@ -12,8 +12,7 @@ DOCBOOKS := z8530book.xml \ ...@@ -12,8 +12,7 @@ DOCBOOKS := z8530book.xml \
kernel-api.xml filesystems.xml lsm.xml usb.xml kgdb.xml \ kernel-api.xml filesystems.xml lsm.xml usb.xml kgdb.xml \
gadget.xml libata.xml mtdnand.xml librs.xml rapidio.xml \ gadget.xml libata.xml mtdnand.xml librs.xml rapidio.xml \
genericirq.xml s390-drivers.xml uio-howto.xml scsi.xml \ genericirq.xml s390-drivers.xml uio-howto.xml scsi.xml \
debugobjects.xml sh.xml regulator.xml \ 80211.xml debugobjects.xml sh.xml regulator.xml \
alsa-driver-api.xml writing-an-alsa-driver.xml \
tracepoint.xml w1.xml \ tracepoint.xml w1.xml \
writing_musb_glue_layer.xml crypto-API.xml iio.xml writing_musb_glue_layer.xml crypto-API.xml iio.xml
......
Documentation/DocBook/alsa-driver-api.tmpl 已删除 100644 → 0
浏览文件 @ 4ce8e6a5
<?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>
Documentation/devicetree/bindings/net/marvell-orion-net.txt
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...@@ -49,6 +49,7 @@ Optional port properties: ...@@ -49,6 +49,7 @@ Optional port properties:
and and
- phy-handle: See ethernet.txt file in the same directory. - phy-handle: See ethernet.txt file in the same directory.
- phy-mode: See ethernet.txt file in the same directory.
or or
......
Documentation/index.rst
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...@@ -18,6 +18,7 @@ Contents: ...@@ -18,6 +18,7 @@ Contents:
media/index media/index
gpu/index gpu/index
80211/index 80211/index
sound/index
Indices and tables Indices and tables
================== ==================
......
Documentation/networking/netdev-FAQ.txt
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...@@ -29,8 +29,8 @@ A: There are always two trees (git repositories) in play. Both are driven ...@@ -29,8 +29,8 @@ A: There are always two trees (git repositories) in play. Both are driven
Linus, and net-next is where the new code goes for the future release. Linus, and net-next is where the new code goes for the future release.
You can find the trees here: You can find the trees here:
http://git.kernel.org/?p=linux/kernel/git/davem/net.git https://git.kernel.org/pub/scm/linux/kernel/git/davem/net.git
http://git.kernel.org/?p=linux/kernel/git/davem/net-next.git https://git.kernel.org/pub/scm/linux/kernel/git/davem/net-next.git
Q: How often do changes from these trees make it to the mainline Linus tree? Q: How often do changes from these trees make it to the mainline Linus tree?
...@@ -76,7 +76,7 @@ Q: So where are we now in this cycle? ...@@ -76,7 +76,7 @@ Q: So where are we now in this cycle?
A: Load the mainline (Linus) page here: A: Load the mainline (Linus) page here:
http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git
and note the top of the "tags" section. If it is rc1, it is early and note the top of the "tags" section. If it is rc1, it is early
in the dev cycle. If it was tagged rc7 a week ago, then a release in the dev cycle. If it was tagged rc7 a week ago, then a release
...@@ -123,7 +123,7 @@ A: Normally Greg Kroah-Hartman collects stable commits himself, but ...@@ -123,7 +123,7 @@ A: Normally Greg Kroah-Hartman collects stable commits himself, but
It contains the patches which Dave has selected, but not yet handed It contains the patches which Dave has selected, but not yet handed
off to Greg. If Greg already has the patch, then it will be here: off to Greg. If Greg already has the patch, then it will be here:
http://git.kernel.org/cgit/linux/kernel/git/stable/stable-queue.git https://git.kernel.org/pub/scm/linux/kernel/git/stable/stable-queue.git
A quick way to find whether the patch is in this stable-queue is A quick way to find whether the patch is in this stable-queue is
to simply clone the repo, and then git grep the mainline commit ID, e.g. to simply clone the repo, and then git grep the mainline commit ID, e.g.
......
Documentation/networking/nf_conntrack-sysctl.txt
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...@@ -33,24 +33,6 @@ nf_conntrack_events - BOOLEAN ...@@ -33,24 +33,6 @@ nf_conntrack_events - BOOLEAN
If this option is enabled, the connection tracking code will If this option is enabled, the connection tracking code will
provide userspace with connection tracking events via ctnetlink. provide userspace with connection tracking events via ctnetlink.
nf_conntrack_events_retry_timeout - INTEGER (seconds)
default 15
This option is only relevant when "reliable connection tracking
events" are used. Normally, ctnetlink is "lossy", that is,
events are normally dropped when userspace listeners can't keep up.
Userspace can request "reliable event mode". When this mode is
active, the conntrack will only be destroyed after the event was
delivered. If event delivery fails, the kernel periodically
re-tries to send the event to userspace.
This is the maximum interval the kernel should use when re-trying
to deliver the destroy event.
A higher number means there will be fewer delivery retries and it
will take longer for a backlog to be processed.
nf_conntrack_expect_max - INTEGER nf_conntrack_expect_max - INTEGER
Maximum size of expectation table. Default value is Maximum size of expectation table. Default value is
nf_conntrack_buckets / 256. Minimum is 1. nf_conntrack_buckets / 256. Minimum is 1.
......
Documentation/sound/alsa-configuration.rst 0 → 100644
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此差异已折叠。
Documentation/sound/alsa/ControlNames.txt 已删除 100644 → 0
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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) */
Documentation/sound/alsa/HD-Audio-Models.txt 已删除 100644 → 0
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Model name Description
---------- -----------
ALC880
======
3stack 3-jack in back and a headphone out
3stack-digout 3-jack in back, a HP out and a SPDIF out
5stack 5-jack in back, 2-jack in front
5stack-digout 5-jack in back, 2-jack in front, a SPDIF out
6stack 6-jack in back, 2-jack in front
6stack-digout 6-jack with a SPDIF out
ALC260
======
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)
Documentation/sound/alsa/VIA82xx-mixer.txt 已删除 100644 → 0
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VIA82xx mixer
=============
On many VIA82xx boards, the 'Input Source Select' mixer control does not work.
Setting it to 'Input2' on such boards will cause recording to hang, or fail
with EIO (input/output error) via OSS emulation. This control should be left
at 'Input1' for such cards.
Documentation/sound/alsa/alsa-parameters.txt 已删除 100644 → 0
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ALSA Kernel Parameters
~~~~~~~~~~~~~~~~~~~~~~
See Documentation/kernel-parameters.txt for general information on
specifying module parameters.
This document may not be entirely up to date and comprehensive. The command
"modinfo -p ${modulename}" shows a current list of all parameters of a loadable
module. Loadable modules, after being loaded into the running kernel, also
reveal their parameters in /sys/module/${modulename}/parameters/. Some of these
parameters may be changed at runtime by the command
"echo -n ${value} > /sys/module/${modulename}/parameters/${parm}".
snd-ad1816a= [HW,ALSA]
snd-ad1848= [HW,ALSA]
snd-ali5451= [HW,ALSA]
snd-als100= [HW,ALSA]
snd-als4000= [HW,ALSA]
snd-azt2320= [HW,ALSA]
snd-cmi8330= [HW,ALSA]
snd-cmipci= [HW,ALSA]
snd-cs4231= [HW,ALSA]
snd-cs4232= [HW,ALSA]
snd-cs4236= [HW,ALSA]
snd-cs4281= [HW,ALSA]
snd-cs46xx= [HW,ALSA]
snd-dt019x= [HW,ALSA]
snd-dummy= [HW,ALSA]
snd-emu10k1= [HW,ALSA]
snd-ens1370= [HW,ALSA]
snd-ens1371= [HW,ALSA]
snd-es968= [HW,ALSA]
snd-es1688= [HW,ALSA]
snd-es18xx= [HW,ALSA]
snd-es1938= [HW,ALSA]
snd-es1968= [HW,ALSA]
snd-fm801= [HW,ALSA]
snd-gusclassic= [HW,ALSA]
snd-gusextreme= [HW,ALSA]
snd-gusmax= [HW,ALSA]
snd-hdsp= [HW,ALSA]
snd-ice1712= [HW,ALSA]
snd-intel8x0= [HW,ALSA]
snd-interwave= [HW,ALSA]
snd-interwave-stb=
[HW,ALSA]
snd-korg1212= [HW,ALSA]
snd-maestro3= [HW,ALSA]
snd-mpu401= [HW,ALSA]
snd-mtpav= [HW,ALSA]
snd-nm256= [HW,ALSA]
snd-opl3sa2= [HW,ALSA]
snd-opti92x-ad1848=
[HW,ALSA]
snd-opti92x-cs4231=
[HW,ALSA]
snd-opti93x= [HW,ALSA]
snd-pmac= [HW,ALSA]
snd-rme32= [HW,ALSA]
snd-rme96= [HW,ALSA]
snd-rme9652= [HW,ALSA]
snd-sb8= [HW,ALSA]
snd-sb16= [HW,ALSA]
snd-sbawe= [HW,ALSA]
snd-serial= [HW,ALSA]
snd-sgalaxy= [HW,ALSA]
snd-sonicvibes= [HW,ALSA]
snd-sun-amd7930=
[HW,ALSA]
snd-sun-cs4231= [HW,ALSA]
snd-trident= [HW,ALSA]
snd-usb-audio= [HW,ALSA,USB]
snd-via82xx= [HW,ALSA]
snd-virmidi= [HW,ALSA]
snd-wavefront= [HW,ALSA]
snd-ymfpci= [HW,ALSA]
Documentation/sound/alsa/seq_oss.html 已删除 100644 → 0
浏览文件 @ 4ce8e6a5
<!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>
Documentation/sound/alsa/Audigy-mixer.txt → Documentation/sound/cards/audigy-mixer.rst
浏览文件 @ a04dae9e
=============================================
Sound Blaster Audigy mixer / default DSP code
=============================================
Sound Blaster Audigy mixer / default DSP code This is based on sb-live-mixer.rst.
===========================================
This is based on SB-Live-mixer.txt.
The EMU10K2 chips have a DSP part which can be programmed to support The EMU10K2 chips have a DSP part which can be programmed to support
various ways of sample processing, which is described here. various ways of sample processing, which is described here.
...@@ -13,8 +13,8 @@ The ALSA driver programs this portion of chip by default code ...@@ -13,8 +13,8 @@ The ALSA driver programs this portion of chip by default code
(can be altered later) which offers the following functionality: (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 These controls are built using the DSP instructions. They offer extended
functionality. Only the default build-in code in the ALSA driver is described 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 ...@@ -26,320 +26,343 @@ is mentioned in multiple controls, the signal is accumulated and can be wrapped
Explanation of used abbreviations: Explanation of used abbreviations:
DAC - digital to analog converter DAC
ADC - analog to digital converter digital to analog converter
I2S - one-way three wire serial bus for digital sound by Philips Semiconductors ADC
(this standard is used for connecting standalone DAC and ADC converters) analog to digital converter
LFE - low frequency effects (subwoofer signal) I2S
AC97 - a chip containing an analog mixer, DAC and ADC converters one-way three wire serial bus for digital sound by Philips Semiconductors
IEC958 - S/PDIF (this standard is used for connecting standalone DAC and ADC converters)
FX-bus - the EMU10K2 chip has an effect bus containing 64 accumulators. LFE
Each of the synthesizer voices can feed its output to these accumulators low frequency effects (subwoofer signal)
and the DSP microcontroller can operate with the resulting sum. 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 name='PCM Front Playback Volume',index=0
----------------------------------------
This control is used to attenuate samples for left and right front PCM FX-bus 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 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 samples for 5.1 playback. The result samples are forwarded to the front DAC PCM
slots of the Philips DAC. slots of the Philips DAC.
name='PCM Surround Playback Volume',index=0 name='PCM Surround Playback Volume',index=0
-------------------------------------------
This control is used to attenuate samples for left and right surround PCM FX-bus 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 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 samples for 5.1 playback. The result samples are forwarded to the surround DAC PCM
slots of the Philips DAC. slots of the Philips DAC.
name='PCM Center Playback Volume',index=0 name='PCM Center Playback Volume',index=0
-----------------------------------------
This control is used to attenuate samples for center PCM FX-bus accumulator. 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 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. is forwarded to the center DAC PCM slot of the Philips DAC.
name='PCM LFE Playback Volume',index=0 name='PCM LFE Playback Volume',index=0
--------------------------------------
This control is used to attenuate sample for LFE PCM FX-bus accumulator. 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 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. is forwarded to the LFE DAC PCM slot of the Philips DAC.
name='PCM Playback Volume',index=0 name='PCM Playback Volume',index=0
----------------------------------
This control is used to attenuate samples for left and right PCM FX-bus 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 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 stereo playback. The result samples are forwarded to the front DAC PCM slots
of the Philips DAC. of the Philips DAC.
name='PCM Capture Volume',index=0 name='PCM Capture Volume',index=0
---------------------------------
This control is used to attenuate samples for left and right PCM FX-bus 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. 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 The result is forwarded to the ADC capture FIFO (thus to the standard capture
PCM device). PCM device).
name='Music Playback Volume',index=0 name='Music Playback Volume',index=0
------------------------------------
This control is used to attenuate samples for left and right MIDI FX-bus 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. 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. The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
name='Music Capture Volume',index=0 name='Music Capture Volume',index=0
-----------------------------------
These controls are used to attenuate samples for left and right MIDI FX-bus 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. 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 The result is forwarded to the ADC capture FIFO (thus to the standard capture
PCM device). PCM device).
name='Mic Playback Volume',index=0 name='Mic Playback Volume',index=0
----------------------------------
This control is used to attenuate samples for left and right Mic input. 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 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 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. capture FIFO (device 1 - 16bit/8KHz mono) too without volume control.
name='Mic Capture Volume',index=0 name='Mic Capture Volume',index=0
---------------------------------
This control is used to attenuate samples for left and right Mic input. 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 The result is forwarded to the ADC capture FIFO (thus to the standard capture
PCM device). PCM device).
name='Audigy CD Playback Volume',index=0 name='Audigy CD Playback Volume',index=0
----------------------------------------
This control is used to attenuate samples from left and right IEC958 TTL 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 digital inputs (usually used by a CDROM drive). The result samples are
forwarded to the front DAC PCM slots of the Philips DAC. forwarded to the front DAC PCM slots of the Philips DAC.
name='Audigy CD Capture Volume',index=0 name='Audigy CD Capture Volume',index=0
---------------------------------------
This control is used to attenuate samples from left and right IEC958 TTL 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 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). forwarded to the ADC capture FIFO (thus to the standard capture PCM device).
name='IEC958 Optical Playback Volume',index=0 name='IEC958 Optical Playback Volume',index=0
---------------------------------------------
This control is used to attenuate samples from left and right IEC958 optical 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 digital input. The result samples are forwarded to the front DAC PCM slots
of the Philips DAC. of the Philips DAC.
name='IEC958 Optical Capture Volume',index=0 name='IEC958 Optical Capture Volume',index=0
--------------------------------------------
This control is used to attenuate samples from left and right IEC958 optical 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 digital inputs. The result samples are forwarded to the ADC capture FIFO
(thus to the standard capture PCM device). (thus to the standard capture PCM device).
name='Line2 Playback Volume',index=0 name='Line2 Playback Volume',index=0
------------------------------------
This control is used to attenuate samples from left and right I2S ADC 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 inputs (on the AudigyDrive). The result samples are forwarded to the front
DAC PCM slots of the Philips DAC. DAC PCM slots of the Philips DAC.
name='Line2 Capture Volume',index=1 name='Line2 Capture Volume',index=1
-----------------------------------
This control is used to attenuate samples from left and right I2S ADC 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 inputs (on the AudigyDrive). The result samples are forwarded to the ADC
capture FIFO (thus to the standard capture PCM device). capture FIFO (thus to the standard capture PCM device).
name='Analog Mix Playback Volume',index=0 name='Analog Mix Playback Volume',index=0
-----------------------------------------
This control is used to attenuate samples from left and right I2S ADC 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 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 DAC PCM slots of the Philips DAC. This contains mix from analog sources
like CD, Line In, Aux, .... like CD, Line In, Aux, ....
name='Analog Mix Capture Volume',index=1 name='Analog Mix Capture Volume',index=1
----------------------------------------
This control is used to attenuate samples from left and right I2S ADC This control is used to attenuate samples from left and right I2S ADC
inputs Philips ADC. The result samples are forwarded to the ADC inputs Philips ADC. The result samples are forwarded to the ADC
capture FIFO (thus to the standard capture PCM device). capture FIFO (thus to the standard capture PCM device).
name='Aux2 Playback Volume',index=0 name='Aux2 Playback Volume',index=0
-----------------------------------
This control is used to attenuate samples from left and right I2S ADC 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 inputs (on the AudigyDrive). The result samples are forwarded to the front
DAC PCM slots of the Philips DAC. DAC PCM slots of the Philips DAC.
name='Aux2 Capture Volume',index=1 name='Aux2 Capture Volume',index=1
----------------------------------
This control is used to attenuate samples from left and right I2S ADC 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 inputs (on the AudigyDrive). The result samples are forwarded to the ADC
capture FIFO (thus to the standard capture PCM device). capture FIFO (thus to the standard capture PCM device).
name='Front Playback Volume',index=0 name='Front Playback Volume',index=0
------------------------------------
All stereo signals are mixed together and mirrored to surround, center and LFE. 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 control is used to attenuate samples for left and right front speakers of
this mix. this mix.
name='Surround Playback Volume',index=0 name='Surround Playback Volume',index=0
---------------------------------------
All stereo signals are mixed together and mirrored to surround, center and LFE. 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 control is used to attenuate samples for left and right surround speakers of
this mix. this mix.
name='Center Playback Volume',index=0 name='Center Playback Volume',index=0
-------------------------------------
All stereo signals are mixed together and mirrored to surround, center and LFE. 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. This control is used to attenuate sample for center speaker of this mix.
name='LFE Playback Volume',index=0 name='LFE Playback Volume',index=0
----------------------------------
All stereo signals are mixed together and mirrored to surround, center and LFE. 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. This control is used to attenuate sample for LFE speaker of this mix.
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 This control turns the tone control on or off. The samples for front, rear
and center / LFE outputs are affected. 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!! This control sets the bass intensity. There is no neutral value!!
When the tone control code is activated, the samples are always modified. When the tone control code is activated, the samples are always modified.
The closest value to pure signal is 20. 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!! This control sets the treble intensity. There is no neutral value!!
When the tone control code is activated, the samples are always modified. When the tone control code is activated, the samples are always modified.
The closest value to pure signal is 20. The closest value to pure signal is 20.
name='Master Playback Volume',index=0 name='Master Playback Volume',index=0
-------------------------------------
This control is used to attenuate samples for front, surround, center and This control is used to attenuate samples for front, surround, center and
LFE outputs. LFE outputs.
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 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 output are taken only from the raw FX8010 PCM, otherwise standard front
PCM samples are taken. PCM samples are taken.
2) PCM stream related controls PCM stream related controls
------------------------------ ===========================
name='EMU10K1 PCM Volume',index 0-31 name='EMU10K1 PCM Volume',index 0-31
------------------------------------
Channel volume attenuation in range 0-0xffff. The maximum value (no Channel volume attenuation in range 0-0xffff. The maximum value (no
attenuation) is default. The channel mapping for three values is attenuation) is default. The channel mapping for three values is
as follows: as follows:
0 - mono, default 0xffff (no attenuation) * 0 - mono, default 0xffff (no attenuation)
1 - left, default 0xffff (no attenuation) * 1 - left, default 0xffff (no attenuation)
2 - right, default 0xffff (no attenuation) * 2 - right, default 0xffff (no attenuation)
name='EMU10K1 PCM Send Routing',index 0-31 name='EMU10K1 PCM Send Routing',index 0-31
------------------------------------------
This control specifies the destination - FX-bus accumulators. There 24 This control specifies the destination - FX-bus accumulators. There 24
values with this mapping: values with this mapping:
0 - mono, A 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 * 1 - mono, B destination (FX-bus 0-63), default 1
2 - mono, C destination (FX-bus 0-63), default 2 * 2 - mono, C destination (FX-bus 0-63), default 2
3 - mono, D destination (FX-bus 0-63), default 3 * 3 - mono, D destination (FX-bus 0-63), default 3
4 - mono, E destination (FX-bus 0-63), default 0 * 4 - mono, E destination (FX-bus 0-63), default 0
5 - mono, F 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 * 6 - mono, G destination (FX-bus 0-63), default 0
7 - mono, H 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 * 8 - left, A destination (FX-bus 0-63), default 0
9 - left, B destination (FX-bus 0-63), default 1 * 9 - left, B destination (FX-bus 0-63), default 1
10 - left, C destination (FX-bus 0-63), default 2 * 10 - left, C destination (FX-bus 0-63), default 2
11 - left, D destination (FX-bus 0-63), default 3 * 11 - left, D destination (FX-bus 0-63), default 3
12 - left, E destination (FX-bus 0-63), default 0 * 12 - left, E destination (FX-bus 0-63), default 0
13 - left, F 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 * 14 - left, G destination (FX-bus 0-63), default 0
15 - left, H 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 * 16 - right, A destination (FX-bus 0-63), default 0
17 - right, B destination (FX-bus 0-63), default 1 * 17 - right, B destination (FX-bus 0-63), default 1
18 - right, C destination (FX-bus 0-63), default 2 * 18 - right, C destination (FX-bus 0-63), default 2
19 - right, D destination (FX-bus 0-63), default 3 * 19 - right, D destination (FX-bus 0-63), default 3
20 - right, E destination (FX-bus 0-63), default 0 * 20 - right, E destination (FX-bus 0-63), default 0
21 - right, F 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 * 22 - right, G destination (FX-bus 0-63), default 0
23 - right, H 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 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). 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. It specifies the attenuation (amount) for given destination in range 0-255.
The channel mapping is following: The channel mapping is following:
0 - mono, A destination attn, default 255 (no attenuation) * 0 - mono, A destination attn, default 255 (no attenuation)
1 - mono, B destination attn, default 255 (no attenuation) * 1 - mono, B destination attn, default 255 (no attenuation)
2 - mono, C destination attn, default 0 (mute) * 2 - mono, C destination attn, default 0 (mute)
3 - mono, D destination attn, default 0 (mute) * 3 - mono, D destination attn, default 0 (mute)
4 - mono, E destination attn, default 0 (mute) * 4 - mono, E destination attn, default 0 (mute)
5 - mono, F destination attn, default 0 (mute) * 5 - mono, F destination attn, default 0 (mute)
6 - mono, G destination attn, default 0 (mute) * 6 - mono, G destination attn, default 0 (mute)
7 - mono, H destination attn, default 0 (mute) * 7 - mono, H destination attn, default 0 (mute)
8 - left, A destination attn, default 255 (no attenuation) * 8 - left, A destination attn, default 255 (no attenuation)
9 - left, B destination attn, default 0 (mute) * 9 - left, B destination attn, default 0 (mute)
10 - left, C destination attn, default 0 (mute) * 10 - left, C destination attn, default 0 (mute)
11 - left, D destination attn, default 0 (mute) * 11 - left, D destination attn, default 0 (mute)
12 - left, E destination attn, default 0 (mute) * 12 - left, E destination attn, default 0 (mute)
13 - left, F destination attn, default 0 (mute) * 13 - left, F destination attn, default 0 (mute)
14 - left, G destination attn, default 0 (mute) * 14 - left, G destination attn, default 0 (mute)
15 - left, H destination attn, default 0 (mute) * 15 - left, H destination attn, default 0 (mute)
16 - right, A destination attn, default 0 (mute) * 16 - right, A destination attn, default 0 (mute)
17 - right, B destination attn, default 255 (no attenuation) * 17 - right, B destination attn, default 255 (no attenuation)
18 - right, C destination attn, default 0 (mute) * 18 - right, C destination attn, default 0 (mute)
19 - right, D destination attn, default 0 (mute) * 19 - right, D destination attn, default 0 (mute)
20 - right, E destination attn, default 0 (mute) * 20 - right, E destination attn, default 0 (mute)
21 - right, F destination attn, default 0 (mute) * 21 - right, F destination attn, default 0 (mute)
22 - right, G destination attn, default 0 (mute) * 22 - right, G destination attn, default 0 (mute)
23 - right, H destination attn, default 0 (mute) * 23 - right, H destination attn, default 0 (mute)
4) MANUALS/PATENTS: MANUALS/PATENTS
------------------- ===============
ftp://opensource.creative.com/pub/doc ftp://opensource.creative.com/pub/doc
------------------------------------- -------------------------------------
Files: LM4545.pdf
LM4545.pdf AC97 Codec 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 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 9901814 (A1)
Processor with Instruction Set for Audio Effects (Jan. 14, 1999)
WO 9901953 (A1) Audio Effects Processor having Decoupled Instruction WO 9901953 (A1)
Execution and Audio Data Sequencing (Jan. 14, 1999) Audio Effects Processor having Decoupled Instruction
Execution and Audio Data Sequencing (Jan. 14, 1999)
US Patents (http://www.uspto.gov/) 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 US 5928342
digital sound samples can be concurrently loaded Audio Effects Processor integrated on a single chip
with a multiport memory onto which multiple asynchronous
US 5930158 Processor with Instruction Set for Audio Effects (Jul. 27, 1999) digital sound samples can be concurrently loaded
(Jul. 27, 1999)
US 6032235 Memory initialization circuit (Tram) (Feb. 29, 2000)
US 5930158
US 6138207 Interpolation looping of audio samples in cache connected to (Oct. 24, 2000) Processor with Instruction Set for Audio Effects (Jul. 27, 1999)
system bus with prioritization and modification of bus transfers
in accordance with loop ends and minimum block sizes US 6032235
Memory initialization circuit (Tram) (Feb. 29, 2000)
US 6151670 Method for conserving memory storage using a (Nov. 21, 2000)
pool of short term memory registers US 6138207
Interpolation looping of audio samples in cache connected to
US 6195715 Interrupt control for multiple programs communicating with (Feb. 27, 2001) system bus with prioritization and modification of bus transfers
a common interrupt by associating programs to GP registers, in accordance with loop ends and minimum block sizes
defining interrupt register, polling GP registers, and invoking (Oct. 24, 2000)
callback routine associated with defined interrupt register
US 6151670
Method for conserving memory storage using a
pool of short term memory registers
(Nov. 21, 2000)
US 6195715
Interrupt control for multiple programs communicating with
a common interrupt by associating programs to GP registers,
defining interrupt register, polling GP registers, and invoking
callback routine associated with defined interrupt register
(Feb. 27, 2001)
Documentation/sound/alsa/Audiophile-Usb.txt → Documentation/sound/cards/audiophile-usb.rst
浏览文件 @ a04dae9e
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 This document is a guide to using the M-Audio Audiophile USB (tm) device with
ALSA and JACK. ALSA and JACK.
History History
======= =======
* v1.4 - Thibault Le Meur (2007-07-11) * v1.4 - Thibault Le Meur (2007-07-11)
- Added Low Endianness nature of 16bits-modes
found by Hakan Lennestal <Hakan.Lennestal@brfsodrahamn.se> - Added Low Endianness nature of 16bits-modes
- Modifying document structure found by Hakan Lennestal <Hakan.Lennestal@brfsodrahamn.se>
- Modifying document structure
* v1.5 - Thibault Le Meur (2007-07-12) * 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 This part is a reminder of important facts about the functions and limitations
of the device. of the device.
The device has 4 audio interfaces, and 2 MIDI ports: The device has 4 audio interfaces, and 2 MIDI ports:
* Analog Stereo Input (Ai) * Analog Stereo Input (Ai)
- This port supports 2 pairs of line-level audio inputs (1/4" TS and RCA) - 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 - When the 1/4" TS (jack) connectors are connected, the RCA connectors
are disabled are disabled
* Analog Stereo Output (Ao) * Analog Stereo Output (Ao)
* Digital Stereo Input (Di) * Digital Stereo Input (Di)
* Digital Stereo Output (Do) * Digital Stereo Output (Do)
...@@ -34,56 +43,69 @@ The device has 4 audio interfaces, and 2 MIDI ports: ...@@ -34,56 +43,69 @@ The device has 4 audio interfaces, and 2 MIDI ports:
* Midi Out (Mo) * Midi Out (Mo)
The internal DAC/ADC has the following characteristics: The internal DAC/ADC has the following characteristics:
* sample depth of 16 or 24 bits * sample depth of 16 or 24 bits
* sample rate from 8kHz to 96kHz * sample rate from 8kHz to 96kHz
* Two interfaces can't use different sample depths at the same time. * Two interfaces can't use different sample depths at the same time.
Moreover, the Audiophile USB documentation gives the following Warning: 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 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: activated at the same time depending on the audio mode selected:
* 16-bit/48kHz ==> 4 channels in + 4 channels out * 16-bit/48kHz ==> 4 channels in + 4 channels out
- Ai+Ao+Di+Do - Ai+Ao+Di+Do
* 24-bit/48kHz ==> 4 channels in + 2 channels out, * 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 - 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) * 24-bit/96kHz ==> 2 channels in _or_ 2 channels out (half duplex only)
- Ai or Ao or Di or Do - Ai or Ao or Di or Do
Important facts about the Digital interface: Important facts about the Digital interface:
-------------------------------------------- --------------------------------------------
* The Do port additionally supports surround-encoded AC-3 and DTS passthrough, * 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 - 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 * 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 - 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 - 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 The Audiophile USB MIDI ports will be automatically supported once the
following modules have been loaded: following modules have been loaded:
* snd-usb-audio * snd-usb-audio
* snd-seq-midi * snd-seq-midi
No additional setting is required. 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 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 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 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 The default behavior of the snd-usb-audio driver is to list the device
capabilities at startup and activate the required mode when required capabilities at startup and activate the required mode when required
...@@ -101,6 +123,7 @@ Default Alsa driver mode can lead to device misconfigurations. ...@@ -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 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 Audiophile interfaces are mapped to alsa pcm devices in the following
way (I suppose the device's index is 1): way (I suppose the device's index is 1):
* hw:1,0 is Ao in playback and Di in capture * hw:1,0 is Ao in playback and Di in capture
* hw:1,1 is Do in playback and Ai in capture * hw:1,1 is Do in playback and Ai in capture
* hw:1,2 is Do in AC3/DTS passthrough mode * 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 ...@@ -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. is reported to be big endian in this default driver mode.
Examples: 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 % 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 % 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 % 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 % 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 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. 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 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 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 ...@@ -137,34 +168,39 @@ from the Ai interface sounds distorted (as if boosted with an excessive high
volume gain). volume gain).
For people having this problem, the snd-usb-audio module has a new module 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) 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 As far as the Audiophile USB device is concerned, this value let the user
specify: specify:
* the sample depth * the sample depth
* the sample rate * the sample rate
* whether the Di port is used or not * 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 the same behaviour as when the parameter is omitted (see paragraph "Default
Alsa driver mode" above) Alsa driver mode" above)
Others modes are described in the following subsections. Others modes are described in the following subsections.
3.2.1.1 - 16-bit modes 16-bit modes
~~~~~~~~~~~~
The two supported modes are: The two supported modes are:
* device_setup=0x01 * ``device_setup=0x01``
- 16bits 48kHz mode with Di disabled - 16bits 48kHz mode with Di disabled
- Ai,Ao,Do can be used at the same time - Ai,Ao,Do can be used at the same time
- hw:1,0 is not available in capture mode - hw:1,0 is not available in capture mode
- hw:1,2 is not available - hw:1,2 is not available
* device_setup=0x11 * ``device_setup=0x11``
- 16bits 48kHz mode with Di enabled - 16bits 48kHz mode with Di enabled
- Ai,Ao,Di,Do can be used at the same time - Ai,Ao,Di,Do can be used at the same time
- hw:1,0 is available in capture mode - hw:1,0 is available in capture mode
...@@ -173,33 +209,43 @@ The two supported modes are: ...@@ -173,33 +209,43 @@ The two supported modes are:
In this modes the device operates only at 16bits-modes. Before kernel 2.6.23, 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 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: 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 % 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. 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 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 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 Alsa now reports Little-Endian interfaces. Thus playing a file now is as simple as
using: using:
::
% aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_LE test_S16_LE.raw % 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: The three supported modes are:
* device_setup=0x09 * ``device_setup=0x09``
- 24bits 48kHz mode with Di disabled - 24bits 48kHz mode with Di disabled
- Ai,Ao,Do can be used at the same time - Ai,Ao,Do can be used at the same time
- hw:1,0 is not available in capture mode - hw:1,0 is not available in capture mode
- hw:1,2 is not available - hw:1,2 is not available
* device_setup=0x19 * ``device_setup=0x19``
- 24bits 48kHz mode with Di enabled - 24bits 48kHz mode with Di enabled
- 3 ports from {Ai,Ao,Di,Do} can be used at the same time - 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 - hw:1,0 is available in capture mode and an active digital source must be
connected to Di connected to Di
- hw:1,2 is not available - hw:1,2 is not available
* device_setup=0x0D or 0x10 * ``device_setup=0x0D`` or ``0x10``
- 24bits 96kHz mode - 24bits 96kHz mode
- Di is enabled by default for this mode but does not need to be connected - Di is enabled by default for this mode but does not need to be connected
to an active source to an active source
...@@ -210,29 +256,35 @@ The three supported modes are: ...@@ -210,29 +256,35 @@ The three supported modes are:
In these modes the device is only Big-Endian compliant (see "Default Alsa driver In these modes the device is only Big-Endian compliant (see "Default Alsa driver
mode" above for an aplay command example) 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. 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 - 16bits 48kHz mode with only the Do port enabled
- AC3 with DTS passthru - AC3 with DTS passthru
- Caution with this setup the Do port is mapped to the pcm device hw:1,0 - 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: 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 % 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: 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 -r snd-usb-audio
# modprobe snd-usb-audio index=1 device_setup=0x09 # modprobe snd-usb-audio index=1 device_setup=0x09
* Or while configuring the modules options in your modules configuration file * 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 alias snd-card-1 snd-usb-audio
options snd-usb-audio index=1 device_setup=0x09 options snd-usb-audio index=1 device_setup=0x09
...@@ -250,26 +302,31 @@ CAUTION when initializing the device ...@@ -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 * 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 to another mode (possibly with another sample-depth), please use also the following
procedure: procedure:
- first turn off the device - first turn off the device
- de-register the snd-usb-audio module (modprobe -r) - de-register the snd-usb-audio module (modprobe -r)
- change the device_setup parameter by changing the device_setup - 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 - turn on the device
* A workaround for this last issue has been applied to kernel 2.6.23, but it may not * 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. 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 This section is for hackers, wanting to understand details about the device
internals and how Alsa supports it. 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 If you want to understand the device_setup magic numbers for the Audiophile
USB, you need some very basic understanding of binary computation. However, 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. 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: The device_setup is one byte long and its structure is the following:
::
+---+---+---+---+---+---+---+---+ +---+---+---+---+---+---+---+---+
| b7| b6| b5| b4| b3| b2| b1| b0| | 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: ...@@ -278,38 +335,55 @@ The device_setup is one byte long and its structure is the following:
+---+---+---+---+---+---+---+---+ +---+---+---+---+---+---+---+---+
Where: Where:
* b0 is the "SET" bit
* b0 is the ``SET`` bit
- it MUST be set if device_setup is initialized - 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 - it is set only for Digital output with DTS/AC3
- this setup is not tested - this setup is not tested
* b2 is the Rate selection flag * 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 - Otherwise the sample rate range is 8-48kHz
* b3 is the bit depth selection flag * 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 - Otherwise they are 16bits long
- Note that b2 implies b3 as the 96kHz mode is only supported for 24 bits - Note that b2 implies b3 as the 96kHz mode is only supported for 24 bits
samples samples
* b4 is the Digital input flag * 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 connected
- You shouldn't enable Di if no source is seen on the port (this leads to - You shouldn't enable Di if no source is seen on the port (this leads to
synchronization issues) synchronization issues)
- b4 is implied by b2 (since only one port is enabled at a time no synch - b4 is implied by b2 (since only one port is enabled at a time no synch
error can occur) 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 - might become Ao, Do, Ai, for b7, b6, b4 respectively
Caution: Caution:
* there is no check on the value you will give to device_setup * 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 - 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 b2 implies b3. But _there_will_be_no_warning_ in /var/log/messages
* Hardware constraints due to the USB bus limitation aren't checked * Hardware constraints due to the USB bus limitation aren't checked
- choosing b2 will prepare all interfaces for 24bits/96kHz but you'll - choosing b2 will prepare all interfaces for 24bits/96kHz but you'll
only be able to use one at the same time 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 You may safely skip this section if you're not interested in driver
hacking. hacking.
...@@ -319,46 +393,72 @@ data I got by usb-snooping the windows and Linux drivers. ...@@ -319,46 +393,72 @@ data I got by usb-snooping the windows and Linux drivers.
The M-Audio Audiophile USB has 7 USB Interfaces: The M-Audio Audiophile USB has 7 USB Interfaces:
a "USB interface": a "USB interface":
* USB Interface nb.0 * USB Interface nb.0
* USB Interface nb.1 * USB Interface nb.1
- Audio Control function - Audio Control function
* USB Interface nb.2 * USB Interface nb.2
- Analog Output - Analog Output
* USB Interface nb.3 * USB Interface nb.3
- Digital Output - Digital Output
* USB Interface nb.4 * USB Interface nb.4
- Analog Input - Analog Input
* USB Interface nb.5 * USB Interface nb.5
- Digital Input - Digital Input
* USB Interface nb.6 * USB Interface nb.6
- MIDI interface compliant with the MIDIMAN quirk - MIDI interface compliant with the MIDIMAN quirk
Each interface has 5 altsettings (AltSet 1,2,3,4,5) except: Each interface has 5 altsettings (AltSet 1,2,3,4,5) except:
* Interface 3 (Digital Out) has an extra Alset nb.6 * Interface 3 (Digital Out) has an extra Alset nb.6
* Interface 5 (Digital In) does not have Alset nb.3 and 5 * Interface 5 (Digital In) does not have Alset nb.3 and 5
Here is a short description of the AltSettings capabilities: 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 - 24-bit depth, 48.1-96kHz sample mode
- Adaptive playback (Ao and Do), Synch capture (Ai), or Asynch capture (Di) - 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 - 24-bit depth, 8-48kHz sample mode
- Asynch capture and playback (Ao,Ai,Do,Di) - Asynch capture and playback (Ao,Ai,Do,Di)
* AltSettings 3 corresponds to
* AltSettings 3 corresponds to
- 24-bit depth, 8-48kHz sample mode - 24-bit depth, 8-48kHz sample mode
- Synch capture (Ai) and Adaptive playback (Ao,Do) - Synch capture (Ai) and Adaptive playback (Ao,Do)
* AltSettings 4 corresponds to
* AltSettings 4 corresponds to
- 16-bit depth, 8-48kHz sample mode - 16-bit depth, 8-48kHz sample mode
- Asynch capture and playback (Ao,Ai,Do,Di) - Asynch capture and playback (Ao,Ai,Do,Di)
* AltSettings 5 corresponds to
* AltSettings 5 corresponds to
- 16-bit depth, 8-48kHz sample mode - 16-bit depth, 8-48kHz sample mode
- Synch capture (Ai) and Adaptive playback (Ao,Do) - Synch capture (Ai) and Adaptive playback (Ao,Do)
* AltSettings 6 corresponds to
* AltSettings 6 corresponds to
- 16-bit depth, 8-48kHz sample mode - 16-bit depth, 8-48kHz sample mode
- Synch playback (Do), audio format type III IEC1937_AC-3 - Synch playback (Do), audio format type III IEC1937_AC-3
In order to ensure a correct initialization of the device, the driver 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 * if DTS is chosen, only Interface 2 with AltSet nb.6 must be
registered registered
* if 96KHz only AltSets nb.1 of each interface must be selected * if 96KHz only AltSets nb.1 of each interface must be selected
...@@ -371,20 +471,21 @@ _must_know_ how the device will be used: ...@@ -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 When device_setup is given as a parameter to the snd-usb-audio module, the
parse_audio_endpoints function uses a quirk called 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. 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. This section deals with support of the Audiophile USB device in Jack.
There are 2 main potential issues when using Jackd with the device: There are 2 main potential issues when using Jackd with the device:
* support for Big-Endian devices in 24-bit modes * support for Big-Endian devices in 24-bit modes
* support for 4-in / 4-out channels * 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 Jack supports big endian devices only in recent versions (thanks to
Andreas Steinmetz for his first big-endian patch). I can't remember Andreas Steinmetz for his first big-endian patch). I can't remember
...@@ -396,29 +497,35 @@ are now Little Endians ;-) ). ...@@ -396,29 +497,35 @@ are now Little Endians ;-) ).
You can run jackd with the following command for playback with Ao and You can run jackd with the following command for playback with Ao and
record with Ai: record with Ai:
::
% jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1 % 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 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 For instance here is one way to run Jack with 2 playback channels on Ao and 2
capture channels from Ai: capture channels from Ai:
::
% jackd -R -dalsa -dplughw:1 -r48000 -p256 -n2 -D -Cplughw:1,1 % jackd -R -dalsa -dplughw:1 -r48000 -p256 -n2 -D -Cplughw:1,1
However you may see the following warning message: However you may see the following warning message:
"You appear to be using the ALSA software "plug" layer, probably a result of 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. 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." 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 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). input (Di or Ai) and 1 stereo output (Ao or Do).
This is due to the following restrictions: This is due to the following restrictions:
* Jack can only open one capture device and one playback device at a time * 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 * The Audiophile USB is seen as 2 (or three) Alsa devices: hw:1,0, hw:1,1
(and optionally hw:1,2) (and optionally hw:1,2)
...@@ -432,6 +539,7 @@ It is related to another device (ice1712) but can be adapted to suit ...@@ -432,6 +539,7 @@ It is related to another device (ice1712) but can be adapted to suit
the Audiophile USB. the Audiophile USB.
Enabling multiple Audiophile USB interfaces for Jackd will certainly require: Enabling multiple Audiophile USB interfaces for Jackd will certainly require:
* Making sure your Jackd version has the MMAP_COMPLEX patch (see the ice1712 page) * 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) * (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 * define a multi device (combination of hw:1,0 and hw:1,1) in your .asoundrc
......
Documentation/sound/alsa/Bt87x.txt → Documentation/sound/cards/bt87x.rst
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=================
ALSA BT87x Driver
=================
Intro Intro
===== =====
You might have noticed that the bt878 grabber cards have actually You might have noticed that the bt878 grabber cards have actually
_two_ PCI functions: *two* PCI functions:
::
$ lspci $ lspci
[ ... ] [ ... ]
00:0a.0 Multimedia video controller: Brooktree Corporation Bt878 (rev 02) 00:0a.0 Multimedia video controller: Brooktree Corporation Bt878 (rev 02)
00:0a.1 Multimedia 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 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 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 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 into your sound card's line-in you probably don't need this driver if all
you want to do is just watching TV... 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, ...@@ -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. and it refuses to load for cards it doesn't know.
If the driver complains ("Unknown TV card found, the audio driver will 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 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). 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 If you have an unknown card, please mail the ID and board name to
......
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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
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