This is a really obsoleted information, effectively just listing the module names. Let's get rid of it for avoiding confusions.

Signed-off-by: Takashi Iwai tiwai@suse.de --- Documentation/sound/alsa/alsa-parameters.txt | 135 --------------------------- 1 file changed, 135 deletions(-) delete mode 100644 Documentation/sound/alsa/alsa-parameters.txt

diff --git a/Documentation/sound/alsa/alsa-parameters.txt b/Documentation/sound/alsa/alsa-parameters.txt deleted file mode 100644 index 0fa40679b080..000000000000 --- a/Documentation/sound/alsa/alsa-parameters.txt +++ /dev/null @@ -1,135 +0,0 @@ - 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]

Another simple conversion from DocBook to ReST. This required a few manual fixups and reformats, but the most of contents are kept as is.

Signed-off-by: Takashi Iwai tiwai@suse.de --- Documentation/DocBook/Makefile | 3 +- Documentation/DocBook/writing-an-alsa-driver.tmpl | 6206 -------------------- Documentation/sound/kernel-api/index.rst | 1 + .../sound/kernel-api/writing-an-alsa-driver.rst | 4219 +++++++++++++ 4 files changed, 4221 insertions(+), 6208 deletions(-) delete mode 100644 Documentation/DocBook/writing-an-alsa-driver.tmpl create mode 100644 Documentation/sound/kernel-api/writing-an-alsa-driver.rst

diff --git a/Documentation/DocBook/Makefile b/Documentation/DocBook/Makefile index e173497959fa..72f78ae46c10 100644 --- a/Documentation/DocBook/Makefile +++ b/Documentation/DocBook/Makefile @@ -12,8 +12,7 @@ DOCBOOKS := z8530book.xml \ kernel-api.xml filesystems.xml lsm.xml usb.xml kgdb.xml \ gadget.xml libata.xml mtdnand.xml librs.xml rapidio.xml \ genericirq.xml s390-drivers.xml uio-howto.xml scsi.xml \ - debugobjects.xml sh.xml regulator.xml \ - writing-an-alsa-driver.xml \ + 80211.xml debugobjects.xml sh.xml regulator.xml \ tracepoint.xml w1.xml \ writing_musb_glue_layer.xml crypto-API.xml iio.xml

diff --git a/Documentation/DocBook/writing-an-alsa-driver.tmpl b/Documentation/DocBook/writing-an-alsa-driver.tmpl deleted file mode 100644 index a27ab9f53fb6..000000000000 --- a/Documentation/DocBook/writing-an-alsa-driver.tmpl +++ /dev/null @@ -1,6206 +0,0 @@ -<?xml version="1.0" encoding="UTF-8"?> -<!DOCTYPE book PUBLIC "-//OASIS//DTD DocBook XML V4.1.2//EN" - "http://www.oasis-open.org/docbook/xml/4.1.2/docbookx.dtd" []> - -<!-- ****************************************************** --> -<!-- Header --> -<!-- ****************************************************** --> -<book id="Writing-an-ALSA-Driver"> - <bookinfo> - <title>Writing an ALSA Driver</title> - <author> - <firstname>Takashi</firstname> - <surname>Iwai</surname> - <affiliation> - <address> - <email>tiwai@suse.de</email> - </address> - </affiliation> - </author> - - <date>Oct 15, 2007</date> - <edition>0.3.7</edition> - - <abstract> - <para> - This document describes how to write an ALSA (Advanced Linux - Sound Architecture) driver. - </para> - </abstract> - - <legalnotice> - <para> - Copyright (c) 2002-2005 Takashi Iwai <email>tiwai@suse.de</email> - </para> - - <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> - -<!-- ****************************************************** --> -<!-- Preface --> -<!-- ****************************************************** --> - <preface id="preface"> - <title>Preface</title> - <para> - This document describes how to write an - <ulink url="http://www.alsa-project.org/"><citetitle> - ALSA (Advanced Linux Sound Architecture)</citetitle></ulink> - driver. The document focuses mainly on PCI soundcards. - In the case of other device types, the API might - be different, too. However, at least the ALSA kernel API is - consistent, and therefore it would be still a bit help for - writing them. - </para> - - <para> - This document targets people who already have enough - C language skills and have basic linux kernel programming - knowledge. This document doesn't explain the general - topic of linux kernel coding and doesn't cover low-level - driver implementation details. It only describes - the standard way to write a PCI sound driver on ALSA. - </para> - - <para> - If you are already familiar with the older ALSA ver.0.5.x API, you - can check the drivers such as <filename>sound/pci/es1938.c</filename> or - <filename>sound/pci/maestro3.c</filename> which have also almost the same - code-base in the ALSA 0.5.x tree, so you can compare the differences. - </para> - - <para> - This document is still a draft version. Any feedback and - corrections, please!! - </para> - </preface> - - -<!-- ****************************************************** --> -<!-- File Tree Structure --> -<!-- ****************************************************** --> - <chapter id="file-tree"> - <title>File Tree Structure</title> - - <section id="file-tree-general"> - <title>General</title> - <para> - The ALSA drivers are provided in two ways. - </para> - - <para> - One is the trees provided as a tarball or via cvs from the - ALSA's ftp site, and another is the 2.6 (or later) Linux kernel - tree. To synchronize both, the ALSA driver tree is split into - two different trees: alsa-kernel and alsa-driver. The former - contains purely the source code for the Linux 2.6 (or later) - tree. This tree is designed only for compilation on 2.6 or - later environment. The latter, alsa-driver, contains many subtle - files for compiling ALSA drivers outside of the Linux kernel tree, - wrapper functions for older 2.2 and 2.4 kernels, to adapt the latest kernel API, - and additional drivers which are still in development or in - tests. The drivers in alsa-driver tree will be moved to - alsa-kernel (and eventually to the 2.6 kernel tree) when they are - finished and confirmed to work fine. - </para> - - <para> - The file tree structure of ALSA driver is depicted below. Both - alsa-kernel and alsa-driver have almost the same file - structure, except for <quote>core</quote> directory. It's - named as <quote>acore</quote> in alsa-driver tree. - - <example> - <title>ALSA File Tree Structure</title> - <literallayout> - sound - /core - /oss - /seq - /oss - /instr - /ioctl32 - /include - /drivers - /mpu401 - /opl3 - /i2c - /l3 - /synth - /emux - /pci - /(cards) - /isa - /(cards) - /arm - /ppc - /sparc - /usb - /pcmcia /(cards) - /oss - </literallayout> - </example> - </para> - </section> - - <section id="file-tree-core-directory"> - <title>core directory</title> - <para> - This directory contains the middle layer which is the heart - of ALSA drivers. In this directory, the native ALSA modules are - stored. The sub-directories contain different modules and are - dependent upon the kernel config. - </para> - - <section id="file-tree-core-directory-oss"> - <title>core/oss</title> - - <para> - The codes for PCM and mixer OSS emulation modules are stored - in this directory. The rawmidi OSS emulation is included in - the ALSA rawmidi code since it's quite small. The sequencer - code is stored in <filename>core/seq/oss</filename> directory (see - <link linkend="file-tree-core-directory-seq-oss"><citetitle> - below</citetitle></link>). - </para> - </section> - - <section id="file-tree-core-directory-ioctl32"> - <title>core/ioctl32</title> - - <para> - This directory contains the 32bit-ioctl wrappers for 64bit - architectures such like x86-64, ppc64 and sparc64. For 32bit - and alpha architectures, these are not compiled. - </para> - </section> - - <section id="file-tree-core-directory-seq"> - <title>core/seq</title> - <para> - This directory and its sub-directories are for the ALSA - sequencer. This directory contains the sequencer core and - primary sequencer modules such like snd-seq-midi, - snd-seq-virmidi, etc. They are compiled only when - <constant>CONFIG_SND_SEQUENCER</constant> is set in the kernel - config. - </para> - </section> - - <section id="file-tree-core-directory-seq-oss"> - <title>core/seq/oss</title> - <para> - This contains the OSS sequencer emulation codes. - </para> - </section> - - <section id="file-tree-core-directory-deq-instr"> - <title>core/seq/instr</title> - <para> - This directory contains the modules for the sequencer - instrument layer. - </para> - </section> - </section> - - <section id="file-tree-include-directory"> - <title>include directory</title> - <para> - This is the place for the public header files of ALSA drivers, - which are to be exported to user-space, or included by - several files at different directories. Basically, the private - header files should not be placed in this directory, but you may - still find files there, due to historical reasons :) - </para> - </section> - - <section id="file-tree-drivers-directory"> - <title>drivers directory</title> - <para> - This directory contains code shared among different drivers - on different architectures. They are hence supposed not to be - architecture-specific. - For example, the dummy pcm driver and the serial MIDI - driver are found in this directory. In the sub-directories, - there is code for components which are independent from - bus and cpu architectures. - </para> - - <section id="file-tree-drivers-directory-mpu401"> - <title>drivers/mpu401</title> - <para> - The MPU401 and MPU401-UART modules are stored here. - </para> - </section> - - <section id="file-tree-drivers-directory-opl3"> - <title>drivers/opl3 and opl4</title> - <para> - The OPL3 and OPL4 FM-synth stuff is found here. - </para> - </section> - </section> - - <section id="file-tree-i2c-directory"> - <title>i2c directory</title> - <para> - This contains the ALSA i2c components. - </para> - - <para> - Although there is a standard i2c layer on Linux, ALSA has its - own i2c code for some cards, because the soundcard needs only a - simple operation and the standard i2c API is too complicated for - such a purpose. - </para> - - <section id="file-tree-i2c-directory-l3"> - <title>i2c/l3</title> - <para> - This is a sub-directory for ARM L3 i2c. - </para> - </section> - </section> - - <section id="file-tree-synth-directory"> - <title>synth directory</title> - <para> - This contains the synth middle-level modules. - </para> - - <para> - So far, there is only Emu8000/Emu10k1 synth driver under - the <filename>synth/emux</filename> sub-directory. - </para> - </section> - - <section id="file-tree-pci-directory"> - <title>pci directory</title> - <para> - This directory and its sub-directories hold the top-level card modules - for PCI soundcards and the code specific to the PCI BUS. - </para> - - <para> - The drivers compiled from a single file are stored directly - in the pci directory, while the drivers with several source files are - stored on their own sub-directory (e.g. emu10k1, ice1712). - </para> - </section> - - <section id="file-tree-isa-directory"> - <title>isa directory</title> - <para> - This directory and its sub-directories hold the top-level card modules - for ISA soundcards. - </para> - </section> - - <section id="file-tree-arm-ppc-sparc-directories"> - <title>arm, ppc, and sparc directories</title> - <para> - They are used for top-level card modules which are - specific to one of these architectures. - </para> - </section> - - <section id="file-tree-usb-directory"> - <title>usb directory</title> - <para> - This directory contains the USB-audio driver. In the latest version, the - USB MIDI driver is integrated in the usb-audio driver. - </para> - </section> - - <section id="file-tree-pcmcia-directory"> - <title>pcmcia directory</title> - <para> - The PCMCIA, especially PCCard drivers will go here. CardBus - drivers will be in the pci directory, because their API is identical - to that of standard PCI cards. - </para> - </section> - - <section id="file-tree-oss-directory"> - <title>oss directory</title> - <para> - The OSS/Lite source files are stored here in Linux 2.6 (or - later) tree. In the ALSA driver tarball, this directory is empty, - of course :) - </para> - </section> - </chapter> - - -<!-- ****************************************************** --> -<!-- Basic Flow for PCI Drivers --> -<!-- ****************************************************** --> - <chapter id="basic-flow"> - <title>Basic Flow for PCI Drivers</title> - - <section id="basic-flow-outline"> - <title>Outline</title> - <para> - The minimum flow for PCI soundcards is as follows: - - <itemizedlist> - <listitem><para>define the PCI ID table (see the section - <link linkend="pci-resource-entries"><citetitle>PCI Entries - </citetitle></link>).</para></listitem> - <listitem><para>create <function>probe()</function> callback.</para></listitem> - <listitem><para>create <function>remove()</function> callback.</para></listitem> - <listitem><para>create a <structname>pci_driver</structname> structure - containing the three pointers above.</para></listitem> - <listitem><para>create an <function>init()</function> function just calling - the <function>pci_register_driver()</function> to register the pci_driver table - defined above.</para></listitem> - <listitem><para>create an <function>exit()</function> function to call - the <function>pci_unregister_driver()</function> function.</para></listitem> - </itemizedlist> - </para> - </section> - - <section id="basic-flow-example"> - <title>Full Code Example</title> - <para> - The code example is shown below. Some parts are kept - unimplemented at this moment but will be filled in the - next sections. The numbers in the comment lines of the - <function>snd_mychip_probe()</function> function - refer to details explained in the following section. - - <example> - <title>Basic Flow for PCI Drivers - Example</title> - <programlisting> -<![CDATA[ - #include <linux/init.h> - #include <linux/pci.h> - #include <linux/slab.h> - #include <sound/core.h> - #include <sound/initval.h> - - /* module parameters (see "Module Parameters") */ - /* SNDRV_CARDS: maximum number of cards supported by this module */ - static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; - static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; - static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; - - /* definition of the chip-specific record */ - struct mychip { - struct snd_card *card; - /* the rest of the implementation will be in section - * "PCI Resource Management" - */ - }; - - /* chip-specific destructor - * (see "PCI Resource Management") - */ - static int snd_mychip_free(struct mychip *chip) - { - .... /* will be implemented later... */ - } - - /* component-destructor - * (see "Management of Cards and Components") - */ - static int snd_mychip_dev_free(struct snd_device *device) - { - return snd_mychip_free(device->device_data); - } - - /* chip-specific constructor - * (see "Management of Cards and Components") - */ - static int snd_mychip_create(struct snd_card *card, - struct pci_dev *pci, - struct mychip **rchip) - { - struct mychip *chip; - int err; - static struct snd_device_ops ops = { - .dev_free = snd_mychip_dev_free, - }; - - *rchip = NULL; - - /* check PCI availability here - * (see "PCI Resource Management") - */ - .... - - /* allocate a chip-specific data with zero filled */ - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (chip == NULL) - return -ENOMEM; - - chip->card = card; - - /* rest of initialization here; will be implemented - * later, see "PCI Resource Management" - */ - .... - - err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); - if (err < 0) { - snd_mychip_free(chip); - return err; - } - - *rchip = chip; - return 0; - } - - /* constructor -- see "Constructor" sub-section */ - static int snd_mychip_probe(struct pci_dev *pci, - const struct pci_device_id *pci_id) - { - static int dev; - struct snd_card *card; - struct mychip *chip; - int err; - - /* (1) */ - if (dev >= SNDRV_CARDS) - return -ENODEV; - if (!enable[dev]) { - dev++; - return -ENOENT; - } - - /* (2) */ - err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, - 0, &card); - if (err < 0) - return err; - - /* (3) */ - err = snd_mychip_create(card, pci, &chip); - if (err < 0) { - snd_card_free(card); - return err; - } - - /* (4) */ - strcpy(card->driver, "My Chip"); - strcpy(card->shortname, "My Own Chip 123"); - sprintf(card->longname, "%s at 0x%lx irq %i", - card->shortname, chip->ioport, chip->irq); - - /* (5) */ - .... /* implemented later */ - - /* (6) */ - err = snd_card_register(card); - if (err < 0) { - snd_card_free(card); - return err; - } - - /* (7) */ - pci_set_drvdata(pci, card); - dev++; - return 0; - } - - /* destructor -- see the "Destructor" sub-section */ - static void snd_mychip_remove(struct pci_dev *pci) - { - snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); - } -]]> - </programlisting> - </example> - </para> - </section> - - <section id="basic-flow-constructor"> - <title>Constructor</title> - <para> - The real constructor of PCI drivers is the <function>probe</function> callback. - The <function>probe</function> callback and other component-constructors which are called - from the <function>probe</function> callback cannot be used with - the <parameter>__init</parameter> prefix - because any PCI device could be a hotplug device. - </para> - - <para> - In the <function>probe</function> callback, the following scheme is often used. - </para> - - <section id="basic-flow-constructor-device-index"> - <title>1) Check and increment the device index.</title> - <para> - <informalexample> - <programlisting> -<![CDATA[ - static int dev; - .... - if (dev >= SNDRV_CARDS) - return -ENODEV; - if (!enable[dev]) { - dev++; - return -ENOENT; - } -]]> - </programlisting> - </informalexample> - - where enable[dev] is the module option. - </para> - - <para> - Each time the <function>probe</function> callback is called, check the - availability of the device. If not available, simply increment - the device index and returns. dev will be incremented also - later (<link - linkend="basic-flow-constructor-set-pci"><citetitle>step - 7</citetitle></link>). - </para> - </section> - - <section id="basic-flow-constructor-create-card"> - <title>2) Create a card instance</title> - <para> - <informalexample> - <programlisting> -<![CDATA[ - struct snd_card *card; - int err; - .... - err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, - 0, &card); -]]> - </programlisting> - </informalexample> - </para> - - <para> - The details will be explained in the section - <link linkend="card-management-card-instance"><citetitle> - Management of Cards and Components</citetitle></link>. - </para> - </section> - - <section id="basic-flow-constructor-create-main"> - <title>3) Create a main component</title> - <para> - In this part, the PCI resources are allocated. - - <informalexample> - <programlisting> -<![CDATA[ - struct mychip *chip; - .... - err = snd_mychip_create(card, pci, &chip); - if (err < 0) { - snd_card_free(card); - return err; - } -]]> - </programlisting> - </informalexample> - - The details will be explained in the section <link - linkend="pci-resource"><citetitle>PCI Resource - Management</citetitle></link>. - </para> - </section> - - <section id="basic-flow-constructor-main-component"> - <title>4) Set the driver ID and name strings.</title> - <para> - <informalexample> - <programlisting> -<![CDATA[ - strcpy(card->driver, "My Chip"); - strcpy(card->shortname, "My Own Chip 123"); - sprintf(card->longname, "%s at 0x%lx irq %i", - card->shortname, chip->ioport, chip->irq); -]]> - </programlisting> - </informalexample> - - The driver field holds the minimal ID string of the - chip. This is used by alsa-lib's configurator, so keep it - simple but unique. - Even the same driver can have different driver IDs to - distinguish the functionality of each chip type. - </para> - - <para> - The shortname field is a string shown as more verbose - name. The longname field contains the information - shown in <filename>/proc/asound/cards</filename>. - </para> - </section> - - <section id="basic-flow-constructor-create-other"> - <title>5) Create other components, such as mixer, MIDI, etc.</title> - <para> - Here you define the basic components such as - <link linkend="pcm-interface"><citetitle>PCM</citetitle></link>, - mixer (e.g. <link linkend="api-ac97"><citetitle>AC97</citetitle></link>), - MIDI (e.g. <link linkend="midi-interface"><citetitle>MPU-401</citetitle></link>), - and other interfaces. - Also, if you want a <link linkend="proc-interface"><citetitle>proc - file</citetitle></link>, define it here, too. - </para> - </section> - - <section id="basic-flow-constructor-register-card"> - <title>6) Register the card instance.</title> - <para> - <informalexample> - <programlisting> -<![CDATA[ - err = snd_card_register(card); - if (err < 0) { - snd_card_free(card); - return err; - } -]]> - </programlisting> - </informalexample> - </para> - - <para> - Will be explained in the section <link - linkend="card-management-registration"><citetitle>Management - of Cards and Components</citetitle></link>, too. - </para> - </section> - - <section id="basic-flow-constructor-set-pci"> - <title>7) Set the PCI driver data and return zero.</title> - <para> - <informalexample> - <programlisting> -<![CDATA[ - pci_set_drvdata(pci, card); - dev++; - return 0; -]]> - </programlisting> - </informalexample> - - In the above, the card record is stored. This pointer is - used in the remove callback and power-management - callbacks, too. - </para> - </section> - </section> - - <section id="basic-flow-destructor"> - <title>Destructor</title> - <para> - The destructor, remove callback, simply releases the card - instance. Then the ALSA middle layer will release all the - attached components automatically. - </para> - - <para> - It would be typically like the following: - - <informalexample> - <programlisting> -<![CDATA[ - static void snd_mychip_remove(struct pci_dev *pci) - { - snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); - } -]]> - </programlisting> - </informalexample> - - The above code assumes that the card pointer is set to the PCI - driver data. - </para> - </section> - - <section id="basic-flow-header-files"> - <title>Header Files</title> - <para> - For the above example, at least the following include files - are necessary. - - <informalexample> - <programlisting> -<![CDATA[ - #include <linux/init.h> - #include <linux/pci.h> - #include <linux/slab.h> - #include <sound/core.h> - #include <sound/initval.h> -]]> - </programlisting> - </informalexample> - - where the last one is necessary only when module options are - defined in the source file. If the code is split into several - files, the files without module options don't need them. - </para> - - <para> - In addition to these headers, you'll need - <filename>&lt;linux/interrupt.h&gt;</filename> for interrupt - handling, and <filename>&lt;asm/io.h&gt;</filename> for I/O - access. If you use the <function>mdelay()</function> or - <function>udelay()</function> functions, you'll need to include - <filename>&lt;linux/delay.h&gt;</filename> too. - </para> - - <para> - The ALSA interfaces like the PCM and control APIs are defined in other - <filename>&lt;sound/xxx.h&gt;</filename> header files. - They have to be included after - <filename>&lt;sound/core.h&gt;</filename>. - </para> - - </section> - </chapter> - - -<!-- ****************************************************** --> -<!-- Management of Cards and Components --> -<!-- ****************************************************** --> - <chapter id="card-management"> - <title>Management of Cards and Components</title> - - <section id="card-management-card-instance"> - <title>Card Instance</title> - <para> - For each soundcard, a <quote>card</quote> record must be allocated. - </para> - - <para> - A card record is the headquarters of the soundcard. It manages - the whole list of devices (components) on the soundcard, such as - PCM, mixers, MIDI, synthesizer, and so on. Also, the card - record holds the ID and the name strings of the card, manages - the root of proc files, and controls the power-management states - and hotplug disconnections. The component list on the card - record is used to manage the correct release of resources at - destruction. - </para> - - <para> - As mentioned above, to create a card instance, call - <function>snd_card_new()</function>. - - <informalexample> - <programlisting> -<![CDATA[ - struct snd_card *card; - int err; - err = snd_card_new(&pci->dev, index, id, module, extra_size, &card); -]]> - </programlisting> - </informalexample> - </para> - - <para> - The function takes six arguments: the parent device pointer, - the card-index number, the id string, the module pointer (usually - <constant>THIS_MODULE</constant>), - the size of extra-data space, and the pointer to return the - card instance. The extra_size argument is used to - allocate card-&gt;private_data for the - chip-specific data. Note that these data - are allocated by <function>snd_card_new()</function>. - </para> - - <para> - The first argument, the pointer of struct - <structname>device</structname>, specifies the parent device. - For PCI devices, typically &amp;pci-&gt; is passed there. - </para> - </section> - - <section id="card-management-component"> - <title>Components</title> - <para> - After the card is created, you can attach the components - (devices) to the card instance. In an ALSA driver, a component is - represented as a struct <structname>snd_device</structname> object. - A component can be a PCM instance, a control interface, a raw - MIDI interface, etc. Each such instance has one component - entry. - </para> - - <para> - A component can be created via - <function>snd_device_new()</function> function. - - <informalexample> - <programlisting> -<![CDATA[ - snd_device_new(card, SNDRV_DEV_XXX, chip, &ops); -]]> - </programlisting> - </informalexample> - </para> - - <para> - This takes the card pointer, the device-level - (<constant>SNDRV_DEV_XXX</constant>), the data pointer, and the - callback pointers (<parameter>&amp;ops</parameter>). The - device-level defines the type of components and the order of - registration and de-registration. For most components, the - device-level is already defined. For a user-defined component, - you can use <constant>SNDRV_DEV_LOWLEVEL</constant>. - </para> - - <para> - This function itself doesn't allocate the data space. The data - must be allocated manually beforehand, and its pointer is passed - as the argument. This pointer (<parameter>chip</parameter> in the - above example) is used as the identifier for the instance. - </para> - - <para> - Each pre-defined ALSA component such as ac97 and pcm calls - <function>snd_device_new()</function> inside its - constructor. The destructor for each component is defined in the - callback pointers. Hence, you don't need to take care of - calling a destructor for such a component. - </para> - - <para> - If you wish to create your own component, you need to - set the destructor function to the dev_free callback in - the <parameter>ops</parameter>, so that it can be released - automatically via <function>snd_card_free()</function>. - The next example will show an implementation of chip-specific - data. - </para> - </section> - - <section id="card-management-chip-specific"> - <title>Chip-Specific Data</title> - <para> - Chip-specific information, e.g. the I/O port address, its - resource pointer, or the irq number, is stored in the - chip-specific record. - - <informalexample> - <programlisting> -<![CDATA[ - struct mychip { - .... - }; -]]> - </programlisting> - </informalexample> - </para> - - <para> - In general, there are two ways of allocating the chip record. - </para> - - <section id="card-management-chip-specific-snd-card-new"> - <title>1. Allocating via <function>snd_card_new()</function>.</title> - <para> - As mentioned above, you can pass the extra-data-length - to the 5th argument of <function>snd_card_new()</function>, i.e. - - <informalexample> - <programlisting> -<![CDATA[ - err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, - sizeof(struct mychip), &card); -]]> - </programlisting> - </informalexample> - - struct <structname>mychip</structname> is the type of the chip record. - </para> - - <para> - In return, the allocated record can be accessed as - - <informalexample> - <programlisting> -<![CDATA[ - struct mychip *chip = card->private_data; -]]> - </programlisting> - </informalexample> - - With this method, you don't have to allocate twice. - The record is released together with the card instance. - </para> - </section> - - <section id="card-management-chip-specific-allocate-extra"> - <title>2. Allocating an extra device.</title> - - <para> - After allocating a card instance via - <function>snd_card_new()</function> (with - <constant>0</constant> on the 4th arg), call - <function>kzalloc()</function>. - - <informalexample> - <programlisting> -<![CDATA[ - struct snd_card *card; - struct mychip *chip; - err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, - 0, &card); - ..... - chip = kzalloc(sizeof(*chip), GFP_KERNEL); -]]> - </programlisting> - </informalexample> - </para> - - <para> - The chip record should have the field to hold the card - pointer at least, - - <informalexample> - <programlisting> -<![CDATA[ - struct mychip { - struct snd_card *card; - .... - }; -]]> - </programlisting> - </informalexample> - </para> - - <para> - Then, set the card pointer in the returned chip instance. - - <informalexample> - <programlisting> -<![CDATA[ - chip->card = card; -]]> - </programlisting> - </informalexample> - </para> - - <para> - Next, initialize the fields, and register this chip - record as a low-level device with a specified - <parameter>ops</parameter>, - - <informalexample> - <programlisting> -<![CDATA[ - static struct snd_device_ops ops = { - .dev_free = snd_mychip_dev_free, - }; - .... - snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); -]]> - </programlisting> - </informalexample> - - <function>snd_mychip_dev_free()</function> is the - device-destructor function, which will call the real - destructor. - </para> - - <para> - <informalexample> - <programlisting> -<![CDATA[ - static int snd_mychip_dev_free(struct snd_device *device) - { - return snd_mychip_free(device->device_data); - } -]]> - </programlisting> - </informalexample> - - where <function>snd_mychip_free()</function> is the real destructor. - </para> - </section> - </section> - - <section id="card-management-registration"> - <title>Registration and Release</title> - <para> - After all components are assigned, register the card instance - by calling <function>snd_card_register()</function>. Access - to the device files is enabled at this point. That is, before - <function>snd_card_register()</function> is called, the - components are safely inaccessible from external side. If this - call fails, exit the probe function after releasing the card via - <function>snd_card_free()</function>. - </para> - - <para> - For releasing the card instance, you can call simply - <function>snd_card_free()</function>. As mentioned earlier, all - components are released automatically by this call. - </para> - - <para> - For a device which allows hotplugging, you can use - <function>snd_card_free_when_closed</function>. This one will - postpone the destruction until all devices are closed. - </para> - - </section> - - </chapter> - - -<!-- ****************************************************** --> -<!-- PCI Resource Management --> -<!-- ****************************************************** --> - <chapter id="pci-resource"> - <title>PCI Resource Management</title> - - <section id="pci-resource-example"> - <title>Full Code Example</title> - <para> - In this section, we'll complete the chip-specific constructor, - destructor and PCI entries. Example code is shown first, - below. - - <example> - <title>PCI Resource Management Example</title> - <programlisting> -<![CDATA[ - struct mychip { - struct snd_card *card; - struct pci_dev *pci; - - unsigned long port; - int irq; - }; - - static int snd_mychip_free(struct mychip *chip) - { - /* disable hardware here if any */ - .... /* (not implemented in this document) */ - - /* release the irq */ - if (chip->irq >= 0) - free_irq(chip->irq, chip); - /* release the I/O ports & memory */ - pci_release_regions(chip->pci); - /* disable the PCI entry */ - pci_disable_device(chip->pci); - /* release the data */ - kfree(chip); - return 0; - } - - /* chip-specific constructor */ - static int snd_mychip_create(struct snd_card *card, - struct pci_dev *pci, - struct mychip **rchip) - { - struct mychip *chip; - int err; - static struct snd_device_ops ops = { - .dev_free = snd_mychip_dev_free, - }; - - *rchip = NULL; - - /* initialize the PCI entry */ - err = pci_enable_device(pci); - if (err < 0) - return err; - /* check PCI availability (28bit DMA) */ - if (pci_set_dma_mask(pci, DMA_BIT_MASK(28)) < 0 || - pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(28)) < 0) { - printk(KERN_ERR "error to set 28bit mask DMA\n"); - pci_disable_device(pci); - return -ENXIO; - } - - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (chip == NULL) { - pci_disable_device(pci); - return -ENOMEM; - } - - /* initialize the stuff */ - chip->card = card; - chip->pci = pci; - chip->irq = -1; - - /* (1) PCI resource allocation */ - err = pci_request_regions(pci, "My Chip"); - if (err < 0) { - kfree(chip); - pci_disable_device(pci); - return err; - } - chip->port = pci_resource_start(pci, 0); - if (request_irq(pci->irq, snd_mychip_interrupt, - IRQF_SHARED, KBUILD_MODNAME, chip)) { - printk(KERN_ERR "cannot grab irq %d\n", pci->irq); - snd_mychip_free(chip); - return -EBUSY; - } - chip->irq = pci->irq; - - /* (2) initialization of the chip hardware */ - .... /* (not implemented in this document) */ - - err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); - if (err < 0) { - snd_mychip_free(chip); - return err; - } - - *rchip = chip; - return 0; - } - - /* PCI IDs */ - static struct pci_device_id snd_mychip_ids[] = { - { PCI_VENDOR_ID_FOO, PCI_DEVICE_ID_BAR, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, - .... - { 0, } - }; - MODULE_DEVICE_TABLE(pci, snd_mychip_ids); - - /* pci_driver definition */ - static struct pci_driver driver = { - .name = KBUILD_MODNAME, - .id_table = snd_mychip_ids, - .probe = snd_mychip_probe, - .remove = snd_mychip_remove, - }; - - /* module initialization */ - static int __init alsa_card_mychip_init(void) - { - return pci_register_driver(&driver); - } - - /* module clean up */ - static void __exit alsa_card_mychip_exit(void) - { - pci_unregister_driver(&driver); - } - - module_init(alsa_card_mychip_init) - module_exit(alsa_card_mychip_exit) - - EXPORT_NO_SYMBOLS; /* for old kernels only */ -]]> - </programlisting> - </example> - </para> - </section> - - <section id="pci-resource-some-haftas"> - <title>Some Hafta's</title> - <para> - The allocation of PCI resources is done in the - <function>probe()</function> function, and usually an extra - <function>xxx_create()</function> function is written for this - purpose. - </para> - - <para> - In the case of PCI devices, you first have to call - the <function>pci_enable_device()</function> function before - allocating resources. Also, you need to set the proper PCI DMA - mask to limit the accessed I/O range. In some cases, you might - need to call <function>pci_set_master()</function> function, - too. - </para> - - <para> - Suppose the 28bit mask, and the code to be added would be like: - - <informalexample> - <programlisting> -<![CDATA[ - err = pci_enable_device(pci); - if (err < 0) - return err; - if (pci_set_dma_mask(pci, DMA_BIT_MASK(28)) < 0 || - pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(28)) < 0) { - printk(KERN_ERR "error to set 28bit mask DMA\n"); - pci_disable_device(pci); - return -ENXIO; - } - -]]> - </programlisting> - </informalexample> - </para> - </section> - - <section id="pci-resource-resource-allocation"> - <title>Resource Allocation</title> - <para> - The allocation of I/O ports and irqs is done via standard kernel - functions. Unlike ALSA ver.0.5.x., there are no helpers for - that. And these resources must be released in the destructor - function (see below). Also, on ALSA 0.9.x, you don't need to - allocate (pseudo-)DMA for PCI like in ALSA 0.5.x. - </para> - - <para> - Now assume that the PCI device has an I/O port with 8 bytes - and an interrupt. Then struct <structname>mychip</structname> will have the - following fields: - - <informalexample> - <programlisting> -<![CDATA[ - struct mychip { - struct snd_card *card; - - unsigned long port; - int irq; - }; -]]> - </programlisting> - </informalexample> - </para> - - <para> - For an I/O port (and also a memory region), you need to have - the resource pointer for the standard resource management. For - an irq, you have to keep only the irq number (integer). But you - need to initialize this number as -1 before actual allocation, - since irq 0 is valid. The port address and its resource pointer - can be initialized as null by - <function>kzalloc()</function> automatically, so you - don't have to take care of resetting them. - </para> - - <para> - The allocation of an I/O port is done like this: - - <informalexample> - <programlisting> -<![CDATA[ - err = pci_request_regions(pci, "My Chip"); - if (err < 0) { - kfree(chip); - pci_disable_device(pci); - return err; - } - chip->port = pci_resource_start(pci, 0); -]]> - </programlisting> - </informalexample> - </para> - - <para> - <!-- obsolete --> - It will reserve the I/O port region of 8 bytes of the given - PCI device. The returned value, chip-&gt;res_port, is allocated - via <function>kmalloc()</function> by - <function>request_region()</function>. The pointer must be - released via <function>kfree()</function>, but there is a - problem with this. This issue will be explained later. - </para> - - <para> - The allocation of an interrupt source is done like this: - - <informalexample> - <programlisting> -<![CDATA[ - if (request_irq(pci->irq, snd_mychip_interrupt, - IRQF_SHARED, KBUILD_MODNAME, chip)) { - printk(KERN_ERR "cannot grab irq %d\n", pci->irq); - snd_mychip_free(chip); - return -EBUSY; - } - chip->irq = pci->irq; -]]> - </programlisting> - </informalexample> - - where <function>snd_mychip_interrupt()</function> is the - interrupt handler defined <link - linkend="pcm-interface-interrupt-handler"><citetitle>later</citetitle></link>. - Note that chip-&gt;irq should be defined - only when <function>request_irq()</function> succeeded. - </para> - - <para> - On the PCI bus, interrupts can be shared. Thus, - <constant>IRQF_SHARED</constant> is used as the interrupt flag of - <function>request_irq()</function>. - </para> - - <para> - The last argument of <function>request_irq()</function> is the - data pointer passed to the interrupt handler. Usually, the - chip-specific record is used for that, but you can use what you - like, too. - </para> - - <para> - I won't give details about the interrupt handler at this - point, but at least its appearance can be explained now. The - interrupt handler looks usually like the following: - - <informalexample> - <programlisting> -<![CDATA[ - static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id) - { - struct mychip *chip = dev_id; - .... - return IRQ_HANDLED; - } -]]> - </programlisting> - </informalexample> - </para> - - <para> - Now let's write the corresponding destructor for the resources - above. The role of destructor is simple: disable the hardware - (if already activated) and release the resources. So far, we - have no hardware part, so the disabling code is not written here. - </para> - - <para> - To release the resources, the <quote>check-and-release</quote> - method is a safer way. For the interrupt, do like this: - - <informalexample> - <programlisting> -<![CDATA[ - if (chip->irq >= 0) - free_irq(chip->irq, chip); -]]> - </programlisting> - </informalexample> - - Since the irq number can start from 0, you should initialize - chip-&gt;irq with a negative value (e.g. -1), so that you can - check the validity of the irq number as above. - </para> - - <para> - When you requested I/O ports or memory regions via - <function>pci_request_region()</function> or - <function>pci_request_regions()</function> like in this example, - release the resource(s) using the corresponding function, - <function>pci_release_region()</function> or - <function>pci_release_regions()</function>. - - <informalexample> - <programlisting> -<![CDATA[ - pci_release_regions(chip->pci); -]]> - </programlisting> - </informalexample> - </para> - - <para> - When you requested manually via <function>request_region()</function> - or <function>request_mem_region</function>, you can release it via - <function>release_resource()</function>. Suppose that you keep - the resource pointer returned from <function>request_region()</function> - in chip-&gt;res_port, the release procedure looks like: - - <informalexample> - <programlisting> -<![CDATA[ - release_and_free_resource(chip->res_port); -]]> - </programlisting> - </informalexample> - </para> - - <para> - Don't forget to call <function>pci_disable_device()</function> - before the end. - </para> - - <para> - And finally, release the chip-specific record. - - <informalexample> - <programlisting> -<![CDATA[ - kfree(chip); -]]> - </programlisting> - </informalexample> - </para> - - <para> - We didn't implement the hardware disabling part in the above. - If you need to do this, please note that the destructor may be - called even before the initialization of the chip is completed. - It would be better to have a flag to skip hardware disabling - if the hardware was not initialized yet. - </para> - - <para> - When the chip-data is assigned to the card using - <function>snd_device_new()</function> with - <constant>SNDRV_DEV_LOWLELVEL</constant> , its destructor is - called at the last. That is, it is assured that all other - components like PCMs and controls have already been released. - You don't have to stop PCMs, etc. explicitly, but just - call low-level hardware stopping. - </para> - - <para> - The management of a memory-mapped region is almost as same as - the management of an I/O port. You'll need three fields like - the following: - - <informalexample> - <programlisting> -<![CDATA[ - struct mychip { - .... - unsigned long iobase_phys; - void __iomem *iobase_virt; - }; -]]> - </programlisting> - </informalexample> - - and the allocation would be like below: - - <informalexample> - <programlisting> -<![CDATA[ - if ((err = pci_request_regions(pci, "My Chip")) < 0) { - kfree(chip); - return err; - } - chip->iobase_phys = pci_resource_start(pci, 0); - chip->iobase_virt = ioremap_nocache(chip->iobase_phys, - pci_resource_len(pci, 0)); -]]> - </programlisting> - </informalexample> - - and the corresponding destructor would be: - - <informalexample> - <programlisting> -<![CDATA[ - static int snd_mychip_free(struct mychip *chip) - { - .... - if (chip->iobase_virt) - iounmap(chip->iobase_virt); - .... - pci_release_regions(chip->pci); - .... - } -]]> - </programlisting> - </informalexample> - </para> - - </section> - - <section id="pci-resource-entries"> - <title>PCI Entries</title> - <para> - So far, so good. Let's finish the missing PCI - stuff. At first, we need a - <structname>pci_device_id</structname> table for this - chipset. It's a table of PCI vendor/device ID number, and some - masks. - </para> - - <para> - For example, - - <informalexample> - <programlisting> -<![CDATA[ - static struct pci_device_id snd_mychip_ids[] = { - { PCI_VENDOR_ID_FOO, PCI_DEVICE_ID_BAR, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, - .... - { 0, } - }; - MODULE_DEVICE_TABLE(pci, snd_mychip_ids); -]]> - </programlisting> - </informalexample> - </para> - - <para> - The first and second fields of - the <structname>pci_device_id</structname> structure are the vendor and - device IDs. If you have no reason to filter the matching - devices, you can leave the remaining fields as above. The last - field of the <structname>pci_device_id</structname> struct contains - private data for this entry. You can specify any value here, for - example, to define specific operations for supported device IDs. - Such an example is found in the intel8x0 driver. - </para> - - <para> - The last entry of this list is the terminator. You must - specify this all-zero entry. - </para> - - <para> - Then, prepare the <structname>pci_driver</structname> record: - - <informalexample> - <programlisting> -<![CDATA[ - static struct pci_driver driver = { - .name = KBUILD_MODNAME, - .id_table = snd_mychip_ids, - .probe = snd_mychip_probe, - .remove = snd_mychip_remove, - }; -]]> - </programlisting> - </informalexample> - </para> - - <para> - The <structfield>probe</structfield> and - <structfield>remove</structfield> functions have already - been defined in the previous sections. - The <structfield>name</structfield> - field is the name string of this device. Note that you must not - use a slash <quote>/</quote> in this string. - </para> - - <para> - And at last, the module entries: - - <informalexample> - <programlisting> -<![CDATA[ - static int __init alsa_card_mychip_init(void) - { - return pci_register_driver(&driver); - } - - static void __exit alsa_card_mychip_exit(void) - { - pci_unregister_driver(&driver); - } - - module_init(alsa_card_mychip_init) - module_exit(alsa_card_mychip_exit) -]]> - </programlisting> - </informalexample> - </para> - - <para> - Note that these module entries are tagged with - <parameter>__init</parameter> and - <parameter>__exit</parameter> prefixes. - </para> - - <para> - Oh, one thing was forgotten. If you have no exported symbols, - you need to declare it in 2.2 or 2.4 kernels (it's not necessary in 2.6 kernels). - - <informalexample> - <programlisting> -<![CDATA[ - EXPORT_NO_SYMBOLS; -]]> - </programlisting> - </informalexample> - - That's all! - </para> - </section> - </chapter> - - -<!-- ****************************************************** --> -<!-- PCM Interface --> -<!-- ****************************************************** --> - <chapter id="pcm-interface"> - <title>PCM Interface</title> - - <section id="pcm-interface-general"> - <title>General</title> - <para> - The PCM middle layer of ALSA is quite powerful and it is only - necessary for each driver to implement the low-level functions - to access its hardware. - </para> - - <para> - For accessing to the PCM layer, you need to include - <filename>&lt;sound/pcm.h&gt;</filename> first. In addition, - <filename>&lt;sound/pcm_params.h&gt;</filename> might be needed - if you access to some functions related with hw_param. - </para> - - <para> - Each card device can have up to four pcm instances. A pcm - instance corresponds to a pcm device file. The limitation of - number of instances comes only from the available bit size of - the Linux's device numbers. Once when 64bit device number is - used, we'll have more pcm instances available. - </para> - - <para> - A pcm instance consists of pcm playback and capture streams, - and each pcm stream consists of one or more pcm substreams. Some - soundcards support multiple playback functions. For example, - emu10k1 has a PCM playback of 32 stereo substreams. In this case, at - each open, a free substream is (usually) automatically chosen - and opened. Meanwhile, when only one substream exists and it was - already opened, the successful open will either block - or error with <constant>EAGAIN</constant> according to the - file open mode. But you don't have to care about such details in your - driver. The PCM middle layer will take care of such work. - </para> - </section> - - <section id="pcm-interface-example"> - <title>Full Code Example</title> - <para> - The example code below does not include any hardware access - routines but shows only the skeleton, how to build up the PCM - interfaces. - - <example> - <title>PCM Example Code</title> - <programlisting> -<![CDATA[ - #include <sound/pcm.h> - .... - - /* hardware definition */ - static struct snd_pcm_hardware snd_mychip_playback_hw = { - .info = (SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP_VALID), - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rates = SNDRV_PCM_RATE_8000_48000, - .rate_min = 8000, - .rate_max = 48000, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 32768, - .period_bytes_min = 4096, - .period_bytes_max = 32768, - .periods_min = 1, - .periods_max = 1024, - }; - - /* hardware definition */ - static struct snd_pcm_hardware snd_mychip_capture_hw = { - .info = (SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP_VALID), - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rates = SNDRV_PCM_RATE_8000_48000, - .rate_min = 8000, - .rate_max = 48000, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 32768, - .period_bytes_min = 4096, - .period_bytes_max = 32768, - .periods_min = 1, - .periods_max = 1024, - }; - - /* open callback */ - static int snd_mychip_playback_open(struct snd_pcm_substream *substream) - { - struct mychip *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - - runtime->hw = snd_mychip_playback_hw; - /* more hardware-initialization will be done here */ - .... - return 0; - } - - /* close callback */ - static int snd_mychip_playback_close(struct snd_pcm_substream *substream) - { - struct mychip *chip = snd_pcm_substream_chip(substream); - /* the hardware-specific codes will be here */ - .... - return 0; - - } - - /* open callback */ - static int snd_mychip_capture_open(struct snd_pcm_substream *substream) - { - struct mychip *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - - runtime->hw = snd_mychip_capture_hw; - /* more hardware-initialization will be done here */ - .... - return 0; - } - - /* close callback */ - static int snd_mychip_capture_close(struct snd_pcm_substream *substream) - { - struct mychip *chip = snd_pcm_substream_chip(substream); - /* the hardware-specific codes will be here */ - .... - return 0; - - } - - /* hw_params callback */ - static int snd_mychip_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) - { - return snd_pcm_lib_malloc_pages(substream, - params_buffer_bytes(hw_params)); - } - - /* hw_free callback */ - static int snd_mychip_pcm_hw_free(struct snd_pcm_substream *substream) - { - return snd_pcm_lib_free_pages(substream); - } - - /* prepare callback */ - static int snd_mychip_pcm_prepare(struct snd_pcm_substream *substream) - { - struct mychip *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - - /* set up the hardware with the current configuration - * for example... - */ - mychip_set_sample_format(chip, runtime->format); - mychip_set_sample_rate(chip, runtime->rate); - mychip_set_channels(chip, runtime->channels); - mychip_set_dma_setup(chip, runtime->dma_addr, - chip->buffer_size, - chip->period_size); - return 0; - } - - /* trigger callback */ - static int snd_mychip_pcm_trigger(struct snd_pcm_substream *substream, - int cmd) - { - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - /* do something to start the PCM engine */ - .... - break; - case SNDRV_PCM_TRIGGER_STOP: - /* do something to stop the PCM engine */ - .... - break; - default: - return -EINVAL; - } - } - - /* pointer callback */ - static snd_pcm_uframes_t - snd_mychip_pcm_pointer(struct snd_pcm_substream *substream) - { - struct mychip *chip = snd_pcm_substream_chip(substream); - unsigned int current_ptr; - - /* get the current hardware pointer */ - current_ptr = mychip_get_hw_pointer(chip); - return current_ptr; - } - - /* operators */ - static struct snd_pcm_ops snd_mychip_playback_ops = { - .open = snd_mychip_playback_open, - .close = snd_mychip_playback_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_mychip_pcm_hw_params, - .hw_free = snd_mychip_pcm_hw_free, - .prepare = snd_mychip_pcm_prepare, - .trigger = snd_mychip_pcm_trigger, - .pointer = snd_mychip_pcm_pointer, - }; - - /* operators */ - static struct snd_pcm_ops snd_mychip_capture_ops = { - .open = snd_mychip_capture_open, - .close = snd_mychip_capture_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_mychip_pcm_hw_params, - .hw_free = snd_mychip_pcm_hw_free, - .prepare = snd_mychip_pcm_prepare, - .trigger = snd_mychip_pcm_trigger, - .pointer = snd_mychip_pcm_pointer, - }; - - /* - * definitions of capture are omitted here... - */ - - /* create a pcm device */ - static int snd_mychip_new_pcm(struct mychip *chip) - { - struct snd_pcm *pcm; - int err; - - err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, &pcm); - if (err < 0) - return err; - pcm->private_data = chip; - strcpy(pcm->name, "My Chip"); - chip->pcm = pcm; - /* set operators */ - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, - &snd_mychip_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, - &snd_mychip_capture_ops); - /* pre-allocation of buffers */ - /* NOTE: this may fail */ - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), - 64*1024, 64*1024); - return 0; - } -]]> - </programlisting> - </example> - </para> - </section> - - <section id="pcm-interface-constructor"> - <title>Constructor</title> - <para> - A pcm instance is allocated by the <function>snd_pcm_new()</function> - function. It would be better to create a constructor for pcm, - namely, - - <informalexample> - <programlisting> -<![CDATA[ - static int snd_mychip_new_pcm(struct mychip *chip) - { - struct snd_pcm *pcm; - int err; - - err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, &pcm); - if (err < 0) - return err; - pcm->private_data = chip; - strcpy(pcm->name, "My Chip"); - chip->pcm = pcm; - .... - return 0; - } -]]> - </programlisting> - </informalexample> - </para> - - <para> - The <function>snd_pcm_new()</function> function takes four - arguments. The first argument is the card pointer to which this - pcm is assigned, and the second is the ID string. - </para> - - <para> - The third argument (<parameter>index</parameter>, 0 in the - above) is the index of this new pcm. It begins from zero. If - you create more than one pcm instances, specify the - different numbers in this argument. For example, - <parameter>index</parameter> = 1 for the second PCM device. - </para> - - <para> - The fourth and fifth arguments are the number of substreams - for playback and capture, respectively. Here 1 is used for - both arguments. When no playback or capture substreams are available, - pass 0 to the corresponding argument. - </para> - - <para> - If a chip supports multiple playbacks or captures, you can - specify more numbers, but they must be handled properly in - open/close, etc. callbacks. When you need to know which - substream you are referring to, then it can be obtained from - struct <structname>snd_pcm_substream</structname> data passed to each callback - as follows: - - <informalexample> - <programlisting> -<![CDATA[ - struct snd_pcm_substream *substream; - int index = substream->number; -]]> - </programlisting> - </informalexample> - </para> - - <para> - After the pcm is created, you need to set operators for each - pcm stream. - - <informalexample> - <programlisting> -<![CDATA[ - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, - &snd_mychip_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, - &snd_mychip_capture_ops); -]]> - </programlisting> - </informalexample> - </para> - - <para> - The operators are defined typically like this: - - <informalexample> - <programlisting> -<![CDATA[ - static struct snd_pcm_ops snd_mychip_playback_ops = { - .open = snd_mychip_pcm_open, - .close = snd_mychip_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_mychip_pcm_hw_params, - .hw_free = snd_mychip_pcm_hw_free, - .prepare = snd_mychip_pcm_prepare, - .trigger = snd_mychip_pcm_trigger, - .pointer = snd_mychip_pcm_pointer, - }; -]]> - </programlisting> - </informalexample> - - All the callbacks are described in the - <link linkend="pcm-interface-operators"><citetitle> - Operators</citetitle></link> subsection. - </para> - - <para> - After setting the operators, you probably will want to - pre-allocate the buffer. For the pre-allocation, simply call - the following: - - <informalexample> - <programlisting> -<![CDATA[ - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), - 64*1024, 64*1024); -]]> - </programlisting> - </informalexample> - - It will allocate a buffer up to 64kB as default. - Buffer management details will be described in the later section <link - linkend="buffer-and-memory"><citetitle>Buffer and Memory - Management</citetitle></link>. - </para> - - <para> - Additionally, you can set some extra information for this pcm - in pcm-&gt;info_flags. - The available values are defined as - <constant>SNDRV_PCM_INFO_XXX</constant> in - <filename>&lt;sound/asound.h&gt;</filename>, which is used for - the hardware definition (described later). When your soundchip - supports only half-duplex, specify like this: - - <informalexample> - <programlisting> -<![CDATA[ - pcm->info_flags = SNDRV_PCM_INFO_HALF_DUPLEX; -]]> - </programlisting> - </informalexample> - </para> - </section> - - <section id="pcm-interface-destructor"> - <title>... And the Destructor?</title> - <para> - The destructor for a pcm instance is not always - necessary. Since the pcm device will be released by the middle - layer code automatically, you don't have to call the destructor - explicitly. - </para> - - <para> - The destructor would be necessary if you created - special records internally and needed to release them. In such a - case, set the destructor function to - pcm-&gt;private_free: - - <example> - <title>PCM Instance with a Destructor</title> - <programlisting> -<![CDATA[ - static void mychip_pcm_free(struct snd_pcm *pcm) - { - struct mychip *chip = snd_pcm_chip(pcm); - /* free your own data */ - kfree(chip->my_private_pcm_data); - /* do what you like else */ - .... - } - - static int snd_mychip_new_pcm(struct mychip *chip) - { - struct snd_pcm *pcm; - .... - /* allocate your own data */ - chip->my_private_pcm_data = kmalloc(...); - /* set the destructor */ - pcm->private_data = chip; - pcm->private_free = mychip_pcm_free; - .... - } -]]> - </programlisting> - </example> - </para> - </section> - - <section id="pcm-interface-runtime"> - <title>Runtime Pointer - The Chest of PCM Information</title> - <para> - When the PCM substream is opened, a PCM runtime instance is - allocated and assigned to the substream. This pointer is - accessible via <constant>substream-&gt;runtime</constant>. - This runtime pointer holds most information you need - to control the PCM: the copy of hw_params and sw_params configurations, the buffer - pointers, mmap records, spinlocks, etc. - </para> - - <para> - The definition of runtime instance is found in - <filename>&lt;sound/pcm.h&gt;</filename>. Here are - the contents of this file: - <informalexample> - <programlisting> -<![CDATA[ -struct _snd_pcm_runtime { - /* -- Status -- */ - struct snd_pcm_substream *trigger_master; - snd_timestamp_t trigger_tstamp; /* trigger timestamp */ - int overrange; - snd_pcm_uframes_t avail_max; - snd_pcm_uframes_t hw_ptr_base; /* Position at buffer restart */ - snd_pcm_uframes_t hw_ptr_interrupt; /* Position at interrupt time*/ - - /* -- HW params -- */ - snd_pcm_access_t access; /* access mode */ - snd_pcm_format_t format; /* SNDRV_PCM_FORMAT_* */ - snd_pcm_subformat_t subformat; /* subformat */ - unsigned int rate; /* rate in Hz */ - unsigned int channels; /* channels */ - snd_pcm_uframes_t period_size; /* period size */ - unsigned int periods; /* periods */ - snd_pcm_uframes_t buffer_size; /* buffer size */ - unsigned int tick_time; /* tick time */ - snd_pcm_uframes_t min_align; /* Min alignment for the format */ - size_t byte_align; - unsigned int frame_bits; - unsigned int sample_bits; - unsigned int info; - unsigned int rate_num; - unsigned int rate_den; - - /* -- SW params -- */ - struct timespec tstamp_mode; /* mmap timestamp is updated */ - unsigned int period_step; - unsigned int sleep_min; /* min ticks to sleep */ - snd_pcm_uframes_t start_threshold; - snd_pcm_uframes_t stop_threshold; - snd_pcm_uframes_t silence_threshold; /* Silence filling happens when - noise is nearest than this */ - snd_pcm_uframes_t silence_size; /* Silence filling size */ - snd_pcm_uframes_t boundary; /* pointers wrap point */ - - snd_pcm_uframes_t silenced_start; - snd_pcm_uframes_t silenced_size; - - snd_pcm_sync_id_t sync; /* hardware synchronization ID */ - - /* -- mmap -- */ - volatile struct snd_pcm_mmap_status *status; - volatile struct snd_pcm_mmap_control *control; - atomic_t mmap_count; - - /* -- locking / scheduling -- */ - spinlock_t lock; - wait_queue_head_t sleep; - struct timer_list tick_timer; - struct fasync_struct *fasync; - - /* -- private section -- */ - void *private_data; - void (*private_free)(struct snd_pcm_runtime *runtime); - - /* -- hardware description -- */ - struct snd_pcm_hardware hw; - struct snd_pcm_hw_constraints hw_constraints; - - /* -- timer -- */ - unsigned int timer_resolution; /* timer resolution */ - - /* -- DMA -- */ - unsigned char *dma_area; /* DMA area */ - dma_addr_t dma_addr; /* physical bus address (not accessible from main CPU) */ - size_t dma_bytes; /* size of DMA area */ - - struct snd_dma_buffer *dma_buffer_p; /* allocated buffer */ - -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) - /* -- OSS things -- */ - struct snd_pcm_oss_runtime oss; -#endif -}; -]]> - </programlisting> - </informalexample> - </para> - - <para> - For the operators (callbacks) of each sound driver, most of - these records are supposed to be read-only. Only the PCM - middle-layer changes / updates them. The exceptions are - the hardware description (hw) DMA buffer information and the - private data. Besides, if you use the standard buffer allocation - method via <function>snd_pcm_lib_malloc_pages()</function>, - you don't need to set the DMA buffer information by yourself. - </para> - - <para> - In the sections below, important records are explained. - </para> - - <section id="pcm-interface-runtime-hw"> - <title>Hardware Description</title> - <para> - The hardware descriptor (struct <structname>snd_pcm_hardware</structname>) - contains the definitions of the fundamental hardware - configuration. Above all, you'll need to define this in - <link linkend="pcm-interface-operators-open-callback"><citetitle> - the open callback</citetitle></link>. - Note that the runtime instance holds the copy of the - descriptor, not the pointer to the existing descriptor. That - is, in the open callback, you can modify the copied descriptor - (<constant>runtime-&gt;hw</constant>) as you need. For example, if the maximum - number of channels is 1 only on some chip models, you can - still use the same hardware descriptor and change the - channels_max later: - <informalexample> - <programlisting> -<![CDATA[ - struct snd_pcm_runtime *runtime = substream->runtime; - ... - runtime->hw = snd_mychip_playback_hw; /* common definition */ - if (chip->model == VERY_OLD_ONE) - runtime->hw.channels_max = 1; -]]> - </programlisting> - </informalexample> - </para> - - <para> - Typically, you'll have a hardware descriptor as below: - <informalexample> - <programlisting> -<![CDATA[ - static struct snd_pcm_hardware snd_mychip_playback_hw = { - .info = (SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP_VALID), - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rates = SNDRV_PCM_RATE_8000_48000, - .rate_min = 8000, - .rate_max = 48000, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 32768, - .period_bytes_min = 4096, - .period_bytes_max = 32768, - .periods_min = 1, - .periods_max = 1024, - }; -]]> - </programlisting> - </informalexample> - </para> - - <para> - <itemizedlist> - <listitem><para> - The <structfield>info</structfield> field contains the type and - capabilities of this pcm. The bit flags are defined in - <filename>&lt;sound/asound.h&gt;</filename> as - <constant>SNDRV_PCM_INFO_XXX</constant>. Here, at least, you - have to specify whether the mmap is supported and which - interleaved format is supported. - When the hardware supports mmap, add the - <constant>SNDRV_PCM_INFO_MMAP</constant> flag here. When the - hardware supports the interleaved or the non-interleaved - formats, <constant>SNDRV_PCM_INFO_INTERLEAVED</constant> or - <constant>SNDRV_PCM_INFO_NONINTERLEAVED</constant> flag must - be set, respectively. If both are supported, you can set both, - too. - </para> - - <para> - In the above example, <constant>MMAP_VALID</constant> and - <constant>BLOCK_TRANSFER</constant> are specified for the OSS mmap - mode. Usually both are set. Of course, - <constant>MMAP_VALID</constant> is set only if the mmap is - really supported. - </para> - - <para> - The other possible flags are - <constant>SNDRV_PCM_INFO_PAUSE</constant> and - <constant>SNDRV_PCM_INFO_RESUME</constant>. The - <constant>PAUSE</constant> bit means that the pcm supports the - <quote>pause</quote> operation, while the - <constant>RESUME</constant> bit means that the pcm supports - the full <quote>suspend/resume</quote> operation. - If the <constant>PAUSE</constant> flag is set, - the <structfield>trigger</structfield> callback below - must handle the corresponding (pause push/release) commands. - The suspend/resume trigger commands can be defined even without - the <constant>RESUME</constant> flag. See <link - linkend="power-management"><citetitle> - Power Management</citetitle></link> section for details. - </para> - - <para> - When the PCM substreams can be synchronized (typically, - synchronized start/stop of a playback and a capture streams), - you can give <constant>SNDRV_PCM_INFO_SYNC_START</constant>, - too. In this case, you'll need to check the linked-list of - PCM substreams in the trigger callback. This will be - described in the later section. - </para> - </listitem> - - <listitem> - <para> - <structfield>formats</structfield> field contains the bit-flags - of supported formats (<constant>SNDRV_PCM_FMTBIT_XXX</constant>). - If the hardware supports more than one format, give all or'ed - bits. In the example above, the signed 16bit little-endian - format is specified. - </para> - </listitem> - - <listitem> - <para> - <structfield>rates</structfield> field contains the bit-flags of - supported rates (<constant>SNDRV_PCM_RATE_XXX</constant>). - When the chip supports continuous rates, pass - <constant>CONTINUOUS</constant> bit additionally. - The pre-defined rate bits are provided only for typical - rates. If your chip supports unconventional rates, you need to add - the <constant>KNOT</constant> bit and set up the hardware - constraint manually (explained later). - </para> - </listitem> - - <listitem> - <para> - <structfield>rate_min</structfield> and - <structfield>rate_max</structfield> define the minimum and - maximum sample rate. This should correspond somehow to - <structfield>rates</structfield> bits. - </para> - </listitem> - - <listitem> - <para> - <structfield>channel_min</structfield> and - <structfield>channel_max</structfield> - define, as you might already expected, the minimum and maximum - number of channels. - </para> - </listitem> - - <listitem> - <para> - <structfield>buffer_bytes_max</structfield> defines the - maximum buffer size in bytes. There is no - <structfield>buffer_bytes_min</structfield> field, since - it can be calculated from the minimum period size and the - minimum number of periods. - Meanwhile, <structfield>period_bytes_min</structfield> and - define the minimum and maximum size of the period in bytes. - <structfield>periods_max</structfield> and - <structfield>periods_min</structfield> define the maximum and - minimum number of periods in the buffer. - </para> - - <para> - The <quote>period</quote> is a term that corresponds to - a fragment in the OSS world. The period defines the size at - which a PCM interrupt is generated. This size strongly - depends on the hardware. - Generally, the smaller period size will give you more - interrupts, that is, more controls. - In the case of capture, this size defines the input latency. - On the other hand, the whole buffer size defines the - output latency for the playback direction. - </para> - </listitem> - - <listitem> - <para> - There is also a field <structfield>fifo_size</structfield>. - This specifies the size of the hardware FIFO, but currently it - is neither used in the driver nor in the alsa-lib. So, you - can ignore this field. - </para> - </listitem> - </itemizedlist> - </para> - </section> - - <section id="pcm-interface-runtime-config"> - <title>PCM Configurations</title> - <para> - Ok, let's go back again to the PCM runtime records. - The most frequently referred records in the runtime instance are - the PCM configurations. - The PCM configurations are stored in the runtime instance - after the application sends <type>hw_params</type> data via - alsa-lib. There are many fields copied from hw_params and - sw_params structs. For example, - <structfield>format</structfield> holds the format type - chosen by the application. This field contains the enum value - <constant>SNDRV_PCM_FORMAT_XXX</constant>. - </para> - - <para> - One thing to be noted is that the configured buffer and period - sizes are stored in <quote>frames</quote> in the runtime. - In the ALSA world, 1 frame = channels * samples-size. - For conversion between frames and bytes, you can use the - <function>frames_to_bytes()</function> and - <function>bytes_to_frames()</function> helper functions. - <informalexample> - <programlisting> -<![CDATA[ - period_bytes = frames_to_bytes(runtime, runtime->period_size); -]]> - </programlisting> - </informalexample> - </para> - - <para> - Also, many software parameters (sw_params) are - stored in frames, too. Please check the type of the field. - <type>snd_pcm_uframes_t</type> is for the frames as unsigned - integer while <type>snd_pcm_sframes_t</type> is for the frames - as signed integer. - </para> - </section> - - <section id="pcm-interface-runtime-dma"> - <title>DMA Buffer Information</title> - <para> - The DMA buffer is defined by the following four fields, - <structfield>dma_area</structfield>, - <structfield>dma_addr</structfield>, - <structfield>dma_bytes</structfield> and - <structfield>dma_private</structfield>. - The <structfield>dma_area</structfield> holds the buffer - pointer (the logical address). You can call - <function>memcpy</function> from/to - this pointer. Meanwhile, <structfield>dma_addr</structfield> - holds the physical address of the buffer. This field is - specified only when the buffer is a linear buffer. - <structfield>dma_bytes</structfield> holds the size of buffer - in bytes. <structfield>dma_private</structfield> is used for - the ALSA DMA allocator. - </para> - - <para> - If you use a standard ALSA function, - <function>snd_pcm_lib_malloc_pages()</function>, for - allocating the buffer, these fields are set by the ALSA middle - layer, and you should <emphasis>not</emphasis> change them by - yourself. You can read them but not write them. - On the other hand, if you want to allocate the buffer by - yourself, you'll need to manage it in hw_params callback. - At least, <structfield>dma_bytes</structfield> is mandatory. - <structfield>dma_area</structfield> is necessary when the - buffer is mmapped. If your driver doesn't support mmap, this - field is not necessary. <structfield>dma_addr</structfield> - is also optional. You can use - <structfield>dma_private</structfield> as you like, too. - </para> - </section> - - <section id="pcm-interface-runtime-status"> - <title>Running Status</title> - <para> - The running status can be referred via <constant>runtime-&gt;status</constant>. - This is the pointer to the struct <structname>snd_pcm_mmap_status</structname> - record. For example, you can get the current DMA hardware - pointer via <constant>runtime-&gt;status-&gt;hw_ptr</constant>. - </para> - - <para> - The DMA application pointer can be referred via - <constant>runtime-&gt;control</constant>, which points to the - struct <structname>snd_pcm_mmap_control</structname> record. - However, accessing directly to this value is not recommended. - </para> - </section> - - <section id="pcm-interface-runtime-private"> - <title>Private Data</title> - <para> - You can allocate a record for the substream and store it in - <constant>runtime-&gt;private_data</constant>. Usually, this - is done in - <link linkend="pcm-interface-operators-open-callback"><citetitle> - the open callback</citetitle></link>. - Don't mix this with <constant>pcm-&gt;private_data</constant>. - The <constant>pcm-&gt;private_data</constant> usually points to the - chip instance assigned statically at the creation of PCM, while the - <constant>runtime-&gt;private_data</constant> points to a dynamic - data structure created at the PCM open callback. - - <informalexample> - <programlisting> -<![CDATA[ - static int snd_xxx_open(struct snd_pcm_substream *substream) - { - struct my_pcm_data *data; - .... - data = kmalloc(sizeof(*data), GFP_KERNEL); - substream->runtime->private_data = data; - .... - } -]]> - </programlisting> - </informalexample> - </para> - - <para> - The allocated object must be released in - <link linkend="pcm-interface-operators-open-callback"><citetitle> - the close callback</citetitle></link>. - </para> - </section> - - </section> - - <section id="pcm-interface-operators"> - <title>Operators</title> - <para> - OK, now let me give details about each pcm callback - (<parameter>ops</parameter>). In general, every callback must - return 0 if successful, or a negative error number - such as <constant>-EINVAL</constant>. To choose an appropriate - error number, it is advised to check what value other parts of - the kernel return when the same kind of request fails. - </para> - - <para> - The callback function takes at least the argument with - <structname>snd_pcm_substream</structname> pointer. To retrieve - the chip record from the given substream instance, you can use the - following macro. - - <informalexample> - <programlisting> -<![CDATA[ - int xxx() { - struct mychip *chip = snd_pcm_substream_chip(substream); - .... - } -]]> - </programlisting> - </informalexample> - - The macro reads <constant>substream-&gt;private_data</constant>, - which is a copy of <constant>pcm-&gt;private_data</constant>. - You can override the former if you need to assign different data - records per PCM substream. For example, the cmi8330 driver assigns - different private_data for playback and capture directions, - because it uses two different codecs (SB- and AD-compatible) for - different directions. - </para> - - <section id="pcm-interface-operators-open-callback"> - <title>open callback</title> - <para> - <informalexample> - <programlisting> -<![CDATA[ - static int snd_xxx_open(struct snd_pcm_substream *substream); -]]> - </programlisting> - </informalexample> - - This is called when a pcm substream is opened. - </para> - - <para> - At least, here you have to initialize the runtime-&gt;hw - record. Typically, this is done by like this: - - <informalexample> - <programlisting> -<![CDATA[ - static int snd_xxx_open(struct snd_pcm_substream *substream) - { - struct mychip *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - - runtime->hw = snd_mychip_playback_hw; - return 0; - } -]]> - </programlisting> - </informalexample> - - where <parameter>snd_mychip_playback_hw</parameter> is the - pre-defined hardware description. - </para> - - <para> - You can allocate a private data in this callback, as described - in <link linkend="pcm-interface-runtime-private"><citetitle> - Private Data</citetitle></link> section. - </para> - - <para> - If the hardware configuration needs more constraints, set the - hardware constraints here, too. - See <link linkend="pcm-interface-constraints"><citetitle> - Constraints</citetitle></link> for more details. - </para> - </section> - - <section id="pcm-interface-operators-close-callback"> - <title>close callback</title> - <para> - <informalexample> - <programlisting> -<![CDATA[ - static int snd_xxx_close(struct snd_pcm_substream *substream); -]]> - </programlisting> - </informalexample> - - Obviously, this is called when a pcm substream is closed. - </para> - - <para> - Any private instance for a pcm substream allocated in the - open callback will be released here. - - <informalexample> - <programlisting> -<![CDATA[ - static int snd_xxx_close(struct snd_pcm_substream *substream) - { - .... - kfree(substream->runtime->private_data); - .... - } -]]> - </programlisting> - </informalexample> - </para> - </section> - - <section id="pcm-interface-operators-ioctl-callback"> - <title>ioctl callback</title> - <para> - This is used for any special call to pcm ioctls. But - usually you can pass a generic ioctl callback, - <function>snd_pcm_lib_ioctl</function>. - </para> - </section> - - <section id="pcm-interface-operators-hw-params-callback"> - <title>hw_params callback</title> - <para> - <informalexample> - <programlisting> -<![CDATA[ - static int snd_xxx_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params); -]]> - </programlisting> - </informalexample> - </para> - - <para> - This is called when the hardware parameter - (<structfield>hw_params</structfield>) is set - up by the application, - that is, once when the buffer size, the period size, the - format, etc. are defined for the pcm substream. - </para> - - <para> - Many hardware setups should be done in this callback, - including the allocation of buffers. - </para> - - <para> - Parameters to be initialized are retrieved by - <function>params_xxx()</function> macros. To allocate - buffer, you can call a helper function, - - <informalexample> - <programlisting> -<![CDATA[ - snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); -]]> - </programlisting> - </informalexample> - - <function>snd_pcm_lib_malloc_pages()</function> is available - only when the DMA buffers have been pre-allocated. - See the section <link - linkend="buffer-and-memory-buffer-types"><citetitle> - Buffer Types</citetitle></link> for more details. - </para> - - <para> - Note that this and <structfield>prepare</structfield> callbacks - may be called multiple times per initialization. - For example, the OSS emulation may - call these callbacks at each change via its ioctl. - </para> - - <para> - Thus, you need to be careful not to allocate the same buffers - many times, which will lead to memory leaks! Calling the - helper function above many times is OK. It will release the - previous buffer automatically when it was already allocated. - </para> - - <para> - Another note is that this callback is non-atomic - (schedulable) as default, i.e. when no - <structfield>nonatomic</structfield> flag set. - This is important, because the - <structfield>trigger</structfield> callback - is atomic (non-schedulable). That is, mutexes or any - schedule-related functions are not available in - <structfield>trigger</structfield> callback. - Please see the subsection - <link linkend="pcm-interface-atomicity"><citetitle> - Atomicity</citetitle></link> for details. - </para> - </section> - - <section id="pcm-interface-operators-hw-free-callback"> - <title>hw_free callback</title> - <para> - <informalexample> - <programlisting> -<![CDATA[ - static int snd_xxx_hw_free(struct snd_pcm_substream *substream); -]]> - </programlisting> - </informalexample> - </para> - - <para> - This is called to release the resources allocated via - <structfield>hw_params</structfield>. For example, releasing the - buffer via - <function>snd_pcm_lib_malloc_pages()</function> is done by - calling the following: - - <informalexample> - <programlisting> -<![CDATA[ - snd_pcm_lib_free_pages(substream); -]]> - </programlisting> - </informalexample> - </para> - - <para> - This function is always called before the close callback is called. - Also, the callback may be called multiple times, too. - Keep track whether the resource was already released. - </para> - </section> - - <section id="pcm-interface-operators-prepare-callback"> - <title>prepare callback</title> - <para> - <informalexample> - <programlisting> -<![CDATA[ - static int snd_xxx_prepare(struct snd_pcm_substream *substream); -]]> - </programlisting> - </informalexample> - </para> - - <para> - This callback is called when the pcm is - <quote>prepared</quote>. You can set the format type, sample - rate, etc. here. The difference from - <structfield>hw_params</structfield> is that the - <structfield>prepare</structfield> callback will be called each - time - <function>snd_pcm_prepare()</function> is called, i.e. when - recovering after underruns, etc. - </para> - - <para> - Note that this callback is now non-atomic. - You can use schedule-related functions safely in this callback. - </para> - - <para> - In this and the following callbacks, you can refer to the - values via the runtime record, - substream-&gt;runtime. - For example, to get the current - rate, format or channels, access to - runtime-&gt;rate, - runtime-&gt;format or - runtime-&gt;channels, respectively. - The physical address of the allocated buffer is set to - runtime-&gt;dma_area. The buffer and period sizes are - in runtime-&gt;buffer_size and runtime-&gt;period_size, - respectively. - </para> - - <para> - Be careful that this callback will be called many times at - each setup, too. - </para> - </section> - - <section id="pcm-interface-operators-trigger-callback"> - <title>trigger callback</title> - <para> - <informalexample> - <programlisting> -<![CDATA[ - static int snd_xxx_trigger(struct snd_pcm_substream *substream, int cmd); -]]> - </programlisting> - </informalexample> - - This is called when the pcm is started, stopped or paused. - </para> - - <para> - Which action is specified in the second argument, - <constant>SNDRV_PCM_TRIGGER_XXX</constant> in - <filename>&lt;sound/pcm.h&gt;</filename>. At least, - the <constant>START</constant> and <constant>STOP</constant> - commands must be defined in this callback. - - <informalexample> - <programlisting> -<![CDATA[ - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - /* do something to start the PCM engine */ - break; - case SNDRV_PCM_TRIGGER_STOP: - /* do something to stop the PCM engine */ - break; - default: - return -EINVAL; - } -]]> - </programlisting> - </informalexample> - </para> - - <para> - When the pcm supports the pause operation (given in the info - field of the hardware table), the <constant>PAUSE_PUSH</constant> - and <constant>PAUSE_RELEASE</constant> commands must be - handled here, too. The former is the command to pause the pcm, - and the latter to restart the pcm again. - </para> - - <para> - When the pcm supports the suspend/resume operation, - regardless of full or partial suspend/resume support, - the <constant>SUSPEND</constant> and <constant>RESUME</constant> - commands must be handled, too. - These commands are issued when the power-management status is - changed. Obviously, the <constant>SUSPEND</constant> and - <constant>RESUME</constant> commands - suspend and resume the pcm substream, and usually, they - are identical to the <constant>STOP</constant> and - <constant>START</constant> commands, respectively. - See the <link linkend="power-management"><citetitle> - Power Management</citetitle></link> section for details. - </para> - - <para> - As mentioned, this callback is atomic as default unless - <structfield>nonatomic</structfield> flag set, and - you cannot call functions which may sleep. - The trigger callback should be as minimal as possible, - just really triggering the DMA. The other stuff should be - initialized hw_params and prepare callbacks properly - beforehand. - </para> - </section> - - <section id="pcm-interface-operators-pointer-callback"> - <title>pointer callback</title> - <para> - <informalexample> - <programlisting> -<![CDATA[ - static snd_pcm_uframes_t snd_xxx_pointer(struct snd_pcm_substream *substream) -]]> - </programlisting> - </informalexample> - - This callback is called when the PCM middle layer inquires - the current hardware position on the buffer. The position must - be returned in frames, - ranging from 0 to buffer_size - 1. - </para> - - <para> - This is called usually from the buffer-update routine in the - pcm middle layer, which is invoked when - <function>snd_pcm_period_elapsed()</function> is called in the - interrupt routine. Then the pcm middle layer updates the - position and calculates the available space, and wakes up the - sleeping poll threads, etc. - </para> - - <para> - This callback is also atomic as default. - </para> - </section> - - <section id="pcm-interface-operators-copy-silence"> - <title>copy and silence callbacks</title> - <para> - These callbacks are not mandatory, and can be omitted in - most cases. These callbacks are used when the hardware buffer - cannot be in the normal memory space. Some chips have their - own buffer on the hardware which is not mappable. In such a - case, you have to transfer the data manually from the memory - buffer to the hardware buffer. Or, if the buffer is - non-contiguous on both physical and virtual memory spaces, - these callbacks must be defined, too. - </para> - - <para> - If these two callbacks are defined, copy and set-silence - operations are done by them. The detailed will be described in - the later section <link - linkend="buffer-and-memory"><citetitle>Buffer and Memory - Management</citetitle></link>. - </para> - </section> - - <section id="pcm-interface-operators-ack"> - <title>ack callback</title> - <para> - This callback is also not mandatory. This callback is called - when the appl_ptr is updated in read or write operations. - Some drivers like emu10k1-fx and cs46xx need to track the - current appl_ptr for the internal buffer, and this callback - is useful only for such a purpose. - </para> - <para> - This callback is atomic as default. - </para> - </section> - - <section id="pcm-interface-operators-page-callback"> - <title>page callback</title> - - <para> - This callback is optional too. This callback is used - mainly for non-contiguous buffers. The mmap calls this - callback to get the page address. Some examples will be - explained in the later section <link - linkend="buffer-and-memory"><citetitle>Buffer and Memory - Management</citetitle></link>, too. - </para> - </section> - </section> - - <section id="pcm-interface-interrupt-handler"> - <title>Interrupt Handler</title> - <para> - The rest of pcm stuff is the PCM interrupt handler. The - role of PCM interrupt handler in the sound driver is to update - the buffer position and to tell the PCM middle layer when the - buffer position goes across the prescribed period size. To - inform this, call the <function>snd_pcm_period_elapsed()</function> - function. - </para> - - <para> - There are several types of sound chips to generate the interrupts. - </para> - - <section id="pcm-interface-interrupt-handler-boundary"> - <title>Interrupts at the period (fragment) boundary</title> - <para> - This is the most frequently found type: the hardware - generates an interrupt at each period boundary. - In this case, you can call - <function>snd_pcm_period_elapsed()</function> at each - interrupt. - </para> - - <para> - <function>snd_pcm_period_elapsed()</function> takes the - substream pointer as its argument. Thus, you need to keep the - substream pointer accessible from the chip instance. For - example, define substream field in the chip record to hold the - current running substream pointer, and set the pointer value - at open callback (and reset at close callback). - </para> - - <para> - If you acquire a spinlock in the interrupt handler, and the - lock is used in other pcm callbacks, too, then you have to - release the lock before calling - <function>snd_pcm_period_elapsed()</function>, because - <function>snd_pcm_period_elapsed()</function> calls other pcm - callbacks inside. - </para> - - <para> - Typical code would be like: - - <example> - <title>Interrupt Handler Case #1</title> - <programlisting> -<![CDATA[ - static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id) - { - struct mychip *chip = dev_id; - spin_lock(&chip->lock); - .... - if (pcm_irq_invoked(chip)) { - /* call updater, unlock before it */ - spin_unlock(&chip->lock); - snd_pcm_period_elapsed(chip->substream); - spin_lock(&chip->lock); - /* acknowledge the interrupt if necessary */ - } - .... - spin_unlock(&chip->lock); - return IRQ_HANDLED; - } -]]> - </programlisting> - </example> - </para> - </section> - - <section id="pcm-interface-interrupt-handler-timer"> - <title>High frequency timer interrupts</title> - <para> - This happens when the hardware doesn't generate interrupts - at the period boundary but issues timer interrupts at a fixed - timer rate (e.g. es1968 or ymfpci drivers). - In this case, you need to check the current hardware - position and accumulate the processed sample length at each - interrupt. When the accumulated size exceeds the period - size, call - <function>snd_pcm_period_elapsed()</function> and reset the - accumulator. - </para> - - <para> - Typical code would be like the following. - - <example> - <title>Interrupt Handler Case #2</title> - <programlisting> -<![CDATA[ - static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id) - { - struct mychip *chip = dev_id; - spin_lock(&chip->lock); - .... - if (pcm_irq_invoked(chip)) { - unsigned int last_ptr, size; - /* get the current hardware pointer (in frames) */ - last_ptr = get_hw_ptr(chip); - /* calculate the processed frames since the - * last update - */ - if (last_ptr < chip->last_ptr) - size = runtime->buffer_size + last_ptr - - chip->last_ptr; - else - size = last_ptr - chip->last_ptr; - /* remember the last updated point */ - chip->last_ptr = last_ptr; - /* accumulate the size */ - chip->size += size; - /* over the period boundary? */ - if (chip->size >= runtime->period_size) { - /* reset the accumulator */ - chip->size %= runtime->period_size; - /* call updater */ - spin_unlock(&chip->lock); - snd_pcm_period_elapsed(substream); - spin_lock(&chip->lock); - } - /* acknowledge the interrupt if necessary */ - } - .... - spin_unlock(&chip->lock); - return IRQ_HANDLED; - } -]]> - </programlisting> - </example> - </para> - </section> - - <section id="pcm-interface-interrupt-handler-both"> - <title>On calling <function>snd_pcm_period_elapsed()</function></title> - <para> - In both cases, even if more than one period are elapsed, you - don't have to call - <function>snd_pcm_period_elapsed()</function> many times. Call - only once. And the pcm layer will check the current hardware - pointer and update to the latest status. - </para> - </section> - </section> - - <section id="pcm-interface-atomicity"> - <title>Atomicity</title> - <para> - One of the most important (and thus difficult to debug) problems - in kernel programming are race conditions. - In the Linux kernel, they are usually avoided via spin-locks, mutexes - or semaphores. In general, if a race condition can happen - in an interrupt handler, it has to be managed atomically, and you - have to use a spinlock to protect the critical session. If the - critical section is not in interrupt handler code and - if taking a relatively long time to execute is acceptable, you - should use mutexes or semaphores instead. - </para> - - <para> - As already seen, some pcm callbacks are atomic and some are - not. For example, the <parameter>hw_params</parameter> callback is - non-atomic, while <parameter>trigger</parameter> callback is - atomic. This means, the latter is called already in a spinlock - held by the PCM middle layer. Please take this atomicity into - account when you choose a locking scheme in the callbacks. - </para> - - <para> - In the atomic callbacks, you cannot use functions which may call - <function>schedule</function> or go to - <function>sleep</function>. Semaphores and mutexes can sleep, - and hence they cannot be used inside the atomic callbacks - (e.g. <parameter>trigger</parameter> callback). - To implement some delay in such a callback, please use - <function>udelay()</function> or <function>mdelay()</function>. - </para> - - <para> - All three atomic callbacks (trigger, pointer, and ack) are - called with local interrupts disabled. - </para> - - <para> - The recent changes in PCM core code, however, allow all PCM - operations to be non-atomic. This assumes that the all caller - sides are in non-atomic contexts. For example, the function - <function>snd_pcm_period_elapsed()</function> is called - typically from the interrupt handler. But, if you set up the - driver to use a threaded interrupt handler, this call can be in - non-atomic context, too. In such a case, you can set - <structfield>nonatomic</structfield> filed of - <structname>snd_pcm</structname> object after creating it. - When this flag is set, mutex and rwsem are used internally in - the PCM core instead of spin and rwlocks, so that you can call - all PCM functions safely in a non-atomic context. - </para> - - </section> - <section id="pcm-interface-constraints"> - <title>Constraints</title> - <para> - If your chip supports unconventional sample rates, or only the - limited samples, you need to set a constraint for the - condition. - </para> - - <para> - For example, in order to restrict the sample rates in the some - supported values, use - <function>snd_pcm_hw_constraint_list()</function>. - You need to call this function in the open callback. - - <example> - <title>Example of Hardware Constraints</title> - <programlisting> -<![CDATA[ - static unsigned int rates[] = - {4000, 10000, 22050, 44100}; - static struct snd_pcm_hw_constraint_list constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, - }; - - static int snd_mychip_pcm_open(struct snd_pcm_substream *substream) - { - int err; - .... - err = snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &constraints_rates); - if (err < 0) - return err; - .... - } -]]> - </programlisting> - </example> - </para> - - <para> - There are many different constraints. - Look at <filename>sound/pcm.h</filename> for a complete list. - You can even define your own constraint rules. - For example, let's suppose my_chip can manage a substream of 1 channel - if and only if the format is S16_LE, otherwise it supports any format - specified in the <structname>snd_pcm_hardware</structname> structure (or in any - other constraint_list). You can build a rule like this: - - <example> - <title>Example of Hardware Constraints for Channels</title> - <programlisting> -<![CDATA[ - static int hw_rule_channels_by_format(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) - { - struct snd_interval *c = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_interval ch; - - snd_interval_any(&ch); - if (f->bits[0] == SNDRV_PCM_FMTBIT_S16_LE) { - ch.min = ch.max = 1; - ch.integer = 1; - return snd_interval_refine(c, &ch); - } - return 0; - } -]]> - </programlisting> - </example> - </para> - - <para> - Then you need to call this function to add your rule: - - <informalexample> - <programlisting> -<![CDATA[ - snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - hw_rule_channels_by_format, NULL, - SNDRV_PCM_HW_PARAM_FORMAT, -1); -]]> - </programlisting> - </informalexample> - </para> - - <para> - The rule function is called when an application sets the PCM - format, and it refines the number of channels accordingly. - But an application may set the number of channels before - setting the format. Thus you also need to define the inverse rule: - - <example> - <title>Example of Hardware Constraints for Formats</title> - <programlisting> -<![CDATA[ - static int hw_rule_format_by_channels(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) - { - struct snd_interval *c = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_mask fmt; - - snd_mask_any(&fmt); /* Init the struct */ - if (c->min < 2) { - fmt.bits[0] &= SNDRV_PCM_FMTBIT_S16_LE; - return snd_mask_refine(f, &fmt); - } - return 0; - } -]]> - </programlisting> - </example> - </para> - - <para> - ...and in the open callback: - <informalexample> - <programlisting> -<![CDATA[ - snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, - hw_rule_format_by_channels, NULL, - SNDRV_PCM_HW_PARAM_CHANNELS, -1); -]]> - </programlisting> - </informalexample> - </para> - - <para> - I won't give more details here, rather I - would like to say, <quote>Luke, use the source.</quote> - </para> - </section> - - </chapter> - - -<!-- ****************************************************** --> -<!-- Control Interface --> -<!-- ****************************************************** --> - <chapter id="control-interface"> - <title>Control Interface</title> - - <section id="control-interface-general"> - <title>General</title> - <para> - The control interface is used widely for many switches, - sliders, etc. which are accessed from user-space. Its most - important use is the mixer interface. In other words, since ALSA - 0.9.x, all the mixer stuff is implemented on the control kernel API. - </para> - - <para> - ALSA has a well-defined AC97 control module. If your chip - supports only the AC97 and nothing else, you can skip this - section. - </para> - - <para> - The control API is defined in - <filename>&lt;sound/control.h&gt;</filename>. - Include this file if you want to add your own controls. - </para> - </section> - - <section id="control-interface-definition"> - <title>Definition of Controls</title> - <para> - To create a new control, you need to define the - following three - callbacks: <structfield>info</structfield>, - <structfield>get</structfield> and - <structfield>put</structfield>. Then, define a - struct <structname>snd_kcontrol_new</structname> record, such as: - - <example> - <title>Definition of a Control</title> - <programlisting> -<![CDATA[ - static struct snd_kcontrol_new my_control = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "PCM Playback Switch", - .index = 0, - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, - .private_value = 0xffff, - .info = my_control_info, - .get = my_control_get, - .put = my_control_put - }; -]]> - </programlisting> - </example> - </para> - - <para> - The <structfield>iface</structfield> field specifies the control - type, <constant>SNDRV_CTL_ELEM_IFACE_XXX</constant>, which - is usually <constant>MIXER</constant>. - Use <constant>CARD</constant> for global controls that are not - logically part of the mixer. - If the control is closely associated with some specific device on - the sound card, use <constant>HWDEP</constant>, - <constant>PCM</constant>, <constant>RAWMIDI</constant>, - <constant>TIMER</constant>, or <constant>SEQUENCER</constant>, and - specify the device number with the - <structfield>device</structfield> and - <structfield>subdevice</structfield> fields. - </para> - - <para> - The <structfield>name</structfield> is the name identifier - string. Since ALSA 0.9.x, the control name is very important, - because its role is classified from its name. There are - pre-defined standard control names. The details are described in - the <link linkend="control-interface-control-names"><citetitle> - Control Names</citetitle></link> subsection. - </para> - - <para> - The <structfield>index</structfield> field holds the index number - of this control. If there are several different controls with - the same name, they can be distinguished by the index - number. This is the case when - several codecs exist on the card. If the index is zero, you can - omit the definition above. - </para> - - <para> - The <structfield>access</structfield> field contains the access - type of this control. Give the combination of bit masks, - <constant>SNDRV_CTL_ELEM_ACCESS_XXX</constant>, there. - The details will be explained in - the <link linkend="control-interface-access-flags"><citetitle> - Access Flags</citetitle></link> subsection. - </para> - - <para> - The <structfield>private_value</structfield> field contains - an arbitrary long integer value for this record. When using - the generic <structfield>info</structfield>, - <structfield>get</structfield> and - <structfield>put</structfield> callbacks, you can pass a value - through this field. If several small numbers are necessary, you can - combine them in bitwise. Or, it's possible to give a pointer - (casted to unsigned long) of some record to this field, too. - </para> - - <para> - The <structfield>tlv</structfield> field can be used to provide - metadata about the control; see the - <link linkend="control-interface-tlv"> - <citetitle>Metadata</citetitle></link> subsection. - </para> - - <para> - The other three are - <link linkend="control-interface-callbacks"><citetitle> - callback functions</citetitle></link>. - </para> - </section> - - <section id="control-interface-control-names"> - <title>Control Names</title> - <para> - There are some standards to define the control names. A - control is usually defined from the three parts as - <quote>SOURCE DIRECTION FUNCTION</quote>. - </para> - - <para> - The first, <constant>SOURCE</constant>, specifies the source - of the control, and is a string such as <quote>Master</quote>, - <quote>PCM</quote>, <quote>CD</quote> and - <quote>Line</quote>. There are many pre-defined sources. - </para> - - <para> - The second, <constant>DIRECTION</constant>, is one of the - following strings according to the direction of the control: - <quote>Playback</quote>, <quote>Capture</quote>, <quote>Bypass - Playback</quote> and <quote>Bypass Capture</quote>. Or, it can - be omitted, meaning both playback and capture directions. - </para> - - <para> - The third, <constant>FUNCTION</constant>, is one of the - following strings according to the function of the control: - <quote>Switch</quote>, <quote>Volume</quote> and - <quote>Route</quote>. - </para> - - <para> - The example of control names are, thus, <quote>Master Capture - Switch</quote> or <quote>PCM Playback Volume</quote>. - </para> - - <para> - There are some exceptions: - </para> - - <section id="control-interface-control-names-global"> - <title>Global capture and playback</title> - <para> - <quote>Capture Source</quote>, <quote>Capture Switch</quote> - and <quote>Capture Volume</quote> are used for the global - capture (input) source, switch and volume. Similarly, - <quote>Playback Switch</quote> and <quote>Playback - Volume</quote> are used for the global output gain switch and - volume. - </para> - </section> - - <section id="control-interface-control-names-tone"> - <title>Tone-controls</title> - <para> - tone-control switch and volumes are specified like - <quote>Tone Control - XXX</quote>, e.g. <quote>Tone Control - - Switch</quote>, <quote>Tone Control - Bass</quote>, - <quote>Tone Control - Center</quote>. - </para> - </section> - - <section id="control-interface-control-names-3d"> - <title>3D controls</title> - <para> - 3D-control switches and volumes are specified like <quote>3D - Control - XXX</quote>, e.g. <quote>3D Control - - Switch</quote>, <quote>3D Control - Center</quote>, <quote>3D - Control - Space</quote>. - </para> - </section> - - <section id="control-interface-control-names-mic"> - <title>Mic boost</title> - <para> - Mic-boost switch is set as <quote>Mic Boost</quote> or - <quote>Mic Boost (6dB)</quote>. - </para> - - <para> - More precise information can be found in - <filename>Documentation/sound/alsa/ControlNames.txt</filename>. - </para> - </section> - </section> - - <section id="control-interface-access-flags"> - <title>Access Flags</title> - - <para> - The access flag is the bitmask which specifies the access type - of the given control. The default access type is - <constant>SNDRV_CTL_ELEM_ACCESS_READWRITE</constant>, - which means both read and write are allowed to this control. - When the access flag is omitted (i.e. = 0), it is - considered as <constant>READWRITE</constant> access as default. - </para> - - <para> - When the control is read-only, pass - <constant>SNDRV_CTL_ELEM_ACCESS_READ</constant> instead. - In this case, you don't have to define - the <structfield>put</structfield> callback. - Similarly, when the control is write-only (although it's a rare - case), you can use the <constant>WRITE</constant> flag instead, and - you don't need the <structfield>get</structfield> callback. - </para> - - <para> - If the control value changes frequently (e.g. the VU meter), - <constant>VOLATILE</constant> flag should be given. This means - that the control may be changed without - <link linkend="control-interface-change-notification"><citetitle> - notification</citetitle></link>. Applications should poll such - a control constantly. - </para> - - <para> - When the control is inactive, set - the <constant>INACTIVE</constant> flag, too. - There are <constant>LOCK</constant> and - <constant>OWNER</constant> flags to change the write - permissions. - </para> - - </section> - - <section id="control-interface-callbacks"> - <title>Callbacks</title> - - <section id="control-interface-callbacks-info"> - <title>info callback</title> - <para> - The <structfield>info</structfield> callback is used to get - detailed information on this control. This must store the - values of the given struct <structname>snd_ctl_elem_info</structname> - object. For example, for a boolean control with a single - element: - - <example> - <title>Example of info callback</title> - <programlisting> -<![CDATA[ - static int snd_myctl_mono_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) - { - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; - } -]]> - </programlisting> - </example> - </para> - - <para> - The <structfield>type</structfield> field specifies the type - of the control. There are <constant>BOOLEAN</constant>, - <constant>INTEGER</constant>, <constant>ENUMERATED</constant>, - <constant>BYTES</constant>, <constant>IEC958</constant> and - <constant>INTEGER64</constant>. The - <structfield>count</structfield> field specifies the - number of elements in this control. For example, a stereo - volume would have count = 2. The - <structfield>value</structfield> field is a union, and - the values stored are depending on the type. The boolean and - integer types are identical. - </para> - - <para> - The enumerated type is a bit different from others. You'll - need to set the string for the currently given item index. - - <informalexample> - <programlisting> -<![CDATA[ - static int snd_myctl_enum_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) - { - static char *texts[4] = { - "First", "Second", "Third", "Fourth" - }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item > 3) - uinfo->value.enumerated.item = 3; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - return 0; - } -]]> - </programlisting> - </informalexample> - </para> - - <para> - The above callback can be simplified with a helper function, - <function>snd_ctl_enum_info</function>. The final code - looks like below. - (You can pass ARRAY_SIZE(texts) instead of 4 in the third - argument; it's a matter of taste.) - - <informalexample> - <programlisting> -<![CDATA[ - static int snd_myctl_enum_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) - { - static char *texts[4] = { - "First", "Second", "Third", "Fourth" - }; - return snd_ctl_enum_info(uinfo, 1, 4, texts); - } -]]> - </programlisting> - </informalexample> - </para> - - <para> - Some common info callbacks are available for your convenience: - <function>snd_ctl_boolean_mono_info()</function> and - <function>snd_ctl_boolean_stereo_info()</function>. - Obviously, the former is an info callback for a mono channel - boolean item, just like <function>snd_myctl_mono_info</function> - above, and the latter is for a stereo channel boolean item. - </para> - - </section> - - <section id="control-interface-callbacks-get"> - <title>get callback</title> - - <para> - This callback is used to read the current value of the - control and to return to user-space. - </para> - - <para> - For example, - - <example> - <title>Example of get callback</title> - <programlisting> -<![CDATA[ - static int snd_myctl_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) - { - struct mychip *chip = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = get_some_value(chip); - return 0; - } -]]> - </programlisting> - </example> - </para> - - <para> - The <structfield>value</structfield> field depends on - the type of control as well as on the info callback. For example, - the sb driver uses this field to store the register offset, - the bit-shift and the bit-mask. The - <structfield>private_value</structfield> field is set as follows: - <informalexample> - <programlisting> -<![CDATA[ - .private_value = reg | (shift << 16) | (mask << 24) -]]> - </programlisting> - </informalexample> - and is retrieved in callbacks like - <informalexample> - <programlisting> -<![CDATA[ - static int snd_sbmixer_get_single(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) - { - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 16) & 0xff; - int mask = (kcontrol->private_value >> 24) & 0xff; - .... - } -]]> - </programlisting> - </informalexample> - </para> - - <para> - In the <structfield>get</structfield> callback, - you have to fill all the elements if the - control has more than one elements, - i.e. <structfield>count</structfield> &gt; 1. - In the example above, we filled only one element - (<structfield>value.integer.value[0]</structfield>) since it's - assumed as <structfield>count</structfield> = 1. - </para> - </section> - - <section id="control-interface-callbacks-put"> - <title>put callback</title> - - <para> - This callback is used to write a value from user-space. - </para> - - <para> - For example, - - <example> - <title>Example of put callback</title> - <programlisting> -<![CDATA[ - static int snd_myctl_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) - { - struct mychip *chip = snd_kcontrol_chip(kcontrol); - int changed = 0; - if (chip->current_value != - ucontrol->value.integer.value[0]) { - change_current_value(chip, - ucontrol->value.integer.value[0]); - changed = 1; - } - return changed; - } -]]> - </programlisting> - </example> - - As seen above, you have to return 1 if the value is - changed. If the value is not changed, return 0 instead. - If any fatal error happens, return a negative error code as - usual. - </para> - - <para> - As in the <structfield>get</structfield> callback, - when the control has more than one elements, - all elements must be evaluated in this callback, too. - </para> - </section> - - <section id="control-interface-callbacks-all"> - <title>Callbacks are not atomic</title> - <para> - All these three callbacks are basically not atomic. - </para> - </section> - </section> - - <section id="control-interface-constructor"> - <title>Constructor</title> - <para> - When everything is ready, finally we can create a new - control. To create a control, there are two functions to be - called, <function>snd_ctl_new1()</function> and - <function>snd_ctl_add()</function>. - </para> - - <para> - In the simplest way, you can do like this: - - <informalexample> - <programlisting> -<![CDATA[ - err = snd_ctl_add(card, snd_ctl_new1(&my_control, chip)); - if (err < 0) - return err; -]]> - </programlisting> - </informalexample> - - where <parameter>my_control</parameter> is the - struct <structname>snd_kcontrol_new</structname> object defined above, and chip - is the object pointer to be passed to - kcontrol-&gt;private_data - which can be referred to in callbacks. - </para> - - <para> - <function>snd_ctl_new1()</function> allocates a new - <structname>snd_kcontrol</structname> instance, - and <function>snd_ctl_add</function> assigns the given - control component to the card. - </para> - </section> - - <section id="control-interface-change-notification"> - <title>Change Notification</title> - <para> - If you need to change and update a control in the interrupt - routine, you can call <function>snd_ctl_notify()</function>. For - example, - - <informalexample> - <programlisting> -<![CDATA[ - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, id_pointer); -]]> - </programlisting> - </informalexample> - - This function takes the card pointer, the event-mask, and the - control id pointer for the notification. The event-mask - specifies the types of notification, for example, in the above - example, the change of control values is notified. - The id pointer is the pointer of struct <structname>snd_ctl_elem_id</structname> - to be notified. - You can find some examples in <filename>es1938.c</filename> or - <filename>es1968.c</filename> for hardware volume interrupts. - </para> - </section> - - <section id="control-interface-tlv"> - <title>Metadata</title> - <para> - To provide information about the dB values of a mixer control, use - on of the <constant>DECLARE_TLV_xxx</constant> macros from - <filename>&lt;sound/tlv.h&gt;</filename> to define a variable - containing this information, set the<structfield>tlv.p - </structfield> field to point to this variable, and include the - <constant>SNDRV_CTL_ELEM_ACCESS_TLV_READ</constant> flag in the - <structfield>access</structfield> field; like this: - <informalexample> - <programlisting> -<![CDATA[ - static DECLARE_TLV_DB_SCALE(db_scale_my_control, -4050, 150, 0); - - static struct snd_kcontrol_new my_control = { - ... - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ, - ... - .tlv.p = db_scale_my_control, - }; -]]> - </programlisting> - </informalexample> - </para> - - <para> - The <function>DECLARE_TLV_DB_SCALE</function> macro defines - information about a mixer control where each step in the control's - value changes the dB value by a constant dB amount. - The first parameter is the name of the variable to be defined. - The second parameter is the minimum value, in units of 0.01 dB. - The third parameter is the step size, in units of 0.01 dB. - Set the fourth parameter to 1 if the minimum value actually mutes - the control. - </para> - - <para> - The <function>DECLARE_TLV_DB_LINEAR</function> macro defines - information about a mixer control where the control's value affects - the output linearly. - The first parameter is the name of the variable to be defined. - The second parameter is the minimum value, in units of 0.01 dB. - The third parameter is the maximum value, in units of 0.01 dB. - If the minimum value mutes the control, set the second parameter to - <constant>TLV_DB_GAIN_MUTE</constant>. - </para> - </section> - - </chapter> - - -<!-- ****************************************************** --> -<!-- API for AC97 Codec --> -<!-- ****************************************************** --> - <chapter id="api-ac97"> - <title>API for AC97 Codec</title> - - <section> - <title>General</title> - <para> - The ALSA AC97 codec layer is a well-defined one, and you don't - have to write much code to control it. Only low-level control - routines are necessary. The AC97 codec API is defined in - <filename>&lt;sound/ac97_codec.h&gt;</filename>. - </para> - </section> - - <section id="api-ac97-example"> - <title>Full Code Example</title> - <para> - <example> - <title>Example of AC97 Interface</title> - <programlisting> -<![CDATA[ - struct mychip { - .... - struct snd_ac97 *ac97; - .... - }; - - static unsigned short snd_mychip_ac97_read(struct snd_ac97 *ac97, - unsigned short reg) - { - struct mychip *chip = ac97->private_data; - .... - /* read a register value here from the codec */ - return the_register_value; - } - - static void snd_mychip_ac97_write(struct snd_ac97 *ac97, - unsigned short reg, unsigned short val) - { - struct mychip *chip = ac97->private_data; - .... - /* write the given register value to the codec */ - } - - static int snd_mychip_ac97(struct mychip *chip) - { - struct snd_ac97_bus *bus; - struct snd_ac97_template ac97; - int err; - static struct snd_ac97_bus_ops ops = { - .write = snd_mychip_ac97_write, - .read = snd_mychip_ac97_read, - }; - - err = snd_ac97_bus(chip->card, 0, &ops, NULL, &bus); - if (err < 0) - return err; - memset(&ac97, 0, sizeof(ac97)); - ac97.private_data = chip; - return snd_ac97_mixer(bus, &ac97, &chip->ac97); - } - -]]> - </programlisting> - </example> - </para> - </section> - - <section id="api-ac97-constructor"> - <title>Constructor</title> - <para> - To create an ac97 instance, first call <function>snd_ac97_bus</function> - with an <type>ac97_bus_ops_t</type> record with callback functions. - - <informalexample> - <programlisting> -<![CDATA[ - struct snd_ac97_bus *bus; - static struct snd_ac97_bus_ops ops = { - .write = snd_mychip_ac97_write, - .read = snd_mychip_ac97_read, - }; - - snd_ac97_bus(card, 0, &ops, NULL, &pbus); -]]> - </programlisting> - </informalexample> - - The bus record is shared among all belonging ac97 instances. - </para> - - <para> - And then call <function>snd_ac97_mixer()</function> with an - struct <structname>snd_ac97_template</structname> - record together with the bus pointer created above. - - <informalexample> - <programlisting> -<![CDATA[ - struct snd_ac97_template ac97; - int err; - - memset(&ac97, 0, sizeof(ac97)); - ac97.private_data = chip; - snd_ac97_mixer(bus, &ac97, &chip->ac97); -]]> - </programlisting> - </informalexample> - - where chip-&gt;ac97 is a pointer to a newly created - <type>ac97_t</type> instance. - In this case, the chip pointer is set as the private data, so that - the read/write callback functions can refer to this chip instance. - This instance is not necessarily stored in the chip - record. If you need to change the register values from the - driver, or need the suspend/resume of ac97 codecs, keep this - pointer to pass to the corresponding functions. - </para> - </section> - - <section id="api-ac97-callbacks"> - <title>Callbacks</title> - <para> - The standard callbacks are <structfield>read</structfield> and - <structfield>write</structfield>. Obviously they - correspond to the functions for read and write accesses to the - hardware low-level codes. - </para> - - <para> - The <structfield>read</structfield> callback returns the - register value specified in the argument. - - <informalexample> - <programlisting> -<![CDATA[ - static unsigned short snd_mychip_ac97_read(struct snd_ac97 *ac97, - unsigned short reg) - { - struct mychip *chip = ac97->private_data; - .... - return the_register_value; - } -]]> - </programlisting> - </informalexample> - - Here, the chip can be cast from ac97-&gt;private_data. - </para> - - <para> - Meanwhile, the <structfield>write</structfield> callback is - used to set the register value. - - <informalexample> - <programlisting> -<![CDATA[ - static void snd_mychip_ac97_write(struct snd_ac97 *ac97, - unsigned short reg, unsigned short val) -]]> - </programlisting> - </informalexample> - </para> - - <para> - These callbacks are non-atomic like the control API callbacks. - </para> - - <para> - There are also other callbacks: - <structfield>reset</structfield>, - <structfield>wait</structfield> and - <structfield>init</structfield>. - </para> - - <para> - The <structfield>reset</structfield> callback is used to reset - the codec. If the chip requires a special kind of reset, you can - define this callback. - </para> - - <para> - The <structfield>wait</structfield> callback is used to - add some waiting time in the standard initialization of the codec. If the - chip requires the extra waiting time, define this callback. - </para> - - <para> - The <structfield>init</structfield> callback is used for - additional initialization of the codec. - </para> - </section> - - <section id="api-ac97-updating-registers"> - <title>Updating Registers in The Driver</title> - <para> - If you need to access to the codec from the driver, you can - call the following functions: - <function>snd_ac97_write()</function>, - <function>snd_ac97_read()</function>, - <function>snd_ac97_update()</function> and - <function>snd_ac97_update_bits()</function>. - </para> - - <para> - Both <function>snd_ac97_write()</function> and - <function>snd_ac97_update()</function> functions are used to - set a value to the given register - (<constant>AC97_XXX</constant>). The difference between them is - that <function>snd_ac97_update()</function> doesn't write a - value if the given value has been already set, while - <function>snd_ac97_write()</function> always rewrites the - value. - - <informalexample> - <programlisting> -<![CDATA[ - snd_ac97_write(ac97, AC97_MASTER, 0x8080); - snd_ac97_update(ac97, AC97_MASTER, 0x8080); -]]> - </programlisting> - </informalexample> - </para> - - <para> - <function>snd_ac97_read()</function> is used to read the value - of the given register. For example, - - <informalexample> - <programlisting> -<![CDATA[ - value = snd_ac97_read(ac97, AC97_MASTER); -]]> - </programlisting> - </informalexample> - </para> - - <para> - <function>snd_ac97_update_bits()</function> is used to update - some bits in the given register. - - <informalexample> - <programlisting> -<![CDATA[ - snd_ac97_update_bits(ac97, reg, mask, value); -]]> - </programlisting> - </informalexample> - </para> - - <para> - Also, there is a function to change the sample rate (of a - given register such as - <constant>AC97_PCM_FRONT_DAC_RATE</constant>) when VRA or - DRA is supported by the codec: - <function>snd_ac97_set_rate()</function>. - - <informalexample> - <programlisting> -<![CDATA[ - snd_ac97_set_rate(ac97, AC97_PCM_FRONT_DAC_RATE, 44100); -]]> - </programlisting> - </informalexample> - </para> - - <para> - The following registers are available to set the rate: - <constant>AC97_PCM_MIC_ADC_RATE</constant>, - <constant>AC97_PCM_FRONT_DAC_RATE</constant>, - <constant>AC97_PCM_LR_ADC_RATE</constant>, - <constant>AC97_SPDIF</constant>. When - <constant>AC97_SPDIF</constant> is specified, the register is - not really changed but the corresponding IEC958 status bits will - be updated. - </para> - </section> - - <section id="api-ac97-clock-adjustment"> - <title>Clock Adjustment</title> - <para> - In some chips, the clock of the codec isn't 48000 but using a - PCI clock (to save a quartz!). In this case, change the field - bus-&gt;clock to the corresponding - value. For example, intel8x0 - and es1968 drivers have their own function to read from the clock. - </para> - </section> - - <section id="api-ac97-proc-files"> - <title>Proc Files</title> - <para> - The ALSA AC97 interface will create a proc file such as - <filename>/proc/asound/card0/codec97#0/ac97#0-0</filename> and - <filename>ac97#0-0+regs</filename>. You can refer to these files to - see the current status and registers of the codec. - </para> - </section> - - <section id="api-ac97-multiple-codecs"> - <title>Multiple Codecs</title> - <para> - When there are several codecs on the same card, you need to - call <function>snd_ac97_mixer()</function> multiple times with - ac97.num=1 or greater. The <structfield>num</structfield> field - specifies the codec number. - </para> - - <para> - If you set up multiple codecs, you either need to write - different callbacks for each codec or check - ac97-&gt;num in the callback routines. - </para> - </section> - - </chapter> - - -<!-- ****************************************************** --> -<!-- MIDI (MPU401-UART) Interface --> -<!-- ****************************************************** --> - <chapter id="midi-interface"> - <title>MIDI (MPU401-UART) Interface</title> - - <section id="midi-interface-general"> - <title>General</title> - <para> - Many soundcards have built-in MIDI (MPU401-UART) - interfaces. When the soundcard supports the standard MPU401-UART - interface, most likely you can use the ALSA MPU401-UART API. The - MPU401-UART API is defined in - <filename>&lt;sound/mpu401.h&gt;</filename>. - </para> - - <para> - Some soundchips have a similar but slightly different - implementation of mpu401 stuff. For example, emu10k1 has its own - mpu401 routines. - </para> - </section> - - <section id="midi-interface-constructor"> - <title>Constructor</title> - <para> - To create a rawmidi object, call - <function>snd_mpu401_uart_new()</function>. - - <informalexample> - <programlisting> -<![CDATA[ - struct snd_rawmidi *rmidi; - snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, port, info_flags, - irq, &rmidi); -]]> - </programlisting> - </informalexample> - </para> - - <para> - The first argument is the card pointer, and the second is the - index of this component. You can create up to 8 rawmidi - devices. - </para> - - <para> - The third argument is the type of the hardware, - <constant>MPU401_HW_XXX</constant>. If it's not a special one, - you can use <constant>MPU401_HW_MPU401</constant>. - </para> - - <para> - The 4th argument is the I/O port address. Many - backward-compatible MPU401 have an I/O port such as 0x330. Or, it - might be a part of its own PCI I/O region. It depends on the - chip design. - </para> - - <para> - The 5th argument is a bitflag for additional information. - When the I/O port address above is part of the PCI I/O - region, the MPU401 I/O port might have been already allocated - (reserved) by the driver itself. In such a case, pass a bit flag - <constant>MPU401_INFO_INTEGRATED</constant>, - and the mpu401-uart layer will allocate the I/O ports by itself. - </para> - - <para> - When the controller supports only the input or output MIDI stream, - pass the <constant>MPU401_INFO_INPUT</constant> or - <constant>MPU401_INFO_OUTPUT</constant> bitflag, respectively. - Then the rawmidi instance is created as a single stream. - </para> - - <para> - <constant>MPU401_INFO_MMIO</constant> bitflag is used to change - the access method to MMIO (via readb and writeb) instead of - iob and outb. In this case, you have to pass the iomapped address - to <function>snd_mpu401_uart_new()</function>. - </para> - - <para> - When <constant>MPU401_INFO_TX_IRQ</constant> is set, the output - stream isn't checked in the default interrupt handler. The driver - needs to call <function>snd_mpu401_uart_interrupt_tx()</function> - by itself to start processing the output stream in the irq handler. - </para> - - <para> - If the MPU-401 interface shares its interrupt with the other logical - devices on the card, set <constant>MPU401_INFO_IRQ_HOOK</constant> - (see <link linkend="midi-interface-interrupt-handler"><citetitle> - below</citetitle></link>). - </para> - - <para> - Usually, the port address corresponds to the command port and - port + 1 corresponds to the data port. If not, you may change - the <structfield>cport</structfield> field of - struct <structname>snd_mpu401</structname> manually - afterward. However, <structname>snd_mpu401</structname> pointer is not - returned explicitly by - <function>snd_mpu401_uart_new()</function>. You need to cast - rmidi-&gt;private_data to - <structname>snd_mpu401</structname> explicitly, - - <informalexample> - <programlisting> -<![CDATA[ - struct snd_mpu401 *mpu; - mpu = rmidi->private_data; -]]> - </programlisting> - </informalexample> - - and reset the cport as you like: - - <informalexample> - <programlisting> -<![CDATA[ - mpu->cport = my_own_control_port; -]]> - </programlisting> - </informalexample> - </para> - - <para> - The 6th argument specifies the ISA irq number that will be - allocated. If no interrupt is to be allocated (because your - code is already allocating a shared interrupt, or because the - device does not use interrupts), pass -1 instead. - For a MPU-401 device without an interrupt, a polling timer - will be used instead. - </para> - </section> - - <section id="midi-interface-interrupt-handler"> - <title>Interrupt Handler</title> - <para> - When the interrupt is allocated in - <function>snd_mpu401_uart_new()</function>, an exclusive ISA - interrupt handler is automatically used, hence you don't have - anything else to do than creating the mpu401 stuff. Otherwise, you - have to set <constant>MPU401_INFO_IRQ_HOOK</constant>, and call - <function>snd_mpu401_uart_interrupt()</function> explicitly from your - own interrupt handler when it has determined that a UART interrupt - has occurred. - </para> - - <para> - In this case, you need to pass the private_data of the - returned rawmidi object from - <function>snd_mpu401_uart_new()</function> as the second - argument of <function>snd_mpu401_uart_interrupt()</function>. - - <informalexample> - <programlisting> -<![CDATA[ - snd_mpu401_uart_interrupt(irq, rmidi->private_data, regs); -]]> - </programlisting> - </informalexample> - </para> - </section> - - </chapter> - - -<!-- ****************************************************** --> -<!-- RawMIDI Interface --> -<!-- ****************************************************** --> - <chapter id="rawmidi-interface"> - <title>RawMIDI Interface</title> - - <section id="rawmidi-interface-overview"> - <title>Overview</title> - - <para> - The raw MIDI interface is used for hardware MIDI ports that can - be accessed as a byte stream. It is not used for synthesizer - chips that do not directly understand MIDI. - </para> - - <para> - ALSA handles file and buffer management. All you have to do is - to write some code to move data between the buffer and the - hardware. - </para> - - <para> - The rawmidi API is defined in - <filename>&lt;sound/rawmidi.h&gt;</filename>. - </para> - </section> - - <section id="rawmidi-interface-constructor"> - <title>Constructor</title> - - <para> - To create a rawmidi device, call the - <function>snd_rawmidi_new</function> function: - <informalexample> - <programlisting> -<![CDATA[ - struct snd_rawmidi *rmidi; - err = snd_rawmidi_new(chip->card, "MyMIDI", 0, outs, ins, &rmidi); - if (err < 0) - return err; - rmidi->private_data = chip; - strcpy(rmidi->name, "My MIDI"); - rmidi->info_flags = SNDRV_RAWMIDI_INFO_OUTPUT | - SNDRV_RAWMIDI_INFO_INPUT | - SNDRV_RAWMIDI_INFO_DUPLEX; -]]> - </programlisting> - </informalexample> - </para> - - <para> - The first argument is the card pointer, the second argument is - the ID string. - </para> - - <para> - The third argument is the index of this component. You can - create up to 8 rawmidi devices. - </para> - - <para> - The fourth and fifth arguments are the number of output and - input substreams, respectively, of this device (a substream is - the equivalent of a MIDI port). - </para> - - <para> - Set the <structfield>info_flags</structfield> field to specify - the capabilities of the device. - Set <constant>SNDRV_RAWMIDI_INFO_OUTPUT</constant> if there is - at least one output port, - <constant>SNDRV_RAWMIDI_INFO_INPUT</constant> if there is at - least one input port, - and <constant>SNDRV_RAWMIDI_INFO_DUPLEX</constant> if the device - can handle output and input at the same time. - </para> - - <para> - After the rawmidi device is created, you need to set the - operators (callbacks) for each substream. There are helper - functions to set the operators for all the substreams of a device: - <informalexample> - <programlisting> -<![CDATA[ - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_mymidi_output_ops); - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_mymidi_input_ops); -]]> - </programlisting> - </informalexample> - </para> - - <para> - The operators are usually defined like this: - <informalexample> - <programlisting> -<![CDATA[ - static struct snd_rawmidi_ops snd_mymidi_output_ops = { - .open = snd_mymidi_output_open, - .close = snd_mymidi_output_close, - .trigger = snd_mymidi_output_trigger, - }; -]]> - </programlisting> - </informalexample> - These callbacks are explained in the <link - linkend="rawmidi-interface-callbacks"><citetitle>Callbacks</citetitle></link> - section. - </para> - - <para> - If there are more than one substream, you should give a - unique name to each of them: - <informalexample> - <programlisting> -<![CDATA[ - struct snd_rawmidi_substream *substream; - list_for_each_entry(substream, - &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams, - list { - sprintf(substream->name, "My MIDI Port %d", substream->number + 1); - } - /* same for SNDRV_RAWMIDI_STREAM_INPUT */ -]]> - </programlisting> - </informalexample> - </para> - </section> - - <section id="rawmidi-interface-callbacks"> - <title>Callbacks</title> - - <para> - In all the callbacks, the private data that you've set for the - rawmidi device can be accessed as - substream-&gt;rmidi-&gt;private_data. - <!-- <code> isn't available before DocBook 4.3 --> - </para> - - <para> - If there is more than one port, your callbacks can determine the - port index from the struct snd_rawmidi_substream data passed to each - callback: - <informalexample> - <programlisting> -<![CDATA[ - struct snd_rawmidi_substream *substream; - int index = substream->number; -]]> - </programlisting> - </informalexample> - </para> - - <section id="rawmidi-interface-op-open"> - <title><function>open</function> callback</title> - - <informalexample> - <programlisting> -<![CDATA[ - static int snd_xxx_open(struct snd_rawmidi_substream *substream); -]]> - </programlisting> - </informalexample> - - <para> - This is called when a substream is opened. - You can initialize the hardware here, but you shouldn't - start transmitting/receiving data yet. - </para> - </section> - - <section id="rawmidi-interface-op-close"> - <title><function>close</function> callback</title> - - <informalexample> - <programlisting> -<![CDATA[ - static int snd_xxx_close(struct snd_rawmidi_substream *substream); -]]> - </programlisting> - </informalexample> - - <para> - Guess what. - </para> - - <para> - The <function>open</function> and <function>close</function> - callbacks of a rawmidi device are serialized with a mutex, - and can sleep. - </para> - </section> - - <section id="rawmidi-interface-op-trigger-out"> - <title><function>trigger</function> callback for output - substreams</title> - - <informalexample> - <programlisting> -<![CDATA[ - static void snd_xxx_output_trigger(struct snd_rawmidi_substream *substream, int up); -]]> - </programlisting> - </informalexample> - - <para> - This is called with a nonzero <parameter>up</parameter> - parameter when there is some data in the substream buffer that - must be transmitted. - </para> - - <para> - To read data from the buffer, call - <function>snd_rawmidi_transmit_peek</function>. It will - return the number of bytes that have been read; this will be - less than the number of bytes requested when there are no more - data in the buffer. - After the data have been transmitted successfully, call - <function>snd_rawmidi_transmit_ack</function> to remove the - data from the substream buffer: - <informalexample> - <programlisting> -<![CDATA[ - unsigned char data; - while (snd_rawmidi_transmit_peek(substream, &data, 1) == 1) { - if (snd_mychip_try_to_transmit(data)) - snd_rawmidi_transmit_ack(substream, 1); - else - break; /* hardware FIFO full */ - } -]]> - </programlisting> - </informalexample> - </para> - - <para> - If you know beforehand that the hardware will accept data, you - can use the <function>snd_rawmidi_transmit</function> function - which reads some data and removes them from the buffer at once: - <informalexample> - <programlisting> -<![CDATA[ - while (snd_mychip_transmit_possible()) { - unsigned char data; - if (snd_rawmidi_transmit(substream, &data, 1) != 1) - break; /* no more data */ - snd_mychip_transmit(data); - } -]]> - </programlisting> - </informalexample> - </para> - - <para> - If you know beforehand how many bytes you can accept, you can - use a buffer size greater than one with the - <function>snd_rawmidi_transmit*</function> functions. - </para> - - <para> - The <function>trigger</function> callback must not sleep. If - the hardware FIFO is full before the substream buffer has been - emptied, you have to continue transmitting data later, either - in an interrupt handler, or with a timer if the hardware - doesn't have a MIDI transmit interrupt. - </para> - - <para> - The <function>trigger</function> callback is called with a - zero <parameter>up</parameter> parameter when the transmission - of data should be aborted. - </para> - </section> - - <section id="rawmidi-interface-op-trigger-in"> - <title><function>trigger</function> callback for input - substreams</title> - - <informalexample> - <programlisting> -<![CDATA[ - static void snd_xxx_input_trigger(struct snd_rawmidi_substream *substream, int up); -]]> - </programlisting> - </informalexample> - - <para> - This is called with a nonzero <parameter>up</parameter> - parameter to enable receiving data, or with a zero - <parameter>up</parameter> parameter do disable receiving data. - </para> - - <para> - The <function>trigger</function> callback must not sleep; the - actual reading of data from the device is usually done in an - interrupt handler. - </para> - - <para> - When data reception is enabled, your interrupt handler should - call <function>snd_rawmidi_receive</function> for all received - data: - <informalexample> - <programlisting> -<![CDATA[ - void snd_mychip_midi_interrupt(...) - { - while (mychip_midi_available()) { - unsigned char data; - data = mychip_midi_read(); - snd_rawmidi_receive(substream, &data, 1); - } - } -]]> - </programlisting> - </informalexample> - </para> - </section> - - <section id="rawmidi-interface-op-drain"> - <title><function>drain</function> callback</title> - - <informalexample> - <programlisting> -<![CDATA[ - static void snd_xxx_drain(struct snd_rawmidi_substream *substream); -]]> - </programlisting> - </informalexample> - - <para> - This is only used with output substreams. This function should wait - until all data read from the substream buffer have been transmitted. - This ensures that the device can be closed and the driver unloaded - without losing data. - </para> - - <para> - This callback is optional. If you do not set - <structfield>drain</structfield> in the struct snd_rawmidi_ops - structure, ALSA will simply wait for 50&nbsp;milliseconds - instead. - </para> - </section> - </section> - - </chapter> - - -<!-- ****************************************************** --> -<!-- Miscellaneous Devices --> -<!-- ****************************************************** --> - <chapter id="misc-devices"> - <title>Miscellaneous Devices</title> - - <section id="misc-devices-opl3"> - <title>FM OPL3</title> - <para> - The FM OPL3 is still used in many chips (mainly for backward - compatibility). ALSA has a nice OPL3 FM control layer, too. The - OPL3 API is defined in - <filename>&lt;sound/opl3.h&gt;</filename>. - </para> - - <para> - FM registers can be directly accessed through the direct-FM API, - defined in <filename>&lt;sound/asound_fm.h&gt;</filename>. In - ALSA native mode, FM registers are accessed through - the Hardware-Dependent Device direct-FM extension API, whereas in - OSS compatible mode, FM registers can be accessed with the OSS - direct-FM compatible API in <filename>/dev/dmfmX</filename> device. - </para> - - <para> - To create the OPL3 component, you have two functions to - call. The first one is a constructor for the <type>opl3_t</type> - instance. - - <informalexample> - <programlisting> -<![CDATA[ - struct snd_opl3 *opl3; - snd_opl3_create(card, lport, rport, OPL3_HW_OPL3_XXX, - integrated, &opl3); -]]> - </programlisting> - </informalexample> - </para> - - <para> - The first argument is the card pointer, the second one is the - left port address, and the third is the right port address. In - most cases, the right port is placed at the left port + 2. - </para> - - <para> - The fourth argument is the hardware type. - </para> - - <para> - When the left and right ports have been already allocated by - the card driver, pass non-zero to the fifth argument - (<parameter>integrated</parameter>). Otherwise, the opl3 module will - allocate the specified ports by itself. - </para> - - <para> - When the accessing the hardware requires special method - instead of the standard I/O access, you can create opl3 instance - separately with <function>snd_opl3_new()</function>. - - <informalexample> - <programlisting> -<![CDATA[ - struct snd_opl3 *opl3; - snd_opl3_new(card, OPL3_HW_OPL3_XXX, &opl3); -]]> - </programlisting> - </informalexample> - </para> - - <para> - Then set <structfield>command</structfield>, - <structfield>private_data</structfield> and - <structfield>private_free</structfield> for the private - access function, the private data and the destructor. - The l_port and r_port are not necessarily set. Only the - command must be set properly. You can retrieve the data - from the opl3-&gt;private_data field. - </para> - - <para> - After creating the opl3 instance via <function>snd_opl3_new()</function>, - call <function>snd_opl3_init()</function> to initialize the chip to the - proper state. Note that <function>snd_opl3_create()</function> always - calls it internally. - </para> - - <para> - If the opl3 instance is created successfully, then create a - hwdep device for this opl3. - - <informalexample> - <programlisting> -<![CDATA[ - struct snd_hwdep *opl3hwdep; - snd_opl3_hwdep_new(opl3, 0, 1, &opl3hwdep); -]]> - </programlisting> - </informalexample> - </para> - - <para> - The first argument is the <type>opl3_t</type> instance you - created, and the second is the index number, usually 0. - </para> - - <para> - The third argument is the index-offset for the sequencer - client assigned to the OPL3 port. When there is an MPU401-UART, - give 1 for here (UART always takes 0). - </para> - </section> - - <section id="misc-devices-hardware-dependent"> - <title>Hardware-Dependent Devices</title> - <para> - Some chips need user-space access for special - controls or for loading the micro code. In such a case, you can - create a hwdep (hardware-dependent) device. The hwdep API is - defined in <filename>&lt;sound/hwdep.h&gt;</filename>. You can - find examples in opl3 driver or - <filename>isa/sb/sb16_csp.c</filename>. - </para> - - <para> - The creation of the <type>hwdep</type> instance is done via - <function>snd_hwdep_new()</function>. - - <informalexample> - <programlisting> -<![CDATA[ - struct snd_hwdep *hw; - snd_hwdep_new(card, "My HWDEP", 0, &hw); -]]> - </programlisting> - </informalexample> - - where the third argument is the index number. - </para> - - <para> - You can then pass any pointer value to the - <parameter>private_data</parameter>. - If you assign a private data, you should define the - destructor, too. The destructor function is set in - the <structfield>private_free</structfield> field. - - <informalexample> - <programlisting> -<![CDATA[ - struct mydata *p = kmalloc(sizeof(*p), GFP_KERNEL); - hw->private_data = p; - hw->private_free = mydata_free; -]]> - </programlisting> - </informalexample> - - and the implementation of the destructor would be: - - <informalexample> - <programlisting> -<![CDATA[ - static void mydata_free(struct snd_hwdep *hw) - { - struct mydata *p = hw->private_data; - kfree(p); - } -]]> - </programlisting> - </informalexample> - </para> - - <para> - The arbitrary file operations can be defined for this - instance. The file operators are defined in - the <parameter>ops</parameter> table. For example, assume that - this chip needs an ioctl. - - <informalexample> - <programlisting> -<![CDATA[ - hw->ops.open = mydata_open; - hw->ops.ioctl = mydata_ioctl; - hw->ops.release = mydata_release; -]]> - </programlisting> - </informalexample> - - And implement the callback functions as you like. - </para> - </section> - - <section id="misc-devices-IEC958"> - <title>IEC958 (S/PDIF)</title> - <para> - Usually the controls for IEC958 devices are implemented via - the control interface. There is a macro to compose a name string for - IEC958 controls, <function>SNDRV_CTL_NAME_IEC958()</function> - defined in <filename>&lt;include/asound.h&gt;</filename>. - </para> - - <para> - There are some standard controls for IEC958 status bits. These - controls use the type <type>SNDRV_CTL_ELEM_TYPE_IEC958</type>, - and the size of element is fixed as 4 bytes array - (value.iec958.status[x]). For the <structfield>info</structfield> - callback, you don't specify - the value field for this type (the count field must be set, - though). - </para> - - <para> - <quote>IEC958 Playback Con Mask</quote> is used to return the - bit-mask for the IEC958 status bits of consumer mode. Similarly, - <quote>IEC958 Playback Pro Mask</quote> returns the bitmask for - professional mode. They are read-only controls, and are defined - as MIXER controls (iface = - <constant>SNDRV_CTL_ELEM_IFACE_MIXER</constant>). - </para> - - <para> - Meanwhile, <quote>IEC958 Playback Default</quote> control is - defined for getting and setting the current default IEC958 - bits. Note that this one is usually defined as a PCM control - (iface = <constant>SNDRV_CTL_ELEM_IFACE_PCM</constant>), - although in some places it's defined as a MIXER control. - </para> - - <para> - In addition, you can define the control switches to - enable/disable or to set the raw bit mode. The implementation - will depend on the chip, but the control should be named as - <quote>IEC958 xxx</quote>, preferably using - the <function>SNDRV_CTL_NAME_IEC958()</function> macro. - </para> - - <para> - You can find several cases, for example, - <filename>pci/emu10k1</filename>, - <filename>pci/ice1712</filename>, or - <filename>pci/cmipci.c</filename>. - </para> - </section> - - </chapter> - - -<!-- ****************************************************** --> -<!-- Buffer and Memory Management --> -<!-- ****************************************************** --> - <chapter id="buffer-and-memory"> - <title>Buffer and Memory Management</title> - - <section id="buffer-and-memory-buffer-types"> - <title>Buffer Types</title> - <para> - ALSA provides several different buffer allocation functions - depending on the bus and the architecture. All these have a - consistent API. The allocation of physically-contiguous pages is - done via - <function>snd_malloc_xxx_pages()</function> function, where xxx - is the bus type. - </para> - - <para> - The allocation of pages with fallback is - <function>snd_malloc_xxx_pages_fallback()</function>. This - function tries to allocate the specified pages but if the pages - are not available, it tries to reduce the page sizes until - enough space is found. - </para> - - <para> - The release the pages, call - <function>snd_free_xxx_pages()</function> function. - </para> - - <para> - Usually, ALSA drivers try to allocate and reserve - a large contiguous physical space - at the time the module is loaded for the later use. - This is called <quote>pre-allocation</quote>. - As already written, you can call the following function at - pcm instance construction time (in the case of PCI bus). - - <informalexample> - <programlisting> -<![CDATA[ - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(pci), size, max); -]]> - </programlisting> - </informalexample> - - where <parameter>size</parameter> is the byte size to be - pre-allocated and the <parameter>max</parameter> is the maximum - size to be changed via the <filename>prealloc</filename> proc file. - The allocator will try to get an area as large as possible - within the given size. - </para> - - <para> - The second argument (type) and the third argument (device pointer) - are dependent on the bus. - In the case of the ISA bus, pass <function>snd_dma_isa_data()</function> - as the third argument with <constant>SNDRV_DMA_TYPE_DEV</constant> type. - For the continuous buffer unrelated to the bus can be pre-allocated - with <constant>SNDRV_DMA_TYPE_CONTINUOUS</constant> type and the - <function>snd_dma_continuous_data(GFP_KERNEL)</function> device pointer, - where <constant>GFP_KERNEL</constant> is the kernel allocation flag to - use. - For the PCI scatter-gather buffers, use - <constant>SNDRV_DMA_TYPE_DEV_SG</constant> with - <function>snd_dma_pci_data(pci)</function> - (see the - <link linkend="buffer-and-memory-non-contiguous"><citetitle>Non-Contiguous Buffers - </citetitle></link> section). - </para> - - <para> - Once the buffer is pre-allocated, you can use the - allocator in the <structfield>hw_params</structfield> callback: - - <informalexample> - <programlisting> -<![CDATA[ - snd_pcm_lib_malloc_pages(substream, size); -]]> - </programlisting> - </informalexample> - - Note that you have to pre-allocate to use this function. - </para> - </section> - - <section id="buffer-and-memory-external-hardware"> - <title>External Hardware Buffers</title> - <para> - Some chips have their own hardware buffers and the DMA - transfer from the host memory is not available. In such a case, - you need to either 1) copy/set the audio data directly to the - external hardware buffer, or 2) make an intermediate buffer and - copy/set the data from it to the external hardware buffer in - interrupts (or in tasklets, preferably). - </para> - - <para> - The first case works fine if the external hardware buffer is large - enough. This method doesn't need any extra buffers and thus is - more effective. You need to define the - <structfield>copy</structfield> and - <structfield>silence</structfield> callbacks for - the data transfer. However, there is a drawback: it cannot - be mmapped. The examples are GUS's GF1 PCM or emu8000's - wavetable PCM. - </para> - - <para> - The second case allows for mmap on the buffer, although you have - to handle an interrupt or a tasklet to transfer the data - from the intermediate buffer to the hardware buffer. You can find an - example in the vxpocket driver. - </para> - - <para> - Another case is when the chip uses a PCI memory-map - region for the buffer instead of the host memory. In this case, - mmap is available only on certain architectures like the Intel one. - In non-mmap mode, the data cannot be transferred as in the normal - way. Thus you need to define the <structfield>copy</structfield> and - <structfield>silence</structfield> callbacks as well, - as in the cases above. The examples are found in - <filename>rme32.c</filename> and <filename>rme96.c</filename>. - </para> - - <para> - The implementation of the <structfield>copy</structfield> and - <structfield>silence</structfield> callbacks depends upon - whether the hardware supports interleaved or non-interleaved - samples. The <structfield>copy</structfield> callback is - defined like below, a bit - differently depending whether the direction is playback or - capture: - - <informalexample> - <programlisting> -<![CDATA[ - static int playback_copy(struct snd_pcm_substream *substream, int channel, - snd_pcm_uframes_t pos, void *src, snd_pcm_uframes_t count); - static int capture_copy(struct snd_pcm_substream *substream, int channel, - snd_pcm_uframes_t pos, void *dst, snd_pcm_uframes_t count); -]]> - </programlisting> - </informalexample> - </para> - - <para> - In the case of interleaved samples, the second argument - (<parameter>channel</parameter>) is not used. The third argument - (<parameter>pos</parameter>) points the - current position offset in frames. - </para> - - <para> - The meaning of the fourth argument is different between - playback and capture. For playback, it holds the source data - pointer, and for capture, it's the destination data pointer. - </para> - - <para> - The last argument is the number of frames to be copied. - </para> - - <para> - What you have to do in this callback is again different - between playback and capture directions. In the - playback case, you copy the given amount of data - (<parameter>count</parameter>) at the specified pointer - (<parameter>src</parameter>) to the specified offset - (<parameter>pos</parameter>) on the hardware buffer. When - coded like memcpy-like way, the copy would be like: - - <informalexample> - <programlisting> -<![CDATA[ - my_memcpy(my_buffer + frames_to_bytes(runtime, pos), src, - frames_to_bytes(runtime, count)); -]]> - </programlisting> - </informalexample> - </para> - - <para> - For the capture direction, you copy the given amount of - data (<parameter>count</parameter>) at the specified offset - (<parameter>pos</parameter>) on the hardware buffer to the - specified pointer (<parameter>dst</parameter>). - - <informalexample> - <programlisting> -<![CDATA[ - my_memcpy(dst, my_buffer + frames_to_bytes(runtime, pos), - frames_to_bytes(runtime, count)); -]]> - </programlisting> - </informalexample> - - Note that both the position and the amount of data are given - in frames. - </para> - - <para> - In the case of non-interleaved samples, the implementation - will be a bit more complicated. - </para> - - <para> - You need to check the channel argument, and if it's -1, copy - the whole channels. Otherwise, you have to copy only the - specified channel. Please check - <filename>isa/gus/gus_pcm.c</filename> as an example. - </para> - - <para> - The <structfield>silence</structfield> callback is also - implemented in a similar way. - - <informalexample> - <programlisting> -<![CDATA[ - static int silence(struct snd_pcm_substream *substream, int channel, - snd_pcm_uframes_t pos, snd_pcm_uframes_t count); -]]> - </programlisting> - </informalexample> - </para> - - <para> - The meanings of arguments are the same as in the - <structfield>copy</structfield> - callback, although there is no <parameter>src/dst</parameter> - argument. In the case of interleaved samples, the channel - argument has no meaning, as well as on - <structfield>copy</structfield> callback. - </para> - - <para> - The role of <structfield>silence</structfield> callback is to - set the given amount - (<parameter>count</parameter>) of silence data at the - specified offset (<parameter>pos</parameter>) on the hardware - buffer. Suppose that the data format is signed (that is, the - silent-data is 0), and the implementation using a memset-like - function would be like: - - <informalexample> - <programlisting> -<![CDATA[ - my_memcpy(my_buffer + frames_to_bytes(runtime, pos), 0, - frames_to_bytes(runtime, count)); -]]> - </programlisting> - </informalexample> - </para> - - <para> - In the case of non-interleaved samples, again, the - implementation becomes a bit more complicated. See, for example, - <filename>isa/gus/gus_pcm.c</filename>. - </para> - </section> - - <section id="buffer-and-memory-non-contiguous"> - <title>Non-Contiguous Buffers</title> - <para> - If your hardware supports the page table as in emu10k1 or the - buffer descriptors as in via82xx, you can use the scatter-gather - (SG) DMA. ALSA provides an interface for handling SG-buffers. - The API is provided in <filename>&lt;sound/pcm.h&gt;</filename>. - </para> - - <para> - For creating the SG-buffer handler, call - <function>snd_pcm_lib_preallocate_pages()</function> or - <function>snd_pcm_lib_preallocate_pages_for_all()</function> - with <constant>SNDRV_DMA_TYPE_DEV_SG</constant> - in the PCM constructor like other PCI pre-allocator. - You need to pass <function>snd_dma_pci_data(pci)</function>, - where pci is the struct <structname>pci_dev</structname> pointer - of the chip as well. - The <type>struct snd_sg_buf</type> instance is created as - substream-&gt;dma_private. You can cast - the pointer like: - - <informalexample> - <programlisting> -<![CDATA[ - struct snd_sg_buf *sgbuf = (struct snd_sg_buf *)substream->dma_private; -]]> - </programlisting> - </informalexample> - </para> - - <para> - Then call <function>snd_pcm_lib_malloc_pages()</function> - in the <structfield>hw_params</structfield> callback - as well as in the case of normal PCI buffer. - The SG-buffer handler will allocate the non-contiguous kernel - pages of the given size and map them onto the virtually contiguous - memory. The virtual pointer is addressed in runtime-&gt;dma_area. - The physical address (runtime-&gt;dma_addr) is set to zero, - because the buffer is physically non-contiguous. - The physical address table is set up in sgbuf-&gt;table. - You can get the physical address at a certain offset via - <function>snd_pcm_sgbuf_get_addr()</function>. - </para> - - <para> - When a SG-handler is used, you need to set - <function>snd_pcm_sgbuf_ops_page</function> as - the <structfield>page</structfield> callback. - (See <link linkend="pcm-interface-operators-page-callback"> - <citetitle>page callback section</citetitle></link>.) - </para> - - <para> - To release the data, call - <function>snd_pcm_lib_free_pages()</function> in the - <structfield>hw_free</structfield> callback as usual. - </para> - </section> - - <section id="buffer-and-memory-vmalloced"> - <title>Vmalloc'ed Buffers</title> - <para> - It's possible to use a buffer allocated via - <function>vmalloc</function>, for example, for an intermediate - buffer. Since the allocated pages are not contiguous, you need - to set the <structfield>page</structfield> callback to obtain - the physical address at every offset. - </para> - - <para> - The implementation of <structfield>page</structfield> callback - would be like this: - - <informalexample> - <programlisting> -<![CDATA[ - #include <linux/vmalloc.h> - - /* get the physical page pointer on the given offset */ - static struct page *mychip_page(struct snd_pcm_substream *substream, - unsigned long offset) - { - void *pageptr = substream->runtime->dma_area + offset; - return vmalloc_to_page(pageptr); - } -]]> - </programlisting> - </informalexample> - </para> - </section> - - </chapter> - - -<!-- ****************************************************** --> -<!-- Proc Interface --> -<!-- ****************************************************** --> - <chapter id="proc-interface"> - <title>Proc Interface</title> - <para> - ALSA provides an easy interface for procfs. The proc files are - very useful for debugging. I recommend you set up proc files if - you write a driver and want to get a running status or register - dumps. The API is found in - <filename>&lt;sound/info.h&gt;</filename>. - </para> - - <para> - To create a proc file, call - <function>snd_card_proc_new()</function>. - - <informalexample> - <programlisting> -<![CDATA[ - struct snd_info_entry *entry; - int err = snd_card_proc_new(card, "my-file", &entry); -]]> - </programlisting> - </informalexample> - - where the second argument specifies the name of the proc file to be - created. The above example will create a file - <filename>my-file</filename> under the card directory, - e.g. <filename>/proc/asound/card0/my-file</filename>. - </para> - - <para> - Like other components, the proc entry created via - <function>snd_card_proc_new()</function> will be registered and - released automatically in the card registration and release - functions. - </para> - - <para> - When the creation is successful, the function stores a new - instance in the pointer given in the third argument. - It is initialized as a text proc file for read only. To use - this proc file as a read-only text file as it is, set the read - callback with a private data via - <function>snd_info_set_text_ops()</function>. - - <informalexample> - <programlisting> -<![CDATA[ - snd_info_set_text_ops(entry, chip, my_proc_read); -]]> - </programlisting> - </informalexample> - - where the second argument (<parameter>chip</parameter>) is the - private data to be used in the callbacks. The third parameter - specifies the read buffer size and the fourth - (<parameter>my_proc_read</parameter>) is the callback function, which - is defined like - - <informalexample> - <programlisting> -<![CDATA[ - static void my_proc_read(struct snd_info_entry *entry, - struct snd_info_buffer *buffer); -]]> - </programlisting> - </informalexample> - - </para> - - <para> - In the read callback, use <function>snd_iprintf()</function> for - output strings, which works just like normal - <function>printf()</function>. For example, - - <informalexample> - <programlisting> -<![CDATA[ - static void my_proc_read(struct snd_info_entry *entry, - struct snd_info_buffer *buffer) - { - struct my_chip *chip = entry->private_data; - - snd_iprintf(buffer, "This is my chip!\n"); - snd_iprintf(buffer, "Port = %ld\n", chip->port); - } -]]> - </programlisting> - </informalexample> - </para> - - <para> - The file permissions can be changed afterwards. As default, it's - set as read only for all users. If you want to add write - permission for the user (root as default), do as follows: - - <informalexample> - <programlisting> -<![CDATA[ - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; -]]> - </programlisting> - </informalexample> - - and set the write buffer size and the callback - - <informalexample> - <programlisting> -<![CDATA[ - entry->c.text.write = my_proc_write; -]]> - </programlisting> - </informalexample> - </para> - - <para> - For the write callback, you can use - <function>snd_info_get_line()</function> to get a text line, and - <function>snd_info_get_str()</function> to retrieve a string from - the line. Some examples are found in - <filename>core/oss/mixer_oss.c</filename>, core/oss/and - <filename>pcm_oss.c</filename>. - </para> - - <para> - For a raw-data proc-file, set the attributes as follows: - - <informalexample> - <programlisting> -<![CDATA[ - static struct snd_info_entry_ops my_file_io_ops = { - .read = my_file_io_read, - }; - - entry->content = SNDRV_INFO_CONTENT_DATA; - entry->private_data = chip; - entry->c.ops = &my_file_io_ops; - entry->size = 4096; - entry->mode = S_IFREG | S_IRUGO; -]]> - </programlisting> - </informalexample> - - For the raw data, <structfield>size</structfield> field must be - set properly. This specifies the maximum size of the proc file access. - </para> - - <para> - The read/write callbacks of raw mode are more direct than the text mode. - You need to use a low-level I/O functions such as - <function>copy_from/to_user()</function> to transfer the - data. - - <informalexample> - <programlisting> -<![CDATA[ - static ssize_t my_file_io_read(struct snd_info_entry *entry, - void *file_private_data, - struct file *file, - char *buf, - size_t count, - loff_t pos) - { - if (copy_to_user(buf, local_data + pos, count)) - return -EFAULT; - return count; - } -]]> - </programlisting> - </informalexample> - - If the size of the info entry has been set up properly, - <structfield>count</structfield> and <structfield>pos</structfield> are - guaranteed to fit within 0 and the given size. - You don't have to check the range in the callbacks unless any - other condition is required. - - </para> - - </chapter> - - -<!-- ****************************************************** --> -<!-- Power Management --> -<!-- ****************************************************** --> - <chapter id="power-management"> - <title>Power Management</title> - <para> - If the chip is supposed to work with suspend/resume - functions, you need to add power-management code to the - driver. The additional code for power-management should be - <function>ifdef</function>'ed with - <constant>CONFIG_PM</constant>. - </para> - - <para> - If the driver <emphasis>fully</emphasis> supports suspend/resume - that is, the device can be - properly resumed to its state when suspend was called, - you can set the <constant>SNDRV_PCM_INFO_RESUME</constant> flag - in the pcm info field. Usually, this is possible when the - registers of the chip can be safely saved and restored to - RAM. If this is set, the trigger callback is called with - <constant>SNDRV_PCM_TRIGGER_RESUME</constant> after the resume - callback completes. - </para> - - <para> - Even if the driver doesn't support PM fully but - partial suspend/resume is still possible, it's still worthy to - implement suspend/resume callbacks. In such a case, applications - would reset the status by calling - <function>snd_pcm_prepare()</function> and restart the stream - appropriately. Hence, you can define suspend/resume callbacks - below but don't set <constant>SNDRV_PCM_INFO_RESUME</constant> - info flag to the PCM. - </para> - - <para> - Note that the trigger with SUSPEND can always be called when - <function>snd_pcm_suspend_all</function> is called, - regardless of the <constant>SNDRV_PCM_INFO_RESUME</constant> flag. - The <constant>RESUME</constant> flag affects only the behavior - of <function>snd_pcm_resume()</function>. - (Thus, in theory, - <constant>SNDRV_PCM_TRIGGER_RESUME</constant> isn't needed - to be handled in the trigger callback when no - <constant>SNDRV_PCM_INFO_RESUME</constant> flag is set. But, - it's better to keep it for compatibility reasons.) - </para> - <para> - In the earlier version of ALSA drivers, a common - power-management layer was provided, but it has been removed. - The driver needs to define the suspend/resume hooks according to - the bus the device is connected to. In the case of PCI drivers, the - callbacks look like below: - - <informalexample> - <programlisting> -<![CDATA[ - #ifdef CONFIG_PM - static int snd_my_suspend(struct pci_dev *pci, pm_message_t state) - { - .... /* do things for suspend */ - return 0; - } - static int snd_my_resume(struct pci_dev *pci) - { - .... /* do things for suspend */ - return 0; - } - #endif -]]> - </programlisting> - </informalexample> - </para> - - <para> - The scheme of the real suspend job is as follows. - - <orderedlist> - <listitem><para>Retrieve the card and the chip data.</para></listitem> - <listitem><para>Call <function>snd_power_change_state()</function> with - <constant>SNDRV_CTL_POWER_D3hot</constant> to change the - power status.</para></listitem> - <listitem><para>Call <function>snd_pcm_suspend_all()</function> to suspend the running PCM streams.</para></listitem> - <listitem><para>If AC97 codecs are used, call - <function>snd_ac97_suspend()</function> for each codec.</para></listitem> - <listitem><para>Save the register values if necessary.</para></listitem> - <listitem><para>Stop the hardware if necessary.</para></listitem> - <listitem><para>Disable the PCI device by calling - <function>pci_disable_device()</function>. Then, call - <function>pci_save_state()</function> at last.</para></listitem> - </orderedlist> - </para> - - <para> - A typical code would be like: - - <informalexample> - <programlisting> -<![CDATA[ - static int mychip_suspend(struct pci_dev *pci, pm_message_t state) - { - /* (1) */ - struct snd_card *card = pci_get_drvdata(pci); - struct mychip *chip = card->private_data; - /* (2) */ - snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - /* (3) */ - snd_pcm_suspend_all(chip->pcm); - /* (4) */ - snd_ac97_suspend(chip->ac97); - /* (5) */ - snd_mychip_save_registers(chip); - /* (6) */ - snd_mychip_stop_hardware(chip); - /* (7) */ - pci_disable_device(pci); - pci_save_state(pci); - return 0; - } -]]> - </programlisting> - </informalexample> - </para> - - <para> - The scheme of the real resume job is as follows. - - <orderedlist> - <listitem><para>Retrieve the card and the chip data.</para></listitem> - <listitem><para>Set up PCI. First, call <function>pci_restore_state()</function>. - Then enable the pci device again by calling <function>pci_enable_device()</function>. - Call <function>pci_set_master()</function> if necessary, too.</para></listitem> - <listitem><para>Re-initialize the chip.</para></listitem> - <listitem><para>Restore the saved registers if necessary.</para></listitem> - <listitem><para>Resume the mixer, e.g. calling - <function>snd_ac97_resume()</function>.</para></listitem> - <listitem><para>Restart the hardware (if any).</para></listitem> - <listitem><para>Call <function>snd_power_change_state()</function> with - <constant>SNDRV_CTL_POWER_D0</constant> to notify the processes.</para></listitem> - </orderedlist> - </para> - - <para> - A typical code would be like: - - <informalexample> - <programlisting> -<![CDATA[ - static int mychip_resume(struct pci_dev *pci) - { - /* (1) */ - struct snd_card *card = pci_get_drvdata(pci); - struct mychip *chip = card->private_data; - /* (2) */ - pci_restore_state(pci); - pci_enable_device(pci); - pci_set_master(pci); - /* (3) */ - snd_mychip_reinit_chip(chip); - /* (4) */ - snd_mychip_restore_registers(chip); - /* (5) */ - snd_ac97_resume(chip->ac97); - /* (6) */ - snd_mychip_restart_chip(chip); - /* (7) */ - snd_power_change_state(card, SNDRV_CTL_POWER_D0); - return 0; - } -]]> - </programlisting> - </informalexample> - </para> - - <para> - As shown in the above, it's better to save registers after - suspending the PCM operations via - <function>snd_pcm_suspend_all()</function> or - <function>snd_pcm_suspend()</function>. It means that the PCM - streams are already stopped when the register snapshot is - taken. But, remember that you don't have to restart the PCM - stream in the resume callback. It'll be restarted via - trigger call with <constant>SNDRV_PCM_TRIGGER_RESUME</constant> - when necessary. - </para> - - <para> - OK, we have all callbacks now. Let's set them up. In the - initialization of the card, make sure that you can get the chip - data from the card instance, typically via - <structfield>private_data</structfield> field, in case you - created the chip data individually. - - <informalexample> - <programlisting> -<![CDATA[ - static int snd_mychip_probe(struct pci_dev *pci, - const struct pci_device_id *pci_id) - { - .... - struct snd_card *card; - struct mychip *chip; - int err; - .... - err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, - 0, &card); - .... - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - .... - card->private_data = chip; - .... - } -]]> - </programlisting> - </informalexample> - - When you created the chip data with - <function>snd_card_new()</function>, it's anyway accessible - via <structfield>private_data</structfield> field. - - <informalexample> - <programlisting> -<![CDATA[ - static int snd_mychip_probe(struct pci_dev *pci, - const struct pci_device_id *pci_id) - { - .... - struct snd_card *card; - struct mychip *chip; - int err; - .... - err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, - sizeof(struct mychip), &card); - .... - chip = card->private_data; - .... - } -]]> - </programlisting> - </informalexample> - - </para> - - <para> - If you need a space to save the registers, allocate the - buffer for it here, too, since it would be fatal - if you cannot allocate a memory in the suspend phase. - The allocated buffer should be released in the corresponding - destructor. - </para> - - <para> - And next, set suspend/resume callbacks to the pci_driver. - - <informalexample> - <programlisting> -<![CDATA[ - static struct pci_driver driver = { - .name = KBUILD_MODNAME, - .id_table = snd_my_ids, - .probe = snd_my_probe, - .remove = snd_my_remove, - #ifdef CONFIG_PM - .suspend = snd_my_suspend, - .resume = snd_my_resume, - #endif - }; -]]> - </programlisting> - </informalexample> - </para> - - </chapter> - - -<!-- ****************************************************** --> -<!-- Module Parameters --> -<!-- ****************************************************** --> - <chapter id="module-parameters"> - <title>Module Parameters</title> - <para> - There are standard module options for ALSA. At least, each - module should have the <parameter>index</parameter>, - <parameter>id</parameter> and <parameter>enable</parameter> - options. - </para> - - <para> - If the module supports multiple cards (usually up to - 8 = <constant>SNDRV_CARDS</constant> cards), they should be - arrays. The default initial values are defined already as - constants for easier programming: - - <informalexample> - <programlisting> -<![CDATA[ - static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; - static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; - static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; -]]> - </programlisting> - </informalexample> - </para> - - <para> - If the module supports only a single card, they could be single - variables, instead. <parameter>enable</parameter> option is not - always necessary in this case, but it would be better to have a - dummy option for compatibility. - </para> - - <para> - The module parameters must be declared with the standard - <function>module_param()()</function>, - <function>module_param_array()()</function> and - <function>MODULE_PARM_DESC()</function> macros. - </para> - - <para> - The typical coding would be like below: - - <informalexample> - <programlisting> -<![CDATA[ - #define CARD_NAME "My Chip" - - module_param_array(index, int, NULL, 0444); - MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); - module_param_array(id, charp, NULL, 0444); - MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); - module_param_array(enable, bool, NULL, 0444); - MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); -]]> - </programlisting> - </informalexample> - </para> - - <para> - Also, don't forget to define the module description, classes, - license and devices. Especially, the recent modprobe requires to - define the module license as GPL, etc., otherwise the system is - shown as <quote>tainted</quote>. - - <informalexample> - <programlisting> -<![CDATA[ - MODULE_DESCRIPTION("My Chip"); - MODULE_LICENSE("GPL"); - MODULE_SUPPORTED_DEVICE("{{Vendor,My Chip Name}}"); -]]> - </programlisting> - </informalexample> - </para> - - </chapter> - - -<!-- ****************************************************** --> -<!-- How To Put Your Driver --> -<!-- ****************************************************** --> - <chapter id="how-to-put-your-driver"> - <title>How To Put Your Driver Into ALSA Tree</title> - <section> - <title>General</title> - <para> - So far, you've learned how to write the driver codes. - And you might have a question now: how to put my own - driver into the ALSA driver tree? - Here (finally :) the standard procedure is described briefly. - </para> - - <para> - Suppose that you create a new PCI driver for the card - <quote>xyz</quote>. The card module name would be - snd-xyz. The new driver is usually put into the alsa-driver - tree, <filename>alsa-driver/pci</filename> directory in - the case of PCI cards. - Then the driver is evaluated, audited and tested - by developers and users. After a certain time, the driver - will go to the alsa-kernel tree (to the corresponding directory, - such as <filename>alsa-kernel/pci</filename>) and eventually - will be integrated into the Linux 2.6 tree (the directory would be - <filename>linux/sound/pci</filename>). - </para> - - <para> - In the following sections, the driver code is supposed - to be put into alsa-driver tree. The two cases are covered: - a driver consisting of a single source file and one consisting - of several source files. - </para> - </section> - - <section> - <title>Driver with A Single Source File</title> - <para> - <orderedlist> - <listitem> - <para> - Modify alsa-driver/pci/Makefile - </para> - - <para> - Suppose you have a file xyz.c. Add the following - two lines - <informalexample> - <programlisting> -<![CDATA[ - snd-xyz-objs := xyz.o - obj-$(CONFIG_SND_XYZ) += snd-xyz.o -]]> - </programlisting> - </informalexample> - </para> - </listitem> - - <listitem> - <para> - Create the Kconfig entry - </para> - - <para> - Add the new entry of Kconfig for your xyz driver. - <informalexample> - <programlisting> -<![CDATA[ - config SND_XYZ - tristate "Foobar XYZ" - depends on SND - select SND_PCM - help - Say Y here to include support for Foobar XYZ soundcard. - - To compile this driver as a module, choose M here: the module - will be called snd-xyz. -]]> - </programlisting> - </informalexample> - - the line, select SND_PCM, specifies that the driver xyz supports - PCM. In addition to SND_PCM, the following components are - supported for select command: - SND_RAWMIDI, SND_TIMER, SND_HWDEP, SND_MPU401_UART, - SND_OPL3_LIB, SND_OPL4_LIB, SND_VX_LIB, SND_AC97_CODEC. - Add the select command for each supported component. - </para> - - <para> - Note that some selections imply the lowlevel selections. - For example, PCM includes TIMER, MPU401_UART includes RAWMIDI, - AC97_CODEC includes PCM, and OPL3_LIB includes HWDEP. - You don't need to give the lowlevel selections again. - </para> - - <para> - For the details of Kconfig script, refer to the kbuild - documentation. - </para> - - </listitem> - - <listitem> - <para> - Run cvscompile script to re-generate the configure script and - build the whole stuff again. - </para> - </listitem> - </orderedlist> - </para> - </section> - - <section> - <title>Drivers with Several Source Files</title> - <para> - Suppose that the driver snd-xyz have several source files. - They are located in the new subdirectory, - pci/xyz. - - <orderedlist> - <listitem> - <para> - Add a new directory (<filename>xyz</filename>) in - <filename>alsa-driver/pci/Makefile</filename> as below - - <informalexample> - <programlisting> -<![CDATA[ - obj-$(CONFIG_SND) += xyz/ -]]> - </programlisting> - </informalexample> - </para> - </listitem> - - <listitem> - <para> - Under the directory <filename>xyz</filename>, create a Makefile - - <example> - <title>Sample Makefile for a driver xyz</title> - <programlisting> -<![CDATA[ - ifndef SND_TOPDIR - SND_TOPDIR=../.. - endif - - include $(SND_TOPDIR)/toplevel.config - include $(SND_TOPDIR)/Makefile.conf - - snd-xyz-objs := xyz.o abc.o def.o - - obj-$(CONFIG_SND_XYZ) += snd-xyz.o - - include $(SND_TOPDIR)/Rules.make -]]> - </programlisting> - </example> - </para> - </listitem> - - <listitem> - <para> - Create the Kconfig entry - </para> - - <para> - This procedure is as same as in the last section. - </para> - </listitem> - - <listitem> - <para> - Run cvscompile script to re-generate the configure script and - build the whole stuff again. - </para> - </listitem> - </orderedlist> - </para> - </section> - - </chapter> - -<!-- ****************************************************** --> -<!-- Useful Functions --> -<!-- ****************************************************** --> - <chapter id="useful-functions"> - <title>Useful Functions</title> - - <section id="useful-functions-snd-printk"> - <title><function>snd_printk()</function> and friends</title> - <para> - ALSA provides a verbose version of the - <function>printk()</function> function. If a kernel config - <constant>CONFIG_SND_VERBOSE_PRINTK</constant> is set, this - function prints the given message together with the file name - and the line of the caller. The <constant>KERN_XXX</constant> - prefix is processed as - well as the original <function>printk()</function> does, so it's - recommended to add this prefix, e.g. - - <informalexample> - <programlisting> -<![CDATA[ - snd_printk(KERN_ERR "Oh my, sorry, it's extremely bad!\n"); -]]> - </programlisting> - </informalexample> - </para> - - <para> - There are also <function>printk()</function>'s for - debugging. <function>snd_printd()</function> can be used for - general debugging purposes. If - <constant>CONFIG_SND_DEBUG</constant> is set, this function is - compiled, and works just like - <function>snd_printk()</function>. If the ALSA is compiled - without the debugging flag, it's ignored. - </para> - - <para> - <function>snd_printdd()</function> is compiled in only when - <constant>CONFIG_SND_DEBUG_VERBOSE</constant> is set. Please note - that <constant>CONFIG_SND_DEBUG_VERBOSE</constant> is not set as default - even if you configure the alsa-driver with - <option>--with-debug=full</option> option. You need to give - explicitly <option>--with-debug=detect</option> option instead. - </para> - </section> - - <section id="useful-functions-snd-bug"> - <title><function>snd_BUG()</function></title> - <para> - It shows the <computeroutput>BUG?</computeroutput> message and - stack trace as well as <function>snd_BUG_ON</function> at the point. - It's useful to show that a fatal error happens there. - </para> - <para> - When no debug flag is set, this macro is ignored. - </para> - </section> - - <section id="useful-functions-snd-bug-on"> - <title><function>snd_BUG_ON()</function></title> - <para> - <function>snd_BUG_ON()</function> macro is similar with - <function>WARN_ON()</function> macro. For example, - - <informalexample> - <programlisting> -<![CDATA[ - snd_BUG_ON(!pointer); -]]> - </programlisting> - </informalexample> - - or it can be used as the condition, - <informalexample> - <programlisting> -<![CDATA[ - if (snd_BUG_ON(non_zero_is_bug)) - return -EINVAL; -]]> - </programlisting> - </informalexample> - - </para> - - <para> - The macro takes an conditional expression to evaluate. - When <constant>CONFIG_SND_DEBUG</constant>, is set, if the - expression is non-zero, it shows the warning message such as - <computeroutput>BUG? (xxx)</computeroutput> - normally followed by stack trace. - - In both cases it returns the evaluated value. - </para> - - </section> - - </chapter> - - -<!-- ****************************************************** --> -<!-- Acknowledgments --> -<!-- ****************************************************** --> - <chapter id="acknowledgments"> - <title>Acknowledgments</title> - <para> - I would like to thank Phil Kerr for his help for improvement and - corrections of this document. - </para> - <para> - Kevin Conder reformatted the original plain-text to the - DocBook format. - </para> - <para> - Giuliano Pochini corrected typos and contributed the example codes - in the hardware constraints section. - </para> - </chapter> -</book> diff --git a/Documentation/sound/kernel-api/index.rst b/Documentation/sound/kernel-api/index.rst index 73c13497dec7..d0e6df35b4b4 100644 --- a/Documentation/sound/kernel-api/index.rst +++ b/Documentation/sound/kernel-api/index.rst @@ -5,3 +5,4 @@ ALSA Kernel API Documentation :maxdepth: 2

alsa-driver-api + writing-an-alsa-driver diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst new file mode 100644 index 000000000000..95c5443eff38 --- /dev/null +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -0,0 +1,4219 @@ +====================== +Writing an ALSA Driver +====================== + +:Author: Takashi Iwai tiwai@suse.de +:Date: Oct 15, 2007 +:Edition: 0.3.7 + +Preface +======= + +This document describes how to write an `ALSA (Advanced Linux Sound +Architecture) http://www.alsa-project.org/`__ driver. The document +focuses mainly on PCI soundcards. In the case of other device types, the +API might be different, too. However, at least the ALSA kernel API is +consistent, and therefore it would be still a bit help for writing them. + +This document targets people who already have enough C language skills +and have basic linux kernel programming knowledge. This document doesn't +explain the general topic of linux kernel coding and doesn't cover +low-level driver implementation details. It only describes the standard +way to write a PCI sound driver on ALSA. + +If you are already familiar with the older ALSA ver.0.5.x API, you can +check the drivers such as ``sound/pci/es1938.c`` or +``sound/pci/maestro3.c`` which have also almost the same code-base in +the ALSA 0.5.x tree, so you can compare the differences. + +This document is still a draft version. Any feedback and corrections, +please!! + +File Tree Structure +=================== + +General +------- + +The ALSA drivers are provided in two ways. + +One is the trees provided as a tarball or via cvs from the ALSA's ftp +site, and another is the 2.6 (or later) Linux kernel tree. To +synchronize both, the ALSA driver tree is split into two different +trees: alsa-kernel and alsa-driver. The former contains purely the +source code for the Linux 2.6 (or later) tree. This tree is designed +only for compilation on 2.6 or later environment. The latter, +alsa-driver, contains many subtle files for compiling ALSA drivers +outside of the Linux kernel tree, wrapper functions for older 2.2 and +2.4 kernels, to adapt the latest kernel API, and additional drivers +which are still in development or in tests. The drivers in alsa-driver +tree will be moved to alsa-kernel (and eventually to the 2.6 kernel +tree) when they are finished and confirmed to work fine. + +The file tree structure of ALSA driver is depicted below. Both +alsa-kernel and alsa-driver have almost the same file structure, except +for “core” directory. It's named as “acore” in alsa-driver tree. + +:: + + sound + /core + /oss + /seq + /oss + /instr + /ioctl32 + /include + /drivers + /mpu401 + /opl3 + /i2c + /l3 + /synth + /emux + /pci + /(cards) + /isa + /(cards) + /arm + /ppc + /sparc + /usb + /pcmcia /(cards) + /oss + + +core directory +-------------- + +This directory contains the middle layer which is the heart of ALSA +drivers. In this directory, the native ALSA modules are stored. The +sub-directories contain different modules and are dependent upon the +kernel config. + +core/oss +~~~~~~~~ + +The codes for PCM and mixer OSS emulation modules are stored in this +directory. The rawmidi OSS emulation is included in the ALSA rawmidi +code since it's quite small. The sequencer code is stored in +``core/seq/oss`` directory (see `below <#core-seq-oss>`__). + +core/ioctl32 +~~~~~~~~~~~~ + +This directory contains the 32bit-ioctl wrappers for 64bit architectures +such like x86-64, ppc64 and sparc64. For 32bit and alpha architectures, +these are not compiled. + +core/seq +~~~~~~~~ + +This directory and its sub-directories are for the ALSA sequencer. This +directory contains the sequencer core and primary sequencer modules such +like snd-seq-midi, snd-seq-virmidi, etc. They are compiled only when +``CONFIG_SND_SEQUENCER`` is set in the kernel config. + +core/seq/oss +~~~~~~~~~~~~ + +This contains the OSS sequencer emulation codes. + +core/seq/instr +~~~~~~~~~~~~~~ + +This directory contains the modules for the sequencer instrument layer. + +include directory +----------------- + +This is the place for the public header files of ALSA drivers, which are +to be exported to user-space, or included by several files at different +directories. Basically, the private header files should not be placed in +this directory, but you may still find files there, due to historical +reasons :) + +drivers directory +----------------- + +This directory contains code shared among different drivers on different +architectures. They are hence supposed not to be architecture-specific. +For example, the dummy pcm driver and the serial MIDI driver are found +in this directory. In the sub-directories, there is code for components +which are independent from bus and cpu architectures. + +drivers/mpu401 +~~~~~~~~~~~~~~ + +The MPU401 and MPU401-UART modules are stored here. + +drivers/opl3 and opl4 +~~~~~~~~~~~~~~~~~~~~~ + +The OPL3 and OPL4 FM-synth stuff is found here. + +i2c directory +------------- + +This contains the ALSA i2c components. + +Although there is a standard i2c layer on Linux, ALSA has its own i2c +code for some cards, because the soundcard needs only a simple operation +and the standard i2c API is too complicated for such a purpose. + +i2c/l3 +~~~~~~ + +This is a sub-directory for ARM L3 i2c. + +synth directory +--------------- + +This contains the synth middle-level modules. + +So far, there is only Emu8000/Emu10k1 synth driver under the +``synth/emux`` sub-directory. + +pci directory +------------- + +This directory and its sub-directories hold the top-level card modules +for PCI soundcards and the code specific to the PCI BUS. + +The drivers compiled from a single file are stored directly in the pci +directory, while the drivers with several source files are stored on +their own sub-directory (e.g. emu10k1, ice1712). + +isa directory +------------- + +This directory and its sub-directories hold the top-level card modules +for ISA soundcards. + +arm, ppc, and sparc directories +------------------------------- + +They are used for top-level card modules which are specific to one of +these architectures. + +usb directory +------------- + +This directory contains the USB-audio driver. In the latest version, the +USB MIDI driver is integrated in the usb-audio driver. + +pcmcia directory +---------------- + +The PCMCIA, especially PCCard drivers will go here. CardBus drivers will +be in the pci directory, because their API is identical to that of +standard PCI cards. + +oss directory +------------- + +The OSS/Lite source files are stored here in Linux 2.6 (or later) tree. +In the ALSA driver tarball, this directory is empty, of course :) + +Basic Flow for PCI Drivers +========================== + +Outline +------- + +The minimum flow for PCI soundcards is as follows: + +- define the PCI ID table (see the section `PCI Entries`_). + +- create ``probe`` callback. + +- create ``remove`` callback. + +- create a :c:type:`struct pci_driver <pci_driver>` structure + containing the three pointers above. + +- create an ``init`` function just calling the + :c:func:`pci_register_driver()` to register the pci_driver + table defined above. + +- create an ``exit`` function to call the + :c:func:`pci_unregister_driver()` function. + +Full Code Example +----------------- + +The code example is shown below. Some parts are kept unimplemented at +this moment but will be filled in the next sections. The numbers in the +comment lines of the :c:func:`snd_mychip_probe()` function refer +to details explained in the following section. + +:: + + #include <linux/init.h> + #include <linux/pci.h> + #include <linux/slab.h> + #include <sound/core.h> + #include <sound/initval.h> + + /* module parameters (see "Module Parameters") */ + /* SNDRV_CARDS: maximum number of cards supported by this module */ + static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; + static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; + static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + + /* definition of the chip-specific record */ + struct mychip { + struct snd_card *card; + /* the rest of the implementation will be in section + * "PCI Resource Management" + */ + }; + + /* chip-specific destructor + * (see "PCI Resource Management") + */ + static int snd_mychip_free(struct mychip *chip) + { + .... /* will be implemented later... */ + } + + /* component-destructor + * (see "Management of Cards and Components") + */ + static int snd_mychip_dev_free(struct snd_device *device) + { + return snd_mychip_free(device->device_data); + } + + /* chip-specific constructor + * (see "Management of Cards and Components") + */ + static int snd_mychip_create(struct snd_card *card, + struct pci_dev *pci, + struct mychip **rchip) + { + struct mychip *chip; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_mychip_dev_free, + }; + + *rchip = NULL; + + /* check PCI availability here + * (see "PCI Resource Management") + */ + .... + + /* allocate a chip-specific data with zero filled */ + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) + return -ENOMEM; + + chip->card = card; + + /* rest of initialization here; will be implemented + * later, see "PCI Resource Management" + */ + .... + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_mychip_free(chip); + return err; + } + + *rchip = chip; + return 0; + } + + /* constructor -- see "Driver Constructor" sub-section */ + static int snd_mychip_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) + { + static int dev; + struct snd_card *card; + struct mychip *chip; + int err; + + /* (1) */ + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + /* (2) */ + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, + 0, &card); + if (err < 0) + return err; + + /* (3) */ + err = snd_mychip_create(card, pci, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + + /* (4) */ + strcpy(card->driver, "My Chip"); + strcpy(card->shortname, "My Own Chip 123"); + sprintf(card->longname, "%s at 0x%lx irq %i", + card->shortname, chip->ioport, chip->irq); + + /* (5) */ + .... /* implemented later */ + + /* (6) */ + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + + /* (7) */ + pci_set_drvdata(pci, card); + dev++; + return 0; + } + + /* destructor -- see the "Destructor" sub-section */ + static void snd_mychip_remove(struct pci_dev *pci) + { + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); + } + + + +Driver Constructor +------------------ + +The real constructor of PCI drivers is the ``probe`` callback. The +``probe`` callback and other component-constructors which are called +from the ``probe`` callback cannot be used with the ``__init`` prefix +because any PCI device could be a hotplug device. + +In the ``probe`` callback, the following scheme is often used. + +1) Check and increment the device index. +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +:: + + static int dev; + .... + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + +where ``enable[dev]`` is the module option. + +Each time the ``probe`` callback is called, check the availability of +the device. If not available, simply increment the device index and +returns. dev will be incremented also later (`step 7 +<#set-the-pci-driver-data-and-return-zero>`__). + +2) Create a card instance +~~~~~~~~~~~~~~~~~~~~~~~~~ + +:: + + struct snd_card *card; + int err; + .... + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, + 0, &card); + + +The details will be explained in the section `Management of Cards and +Components`_. + +3) Create a main component +~~~~~~~~~~~~~~~~~~~~~~~~~~ + +In this part, the PCI resources are allocated. + +:: + + struct mychip *chip; + .... + err = snd_mychip_create(card, pci, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + +The details will be explained in the section `PCI Resource +Management`_. + +4) Set the driver ID and name strings. +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +:: + + strcpy(card->driver, "My Chip"); + strcpy(card->shortname, "My Own Chip 123"); + sprintf(card->longname, "%s at 0x%lx irq %i", + card->shortname, chip->ioport, chip->irq); + +The driver field holds the minimal ID string of the chip. This is used +by alsa-lib's configurator, so keep it simple but unique. Even the +same driver can have different driver IDs to distinguish the +functionality of each chip type. + +The shortname field is a string shown as more verbose name. The longname +field contains the information shown in ``/proc/asound/cards``. + +5) Create other components, such as mixer, MIDI, etc. +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +Here you define the basic components such as `PCM <#PCM-Interface>`__, +mixer (e.g. `AC97 <#API-for-AC97-Codec>`__), MIDI (e.g. +`MPU-401 <#MIDI-MPU401-UART-Interface>`__), and other interfaces. +Also, if you want a `proc file <#Proc-Interface>`__, define it here, +too. + +6) Register the card instance. +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +:: + + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + +Will be explained in the section `Management of Cards and +Components`_, too. + +7) Set the PCI driver data and return zero. +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +:: + + pci_set_drvdata(pci, card); + dev++; + return 0; + +In the above, the card record is stored. This pointer is used in the +remove callback and power-management callbacks, too. + +Destructor +---------- + +The destructor, remove callback, simply releases the card instance. Then +the ALSA middle layer will release all the attached components +automatically. + +It would be typically like the following: + +:: + + static void snd_mychip_remove(struct pci_dev *pci) + { + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); + } + + +The above code assumes that the card pointer is set to the PCI driver +data. + +Header Files +------------ + +For the above example, at least the following include files are +necessary. + +:: + + #include <linux/init.h> + #include <linux/pci.h> + #include <linux/slab.h> + #include <sound/core.h> + #include <sound/initval.h> + +where the last one is necessary only when module options are defined +in the source file. If the code is split into several files, the files +without module options don't need them. + +In addition to these headers, you'll need ``<linux/interrupt.h>`` for +interrupt handling, and ``<asm/io.h>`` for I/O access. If you use the +:c:func:`mdelay()` or :c:func:`udelay()` functions, you'll need +to include ``<linux/delay.h>`` too. + +The ALSA interfaces like the PCM and control APIs are defined in other +``<sound/xxx.h>`` header files. They have to be included after +``<sound/core.h>``. + +Management of Cards and Components +================================== + +Card Instance +------------- + +For each soundcard, a “card” record must be allocated. + +A card record is the headquarters of the soundcard. It manages the whole +list of devices (components) on the soundcard, such as PCM, mixers, +MIDI, synthesizer, and so on. Also, the card record holds the ID and the +name strings of the card, manages the root of proc files, and controls +the power-management states and hotplug disconnections. The component +list on the card record is used to manage the correct release of +resources at destruction. + +As mentioned above, to create a card instance, call +:c:func:`snd_card_new()`. + +:: + + struct snd_card *card; + int err; + err = snd_card_new(&pci->dev, index, id, module, extra_size, &card); + + +The function takes six arguments: the parent device pointer, the +card-index number, the id string, the module pointer (usually +``THIS_MODULE``), the size of extra-data space, and the pointer to +return the card instance. The extra_size argument is used to allocate +card->private_data for the chip-specific data. Note that these data are +allocated by :c:func:`snd_card_new()`. + +The first argument, the pointer of struct :c:type:`struct device +<device>`, specifies the parent device. For PCI devices, typically +``&pci->`` is passed there. + +Components +---------- + +After the card is created, you can attach the components (devices) to +the card instance. In an ALSA driver, a component is represented as a +:c:type:`struct snd_device <snd_device>` object. A component +can be a PCM instance, a control interface, a raw MIDI interface, etc. +Each such instance has one component entry. + +A component can be created via :c:func:`snd_device_new()` +function. + +:: + + snd_device_new(card, SNDRV_DEV_XXX, chip, &ops); + +This takes the card pointer, the device-level (``SNDRV_DEV_XXX``), the +data pointer, and the callback pointers (``&ops``). The device-level +defines the type of components and the order of registration and +de-registration. For most components, the device-level is already +defined. For a user-defined component, you can use +``SNDRV_DEV_LOWLEVEL``. + +This function itself doesn't allocate the data space. The data must be +allocated manually beforehand, and its pointer is passed as the +argument. This pointer (``chip`` in the above example) is used as the +identifier for the instance. + +Each pre-defined ALSA component such as ac97 and pcm calls +:c:func:`snd_device_new()` inside its constructor. The destructor +for each component is defined in the callback pointers. Hence, you don't +need to take care of calling a destructor for such a component. + +If you wish to create your own component, you need to set the destructor +function to the dev_free callback in the ``ops``, so that it can be +released automatically via :c:func:`snd_card_free()`. The next +example will show an implementation of chip-specific data. + +Chip-Specific Data +------------------ + +Chip-specific information, e.g. the I/O port address, its resource +pointer, or the irq number, is stored in the chip-specific record. + +:: + + struct mychip { + .... + }; + + +In general, there are two ways of allocating the chip record. + +1. Allocating via :c:func:`snd_card_new()`. +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +As mentioned above, you can pass the extra-data-length to the 5th +argument of :c:func:`snd_card_new()`, i.e. + +:: + + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, + sizeof(struct mychip), &card); + +:c:type:`struct mychip <mychip>` is the type of the chip record. + +In return, the allocated record can be accessed as + +:: + + struct mychip *chip = card->private_data; + +With this method, you don't have to allocate twice. The record is +released together with the card instance. + +2. Allocating an extra device. +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +After allocating a card instance via :c:func:`snd_card_new()` +(with ``0`` on the 4th arg), call :c:func:`kzalloc()`. + +:: + + struct snd_card *card; + struct mychip *chip; + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, + 0, &card); + ..... + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + +The chip record should have the field to hold the card pointer at least, + +:: + + struct mychip { + struct snd_card *card; + .... + }; + + +Then, set the card pointer in the returned chip instance. + +:: + + chip->card = card; + +Next, initialize the fields, and register this chip record as a +low-level device with a specified ``ops``, + +:: + + static struct snd_device_ops ops = { + .dev_free = snd_mychip_dev_free, + }; + .... + snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + +:c:func:`snd_mychip_dev_free()` is the device-destructor +function, which will call the real destructor. + +:: + + static int snd_mychip_dev_free(struct snd_device *device) + { + return snd_mychip_free(device->device_data); + } + +where :c:func:`snd_mychip_free()` is the real destructor. + +Registration and Release +------------------------ + +After all components are assigned, register the card instance by calling +:c:func:`snd_card_register()`. Access to the device files is +enabled at this point. That is, before +:c:func:`snd_card_register()` is called, the components are safely +inaccessible from external side. If this call fails, exit the probe +function after releasing the card via :c:func:`snd_card_free()`. + +For releasing the card instance, you can call simply +:c:func:`snd_card_free()`. As mentioned earlier, all components +are released automatically by this call. + +For a device which allows hotplugging, you can use +:c:func:`snd_card_free_when_closed()`. This one will postpone +the destruction until all devices are closed. + +PCI Resource Management +======================= + +Full Code Example +----------------- + +In this section, we'll complete the chip-specific constructor, +destructor and PCI entries. Example code is shown first, below. + +:: + + struct mychip { + struct snd_card *card; + struct pci_dev *pci; + + unsigned long port; + int irq; + }; + + static int snd_mychip_free(struct mychip *chip) + { + /* disable hardware here if any */ + .... /* (not implemented in this document) */ + + /* release the irq */ + if (chip->irq >= 0) + free_irq(chip->irq, chip); + /* release the I/O ports & memory */ + pci_release_regions(chip->pci); + /* disable the PCI entry */ + pci_disable_device(chip->pci); + /* release the data */ + kfree(chip); + return 0; + } + + /* chip-specific constructor */ + static int snd_mychip_create(struct snd_card *card, + struct pci_dev *pci, + struct mychip **rchip) + { + struct mychip *chip; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_mychip_dev_free, + }; + + *rchip = NULL; + + /* initialize the PCI entry */ + err = pci_enable_device(pci); + if (err < 0) + return err; + /* check PCI availability (28bit DMA) */ + if (pci_set_dma_mask(pci, DMA_BIT_MASK(28)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(28)) < 0) { + printk(KERN_ERR "error to set 28bit mask DMA\n"); + pci_disable_device(pci); + return -ENXIO; + } + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) { + pci_disable_device(pci); + return -ENOMEM; + } + + /* initialize the stuff */ + chip->card = card; + chip->pci = pci; + chip->irq = -1; + + /* (1) PCI resource allocation */ + err = pci_request_regions(pci, "My Chip"); + if (err < 0) { + kfree(chip); + pci_disable_device(pci); + return err; + } + chip->port = pci_resource_start(pci, 0); + if (request_irq(pci->irq, snd_mychip_interrupt, + IRQF_SHARED, KBUILD_MODNAME, chip)) { + printk(KERN_ERR "cannot grab irq %d\n", pci->irq); + snd_mychip_free(chip); + return -EBUSY; + } + chip->irq = pci->irq; + + /* (2) initialization of the chip hardware */ + .... /* (not implemented in this document) */ + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_mychip_free(chip); + return err; + } + + *rchip = chip; + return 0; + } + + /* PCI IDs */ + static struct pci_device_id snd_mychip_ids[] = { + { PCI_VENDOR_ID_FOO, PCI_DEVICE_ID_BAR, + PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, + .... + { 0, } + }; + MODULE_DEVICE_TABLE(pci, snd_mychip_ids); + + /* pci_driver definition */ + static struct pci_driver driver = { + .name = KBUILD_MODNAME, + .id_table = snd_mychip_ids, + .probe = snd_mychip_probe, + .remove = snd_mychip_remove, + }; + + /* module initialization */ + static int __init alsa_card_mychip_init(void) + { + return pci_register_driver(&driver); + } + + /* module clean up */ + static void __exit alsa_card_mychip_exit(void) + { + pci_unregister_driver(&driver); + } + + module_init(alsa_card_mychip_init) + module_exit(alsa_card_mychip_exit) + + EXPORT_NO_SYMBOLS; /* for old kernels only */ + +Some Hafta's +------------ + +The allocation of PCI resources is done in the ``probe`` function, and +usually an extra :c:func:`xxx_create()` function is written for this +purpose. + +In the case of PCI devices, you first have to call the +:c:func:`pci_enable_device()` function before allocating +resources. Also, you need to set the proper PCI DMA mask to limit the +accessed I/O range. In some cases, you might need to call +:c:func:`pci_set_master()` function, too. + +Suppose the 28bit mask, and the code to be added would be like: + +:: + + err = pci_enable_device(pci); + if (err < 0) + return err; + if (pci_set_dma_mask(pci, DMA_BIT_MASK(28)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(28)) < 0) { + printk(KERN_ERR "error to set 28bit mask DMA\n"); + pci_disable_device(pci); + return -ENXIO; + } + + +Resource Allocation +------------------- + +The allocation of I/O ports and irqs is done via standard kernel +functions. Unlike ALSA ver.0.5.x., there are no helpers for that. And +these resources must be released in the destructor function (see below). +Also, on ALSA 0.9.x, you don't need to allocate (pseudo-)DMA for PCI +like in ALSA 0.5.x. + +Now assume that the PCI device has an I/O port with 8 bytes and an +interrupt. Then :c:type:`struct mychip <mychip>` will have the +following fields: + +:: + + struct mychip { + struct snd_card *card; + + unsigned long port; + int irq; + }; + + +For an I/O port (and also a memory region), you need to have the +resource pointer for the standard resource management. For an irq, you +have to keep only the irq number (integer). But you need to initialize +this number as -1 before actual allocation, since irq 0 is valid. The +port address and its resource pointer can be initialized as null by +:c:func:`kzalloc()` automatically, so you don't have to take care of +resetting them. + +The allocation of an I/O port is done like this: + +:: + + err = pci_request_regions(pci, "My Chip"); + if (err < 0) { + kfree(chip); + pci_disable_device(pci); + return err; + } + chip->port = pci_resource_start(pci, 0); + +It will reserve the I/O port region of 8 bytes of the given PCI device. +The returned value, ``chip->res_port``, is allocated via +:c:func:`kmalloc()` by :c:func:`request_region()`. The pointer +must be released via :c:func:`kfree()`, but there is a problem with +this. This issue will be explained later. + +The allocation of an interrupt source is done like this: + +:: + + if (request_irq(pci->irq, snd_mychip_interrupt, + IRQF_SHARED, KBUILD_MODNAME, chip)) { + printk(KERN_ERR "cannot grab irq %d\n", pci->irq); + snd_mychip_free(chip); + return -EBUSY; + } + chip->irq = pci->irq; + +where :c:func:`snd_mychip_interrupt()` is the interrupt handler +defined `later <#pcm-interface-interrupt-handler>`__. Note that +``chip->irq`` should be defined only when :c:func:`request_irq()` +succeeded. + +On the PCI bus, interrupts can be shared. Thus, ``IRQF_SHARED`` is used +as the interrupt flag of :c:func:`request_irq()`. + +The last argument of :c:func:`request_irq()` is the data pointer +passed to the interrupt handler. Usually, the chip-specific record is +used for that, but you can use what you like, too. + +I won't give details about the interrupt handler at this point, but at +least its appearance can be explained now. The interrupt handler looks +usually like the following: + +:: + + static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id) + { + struct mychip *chip = dev_id; + .... + return IRQ_HANDLED; + } + + +Now let's write the corresponding destructor for the resources above. +The role of destructor is simple: disable the hardware (if already +activated) and release the resources. So far, we have no hardware part, +so the disabling code is not written here. + +To release the resources, the “check-and-release” method is a safer way. +For the interrupt, do like this: + +:: + + if (chip->irq >= 0) + free_irq(chip->irq, chip); + +Since the irq number can start from 0, you should initialize +``chip->irq`` with a negative value (e.g. -1), so that you can check +the validity of the irq number as above. + +When you requested I/O ports or memory regions via +:c:func:`pci_request_region()` or +:c:func:`pci_request_regions()` like in this example, release the +resource(s) using the corresponding function, +:c:func:`pci_release_region()` or +:c:func:`pci_release_regions()`. + +:: + + pci_release_regions(chip->pci); + +When you requested manually via :c:func:`request_region()` or +:c:func:`request_mem_region()`, you can release it via +:c:func:`release_resource()`. Suppose that you keep the resource +pointer returned from :c:func:`request_region()` in +chip->res_port, the release procedure looks like: + +:: + + release_and_free_resource(chip->res_port); + +Don't forget to call :c:func:`pci_disable_device()` before the +end. + +And finally, release the chip-specific record. + +:: + + kfree(chip); + +We didn't implement the hardware disabling part in the above. If you +need to do this, please note that the destructor may be called even +before the initialization of the chip is completed. It would be better +to have a flag to skip hardware disabling if the hardware was not +initialized yet. + +When the chip-data is assigned to the card using +:c:func:`snd_device_new()` with ``SNDRV_DEV_LOWLELVEL`` , its +destructor is called at the last. That is, it is assured that all other +components like PCMs and controls have already been released. You don't +have to stop PCMs, etc. explicitly, but just call low-level hardware +stopping. + +The management of a memory-mapped region is almost as same as the +management of an I/O port. You'll need three fields like the +following: + +:: + + struct mychip { + .... + unsigned long iobase_phys; + void __iomem *iobase_virt; + }; + +and the allocation would be like below: + +:: + + if ((err = pci_request_regions(pci, "My Chip")) < 0) { + kfree(chip); + return err; + } + chip->iobase_phys = pci_resource_start(pci, 0); + chip->iobase_virt = ioremap_nocache(chip->iobase_phys, + pci_resource_len(pci, 0)); + +and the corresponding destructor would be: + +:: + + static int snd_mychip_free(struct mychip *chip) + { + .... + if (chip->iobase_virt) + iounmap(chip->iobase_virt); + .... + pci_release_regions(chip->pci); + .... + } + +PCI Entries +----------- + +So far, so good. Let's finish the missing PCI stuff. At first, we need a +:c:type:`struct pci_device_id <pci_device_id>` table for +this chipset. It's a table of PCI vendor/device ID number, and some +masks. + +For example, + +:: + + static struct pci_device_id snd_mychip_ids[] = { + { PCI_VENDOR_ID_FOO, PCI_DEVICE_ID_BAR, + PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, + .... + { 0, } + }; + MODULE_DEVICE_TABLE(pci, snd_mychip_ids); + +The first and second fields of the :c:type:`struct pci_device_id +<pci_device_id>` structure are the vendor and device IDs. If you +have no reason to filter the matching devices, you can leave the +remaining fields as above. The last field of the :c:type:`struct +pci_device_id <pci_device_id>` struct contains private data +for this entry. You can specify any value here, for example, to define +specific operations for supported device IDs. Such an example is found +in the intel8x0 driver. + +The last entry of this list is the terminator. You must specify this +all-zero entry. + +Then, prepare the :c:type:`struct pci_driver <pci_driver>` +record: + +:: + + static struct pci_driver driver = { + .name = KBUILD_MODNAME, + .id_table = snd_mychip_ids, + .probe = snd_mychip_probe, + .remove = snd_mychip_remove, + }; + +The ``probe`` and ``remove`` functions have already been defined in +the previous sections. The ``name`` field is the name string of this +device. Note that you must not use a slash “/” in this string. + +And at last, the module entries: + +:: + + static int __init alsa_card_mychip_init(void) + { + return pci_register_driver(&driver); + } + + static void __exit alsa_card_mychip_exit(void) + { + pci_unregister_driver(&driver); + } + + module_init(alsa_card_mychip_init) + module_exit(alsa_card_mychip_exit) + +Note that these module entries are tagged with ``__init`` and ``__exit`` +prefixes. + +Oh, one thing was forgotten. If you have no exported symbols, you need +to declare it in 2.2 or 2.4 kernels (it's not necessary in 2.6 kernels). + +:: + + EXPORT_NO_SYMBOLS; + +That's all! + +PCM Interface +============= + +General +------- + +The PCM middle layer of ALSA is quite powerful and it is only necessary +for each driver to implement the low-level functions to access its +hardware. + +For accessing to the PCM layer, you need to include ``<sound/pcm.h>`` +first. In addition, ``<sound/pcm_params.h>`` might be needed if you +access to some functions related with hw_param. + +Each card device can have up to four pcm instances. A pcm instance +corresponds to a pcm device file. The limitation of number of instances +comes only from the available bit size of the Linux's device numbers. +Once when 64bit device number is used, we'll have more pcm instances +available. + +A pcm instance consists of pcm playback and capture streams, and each +pcm stream consists of one or more pcm substreams. Some soundcards +support multiple playback functions. For example, emu10k1 has a PCM +playback of 32 stereo substreams. In this case, at each open, a free +substream is (usually) automatically chosen and opened. Meanwhile, when +only one substream exists and it was already opened, the successful open +will either block or error with ``EAGAIN`` according to the file open +mode. But you don't have to care about such details in your driver. The +PCM middle layer will take care of such work. + +Full Code Example +----------------- + +The example code below does not include any hardware access routines but +shows only the skeleton, how to build up the PCM interfaces. + +:: + + #include <sound/pcm.h> + .... + + /* hardware definition */ + static struct snd_pcm_hardware snd_mychip_playback_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 32768, + .period_bytes_min = 4096, + .period_bytes_max = 32768, + .periods_min = 1, + .periods_max = 1024, + }; + + /* hardware definition */ + static struct snd_pcm_hardware snd_mychip_capture_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 32768, + .period_bytes_min = 4096, + .period_bytes_max = 32768, + .periods_min = 1, + .periods_max = 1024, + }; + + /* open callback */ + static int snd_mychip_playback_open(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_mychip_playback_hw; + /* more hardware-initialization will be done here */ + .... + return 0; + } + + /* close callback */ + static int snd_mychip_playback_close(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + /* the hardware-specific codes will be here */ + .... + return 0; + + } + + /* open callback */ + static int snd_mychip_capture_open(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_mychip_capture_hw; + /* more hardware-initialization will be done here */ + .... + return 0; + } + + /* close callback */ + static int snd_mychip_capture_close(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + /* the hardware-specific codes will be here */ + .... + return 0; + + } + + /* hw_params callback */ + static int snd_mychip_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) + { + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + } + + /* hw_free callback */ + static int snd_mychip_pcm_hw_free(struct snd_pcm_substream *substream) + { + return snd_pcm_lib_free_pages(substream); + } + + /* prepare callback */ + static int snd_mychip_pcm_prepare(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + /* set up the hardware with the current configuration + * for example... + */ + mychip_set_sample_format(chip, runtime->format); + mychip_set_sample_rate(chip, runtime->rate); + mychip_set_channels(chip, runtime->channels); + mychip_set_dma_setup(chip, runtime->dma_addr, + chip->buffer_size, + chip->period_size); + return 0; + } + + /* trigger callback */ + static int snd_mychip_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) + { + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* do something to start the PCM engine */ + .... + break; + case SNDRV_PCM_TRIGGER_STOP: + /* do something to stop the PCM engine */ + .... + break; + default: + return -EINVAL; + } + } + + /* pointer callback */ + static snd_pcm_uframes_t + snd_mychip_pcm_pointer(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + unsigned int current_ptr; + + /* get the current hardware pointer */ + current_ptr = mychip_get_hw_pointer(chip); + return current_ptr; + } + + /* operators */ + static struct snd_pcm_ops snd_mychip_playback_ops = { + .open = snd_mychip_playback_open, + .close = snd_mychip_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_mychip_pcm_hw_params, + .hw_free = snd_mychip_pcm_hw_free, + .prepare = snd_mychip_pcm_prepare, + .trigger = snd_mychip_pcm_trigger, + .pointer = snd_mychip_pcm_pointer, + }; + + /* operators */ + static struct snd_pcm_ops snd_mychip_capture_ops = { + .open = snd_mychip_capture_open, + .close = snd_mychip_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_mychip_pcm_hw_params, + .hw_free = snd_mychip_pcm_hw_free, + .prepare = snd_mychip_pcm_prepare, + .trigger = snd_mychip_pcm_trigger, + .pointer = snd_mychip_pcm_pointer, + }; + + /* + * definitions of capture are omitted here... + */ + + /* create a pcm device */ + static int snd_mychip_new_pcm(struct mychip *chip) + { + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, &pcm); + if (err < 0) + return err; + pcm->private_data = chip; + strcpy(pcm->name, "My Chip"); + chip->pcm = pcm; + /* set operators */ + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_mychip_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_mychip_capture_ops); + /* pre-allocation of buffers */ + /* NOTE: this may fail */ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(chip->pci), + 64*1024, 64*1024); + return 0; + } + + +PCM Constructor +--------------- + +A pcm instance is allocated by the :c:func:`snd_pcm_new()` +function. It would be better to create a constructor for pcm, namely, + +:: + + static int snd_mychip_new_pcm(struct mychip *chip) + { + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, &pcm); + if (err < 0) + return err; + pcm->private_data = chip; + strcpy(pcm->name, "My Chip"); + chip->pcm = pcm; + .... + return 0; + } + +The :c:func:`snd_pcm_new()` function takes four arguments. The +first argument is the card pointer to which this pcm is assigned, and +the second is the ID string. + +The third argument (``index``, 0 in the above) is the index of this new +pcm. It begins from zero. If you create more than one pcm instances, +specify the different numbers in this argument. For example, ``index = +1`` for the second PCM device. + +The fourth and fifth arguments are the number of substreams for playback +and capture, respectively. Here 1 is used for both arguments. When no +playback or capture substreams are available, pass 0 to the +corresponding argument. + +If a chip supports multiple playbacks or captures, you can specify more +numbers, but they must be handled properly in open/close, etc. +callbacks. When you need to know which substream you are referring to, +then it can be obtained from :c:type:`struct snd_pcm_substream +<snd_pcm_substream>` data passed to each callback as follows: + +:: + + struct snd_pcm_substream *substream; + int index = substream->number; + + +After the pcm is created, you need to set operators for each pcm stream. + +:: + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_mychip_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_mychip_capture_ops); + +The operators are defined typically like this: + +:: + + static struct snd_pcm_ops snd_mychip_playback_ops = { + .open = snd_mychip_pcm_open, + .close = snd_mychip_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_mychip_pcm_hw_params, + .hw_free = snd_mychip_pcm_hw_free, + .prepare = snd_mychip_pcm_prepare, + .trigger = snd_mychip_pcm_trigger, + .pointer = snd_mychip_pcm_pointer, + }; + +All the callbacks are described in the Operators_ subsection. + +After setting the operators, you probably will want to pre-allocate the +buffer. For the pre-allocation, simply call the following: + +:: + + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(chip->pci), + 64*1024, 64*1024); + +It will allocate a buffer up to 64kB as default. Buffer management +details will be described in the later section `Buffer and Memory +Management`_. + +Additionally, you can set some extra information for this pcm in +``pcm->info_flags``. The available values are defined as +``SNDRV_PCM_INFO_XXX`` in ``<sound/asound.h>``, which is used for the +hardware definition (described later). When your soundchip supports only +half-duplex, specify like this: + +:: + + pcm->info_flags = SNDRV_PCM_INFO_HALF_DUPLEX; + + +... And the Destructor? +----------------------- + +The destructor for a pcm instance is not always necessary. Since the pcm +device will be released by the middle layer code automatically, you +don't have to call the destructor explicitly. + +The destructor would be necessary if you created special records +internally and needed to release them. In such a case, set the +destructor function to ``pcm->private_free``: + +:: + + static void mychip_pcm_free(struct snd_pcm *pcm) + { + struct mychip *chip = snd_pcm_chip(pcm); + /* free your own data */ + kfree(chip->my_private_pcm_data); + /* do what you like else */ + .... + } + + static int snd_mychip_new_pcm(struct mychip *chip) + { + struct snd_pcm *pcm; + .... + /* allocate your own data */ + chip->my_private_pcm_data = kmalloc(...); + /* set the destructor */ + pcm->private_data = chip; + pcm->private_free = mychip_pcm_free; + .... + } + + + +Runtime Pointer - The Chest of PCM Information +---------------------------------------------- + +When the PCM substream is opened, a PCM runtime instance is allocated +and assigned to the substream. This pointer is accessible via +``substream->runtime``. This runtime pointer holds most information you +need to control the PCM: the copy of hw_params and sw_params +configurations, the buffer pointers, mmap records, spinlocks, etc. + +The definition of runtime instance is found in ``<sound/pcm.h>``. Here +are the contents of this file: + +:: + + struct _snd_pcm_runtime { + /* -- Status -- */ + struct snd_pcm_substream *trigger_master; + snd_timestamp_t trigger_tstamp; /* trigger timestamp */ + int overrange; + snd_pcm_uframes_t avail_max; + snd_pcm_uframes_t hw_ptr_base; /* Position at buffer restart */ + snd_pcm_uframes_t hw_ptr_interrupt; /* Position at interrupt time*/ + + /* -- HW params -- */ + snd_pcm_access_t access; /* access mode */ + snd_pcm_format_t format; /* SNDRV_PCM_FORMAT_* */ + snd_pcm_subformat_t subformat; /* subformat */ + unsigned int rate; /* rate in Hz */ + unsigned int channels; /* channels */ + snd_pcm_uframes_t period_size; /* period size */ + unsigned int periods; /* periods */ + snd_pcm_uframes_t buffer_size; /* buffer size */ + unsigned int tick_time; /* tick time */ + snd_pcm_uframes_t min_align; /* Min alignment for the format */ + size_t byte_align; + unsigned int frame_bits; + unsigned int sample_bits; + unsigned int info; + unsigned int rate_num; + unsigned int rate_den; + + /* -- SW params -- */ + struct timespec tstamp_mode; /* mmap timestamp is updated */ + unsigned int period_step; + unsigned int sleep_min; /* min ticks to sleep */ + snd_pcm_uframes_t start_threshold; + snd_pcm_uframes_t stop_threshold; + snd_pcm_uframes_t silence_threshold; /* Silence filling happens when + noise is nearest than this */ + snd_pcm_uframes_t silence_size; /* Silence filling size */ + snd_pcm_uframes_t boundary; /* pointers wrap point */ + + snd_pcm_uframes_t silenced_start; + snd_pcm_uframes_t silenced_size; + + snd_pcm_sync_id_t sync; /* hardware synchronization ID */ + + /* -- mmap -- */ + volatile struct snd_pcm_mmap_status *status; + volatile struct snd_pcm_mmap_control *control; + atomic_t mmap_count; + + /* -- locking / scheduling -- */ + spinlock_t lock; + wait_queue_head_t sleep; + struct timer_list tick_timer; + struct fasync_struct *fasync; + + /* -- private section -- */ + void *private_data; + void (*private_free)(struct snd_pcm_runtime *runtime); + + /* -- hardware description -- */ + struct snd_pcm_hardware hw; + struct snd_pcm_hw_constraints hw_constraints; + + /* -- timer -- */ + unsigned int timer_resolution; /* timer resolution */ + + /* -- DMA -- */ + unsigned char *dma_area; /* DMA area */ + dma_addr_t dma_addr; /* physical bus address (not accessible from main CPU) */ + size_t dma_bytes; /* size of DMA area */ + + struct snd_dma_buffer *dma_buffer_p; /* allocated buffer */ + + #if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) + /* -- OSS things -- */ + struct snd_pcm_oss_runtime oss; + #endif + }; + + +For the operators (callbacks) of each sound driver, most of these +records are supposed to be read-only. Only the PCM middle-layer changes +/ updates them. The exceptions are the hardware description (hw) DMA +buffer information and the private data. Besides, if you use the +standard buffer allocation method via +:c:func:`snd_pcm_lib_malloc_pages()`, you don't need to set the +DMA buffer information by yourself. + +In the sections below, important records are explained. + +Hardware Description +~~~~~~~~~~~~~~~~~~~~ + +The hardware descriptor (:c:type:`struct snd_pcm_hardware +<snd_pcm_hardware>`) contains the definitions of the fundamental +hardware configuration. Above all, you'll need to define this in the +`PCM open callback`_. Note that the runtime instance holds the copy of +the descriptor, not the pointer to the existing descriptor. That is, +in the open callback, you can modify the copied descriptor +(``runtime->hw``) as you need. For example, if the maximum number of +channels is 1 only on some chip models, you can still use the same +hardware descriptor and change the channels_max later: + +:: + + struct snd_pcm_runtime *runtime = substream->runtime; + ... + runtime->hw = snd_mychip_playback_hw; /* common definition */ + if (chip->model == VERY_OLD_ONE) + runtime->hw.channels_max = 1; + +Typically, you'll have a hardware descriptor as below: + +:: + + static struct snd_pcm_hardware snd_mychip_playback_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 32768, + .period_bytes_min = 4096, + .period_bytes_max = 32768, + .periods_min = 1, + .periods_max = 1024, + }; + +- The ``info`` field contains the type and capabilities of this + pcm. The bit flags are defined in ``<sound/asound.h>`` as + ``SNDRV_PCM_INFO_XXX``. Here, at least, you have to specify whether + the mmap is supported and which interleaved format is + supported. When the hardware supports mmap, add the + ``SNDRV_PCM_INFO_MMAP`` flag here. When the hardware supports the + interleaved or the non-interleaved formats, + ``SNDRV_PCM_INFO_INTERLEAVED`` or ``SNDRV_PCM_INFO_NONINTERLEAVED`` + flag must be set, respectively. If both are supported, you can set + both, too. + + In the above example, ``MMAP_VALID`` and ``BLOCK_TRANSFER`` are + specified for the OSS mmap mode. Usually both are set. Of course, + ``MMAP_VALID`` is set only if the mmap is really supported. + + The other possible flags are ``SNDRV_PCM_INFO_PAUSE`` and + ``SNDRV_PCM_INFO_RESUME``. The ``PAUSE`` bit means that the pcm + supports the “pause” operation, while the ``RESUME`` bit means that + the pcm supports the full “suspend/resume” operation. If the + ``PAUSE`` flag is set, the ``trigger`` callback below must handle + the corresponding (pause push/release) commands. The suspend/resume + trigger commands can be defined even without the ``RESUME`` + flag. See `Power Management`_ section for details. + + When the PCM substreams can be synchronized (typically, + synchronized start/stop of a playback and a capture streams), you + can give ``SNDRV_PCM_INFO_SYNC_START``, too. In this case, you'll + need to check the linked-list of PCM substreams in the trigger + callback. This will be described in the later section. + +- ``formats`` field contains the bit-flags of supported formats + (``SNDRV_PCM_FMTBIT_XXX``). If the hardware supports more than one + format, give all or'ed bits. In the example above, the signed 16bit + little-endian format is specified. + +- ``rates`` field contains the bit-flags of supported rates + (``SNDRV_PCM_RATE_XXX``). When the chip supports continuous rates, + pass ``CONTINUOUS`` bit additionally. The pre-defined rate bits are + provided only for typical rates. If your chip supports + unconventional rates, you need to add the ``KNOT`` bit and set up + the hardware constraint manually (explained later). + +- ``rate_min`` and ``rate_max`` define the minimum and maximum sample + rate. This should correspond somehow to ``rates`` bits. + +- ``channel_min`` and ``channel_max`` define, as you might already + expected, the minimum and maximum number of channels. + +- ``buffer_bytes_max`` defines the maximum buffer size in + bytes. There is no ``buffer_bytes_min`` field, since it can be + calculated from the minimum period size and the minimum number of + periods. Meanwhile, ``period_bytes_min`` and define the minimum and + maximum size of the period in bytes. ``periods_max`` and + ``periods_min`` define the maximum and minimum number of periods in + the buffer. + + The “period” is a term that corresponds to a fragment in the OSS + world. The period defines the size at which a PCM interrupt is + generated. This size strongly depends on the hardware. Generally, + the smaller period size will give you more interrupts, that is, + more controls. In the case of capture, this size defines the input + latency. On the other hand, the whole buffer size defines the + output latency for the playback direction. + +- There is also a field ``fifo_size``. This specifies the size of the + hardware FIFO, but currently it is neither used in the driver nor + in the alsa-lib. So, you can ignore this field. + +PCM Configurations +~~~~~~~~~~~~~~~~~~ + +Ok, let's go back again to the PCM runtime records. The most +frequently referred records in the runtime instance are the PCM +configurations. The PCM configurations are stored in the runtime +instance after the application sends ``hw_params`` data via +alsa-lib. There are many fields copied from hw_params and sw_params +structs. For example, ``format`` holds the format type chosen by the +application. This field contains the enum value +``SNDRV_PCM_FORMAT_XXX``. + +One thing to be noted is that the configured buffer and period sizes +are stored in “frames” in the runtime. In the ALSA world, ``1 frame = +channels * samples-size``. For conversion between frames and bytes, +you can use the :c:func:`frames_to_bytes()` and +:c:func:`bytes_to_frames()` helper functions. + +:: + + period_bytes = frames_to_bytes(runtime, runtime->period_size); + +Also, many software parameters (sw_params) are stored in frames, too. +Please check the type of the field. ``snd_pcm_uframes_t`` is for the +frames as unsigned integer while ``snd_pcm_sframes_t`` is for the +frames as signed integer. + +DMA Buffer Information +~~~~~~~~~~~~~~~~~~~~~~ + +The DMA buffer is defined by the following four fields, ``dma_area``, +``dma_addr``, ``dma_bytes`` and ``dma_private``. The ``dma_area`` +holds the buffer pointer (the logical address). You can call +:c:func:`memcpy()` from/to this pointer. Meanwhile, ``dma_addr`` holds +the physical address of the buffer. This field is specified only when +the buffer is a linear buffer. ``dma_bytes`` holds the size of buffer +in bytes. ``dma_private`` is used for the ALSA DMA allocator. + +If you use a standard ALSA function, +:c:func:`snd_pcm_lib_malloc_pages()`, for allocating the buffer, +these fields are set by the ALSA middle layer, and you should *not* +change them by yourself. You can read them but not write them. On the +other hand, if you want to allocate the buffer by yourself, you'll +need to manage it in hw_params callback. At least, ``dma_bytes`` is +mandatory. ``dma_area`` is necessary when the buffer is mmapped. If +your driver doesn't support mmap, this field is not +necessary. ``dma_addr`` is also optional. You can use dma_private as +you like, too. + +Running Status +~~~~~~~~~~~~~~ + +The running status can be referred via ``runtime->status``. This is +the pointer to the :c:type:`struct snd_pcm_mmap_status +<snd_pcm_mmap_status>` record. For example, you can get the current +DMA hardware pointer via ``runtime->status->hw_ptr``. + +The DMA application pointer can be referred via ``runtime->control``, +which points to the :c:type:`struct snd_pcm_mmap_control +<snd_pcm_mmap_control>` record. However, accessing directly to +this value is not recommended. + +Private Data +~~~~~~~~~~~~ + +You can allocate a record for the substream and store it in +``runtime->private_data``. Usually, this is done in the `PCM open +callback`_. Don't mix this with ``pcm->private_data``. The +``pcm->private_data`` usually points to the chip instance assigned +statically at the creation of PCM, while the ``runtime->private_data`` +points to a dynamic data structure created at the PCM open +callback. + +:: + + static int snd_xxx_open(struct snd_pcm_substream *substream) + { + struct my_pcm_data *data; + .... + data = kmalloc(sizeof(*data), GFP_KERNEL); + substream->runtime->private_data = data; + .... + } + + +The allocated object must be released in the `close callback`_. + +Operators +--------- + +OK, now let me give details about each pcm callback (``ops``). In +general, every callback must return 0 if successful, or a negative +error number such as ``-EINVAL``. To choose an appropriate error +number, it is advised to check what value other parts of the kernel +return when the same kind of request fails. + +The callback function takes at least the argument with :c:type:`struct +snd_pcm_substream <snd_pcm_substream>` pointer. To retrieve the chip +record from the given substream instance, you can use the following +macro. + +:: + + int xxx() { + struct mychip *chip = snd_pcm_substream_chip(substream); + .... + } + +The macro reads ``substream->private_data``, which is a copy of +``pcm->private_data``. You can override the former if you need to +assign different data records per PCM substream. For example, the +cmi8330 driver assigns different ``private_data`` for playback and +capture directions, because it uses two different codecs (SB- and +AD-compatible) for different directions. + +PCM open callback +~~~~~~~~~~~~~~~~~ + +:: + + static int snd_xxx_open(struct snd_pcm_substream *substream); + +This is called when a pcm substream is opened. + +At least, here you have to initialize the ``runtime->hw`` +record. Typically, this is done by like this: + +:: + + static int snd_xxx_open(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_mychip_playback_hw; + return 0; + } + +where ``snd_mychip_playback_hw`` is the pre-defined hardware +description. + +You can allocate a private data in this callback, as described in +`Private Data`_ section. + +If the hardware configuration needs more constraints, set the hardware +constraints here, too. See Constraints_ for more details. + +close callback +~~~~~~~~~~~~~~ + +:: + + static int snd_xxx_close(struct snd_pcm_substream *substream); + + +Obviously, this is called when a pcm substream is closed. + +Any private instance for a pcm substream allocated in the ``open`` +callback will be released here. + +:: + + static int snd_xxx_close(struct snd_pcm_substream *substream) + { + .... + kfree(substream->runtime->private_data); + .... + } + +ioctl callback +~~~~~~~~~~~~~~ + +This is used for any special call to pcm ioctls. But usually you can +pass a generic ioctl callback, :c:func:`snd_pcm_lib_ioctl()`. + +hw_params callback +~~~~~~~~~~~~~~~~~~~ + +:: + + static int snd_xxx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params); + +This is called when the hardware parameter (``hw_params``) is set up +by the application, that is, once when the buffer size, the period +size, the format, etc. are defined for the pcm substream. + +Many hardware setups should be done in this callback, including the +allocation of buffers. + +Parameters to be initialized are retrieved by +:c:func:`params_xxx()` macros. To allocate buffer, you can call a +helper function, + +:: + + snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); + +:c:func:`snd_pcm_lib_malloc_pages()` is available only when the +DMA buffers have been pre-allocated. See the section `Buffer Types`_ +for more details. + +Note that this and ``prepare`` callbacks may be called multiple times +per initialization. For example, the OSS emulation may call these +callbacks at each change via its ioctl. + +Thus, you need to be careful not to allocate the same buffers many +times, which will lead to memory leaks! Calling the helper function +above many times is OK. It will release the previous buffer +automatically when it was already allocated. + +Another note is that this callback is non-atomic (schedulable) as +default, i.e. when no ``nonatomic`` flag set. This is important, +because the ``trigger`` callback is atomic (non-schedulable). That is, +mutexes or any schedule-related functions are not available in +``trigger`` callback. Please see the subsection Atomicity_ for +details. + +hw_free callback +~~~~~~~~~~~~~~~~~ + +:: + + static int snd_xxx_hw_free(struct snd_pcm_substream *substream); + +This is called to release the resources allocated via +``hw_params``. For example, releasing the buffer via +:c:func:`snd_pcm_lib_malloc_pages()` is done by calling the +following: + +:: + + snd_pcm_lib_free_pages(substream); + +This function is always called before the close callback is called. +Also, the callback may be called multiple times, too. Keep track +whether the resource was already released. + +prepare callback +~~~~~~~~~~~~~~~~ + +:: + + static int snd_xxx_prepare(struct snd_pcm_substream *substream); + +This callback is called when the pcm is “prepared”. You can set the +format type, sample rate, etc. here. The difference from ``hw_params`` +is that the ``prepare`` callback will be called each time +:c:func:`snd_pcm_prepare()` is called, i.e. when recovering after +underruns, etc. + +Note that this callback is now non-atomic. You can use +schedule-related functions safely in this callback. + +In this and the following callbacks, you can refer to the values via +the runtime record, ``substream->runtime``. For example, to get the +current rate, format or channels, access to ``runtime->rate``, +``runtime->format`` or ``runtime->channels``, respectively. The +physical address of the allocated buffer is set to +``runtime->dma_area``. The buffer and period sizes are in +``runtime->buffer_size`` and ``runtime->period_size``, respectively. + +Be careful that this callback will be called many times at each setup, +too. + +trigger callback +~~~~~~~~~~~~~~~~ + +:: + + static int snd_xxx_trigger(struct snd_pcm_substream *substream, int cmd); + +This is called when the pcm is started, stopped or paused. + +Which action is specified in the second argument, +``SNDRV_PCM_TRIGGER_XXX`` in ``<sound/pcm.h>``. At least, the ``START`` +and ``STOP`` commands must be defined in this callback. + +:: + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* do something to start the PCM engine */ + break; + case SNDRV_PCM_TRIGGER_STOP: + /* do something to stop the PCM engine */ + break; + default: + return -EINVAL; + } + +When the pcm supports the pause operation (given in the info field of +the hardware table), the ``PAUSE_PUSH`` and ``PAUSE_RELEASE`` commands +must be handled here, too. The former is the command to pause the pcm, +and the latter to restart the pcm again. + +When the pcm supports the suspend/resume operation, regardless of full +or partial suspend/resume support, the ``SUSPEND`` and ``RESUME`` +commands must be handled, too. These commands are issued when the +power-management status is changed. Obviously, the ``SUSPEND`` and +``RESUME`` commands suspend and resume the pcm substream, and usually, +they are identical to the ``STOP`` and ``START`` commands, respectively. +See the `Power Management`_ section for details. + +As mentioned, this callback is atomic as default unless ``nonatomic`` +flag set, and you cannot call functions which may sleep. The +``trigger`` callback should be as minimal as possible, just really +triggering the DMA. The other stuff should be initialized +``hw_params`` and ``prepare`` callbacks properly beforehand. + +pointer callback +~~~~~~~~~~~~~~~~ + +:: + + static snd_pcm_uframes_t snd_xxx_pointer(struct snd_pcm_substream *substream) + +This callback is called when the PCM middle layer inquires the current +hardware position on the buffer. The position must be returned in +frames, ranging from 0 to ``buffer_size - 1``. + +This is called usually from the buffer-update routine in the pcm +middle layer, which is invoked when :c:func:`snd_pcm_period_elapsed()` +is called in the interrupt routine. Then the pcm middle layer updates +the position and calculates the available space, and wakes up the +sleeping poll threads, etc. + +This callback is also atomic as default. + +copy and silence callbacks +~~~~~~~~~~~~~~~~~~~~~~~~~~ + +These callbacks are not mandatory, and can be omitted in most cases. +These callbacks are used when the hardware buffer cannot be in the +normal memory space. Some chips have their own buffer on the hardware +which is not mappable. In such a case, you have to transfer the data +manually from the memory buffer to the hardware buffer. Or, if the +buffer is non-contiguous on both physical and virtual memory spaces, +these callbacks must be defined, too. + +If these two callbacks are defined, copy and set-silence operations +are done by them. The detailed will be described in the later section +`Buffer and Memory Management`_. + +ack callback +~~~~~~~~~~~~ + +This callback is also not mandatory. This callback is called when the +``appl_ptr`` is updated in read or write operations. Some drivers like +emu10k1-fx and cs46xx need to track the current ``appl_ptr`` for the +internal buffer, and this callback is useful only for such a purpose. + +This callback is atomic as default. + +page callback +~~~~~~~~~~~~~ + +This callback is optional too. This callback is used mainly for +non-contiguous buffers. The mmap calls this callback to get the page +address. Some examples will be explained in the later section `Buffer +and Memory Management`_, too. + +PCM Interrupt Handler +--------------------- + +The rest of pcm stuff is the PCM interrupt handler. The role of PCM +interrupt handler in the sound driver is to update the buffer position +and to tell the PCM middle layer when the buffer position goes across +the prescribed period size. To inform this, call the +:c:func:`snd_pcm_period_elapsed()` function. + +There are several types of sound chips to generate the interrupts. + +Interrupts at the period (fragment) boundary +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +This is the most frequently found type: the hardware generates an +interrupt at each period boundary. In this case, you can call +:c:func:`snd_pcm_period_elapsed()` at each interrupt. + +:c:func:`snd_pcm_period_elapsed()` takes the substream pointer as +its argument. Thus, you need to keep the substream pointer accessible +from the chip instance. For example, define ``substream`` field in the +chip record to hold the current running substream pointer, and set the +pointer value at ``open`` callback (and reset at ``close`` callback). + +If you acquire a spinlock in the interrupt handler, and the lock is used +in other pcm callbacks, too, then you have to release the lock before +calling :c:func:`snd_pcm_period_elapsed()`, because +:c:func:`snd_pcm_period_elapsed()` calls other pcm callbacks +inside. + +Typical code would be like: + +:: + + + static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id) + { + struct mychip *chip = dev_id; + spin_lock(&chip->lock); + .... + if (pcm_irq_invoked(chip)) { + /* call updater, unlock before it */ + spin_unlock(&chip->lock); + snd_pcm_period_elapsed(chip->substream); + spin_lock(&chip->lock); + /* acknowledge the interrupt if necessary */ + } + .... + spin_unlock(&chip->lock); + return IRQ_HANDLED; + } + + + +High frequency timer interrupts +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +This happens when the hardware doesn't generate interrupts at the period +boundary but issues timer interrupts at a fixed timer rate (e.g. es1968 +or ymfpci drivers). In this case, you need to check the current hardware +position and accumulate the processed sample length at each interrupt. +When the accumulated size exceeds the period size, call +:c:func:`snd_pcm_period_elapsed()` and reset the accumulator. + +Typical code would be like the following. + +:: + + + static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id) + { + struct mychip *chip = dev_id; + spin_lock(&chip->lock); + .... + if (pcm_irq_invoked(chip)) { + unsigned int last_ptr, size; + /* get the current hardware pointer (in frames) */ + last_ptr = get_hw_ptr(chip); + /* calculate the processed frames since the + * last update + */ + if (last_ptr < chip->last_ptr) + size = runtime->buffer_size + last_ptr + - chip->last_ptr; + else + size = last_ptr - chip->last_ptr; + /* remember the last updated point */ + chip->last_ptr = last_ptr; + /* accumulate the size */ + chip->size += size; + /* over the period boundary? */ + if (chip->size >= runtime->period_size) { + /* reset the accumulator */ + chip->size %= runtime->period_size; + /* call updater */ + spin_unlock(&chip->lock); + snd_pcm_period_elapsed(substream); + spin_lock(&chip->lock); + } + /* acknowledge the interrupt if necessary */ + } + .... + spin_unlock(&chip->lock); + return IRQ_HANDLED; + } + + + +On calling :c:func:`snd_pcm_period_elapsed()` +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +In both cases, even if more than one period are elapsed, you don't have +to call :c:func:`snd_pcm_period_elapsed()` many times. Call only +once. And the pcm layer will check the current hardware pointer and +update to the latest status. + +Atomicity +--------- + +One of the most important (and thus difficult to debug) problems in +kernel programming are race conditions. In the Linux kernel, they are +usually avoided via spin-locks, mutexes or semaphores. In general, if a +race condition can happen in an interrupt handler, it has to be managed +atomically, and you have to use a spinlock to protect the critical +session. If the critical section is not in interrupt handler code and if +taking a relatively long time to execute is acceptable, you should use +mutexes or semaphores instead. + +As already seen, some pcm callbacks are atomic and some are not. For +example, the ``hw_params`` callback is non-atomic, while ``trigger`` +callback is atomic. This means, the latter is called already in a +spinlock held by the PCM middle layer. Please take this atomicity into +account when you choose a locking scheme in the callbacks. + +In the atomic callbacks, you cannot use functions which may call +:c:func:`schedule()` or go to :c:func:`sleep()`. Semaphores and +mutexes can sleep, and hence they cannot be used inside the atomic +callbacks (e.g. ``trigger`` callback). To implement some delay in such a +callback, please use :c:func:`udelay()` or :c:func:`mdelay()`. + +All three atomic callbacks (trigger, pointer, and ack) are called with +local interrupts disabled. + +The recent changes in PCM core code, however, allow all PCM operations +to be non-atomic. This assumes that the all caller sides are in +non-atomic contexts. For example, the function +:c:func:`snd_pcm_period_elapsed()` is called typically from the +interrupt handler. But, if you set up the driver to use a threaded +interrupt handler, this call can be in non-atomic context, too. In such +a case, you can set ``nonatomic`` filed of :c:type:`struct snd_pcm +<snd_pcm>` object after creating it. When this flag is set, mutex +and rwsem are used internally in the PCM core instead of spin and +rwlocks, so that you can call all PCM functions safely in a non-atomic +context. + +Constraints +----------- + +If your chip supports unconventional sample rates, or only the limited +samples, you need to set a constraint for the condition. + +For example, in order to restrict the sample rates in the some supported +values, use :c:func:`snd_pcm_hw_constraint_list()`. You need to +call this function in the open callback. + +:: + + static unsigned int rates[] = + {4000, 10000, 22050, 44100}; + static struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, + }; + + static int snd_mychip_pcm_open(struct snd_pcm_substream *substream) + { + int err; + .... + err = snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + if (err < 0) + return err; + .... + } + + + +There are many different constraints. Look at ``sound/pcm.h`` for a +complete list. You can even define your own constraint rules. For +example, let's suppose my_chip can manage a substream of 1 channel if +and only if the format is ``S16_LE``, otherwise it supports any format +specified in the :c:type:`struct snd_pcm_hardware +<snd_pcm_hardware>` structure (or in any other +constraint_list). You can build a rule like this: + +:: + + static int hw_rule_channels_by_format(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) + { + struct snd_interval *c = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + struct snd_interval ch; + + snd_interval_any(&ch); + if (f->bits[0] == SNDRV_PCM_FMTBIT_S16_LE) { + ch.min = ch.max = 1; + ch.integer = 1; + return snd_interval_refine(c, &ch); + } + return 0; + } + + +Then you need to call this function to add your rule: + +:: + + snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + hw_rule_channels_by_format, NULL, + SNDRV_PCM_HW_PARAM_FORMAT, -1); + +The rule function is called when an application sets the PCM format, and +it refines the number of channels accordingly. But an application may +set the number of channels before setting the format. Thus you also need +to define the inverse rule: + +:: + + static int hw_rule_format_by_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) + { + struct snd_interval *c = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + struct snd_mask fmt; + + snd_mask_any(&fmt); /* Init the struct */ + if (c->min < 2) { + fmt.bits[0] &= SNDRV_PCM_FMTBIT_S16_LE; + return snd_mask_refine(f, &fmt); + } + return 0; + } + + +... and in the open callback: + +:: + + snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + hw_rule_format_by_channels, NULL, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + +I won't give more details here, rather I would like to say, “Luke, use +the source.” + +Control Interface +================= + +General +------- + +The control interface is used widely for many switches, sliders, etc. +which are accessed from user-space. Its most important use is the mixer +interface. In other words, since ALSA 0.9.x, all the mixer stuff is +implemented on the control kernel API. + +ALSA has a well-defined AC97 control module. If your chip supports only +the AC97 and nothing else, you can skip this section. + +The control API is defined in ``<sound/control.h>``. Include this file +if you want to add your own controls. + +Definition of Controls +---------------------- + +To create a new control, you need to define the following three +callbacks: ``info``, ``get`` and ``put``. Then, define a +:c:type:`struct snd_kcontrol_new <snd_kcontrol_new>` record, such as: + +:: + + + static struct snd_kcontrol_new my_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Switch", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = 0xffff, + .info = my_control_info, + .get = my_control_get, + .put = my_control_put + }; + + +The ``iface`` field specifies the control type, +``SNDRV_CTL_ELEM_IFACE_XXX``, which is usually ``MIXER``. Use ``CARD`` +for global controls that are not logically part of the mixer. If the +control is closely associated with some specific device on the sound +card, use ``HWDEP``, ``PCM``, ``RAWMIDI``, ``TIMER``, or ``SEQUENCER``, +and specify the device number with the ``device`` and ``subdevice`` +fields. + +The ``name`` is the name identifier string. Since ALSA 0.9.x, the +control name is very important, because its role is classified from +its name. There are pre-defined standard control names. The details +are described in the `Control Names`_ subsection. + +The ``index`` field holds the index number of this control. If there +are several different controls with the same name, they can be +distinguished by the index number. This is the case when several +codecs exist on the card. If the index is zero, you can omit the +definition above. + +The ``access`` field contains the access type of this control. Give +the combination of bit masks, ``SNDRV_CTL_ELEM_ACCESS_XXX``, +there. The details will be explained in the `Access Flags`_ +subsection. + +The ``private_value`` field contains an arbitrary long integer value +for this record. When using the generic ``info``, ``get`` and ``put`` +callbacks, you can pass a value through this field. If several small +numbers are necessary, you can combine them in bitwise. Or, it's +possible to give a pointer (casted to unsigned long) of some record to +this field, too. + +The ``tlv`` field can be used to provide metadata about the control; +see the `Metadata`_ subsection. + +The other three are `Control Callbacks`_. + +Control Names +------------- + +There are some standards to define the control names. A control is +usually defined from the three parts as “SOURCE DIRECTION FUNCTION”. + +The first, ``SOURCE``, specifies the source of the control, and is a +string such as “Master”, “PCM”, “CD” and “Line”. There are many +pre-defined sources. + +The second, ``DIRECTION``, is one of the following strings according to +the direction of the control: “Playback”, “Capture”, “Bypass Playback” +and “Bypass Capture”. Or, it can be omitted, meaning both playback and +capture directions. + +The third, ``FUNCTION``, is one of the following strings according to +the function of the control: “Switch”, “Volume” and “Route”. + +The example of control names are, thus, “Master Capture Switch” or “PCM +Playback Volume”. + +There are some exceptions: + +Global capture and playback +~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +“Capture Source”, “Capture Switch” and “Capture Volume” are used for the +global capture (input) source, switch and volume. Similarly, “Playback +Switch” and “Playback Volume” are used for the global output gain switch +and volume. + +Tone-controls +~~~~~~~~~~~~~ + +tone-control switch and volumes are specified like “Tone Control - XXX”, +e.g. “Tone Control - Switch”, “Tone Control - Bass”, “Tone Control - +Center”. + +3D controls +~~~~~~~~~~~ + +3D-control switches and volumes are specified like “3D Control - XXX”, +e.g. “3D Control - Switch”, “3D Control - Center”, “3D Control - Space”. + +Mic boost +~~~~~~~~~ + +Mic-boost switch is set as “Mic Boost” or “Mic Boost (6dB)”. + +More precise information can be found in +``Documentation/sound/alsa/ControlNames.txt``. + +Access Flags +------------ + +The access flag is the bitmask which specifies the access type of the +given control. The default access type is +``SNDRV_CTL_ELEM_ACCESS_READWRITE``, which means both read and write are +allowed to this control. When the access flag is omitted (i.e. = 0), it +is considered as ``READWRITE`` access as default. + +When the control is read-only, pass ``SNDRV_CTL_ELEM_ACCESS_READ`` +instead. In this case, you don't have to define the ``put`` callback. +Similarly, when the control is write-only (although it's a rare case), +you can use the ``WRITE`` flag instead, and you don't need the ``get`` +callback. + +If the control value changes frequently (e.g. the VU meter), +``VOLATILE`` flag should be given. This means that the control may be +changed without `Change notification`_. Applications should poll such +a control constantly. + +When the control is inactive, set the ``INACTIVE`` flag, too. There are +``LOCK`` and ``OWNER`` flags to change the write permissions. + +Control Callbacks +----------------- + +info callback +~~~~~~~~~~~~~ + +The ``info`` callback is used to get detailed information on this +control. This must store the values of the given :c:type:`struct +snd_ctl_elem_info <snd_ctl_elem_info>` object. For example, +for a boolean control with a single element: + +:: + + + static int snd_myctl_mono_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) + { + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; + } + + + +The ``type`` field specifies the type of the control. There are +``BOOLEAN``, ``INTEGER``, ``ENUMERATED``, ``BYTES``, ``IEC958`` and +``INTEGER64``. The ``count`` field specifies the number of elements in +this control. For example, a stereo volume would have count = 2. The +``value`` field is a union, and the values stored are depending on the +type. The boolean and integer types are identical. + +The enumerated type is a bit different from others. You'll need to set +the string for the currently given item index. + +:: + + static int snd_myctl_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) + { + static char *texts[4] = { + "First", "Second", "Third", "Fourth" + }; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 4; + if (uinfo->value.enumerated.item > 3) + uinfo->value.enumerated.item = 3; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + return 0; + } + +The above callback can be simplified with a helper function, +:c:func:`snd_ctl_enum_info()`. The final code looks like below. +(You can pass ``ARRAY_SIZE(texts)`` instead of 4 in the third argument; +it's a matter of taste.) + +:: + + static int snd_myctl_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) + { + static char *texts[4] = { + "First", "Second", "Third", "Fourth" + }; + return snd_ctl_enum_info(uinfo, 1, 4, texts); + } + + +Some common info callbacks are available for your convenience: +:c:func:`snd_ctl_boolean_mono_info()` and +:c:func:`snd_ctl_boolean_stereo_info()`. Obviously, the former +is an info callback for a mono channel boolean item, just like +:c:func:`snd_myctl_mono_info()` above, and the latter is for a +stereo channel boolean item. + +get callback +~~~~~~~~~~~~ + +This callback is used to read the current value of the control and to +return to user-space. + +For example, + +:: + + + static int snd_myctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) + { + struct mychip *chip = snd_kcontrol_chip(kcontrol); + ucontrol->value.integer.value[0] = get_some_value(chip); + return 0; + } + + + +The ``value`` field depends on the type of control as well as on the +info callback. For example, the sb driver uses this field to store the +register offset, the bit-shift and the bit-mask. The ``private_value`` +field is set as follows: + +:: + + .private_value = reg | (shift << 16) | (mask << 24) + +and is retrieved in callbacks like + +:: + + static int snd_sbmixer_get_single(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) + { + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 16) & 0xff; + int mask = (kcontrol->private_value >> 24) & 0xff; + .... + } + +In the ``get`` callback, you have to fill all the elements if the +control has more than one elements, i.e. ``count > 1``. In the example +above, we filled only one element (``value.integer.value[0]``) since +it's assumed as ``count = 1``. + +put callback +~~~~~~~~~~~~ + +This callback is used to write a value from user-space. + +For example, + +:: + + + static int snd_myctl_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) + { + struct mychip *chip = snd_kcontrol_chip(kcontrol); + int changed = 0; + if (chip->current_value != + ucontrol->value.integer.value[0]) { + change_current_value(chip, + ucontrol->value.integer.value[0]); + changed = 1; + } + return changed; + } + + + +As seen above, you have to return 1 if the value is changed. If the +value is not changed, return 0 instead. If any fatal error happens, +return a negative error code as usual. + +As in the ``get`` callback, when the control has more than one +elements, all elements must be evaluated in this callback, too. + +Callbacks are not atomic +~~~~~~~~~~~~~~~~~~~~~~~~ + +All these three callbacks are basically not atomic. + +Control Constructor +------------------- + +When everything is ready, finally we can create a new control. To create +a control, there are two functions to be called, +:c:func:`snd_ctl_new1()` and :c:func:`snd_ctl_add()`. + +In the simplest way, you can do like this: + +:: + + err = snd_ctl_add(card, snd_ctl_new1(&my_control, chip)); + if (err < 0) + return err; + +where ``my_control`` is the :c:type:`struct snd_kcontrol_new +<snd_kcontrol_new>` object defined above, and chip is the object +pointer to be passed to kcontrol->private_data which can be referred +to in callbacks. + +:c:func:`snd_ctl_new1()` allocates a new :c:type:`struct +snd_kcontrol <snd_kcontrol>` instance, and +:c:func:`snd_ctl_add()` assigns the given control component to the +card. + +Change Notification +------------------- + +If you need to change and update a control in the interrupt routine, you +can call :c:func:`snd_ctl_notify()`. For example, + +:: + + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, id_pointer); + +This function takes the card pointer, the event-mask, and the control id +pointer for the notification. The event-mask specifies the types of +notification, for example, in the above example, the change of control +values is notified. The id pointer is the pointer of :c:type:`struct +snd_ctl_elem_id <snd_ctl_elem_id>` to be notified. You can +find some examples in ``es1938.c`` or ``es1968.c`` for hardware volume +interrupts. + +Metadata +-------- + +To provide information about the dB values of a mixer control, use on of +the ``DECLARE_TLV_xxx`` macros from ``<sound/tlv.h>`` to define a +variable containing this information, set the ``tlv.p`` field to point to +this variable, and include the ``SNDRV_CTL_ELEM_ACCESS_TLV_READ`` flag +in the ``access`` field; like this: + +:: + + static DECLARE_TLV_DB_SCALE(db_scale_my_control, -4050, 150, 0); + + static struct snd_kcontrol_new my_control = { + ... + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + ... + .tlv.p = db_scale_my_control, + }; + + +The :c:func:`DECLARE_TLV_DB_SCALE()` macro defines information +about a mixer control where each step in the control's value changes the +dB value by a constant dB amount. The first parameter is the name of the +variable to be defined. The second parameter is the minimum value, in +units of 0.01 dB. The third parameter is the step size, in units of 0.01 +dB. Set the fourth parameter to 1 if the minimum value actually mutes +the control. + +The :c:func:`DECLARE_TLV_DB_LINEAR()` macro defines information +about a mixer control where the control's value affects the output +linearly. The first parameter is the name of the variable to be defined. +The second parameter is the minimum value, in units of 0.01 dB. The +third parameter is the maximum value, in units of 0.01 dB. If the +minimum value mutes the control, set the second parameter to +``TLV_DB_GAIN_MUTE``. + +API for AC97 Codec +================== + +General +------- + +The ALSA AC97 codec layer is a well-defined one, and you don't have to +write much code to control it. Only low-level control routines are +necessary. The AC97 codec API is defined in ``<sound/ac97_codec.h>``. + +Full Code Example +----------------- + +:: + + struct mychip { + .... + struct snd_ac97 *ac97; + .... + }; + + static unsigned short snd_mychip_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) + { + struct mychip *chip = ac97->private_data; + .... + /* read a register value here from the codec */ + return the_register_value; + } + + static void snd_mychip_ac97_write(struct snd_ac97 *ac97, + unsigned short reg, unsigned short val) + { + struct mychip *chip = ac97->private_data; + .... + /* write the given register value to the codec */ + } + + static int snd_mychip_ac97(struct mychip *chip) + { + struct snd_ac97_bus *bus; + struct snd_ac97_template ac97; + int err; + static struct snd_ac97_bus_ops ops = { + .write = snd_mychip_ac97_write, + .read = snd_mychip_ac97_read, + }; + + err = snd_ac97_bus(chip->card, 0, &ops, NULL, &bus); + if (err < 0) + return err; + memset(&ac97, 0, sizeof(ac97)); + ac97.private_data = chip; + return snd_ac97_mixer(bus, &ac97, &chip->ac97); + } + + +AC97 Constructor +---------------- + +To create an ac97 instance, first call :c:func:`snd_ac97_bus()` +with an ``ac97_bus_ops_t`` record with callback functions. + +:: + + struct snd_ac97_bus *bus; + static struct snd_ac97_bus_ops ops = { + .write = snd_mychip_ac97_write, + .read = snd_mychip_ac97_read, + }; + + snd_ac97_bus(card, 0, &ops, NULL, &pbus); + +The bus record is shared among all belonging ac97 instances. + +And then call :c:func:`snd_ac97_mixer()` with an :c:type:`struct +snd_ac97_template <snd_ac97_template>` record together with +the bus pointer created above. + +:: + + struct snd_ac97_template ac97; + int err; + + memset(&ac97, 0, sizeof(ac97)); + ac97.private_data = chip; + snd_ac97_mixer(bus, &ac97, &chip->ac97); + +where chip->ac97 is a pointer to a newly created ``ac97_t`` +instance. In this case, the chip pointer is set as the private data, +so that the read/write callback functions can refer to this chip +instance. This instance is not necessarily stored in the chip +record. If you need to change the register values from the driver, or +need the suspend/resume of ac97 codecs, keep this pointer to pass to +the corresponding functions. + +AC97 Callbacks +-------------- + +The standard callbacks are ``read`` and ``write``. Obviously they +correspond to the functions for read and write accesses to the +hardware low-level codes. + +The ``read`` callback returns the register value specified in the +argument. + +:: + + static unsigned short snd_mychip_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) + { + struct mychip *chip = ac97->private_data; + .... + return the_register_value; + } + +Here, the chip can be cast from ``ac97->private_data``. + +Meanwhile, the ``write`` callback is used to set the register +value + +:: + + static void snd_mychip_ac97_write(struct snd_ac97 *ac97, + unsigned short reg, unsigned short val) + + +These callbacks are non-atomic like the control API callbacks. + +There are also other callbacks: ``reset``, ``wait`` and ``init``. + +The ``reset`` callback is used to reset the codec. If the chip +requires a special kind of reset, you can define this callback. + +The ``wait`` callback is used to add some waiting time in the standard +initialization of the codec. If the chip requires the extra waiting +time, define this callback. + +The ``init`` callback is used for additional initialization of the +codec. + +Updating Registers in The Driver +-------------------------------- + +If you need to access to the codec from the driver, you can call the +following functions: :c:func:`snd_ac97_write()`, +:c:func:`snd_ac97_read()`, :c:func:`snd_ac97_update()` and +:c:func:`snd_ac97_update_bits()`. + +Both :c:func:`snd_ac97_write()` and +:c:func:`snd_ac97_update()` functions are used to set a value to +the given register (``AC97_XXX``). The difference between them is that +:c:func:`snd_ac97_update()` doesn't write a value if the given +value has been already set, while :c:func:`snd_ac97_write()` +always rewrites the value. + +:: + + snd_ac97_write(ac97, AC97_MASTER, 0x8080); + snd_ac97_update(ac97, AC97_MASTER, 0x8080); + +:c:func:`snd_ac97_read()` is used to read the value of the given +register. For example, + +:: + + value = snd_ac97_read(ac97, AC97_MASTER); + +:c:func:`snd_ac97_update_bits()` is used to update some bits in +the given register. + +:: + + snd_ac97_update_bits(ac97, reg, mask, value); + +Also, there is a function to change the sample rate (of a given register +such as ``AC97_PCM_FRONT_DAC_RATE``) when VRA or DRA is supported by the +codec: :c:func:`snd_ac97_set_rate()`. + +:: + + snd_ac97_set_rate(ac97, AC97_PCM_FRONT_DAC_RATE, 44100); + + +The following registers are available to set the rate: +``AC97_PCM_MIC_ADC_RATE``, ``AC97_PCM_FRONT_DAC_RATE``, +``AC97_PCM_LR_ADC_RATE``, ``AC97_SPDIF``. When ``AC97_SPDIF`` is +specified, the register is not really changed but the corresponding +IEC958 status bits will be updated. + +Clock Adjustment +---------------- + +In some chips, the clock of the codec isn't 48000 but using a PCI clock +(to save a quartz!). In this case, change the field ``bus->clock`` to +the corresponding value. For example, intel8x0 and es1968 drivers have +their own function to read from the clock. + +Proc Files +---------- + +The ALSA AC97 interface will create a proc file such as +``/proc/asound/card0/codec97#0/ac97#0-0`` and ``ac97#0-0+regs``. You +can refer to these files to see the current status and registers of +the codec. + +Multiple Codecs +--------------- + +When there are several codecs on the same card, you need to call +:c:func:`snd_ac97_mixer()` multiple times with ``ac97.num=1`` or +greater. The ``num`` field specifies the codec number. + +If you set up multiple codecs, you either need to write different +callbacks for each codec or check ``ac97->num`` in the callback +routines. + +MIDI (MPU401-UART) Interface +============================ + +General +------- + +Many soundcards have built-in MIDI (MPU401-UART) interfaces. When the +soundcard supports the standard MPU401-UART interface, most likely you +can use the ALSA MPU401-UART API. The MPU401-UART API is defined in +``<sound/mpu401.h>``. + +Some soundchips have a similar but slightly different implementation of +mpu401 stuff. For example, emu10k1 has its own mpu401 routines. + +MIDI Constructor +---------------- + +To create a rawmidi object, call :c:func:`snd_mpu401_uart_new()`. + +:: + + struct snd_rawmidi *rmidi; + snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, port, info_flags, + irq, &rmidi); + + +The first argument is the card pointer, and the second is the index of +this component. You can create up to 8 rawmidi devices. + +The third argument is the type of the hardware, ``MPU401_HW_XXX``. If +it's not a special one, you can use ``MPU401_HW_MPU401``. + +The 4th argument is the I/O port address. Many backward-compatible +MPU401 have an I/O port such as 0x330. Or, it might be a part of its own +PCI I/O region. It depends on the chip design. + +The 5th argument is a bitflag for additional information. When the I/O +port address above is part of the PCI I/O region, the MPU401 I/O port +might have been already allocated (reserved) by the driver itself. In +such a case, pass a bit flag ``MPU401_INFO_INTEGRATED``, and the +mpu401-uart layer will allocate the I/O ports by itself. + +When the controller supports only the input or output MIDI stream, pass +the ``MPU401_INFO_INPUT`` or ``MPU401_INFO_OUTPUT`` bitflag, +respectively. Then the rawmidi instance is created as a single stream. + +``MPU401_INFO_MMIO`` bitflag is used to change the access method to MMIO +(via readb and writeb) instead of iob and outb. In this case, you have +to pass the iomapped address to :c:func:`snd_mpu401_uart_new()`. + +When ``MPU401_INFO_TX_IRQ`` is set, the output stream isn't checked in +the default interrupt handler. The driver needs to call +:c:func:`snd_mpu401_uart_interrupt_tx()` by itself to start +processing the output stream in the irq handler. + +If the MPU-401 interface shares its interrupt with the other logical +devices on the card, set ``MPU401_INFO_IRQ_HOOK`` (see +`below <#MIDI-Interrupt-Handler>`__). + +Usually, the port address corresponds to the command port and port + 1 +corresponds to the data port. If not, you may change the ``cport`` +field of :c:type:`struct snd_mpu401 <snd_mpu401>` manually afterward. +However, :c:type:`struct snd_mpu401 <snd_mpu401>` pointer is +not returned explicitly by :c:func:`snd_mpu401_uart_new()`. You +need to cast ``rmidi->private_data`` to :c:type:`struct snd_mpu401 +<snd_mpu401>` explicitly, + +:: + + struct snd_mpu401 *mpu; + mpu = rmidi->private_data; + +and reset the ``cport`` as you like: + +:: + + mpu->cport = my_own_control_port; + +The 6th argument specifies the ISA irq number that will be allocated. If +no interrupt is to be allocated (because your code is already allocating +a shared interrupt, or because the device does not use interrupts), pass +-1 instead. For a MPU-401 device without an interrupt, a polling timer +will be used instead. + +MIDI Interrupt Handler +---------------------- + +When the interrupt is allocated in +:c:func:`snd_mpu401_uart_new()`, an exclusive ISA interrupt +handler is automatically used, hence you don't have anything else to do +than creating the mpu401 stuff. Otherwise, you have to set +``MPU401_INFO_IRQ_HOOK``, and call +:c:func:`snd_mpu401_uart_interrupt()` explicitly from your own +interrupt handler when it has determined that a UART interrupt has +occurred. + +In this case, you need to pass the private_data of the returned rawmidi +object from :c:func:`snd_mpu401_uart_new()` as the second +argument of :c:func:`snd_mpu401_uart_interrupt()`. + +:: + + snd_mpu401_uart_interrupt(irq, rmidi->private_data, regs); + + +RawMIDI Interface +================= + +Overview +-------- + +The raw MIDI interface is used for hardware MIDI ports that can be +accessed as a byte stream. It is not used for synthesizer chips that do +not directly understand MIDI. + +ALSA handles file and buffer management. All you have to do is to write +some code to move data between the buffer and the hardware. + +The rawmidi API is defined in ``<sound/rawmidi.h>``. + +RawMIDI Constructor +------------------- + +To create a rawmidi device, call the :c:func:`snd_rawmidi_new()` +function: + +:: + + struct snd_rawmidi *rmidi; + err = snd_rawmidi_new(chip->card, "MyMIDI", 0, outs, ins, &rmidi); + if (err < 0) + return err; + rmidi->private_data = chip; + strcpy(rmidi->name, "My MIDI"); + rmidi->info_flags = SNDRV_RAWMIDI_INFO_OUTPUT | + SNDRV_RAWMIDI_INFO_INPUT | + SNDRV_RAWMIDI_INFO_DUPLEX; + +The first argument is the card pointer, the second argument is the ID +string. + +The third argument is the index of this component. You can create up to +8 rawmidi devices. + +The fourth and fifth arguments are the number of output and input +substreams, respectively, of this device (a substream is the equivalent +of a MIDI port). + +Set the ``info_flags`` field to specify the capabilities of the +device. Set ``SNDRV_RAWMIDI_INFO_OUTPUT`` if there is at least one +output port, ``SNDRV_RAWMIDI_INFO_INPUT`` if there is at least one +input port, and ``SNDRV_RAWMIDI_INFO_DUPLEX`` if the device can handle +output and input at the same time. + +After the rawmidi device is created, you need to set the operators +(callbacks) for each substream. There are helper functions to set the +operators for all the substreams of a device: + +:: + + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_mymidi_output_ops); + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_mymidi_input_ops); + +The operators are usually defined like this: + +:: + + static struct snd_rawmidi_ops snd_mymidi_output_ops = { + .open = snd_mymidi_output_open, + .close = snd_mymidi_output_close, + .trigger = snd_mymidi_output_trigger, + }; + +These callbacks are explained in the `RawMIDI Callbacks`_ section. + +If there are more than one substream, you should give a unique name to +each of them: + +:: + + struct snd_rawmidi_substream *substream; + list_for_each_entry(substream, + &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams, + list { + sprintf(substream->name, "My MIDI Port %d", substream->number + 1); + } + /* same for SNDRV_RAWMIDI_STREAM_INPUT */ + +RawMIDI Callbacks +----------------- + +In all the callbacks, the private data that you've set for the rawmidi +device can be accessed as ``substream->rmidi->private_data``. + +If there is more than one port, your callbacks can determine the port +index from the struct snd_rawmidi_substream data passed to each +callback: + +:: + + struct snd_rawmidi_substream *substream; + int index = substream->number; + +RawMIDI open callback +~~~~~~~~~~~~~~~~~~~~~ + +:: + + static int snd_xxx_open(struct snd_rawmidi_substream *substream); + + +This is called when a substream is opened. You can initialize the +hardware here, but you shouldn't start transmitting/receiving data yet. + +RawMIDI close callback +~~~~~~~~~~~~~~~~~~~~~~ + +:: + + static int snd_xxx_close(struct snd_rawmidi_substream *substream); + +Guess what. + +The ``open`` and ``close`` callbacks of a rawmidi device are +serialized with a mutex, and can sleep. + +Rawmidi trigger callback for output substreams +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +:: + + static void snd_xxx_output_trigger(struct snd_rawmidi_substream *substream, int up); + + +This is called with a nonzero ``up`` parameter when there is some data +in the substream buffer that must be transmitted. + +To read data from the buffer, call +:c:func:`snd_rawmidi_transmit_peek()`. It will return the number +of bytes that have been read; this will be less than the number of bytes +requested when there are no more data in the buffer. After the data have +been transmitted successfully, call +:c:func:`snd_rawmidi_transmit_ack()` to remove the data from the +substream buffer: + +:: + + unsigned char data; + while (snd_rawmidi_transmit_peek(substream, &data, 1) == 1) { + if (snd_mychip_try_to_transmit(data)) + snd_rawmidi_transmit_ack(substream, 1); + else + break; /* hardware FIFO full */ + } + +If you know beforehand that the hardware will accept data, you can use +the :c:func:`snd_rawmidi_transmit()` function which reads some +data and removes them from the buffer at once: + +:: + + while (snd_mychip_transmit_possible()) { + unsigned char data; + if (snd_rawmidi_transmit(substream, &data, 1) != 1) + break; /* no more data */ + snd_mychip_transmit(data); + } + +If you know beforehand how many bytes you can accept, you can use a +buffer size greater than one with the +:c:func:`snd_rawmidi_transmit*()` functions. + +The ``trigger`` callback must not sleep. If the hardware FIFO is full +before the substream buffer has been emptied, you have to continue +transmitting data later, either in an interrupt handler, or with a +timer if the hardware doesn't have a MIDI transmit interrupt. + +The ``trigger`` callback is called with a zero ``up`` parameter when +the transmission of data should be aborted. + +RawMIDI trigger callback for input substreams +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +:: + + static void snd_xxx_input_trigger(struct snd_rawmidi_substream *substream, int up); + + +This is called with a nonzero ``up`` parameter to enable receiving data, +or with a zero ``up`` parameter do disable receiving data. + +The ``trigger`` callback must not sleep; the actual reading of data +from the device is usually done in an interrupt handler. + +When data reception is enabled, your interrupt handler should call +:c:func:`snd_rawmidi_receive()` for all received data: + +:: + + void snd_mychip_midi_interrupt(...) + { + while (mychip_midi_available()) { + unsigned char data; + data = mychip_midi_read(); + snd_rawmidi_receive(substream, &data, 1); + } + } + + +drain callback +~~~~~~~~~~~~~~ + +:: + + static void snd_xxx_drain(struct snd_rawmidi_substream *substream); + + +This is only used with output substreams. This function should wait +until all data read from the substream buffer have been transmitted. +This ensures that the device can be closed and the driver unloaded +without losing data. + +This callback is optional. If you do not set ``drain`` in the struct +snd_rawmidi_ops structure, ALSA will simply wait for 50 milliseconds +instead. + +Miscellaneous Devices +===================== + +FM OPL3 +------- + +The FM OPL3 is still used in many chips (mainly for backward +compatibility). ALSA has a nice OPL3 FM control layer, too. The OPL3 API +is defined in ``<sound/opl3.h>``. + +FM registers can be directly accessed through the direct-FM API, defined +in ``<sound/asound_fm.h>``. In ALSA native mode, FM registers are +accessed through the Hardware-Dependent Device direct-FM extension API, +whereas in OSS compatible mode, FM registers can be accessed with the +OSS direct-FM compatible API in ``/dev/dmfmX`` device. + +To create the OPL3 component, you have two functions to call. The first +one is a constructor for the ``opl3_t`` instance. + +:: + + struct snd_opl3 *opl3; + snd_opl3_create(card, lport, rport, OPL3_HW_OPL3_XXX, + integrated, &opl3); + +The first argument is the card pointer, the second one is the left port +address, and the third is the right port address. In most cases, the +right port is placed at the left port + 2. + +The fourth argument is the hardware type. + +When the left and right ports have been already allocated by the card +driver, pass non-zero to the fifth argument (``integrated``). Otherwise, +the opl3 module will allocate the specified ports by itself. + +When the accessing the hardware requires special method instead of the +standard I/O access, you can create opl3 instance separately with +:c:func:`snd_opl3_new()`. + +:: + + struct snd_opl3 *opl3; + snd_opl3_new(card, OPL3_HW_OPL3_XXX, &opl3); + +Then set ``command``, ``private_data`` and ``private_free`` for the +private access function, the private data and the destructor. The +``l_port`` and ``r_port`` are not necessarily set. Only the command +must be set properly. You can retrieve the data from the +``opl3->private_data`` field. + +After creating the opl3 instance via :c:func:`snd_opl3_new()`, +call :c:func:`snd_opl3_init()` to initialize the chip to the +proper state. Note that :c:func:`snd_opl3_create()` always calls +it internally. + +If the opl3 instance is created successfully, then create a hwdep device +for this opl3. + +:: + + struct snd_hwdep *opl3hwdep; + snd_opl3_hwdep_new(opl3, 0, 1, &opl3hwdep); + +The first argument is the ``opl3_t`` instance you created, and the +second is the index number, usually 0. + +The third argument is the index-offset for the sequencer client assigned +to the OPL3 port. When there is an MPU401-UART, give 1 for here (UART +always takes 0). + +Hardware-Dependent Devices +-------------------------- + +Some chips need user-space access for special controls or for loading +the micro code. In such a case, you can create a hwdep +(hardware-dependent) device. The hwdep API is defined in +``<sound/hwdep.h>``. You can find examples in opl3 driver or +``isa/sb/sb16_csp.c``. + +The creation of the ``hwdep`` instance is done via +:c:func:`snd_hwdep_new()`. + +:: + + struct snd_hwdep *hw; + snd_hwdep_new(card, "My HWDEP", 0, &hw); + +where the third argument is the index number. + +You can then pass any pointer value to the ``private_data``. If you +assign a private data, you should define the destructor, too. The +destructor function is set in the ``private_free`` field. + +:: + + struct mydata *p = kmalloc(sizeof(*p), GFP_KERNEL); + hw->private_data = p; + hw->private_free = mydata_free; + +and the implementation of the destructor would be: + +:: + + static void mydata_free(struct snd_hwdep *hw) + { + struct mydata *p = hw->private_data; + kfree(p); + } + +The arbitrary file operations can be defined for this instance. The file +operators are defined in the ``ops`` table. For example, assume that +this chip needs an ioctl. + +:: + + hw->ops.open = mydata_open; + hw->ops.ioctl = mydata_ioctl; + hw->ops.release = mydata_release; + +And implement the callback functions as you like. + +IEC958 (S/PDIF) +--------------- + +Usually the controls for IEC958 devices are implemented via the control +interface. There is a macro to compose a name string for IEC958 +controls, :c:func:`SNDRV_CTL_NAME_IEC958()` defined in +``<include/asound.h>``. + +There are some standard controls for IEC958 status bits. These controls +use the type ``SNDRV_CTL_ELEM_TYPE_IEC958``, and the size of element is +fixed as 4 bytes array (value.iec958.status[x]). For the ``info`` +callback, you don't specify the value field for this type (the count +field must be set, though). + +“IEC958 Playback Con Mask” is used to return the bit-mask for the IEC958 +status bits of consumer mode. Similarly, “IEC958 Playback Pro Mask” +returns the bitmask for professional mode. They are read-only controls, +and are defined as MIXER controls (iface = +``SNDRV_CTL_ELEM_IFACE_MIXER``). + +Meanwhile, “IEC958 Playback Default” control is defined for getting and +setting the current default IEC958 bits. Note that this one is usually +defined as a PCM control (iface = ``SNDRV_CTL_ELEM_IFACE_PCM``), +although in some places it's defined as a MIXER control. + +In addition, you can define the control switches to enable/disable or to +set the raw bit mode. The implementation will depend on the chip, but +the control should be named as “IEC958 xxx”, preferably using the +:c:func:`SNDRV_CTL_NAME_IEC958()` macro. + +You can find several cases, for example, ``pci/emu10k1``, +``pci/ice1712``, or ``pci/cmipci.c``. + +Buffer and Memory Management +============================ + +Buffer Types +------------ + +ALSA provides several different buffer allocation functions depending on +the bus and the architecture. All these have a consistent API. The +allocation of physically-contiguous pages is done via +:c:func:`snd_malloc_xxx_pages()` function, where xxx is the bus +type. + +The allocation of pages with fallback is +:c:func:`snd_malloc_xxx_pages_fallback()`. This function tries +to allocate the specified pages but if the pages are not available, it +tries to reduce the page sizes until enough space is found. + +The release the pages, call :c:func:`snd_free_xxx_pages()` +function. + +Usually, ALSA drivers try to allocate and reserve a large contiguous +physical space at the time the module is loaded for the later use. This +is called “pre-allocation”. As already written, you can call the +following function at pcm instance construction time (in the case of PCI +bus). + +:: + + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(pci), size, max); + +where ``size`` is the byte size to be pre-allocated and the ``max`` is +the maximum size to be changed via the ``prealloc`` proc file. The +allocator will try to get an area as large as possible within the +given size. + +The second argument (type) and the third argument (device pointer) are +dependent on the bus. In the case of the ISA bus, pass +:c:func:`snd_dma_isa_data()` as the third argument with +``SNDRV_DMA_TYPE_DEV`` type. For the continuous buffer unrelated to the +bus can be pre-allocated with ``SNDRV_DMA_TYPE_CONTINUOUS`` type and the +``snd_dma_continuous_data(GFP_KERNEL)`` device pointer, where +``GFP_KERNEL`` is the kernel allocation flag to use. For the PCI +scatter-gather buffers, use ``SNDRV_DMA_TYPE_DEV_SG`` with +``snd_dma_pci_data(pci)`` (see the `Non-Contiguous Buffers`_ +section). + +Once the buffer is pre-allocated, you can use the allocator in the +``hw_params`` callback: + +:: + + snd_pcm_lib_malloc_pages(substream, size); + +Note that you have to pre-allocate to use this function. + +External Hardware Buffers +------------------------- + +Some chips have their own hardware buffers and the DMA transfer from the +host memory is not available. In such a case, you need to either 1) +copy/set the audio data directly to the external hardware buffer, or 2) +make an intermediate buffer and copy/set the data from it to the +external hardware buffer in interrupts (or in tasklets, preferably). + +The first case works fine if the external hardware buffer is large +enough. This method doesn't need any extra buffers and thus is more +effective. You need to define the ``copy`` and ``silence`` callbacks +for the data transfer. However, there is a drawback: it cannot be +mmapped. The examples are GUS's GF1 PCM or emu8000's wavetable PCM. + +The second case allows for mmap on the buffer, although you have to +handle an interrupt or a tasklet to transfer the data from the +intermediate buffer to the hardware buffer. You can find an example in +the vxpocket driver. + +Another case is when the chip uses a PCI memory-map region for the +buffer instead of the host memory. In this case, mmap is available only +on certain architectures like the Intel one. In non-mmap mode, the data +cannot be transferred as in the normal way. Thus you need to define the +``copy`` and ``silence`` callbacks as well, as in the cases above. The +examples are found in ``rme32.c`` and ``rme96.c``. + +The implementation of the ``copy`` and ``silence`` callbacks depends +upon whether the hardware supports interleaved or non-interleaved +samples. The ``copy`` callback is defined like below, a bit +differently depending whether the direction is playback or capture: + +:: + + static int playback_copy(struct snd_pcm_substream *substream, int channel, + snd_pcm_uframes_t pos, void *src, snd_pcm_uframes_t count); + static int capture_copy(struct snd_pcm_substream *substream, int channel, + snd_pcm_uframes_t pos, void *dst, snd_pcm_uframes_t count); + +In the case of interleaved samples, the second argument (``channel``) is +not used. The third argument (``pos``) points the current position +offset in frames. + +The meaning of the fourth argument is different between playback and +capture. For playback, it holds the source data pointer, and for +capture, it's the destination data pointer. + +The last argument is the number of frames to be copied. + +What you have to do in this callback is again different between playback +and capture directions. In the playback case, you copy the given amount +of data (``count``) at the specified pointer (``src``) to the specified +offset (``pos``) on the hardware buffer. When coded like memcpy-like +way, the copy would be like: + +:: + + my_memcpy(my_buffer + frames_to_bytes(runtime, pos), src, + frames_to_bytes(runtime, count)); + +For the capture direction, you copy the given amount of data (``count``) +at the specified offset (``pos``) on the hardware buffer to the +specified pointer (``dst``). + +:: + + my_memcpy(dst, my_buffer + frames_to_bytes(runtime, pos), + frames_to_bytes(runtime, count)); + +Note that both the position and the amount of data are given in frames. + +In the case of non-interleaved samples, the implementation will be a bit +more complicated. + +You need to check the channel argument, and if it's -1, copy the whole +channels. Otherwise, you have to copy only the specified channel. Please +check ``isa/gus/gus_pcm.c`` as an example. + +The ``silence`` callback is also implemented in a similar way + +:: + + static int silence(struct snd_pcm_substream *substream, int channel, + snd_pcm_uframes_t pos, snd_pcm_uframes_t count); + +The meanings of arguments are the same as in the ``copy`` callback, +although there is no ``src/dst`` argument. In the case of interleaved +samples, the channel argument has no meaning, as well as on ``copy`` +callback. + +The role of ``silence`` callback is to set the given amount +(``count``) of silence data at the specified offset (``pos``) on the +hardware buffer. Suppose that the data format is signed (that is, the +silent-data is 0), and the implementation using a memset-like function +would be like: + +:: + + my_memcpy(my_buffer + frames_to_bytes(runtime, pos), 0, + frames_to_bytes(runtime, count)); + +In the case of non-interleaved samples, again, the implementation +becomes a bit more complicated. See, for example, ``isa/gus/gus_pcm.c``. + +Non-Contiguous Buffers +---------------------- + +If your hardware supports the page table as in emu10k1 or the buffer +descriptors as in via82xx, you can use the scatter-gather (SG) DMA. ALSA +provides an interface for handling SG-buffers. The API is provided in +``<sound/pcm.h>``. + +For creating the SG-buffer handler, call +:c:func:`snd_pcm_lib_preallocate_pages()` or +:c:func:`snd_pcm_lib_preallocate_pages_for_all()` with +``SNDRV_DMA_TYPE_DEV_SG`` in the PCM constructor like other PCI +pre-allocator. You need to pass ``snd_dma_pci_data(pci)``, where pci is +the :c:type:`struct pci_dev <pci_dev>` pointer of the chip as +well. The ``struct snd_sg_buf`` instance is created as +``substream->dma_private``. You can cast the pointer like: + +:: + + struct snd_sg_buf *sgbuf = (struct snd_sg_buf *)substream->dma_private; + +Then call :c:func:`snd_pcm_lib_malloc_pages()` in the ``hw_params`` +callback as well as in the case of normal PCI buffer. The SG-buffer +handler will allocate the non-contiguous kernel pages of the given size +and map them onto the virtually contiguous memory. The virtual pointer +is addressed in runtime->dma_area. The physical address +(``runtime->dma_addr``) is set to zero, because the buffer is +physically non-contiguous. The physical address table is set up in +``sgbuf->table``. You can get the physical address at a certain offset +via :c:func:`snd_pcm_sgbuf_get_addr()`. + +When a SG-handler is used, you need to set +:c:func:`snd_pcm_sgbuf_ops_page()` as the ``page`` callback. (See +`page callback`_ section.) + +To release the data, call :c:func:`snd_pcm_lib_free_pages()` in +the ``hw_free`` callback as usual. + +Vmalloc'ed Buffers +------------------ + +It's possible to use a buffer allocated via :c:func:`vmalloc()`, for +example, for an intermediate buffer. Since the allocated pages are not +contiguous, you need to set the ``page`` callback to obtain the physical +address at every offset. + +The implementation of ``page`` callback would be like this: + +:: + + #include <linux/vmalloc.h> + + /* get the physical page pointer on the given offset */ + static struct page *mychip_page(struct snd_pcm_substream *substream, + unsigned long offset) + { + void *pageptr = substream->runtime->dma_area + offset; + return vmalloc_to_page(pageptr); + } + +Proc Interface +============== + +ALSA provides an easy interface for procfs. The proc files are very +useful for debugging. I recommend you set up proc files if you write a +driver and want to get a running status or register dumps. The API is +found in ``<sound/info.h>``. + +To create a proc file, call :c:func:`snd_card_proc_new()`. + +:: + + struct snd_info_entry *entry; + int err = snd_card_proc_new(card, "my-file", &entry); + +where the second argument specifies the name of the proc file to be +created. The above example will create a file ``my-file`` under the +card directory, e.g. ``/proc/asound/card0/my-file``. + +Like other components, the proc entry created via +:c:func:`snd_card_proc_new()` will be registered and released +automatically in the card registration and release functions. + +When the creation is successful, the function stores a new instance in +the pointer given in the third argument. It is initialized as a text +proc file for read only. To use this proc file as a read-only text file +as it is, set the read callback with a private data via +:c:func:`snd_info_set_text_ops()`. + +:: + + snd_info_set_text_ops(entry, chip, my_proc_read); + +where the second argument (``chip``) is the private data to be used in +the callbacks. The third parameter specifies the read buffer size and +the fourth (``my_proc_read``) is the callback function, which is +defined like + +:: + + static void my_proc_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer); + +In the read callback, use :c:func:`snd_iprintf()` for output +strings, which works just like normal :c:func:`printf()`. For +example, + +:: + + static void my_proc_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) + { + struct my_chip *chip = entry->private_data; + + snd_iprintf(buffer, "This is my chip!\n"); + snd_iprintf(buffer, "Port = %ld\n", chip->port); + } + +The file permissions can be changed afterwards. As default, it's set as +read only for all users. If you want to add write permission for the +user (root as default), do as follows: + +:: + + entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + +and set the write buffer size and the callback + +:: + + entry->c.text.write = my_proc_write; + +For the write callback, you can use :c:func:`snd_info_get_line()` +to get a text line, and :c:func:`snd_info_get_str()` to retrieve +a string from the line. Some examples are found in +``core/oss/mixer_oss.c``, core/oss/and ``pcm_oss.c``. + +For a raw-data proc-file, set the attributes as follows: + +:: + + static struct snd_info_entry_ops my_file_io_ops = { + .read = my_file_io_read, + }; + + entry->content = SNDRV_INFO_CONTENT_DATA; + entry->private_data = chip; + entry->c.ops = &my_file_io_ops; + entry->size = 4096; + entry->mode = S_IFREG | S_IRUGO; + +For the raw data, ``size`` field must be set properly. This specifies +the maximum size of the proc file access. + +The read/write callbacks of raw mode are more direct than the text mode. +You need to use a low-level I/O functions such as +:c:func:`copy_from/to_user()` to transfer the data. + +:: + + static ssize_t my_file_io_read(struct snd_info_entry *entry, + void *file_private_data, + struct file *file, + char *buf, + size_t count, + loff_t pos) + { + if (copy_to_user(buf, local_data + pos, count)) + return -EFAULT; + return count; + } + +If the size of the info entry has been set up properly, ``count`` and +``pos`` are guaranteed to fit within 0 and the given size. You don't +have to check the range in the callbacks unless any other condition is +required. + +Power Management +================ + +If the chip is supposed to work with suspend/resume functions, you need +to add power-management code to the driver. The additional code for +power-management should be ifdef-ed with ``CONFIG_PM``. + +If the driver *fully* supports suspend/resume that is, the device can be +properly resumed to its state when suspend was called, you can set the +``SNDRV_PCM_INFO_RESUME`` flag in the pcm info field. Usually, this is +possible when the registers of the chip can be safely saved and restored +to RAM. If this is set, the trigger callback is called with +``SNDRV_PCM_TRIGGER_RESUME`` after the resume callback completes. + +Even if the driver doesn't support PM fully but partial suspend/resume +is still possible, it's still worthy to implement suspend/resume +callbacks. In such a case, applications would reset the status by +calling :c:func:`snd_pcm_prepare()` and restart the stream +appropriately. Hence, you can define suspend/resume callbacks below but +don't set ``SNDRV_PCM_INFO_RESUME`` info flag to the PCM. + +Note that the trigger with SUSPEND can always be called when +:c:func:`snd_pcm_suspend_all()` is called, regardless of the +``SNDRV_PCM_INFO_RESUME`` flag. The ``RESUME`` flag affects only the +behavior of :c:func:`snd_pcm_resume()`. (Thus, in theory, +``SNDRV_PCM_TRIGGER_RESUME`` isn't needed to be handled in the trigger +callback when no ``SNDRV_PCM_INFO_RESUME`` flag is set. But, it's better +to keep it for compatibility reasons.) + +In the earlier version of ALSA drivers, a common power-management layer +was provided, but it has been removed. The driver needs to define the +suspend/resume hooks according to the bus the device is connected to. In +the case of PCI drivers, the callbacks look like below: + +:: + + #ifdef CONFIG_PM + static int snd_my_suspend(struct pci_dev *pci, pm_message_t state) + { + .... /* do things for suspend */ + return 0; + } + static int snd_my_resume(struct pci_dev *pci) + { + .... /* do things for suspend */ + return 0; + } + #endif + +The scheme of the real suspend job is as follows. + +1. Retrieve the card and the chip data. + +2. Call :c:func:`snd_power_change_state()` with + ``SNDRV_CTL_POWER_D3hot`` to change the power status. + +3. Call :c:func:`snd_pcm_suspend_all()` to suspend the running + PCM streams. + +4. If AC97 codecs are used, call :c:func:`snd_ac97_suspend()` for + each codec. + +5. Save the register values if necessary. + +6. Stop the hardware if necessary. + +7. Disable the PCI device by calling + :c:func:`pci_disable_device()`. Then, call + :c:func:`pci_save_state()` at last. + +A typical code would be like: + +:: + + static int mychip_suspend(struct pci_dev *pci, pm_message_t state) + { + /* (1) */ + struct snd_card *card = pci_get_drvdata(pci); + struct mychip *chip = card->private_data; + /* (2) */ + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + /* (3) */ + snd_pcm_suspend_all(chip->pcm); + /* (4) */ + snd_ac97_suspend(chip->ac97); + /* (5) */ + snd_mychip_save_registers(chip); + /* (6) */ + snd_mychip_stop_hardware(chip); + /* (7) */ + pci_disable_device(pci); + pci_save_state(pci); + return 0; + } + + +The scheme of the real resume job is as follows. + +1. Retrieve the card and the chip data. + +2. Set up PCI. First, call :c:func:`pci_restore_state()`. Then + enable the pci device again by calling + :c:func:`pci_enable_device()`. Call + :c:func:`pci_set_master()` if necessary, too. + +3. Re-initialize the chip. + +4. Restore the saved registers if necessary. + +5. Resume the mixer, e.g. calling :c:func:`snd_ac97_resume()`. + +6. Restart the hardware (if any). + +7. Call :c:func:`snd_power_change_state()` with + ``SNDRV_CTL_POWER_D0`` to notify the processes. + +A typical code would be like: + +:: + + static int mychip_resume(struct pci_dev *pci) + { + /* (1) */ + struct snd_card *card = pci_get_drvdata(pci); + struct mychip *chip = card->private_data; + /* (2) */ + pci_restore_state(pci); + pci_enable_device(pci); + pci_set_master(pci); + /* (3) */ + snd_mychip_reinit_chip(chip); + /* (4) */ + snd_mychip_restore_registers(chip); + /* (5) */ + snd_ac97_resume(chip->ac97); + /* (6) */ + snd_mychip_restart_chip(chip); + /* (7) */ + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; + } + +As shown in the above, it's better to save registers after suspending +the PCM operations via :c:func:`snd_pcm_suspend_all()` or +:c:func:`snd_pcm_suspend()`. It means that the PCM streams are +already stopped when the register snapshot is taken. But, remember that +you don't have to restart the PCM stream in the resume callback. It'll +be restarted via trigger call with ``SNDRV_PCM_TRIGGER_RESUME`` when +necessary. + +OK, we have all callbacks now. Let's set them up. In the initialization +of the card, make sure that you can get the chip data from the card +instance, typically via ``private_data`` field, in case you created the +chip data individually. + +:: + + static int snd_mychip_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) + { + .... + struct snd_card *card; + struct mychip *chip; + int err; + .... + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, + 0, &card); + .... + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + .... + card->private_data = chip; + .... + } + +When you created the chip data with :c:func:`snd_card_new()`, it's +anyway accessible via ``private_data`` field. + +:: + + static int snd_mychip_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) + { + .... + struct snd_card *card; + struct mychip *chip; + int err; + .... + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, + sizeof(struct mychip), &card); + .... + chip = card->private_data; + .... + } + +If you need a space to save the registers, allocate the buffer for it +here, too, since it would be fatal if you cannot allocate a memory in +the suspend phase. The allocated buffer should be released in the +corresponding destructor. + +And next, set suspend/resume callbacks to the pci_driver. + +:: + + static struct pci_driver driver = { + .name = KBUILD_MODNAME, + .id_table = snd_my_ids, + .probe = snd_my_probe, + .remove = snd_my_remove, + #ifdef CONFIG_PM + .suspend = snd_my_suspend, + .resume = snd_my_resume, + #endif + }; + +Module Parameters +================= + +There are standard module options for ALSA. At least, each module should +have the ``index``, ``id`` and ``enable`` options. + +If the module supports multiple cards (usually up to 8 = ``SNDRV_CARDS`` +cards), they should be arrays. The default initial values are defined +already as constants for easier programming: + +:: + + static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; + static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; + static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +If the module supports only a single card, they could be single +variables, instead. ``enable`` option is not always necessary in this +case, but it would be better to have a dummy option for compatibility. + +The module parameters must be declared with the standard +``module_param()()``, ``module_param_array()()`` and +:c:func:`MODULE_PARM_DESC()` macros. + +The typical coding would be like below: + +:: + + #define CARD_NAME "My Chip" + + module_param_array(index, int, NULL, 0444); + MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); + module_param_array(id, charp, NULL, 0444); + MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); + module_param_array(enable, bool, NULL, 0444); + MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); + +Also, don't forget to define the module description, classes, license +and devices. Especially, the recent modprobe requires to define the +module license as GPL, etc., otherwise the system is shown as “tainted”. + +:: + + MODULE_DESCRIPTION("My Chip"); + MODULE_LICENSE("GPL"); + MODULE_SUPPORTED_DEVICE("{{Vendor,My Chip Name}}"); + + +How To Put Your Driver Into ALSA Tree +===================================== + +General +------- + +So far, you've learned how to write the driver codes. And you might have +a question now: how to put my own driver into the ALSA driver tree? Here +(finally :) the standard procedure is described briefly. + +Suppose that you create a new PCI driver for the card “xyz”. The card +module name would be snd-xyz. The new driver is usually put into the +alsa-driver tree, ``alsa-driver/pci`` directory in the case of PCI +cards. Then the driver is evaluated, audited and tested by developers +and users. After a certain time, the driver will go to the alsa-kernel +tree (to the corresponding directory, such as ``alsa-kernel/pci``) and +eventually will be integrated into the Linux 2.6 tree (the directory +would be ``linux/sound/pci``). + +In the following sections, the driver code is supposed to be put into +alsa-driver tree. The two cases are covered: a driver consisting of a +single source file and one consisting of several source files. + +Driver with A Single Source File +-------------------------------- + +1. Modify alsa-driver/pci/Makefile + + Suppose you have a file xyz.c. Add the following two lines + +:: + + snd-xyz-objs := xyz.o + obj-$(CONFIG_SND_XYZ) += snd-xyz.o + +2. Create the Kconfig entry + + Add the new entry of Kconfig for your xyz driver. config SND_XYZ + tristate "Foobar XYZ" depends on SND select SND_PCM help Say Y here + to include support for Foobar XYZ soundcard. To compile this driver + as a module, choose M here: the module will be called snd-xyz. the + line, select SND_PCM, specifies that the driver xyz supports PCM. In + addition to SND_PCM, the following components are supported for + select command: SND_RAWMIDI, SND_TIMER, SND_HWDEP, + SND_MPU401_UART, SND_OPL3_LIB, SND_OPL4_LIB, SND_VX_LIB, + SND_AC97_CODEC. Add the select command for each supported + component. + + Note that some selections imply the lowlevel selections. For example, + PCM includes TIMER, MPU401_UART includes RAWMIDI, AC97_CODEC + includes PCM, and OPL3_LIB includes HWDEP. You don't need to give + the lowlevel selections again. + + For the details of Kconfig script, refer to the kbuild documentation. + +3. Run cvscompile script to re-generate the configure script and build + the whole stuff again. + +Drivers with Several Source Files +--------------------------------- + +Suppose that the driver snd-xyz have several source files. They are +located in the new subdirectory, pci/xyz. + +1. Add a new directory (``xyz``) in ``alsa-driver/pci/Makefile`` as + below + +:: + + obj-$(CONFIG_SND) += xyz/ + + +2. Under the directory ``xyz``, create a Makefile + +:: + + ifndef SND_TOPDIR + SND_TOPDIR=../.. + endif + + include $(SND_TOPDIR)/toplevel.config + include $(SND_TOPDIR)/Makefile.conf + + snd-xyz-objs := xyz.o abc.o def.o + + obj-$(CONFIG_SND_XYZ) += snd-xyz.o + + include $(SND_TOPDIR)/Rules.make + +3. Create the Kconfig entry + + This procedure is as same as in the last section. + +4. Run cvscompile script to re-generate the configure script and build + the whole stuff again. + +Useful Functions +================ + +:c:func:`snd_printk()` and friends +--------------------------------------- + +ALSA provides a verbose version of the :c:func:`printk()` function. +If a kernel config ``CONFIG_SND_VERBOSE_PRINTK`` is set, this function +prints the given message together with the file name and the line of the +caller. The ``KERN_XXX`` prefix is processed as well as the original +:c:func:`printk()` does, so it's recommended to add this prefix, +e.g. snd_printk(KERN_ERR "Oh my, sorry, it's extremely bad!\n"); + +There are also :c:func:`printk()`'s for debugging. +:c:func:`snd_printd()` can be used for general debugging purposes. +If ``CONFIG_SND_DEBUG`` is set, this function is compiled, and works +just like :c:func:`snd_printk()`. If the ALSA is compiled without +the debugging flag, it's ignored. + +:c:func:`snd_printdd()` is compiled in only when +``CONFIG_SND_DEBUG_VERBOSE`` is set. Please note that +``CONFIG_SND_DEBUG_VERBOSE`` is not set as default even if you configure +the alsa-driver with ``--with-debug=full`` option. You need to give +explicitly ``--with-debug=detect`` option instead. + +:c:func:`snd_BUG()` +------------------------ + +It shows the ``BUG?`` message and stack trace as well as +:c:func:`snd_BUG_ON()` at the point. It's useful to show that a +fatal error happens there. + +When no debug flag is set, this macro is ignored. + +:c:func:`snd_BUG_ON()` +---------------------------- + +:c:func:`snd_BUG_ON()` macro is similar with +:c:func:`WARN_ON()` macro. For example, snd_BUG_ON(!pointer); or +it can be used as the condition, if (snd_BUG_ON(non_zero_is_bug)) +return -EINVAL; + +The macro takes an conditional expression to evaluate. When +``CONFIG_SND_DEBUG``, is set, if the expression is non-zero, it shows +the warning message such as ``BUG? (xxx)`` normally followed by stack +trace. In both cases it returns the evaluated value. + +Acknowledgments +=============== + +I would like to thank Phil Kerr for his help for improvement and +corrections of this document. + +Kevin Conder reformatted the original plain-text to the DocBook format. + +Giuliano Pochini corrected typos and contributed the example codes in +the hardware constraints section.

A simple reformat with the description list of ReST, and the content was kept as is, but renamed as Documentation/sound/hd-audio/models.rst.

Signed-off-by: Takashi Iwai tiwai@suse.de --- Documentation/sound/alsa/HD-Audio-Models.txt | 324 ----------------- Documentation/sound/hd-audio/index.rst | 1 + Documentation/sound/hd-audio/models.rst | 518 +++++++++++++++++++++++++++ 3 files changed, 519 insertions(+), 324 deletions(-) delete mode 100644 Documentation/sound/alsa/HD-Audio-Models.txt create mode 100644 Documentation/sound/hd-audio/models.rst

diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt deleted file mode 100644 index ec099d4343f2..000000000000 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ /dev/null @@ -1,324 +0,0 @@ - Model name Description - ---------- ----------- -ALC880 -====== - 3stack 3-jack in back and a headphone out - 3stack-digout 3-jack in back, a HP out and a SPDIF out - 5stack 5-jack in back, 2-jack in front - 5stack-digout 5-jack in back, 2-jack in front, a SPDIF out - 6stack 6-jack in back, 2-jack in front - 6stack-digout 6-jack with a SPDIF out - -ALC260 -====== - gpio1 Enable GPIO1 - coef Enable EAPD via COEF table - fujitsu Quirk for FSC S7020 - fujitsu-jwse Quirk for FSC S7020 with jack modes and HP mic support - -ALC262 -====== - inv-dmic Inverted internal mic workaround - -ALC267/268 -========== - inv-dmic Inverted internal mic workaround - hp-eapd Disable HP EAPD on NID 0x15 - -ALC22x/23x/25x/269/27x/28x/29x (and vendor-specific ALC3xxx models) -====== - laptop-amic Laptops with analog-mic input - laptop-dmic Laptops with digital-mic input - alc269-dmic Enable ALC269(VA) digital mic workaround - alc271-dmic Enable ALC271X digital mic workaround - inv-dmic Inverted internal mic workaround - headset-mic Indicates a combined headset (headphone+mic) jack - headset-mode More comprehensive headset support for ALC269 & co - headset-mode-no-hp-mic Headset mode support without headphone mic - lenovo-dock Enables docking station I/O for some Lenovos - hp-gpio-led GPIO LED support on HP laptops - dell-headset-multi Headset jack, which can also be used as mic-in - dell-headset-dock Headset jack (without mic-in), and also dock I/O - alc283-dac-wcaps Fixups for Chromebook with ALC283 - alc283-sense-combo Combo jack sensing on ALC283 - tpt440-dock Pin configs for Lenovo Thinkpad Dock support - -ALC66x/67x/892 -============== - mario Chromebook mario model fixup - asus-mode1 ASUS - asus-mode2 ASUS - asus-mode3 ASUS - asus-mode4 ASUS - asus-mode5 ASUS - asus-mode6 ASUS - asus-mode7 ASUS - asus-mode8 ASUS - inv-dmic Inverted internal mic workaround - dell-headset-multi Headset jack, which can also be used as mic-in - -ALC680 -====== - N/A - -ALC88x/898/1150 -====================== - acer-aspire-4930g Acer Aspire 4930G/5930G/6530G/6930G/7730G - acer-aspire-8930g Acer Aspire 8330G/6935G - acer-aspire Acer Aspire others - inv-dmic Inverted internal mic workaround - no-primary-hp VAIO Z/VGC-LN51JGB workaround (for fixed speaker DAC) - -ALC861/660 -========== - N/A - -ALC861VD/660VD -============== - N/A - -CMI9880 -======= - minimal 3-jack in back - min_fp 3-jack in back, 2-jack in front - full 6-jack in back, 2-jack in front - full_dig 6-jack in back, 2-jack in front, SPDIF I/O - allout 5-jack in back, 2-jack in front, SPDIF out - auto auto-config reading BIOS (default) - -AD1882 / AD1882A -================ - 3stack 3-stack mode - 3stack-automute 3-stack with automute front HP (default) - 6stack 6-stack mode - -AD1884A / AD1883 / AD1984A / AD1984B -==================================== - desktop 3-stack desktop (default) - laptop laptop with HP jack sensing - mobile mobile devices with HP jack sensing - thinkpad Lenovo Thinkpad X300 - touchsmart HP Touchsmart - -AD1884 -====== - N/A - -AD1981 -====== - basic 3-jack (default) - hp HP nx6320 - thinkpad Lenovo Thinkpad T60/X60/Z60 - toshiba Toshiba U205 - -AD1983 -====== - N/A - -AD1984 -====== - basic default configuration - thinkpad Lenovo Thinkpad T61/X61 - dell_desktop Dell T3400 - -AD1986A -======= - 3stack 3-stack, shared surrounds - laptop 2-channel only (FSC V2060, Samsung M50) - laptop-imic 2-channel with built-in mic - eapd Turn on EAPD constantly - -AD1988/AD1988B/AD1989A/AD1989B -============================== - 6stack 6-jack - 6stack-dig ditto with SPDIF - 3stack 3-jack - 3stack-dig ditto with SPDIF - laptop 3-jack with hp-jack automute - laptop-dig ditto with SPDIF - auto auto-config reading BIOS (default) - -Conexant 5045 -============= - laptop-hpsense Laptop with HP sense (old model laptop) - laptop-micsense Laptop with Mic sense (old model fujitsu) - laptop-hpmicsense Laptop with HP and Mic senses - benq Benq R55E - laptop-hp530 HP 530 laptop - test for testing/debugging purpose, almost all controls - can be adjusted. Appearing only when compiled with - $CONFIG_SND_DEBUG=y - -Conexant 5047 -============= - laptop Basic Laptop config - laptop-hp Laptop config for some HP models (subdevice 30A5) - laptop-eapd Laptop config with EAPD support - test for testing/debugging purpose, almost all controls - can be adjusted. Appearing only when compiled with - $CONFIG_SND_DEBUG=y - -Conexant 5051 -============= - laptop Basic Laptop config (default) - hp HP Spartan laptop - hp-dv6736 HP dv6736 - hp-f700 HP Compaq Presario F700 - ideapad Lenovo IdeaPad laptop - toshiba Toshiba Satellite M300 - -Conexant 5066 -============= - laptop Basic Laptop config (default) - hp-laptop HP laptops, e g G60 - asus Asus K52JU, Lenovo G560 - dell-laptop Dell laptops - dell-vostro Dell Vostro - olpc-xo-1_5 OLPC XO 1.5 - ideapad Lenovo IdeaPad U150 - thinkpad Lenovo Thinkpad - -STAC9200 -======== - ref Reference board - oqo OQO Model 2 - dell-d21 Dell (unknown) - dell-d22 Dell (unknown) - dell-d23 Dell (unknown) - dell-m21 Dell Inspiron 630m, Dell Inspiron 640m - dell-m22 Dell Latitude D620, Dell Latitude D820 - dell-m23 Dell XPS M1710, Dell Precision M90 - dell-m24 Dell Latitude 120L - dell-m25 Dell Inspiron E1505n - dell-m26 Dell Inspiron 1501 - dell-m27 Dell Inspiron E1705/9400 - gateway-m4 Gateway laptops with EAPD control - gateway-m4-2 Gateway laptops with EAPD control - panasonic Panasonic CF-74 - auto BIOS setup (default) - -STAC9205/9254 -============= - ref Reference board - dell-m42 Dell (unknown) - dell-m43 Dell Precision - dell-m44 Dell Inspiron - eapd Keep EAPD on (e.g. Gateway T1616) - auto BIOS setup (default) - -STAC9220/9221 -============= - ref Reference board - 3stack D945 3stack - 5stack D945 5stack + SPDIF - intel-mac-v1 Intel Mac Type 1 - intel-mac-v2 Intel Mac Type 2 - intel-mac-v3 Intel Mac Type 3 - intel-mac-v4 Intel Mac Type 4 - intel-mac-v5 Intel Mac Type 5 - intel-mac-auto Intel Mac (detect type according to subsystem id) - macmini Intel Mac Mini (equivalent with type 3) - macbook Intel Mac Book (eq. type 5) - macbook-pro-v1 Intel Mac Book Pro 1st generation (eq. type 3) - macbook-pro Intel Mac Book Pro 2nd generation (eq. type 3) - imac-intel Intel iMac (eq. type 2) - imac-intel-20 Intel iMac (newer version) (eq. type 3) - ecs202 ECS/PC chips - dell-d81 Dell (unknown) - dell-d82 Dell (unknown) - dell-m81 Dell (unknown) - dell-m82 Dell XPS M1210 - auto BIOS setup (default) - -STAC9202/9250/9251 -================== - ref Reference board, base config - m1 Some Gateway MX series laptops (NX560XL) - m1-2 Some Gateway MX series laptops (MX6453) - m2 Some Gateway MX series laptops (M255) - m2-2 Some Gateway MX series laptops - m3 Some Gateway MX series laptops - m5 Some Gateway MX series laptops (MP6954) - m6 Some Gateway NX series laptops - auto BIOS setup (default) - -STAC9227/9228/9229/927x -======================= - ref Reference board - ref-no-jd Reference board without HP/Mic jack detection - 3stack D965 3stack - 5stack D965 5stack + SPDIF - 5stack-no-fp D965 5stack without front panel - dell-3stack Dell Dimension E520 - dell-bios Fixes with Dell BIOS setup - dell-bios-amic Fixes with Dell BIOS setup including analog mic - volknob Fixes with volume-knob widget 0x24 - auto BIOS setup (default) - -STAC92HD71B* -============ - ref Reference board - dell-m4-1 Dell desktops - dell-m4-2 Dell desktops - dell-m4-3 Dell desktops - hp-m4 HP mini 1000 - hp-dv5 HP dv series - hp-hdx HP HDX series - hp-dv4-1222nr HP dv4-1222nr (with LED support) - auto BIOS setup (default) - -STAC92HD73* -=========== - ref Reference board - no-jd BIOS setup but without jack-detection - intel Intel DG45* mobos - dell-m6-amic Dell desktops/laptops with analog mics - dell-m6-dmic Dell desktops/laptops with digital mics - dell-m6 Dell desktops/laptops with both type of mics - dell-eq Dell desktops/laptops - alienware Alienware M17x - auto BIOS setup (default) - -STAC92HD83* -=========== - ref Reference board - mic-ref Reference board with power management for ports - dell-s14 Dell laptop - dell-vostro-3500 Dell Vostro 3500 laptop - hp-dv7-4000 HP dv-7 4000 - hp_cNB11_intquad HP CNB models with 4 speakers - hp-zephyr HP Zephyr - hp-led HP with broken BIOS for mute LED - hp-inv-led HP with broken BIOS for inverted mute LED - hp-mic-led HP with mic-mute LED - headset-jack Dell Latitude with a 4-pin headset jack - hp-envy-bass Pin fixup for HP Envy bass speaker (NID 0x0f) - hp-envy-ts-bass Pin fixup for HP Envy TS bass speaker (NID 0x10) - hp-bnb13-eq Hardware equalizer setup for HP laptops - auto BIOS setup (default) - -STAC92HD95 -========== - hp-led LED support for HP laptops - hp-bass Bass HPF setup for HP Spectre 13 - -STAC9872 -======== - vaio VAIO laptop without SPDIF - auto BIOS setup (default) - -Cirrus Logic CS4206/4207 -======================== - mbp55 MacBook Pro 5,5 - imac27 IMac 27 Inch - auto BIOS setup (default) - -Cirrus Logic CS4208 -=================== - mba6 MacBook Air 6,1 and 6,2 - gpio0 Enable GPIO 0 amp - auto BIOS setup (default) - -VIA VT17xx/VT18xx/VT20xx -======================== - auto BIOS setup (default) diff --git a/Documentation/sound/hd-audio/index.rst b/Documentation/sound/hd-audio/index.rst index f2dc29019f12..1e8376b0066c 100644 --- a/Documentation/sound/hd-audio/index.rst +++ b/Documentation/sound/hd-audio/index.rst @@ -5,3 +5,4 @@ HD-Audio :maxdepth: 2

notes + models diff --git a/Documentation/sound/hd-audio/models.rst b/Documentation/sound/hd-audio/models.rst new file mode 100644 index 000000000000..5338673c88d9 --- /dev/null +++ b/Documentation/sound/hd-audio/models.rst @@ -0,0 +1,518 @@ +============================== +HD-Audio Codec-Specific Models +============================== + +ALC880 +====== +3stack + 3-jack in back and a headphone out +3stack-digout + 3-jack in back, a HP out and a SPDIF out +5stack + 5-jack in back, 2-jack in front +5stack-digout + 5-jack in back, 2-jack in front, a SPDIF out +6stack + 6-jack in back, 2-jack in front +6stack-digout + 6-jack with a SPDIF out + +ALC260 +====== +gpio1 + Enable GPIO1 +coef + Enable EAPD via COEF table +fujitsu + Quirk for FSC S7020 +fujitsu-jwse + Quirk for FSC S7020 with jack modes and HP mic support + +ALC262 +====== +inv-dmic + Inverted internal mic workaround + +ALC267/268 +========== +inv-dmic + Inverted internal mic workaround +hp-eapd + Disable HP EAPD on NID 0x15 + +ALC22x/23x/25x/269/27x/28x/29x (and vendor-specific ALC3xxx models) +=================================================================== +laptop-amic + Laptops with analog-mic input +laptop-dmic + Laptops with digital-mic input +alc269-dmic + Enable ALC269(VA) digital mic workaround +alc271-dmic + Enable ALC271X digital mic workaround +inv-dmic + Inverted internal mic workaround +headset-mic + Indicates a combined headset (headphone+mic) jack +headset-mode + More comprehensive headset support for ALC269 & co +headset-mode-no-hp-mic + Headset mode support without headphone mic +lenovo-dock + Enables docking station I/O for some Lenovos +hp-gpio-led + GPIO LED support on HP laptops +dell-headset-multi + Headset jack, which can also be used as mic-in +dell-headset-dock + Headset jack (without mic-in), and also dock I/O +alc283-dac-wcaps + Fixups for Chromebook with ALC283 +alc283-sense-combo + Combo jack sensing on ALC283 +tpt440-dock + Pin configs for Lenovo Thinkpad Dock support + +ALC66x/67x/892 +============== +mario + Chromebook mario model fixup +asus-mode1 + ASUS +asus-mode2 + ASUS +asus-mode3 + ASUS +asus-mode4 + ASUS +asus-mode5 + ASUS +asus-mode6 + ASUS +asus-mode7 + ASUS +asus-mode8 + ASUS +inv-dmic + Inverted internal mic workaround +dell-headset-multi + Headset jack, which can also be used as mic-in + +ALC680 +====== +N/A + +ALC88x/898/1150 +====================== +acer-aspire-4930g + Acer Aspire 4930G/5930G/6530G/6930G/7730G +acer-aspire-8930g + Acer Aspire 8330G/6935G +acer-aspire + Acer Aspire others +inv-dmic + Inverted internal mic workaround +no-primary-hp + VAIO Z/VGC-LN51JGB workaround (for fixed speaker DAC) + +ALC861/660 +========== +N/A + +ALC861VD/660VD +============== +N/A + +CMI9880 +======= +minimal + 3-jack in back +min_fp + 3-jack in back, 2-jack in front +full + 6-jack in back, 2-jack in front +full_dig + 6-jack in back, 2-jack in front, SPDIF I/O +allout + 5-jack in back, 2-jack in front, SPDIF out +auto + auto-config reading BIOS (default) + +AD1882 / AD1882A +================ +3stack + 3-stack mode +3stack-automute + 3-stack with automute front HP (default) +6stack + 6-stack mode + +AD1884A / AD1883 / AD1984A / AD1984B +==================================== +desktop 3-stack desktop (default) +laptop laptop with HP jack sensing +mobile mobile devices with HP jack sensing +thinkpad Lenovo Thinkpad X300 +touchsmart HP Touchsmart + +AD1884 +====== +N/A + +AD1981 +====== +basic 3-jack (default) +hp HP nx6320 +thinkpad Lenovo Thinkpad T60/X60/Z60 +toshiba Toshiba U205 + +AD1983 +====== +N/A + +AD1984 +====== +basic default configuration +thinkpad Lenovo Thinkpad T61/X61 +dell_desktop Dell T3400 + +AD1986A +======= +3stack + 3-stack, shared surrounds +laptop + 2-channel only (FSC V2060, Samsung M50) +laptop-imic + 2-channel with built-in mic +eapd + Turn on EAPD constantly + +AD1988/AD1988B/AD1989A/AD1989B +============================== +6stack + 6-jack +6stack-dig + ditto with SPDIF +3stack + 3-jack +3stack-dig + ditto with SPDIF +laptop + 3-jack with hp-jack automute +laptop-dig + ditto with SPDIF +auto + auto-config reading BIOS (default) + +Conexant 5045 +============= +laptop-hpsense + Laptop with HP sense (old model laptop) +laptop-micsense + Laptop with Mic sense (old model fujitsu) +laptop-hpmicsense + Laptop with HP and Mic senses +benq + Benq R55E +laptop-hp530 + HP 530 laptop +test + for testing/debugging purpose, almost all controls can be + adjusted. Appearing only when compiled with $CONFIG_SND_DEBUG=y + +Conexant 5047 +============= +laptop + Basic Laptop config +laptop-hp + Laptop config for some HP models (subdevice 30A5) +laptop-eapd + Laptop config with EAPD support +test + for testing/debugging purpose, almost all controls can be + adjusted. Appearing only when compiled with $CONFIG_SND_DEBUG=y + +Conexant 5051 +============= +laptop + Basic Laptop config (default) +hp + HP Spartan laptop +hp-dv6736 + HP dv6736 +hp-f700 + HP Compaq Presario F700 +ideapad + Lenovo IdeaPad laptop +toshiba + Toshiba Satellite M300 + +Conexant 5066 +============= +laptop + Basic Laptop config (default) +hp-laptop + HP laptops, e g G60 +asus + Asus K52JU, Lenovo G560 +dell-laptop + Dell laptops +dell-vostro + Dell Vostro +olpc-xo-1_5 + OLPC XO 1.5 +ideapad + Lenovo IdeaPad U150 +thinkpad + Lenovo Thinkpad + +STAC9200 +======== +ref + Reference board +oqo + OQO Model 2 +dell-d21 + Dell (unknown) +dell-d22 + Dell (unknown) +dell-d23 + Dell (unknown) +dell-m21 + Dell Inspiron 630m, Dell Inspiron 640m +dell-m22 + Dell Latitude D620, Dell Latitude D820 +dell-m23 + Dell XPS M1710, Dell Precision M90 +dell-m24 + Dell Latitude 120L +dell-m25 + Dell Inspiron E1505n +dell-m26 + Dell Inspiron 1501 +dell-m27 + Dell Inspiron E1705/9400 +gateway-m4 + Gateway laptops with EAPD control +gateway-m4-2 + Gateway laptops with EAPD control +panasonic + Panasonic CF-74 +auto + BIOS setup (default) + +STAC9205/9254 +============= +ref + Reference board +dell-m42 + Dell (unknown) +dell-m43 + Dell Precision +dell-m44 + Dell Inspiron +eapd + Keep EAPD on (e.g. Gateway T1616) +auto + BIOS setup (default) + +STAC9220/9221 +============= +ref + Reference board +3stack + D945 3stack +5stack + D945 5stack + SPDIF +intel-mac-v1 + Intel Mac Type 1 +intel-mac-v2 + Intel Mac Type 2 +intel-mac-v3 + Intel Mac Type 3 +intel-mac-v4 + Intel Mac Type 4 +intel-mac-v5 + Intel Mac Type 5 +intel-mac-auto + Intel Mac (detect type according to subsystem id) +macmini + Intel Mac Mini (equivalent with type 3) +macbook + Intel Mac Book (eq. type 5) +macbook-pro-v1 + Intel Mac Book Pro 1st generation (eq. type 3) +macbook-pro + Intel Mac Book Pro 2nd generation (eq. type 3) +imac-intel + Intel iMac (eq. type 2) +imac-intel-20 + Intel iMac (newer version) (eq. type 3) +ecs202 + ECS/PC chips +dell-d81 + Dell (unknown) +dell-d82 + Dell (unknown) +dell-m81 + Dell (unknown) +dell-m82 + Dell XPS M1210 +auto + BIOS setup (default) + +STAC9202/9250/9251 +================== +ref + Reference board, base config +m1 + Some Gateway MX series laptops (NX560XL) +m1-2 + Some Gateway MX series laptops (MX6453) +m2 + Some Gateway MX series laptops (M255) +m2-2 + Some Gateway MX series laptops +m3 + Some Gateway MX series laptops +m5 + Some Gateway MX series laptops (MP6954) +m6 + Some Gateway NX series laptops +auto + BIOS setup (default) + +STAC9227/9228/9229/927x +======================= +ref + Reference board +ref-no-jd + Reference board without HP/Mic jack detection +3stack + D965 3stack +5stack + D965 5stack + SPDIF +5stack-no-fp + D965 5stack without front panel +dell-3stack + Dell Dimension E520 +dell-bios + Fixes with Dell BIOS setup +dell-bios-amic + Fixes with Dell BIOS setup including analog mic +volknob + Fixes with volume-knob widget 0x24 +auto + BIOS setup (default) + +STAC92HD71B* +============ +ref + Reference board +dell-m4-1 + Dell desktops +dell-m4-2 + Dell desktops +dell-m4-3 + Dell desktops +hp-m4 + HP mini 1000 +hp-dv5 + HP dv series +hp-hdx + HP HDX series +hp-dv4-1222nr + HP dv4-1222nr (with LED support) +auto + BIOS setup (default) + +STAC92HD73* +=========== +ref + Reference board +no-jd + BIOS setup but without jack-detection +intel + Intel DG45* mobos +dell-m6-amic + Dell desktops/laptops with analog mics +dell-m6-dmic + Dell desktops/laptops with digital mics +dell-m6 + Dell desktops/laptops with both type of mics +dell-eq + Dell desktops/laptops +alienware + Alienware M17x +auto + BIOS setup (default) + +STAC92HD83* +=========== +ref + Reference board +mic-ref + Reference board with power management for ports +dell-s14 + Dell laptop +dell-vostro-3500 + Dell Vostro 3500 laptop +hp-dv7-4000 + HP dv-7 4000 +hp_cNB11_intquad + HP CNB models with 4 speakers +hp-zephyr + HP Zephyr +hp-led + HP with broken BIOS for mute LED +hp-inv-led + HP with broken BIOS for inverted mute LED +hp-mic-led + HP with mic-mute LED +headset-jack + Dell Latitude with a 4-pin headset jack +hp-envy-bass + Pin fixup for HP Envy bass speaker (NID 0x0f) +hp-envy-ts-bass + Pin fixup for HP Envy TS bass speaker (NID 0x10) +hp-bnb13-eq + Hardware equalizer setup for HP laptops +auto + BIOS setup (default) + +STAC92HD95 +========== +hp-led + LED support for HP laptops +hp-bass + Bass HPF setup for HP Spectre 13 + +STAC9872 +======== +vaio + VAIO laptop without SPDIF +auto + BIOS setup (default) + +Cirrus Logic CS4206/4207 +======================== +mbp55 + MacBook Pro 5,5 +imac27 + IMac 27 Inch +auto + BIOS setup (default) + +Cirrus Logic CS4208 +=================== +mba6 + MacBook Air 6,1 and 6,2 +gpio0 + Enable GPIO 0 amp +auto + BIOS setup (default) + +VIA VT17xx/VT18xx/VT20xx +======================== +auto + BIOS setup (default)

A simple conversion from a plain text file. Put to hd-audio subdirectory.

Signed-off-by: Takashi Iwai tiwai@suse.de --- .../dp-mst.rst} | 30 ++++++++++++++-------- Documentation/sound/hd-audio/index.rst | 1 + 2 files changed, 21 insertions(+), 10 deletions(-) rename Documentation/sound/{alsa/HD-Audio-DP-MST-audio.txt => hd-audio/dp-mst.rst} (69%)

diff --git a/Documentation/sound/alsa/HD-Audio-DP-MST-audio.txt b/Documentation/sound/hd-audio/dp-mst.rst similarity index 69% rename from Documentation/sound/alsa/HD-Audio-DP-MST-audio.txt rename to Documentation/sound/hd-audio/dp-mst.rst index 82744ac3513d..58b72437e6c3 100644 --- a/Documentation/sound/alsa/HD-Audio-DP-MST-audio.txt +++ b/Documentation/sound/hd-audio/dp-mst.rst @@ -1,3 +1,7 @@ +======================= +HD-Audio DP-MST Support +======================= + To support DP MST audio, HD Audio hdmi codec driver introduces virtual pin and dynamic pcm assignment.

@@ -44,10 +48,12 @@ Build Jack ----------

- dyn_pcm_assign -Will not use hda_jack but use snd_jack in spec->pcm_rec[pcm_idx].jack directly. + + Will not use hda_jack but use snd_jack in spec->pcm_rec[pcm_idx].jack directly.

- !dyn_pcm_assign -Use hda_jack and assign spec->pcm_rec[pcm_idx].jack = jack->jack statically. + + Use hda_jack and assign spec->pcm_rec[pcm_idx].jack = jack->jack statically.

Unsolicited Event Enabling @@ -58,16 +64,20 @@ Enable unsolicited event if !acomp. Monitor Hotplug Event Handling ------------------------------ - acomp -pin_eld_notify() -> check_presence_and_report() -> hdmi_present_sense() -> -sync_eld_via_acomp(). -Use directly snd_jack_report() on spec->pcm_rec[pcm_idx].jack for -both dyn_pcm_assign and !dyn_pcm_assign + + pin_eld_notify() -> check_presence_and_report() -> hdmi_present_sense() -> + sync_eld_via_acomp(). + + Use directly snd_jack_report() on spec->pcm_rec[pcm_idx].jack for + both dyn_pcm_assign and !dyn_pcm_assign

- !acomp -Hdmi_unsol_event() -> hdmi_intrinsic_event() -> check_presence_and_report() -> -hdmi_present_sense() -> hdmi_prepsent_sense_via_verbs() -Use directly snd_jack_report() on spec->pcm_rec[pcm_idx].jack for dyn_pcm_assign. -Use hda_jack mechanism to handle jack events. + + hdmi_unsol_event() -> hdmi_intrinsic_event() -> check_presence_and_report() -> + hdmi_present_sense() -> hdmi_prepsent_sense_via_verbs() + + Use directly snd_jack_report() on spec->pcm_rec[pcm_idx].jack for dyn_pcm_assign. + Use hda_jack mechanism to handle jack events.

Others to be added later diff --git a/Documentation/sound/hd-audio/index.rst b/Documentation/sound/hd-audio/index.rst index c6efd55a219f..f8a72ffffe66 100644 --- a/Documentation/sound/hd-audio/index.rst +++ b/Documentation/sound/hd-audio/index.rst @@ -7,3 +7,4 @@ HD-Audio notes models controls + dp-mst

A simple conversion from a text file.

A new subidrectory, Documentation/sound/designs, was created to put this document. The other API design and implementation docuemnts will be put to that directory in later commits.

Signed-off-by: Takashi Iwai tiwai@suse.de --- Documentation/sound/designs/index.rst | 7 ++ .../{alsa/Procfile.txt => designs/procfile.rst} | 106 +++++++++++---------- Documentation/sound/index.rst | 1 + 3 files changed, 63 insertions(+), 51 deletions(-) create mode 100644 Documentation/sound/designs/index.rst rename Documentation/sound/{alsa/Procfile.txt => designs/procfile.rst} (71%)

diff --git a/Documentation/sound/designs/index.rst b/Documentation/sound/designs/index.rst new file mode 100644 index 000000000000..82d19fe3c380 --- /dev/null +++ b/Documentation/sound/designs/index.rst @@ -0,0 +1,7 @@ +Designs and Implementations +=========================== + +.. toctree:: + :maxdepth: 2 + + procfile diff --git a/Documentation/sound/alsa/Procfile.txt b/Documentation/sound/designs/procfile.rst similarity index 71% rename from Documentation/sound/alsa/Procfile.txt rename to Documentation/sound/designs/procfile.rst index 7f8a0d325905..29a466851fd2 100644 --- a/Documentation/sound/alsa/Procfile.txt +++ b/Documentation/sound/designs/procfile.rst @@ -1,20 +1,22 @@ - Proc Files of ALSA Drivers - ========================== - Takashi Iwai tiwai@suse.de +========================== +Proc Files of ALSA Drivers +========================== + +Takashi Iwai tiwai@suse.de

General -------- +=======

ALSA has its own proc tree, /proc/asound. Many useful information are found in this tree. When you encounter a problem and need debugging, check the files listed in the following sections.

Each card has its subtree cardX, where X is from 0 to 7. The -card-specific files are stored in the card* subdirectories. +card-specific files are stored in the ``card*`` subdirectories.

Global Information ------------------- +==================

cards Shows the list of currently configured ALSA drivers, @@ -31,15 +33,15 @@ devices

meminfo Shows the status of allocated pages via ALSA drivers. - Appears only when CONFIG_SND_DEBUG=y. + Appears only when ``CONFIG_SND_DEBUG=y``.

hwdep Lists the currently available hwdep devices in format of - <card>-<device>: <name> + ``<card>-<device>: <name>``

pcm Lists the currently available PCM devices in format of - <card>-<device>: <id>: <name> : <sub-streams> + ``<card>-<device>: <id>: <name> : <sub-streams>``

timer Lists the currently available timer devices @@ -54,23 +56,23 @@ oss/sndstat

Card Specific Files -------------------- +===================

-The card-specific files are found in /proc/asound/card* directories. +The card-specific files are found in ``/proc/asound/card*`` directories. Some drivers (e.g. cmipci) have their own proc entries for the -register dump, etc (e.g. /proc/asound/card*/cmipci shows the register +register dump, etc (e.g. ``/proc/asound/card*/cmipci`` shows the register dump). These files would be really helpful for debugging.

When PCM devices are available on this card, you can see directories like pcm0p or pcm1c. They hold the PCM information for each PCM -stream. The number after 'pcm' is the PCM device number from 0, and -the last 'p' or 'c' means playback or capture direction. The files in +stream. The number after ``pcm`` is the PCM device number from 0, and +the last ``p`` or ``c`` means playback or capture direction. The files in this subtree is described later.

-The status of MIDI I/O is found in midi* files. It shows the device +The status of MIDI I/O is found in ``midi*`` files. It shows the device name and the received/transmitted bytes through the MIDI device.

-When the card is equipped with AC97 codecs, there are codec97#* +When the card is equipped with AC97 codecs, there are ``codec97#*`` subdirectories (described later).

When the OSS mixer emulation is enabled (and the module is loaded), @@ -81,26 +83,27 @@ details.

PCM Proc Files --------------- +==============

-card*/pcm*/info +``card*/pcm*/info`` The general information of this PCM device: card #, device #, substreams, etc.

-card*/pcm*/xrun_debug - This file appears when CONFIG_SND_DEBUG=y and - CONFIG_PCM_XRUN_DEBUG=y. +``card*/pcm*/xrun_debug`` + This file appears when ``CONFIG_SND_DEBUG=y`` and + ``CONFIG_PCM_XRUN_DEBUG=y``. This shows the status of xrun (= buffer overrun/xrun) and invalid PCM position debug/check of ALSA PCM middle layer. It takes an integer value, can be changed by writing to this - file, such as + file, such as::

# echo 5 > /proc/asound/card0/pcm0p/xrun_debug

The value consists of the following bit flags: - bit 0 = Enable XRUN/jiffies debug messages - bit 1 = Show stack trace at XRUN / jiffies check - bit 2 = Enable additional jiffies check + + * bit 0 = Enable XRUN/jiffies debug messages + * bit 1 = Show stack trace at XRUN / jiffies check + * bit 2 = Enable additional jiffies check

When the bit 0 is set, the driver will show the messages to kernel log when an xrun is detected. The debug message is @@ -117,72 +120,74 @@ card*/pcm*/xrun_debug buggy) hardware that doesn't give smooth pointer updates. This feature is enabled via the bit 2.

-card*/pcm*/sub*/info +``card*/pcm*/sub*/info`` The general information of this PCM sub-stream.

-card*/pcm*/sub*/status +``card*/pcm*/sub*/status`` The current status of this PCM sub-stream, elapsed time, H/W position, etc.

-card*/pcm*/sub*/hw_params +``card*/pcm*/sub*/hw_params`` The hardware parameters set for this sub-stream.

-card*/pcm*/sub*/sw_params +``card*/pcm*/sub*/sw_params`` The soft parameters set for this sub-stream.

-card*/pcm*/sub*/prealloc +``card*/pcm*/sub*/prealloc`` The buffer pre-allocation information.

-card*/pcm*/sub*/xrun_injection +``card*/pcm*/sub*/xrun_injection`` Triggers an XRUN to the running stream when any value is written to this proc file. Used for fault injection. This entry is write-only.

AC97 Codec Information ----------------------- +======================

-card*/codec97#*/ac97#?-? +``card*/codec97#*/ac97#?-?`` Shows the general information of this AC97 codec chip, such as name, capabilities, set up.

-card*/codec97#0/ac97#?-?+regs +``card*/codec97#0/ac97#?-?+regs`` Shows the AC97 register dump. Useful for debugging.

When CONFIG_SND_DEBUG is enabled, you can write to this file for changing an AC97 register directly. Pass two hex numbers. For example,

+:: + # echo 02 9f1f > /proc/asound/card0/codec97#0/ac97#0-0+regs

USB Audio Streams ------------------ +=================

-card*/stream* +``card*/stream*`` Shows the assignment and the current status of each audio stream of the given card. This information is very useful for debugging.

HD-Audio Codecs ---------------- +===============

-card*/codec#* +``card*/codec#*`` Shows the general codec information and the attribute of each widget node.

-card*/eld#* +``card*/eld#*`` Available for HDMI or DisplayPort interfaces. Shows ELD(EDID Like Data) info retrieved from the attached HDMI sink, and describes its audio capabilities and configurations.

- Some ELD fields may be modified by doing `echo name hex_value > eld#*`. + Some ELD fields may be modified by doing ``echo name hex_value > eld#*``. Only do this if you are sure the HDMI sink provided value is wrong. And if that makes your HDMI audio work, please report to us so that we can fix it in future kernel releases.

Sequencer Information ---------------------- +=====================

seq/drivers Lists the currently available ALSA sequencer drivers. @@ -203,7 +208,7 @@ seq/oss

Help For Debugging? -------------------- +===================

When the problem is related with PCM, first try to turn on xrun_debug mode. This will give you the kernel messages when and where xrun @@ -211,24 +216,23 @@ happened.

If it's really a bug, report it with the following information:

- - the name of the driver/card, show in /proc/asound/cards - - the register dump, if available (e.g. card*/cmipci) +- the name of the driver/card, show in ``/proc/asound/cards`` +- the register dump, if available (e.g. ``card*/cmipci``)

when it's a PCM problem,

- - set-up of PCM, shown in hw_parms, sw_params, and status in the PCM - sub-stream directory +- set-up of PCM, shown in hw_parms, sw_params, and status in the PCM + sub-stream directory

when it's a mixer problem,

- - AC97 proc files, codec97#*/* files +- AC97 proc files, ``codec97#*/*`` files

for USB audio/midi,

- - output of lsusb -v - - stream* files in card directory +- output of ``lsusb -v`` +- ``stream*`` files in card directory

The ALSA bug-tracking system is found at: - - https://bugtrack.alsa-project.org/alsa-bug/ +https://bugtrack.alsa-project.org/alsa-bug/ diff --git a/Documentation/sound/index.rst b/Documentation/sound/index.rst index 64fe47af05b6..e9fbbfff9d0d 100644 --- a/Documentation/sound/index.rst +++ b/Documentation/sound/index.rst @@ -6,6 +6,7 @@ Linux Sound Subsystem Documentation :maxdepth: 2

kernel-api/index + designs/index alsa-configuration hd-audio/index

A simple conversion from a text file. Put to designs subdirectory.

Signed-off-by: Takashi Iwai tiwai@suse.de --- .../channel-mapping-api.rst} | 77 ++++++++++++---------- Documentation/sound/designs/index.rst | 1 + 2 files changed, 45 insertions(+), 33 deletions(-) rename Documentation/sound/{alsa/Channel-Mapping-API.txt => designs/channel-mapping-api.rst} (75%)

diff --git a/Documentation/sound/alsa/Channel-Mapping-API.txt b/Documentation/sound/designs/channel-mapping-api.rst similarity index 75% rename from Documentation/sound/alsa/Channel-Mapping-API.txt rename to Documentation/sound/designs/channel-mapping-api.rst index 3c43d1a4ca0e..58e6312a43c0 100644 --- a/Documentation/sound/alsa/Channel-Mapping-API.txt +++ b/Documentation/sound/designs/channel-mapping-api.rst @@ -1,9 +1,11 @@ +============================ ALSA PCM channel-mapping API ============================ - Takashi Iwai tiwai@suse.de

-GENERAL -------- +Takashi Iwai tiwai@suse.de + +General +=======

The channel mapping API allows user to query the possible channel maps and the current channel map, also optionally to modify the channel map @@ -11,9 +13,9 @@ of the current stream.

A channel map is an array of position for each PCM channel. Typically, a stereo PCM stream has a channel map of - { front_left, front_right } +``{ front_left, front_right }`` while a 4.0 surround PCM stream has a channel map of - { front left, front right, rear left, rear right }. +``{ front left, front right, rear left, rear right }.``

The problem, so far, was that we had no standard channel map explicitly, and applications had no way to know which channel @@ -29,8 +31,8 @@ specification. These are the main motivations for the new channel mapping API.

-DESIGN ------- +Design +======

Actually, "the channel mapping API" doesn't introduce anything new in the kernel/user-space ABI perspective. It uses only the existing @@ -39,10 +41,11 @@ control element features. As a ground design, each PCM substream may contain a control element providing the channel mapping information and configuration. This element is specified by: - iface = SNDRV_CTL_ELEM_IFACE_PCM - name = "Playback Channel Map" or "Capture Channel Map" - device = the same device number for the assigned PCM substream - index = the same index number for the assigned PCM substream + +* iface = SNDRV_CTL_ELEM_IFACE_PCM +* name = "Playback Channel Map" or "Capture Channel Map" +* device = the same device number for the assigned PCM substream +* index = the same index number for the assigned PCM substream

Note the name is different depending on the PCM substream direction.

@@ -50,32 +53,35 @@ Each control element provides at least the TLV read operation and the read operation. Optionally, the write operation can be provided to allow user to change the channel map dynamically.

-* TLV +TLV +---

The TLV operation gives the list of available channel maps. A list item of a channel map is usually a TLV of - type data-bytes ch0 ch1 ch2... +``type data-bytes ch0 ch1 ch2...`` where type is the TLV type value, the second argument is the total bytes (not the numbers) of channel values, and the rest are the position value for each channel.

-As a TLV type, either SNDRV_CTL_TLVT_CHMAP_FIXED, -SNDRV_CTL_TLV_CHMAP_VAR or SNDRV_CTL_TLVT_CHMAP_PAIRED can be used. -The _FIXED type is for a channel map with the fixed channel position -while the latter two are for flexible channel positions. _VAR type is -for a channel map where all channels are freely swappable and _PAIRED +As a TLV type, either ``SNDRV_CTL_TLVT_CHMAP_FIXED``, +``SNDRV_CTL_TLV_CHMAP_VAR`` or ``SNDRV_CTL_TLVT_CHMAP_PAIRED`` can be used. +The ``_FIXED`` type is for a channel map with the fixed channel position +while the latter two are for flexible channel positions. ``_VAR`` type is +for a channel map where all channels are freely swappable and ``_PAIRED`` type is where pair-wise channels are swappable. For example, when you -have {FL/FR/RL/RR} channel map, _PAIRED type would allow you to swap -only {RL/RR/FL/FR} while _VAR type would allow even swapping FL and +have {FL/FR/RL/RR} channel map, ``_PAIRED`` type would allow you to swap +only {RL/RR/FL/FR} while ``_VAR`` type would allow even swapping FL and RR.

-These new TLV types are defined in sound/tlv.h. +These new TLV types are defined in ``sound/tlv.h``.

-The available channel position values are defined in sound/asound.h, +The available channel position values are defined in ``sound/asound.h``, here is a cut:

-/* channel positions */ -enum { +:: + + /* channel positions */ + enum { SNDRV_CHMAP_UNKNOWN = 0, SNDRV_CHMAP_NA, /* N/A, silent */ SNDRV_CHMAP_MONO, /* mono stream */ @@ -107,11 +113,13 @@ enum { SNDRV_CHMAP_TRR, /* top rear right */ SNDRV_CHMAP_TRC, /* top rear center */ SNDRV_CHMAP_LAST = SNDRV_CHMAP_TRC, -}; + };

When a PCM stream can provide more than one channel map, you can provide multiple channel maps in a TLV container type. The TLV data to be returned will contain such as: +:: + SNDRV_CTL_TLVT_CONTAINER 96 SNDRV_CTL_TLVT_CHMAP_FIXED 4 SNDRV_CHMAP_FC SNDRV_CTL_TLVT_CHMAP_FIXED 8 SNDRV_CHMAP_FL SNDRV_CHMAP_FR @@ -120,19 +128,21 @@ to be returned will contain such as:

The channel position is provided in LSB 16bits. The upper bits are used for bit flags. +::

-#define SNDRV_CHMAP_POSITION_MASK 0xffff -#define SNDRV_CHMAP_PHASE_INVERSE (0x01 << 16) -#define SNDRV_CHMAP_DRIVER_SPEC (0x02 << 16) + #define SNDRV_CHMAP_POSITION_MASK 0xffff + #define SNDRV_CHMAP_PHASE_INVERSE (0x01 << 16) + #define SNDRV_CHMAP_DRIVER_SPEC (0x02 << 16)

-SNDRV_CHMAP_PHASE_INVERSE indicates the channel is phase inverted, +``SNDRV_CHMAP_PHASE_INVERSE`` indicates the channel is phase inverted, (thus summing left and right channels would result in almost silence). Some digital mic devices have this.

-When SNDRV_CHMAP_DRIVER_SPEC is set, all the channel position values +When ``SNDRV_CHMAP_DRIVER_SPEC`` is set, all the channel position values don't follow the standard definition above but driver-specific.

-* READ OPERATION +Read Operation +--------------

The control read operation is for providing the current channel map of the given stream. The control element returns an integer array @@ -140,9 +150,10 @@ containing the position of each channel.

When this is performed before the number of the channel is specified (i.e. hw_params is set), it should return all channels set to -UNKNOWN. +``UNKNOWN``.

-* WRITE OPERATION +Write Operation +---------------

The control write operation is optional, and only for devices that can change the channel configuration on the fly, such as HDMI. User needs diff --git a/Documentation/sound/designs/index.rst b/Documentation/sound/designs/index.rst index 362e1c23d51f..dd4fcbcb2b92 100644 --- a/Documentation/sound/designs/index.rst +++ b/Documentation/sound/designs/index.rst @@ -4,5 +4,6 @@ Designs and Implementations .. toctree:: :maxdepth: 2

+ channel-mapping-api procfile powersave

Converted from an ancient plain HTML document. It's much readable now! Put to designs subdirectory with a slight rename.

Signed-off-by: Takashi Iwai tiwai@suse.de --- Documentation/sound/alsa/seq_oss.html | 409 -------------------------------- Documentation/sound/designs/index.rst | 1 + Documentation/sound/designs/seq-oss.rst | 371 +++++++++++++++++++++++++++++ 3 files changed, 372 insertions(+), 409 deletions(-) delete mode 100644 Documentation/sound/alsa/seq_oss.html create mode 100644 Documentation/sound/designs/seq-oss.rst

diff --git a/Documentation/sound/alsa/seq_oss.html b/Documentation/sound/alsa/seq_oss.html deleted file mode 100644 index 9663b45f6fde..000000000000 --- a/Documentation/sound/alsa/seq_oss.html +++ /dev/null @@ -1,409 +0,0 @@ -<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN"> -<HTML> -<HEAD> - <TITLE>OSS Sequencer Emulation on ALSA</TITLE> -</HEAD> -<BODY> - -<CENTER> -<H1> - -<HR WIDTH="100%"></H1></CENTER> - -<CENTER> -<H1> -OSS Sequencer Emulation on ALSA</H1></CENTER> - -<HR WIDTH="100%"> -<P>Copyright (c) 1998,1999 by Takashi Iwai -<TT><A HREF="mailto:iwai@ww.uni-erlangen.de">&lt;iwai@ww.uni-erlangen.de></A></TT> -<P>ver.0.1.8; Nov. 16, 1999 -<H2> - -<HR WIDTH="100%"></H2> - -<H2> -1. Description</H2> -This directory contains the OSS sequencer emulation driver on ALSA. Note -that this program is still in the development state. -<P>What this does - it provides the emulation of the OSS sequencer, access -via -<TT>/dev/sequencer</TT> and <TT>/dev/music</TT> devices. -The most of applications using OSS can run if the appropriate ALSA -sequencer is prepared. -<P>The following features are emulated by this driver: -<UL> -<LI> -Normal sequencer and MIDI events:</LI> - -<BR>They are converted to the ALSA sequencer events, and sent to the corresponding -port. -<LI> -Timer events:</LI> - -<BR>The timer is not selectable by ioctl. The control rate is fixed to -100 regardless of HZ. That is, even on Alpha system, a tick is always -1/100 second. The base rate and tempo can be changed in <TT>/dev/music</TT>. - -<LI> -Patch loading:</LI> - -<BR>It purely depends on the synth drivers whether it's supported since -the patch loading is realized by callback to the synth driver. -<LI> -I/O controls:</LI> - -<BR>Most of controls are accepted. Some controls -are dependent on the synth driver, as well as even on original OSS.</UL> -Furthermore, you can find the following advanced features: -<UL> -<LI> -Better queue mechanism:</LI> - -<BR>The events are queued before processing them. -<LI> -Multiple applications:</LI> - -<BR>You can run two or more applications simultaneously (even for OSS sequencer)! -However, each MIDI device is exclusive - that is, if a MIDI device is opened -once by some application, other applications can't use it. No such a restriction -in synth devices. -<LI> -Real-time event processing:</LI> - -<BR>The events can be processed in real time without using out of bound -ioctl. To switch to real-time mode, send ABSTIME 0 event. The followed -events will be processed in real-time without queued. To switch off the -real-time mode, send RELTIME 0 event. -<LI> -<TT>/proc</TT> interface:</LI> - -<BR>The status of applications and devices can be shown via <TT>/proc/asound/seq/oss</TT> -at any time. In the later version, configuration will be changed via <TT>/proc</TT> -interface, too.</UL> - -<H2> -2. Installation</H2> -Run configure script with both sequencer support (<TT>--with-sequencer=yes</TT>) -and OSS emulation (<TT>--with-oss=yes</TT>) options. A module <TT>snd-seq-oss.o</TT> -will be created. If the synth module of your sound card supports for OSS -emulation (so far, only Emu8000 driver), this module will be loaded automatically. -Otherwise, you need to load this module manually. -<P>At beginning, this module probes all the MIDI ports which have been -already connected to the sequencer. Once after that, the creation and deletion -of ports are watched by announcement mechanism of ALSA sequencer. -<P>The available synth and MIDI devices can be found in proc interface. -Run "<TT>cat /proc/asound/seq/oss</TT>", and check the devices. For example, -if you use an AWE64 card, you'll see like the following: -<PRE>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; OSS sequencer emulation version 0.1.8 -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ALSA client number 63 -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ALSA receiver port 0 - -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Number of applications: 0 - -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Number of synth devices: 1 - -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; synth 0: [EMU8000] -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; type 0x1 : subtype 0x20 : voices 32 -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; capabilties : ioctl enabled / load_patch enabled - -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Number of MIDI devices: 3 - -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; midi 0: [Emu8000 Port-0] ALSA port 65:0 -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; capability write / opened none - -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; midi 1: [Emu8000 Port-1] ALSA port 65:1 -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; capability write / opened none - -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; midi 2: [0: MPU-401 (UART)] ALSA port 64:0 -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; capability read/write / opened none</PRE> -Note that the device number may be different from the information of -<TT>/proc/asound/oss-devices</TT> -or ones of the original OSS driver. Use the device number listed in <TT>/proc/asound/seq/oss</TT> -to play via OSS sequencer emulation. -<H2> -3. Using Synthesizer Devices</H2> -Run your favorite program. I've tested playmidi-2.4, awemidi-0.4.3, gmod-3.1 -and xmp-1.1.5. You can load samples via <TT>/dev/sequencer</TT> like sfxload, -too. -<P>If the lowlevel driver supports multiple access to synth devices (like -Emu8000 driver), two or more applications are allowed to run at the same -time. -<H2> -4. Using MIDI Devices</H2> -So far, only MIDI output was tested. MIDI input was not checked at all, -but hopefully it will work. Use the device number listed in <TT>/proc/asound/seq/oss</TT>. -Be aware that these numbers are mostly different from the list in -<TT>/proc/asound/oss-devices</TT>. -<H2> -5. Module Options</H2> -The following module options are available: -<UL> -<LI> -<TT>maxqlen</TT></LI> - -<BR>specifies the maximum read/write queue length. This queue is private -for OSS sequencer, so that it is independent from the queue length of ALSA -sequencer. Default value is 1024. -<LI> -<TT>seq_oss_debug</TT></LI> - -<BR>specifies the debug level and accepts zero (= no debug message) or -positive integer. Default value is 0.</UL> - -<H2> -6. Queue Mechanism</H2> -OSS sequencer emulation uses an ALSA priority queue. The -events from <TT>/dev/sequencer</TT> are processed and put onto the queue -specified by module option. -<P>All the events from <TT>/dev/sequencer</TT> are parsed at beginning. -The timing events are also parsed at this moment, so that the events may -be processed in real-time. Sending an event ABSTIME 0 switches the operation -mode to real-time mode, and sending an event RELTIME 0 switches it off. -In the real-time mode, all events are dispatched immediately. -<P>The queued events are dispatched to the corresponding ALSA sequencer -ports after scheduled time by ALSA sequencer dispatcher. -<P>If the write-queue is full, the application sleeps until a certain amount -(as default one half) becomes empty in blocking mode. The synchronization -to write timing was implemented, too. -<P>The input from MIDI devices or echo-back events are stored on read FIFO -queue. If application reads <TT>/dev/sequencer</TT> in blocking mode, the -process will be awaked. - -<H2> -7. Interface to Synthesizer Device</H2> - -<H3> -7.1. Registration</H3> -To register an OSS synthesizer device, use <TT>snd_seq_oss_synth_register</TT> -function. -<PRE>int snd_seq_oss_synth_register(char *name, int type, int subtype, int nvoices, -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; snd_seq_oss_callback_t *oper, void *private_data)</PRE> -The arguments <TT>name</TT>, <TT>type</TT>, <TT>subtype</TT> and -<TT>nvoices</TT> -are used for making the appropriate synth_info structure for ioctl. The -return value is an index number of this device. This index must be remembered -for unregister. If registration is failed, -errno will be returned. -<P>To release this device, call <TT>snd_seq_oss_synth_unregister function</TT>: -<PRE>int snd_seq_oss_synth_unregister(int index),</PRE> -where the <TT>index</TT> is the index number returned by register function. -<H3> -7.2. Callbacks</H3> -OSS synthesizer devices have capability for sample downloading and ioctls -like sample reset. In OSS emulation, these special features are realized -by using callbacks. The registration argument oper is used to specify these -callbacks. The following callback functions must be defined: -<PRE>snd_seq_oss_callback_t: -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int (*open)(snd_seq_oss_arg_t *p, void *closure); -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int (*close)(snd_seq_oss_arg_t *p); -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int (*ioctl)(snd_seq_oss_arg_t *p, unsigned int cmd, unsigned long arg); -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int (*load_patch)(snd_seq_oss_arg_t *p, int format, const char *buf, int offs, int count); -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int (*reset)(snd_seq_oss_arg_t *p); -Except for <TT>open</TT> and <TT>close</TT> callbacks, they are allowed -to be NULL. -<P>Each callback function takes the argument type snd_seq_oss_arg_t as the -first argument. -<PRE>struct snd_seq_oss_arg_t { -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int app_index; -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int file_mode; -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int seq_mode; -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; snd_seq_addr_t addr; -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; void *private_data; -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int event_passing; -};</PRE> -The first three fields, <TT>app_index</TT>, <TT>file_mode</TT> and -<TT>seq_mode</TT> -are initialized by OSS sequencer. The <TT>app_index</TT> is the application -index which is unique to each application opening OSS sequencer. The -<TT>file_mode</TT> -is bit-flags indicating the file operation mode. See -<TT>seq_oss.h</TT> -for its meaning. The <TT>seq_mode</TT> is sequencer operation mode. In -the current version, only <TT>SND_OSSSEQ_MODE_SYNTH</TT> is used. -<P>The next two fields, <TT>addr</TT> and <TT>private_data</TT>, must be -filled by the synth driver at open callback. The <TT>addr</TT> contains -the address of ALSA sequencer port which is assigned to this device. If -the driver allocates memory for <TT>private_data</TT>, it must be released -in close callback by itself. -<P>The last field, <TT>event_passing</TT>, indicates how to translate note-on -/ off events. In <TT>PROCESS_EVENTS</TT> mode, the note 255 is regarded -as velocity change, and key pressure event is passed to the port. In <TT>PASS_EVENTS</TT> -mode, all note on/off events are passed to the port without modified. <TT>PROCESS_KEYPRESS</TT> -mode checks the note above 128 and regards it as key pressure event (mainly -for Emu8000 driver). -<H4> -7.2.1. Open Callback</H4> -The <TT>open</TT> is called at each time this device is opened by an application -using OSS sequencer. This must not be NULL. Typically, the open callback -does the following procedure: -<OL> -<LI> -Allocate private data record.</LI> - -<LI> -Create an ALSA sequencer port.</LI> - -<LI> -Set the new port address on arg->addr.</LI> - -<LI> -Set the private data record pointer on arg->private_data.</LI> -</OL> -Note that the type bit-flags in port_info of this synth port must NOT contain -<TT>TYPE_MIDI_GENERIC</TT> -bit. Instead, <TT>TYPE_SPECIFIC</TT> should be used. Also, <TT>CAP_SUBSCRIPTION</TT> -bit should NOT be included, too. This is necessary to tell it from other -normal MIDI devices. If the open procedure succeeded, return zero. Otherwise, -return -errno. -<H4> -7.2.2 Ioctl Callback</H4> -The <TT>ioctl</TT> callback is called when the sequencer receives device-specific -ioctls. The following two ioctls should be processed by this callback: -<UL> -<LI> -<TT>IOCTL_SEQ_RESET_SAMPLES</TT></LI> - -<BR>reset all samples on memory -- return 0 -<LI> -<TT>IOCTL_SYNTH_MEMAVL</TT></LI> - -<BR>return the available memory size -<LI> -<TT>FM_4OP_ENABLE</TT></LI> - -<BR>can be ignored usually</UL> -The other ioctls are processed inside the sequencer without passing to -the lowlevel driver. -<H4> -7.2.3 Load_Patch Callback</H4> -The <TT>load_patch</TT> callback is used for sample-downloading. This callback -must read the data on user-space and transfer to each device. Return 0 -if succeeded, and -errno if failed. The format argument is the patch key -in patch_info record. The buf is user-space pointer where patch_info record -is stored. The offs can be ignored. The count is total data size of this -sample data. -<H4> -7.2.4 Close Callback</H4> -The <TT>close</TT> callback is called when this device is closed by the -application. If any private data was allocated in open callback, it must -be released in the close callback. The deletion of ALSA port should be -done here, too. This callback must not be NULL. -<H4> -7.2.5 Reset Callback</H4> -The <TT>reset</TT> callback is called when sequencer device is reset or -closed by applications. The callback should turn off the sounds on the -relevant port immediately, and initialize the status of the port. If this -callback is undefined, OSS seq sends a <TT>HEARTBEAT</TT> event to the -port. -<H3> -7.3 Events</H3> -Most of the events are processed by sequencer and translated to the adequate -ALSA sequencer events, so that each synth device can receive by input_event -callback of ALSA sequencer port. The following ALSA events should be implemented -by the driver: -<BR>&nbsp; -<TABLE BORDER WIDTH="75%" NOSAVE > -<TR NOSAVE> -<TD NOSAVE><B>ALSA event</B></TD> - -<TD><B>Original OSS events</B></TD> -</TR> - -<TR> -<TD>NOTEON</TD> - -<TD>SEQ_NOTEON -<BR>MIDI_NOTEON</TD> -</TR> - -<TR> -<TD>NOTE</TD> - -<TD>SEQ_NOTEOFF -<BR>MIDI_NOTEOFF</TD> -</TR> - -<TR NOSAVE> -<TD NOSAVE>KEYPRESS</TD> - -<TD>MIDI_KEY_PRESSURE</TD> -</TR> - -<TR NOSAVE> -<TD>CHANPRESS</TD> - -<TD NOSAVE>SEQ_AFTERTOUCH -<BR>MIDI_CHN_PRESSURE</TD> -</TR> - -<TR NOSAVE> -<TD NOSAVE>PGMCHANGE</TD> - -<TD NOSAVE>SEQ_PGMCHANGE -<BR>MIDI_PGM_CHANGE</TD> -</TR> - -<TR> -<TD>PITCHBEND</TD> - -<TD>SEQ_CONTROLLER(CTRL_PITCH_BENDER) -<BR>MIDI_PITCH_BEND</TD> -</TR> - -<TR> -<TD>CONTROLLER</TD> - -<TD>MIDI_CTL_CHANGE -<BR>SEQ_BALANCE (with CTL_PAN)</TD> -</TR> - -<TR> -<TD>CONTROL14</TD> - -<TD>SEQ_CONTROLLER</TD> -</TR> - -<TR> -<TD>REGPARAM</TD> - -<TD>SEQ_CONTROLLER(CTRL_PITCH_BENDER_RANGE)</TD> -</TR> - -<TR> -<TD>SYSEX</TD> - -<TD>SEQ_SYSEX</TD> -</TR> -</TABLE> - -<P>The most of these behavior can be realized by MIDI emulation driver -included in the Emu8000 lowlevel driver. In the future release, this module -will be independent. -<P>Some OSS events (<TT>SEQ_PRIVATE</TT> and <TT>SEQ_VOLUME</TT> events) are passed as event -type SND_SEQ_OSS_PRIVATE. The OSS sequencer passes these event 8 byte -packets without any modification. The lowlevel driver should process these -events appropriately. -<H2> -8. Interface to MIDI Device</H2> -Since the OSS emulation probes the creation and deletion of ALSA MIDI sequencer -ports automatically by receiving announcement from ALSA sequencer, the -MIDI devices don't need to be registered explicitly like synth devices. -However, the MIDI port_info registered to ALSA sequencer must include a group -name <TT>SND_SEQ_GROUP_DEVICE</TT> and a capability-bit <TT>CAP_READ</TT> or -<TT>CAP_WRITE</TT>. Also, subscription capabilities, <TT>CAP_SUBS_READ</TT> or <TT>CAP_SUBS_WRITE</TT>, -must be defined, too. If these conditions are not satisfied, the port is not -registered as OSS sequencer MIDI device. -<P>The events via MIDI devices are parsed in OSS sequencer and converted -to the corresponding ALSA sequencer events. The input from MIDI sequencer -is also converted to MIDI byte events by OSS sequencer. This works just -a reverse way of seq_midi module. -<H2> -9. Known Problems / TODO's</H2> - -<UL> -<LI> -Patch loading via ALSA instrument layer is not implemented yet.</LI> -</UL> - -</BODY> -</HTML> diff --git a/Documentation/sound/designs/index.rst b/Documentation/sound/designs/index.rst index 5b3033c556d0..0ca3a6b0a5ce 100644 --- a/Documentation/sound/designs/index.rst +++ b/Documentation/sound/designs/index.rst @@ -8,3 +8,4 @@ Designs and Implementations procfile powersave oss-emulation + seq-oss diff --git a/Documentation/sound/designs/seq-oss.rst b/Documentation/sound/designs/seq-oss.rst new file mode 100644 index 000000000000..e82ffe0e7f43 --- /dev/null +++ b/Documentation/sound/designs/seq-oss.rst @@ -0,0 +1,371 @@ +=============================== +OSS Sequencer Emulation on ALSA +=============================== + +Copyright (c) 1998,1999 by Takashi Iwai + +ver.0.1.8; Nov. 16, 1999 + +Description +=========== + +This directory contains the OSS sequencer emulation driver on ALSA. Note +that this program is still in the development state. + +What this does - it provides the emulation of the OSS sequencer, access +via ``/dev/sequencer`` and ``/dev/music`` devices. +The most of applications using OSS can run if the appropriate ALSA +sequencer is prepared. + +The following features are emulated by this driver: + +* Normal sequencer and MIDI events: + + They are converted to the ALSA sequencer events, and sent to the + corresponding port. + +* Timer events: + + The timer is not selectable by ioctl. The control rate is fixed to + 100 regardless of HZ. That is, even on Alpha system, a tick is always + 1/100 second. The base rate and tempo can be changed in ``/dev/music``. + +* Patch loading: + + It purely depends on the synth drivers whether it's supported since + the patch loading is realized by callback to the synth driver. + +* I/O controls: + + Most of controls are accepted. Some controls + are dependent on the synth driver, as well as even on original OSS. + +Furthermore, you can find the following advanced features: + +* Better queue mechanism: + + The events are queued before processing them. + +* Multiple applications: + + You can run two or more applications simultaneously (even for OSS + sequencer)! + However, each MIDI device is exclusive - that is, if a MIDI device + is opened once by some application, other applications can't use + it. No such a restriction in synth devices. + +* Real-time event processing: + + The events can be processed in real time without using out of bound + ioctl. To switch to real-time mode, send ABSTIME 0 event. The followed + events will be processed in real-time without queued. To switch off the + real-time mode, send RELTIME 0 event. + +* ``/proc`` interface: + + The status of applications and devices can be shown via + ``/proc/asound/seq/oss`` at any time. In the later version, + configuration will be changed via ``/proc`` interface, too. + + +Installation +============ + +Run configure script with both sequencer support (``--with-sequencer=yes``) +and OSS emulation (``--with-oss=yes``) options. A module ``snd-seq-oss.o`` +will be created. If the synth module of your sound card supports for OSS +emulation (so far, only Emu8000 driver), this module will be loaded +automatically. +Otherwise, you need to load this module manually. + +At beginning, this module probes all the MIDI ports which have been +already connected to the sequencer. Once after that, the creation and deletion +of ports are watched by announcement mechanism of ALSA sequencer. + +The available synth and MIDI devices can be found in proc interface. +Run ``cat /proc/asound/seq/oss``, and check the devices. For example, +if you use an AWE64 card, you'll see like the following: +:: + + OSS sequencer emulation version 0.1.8 + ALSA client number 63 + ALSA receiver port 0 + + Number of applications: 0 + + Number of synth devices: 1 + synth 0: [EMU8000] + type 0x1 : subtype 0x20 : voices 32 + capabilties : ioctl enabled / load_patch enabled + + Number of MIDI devices: 3 + midi 0: [Emu8000 Port-0] ALSA port 65:0 + capability write / opened none + + midi 1: [Emu8000 Port-1] ALSA port 65:1 + capability write / opened none + + midi 2: [0: MPU-401 (UART)] ALSA port 64:0 + capability read/write / opened none + +Note that the device number may be different from the information of +``/proc/asound/oss-devices`` or ones of the original OSS driver. +Use the device number listed in ``/proc/asound/seq/oss`` +to play via OSS sequencer emulation. + +Using Synthesizer Devices +========================= + +Run your favorite program. I've tested playmidi-2.4, awemidi-0.4.3, gmod-3.1 +and xmp-1.1.5. You can load samples via ``/dev/sequencer`` like sfxload, +too. + +If the lowlevel driver supports multiple access to synth devices (like +Emu8000 driver), two or more applications are allowed to run at the same +time. + +Using MIDI Devices +================== + +So far, only MIDI output was tested. MIDI input was not checked at all, +but hopefully it will work. Use the device number listed in +``/proc/asound/seq/oss``. +Be aware that these numbers are mostly different from the list in +``/proc/asound/oss-devices``. + +Module Options +============== + +The following module options are available: + +maxqlen + specifies the maximum read/write queue length. This queue is private + for OSS sequencer, so that it is independent from the queue length of ALSA + sequencer. Default value is 1024. + +seq_oss_debug + specifies the debug level and accepts zero (= no debug message) or + positive integer. Default value is 0. + +Queue Mechanism +=============== + +OSS sequencer emulation uses an ALSA priority queue. The +events from ``/dev/sequencer`` are processed and put onto the queue +specified by module option. + +All the events from ``/dev/sequencer`` are parsed at beginning. +The timing events are also parsed at this moment, so that the events may +be processed in real-time. Sending an event ABSTIME 0 switches the operation +mode to real-time mode, and sending an event RELTIME 0 switches it off. +In the real-time mode, all events are dispatched immediately. + +The queued events are dispatched to the corresponding ALSA sequencer +ports after scheduled time by ALSA sequencer dispatcher. + +If the write-queue is full, the application sleeps until a certain amount +(as default one half) becomes empty in blocking mode. The synchronization +to write timing was implemented, too. + +The input from MIDI devices or echo-back events are stored on read FIFO +queue. If application reads ``/dev/sequencer`` in blocking mode, the +process will be awaked. + +Interface to Synthesizer Device +=============================== + +Registration +------------ + +To register an OSS synthesizer device, use snd_seq_oss_synth_register() +function: +:: + + int snd_seq_oss_synth_register(char *name, int type, int subtype, int nvoices, + snd_seq_oss_callback_t *oper, void *private_data) + +The arguments ``name``, ``type``, ``subtype`` and ``nvoices`` +are used for making the appropriate synth_info structure for ioctl. The +return value is an index number of this device. This index must be remembered +for unregister. If registration is failed, -errno will be returned. + +To release this device, call snd_seq_oss_synth_unregister() function: +:: + + int snd_seq_oss_synth_unregister(int index) + +where the ``index`` is the index number returned by register function. + +Callbacks +--------- + +OSS synthesizer devices have capability for sample downloading and ioctls +like sample reset. In OSS emulation, these special features are realized +by using callbacks. The registration argument oper is used to specify these +callbacks. The following callback functions must be defined: +:: + + snd_seq_oss_callback_t: + int (*open)(snd_seq_oss_arg_t *p, void *closure); + int (*close)(snd_seq_oss_arg_t *p); + int (*ioctl)(snd_seq_oss_arg_t *p, unsigned int cmd, unsigned long arg); + int (*load_patch)(snd_seq_oss_arg_t *p, int format, const char *buf, int offs, int count); + int (*reset)(snd_seq_oss_arg_t *p); + +Except for ``open`` and ``close`` callbacks, they are allowed to be NULL. + +Each callback function takes the argument type ``snd_seq_oss_arg_t`` as the +first argument. +:: + + struct snd_seq_oss_arg_t { + int app_index; + int file_mode; + int seq_mode; + snd_seq_addr_t addr; + void *private_data; + int event_passing; + }; + +The first three fields, ``app_index``, ``file_mode`` and ``seq_mode`` +are initialized by OSS sequencer. The ``app_index`` is the application +index which is unique to each application opening OSS sequencer. The +``file_mode`` is bit-flags indicating the file operation mode. See +``seq_oss.h`` for its meaning. The ``seq_mode`` is sequencer operation +mode. In the current version, only ``SND_OSSSEQ_MODE_SYNTH`` is used. + +The next two fields, ``addr`` and ``private_data``, must be +filled by the synth driver at open callback. The ``addr`` contains +the address of ALSA sequencer port which is assigned to this device. If +the driver allocates memory for ``private_data``, it must be released +in close callback by itself. + +The last field, ``event_passing``, indicates how to translate note-on +/ off events. In ``PROCESS_EVENTS`` mode, the note 255 is regarded +as velocity change, and key pressure event is passed to the port. In +``PASS_EVENTS`` mode, all note on/off events are passed to the port +without modified. ``PROCESS_KEYPRESS`` mode checks the note above 128 +and regards it as key pressure event (mainly for Emu8000 driver). + +Open Callback +------------- + +The ``open`` is called at each time this device is opened by an application +using OSS sequencer. This must not be NULL. Typically, the open callback +does the following procedure: + +#. Allocate private data record. +#. Create an ALSA sequencer port. +#. Set the new port address on ``arg->addr``. +#. Set the private data record pointer on ``arg->private_data``. + +Note that the type bit-flags in port_info of this synth port must NOT contain +``TYPE_MIDI_GENERIC`` +bit. Instead, ``TYPE_SPECIFIC`` should be used. Also, ``CAP_SUBSCRIPTION`` +bit should NOT be included, too. This is necessary to tell it from other +normal MIDI devices. If the open procedure succeeded, return zero. Otherwise, +return -errno. + +Ioctl Callback +-------------- + +The ``ioctl`` callback is called when the sequencer receives device-specific +ioctls. The following two ioctls should be processed by this callback: + +IOCTL_SEQ_RESET_SAMPLES + reset all samples on memory -- return 0 + +IOCTL_SYNTH_MEMAVL + return the available memory size + +FM_4OP_ENABLE + can be ignored usually + +The other ioctls are processed inside the sequencer without passing to +the lowlevel driver. + +Load_Patch Callback +------------------- + +The ``load_patch`` callback is used for sample-downloading. This callback +must read the data on user-space and transfer to each device. Return 0 +if succeeded, and -errno if failed. The format argument is the patch key +in patch_info record. The buf is user-space pointer where patch_info record +is stored. The offs can be ignored. The count is total data size of this +sample data. + +Close Callback +-------------- + +The ``close`` callback is called when this device is closed by the +application. If any private data was allocated in open callback, it must +be released in the close callback. The deletion of ALSA port should be +done here, too. This callback must not be NULL. + +Reset Callback +-------------- + +The ``reset`` callback is called when sequencer device is reset or +closed by applications. The callback should turn off the sounds on the +relevant port immediately, and initialize the status of the port. If this +callback is undefined, OSS seq sends a ``HEARTBEAT`` event to the +port. + +Events +====== + +Most of the events are processed by sequencer and translated to the adequate +ALSA sequencer events, so that each synth device can receive by input_event +callback of ALSA sequencer port. The following ALSA events should be +implemented by the driver: + +============= =================== +ALSA event Original OSS events +============= =================== +NOTEON SEQ_NOTEON, MIDI_NOTEON +NOTE SEQ_NOTEOFF, MIDI_NOTEOFF +KEYPRESS MIDI_KEY_PRESSURE +CHANPRESS SEQ_AFTERTOUCH, MIDI_CHN_PRESSURE +PGMCHANGE SEQ_PGMCHANGE, MIDI_PGM_CHANGE +PITCHBEND SEQ_CONTROLLER(CTRL_PITCH_BENDER), + MIDI_PITCH_BEND +CONTROLLER MIDI_CTL_CHANGE, + SEQ_BALANCE (with CTL_PAN) +CONTROL14 SEQ_CONTROLLER +REGPARAM SEQ_CONTROLLER(CTRL_PITCH_BENDER_RANGE) +SYSEX SEQ_SYSEX +============= =================== + +The most of these behavior can be realized by MIDI emulation driver +included in the Emu8000 lowlevel driver. In the future release, this module +will be independent. + +Some OSS events (``SEQ_PRIVATE`` and ``SEQ_VOLUME`` events) are passed as event +type SND_SEQ_OSS_PRIVATE. The OSS sequencer passes these event 8 byte +packets without any modification. The lowlevel driver should process these +events appropriately. + +Interface to MIDI Device +======================== + +Since the OSS emulation probes the creation and deletion of ALSA MIDI +sequencer ports automatically by receiving announcement from ALSA +sequencer, the MIDI devices don't need to be registered explicitly +like synth devices. +However, the MIDI port_info registered to ALSA sequencer must include +a group name ``SND_SEQ_GROUP_DEVICE`` and a capability-bit +``CAP_READ`` or ``CAP_WRITE``. Also, subscription capabilities, +``CAP_SUBS_READ`` or ``CAP_SUBS_WRITE``, must be defined, too. If +these conditions are not satisfied, the port is not registered as OSS +sequencer MIDI device. + +The events via MIDI devices are parsed in OSS sequencer and converted +to the corresponding ALSA sequencer events. The input from MIDI sequencer +is also converted to MIDI byte events by OSS sequencer. This works just +a reverse way of seq_midi module. + +Known Problems / TODO's +======================= + +* Patch loading via ALSA instrument layer is not implemented yet. +

A simple conversion from a plain text file. Put to designs subdirectory.

Signed-off-by: Takashi Iwai tiwai@suse.de --- .../compress-offload.rst} | 127 +++++++++++---------- Documentation/sound/designs/index.rst | 1 + 2 files changed, 70 insertions(+), 58 deletions(-) rename Documentation/sound/{alsa/compress_offload.txt => designs/compress-offload.rst} (73%)

diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/designs/compress-offload.rst similarity index 73% rename from Documentation/sound/alsa/compress_offload.txt rename to Documentation/sound/designs/compress-offload.rst index 8ba556a131c3..ad4bfbdacc83 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/designs/compress-offload.rst @@ -1,10 +1,14 @@ - compress_offload.txt - ===================== - Pierre-Louis.Bossart pierre-louis.bossart@linux.intel.com - Vinod Koul vinod.koul@linux.intel.com +========================= +ALSA Compress-Offload API +========================= + +Pierre-Louis.Bossart pierre-louis.bossart@linux.intel.com + +Vinod Koul vinod.koul@linux.intel.com

-Overview

+Overview +======== Since its early days, the ALSA API was defined with PCM support or constant bitrates payloads such as IEC61937 in mind. Arguments and returned values in frames are the norm, making it a challenge to @@ -27,8 +31,9 @@ Intel Moorestown SOC, with many corrections required to upstream the API in the mainline kernel instead of the staging tree and make it usable by others.

-Requirements

+Requirements +============ The main requirements are:

- separation between byte counts and time. Compressed formats may have @@ -63,7 +68,7 @@ The main requirements are: streaming compressed data to a DSP, with the assumption that the decoded samples are routed to a physical output or logical back-end.

- - Complexity hiding. Existing user-space multimedia frameworks all +- Complexity hiding. Existing user-space multimedia frameworks all have existing enums/structures for each compressed format. This new API assumes the existence of a platform-specific compatibility layer to expose, translate and make use of the capabilities of the audio @@ -72,7 +77,7 @@ The main requirements are:

Design - +====== The new API shares a number of concepts with the PCM API for flow control. Start, pause, resume, drain and stop commands have the same semantics no matter what the content is. @@ -95,43 +100,44 @@ mandatory routines and possibly make use of optional ones.

The main additions are

-- get_caps -This routine returns the list of audio formats supported. Querying the -codecs on a capture stream will return encoders, decoders will be -listed for playback streams. - -- get_codec_caps For each codec, this routine returns a list of -capabilities. The intent is to make sure all the capabilities -correspond to valid settings, and to minimize the risks of -configuration failures. For example, for a complex codec such as AAC, -the number of channels supported may depend on a specific profile. If -the capabilities were exposed with a single descriptor, it may happen -that a specific combination of profiles/channels/formats may not be -supported. Likewise, embedded DSPs have limited memory and cpu cycles, -it is likely that some implementations make the list of capabilities -dynamic and dependent on existing workloads. In addition to codec -settings, this routine returns the minimum buffer size handled by the -implementation. This information can be a function of the DMA buffer -sizes, the number of bytes required to synchronize, etc, and can be -used by userspace to define how much needs to be written in the ring -buffer before playback can start. - -- set_params -This routine sets the configuration chosen for a specific codec. The -most important field in the parameters is the codec type; in most -cases decoders will ignore other fields, while encoders will strictly -comply to the settings - -- get_params -This routines returns the actual settings used by the DSP. Changes to -the settings should remain the exception. - -- get_timestamp -The timestamp becomes a multiple field structure. It lists the number -of bytes transferred, the number of samples processed and the number -of samples rendered/grabbed. All these values can be used to determine -the average bitrate, figure out if the ring buffer needs to be -refilled or the delay due to decoding/encoding/io on the DSP. +get_caps + This routine returns the list of audio formats supported. Querying the + codecs on a capture stream will return encoders, decoders will be + listed for playback streams. + +get_codec_caps + For each codec, this routine returns a list of + capabilities. The intent is to make sure all the capabilities + correspond to valid settings, and to minimize the risks of + configuration failures. For example, for a complex codec such as AAC, + the number of channels supported may depend on a specific profile. If + the capabilities were exposed with a single descriptor, it may happen + that a specific combination of profiles/channels/formats may not be + supported. Likewise, embedded DSPs have limited memory and cpu cycles, + it is likely that some implementations make the list of capabilities + dynamic and dependent on existing workloads. In addition to codec + settings, this routine returns the minimum buffer size handled by the + implementation. This information can be a function of the DMA buffer + sizes, the number of bytes required to synchronize, etc, and can be + used by userspace to define how much needs to be written in the ring + buffer before playback can start. + +set_params + This routine sets the configuration chosen for a specific codec. The + most important field in the parameters is the codec type; in most + cases decoders will ignore other fields, while encoders will strictly + comply to the settings + +get_params + This routines returns the actual settings used by the DSP. Changes to + the settings should remain the exception. + +get_timestamp + The timestamp becomes a multiple field structure. It lists the number + of bytes transferred, the number of samples processed and the number + of samples rendered/grabbed. All these values can be used to determine + the average bitrate, figure out if the ring buffer needs to be + refilled or the delay due to decoding/encoding/io on the DSP.

Note that the list of codecs/profiles/modes was derived from the OpenMAX AL specification instead of reinventing the wheel. @@ -145,6 +151,7 @@ Modifications include: - Addition of encoding options when required (derived from OpenMAX IL) - Addition of rateControlSupported (missing in OpenMAX AL)

+ Gapless Playback ================ When playing thru an album, the decoders have the ability to skip the encoder @@ -162,19 +169,19 @@ switch from one track to another and start using data for second track.

The main additions are:

-- set_metadata -This routine sets the encoder delay and encoder padding. This can be used by -decoder to strip the silence. This needs to be set before the data in the track -is written. +set_metadata + This routine sets the encoder delay and encoder padding. This can be used by + decoder to strip the silence. This needs to be set before the data in the track + is written.

-- set_next_track -This routine tells DSP that metadata and write operation sent after this would -correspond to subsequent track +set_next_track + This routine tells DSP that metadata and write operation sent after this would + correspond to subsequent track

-- partial drain -This is called when end of file is reached. The userspace can inform DSP that -EOF is reached and now DSP can start skipping padding delay. Also next write -data would belong to next track +partial drain + This is called when end of file is reached. The userspace can inform DSP that + EOF is reached and now DSP can start skipping padding delay. Also next write + data would belong to next track

Sequence flow for gapless would be: - Open @@ -189,10 +196,12 @@ Sequence flow for gapless would be: - then call partial_drain to flush most of buffer in DSP - Fill data of the next track - DSP switches to second track + (note: order for partial_drain and write for next track can be reversed as well)

-Not supported:

+Not supported +============= - Support for VoIP/circuit-switched calls is not the target of this API. Support for dynamic bit-rate changes would require a tight coupling between the DSP and the host stack, limiting power savings. @@ -225,7 +234,9 @@ Not supported: rendered output in time, this does not deal with underrun/overrun and maybe dealt in user-library

-Credits: + +Credits +======= - Mark Brown and Liam Girdwood for discussions on the need for this API - Harsha Priya for her work on intel_sst compressed API - Rakesh Ughreja for valuable feedback diff --git a/Documentation/sound/designs/index.rst b/Documentation/sound/designs/index.rst index e53a5fac0acf..f7ca11307033 100644 --- a/Documentation/sound/designs/index.rst +++ b/Documentation/sound/designs/index.rst @@ -6,6 +6,7 @@ Designs and Implementations

control-names channel-mapping-api + compress-offload procfile powersave oss-emulation

A simple conversion from a plain text file. Put to designs subdirectory.

Signed-off-by: Takashi Iwai tiwai@suse.de --- Documentation/sound/designs/index.rst | 1 + .../{alsa/Jack-Controls.txt => designs/jack-controls.rst} | 13 +++++++++---- 2 files changed, 10 insertions(+), 4 deletions(-) rename Documentation/sound/{alsa/Jack-Controls.txt => designs/jack-controls.rst} (86%)

diff --git a/Documentation/sound/designs/index.rst b/Documentation/sound/designs/index.rst index 798b1a44bbbd..04dcdae3e4f2 100644 --- a/Documentation/sound/designs/index.rst +++ b/Documentation/sound/designs/index.rst @@ -8,6 +8,7 @@ Designs and Implementations channel-mapping-api compress-offload timestamping + jack-controls procfile powersave oss-emulation diff --git a/Documentation/sound/alsa/Jack-Controls.txt b/Documentation/sound/designs/jack-controls.rst similarity index 86% rename from Documentation/sound/alsa/Jack-Controls.txt rename to Documentation/sound/designs/jack-controls.rst index fe1c5e0c8555..ae25b1531bb0 100644 --- a/Documentation/sound/alsa/Jack-Controls.txt +++ b/Documentation/sound/designs/jack-controls.rst @@ -1,3 +1,7 @@ +================== +ALSA Jack Controls +================== + Why we need Jack kcontrols ==========================

@@ -29,11 +33,12 @@ How to use jack kcontrols =========================

In order to keep compatibility, snd_jack_new() has been modified by -adding two params :- +adding two params:

- - @initial_kctl: if true, create a kcontrol and add it to the jack - list. - - @phantom_jack: Don't create a input device for phantom jacks. +initial_kctl + if true, create a kcontrol and add it to the jack list. +phantom_jack + Don't create a input device for phantom jacks.

HDA jacks can set phantom_jack to true in order to create a phantom jack and set initial_kctl to true to create an initial kcontrol with

A simple conversion from a plain text file. Put to cards subdirectory.

Signed-off-by: Takashi Iwai tiwai@suse.de --- .../sound/{alsa/CMIPCI.txt => cards/cmipci.rst} | 62 ++++++++++++++-------- Documentation/sound/cards/index.rst | 1 + 2 files changed, 41 insertions(+), 22 deletions(-) rename Documentation/sound/{alsa/CMIPCI.txt => cards/cmipci.rst} (86%)

diff --git a/Documentation/sound/alsa/CMIPCI.txt b/Documentation/sound/cards/cmipci.rst similarity index 86% rename from Documentation/sound/alsa/CMIPCI.txt rename to Documentation/sound/cards/cmipci.rst index 4e36e6e809ca..9ea1de6ec4ce 100644 --- a/Documentation/sound/alsa/CMIPCI.txt +++ b/Documentation/sound/cards/cmipci.rst @@ -1,7 +1,8 @@ - Brief Notes on C-Media 8338/8738/8768/8770 Driver - ================================================= +================================================= +Brief Notes on C-Media 8338/8738/8768/8770 Driver +=================================================

- Takashi Iwai tiwai@suse.de +Takashi Iwai tiwai@suse.de

Front/Rear Multi-channel Playback @@ -30,19 +31,20 @@ The rear output can be heard only when "Four Channel Mode" switch is disabled. Otherwise no signal will be routed to the rear speakers. As default it's turned on.

-*** WARNING *** -When "Four Channel Mode" switch is off, the output from rear speakers -will be FULL VOLUME regardless of Master and PCM volumes. -This might damage your audio equipment. Please disconnect speakers -before your turn off this switch. -*** WARNING *** +.. WARNING:: + When "Four Channel Mode" switch is off, the output from rear speakers + will be FULL VOLUME regardless of Master and PCM volumes [#]_. + This might damage your audio equipment. Please disconnect speakers + before your turn off this switch.

-[ Well.. I once got the output with correct volume (i.e. same with the + +.. [#] + Well.. I once got the output with correct volume (i.e. same with the front one) and was so excited. It was even with "Four Channel" bit on and "double DAC" mode. Actually I could hear separate 4 channels from front and rear speakers! But.. after reboot, all was gone. It's a very pity that I didn't save the register dump at that - time.. Maybe there is an unknown register to achieve this... ] + time.. Maybe there is an unknown register to achieve this...

If your card has an extra output jack for the rear output, the rear playback should be routed there as default. If not, there is a @@ -73,12 +75,14 @@ cannot operate with full-duplex.

The 4.0 and 5.1 modes are defined as the pcm "surround40" and "surround51" in alsa-lib. For example, you can play a WAV file with 6 channels like +::

% aplay -Dsurround51 sixchannels.wav

For programming the 4/6 channel playback, you need to specify the PCM channels as you like and set the format S16LE. For example, for playback with 4 channels, +::

snd_pcm_hw_params_set_access(pcm, hw, SND_PCM_ACCESS_RW_INTERLEAVED); // or mmap if you like @@ -89,13 +93,15 @@ and use the interleaved 4 channel data.

There are some control switches affecting to the speaker connections:

-"Line-In Mode" - an enum control to change the behavior of line-in +Line-In Mode + an enum control to change the behavior of line-in jack. Either "Line-In", "Rear Output" or "Bass Output" can be selected. The last item is available only with model 039 or newer. When "Rear Output" is chosen, the surround channels 3 and 4 are output to line-in jack. -"Mic-In Mode" - an enum control to change the behavior of mic-in +Mic-In Mode + an enum control to change the behavior of mic-in jack. Either "Mic-In" or "Center/LFE Output" can be selected. When "Center/LFE Output" is chosen, the center and bass @@ -111,11 +117,14 @@ The SPDIF playback and capture are done via the third PCM device (hw:0,2). Usually this is assigned to the PCM device "spdif". The available rates are 44100 and 48000 Hz. For playback with aplay, you can run like below: +::

% aplay -Dhw:0,2 foo.wav

or

+:: + % aplay -Dspdif foo.wav

24bit format is also supported experimentally. @@ -140,31 +149,40 @@ off. (Also don't forget to turn on "IEC958 Output Switch", too.)

Additionally there are relevant control switches:

-"IEC958 Mix Analog" - Mix analog PCM playback and FM-OPL/3 streams and +IEC958 Mix Analog + Mix analog PCM playback and FM-OPL/3 streams and output through SPDIF. This switch appears only on old chip models (CM8738 033 and 037). + Note: without this control you can output PCM to SPDIF. This is "mixing" of streams, so e.g. it's not for AC3 output (see the next section).

-"IEC958 In Select" - Select SPDIF input, the internal CD-in (false) +IEC958 In Select + Select SPDIF input, the internal CD-in (false) and the external input (true).

-"IEC958 Loop" - SPDIF input data is loop back into SPDIF +IEC958 Loop + SPDIF input data is loop back into SPDIF output (aka bypass)

-"IEC958 Copyright" - Set the copyright bit. +IEC958 Copyright + Set the copyright bit.

-"IEC958 5V" - Select 0.5V (coax) or 5V (optical) interface. +IEC958 5V + Select 0.5V (coax) or 5V (optical) interface. On some cards this doesn't work and you need to change the configuration with hardware dip-switch.

-"IEC958 In Monitor" - SPDIF input is routed to DAC. +IEC958 In Monitor + SPDIF input is routed to DAC.

-"IEC958 In Phase Inverse" - Set SPDIF input format as inverse. +IEC958 In Phase Inverse + Set SPDIF input format as inverse. [FIXME: this doesn't work on all chips..]

-"IEC958 In Valid" - Set input validity flag detection. +IEC958 In Valid + Set input validity flag detection.

Note: When "PCM Playback Switch" is on, you'll hear the digital output stream through analog line-out. @@ -217,7 +235,7 @@ to enable MIDI support. Valid I/O ports are 0x300, 0x310, 0x320 and With CMI8738 and newer chips, the MIDI interface is enabled by default and the driver automatically chooses a port address.

-There is _no_ hardware wavetable function on this chip (except for +There is *no* hardware wavetable function on this chip (except for OPL3 synth below). What's said as MIDI synth on Windows is a software synthesizer emulation. On Linux use TiMidity or other softsynth program for diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst index e1f4b78c75d6..67a3073157b9 100644 --- a/Documentation/sound/cards/index.rst +++ b/Documentation/sound/cards/index.rst @@ -5,3 +5,4 @@ Card-Specific Information :maxdepth: 2

joystick + cmipci

Another simple conversion from a plain text file. Put to cards subdirectory.

Signed-off-by: Takashi Iwai tiwai@suse.de --- .../Audigy-mixer.txt => cards/audigy-mixer.rst} | 297 +++++++++++---------- Documentation/sound/cards/index.rst | 2 + 2 files changed, 162 insertions(+), 137 deletions(-) rename Documentation/sound/{alsa/Audigy-mixer.txt => cards/audigy-mixer.rst} (57%)

diff --git a/Documentation/sound/alsa/Audigy-mixer.txt b/Documentation/sound/cards/audigy-mixer.rst similarity index 57% rename from Documentation/sound/alsa/Audigy-mixer.txt rename to Documentation/sound/cards/audigy-mixer.rst index 7f10dc6ff28c..86213234435f 100644 --- a/Documentation/sound/alsa/Audigy-mixer.txt +++ b/Documentation/sound/cards/audigy-mixer.rst @@ -1,8 +1,8 @@ +============================================= +Sound Blaster Audigy mixer / default DSP code +=============================================

- Sound Blaster Audigy mixer / default DSP code - =========================================== - -This is based on SB-Live-mixer.txt. +This is based on sb-live-mixer.rst.

The EMU10K2 chips have a DSP part which can be programmed to support various ways of sample processing, which is described here. @@ -13,8 +13,8 @@ The ALSA driver programs this portion of chip by default code (can be altered later) which offers the following functionality:

-1) Digital mixer controls -------------------------- +Digital mixer controls +======================

These controls are built using the DSP instructions. They offer extended functionality. Only the default build-in code in the ALSA driver is described @@ -26,320 +26,343 @@ is mentioned in multiple controls, the signal is accumulated and can be wrapped

Explanation of used abbreviations:

-DAC - digital to analog converter -ADC - analog to digital converter -I2S - one-way three wire serial bus for digital sound by Philips Semiconductors - (this standard is used for connecting standalone DAC and ADC converters) -LFE - low frequency effects (subwoofer signal) -AC97 - a chip containing an analog mixer, DAC and ADC converters -IEC958 - S/PDIF -FX-bus - the EMU10K2 chip has an effect bus containing 64 accumulators. - Each of the synthesizer voices can feed its output to these accumulators - and the DSP microcontroller can operate with the resulting sum. +DAC + digital to analog converter +ADC + analog to digital converter +I2S + one-way three wire serial bus for digital sound by Philips Semiconductors + (this standard is used for connecting standalone DAC and ADC converters) +LFE + low frequency effects (subwoofer signal) +AC97 + a chip containing an analog mixer, DAC and ADC converters +IEC958 + S/PDIF +FX-bus + the EMU10K2 chip has an effect bus containing 64 accumulators. + Each of the synthesizer voices can feed its output to these accumulators + and the DSP microcontroller can operate with the resulting sum.

name='PCM Front Playback Volume',index=0 - +---------------------------------------- This control is used to attenuate samples for left and right front PCM FX-bus accumulators. ALSA uses accumulators 8 and 9 for left and right front PCM samples for 5.1 playback. The result samples are forwarded to the front DAC PCM slots of the Philips DAC.

name='PCM Surround Playback Volume',index=0 - +------------------------------------------- This control is used to attenuate samples for left and right surround PCM FX-bus accumulators. ALSA uses accumulators 2 and 3 for left and right surround PCM samples for 5.1 playback. The result samples are forwarded to the surround DAC PCM slots of the Philips DAC.

name='PCM Center Playback Volume',index=0 - +----------------------------------------- This control is used to attenuate samples for center PCM FX-bus accumulator. ALSA uses accumulator 6 for center PCM sample for 5.1 playback. The result sample is forwarded to the center DAC PCM slot of the Philips DAC.

name='PCM LFE Playback Volume',index=0 - +-------------------------------------- This control is used to attenuate sample for LFE PCM FX-bus accumulator. ALSA uses accumulator 7 for LFE PCM sample for 5.1 playback. The result sample is forwarded to the LFE DAC PCM slot of the Philips DAC.

name='PCM Playback Volume',index=0 - +---------------------------------- This control is used to attenuate samples for left and right PCM FX-bus accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples for stereo playback. The result samples are forwarded to the front DAC PCM slots of the Philips DAC.

name='PCM Capture Volume',index=0 - +--------------------------------- This control is used to attenuate samples for left and right PCM FX-bus accumulator. ALSA uses accumulators 0 and 1 for left and right PCM. The result is forwarded to the ADC capture FIFO (thus to the standard capture PCM device).

name='Music Playback Volume',index=0 - +------------------------------------ This control is used to attenuate samples for left and right MIDI FX-bus accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples. The result samples are forwarded to the front DAC PCM slots of the AC97 codec.

name='Music Capture Volume',index=0 - +----------------------------------- These controls are used to attenuate samples for left and right MIDI FX-bus accumulator. ALSA uses accumulators 4 and 5 for left and right PCM. The result is forwarded to the ADC capture FIFO (thus to the standard capture PCM device).

name='Mic Playback Volume',index=0 - +---------------------------------- This control is used to attenuate samples for left and right Mic input. For Mic input is used AC97 codec. The result samples are forwarded to the front DAC PCM slots of the Philips DAC. Samples are forwarded to Mic capture FIFO (device 1 - 16bit/8KHz mono) too without volume control.

name='Mic Capture Volume',index=0 - +--------------------------------- This control is used to attenuate samples for left and right Mic input. The result is forwarded to the ADC capture FIFO (thus to the standard capture PCM device).

name='Audigy CD Playback Volume',index=0 - +---------------------------------------- This control is used to attenuate samples from left and right IEC958 TTL digital inputs (usually used by a CDROM drive). The result samples are forwarded to the front DAC PCM slots of the Philips DAC.

name='Audigy CD Capture Volume',index=0 - +--------------------------------------- This control is used to attenuate samples from left and right IEC958 TTL digital inputs (usually used by a CDROM drive). The result samples are forwarded to the ADC capture FIFO (thus to the standard capture PCM device).

name='IEC958 Optical Playback Volume',index=0 - +--------------------------------------------- This control is used to attenuate samples from left and right IEC958 optical digital input. The result samples are forwarded to the front DAC PCM slots of the Philips DAC.

name='IEC958 Optical Capture Volume',index=0 - +-------------------------------------------- This control is used to attenuate samples from left and right IEC958 optical digital inputs. The result samples are forwarded to the ADC capture FIFO (thus to the standard capture PCM device).

name='Line2 Playback Volume',index=0 - +------------------------------------ This control is used to attenuate samples from left and right I2S ADC inputs (on the AudigyDrive). The result samples are forwarded to the front DAC PCM slots of the Philips DAC.

name='Line2 Capture Volume',index=1 - +----------------------------------- This control is used to attenuate samples from left and right I2S ADC inputs (on the AudigyDrive). The result samples are forwarded to the ADC capture FIFO (thus to the standard capture PCM device).

name='Analog Mix Playback Volume',index=0 - +----------------------------------------- This control is used to attenuate samples from left and right I2S ADC inputs from Philips ADC. The result samples are forwarded to the front DAC PCM slots of the Philips DAC. This contains mix from analog sources like CD, Line In, Aux, ....

name='Analog Mix Capture Volume',index=1 - +---------------------------------------- This control is used to attenuate samples from left and right I2S ADC inputs Philips ADC. The result samples are forwarded to the ADC capture FIFO (thus to the standard capture PCM device).

name='Aux2 Playback Volume',index=0 - +----------------------------------- This control is used to attenuate samples from left and right I2S ADC inputs (on the AudigyDrive). The result samples are forwarded to the front DAC PCM slots of the Philips DAC.

name='Aux2 Capture Volume',index=1 - +---------------------------------- This control is used to attenuate samples from left and right I2S ADC inputs (on the AudigyDrive). The result samples are forwarded to the ADC capture FIFO (thus to the standard capture PCM device).

name='Front Playback Volume',index=0 - +------------------------------------ All stereo signals are mixed together and mirrored to surround, center and LFE. This control is used to attenuate samples for left and right front speakers of this mix.

name='Surround Playback Volume',index=0 - +--------------------------------------- All stereo signals are mixed together and mirrored to surround, center and LFE. This control is used to attenuate samples for left and right surround speakers of this mix.

name='Center Playback Volume',index=0 - +------------------------------------- All stereo signals are mixed together and mirrored to surround, center and LFE. This control is used to attenuate sample for center speaker of this mix.

name='LFE Playback Volume',index=0 - +---------------------------------- All stereo signals are mixed together and mirrored to surround, center and LFE. This control is used to attenuate sample for LFE speaker of this mix.

name='Tone Control - Switch',index=0 - +------------------------------------ This control turns the tone control on or off. The samples for front, rear and center / LFE outputs are affected.

name='Tone Control - Bass',index=0 - +---------------------------------- This control sets the bass intensity. There is no neutral value!! When the tone control code is activated, the samples are always modified. The closest value to pure signal is 20.

name='Tone Control - Treble',index=0 - +------------------------------------ This control sets the treble intensity. There is no neutral value!! When the tone control code is activated, the samples are always modified. The closest value to pure signal is 20.

name='Master Playback Volume',index=0 - +------------------------------------- This control is used to attenuate samples for front, surround, center and LFE outputs.

name='IEC958 Optical Raw Playback Switch',index=0 - +------------------------------------------------- If this switch is on, then the samples for the IEC958 (S/PDIF) digital output are taken only from the raw FX8010 PCM, otherwise standard front PCM samples are taken.

-2) PCM stream related controls ------------------------------- +PCM stream related controls +===========================

name='EMU10K1 PCM Volume',index 0-31 - +------------------------------------ Channel volume attenuation in range 0-0xffff. The maximum value (no attenuation) is default. The channel mapping for three values is as follows:

- 0 - mono, default 0xffff (no attenuation) - 1 - left, default 0xffff (no attenuation) - 2 - right, default 0xffff (no attenuation) +* 0 - mono, default 0xffff (no attenuation) +* 1 - left, default 0xffff (no attenuation) +* 2 - right, default 0xffff (no attenuation)

name='EMU10K1 PCM Send Routing',index 0-31 - +------------------------------------------ This control specifies the destination - FX-bus accumulators. There 24 values with this mapping:

- 0 - mono, A destination (FX-bus 0-63), default 0 - 1 - mono, B destination (FX-bus 0-63), default 1 - 2 - mono, C destination (FX-bus 0-63), default 2 - 3 - mono, D destination (FX-bus 0-63), default 3 - 4 - mono, E destination (FX-bus 0-63), default 0 - 5 - mono, F destination (FX-bus 0-63), default 0 - 6 - mono, G destination (FX-bus 0-63), default 0 - 7 - mono, H destination (FX-bus 0-63), default 0 - 8 - left, A destination (FX-bus 0-63), default 0 - 9 - left, B destination (FX-bus 0-63), default 1 - 10 - left, C destination (FX-bus 0-63), default 2 - 11 - left, D destination (FX-bus 0-63), default 3 - 12 - left, E destination (FX-bus 0-63), default 0 - 13 - left, F destination (FX-bus 0-63), default 0 - 14 - left, G destination (FX-bus 0-63), default 0 - 15 - left, H destination (FX-bus 0-63), default 0 - 16 - right, A destination (FX-bus 0-63), default 0 - 17 - right, B destination (FX-bus 0-63), default 1 - 18 - right, C destination (FX-bus 0-63), default 2 - 19 - right, D destination (FX-bus 0-63), default 3 - 20 - right, E destination (FX-bus 0-63), default 0 - 21 - right, F destination (FX-bus 0-63), default 0 - 22 - right, G destination (FX-bus 0-63), default 0 - 23 - right, H destination (FX-bus 0-63), default 0 +* 0 - mono, A destination (FX-bus 0-63), default 0 +* 1 - mono, B destination (FX-bus 0-63), default 1 +* 2 - mono, C destination (FX-bus 0-63), default 2 +* 3 - mono, D destination (FX-bus 0-63), default 3 +* 4 - mono, E destination (FX-bus 0-63), default 0 +* 5 - mono, F destination (FX-bus 0-63), default 0 +* 6 - mono, G destination (FX-bus 0-63), default 0 +* 7 - mono, H destination (FX-bus 0-63), default 0 +* 8 - left, A destination (FX-bus 0-63), default 0 +* 9 - left, B destination (FX-bus 0-63), default 1 +* 10 - left, C destination (FX-bus 0-63), default 2 +* 11 - left, D destination (FX-bus 0-63), default 3 +* 12 - left, E destination (FX-bus 0-63), default 0 +* 13 - left, F destination (FX-bus 0-63), default 0 +* 14 - left, G destination (FX-bus 0-63), default 0 +* 15 - left, H destination (FX-bus 0-63), default 0 +* 16 - right, A destination (FX-bus 0-63), default 0 +* 17 - right, B destination (FX-bus 0-63), default 1 +* 18 - right, C destination (FX-bus 0-63), default 2 +* 19 - right, D destination (FX-bus 0-63), default 3 +* 20 - right, E destination (FX-bus 0-63), default 0 +* 21 - right, F destination (FX-bus 0-63), default 0 +* 22 - right, G destination (FX-bus 0-63), default 0 +* 23 - right, H destination (FX-bus 0-63), default 0

Don't forget that it's illegal to assign a channel to the same FX-bus accumulator more than once (it means 0=0 && 1=0 is an invalid combination).

name='EMU10K1 PCM Send Volume',index 0-31 - +----------------------------------------- It specifies the attenuation (amount) for given destination in range 0-255. The channel mapping is following:

- 0 - mono, A destination attn, default 255 (no attenuation) - 1 - mono, B destination attn, default 255 (no attenuation) - 2 - mono, C destination attn, default 0 (mute) - 3 - mono, D destination attn, default 0 (mute) - 4 - mono, E destination attn, default 0 (mute) - 5 - mono, F destination attn, default 0 (mute) - 6 - mono, G destination attn, default 0 (mute) - 7 - mono, H destination attn, default 0 (mute) - 8 - left, A destination attn, default 255 (no attenuation) - 9 - left, B destination attn, default 0 (mute) - 10 - left, C destination attn, default 0 (mute) - 11 - left, D destination attn, default 0 (mute) - 12 - left, E destination attn, default 0 (mute) - 13 - left, F destination attn, default 0 (mute) - 14 - left, G destination attn, default 0 (mute) - 15 - left, H destination attn, default 0 (mute) - 16 - right, A destination attn, default 0 (mute) - 17 - right, B destination attn, default 255 (no attenuation) - 18 - right, C destination attn, default 0 (mute) - 19 - right, D destination attn, default 0 (mute) - 20 - right, E destination attn, default 0 (mute) - 21 - right, F destination attn, default 0 (mute) - 22 - right, G destination attn, default 0 (mute) - 23 - right, H destination attn, default 0 (mute) - - - -4) MANUALS/PATENTS: -------------------- +* 0 - mono, A destination attn, default 255 (no attenuation) +* 1 - mono, B destination attn, default 255 (no attenuation) +* 2 - mono, C destination attn, default 0 (mute) +* 3 - mono, D destination attn, default 0 (mute) +* 4 - mono, E destination attn, default 0 (mute) +* 5 - mono, F destination attn, default 0 (mute) +* 6 - mono, G destination attn, default 0 (mute) +* 7 - mono, H destination attn, default 0 (mute) +* 8 - left, A destination attn, default 255 (no attenuation) +* 9 - left, B destination attn, default 0 (mute) +* 10 - left, C destination attn, default 0 (mute) +* 11 - left, D destination attn, default 0 (mute) +* 12 - left, E destination attn, default 0 (mute) +* 13 - left, F destination attn, default 0 (mute) +* 14 - left, G destination attn, default 0 (mute) +* 15 - left, H destination attn, default 0 (mute) +* 16 - right, A destination attn, default 0 (mute) +* 17 - right, B destination attn, default 255 (no attenuation) +* 18 - right, C destination attn, default 0 (mute) +* 19 - right, D destination attn, default 0 (mute) +* 20 - right, E destination attn, default 0 (mute) +* 21 - right, F destination attn, default 0 (mute) +* 22 - right, G destination attn, default 0 (mute) +* 23 - right, H destination attn, default 0 (mute) + + + +MANUALS/PATENTS +===============

ftp://opensource.creative.com/pub/doc -------------------------------------

- Files: - LM4545.pdf AC97 Codec +LM4545.pdf + AC97 Codec

- m2049.pdf The EMU10K1 Digital Audio Processor +m2049.pdf + The EMU10K1 Digital Audio Processor

- hog63.ps FX8010 - A DSP Chip Architecture for Audio Effects +hog63.ps + FX8010 - A DSP Chip Architecture for Audio Effects

WIPO Patents ------------ - Patent numbers: - WO 9901813 (A1) Audio Effects Processor with multiple asynchronous (Jan. 14, 1999) - streams

- WO 9901814 (A1) Processor with Instruction Set for Audio Effects (Jan. 14, 1999) +WO 9901813 (A1) + Audio Effects Processor with multiple asynchronous streams + (Jan. 14, 1999) + +WO 9901814 (A1) + Processor with Instruction Set for Audio Effects (Jan. 14, 1999)

- WO 9901953 (A1) Audio Effects Processor having Decoupled Instruction - Execution and Audio Data Sequencing (Jan. 14, 1999) +WO 9901953 (A1) + Audio Effects Processor having Decoupled Instruction + Execution and Audio Data Sequencing (Jan. 14, 1999)

US Patents (http://www.uspto.gov/) ----------------------------------

- US 5925841 Digital Sampling Instrument employing cache memory (Jul. 20, 1999) - - US 5928342 Audio Effects Processor integrated on a single chip (Jul. 27, 1999) - with a multiport memory onto which multiple asynchronous - digital sound samples can be concurrently loaded - - US 5930158 Processor with Instruction Set for Audio Effects (Jul. 27, 1999) - - US 6032235 Memory initialization circuit (Tram) (Feb. 29, 2000) - - US 6138207 Interpolation looping of audio samples in cache connected to (Oct. 24, 2000) - system bus with prioritization and modification of bus transfers - in accordance with loop ends and minimum block sizes - - US 6151670 Method for conserving memory storage using a (Nov. 21, 2000) - pool of short term memory registers - - US 6195715 Interrupt control for multiple programs communicating with (Feb. 27, 2001) - a common interrupt by associating programs to GP registers, - defining interrupt register, polling GP registers, and invoking - callback routine associated with defined interrupt register +US 5925841 + Digital Sampling Instrument employing cache memory (Jul. 20, 1999) + +US 5928342 + Audio Effects Processor integrated on a single chip + with a multiport memory onto which multiple asynchronous + digital sound samples can be concurrently loaded + (Jul. 27, 1999) + +US 5930158 + Processor with Instruction Set for Audio Effects (Jul. 27, 1999) + +US 6032235 + Memory initialization circuit (Tram) (Feb. 29, 2000) + +US 6138207 + Interpolation looping of audio samples in cache connected to + system bus with prioritization and modification of bus transfers + in accordance with loop ends and minimum block sizes + (Oct. 24, 2000) + +US 6151670 + Method for conserving memory storage using a + pool of short term memory registers + (Nov. 21, 2000) + +US 6195715 + Interrupt control for multiple programs communicating with + a common interrupt by associating programs to GP registers, + defining interrupt register, polling GP registers, and invoking + callback routine associated with defined interrupt register + (Feb. 27, 2001) diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst index 294efdb75bc9..c9d7ce4286b2 100644 --- a/Documentation/sound/cards/index.rst +++ b/Documentation/sound/cards/index.rst @@ -7,3 +7,5 @@ Card-Specific Information joystick cmipci sb-live-mixer + audigy-mixer +

Another simple conversion from a plain text file. Put to cards directory.

Signed-off-by: Takashi Iwai tiwai@suse.de --- Documentation/sound/alsa/VIA82xx-mixer.txt | 8 -------- Documentation/sound/cards/index.rst | 1 + Documentation/sound/cards/via82xx-mixer.rst | 8 ++++++++ 3 files changed, 9 insertions(+), 8 deletions(-) delete mode 100644 Documentation/sound/alsa/VIA82xx-mixer.txt create mode 100644 Documentation/sound/cards/via82xx-mixer.rst

diff --git a/Documentation/sound/alsa/VIA82xx-mixer.txt b/Documentation/sound/alsa/VIA82xx-mixer.txt deleted file mode 100644 index 1b0ac06ba95d..000000000000 --- a/Documentation/sound/alsa/VIA82xx-mixer.txt +++ /dev/null @@ -1,8 +0,0 @@ - - VIA82xx mixer - ============= - -On many VIA82xx boards, the 'Input Source Select' mixer control does not work. -Setting it to 'Input2' on such boards will cause recording to hang, or fail -with EIO (input/output error) via OSS emulation. This control should be left -at 'Input1' for such cards. diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst index 251f1d7675f7..4fcb88049ccc 100644 --- a/Documentation/sound/cards/index.rst +++ b/Documentation/sound/cards/index.rst @@ -9,3 +9,4 @@ Card-Specific Information sb-live-mixer audigy-mixer emu10k1-jack + via82xx-mixer diff --git a/Documentation/sound/cards/via82xx-mixer.rst b/Documentation/sound/cards/via82xx-mixer.rst new file mode 100644 index 000000000000..6ee993d4535b --- /dev/null +++ b/Documentation/sound/cards/via82xx-mixer.rst @@ -0,0 +1,8 @@ +============= +VIA82xx mixer +============= + +On many VIA82xx boards, the ``Input Source Select`` mixer control does not work. +Setting it to ``Input2`` on such boards will cause recording to hang, or fail +with EIO (input/output error) via OSS emulation. This control should be left +at ``Input1`` for such cards.

Another simple conversion from a plain text file. Put to cards directory.

Signed-off-by: Takashi Iwai tiwai@suse.de --- Documentation/sound/cards/index.rst | 1 + .../sound/{alsa/MIXART.txt => cards/mixart.rst} | 26 +++++++++++++++------- 2 files changed, 19 insertions(+), 8 deletions(-) rename Documentation/sound/{alsa/MIXART.txt => cards/mixart.rst} (83%)

diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst index e92a4f7e86fc..0006bc547bb8 100644 --- a/Documentation/sound/cards/index.rst +++ b/Documentation/sound/cards/index.rst @@ -11,3 +11,4 @@ Card-Specific Information emu10k1-jack via82xx-mixer audiophile-usb + mixart diff --git a/Documentation/sound/alsa/MIXART.txt b/Documentation/sound/cards/mixart.rst similarity index 83% rename from Documentation/sound/alsa/MIXART.txt rename to Documentation/sound/cards/mixart.rst index 4ee35b4fbe4a..48aba98b088f 100644 --- a/Documentation/sound/alsa/MIXART.txt +++ b/Documentation/sound/cards/mixart.rst @@ -1,5 +1,8 @@ - Alsa driver for Digigram miXart8 and miXart8AES/EBU soundcards - Digigram alsa@digigram.com +============================================================== +Alsa driver for Digigram miXart8 and miXart8AES/EBU soundcards +============================================================== + +Digigram alsa@digigram.com

GENERAL @@ -48,11 +51,15 @@ formats are supported.

Mixer ----- -<Master> and <Master Capture> : analog volume control of playback and capture PCM. -<PCM 0-3> and <PCM Capture> : digital volume control of each analog substream. -<AES 0-3> and <AES Capture> : digital volume control of each AES/EBU substream. -<Monitoring> : Loopback from 'pcm0c' to 'pcm0p' with digital volume -and mute control. +<Master> and <Master Capture> + analog volume control of playback and capture PCM. +<PCM 0-3> and <PCM Capture> + digital volume control of each analog substream. +<AES 0-3> and <AES Capture> + digital volume control of each AES/EBU substream. +<Monitoring> + Loopback from 'pcm0c' to 'pcm0p' with digital volume + and mute control.

Rem : for best audio quality try to keep a 0 attenuation on the PCM and AES volume controls which is set by 219 in the range from 0 to 255 @@ -79,11 +86,14 @@ FIRMWARE For loading the firmware automatically after the module is loaded, use a install command. For example, add the following entry to /etc/modprobe.d/mixart.conf for miXart driver: +::

install snd-mixart /sbin/modprobe --first-time -i snd-mixart && \ /usr/bin/mixartloader + + (for 2.2/2.4 kernels, add "post-install snd-mixart /usr/bin/vxloader" to - /etc/modules.conf, instead.) +/etc/modules.conf, instead.)

The firmware binaries are installed on /usr/share/alsa/firmware (or /usr/local/share/alsa/firmware, depending to the prefix option of

Another simple conversion from a plain text file. Put to cards directory.

Signed-off-by: Takashi Iwai tiwai@suse.de --- Documentation/sound/cards/index.rst | 1 + .../sound/{alsa/README.maya44 => cards/maya44.rst} | 137 ++++++++++++--------- 2 files changed, 81 insertions(+), 57 deletions(-) rename Documentation/sound/{alsa/README.maya44 => cards/maya44.rst} (65%)

diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst index 157372faae06..6c8e4141df9a 100644 --- a/Documentation/sound/cards/index.rst +++ b/Documentation/sound/cards/index.rst @@ -13,3 +13,4 @@ Card-Specific Information audiophile-usb mixart bt87x + maya44 diff --git a/Documentation/sound/alsa/README.maya44 b/Documentation/sound/cards/maya44.rst similarity index 65% rename from Documentation/sound/alsa/README.maya44 rename to Documentation/sound/cards/maya44.rst index 67b2ea1cc31d..bf09a584b443 100644 --- a/Documentation/sound/alsa/README.maya44 +++ b/Documentation/sound/cards/maya44.rst @@ -1,10 +1,18 @@ -NOTE: The following is the original document of Rainer's patch that the -current maya44 code based on. Some contents might be obsoleted, but I -keep here as reference -- tiwai +================================= +Notes on Maya44 USB Audio Support +=================================

----------------------------------------------------------------- +.. note:: + The following is the original document of Rainer's patch that the + current maya44 code based on. Some contents might be obsoleted, but I + keep here as reference -- tiwai + +Feb 14, 2008 + +Rainer Zimmermann mail@lightshed.de

-STATE OF DEVELOPMENT: +STATE OF DEVELOPMENT +====================

This driver is being developed on the initiative of Piotr Makowski (oponek@gmail.com) and financed by Lars Bergmann. Development is carried out by Rainer Zimmermann (mail@lightshed.de). @@ -44,16 +52,17 @@ Things that do not seem to work: - Ardour 2.1 seems to work only via JACK, not using ALSA directly or via OSS. This still needs to be tracked down.

-DRIVER DETAILS: +DRIVER DETAILS +==============

the following files were added:

-pci/ice1724/maya44.c - Maya44 specific code -pci/ice1724/maya44.h -pci/ice1724/ice1724.patch -pci/ice1724/ice1724.h.patch - PROPOSED patch to ice1724.h (see SAMPLING RATES) -i2c/other/wm8776.c - low-level access routines for Wolfson WM8776 codecs -include/wm8776.h +* pci/ice1724/maya44.c - Maya44 specific code +* pci/ice1724/maya44.h +* pci/ice1724/ice1724.patch +* pci/ice1724/ice1724.h.patch - PROPOSED patch to ice1724.h (see SAMPLING RATES) +* i2c/other/wm8776.c - low-level access routines for Wolfson WM8776 codecs +* include/wm8776.h

Note that the wm8776.c code is meant to be card-independent and does not actually register the codec with the ALSA infrastructure. @@ -62,25 +71,26 @@ This is done in maya44.c, mainly because some of the WM8776 controls are used in

the following files were created in pci/ice1724, simply #including the corresponding file from the alsa-kernel tree:

-wtm.h -vt1720_mobo.h -revo.h -prodigy192.h -pontis.h -phase.h -maya44.h -juli.h -aureon.h -amp.h -envy24ht.h -se.h -prodigy_hifi.h +* wtm.h +* vt1720_mobo.h +* revo.h +* prodigy192.h +* pontis.h +* phase.h +* maya44.h +* juli.h +* aureon.h +* amp.h +* envy24ht.h +* se.h +* prodigy_hifi.h

*I hope this is the correct way to do things.*

-SAMPLING RATES: +SAMPLING RATES +==============

The Maya44 card (or more exactly, the Wolfson WM8776 codecs) allow a maximum sampling rate of 192 kHz for playback and 92 kHz for capture.

@@ -98,66 +108,79 @@ I propose some additional code for limiting the sampling rate when setting on a The proposed code (currently deactivated) is in ice1712.h.patch, ice1724.c and maya44.c (in pci/ice1712).

-SOUND DEVICES: +SOUND DEVICES +=============

PCM devices correspond to inputs/outputs as follows (assuming Maya44 is card #0):

-hw:0,0 input - stereo, analog input 1+2 -hw:0,0 output - stereo, analog output 1+2 -hw:0,1 input - stereo, analog input 3+4 OR S/PDIF input -hw:0,1 output - stereo, analog output 3+4 (and SPDIF out) +* hw:0,0 input - stereo, analog input 1+2 +* hw:0,0 output - stereo, analog output 1+2 +* hw:0,1 input - stereo, analog input 3+4 OR S/PDIF input +* hw:0,1 output - stereo, analog output 3+4 (and SPDIF out)

-NAMING OF MIXER CONTROLS: +NAMING OF MIXER CONTROLS +========================

(for more information about the signal flow, please refer to the block diagram on p.24 of the ESI Maya44 manual, or in the ESI windows software).

-PCM: (digital) output level for channel 1+2 -PCM 1: same for channel 3+4 +PCM + (digital) output level for channel 1+2 +PCM 1 + same for channel 3+4 + +Mic Phantom+48V + switch for +48V phantom power for electrostatic microphones on input 1/2.

-Mic Phantom+48V: switch for +48V phantom power for electrostatic microphones on input 1/2. Make sure this is not turned on while any other source is connected to input 1/2. It might damage the source and/or the maya44 card.

-Mic/Line input: if switch is on, input jack 1/2 is microphone input (mono), otherwise line input (stereo). +Mic/Line input + if switch is on, input jack 1/2 is microphone input (mono), otherwise line input (stereo). + +Bypass + analogue bypass from ADC input to output for channel 1+2. Same as "Monitor" in the windows driver. +Bypass 1 + same for channel 3+4.

-Bypass: analogue bypass from ADC input to output for channel 1+2. Same as "Monitor" in the windows driver. -Bypass 1: same for channel 3+4. +Crossmix + cross-mixer from channels 1+2 to channels 3+4 +Crossmix 1 + cross-mixer from channels 3+4 to channels 1+2

-Crossmix: cross-mixer from channels 1+2 to channels 3+4 -Crossmix 1: cross-mixer from channels 3+4 to channels 1+2 +IEC958 Output + switch for S/PDIF output.

-IEC958 Output: switch for S/PDIF output. This is not supported by the ESI windows driver. S/PDIF should output the same signal as channel 3+4. [untested!]

-Digitial output selectors: - +Digitial output selectors These switches allow a direct digital routing from the ADCs to the DACs. Each switch determines where the digital input data to one of the DACs comes from. They are not supported by the ESI windows driver. For normal operation, they should all be set to "PCM out".

-H/W: Output source channel 1 -H/W 1: Output source channel 2 -H/W 2: Output source channel 3 -H/W 3: Output source channel 4 +H/W + Output source channel 1 +H/W 1 + Output source channel 2 +H/W 2 + Output source channel 3 +H/W 3 + Output source channel 4 + +H/W 4 ... H/W 9 + unknown function, left in to enable testing.

-H/W 4 ... H/W 9: unknown function, left in to enable testing. Possibly some of these control S/PDIF output(s). If these turn out to be unused, they will go away in later driver versions.

Selectable values for each of the digital output selectors are: - "PCM out" -> DAC output of the corresponding channel (default setting) - "Input 1"... - "Input 4" -> direct routing from ADC output of the selected input channel -

--------- - -Feb 14, 2008 -Rainer Zimmermann -mail@lightshed.de +PCM out + DAC output of the corresponding channel (default setting) +Input 1 ... Input 4 + direct routing from ADC output of the selected input channel

Yet another simple conversion from a plain text file. Put to cards directory.

Signed-off-by: Takashi Iwai tiwai@suse.de --- Documentation/sound/cards/index.rst | 1 + .../serial-u16550.txt => cards/serial-u16550.rst} | 21 +++++++++++++-------- 2 files changed, 14 insertions(+), 8 deletions(-) rename Documentation/sound/{alsa/serial-u16550.txt => cards/serial-u16550.rst} (92%)

diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst index b143eff93c87..976ef5eb8f89 100644 --- a/Documentation/sound/cards/index.rst +++ b/Documentation/sound/cards/index.rst @@ -15,3 +15,4 @@ Card-Specific Information bt87x maya44 hdspm + serial-u16550 diff --git a/Documentation/sound/alsa/serial-u16550.txt b/Documentation/sound/cards/serial-u16550.rst similarity index 92% rename from Documentation/sound/alsa/serial-u16550.txt rename to Documentation/sound/cards/serial-u16550.rst index c1919559d509..197aeacea3da 100644 --- a/Documentation/sound/alsa/serial-u16550.txt +++ b/Documentation/sound/cards/serial-u16550.rst @@ -1,14 +1,14 @@ - - Serial UART 16450/16550 MIDI driver - =================================== +=================================== +Serial UART 16450/16550 MIDI driver +===================================

The adaptor module parameter allows you to select either:

- 0 - Roland Soundcanvas support (default) - 1 - Midiator MS-124T support (1) - 2 - Midiator MS-124W S/A mode (2) - 3 - MS-124W M/B mode support (3) - 4 - Generic device with multiple input support (4) +* 0 - Roland Soundcanvas support (default) +* 1 - Midiator MS-124T support (1) +* 2 - Midiator MS-124W S/A mode (2) +* 3 - MS-124W M/B mode support (3) +* 4 - Generic device with multiple input support (4)

For the Midiator MS-124W, you must set the physical M-S and A-B switches on the Midiator to match the driver mode you select. @@ -22,11 +22,13 @@ substream. The driver provides no way to send F5 00 (no selection) or to not send the F5 NN command sequence at all; perhaps it ought to.

Usage example for simple serial converter: +::

/sbin/setserial /dev/ttyS0 uart none /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 speed=115200

Usage example for Roland SoundCanvas with 4 MIDI ports: +::

/sbin/setserial /dev/ttyS0 uart none /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 outs=4 @@ -37,6 +39,7 @@ all four MIDI Out connectors. Set the A-B switch and the speed module parameter to match (A=19200, B=9600).

Usage example for MS-124T, with A-B switch in A position: +::

/sbin/setserial /dev/ttyS0 uart none /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=1 \ @@ -47,6 +50,7 @@ the outs module parameter is automatically set to 1. The driver sends the same data to all four MIDI Out connectors at full MIDI speed.

Usage example for S/A mode: +::

/sbin/setserial /dev/ttyS0 uart none /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=2 @@ -63,6 +67,7 @@ at most one byte every 520 us, as compared with the full MIDI data rate of one byte every 320 us per port.

Usage example for M/B mode: +::

/sbin/setserial /dev/ttyS0 uart none /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=3