mirror of
https://github.com/alsa-project/alsa-lib.git
synced 2025-12-16 08:56:42 -05:00
Configuration:
- changed snd_config_get_id function to follow semantic of other get functions
- added snd_config_test_id
- added runtime pointer type (not persistent)
- added snd_config_make_pointer, snd_config_set_pointer, snd_config_get_pointer
- added type/contents checking for callback functions
- changed 'void *private_data' to 'snd_config_t *private_data'
- renamed card_strtype functions to card_driver
Control:
- fixed passing parameters to snd_ctl_async
Async handlers:
- added public snd_async_handler_get_signo function
Documentation:
- moved all documentation to source files
This commit is contained in:
parent
ef035eacfe
commit
c39882f602
52 changed files with 1573 additions and 1042 deletions
220
doc/conf.doxygen
220
doc/conf.doxygen
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@ -1,220 +0,0 @@
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/*! \page conf Configuration files
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<P>Configuration files are using a simple format allowing the modern
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data description like nesting and array assignments.</P>
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\section conf_whitespace Whitespace
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Whitespace is the collective name given to spaces (blanks), horizontal and
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vertical tabs, newline characters, and comments. Whitespace can serve to
|
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indicate where configuration tokens start and end, but beyond this function,
|
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any surplus whitespace is discarded. For example, the two sequences
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|
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\code
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a 1 b 2
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\endcode
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|
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and
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|
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\code
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a 1
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b 2
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\endcode
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|
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are lexically equivalent and parse identically to give the four tokens:
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|
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\code
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a
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1
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b
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2
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\endcode
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|
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The ASCII characters representing whitespace can occur within literal
|
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strings, int which case they are protected from the normal parsing process
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(tey remain as part of the string). For example:
|
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|
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\code
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name "John Smith"
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\endcode
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parses to two tokens, including the single literal-string token "John
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Smith".
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\section conf_linesplicing Line splicing with \
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A special case occurs, if the final newline character encountered is
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preceded by a backslash (\) in the string value definition. The backslash
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and new line are both discarded, allowing two physical lines of text to be
|
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treated as one unit.
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|
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\code
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"John \
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Smith"
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\endcode
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is parsed as "John Smith".
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|
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\section conf_comments Comments
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A single-line comments are defined using character #. The comment can start
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in any position, and extends until the next new line.
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\code
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a 1 # this is a comment
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\endcode
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\section conf_include Include another configuration file
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A new configuration file can be included using <filename> syntax. The global
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configuration directory can be referenced using <confdir:filename> syntax.
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\code
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</etc/alsa1.conf>
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<confdir:pcm/surround.conf>
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\endcode
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\section conf_punctuators Punctuators
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The configuration punctuators (also known as separators) are:
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\code
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{} [] , ; = . ' " new-line form-feed carriage-return whitespace
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\endcode
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\subsection conf_braces Braces
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|
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Open and close braces { } indicate the start and end of a compound
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statement:
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\code
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a {
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b 1
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}
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\endcode
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\subsection conf_brackets Brackets
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Open and close brackets indicate single array definition. The identificators
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are automatically generated starting with zero.
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\code
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a [
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"first"
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"second"
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]
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\endcode
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|
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Above code is equal to
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\code
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a.0 "first"
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a.1 "second"
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\endcode
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\subsection conf_comma_semicolon Comma and semicolon
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The comma (,) or semicolon (;) can separate the value assignments. It is not
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strictly required to use these separators, because any whitespace supplies
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them.
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\code
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a 1;
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b 1,
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\endcode
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\subsection conf_equal Equal sign
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The equal sign (=) separates can separate variable declarations from
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initialization lists:
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\code
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a=1
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b=2
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\endcode
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Using the equal signs is not required, because any whitespace supplies
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them.
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\section conf_assigns Assigns
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The configuration file defines id (key) and value pairs. The id (key) can be
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composed from any ASCII digits or chars from a to z or A to Z, including
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char _. The value can be either a string, integer or real number.
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\subsection conf_single Single assign
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\code
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a 1 # is equal to
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a=1 # is equal to
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a=1; # is equal to
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a 1,
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\endcode
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\subsection conf_compound Compound assign (definition using braces)
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\code
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a {
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b = 1
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}
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a={
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b 1,
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}
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\endcode
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\section conf_compound1 Compound assign (one key definition)
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\code
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a.b 1
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a.b=1
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\endcode
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\subsection conf_array Array assign (definition using brackets)
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\code
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a [
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"first"
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"second"
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]
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\endcode
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\subsection conf_array1 Array assign (one key definition)
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\code
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a.0 "first"
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a.1 "second"
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\endcode
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\section conf_summary Summary
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\code
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# Configuration file syntax
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# Include a new configuration file
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<filename>
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# Simple assign
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name [=] value [,|;]
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# Compound assign (first style)
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name [=] {
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name1 [=] value [,|;]
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...
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}
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# Compound assign (second style)
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name.name1 [=] value [,|;]
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# Array assign (first style)
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name [
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value0 [,|;]
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value1 [,|;]
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...
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]
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# Array assign (second style)
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name.0 [=] value0 [,|;]
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name.1 [=] value1 [,|;]
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\endcode
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*/
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@ -1,56 +0,0 @@
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/*! \page confarg Configuration - runtime arguments
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<P>The ALSA library can accept runtime arguments for some configuration
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blocks. This extension is on top of the basic syntax of the configuration
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files.<P>
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\section confarg_define Defining arguments
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Arguments are specified by id (key) @args and array values containing
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the string names of arguments:
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\code
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@args [ CARD ] # or
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@args.0 CARD
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\endcode
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\section confarg_type Defining argument type and default value
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Arguments type is specified by id (key) @args and argument name. The type
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and default value is specified in the compound:
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\code
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@args.CARD {
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type string
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default "abcd"
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}
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\endcode
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\section confarg_refer Refering argument
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Arguments are refered by dollar-sign ($) and name of argument:
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\code
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card $CARD
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\endcode
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\section confarg_example Example
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\code
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pcm.demo {
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@args [ CARD DEVICE ]
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@args.CARD {
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type string
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default "supersonic"
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}
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@args.DEVICE {
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type integer
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default 0
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}
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type hw
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card $CARD
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device $DEVICE
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}
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\endcode
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*/
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@ -1,106 +0,0 @@
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/*! \page conffunc Configuration - runtime functions
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<P>The ALSA library accepts the runtime modification of configuration.
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The several build-in functions are available.</P>
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<P>The function is refered using id @func and function name. All other
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values in the current compound are used as configuration for the function.
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If compound func.<function_name> is defined in the root leafs, then library
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and function from this compound configuration is used, otherwise the prefix
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'snd_func_' is added to string and the code from the ALSA library is used.
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The definition of function looks like:</P>
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\code
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func.remove_first_char {
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lib "/usr/lib/libasoundextend.so"
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func "extend_remove_first_char"
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}
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\endcode
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\section conffunc_getenv The getenv function
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The getenv function allows to get an environment value. The vars values
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(array) defined the order and names for the environment values. When the
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first environment value is found, then the function replaces the whole
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compound by this result. If no value is found, then the default value is
|
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used, if defined.
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\code
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card {
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@func getenv
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vars [ MY_CARD CARD C ]
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default 0
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}
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\endcode
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\section conffunc_igetenv The igetenv function
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This function is same as getenv function, but the result value is converted
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to integer.
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\section conffunc_concat The concat function
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The concat function merges all given string in the array named string into
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one.
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|
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\code
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filename {
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@func concat
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strings [
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"/usr/share"
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"/sound"
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"/a.wav"
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]
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}
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\endcode
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\section conffunc_datadir The datadir function
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This function return the configuration data directory (usually /usr/share/alsa)
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as string as result. This function requires no other values.
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\section conffunc_refer The refer function
|
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|
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This function substitutes the current compound with the compound named (key
|
||||
name, value string) and filename (key file - optional, value string).
|
||||
|
||||
\code
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{
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||||
@func refer
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file /etc/my-alsa.conf
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name pcm.lastone
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}
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\endcode
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\section conffunc_card_strtype The card_strtype function
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|
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This function converts the given card number (key card, value integer) to card type
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(string).
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|
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\section conffunc_card_id The card_id function
|
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|
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This function returns the card id string for the given card number (key card, value
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integer).
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|
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\section conffunc_pcm_id The pcm_id function
|
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|
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This function returns the pcm id string for the given PCM device (key card,
|
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value integer; key device, value integer; key subdevice (optional), value
|
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integer).
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|
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\section conffunc_private_string The private_string function
|
||||
|
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This function returns the private data as string as result.
|
||||
|
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\section conffunc_private_card_strtype The private_card_strtype function
|
||||
|
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This function converts the private data (int) with card number to card type
|
||||
(string).
|
||||
|
||||
\section conffunc_private_pcm_subdevice The private_pcm_subdevice function
|
||||
|
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This functions returns the subdevice number for the pcm handle specified by
|
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the private data.
|
||||
|
||||
*/
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|
|
@ -5,8 +5,7 @@ GENERATE_MAN = NO
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GENERATE_RTF = NO
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|
||||
CASE_SENSE_NAMES = NO
|
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INPUT = index.doxygen conf.doxygen confarg.doxygen \
|
||||
conffunc.doxygen pcm.doxygen \
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INPUT = index.doxygen \
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||||
../include/asoundlib.h \
|
||||
../include/version.h \
|
||||
../include/global.h \
|
||||
|
|
|
|||
441
doc/pcm.doxygen
441
doc/pcm.doxygen
|
|
@ -1,441 +0,0 @@
|
|||
/*! \page pcm PCM (digital audio) interface
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||||
|
||||
<P>Although abbreviation PCM stands for Pulse Code Modulation, we are
|
||||
understanding it as general digital audio processing with volume samples
|
||||
generated in continuous time periods.</P>
|
||||
|
||||
<P>Digital audio is the most commonly used method of representing
|
||||
sound inside a computer. In this method sound is stored as a sequence of
|
||||
samples taken from the audio signal using constant time intervals.
|
||||
A sample represents volume of the signal at the moment when it
|
||||
was measured. In uncompressed digital audio each sample require one
|
||||
or more bytes of storage. The number of bytes required depends on number
|
||||
of channels (mono, stereo) and sample format (8 or 16 bits, mu-Law, etc.).
|
||||
The length of this interval determines the sampling rate. Commonly used
|
||||
sampling rates are between 8kHz (telephone quality) and
|
||||
48kHz (DAT tapes).</P>
|
||||
|
||||
<P>The physical devices used in digital audio are called the
|
||||
ADC (Analog to Digital Converter) and DAC (Digital to Analog Converter).
|
||||
A device containing both ADC and DAC is commonly known as a codec.
|
||||
The codec device used in a Sound Blaster cards is called a DSP which
|
||||
is somewhat misleading since DSP also stands for Digital Signal Processor
|
||||
(the SB DSP chip is very limited when compared to "true" DSP chips).</P>
|
||||
|
||||
<P>Sampling parameters affect the quality of sound which can be
|
||||
reproduced from the recorded signal. The most fundamental parameter
|
||||
is sampling rate which limits the highest frequency that can be stored.
|
||||
It is well known (Nyquist's Sampling Theorem) that the highest frequency
|
||||
that can be stored in a sampled signal is at most 1/2 of the sampling
|
||||
frequency. For example, an 8 kHz sampling rate permits the recording of
|
||||
a signal in which the highest frequency is less than 4 kHz. Higher frequency
|
||||
signals must be filtered out before feeding them to ADC.</P>
|
||||
|
||||
<P>Sample encoding limits the dynamic range of a recorded signal
|
||||
(difference between the faintest and the loudest signal that can be
|
||||
recorded). In theory the maximum dynamic range of signal is number_of_bits *
|
||||
6dB. This means that 8 bits sampling resolution gives dynamic range of
|
||||
48dB and 16 bit resolution gives 96dB.</P>
|
||||
|
||||
<P>Quality has price. The number of bytes required to store an audio
|
||||
sequence depends on sampling rate, number of channels and sampling
|
||||
resolution. For example just 8000 bytes of memory is required to store
|
||||
one second of sound using 8kHz/8 bits/mono but 48kHz/16bit/stereo takes
|
||||
192 kilobytes. A 64 kbps ISDN channel is required to transfer a
|
||||
8kHz/8bit/mono audio stream in real time, and about 1.5Mbps is required
|
||||
for DAT quality (48kHz/16bit/stereo). On the other hand it is possible
|
||||
to store just 5.46 seconds of sound in a megabyte of memory when using
|
||||
48kHz/16bit/stereo sampling. With 8kHz/8bits/mono it is possible to store
|
||||
131 seconds of sound using the same amount of memory. It is possible
|
||||
to reduce memory and communication costs by compressing the recorded
|
||||
signal but this is beyond the scope of this document. </P>
|
||||
|
||||
\section pcm_general_overview General overview
|
||||
|
||||
ALSA uses the ring buffer to store outgoing (playback) and incoming (capture,
|
||||
record) samples. There are two pointers being mantained to allow
|
||||
a precise communication between application and device pointing to current
|
||||
processed sample by hardware and last processed sample by application.
|
||||
The modern audio chips allow to program the transfer time periods.
|
||||
It means that the stream of samples is divided to small chunks. Device
|
||||
acknowledges to application when the transfer of a chunk is complete.
|
||||
|
||||
\section pcm_transfer Transfer methods in unix environments
|
||||
|
||||
In the unix environment, data chunk acknowledges are received via standard I/O
|
||||
calls or event waiting routines (poll or select function). To accomplish
|
||||
this list, the asynchronous notification of acknowledges should be listed
|
||||
here. The ALSA implementation for these methods is described in
|
||||
the \ref alsa_transfers section.
|
||||
|
||||
\subsection pcm_transfer_io Standard I/O transfers
|
||||
|
||||
The standard I/O transfers are using the read (see 'man 2 read') and write
|
||||
(see 'man 2 write') C functions. There are two basic behaviours of these
|
||||
functions - blocked and non-blocked (see the O_NONBLOCK flag for the
|
||||
standard C open function - see 'man 2 open'). In non-blocked behaviour,
|
||||
these I/O functions never stops, they return -EAGAIN error code, when no
|
||||
data can be transferred (the ring buffer is full in our case). In blocked
|
||||
behaviour, these I/O functions stop and wait until there is a room in the
|
||||
ring buffer (playback) or until there are a new samples (capture). The ALSA
|
||||
implementation can be found in the \ref alsa_pcm_rw section.
|
||||
|
||||
\subsection pcm_transfer_event Event waiting routines
|
||||
|
||||
The poll or select functions (see 'man 2 poll' or 'man 2 select' for further
|
||||
details) allows to receive requests/events from the device while
|
||||
an application is waiting on events from other sources (like keyboard, screen,
|
||||
network etc.), too. The select function is old and deprecated in modern
|
||||
applications, so the ALSA library does not support it. The implemented
|
||||
transfer routines can be found in the \ref alsa_transfers section.
|
||||
|
||||
\subsection pcm_transfer_async Asynchronous notification
|
||||
|
||||
ALSA driver and library knows to handle the asynchronous notifications over
|
||||
the SIGIO signal. This signal allows to interrupt application and transfer
|
||||
data in the signal handler. For further details see the sigaction function
|
||||
('man 2 sigaction'). The section \ref pcm_async describes the ALSA API for
|
||||
this extension. The implemented transfer routines can be found in the
|
||||
\ref alsa_transfers section.
|
||||
|
||||
\section pcm_open_behaviour Blocked and non-blocked open
|
||||
|
||||
The ALSA PCM API uses a different behaviour when the device is opened
|
||||
with blocked or non-blocked mode. The mode can be specified with
|
||||
\a mode argument in \link ::snd_pcm_open() \endlink function.
|
||||
The blocked mode is the default (without \link ::SND_PCM_NONBLOCK \endlink mode).
|
||||
In this mode, the behaviour is that if the resources have already used
|
||||
with another application, then it blocks the caller, until resources are
|
||||
free. The non-blocked behaviour (with \link ::SND_PCM_NONBLOCK \endlink)
|
||||
doesn't block the caller in any way and returns -EBUSY error when the
|
||||
resources are not available. Note that the mode also determines the
|
||||
behaviour of standard I/O calls, returning -EAGAIN when non-blocked mode is
|
||||
used and the ring buffer is full (playback) or empty (capture).
|
||||
The operation mode for I/O calls can be changed later with
|
||||
the \link snd_pcm_nonblock() \endlink function.
|
||||
|
||||
\section pcm_async Asynchronous mode
|
||||
|
||||
There is also possibility to receive asynchronous notification after
|
||||
specified time periods. You may see the \link ::SND_PCM_ASYNC \endlink
|
||||
mode for \link ::snd_pcm_open() \endlink function and
|
||||
\link ::snd_async_add_pcm_handler() \endlink function for further details.
|
||||
|
||||
\section pcm_handshake Handshake between application and library
|
||||
|
||||
The ALSA PCM API design uses the states to determine the communication
|
||||
phase between application and library. The actual state can be determined
|
||||
using \link ::snd_pcm_state() \endlink call. There are these states:
|
||||
|
||||
\par SND_PCM_STATE_OPEN
|
||||
The PCM device is in the open state. After the \link ::snd_pcm_open() \endlink open call,
|
||||
the device is in this state. Also, when \link ::snd_pcm_hw_params() \endlink call fails,
|
||||
then this state is entered to force application calling
|
||||
\link ::snd_pcm_hw_params() \endlink function to set right communication
|
||||
parameters.
|
||||
|
||||
\par SND_PCM_STATE_SETUP
|
||||
The PCM device has accepted communication parameters and it is waiting
|
||||
for \link ::snd_pcm_prepare() \endlink call to prepare the hardware for
|
||||
selected operation (playback or capture).
|
||||
|
||||
\par SND_PCM_STATE_PREPARE
|
||||
The PCM device is prepared for operation. Application can use
|
||||
\link ::snd_pcm_start() \endlink call, write or read data to start
|
||||
the operation.
|
||||
|
||||
\par SND_PCM_STATE_RUNNING
|
||||
The PCM device is running. It processes the samples. The stream can
|
||||
be stopped using the \link ::snd_pcm_drop() \endlink or
|
||||
\link ::snd_pcm_drain \endlink calls.
|
||||
|
||||
\par SND_PCM_STATE_XRUN
|
||||
The PCM device reached overrun (capture) or underrun (playback).
|
||||
You can use the -EPIPE return code from I/O functions
|
||||
(\link ::snd_pcm_writei() \endlink, \link ::snd_pcm_writen() \endlink,
|
||||
\link ::snd_pcm_readi() \endlink, \link ::snd_pcm_readi() \endlink)
|
||||
to determine this state without checking
|
||||
the actual state via \link ::snd_pcm_state() \endlink call. You can recover from
|
||||
this state with \link ::snd_pcm_prepare() \endlink,
|
||||
\link ::snd_pcm_drop() \endlink or \link ::snd_pcm_drain() \endlink calls.
|
||||
|
||||
\par SND_PCM_STATE_DRAINING
|
||||
The device is in this state when application using the capture mode
|
||||
called \link ::snd_pcm_drain() \endlink function. Until all data are
|
||||
read from the internal ring buffer using I/O routines
|
||||
(\link ::snd_pcm_readi() \endlink, \link ::snd_pcm_readn() \endlink),
|
||||
then the device stays in this state.
|
||||
|
||||
\par SND_PCM_STATE_PAUSED
|
||||
The device is in this state when application called
|
||||
the \link ::snd_pcm_pause() \endlink function until the pause is released.
|
||||
Not all hardware supports this feature. Application should check the
|
||||
capability with the \link ::snd_pcm_hw_params_can_pause() \endlink.
|
||||
|
||||
\par SND_PCM_STATE_SUSPENDED
|
||||
The device is in the suspend state provoked with the power management
|
||||
system. The stream can be resumed using \link ::snd_pcm_resume() \endlink
|
||||
call, but not all hardware supports this feature. Application should check
|
||||
the capability with the \link ::snd_pcm_hw_params_can_resume() \endlink.
|
||||
In other case, the calls \link ::snd_pcm_prepare() \endlink,
|
||||
\link ::snd_pcm_drop() \endlink, \link ::snd_pcm_drain() \endlink can be used
|
||||
to leave this state.
|
||||
|
||||
\section pcm_formats PCM formats
|
||||
|
||||
The full list of formats present the \link ::snd_pcm_format_t \endlink type.
|
||||
The 24-bit linear samples uses 32-bit physical space, but the sample is
|
||||
stored in low three bits. Some hardware does not support processing of full
|
||||
range, thus you may get the significative bits for linear samples via
|
||||
\link ::snd_pcm_hw_params_get_sbits \endlink function. The example: ICE1712
|
||||
chips support 32-bit sample processing, but low byte is ignored (playback)
|
||||
or zero (capture). The function \link ::snd_pcm_hw_params_get_sbits() \endlink
|
||||
returns 24 in the case.
|
||||
|
||||
\section alsa_transfers ALSA transfers
|
||||
|
||||
There are two methods to transfer samples in application. The first method
|
||||
is the standard read / write one. The second method, uses the direct audio
|
||||
buffer to communicate with the device while ALSA library manages this space
|
||||
itself. You can find examples of all communication schemes for playback
|
||||
in \ref example_test_pcm "Sine-wave generator example". To complete the
|
||||
list, we should note that \link ::snd_pcm_wait \endlink function contains
|
||||
embedded poll waiting implementation.
|
||||
|
||||
\subsection alsa_pcm_rw Read / Write transfer
|
||||
|
||||
There are two versions of read / write routines. The first expects the
|
||||
interleaved samples at input, and the second one expects non-interleaved
|
||||
(samples in separated buffers) at input. There are these functions for
|
||||
interleaved transfers: \link ::snd_pcm_writei \endlink,
|
||||
\link ::snd_pcm_readi \endlink. For non-interleaved transfers, there are
|
||||
these functions: \link ::snd_pcm_writen \endlink and \link ::snd_pcm_readn
|
||||
\endlink.
|
||||
|
||||
\subsection alsa_mmap_rw Direct Read / Write transfer (via mmaped areas)
|
||||
|
||||
There are two functions for this kind of transfer. Application can get an
|
||||
access to memory areas via \link ::snd_pcm_mmap_begin \endlink function.
|
||||
This functions returns the areas (single area is equal to a channel)
|
||||
containing the direct pointers to memory and sample position description
|
||||
in \link ::snd_pcm_channel_area_t \endlink structure. After application
|
||||
transfers the data in the memory areas, then it must be acknowledged
|
||||
the end of transfer via \link ::snd_pcm_mmap_commit() \endlink function
|
||||
to allow the ALSA library update the pointers to ring buffer. This sort of
|
||||
communication is also called "zero-copy", because the device does not require
|
||||
to copy the samples from application to another place in system memory.
|
||||
|
||||
\par
|
||||
|
||||
If you like to use the compatibility functions in mmap mode, there are
|
||||
read / write routines equaling to standard read / write transfers. Using
|
||||
these functions discards the benefits of direct access to memory region.
|
||||
See the \link ::snd_pcm_mmap_readi() \endlink,
|
||||
\link ::snd_pcm_writei() \endlink, \link ::snd_pcm_readn() \endlink
|
||||
and \link ::snd_pcm_writen() \endlink functions.
|
||||
|
||||
\section pcm_params Managing parameters
|
||||
|
||||
The ALSA PCM device uses two groups of PCM related parameters. The hardware
|
||||
parameters contains the stream description like format, rate, count of
|
||||
channels, ring buffer size etc. The software parameters contains the
|
||||
software (driver) related parameters. The communicatino behaviour can be
|
||||
controlled via these parameters, like automatic start, automatic stop,
|
||||
interrupting (chunk acknowledge) etc. The software parameters can be
|
||||
modified at any time (when valid hardware parameters are set). It includes
|
||||
the running state as well.
|
||||
|
||||
\subsection pcm_hw_params Hardware related parameters
|
||||
|
||||
The ALSA PCM devices use the parameter refining system for hardware
|
||||
parameters - \link ::snd_pcm_hw_params_t \endlink. It means, that
|
||||
application choose the full-range of configurations at first and then
|
||||
application sets single parameters until all parameters are elementary
|
||||
(definite).
|
||||
|
||||
\par Access modes
|
||||
|
||||
ALSA knows about five access modes. The first three can be used for direct
|
||||
communication. The access mode \link ::SND_PCM_ACCESS_MMAP_INTERLEAVED \endlink
|
||||
determines the direct memory area and interleaved sample organization.
|
||||
Interleaved organization means, that samples from channels are mixed together.
|
||||
The access mode \link ::SND_PCM_ACCESS_MMAP_NONINTERLEAVED \endlink
|
||||
determines the direct memory area and non-interleaved sample organization.
|
||||
Each channel has a separate buffer in the case. The complex direct memory
|
||||
organization represents the \link ::SND_PCM_ACCESS_MMAP_COMPLEX \endlink
|
||||
access mode. The sample organization does not fit the interleaved or
|
||||
non-interleaved access modes in the case. The last two access modes
|
||||
describes the read / write access methods.
|
||||
The \link ::SND_PCM_ACCESS_RW_INTERLEAVED \endlink access represents the read /
|
||||
write interleaved access and the \link ::SND_PCM_ACCESS_RW_NONINTERLEAVED \endlink
|
||||
represents the non-interleaved access.
|
||||
|
||||
\par Formats
|
||||
|
||||
The full list of formats is available in \link ::snd_pcm_format_t \endlink
|
||||
enumeration.
|
||||
|
||||
\subsection pcm_sw_params Software related parameters
|
||||
|
||||
These parameters - \link ::snd_pcm_sw_params_t \endlink can be modified at
|
||||
any time including the running state.
|
||||
|
||||
\par Minimum available count of samples
|
||||
|
||||
This parameter controls the wakeup point. If the count of available samples
|
||||
is equal or greater than this value, then application will be activated.
|
||||
|
||||
\par Timestamp mode
|
||||
|
||||
The timestamp mode specifies, if timestamps are activated. Currently, only
|
||||
\link ::SND_PCM_TSTAMP_NONE \endlink and \link ::SND_PCM_TSTAMP_MMAP
|
||||
\endlink modes are known. The mmap mode means that timestamp is taken
|
||||
on every period time boundary.
|
||||
|
||||
\par Minimal sleep
|
||||
|
||||
This parameters means the minimum of ticks to sleep using a standalone
|
||||
timer (usually the system timer). The tick resolution can be obtained
|
||||
via the function \link ::snd_pcm_hw_params_get_tick_time \endlink. This
|
||||
function can be used to fine-tune the transfer acknowledge process. It could
|
||||
be useful especially when some hardware does not support small transfer
|
||||
periods.
|
||||
|
||||
\par Transfer align
|
||||
|
||||
The read / write transfers can be aligned to this sample count. The modulo
|
||||
is ignored by device. Usually, this value is set to one (no align).
|
||||
|
||||
\par Start threshold
|
||||
|
||||
The start threshold parameter is used to determine the start point in
|
||||
stream. For playback, if samples in ring buffer is equal or greater than
|
||||
the start threshold parameters and the stream is not running, the stream will
|
||||
be started automatically from the device. For capture, if the application wants
|
||||
to read count of samples equal or greater then the stream will be started.
|
||||
If you want to use explicit start (\link ::snd_pcm_start \endlink), you can
|
||||
set this value greater than ring buffer size (in samples), but use the
|
||||
constant MAXINT is not a bad idea.
|
||||
|
||||
\par Stop threshold
|
||||
|
||||
Similarly, the stop threshold parameter is used to automatically stop
|
||||
the running stream, when the available samples crosses this boundary.
|
||||
It means, for playback, the empty samples in ring buffer and for capture,
|
||||
the filled (used) samples in ring buffer.
|
||||
|
||||
\par Silence threshold
|
||||
|
||||
The silence threshold specifies count of samples filled with silence
|
||||
ahead of the current application pointer for playback. It is useable
|
||||
for applications when an overrun is possible (like tasks depending on
|
||||
network I/O etc.). If application wants to manage the ahead samples itself,
|
||||
the \link ::snd_pcm_rewind() \endlink function allows to forget the last
|
||||
samples in the stream.
|
||||
|
||||
\section pcm_status Obtaining device status
|
||||
|
||||
The device status is stored in \link ::snd_pcm_status_t \endlink structure.
|
||||
These parameters can be obtained: the current stream state -
|
||||
\link ::snd_pcm_status_get_state \endlink, timestamp of trigger -
|
||||
\link ::snd_pcm_status_get_trigger_tstamp \endlink, timestamp of last
|
||||
update \link ::snd_pcm_status_get_tstamp \endlink, delay in samples -
|
||||
\link ::snd_pcm_status_get_delay \endlink, available count in samples -
|
||||
\link ::snd_pcm_status_get_avail \endlink, maximum available samples -
|
||||
\link ::snd_pcm_status_get_avail_max \endlink, ADC overrange count in
|
||||
samples - \link ::snd_pcm_status_get_overrange \endlink. The last two
|
||||
parameters - avail_max and overrange are reset to zero after the status
|
||||
call.
|
||||
|
||||
\subsection pcm_status_fast Obtaining fast device status
|
||||
|
||||
The function \link ::snd_pcm_avail_update \endlink updates the current
|
||||
available count of samples for writting (playback) or filled samples for
|
||||
reading (capture).
|
||||
<p>
|
||||
The function \link ::snd_pcm_delay \endlink returns the delay in samples.
|
||||
For playback, it means count of samples in the ring buffer before
|
||||
the next sample will be sent to DAC. For capture, it means count of samples
|
||||
in the ring buffer before the next sample will be captured from ADC.
|
||||
|
||||
\section pcm_action Managing the stream state
|
||||
|
||||
These functions directly and indirectly affecting the stream state:
|
||||
|
||||
\par snd_pcm_hw_params
|
||||
The \link ::snd_pcm_hw_params \endlink function brings the stream state
|
||||
to \link ::SND_PCM_STATE_SETUP \endlink
|
||||
if successfully finishes, otherwise the state \link ::SND_PCM_STATE_OPEN
|
||||
\endlink is entered.
|
||||
|
||||
\par snd_pcm_prepare
|
||||
The \link ::snd_pcm_prepare \endlink function enters the
|
||||
\link ::SND_PCM_STATE_PREPARED \endlink after a successfull finish.
|
||||
|
||||
\par snd_pcm_start
|
||||
The \link ::snd_pcm_start \endlink function enters
|
||||
the \link ::SND_PCM_STATE_RUNNING \endlink after a successfull finish.
|
||||
|
||||
\par snd_pcm_drop
|
||||
The \link ::snd_pcm_drop \endlink function enters the
|
||||
\link ::SND_PCM_STATE_SETUP \endlink state.
|
||||
|
||||
\par snd_pcm_drain
|
||||
The \link ::snd_pcm_drain \endlink function enters the
|
||||
\link ::SND_PCM_STATE_DRAINING \endlink, if
|
||||
the capture device has some samples in the ring buffer otherwise
|
||||
\link ::SND_PCM_STATE_SETUP \endlink state is entered.
|
||||
|
||||
\par snd_pcm_pause
|
||||
The \link ::snd_pcm_pause \endlink function enters the
|
||||
\link ::SND_PCM_STATE_PAUSED \endlink or
|
||||
\link ::SND_PCM_STATE_RUNNING \endlink.
|
||||
|
||||
\par snd_pcm_writei, snd_pcm_writen
|
||||
The \link ::snd_pcm_writei \endlink and \link ::snd_pcm_writen \endlink
|
||||
functions can conditionally start the stream -
|
||||
\link ::SND_PCM_STATE_RUNNING \endlink. They depend on the start threshold
|
||||
software parameter.
|
||||
|
||||
\par snd_pcm_readi, snd_pcm_readn
|
||||
The \link ::snd_pcm_readi \endlink and \link ::snd_pcm_readn \endlink
|
||||
functions can conditionally start the stream -
|
||||
\link ::SND_PCM_STATE_RUNNING \endlink. They depend on the start threshold
|
||||
software parameter.
|
||||
|
||||
\section pcm_sync Streams synchronization
|
||||
|
||||
There are two functions allowing link multiple streams together. In the
|
||||
case, the linking means that all operations are synchronized. Because the
|
||||
drivers cannot guarantee the synchronization (sample resolution) on hardware
|
||||
lacking this feature, the \link ::snd_pcm_info_get_sync \endlink function
|
||||
returns synchronization ID - \link ::snd_pcm_sync_id_t \endlink, which is equal
|
||||
for hardware synchronizated streams. When the \link ::snd_pcm_link \endlink
|
||||
function is called, all operations managing the stream state for these two
|
||||
streams are joined. The oposite function is \link ::snd_pcm_unlink \endlink.
|
||||
|
||||
\section pcm_examples Examples
|
||||
|
||||
The full featured examples with cross-links:
|
||||
|
||||
\par Sine-wave generator
|
||||
\ref example_test_pcm "example code"
|
||||
\par
|
||||
This example shows various transfer methods for the playback direction.
|
||||
|
||||
\par Latency measuring tool
|
||||
\ref example_test_latency "example code"
|
||||
\par
|
||||
This example shows the measuring of minimal latency between capture and
|
||||
playback devices.
|
||||
|
||||
*/
|
||||
|
||||
/**
|
||||
* \example ../test/pcm.c
|
||||
* \anchor example_test_pcm
|
||||
*/
|
||||
/**
|
||||
* \example ../test/latency.c
|
||||
* \anchor example_test_latency
|
||||
*/
|
||||
Loading…
Add table
Add a link
Reference in a new issue